Updated FFMPEG, and enabled some new formats, including Atrac3+

Frameworks/FFMPEG/FFMPEG.xcodeproj/project.pbxproj

@@ -222,7 +222,6 @@
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-		830F0D5117FC880E00042E8F /* time_.h in Headers */ = {isa = PBXBuildFile; fileRef = 830F0D5017FC880E00042E8F /* time_.h */; };
 		830F0D5317FC891800042E8F /* file_open.c in Sources */ = {isa = PBXBuildFile; fileRef = 830F0D5217FC891800042E8F /* file_open.c */; };
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@@ -278,6 +277,22 @@
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 		830F0DD217FC934D00042E8F /* spdif.h in Headers */ = {isa = PBXBuildFile; fileRef = 830F0DD017FC934D00042E8F /* spdif.h */; };
 		830F0DD617FC93E400042E8F /* xwma.c in Sources */ = {isa = PBXBuildFile; fileRef = 830F0DD417FC93E400042E8F /* xwma.c */; };
+		832F4BA618CD4F47003E940E /* lls2.c in Sources */ = {isa = PBXBuildFile; fileRef = 832F4BA418CD4F47003E940E /* lls2.c */; };
+		832F4BA718CD4F47003E940E /* lls2.h in Headers */ = {isa = PBXBuildFile; fileRef = 832F4BA518CD4F47003E940E /* lls2.h */; };
+		832F4BAF18CD4F6B003E940E /* atrac3plus_data.h in Headers */ = {isa = PBXBuildFile; fileRef = 832F4BA918CD4F6B003E940E /* atrac3plus_data.h */; };
+		832F4BB018CD4F6B003E940E /* atrac3plus.c in Sources */ = {isa = PBXBuildFile; fileRef = 832F4BAA18CD4F6B003E940E /* atrac3plus.c */; };
+		832F4BB118CD4F6B003E940E /* atrac3plus.h in Headers */ = {isa = PBXBuildFile; fileRef = 832F4BAB18CD4F6B003E940E /* atrac3plus.h */; };
+		832F4BB218CD4F6B003E940E /* atrac3plusdec.c in Sources */ = {isa = PBXBuildFile; fileRef = 832F4BAC18CD4F6B003E940E /* atrac3plusdec.c */; };
+		832F4BB318CD4F6B003E940E /* atrac3plusdsp.c in Sources */ = {isa = PBXBuildFile; fileRef = 832F4BAD18CD4F6B003E940E /* atrac3plusdsp.c */; };
+		832F4BB618CD4FEA003E940E /* macros.h in Headers */ = {isa = PBXBuildFile; fileRef = 832F4BB418CD4FEA003E940E /* macros.h */; settings = {ATTRIBUTES = (Public, ); }; };
+		832F4BB718CD4FEA003E940E /* timestamp.h in Headers */ = {isa = PBXBuildFile; fileRef = 832F4BB518CD4FEA003E940E /* timestamp.h */; };
+		832F4BBA18CD53E9003E940E /* ttadsp.c in Sources */ = {isa = PBXBuildFile; fileRef = 832F4BB818CD53E9003E940E /* ttadsp.c */; };
+		832F4BBB18CD53E9003E940E /* ttadsp.h in Headers */ = {isa = PBXBuildFile; fileRef = 832F4BB918CD53E9003E940E /* ttadsp.h */; };
+		832F4BC018CD5DD9003E940E /* downmix_info.c in Sources */ = {isa = PBXBuildFile; fileRef = 832F4BBE18CD5DD9003E940E /* downmix_info.c */; };
+		832F4BC118CD5DD9003E940E /* downmix_info.h in Headers */ = {isa = PBXBuildFile; fileRef = 832F4BBF18CD5DD9003E940E /* downmix_info.h */; };
+		832F4BC318CD5DE2003E940E /* time_.h in Headers */ = {isa = PBXBuildFile; fileRef = 832F4BC218CD5DE2003E940E /* time_.h */; };
+		832F4BC518CD601B003E940E /* ttadsp_init.c in Sources */ = {isa = PBXBuildFile; fileRef = 832F4BC418CD601B003E940E /* ttadsp_init.c */; };
+		832F4BC718CD6052003E940E /* dcadsp_init.c in Sources */ = {isa = PBXBuildFile; fileRef = 832F4BC618CD6052003E940E /* dcadsp_init.c */; };
 		833C3795180328A300CBA602 /* takdec.c in Sources */ = {isa = PBXBuildFile; fileRef = 833C3794180328A300CBA602 /* takdec.c */; };
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@@ -401,8 +416,6 @@
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-		838491081807CDF400E7332D /* lls.c in Sources */ = {isa = PBXBuildFile; fileRef = 838491061807CDF400E7332D /* lls.c */; };
-		838491091807CDF400E7332D /* lls.h in Headers */ = {isa = PBXBuildFile; fileRef = 838491071807CDF400E7332D /* lls.h */; };
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@@ -430,7 +443,6 @@
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-		8393B80B18052BC200913C76 /* mpegaudioenc.c in Sources */ = {isa = PBXBuildFile; fileRef = 8393B7F718052BC200913C76 /* mpegaudioenc.c */; };
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@@ -665,7 +677,6 @@
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-		830F0D5017FC880E00042E8F /* time_.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = time_.h; sourceTree = "<group>"; };
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@@ -721,6 +732,22 @@
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 		830F0DD417FC93E400042E8F /* xwma.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = xwma.c; sourceTree = "<group>"; };
+		832F4BA418CD4F47003E940E /* lls2.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = lls2.c; sourceTree = "<group>"; };
+		832F4BA518CD4F47003E940E /* lls2.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = lls2.h; sourceTree = "<group>"; };
+		832F4BA918CD4F6B003E940E /* atrac3plus_data.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = atrac3plus_data.h; sourceTree = "<group>"; };
+		832F4BAA18CD4F6B003E940E /* atrac3plus.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = atrac3plus.c; sourceTree = "<group>"; };
+		832F4BAB18CD4F6B003E940E /* atrac3plus.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = atrac3plus.h; sourceTree = "<group>"; };
+		832F4BAC18CD4F6B003E940E /* atrac3plusdec.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = atrac3plusdec.c; sourceTree = "<group>"; };
+		832F4BAD18CD4F6B003E940E /* atrac3plusdsp.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = atrac3plusdsp.c; sourceTree = "<group>"; };
+		832F4BB418CD4FEA003E940E /* macros.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = macros.h; sourceTree = "<group>"; };
+		832F4BB518CD4FEA003E940E /* timestamp.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = timestamp.h; sourceTree = "<group>"; };
+		832F4BB818CD53E9003E940E /* ttadsp.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = ttadsp.c; sourceTree = "<group>"; };
+		832F4BB918CD53E9003E940E /* ttadsp.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ttadsp.h; sourceTree = "<group>"; };
+		832F4BBE18CD5DD9003E940E /* downmix_info.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = downmix_info.c; sourceTree = "<group>"; };
+		832F4BBF18CD5DD9003E940E /* downmix_info.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = downmix_info.h; sourceTree = "<group>"; };
+		832F4BC218CD5DE2003E940E /* time_.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = time_.h; sourceTree = "<group>"; };
+		832F4BC418CD601B003E940E /* ttadsp_init.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = ttadsp_init.c; sourceTree = "<group>"; };
+		832F4BC618CD6052003E940E /* dcadsp_init.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = dcadsp_init.c; sourceTree = "<group>"; };
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@@ -844,8 +871,6 @@
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-		838491061807CDF400E7332D /* lls.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = lls.c; sourceTree = "<group>"; };
-		838491071807CDF400E7332D /* lls.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = lls.h; sourceTree = "<group>"; };
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 		8384910E1807CEC400E7332D /* lpc.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = lpc.c; sourceTree = "<group>"; };
@@ -873,7 +898,6 @@
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@@ -967,6 +991,13 @@
 		830F0B8E17FC4FB900042E8F /* libavcodec */ = {
 			isa = PBXGroup;
 			children = (
+				832F4BB818CD53E9003E940E /* ttadsp.c */,
+				832F4BB918CD53E9003E940E /* ttadsp.h */,
+				832F4BA918CD4F6B003E940E /* atrac3plus_data.h */,
+				832F4BAA18CD4F6B003E940E /* atrac3plus.c */,
+				832F4BAB18CD4F6B003E940E /* atrac3plus.h */,
+				832F4BAC18CD4F6B003E940E /* atrac3plusdec.c */,
+				832F4BAD18CD4F6B003E940E /* atrac3plusdsp.c */,
 				833E5F9D18A31EC4006A85F9 /* twinvq_data.h */,
 				833E5F9E18A31EC4006A85F9 /* twinvq.c */,
 				833E5F9F18A31EC4006A85F9 /* twinvq.h */,
@@ -1074,7 +1105,6 @@
 				8393B7F118052BC200913C76 /* mpegaudiodsp_data.c */,
 				8393B7F318052BC200913C76 /* mpegaudiodsp_float.c */,
 				8393B7F618052BC200913C76 /* mpegaudiodsp.h */,
-				8393B7F718052BC200913C76 /* mpegaudioenc.c */,
 				8393B7F818052BC200913C76 /* mpegaudiotab.h */,
 				833C37D418032E7000CBA602 /* jrevdct.c */,
 				833C37CA18032D4800CBA602 /* golomb.c */,
@@ -1308,8 +1338,13 @@
 		830F0BBA17FC4FB900042E8F /* libavutil */ = {
 			isa = PBXGroup;
 			children = (
-				838491061807CDF400E7332D /* lls.c */,
+				832F4BC218CD5DE2003E940E /* time_.h */,
-				838491071807CDF400E7332D /* lls.h */,
+				832F4BBE18CD5DD9003E940E /* downmix_info.c */,
+				832F4BBF18CD5DD9003E940E /* downmix_info.h */,
+				832F4BB418CD4FEA003E940E /* macros.h */,
+				832F4BB518CD4FEA003E940E /* timestamp.h */,
+				832F4BA418CD4F47003E940E /* lls2.c */,
+				832F4BA518CD4F47003E940E /* lls2.h */,
 				838490721807B07000E7332D /* md5.c */,
 				838490731807B07000E7332D /* md5.h */,
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@@ -1319,7 +1354,6 @@
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-				830F0D5017FC880E00042E8F /* time_.h */,
 				830F0D4A17FC862300042E8F /* time.c */,
 				830F0D4C17FC862400042E8F /* utils.c */,
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@@ -1438,6 +1472,8 @@
 		830F0D0717FC80B400042E8F /* x86 */ = {
 			isa = PBXGroup;
 			children = (
+				832F4BC618CD6052003E940E /* dcadsp_init.c */,
+				832F4BC418CD601B003E940E /* ttadsp_init.c */,
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 				8384910E1807CEC400E7332D /* lpc.c */,
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@@ -1477,12 +1513,13 @@
 			isa = PBXHeadersBuildPhase;
 			buildActionMask = 2147483647;
 			files = (
+				83BCB8E217FCA64400760340 /* avconfig.h in Headers */,
 				830F0D3417FC841B00042E8F /* opt.h in Headers */,
 				830F0C1C17FC523000042E8F /* url.h in Headers */,
-				83BCB8E217FCA64400760340 /* avconfig.h in Headers */,
 				830F0BDC17FC4FB900042E8F /* old_codec_ids.h in Headers */,
 				830F0C0B17FC4FB900042E8F /* pixfmt.h in Headers */,
 				830F0C0517FC4FB900042E8F /* intfloat_readwrite.h in Headers */,
+				832F4BBB18CD53E9003E940E /* ttadsp.h in Headers */,
 				830F0C0417FC4FB900042E8F /* intfloat.h in Headers */,
 				830F0C0717FC4FB900042E8F /* mathematics.h in Headers */,
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@@ -1502,6 +1539,7 @@
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 				830F0BFA17FC4FB900042E8F /* attributes.h in Headers */,
 				830F0BFF17FC4FB900042E8F /* common.h in Headers */,
+				832F4BB618CD4FEA003E940E /* macros.h in Headers */,
 				830F0BFB17FC4FB900042E8F /* avutil.h in Headers */,
 				830F0C0D17FC4FB900042E8F /* samplefmt.h in Headers */,
 				830F0BFD17FC4FB900042E8F /* buffer.h in Headers */,
@@ -1598,11 +1636,12 @@
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 				830F0CCC17FC7F1E00042E8F /* celp_filters.h in Headers */,
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-				830F0D5117FC880E00042E8F /* time_.h in Headers */,
 				830F0BD717FC4FB900042E8F /* get_bits.h in Headers */,
 				830F0C2417FC527400042E8F /* atomic.h in Headers */,
+				832F4BC318CD5DE2003E940E /* time_.h in Headers */,
 				833E5FA118A31EC4006A85F9 /* twinvq_data.h in Headers */,
 				830F0C5D17FC7CEA00042E8F /* id3v1.h in Headers */,
+				832F4BAF18CD4F6B003E940E /* atrac3plus_data.h in Headers */,
 				838490631807AF0100E7332D /* ac3dec.h in Headers */,
 				838490C31807C37300E7332D /* atrac.h in Headers */,
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@@ -1620,7 +1659,6 @@
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-				838491091807CDF400E7332D /* lls.h in Headers */,
 				838490F21807C57B00E7332D /* aacadtsdec.h in Headers */,
 				830F0CD117FC7F1E00042E8F /* dct.h in Headers */,
 				830F0DBC17FC921D00042E8F /* timer.h in Headers */,
@@ -1628,6 +1666,7 @@
 				830F0BD817FC4FB900042E8F /* internal.h in Headers */,
 				830F0BE517FC4FB900042E8F /* wma_common.h in Headers */,
 				830F0CDB17FC7F1E00042E8F /* error_resilience.h in Headers */,
+				832F4BB718CD4FEA003E940E /* timestamp.h in Headers */,
 				830F0C4A17FC7CA300042E8F /* avio_internal.h in Headers */,
 				830F0C7017FC7DB100042E8F /* pixdesc.h in Headers */,
 				833C37C318032CF600CBA602 /* dirac_arith.h in Headers */,
@@ -1638,6 +1677,7 @@
 				830F0CE817FC7F1E00042E8F /* hpeldsp.h in Headers */,
 				830F0D8B17FC8E8B00042E8F /* aacpsdsp.h in Headers */,
 				830F0CD617FC7F1E00042E8F /* dctref.h in Headers */,
+				832F4BA718CD4F47003E940E /* lls2.h in Headers */,
 				838490F61807C57B00E7332D /* aacps_tablegen.h in Headers */,
 				830F0C1917FC523000042E8F /* rdt.h in Headers */,
 				838490941807BC9400E7332D /* dcadsp.h in Headers */,
@@ -1680,8 +1720,10 @@
 				830F0C8117FC7ED100042E8F /* crc.h in Headers */,
 				833C379C180328B300CBA602 /* tak.h in Headers */,
 				830F0DB817FC921D00042E8F /* asm.h in Headers */,
+				832F4BC118CD5DD9003E940E /* downmix_info.h in Headers */,
 				830F0CC517FC7F1E00042E8F /* acelp_pitch_delay.h in Headers */,
 				838490D91807C47E00E7332D /* g729postfilter.h in Headers */,
+				832F4BB118CD4F6B003E940E /* atrac3plus.h in Headers */,
 				830F0D8F17FC8E8B00042E8F /* sbrdsp.h in Headers */,
 				830F0CCE17FC7F1E00042E8F /* celp_math.h in Headers */,
 				8384905D1807AF0100E7332D /* ac3_parser.h in Headers */,
@@ -1755,7 +1797,6 @@
 			buildActionMask = 2147483647;
 			files = (
 				838490DA1807C47E00E7332D /* g726.c in Sources */,
-				838491081807CDF400E7332D /* lls.c in Sources */,
 				833C37A918032AAD00CBA602 /* img2.c in Sources */,
 				838490701807B04200E7332D /* lfg.c in Sources */,
 				838490981807BC9C00E7332D /* dtshddec.c in Sources */,
@@ -1768,6 +1809,7 @@
 				830F0CF417FC7F1E00042E8F /* rdft.c in Sources */,
 				830F0DB217FC8FCE00042E8F /* log2_tab.c in Sources */,
 				830F0BEE17FC4FB900042E8F /* allformats.c in Sources */,
+				832F4BB218CD4F6B003E940E /* atrac3plusdec.c in Sources */,
 				830F0D4817FC85ED00042E8F /* intmath.c in Sources */,
 				8384904F1807898A00E7332D /* ape.c in Sources */,
 				830F0CF017FC7F1E00042E8F /* ratecontrol.c in Sources */,
@@ -1804,6 +1846,7 @@
 				830F0BF217FC4FB900042E8F /* avio.c in Sources */,
 				830C4B5F180C3434007674D6 /* amrwbdec.c in Sources */,
 				838490DC1807C47E00E7332D /* g723_1.c in Sources */,
+				832F4BC518CD601B003E940E /* ttadsp_init.c in Sources */,
 				830F0CE217FC7F1E00042E8F /* frame_thread_encoder.c in Sources */,
 				830C4B61180C343C007674D6 /* amr.c in Sources */,
 				838491151807D1D200E7332D /* tta.c in Sources */,
@@ -1881,6 +1924,7 @@
 				830F0CDE17FC7F1E00042E8F /* faanidct.c in Sources */,
 				833C379B180328B300CBA602 /* tak.c in Sources */,
 				830F0CC917FC7F1E00042E8F /* bit_depth_template.c in Sources */,
+				832F4BC718CD6052003E940E /* dcadsp_init.c in Sources */,
 				8384906C1807AFB800E7332D /* aac_ac3_parser.c in Sources */,
 				8393B7E318052BB000913C76 /* mpeg.c in Sources */,
 				838491111807CEF400E7332D /* sbrdsp_init.c in Sources */,
@@ -1911,11 +1955,13 @@
 				830F0D5F17FC89BD00042E8F /* options.c in Sources */,
 				8393B7FE18052BC200913C76 /* mpegaudiodata.c in Sources */,
 				833C37D318032E4800CBA602 /* dsputil_init.c in Sources */,
+				832F4BA618CD4F47003E940E /* lls2.c in Sources */,
 				830F0DB917FC921D00042E8F /* cpu.c in Sources */,
 				830F0C7D17FC7E4E00042E8F /* mathematics.c in Sources */,
 				830F0BDD17FC4FB900042E8F /* parser.c in Sources */,
 				830F0C5C17FC7CEA00042E8F /* id3v1.c in Sources */,
 				8393B7F918052BC200913C76 /* mpegaudio_parser.c in Sources */,
+				832F4BBA18CD53E9003E940E /* ttadsp.c in Sources */,
 				830F0CFC17FC7F1E00042E8F /* videodsp.c in Sources */,
 				830F0D8C17FC8E8B00042E8F /* dv_profile.c in Sources */,
 				833C37B118032AEF00CBA602 /* diracdsp.c in Sources */,
@@ -1929,7 +1975,9 @@
 				8384904D1807898200E7332D /* apedec.c in Sources */,
 				830F0D2E17FC841B00042E8F /* channel_layout.c in Sources */,
 				830F0CE517FC7F1E00042E8F /* h264chroma.c in Sources */,
+				832F4BB318CD4F6B003E940E /* atrac3plusdsp.c in Sources */,
 				830F0BE717FC4FB900042E8F /* wmadec.c in Sources */,
+				832F4BC018CD5DD9003E940E /* downmix_info.c in Sources */,
 				838490931807BC9400E7332D /* dcadsp.c in Sources */,
 				838490FF1807C58500E7332D /* aacdec.c in Sources */,
 				830F0CC617FC7F1E00042E8F /* acelp_vectors.c in Sources */,
@@ -1944,6 +1992,7 @@
 				830F0D3817FC844E00042E8F /* parseutils.c in Sources */,
 				830F0D2017FC82AB00042E8F /* cpu.c in Sources */,
 				8393B7FC18052BC200913C76 /* mpegaudio.c in Sources */,
+				832F4BB018CD4F6B003E940E /* atrac3plus.c in Sources */,
 				838490D81807C47E00E7332D /* g729postfilter.c in Sources */,
 				833C37C618032CF600CBA602 /* dirac_parser.c in Sources */,
 				830F0CED17FC7F1E00042E8F /* mpeg12data.c in Sources */,
@@ -1954,7 +2003,6 @@
 				830F0BE817FC4FB900042E8F /* wmalosslessdec.c in Sources */,
 				838490621807AF0100E7332D /* ac3dec.c in Sources */,
 				83BCB8E317FCA64400760340 /* timecode.c in Sources */,
-				8393B80B18052BC200913C76 /* mpegaudioenc.c in Sources */,
 				8384910D1807CEB600E7332D /* sbrdsp.c in Sources */,
 				8384911A1807D1D900E7332D /* ttadata.c in Sources */,
 				838490B91807C1C300E7332D /* truespeech.c in Sources */,
@@ -2009,7 +2057,6 @@
 				GCC_PREFIX_HEADER = FFMPEG_Prefix.pch;
 				HEADER_SEARCH_PATHS = (
 					"$(inherited)",
-					/Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/include,
 					"ffmpeg-minimal",
 				);
 				INFOPLIST_FILE = Info.plist;
@@ -2034,7 +2081,6 @@
 				GCC_PREFIX_HEADER = FFMPEG_Prefix.pch;
 				HEADER_SEARCH_PATHS = (
 					"$(inherited)",
-					/Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/include,
 					"ffmpeg-minimal",
 				);
 				INFOPLIST_FILE = Info.plist;
@@ -2049,6 +2095,7 @@
 		1DEB91B208733DA50010E9CD /* Debug */ = {
 			isa = XCBuildConfiguration;
 			buildSettings = {
+				ALWAYS_SEARCH_USER_PATHS = NO;
 				CODE_SIGN_IDENTITY = "Christopher Snowhill";
 				GCC_PREPROCESSOR_DEFINITIONS = "DEBUG=1";
 				GCC_WARN_ABOUT_RETURN_TYPE = YES;
@@ -2063,6 +2110,7 @@
 		1DEB91B308733DA50010E9CD /* Release */ = {
 			isa = XCBuildConfiguration;
 			buildSettings = {
+				ALWAYS_SEARCH_USER_PATHS = NO;
 				CODE_SIGN_IDENTITY = "Christopher Snowhill";
 				DEBUG_INFORMATION_FORMAT = "dwarf-with-dsym";
 				GCC_WARN_ABOUT_RETURN_TYPE = YES;

Frameworks/FFMPEG/ffmpeg-minimal/config.h

@@ -5,6 +5,7 @@
 #define ARCH_X86 1
 #define ARCH_ARM 0
 #define ARCH_PPC 0
+#define ARCH_AARCH64 0
 #define ARCH_ALPHA 0
 #define ARCH_BFIN 0
 #define ARCH_SH4 0
@@ -79,6 +80,7 @@
 #define CONFIG_RISKY 1
 #define CONFIG_FTRAPV 0
 #define CONFIG_SMALL 0
+#define CONFIG_XVMC 0
 
 #define NULL_IF_CONFIG_SMALL(a) a
 
@@ -88,8 +90,9 @@
 #define CONFIG_H264_VAAPI_HWACCEL 0
 #define CONFIG_H264_VDA_HWACCEL 0
 #define CONFIG_H264_VDPAU_HWACCEL 0
+#define CONFIG_MPEG1_XVMC_HWACCEL 0
 #define CONFIG_MPEG1_VDPAU_HWACCEL 0
-#define CONFIG_MPEG1_DXVA2_HWACCEL 0
+#define CONFIG_MPEG2_XVMC_HWACCEL 0
 #define CONFIG_MPEG2_DXVA2_HWACCEL 0
 #define CONFIG_MPEG2_VAAPI_HWACCEL 0
 #define CONFIG_MPEG2_VDPAU_HWACCEL 0
@@ -135,6 +138,7 @@
 #define CONFIG_CAVS_DECODER 0
 #define CONFIG_CDGRAPHICS_DECODER 0
 #define CONFIG_CDXL_DECODER 0
+#define CONFIG_CINEPAK_ENCODER 0
 #define CONFIG_CINEPAK_DECODER 0
 #define CONFIG_CLJR_ENCODER 0
 #define CONFIG_CLJR_DECODER 0
@@ -170,6 +174,7 @@
 #define CONFIG_FFV1_DECODER 0
 #define CONFIG_FFVHUFF_ENCODER 0
 #define CONFIG_FFVHUFF_DECODER 0
+#define CONFIG_FIC_DECODER 0
 #define CONFIG_FLASHSV_ENCODER 0
 #define CONFIG_FLASHSV_DECODER 0
 #define CONFIG_FLASHSV2_ENCODER 0
@@ -194,6 +199,8 @@
 #define CONFIG_H264_CRYSTALHD_DECODER 0
 #define CONFIG_H264_VDA_DECODER 0
 #define CONFIG_H264_VDPAU_DECODER 0
+#define CONFIG_HEVC_DECODER 0
+#define CONFIG_HNM4_VIDEO_DECODER 0
 #define CONFIG_HUFFYUV_ENCODER 0
 #define CONFIG_HUFFYUV_DECODER 0
 #define CONFIG_IDCIN_DECODER 0
@@ -398,6 +405,7 @@
 #define CONFIG_APE_DECODER 1
 #define CONFIG_ATRAC1_DECODER 0
 #define CONFIG_ATRAC3_DECODER 1
+#define CONFIG_ATRAC3P_DECODER 1
 #define CONFIG_BINKAUDIO_DCT_DECODER 0
 #define CONFIG_BINKAUDIO_RDFT_DECODER 0
 #define CONFIG_BMV_AUDIO_DECODER 0
@@ -427,6 +435,7 @@
 #define CONFIG_MP2_ENCODER 0
 #define CONFIG_MP2_DECODER 0
 #define CONFIG_MP2FLOAT_DECODER 1
+#define CONFIG_MP2FIXED_ENCODER 0
 #define CONFIG_MP3_DECODER 0
 #define CONFIG_MP3FLOAT_DECODER 1
 #define CONFIG_MP3ADU_DECODER 0
@@ -550,6 +559,7 @@
 #define CONFIG_ADPCM_G722_DECODER 1
 #define CONFIG_ADPCM_G726_ENCODER 0
 #define CONFIG_ADPCM_G726_DECODER 1
+#define CONFIG_ADPCM_G726LE_DECODER 1
 #define CONFIG_ADPCM_IMA_AMV_DECODER 0
 #define CONFIG_ADPCM_IMA_APC_DECODER 0
 #define CONFIG_ADPCM_IMA_DK3_DECODER 1
@@ -644,8 +654,10 @@
 #define CONFIG_LIBVPX_VP9_ENCODER 0
 #define CONFIG_LIBVPX_VP9_DECODER 0
 #define CONFIG_LIBWAVPACK_ENCODER 0
+#define CONFIG_LIBWEBP_ENCODER 0
 #define CONFIG_LIBX264_ENCODER 0
 #define CONFIG_LIBX264RGB_ENCODER 0
+#define CONFIG_LIBX265_ENCODER 0
 #define CONFIG_LIBXAVS_ENCODER 0
 #define CONFIG_LIBXVID_ENCODER 0
 #define CONFIG_LIBZVBI_TELETEXT_DECODER 0
@@ -673,6 +685,7 @@
 #define CONFIG_H261_PARSER 0
 #define CONFIG_H263_PARSER 0
 #define CONFIG_H264_PARSER 0
+#define CONFIG_HEVC_PARSER 0
 #define CONFIG_MJPEG_PARSER 0
 #define CONFIG_MLP_PARSER 0
 #define CONFIG_MPEG4VIDEO_PARSER 0
@@ -687,6 +700,7 @@
 #define CONFIG_VORBIS_PARSER 0
 #define CONFIG_VP3_PARSER 0
 #define CONFIG_VP8_PARSER 0
+#define CONFIG_VP9_PARSER 0
 
 #define CONFIG_AAC_ADTSTOASC_BSF 1
 #define CONFIG_CHOMP_BSF 0
@@ -809,9 +823,12 @@
 #define CONFIG_H263_DEMUXER 0
 #define CONFIG_H264_MUXER 0
 #define CONFIG_H264_DEMUXER 0
-#define CONFIG_H265_DEMUXER 0
+#define CONFIG_HDS_MUXER 0
+#define CONFIG_HEVC_MUXER 0
+#define CONFIG_HEVC_DEMUXER 0
 #define CONFIG_HLS_MUXER 0
 #define CONFIG_HLS_DEMUXER 0
+#define CONFIG_HNM_DEMUXER 0
 #define CONFIG_ICO_MUXER 0
 #define CONFIG_ICO_DEMUXER 0
 #define CONFIG_IDCIN_DEMUXER 0
@@ -900,6 +917,7 @@
 #define CONFIG_OGG_DEMUXER 0
 #define CONFIG_OMA_MUXER 0
 #define CONFIG_OMA_DEMUXER 0
+#define CONFIG_OPUS_MUXER 0
 #define CONFIG_PAF_DEMUXER 0
 #define CONFIG_PCM_ALAW_MUXER 0
 #define CONFIG_PCM_ALAW_DEMUXER 0
@@ -970,6 +988,7 @@
 #define CONFIG_SAP_DEMUXER 0
 #define CONFIG_SBG_DEMUXER 0
 #define CONFIG_SDP_DEMUXER 0
+#define CONFIG_SDR2_DEMUXER 0
 #define CONFIG_SEGAFILM_DEMUXER 0
 #define CONFIG_SEGMENT_MUXER 0
 #define CONFIG_SHORTEN_DEMUXER 0
@@ -984,6 +1003,7 @@
 #define CONFIG_SOX_DEMUXER 0
 #define CONFIG_SPDIF_MUXER 0
 #define CONFIG_SPDIF_DEMUXER 1
+#define CONFIG_SPEEX_MUXER 0
 #define CONFIG_SRT_MUXER 0
 #define CONFIG_SRT_DEMUXER 0
 #define CONFIG_STR_DEMUXER 0
@@ -1005,6 +1025,7 @@
 #define CONFIG_TTA_DEMUXER 1
 #define CONFIG_TXD_DEMUXER 0
 #define CONFIG_TTY_DEMUXER 0
+#define CONFIG_UNCODEDFRAMECRC_MUXER 0
 #define CONFIG_VC1_MUXER 0
 #define CONFIG_VC1_DEMUXER 0
 #define CONFIG_VC1T_MUXER 0
@@ -1066,6 +1087,7 @@
 #define CONFIG_RTP_PROTOCOL 0
 #define CONFIG_SCTP_PROTOCOL 0
 #define CONFIG_SRTP_PROTOCOL 0
+#define CONFIG_SUBFILE_PROTOCOL 0
 #define CONFIG_TCP_PROTOCOL 0
 #define CONFIG_TLS_PROTOCOL 0
 #define CONFIG_UDP_PROTOCOL 0

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/aac.h

@@ -234,7 +234,7 @@ typedef struct SingleChannelElement {
     int sf_idx[128];                                ///< scalefactor indices (used by encoder)
     uint8_t zeroes[128];                            ///< band is not coded (used by encoder)
     DECLARE_ALIGNED(32, float,   coeffs)[1024];     ///< coefficients for IMDCT
-    DECLARE_ALIGNED(32, float,   saved)[1024];      ///< overlap
+    DECLARE_ALIGNED(32, float,   saved)[1536];      ///< overlap
     DECLARE_ALIGNED(32, float,   ret_buf)[2048];    ///< PCM output buffer
     DECLARE_ALIGNED(16, float,   ltp_state)[3072];  ///< time signal for LTP
     PredictorState predictor_state[MAX_PREDICTORS];

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/aac_ac3_parser.c

@@ -20,7 +20,9 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include "libavutil/channel_layout.h"
 #include "libavutil/common.h"
+#include "libavutil/internal.h"
 #include "parser.h"
 #include "aac_ac3_parser.h"
 
@@ -82,20 +84,27 @@ get_next:
     if (avctx->codec_id != AV_CODEC_ID_AAC) {
         avctx->sample_rate = s->sample_rate;
 
-        /* allow downmixing to stereo (or mono for AC-3) */
+        /* (E-)AC-3: allow downmixing to stereo or mono */
-        AV_NOWARN_DEPRECATED(
+#if FF_API_REQUEST_CHANNELS
-        if(avctx->request_channels > 0 &&
+FF_DISABLE_DEPRECATION_WARNINGS
-                avctx->request_channels < s->channels &&
+        if (avctx->request_channels == 1)
-                (avctx->request_channels <= 2 ||
+            avctx->request_channel_layout = AV_CH_LAYOUT_MONO;
-                (avctx->request_channels == 1 &&
+        else if (avctx->request_channels == 2)
-                (avctx->codec_id == AV_CODEC_ID_AC3 ||
+            avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;
-                 avctx->codec_id == AV_CODEC_ID_EAC3)))) {
+FF_ENABLE_DEPRECATION_WARNINGS
-            avctx->channels = avctx->request_channels;
+#endif
+        if (s->channels > 1 &&
+            avctx->request_channel_layout == AV_CH_LAYOUT_MONO) {
+            avctx->channels       = 1;
+            avctx->channel_layout = AV_CH_LAYOUT_MONO;
+        } else if (s->channels > 2 &&
+                   avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
+            avctx->channels       = 2;
+            avctx->channel_layout = AV_CH_LAYOUT_STEREO;
         } else {
             avctx->channels = s->channels;
             avctx->channel_layout = s->channel_layout;
         }
-        );
         s1->duration = s->samples;
         avctx->audio_service_type = s->service_type;
     }

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/aac_adtstoasc_bsf.c

@@ -88,6 +88,7 @@ static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
             buf_size -= get_bits_count(&gb)/8;
             buf      += get_bits_count(&gb)/8;
         }
+        av_free(avctx->extradata);
         avctx->extradata_size = 2 + pce_size;
         avctx->extradata = av_mallocz(avctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
 

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/aacdec.c

@@ -2,6 +2,7 @@
  * AAC decoder
  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
  *
  * AAC LATM decoder
  * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
@@ -73,6 +74,7 @@
  * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)
  * Y                    Parametric Stereo
  * N                    Direct Stream Transfer
+ * Y                    Enhanced AAC Low Delay (ER AAC ELD)
  *
  * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
  *       - HE AAC v2 comprises LC AAC with Spectral Band Replication and
@@ -104,6 +106,7 @@
 #include <assert.h>
 #include <errno.h>
 #include <math.h>
+#include <stdint.h>
 #include <string.h>
 
 #if ARCH_ARM
@@ -194,6 +197,9 @@ static int frame_configure_elements(AVCodecContext *avctx)
 
     /* get output buffer */
     av_frame_unref(ac->frame);
+    if (!avctx->channels)
+        return 1;
+
     ac->frame->nb_samples = 2048;
     if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
         return ret;
@@ -532,6 +538,25 @@ static int set_default_channel_config(AVCodecContext *avctx,
     *tags = tags_per_config[channel_config];
     memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
            *tags * sizeof(*layout_map));
+
+    /*
+     * AAC specification has 7.1(wide) as a default layout for 8-channel streams.
+     * However, at least Nero AAC encoder encodes 7.1 streams using the default
+     * channel config 7, mapping the side channels of the original audio stream
+     * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
+     * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
+     * the incorrect streams as if they were correct (and as the encoder intended).
+     *
+     * As actual intended 7.1(wide) streams are very rare, default to assuming a
+     * 7.1 layout was intended.
+     */
+    if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
+        av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
+               " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
+               " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
+        layout_map[2][2] = AAC_CHANNEL_SIDE;
+    }
+
     return 0;
 }
 
@@ -794,9 +819,9 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
         case AOT_ER_AAC_LD:
             res_flags = get_bits(gb, 3);
             if (res_flags) {
-                av_log(avctx, AV_LOG_ERROR,
+                avpriv_report_missing_feature(avctx,
-                       "AAC data resilience not supported (flags %x)\n",
+                                              "AAC data resilience (flags %x)",
-                       res_flags);
+                                              res_flags);
                 return AVERROR_PATCHWELCOME;
             }
             break;
@@ -810,15 +835,75 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
     case AOT_ER_AAC_LD:
         ep_config = get_bits(gb, 2);
         if (ep_config) {
-            av_log(avctx, AV_LOG_ERROR,
+            avpriv_report_missing_feature(avctx,
-                   "epConfig %d is not supported.\n",
+                                          "epConfig %d", ep_config);
-                   ep_config);
             return AVERROR_PATCHWELCOME;
         }
     }
     return 0;
 }
 
+static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
+                                     GetBitContext *gb,
+                                     MPEG4AudioConfig *m4ac,
+                                     int channel_config)
+{
+    int ret, ep_config, res_flags;
+    uint8_t layout_map[MAX_ELEM_ID*4][3];
+    int tags = 0;
+    const int ELDEXT_TERM = 0;
+
+    m4ac->ps  = 0;
+    m4ac->sbr = 0;
+
+    if (get_bits1(gb)) { // frameLengthFlag
+        avpriv_request_sample(avctx, "960/120 MDCT window");
+        return AVERROR_PATCHWELCOME;
+    }
+
+    res_flags = get_bits(gb, 3);
+    if (res_flags) {
+        avpriv_report_missing_feature(avctx,
+                                      "AAC data resilience (flags %x)",
+                                      res_flags);
+        return AVERROR_PATCHWELCOME;
+    }
+
+    if (get_bits1(gb)) { // ldSbrPresentFlag
+        avpriv_report_missing_feature(avctx,
+                                      "Low Delay SBR");
+        return AVERROR_PATCHWELCOME;
+    }
+
+    while (get_bits(gb, 4) != ELDEXT_TERM) {
+        int len = get_bits(gb, 4);
+        if (len == 15)
+            len += get_bits(gb, 8);
+        if (len == 15 + 255)
+            len += get_bits(gb, 16);
+        if (get_bits_left(gb) < len * 8 + 4) {
+            av_log(ac->avctx, AV_LOG_ERROR, overread_err);
+            return AVERROR_INVALIDDATA;
+        }
+        skip_bits_long(gb, 8 * len);
+    }
+
+    if ((ret = set_default_channel_config(avctx, layout_map,
+                                          &tags, channel_config)))
+        return ret;
+
+    if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
+        return ret;
+
+    ep_config = get_bits(gb, 2);
+    if (ep_config) {
+        avpriv_report_missing_feature(avctx,
+                                      "epConfig %d", ep_config);
+        return AVERROR_PATCHWELCOME;
+    }
+    return 0;
+}
+
 /**
  * Decode audio specific configuration; reference: table 1.13.
  *
@@ -877,11 +962,16 @@ static int decode_audio_specific_config(AACContext *ac,
                                             m4ac, m4ac->chan_config)) < 0)
             return ret;
         break;
+    case AOT_ER_AAC_ELD:
+        if ((ret = decode_eld_specific_config(ac, avctx, &gb,
+                                              m4ac, m4ac->chan_config)) < 0)
+            return ret;
+        break;
     default:
-        av_log(avctx, AV_LOG_ERROR,
+        avpriv_report_missing_feature(avctx,
-               "Audio object type %s%d is not supported.\n",
+                                      "Audio object type %s%d",
-               m4ac->sbr == 1 ? "SBR+" : "",
+                                      m4ac->sbr == 1 ? "SBR+" : "",
-               m4ac->object_type);
+                                      m4ac->object_type);
         return AVERROR(ENOSYS);
     }
 
@@ -1047,7 +1137,6 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
     ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0 * 32768.0);
     // window initialization
     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
-    ff_kbd_window_init(ff_aac_kbd_long_512,  4.0, 512);
     ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
     ff_init_ff_sine_windows(10);
     ff_init_ff_sine_windows( 9);
@@ -1117,22 +1206,25 @@ static void decode_ltp(LongTermPrediction *ltp,
 static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
                            GetBitContext *gb)
 {
-    if (get_bits1(gb)) {
+    int aot = ac->oc[1].m4ac.object_type;
-        av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
+    if (aot != AOT_ER_AAC_ELD) {
-        return AVERROR_INVALIDDATA;
+        if (get_bits1(gb)) {
+            av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
+            return AVERROR_INVALIDDATA;
+        }
+        ics->window_sequence[1] = ics->window_sequence[0];
+        ics->window_sequence[0] = get_bits(gb, 2);
+        if (aot == AOT_ER_AAC_LD &&
+            ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
+            av_log(ac->avctx, AV_LOG_ERROR,
+                   "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
+                   "window sequence %d found.\n", ics->window_sequence[0]);
+            ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
+            return AVERROR_INVALIDDATA;
+        }
+        ics->use_kb_window[1]   = ics->use_kb_window[0];
+        ics->use_kb_window[0]   = get_bits1(gb);
     }
-    ics->window_sequence[1] = ics->window_sequence[0];
-    ics->window_sequence[0] = get_bits(gb, 2);
-    if (ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD &&
-        ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
-        av_log(ac->avctx, AV_LOG_ERROR,
-               "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
-               "window sequence %d found.\n", ics->window_sequence[0]);
-        ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
-        return AVERROR_INVALIDDATA;
-    }
-    ics->use_kb_window[1]   = ics->use_kb_window[0];
-    ics->use_kb_window[0]   = get_bits1(gb);
     ics->num_window_groups  = 1;
     ics->group_len[0]       = 1;
     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
@@ -1154,30 +1246,33 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
     } else {
         ics->max_sfb               = get_bits(gb, 6);
         ics->num_windows           = 1;
-        if (ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD) {
+        if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
             ics->swb_offset        =     ff_swb_offset_512[ac->oc[1].m4ac.sampling_index];
             ics->num_swb           =    ff_aac_num_swb_512[ac->oc[1].m4ac.sampling_index];
+            ics->tns_max_bands     =  ff_tns_max_bands_512[ac->oc[1].m4ac.sampling_index];
             if (!ics->num_swb || !ics->swb_offset)
                 return AVERROR_BUG;
         } else {
             ics->swb_offset        =    ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
             ics->num_swb           =   ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
+            ics->tns_max_bands     = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
+        }
+        if (aot != AOT_ER_AAC_ELD) {
+            ics->predictor_present     = get_bits1(gb);
+            ics->predictor_reset_group = 0;
         }
-        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
-        ics->predictor_present     = get_bits1(gb);
-        ics->predictor_reset_group = 0;
         if (ics->predictor_present) {
-            if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
+            if (aot == AOT_AAC_MAIN) {
                 if (decode_prediction(ac, ics, gb)) {
                     goto fail;
                 }
-            } else if (ac->oc[1].m4ac.object_type == AOT_AAC_LC ||
+            } else if (aot == AOT_AAC_LC ||
-                       ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC) {
+                       aot == AOT_ER_AAC_LC) {
                 av_log(ac->avctx, AV_LOG_ERROR,
                        "Prediction is not allowed in AAC-LC.\n");
                 goto fail;
             } else {
-                if (ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD) {
+                if (aot == AOT_ER_AAC_LD) {
                     av_log(ac->avctx, AV_LOG_ERROR,
                            "LTP in ER AAC LD not yet implemented.\n");
                     return AVERROR_PATCHWELCOME;
@@ -1332,12 +1427,12 @@ static int decode_pulses(Pulse *pulse, GetBitContext *gb,
         return -1;
     pulse->pos[0]    = swb_offset[pulse_swb];
     pulse->pos[0]   += get_bits(gb, 5);
-    if (pulse->pos[0] > 1023)
+    if (pulse->pos[0] >= swb_offset[num_swb])
         return -1;
     pulse->amp[0]    = get_bits(gb, 4);
     for (i = 1; i < pulse->num_pulse; i++) {
         pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
-        if (pulse->pos[i] > 1023)
+        if (pulse->pos[i] >= swb_offset[num_swb])
             return -1;
         pulse->amp[i] = get_bits(gb, 4);
     }
@@ -1802,9 +1897,15 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
     TemporalNoiseShaping    *tns = &sce->tns;
     IndividualChannelStream *ics = &sce->ics;
     float *out = sce->coeffs;
-    int global_gain, er_syntax, pulse_present = 0;
+    int global_gain, eld_syntax, er_syntax, pulse_present = 0;
     int ret;
 
+    eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
+    er_syntax  = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
+                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
+                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
+                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
+
     /* This assignment is to silence a GCC warning about the variable being used
      * uninitialized when in fact it always is.
      */
@@ -1825,11 +1926,8 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
         return ret;
 
     pulse_present = 0;
-    er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
-                ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
-                ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD;
     if (!scale_flag) {
-        if ((pulse_present = get_bits1(gb))) {
+        if (!eld_syntax && (pulse_present = get_bits1(gb))) {
             if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
                 av_log(ac->avctx, AV_LOG_ERROR,
                        "Pulse tool not allowed in eight short sequence.\n");
@@ -1845,7 +1943,7 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
         if (tns->present && !er_syntax)
             if (decode_tns(ac, tns, gb, ics) < 0)
                 return AVERROR_INVALIDDATA;
-        if (get_bits1(gb)) {
+        if (!eld_syntax && get_bits1(gb)) {
             avpriv_request_sample(ac->avctx, "SSR");
             return AVERROR_PATCHWELCOME;
         }
@@ -1945,8 +2043,9 @@ static void apply_intensity_stereo(AACContext *ac,
 static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
 {
     int i, ret, common_window, ms_present = 0;
+    int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
 
-    common_window = get_bits1(gb);
+    common_window = eld_syntax || get_bits1(gb);
     if (common_window) {
         if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
             return AVERROR_INVALIDDATA;
@@ -2188,10 +2287,12 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
         } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
             ac->oc[1].m4ac.sbr = 1;
             ac->oc[1].m4ac.ps = 1;
+            ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
             output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
                              ac->oc[1].status, 1);
         } else {
             ac->oc[1].m4ac.sbr = 1;
+            ac->avctx->profile = FF_PROFILE_AAC_HE;
         }
         res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
         break;
@@ -2431,19 +2532,81 @@ static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
     float *in    = sce->coeffs;
     float *out   = sce->ret;
     float *saved = sce->saved;
-    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_512 : ff_sine_512;
     float *buf  = ac->buf_mdct;
 
     // imdct
     ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
 
     // window overlapping
-    ac->fdsp.vector_fmul_window(out, saved, buf, lwindow_prev, 256);
+    if (ics->use_kb_window[1]) {
+        // AAC LD uses a low overlap sine window instead of a KBD window
+        memcpy(out, saved, 192 * sizeof(float));
+        ac->fdsp.vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64);
+        memcpy(                     out + 320, buf + 64, 192 * sizeof(float));
+    } else {
+        ac->fdsp.vector_fmul_window(out, saved, buf, ff_sine_512, 256);
+    }
 
     // buffer update
     memcpy(saved, buf + 256, 256 * sizeof(float));
 }
 
+static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
+{
+    float *in    = sce->coeffs;
+    float *out   = sce->ret;
+    float *saved = sce->saved;
+    const float *const window = ff_aac_eld_window;
+    float *buf  = ac->buf_mdct;
+    int i;
+    const int n  = 512;
+    const int n2 = n >> 1;
+    const int n4 = n >> 2;
+
+    // Inverse transform, mapped to the conventional IMDCT by
+    // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
+    // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
+    // International Conference on Audio, Language and Image Processing, ICALIP 2008.
+    // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
+    for (i = 0; i < n2; i+=2) {
+        float temp;
+        temp =  in[i    ]; in[i    ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
+        temp = -in[i + 1]; in[i + 1] =  in[n - 2 - i]; in[n - 2 - i] = temp;
+    }
+    ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
+    for (i = 0; i < n; i+=2) {
+        buf[i] = -buf[i];
+    }
+    // Like with the regular IMDCT at this point we still have the middle half
+    // of a transform but with even symmetry on the left and odd symmetry on
+    // the right
+
+    // window overlapping
+    // The spec says to use samples [0..511] but the reference decoder uses
+    // samples [128..639].
+    for (i = n4; i < n2; i ++) {
+        out[i - n4] =    buf[n2 - 1 - i]       * window[i       - n4] +
+                       saved[      i + n2]     * window[i +   n - n4] +
+                      -saved[  n + n2 - 1 - i] * window[i + 2*n - n4] +
+                      -saved[2*n + n2 + i]     * window[i + 3*n - n4];
+    }
+    for (i = 0; i < n2; i ++) {
+        out[n4 + i] =    buf[i]               * window[i + n2       - n4] +
+                      -saved[      n - 1 - i] * window[i + n2 +   n - n4] +
+                      -saved[  n + i]         * window[i + n2 + 2*n - n4] +
+                       saved[2*n + n - 1 - i] * window[i + n2 + 3*n - n4];
+    }
+    for (i = 0; i < n4; i ++) {
+        out[n2 + n4 + i] =    buf[      i + n2]     * window[i +   n - n4] +
+                           -saved[      n2 - 1 - i] * window[i + 2*n - n4] +
+                           -saved[  n + n2 + i]     * window[i + 3*n - n4];
+    }
+
+    // buffer update
+    memmove(saved + n, saved, 2 * n * sizeof(float));
+    memcpy( saved,       buf,     n * sizeof(float));
+}
+
 /**
  * Apply dependent channel coupling (applied before IMDCT).
  *
@@ -2541,10 +2704,16 @@ static void spectral_to_sample(AACContext *ac)
 {
     int i, type;
     void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
-    if (ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD)
+    switch (ac->oc[1].m4ac.object_type) {
+    case AOT_ER_AAC_LD:
         imdct_and_window = imdct_and_windowing_ld;
-    else
+        break;
+    case AOT_ER_AAC_ELD:
+        imdct_and_window = imdct_and_windowing_eld;
+        break;
+    default:
         imdct_and_window = ac->imdct_and_windowing;
+    }
     for (type = 3; type >= 0; type--) {
         for (i = 0; i < MAX_ELEM_ID; i++) {
             ChannelElement *che = ac->che[type][i];
@@ -2654,8 +2823,9 @@ static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
     int err, i;
     int samples = 1024;
     int chan_config = ac->oc[1].m4ac.chan_config;
+    int aot = ac->oc[1].m4ac.object_type;
 
-    if (ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD)
+    if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
         samples >>= 1;
 
     ac->frame = data;
@@ -2663,6 +2833,10 @@ static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
     if ((err = frame_configure_elements(avctx)) < 0)
         return err;
 
+    // The FF_PROFILE_AAC_* defines are all object_type - 1
+    // This may lead to an undefined profile being signaled
+    ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
+
     ac->tags_mapped = 0;
 
     if (chan_config < 0 || chan_config >= 8) {
@@ -2679,7 +2853,8 @@ static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
                    elem_type, elem_id);
             return AVERROR_INVALIDDATA;
         }
-        skip_bits(gb, 4);
+        if (aot != AOT_ER_AAC_ELD)
+            skip_bits(gb, 4);
         switch (elem_type) {
         case TYPE_SCE:
             err = decode_ics(ac, &che->ch[0], gb, 0, 0);
@@ -2698,6 +2873,7 @@ static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
     spectral_to_sample(ac);
 
     ac->frame->nb_samples = samples;
+    ac->frame->sample_rate = avctx->sample_rate;
     *got_frame_ptr = 1;
 
     skip_bits_long(gb, get_bits_left(gb));
@@ -2731,6 +2907,10 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
     if ((err = frame_configure_elements(avctx)) < 0)
         goto fail;
 
+    // The FF_PROFILE_AAC_* defines are all object_type - 1
+    // This may lead to an undefined profile being signaled
+    ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
+
     ac->tags_mapped = 0;
     // parse
     while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
@@ -2828,22 +3008,6 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
 
     multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
     samples <<= multiplier;
-    /* for dual-mono audio (SCE + SCE) */
-    is_dmono = ac->dmono_mode && sce_count == 2 &&
-               ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
-
-    if (samples)
-        ac->frame->nb_samples = samples;
-    else
-        av_frame_unref(ac->frame);
-    *got_frame_ptr = !!samples;
-
-    if (is_dmono) {
-        if (ac->dmono_mode == 1)
-            ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
-        else if (ac->dmono_mode == 2)
-            ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
-    }
 
     if (ac->oc[1].status && audio_found) {
         avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
@@ -2857,6 +3021,25 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
         if (side && side_size>=4)
             AV_WL32(side, 2*AV_RL32(side));
     }
+
+    *got_frame_ptr = !!samples;
+    if (samples) {
+        ac->frame->nb_samples = samples;
+        ac->frame->sample_rate = avctx->sample_rate;
+    } else
+        av_frame_unref(ac->frame);
+    *got_frame_ptr = !!samples;
+
+    /* for dual-mono audio (SCE + SCE) */
+    is_dmono = ac->dmono_mode && sce_count == 2 &&
+               ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
+    if (is_dmono) {
+        if (ac->dmono_mode == 1)
+            ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
+        else if (ac->dmono_mode == 2)
+            ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
+    }
+
     return 0;
 fail:
     pop_output_configuration(ac);
@@ -2915,6 +3098,7 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
     case AOT_ER_AAC_LC:
     case AOT_ER_AAC_LTP:
     case AOT_ER_AAC_LD:
+    case AOT_ER_AAC_ELD:
         err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
         break;
     default:

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/aacps.c

@@ -323,14 +323,15 @@ static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[8
 }
 
 /** Split one subband into 6 subsubbands with a complex filter */
-static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2], const float (*filter)[8][2], int len)
+static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2],
+                       TABLE_CONST float (*filter)[8][2], int len)
 {
     int i;
     int N = 8;
     LOCAL_ALIGNED_16(float, temp, [8], [2]);
 
     for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(temp, in, filter, 1, N);
+        dsp->hybrid_analysis(temp, in, (const float (*)[8][2]) filter, 1, N);
         out[0][i][0] = temp[6][0];
         out[0][i][1] = temp[6][1];
         out[1][i][0] = temp[7][0];
@@ -346,12 +347,14 @@ static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2], c
     }
 }
 
-static void hybrid4_8_12_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2], const float (*filter)[8][2], int N, int len)
+static void hybrid4_8_12_cx(PSDSPContext *dsp,
+                            float (*in)[2], float (*out)[32][2],
+                            TABLE_CONST float (*filter)[8][2], int N, int len)
 {
     int i;
 
     for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(out[0] + i, in, filter, 32, N);
+        dsp->hybrid_analysis(out[0] + i, in, (const float (*)[8][2]) filter, 32, N);
     }
 }
 
@@ -430,6 +433,7 @@ static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
 #define DECAY_SLOPE      0.05f
 /// Number of frequency bands that can be addressed by the parameter index, b(k)
 static const int   NR_PAR_BANDS[]      = { 20, 34 };
+static const int   NR_IPDOPD_BANDS[]   = { 11, 17 };
 /// Number of frequency bands that can be addressed by the sub subband index, k
 static const int   NR_BANDS[]          = { 71, 91 };
 /// Start frequency band for the all-pass filter decay slope
@@ -686,7 +690,8 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
             memcpy(ap_delay[k][m],   ap_delay[k][m]+numQMFSlots,           5*sizeof(ap_delay[k][m][0]));
         }
         ps->dsp.decorrelate(out[k], delay[k] + PS_MAX_DELAY - 2, ap_delay[k],
-                            phi_fract[is34][k], Q_fract_allpass[is34][k],
+                            phi_fract[is34][k],
+                            (const float (*)[2]) Q_fract_allpass[is34][k],
                             transient_gain[b], g_decay_slope, nL - n0);
     }
     for (; k < SHORT_DELAY_BAND[is34]; k++) {
@@ -764,7 +769,7 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
     int8_t (*ipd_mapped)[PS_MAX_NR_IIDICC] = ipd_mapped_buf;
     int8_t (*opd_mapped)[PS_MAX_NR_IIDICC] = opd_mapped_buf;
     const int8_t *k_to_i = is34 ? k_to_i_34 : k_to_i_20;
-    const float (*H_LUT)[8][4] = (PS_BASELINE || ps->icc_mode < 3) ? HA : HB;
+    TABLE_CONST float (*H_LUT)[8][4] = (PS_BASELINE || ps->icc_mode < 3) ? HA : HB;
 
     //Remapping
     if (ps->num_env_old) {
@@ -825,7 +830,7 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
             h21 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][2];
             h22 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][3];
 
-            if (!PS_BASELINE && ps->enable_ipdopd && 2*b <= NR_PAR_BANDS[is34]) {
+            if (!PS_BASELINE && ps->enable_ipdopd && b < NR_IPDOPD_BANDS[is34]) {
                 //The spec say says to only run this smoother when enable_ipdopd
                 //is set but the reference decoder appears to run it constantly
                 float h11i, h12i, h21i, h22i;
@@ -915,7 +920,7 @@ int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float
         memset(ps->ap_delay + top, 0, (NR_ALLPASS_BANDS[is34] - top)*sizeof(ps->ap_delay[0]));
 
     hybrid_analysis(&ps->dsp, Lbuf, ps->in_buf, L, is34, len);
-    decorrelation(ps, Rbuf, Lbuf, is34);
+    decorrelation(ps, Rbuf, (const float (*)[32][2]) Lbuf, is34);
     stereo_processing(ps, Lbuf, Rbuf, is34);
     hybrid_synthesis(&ps->dsp, L, Lbuf, is34, len);
     hybrid_synthesis(&ps->dsp, R, Rbuf, is34, len);

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/aacps_tablegen.c

@@ -0,0 +1,93 @@
+/*
+ * Generate a header file for hardcoded Parametric Stereo tables
+ *
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdlib.h>
+#define CONFIG_HARDCODED_TABLES 0
+#include "aacps_tablegen.h"
+#include "tableprint.h"
+
+void write_float_3d_array (const void *p, int b, int c, int d)
+{
+    int i;
+    const float *f = p;
+    for (i = 0; i < b; i++) {
+        printf("{\n");
+        write_float_2d_array(f, c, d);
+        printf("},\n");
+        f += c * d;
+    }
+}
+
+void write_float_4d_array (const void *p, int a, int b, int c, int d)
+{
+    int i;
+    const float *f = p;
+    for (i = 0; i < a; i++) {
+        printf("{\n");
+        write_float_3d_array(f, b, c, d);
+        printf("},\n");
+        f += b * c * d;
+    }
+}
+
+int main(void)
+{
+    ps_tableinit();
+
+    write_fileheader();
+
+    printf("static const float pd_re_smooth[8*8*8] = {\n");
+    write_float_array(pd_re_smooth, 8*8*8);
+    printf("};\n");
+    printf("static const float pd_im_smooth[8*8*8] = {\n");
+    write_float_array(pd_im_smooth, 8*8*8);
+    printf("};\n");
+
+    printf("static const float HA[46][8][4] = {\n");
+    write_float_3d_array(HA, 46, 8, 4);
+    printf("};\n");
+    printf("static const float HB[46][8][4] = {\n");
+    write_float_3d_array(HB, 46, 8, 4);
+    printf("};\n");
+
+    printf("static const DECLARE_ALIGNED(16, float, f20_0_8)[8][8][2] = {\n");
+    write_float_3d_array(f20_0_8, 8, 8, 2);
+    printf("};\n");
+    printf("static const DECLARE_ALIGNED(16, float, f34_0_12)[12][8][2] = {\n");
+    write_float_3d_array(f34_0_12, 12, 8, 2);
+    printf("};\n");
+    printf("static const DECLARE_ALIGNED(16, float, f34_1_8)[8][8][2] = {\n");
+    write_float_3d_array(f34_1_8, 8, 8, 2);
+    printf("};\n");
+    printf("static const DECLARE_ALIGNED(16, float, f34_2_4)[4][8][2] = {\n");
+    write_float_3d_array(f34_2_4, 4, 8, 2);
+    printf("};\n");
+
+    printf("static TABLE_CONST DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2] = {\n");
+    write_float_4d_array(Q_fract_allpass, 2, 50, 3, 2);
+    printf("};\n");
+    printf("static const DECLARE_ALIGNED(16, float, phi_fract)[2][50][2] = {\n");
+    write_float_3d_array(phi_fract, 2, 50, 2);
+    printf("};\n");
+
+    return 0;
+}

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/aacps_tablegen.h

@@ -28,6 +28,7 @@
 
 #if CONFIG_HARDCODED_TABLES
 #define ps_tableinit()
+#define TABLE_CONST const
 #include "libavcodec/aacps_tables.h"
 #else
 #include "libavutil/common.h"
@@ -37,6 +38,7 @@
 #define NR_ALLPASS_BANDS20 30
 #define NR_ALLPASS_BANDS34 50
 #define PS_AP_LINKS 3
+#define TABLE_CONST
 static float pd_re_smooth[8*8*8];
 static float pd_im_smooth[8*8*8];
 static float HA[46][8][4];
@@ -45,7 +47,7 @@ static DECLARE_ALIGNED(16, float, f20_0_8) [ 8][8][2];
 static DECLARE_ALIGNED(16, float, f34_0_12)[12][8][2];
 static DECLARE_ALIGNED(16, float, f34_1_8) [ 8][8][2];
 static DECLARE_ALIGNED(16, float, f34_2_4) [ 4][8][2];
-static DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2];
+static TABLE_CONST DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2];
 static DECLARE_ALIGNED(16, float, phi_fract)[2][50][2];
 
 static const float g0_Q8[] = {

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/aacpsdsp.c

@@ -93,7 +93,7 @@ static void ps_hybrid_synthesis_deint_c(float out[2][38][64],
 
 static void ps_decorrelate_c(float (*out)[2], float (*delay)[2],
                              float (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2],
-                             const float phi_fract[2], float (*Q_fract)[2],
+                             const float phi_fract[2], const float (*Q_fract)[2],
                              const float *transient_gain,
                              float g_decay_slope,
                              int len)

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/aacpsdsp.h

@@ -38,7 +38,7 @@ typedef struct PSDSPContext {
                                    int i, int len);
     void (*decorrelate)(float (*out)[2], float (*delay)[2],
                         float (*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2],
-                        const float phi_fract[2], float (*Q_fract)[2],
+                        const float phi_fract[2], const float (*Q_fract)[2],
                         const float *transient_gain,
                         float g_decay_slope,
                         int len);

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/aacpsy.c

@@ -0,0 +1,965 @@
+/*
+ * AAC encoder psychoacoustic model
+ * Copyright (C) 2008 Konstantin Shishkov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder psychoacoustic model
+ */
+
+#include "libavutil/attributes.h"
+#include "libavutil/libm.h"
+
+#include "avcodec.h"
+#include "aactab.h"
+#include "psymodel.h"
+
+/***********************************
+ *              TODOs:
+ * try other bitrate controlling mechanism (maybe use ratecontrol.c?)
+ * control quality for quality-based output
+ **********************************/
+
+/**
+ * constants for 3GPP AAC psychoacoustic model
+ * @{
+ */
+#define PSY_3GPP_THR_SPREAD_HI   1.5f // spreading factor for low-to-hi threshold spreading  (15 dB/Bark)
+#define PSY_3GPP_THR_SPREAD_LOW  3.0f // spreading factor for hi-to-low threshold spreading  (30 dB/Bark)
+/* spreading factor for low-to-hi energy spreading, long block, > 22kbps/channel (20dB/Bark) */
+#define PSY_3GPP_EN_SPREAD_HI_L1 2.0f
+/* spreading factor for low-to-hi energy spreading, long block, <= 22kbps/channel (15dB/Bark) */
+#define PSY_3GPP_EN_SPREAD_HI_L2 1.5f
+/* spreading factor for low-to-hi energy spreading, short block (15 dB/Bark) */
+#define PSY_3GPP_EN_SPREAD_HI_S  1.5f
+/* spreading factor for hi-to-low energy spreading, long block (30dB/Bark) */
+#define PSY_3GPP_EN_SPREAD_LOW_L 3.0f
+/* spreading factor for hi-to-low energy spreading, short block (20dB/Bark) */
+#define PSY_3GPP_EN_SPREAD_LOW_S 2.0f
+
+#define PSY_3GPP_RPEMIN      0.01f
+#define PSY_3GPP_RPELEV      2.0f
+
+#define PSY_3GPP_C1          3.0f           /* log2(8) */
+#define PSY_3GPP_C2          1.3219281f     /* log2(2.5) */
+#define PSY_3GPP_C3          0.55935729f    /* 1 - C2 / C1 */
+
+#define PSY_SNR_1DB          7.9432821e-1f  /* -1dB */
+#define PSY_SNR_25DB         3.1622776e-3f  /* -25dB */
+
+#define PSY_3GPP_SAVE_SLOPE_L  -0.46666667f
+#define PSY_3GPP_SAVE_SLOPE_S  -0.36363637f
+#define PSY_3GPP_SAVE_ADD_L    -0.84285712f
+#define PSY_3GPP_SAVE_ADD_S    -0.75f
+#define PSY_3GPP_SPEND_SLOPE_L  0.66666669f
+#define PSY_3GPP_SPEND_SLOPE_S  0.81818181f
+#define PSY_3GPP_SPEND_ADD_L   -0.35f
+#define PSY_3GPP_SPEND_ADD_S   -0.26111111f
+#define PSY_3GPP_CLIP_LO_L      0.2f
+#define PSY_3GPP_CLIP_LO_S      0.2f
+#define PSY_3GPP_CLIP_HI_L      0.95f
+#define PSY_3GPP_CLIP_HI_S      0.75f
+
+#define PSY_3GPP_AH_THR_LONG    0.5f
+#define PSY_3GPP_AH_THR_SHORT   0.63f
+
+enum {
+    PSY_3GPP_AH_NONE,
+    PSY_3GPP_AH_INACTIVE,
+    PSY_3GPP_AH_ACTIVE
+};
+
+#define PSY_3GPP_BITS_TO_PE(bits) ((bits) * 1.18f)
+
+/* LAME psy model constants */
+#define PSY_LAME_FIR_LEN 21         ///< LAME psy model FIR order
+#define AAC_BLOCK_SIZE_LONG 1024    ///< long block size
+#define AAC_BLOCK_SIZE_SHORT 128    ///< short block size
+#define AAC_NUM_BLOCKS_SHORT 8      ///< number of blocks in a short sequence
+#define PSY_LAME_NUM_SUBBLOCKS 3    ///< Number of sub-blocks in each short block
+
+/**
+ * @}
+ */
+
+/**
+ * information for single band used by 3GPP TS26.403-inspired psychoacoustic model
+ */
+typedef struct AacPsyBand{
+    float energy;       ///< band energy
+    float thr;          ///< energy threshold
+    float thr_quiet;    ///< threshold in quiet
+    float nz_lines;     ///< number of non-zero spectral lines
+    float active_lines; ///< number of active spectral lines
+    float pe;           ///< perceptual entropy
+    float pe_const;     ///< constant part of the PE calculation
+    float norm_fac;     ///< normalization factor for linearization
+    int   avoid_holes;  ///< hole avoidance flag
+}AacPsyBand;
+
+/**
+ * single/pair channel context for psychoacoustic model
+ */
+typedef struct AacPsyChannel{
+    AacPsyBand band[128];               ///< bands information
+    AacPsyBand prev_band[128];          ///< bands information from the previous frame
+
+    float       win_energy;              ///< sliding average of channel energy
+    float       iir_state[2];            ///< hi-pass IIR filter state
+    uint8_t     next_grouping;           ///< stored grouping scheme for the next frame (in case of 8 short window sequence)
+    enum WindowSequence next_window_seq; ///< window sequence to be used in the next frame
+    /* LAME psy model specific members */
+    float attack_threshold;              ///< attack threshold for this channel
+    float prev_energy_subshort[AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS];
+    int   prev_attack;                   ///< attack value for the last short block in the previous sequence
+}AacPsyChannel;
+
+/**
+ * psychoacoustic model frame type-dependent coefficients
+ */
+typedef struct AacPsyCoeffs{
+    float ath;           ///< absolute threshold of hearing per bands
+    float barks;         ///< Bark value for each spectral band in long frame
+    float spread_low[2]; ///< spreading factor for low-to-high threshold spreading in long frame
+    float spread_hi [2]; ///< spreading factor for high-to-low threshold spreading in long frame
+    float min_snr;       ///< minimal SNR
+}AacPsyCoeffs;
+
+/**
+ * 3GPP TS26.403-inspired psychoacoustic model specific data
+ */
+typedef struct AacPsyContext{
+    int chan_bitrate;     ///< bitrate per channel
+    int frame_bits;       ///< average bits per frame
+    int fill_level;       ///< bit reservoir fill level
+    struct {
+        float min;        ///< minimum allowed PE for bit factor calculation
+        float max;        ///< maximum allowed PE for bit factor calculation
+        float previous;   ///< allowed PE of the previous frame
+        float correction; ///< PE correction factor
+    } pe;
+    AacPsyCoeffs psy_coef[2][64];
+    AacPsyChannel *ch;
+}AacPsyContext;
+
+/**
+ * LAME psy model preset struct
+ */
+typedef struct {
+    int   quality;  ///< Quality to map the rest of the vaules to.
+     /* This is overloaded to be both kbps per channel in ABR mode, and
+      * requested quality in constant quality mode.
+      */
+    float st_lrm;   ///< short threshold for L, R, and M channels
+} PsyLamePreset;
+
+/**
+ * LAME psy model preset table for ABR
+ */
+static const PsyLamePreset psy_abr_map[] = {
+/* TODO: Tuning. These were taken from LAME. */
+/* kbps/ch st_lrm   */
+    {  8,  6.60},
+    { 16,  6.60},
+    { 24,  6.60},
+    { 32,  6.60},
+    { 40,  6.60},
+    { 48,  6.60},
+    { 56,  6.60},
+    { 64,  6.40},
+    { 80,  6.00},
+    { 96,  5.60},
+    {112,  5.20},
+    {128,  5.20},
+    {160,  5.20}
+};
+
+/**
+* LAME psy model preset table for constant quality
+*/
+static const PsyLamePreset psy_vbr_map[] = {
+/* vbr_q  st_lrm    */
+    { 0,  4.20},
+    { 1,  4.20},
+    { 2,  4.20},
+    { 3,  4.20},
+    { 4,  4.20},
+    { 5,  4.20},
+    { 6,  4.20},
+    { 7,  4.20},
+    { 8,  4.20},
+    { 9,  4.20},
+    {10,  4.20}
+};
+
+/**
+ * LAME psy model FIR coefficient table
+ */
+static const float psy_fir_coeffs[] = {
+    -8.65163e-18 * 2, -0.00851586 * 2, -6.74764e-18 * 2, 0.0209036 * 2,
+    -3.36639e-17 * 2, -0.0438162 * 2,  -1.54175e-17 * 2, 0.0931738 * 2,
+    -5.52212e-17 * 2, -0.313819 * 2
+};
+
+#if ARCH_MIPS
+#   include "mips/aacpsy_mips.h"
+#endif /* ARCH_MIPS */
+
+/**
+ * Calculate the ABR attack threshold from the above LAME psymodel table.
+ */
+static float lame_calc_attack_threshold(int bitrate)
+{
+    /* Assume max bitrate to start with */
+    int lower_range = 12, upper_range = 12;
+    int lower_range_kbps = psy_abr_map[12].quality;
+    int upper_range_kbps = psy_abr_map[12].quality;
+    int i;
+
+    /* Determine which bitrates the value specified falls between.
+     * If the loop ends without breaking our above assumption of 320kbps was correct.
+     */
+    for (i = 1; i < 13; i++) {
+        if (FFMAX(bitrate, psy_abr_map[i].quality) != bitrate) {
+            upper_range = i;
+            upper_range_kbps = psy_abr_map[i    ].quality;
+            lower_range = i - 1;
+            lower_range_kbps = psy_abr_map[i - 1].quality;
+            break; /* Upper range found */
+        }
+    }
+
+    /* Determine which range the value specified is closer to */
+    if ((upper_range_kbps - bitrate) > (bitrate - lower_range_kbps))
+        return psy_abr_map[lower_range].st_lrm;
+    return psy_abr_map[upper_range].st_lrm;
+}
+
+/**
+ * LAME psy model specific initialization
+ */
+static av_cold void lame_window_init(AacPsyContext *ctx, AVCodecContext *avctx)
+{
+    int i, j;
+
+    for (i = 0; i < avctx->channels; i++) {
+        AacPsyChannel *pch = &ctx->ch[i];
+
+        if (avctx->flags & CODEC_FLAG_QSCALE)
+            pch->attack_threshold = psy_vbr_map[avctx->global_quality / FF_QP2LAMBDA].st_lrm;
+        else
+            pch->attack_threshold = lame_calc_attack_threshold(avctx->bit_rate / avctx->channels / 1000);
+
+        for (j = 0; j < AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS; j++)
+            pch->prev_energy_subshort[j] = 10.0f;
+    }
+}
+
+/**
+ * Calculate Bark value for given line.
+ */
+static av_cold float calc_bark(float f)
+{
+    return 13.3f * atanf(0.00076f * f) + 3.5f * atanf((f / 7500.0f) * (f / 7500.0f));
+}
+
+#define ATH_ADD 4
+/**
+ * Calculate ATH value for given frequency.
+ * Borrowed from Lame.
+ */
+static av_cold float ath(float f, float add)
+{
+    f /= 1000.0f;
+    return    3.64 * pow(f, -0.8)
+            - 6.8  * exp(-0.6  * (f - 3.4) * (f - 3.4))
+            + 6.0  * exp(-0.15 * (f - 8.7) * (f - 8.7))
+            + (0.6 + 0.04 * add) * 0.001 * f * f * f * f;
+}
+
+static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
+    AacPsyContext *pctx;
+    float bark;
+    int i, j, g, start;
+    float prev, minscale, minath, minsnr, pe_min;
+    const int chan_bitrate = ctx->avctx->bit_rate / ctx->avctx->channels;
+    const int bandwidth    = ctx->avctx->cutoff ? ctx->avctx->cutoff : AAC_CUTOFF(ctx->avctx);
+    const float num_bark   = calc_bark((float)bandwidth);
+
+    ctx->model_priv_data = av_mallocz(sizeof(AacPsyContext));
+    pctx = (AacPsyContext*) ctx->model_priv_data;
+
+    pctx->chan_bitrate = chan_bitrate;
+    pctx->frame_bits   = chan_bitrate * AAC_BLOCK_SIZE_LONG / ctx->avctx->sample_rate;
+    pctx->pe.min       =  8.0f * AAC_BLOCK_SIZE_LONG * bandwidth / (ctx->avctx->sample_rate * 2.0f);
+    pctx->pe.max       = 12.0f * AAC_BLOCK_SIZE_LONG * bandwidth / (ctx->avctx->sample_rate * 2.0f);
+    ctx->bitres.size   = 6144 - pctx->frame_bits;
+    ctx->bitres.size  -= ctx->bitres.size % 8;
+    pctx->fill_level   = ctx->bitres.size;
+    minath = ath(3410, ATH_ADD);
+    for (j = 0; j < 2; j++) {
+        AacPsyCoeffs *coeffs = pctx->psy_coef[j];
+        const uint8_t *band_sizes = ctx->bands[j];
+        float line_to_frequency = ctx->avctx->sample_rate / (j ? 256.f : 2048.0f);
+        float avg_chan_bits = chan_bitrate * (j ? 128.0f : 1024.0f) / ctx->avctx->sample_rate;
+        /* reference encoder uses 2.4% here instead of 60% like the spec says */
+        float bark_pe = 0.024f * PSY_3GPP_BITS_TO_PE(avg_chan_bits) / num_bark;
+        float en_spread_low = j ? PSY_3GPP_EN_SPREAD_LOW_S : PSY_3GPP_EN_SPREAD_LOW_L;
+        /* High energy spreading for long blocks <= 22kbps/channel and short blocks are the same. */
+        float en_spread_hi  = (j || (chan_bitrate <= 22.0f)) ? PSY_3GPP_EN_SPREAD_HI_S : PSY_3GPP_EN_SPREAD_HI_L1;
+
+        i = 0;
+        prev = 0.0;
+        for (g = 0; g < ctx->num_bands[j]; g++) {
+            i += band_sizes[g];
+            bark = calc_bark((i-1) * line_to_frequency);
+            coeffs[g].barks = (bark + prev) / 2.0;
+            prev = bark;
+        }
+        for (g = 0; g < ctx->num_bands[j] - 1; g++) {
+            AacPsyCoeffs *coeff = &coeffs[g];
+            float bark_width = coeffs[g+1].barks - coeffs->barks;
+            coeff->spread_low[0] = pow(10.0, -bark_width * PSY_3GPP_THR_SPREAD_LOW);
+            coeff->spread_hi [0] = pow(10.0, -bark_width * PSY_3GPP_THR_SPREAD_HI);
+            coeff->spread_low[1] = pow(10.0, -bark_width * en_spread_low);
+            coeff->spread_hi [1] = pow(10.0, -bark_width * en_spread_hi);
+            pe_min = bark_pe * bark_width;
+            minsnr = exp2(pe_min / band_sizes[g]) - 1.5f;
+            coeff->min_snr = av_clipf(1.0f / minsnr, PSY_SNR_25DB, PSY_SNR_1DB);
+        }
+        start = 0;
+        for (g = 0; g < ctx->num_bands[j]; g++) {
+            minscale = ath(start * line_to_frequency, ATH_ADD);
+            for (i = 1; i < band_sizes[g]; i++)
+                minscale = FFMIN(minscale, ath((start + i) * line_to_frequency, ATH_ADD));
+            coeffs[g].ath = minscale - minath;
+            start += band_sizes[g];
+        }
+    }
+
+    pctx->ch = av_mallocz(sizeof(AacPsyChannel) * ctx->avctx->channels);
+
+    lame_window_init(pctx, ctx->avctx);
+
+    return 0;
+}
+
+/**
+ * IIR filter used in block switching decision
+ */
+static float iir_filter(int in, float state[2])
+{
+    float ret;
+
+    ret = 0.7548f * (in - state[0]) + 0.5095f * state[1];
+    state[0] = in;
+    state[1] = ret;
+    return ret;
+}
+
+/**
+ * window grouping information stored as bits (0 - new group, 1 - group continues)
+ */
+static const uint8_t window_grouping[9] = {
+    0xB6, 0x6C, 0xD8, 0xB2, 0x66, 0xC6, 0x96, 0x36, 0x36
+};
+
+/**
+ * Tell encoder which window types to use.
+ * @see 3GPP TS26.403 5.4.1 "Blockswitching"
+ */
+static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
+                                                 const int16_t *audio,
+                                                 const int16_t *la,
+                                                 int channel, int prev_type)
+{
+    int i, j;
+    int br               = ctx->avctx->bit_rate / ctx->avctx->channels;
+    int attack_ratio     = br <= 16000 ? 18 : 10;
+    AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data;
+    AacPsyChannel *pch  = &pctx->ch[channel];
+    uint8_t grouping     = 0;
+    int next_type        = pch->next_window_seq;
+    FFPsyWindowInfo wi  = { { 0 } };
+
+    if (la) {
+        float s[8], v;
+        int switch_to_eight = 0;
+        float sum = 0.0, sum2 = 0.0;
+        int attack_n = 0;
+        int stay_short = 0;
+        for (i = 0; i < 8; i++) {
+            for (j = 0; j < 128; j++) {
+                v = iir_filter(la[i*128+j], pch->iir_state);
+                sum += v*v;
+            }
+            s[i]  = sum;
+            sum2 += sum;
+        }
+        for (i = 0; i < 8; i++) {
+            if (s[i] > pch->win_energy * attack_ratio) {
+                attack_n        = i + 1;
+                switch_to_eight = 1;
+                break;
+            }
+        }
+        pch->win_energy = pch->win_energy*7/8 + sum2/64;
+
+        wi.window_type[1] = prev_type;
+        switch (prev_type) {
+        case ONLY_LONG_SEQUENCE:
+            wi.window_type[0] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE;
+            next_type = switch_to_eight ? EIGHT_SHORT_SEQUENCE : ONLY_LONG_SEQUENCE;
+            break;
+        case LONG_START_SEQUENCE:
+            wi.window_type[0] = EIGHT_SHORT_SEQUENCE;
+            grouping = pch->next_grouping;
+            next_type = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE;
+            break;
+        case LONG_STOP_SEQUENCE:
+            wi.window_type[0] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE;
+            next_type = switch_to_eight ? EIGHT_SHORT_SEQUENCE : ONLY_LONG_SEQUENCE;
+            break;
+        case EIGHT_SHORT_SEQUENCE:
+            stay_short = next_type == EIGHT_SHORT_SEQUENCE || switch_to_eight;
+            wi.window_type[0] = stay_short ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE;
+            grouping = next_type == EIGHT_SHORT_SEQUENCE ? pch->next_grouping : 0;
+            next_type = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE;
+            break;
+        }
+
+        pch->next_grouping = window_grouping[attack_n];
+        pch->next_window_seq = next_type;
+    } else {
+        for (i = 0; i < 3; i++)
+            wi.window_type[i] = prev_type;
+        grouping = (prev_type == EIGHT_SHORT_SEQUENCE) ? window_grouping[0] : 0;
+    }
+
+    wi.window_shape   = 1;
+    if (wi.window_type[0] != EIGHT_SHORT_SEQUENCE) {
+        wi.num_windows = 1;
+        wi.grouping[0] = 1;
+    } else {
+        int lastgrp = 0;
+        wi.num_windows = 8;
+        for (i = 0; i < 8; i++) {
+            if (!((grouping >> i) & 1))
+                lastgrp = i;
+            wi.grouping[lastgrp]++;
+        }
+    }
+
+    return wi;
+}
+
+/* 5.6.1.2 "Calculation of Bit Demand" */
+static int calc_bit_demand(AacPsyContext *ctx, float pe, int bits, int size,
+                           int short_window)
+{
+    const float bitsave_slope  = short_window ? PSY_3GPP_SAVE_SLOPE_S  : PSY_3GPP_SAVE_SLOPE_L;
+    const float bitsave_add    = short_window ? PSY_3GPP_SAVE_ADD_S    : PSY_3GPP_SAVE_ADD_L;
+    const float bitspend_slope = short_window ? PSY_3GPP_SPEND_SLOPE_S : PSY_3GPP_SPEND_SLOPE_L;
+    const float bitspend_add   = short_window ? PSY_3GPP_SPEND_ADD_S   : PSY_3GPP_SPEND_ADD_L;
+    const float clip_low       = short_window ? PSY_3GPP_CLIP_LO_S     : PSY_3GPP_CLIP_LO_L;
+    const float clip_high      = short_window ? PSY_3GPP_CLIP_HI_S     : PSY_3GPP_CLIP_HI_L;
+    float clipped_pe, bit_save, bit_spend, bit_factor, fill_level;
+
+    ctx->fill_level += ctx->frame_bits - bits;
+    ctx->fill_level  = av_clip(ctx->fill_level, 0, size);
+    fill_level = av_clipf((float)ctx->fill_level / size, clip_low, clip_high);
+    clipped_pe = av_clipf(pe, ctx->pe.min, ctx->pe.max);
+    bit_save   = (fill_level + bitsave_add) * bitsave_slope;
+    assert(bit_save <= 0.3f && bit_save >= -0.05000001f);
+    bit_spend  = (fill_level + bitspend_add) * bitspend_slope;
+    assert(bit_spend <= 0.5f && bit_spend >= -0.1f);
+    /* The bit factor graph in the spec is obviously incorrect.
+     *      bit_spend + ((bit_spend - bit_spend))...
+     * The reference encoder subtracts everything from 1, but also seems incorrect.
+     *      1 - bit_save + ((bit_spend + bit_save))...
+     * Hopefully below is correct.
+     */
+    bit_factor = 1.0f - bit_save + ((bit_spend - bit_save) / (ctx->pe.max - ctx->pe.min)) * (clipped_pe - ctx->pe.min);
+    /* NOTE: The reference encoder attempts to center pe max/min around the current pe. */
+    ctx->pe.max = FFMAX(pe, ctx->pe.max);
+    ctx->pe.min = FFMIN(pe, ctx->pe.min);
+
+    return FFMIN(ctx->frame_bits * bit_factor, ctx->frame_bits + size - bits);
+}
+
+static float calc_pe_3gpp(AacPsyBand *band)
+{
+    float pe, a;
+
+    band->pe           = 0.0f;
+    band->pe_const     = 0.0f;
+    band->active_lines = 0.0f;
+    if (band->energy > band->thr) {
+        a  = log2f(band->energy);
+        pe = a - log2f(band->thr);
+        band->active_lines = band->nz_lines;
+        if (pe < PSY_3GPP_C1) {
+            pe = pe * PSY_3GPP_C3 + PSY_3GPP_C2;
+            a  = a  * PSY_3GPP_C3 + PSY_3GPP_C2;
+            band->active_lines *= PSY_3GPP_C3;
+        }
+        band->pe       = pe * band->nz_lines;
+        band->pe_const = a  * band->nz_lines;
+    }
+
+    return band->pe;
+}
+
+static float calc_reduction_3gpp(float a, float desired_pe, float pe,
+                                 float active_lines)
+{
+    float thr_avg, reduction;
+
+    if(active_lines == 0.0)
+        return 0;
+
+    thr_avg   = exp2f((a - pe) / (4.0f * active_lines));
+    reduction = exp2f((a - desired_pe) / (4.0f * active_lines)) - thr_avg;
+
+    return FFMAX(reduction, 0.0f);
+}
+
+static float calc_reduced_thr_3gpp(AacPsyBand *band, float min_snr,
+                                   float reduction)
+{
+    float thr = band->thr;
+
+    if (band->energy > thr) {
+        thr = sqrtf(thr);
+        thr = sqrtf(thr) + reduction;
+        thr *= thr;
+        thr *= thr;
+
+        /* This deviates from the 3GPP spec to match the reference encoder.
+         * It performs min(thr_reduced, max(thr, energy/min_snr)) only for bands
+         * that have hole avoidance on (active or inactive). It always reduces the
+         * threshold of bands with hole avoidance off.
+         */
+        if (thr > band->energy * min_snr && band->avoid_holes != PSY_3GPP_AH_NONE) {
+            thr = FFMAX(band->thr, band->energy * min_snr);
+            band->avoid_holes = PSY_3GPP_AH_ACTIVE;
+        }
+    }
+
+    return thr;
+}
+
+#ifndef calc_thr_3gpp
+static void calc_thr_3gpp(const FFPsyWindowInfo *wi, const int num_bands, AacPsyChannel *pch,
+                          const uint8_t *band_sizes, const float *coefs)
+{
+    int i, w, g;
+    int start = 0;
+    for (w = 0; w < wi->num_windows*16; w += 16) {
+        for (g = 0; g < num_bands; g++) {
+            AacPsyBand *band = &pch->band[w+g];
+
+            float form_factor = 0.0f;
+            float Temp;
+            band->energy = 0.0f;
+            for (i = 0; i < band_sizes[g]; i++) {
+                band->energy += coefs[start+i] * coefs[start+i];
+                form_factor  += sqrtf(fabs(coefs[start+i]));
+            }
+            Temp = band->energy > 0 ? sqrtf((float)band_sizes[g] / band->energy) : 0;
+            band->thr      = band->energy * 0.001258925f;
+            band->nz_lines = form_factor * sqrtf(Temp);
+
+            start += band_sizes[g];
+        }
+    }
+}
+#endif /* calc_thr_3gpp */
+
+#ifndef psy_hp_filter
+static void psy_hp_filter(const float *firbuf, float *hpfsmpl, const float *psy_fir_coeffs)
+{
+    int i, j;
+    for (i = 0; i < AAC_BLOCK_SIZE_LONG; i++) {
+        float sum1, sum2;
+        sum1 = firbuf[i + (PSY_LAME_FIR_LEN - 1) / 2];
+        sum2 = 0.0;
+        for (j = 0; j < ((PSY_LAME_FIR_LEN - 1) / 2) - 1; j += 2) {
+            sum1 += psy_fir_coeffs[j] * (firbuf[i + j] + firbuf[i + PSY_LAME_FIR_LEN - j]);
+            sum2 += psy_fir_coeffs[j + 1] * (firbuf[i + j + 1] + firbuf[i + PSY_LAME_FIR_LEN - j - 1]);
+        }
+        /* NOTE: The LAME psymodel expects it's input in the range -32768 to 32768.
+         *       Tuning this for normalized floats would be difficult. */
+        hpfsmpl[i] = (sum1 + sum2) * 32768.0f;
+    }
+}
+#endif /* psy_hp_filter */
+
+/**
+ * Calculate band thresholds as suggested in 3GPP TS26.403
+ */
+static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
+                                     const float *coefs, const FFPsyWindowInfo *wi)
+{
+    AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data;
+    AacPsyChannel *pch  = &pctx->ch[channel];
+    int i, w, g;
+    float desired_bits, desired_pe, delta_pe, reduction= NAN, spread_en[128] = {0};
+    float a = 0.0f, active_lines = 0.0f, norm_fac = 0.0f;
+    float pe = pctx->chan_bitrate > 32000 ? 0.0f : FFMAX(50.0f, 100.0f - pctx->chan_bitrate * 100.0f / 32000.0f);
+    const int      num_bands   = ctx->num_bands[wi->num_windows == 8];
+    const uint8_t *band_sizes  = ctx->bands[wi->num_windows == 8];
+    AacPsyCoeffs  *coeffs      = pctx->psy_coef[wi->num_windows == 8];
+    const float avoid_hole_thr = wi->num_windows == 8 ? PSY_3GPP_AH_THR_SHORT : PSY_3GPP_AH_THR_LONG;
+
+    //calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
+    calc_thr_3gpp(wi, num_bands, pch, band_sizes, coefs);
+
+    //modify thresholds and energies - spread, threshold in quiet, pre-echo control
+    for (w = 0; w < wi->num_windows*16; w += 16) {
+        AacPsyBand *bands = &pch->band[w];
+
+        /* 5.4.2.3 "Spreading" & 5.4.3 "Spread Energy Calculation" */
+        spread_en[0] = bands[0].energy;
+        for (g = 1; g < num_bands; g++) {
+            bands[g].thr   = FFMAX(bands[g].thr,    bands[g-1].thr * coeffs[g].spread_hi[0]);
+            spread_en[w+g] = FFMAX(bands[g].energy, spread_en[w+g-1] * coeffs[g].spread_hi[1]);
+        }
+        for (g = num_bands - 2; g >= 0; g--) {
+            bands[g].thr   = FFMAX(bands[g].thr,   bands[g+1].thr * coeffs[g].spread_low[0]);
+            spread_en[w+g] = FFMAX(spread_en[w+g], spread_en[w+g+1] * coeffs[g].spread_low[1]);
+        }
+        //5.4.2.4 "Threshold in quiet"
+        for (g = 0; g < num_bands; g++) {
+            AacPsyBand *band = &bands[g];
+
+            band->thr_quiet = band->thr = FFMAX(band->thr, coeffs[g].ath);
+            //5.4.2.5 "Pre-echo control"
+            if (!(wi->window_type[0] == LONG_STOP_SEQUENCE || (wi->window_type[1] == LONG_START_SEQUENCE && !w)))
+                band->thr = FFMAX(PSY_3GPP_RPEMIN*band->thr, FFMIN(band->thr,
+                                  PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
+
+            /* 5.6.1.3.1 "Preparatory steps of the perceptual entropy calculation" */
+            pe += calc_pe_3gpp(band);
+            a  += band->pe_const;
+            active_lines += band->active_lines;
+
+            /* 5.6.1.3.3 "Selection of the bands for avoidance of holes" */
+            if (spread_en[w+g] * avoid_hole_thr > band->energy || coeffs[g].min_snr > 1.0f)
+                band->avoid_holes = PSY_3GPP_AH_NONE;
+            else
+                band->avoid_holes = PSY_3GPP_AH_INACTIVE;
+        }
+    }
+
+    /* 5.6.1.3.2 "Calculation of the desired perceptual entropy" */
+    ctx->ch[channel].entropy = pe;
+    desired_bits = calc_bit_demand(pctx, pe, ctx->bitres.bits, ctx->bitres.size, wi->num_windows == 8);
+    desired_pe = PSY_3GPP_BITS_TO_PE(desired_bits);
+    /* NOTE: PE correction is kept simple. During initial testing it had very
+     *       little effect on the final bitrate. Probably a good idea to come
+     *       back and do more testing later.
+     */
+    if (ctx->bitres.bits > 0)
+        desired_pe *= av_clipf(pctx->pe.previous / PSY_3GPP_BITS_TO_PE(ctx->bitres.bits),
+                               0.85f, 1.15f);
+    pctx->pe.previous = PSY_3GPP_BITS_TO_PE(desired_bits);
+
+    if (desired_pe < pe) {
+        /* 5.6.1.3.4 "First Estimation of the reduction value" */
+        for (w = 0; w < wi->num_windows*16; w += 16) {
+            reduction = calc_reduction_3gpp(a, desired_pe, pe, active_lines);
+            pe = 0.0f;
+            a  = 0.0f;
+            active_lines = 0.0f;
+            for (g = 0; g < num_bands; g++) {
+                AacPsyBand *band = &pch->band[w+g];
+
+                band->thr = calc_reduced_thr_3gpp(band, coeffs[g].min_snr, reduction);
+                /* recalculate PE */
+                pe += calc_pe_3gpp(band);
+                a  += band->pe_const;
+                active_lines += band->active_lines;
+            }
+        }
+
+        /* 5.6.1.3.5 "Second Estimation of the reduction value" */
+        for (i = 0; i < 2; i++) {
+            float pe_no_ah = 0.0f, desired_pe_no_ah;
+            active_lines = a = 0.0f;
+            for (w = 0; w < wi->num_windows*16; w += 16) {
+                for (g = 0; g < num_bands; g++) {
+                    AacPsyBand *band = &pch->band[w+g];
+
+                    if (band->avoid_holes != PSY_3GPP_AH_ACTIVE) {
+                        pe_no_ah += band->pe;
+                        a        += band->pe_const;
+                        active_lines += band->active_lines;
+                    }
+                }
+            }
+            desired_pe_no_ah = FFMAX(desired_pe - (pe - pe_no_ah), 0.0f);
+            if (active_lines > 0.0f)
+                reduction += calc_reduction_3gpp(a, desired_pe_no_ah, pe_no_ah, active_lines);
+
+            pe = 0.0f;
+            for (w = 0; w < wi->num_windows*16; w += 16) {
+                for (g = 0; g < num_bands; g++) {
+                    AacPsyBand *band = &pch->band[w+g];
+
+                    if (active_lines > 0.0f)
+                        band->thr = calc_reduced_thr_3gpp(band, coeffs[g].min_snr, reduction);
+                    pe += calc_pe_3gpp(band);
+                    band->norm_fac = band->active_lines / band->thr;
+                    norm_fac += band->norm_fac;
+                }
+            }
+            delta_pe = desired_pe - pe;
+            if (fabs(delta_pe) > 0.05f * desired_pe)
+                break;
+        }
+
+        if (pe < 1.15f * desired_pe) {
+            /* 6.6.1.3.6 "Final threshold modification by linearization" */
+            norm_fac = 1.0f / norm_fac;
+            for (w = 0; w < wi->num_windows*16; w += 16) {
+                for (g = 0; g < num_bands; g++) {
+                    AacPsyBand *band = &pch->band[w+g];
+
+                    if (band->active_lines > 0.5f) {
+                        float delta_sfb_pe = band->norm_fac * norm_fac * delta_pe;
+                        float thr = band->thr;
+
+                        thr *= exp2f(delta_sfb_pe / band->active_lines);
+                        if (thr > coeffs[g].min_snr * band->energy && band->avoid_holes == PSY_3GPP_AH_INACTIVE)
+                            thr = FFMAX(band->thr, coeffs[g].min_snr * band->energy);
+                        band->thr = thr;
+                    }
+                }
+            }
+        } else {
+            /* 5.6.1.3.7 "Further perceptual entropy reduction" */
+            g = num_bands;
+            while (pe > desired_pe && g--) {
+                for (w = 0; w < wi->num_windows*16; w+= 16) {
+                    AacPsyBand *band = &pch->band[w+g];
+                    if (band->avoid_holes != PSY_3GPP_AH_NONE && coeffs[g].min_snr < PSY_SNR_1DB) {
+                        coeffs[g].min_snr = PSY_SNR_1DB;
+                        band->thr = band->energy * PSY_SNR_1DB;
+                        pe += band->active_lines * 1.5f - band->pe;
+                    }
+                }
+            }
+            /* TODO: allow more holes (unused without mid/side) */
+        }
+    }
+
+    for (w = 0; w < wi->num_windows*16; w += 16) {
+        for (g = 0; g < num_bands; g++) {
+            AacPsyBand *band     = &pch->band[w+g];
+            FFPsyBand  *psy_band = &ctx->ch[channel].psy_bands[w+g];
+
+            psy_band->threshold = band->thr;
+            psy_band->energy    = band->energy;
+        }
+    }
+
+    memcpy(pch->prev_band, pch->band, sizeof(pch->band));
+}
+
+static void psy_3gpp_analyze(FFPsyContext *ctx, int channel,
+                                   const float **coeffs, const FFPsyWindowInfo *wi)
+{
+    int ch;
+    FFPsyChannelGroup *group = ff_psy_find_group(ctx, channel);
+
+    for (ch = 0; ch < group->num_ch; ch++)
+        psy_3gpp_analyze_channel(ctx, channel + ch, coeffs[ch], &wi[ch]);
+}
+
+static av_cold void psy_3gpp_end(FFPsyContext *apc)
+{
+    AacPsyContext *pctx = (AacPsyContext*) apc->model_priv_data;
+    av_freep(&pctx->ch);
+    av_freep(&apc->model_priv_data);
+}
+
+static void lame_apply_block_type(AacPsyChannel *ctx, FFPsyWindowInfo *wi, int uselongblock)
+{
+    int blocktype = ONLY_LONG_SEQUENCE;
+    if (uselongblock) {
+        if (ctx->next_window_seq == EIGHT_SHORT_SEQUENCE)
+            blocktype = LONG_STOP_SEQUENCE;
+    } else {
+        blocktype = EIGHT_SHORT_SEQUENCE;
+        if (ctx->next_window_seq == ONLY_LONG_SEQUENCE)
+            ctx->next_window_seq = LONG_START_SEQUENCE;
+        if (ctx->next_window_seq == LONG_STOP_SEQUENCE)
+            ctx->next_window_seq = EIGHT_SHORT_SEQUENCE;
+    }
+
+    wi->window_type[0] = ctx->next_window_seq;
+    ctx->next_window_seq = blocktype;
+}
+
+static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
+                                       const float *la, int channel, int prev_type)
+{
+    AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data;
+    AacPsyChannel *pch  = &pctx->ch[channel];
+    int grouping     = 0;
+    int uselongblock = 1;
+    int attacks[AAC_NUM_BLOCKS_SHORT + 1] = { 0 };
+    int i;
+    FFPsyWindowInfo wi = { { 0 } };
+
+    if (la) {
+        float hpfsmpl[AAC_BLOCK_SIZE_LONG];
+        float const *pf = hpfsmpl;
+        float attack_intensity[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS];
+        float energy_subshort[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS];
+        float energy_short[AAC_NUM_BLOCKS_SHORT + 1] = { 0 };
+        const float *firbuf = la + (AAC_BLOCK_SIZE_SHORT/4 - PSY_LAME_FIR_LEN);
+        int att_sum = 0;
+
+        /* LAME comment: apply high pass filter of fs/4 */
+        psy_hp_filter(firbuf, hpfsmpl, psy_fir_coeffs);
+
+        /* Calculate the energies of each sub-shortblock */
+        for (i = 0; i < PSY_LAME_NUM_SUBBLOCKS; i++) {
+            energy_subshort[i] = pch->prev_energy_subshort[i + ((AAC_NUM_BLOCKS_SHORT - 1) * PSY_LAME_NUM_SUBBLOCKS)];
+            assert(pch->prev_energy_subshort[i + ((AAC_NUM_BLOCKS_SHORT - 2) * PSY_LAME_NUM_SUBBLOCKS + 1)] > 0);
+            attack_intensity[i] = energy_subshort[i] / pch->prev_energy_subshort[i + ((AAC_NUM_BLOCKS_SHORT - 2) * PSY_LAME_NUM_SUBBLOCKS + 1)];
+            energy_short[0] += energy_subshort[i];
+        }
+
+        for (i = 0; i < AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS; i++) {
+            float const *const pfe = pf + AAC_BLOCK_SIZE_LONG / (AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS);
+            float p = 1.0f;
+            for (; pf < pfe; pf++)
+                p = FFMAX(p, fabsf(*pf));
+            pch->prev_energy_subshort[i] = energy_subshort[i + PSY_LAME_NUM_SUBBLOCKS] = p;
+            energy_short[1 + i / PSY_LAME_NUM_SUBBLOCKS] += p;
+            /* NOTE: The indexes below are [i + 3 - 2] in the LAME source.
+             *       Obviously the 3 and 2 have some significance, or this would be just [i + 1]
+             *       (which is what we use here). What the 3 stands for is ambiguous, as it is both
+             *       number of short blocks, and the number of sub-short blocks.
+             *       It seems that LAME is comparing each sub-block to sub-block + 1 in the
+             *       previous block.
+             */
+            if (p > energy_subshort[i + 1])
+                p = p / energy_subshort[i + 1];
+            else if (energy_subshort[i + 1] > p * 10.0f)
+                p = energy_subshort[i + 1] / (p * 10.0f);
+            else
+                p = 0.0;
+            attack_intensity[i + PSY_LAME_NUM_SUBBLOCKS] = p;
+        }
+
+        /* compare energy between sub-short blocks */
+        for (i = 0; i < (AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS; i++)
+            if (!attacks[i / PSY_LAME_NUM_SUBBLOCKS])
+                if (attack_intensity[i] > pch->attack_threshold)
+                    attacks[i / PSY_LAME_NUM_SUBBLOCKS] = (i % PSY_LAME_NUM_SUBBLOCKS) + 1;
+
+        /* should have energy change between short blocks, in order to avoid periodic signals */
+        /* Good samples to show the effect are Trumpet test songs */
+        /* GB: tuned (1) to avoid too many short blocks for test sample TRUMPET */
+        /* RH: tuned (2) to let enough short blocks through for test sample FSOL and SNAPS */
+        for (i = 1; i < AAC_NUM_BLOCKS_SHORT + 1; i++) {
+            float const u = energy_short[i - 1];
+            float const v = energy_short[i];
+            float const m = FFMAX(u, v);
+            if (m < 40000) {                          /* (2) */
+                if (u < 1.7f * v && v < 1.7f * u) {   /* (1) */
+                    if (i == 1 && attacks[0] < attacks[i])
+                        attacks[0] = 0;
+                    attacks[i] = 0;
+                }
+            }
+            att_sum += attacks[i];
+        }
+
+        if (attacks[0] <= pch->prev_attack)
+            attacks[0] = 0;
+
+        att_sum += attacks[0];
+        /* 3 below indicates the previous attack happened in the last sub-block of the previous sequence */
+        if (pch->prev_attack == 3 || att_sum) {
+            uselongblock = 0;
+
+            for (i = 1; i < AAC_NUM_BLOCKS_SHORT + 1; i++)
+                if (attacks[i] && attacks[i-1])
+                    attacks[i] = 0;
+        }
+    } else {
+        /* We have no lookahead info, so just use same type as the previous sequence. */
+        uselongblock = !(prev_type == EIGHT_SHORT_SEQUENCE);
+    }
+
+    lame_apply_block_type(pch, &wi, uselongblock);
+
+    wi.window_type[1] = prev_type;
+    if (wi.window_type[0] != EIGHT_SHORT_SEQUENCE) {
+        wi.num_windows  = 1;
+        wi.grouping[0]  = 1;
+        if (wi.window_type[0] == LONG_START_SEQUENCE)
+            wi.window_shape = 0;
+        else
+            wi.window_shape = 1;
+    } else {
+        int lastgrp = 0;
+
+        wi.num_windows = 8;
+        wi.window_shape = 0;
+        for (i = 0; i < 8; i++) {
+            if (!((pch->next_grouping >> i) & 1))
+                lastgrp = i;
+            wi.grouping[lastgrp]++;
+        }
+    }
+
+    /* Determine grouping, based on the location of the first attack, and save for
+     * the next frame.
+     * FIXME: Move this to analysis.
+     * TODO: Tune groupings depending on attack location
+     * TODO: Handle more than one attack in a group
+     */
+    for (i = 0; i < 9; i++) {
+        if (attacks[i]) {
+            grouping = i;
+            break;
+        }
+    }
+    pch->next_grouping = window_grouping[grouping];
+
+    pch->prev_attack = attacks[8];
+
+    return wi;
+}
+
+const FFPsyModel ff_aac_psy_model =
+{
+    .name    = "3GPP TS 26.403-inspired model",
+    .init    = psy_3gpp_init,
+    .window  = psy_lame_window,
+    .analyze = psy_3gpp_analyze,
+    .end     = psy_3gpp_end,
+};

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/aactab.c

@@ -34,7 +34,6 @@
 #include <stdint.h>
 
 DECLARE_ALIGNED(32, float,  ff_aac_kbd_long_1024)[1024];
-DECLARE_ALIGNED(32, float,  ff_aac_kbd_long_512 )[512];
 DECLARE_ALIGNED(32, float,  ff_aac_kbd_short_128)[128];
 
 const uint8_t ff_aac_num_swb_1024[] = {
@@ -1237,7 +1236,494 @@ const uint8_t ff_tns_max_bands_1024[] = {
     31, 31, 34, 40, 42, 51, 46, 46, 42, 42, 42, 39, 39
 };
 
+const uint8_t ff_tns_max_bands_512[] = {
+    0, 0, 0, 31, 32, 37, 31, 31, 0, 0, 0, 0, 0
+};
+
 const uint8_t ff_tns_max_bands_128[] = {
     9, 9, 10, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14
 };
 // @}
+
+const DECLARE_ALIGNED(32, float, ff_aac_eld_window)[1920] = {
+     0.00338834,  0.00567745,  0.00847677,  0.01172641,
+     0.01532555,  0.01917664,  0.02318809,  0.02729259,
+     0.03144503,  0.03560261,  0.03972499,  0.04379783,
+     0.04783094,  0.05183357,  0.05581342,  0.05977723,
+     0.06373173,  0.06768364,  0.07163937,  0.07559976,
+     0.07956096,  0.08352024,  0.08747623,  0.09143035,
+     0.09538618,  0.09934771,  0.10331917,  0.10730456,
+     0.11130697,  0.11532867,  0.11937133,  0.12343922,
+     0.12753911,  0.13167705,  0.13585812,  0.14008529,
+     0.14435986,  0.14868291,  0.15305531,  0.15747594,
+     0.16194193,  0.16645070,  0.17099991,  0.17558633,
+     0.18020600,  0.18485548,  0.18953191,  0.19423322,
+     0.19895800,  0.20370512,  0.20847374,  0.21326312,
+     0.21807244,  0.22290083,  0.22774742,  0.23261210,
+     0.23749542,  0.24239767,  0.24731889,  0.25225887,
+     0.25721719,  0.26219330,  0.26718648,  0.27219630,
+     0.27722262,  0.28226514,  0.28732336,  0.29239628,
+     0.29748247,  0.30258055,  0.30768914,  0.31280508,
+     0.31792385,  0.32304172,  0.32815579,  0.33326397,
+     0.33836470,  0.34345661,  0.34853868,  0.35361188,
+     0.35867865,  0.36374072,  0.36879900,  0.37385347,
+     0.37890349,  0.38394836,  0.38898730,  0.39401912,
+     0.39904236,  0.40405575,  0.40905820,  0.41404819,
+     0.41902398,  0.42398423,  0.42892805,  0.43385441,
+     0.43876210,  0.44365014,  0.44851786,  0.45336632,
+     0.45819759,  0.46301302,  0.46781309,  0.47259722,
+     0.47736435,  0.48211365,  0.48684450,  0.49155594,
+     0.49624679,  0.50091636,  0.50556440,  0.51019132,
+     0.51479771,  0.51938391,  0.52394998,  0.52849587,
+     0.53302151,  0.53752680,  0.54201160,  0.54647575,
+     0.55091916,  0.55534181,  0.55974376,  0.56412513,
+     0.56848615,  0.57282710,  0.57714834,  0.58145030,
+     0.58492489,  0.58918511,  0.59342326,  0.59763936,
+     0.60183347,  0.60600561,  0.61015581,  0.61428412,
+     0.61839056,  0.62247517,  0.62653799,  0.63057912,
+     0.63459872,  0.63859697,  0.64257403,  0.64653001,
+     0.65046495,  0.65437887,  0.65827181,  0.66214383,
+     0.66599499,  0.66982535,  0.67363499,  0.67742394,
+     0.68119219,  0.68493972,  0.68866653,  0.69237258,
+     0.69605778,  0.69972207,  0.70336537,  0.70698758,
+     0.71058862,  0.71416837,  0.71772674,  0.72126361,
+     0.72477889,  0.72827246,  0.73174419,  0.73519392,
+     0.73862141,  0.74202643,  0.74540874,  0.74876817,
+     0.75210458,  0.75541785,  0.75870785,  0.76197437,
+     0.76521709,  0.76843570,  0.77162988,  0.77479939,
+     0.77794403,  0.78106359,  0.78415789,  0.78722670,
+     0.79026979,  0.79328694,  0.79627791,  0.79924244,
+     0.80218027,  0.80509112,  0.80797472,  0.81083081,
+     0.81365915,  0.81645949,  0.81923160,  0.82197528,
+     0.82469037,  0.82737673,  0.83003419,  0.83266262,
+     0.83526186,  0.83783176,  0.84037217,  0.84288297,
+     0.84536401,  0.84781517,  0.85023632,  0.85262739,
+     0.85498836,  0.85731921,  0.85961993,  0.86189052,
+     0.86413101,  0.86634140,  0.86852173,  0.87067211,
+     0.87279275,  0.87488384,  0.87694559,  0.87897824,
+     0.88098206,  0.88295729,  0.88490423,  0.88682332,
+     0.88871519,  0.89058048,  0.89241983,  0.89423391,
+     0.89602338,  0.89778893,  0.89953126,  0.90125142,
+     0.90295086,  0.90463104,  0.90629341,  0.90793946,
+     0.90957067,  0.91118856,  0.91279464,  0.91439073,
+     0.91597898,  0.91756153,  0.91914049,  0.92071690,
+     0.92229070,  0.92386182,  0.92542993,  0.92698946,
+     0.92852960,  0.93003929,  0.93150727,  0.93291739,
+     0.93424863,  0.93547974,  0.93658982,  0.93756587,
+     0.93894072,  0.93922780,  0.93955477,  0.93991290,
+     0.94029104,  0.94067794,  0.94106258,  0.94144084,
+     0.94181549,  0.94218963,  0.94256628,  0.94294662,
+     0.94332998,  0.94371562,  0.94410280,  0.94449122,
+     0.94488106,  0.94527249,  0.94566568,  0.94606074,
+     0.94645772,  0.94685665,  0.94725759,  0.94766054,
+     0.94806547,  0.94847234,  0.94888115,  0.94929190,
+     0.94970469,  0.95011960,  0.95053672,  0.95095604,
+     0.95137751,  0.95180105,  0.95222658,  0.95265413,
+     0.95308380,  0.95351571,  0.95394994,  0.95438653,
+     0.95482538,  0.95526643,  0.95570958,  0.95615486,
+     0.95660234,  0.95705214,  0.95750433,  0.95795892,
+     0.95841582,  0.95887493,  0.95933616,  0.95979949,
+     0.96026500,  0.96073277,  0.96120286,  0.96167526,
+     0.96214986,  0.96262655,  0.96310522,  0.96358586,
+     0.96406853,  0.96455330,  0.96504026,  0.96552936,
+     0.96602051,  0.96651360,  0.96700850,  0.96750520,
+     0.96800376,  0.96850424,  0.96900670,  0.96951112,
+     0.97001738,  0.97052533,  0.97103488,  0.97154597,
+     0.97205867,  0.97257304,  0.97308915,  0.97360694,
+     0.97412631,  0.97464711,  0.97516923,  0.97569262,
+     0.97621735,  0.97674350,  0.97727111,  0.97780016,
+     0.97833051,  0.97886205,  0.97939463,  0.97992823,
+     0.98046291,  0.98099875,  0.98153580,  0.98207405,
+     0.98261337,  0.98315364,  0.98369474,  0.98423664,
+     0.98477941,  0.98532311,  0.98586780,  0.98641348,
+     0.98696003,  0.98750734,  0.98805530,  0.98860389,
+     0.98915320,  0.98970328,  0.99025423,  0.99080602,
+     0.99135855,  0.99191171,  0.99246541,  0.99301962,
+     0.99357443,  0.99412992,  0.99468617,  0.99524320,
+     0.99580092,  0.99635926,  0.99691814,  0.99747748,
+     0.99803721,  0.99859725,  0.99915752,  0.99971793,
+     1.00028215,  1.00084319,  1.00140472,  1.00196665,
+     1.00252889,  1.00309139,  1.00365404,  1.00421679,
+     1.00477954,  1.00534221,  1.00590474,  1.00646713,
+     1.00702945,  1.00759179,  1.00815424,  1.00871678,
+     1.00927930,  1.00984169,  1.01040384,  1.01096575,
+     1.01152747,  1.01208910,  1.01265070,  1.01321226,
+     1.01377365,  1.01433478,  1.01489551,  1.01545584,
+     1.01601582,  1.01657553,  1.01713502,  1.01769427,
+     1.01825316,  1.01881154,  1.01936929,  1.01992639,
+     1.02048289,  1.02103888,  1.02159441,  1.02214945,
+     1.02270387,  1.02325751,  1.02381025,  1.02436204,
+     1.02491295,  1.02546304,  1.02601238,  1.02656092,
+     1.02710853,  1.02765508,  1.02820041,  1.02874449,
+     1.02928737,  1.02982913,  1.03036981,  1.03090937,
+     1.03144768,  1.03198460,  1.03252000,  1.03305384,
+     1.03358617,  1.03411707,  1.03464659,  1.03517470,
+     1.03570128,  1.03622620,  1.03674934,  1.03727066,
+     1.03779024,  1.03830815,  1.03882446,  1.03933914,
+     1.03985206,  1.04036312,  1.04087217,  1.04137920,
+     1.04188428,  1.04238748,  1.04288888,  1.04338845,
+     1.04388610,  1.04438170,  1.04487515,  1.04536645,
+     1.04585569,  1.04634297,  1.04682838,  1.04731192,
+     1.04779350,  1.04827303,  1.04875042,  1.04922568,
+     1.04969891,  1.05017022,  1.05063974,  1.05110746,
+     1.05157332,  1.05203721,  1.05249907,  1.05295889,
+     1.05341676,  1.05387277,  1.05432700,  1.05477948,
+     1.05523018,  1.05567906,  1.05612608,  1.05657124,
+     1.05701459,  1.05745616,  1.05789601,  1.05833426,
+     1.05877109,  1.05920669,  1.05964125,  1.06007444,
+     1.06050542,  1.06093335,  1.06135746,  1.06177909,
+     1.06220164,  1.06262858,  1.06306309,  1.06350050,
+     1.06392837,  1.06433391,  1.06470443,  1.06502996,
+     1.06481076,  1.06469765,  1.06445004,  1.06408002,
+     1.06361382,  1.06307719,  1.06249453,  1.06188365,
+     1.06125612,  1.06062291,  1.05999418,  1.05937132,
+     1.05874726,  1.05811486,  1.05746728,  1.05680000,
+     1.05611070,  1.05539715,  1.05465735,  1.05389329,
+     1.05311083,  1.05231578,  1.05151372,  1.05070811,
+     1.04990044,  1.04909210,  1.04828434,  1.04747647,
+     1.04666590,  1.04585003,  1.04502628,  1.04419009,
+     1.04333499,  1.04245452,  1.04154244,  1.04059452,
+     1.03960846,  1.03858207,  1.03751326,  1.03640189,
+     1.03524976,  1.03405868,  1.03283047,  1.03156812,
+     1.03027574,  1.02895743,  1.02761717,  1.02625804,
+     1.02488222,  1.02349184,  1.02208892,  1.02067450,
+     1.01924861,  1.01781123,  1.01636229,  1.01490045,
+     1.01342315,  1.01192778,  1.01041175,  1.00887284,
+     1.00730915,  1.00571882,  1.00409996,  1.00245032,
+     1.00076734,  0.99904842,  0.99729101,  0.99549380,
+     0.99365664,  0.99177946,  0.98986234,  0.98791024,
+     0.98593294,  0.98394037,  0.98194226,  0.97994532,
+     0.97795324,  0.97596955,  0.97399748,  0.97203326,
+     0.97006624,  0.96808546,  0.96608018,  0.96404416,
+     0.96197556,  0.95987276,  0.95773420,  0.95556018,
+     0.95335291,  0.95111462,  0.94884764,  0.94655663,
+     0.94424858,  0.94193055,  0.93960953,  0.93729154,
+     0.93498157,  0.93268456,  0.93040503,  0.92813771,
+     0.92586755,  0.92357910,  0.92125731,  0.91889642,
+     0.91649998,  0.91407191,  0.91161623,  0.90913975,
+     0.90665202,  0.90416271,  0.90168115,  0.89920934,
+     0.89674189,  0.89427312,  0.89179743,  0.88931147,
+     0.88681415,  0.88430445,  0.88178141,  0.87924528,
+     0.87669753,  0.87413966,  0.87157318,  0.86899958,
+     0.86642037,  0.86383703,  0.86125106,  0.85866393,
+     0.85604236,  0.85344385,  0.85083093,  0.84820550,
+     0.84556943,  0.84292458,  0.84027278,  0.83761586,
+     0.83495565,  0.83229393,  0.82963243,  0.82697135,
+     0.82430933,  0.82164496,  0.81897669,  0.81630017,
+     0.81360822,  0.81089355,  0.80814924,  0.80537741,
+     0.80258920,  0.79979611,  0.79700954,  0.79423813,
+     0.79148780,  0.78876432,  0.78607290,  0.78340590,
+     0.78074288,  0.77806279,  0.77534514,  0.77258187,
+     0.76977737,  0.76693654,  0.76406441,  0.76116851,
+     0.75825892,  0.75534582,  0.75243924,  0.74954634,
+     0.74667135,  0.74381840,  0.74099145,  0.73819147,
+     0.73541641,  0.73266408,  0.72993193,  0.72720913,
+     0.72447661,  0.72171494,  0.71890515,  0.71603932,
+     0.71312056,  0.71015250,  0.70713900,  0.70409084,
+     0.70102565,  0.69796137,  0.69491556,  0.69189772,
+     0.68890931,  0.68595141,  0.68302498,  0.68012852,
+     0.67725801,  0.67440936,  0.67157841,  0.66876081,
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+     0.00020573,  0.00021399,  0.00022233,  0.00023076,
+     0.00023924,  0.00024773,  0.00025621,  0.00026462,
+     0.00027293,  0.00028108,  0.00028904,  0.00029675,
+     0.00030419,  0.00031132,  0.00031810,  0.00032453,
+     0.00033061,  0.00033632,  0.00034169,  0.00034672,
+     0.00035142,  0.00035580,  0.00035988,  0.00036369,
+     0.00036723,  0.00037053,  0.00037361,  0.00037647,
+     0.00037909,  0.00038145,  0.00038352,  0.00038527,
+     0.00038663,  0.00038757,  0.00038801,  0.00038790,
+     0.00038717,  0.00038572,  0.00038350,  0.00038044,
+     0.00037651,  0.00037170,  0.00036597,  0.00035936,
+     0.00035191,  0.00034370,  0.00033480,  0.00032531,
+     0.00031537,  0.00030512,  0.00029470,  0.00028417,
+     0.00027354,  0.00026279,  0.00025191,  0.00024081,
+     0.00022933,  0.00021731,  0.00020458,  0.00019101,
+     0.00017654,  0.00016106,  0.00014452,  0.00012694,
+     0.00010848,  0.00008929,  0.00006953,  0.00004935,
+     0.00002884,  0.00000813, -0.00001268, -0.00003357,
+    -0.00005457, -0.00007574, -0.00009714, -0.00011882,
+    -0.00014082, -0.00016318, -0.00018595, -0.00020912,
+    -0.00023265, -0.00025650, -0.00028060, -0.00030492,
+    -0.00032941, -0.00035400, -0.00037865, -0.00040333,
+    -0.00042804, -0.00045279, -0.00047759, -0.00050243,
+    -0.00052728, -0.00055209, -0.00057685, -0.00060153,
+    -0.00062611, -0.00065056, -0.00067485, -0.00069895,
+    -0.00072287, -0.00074660, -0.00077013, -0.00079345,
+    -0.00081653, -0.00083936, -0.00086192, -0.00088421,
+    -0.00090619, -0.00092786, -0.00094919, -0.00097017,
+    -0.00099077, -0.00101098, -0.00103077, -0.00105012,
+    -0.00106904, -0.00108750, -0.00110549, -0.00112301,
+    -0.00114005, -0.00115660, -0.00117265, -0.00118821,
+    -0.00120325, -0.00121779, -0.00123180, -0.00124528,
+    -0.00125822, -0.00127061, -0.00128243, -0.00129368,
+    -0.00130435, -0.00131445, -0.00132395, -0.00133285,
+    -0.00134113, -0.00134878, -0.00135577, -0.00136215,
+    -0.00136797, -0.00137333, -0.00137834, -0.00138305,
+    -0.00138748, -0.00139163, -0.00139551, -0.00139913,
+    -0.00140249, -0.00140559, -0.00140844, -0.00141102,
+    -0.00141334, -0.00141538, -0.00141714, -0.00141861,
+    -0.00141978, -0.00142064, -0.00142117, -0.00142138,
+    -0.00142125, -0.00142077, -0.00141992, -0.00141870,
+    -0.00141710, -0.00141510, -0.00141268, -0.00140986,
+    -0.00140663, -0.00140301, -0.00139900, -0.00139460,
+    -0.00138981, -0.00138464, -0.00137908, -0.00137313,
+    -0.00136680, -0.00136010, -0.00135301, -0.00134555,
+    -0.00133772, -0.00132952, -0.00132095, -0.00131201,
+    -0.00130272, -0.00129307, -0.00128309, -0.00127277,
+    -0.00126211, -0.00125113, -0.00123981, -0.00122817,
+    -0.00121622, -0.00120397, -0.00119141, -0.00117859,
+    -0.00116552, -0.00115223, -0.00113877, -0.00112517,
+    -0.00111144, -0.00109764, -0.00108377, -0.00106989,
+};

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/aactab.h

@@ -45,8 +45,8 @@
  * @{
  */
 DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_long_1024)[1024];
-DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_long_512 )[512];
 DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_short_128)[128];
+const DECLARE_ALIGNED(32, extern float, ff_aac_eld_window)[1920];
 // @}
 
 /* @name number of scalefactor window bands for long and short transform windows respectively
@@ -75,6 +75,7 @@ extern const uint16_t * const ff_swb_offset_512 [13];
 extern const uint16_t * const ff_swb_offset_128 [13];
 
 extern const uint8_t ff_tns_max_bands_1024[13];
+extern const uint8_t ff_tns_max_bands_512 [13];
 extern const uint8_t ff_tns_max_bands_128 [13];
 
 #endif /* AVCODEC_AACTAB_H */

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/ac3.h

@@ -82,6 +82,38 @@ typedef enum {
     AC3_CHMODE_3F2R
 } AC3ChannelMode;
 
+/** Dolby Surround mode */
+typedef enum AC3DolbySurroundMode {
+    AC3_DSURMOD_NOTINDICATED = 0,
+    AC3_DSURMOD_OFF,
+    AC3_DSURMOD_ON,
+    AC3_DSURMOD_RESERVED
+} AC3DolbySurroundMode;
+
+/** Dolby Surround EX mode */
+typedef enum AC3DolbySurroundEXMode {
+    AC3_DSUREXMOD_NOTINDICATED = 0,
+    AC3_DSUREXMOD_OFF,
+    AC3_DSUREXMOD_ON,
+    AC3_DSUREXMOD_PLIIZ
+} AC3DolbySurroundEXMode;
+
+/** Dolby Headphone mode */
+typedef enum AC3DolbyHeadphoneMode {
+    AC3_DHEADPHONMOD_NOTINDICATED = 0,
+    AC3_DHEADPHONMOD_OFF,
+    AC3_DHEADPHONMOD_ON,
+    AC3_DHEADPHONMOD_RESERVED
+} AC3DolbyHeadphoneMode;
+
+/** Preferred Stereo Downmix mode */
+typedef enum AC3PreferredStereoDownmixMode {
+    AC3_DMIXMOD_NOTINDICATED = 0,
+    AC3_DMIXMOD_LTRT,
+    AC3_DMIXMOD_LORO,
+    AC3_DMIXMOD_DPLII // reserved value in A/52, but used by encoders to indicate DPL2
+} AC3PreferredStereoDownmixMode;
+
 typedef struct AC3BitAllocParameters {
     int sr_code;
     int sr_shift;
@@ -110,6 +142,9 @@ typedef struct AC3HeaderInfo {
     int surround_mix_level;                 ///< Surround mix level index
     uint16_t channel_map;
     int num_blocks;                         ///< number of audio blocks
+#if AV_HAVE_INCOMPATIBLE_LIBAV_ABI
+    int dolby_surround_mode;
+#endif
     /** @} */
 
     /** @name Derived values
@@ -122,6 +157,9 @@ typedef struct AC3HeaderInfo {
     uint16_t frame_size;
     uint64_t channel_layout;
     /** @} */
+#if !AV_HAVE_INCOMPATIBLE_LIBAV_ABI
+    int dolby_surround_mode;
+#endif
 } AC3HeaderInfo;
 
 typedef enum {

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/ac3_parser.c

@@ -47,9 +47,16 @@ static const uint8_t center_levels[4] = { 4, 5, 6, 5 };
 static const uint8_t surround_levels[4] = { 4, 6, 7, 6 };
 
 
-int avpriv_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
+int avpriv_ac3_parse_header2(GetBitContext *gbc, AC3HeaderInfo **phdr)
 {
     int frame_size_code;
+    AC3HeaderInfo *hdr;
+
+    if (!*phdr)
+        *phdr = av_mallocz(sizeof(AC3HeaderInfo));
+    if (!*phdr)
+        return AVERROR(ENOMEM);
+    hdr = *phdr;
 
     memset(hdr, 0, sizeof(*hdr));
 
@@ -68,6 +75,9 @@ int avpriv_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
     hdr->center_mix_level   = 5;  // -4.5dB
     hdr->surround_mix_level = 6;  // -6.0dB
 
+    /* set default dolby surround mode */
+    hdr->dolby_surround_mode = AC3_DSURMOD_NOTINDICATED;
+
     if(hdr->bitstream_id <= 10) {
         /* Normal AC-3 */
         hdr->crc1 = get_bits(gbc, 16);
@@ -85,7 +95,7 @@ int avpriv_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
         hdr->channel_mode = get_bits(gbc, 3);
 
         if(hdr->channel_mode == AC3_CHMODE_STEREO) {
-            skip_bits(gbc, 2); // skip dsurmod
+            hdr->dolby_surround_mode = get_bits(gbc, 2);
         } else {
             if((hdr->channel_mode & 1) && hdr->channel_mode != AC3_CHMODE_MONO)
                 hdr->  center_mix_level =   center_levels[get_bits(gbc, 2)];
@@ -141,6 +151,15 @@ int avpriv_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
     return 0;
 }
 
+int avpriv_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
+{
+    AC3HeaderInfo tmp, *ptmp = &tmp;
+    int ret = avpriv_ac3_parse_header2(gbc, &ptmp);
+
+    memcpy(hdr, ptmp, ((intptr_t)&tmp.channel_layout) - ((intptr_t)&tmp) + sizeof(uint64_t));
+    return ret;
+}
+
 static int ac3_sync(uint64_t state, AACAC3ParseContext *hdr_info,
         int *need_next_header, int *new_frame_start)
 {
@@ -149,11 +168,11 @@ static int ac3_sync(uint64_t state, AACAC3ParseContext *hdr_info,
         uint64_t u64;
         uint8_t  u8[8];
     } tmp = { av_be2ne64(state) };
-    AC3HeaderInfo hdr;
+    AC3HeaderInfo hdr, *phdr = &hdr;
     GetBitContext gbc;
 
     init_get_bits(&gbc, tmp.u8+8-AC3_HEADER_SIZE, 54);
-    err = avpriv_ac3_parse_header(&gbc, &hdr);
+    err = avpriv_ac3_parse_header2(&gbc, &phdr);
 
     if(err < 0)
         return 0;

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/ac3_parser.h

@@ -31,11 +31,14 @@
  * Parse the header up to the lfeon element, which is the first 52 or 54 bits
  * depending on the audio coding mode.
  * @param[in]  gbc BitContext containing the first 54 bits of the frame.
- * @param[out] hdr Pointer to struct where header info is written.
+ * @param[out] hdr Pointer to Pointer to struct where header info is written.
+ *                 will be allocated if NULL
  * @return Returns 0 on success, -1 if there is a sync word mismatch,
  * -2 if the bsid (version) element is invalid, -3 if the fscod (sample rate)
  * element is invalid, or -4 if the frmsizecod (bit rate) element is invalid.
  */
+int avpriv_ac3_parse_header2(GetBitContext *gbc, AC3HeaderInfo **hdr);
+
 int avpriv_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr);
 
 #endif /* AVCODEC_AC3_PARSER_H */

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/ac3dec.c

@@ -29,7 +29,9 @@
 #include <math.h>
 #include <string.h>
 
+#include "libavutil/channel_layout.h"
 #include "libavutil/crc.h"
+#include "libavutil/downmix_info.h"
 #include "libavutil/opt.h"
 #include "libavutil/internal.h"
 #include "internal.h"
@@ -77,6 +79,15 @@ static const float gain_levels[9] = {
     LEVEL_MINUS_9DB
 };
 
+/** Adjustments in dB gain (LFE, +10 to -21 dB) */
+static const float gain_levels_lfe[32] = {
+    3.162275, 2.818382, 2.511886, 2.238719, 1.995261, 1.778278, 1.584893,
+    1.412536, 1.258924, 1.122018, 1.000000, 0.891251, 0.794328, 0.707946,
+    0.630957, 0.562341, 0.501187, 0.446683, 0.398107, 0.354813, 0.316227,
+    0.281838, 0.251188, 0.223872, 0.199526, 0.177828, 0.158489, 0.141253,
+    0.125892, 0.112201, 0.100000, 0.089125
+};
+
 /**
  * Table for default stereo downmixing coefficients
  * reference: Section 7.8.2 Downmixing Into Two Channels
@@ -179,13 +190,20 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
 
     /* allow downmixing to stereo or mono */
-    AV_NOWARN_DEPRECATED(
+#if FF_API_REQUEST_CHANNELS
-    if (avctx->channels > 0 && avctx->request_channels > 0 &&
+FF_DISABLE_DEPRECATION_WARNINGS
-            avctx->request_channels < avctx->channels &&
+    if (avctx->request_channels == 1)
-            avctx->request_channels <= 2) {
+        avctx->request_channel_layout = AV_CH_LAYOUT_MONO;
-        avctx->channels = avctx->request_channels;
+    else if (avctx->request_channels == 2)
-    }
+        avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;
-    );
+FF_ENABLE_DEPRECATION_WARNINGS
+#endif
+    if (avctx->channels > 1 &&
+        avctx->request_channel_layout == AV_CH_LAYOUT_MONO)
+        avctx->channels = 1;
+    else if (avctx->channels > 2 &&
+             avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
+        avctx->channels = 2;
     s->downmixed = 1;
 
     for (i = 0; i < AC3_MAX_CHANNELS; i++) {
@@ -220,12 +238,26 @@ static int ac3_parse_header(AC3DecodeContext *s)
 
     skip_bits(gbc, 2); //skip copyright bit and original bitstream bit
 
-    /* skip the timecodes (or extra bitstream information for Alternate Syntax)
+    /* skip the timecodes or parse the Alternate Bit Stream Syntax */
-       TODO: read & use the xbsi1 downmix levels */
+    if (s->bitstream_id != 6) {
-    if (get_bits1(gbc))
+        if (get_bits1(gbc))
-        skip_bits(gbc, 14); //skip timecode1 / xbsi1
+            skip_bits(gbc, 14); //skip timecode1
-    if (get_bits1(gbc))
+        if (get_bits1(gbc))
-        skip_bits(gbc, 14); //skip timecode2 / xbsi2
+            skip_bits(gbc, 14); //skip timecode2
+    } else {
+        if (get_bits1(gbc)) {
+            s->preferred_downmix       = get_bits(gbc, 2);
+            s->center_mix_level_ltrt   = get_bits(gbc, 3);
+            s->surround_mix_level_ltrt = av_clip(get_bits(gbc, 3), 3, 7);
+            s->center_mix_level        = get_bits(gbc, 3);
+            s->surround_mix_level      = av_clip(get_bits(gbc, 3), 3, 7);
+        }
+        if (get_bits1(gbc)) {
+            s->dolby_surround_ex_mode = get_bits(gbc, 2);
+            s->dolby_headphone_mode   = get_bits(gbc, 2);
+            skip_bits(gbc, 10); // skip adconvtyp (1), xbsi2 (8), encinfo (1)
+        }
+    }
 
     /* skip additional bitstream info */
     if (get_bits1(gbc)) {
@@ -243,18 +275,18 @@ static int ac3_parse_header(AC3DecodeContext *s)
  */
 static int parse_frame_header(AC3DecodeContext *s)
 {
-    AC3HeaderInfo hdr;
+    AC3HeaderInfo hdr, *phdr=&hdr;
     int err;
 
-    err = avpriv_ac3_parse_header(&s->gbc, &hdr);
+    err = avpriv_ac3_parse_header2(&s->gbc, &phdr);
     if (err)
         return err;
 
     /* get decoding parameters from header info */
     s->bit_alloc_params.sr_code     = hdr.sr_code;
+    s->bitstream_id                 = hdr.bitstream_id;
     s->bitstream_mode               = hdr.bitstream_mode;
     s->channel_mode                 = hdr.channel_mode;
-    s->channel_layout               = hdr.channel_layout;
     s->lfe_on                       = hdr.lfe_on;
     s->bit_alloc_params.sr_shift    = hdr.sr_shift;
     s->sample_rate                  = hdr.sample_rate;
@@ -263,11 +295,18 @@ static int parse_frame_header(AC3DecodeContext *s)
     s->fbw_channels                 = s->channels - s->lfe_on;
     s->lfe_ch                       = s->fbw_channels + 1;
     s->frame_size                   = hdr.frame_size;
+    s->preferred_downmix            = AC3_DMIXMOD_NOTINDICATED;
     s->center_mix_level             = hdr.center_mix_level;
+    s->center_mix_level_ltrt        = 4; // -3.0dB
     s->surround_mix_level           = hdr.surround_mix_level;
+    s->surround_mix_level_ltrt      = 4; // -3.0dB
+    s->lfe_mix_level_exists         = 0;
     s->num_blocks                   = hdr.num_blocks;
     s->frame_type                   = hdr.frame_type;
     s->substreamid                  = hdr.substreamid;
+    s->dolby_surround_mode          = hdr.dolby_surround_mode;
+    s->dolby_surround_ex_mode       = AC3_DSUREXMOD_NOTINDICATED;
+    s->dolby_headphone_mode         = AC3_DHEADPHONMOD_NOTINDICATED;
 
     if (s->lfe_on) {
         s->start_freq[s->lfe_ch]     = 0;
@@ -276,7 +315,7 @@ static int parse_frame_header(AC3DecodeContext *s)
         s->channel_in_cpl[s->lfe_ch] = 0;
     }
 
-    if (hdr.bitstream_id <= 10) {
+    if (s->bitstream_id <= 10) {
         s->eac3                  = 0;
         s->snr_offset_strategy   = 2;
         s->block_switch_syntax   = 1;
@@ -491,6 +530,10 @@ static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, ma
             break;
         default: /* 6 to 15 */
             /* Shift mantissa and sign-extend it. */
+            if (bap > 15) {
+                av_log(s->avctx, AV_LOG_ERROR, "bap %d is invalid in plain AC-3\n", bap);
+                bap = 15;
+            }
             mantissa = get_sbits(gbc, quantization_tab[bap]);
             mantissa <<= 24 - quantization_tab[bap];
             break;
@@ -750,8 +793,13 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
     i = !s->channel_mode;
     do {
         if (get_bits1(gbc)) {
-            s->dynamic_range[i] = ((dynamic_range_tab[get_bits(gbc, 8)] - 1.0) *
+            /* Allow asymmetric application of DRC when drc_scale > 1.
-                                  s->drc_scale) + 1.0;
+               Amplification of quiet sounds is enhanced */
+            float range = dynamic_range_tab[get_bits(gbc, 8)];
+            if (range > 1.0 || s->drc_scale <= 1.0)
+                s->dynamic_range[i] = powf(range, s->drc_scale);
+            else
+                s->dynamic_range[i] = range;
         } else if (blk == 0) {
             s->dynamic_range[i] = 1.0f;
         }
@@ -1274,6 +1322,8 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
     int blk, ch, err, ret;
     const uint8_t *channel_map;
     const float *output[AC3_MAX_CHANNELS];
+    enum AVMatrixEncoding matrix_encoding;
+    AVDownmixInfo *downmix_info;
 
     /* copy input buffer to decoder context to avoid reading past the end
        of the buffer, which can be caused by a damaged input stream. */
@@ -1333,6 +1383,8 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
             if (av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2],
                        s->frame_size - 2)) {
                 av_log(avctx, AV_LOG_ERROR, "frame CRC mismatch\n");
+                if (avctx->err_recognition & AV_EF_EXPLODE)
+                    return AVERROR_INVALIDDATA;
                 err = AAC_AC3_PARSE_ERROR_CRC;
             }
         }
@@ -1350,16 +1402,15 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
         s->output_mode  = s->channel_mode;
         if (s->lfe_on)
             s->output_mode |= AC3_OUTPUT_LFEON;
-        AV_NOWARN_DEPRECATED(
+        if (s->channels > 1 &&
-        if (avctx->request_channels > 0 && avctx->request_channels <= 2 &&
+            avctx->request_channel_layout == AV_CH_LAYOUT_MONO) {
-                avctx->request_channels < s->channels) {
+            s->out_channels = 1;
-            s->out_channels = avctx->request_channels;
+            s->output_mode  = AC3_CHMODE_MONO;
-            s->output_mode  = avctx->request_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO;
+        } else if (s->channels > 2 &&
-            s->channel_layout = avpriv_ac3_channel_layout_tab[s->output_mode];
+                   avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
+            s->out_channels = 2;
+            s->output_mode  = AC3_CHMODE_STEREO;
         }
-        );
-        avctx->channels       = s->out_channels;
-        avctx->channel_layout = s->channel_layout;
 
         s->loro_center_mix_level   = gain_levels[s->  center_mix_level];
         s->loro_surround_mix_level = gain_levels[s->surround_mix_level];
@@ -1375,6 +1426,9 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
         return AVERROR_INVALIDDATA;
     }
     avctx->channels = s->out_channels;
+    avctx->channel_layout = avpriv_ac3_channel_layout_tab[s->output_mode & ~AC3_OUTPUT_LFEON];
+    if (s->output_mode & AC3_OUTPUT_LFEON)
+        avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
 
     /* set audio service type based on bitstream mode for AC-3 */
     avctx->audio_service_type = s->bitstream_mode;
@@ -1418,6 +1472,62 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
     for (ch = 0; ch < s->out_channels; ch++)
         memcpy(s->output[ch], output[ch], sizeof(**output) * AC3_BLOCK_SIZE);
 
+    /*
+     * AVMatrixEncoding
+     *
+     * Check whether the input layout is compatible, and make sure we're not
+     * downmixing (else the matrix encoding is no longer applicable).
+     */
+    matrix_encoding = AV_MATRIX_ENCODING_NONE;
+    if (s->channel_mode == AC3_CHMODE_STEREO &&
+        s->channel_mode == (s->output_mode & ~AC3_OUTPUT_LFEON)) {
+        if (s->dolby_surround_mode == AC3_DSURMOD_ON)
+            matrix_encoding = AV_MATRIX_ENCODING_DOLBY;
+        else if (s->dolby_headphone_mode == AC3_DHEADPHONMOD_ON)
+            matrix_encoding = AV_MATRIX_ENCODING_DOLBYHEADPHONE;
+    } else if (s->channel_mode >= AC3_CHMODE_2F2R &&
+               s->channel_mode == (s->output_mode & ~AC3_OUTPUT_LFEON)) {
+        switch (s->dolby_surround_ex_mode) {
+        case AC3_DSUREXMOD_ON: // EX or PLIIx
+            matrix_encoding = AV_MATRIX_ENCODING_DOLBYEX;
+            break;
+        case AC3_DSUREXMOD_PLIIZ:
+            matrix_encoding = AV_MATRIX_ENCODING_DPLIIZ;
+            break;
+        default: // not indicated or off
+            break;
+        }
+    }
+    if ((ret = ff_side_data_update_matrix_encoding(frame, matrix_encoding)) < 0)
+        return ret;
+
+    /* AVDownmixInfo */
+    if ((downmix_info = av_downmix_info_update_side_data(frame))) {
+        switch (s->preferred_downmix) {
+        case AC3_DMIXMOD_LTRT:
+            downmix_info->preferred_downmix_type = AV_DOWNMIX_TYPE_LTRT;
+            break;
+        case AC3_DMIXMOD_LORO:
+            downmix_info->preferred_downmix_type = AV_DOWNMIX_TYPE_LORO;
+            break;
+        case AC3_DMIXMOD_DPLII:
+            downmix_info->preferred_downmix_type = AV_DOWNMIX_TYPE_DPLII;
+            break;
+        default:
+            downmix_info->preferred_downmix_type = AV_DOWNMIX_TYPE_UNKNOWN;
+            break;
+        }
+        downmix_info->center_mix_level        = gain_levels[s->       center_mix_level];
+        downmix_info->center_mix_level_ltrt   = gain_levels[s->  center_mix_level_ltrt];
+        downmix_info->surround_mix_level      = gain_levels[s->     surround_mix_level];
+        downmix_info->surround_mix_level_ltrt = gain_levels[s->surround_mix_level_ltrt];
+        if (s->lfe_mix_level_exists)
+            downmix_info->lfe_mix_level       = gain_levels_lfe[s->lfe_mix_level];
+        else
+            downmix_info->lfe_mix_level       = 0.0; // -inf dB
+    } else
+        return AVERROR(ENOMEM);
+
     *got_frame_ptr = 1;
 
     return FFMIN(buf_size, s->frame_size);
@@ -1438,7 +1548,7 @@ static av_cold int ac3_decode_end(AVCodecContext *avctx)
 #define OFFSET(x) offsetof(AC3DecodeContext, x)
 #define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM)
 static const AVOption options[] = {
-    { "drc_scale", "percentage of dynamic range compression to apply", OFFSET(drc_scale), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, 0.0, 1.0, PAR },
+    { "drc_scale", "percentage of dynamic range compression to apply", OFFSET(drc_scale), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, 0.0, 6.0, PAR },
 
 {"dmix_mode", "Preferred Stereo Downmix Mode", OFFSET(preferred_stereo_downmix), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 2, 0, "dmix_mode"},
 {"ltrt_cmixlev",   "Lt/Rt Center Mix Level",   OFFSET(ltrt_center_mix_level),    AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/ac3dec.h

@@ -79,14 +79,22 @@ typedef struct AC3DecodeContext {
     int bit_rate;                           ///< stream bit rate, in bits-per-second
     int sample_rate;                        ///< sample frequency, in Hz
     int num_blocks;                         ///< number of audio blocks
+    int bitstream_id;                       ///< bitstream id                           (bsid)
     int bitstream_mode;                     ///< bitstream mode                         (bsmod)
     int channel_mode;                       ///< channel mode                           (acmod)
-    int channel_layout;                     ///< channel layout
     int lfe_on;                             ///< lfe channel in use
     int channel_map;                        ///< custom channel map
+    int preferred_downmix;                  ///< Preferred 2-channel downmix mode       (dmixmod)
     int center_mix_level;                   ///< Center mix level index
+    int center_mix_level_ltrt;              ///< Center mix level index for Lt/Rt       (ltrtcmixlev)
     int surround_mix_level;                 ///< Surround mix level index
+    int surround_mix_level_ltrt;            ///< Surround mix level index for Lt/Rt     (ltrtsurmixlev)
+    int lfe_mix_level_exists;               ///< indicates if lfemixlevcod is specified (lfemixlevcode)
+    int lfe_mix_level;                      ///< LFE mix level index                    (lfemixlevcod)
     int eac3;                               ///< indicates if current frame is E-AC-3
+    int dolby_surround_mode;                ///< dolby surround mode                    (dsurmod)
+    int dolby_surround_ex_mode;             ///< dolby surround ex mode                 (dsurexmod)
+    int dolby_headphone_mode;               ///< dolby headphone mode                   (dheadphonmod)
 ///@}
 
     int preferred_stereo_downmix;

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/ac3dsp.c

@@ -239,6 +239,19 @@ static void ac3_downmix_c(float **samples, float (*matrix)[2],
     }
 }
 
+static void apply_window_int16_c(int16_t *output, const int16_t *input,
+                                 const int16_t *window, unsigned int len)
+{
+    int i;
+    int len2 = len >> 1;
+
+    for (i = 0; i < len2; i++) {
+        int16_t w       = window[i];
+        output[i]       = (MUL16(input[i],       w) + (1 << 14)) >> 15;
+        output[len-i-1] = (MUL16(input[len-i-1], w) + (1 << 14)) >> 15;
+    }
+}
+
 av_cold void ff_ac3dsp_init(AC3DSPContext *c, int bit_exact)
 {
     c->ac3_exponent_min = ac3_exponent_min_c;
@@ -253,6 +266,7 @@ av_cold void ff_ac3dsp_init(AC3DSPContext *c, int bit_exact)
     c->sum_square_butterfly_int32 = ac3_sum_square_butterfly_int32_c;
     c->sum_square_butterfly_float = ac3_sum_square_butterfly_float_c;
     c->downmix = ac3_downmix_c;
+    c->apply_window_int16 = apply_window_int16_c;
 
     if (ARCH_ARM)
         ff_ac3dsp_init_arm(c, bit_exact);

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/ac3dsp.h

@@ -134,6 +134,20 @@ typedef struct AC3DSPContext {
 
     void (*downmix)(float **samples, float (*matrix)[2], int out_ch,
                     int in_ch, int len);
+
+    /**
+     * Apply symmetric window in 16-bit fixed-point.
+     * @param output destination array
+     *               constraints: 16-byte aligned
+     * @param input  source array
+     *               constraints: 16-byte aligned
+     * @param window window array
+     *               constraints: 16-byte aligned, at least len/2 elements
+     * @param len    full window length
+     *               constraints: multiple of ? greater than zero
+     */
+    void (*apply_window_int16)(int16_t *output, const int16_t *input,
+                               const int16_t *window, unsigned int len);
 } AC3DSPContext;
 
 void ff_ac3dsp_init    (AC3DSPContext *c, int bit_exact);

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/ac3tab.h

@@ -22,8 +22,9 @@
 #ifndef AVCODEC_AC3TAB_H
 #define AVCODEC_AC3TAB_H
 
-#include "config.h"
+#include <stdint.h>
-#include "libavutil/common.h"
+
+#include "libavutil/internal.h"
 #include "ac3.h"
 
 #if CONFIG_HARDCODED_TABLES

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/adpcm.c

@@ -1317,7 +1317,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
                                                        byte & 0x0F, 4, 0);
             }
         } else if (avctx->codec->id == AV_CODEC_ID_ADPCM_SBPRO_3) {
-            for (n = nb_samples / 3; n > 0; n--) {
+            for (n = (nb_samples<<st) / 3; n > 0; n--) {
                 int byte = bytestream2_get_byteu(&gb);
                 *samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
                                                         byte >> 5        , 3, 0);

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/adpcm_data.c

@@ -47,7 +47,7 @@ static const int8_t adpcm_index_table5[32] = {
     -1, -1, -1, -1, -1, -1, -1, -1, 1, 2, 4, 6, 8, 10, 13, 16,
 };
 
-const int8_t *ff_adpcm_index_tables[4] = {
+const int8_t * const ff_adpcm_index_tables[4] = {
     &adpcm_index_table2[0],
     &adpcm_index_table3[0],
     &ff_adpcm_index_table[0],

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/adpcm_data.h

@@ -31,7 +31,7 @@
 static const uint8_t ff_adpcm_ima_block_sizes[4]   = {  4, 12, 4, 20 };
 static const uint8_t ff_adpcm_ima_block_samples[4] = { 16, 32, 8, 32 };
 
-extern const int8_t *ff_adpcm_index_tables[4];
+extern const int8_t * const ff_adpcm_index_tables[4];
 extern const int8_t  ff_adpcm_index_table[16];
 extern const int16_t ff_adpcm_step_table[89];
 extern const int16_t ff_adpcm_oki_step_table[49];

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/allcodecs.c

@@ -24,8 +24,9 @@
  * Provide registration of all codecs, parsers and bitstream filters for libavcodec.
  */
 
-#include "avcodec.h"
 #include "config.h"
+#include "avcodec.h"
+#include "version.h"
 
 #define REGISTER_HWACCEL(X, x)                                          \
     {                                                                   \
@@ -79,7 +80,9 @@ void avcodec_register_all(void)
     REGISTER_HWACCEL(H264_VAAPI,        h264_vaapi);
     REGISTER_HWACCEL(H264_VDA,          h264_vda);
     REGISTER_HWACCEL(H264_VDPAU,        h264_vdpau);
+    REGISTER_HWACCEL(MPEG1_XVMC,        mpeg1_xvmc);
     REGISTER_HWACCEL(MPEG1_VDPAU,       mpeg1_vdpau);
+    REGISTER_HWACCEL(MPEG2_XVMC,        mpeg2_xvmc);
     REGISTER_HWACCEL(MPEG2_DXVA2,       mpeg2_dxva2);
     REGISTER_HWACCEL(MPEG2_VAAPI,       mpeg2_vaapi);
     REGISTER_HWACCEL(MPEG2_VDPAU,       mpeg2_vdpau);
@@ -119,7 +122,7 @@ void avcodec_register_all(void)
     REGISTER_DECODER(CAVS,              cavs);
     REGISTER_DECODER(CDGRAPHICS,        cdgraphics);
     REGISTER_DECODER(CDXL,              cdxl);
-    REGISTER_DECODER(CINEPAK,           cinepak);
+    REGISTER_ENCDEC (CINEPAK,           cinepak);
     REGISTER_ENCDEC (CLJR,              cljr);
     REGISTER_DECODER(CLLC,              cllc);
     REGISTER_ENCDEC (COMFORTNOISE,      comfortnoise);
@@ -147,6 +150,7 @@ void avcodec_register_all(void)
     REGISTER_DECODER(EXR,               exr);
     REGISTER_ENCDEC (FFV1,              ffv1);
     REGISTER_ENCDEC (FFVHUFF,           ffvhuff);
+    REGISTER_DECODER(FIC,               fic);
     REGISTER_ENCDEC (FLASHSV,           flashsv);
     REGISTER_ENCDEC (FLASHSV2,          flashsv2);
     REGISTER_DECODER(FLIC,              flic);
@@ -164,6 +168,8 @@ void avcodec_register_all(void)
     REGISTER_DECODER(H264_CRYSTALHD,    h264_crystalhd);
     REGISTER_DECODER(H264_VDA,          h264_vda);
     REGISTER_DECODER(H264_VDPAU,        h264_vdpau);
+    REGISTER_DECODER(HEVC,              hevc);
+    REGISTER_DECODER(HNM4_VIDEO,        hnm4_video);
     REGISTER_ENCDEC (HUFFYUV,           huffyuv);
     REGISTER_DECODER(IDCIN,             idcin);
     REGISTER_DECODER(IFF_BYTERUN1,      iff_byterun1);
@@ -187,7 +193,9 @@ void avcodec_register_all(void)
     REGISTER_DECODER(MJPEGB,            mjpegb);
     REGISTER_DECODER(MMVIDEO,           mmvideo);
     REGISTER_DECODER(MOTIONPIXELS,      motionpixels);
+#if FF_API_XVMC
     REGISTER_DECODER(MPEG_XVMC,         mpeg_xvmc);
+#endif /* FF_API_XVMC */
     REGISTER_ENCDEC (MPEG1VIDEO,        mpeg1video);
     REGISTER_ENCDEC (MPEG2VIDEO,        mpeg2video);
     REGISTER_ENCDEC (MPEG4,             mpeg4);
@@ -320,6 +328,7 @@ void avcodec_register_all(void)
     REGISTER_DECODER(APE,               ape);
     REGISTER_DECODER(ATRAC1,            atrac1);
     REGISTER_DECODER(ATRAC3,            atrac3);
+    REGISTER_DECODER(ATRAC3P,           atrac3p);
     REGISTER_DECODER(BINKAUDIO_DCT,     binkaudio_dct);
     REGISTER_DECODER(BINKAUDIO_RDFT,    binkaudio_rdft);
     REGISTER_DECODER(BMV_AUDIO,         bmv_audio);
@@ -344,6 +353,7 @@ void avcodec_register_all(void)
     REGISTER_DECODER(MP1FLOAT,          mp1float);
     REGISTER_ENCDEC (MP2,               mp2);
     REGISTER_DECODER(MP2FLOAT,          mp2float);
+    REGISTER_ENCODER(MP2FIXED,          mp2fixed);
     REGISTER_DECODER(MP3,               mp3);
     REGISTER_DECODER(MP3FLOAT,          mp3float);
     REGISTER_DECODER(MP3ADU,            mp3adu);
@@ -431,6 +441,7 @@ void avcodec_register_all(void)
     REGISTER_DECODER(ADPCM_EA_XAS,      adpcm_ea_xas);
     REGISTER_ENCDEC (ADPCM_G722,        adpcm_g722);
     REGISTER_ENCDEC (ADPCM_G726,        adpcm_g726);
+    REGISTER_DECODER(ADPCM_G726LE,      adpcm_g726le);
     REGISTER_DECODER(ADPCM_IMA_AMV,     adpcm_ima_amv);
     REGISTER_DECODER(ADPCM_IMA_APC,     adpcm_ima_apc);
     REGISTER_DECODER(ADPCM_IMA_DK3,     adpcm_ima_dk3);
@@ -501,8 +512,10 @@ void avcodec_register_all(void)
     REGISTER_ENCDEC (LIBVPX_VP8,        libvpx_vp8);
     REGISTER_ENCDEC (LIBVPX_VP9,        libvpx_vp9);
     REGISTER_ENCODER(LIBWAVPACK,        libwavpack);
+    REGISTER_ENCODER(LIBWEBP,           libwebp);
     REGISTER_ENCODER(LIBX264,           libx264);
     REGISTER_ENCODER(LIBX264RGB,        libx264rgb);
+    REGISTER_ENCODER(LIBX265,           libx265);
     REGISTER_ENCODER(LIBXAVS,           libxavs);
     REGISTER_ENCODER(LIBXVID,           libxvid);
     REGISTER_DECODER(LIBZVBI_TELETEXT,  libzvbi_teletext);
@@ -532,6 +545,7 @@ void avcodec_register_all(void)
     REGISTER_PARSER(H261,               h261);
     REGISTER_PARSER(H263,               h263);
     REGISTER_PARSER(H264,               h264);
+    REGISTER_PARSER(HEVC,               hevc);
     REGISTER_PARSER(MJPEG,              mjpeg);
     REGISTER_PARSER(MLP,                mlp);
     REGISTER_PARSER(MPEG4VIDEO,         mpeg4video);
@@ -546,6 +560,7 @@ void avcodec_register_all(void)
     REGISTER_PARSER(VORBIS,             vorbis);
     REGISTER_PARSER(VP3,                vp3);
     REGISTER_PARSER(VP8,                vp8);
+    REGISTER_PARSER(VP9,                vp9);
 
     /* bitstream filters */
     REGISTER_BSF(AAC_ADTSTOASC,         aac_adtstoasc);
@@ -555,7 +570,6 @@ void avcodec_register_all(void)
     REGISTER_BSF(IMX_DUMP_HEADER,       imx_dump_header);
     REGISTER_BSF(MJPEG2JPEG,            mjpeg2jpeg);
     REGISTER_BSF(MJPEGA_DUMP_HEADER,    mjpega_dump_header);
-    REGISTER_BSF(MP3_HEADER_COMPRESS,   mp3_header_compress);
     REGISTER_BSF(MP3_HEADER_DECOMPRESS, mp3_header_decompress);
     REGISTER_BSF(MOV2TEXTSUB,           mov2textsub);
     REGISTER_BSF(NOISE,                 noise);

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/apedec.c

@@ -605,10 +605,14 @@ static void decode_array_0000(APEContext *ctx, GetBitContext *gb,
         rice->ksum += out[i];
     }
     rice->k = av_log2(rice->ksum / 10) + 1;
+    if (rice->k >= 24)
+        return;
     for (; i < 64; i++) {
         out[i] = get_rice_ook(&ctx->gb, rice->k);
         rice->ksum += out[i];
         rice->k = av_log2(rice->ksum / ((i + 1) * 2)) + 1;
+        if (rice->k >= 24)
+            return;
     }
     ksummax = 1 << (rice->k + 7);
     ksummin = rice->k ? (1 << (rice->k + 6)) : 0;
@@ -1436,7 +1440,7 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
         }
         if (s->fileversion < 3950) // previous versions overread two bytes
             buf_size += 2;
-        av_fast_malloc(&s->data, (unsigned int *) &s->data_size, buf_size);
+        av_fast_padded_malloc(&s->data, &s->data_size, buf_size);
         if (!s->data)
             return AVERROR(ENOMEM);
         s->dsp.bswap_buf((uint32_t*)s->data, (const uint32_t*)buf, buf_size >> 2);
@@ -1458,7 +1462,8 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
             }
             s->ptr += offset;
         } else {
-            init_get_bits(&s->gb, s->ptr, (s->data_end - s->ptr) * 8);
+            if ((ret = init_get_bits8(&s->gb, s->ptr, s->data_end - s->ptr)) < 0)
+                return ret;
             if (s->fileversion > 3800)
                 skip_bits_long(&s->gb, offset * 8);
             else

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/atrac3.c

@@ -116,9 +116,6 @@ typedef struct ATRAC3Context {
 static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
 static VLC_TYPE atrac3_vlc_table[4096][2];
 static VLC   spectral_coeff_tab[7];
-static float gain_tab1[16];
-static float gain_tab2[31];
-
 
 /**
  * Regular 512 points IMDCT without overlapping, with the exception of the
@@ -175,7 +172,7 @@ static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
     return off;
 }
 
-static av_cold void init_atrac3_window(void)
+static av_cold void init_imdct_window(void)
 {
     int i, j;
 
@@ -780,7 +777,7 @@ static av_cold void atrac3_init_static_data(void)
 {
     int i;
 
-    init_atrac3_window();
+    init_imdct_window();
     ff_atrac_generate_tables();
 
     /* Initialize the VLC tables. */
@@ -792,13 +789,6 @@ static av_cold void atrac3_init_static_data(void)
                  huff_bits[i],  1, 1,
                  huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
     }
-
-    /* Generate gain tables */
-    for (i = 0; i < 16; i++)
-        gain_tab1[i] = exp2f (4 - i);
-
-    for (i = -15; i < 16; i++)
-        gain_tab2[i + 15] = exp2f (i * -0.125);
 }
 
 static av_cold int atrac3_decode_init(AVCodecContext *avctx)

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/atrac3plus.h

@@ -0,0 +1,240 @@
+/*
+ * ATRAC3+ compatible decoder
+ *
+ * Copyright (c) 2010-2013 Maxim Poliakovski
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Global structures, constants and data for ATRAC3+ decoder.
+ */
+
+#ifndef AVCODEC_ATRAC3PLUS_H
+#define AVCODEC_ATRAC3PLUS_H
+
+#include <stdint.h>
+
+#include "libavutil/float_dsp.h"
+#include "atrac.h"
+#include "avcodec.h"
+#include "fft.h"
+#include "get_bits.h"
+
+/** Global unit sizes */
+#define ATRAC3P_SUBBANDS        16  ///< number of PQF subbands
+#define ATRAC3P_SUBBAND_SAMPLES 128 ///< number of samples per subband
+#define ATRAC3P_FRAME_SAMPLES   (ATRAC3P_SUBBAND_SAMPLES * ATRAC3P_SUBBANDS)
+
+#define ATRAC3P_PQF_FIR_LEN     12  ///< length of the prototype FIR of the PQF
+
+/** Global constants */
+#define ATRAC3P_POWER_COMP_OFF  15  ///< disable power compensation
+
+/** ATRAC3+ channel unit types */
+enum Atrac3pChannelUnitTypes {
+    CH_UNIT_MONO       = 0, ///< unit containing one coded channel
+    CH_UNIT_STEREO     = 1, ///< unit containing two jointly-coded channels
+    CH_UNIT_EXTENSION  = 2, ///< unit containing extension information
+    CH_UNIT_TERMINATOR = 3  ///< unit sequence terminator
+};
+
+/** Per-channel IPQF history */
+typedef struct Atrac3pIPQFChannelCtx {
+    DECLARE_ALIGNED(32, float, buf1)[ATRAC3P_PQF_FIR_LEN * 2][8];
+    DECLARE_ALIGNED(32, float, buf2)[ATRAC3P_PQF_FIR_LEN * 2][8];
+    int pos;
+} Atrac3pIPQFChannelCtx;
+
+/** Amplitude envelope of a group of sine waves */
+typedef struct Atrac3pWaveEnvelope {
+    int has_start_point;    ///< indicates start point within the GHA window
+    int has_stop_point;     ///< indicates stop point within the GHA window
+    int start_pos;          ///< start position expressed in n*4 samples
+    int stop_pos;           ///< stop  position expressed in n*4 samples
+} Atrac3pWaveEnvelope;
+
+/** Parameters of a group of sine waves */
+typedef struct Atrac3pWavesData {
+    Atrac3pWaveEnvelope pend_env;   ///< pending envelope from the previous frame
+    Atrac3pWaveEnvelope curr_env;   ///< group envelope from the current frame
+    int num_wavs;           ///< number of sine waves in the group
+    int start_index;        ///< start index into global tones table for that subband
+} Atrac3pWavesData;
+
+/** Parameters of a single sine wave */
+typedef struct Atrac3pWaveParam {
+    int   freq_index;   ///< wave frequency index
+    int   amp_sf;       ///< quantized amplitude scale factor
+    int   amp_index;    ///< quantized amplitude index
+    int   phase_index;  ///< quantized phase index
+} Atrac3pWaveParam;
+
+/** Sound channel parameters */
+typedef struct Atrac3pChanParams {
+    int ch_num;
+    int num_coded_vals;         ///< number of transmitted quant unit values
+    int fill_mode;
+    int split_point;
+    int table_type;             ///< table type: 0 - tone?, 1- noise?
+    int qu_wordlen[32];         ///< array of word lengths for each quant unit
+    int qu_sf_idx[32];          ///< array of scale factor indexes for each quant unit
+    int qu_tab_idx[32];         ///< array of code table indexes for each quant unit
+    int16_t spectrum[2048];     ///< decoded IMDCT spectrum
+    uint8_t power_levs[5];      ///< power compensation levels
+
+    /* imdct window shape history (2 frames) for overlapping. */
+    uint8_t wnd_shape_hist[2][ATRAC3P_SUBBANDS];    ///< IMDCT window shape, 0=sine/1=steep
+    uint8_t *wnd_shape;         ///< IMDCT window shape for current frame
+    uint8_t *wnd_shape_prev;    ///< IMDCT window shape for previous frame
+
+    /* gain control data history (2 frames) for overlapping. */
+    AtracGainInfo gain_data_hist[2][ATRAC3P_SUBBANDS];  ///< gain control data for all subbands
+    AtracGainInfo *gain_data;       ///< gain control data for next frame
+    AtracGainInfo *gain_data_prev;  ///< gain control data for previous frame
+    int num_gain_subbands;      ///< number of subbands with gain control data
+
+    /* tones data history (2 frames) for overlapping. */
+    Atrac3pWavesData tones_info_hist[2][ATRAC3P_SUBBANDS];
+    Atrac3pWavesData *tones_info;
+    Atrac3pWavesData *tones_info_prev;
+} Atrac3pChanParams;
+
+/* Per-unit sine wave parameters */
+typedef struct Atrac3pWaveSynthParams {
+    int tones_present;                      ///< 1 - tones info present
+    int amplitude_mode;                     ///< 1 - low range, 0 - high range
+    int num_tone_bands;                     ///< number of PQF bands with tones
+    uint8_t tone_sharing[ATRAC3P_SUBBANDS]; ///< 1 - subband-wise tone sharing flags
+    uint8_t tone_master[ATRAC3P_SUBBANDS];  ///< 1 - subband-wise tone channel swapping
+    uint8_t phase_shift[ATRAC3P_SUBBANDS];  ///< 1 - subband-wise 180° phase shifting
+    int tones_index;                        ///< total sum of tones in this unit
+    Atrac3pWaveParam waves[48];
+} Atrac3pWaveSynthParams;
+
+/** Channel unit parameters */
+typedef struct Atrac3pChanUnitCtx {
+    /* channel unit variables */
+    int unit_type;                          ///< unit type (mono/stereo)
+    int num_quant_units;
+    int num_subbands;
+    int used_quant_units;                   ///< number of quant units with coded spectrum
+    int num_coded_subbands;                 ///< number of subbands with coded spectrum
+    int mute_flag;                          ///< mute flag
+    int use_full_table;                     ///< 1 - full table list, 0 - restricted one
+    int noise_present;                      ///< 1 - global noise info present
+    int noise_level_index;                  ///< global noise level index
+    int noise_table_index;                  ///< global noise RNG table index
+    uint8_t swap_channels[ATRAC3P_SUBBANDS];    ///< 1 - perform subband-wise channel swapping
+    uint8_t negate_coeffs[ATRAC3P_SUBBANDS];    ///< 1 - subband-wise IMDCT coefficients negation
+    Atrac3pChanParams channels[2];
+
+    /* Variables related to GHA tones */
+    Atrac3pWaveSynthParams wave_synth_hist[2];     ///< waves synth history for two frames
+    Atrac3pWaveSynthParams *waves_info;
+    Atrac3pWaveSynthParams *waves_info_prev;
+
+    Atrac3pIPQFChannelCtx ipqf_ctx[2];
+    DECLARE_ALIGNED(32, float, prev_buf)[2][ATRAC3P_FRAME_SAMPLES]; ///< overlapping buffer
+} Atrac3pChanUnitCtx;
+
+/**
+ * Initialize VLC tables for bitstream parsing.
+ */
+void ff_atrac3p_init_vlcs(void);
+
+/**
+ * Decode bitstream data of a channel unit.
+ *
+ * @param[in]     gb            the GetBit context
+ * @param[in,out] ctx           ptr to the channel unit context
+ * @param[in]     num_channels  number of channels to process
+ * @param[in]     avctx         ptr to the AVCodecContext
+ * @return result code: 0 = OK, otherwise - error code
+ */
+int ff_atrac3p_decode_channel_unit(GetBitContext *gb, Atrac3pChanUnitCtx *ctx,
+                                   int num_channels, AVCodecContext *avctx);
+
+/**
+ * Initialize IMDCT transform.
+ *
+ * @param[in]   avctx      ptr to the AVCodecContext
+ * @param[in]   mdct_ctx   pointer to MDCT transform context
+ */
+void ff_atrac3p_init_imdct(AVCodecContext *avctx, FFTContext *mdct_ctx);
+
+/**
+ * Initialize sine waves synthesizer.
+ */
+void ff_atrac3p_init_wave_synth(void);
+
+/**
+ * Synthesize sine waves for a particular subband.
+ *
+ * @param[in]   ch_unit   pointer to the channel unit context
+ * @param[in]   fdsp      pointer to float DSP context
+ * @param[in]   ch_num    which channel to process
+ * @param[in]   sb        which subband to process
+ * @param[out]  out       receives processed data
+ */
+void ff_atrac3p_generate_tones(Atrac3pChanUnitCtx *ch_unit, AVFloatDSPContext *fdsp,
+                               int ch_num, int sb, float *out);
+
+/**
+ * Perform power compensation aka noise dithering.
+ *
+ * @param[in]      ctx         ptr to the channel context
+ * @param[in]      ch_index    which channel to process
+ * @param[in,out]  sp          ptr to channel spectrum to process
+ * @param[in]      rng_index   indicates which RNG table to use
+ * @param[in]      sb_num      which subband to process
+ */
+void ff_atrac3p_power_compensation(Atrac3pChanUnitCtx *ctx, int ch_index,
+                                   float *sp, int rng_index, int sb_num);
+
+/**
+ * Regular IMDCT and windowing without overlapping,
+ * with spectrum reversal in the odd subbands.
+ *
+ * @param[in]   fdsp       pointer to float DSP context
+ * @param[in]   mdct_ctx   pointer to MDCT transform context
+ * @param[in]   pIn        float input
+ * @param[out]  pOut       float output
+ * @param[in]   wind_id    which MDCT window to apply
+ * @param[in]   sb         subband number
+ */
+void ff_atrac3p_imdct(AVFloatDSPContext *fdsp, FFTContext *mdct_ctx, float *pIn,
+                      float *pOut, int wind_id, int sb);
+
+/**
+ * Subband synthesis filter based on the polyphase quadrature (pseudo-QMF)
+ * filter bank.
+ *
+ * @param[in]      dct_ctx   ptr to the pre-initialized IDCT context
+ * @param[in,out]  hist      ptr to the filter history
+ * @param[in]      in        input data to process
+ * @param[out]     out       receives processed data
+ */
+void ff_atrac3p_ipqf(FFTContext *dct_ctx, Atrac3pIPQFChannelCtx *hist,
+                     const float *in, float *out);
+
+extern const uint16_t ff_atrac3p_qu_to_spec_pos[33];
+extern const float ff_atrac3p_sf_tab[64];
+extern const float ff_atrac3p_mant_tab[8];
+
+#endif /* AVCODEC_ATRAC3PLUS_H */

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/atrac3plusdec.c

@@ -0,0 +1,396 @@
+/*
+ * ATRAC3+ compatible decoder
+ *
+ * Copyright (c) 2010-2013 Maxim Poliakovski
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Sony ATRAC3+ compatible decoder.
+ *
+ * Container formats used to store its data:
+ * RIFF WAV (.at3) and Sony OpenMG (.oma, .aa3).
+ *
+ * Technical description of this codec can be found here:
+ * http://wiki.multimedia.cx/index.php?title=ATRAC3plus
+ *
+ * Kudos to Benjamin Larsson and Michael Karcher
+ * for their precious technical help!
+ */
+
+#include <stdint.h>
+#include <string.h>
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/float_dsp.h"
+#include "avcodec.h"
+#include "get_bits.h"
+#include "internal.h"
+#include "atrac.h"
+#include "atrac3plus.h"
+
+typedef struct ATRAC3PContext {
+    GetBitContext gb;
+    AVFloatDSPContext fdsp;
+
+    DECLARE_ALIGNED(32, float, samples)[2][ATRAC3P_FRAME_SAMPLES];  ///< quantized MDCT spectrum
+    DECLARE_ALIGNED(32, float, mdct_buf)[2][ATRAC3P_FRAME_SAMPLES]; ///< output of the IMDCT
+    DECLARE_ALIGNED(32, float, time_buf)[2][ATRAC3P_FRAME_SAMPLES]; ///< output of the gain compensation
+    DECLARE_ALIGNED(32, float, outp_buf)[2][ATRAC3P_FRAME_SAMPLES];
+
+    AtracGCContext gainc_ctx;   ///< gain compensation context
+    FFTContext mdct_ctx;
+    FFTContext ipqf_dct_ctx;    ///< IDCT context used by IPQF
+
+    Atrac3pChanUnitCtx *ch_units;   ///< global channel units
+
+    int num_channel_blocks;     ///< number of channel blocks
+    uint8_t channel_blocks[5];  ///< channel configuration descriptor
+    uint64_t my_channel_layout; ///< current channel layout
+} ATRAC3PContext;
+
+static av_cold int atrac3p_decode_close(AVCodecContext *avctx)
+{
+    av_free(((ATRAC3PContext *)(avctx->priv_data))->ch_units);
+
+    return 0;
+}
+
+static av_cold int set_channel_params(ATRAC3PContext *ctx,
+                                      AVCodecContext *avctx)
+{
+    memset(ctx->channel_blocks, 0, sizeof(ctx->channel_blocks));
+
+    switch (avctx->channels) {
+    case 1:
+        if (avctx->channel_layout != AV_CH_FRONT_LEFT)
+            avctx->channel_layout = AV_CH_LAYOUT_MONO;
+
+        ctx->num_channel_blocks = 1;
+        ctx->channel_blocks[0]  = CH_UNIT_MONO;
+        break;
+    case 2:
+        avctx->channel_layout   = AV_CH_LAYOUT_STEREO;
+        ctx->num_channel_blocks = 1;
+        ctx->channel_blocks[0]  = CH_UNIT_STEREO;
+        break;
+    case 3:
+        avctx->channel_layout   = AV_CH_LAYOUT_SURROUND;
+        ctx->num_channel_blocks = 2;
+        ctx->channel_blocks[0]  = CH_UNIT_STEREO;
+        ctx->channel_blocks[1]  = CH_UNIT_MONO;
+        break;
+    case 4:
+        avctx->channel_layout   = AV_CH_LAYOUT_4POINT0;
+        ctx->num_channel_blocks = 3;
+        ctx->channel_blocks[0]  = CH_UNIT_STEREO;
+        ctx->channel_blocks[1]  = CH_UNIT_MONO;
+        ctx->channel_blocks[2]  = CH_UNIT_MONO;
+        break;
+    case 6:
+        avctx->channel_layout   = AV_CH_LAYOUT_5POINT1_BACK;
+        ctx->num_channel_blocks = 4;
+        ctx->channel_blocks[0]  = CH_UNIT_STEREO;
+        ctx->channel_blocks[1]  = CH_UNIT_MONO;
+        ctx->channel_blocks[2]  = CH_UNIT_STEREO;
+        ctx->channel_blocks[3]  = CH_UNIT_MONO;
+        break;
+    case 7:
+        avctx->channel_layout   = AV_CH_LAYOUT_6POINT1_BACK;
+        ctx->num_channel_blocks = 5;
+        ctx->channel_blocks[0]  = CH_UNIT_STEREO;
+        ctx->channel_blocks[1]  = CH_UNIT_MONO;
+        ctx->channel_blocks[2]  = CH_UNIT_STEREO;
+        ctx->channel_blocks[3]  = CH_UNIT_MONO;
+        ctx->channel_blocks[4]  = CH_UNIT_MONO;
+        break;
+    case 8:
+        avctx->channel_layout   = AV_CH_LAYOUT_7POINT1;
+        ctx->num_channel_blocks = 5;
+        ctx->channel_blocks[0]  = CH_UNIT_STEREO;
+        ctx->channel_blocks[1]  = CH_UNIT_MONO;
+        ctx->channel_blocks[2]  = CH_UNIT_STEREO;
+        ctx->channel_blocks[3]  = CH_UNIT_STEREO;
+        ctx->channel_blocks[4]  = CH_UNIT_MONO;
+        break;
+    default:
+        av_log(avctx, AV_LOG_ERROR,
+               "Unsupported channel count: %d!\n", avctx->channels);
+        return AVERROR_INVALIDDATA;
+    }
+
+    return 0;
+}
+
+static av_cold int atrac3p_decode_init(AVCodecContext *avctx)
+{
+    ATRAC3PContext *ctx = avctx->priv_data;
+    int i, ch, ret;
+
+    if (!avctx->block_align) {
+        av_log(avctx, AV_LOG_ERROR, "block_align is not set\n");
+        return AVERROR(EINVAL);
+    }
+
+    ff_atrac3p_init_vlcs();
+
+    avpriv_float_dsp_init(&ctx->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+
+    /* initialize IPQF */
+    ff_mdct_init(&ctx->ipqf_dct_ctx, 5, 1, 32.0 / 32768.0);
+
+    ff_atrac3p_init_imdct(avctx, &ctx->mdct_ctx);
+
+    ff_atrac_init_gain_compensation(&ctx->gainc_ctx, 6, 2);
+
+    ff_atrac3p_init_wave_synth();
+
+    if ((ret = set_channel_params(ctx, avctx)) < 0)
+        return ret;
+
+    ctx->my_channel_layout = avctx->channel_layout;
+
+    ctx->ch_units = av_mallocz(sizeof(*ctx->ch_units) *
+                               ctx->num_channel_blocks);
+    if (!ctx->ch_units) {
+        atrac3p_decode_close(avctx);
+        return AVERROR(ENOMEM);
+    }
+
+    for (i = 0; i < ctx->num_channel_blocks; i++) {
+        for (ch = 0; ch < 2; ch++) {
+            ctx->ch_units[i].channels[ch].ch_num          = ch;
+            ctx->ch_units[i].channels[ch].wnd_shape       = &ctx->ch_units[i].channels[ch].wnd_shape_hist[0][0];
+            ctx->ch_units[i].channels[ch].wnd_shape_prev  = &ctx->ch_units[i].channels[ch].wnd_shape_hist[1][0];
+            ctx->ch_units[i].channels[ch].gain_data       = &ctx->ch_units[i].channels[ch].gain_data_hist[0][0];
+            ctx->ch_units[i].channels[ch].gain_data_prev  = &ctx->ch_units[i].channels[ch].gain_data_hist[1][0];
+            ctx->ch_units[i].channels[ch].tones_info      = &ctx->ch_units[i].channels[ch].tones_info_hist[0][0];
+            ctx->ch_units[i].channels[ch].tones_info_prev = &ctx->ch_units[i].channels[ch].tones_info_hist[1][0];
+        }
+
+        ctx->ch_units[i].waves_info      = &ctx->ch_units[i].wave_synth_hist[0];
+        ctx->ch_units[i].waves_info_prev = &ctx->ch_units[i].wave_synth_hist[1];
+    }
+
+    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+
+    return 0;
+}
+
+static void decode_residual_spectrum(Atrac3pChanUnitCtx *ctx,
+                                     float out[2][ATRAC3P_FRAME_SAMPLES],
+                                     int num_channels,
+                                     AVCodecContext *avctx)
+{
+    int i, sb, ch, qu, nspeclines, RNG_index;
+    float *dst, q;
+    int16_t *src;
+    /* calculate RNG table index for each subband */
+    int sb_RNG_index[ATRAC3P_SUBBANDS] = { 0 };
+
+    if (ctx->mute_flag) {
+        for (ch = 0; ch < num_channels; ch++)
+            memset(out[ch], 0, ATRAC3P_FRAME_SAMPLES * sizeof(*out[ch]));
+        return;
+    }
+
+    for (qu = 0, RNG_index = 0; qu < ctx->used_quant_units; qu++)
+        RNG_index += ctx->channels[0].qu_sf_idx[qu] +
+                     ctx->channels[1].qu_sf_idx[qu];
+
+    for (sb = 0; sb < ctx->num_coded_subbands; sb++, RNG_index += 128)
+        sb_RNG_index[sb] = RNG_index & 0x3FC;
+
+    /* inverse quant and power compensation */
+    for (ch = 0; ch < num_channels; ch++) {
+        /* clear channel's residual spectrum */
+        memset(out[ch], 0, ATRAC3P_FRAME_SAMPLES * sizeof(*out[ch]));
+
+        for (qu = 0; qu < ctx->used_quant_units; qu++) {
+            src        = &ctx->channels[ch].spectrum[ff_atrac3p_qu_to_spec_pos[qu]];
+            dst        = &out[ch][ff_atrac3p_qu_to_spec_pos[qu]];
+            nspeclines = ff_atrac3p_qu_to_spec_pos[qu + 1] -
+                         ff_atrac3p_qu_to_spec_pos[qu];
+
+            if (ctx->channels[ch].qu_wordlen[qu] > 0) {
+                q = ff_atrac3p_sf_tab[ctx->channels[ch].qu_sf_idx[qu]] *
+                    ff_atrac3p_mant_tab[ctx->channels[ch].qu_wordlen[qu]];
+                for (i = 0; i < nspeclines; i++)
+                    dst[i] = src[i] * q;
+            }
+        }
+
+        for (sb = 0; sb < ctx->num_coded_subbands; sb++)
+            ff_atrac3p_power_compensation(ctx, ch, &out[ch][0],
+                                          sb_RNG_index[sb], sb);
+    }
+
+    if (ctx->unit_type == CH_UNIT_STEREO) {
+        for (sb = 0; sb < ctx->num_coded_subbands; sb++) {
+            if (ctx->swap_channels[sb]) {
+                for (i = 0; i < ATRAC3P_SUBBAND_SAMPLES; i++)
+                    FFSWAP(float, out[0][sb * ATRAC3P_SUBBAND_SAMPLES + i],
+                                  out[1][sb * ATRAC3P_SUBBAND_SAMPLES + i]);
+            }
+
+            /* flip coefficients' sign if requested */
+            if (ctx->negate_coeffs[sb])
+                for (i = 0; i < ATRAC3P_SUBBAND_SAMPLES; i++)
+                    out[1][sb * ATRAC3P_SUBBAND_SAMPLES + i] = -(out[1][sb * ATRAC3P_SUBBAND_SAMPLES + i]);
+        }
+    }
+}
+
+static void reconstruct_frame(ATRAC3PContext *ctx, Atrac3pChanUnitCtx *ch_unit,
+                              int num_channels, AVCodecContext *avctx)
+{
+    int ch, sb;
+
+    for (ch = 0; ch < num_channels; ch++) {
+        for (sb = 0; sb < ch_unit->num_subbands; sb++) {
+            /* inverse transform and windowing */
+            ff_atrac3p_imdct(&ctx->fdsp, &ctx->mdct_ctx,
+                             &ctx->samples[ch][sb * ATRAC3P_SUBBAND_SAMPLES],
+                             &ctx->mdct_buf[ch][sb * ATRAC3P_SUBBAND_SAMPLES],
+                             (ch_unit->channels[ch].wnd_shape_prev[sb] << 1) +
+                             ch_unit->channels[ch].wnd_shape[sb], sb);
+
+            /* gain compensation and overlapping */
+            ff_atrac_gain_compensation(&ctx->gainc_ctx,
+                                       &ctx->mdct_buf[ch][sb * ATRAC3P_SUBBAND_SAMPLES],
+                                       &ch_unit->prev_buf[ch][sb * ATRAC3P_SUBBAND_SAMPLES],
+                                       &ch_unit->channels[ch].gain_data_prev[sb],
+                                       &ch_unit->channels[ch].gain_data[sb],
+                                       ATRAC3P_SUBBAND_SAMPLES,
+                                       &ctx->time_buf[ch][sb * ATRAC3P_SUBBAND_SAMPLES]);
+        }
+
+        /* zero unused subbands in both output and overlapping buffers */
+        memset(&ch_unit->prev_buf[ch][ch_unit->num_subbands * ATRAC3P_SUBBAND_SAMPLES],
+               0,
+               (ATRAC3P_SUBBANDS - ch_unit->num_subbands) *
+               ATRAC3P_SUBBAND_SAMPLES *
+               sizeof(ch_unit->prev_buf[ch][ch_unit->num_subbands * ATRAC3P_SUBBAND_SAMPLES]));
+        memset(&ctx->time_buf[ch][ch_unit->num_subbands * ATRAC3P_SUBBAND_SAMPLES],
+               0,
+               (ATRAC3P_SUBBANDS - ch_unit->num_subbands) *
+               ATRAC3P_SUBBAND_SAMPLES *
+               sizeof(ctx->time_buf[ch][ch_unit->num_subbands * ATRAC3P_SUBBAND_SAMPLES]));
+
+        /* resynthesize and add tonal signal */
+        if (ch_unit->waves_info->tones_present ||
+            ch_unit->waves_info_prev->tones_present) {
+            for (sb = 0; sb < ch_unit->num_subbands; sb++)
+                if (ch_unit->channels[ch].tones_info[sb].num_wavs ||
+                    ch_unit->channels[ch].tones_info_prev[sb].num_wavs) {
+                    ff_atrac3p_generate_tones(ch_unit, &ctx->fdsp, ch, sb,
+                                              &ctx->time_buf[ch][sb * 128]);
+                }
+        }
+
+        /* subband synthesis and acoustic signal output */
+        ff_atrac3p_ipqf(&ctx->ipqf_dct_ctx, &ch_unit->ipqf_ctx[ch],
+                        &ctx->time_buf[ch][0], &ctx->outp_buf[ch][0]);
+    }
+
+    /* swap window shape and gain control buffers. */
+    for (ch = 0; ch < num_channels; ch++) {
+        FFSWAP(uint8_t *, ch_unit->channels[ch].wnd_shape,
+               ch_unit->channels[ch].wnd_shape_prev);
+        FFSWAP(AtracGainInfo *, ch_unit->channels[ch].gain_data,
+               ch_unit->channels[ch].gain_data_prev);
+        FFSWAP(Atrac3pWavesData *, ch_unit->channels[ch].tones_info,
+               ch_unit->channels[ch].tones_info_prev);
+    }
+
+    FFSWAP(Atrac3pWaveSynthParams *, ch_unit->waves_info, ch_unit->waves_info_prev);
+}
+
+static int atrac3p_decode_frame(AVCodecContext *avctx, void *data,
+                                int *got_frame_ptr, AVPacket *avpkt)
+{
+    ATRAC3PContext *ctx = avctx->priv_data;
+    AVFrame *frame      = data;
+    int i, ret, ch_unit_id, ch_block = 0, out_ch_index = 0, channels_to_process;
+    float **samples_p = (float **)frame->extended_data;
+
+    frame->nb_samples = ATRAC3P_FRAME_SAMPLES;
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+        return ret;
+    }
+
+    if ((ret = init_get_bits8(&ctx->gb, avpkt->data, avpkt->size)) < 0)
+        return ret;
+
+    if (get_bits1(&ctx->gb)) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid start bit!\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    while (get_bits_left(&ctx->gb) >= 2 &&
+           (ch_unit_id = get_bits(&ctx->gb, 2)) != CH_UNIT_TERMINATOR) {
+        if (ch_unit_id == CH_UNIT_EXTENSION) {
+            avpriv_report_missing_feature(avctx, "Channel unit extension");
+            return AVERROR_PATCHWELCOME;
+        }
+        if (ch_block >= ctx->num_channel_blocks ||
+            ctx->channel_blocks[ch_block] != ch_unit_id) {
+            av_log(avctx, AV_LOG_ERROR,
+                   "Frame data doesn't match channel configuration!\n");
+            return AVERROR_INVALIDDATA;
+        }
+
+        ctx->ch_units[ch_block].unit_type = ch_unit_id;
+        channels_to_process               = ch_unit_id + 1;
+
+        if ((ret = ff_atrac3p_decode_channel_unit(&ctx->gb,
+                                                  &ctx->ch_units[ch_block],
+                                                  channels_to_process,
+                                                  avctx)) < 0)
+            return ret;
+
+        decode_residual_spectrum(&ctx->ch_units[ch_block], ctx->samples,
+                                 channels_to_process, avctx);
+        reconstruct_frame(ctx, &ctx->ch_units[ch_block],
+                          channels_to_process, avctx);
+
+        for (i = 0; i < channels_to_process; i++)
+            memcpy(samples_p[out_ch_index + i], ctx->outp_buf[i],
+                   ATRAC3P_FRAME_SAMPLES * sizeof(**samples_p));
+
+        ch_block++;
+        out_ch_index += channels_to_process;
+    }
+
+    *got_frame_ptr = 1;
+
+    return avctx->block_align;
+}
+
+AVCodec ff_atrac3p_decoder = {
+    .name           = "atrac3plus",
+    .long_name      = NULL_IF_CONFIG_SMALL("ATRAC3+ (Adaptive TRansform Acoustic Coding 3+)"),
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_ATRAC3P,
+    .priv_data_size = sizeof(ATRAC3PContext),
+    .init           = atrac3p_decode_init,
+    .close          = atrac3p_decode_close,
+    .decode         = atrac3p_decode_frame,
+};

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/atrac3plusdsp.c

@@ -0,0 +1,638 @@
+/*
+ * ATRAC3+ compatible decoder
+ *
+ * Copyright (c) 2010-2013 Maxim Poliakovski
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ *  @file
+ *  DSP functions for ATRAC3+ decoder.
+ */
+
+#include <math.h>
+
+#include "libavutil/float_dsp.h"
+#include "avcodec.h"
+#include "sinewin.h"
+#include "fft.h"
+#include "atrac3plus.h"
+
+/**
+ *  Map quant unit number to its position in the spectrum.
+ *  To get the number of spectral lines in each quant unit do the following:
+ *  num_specs = qu_to_spec_pos[i+1] - qu_to_spec_pos[i]
+ */
+const uint16_t ff_atrac3p_qu_to_spec_pos[33] = {
+      0,    16,   32,   48,   64,   80,   96,  112,
+    128,   160,  192,  224,  256,  288,  320,  352,
+    384,   448,  512,  576,  640,  704,  768,  896,
+    1024, 1152, 1280, 1408, 1536, 1664, 1792, 1920,
+    2048
+};
+
+/* Scalefactors table. */
+/* Approx. Equ: pow(2.0, (i - 16.0 + 0.501783948) / 3.0) */
+const float ff_atrac3p_sf_tab[64] = {
+    0.027852058,  0.0350914, 0.044212341, 0.055704117,  0.0701828,
+    0.088424683, 0.11140823,   0.1403656,  0.17684937, 0.22281647, 0.2807312, 0.35369873,
+    0.44563293,   0.5614624,  0.70739746,  0.89126587,  1.1229248, 1.4147949,  1.7825317,
+    2.2458496,    2.8295898,   3.5650635,   4.4916992,  5.6591797,  7.130127,  8.9833984,
+    11.318359,    14.260254,   17.966797,   22.636719,  28.520508, 35.933594,  45.273438,
+    57.041016,    71.867188,   90.546875,   114.08203,  143.73438, 181.09375,  228.16406,
+    287.46875,     362.1875,   456.32812,    574.9375,    724.375, 912.65625,   1149.875,
+    1448.75,      1825.3125,     2299.75,      2897.5,   3650.625,    4599.5,     5795.0,
+    7301.25,         9199.0,     11590.0,     14602.5,    18398.0,   23180.0,    29205.0,
+    36796.0,        46360.0,     58410.0
+};
+
+/* Mantissa table. */
+/* pow(10, x * log10(2) + 0.05) / 2 / ([1,2,3,5,7,15,31] + 0.5) */
+const float ff_atrac3p_mant_tab[8] = {
+    0.0,
+    0.74801636,
+    0.44882202,
+    0.32058716,
+    0.20400238,
+    0.1496048,
+    0.07239151,
+    0.035619736
+};
+
+#define ATRAC3P_MDCT_SIZE (ATRAC3P_SUBBAND_SAMPLES * 2)
+
+av_cold void ff_atrac3p_init_imdct(AVCodecContext *avctx, FFTContext *mdct_ctx)
+{
+    ff_init_ff_sine_windows(7);
+    ff_init_ff_sine_windows(6);
+
+    /* Initialize the MDCT transform. */
+    ff_mdct_init(mdct_ctx, 8, 1, -1.0);
+}
+
+#define TWOPI (2 * M_PI)
+
+#define DEQUANT_PHASE(ph) (((ph) & 0x1F) << 6)
+
+static DECLARE_ALIGNED(32, float, sine_table)[2048]; ///< wave table
+static DECLARE_ALIGNED(32, float, hann_window)[256]; ///< Hann windowing function
+static float amp_sf_tab[64];   ///< scalefactors for quantized amplitudes
+
+av_cold void ff_atrac3p_init_wave_synth(void)
+{
+    int i;
+
+    /* generate sine wave table */
+    for (i = 0; i < 2048; i++)
+        sine_table[i] = sin(TWOPI * i / 2048);
+
+    /* generate Hann window */
+    for (i = 0; i < 256; i++)
+        hann_window[i] = (1.0f - cos(TWOPI * i / 256.0f)) * 0.5f;
+
+    /* generate amplitude scalefactors table */
+    for (i = 0; i < 64; i++)
+        amp_sf_tab[i] = pow(2.0f, ((double)i - 3) / 4.0f);
+}
+
+/**
+ *  Synthesize sine waves according to given parameters.
+ *
+ *  @param[in]    synth_param   ptr to common synthesis parameters
+ *  @param[in]    waves_info    parameters for each sine wave
+ *  @param[in]    envelope      envelope data for all waves in a group
+ *  @param[in]    phase_shift   flag indicates 180° phase shift
+ *  @param[in]    reg_offset    region offset for trimming envelope data
+ *  @param[out]   out           receives sythesized data
+ */
+static void waves_synth(Atrac3pWaveSynthParams *synth_param,
+                        Atrac3pWavesData *waves_info,
+                        Atrac3pWaveEnvelope *envelope,
+                        int phase_shift, int reg_offset, float *out)
+{
+    int i, wn, inc, pos;
+    double amp;
+    Atrac3pWaveParam *wave_param = &synth_param->waves[waves_info->start_index];
+
+    for (wn = 0; wn < waves_info->num_wavs; wn++, wave_param++) {
+        /* amplitude dequantization */
+        amp = amp_sf_tab[wave_param->amp_sf] *
+              (!synth_param->amplitude_mode
+               ? (wave_param->amp_index + 1) / 15.13f
+               : 1.0f);
+
+        inc = wave_param->freq_index;
+        pos = DEQUANT_PHASE(wave_param->phase_index) - (reg_offset ^ 128) * inc & 2047;
+
+        /* waveform generation */
+        for (i = 0; i < 128; i++) {
+            out[i] += sine_table[pos] * amp;
+            pos     = (pos + inc) & 2047;
+        }
+    }
+
+    /* fade in with steep Hann window if requested */
+    if (envelope->has_start_point) {
+        pos = (envelope->start_pos << 2) - reg_offset;
+        if (pos > 0 && pos <= 128) {
+            memset(out, 0, pos * sizeof(*out));
+            if (!envelope->has_stop_point ||
+                envelope->start_pos != envelope->stop_pos) {
+                out[pos + 0] *= hann_window[0];
+                out[pos + 1] *= hann_window[32];
+                out[pos + 2] *= hann_window[64];
+                out[pos + 3] *= hann_window[96];
+            }
+        }
+    }
+
+    /* fade out with steep Hann window if requested */
+    if (envelope->has_stop_point) {
+        pos = (envelope->stop_pos + 1 << 2) - reg_offset;
+        if (pos > 0 && pos <= 128) {
+            out[pos - 4] *= hann_window[96];
+            out[pos - 3] *= hann_window[64];
+            out[pos - 2] *= hann_window[32];
+            out[pos - 1] *= hann_window[0];
+            memset(&out[pos], 0, (128 - pos) * sizeof(out[pos]));
+        }
+    }
+}
+
+void ff_atrac3p_generate_tones(Atrac3pChanUnitCtx *ch_unit, AVFloatDSPContext *fdsp,
+                               int ch_num, int sb, float *out)
+{
+    DECLARE_ALIGNED(32, float, wavreg1)[128] = { 0 };
+    DECLARE_ALIGNED(32, float, wavreg2)[128] = { 0 };
+    int i, reg1_env_nonzero, reg2_env_nonzero;
+    Atrac3pWavesData *tones_now  = &ch_unit->channels[ch_num].tones_info_prev[sb];
+    Atrac3pWavesData *tones_next = &ch_unit->channels[ch_num].tones_info[sb];
+
+    /* reconstruct full envelopes for both overlapping regions
+     * from truncated bitstream data */
+    if (tones_next->pend_env.has_start_point &&
+        tones_next->pend_env.start_pos < tones_next->pend_env.stop_pos) {
+        tones_next->curr_env.has_start_point = 1;
+        tones_next->curr_env.start_pos       = tones_next->pend_env.start_pos + 32;
+    } else if (tones_now->pend_env.has_start_point) {
+        tones_next->curr_env.has_start_point = 1;
+        tones_next->curr_env.start_pos       = tones_now->pend_env.start_pos;
+    } else {
+        tones_next->curr_env.has_start_point = 0;
+        tones_next->curr_env.start_pos       = 0;
+    }
+
+    if (tones_now->pend_env.has_stop_point &&
+        tones_now->pend_env.stop_pos >= tones_next->curr_env.start_pos) {
+        tones_next->curr_env.has_stop_point = 1;
+        tones_next->curr_env.stop_pos       = tones_now->pend_env.stop_pos;
+    } else if (tones_next->pend_env.has_stop_point) {
+        tones_next->curr_env.has_stop_point = 1;
+        tones_next->curr_env.stop_pos       = tones_next->pend_env.stop_pos + 32;
+    } else {
+        tones_next->curr_env.has_stop_point = 0;
+        tones_next->curr_env.stop_pos       = 64;
+    }
+
+    /* is the visible part of the envelope non-zero? */
+    reg1_env_nonzero = (tones_now->curr_env.stop_pos    < 32) ? 0 : 1;
+    reg2_env_nonzero = (tones_next->curr_env.start_pos >= 32) ? 0 : 1;
+
+    /* synthesize waves for both overlapping regions */
+    if (tones_now->num_wavs && reg1_env_nonzero)
+        waves_synth(ch_unit->waves_info_prev, tones_now, &tones_now->curr_env,
+                    ch_unit->waves_info_prev->phase_shift[sb] & ch_num,
+                    128, wavreg1);
+
+    if (tones_next->num_wavs && reg2_env_nonzero)
+        waves_synth(ch_unit->waves_info, tones_next, &tones_next->curr_env,
+                    ch_unit->waves_info->phase_shift[sb] & ch_num, 0, wavreg2);
+
+    /* Hann windowing for non-faded wave signals */
+    if (tones_now->num_wavs && tones_next->num_wavs &&
+        reg1_env_nonzero && reg2_env_nonzero) {
+        fdsp->vector_fmul(wavreg1, wavreg1, &hann_window[128], 128);
+        fdsp->vector_fmul(wavreg2, wavreg2,  hann_window,      128);
+    } else {
+        if (tones_now->num_wavs && !tones_now->curr_env.has_stop_point)
+            fdsp->vector_fmul(wavreg1, wavreg1, &hann_window[128], 128);
+
+        if (tones_next->num_wavs && !tones_next->curr_env.has_start_point)
+            fdsp->vector_fmul(wavreg2, wavreg2, hann_window, 128);
+    }
+
+    /* Overlap and add to residual */
+    for (i = 0; i < 128; i++)
+        out[i] += wavreg1[i] + wavreg2[i];
+}
+
+static const int subband_to_powgrp[ATRAC3P_SUBBANDS] = {
+    0, 1, 1, 2, 2, 2, 3, 3, 3, 3, 3, 4, 4, 4, 4, 4
+};
+
+/* noise table for power compensation */
+static const float noise_tab[1024] = {
+    -0.01358032,  -0.05593872,   0.01696777,  -0.14871216,  -0.26412964,  -0.09893799,   0.25723267,
+     0.02008057,  -0.72235107,  -0.44351196,  -0.22985840,   0.16833496,   0.46902466,   0.05917358,
+    -0.15179443,   0.41299438,  -0.01287842,   0.13360596,   0.43557739,  -0.09530640,  -0.58422852,
+     0.39266968,  -0.08343506,  -0.25604248,   0.22848511,   0.26013184,  -0.65588379,   0.17288208,
+    -0.08673096,  -0.05203247,   0.07299805,  -0.28665161,  -0.35806274,   0.06552124,  -0.09387207,
+     0.21099854,  -0.28347778,  -0.72402954,   0.05050659,  -0.10635376,  -0.18853760,   0.29724121,
+     0.20703125,  -0.29791260,  -0.37634277,   0.47970581,  -0.09976196,   0.32641602,  -0.29248047,
+    -0.28237915,   0.26028442,  -0.36157227,   0.22042847,  -0.03222656,  -0.37268066,  -0.03759766,
+     0.09909058,   0.23284912,   0.19320679,   0.14453125,  -0.02139282,  -0.19702148,   0.31533813,
+    -0.16741943,   0.35031128,  -0.35656738,  -0.66128540,  -0.00701904,   0.20898438,   0.26837158,
+    -0.33706665,  -0.04568481,   0.12600708,   0.10284424,   0.07321167,  -0.18280029,   0.38101196,
+     0.21301270,   0.04541016,   0.01156616,  -0.26391602,  -0.02346802,  -0.22125244,   0.29760742,
+    -0.36233521,  -0.31314087,  -0.13967896,  -0.11276245,  -0.19433594,   0.34490967,   0.02343750,
+     0.21963501,  -0.02777100,  -0.67678833,  -0.08999634,   0.14233398,  -0.27697754,   0.51422119,
+    -0.05047607,   0.48327637,   0.37167358,  -0.60806274,   0.18728638,  -0.15191650,   0.00637817,
+     0.02832031,  -0.15618896,   0.60644531,   0.21826172,   0.06384277,  -0.31863403,   0.08816528,
+     0.15447998,  -0.07015991,  -0.08154297,  -0.40966797,  -0.39785767,  -0.11709595,   0.22052002,
+     0.18466187,  -0.17257690,   0.03759766,  -0.06195068,   0.00433350,   0.12176514,   0.34011841,
+     0.25610352,  -0.05294800,   0.41033936,   0.16854858,  -0.76187134,   0.13845825,  -0.19418335,
+    -0.21524048,  -0.44412231,  -0.08160400,  -0.28195190,  -0.01873779,   0.15524292,  -0.37438965,
+    -0.44860840,   0.43096924,  -0.24746704,   0.49856567,   0.14859009,   0.38159180,   0.20541382,
+    -0.39175415,  -0.65850830,  -0.43716431,   0.13037109,  -0.05111694,   0.39956665,   0.21447754,
+    -0.04861450,   0.33654785,   0.10589600,  -0.88085938,  -0.30822754,   0.38577271,   0.30047607,
+     0.38836670,   0.09118652,  -0.36477661,  -0.01641846,  -0.23031616,   0.26058960,   0.18859863,
+    -0.21868896,  -0.17861938,  -0.29754639,   0.09777832,   0.10806274,  -0.51605225,   0.00076294,
+     0.13259888,   0.11090088,  -0.24084473,   0.24957275,   0.01379395,  -0.04141235,  -0.04937744,
+     0.57394409,   0.27410889,   0.27587891,   0.45013428,  -0.32592773,   0.11160278,  -0.00970459,
+     0.29092407,   0.03356934,  -0.70925903,   0.04882812,   0.43499756,   0.07720947,  -0.27554321,
+    -0.01742554,  -0.08413696,  -0.04028320,  -0.52850342,  -0.07330322,   0.05181885,   0.21362305,
+    -0.18765259,   0.07058716,  -0.03009033,   0.32662964,   0.27023315,  -0.28002930,   0.17568970,
+     0.03338623,   0.30242920,  -0.03921509,   0.32174683,  -0.23733521,   0.08575439,  -0.38269043,
+     0.09194946,  -0.07238770,   0.17941284,  -0.51278687,  -0.25146484,   0.19790649,  -0.19195557,
+     0.16549683,   0.42456055,   0.39129639,  -0.02868652,   0.17980957,   0.24902344,  -0.76583862,
+    -0.20959473,   0.61013794,   0.37011719,   0.36859131,  -0.04486084,   0.10678101,  -0.15994263,
+    -0.05328369,   0.28463745,  -0.06420898,  -0.36987305,  -0.28009033,  -0.11764526,   0.04312134,
+    -0.08038330,   0.04885864,  -0.03067017,  -0.00042725,   0.34289551,  -0.00988770,   0.34838867,
+     0.32516479,  -0.16271973,   0.38269043,   0.03240967,   0.12417603,  -0.14331055,  -0.34902954,
+    -0.18325806,   0.29421997,   0.44284058,   0.75170898,  -0.67245483,  -0.12176514,   0.27914429,
+    -0.29806519,   0.19863892,   0.30087280,   0.22680664,  -0.36633301,  -0.32534790,  -0.57553101,
+    -0.16641235,   0.43811035,   0.08331299,   0.15942383,   0.26516724,  -0.24240112,  -0.11761475,
+    -0.16827393,  -0.14260864,   0.46343994,   0.11804199,  -0.55514526,  -0.02520752,  -0.14309692,
+     0.00448608,   0.02749634,  -0.30545044,   0.70965576,   0.45108032,   0.66439819,  -0.68255615,
+    -0.12496948,   0.09146118,  -0.21109009,  -0.23791504,   0.79943848,  -0.35205078,  -0.24963379,
+     0.18719482,  -0.19079590,   0.07458496,   0.07623291,  -0.28781128,  -0.37121582,  -0.19580078,
+    -0.01773071,  -0.16717529,   0.13040161,   0.14672852,   0.42379761,   0.03582764,   0.11431885,
+     0.05145264,   0.44702148,   0.08963013,   0.01367188,  -0.54519653,  -0.12692261,   0.21176147,
+     0.04925537,   0.30670166,  -0.11029053,   0.19555664,  -0.27740479,   0.23043823,   0.15554810,
+    -0.19299316,  -0.25729370,   0.17800903,  -0.03579712,  -0.05065918,  -0.06933594,  -0.09500122,
+    -0.07821655,   0.23889160,  -0.31900024,   0.03073120,  -0.00415039,   0.61315918,   0.37176514,
+    -0.13442993,  -0.15536499,  -0.19216919,  -0.37899780,   0.19992065,   0.02630615,  -0.12573242,
+     0.25927734,  -0.02447510,   0.29629517,  -0.40731812,  -0.17333984,   0.24310303,  -0.10607910,
+     0.14828491,   0.08792114,  -0.18743896,  -0.05572510,  -0.04833984,   0.10473633,  -0.29028320,
+    -0.67687988,  -0.28170776,  -0.41687012,   0.05413818,  -0.23284912,   0.09555054,  -0.08969116,
+    -0.15112305,   0.12738037,   0.35986328,   0.28948975,   0.30691528,   0.23956299,   0.06973267,
+    -0.31198120,  -0.18450928,   0.22280884,  -0.21600342,   0.23522949,  -0.61840820,  -0.13012695,
+     0.26412964,   0.47320557,  -0.26440430,   0.38757324,   0.17352295,  -0.26104736,  -0.25866699,
+    -0.12274170,  -0.29733276,   0.07687378,   0.18588257,  -0.08880615,   0.31185913,   0.05313110,
+    -0.10885620,  -0.14901733,  -0.22323608,  -0.08538818,   0.19812012,   0.19732666,  -0.18927002,
+     0.29058838,   0.25555420,  -0.48599243,   0.18768311,   0.01345825,   0.34887695,   0.21530151,
+     0.19857788,   0.18661499,  -0.01394653,  -0.09063721,  -0.38781738,   0.27160645,  -0.20379639,
+    -0.32119751,  -0.23889160,   0.27096558,   0.24951172,   0.07922363,   0.07479858,  -0.50946045,
+     0.10220337,   0.58364868,  -0.19503784,  -0.18560791,  -0.01165771,   0.47195435,   0.22430420,
+    -0.38635254,  -0.03732300,  -0.09179688,   0.06991577,   0.15106201,   0.20605469,  -0.05969238,
+    -0.41821289,   0.12231445,  -0.04672241,  -0.05117798,  -0.11523438,  -0.51849365,  -0.04077148,
+     0.44284058,  -0.64086914,   0.17019653,   0.02236938,   0.22848511,  -0.23214722,  -0.32354736,
+    -0.14068604,  -0.29690552,  -0.19891357,   0.02774048,  -0.20965576,  -0.52191162,  -0.19299316,
+    -0.07290649,   0.49053955,  -0.22302246,   0.05642700,   0.13122559,  -0.20819092,  -0.83590698,
+    -0.08181763,   0.26797485,  -0.00091553,  -0.09457397,   0.17089844,  -0.27020264,   0.30270386,
+     0.05496216,   0.09564209,  -0.08590698,   0.02130127,   0.35931396,   0.21728516,  -0.15396118,
+    -0.05053711,   0.02719116,   0.16302490,   0.43212891,   0.10229492,  -0.40820312,   0.21646118,
+     0.08435059,  -0.11145020,  -0.39962769,  -0.05618286,  -0.10223389,  -0.60839844,   0.33724976,
+    -0.06341553,  -0.47369385,  -0.32852173,   0.05242920,   0.19635010,  -0.19137573,  -0.67901611,
+     0.16180420,   0.05133057,  -0.22283936,   0.09646606,   0.24288940,  -0.45007324,   0.08804321,
+     0.14053345,   0.22619629,  -0.01000977,   0.36355591,  -0.19863892,  -0.30364990,  -0.24118042,
+    -0.57461548,   0.26498413,   0.04345703,  -0.09796143,  -0.47714233,  -0.23739624,   0.18737793,
+     0.08926392,  -0.02795410,   0.00305176,  -0.08700562,  -0.38711548,   0.03222656,   0.10940552,
+    -0.41906738,  -0.01620483,  -0.47061157,   0.37985229,  -0.21624756,   0.47976685,  -0.20046997,
+    -0.62533569,  -0.26907349,  -0.02877808,   0.00671387,  -0.29071045,  -0.24685669,  -0.15722656,
+    -0.26055908,   0.29968262,   0.28225708,  -0.08990479,  -0.16748047,  -0.46759033,  -0.25067139,
+    -0.25183105,  -0.45932007,   0.05828857,   0.29006958,   0.23840332,  -0.17974854,   0.26931763,
+     0.10696411,  -0.06848145,  -0.17126465,  -0.10522461,  -0.55386353,  -0.42306519,  -0.07608032,
+     0.24380493,   0.38586426,   0.16882324,   0.26751709,   0.17303467,   0.35809326,  -0.22094727,
+    -0.30703735,  -0.28497314,  -0.04321289,   0.15219116,  -0.17071533,  -0.39334106,   0.03439331,
+    -0.10809326,  -0.30590820,   0.26449585,  -0.07412720,   0.13638306,  -0.01062012,   0.27996826,
+     0.04397583,  -0.05557251,  -0.56933594,   0.03363037,  -0.00949097,   0.52642822,  -0.44329834,
+     0.28308105,  -0.05499268,  -0.23312378,  -0.29870605,  -0.05123901,   0.26831055,  -0.35238647,
+    -0.30993652,   0.34646606,  -0.19775391,   0.44595337,   0.13769531,   0.45358276,   0.19961548,
+     0.42681885,   0.15722656,   0.00128174,   0.23757935,   0.40988159,   0.25164795,  -0.00732422,
+    -0.12405396,  -0.43420410,  -0.00402832,   0.34243774,   0.36264038,   0.18807983,  -0.09301758,
+    -0.10296631,   0.05532837,  -0.31652832,   0.14337158,   0.35040283,   0.32540894,   0.05728149,
+    -0.12030029,  -0.25942993,  -0.20312500,  -0.16491699,  -0.46051025,  -0.08004761,   0.50772095,
+     0.16168213,   0.28439331,   0.08105469,  -0.19104004,   0.38589478,  -0.16400146,  -0.25454712,
+     0.20281982,  -0.20730591,  -0.06311035,   0.32937622,   0.15032959,  -0.05340576,   0.30487061,
+    -0.11648560,   0.38009644,  -0.20062256,   0.43466187,   0.01150513,   0.35754395,  -0.13146973,
+     0.67489624,   0.05212402,   0.27914429,  -0.39431763,   0.75308228,  -0.13366699,   0.24453735,
+     0.42248535,  -0.65905762,  -0.00546265,  -0.03491211,  -0.13659668,  -0.08294678,  -0.45666504,
+     0.27188110,   0.12731934,   0.61148071,   0.10449219,  -0.28836060,   0.00091553,   0.24618530,
+     0.13119507,   0.05685425,   0.17355347,   0.42034912,   0.08514404,   0.24536133,   0.18951416,
+    -0.19107056,  -0.15036011,   0.02334595,   0.54986572,   0.32321167,  -0.16104126,  -0.03054810,
+     0.43594360,   0.17309570,   0.61053467,   0.24731445,   0.33334351,   0.15240479,   0.15588379,
+     0.36425781,  -0.30407715,  -0.13302612,   0.00427246,   0.04171753,  -0.33178711,   0.34216309,
+    -0.12463379,  -0.02764893,   0.05905151,  -0.31436157,   0.16531372,   0.34542847,  -0.03292847,
+     0.12527466,  -0.12313843,  -0.13171387,   0.04757690,  -0.45095825,  -0.19085693,   0.35342407,
+    -0.23239136,  -0.34387207,   0.11264038,  -0.15740967,   0.05273438,   0.74942017,   0.21505737,
+     0.08514404,  -0.42391968,  -0.19531250,   0.35293579,   0.25305176,   0.15731812,  -0.70324707,
+    -0.21591187,   0.35604858,   0.14132690,   0.11724854,   0.15853882,  -0.24597168,   0.07019043,
+     0.02127075,   0.12658691,   0.06390381,  -0.12292480,   0.15441895,  -0.47640991,   0.06195068,
+     0.58981323,  -0.15151978,  -0.03604126,  -0.45059204,  -0.01672363,  -0.46997070,   0.25750732,
+     0.18084717,   0.06661987,   0.13253784,   0.67828369,   0.11370850,   0.11325073,  -0.04611206,
+    -0.07791138,  -0.36544800,  -0.06747437,  -0.31594849,   0.16131592,   0.41983032,   0.11071777,
+    -0.36889648,   0.30963135,  -0.37875366,   0.58508301,   0.00393677,   0.12338257,   0.03424072,
+    -0.21728516,  -0.12838745,  -0.46981812,   0.05868530,  -0.25015259,   0.27407837,   0.65240479,
+    -0.34429932,  -0.15179443,   0.14056396,   0.33505249,   0.28826904,   0.09921265,   0.34390259,
+     0.13656616,  -0.23608398,   0.00863647,   0.02627563,  -0.19119263,   0.19775391,  -0.07214355,
+     0.07809448,   0.03454590,  -0.03417969,   0.00033569,  -0.23095703,   0.18673706,   0.05798340,
+     0.03814697,  -0.04318237,   0.05487061,   0.08633423,   0.55950928,  -0.06347656,   0.10333252,
+     0.25305176,   0.05853271,   0.12246704,  -0.25543213,  -0.34262085,  -0.36437988,  -0.21304321,
+    -0.05093384,   0.02777100,   0.07620239,  -0.21215820,  -0.09326172,   0.19021606,  -0.40579224,
+    -0.01193237,   0.19845581,  -0.35336304,  -0.07397461,   0.20104980,   0.08615112,  -0.44375610,
+     0.11419678,   0.24453735,  -0.16555786,  -0.05081177,  -0.01406860,   0.27893066,  -0.18692017,
+     0.07473755,   0.03451538,  -0.39733887,   0.21548462,  -0.22534180,  -0.39651489,  -0.04989624,
+    -0.57662964,   0.06390381,   0.62020874,  -0.13470459,   0.04345703,  -0.21862793,  -0.02789307,
+     0.51696777,  -0.27587891,   0.39004517,   0.09857178,  -0.00738525,   0.31317139,   0.00048828,
+    -0.46572876,   0.29531860,  -0.10009766,  -0.27856445,   0.03594971,   0.25048828,  -0.74584961,
+    -0.25350952,  -0.03302002,   0.31188965,   0.01571655,   0.46710205,   0.21591187,   0.07260132,
+    -0.42132568,  -0.53900146,  -0.13674927,  -0.16571045,  -0.34454346,   0.12359619,  -0.11184692,
+     0.00967407,   0.34576416,  -0.05761719,   0.34848022,   0.17645264,  -0.39395142,   0.10339355,
+     0.18215942,   0.20697021,   0.59109497,  -0.11560059,  -0.07385254,   0.10397339,   0.35437012,
+    -0.22863770,   0.01794434,   0.17559814,  -0.17495728,   0.12142944,   0.10928345,  -1.00000000,
+    -0.01379395,   0.21237183,  -0.27035522,   0.27319336,  -0.37066650,   0.41354370,  -0.40054321,
+     0.00689697,   0.26321411,   0.39266968,   0.65298462,   0.41625977,  -0.13909912,   0.78375244,
+    -0.30941772,   0.20169067,  -0.39367676,   0.94021606,  -0.24066162,   0.05557251,  -0.24533081,
+    -0.05444336,  -0.76754761,  -0.19375610,  -0.11041260,  -0.17532349,   0.16006470,   0.02188110,
+     0.17465210,  -0.04342651,  -0.56777954,  -0.40988159,   0.26687622,   0.11700439,  -0.00344849,
+    -0.05395508,   0.37426758,  -0.40719604,  -0.15032959,  -0.01660156,   0.04196167,  -0.04559326,
+    -0.12969971,   0.12011719,   0.08419800,  -0.11199951,   0.35174561,   0.10275269,  -0.25686646,
+     0.48446655,   0.03225708,   0.28408813,  -0.18701172,   0.36282349,  -0.03280640,   0.32302856,
+     0.17233276,   0.48269653,   0.31112671,  -0.04946899,   0.12774658,   0.52685547,   0.10211182,
+     0.05953979,   0.05999756,   0.20144653,   0.00744629,   0.27316284,   0.24377441,   0.39672852,
+     0.01702881,  -0.35513306,   0.11364746,  -0.13555908,   0.48880005,  -0.15417480,  -0.09149170,
+    -0.02615356,   0.46246338,  -0.72250366,   0.22332764,   0.23849487,  -0.25686646,  -0.08514404,
+    -0.02062988,  -0.34494019,  -0.02297974,  -0.80386353,  -0.08074951,  -0.12689209,  -0.06896973,
+     0.24099731,  -0.35650635,  -0.09558105,   0.29254150,   0.23132324,  -0.16726685,   0.00000000,
+    -0.24237061,   0.30899048,   0.29504395,  -0.20898438,   0.17059326,  -0.07672119,  -0.14395142,
+     0.05572510,   0.20602417,  -0.51550293,  -0.03167725,  -0.48840332,  -0.20425415,   0.14144897,
+     0.07275391,  -0.76669312,  -0.22488403,   0.20651245,   0.03259277,   0.00085449,   0.03039551,
+     0.47555542,   0.38351440
+};
+
+/** Noise level table for power compensation.
+ *  Equ: pow(2.0f, (double)(6 - i) / 3.0f) where i = 0...15 */
+static const float pwc_levs[16] = {
+    3.96875, 3.15625,     2.5,    2.0, 1.59375,   1.25,     1.0, 0.78125,
+    0.625,       0.5, 0.40625, 0.3125,    0.25, 0.1875, 0.15625, 0.0
+};
+
+/** Map subband number to quant unit number. */
+static const int subband_to_qu[17] = {
+    0, 8, 12, 16, 18, 20, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32
+};
+
+void ff_atrac3p_power_compensation(Atrac3pChanUnitCtx *ctx, int ch_index,
+                                   float *sp, int rng_index, int sb)
+{
+    AtracGainInfo *g1, *g2;
+    float pwcsp[ATRAC3P_SUBBAND_SAMPLES], *dst, grp_lev, qu_lev;
+    int i, gain_lev, gcv = 0, qu, nsp;
+    int swap_ch = (ctx->unit_type == CH_UNIT_STEREO && ctx->swap_channels[sb]) ? 1 : 0;
+
+    if (ctx->channels[ch_index ^ swap_ch].power_levs[subband_to_powgrp[sb]] == ATRAC3P_POWER_COMP_OFF)
+        return;
+
+    /* generate initial noise spectrum */
+    for (i = 0; i < ATRAC3P_SUBBAND_SAMPLES; i++, rng_index++)
+        pwcsp[i] = noise_tab[rng_index & 0x3FF];
+
+    /* check gain control information */
+    g1 = &ctx->channels[ch_index ^ swap_ch].gain_data[sb];
+    g2 = &ctx->channels[ch_index ^ swap_ch].gain_data_prev[sb];
+
+    gain_lev = (g1->num_points > 0) ? (6 - g1->lev_code[0]) : 0;
+
+    for (i = 0; i < g2->num_points; i++)
+        gcv = FFMAX(gcv, gain_lev - (g2->lev_code[i] - 6));
+
+    for (i = 0; i < g1->num_points; i++)
+        gcv = FFMAX(gcv, 6 - g1->lev_code[i]);
+
+    grp_lev = pwc_levs[ctx->channels[ch_index ^ swap_ch].power_levs[subband_to_powgrp[sb]]] / (1 << gcv);
+
+    /* skip the lowest two quant units (frequencies 0...351 Hz) for subband 0 */
+    for (qu = subband_to_qu[sb] + (!sb ? 2 : 0); qu < subband_to_qu[sb + 1]; qu++) {
+        if (ctx->channels[ch_index].qu_wordlen[qu] <= 0)
+            continue;
+
+        qu_lev = ff_atrac3p_sf_tab[ctx->channels[ch_index].qu_sf_idx[qu]] *
+                 ff_atrac3p_mant_tab[ctx->channels[ch_index].qu_wordlen[qu]] /
+                 (1 << ctx->channels[ch_index].qu_wordlen[qu]) * grp_lev;
+
+        dst = &sp[ff_atrac3p_qu_to_spec_pos[qu]];
+        nsp = ff_atrac3p_qu_to_spec_pos[qu + 1] - ff_atrac3p_qu_to_spec_pos[qu];
+
+        for (i = 0; i < nsp; i++)
+            dst[i] += pwcsp[i] * qu_lev;
+    }
+}
+
+void ff_atrac3p_imdct(AVFloatDSPContext *fdsp, FFTContext *mdct_ctx, float *pIn,
+                      float *pOut, int wind_id, int sb)
+{
+    int i;
+
+    if (sb & 1)
+        for (i = 0; i < ATRAC3P_SUBBAND_SAMPLES / 2; i++)
+            FFSWAP(float, pIn[i], pIn[ATRAC3P_SUBBAND_SAMPLES - 1 - i]);
+
+    mdct_ctx->imdct_calc(mdct_ctx, pOut, pIn);
+
+    /* Perform windowing on the output.
+     * ATRAC3+ uses two different MDCT windows:
+     * - The first one is just the plain sine window of size 256
+     * - The 2nd one is the plain sine window of size 128
+     *   wrapped into zero (at the start) and one (at the end) regions.
+     *   Both regions are 32 samples long. */
+    if (wind_id & 2) { /* 1st half: steep window */
+        memset(pOut, 0, sizeof(float) * 32);
+        fdsp->vector_fmul(&pOut[32], &pOut[32], ff_sine_64, 64);
+    } else /* 1st half: simple sine window */
+        fdsp->vector_fmul(pOut, pOut, ff_sine_128, ATRAC3P_MDCT_SIZE / 2);
+
+    if (wind_id & 1) { /* 2nd half: steep window */
+        fdsp->vector_fmul_reverse(&pOut[160], &pOut[160], ff_sine_64, 64);
+        memset(&pOut[224], 0, sizeof(float) * 32);
+    } else /* 2nd half: simple sine window */
+        fdsp->vector_fmul_reverse(&pOut[128], &pOut[128], ff_sine_128,
+                                  ATRAC3P_MDCT_SIZE / 2);
+}
+
+/* lookup table for fast modulo 23 op required for cyclic buffers of the IPQF */
+static const int mod23_lut[26] = {
+    23,  0,  1,  2,  3,  4,  5,  6,  7,  8,  9, 10, 11,
+    12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 0
+};
+
+/* First half of the 384-tap IPQF filtering coefficients. */
+static const float ipqf_coeffs1[ATRAC3P_PQF_FIR_LEN][16] = {
+    { -5.8336207e-7,    -8.0604229e-7,    -4.2005411e-7,    -4.4400572e-8,
+       3.226247e-8,      3.530856e-8,      1.2660377e-8,     0.000010516783,
+      -0.000011838618,   6.005389e-7,      0.0000014333754,  0.0000023108685,
+       0.0000032569742,  0.0000046192422,  0.0000063894258,  0.0000070302972 },
+    { -0.0000091622824, -0.000010502935,  -0.0000079212787, -0.0000041712024,
+      -0.0000026336629, -0.0000015432918, -5.7168614e-7,     0.0000018111954,
+       0.000023530851,   0.00002780562,    0.000032302323,   0.000036968919,
+       0.000041575615,   0.000045337845,   0.000046043948,   0.000048585582 },
+    { -0.000064464548,  -0.000068306952,  -0.000073081472,  -0.00007612785,
+      -0.000074850752,  -0.000070208509,  -0.000062285151,  -0.000058270442,
+      -0.000056296329,  -0.000049888811,  -0.000035615325,  -0.000018532943,
+       0.0000016657353,  0.00002610587,    0.000053397067,   0.00008079566 },
+    { -0.00054488552,   -0.00052537228,   -0.00049731287,   -0.00045778,
+      -0.00040612387,   -0.00034301577,   -0.00026866337,   -0.00018248901,
+      -0.000084307925,   0.000025081157,   0.00014135583,    0.00026649953,
+       0.00039945057,    0.00053928449,    0.00068422867,    0.00083093712 },
+    { -0.0014771431,    -0.001283227,     -0.0010566821,    -0.00079780724,
+      -0.00050782406,   -0.00018855913,    0.00015771533,    0.00052769453,
+       0.00091862219,    0.001326357,      0.0017469483,     0.0021754825,
+       0.0026067684,     0.0030352892,     0.0034549395,     0.0038591374 },
+    { -0.0022995141,    -0.001443546,     -0.00049266568,    0.00055068987,
+       0.001682895,      0.0028992873,     0.0041943151,     0.0055614738,
+       0.0069935122,     0.0084823566,     0.010018963,      0.011593862,
+       0.013196872,      0.014817309,      0.016444042,      0.018065533 },
+    { -0.034426283,     -0.034281436,     -0.033992987,     -0.033563249,
+      -0.032995768,     -0.032295227,     -0.031467363,     -0.030518902,
+      -0.02945766,      -0.028291954,     -0.027031265,     -0.025685543,
+      -0.024265358,     -0.022781773,     -0.021246184,     -0.019670162 },
+    { -0.0030586775,    -0.0037203205,    -0.0042847847,    -0.0047529764,
+      -0.0051268316,    -0.0054091476,    -0.0056034233,    -0.005714261,
+      -0.0057445862,    -0.0057025906,    -0.0055920109,    -0.0054194843,
+      -0.0051914565,    -0.0049146507,    -0.0045959447,    -0.0042418269 },
+    { -0.0016376863,    -0.0017651899,    -0.0018608454,    -0.0019252141,
+      -0.0019593791,    -0.0019653172,    -0.0019450618,    -0.0018990048,
+      -0.00183808,      -0.0017501717,    -0.0016481078,    -0.0015320742,
+      -0.0014046903,    -0.0012685474,    -0.001125814,     -0.00097943726 },
+    { -0.00055432378,   -0.00055472925,   -0.00054783461,   -0.00053276919,
+      -0.00051135791,   -0.00048466062,   -0.00045358928,   -0.00042499689,
+      -0.00036942671,   -0.0003392619,    -0.00030001783,   -0.00025986304,
+      -0.0002197204,    -0.00018116167,   -0.00014691355,   -0.00011279432 },
+    { -0.000064147389,  -0.00006174868,   -0.000054267788,  -0.000047133824,
+      -0.000042927582,  -0.000039477309,  -0.000036340745,  -0.000029687517,
+      -0.000049787737,  -0.000041577889,  -0.000033864744,  -0.000026534748,
+      -0.000019841305,  -0.000014789486,  -0.000013131184,  -0.0000099198869 },
+    { -0.0000062990207, -0.0000072701259, -0.000011984052,  -0.000017348082,
+      -0.000019907106,  -0.000021348773,  -0.000021961965,  -0.000012203576,
+      -0.000010840992,   4.6299544e-7,     5.2588763e-7,     2.7792686e-7,
+      -2.3649704e-7,    -0.0000010897784, -9.171448e-7,     -5.22682e-7 }
+};
+
+/* Second half of the 384-tap IPQF filtering coefficients. */
+static const float ipqf_coeffs2[ATRAC3P_PQF_FIR_LEN][16] = {
+    {  5.22682e-7,       9.171448e-7,      0.0000010897784,  2.3649704e-7,
+      -2.7792686e-7,    -5.2588763e-7,    -4.6299544e-7,     0.000010840992,
+      -0.000012203576,  -0.000021961965,  -0.000021348773,  -0.000019907106,
+      -0.000017348082,  -0.000011984052,  -0.0000072701259, -0.0000062990207 },
+    {  0.0000099198869,  0.000013131184,   0.000014789486,   0.000019841305,
+       0.000026534748,   0.000033864744,   0.000041577889,   0.000049787737,
+      -0.000029687517,  -0.000036340745,  -0.000039477309,  -0.000042927582,
+      -0.000047133824,  -0.000054267788,  -0.00006174868,   -0.000064147389 },
+    {  0.00011279432,    0.00014691355,    0.00018116167,    0.0002197204,
+       0.00025986304,    0.00030001783,    0.0003392619,     0.00036942671,
+      -0.00042499689,   -0.00045358928,   -0.00048466062,   -0.00051135791,
+      -0.00053276919,   -0.00054783461,   -0.00055472925,   -0.00055432378 },
+    {  0.00097943726,    0.001125814,      0.0012685474,     0.0014046903,
+       0.0015320742,     0.0016481078,     0.0017501717,     0.00183808,
+      -0.0018990048,    -0.0019450618,    -0.0019653172,    -0.0019593791,
+      -0.0019252141,    -0.0018608454,    -0.0017651899,    -0.0016376863 },
+    {  0.0042418269,     0.0045959447,     0.0049146507,     0.0051914565,
+       0.0054194843,     0.0055920109,     0.0057025906,     0.0057445862,
+      -0.005714261,     -0.0056034233,    -0.0054091476,    -0.0051268316,
+      -0.0047529764,    -0.0042847847,    -0.0037203205,    -0.0030586775 },
+    {  0.019670162,      0.021246184,      0.022781773,      0.024265358,
+       0.025685543,      0.027031265,      0.028291954,      0.02945766,
+      -0.030518902,     -0.031467363,     -0.032295227,     -0.032995768,
+      -0.033563249,     -0.033992987,     -0.034281436,     -0.034426283 },
+    { -0.018065533,     -0.016444042,     -0.014817309,     -0.013196872,
+      -0.011593862,     -0.010018963,     -0.0084823566,    -0.0069935122,
+       0.0055614738,     0.0041943151,     0.0028992873,     0.001682895,
+       0.00055068987,   -0.00049266568,   -0.001443546,     -0.0022995141 },
+    { -0.0038591374,    -0.0034549395,    -0.0030352892,    -0.0026067684,
+      -0.0021754825,    -0.0017469483,    -0.001326357,     -0.00091862219,
+       0.00052769453,    0.00015771533,   -0.00018855913,   -0.00050782406,
+      -0.00079780724,   -0.0010566821,    -0.001283227,     -0.0014771431 },
+    { -0.00083093712,   -0.00068422867,   -0.00053928449,   -0.00039945057,
+      -0.00026649953,   -0.00014135583,   -0.000025081157,   0.000084307925,
+      -0.00018248901,   -0.00026866337,   -0.00034301577,   -0.00040612387,
+      -0.00045778,      -0.00049731287,   -0.00052537228,   -0.00054488552 },
+    { -0.00008079566,   -0.000053397067,  -0.00002610587,   -0.0000016657353,
+       0.000018532943,   0.000035615325,   0.000049888811,   0.000056296329,
+      -0.000058270442,  -0.000062285151,  -0.000070208509,  -0.000074850752,
+      -0.00007612785,   -0.000073081472,  -0.000068306952,  -0.000064464548 },
+    { -0.000048585582,  -0.000046043948,  -0.000045337845,  -0.000041575615,
+      -0.000036968919,  -0.000032302323,  -0.00002780562,   -0.000023530851,
+       0.0000018111954, -5.7168614e-7,    -0.0000015432918, -0.0000026336629,
+      -0.0000041712024, -0.0000079212787, -0.000010502935,  -0.0000091622824 },
+    { -0.0000070302972, -0.0000063894258, -0.0000046192422, -0.0000032569742,
+      -0.0000023108685, -0.0000014333754, -6.005389e-7,      0.000011838618,
+       0.000010516783,   1.2660377e-8,     3.530856e-8,      3.226247e-8,
+      -4.4400572e-8,    -4.2005411e-7,    -8.0604229e-7,    -5.8336207e-7 }
+};
+
+void ff_atrac3p_ipqf(FFTContext *dct_ctx, Atrac3pIPQFChannelCtx *hist,
+                     const float *in, float *out)
+{
+    int i, s, sb, t, pos_now, pos_next;
+    DECLARE_ALIGNED(32, float, idct_in)[ATRAC3P_SUBBANDS];
+    DECLARE_ALIGNED(32, float, idct_out)[ATRAC3P_SUBBANDS];
+
+    memset(out, 0, ATRAC3P_FRAME_SAMPLES * sizeof(*out));
+
+    for (s = 0; s < ATRAC3P_SUBBAND_SAMPLES; s++) {
+        /* pick up one sample from each subband */
+        for (sb = 0; sb < ATRAC3P_SUBBANDS; sb++)
+            idct_in[sb] = in[sb * ATRAC3P_SUBBAND_SAMPLES + s];
+
+        /* Calculate the sine and cosine part of the PQF using IDCT-IV */
+        dct_ctx->imdct_half(dct_ctx, idct_out, idct_in);
+
+        /* append the result to the history */
+        for (i = 0; i < 8; i++) {
+            hist->buf1[hist->pos][i] = idct_out[i + 8];
+            hist->buf2[hist->pos][i] = idct_out[7 - i];
+        }
+
+        pos_now  = hist->pos;
+        pos_next = mod23_lut[pos_now + 2]; // pos_next = (pos_now + 1) % 23;
+
+        for (t = 0; t < ATRAC3P_PQF_FIR_LEN; t++) {
+            for (i = 0; i < 8; i++) {
+                out[s * 16 + i + 0] += hist->buf1[pos_now][i]  * ipqf_coeffs1[t][i] +
+                                       hist->buf2[pos_next][i] * ipqf_coeffs2[t][i];
+                out[s * 16 + i + 8] += hist->buf1[pos_now][7 - i]  * ipqf_coeffs1[t][i + 8] +
+                                       hist->buf2[pos_next][7 - i] * ipqf_coeffs2[t][i + 8];
+            }
+
+            pos_now  = mod23_lut[pos_next + 2]; // pos_now  = (pos_now  + 2) % 23;
+            pos_next = mod23_lut[pos_now + 2];  // pos_next = (pos_next + 2) % 23;
+        }
+
+        hist->pos = mod23_lut[hist->pos]; // hist->pos = (hist->pos - 1) % 23;
+    }
+}

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/avcodec.h

@@ -57,6 +57,15 @@
 
 #include "version.h"
 
+#if FF_API_FAST_MALLOC
+// to provide fast_*alloc
+#ifdef __FRAMEWORK__
+#include "mem.h"
+#else
+#include "libavutil/mem.h"
+#endif
+#endif
+
 /**
  * @defgroup libavc Encoding/Decoding Library
  * @{
@@ -120,7 +129,9 @@ enum AVCodecID {
     /* video codecs */
     AV_CODEC_ID_MPEG1VIDEO,
     AV_CODEC_ID_MPEG2VIDEO, ///< preferred ID for MPEG-1/2 video decoding
+#if FF_API_XVMC
     AV_CODEC_ID_MPEG2VIDEO_XVMC,
+#endif /* FF_API_XVMC */
     AV_CODEC_ID_H261,
     AV_CODEC_ID_H263,
     AV_CODEC_ID_RV10,
@@ -291,6 +302,9 @@ enum AVCodecID {
     AV_CODEC_ID_ESCAPE130_DEPRECATED,
     AV_CODEC_ID_G2M_DEPRECATED,
     AV_CODEC_ID_WEBP_DEPRECATED,
+    AV_CODEC_ID_HNM4_VIDEO,
+    AV_CODEC_ID_HEVC_DEPRECATED,
+    AV_CODEC_ID_FIC,
 
     AV_CODEC_ID_BRENDER_PIX= MKBETAG('B','P','I','X'),
     AV_CODEC_ID_Y41P       = MKBETAG('Y','4','1','P'),
@@ -317,7 +331,8 @@ enum AVCodecID {
     AV_CODEC_ID_SNOW       = MKBETAG('S','N','O','W'),
     AV_CODEC_ID_WEBP       = MKBETAG('W','E','B','P'),
     AV_CODEC_ID_SMVJPEG    = MKBETAG('S','M','V','J'),
-    AV_CODEC_ID_H265       = MKBETAG('H','2','6','5'),
+    AV_CODEC_ID_HEVC       = MKBETAG('H','2','6','5'),
+#define AV_CODEC_ID_H265 AV_CODEC_ID_HEVC
 
     /* various PCM "codecs" */
     AV_CODEC_ID_FIRST_AUDIO = 0x10000,     ///< A dummy id pointing at the start of audio codecs
@@ -391,6 +406,7 @@ enum AVCodecID {
     AV_CODEC_ID_ADPCM_IMA_OKI = MKBETAG('O','K','I',' '),
     AV_CODEC_ID_ADPCM_DTK  = MKBETAG('D','T','K',' '),
     AV_CODEC_ID_ADPCM_IMA_RAD = MKBETAG('R','A','D',' '),
+    AV_CODEC_ID_ADPCM_G726LE = MKBETAG('6','2','7','G'),
 
     /* AMR */
     AV_CODEC_ID_AMR_NB = 0x12000,
@@ -517,6 +533,7 @@ enum AVCodecID {
     AV_CODEC_ID_OTF        = MKBETAG( 0 ,'O','T','F'),
     AV_CODEC_ID_SMPTE_KLV  = MKBETAG('K','L','V','A'),
     AV_CODEC_ID_DVD_NAV    = MKBETAG('D','N','A','V'),
+    AV_CODEC_ID_TIMED_ID3  = MKBETAG('T','I','D','3'),
 
 
     AV_CODEC_ID_PROBE = 0x19000, ///< codec_id is not known (like AV_CODEC_ID_NONE) but lavf should attempt to identify it
@@ -639,16 +656,26 @@ enum AVColorPrimaries{
     AVCOL_PRI_SMPTE170M   = 6, ///< also ITU-R BT601-6 525 / ITU-R BT1358 525 / ITU-R BT1700 NTSC
     AVCOL_PRI_SMPTE240M   = 7, ///< functionally identical to above
     AVCOL_PRI_FILM        = 8,
+    AVCOL_PRI_BT2020      = 9, ///< ITU-R BT2020
     AVCOL_PRI_NB             , ///< Not part of ABI
 };
 
 enum AVColorTransferCharacteristic{
-    AVCOL_TRC_BT709       = 1, ///< also ITU-R BT1361
+    AVCOL_TRC_BT709        =  1, ///< also ITU-R BT1361
-    AVCOL_TRC_UNSPECIFIED = 2,
+    AVCOL_TRC_UNSPECIFIED  =  2,
-    AVCOL_TRC_GAMMA22     = 4, ///< also ITU-R BT470M / ITU-R BT1700 625 PAL & SECAM
+    AVCOL_TRC_GAMMA22      =  4, ///< also ITU-R BT470M / ITU-R BT1700 625 PAL & SECAM
-    AVCOL_TRC_GAMMA28     = 5, ///< also ITU-R BT470BG
+    AVCOL_TRC_GAMMA28      =  5, ///< also ITU-R BT470BG
-    AVCOL_TRC_SMPTE240M   = 7,
+    AVCOL_TRC_SMPTE170M    =  6, ///< also ITU-R BT601-6 525 or 625 / ITU-R BT1358 525 or 625 / ITU-R BT1700 NTSC
-    AVCOL_TRC_NB             , ///< Not part of ABI
+    AVCOL_TRC_SMPTE240M    =  7,
+    AVCOL_TRC_LINEAR       =  8, ///< "Linear transfer characteristics"
+    AVCOL_TRC_LOG          =  9, ///< "Logarithmic transfer characteristic (100:1 range)"
+    AVCOL_TRC_LOG_SQRT     = 10, ///< "Logarithmic transfer characteristic (100 * Sqrt( 10 ) : 1 range)"
+    AVCOL_TRC_IEC61966_2_4 = 11, ///< IEC 61966-2-4
+    AVCOL_TRC_BT1361_ECG   = 12, ///< ITU-R BT1361 Extended Colour Gamut
+    AVCOL_TRC_IEC61966_2_1 = 13, ///< IEC 61966-2-1 (sRGB or sYCC)
+    AVCOL_TRC_BT2020_10    = 14, ///< ITU-R BT2020 for 10 bit system
+    AVCOL_TRC_BT2020_12    = 15, ///< ITU-R BT2020 for 12 bit system
+    AVCOL_TRC_NB               , ///< Not part of ABI
 };
 
 /**
@@ -690,7 +717,12 @@ typedef struct RcOverride{
     float quality_factor;
 } RcOverride;
 
+#if FF_API_MAX_BFRAMES
+/**
+ * @deprecated there is no libavcodec-wide limit on the number of B-frames
+ */
 #define FF_MAX_B_FRAMES 16
+#endif
 
 /* encoding support
    These flags can be passed in AVCodecContext.flags before initialization.
@@ -704,6 +736,7 @@ typedef struct RcOverride{
 #define CODEC_FLAG_UNALIGNED 0x0001
 #define CODEC_FLAG_QSCALE 0x0002  ///< Use fixed qscale.
 #define CODEC_FLAG_4MV    0x0004  ///< 4 MV per MB allowed / advanced prediction for H.263.
+#define CODEC_FLAG_OUTPUT_CORRUPT 0x0008 ///< Output even those frames that might be corrupted
 #define CODEC_FLAG_QPEL   0x0010  ///< Use qpel MC.
 #define CODEC_FLAG_GMC    0x0020  ///< Use GMC.
 #define CODEC_FLAG_MV0    0x0040  ///< Always try a MB with MV=<0,0>.
@@ -716,7 +749,13 @@ typedef struct RcOverride{
 #define CODEC_FLAG_PASS1           0x0200   ///< Use internal 2pass ratecontrol in first pass mode.
 #define CODEC_FLAG_PASS2           0x0400   ///< Use internal 2pass ratecontrol in second pass mode.
 #define CODEC_FLAG_GRAY            0x2000   ///< Only decode/encode grayscale.
-#define CODEC_FLAG_EMU_EDGE        0x4000   ///< Don't draw edges.
+#if FF_API_EMU_EDGE
+/**
+ * @deprecated edges are not used/required anymore. I.e. this flag is now always
+ * set.
+ */
+#define CODEC_FLAG_EMU_EDGE        0x4000
+#endif
 #define CODEC_FLAG_PSNR            0x8000   ///< error[?] variables will be set during encoding.
 #define CODEC_FLAG_TRUNCATED       0x00010000 /** Input bitstream might be truncated at a random
                                                   location instead of only at frame boundaries. */
@@ -754,8 +793,16 @@ typedef struct RcOverride{
  */
 #define CODEC_CAP_DR1             0x0002
 #define CODEC_CAP_TRUNCATED       0x0008
-/* Codec can export data for HW decoding (XvMC). */
+#if FF_API_XVMC
+/* Codec can export data for HW decoding. This flag indicates that
+ * the codec would call get_format() with list that might contain HW accelerated
+ * pixel formats (XvMC, VDPAU, VAAPI, etc). The application can pick any of them
+ * including raw image format.
+ * The application can use the passed context to determine bitstream version,
+ * chroma format, resolution etc.
+ */
 #define CODEC_CAP_HWACCEL         0x0010
+#endif /* FF_API_XVMC */
 /**
  * Encoder or decoder requires flushing with NULL input at the end in order to
  * give the complete and correct output.
@@ -812,12 +859,12 @@ typedef struct RcOverride{
  * Codec should fill in channel configuration and samplerate instead of container
  */
 #define CODEC_CAP_CHANNEL_CONF     0x0400
-
+#if FF_API_NEG_LINESIZES
 /**
- * Codec is able to deal with negative linesizes
+ * @deprecated no codecs use this capability
  */
 #define CODEC_CAP_NEG_LINESIZES    0x0800
-
+#endif
 /**
  * Codec supports frame-level multithreading.
  */
@@ -847,6 +894,7 @@ typedef struct RcOverride{
  */
 #define CODEC_CAP_LOSSLESS         0x80000000
 
+#if FF_API_MB_TYPE
 //The following defines may change, don't expect compatibility if you use them.
 #define MB_TYPE_INTRA4x4   0x0001
 #define MB_TYPE_INTRA16x16 0x0002 //FIXME H.264-specific
@@ -870,6 +918,7 @@ typedef struct RcOverride{
 #define MB_TYPE_QUANT      0x00010000
 #define MB_TYPE_CBP        0x00020000
 //Note bits 24-31 are reserved for codec specific use (h264 ref0, mpeg1 0mv, ...)
+#endif
 
 /**
  * Pan Scan area.
@@ -900,10 +949,12 @@ typedef struct AVPanScan{
     int16_t position[3][2];
 }AVPanScan;
 
+#if FF_API_QSCALE_TYPE
 #define FF_QSCALE_TYPE_MPEG1 0
 #define FF_QSCALE_TYPE_MPEG2 1
 #define FF_QSCALE_TYPE_H264  2
 #define FF_QSCALE_TYPE_VP56  3
+#endif
 
 #if FF_API_GET_BUFFER
 #define FF_BUFFER_TYPE_INTERNAL 1
@@ -1026,6 +1077,13 @@ enum AVPacketSideDataType {
      * follow the timestamp specifier of a WebVTT cue.
      */
     AV_PKT_DATA_WEBVTT_SETTINGS,
+
+    /**
+     * A list of zero terminated key/value strings. There is no end marker for
+     * the list, so it is required to rely on the side data size to stop. This
+     * side data includes updated metadata which appeared in the stream.
+     */
+    AV_PKT_DATA_METADATA_UPDATE,
 };
 
 /**
@@ -1336,7 +1394,9 @@ typedef struct AVCodecContext {
      */
     int coded_width, coded_height;
 
+#if FF_API_ASPECT_EXTENDED
 #define FF_ASPECT_EXTENDED 15
+#endif
 
     /**
      * the number of pictures in a group of pictures, or 0 for intra_only
@@ -1663,12 +1723,15 @@ typedef struct AVCodecContext {
 #define SLICE_FLAG_ALLOW_FIELD    0x0002 ///< allow draw_horiz_band() with field slices (MPEG2 field pics)
 #define SLICE_FLAG_ALLOW_PLANE    0x0004 ///< allow draw_horiz_band() with 1 component at a time (SVQ1)
 
+#if FF_API_XVMC
     /**
      * XVideo Motion Acceleration
      * - encoding: forbidden
      * - decoding: set by decoder
+     * @deprecated XvMC doesn't need it anymore.
      */
-    int xvmc_acceleration;
+    attribute_deprecated int xvmc_acceleration;
+#endif /* FF_API_XVMC */
 
     /**
      * macroblock decision mode
@@ -2115,12 +2178,11 @@ typedef struct AVCodecContext {
      * If AV_GET_BUFFER_FLAG_REF is set in flags then the frame may be reused
      * (read and/or written to if it is writable) later by libavcodec.
      *
-     * If CODEC_FLAG_EMU_EDGE is not set in s->flags, the buffer must contain an
-     * edge of the size returned by avcodec_get_edge_width() on all sides.
-     *
      * avcodec_align_dimensions2() should be used to find the required width and
      * height, as they normally need to be rounded up to the next multiple of 16.
      *
+     * Some decoders do not support linesizes changing between frames.
+     *
      * If frame multithreading is used and thread_safe_callbacks is set,
      * this callback may be called from a different thread, but not from more
      * than one at once. Does not need to be reentrant.
@@ -2403,12 +2465,16 @@ typedef struct AVCodecContext {
      */
     int workaround_bugs;
 #define FF_BUG_AUTODETECT       1  ///< autodetection
+#if FF_API_OLD_MSMPEG4
 #define FF_BUG_OLD_MSMPEG4      2
+#endif
 #define FF_BUG_XVID_ILACE       4
 #define FF_BUG_UMP4             8
 #define FF_BUG_NO_PADDING       16
 #define FF_BUG_AMV              32
+#if FF_API_AC_VLC
 #define FF_BUG_AC_VLC           0  ///< Will be removed, libavcodec can now handle these non-compliant files by default.
+#endif
 #define FF_BUG_QPEL_CHROMA      64
 #define FF_BUG_STD_QPEL         128
 #define FF_BUG_QPEL_CHROMA2     256
@@ -2458,7 +2524,12 @@ typedef struct AVCodecContext {
 #define FF_DEBUG_BITSTREAM   4
 #define FF_DEBUG_MB_TYPE     8
 #define FF_DEBUG_QP          16
+#if FF_API_DEBUG_MV
+/**
+ * @deprecated this option does nothing
+ */
 #define FF_DEBUG_MV          32
+#endif
 #define FF_DEBUG_DCT_COEFF   0x00000040
 #define FF_DEBUG_SKIP        0x00000080
 #define FF_DEBUG_STARTCODE   0x00000100
@@ -2466,13 +2537,17 @@ typedef struct AVCodecContext {
 #define FF_DEBUG_ER          0x00000400
 #define FF_DEBUG_MMCO        0x00000800
 #define FF_DEBUG_BUGS        0x00001000
-#define FF_DEBUG_VIS_QP      0x00002000
+#if FF_API_DEBUG_MV
-#define FF_DEBUG_VIS_MB_TYPE 0x00004000
+#define FF_DEBUG_VIS_QP      0x00002000 ///< only access through AVOptions from outside libavcodec
+#define FF_DEBUG_VIS_MB_TYPE 0x00004000 ///< only access through AVOptions from outside libavcodec
+#endif
 #define FF_DEBUG_BUFFERS     0x00008000
 #define FF_DEBUG_THREADS     0x00010000
 
+#if FF_API_DEBUG_MV
     /**
      * debug
+     * Code outside libavcodec should access this field using AVOptions
      * - encoding: Set by user.
      * - decoding: Set by user.
      */
@@ -2480,6 +2555,7 @@ typedef struct AVCodecContext {
 #define FF_DEBUG_VIS_MV_P_FOR  0x00000001 //visualize forward predicted MVs of P frames
 #define FF_DEBUG_VIS_MV_B_FOR  0x00000002 //visualize forward predicted MVs of B frames
 #define FF_DEBUG_VIS_MV_B_BACK 0x00000004 //visualize backward predicted MVs of B frames
+#endif
 
     /**
      * Error recognition; may misdetect some more or less valid parts as errors.
@@ -2487,14 +2563,21 @@ typedef struct AVCodecContext {
      * - decoding: Set by user.
      */
     int err_recognition;
-#define AV_EF_CRCCHECK  (1<<0)
-#define AV_EF_BITSTREAM (1<<1)
-#define AV_EF_BUFFER    (1<<2)
-#define AV_EF_EXPLODE   (1<<3)
 
-#define AV_EF_CAREFUL    (1<<16)
+/**
-#define AV_EF_COMPLIANT  (1<<17)
+ * Verify checksums embedded in the bitstream (could be of either encoded or
-#define AV_EF_AGGRESSIVE (1<<18)
+ * decoded data, depending on the codec) and print an error message on mismatch.
+ * If AV_EF_EXPLODE is also set, a mismatching checksum will result in the
+ * decoder returning an error.
+ */
+#define AV_EF_CRCCHECK  (1<<0)
+#define AV_EF_BITSTREAM (1<<1)          ///< detect bitstream specification deviations
+#define AV_EF_BUFFER    (1<<2)          ///< detect improper bitstream length
+#define AV_EF_EXPLODE   (1<<3)          ///< abort decoding on minor error detection
+
+#define AV_EF_CAREFUL    (1<<16)        ///< consider things that violate the spec, are fast to calculate and have not been seen in the wild as errors
+#define AV_EF_COMPLIANT  (1<<17)        ///< consider all spec non compliancies as errors
+#define AV_EF_AGGRESSIVE (1<<18)        ///< consider things that a sane encoder should not do as an error
 
 
     /**
@@ -2566,7 +2649,9 @@ typedef struct AVCodecContext {
 #define FF_IDCT_SIMPLEVIS     18
 #define FF_IDCT_FAAN          20
 #define FF_IDCT_SIMPLENEON    22
+#if FF_API_ARCH_ALPHA
 #define FF_IDCT_SIMPLEALPHA   23
+#endif
 
     /**
      * bits per sample/pixel from the demuxer (needed for huffyuv).
@@ -2596,7 +2681,7 @@ typedef struct AVCodecContext {
     /**
      * the picture in the bitstream
      * - encoding: Set by libavcodec.
-     * - decoding: Set by libavcodec.
+     * - decoding: unused
      */
     AVFrame *coded_frame;
 
@@ -2668,13 +2753,13 @@ typedef struct AVCodecContext {
      */
     int (*execute2)(struct AVCodecContext *c, int (*func)(struct AVCodecContext *c2, void *arg, int jobnr, int threadnr), void *arg2, int *ret, int count);
 
+#if FF_API_THREAD_OPAQUE
     /**
-     * thread opaque
+     * @deprecated this field should not be used from outside of lavc
-     * Can be used by execute() to store some per AVCodecContext stuff.
-     * - encoding: set by execute()
-     * - decoding: set by execute()
      */
+    attribute_deprecated
     void *thread_opaque;
+#endif
 
     /**
      * noise vs. sse weight for the nsse comparsion function
@@ -2761,6 +2846,11 @@ typedef struct AVCodecContext {
 #define FF_PROFILE_JPEG2000_DCINEMA_2K              3
 #define FF_PROFILE_JPEG2000_DCINEMA_4K              4
 
+
+#define FF_PROFILE_HEVC_MAIN                        1
+#define FF_PROFILE_HEVC_MAIN_10                     2
+#define FF_PROFILE_HEVC_MAIN_STILL_PICTURE          3
+
     /**
      * level
      * - encoding: Set by user.
@@ -2801,21 +2891,22 @@ typedef struct AVCodecContext {
     uint8_t *subtitle_header;
     int subtitle_header_size;
 
+#if FF_API_ERROR_RATE
     /**
-     * Simulates errors in the bitstream to test error concealment.
+     * @deprecated use the 'error_rate' private AVOption of the mpegvideo
-     * - encoding: Set by user.
+     * encoders
-     * - decoding: unused
      */
+    attribute_deprecated
     int error_rate;
+#endif
 
+#if FF_API_CODEC_PKT
     /**
-     * Current packet as passed into the decoder, to avoid having
+     * @deprecated this field is not supposed to be accessed from outside lavc
-     * to pass the packet into every function. Currently only valid
-     * inside lavc and get/release_buffer callbacks.
-     * - decoding: set by avcodec_decode_*, read by get_buffer() for setting pkt_pts
-     * - encoding: unused
      */
+    attribute_deprecated
     AVPacket *pkt;
+#endif
 
     /**
      * VBV delay coded in the last frame (in periods of a 27 MHz clock).
@@ -2896,6 +2987,34 @@ typedef struct AVCodecContext {
      * - encoding: unused
      */
     int skip_alpha;
+
+    /**
+     * Number of samples to skip after a discontinuity
+     * - decoding: unused
+     * - encoding: set by libavcodec
+     */
+    int seek_preroll;
+
+#if !FF_API_DEBUG_MV
+    /**
+     * debug motion vectors
+     * Code outside libavcodec should access this field using AVOptions
+     * - encoding: Set by user.
+     * - decoding: Set by user.
+     */
+    int debug_mv;
+#define FF_DEBUG_VIS_MV_P_FOR  0x00000001 //visualize forward predicted MVs of P frames
+#define FF_DEBUG_VIS_MV_B_FOR  0x00000002 //visualize forward predicted MVs of B frames
+#define FF_DEBUG_VIS_MV_B_BACK 0x00000004 //visualize backward predicted MVs of B frames
+#endif
+
+    /**
+     * custom intra quantization matrix
+     * Code outside libavcodec should access this field using av_codec_g/set_chroma_intra_matrix()
+     * - encoding: Set by user, can be NULL.
+     * - decoding: unused.
+     */
+    uint16_t *chroma_intra_matrix;
 } AVCodecContext;
 
 AVRational av_codec_get_pkt_timebase         (const AVCodecContext *avctx);
@@ -2907,6 +3026,12 @@ void                     av_codec_set_codec_descriptor(AVCodecContext *avctx, co
 int  av_codec_get_lowres(const AVCodecContext *avctx);
 void av_codec_set_lowres(AVCodecContext *avctx, int val);
 
+int  av_codec_get_seek_preroll(const AVCodecContext *avctx);
+void av_codec_set_seek_preroll(AVCodecContext *avctx, int val);
+
+uint16_t *av_codec_get_chroma_intra_matrix(const AVCodecContext *avctx);
+void av_codec_set_chroma_intra_matrix(AVCodecContext *avctx, uint16_t *val);
+
 /**
  * AVProfile.
  */
@@ -3014,6 +3139,10 @@ typedef struct AVCodec {
     void (*flush)(AVCodecContext *);
 } AVCodec;
 
+int av_codec_get_max_lowres(const AVCodec *codec);
+
+struct MpegEncContext;
+
 /**
  * AVHWAccel.
  */
@@ -3075,6 +3204,7 @@ typedef struct AVHWAccel {
      *
      * Meaningful slice information (codec specific) is guaranteed to
      * be parsed at this point. This function is mandatory.
+     * The only exception is XvMC, that works on MB level.
      *
      * @param avctx the codec context
      * @param buf the slice data buffer base
@@ -3102,6 +3232,17 @@ typedef struct AVHWAccel {
      * AVCodecContext.release_buffer().
      */
     int priv_data_size;
+
+    /**
+     * Called for every Macroblock in a slice.
+     *
+     * XvMC uses it to replace the ff_MPV_decode_mb().
+     * Instead of decoding to raw picture, MB parameters are
+     * stored in an array provided by the video driver.
+     *
+     * @param s the mpeg context
+     */
+    void (*decode_mb)(struct MpegEncContext *s);
 } AVHWAccel;
 
 /**
@@ -3225,40 +3366,6 @@ void avcodec_register(AVCodec *codec);
  */
 void avcodec_register_all(void);
 
-
-#if FF_API_ALLOC_CONTEXT
-/**
- * Allocate an AVCodecContext and set its fields to default values.  The
- * resulting struct can be deallocated by simply calling av_free().
- *
- * @return An AVCodecContext filled with default values or NULL on failure.
- * @see avcodec_get_context_defaults
- *
- * @deprecated use avcodec_alloc_context3()
- */
-attribute_deprecated
-AVCodecContext *avcodec_alloc_context(void);
-
-/** THIS FUNCTION IS NOT YET PART OF THE PUBLIC API!
- *  we WILL change its arguments and name a few times! */
-attribute_deprecated
-AVCodecContext *avcodec_alloc_context2(enum AVMediaType);
-
-/**
- * Set the fields of the given AVCodecContext to default values.
- *
- * @param s The AVCodecContext of which the fields should be set to default values.
- * @deprecated use avcodec_get_context_defaults3
- */
-attribute_deprecated
-void avcodec_get_context_defaults(AVCodecContext *s);
-
-/** THIS FUNCTION IS NOT YET PART OF THE PUBLIC API!
- *  we WILL change its arguments and name a few times! */
-attribute_deprecated
-void avcodec_get_context_defaults2(AVCodecContext *s, enum AVMediaType);
-#endif
-
 /**
  * Allocate an AVCodecContext and set its fields to default values.  The
  * resulting struct can be deallocated by calling avcodec_close() on it followed
@@ -3324,20 +3431,21 @@ const AVClass *avcodec_get_subtitle_rect_class(void);
  */
 int avcodec_copy_context(AVCodecContext *dest, const AVCodecContext *src);
 
+#if FF_API_AVFRAME_LAVC
 /**
- * Allocate an AVFrame and set its fields to default values.  The resulting
+ * @deprecated use av_frame_alloc()
- * struct must be freed using avcodec_free_frame().
- *
- * @return An AVFrame filled with default values or NULL on failure.
- * @see avcodec_get_frame_defaults
  */
+attribute_deprecated
 AVFrame *avcodec_alloc_frame(void);
 
 /**
  * Set the fields of the given AVFrame to default values.
  *
  * @param frame The AVFrame of which the fields should be set to default values.
+ *
+ * @deprecated use av_frame_unref()
  */
+attribute_deprecated
 void avcodec_get_frame_defaults(AVFrame *frame);
 
 /**
@@ -3349,41 +3457,11 @@ void avcodec_get_frame_defaults(AVFrame *frame);
  * @warning this function does NOT free the data buffers themselves
  * (it does not know how, since they might have been allocated with
  *  a custom get_buffer()).
- */
-void avcodec_free_frame(AVFrame **frame);
-
-#if FF_API_AVCODEC_OPEN
-/**
- * Initialize the AVCodecContext to use the given AVCodec. Prior to using this
- * function the context has to be allocated.
  *
- * The functions avcodec_find_decoder_by_name(), avcodec_find_encoder_by_name(),
+ * @deprecated use av_frame_free()
- * avcodec_find_decoder() and avcodec_find_encoder() provide an easy way for
- * retrieving a codec.
- *
- * @warning This function is not thread safe!
- *
- * @code
- * avcodec_register_all();
- * codec = avcodec_find_decoder(AV_CODEC_ID_H264);
- * if (!codec)
- *     exit(1);
- *
- * context = avcodec_alloc_context3(codec);
- *
- * if (avcodec_open(context, codec) < 0)
- *     exit(1);
- * @endcode
- *
- * @param avctx The context which will be set up to use the given codec.
- * @param codec The codec to use within the context.
- * @return zero on success, a negative value on error
- * @see avcodec_alloc_context3, avcodec_find_decoder, avcodec_find_encoder, avcodec_close
- *
- * @deprecated use avcodec_open2
  */
 attribute_deprecated
-int avcodec_open(AVCodecContext *avctx, AVCodec *codec);
+void avcodec_free_frame(AVFrame **frame);
 #endif
 
 /**
@@ -3575,6 +3653,24 @@ int av_packet_merge_side_data(AVPacket *pkt);
 
 int av_packet_split_side_data(AVPacket *pkt);
 
+/**
+ * Pack a dictionary for use in side_data.
+ *
+ * @param dict The dictionary to pack.
+ * @param size pointer to store the size of the returned data
+ * @return pointer to data if successful, NULL otherwise
+ */
+uint8_t *av_packet_pack_dictionary(AVDictionary *dict, int *size);
+/**
+ * Unpack a dictionary from side_data.
+ *
+ * @param data data from side_data
+ * @param size size of the data
+ * @param dict the metadata storage dictionary
+ * @return 0 on success, < 0 on failure
+ */
+int av_packet_unpack_dictionary(const uint8_t *data, int size, AVDictionary **dict);
+
 
 /**
  * Convenience function to free all the side data stored.
@@ -3674,14 +3770,20 @@ attribute_deprecated int avcodec_default_reget_buffer(AVCodecContext *s, AVFrame
  */
 int avcodec_default_get_buffer2(AVCodecContext *s, AVFrame *frame, int flags);
 
+#if FF_API_EMU_EDGE
 /**
  * Return the amount of padding in pixels which the get_buffer callback must
  * provide around the edge of the image for codecs which do not have the
  * CODEC_FLAG_EMU_EDGE flag.
  *
  * @return Required padding in pixels.
+ *
+ * @deprecated CODEC_FLAG_EMU_EDGE is deprecated, so this function is no longer
+ * needed
  */
+attribute_deprecated
 unsigned avcodec_get_edge_width(void);
+#endif
 
 /**
  * Modify width and height values so that they will result in a memory
@@ -3689,8 +3791,6 @@ unsigned avcodec_get_edge_width(void);
  * padding.
  *
  * May only be used if a codec with CODEC_CAP_DR1 has been opened.
- * If CODEC_FLAG_EMU_EDGE is not set, the dimensions must have been increased
- * according to avcodec_get_edge_width() before.
  */
 void avcodec_align_dimensions(AVCodecContext *s, int *width, int *height);
 
@@ -3700,8 +3800,6 @@ void avcodec_align_dimensions(AVCodecContext *s, int *width, int *height);
  * line sizes are a multiple of the respective linesize_align[i].
  *
  * May only be used if a codec with CODEC_CAP_DR1 has been opened.
- * If CODEC_FLAG_EMU_EDGE is not set, the dimensions must have been increased
- * according to avcodec_get_edge_width() before.
  */
 void avcodec_align_dimensions2(AVCodecContext *s, int *width, int *height,
                                int linesize_align[AV_NUM_DATA_POINTERS]);
@@ -3792,19 +3890,25 @@ attribute_deprecated int avcodec_decode_audio3(AVCodecContext *avctx, int16_t *s
  * Decode the audio frame of size avpkt->size from avpkt->data into frame.
  *
  * Some decoders may support multiple frames in a single AVPacket. Such
- * decoders would then just decode the first frame. In this case,
+ * decoders would then just decode the first frame and the return value would be
- * avcodec_decode_audio4 has to be called again with an AVPacket containing
+ * less than the packet size. In this case, avcodec_decode_audio4 has to be
- * the remaining data in order to decode the second frame, etc...
+ * called again with an AVPacket containing the remaining data in order to
- * Even if no frames are returned, the packet needs to be fed to the decoder
+ * decode the second frame, etc...  Even if no frames are returned, the packet
- * with remaining data until it is completely consumed or an error occurs.
+ * needs to be fed to the decoder with remaining data until it is completely
+ * consumed or an error occurs.
+ *
+ * Some decoders (those marked with CODEC_CAP_DELAY) have a delay between input
+ * and output. This means that for some packets they will not immediately
+ * produce decoded output and need to be flushed at the end of decoding to get
+ * all the decoded data. Flushing is done by calling this function with packets
+ * with avpkt->data set to NULL and avpkt->size set to 0 until it stops
+ * returning samples. It is safe to flush even those decoders that are not
+ * marked with CODEC_CAP_DELAY, then no samples will be returned.
  *
  * @warning The input buffer, avpkt->data must be FF_INPUT_BUFFER_PADDING_SIZE
  *          larger than the actual read bytes because some optimized bitstream
  *          readers read 32 or 64 bits at once and could read over the end.
  *
- * @note You might have to align the input buffer. The alignment requirements
- *       depend on the CPU and the decoder.
- *
  * @param      avctx the codec context
  * @param[out] frame The AVFrame in which to store decoded audio samples.
  *                   The decoder will allocate a buffer for the decoded frame by
@@ -3819,7 +3923,10 @@ attribute_deprecated int avcodec_decode_audio3(AVCodecContext *avctx, int16_t *s
  *                   next call to this function or until closing or flushing the
  *                   decoder. The caller may not write to it.
  * @param[out] got_frame_ptr Zero if no frame could be decoded, otherwise it is
- *                           non-zero.
+ *                           non-zero. Note that this field being set to zero
+ *                           does not mean that an error has occurred. For
+ *                           decoders with CODEC_CAP_DELAY set, no given decode
+ *                           call is guaranteed to produce a frame.
  * @param[in]  avpkt The input AVPacket containing the input buffer.
  *                   At least avpkt->data and avpkt->size should be set. Some
  *                   decoders might also require additional fields to be set.
@@ -3842,13 +3949,6 @@ int avcodec_decode_audio4(AVCodecContext *avctx, AVFrame *frame,
  * @warning The end of the input buffer buf should be set to 0 to ensure that
  * no overreading happens for damaged MPEG streams.
  *
- * @note You might have to align the input buffer avpkt->data.
- * The alignment requirements depend on the CPU: on some CPUs it isn't
- * necessary at all, on others it won't work at all if not aligned and on others
- * it will work but it will have an impact on performance.
- *
- * In practice, avpkt->data should have 4 byte alignment at minimum.
- *
  * @note Codecs which have the CODEC_CAP_DELAY capability set have a delay
  * between input and output, these need to be fed with avpkt->data=NULL,
  * avpkt->size=0 at the end to return the remaining frames.
@@ -3891,6 +3991,14 @@ int avcodec_decode_video2(AVCodecContext *avctx, AVFrame *picture,
  * and reusing a get_buffer written for video codecs would probably perform badly
  * due to a potentially very different allocation pattern.
  *
+ * Some decoders (those marked with CODEC_CAP_DELAY) have a delay between input
+ * and output. This means that for some packets they will not immediately
+ * produce decoded output and need to be flushed at the end of decoding to get
+ * all the decoded data. Flushing is done by calling this function with packets
+ * with avpkt->data set to NULL and avpkt->size set to 0 until it stops
+ * returning subtitles. It is safe to flush even those decoders that are not
+ * marked with CODEC_CAP_DELAY, then no subtitles will be returned.
+ *
  * @param avctx the codec context
  * @param[out] sub The AVSubtitle in which the decoded subtitle will be stored, must be
                    freed with avsubtitle_free if *got_sub_ptr is set.
@@ -4655,12 +4763,20 @@ enum AVPixelFormat avcodec_default_get_format(struct AVCodecContext *s, const en
  * @}
  */
 
+#if FF_API_SET_DIMENSIONS
+/**
+ * @deprecated this function is not supposed to be used from outside of lavc
+ */
+attribute_deprecated
 void avcodec_set_dimensions(AVCodecContext *s, int width, int height);
+#endif
 
 /**
  * Put a string representing the codec tag codec_tag in buf.
  *
+ * @param buf       buffer to place codec tag in
  * @param buf_size size in bytes of buf
+ * @param codec_tag codec tag to assign
  * @return the length of the string that would have been generated if
  * enough space had been available, excluding the trailing null
  */
@@ -4851,27 +4967,6 @@ AVBitStreamFilter *av_bitstream_filter_next(AVBitStreamFilter *f);
 
 /* memory */
 
-/**
- * Reallocate the given block if it is not large enough, otherwise do nothing.
- *
- * @see av_realloc
- */
-void *av_fast_realloc(void *ptr, unsigned int *size, size_t min_size);
-
-/**
- * Allocate a buffer, reusing the given one if large enough.
- *
- * Contrary to av_fast_realloc the current buffer contents might not be
- * preserved and on error the old buffer is freed, thus no special
- * handling to avoid memleaks is necessary.
- *
- * @param ptr pointer to pointer to already allocated buffer, overwritten with pointer to new buffer
- * @param size size of the buffer *ptr points to
- * @param min_size minimum size of *ptr buffer after returning, *ptr will be NULL and
- *                 *size 0 if an error occurred.
- */
-void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size);
-
 /**
  * Same behaviour av_fast_malloc but the buffer has additional
  * FF_INPUT_BUFFER_PADDING_SIZE at the end which will always be 0.

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/avpacket.c

@@ -237,6 +237,7 @@ int av_copy_packet_side_data(AVPacket *pkt, AVPacket *src)
             pkt->side_data[i].type = src->side_data[i].type;
         }
     }
+    pkt->side_data_elems = src->side_data_elems;
     return 0;
 
 failed_alloc:
@@ -380,7 +381,7 @@ FF_ENABLE_DEPRECATION_WARNINGS
 int av_packet_split_side_data(AVPacket *pkt){
     if (!pkt->side_data_elems && pkt->size >12 && AV_RB64(pkt->data + pkt->size - 8) == FF_MERGE_MARKER){
         int i;
-        unsigned int size, orig_pktsize = pkt->size;
+        unsigned int size;
         uint8_t *p;
 
         p = pkt->data + pkt->size - 8 - 5;
@@ -401,7 +402,7 @@ int av_packet_split_side_data(AVPacket *pkt){
         for (i=0; ; i++){
             size= AV_RB32(p);
             av_assert0(size<=INT_MAX && p - pkt->data >= size);
-            pkt->side_data[i].data = av_malloc(size + FF_INPUT_BUFFER_PADDING_SIZE);
+            pkt->side_data[i].data = av_mallocz(size + FF_INPUT_BUFFER_PADDING_SIZE);
             pkt->side_data[i].size = size;
             pkt->side_data[i].type = p[4]&127;
             if (!pkt->side_data[i].data)
@@ -413,19 +414,72 @@ int av_packet_split_side_data(AVPacket *pkt){
             p-= size+5;
         }
         pkt->size -= 8;
-        /* FFMIN() prevents overflow in case the packet wasn't allocated with
-         * proper padding.
-         * If the side data is smaller than the buffer padding size, the
-         * remaining bytes should have already been filled with zeros by the
-         * original packet allocation anyway. */
-        memset(pkt->data + pkt->size, 0,
-               FFMIN(orig_pktsize - pkt->size, FF_INPUT_BUFFER_PADDING_SIZE));
         pkt->side_data_elems = i+1;
         return 1;
     }
     return 0;
 }
 
+uint8_t *av_packet_pack_dictionary(AVDictionary *dict, int *size)
+{
+    AVDictionaryEntry *t = NULL;
+    uint8_t *data = NULL;
+    *size = 0;
+
+    if (!dict)
+        return NULL;
+
+    while ((t = av_dict_get(dict, "", t, AV_DICT_IGNORE_SUFFIX))) {
+        const size_t keylen   = strlen(t->key);
+        const size_t valuelen = strlen(t->value);
+        const size_t new_size = *size + keylen + 1 + valuelen + 1;
+        uint8_t *const new_data = av_realloc(data, new_size);
+
+        if (!new_data)
+            goto fail;
+        data = new_data;
+        if (new_size > INT_MAX)
+            goto fail;
+
+        memcpy(data + *size, t->key, keylen + 1);
+        memcpy(data + *size + keylen + 1, t->value, valuelen + 1);
+
+        *size = new_size;
+    }
+
+    return data;
+
+fail:
+    av_freep(&data);
+    *size = 0;
+    return NULL;
+}
+
+int av_packet_unpack_dictionary(const uint8_t *data, int size, AVDictionary **dict)
+{
+    const uint8_t *end = data + size;
+    int ret = 0;
+
+    if (!dict || !data || !size)
+        return ret;
+    if (size && end[-1])
+        return AVERROR_INVALIDDATA;
+    while (data < end) {
+        const uint8_t *key = data;
+        const uint8_t *val = data + strlen(key) + 1;
+
+        if (val >= end)
+            return AVERROR_INVALIDDATA;
+
+        ret = av_dict_set(dict, key, val, 0);
+        if (ret < 0)
+            break;
+        data = val + strlen(val) + 1;
+    }
+
+    return ret;
+}
+
 int av_packet_shrink_side_data(AVPacket *pkt, enum AVPacketSideDataType type,
                                int size)
 {

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/bitstream.c

@@ -112,8 +112,11 @@ static int alloc_table(VLC *vlc, int size, int use_static)
             abort(); // cannot do anything, init_vlc() is used with too little memory
         vlc->table_allocated += (1 << vlc->bits);
         vlc->table = av_realloc_f(vlc->table, vlc->table_allocated, sizeof(VLC_TYPE) * 2);
-        if (!vlc->table)
+        if (!vlc->table) {
+            vlc->table_allocated = 0;
+            vlc->table_size = 0;
             return AVERROR(ENOMEM);
+        }
     }
     return index;
 }
@@ -229,6 +232,7 @@ static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,
             /* note: realloc has been done, so reload tables */
             table = &vlc->table[table_index];
             table[j][0] = index; //code
+            av_assert0(table[j][0] == index);
             i = k-1;
         }
     }

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/codec_desc.c

@@ -21,10 +21,10 @@
 
 #include <string.h>
 
-#include "avcodec.h"
-
 #include "libavutil/common.h"
 #include "libavutil/internal.h"
+#include "avcodec.h"
+#include "version.h"
 
 static const AVCodecDescriptor codec_descriptors[] = {
     /* video codecs */
@@ -42,6 +42,7 @@ static const AVCodecDescriptor codec_descriptors[] = {
         .long_name = NULL_IF_CONFIG_SMALL("MPEG-2 video"),
         .props     = AV_CODEC_PROP_LOSSY,
     },
+#if FF_API_XVMC
     {
         .id        = AV_CODEC_ID_MPEG2VIDEO_XVMC,
         .type      = AVMEDIA_TYPE_VIDEO,
@@ -49,6 +50,7 @@ static const AVCodecDescriptor codec_descriptors[] = {
         .long_name = NULL_IF_CONFIG_SMALL("MPEG-1/2 video XvMC (X-Video Motion Compensation)"),
         .props     = AV_CODEC_PROP_LOSSY,
     },
+#endif /* FF_API_XVMC */
     {
         .id        = AV_CODEC_ID_H261,
         .type      = AVMEDIA_TYPE_VIDEO,
@@ -225,13 +227,6 @@ static const AVCodecDescriptor codec_descriptors[] = {
         .long_name = NULL_IF_CONFIG_SMALL("H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10"),
         .props     = AV_CODEC_PROP_LOSSY | AV_CODEC_PROP_LOSSLESS,
     },
-    {
-        .id        = AV_CODEC_ID_H265,
-        .type      = AVMEDIA_TYPE_VIDEO,
-        .name      = "h265",
-        .long_name = NULL_IF_CONFIG_SMALL("H.265 / HEVC"),
-        .props     = AV_CODEC_PROP_LOSSY | AV_CODEC_PROP_LOSSLESS,
-    },
     {
         .id        = AV_CODEC_ID_INDEO3,
         .type      = AVMEDIA_TYPE_VIDEO,
@@ -1394,6 +1389,27 @@ static const AVCodecDescriptor codec_descriptors[] = {
         .props     = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSY |
                      AV_CODEC_PROP_LOSSLESS,
     },
+    {
+        .id        = AV_CODEC_ID_HNM4_VIDEO,
+        .type      = AVMEDIA_TYPE_VIDEO,
+        .name      = "hnm4video",
+        .long_name = NULL_IF_CONFIG_SMALL("HNM 4 video"),
+        .props     = AV_CODEC_PROP_LOSSY,
+    },
+    {
+        .id        = AV_CODEC_ID_HEVC,
+        .type      = AVMEDIA_TYPE_VIDEO,
+        .name      = "hevc",
+        .long_name = NULL_IF_CONFIG_SMALL("H.265 / HEVC (High Efficiency Video Coding)"),
+        .props     = AV_CODEC_PROP_LOSSY,
+    },
+    {
+        .id        = AV_CODEC_ID_FIC,
+        .type      = AVMEDIA_TYPE_VIDEO,
+        .name      = "fic",
+        .long_name = NULL_IF_CONFIG_SMALL("Mirillis FIC"),
+        .props     = AV_CODEC_PROP_LOSSY,
+    },
 
     /* various PCM "codecs" */
     {
@@ -1853,6 +1869,13 @@ static const AVCodecDescriptor codec_descriptors[] = {
         .long_name = NULL_IF_CONFIG_SMALL("ADPCM IMA Radical"),
         .props     = AV_CODEC_PROP_LOSSY,
     },
+    {
+        .id        = AV_CODEC_ID_ADPCM_G726LE,
+        .type      = AVMEDIA_TYPE_AUDIO,
+        .name      = "adpcm_g726le",
+        .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM little-endian"),
+        .props     = AV_CODEC_PROP_LOSSY,
+    },
 
     /* AMR */
     {
@@ -2606,6 +2629,12 @@ static const AVCodecDescriptor codec_descriptors[] = {
         .name      = "dvd_nav_packet",
         .long_name = NULL_IF_CONFIG_SMALL("DVD Nav packet"),
     },
+    {
+        .id        = AV_CODEC_ID_TIMED_ID3,
+        .type      = AVMEDIA_TYPE_DATA,
+        .name      = "timed_id3",
+        .long_name = NULL_IF_CONFIG_SMALL("timed ID3 metadata"),
+    },
 
 };
 

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/dca.c

@@ -25,8 +25,10 @@
 #include <stdint.h>
 #include <string.h>
 
-#include "put_bits.h"
+#include "libavutil/error.h"
+
 #include "dca.h"
+#include "put_bits.h"
 
 const uint32_t avpriv_dca_sample_rates[16] =
 {

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/dcadata.h

@@ -7305,224 +7305,222 @@ DECLARE_ALIGNED(16, static const float, fir_32bands_nonperfect)[] =
 +1.390191784E-007
 };
 
+/* pre-scale lfe fir coefficients */
+#define SCALE(c) ((c) / (256.0f * 32768.0f))
 DECLARE_ALIGNED(16, static const float, lfe_fir_64)[] =
 {
-    2.658434386830777e-4, 9.029330685734748e-3,
+    SCALE(2.658434386830777e-4), SCALE(9.029330685734748e-3),
-    7.939263433218002e-2, 2.425158768892288e-1,
+    SCALE(7.939263433218002e-2), SCALE(2.425158768892288e-1),
-    3.430179357528686e-1, 2.398228943347931e-1,
+    SCALE(3.430179357528686e-1), SCALE(2.398228943347931e-1),
-    7.746443897485733e-2, 8.622321300208569e-3,
+    SCALE(7.746443897485733e-2), SCALE(8.622321300208569e-3),
-    8.179365249816328e-5, 9.450953453779220e-3,
+    SCALE(8.179365249816328e-5), SCALE(9.450953453779220e-3),
-    8.134882897138596e-2, 2.451938837766648e-1,
+    SCALE(8.134882897138596e-2), SCALE(2.451938837766648e-1),
-    3.429597318172455e-1, 2.371159791946411e-1,
+    SCALE(3.429597318172455e-1), SCALE(2.371159791946411e-1),
-    7.556436210870743e-2, 8.229630999267101e-3,
+    SCALE(7.556436210870743e-2), SCALE(8.229630999267101e-3),
-    9.439323912374676e-5, 9.887560270726680e-3,
+    SCALE(9.439323912374676e-5), SCALE(9.887560270726680e-3),
-    8.333285897970200e-2, 2.478559017181396e-1,
+    SCALE(8.333285897970200e-2), SCALE(2.478559017181396e-1),
-    3.428434133529663e-1, 2.343961596488952e-1,
+    SCALE(3.428434133529663e-1), SCALE(2.343961596488952e-1),
-    7.369252294301987e-2, 7.850865833461285e-3,
+    SCALE(7.369252294301987e-2), SCALE(7.850865833461285e-3),
-    1.082170274457894e-4, 1.033949479460716e-2,
+    SCALE(1.082170274457894e-4), SCALE(1.033949479460716e-2),
-    8.534456789493561e-2, 2.505008876323700e-1,
+    SCALE(8.534456789493561e-2), SCALE(2.505008876323700e-1),
-    3.426689505577088e-1, 2.316644787788391e-1,
+    SCALE(3.426689505577088e-1), SCALE(2.316644787788391e-1),
-    7.184901088476181e-2, 7.485736627131701e-3,
+    SCALE(7.184901088476181e-2), SCALE(7.485736627131701e-3),
-    1.233371440321207e-4, 1.080708485096693e-2,
+    SCALE(1.233371440321207e-4), SCALE(1.080708485096693e-2),
-    8.738376945257187e-2, 2.531278133392334e-1,
+    SCALE(8.738376945257187e-2), SCALE(2.531278133392334e-1),
-    3.424364924430847e-1, 2.289219647645950e-1,
+    SCALE(3.424364924430847e-1), SCALE(2.289219647645950e-1),
-    7.003392279148102e-2, 7.133882027119398e-3,
+    SCALE(7.003392279148102e-2), SCALE(7.133882027119398e-3),
-    1.397485757479444e-4, 1.129068247973919e-2,
+    SCALE(1.397485757479444e-4), SCALE(1.129068247973919e-2),
-    8.945026248693466e-2, 2.557355761528015e-1,
+    SCALE(8.945026248693466e-2), SCALE(2.557355761528015e-1),
-    3.421461284160614e-1, 2.261696159839630e-1,
+    SCALE(3.421461284160614e-1), SCALE(2.261696159839630e-1),
-    6.824731826782227e-2, 6.794991903007030e-3,
+    SCALE(6.824731826782227e-2), SCALE(6.794991903007030e-3),
-    1.575958012836054e-4, 1.179065089672804e-2,
+    SCALE(1.575958012836054e-4), SCALE(1.179065089672804e-2),
-    9.154383838176728e-2, 2.583232223987580e-1,
+    SCALE(9.154383838176728e-2), SCALE(2.583232223987580e-1),
-    3.417979776859284e-1, 2.234084606170654e-1,
+    SCALE(3.417979776859284e-1), SCALE(2.234084606170654e-1),
-    6.648923456668854e-2, 6.468691397458315e-3,
+    SCALE(6.648923456668854e-2), SCALE(6.468691397458315e-3),
-    1.769922382663936e-4, 1.230732165277004e-2,
+    SCALE(1.769922382663936e-4), SCALE(1.230732165277004e-2),
-    9.366425126791000e-2, 2.608896791934967e-1,
+    SCALE(9.366425126791000e-2), SCALE(2.608896791934967e-1),
-    3.413922190666198e-1, 2.206395119428635e-1,
+    SCALE(3.413922190666198e-1), SCALE(2.206395119428635e-1),
-    6.475970894098282e-2, 6.154712289571762e-3,
+    SCALE(6.475970894098282e-2), SCALE(6.154712289571762e-3),
-    1.981738605536520e-4, 1.284105982631445e-2,
+    SCALE(1.981738605536520e-4), SCALE(1.284105982631445e-2),
-    9.581124037504196e-2, 2.634339034557342e-1,
+    SCALE(9.581124037504196e-2), SCALE(2.634339034557342e-1),
-    3.409290313720703e-1, 2.178637981414795e-1,
+    SCALE(3.409290313720703e-1), SCALE(2.178637981414795e-1),
-    6.305878609418869e-2, 5.852684378623962e-3,
+    SCALE(6.305878609418869e-2), SCALE(5.852684378623962e-3),
-    2.211847313446924e-4, 1.339218579232693e-2,
+    SCALE(2.211847313446924e-4), SCALE(1.339218579232693e-2),
-    9.798453748226166e-2, 2.659549415111542e-1,
+    SCALE(9.798453748226166e-2), SCALE(2.659549415111542e-1),
-    3.404086530208588e-1, 2.150822728872299e-1,
+    SCALE(3.404086530208588e-1), SCALE(2.150822728872299e-1),
-    6.138643622398376e-2, 5.562345497310162e-3,
+    SCALE(6.138643622398376e-2), SCALE(5.562345497310162e-3),
-    2.460231189616024e-4, 1.396108977496624e-2,
+    SCALE(2.460231189616024e-4), SCALE(1.396108977496624e-2),
-    1.001838669180870e-1, 2.684516608715058e-1,
+    SCALE(1.001838669180870e-1), SCALE(2.684516608715058e-1),
-    3.398312926292420e-1, 2.122959494590759e-1,
+    SCALE(3.398312926292420e-1), SCALE(2.122959494590759e-1),
-    5.974265560507774e-2, 5.283284001052380e-3,
+    SCALE(5.974265560507774e-2), SCALE(5.283284001052380e-3),
-    2.726115926634520e-4, 1.454808749258518e-2,
+    SCALE(2.726115926634520e-4), SCALE(1.454808749258518e-2),
-    1.024089083075523e-1, 2.709231376647949e-1,
+    SCALE(1.024089083075523e-1), SCALE(2.709231376647949e-1),
-    3.391972482204438e-1, 2.095058411359787e-1,
+    SCALE(3.391972482204438e-1), SCALE(2.095058411359787e-1),
-    5.812742188572884e-2, 5.015311297029257e-3,
+    SCALE(5.812742188572884e-2), SCALE(5.015311297029257e-3),
-    3.013863170053810e-4, 1.515355054289102e-2,
+    SCALE(3.013863170053810e-4), SCALE(1.515355054289102e-2),
-    1.046593263745308e-1, 2.733682692050934e-1,
+    SCALE(1.046593263745308e-1), SCALE(2.733682692050934e-1),
-    3.385068178176880e-1, 2.067128717899322e-1,
+    SCALE(3.385068178176880e-1), SCALE(2.067128717899322e-1),
-    5.654069408774376e-2, 4.758012015372515e-3,
+    SCALE(5.654069408774376e-2), SCALE(4.758012015372515e-3),
-    3.328395541757345e-4, 1.577781140804291e-2,
+    SCALE(3.328395541757345e-4), SCALE(1.577781140804291e-2),
-    1.069347932934761e-1, 2.757860720157624e-1,
+    SCALE(1.069347932934761e-1), SCALE(2.757860720157624e-1),
-    3.377602994441986e-1, 2.039180546998978e-1,
+    SCALE(3.377602994441986e-1), SCALE(2.039180546998978e-1),
-    5.498242005705833e-2, 4.511159844696522e-3,
+    SCALE(5.498242005705833e-2), SCALE(4.511159844696522e-3),
-    3.658991190604866e-4, 1.642123050987720e-2,
+    SCALE(3.658991190604866e-4), SCALE(1.642123050987720e-2),
-    1.092349365353584e-1, 2.781755328178406e-1,
+    SCALE(1.092349365353584e-1), SCALE(2.781755328178406e-1),
-    3.369580209255218e-1, 2.011223286390304e-1,
+    SCALE(3.369580209255218e-1), SCALE(2.011223286390304e-1),
-    5.345252528786659e-2, 4.274417180567980e-3,
+    SCALE(5.345252528786659e-2), SCALE(4.274417180567980e-3),
-    4.018281470052898e-4, 1.708412915468216e-2,
+    SCALE(4.018281470052898e-4), SCALE(1.708412915468216e-2),
-    1.115593686699867e-1, 2.805356979370117e-1,
+    SCALE(1.115593686699867e-1), SCALE(2.805356979370117e-1),
-    3.361004292964936e-1, 1.983266174793244e-1,
+    SCALE(3.361004292964936e-1), SCALE(1.983266174793244e-1),
-    5.195093154907227e-2, 4.047499038279056e-3,
+    SCALE(5.195093154907227e-2), SCALE(4.047499038279056e-3),
-    4.401875485200435e-4, 1.776690222322941e-2,
+    SCALE(4.401875485200435e-4), SCALE(1.776690222322941e-2),
-    1.139076948165894e-1, 2.828655838966370e-1,
+    SCALE(1.139076948165894e-1), SCALE(2.828655838966370e-1),
-    3.351879119873047e-1, 1.955319195985794e-1,
+    SCALE(3.351879119873047e-1), SCALE(1.955319195985794e-1),
-    5.047753453254700e-2, 3.830091329291463e-3,
+    SCALE(5.047753453254700e-2), SCALE(3.830091329291463e-3),
-    4.812776169274002e-4, 1.846982724964619e-2,
+    SCALE(4.812776169274002e-4), SCALE(1.846982724964619e-2),
-    1.162794977426529e-1, 2.851640880107880e-1,
+    SCALE(1.162794977426529e-1), SCALE(2.851640880107880e-1),
-    3.342207968235016e-1, 1.927391141653061e-1,
+    SCALE(3.342207968235016e-1), SCALE(1.927391141653061e-1),
-    4.903224110603333e-2, 3.621967276558280e-3,
+    SCALE(4.903224110603333e-2), SCALE(3.621967276558280e-3),
-    5.252459668554366e-4, 1.919330470263958e-2,
+    SCALE(5.252459668554366e-4), SCALE(1.919330470263958e-2),
-    1.186743453145027e-1, 2.874303460121155e-1,
+    SCALE(1.186743453145027e-1), SCALE(2.874303460121155e-1),
-    3.331996202468872e-1, 1.899491697549820e-1,
+    SCALE(3.331996202468872e-1), SCALE(1.899491697549820e-1),
-    4.761491715908051e-2, 3.422776935622096e-3,
+    SCALE(4.761491715908051e-2), SCALE(3.422776935622096e-3),
-    5.721592460758984e-4, 1.993762329220772e-2,
+    SCALE(5.721592460758984e-4), SCALE(1.993762329220772e-2),
-    1.210917681455612e-1, 2.896633744239807e-1,
+    SCALE(1.210917681455612e-1), SCALE(2.896633744239807e-1),
-    3.321248590946198e-1, 1.871629506349564e-1,
+    SCALE(3.321248590946198e-1), SCALE(1.871629506349564e-1),
-    4.622544348239899e-2, 3.232272574678064e-3,
+    SCALE(4.622544348239899e-2), SCALE(3.232272574678064e-3),
-    6.222130032256246e-4, 2.070316113531590e-2,
+    SCALE(6.222130032256246e-4), SCALE(2.070316113531590e-2),
-    1.235313042998314e-1, 2.918621897697448e-1,
+    SCALE(1.235313042998314e-1), SCALE(2.918621897697448e-1),
-    3.309969604015350e-1, 1.843813359737396e-1,
+    SCALE(3.309969604015350e-1), SCALE(1.843813359737396e-1),
-    4.486365616321564e-2, 3.050152910873294e-3,
+    SCALE(4.486365616321564e-2), SCALE(3.050152910873294e-3),
-    6.755515350960195e-4, 2.149021252989769e-2,
+    SCALE(6.755515350960195e-4), SCALE(2.149021252989769e-2),
-    1.259924471378326e-1, 2.940258979797364e-1,
+    SCALE(1.259924471378326e-1), SCALE(2.940258979797364e-1),
-    3.298164308071136e-1, 1.816052496433258e-1,
+    SCALE(3.298164308071136e-1), SCALE(1.816052496433258e-1),
-    4.352942481637001e-2, 2.876190468668938e-3,
+    SCALE(4.352942481637001e-2), SCALE(2.876190468668938e-3),
-    7.324148900806904e-4, 2.229913882911205e-2,
+    SCALE(7.324148900806904e-4), SCALE(2.229913882911205e-2),
-    1.284746825695038e-1, 2.961534857749939e-1,
+    SCALE(1.284746825695038e-1), SCALE(2.961534857749939e-1),
-    3.285838961601258e-1, 1.788355410099030e-1,
+    SCALE(3.285838961601258e-1), SCALE(1.788355410099030e-1),
-    4.222255200147629e-2, 2.710093278437853e-3,
+    SCALE(4.222255200147629e-2), SCALE(2.710093278437853e-3),
-    7.928516715764999e-4, 2.313023805618286e-2,
+    SCALE(7.928516715764999e-4), SCALE(2.313023805618286e-2),
-    1.309774816036224e-1, 2.982441186904907e-1,
+    SCALE(1.309774816036224e-1), SCALE(2.982441186904907e-1),
-    3.272998929023742e-1, 1.760730892419815e-1,
+    SCALE(3.272998929023742e-1), SCALE(1.760730892419815e-1),
-    4.094288870692253e-2, 2.551567042246461e-3,
+    SCALE(4.094288870692253e-2), SCALE(2.551567042246461e-3),
-    8.570110658183694e-4, 2.398385666310787e-2,
+    SCALE(8.570110658183694e-4), SCALE(2.398385666310787e-2),
-    1.335003077983856e-1, 3.002967536449432e-1,
+    SCALE(1.335003077983856e-1), SCALE(3.002967536449432e-1),
-    3.259649574756622e-1, 1.733186990022659e-1,
+    SCALE(3.259649574756622e-1), SCALE(1.733186990022659e-1),
-    3.969023004174232e-2, 2.400433411821723e-3,
+    SCALE(3.969023004174232e-2), SCALE(2.400433411821723e-3),
-    9.251192095689476e-4, 2.486028522253036e-2,
+    SCALE(9.251192095689476e-4), SCALE(2.486028522253036e-2),
-    1.360425949096680e-1, 3.023106753826142e-1,
+    SCALE(1.360425949096680e-1), SCALE(3.023106753826142e-1),
-    3.245797157287598e-1, 1.705732345581055e-1,
+    SCALE(3.245797157287598e-1), SCALE(1.705732345581055e-1),
-    3.846437484025955e-2, 2.256359672173858e-3,
+    SCALE(3.846437484025955e-2), SCALE(2.256359672173858e-3),
-    9.974770946428180e-4, 2.575986087322235e-2,
+    SCALE(9.974770946428180e-4), SCALE(2.575986087322235e-2),
-    1.386037617921829e-1, 3.042849004268646e-1,
+    SCALE(1.386037617921829e-1), SCALE(3.042849004268646e-1),
-    3.231448531150818e-1, 1.678375005722046e-1,
+    SCALE(3.231448531150818e-1), SCALE(1.678375005722046e-1),
-    3.726511076092720e-2, 2.119151875376701e-3,
+    SCALE(3.726511076092720e-2), SCALE(2.119151875376701e-3),
-    1.073930296115577e-3, 2.668286114931106e-2,
+    SCALE(1.073930296115577e-3), SCALE(2.668286114931106e-2),
-    1.411831974983215e-1, 3.062185347080230e-1,
+    SCALE(1.411831974983215e-1), SCALE(3.062185347080230e-1),
-    3.216609656810760e-1, 1.651122719049454e-1,
+    SCALE(3.216609656810760e-1), SCALE(1.651122719049454e-1),
-    3.609224036335945e-2, 1.988604199141264e-3,
+    SCALE(3.609224036335945e-2), SCALE(1.988604199141264e-3),
-    1.155023579485714e-3, 2.762960828840732e-2,
+    SCALE(1.155023579485714e-3), SCALE(2.762960828840732e-2),
-    1.437802612781525e-1, 3.081108033657074e-1,
+    SCALE(1.437802612781525e-1), SCALE(3.081108033657074e-1),
-    3.201287388801574e-1, 1.623983532190323e-1,
+    SCALE(3.201287388801574e-1), SCALE(1.623983532190323e-1),
-    3.494550660252571e-2, 1.864377525635064e-3,
+    SCALE(3.494550660252571e-2), SCALE(1.864377525635064e-3),
-    1.240676851011813e-3, 2.860039286315441e-2,
+    SCALE(1.240676851011813e-3), SCALE(2.860039286315441e-2),
-    1.463943719863892e-1, 3.099608123302460e-1,
+    SCALE(1.463943719863892e-1), SCALE(3.099608123302460e-1),
-    3.185488879680634e-1, 1.596965193748474e-1,
+    SCALE(3.185488879680634e-1), SCALE(1.596965193748474e-1),
-    3.382468968629837e-2, 1.746327499859035e-3,
+    SCALE(3.382468968629837e-2), SCALE(1.746327499859035e-3),
-    1.331258914433420e-3, 2.959549613296986e-2,
+    SCALE(1.331258914433420e-3), SCALE(2.959549613296986e-2),
-    1.490248143672943e-1, 3.117676973342896e-1,
+    SCALE(1.490248143672943e-1), SCALE(3.117676973342896e-1),
-    3.169221282005310e-1, 1.570075154304504e-1,
+    SCALE(3.169221282005310e-1), SCALE(1.570075154304504e-1),
-    3.272953629493714e-2, 1.634211512282491e-3,
+    SCALE(3.272953629493714e-2), SCALE(1.634211512282491e-3),
-    1.426893868483603e-3, 3.061520494520664e-2,
+    SCALE(1.426893868483603e-3), SCALE(3.061520494520664e-2),
-    1.516709625720978e-1, 3.135308027267456e-1,
+    SCALE(1.516709625720978e-1), SCALE(3.135308027267456e-1),
-    3.152491748332978e-1, 1.543320864439010e-1,
+    SCALE(3.152491748332978e-1), SCALE(1.543320864439010e-1),
-    3.165979683399200e-2, 1.527829794213176e-3,
+    SCALE(3.165979683399200e-2), SCALE(1.527829794213176e-3),
 };
 
 DECLARE_ALIGNED(16, static const float, lfe_fir_128)[] =
 {
-    0.00053168571, 0.15878495574, 0.68603444099, 0.15492856503,
+    SCALE(0.00053168571), SCALE(0.15878495574), SCALE(0.68603444099), SCALE(0.15492856503),
-    0.00016358691, 0.16269733012, 0.68591803312, 0.15112841129,
+    SCALE(0.00016358691), SCALE(0.16269733012), SCALE(0.68591803312), SCALE(0.15112841129),
-    0.00018878609, 0.16666537523, 0.68568539619, 0.14738474786,
+    SCALE(0.00018878609), SCALE(0.16666537523), SCALE(0.68568539619), SCALE(0.14738474786),
-    0.00021643363, 0.17068879306, 0.68533653021, 0.14369773865,
+    SCALE(0.00021643363), SCALE(0.17068879306), SCALE(0.68533653021), SCALE(0.14369773865),
-    0.00024667382, 0.17476719618, 0.68487155437, 0.14006754756,
+    SCALE(0.00024667382), SCALE(0.17476719618), SCALE(0.68487155437), SCALE(0.14006754756),
-    0.00027949660, 0.17890018225, 0.68429082632, 0.13649433851,
+    SCALE(0.00027949660), SCALE(0.17890018225), SCALE(0.68429082632), SCALE(0.13649433851),
-    0.00031519096, 0.18308731914, 0.68359452486, 0.13297818601,
+    SCALE(0.00031519096), SCALE(0.18308731914), SCALE(0.68359452486), SCALE(0.13297818601),
-    0.00035398375, 0.18732811511, 0.68278300762, 0.12951917946,
+    SCALE(0.00035398375), SCALE(0.18732811511), SCALE(0.68278300762), SCALE(0.12951917946),
-    0.00039634691, 0.19162209332, 0.68185669184, 0.12611730397,
+    SCALE(0.00039634691), SCALE(0.19162209332), SCALE(0.68185669184), SCALE(0.12611730397),
-    0.00044236859, 0.19596865773, 0.68081587553, 0.12277261168,
+    SCALE(0.00044236859), SCALE(0.19596865773), SCALE(0.68081587553), SCALE(0.12277261168),
-    0.00049204525, 0.20036731660, 0.67966115475, 0.11948505789,
+    SCALE(0.00049204525), SCALE(0.20036731660), SCALE(0.67966115475), SCALE(0.11948505789),
-    0.00054522208, 0.20481738448, 0.67839306593, 0.11625462025,
+    SCALE(0.00054522208), SCALE(0.20481738448), SCALE(0.67839306593), SCALE(0.11625462025),
-    0.00060277141, 0.20931822062, 0.67701220512, 0.11308115721,
+    SCALE(0.00060277141), SCALE(0.20931822062), SCALE(0.67701220512), SCALE(0.11308115721),
-    0.00066567765, 0.21386915445, 0.67551922798, 0.10996460915,
+    SCALE(0.00066567765), SCALE(0.21386915445), SCALE(0.67551922798), SCALE(0.10996460915),
-    0.00073179678, 0.21846942604, 0.67391467094, 0.10690483451,
+    SCALE(0.00073179678), SCALE(0.21846942604), SCALE(0.67391467094), SCALE(0.10690483451),
-    0.00080365466, 0.22311829031, 0.67219948769, 0.10390164703,
+    SCALE(0.00080365466), SCALE(0.22311829031), SCALE(0.67219948769), SCALE(0.10390164703),
-    0.00088037323, 0.22781492770, 0.67037439346, 0.10095486045,
+    SCALE(0.00088037323), SCALE(0.22781492770), SCALE(0.67037439346), SCALE(0.10095486045),
-    0.00096255314, 0.23255851865, 0.66844022274, 0.09806428105,
+    SCALE(0.00096255314), SCALE(0.23255851865), SCALE(0.66844022274), SCALE(0.09806428105),
-    0.00105048984, 0.23734821379, 0.66639786959, 0.09522963315,
+    SCALE(0.00105048984), SCALE(0.23734821379), SCALE(0.66639786959), SCALE(0.09522963315),
-    0.00114431616, 0.24218304455, 0.66424828768, 0.09245070815,
+    SCALE(0.00114431616), SCALE(0.24218304455), SCALE(0.66424828768), SCALE(0.09245070815),
-    0.00124442333, 0.24706205726, 0.66199249029, 0.08972713351,
+    SCALE(0.00124442333), SCALE(0.24706205726), SCALE(0.66199249029), SCALE(0.08972713351),
-    0.00135110028, 0.25198432803, 0.65963155031, 0.08705867827,
+    SCALE(0.00135110028), SCALE(0.25198432803), SCALE(0.65963155031), SCALE(0.08705867827),
-    0.00146482687, 0.25694879889, 0.65716648102, 0.08444493264,
+    SCALE(0.00146482687), SCALE(0.25694879889), SCALE(0.65716648102), SCALE(0.08444493264),
-    0.00158570008, 0.26195442677, 0.65459835529, 0.08188561350,
+    SCALE(0.00158570008), SCALE(0.26195442677), SCALE(0.65459835529), SCALE(0.08188561350),
-    0.00171401864, 0.26700007915, 0.65192854404, 0.07938029617,
+    SCALE(0.00171401864), SCALE(0.26700007915), SCALE(0.65192854404), SCALE(0.07938029617),
-    0.00185023469, 0.27208462358, 0.64915806055, 0.07692859322,
+    SCALE(0.00185023469), SCALE(0.27208462358), SCALE(0.64915806055), SCALE(0.07692859322),
-    0.00199495023, 0.27720692754, 0.64628833532, 0.07453006506,
+    SCALE(0.00199495023), SCALE(0.27720692754), SCALE(0.64628833532), SCALE(0.07453006506),
-    0.00214785640, 0.28236576915, 0.64332056046, 0.07218432426,
+    SCALE(0.00214785640), SCALE(0.28236576915), SCALE(0.64332056046), SCALE(0.07218432426),
-    0.00231004250, 0.28755992651, 0.64025616646, 0.06989086419,
+    SCALE(0.00231004250), SCALE(0.28755992651), SCALE(0.64025616646), SCALE(0.06989086419),
-    0.00248134881, 0.29278811812, 0.63709646463, 0.06764923781,
+    SCALE(0.00248134881), SCALE(0.29278811812), SCALE(0.63709646463), SCALE(0.06764923781),
-    0.00266251224, 0.29804900289, 0.63384294510, 0.06545893103,
+    SCALE(0.00266251224), SCALE(0.29804900289), SCALE(0.63384294510), SCALE(0.06545893103),
-    0.00285378192, 0.30334126949, 0.63049703836, 0.06331945211,
+    SCALE(0.00285378192), SCALE(0.30334126949), SCALE(0.63049703836), SCALE(0.06331945211),
-    0.00305565330, 0.30866351724, 0.62706029415, 0.06123027951,
+    SCALE(0.00305565330), SCALE(0.30866351724), SCALE(0.62706029415), SCALE(0.06123027951),
-    0.00326841651, 0.31401440501, 0.62353414297, 0.05919086933,
+    SCALE(0.00326841651), SCALE(0.31401440501), SCALE(0.62353414297), SCALE(0.05919086933),
-    0.00349264755, 0.31939238310, 0.61992025375, 0.05720067024,
+    SCALE(0.00349264755), SCALE(0.31939238310), SCALE(0.61992025375), SCALE(0.05720067024),
-    0.00372874714, 0.32479602098, 0.61622029543, 0.05525910854,
+    SCALE(0.00372874714), SCALE(0.32479602098), SCALE(0.61622029543), SCALE(0.05525910854),
-    0.00397720048, 0.33022382855, 0.61243581772, 0.05336561054,
+    SCALE(0.00397720048), SCALE(0.33022382855), SCALE(0.61243581772), SCALE(0.05336561054),
-    0.00423829490, 0.33567428589, 0.60856848955, 0.05151961371,
+    SCALE(0.00423829490), SCALE(0.33567428589), SCALE(0.60856848955), SCALE(0.05151961371),
-    0.00451271003, 0.34114575386, 0.60462015867, 0.04972046614,
+    SCALE(0.00451271003), SCALE(0.34114575386), SCALE(0.60462015867), SCALE(0.04972046614),
-    0.00480085658, 0.34663668275, 0.60059231520, 0.04796761274,
+    SCALE(0.00480085658), SCALE(0.34663668275), SCALE(0.60059231520), SCALE(0.04796761274),
-    0.00510312291, 0.35214546323, 0.59648692608, 0.04626038298,
+    SCALE(0.00510312291), SCALE(0.35214546323), SCALE(0.59648692608), SCALE(0.04626038298),
-    0.00542017492, 0.35767036676, 0.59230577946, 0.04459818453,
+    SCALE(0.00542017492), SCALE(0.35767036676), SCALE(0.59230577946), SCALE(0.04459818453),
-    0.00575236930, 0.36320972443, 0.58805054426, 0.04298033938,
+    SCALE(0.00575236930), SCALE(0.36320972443), SCALE(0.58805054426), SCALE(0.04298033938),
-    0.00610029325, 0.36876192689, 0.58372318745, 0.04140623659,
+    SCALE(0.00610029325), SCALE(0.36876192689), SCALE(0.58372318745), SCALE(0.04140623659),
-    0.00646453211, 0.37432509661, 0.57932555676, 0.03987516090,
+    SCALE(0.00646453211), SCALE(0.37432509661), SCALE(0.57932555676), SCALE(0.03987516090),
-    0.00684553990, 0.37989753485, 0.57485944033, 0.03838652745,
+    SCALE(0.00684553990), SCALE(0.37989753485), SCALE(0.57485944033), SCALE(0.03838652745),
-    0.00724391919, 0.38547745347, 0.57032698393, 0.03693958372,
+    SCALE(0.00724391919), SCALE(0.38547745347), SCALE(0.57032698393), SCALE(0.03693958372),
-    0.00766016589, 0.39106300473, 0.56572991610, 0.03553372994,
+    SCALE(0.00766016589), SCALE(0.39106300473), SCALE(0.56572991610), SCALE(0.03553372994),
-    0.00809498038, 0.39665243030, 0.56107026339, 0.03416819125,
+    SCALE(0.00809498038), SCALE(0.39665243030), SCALE(0.56107026339), SCALE(0.03416819125),
-    0.00854881573, 0.40224379301, 0.55634999275, 0.03284239396,
+    SCALE(0.00854881573), SCALE(0.40224379301), SCALE(0.55634999275), SCALE(0.03284239396),
-    0.00902230106, 0.40783521533, 0.55157101154, 0.03155555204,
+    SCALE(0.00902230106), SCALE(0.40783521533), SCALE(0.55157101154), SCALE(0.03155555204),
-    0.00951600447, 0.41342487931, 0.54673534632, 0.03030703776,
+    SCALE(0.00951600447), SCALE(0.41342487931), SCALE(0.54673534632), SCALE(0.03030703776),
-    0.01003060210, 0.41901078820, 0.54184508324, 0.02909611352,
+    SCALE(0.01003060210), SCALE(0.41901078820), SCALE(0.54184508324), SCALE(0.02909611352),
-    0.01056654565, 0.42459106445, 0.53690224886, 0.02792212367,
+    SCALE(0.01056654565), SCALE(0.42459106445), SCALE(0.53690224886), SCALE(0.02792212367),
-    0.01112466771, 0.43016362190, 0.53190881014, 0.02678431384,
+    SCALE(0.01112466771), SCALE(0.43016362190), SCALE(0.53190881014), SCALE(0.02678431384),
-    0.01170534454, 0.43572667241, 0.52686679363, 0.02568206564,
+    SCALE(0.01170534454), SCALE(0.43572667241), SCALE(0.52686679363), SCALE(0.02568206564),
-    0.01230939943, 0.44127810001, 0.52177828550, 0.02461459488,
+    SCALE(0.01230939943), SCALE(0.44127810001), SCALE(0.52177828550), SCALE(0.02461459488),
-    0.01293735672, 0.44681602716, 0.51664537191, 0.02358125709,
+    SCALE(0.01293735672), SCALE(0.44681602716), SCALE(0.51664537191), SCALE(0.02358125709),
-    0.01358995494, 0.45233830810, 0.51147013903, 0.02258131653,
+    SCALE(0.01358995494), SCALE(0.45233830810), SCALE(0.51147013903), SCALE(0.02258131653),
-    0.01426773332, 0.45784294605, 0.50625455379, 0.02161412500,
+    SCALE(0.01426773332), SCALE(0.45784294605), SCALE(0.50625455379), SCALE(0.02161412500),
-    0.01497144438, 0.46332800388, 0.50100076199, 0.02067894675,
+    SCALE(0.01497144438), SCALE(0.46332800388), SCALE(0.50100076199), SCALE(0.02067894675),
-    0.01570170000, 0.46879136562, 0.49571081996, 0.01977507770,
+    SCALE(0.01570170000), SCALE(0.46879136562), SCALE(0.49571081996), SCALE(0.01977507770),
-    0.01645922661, 0.47423094511, 0.49038675427, 0.01890186779,
+    SCALE(0.01645922661), SCALE(0.47423094511), SCALE(0.49038675427), SCALE(0.01890186779),
-    0.01724460535, 0.47964480519, 0.48503074050, 0.01805862412,
+    SCALE(0.01724460535), SCALE(0.47964480519), SCALE(0.48503074050), SCALE(0.01805862412),
 };
+#undef SCALE
 
-/* 10^-(dB/20), with dB being a list of dB values ranging from 0 to -72 */
+/*
-/* do a 20*log10(dca_downmix_coeffs) to reconvert the values */
+ * D.11 Look-up Table for Downmix Scale Factors
-
+ *
-static const float dca_downmix_coeffs[65] = {
+ * Note that the range of the entries in DmixTable[] is between -60 dB and 0 dB
-  1.000000000000000, 0.988553094656939, 0.971627951577106, 0.944060876285923, 0.917275935389780, 0.891250938133746,
+ * with addition of -inf (|DMixCoeff| = 0), which is coded with a DmixCode = 0.
-  0.865964323360065, 0.841395141645195, 0.817523037943650, 0.794328234724281, 0.771791515585012, 0.749894209332456,
+ * Furthermore, the range [-60 to 0] is subdivided into 3 regions, each with a
-  0.728618174513228, 0.707945784384138, 0.687859912308808, 0.668343917568615, 0.649381631576211, 0.630957344480193,
+ * different grid resolution:
-  0.613055792149821, 0.595662143529010, 0.578761988349121, 0.562341325190349, 0.546386549881854, 0.530884444230988,
+ *
-  0.515822165072306, 0.501187233627272, 0.446683592150963, 0.398107170553497, 0.354813389233575, 0.316227766016838,
+ * 1) [-60.000 to -30] with resolution of 0.500 dB
-  0.281838293126445, 0.251188643150958, 0.223872113856834, 0.199526231496888, 0.177827941003892, 0.158489319246111,
+ * 2) [-29.750 to -15] with resolution of 0.250 dB
-  0.141253754462275, 0.125892541179417, 0.112201845430196, 0.100000000000000, 0.089125093813374, 0.079432823472428,
+ * 3) [-14.875 to   0] with resolution of 0.125 dB
-  0.070794578438414, 0.063095734448019, 0.053088444423099, 0.044668359215096, 0.037583740428844, 0.031622776601684,
+ */
-  0.026607250597988, 0.022387211385683, 0.018836490894898, 0.015848931924611, 0.013335214321633, 0.011220184543020,
+static const float dca_dmixtable[241] = {
-  0.009440608762859, 0.007943282347243, 0.005623413251903, 0.003981071705535, 0.002818382931264, 0.001995262314969,
-  0.001412537544623, 0.001000000000000, 0.000501187233627, 0.000251188643151, 0.000000000000000,
-};
-
-static const float dca_downmix_scale_factors[241] = {
     0.001000, 0.001059, 0.001122, 0.001189, 0.001259, 0.001334, 0.001413, 0.001496,
     0.001585, 0.001679, 0.001778, 0.001884, 0.001995, 0.002113, 0.002239, 0.002371,
     0.002512, 0.002661, 0.002818, 0.002985, 0.003162, 0.003350, 0.003548, 0.003758,
@@ -7553,20 +7551,20 @@ static const float dca_downmix_scale_factors[241] = {
     0.707107, 0.718208, 0.728618, 0.739180, 0.749894, 0.760764, 0.771792, 0.782979,
     0.794328, 0.805842, 0.817523, 0.829373, 0.841395, 0.853591, 0.865964, 0.878517,
     0.891251, 0.904170, 0.917276, 0.930572, 0.944061, 0.957745, 0.971628, 0.985712,
-    1.000000
+    1.000000,
 };
 
-static const uint8_t dca_default_coeffs[10][5][2] = {
+static const float dca_default_coeffs[10][6][2] = {
-    { { 13, 13 },                                                 },
+    { { 0.707107, 0.707107 }, { 0.000000, 0.000000 },                                                                                                 }, // A [LFE]
-    { {  0, 64 }, { 64,  0 },                                     },
+    { { 1.000000, 0.000000 }, { 0.000000, 1.000000 }, { 0.000000, 0.000000 },                                                                         }, // A + B (dual mono) [LFE]
-    { {  0, 64 }, { 64,  0 },                                     },
+    { { 1.000000, 0.000000 }, { 0.000000, 1.000000 }, { 0.000000, 0.000000 },                                                                         }, // L + R (stereo) [LFE]
-    { {  0, 64 }, { 64,  0 },                                     },
+    { { 1.000000, 0.000000 }, { 0.000000, 1.000000 }, { 0.000000, 0.000000 },                                                                         }, // (L+R) + (L-R) (sum-difference) [LFE]
-    { {  0, 64 }, { 64,  0 },                                     },
+    { { 1.000000, 0.000000 }, { 0.000000, 1.000000 }, { 0.000000, 0.000000 },                                                                         }, // LT + RT (left and right total) [LFE]
-    { {  6,  6 }, {  0, 25 }, { 25,  0 },                         },
+    { { 0.501187, 0.501187 }, { 0.707107, 0.000000 }, { 0.000000, 0.707107 }, { 0.000000, 0.000000 },                                                 }, // C + L + R [LFE]
-    { {  0, 25 }, { 25,  0 }, { 13, 13 },                         },
+    { { 0.707107, 0.000000 }, { 0.000000, 0.707107 }, { 0.501187, 0.501187 }, { 0.000000, 0.000000 },                                                 }, // L + R + S [LFE]
-    { {  6,  6 }, {  0, 25 }, { 25,  0 }, { 13, 13 },             },
+    { { 0.501187, 0.501187 }, { 0.707107, 0.000000 }, { 0.000000, 0.707107 }, { 0.501187, 0.501187 }, { 0.000000, 0.000000 },                         }, // C + L + R + S [LFE]
-    { {  0, 25 }, { 25,  0 }, {  0, 13 }, { 13,  0 },             },
+    { { 0.707107, 0.000000 }, { 0.000000, 0.707107 }, { 0.501187, 0.000000 }, { 0.000000, 0.501187 }, { 0.000000, 0.000000 },                         }, // L + R + SL + SR [LFE]
-    { {  6,  6 }, {  0, 25 }, { 25,  0 }, {  0, 13 }, { 13,  0 }, },
+    { { 0.501187, 0.501187 }, { 0.707107, 0.000000 }, { 0.000000, 0.707107 }, { 0.501187, 0.000000 }, { 0.000000, 0.501187 }, { 0.000000, 0.000000 }, }, // C + L + R + SL + SR [LFE]
 };
 
 /* downmix coeffs

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/dcadec.c

@@ -32,6 +32,7 @@
 #include "libavutil/internal.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/mathematics.h"
+#include "libavutil/opt.h"
 #include "libavutil/samplefmt.h"
 #include "avcodec.h"
 #include "fft.h"
@@ -52,7 +53,6 @@
 //#define TRACE
 
 #define DCA_PRIM_CHANNELS_MAX  (7)
-#define DCA_SUBBANDS          (64)
 #define DCA_ABITS_MAX         (32)      /* Should be 28 */
 #define DCA_SUBSUBFRAMES_MAX   (4)
 #define DCA_SUBFRAMES_MAX     (16)
@@ -325,6 +325,8 @@ static const int8_t dca_channel_reorder_nolfe_xch[][9] = {
 
 #define DCA_BUFFER_PADDING_SIZE   1024
 
+#define DCA_NSYNCAUX        0x9A1105A0
+
 /** Bit allocation */
 typedef struct {
     int offset;                 ///< code values offset
@@ -346,6 +348,7 @@ static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,
 }
 
 typedef struct {
+    const AVClass *class;       ///< class for AVOptions
     AVCodecContext *avctx;
     /* Frame header */
     int frame_type;             ///< type of the current frame
@@ -358,7 +361,6 @@ typedef struct {
     int bit_rate;               ///< transmission bit rate
     int bit_rate_index;         ///< transmission bit rate index
 
-    int downmix;                ///< embedded downmix enabled
     int dynrange;               ///< embedded dynamic range flag
     int timestamp;              ///< embedded time stamp flag
     int aux_data;               ///< auxiliary data flag
@@ -397,13 +399,20 @@ typedef struct {
     int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];      ///< prediction VQ coefs
     int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];           ///< bit allocation index
     int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< transition mode (transients)
-    int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2];    ///< scale factors (2 if transient)
+    int32_t scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2];///< scale factors (2 if transient)
     int joint_huff[DCA_PRIM_CHANNELS_MAX];                       ///< joint subband scale factors codebook
     int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
-    int downmix_coef[DCA_PRIM_CHANNELS_MAX][2];                  ///< stereo downmix coefficients
+    float downmix_coef[DCA_PRIM_CHANNELS_MAX + 1][2];            ///< stereo downmix coefficients
     int dynrange_coef;                                           ///< dynamic range coefficient
 
-    int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];       ///< VQ encoded high frequency subbands
+    /* Core substream's embedded downmix coefficients (cf. ETSI TS 102 114 V1.4.1)
+     * Input:  primary audio channels (incl. LFE if present)
+     * Output: downmix audio channels (up to 4, no LFE) */
+    uint8_t  core_downmix;                                       ///< embedded downmix coefficients available
+    uint8_t  core_downmix_amode;                                 ///< audio channel arrangement of embedded downmix
+    uint16_t core_downmix_codes[DCA_PRIM_CHANNELS_MAX + 1][4];   ///< embedded downmix coefficients (9-bit codes)
+
+    int32_t  high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];  ///< VQ encoded high frequency subbands
 
     float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)];      ///< Low frequency effect data
     int lfe_scale_factor;
@@ -437,6 +446,7 @@ typedef struct {
     /* XCh extension information */
     int xch_present;            ///< XCh extension present and valid
     int xch_base_channel;       ///< index of first (only) channel containing XCH data
+    int xch_disable;            ///< whether the XCh extension should be decoded or not
 
     /* XXCH extension information */
     int xxch_chset;
@@ -589,7 +599,7 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
         if (get_bits1(&s->gb)) {
             embedded_downmix = get_bits1(&s->gb);
             scale_factor     =
-               1.0f / dca_downmix_scale_factors[(get_bits(&s->gb, 6) - 1) << 2];
+               1.0f / dca_dmixtable[(get_bits(&s->gb, 6) - 1) << 2];
 
             s->xxch_dmix_sf[s->xxch_chset] = scale_factor;
 
@@ -610,7 +620,7 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
 
                         coeff = get_bits(&s->gb, 7);
                         sign  = (coeff & 64) ? 1.0 : -1.0;
-                        mag   = dca_downmix_scale_factors[((coeff & 63) - 1) << 2];
+                        mag   = dca_dmixtable[((coeff & 63) - 1) << 2];
                         ichan = dca_xxch2index(s, 1 << i);
                         s->xxch_dmix_coeff[j][ichan] = sign * mag;
                     }
@@ -724,7 +734,7 @@ static int dca_parse_frame_header(DCAContext *s)
     if (!s->bit_rate)
         return AVERROR_INVALIDDATA;
 
-    s->downmix           = get_bits(&s->gb, 1); /* note: this is FixedBit == 0 */
+    skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)
     s->dynrange          = get_bits(&s->gb, 1);
     s->timestamp         = get_bits(&s->gb, 1);
     s->aux_data          = get_bits(&s->gb, 1);
@@ -771,7 +781,6 @@ static int dca_parse_frame_header(DCAContext *s)
            s->sample_rate);
     av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i bits/s\n",
            s->bit_rate);
-    av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix);
     av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
     av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
     av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);
@@ -954,33 +963,6 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
         }
     }
 
-    /* Stereo downmix coefficients */
-    if (!base_channel && s->prim_channels > 2) {
-        if (s->downmix) {
-            for (j = base_channel; j < s->prim_channels; j++) {
-                s->downmix_coef[j][0] = get_bits(&s->gb, 7);
-                s->downmix_coef[j][1] = get_bits(&s->gb, 7);
-            }
-        } else {
-            int am = s->amode & DCA_CHANNEL_MASK;
-            if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {
-                av_log(s->avctx, AV_LOG_ERROR,
-                       "Invalid channel mode %d\n", am);
-                return AVERROR_INVALIDDATA;
-            }
-            if (s->prim_channels > FF_ARRAY_ELEMS(dca_default_coeffs[0])) {
-                avpriv_request_sample(s->avctx, "Downmixing %d channels",
-                                      s->prim_channels);
-                return AVERROR_PATCHWELCOME;
-            }
-
-            for (j = base_channel; j < s->prim_channels; j++) {
-                s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];
-                s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];
-            }
-        }
-    }
-
     /* Dynamic range coefficient */
     if (!base_channel && s->dynrange)
         s->dynrange_coef = get_bits(&s->gb, 8);
@@ -1080,16 +1062,6 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
             av_log(s->avctx, AV_LOG_DEBUG, "\n");
         }
     }
-    if (!base_channel && s->prim_channels > 2 && s->downmix) {
-        av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");
-        for (j = 0; j < s->prim_channels; j++) {
-            av_log(s->avctx, AV_LOG_DEBUG, "Channel 0, %d = %f\n", j,
-                   dca_downmix_coeffs[s->downmix_coef[j][0]]);
-            av_log(s->avctx, AV_LOG_DEBUG, "Channel 1, %d = %f\n", j,
-                   dca_downmix_coeffs[s->downmix_coef[j][1]]);
-        }
-        av_log(s->avctx, AV_LOG_DEBUG, "\n");
-    }
     for (j = base_channel; j < s->prim_channels; j++)
         for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
             av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
@@ -1132,7 +1104,7 @@ static void qmf_32_subbands(DCAContext *s, int chans,
 
 static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
                                   int num_deci_sample, float *samples_in,
-                                  float *samples_out, float scale)
+                                  float *samples_out)
 {
     /* samples_in: An array holding decimated samples.
      *   Samples in current subframe starts from samples_in[0],
@@ -1142,23 +1114,23 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
      * samples_out: An array holding interpolated samples
      */
 
-    int decifactor;
+    int idx;
     const float *prCoeff;
     int deciindex;
 
     /* Select decimation filter */
     if (decimation_select == 1) {
-        decifactor = 64;
+        idx = 1;
         prCoeff = lfe_fir_128;
     } else {
-        decifactor = 32;
+        idx = 0;
         prCoeff = lfe_fir_64;
     }
     /* Interpolation */
     for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
-        s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor, scale);
+        s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff);
         samples_in++;
-        samples_out += 2 * decifactor;
+        samples_out += 2 * 32 * (1 + idx);
     }
 }
 
@@ -1184,29 +1156,23 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
         op2                                     \
     }
 
-static void dca_downmix(float **samples, int srcfmt,
+static void dca_downmix(float **samples, int srcfmt, int lfe_present,
-                        int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],
+                        float coef[DCA_PRIM_CHANNELS_MAX + 1][2],
                         const int8_t *channel_mapping)
 {
     int c, l, r, sl, sr, s;
     int i;
     float t, u, v;
-    float coef[DCA_PRIM_CHANNELS_MAX][2];
-
-    for (i = 0; i < DCA_PRIM_CHANNELS_MAX; i++) {
-        coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];
-        coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];
-    }
 
     switch (srcfmt) {
     case DCA_MONO:
-    case DCA_CHANNEL:
-    case DCA_STEREO_TOTAL:
-    case DCA_STEREO_SUMDIFF:
     case DCA_4F2R:
         av_log(NULL, AV_LOG_ERROR, "Not implemented!\n");
         break;
+    case DCA_CHANNEL:
     case DCA_STEREO:
+    case DCA_STEREO_TOTAL:
+    case DCA_STEREO_SUMDIFF:
         break;
     case DCA_3F:
         c = channel_mapping[0];
@@ -1241,6 +1207,14 @@ static void dca_downmix(float **samples, int srcfmt,
                           MIX_REAR2(samples, sl, sr, 3, coef));
         break;
     }
+    if (lfe_present) {
+        int lf_buf = dca_lfe_index[srcfmt];
+        int lf_idx = dca_channels [srcfmt];
+        for (i = 0; i < 256; i++) {
+            samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];
+            samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];
+        }
+    }
 }
 
 
@@ -1271,16 +1245,6 @@ static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)
 static const uint8_t abits_sizes[7]  = { 7, 10, 12, 13, 15, 17, 19 };
 static const uint8_t abits_levels[7] = { 3,  5,  7,  9, 13, 17, 25 };
 
-#ifndef int8x8_fmul_int32
-static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
-{
-    float fscale = scale / 16.0;
-    int i;
-    for (i = 0; i < 8; i++)
-        dst[i] = src[i] * fscale;
-}
-#endif
-
 static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
 {
     int k, l;
@@ -1377,16 +1341,27 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
              */
             if (s->prediction_mode[k][l]) {
                 int n;
-                for (m = 0; m < 8; m++) {
+                if (s->predictor_history)
-                    for (n = 1; n <= 4; n++)
+                    subband_samples[k][l][0] += (adpcm_vb[s->prediction_vq[k][l]][0] *
+                                                 s->subband_samples_hist[k][l][3] +
+                                                 adpcm_vb[s->prediction_vq[k][l]][1] *
+                                                 s->subband_samples_hist[k][l][2] +
+                                                 adpcm_vb[s->prediction_vq[k][l]][2] *
+                                                 s->subband_samples_hist[k][l][1] +
+                                                 adpcm_vb[s->prediction_vq[k][l]][3] *
+                                                 s->subband_samples_hist[k][l][0]) *
+                                                (1.0f / 8192);
+                for (m = 1; m < 8; m++) {
+                    float sum = adpcm_vb[s->prediction_vq[k][l]][0] *
+                                subband_samples[k][l][m - 1];
+                    for (n = 2; n <= 4; n++)
                         if (m >= n)
-                            subband_samples[k][l][m] +=
+                            sum += adpcm_vb[s->prediction_vq[k][l]][n - 1] *
-                                (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
+                                   subband_samples[k][l][m - n];
-                                 subband_samples[k][l][m - n] / 8192);
                         else if (s->predictor_history)
-                            subband_samples[k][l][m] +=
+                            sum += adpcm_vb[s->prediction_vq[k][l]][n - 1] *
-                                (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
+                                   s->subband_samples_hist[k][l][m - n + 4];
-                                 s->subband_samples_hist[k][l][m - n + 4] / 8192);
+                    subband_samples[k][l][m] += sum * (1.0f / 8192);
                 }
             }
         }
@@ -1394,20 +1369,16 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
         /*
          * Decode VQ encoded high frequencies
          */
-        for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {
+        if (s->subband_activity[k] > s->vq_start_subband[k]) {
-            /* 1 vector -> 32 samples but we only need the 8 samples
-             * for this subsubframe. */
-            int hfvq = s->high_freq_vq[k][l];
-
             if (!s->debug_flag & 0x01) {
                 av_log(s->avctx, AV_LOG_DEBUG,
                        "Stream with high frequencies VQ coding\n");
                 s->debug_flag |= 0x01;
             }
-
+            s->dcadsp.decode_hf(subband_samples[k], s->high_freq_vq[k],
-            int8x8_fmul_int32(subband_samples[k][l],
+                                high_freq_vq, subsubframe * 8,
-                              &high_freq_vq[hfvq][subsubframe * 8],
+                                s->scale_factor[k], s->vq_start_subband[k],
-                              s->scale_factor[k][l][0]);
+                                s->subband_activity[k]);
         }
     }
 
@@ -1426,9 +1397,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
     /* Backup predictor history for adpcm */
     for (k = base_channel; k < s->prim_channels; k++)
         for (l = 0; l < s->vq_start_subband[k]; l++)
-            memcpy(s->subband_samples_hist[k][l],
+            AV_COPY128(s->subband_samples_hist[k][l], &subband_samples[k][l][4]);
-                   &subband_samples[k][l][4],
-                   4 * sizeof(subband_samples[0][0][0]));
 
     return 0;
 }
@@ -1449,28 +1418,29 @@ static int dca_filter_channels(DCAContext *s, int block_index)
                             M_SQRT1_2 / 32768.0 /* pcm_to_double[s->source_pcm_res] */);
     }
 
-    /* Down mixing */
-    AV_NOWARN_DEPRECATED( request_channels = s->avctx->request_channels; );
-    if (request_channels == 2 && s->prim_channels > 2) {
-        dca_downmix(s->samples_chanptr, s->amode, s->downmix_coef, s->channel_order_tab);
-    }
-
     /* Generate LFE samples for this subsubframe FIXME!!! */
-    if (s->output & DCA_LFE) {
+    if (s->lfe) {
         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
                               s->lfe_data + 2 * s->lfe * (block_index + 4),
-                              s->samples_chanptr[s->lfe_index],
+                              s->samples_chanptr[s->lfe_index]);
-                              1.0 / (256.0 * 32768.0));
         /* Outputs 20bits pcm samples */
     }
 
+    /* Downmixing to Stereo */
+    if (s->prim_channels + !!s->lfe > 2 &&
+        s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
+        dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
+                    s->channel_order_tab);
+    }
+
     return 0;
 }
 
 
 static int dca_subframe_footer(DCAContext *s, int base_channel)
 {
-    int aux_data_count = 0, i;
+    int in, out, aux_data_count, aux_data_end, reserved;
+    uint32_t nsyncaux;
 
     /*
      * Unpack optional information
@@ -1481,13 +1451,89 @@ static int dca_subframe_footer(DCAContext *s, int base_channel)
         if (s->timestamp)
             skip_bits_long(&s->gb, 32);
 
-        if (s->aux_data)
+        if (s->aux_data) {
             aux_data_count = get_bits(&s->gb, 6);
 
-        for (i = 0; i < aux_data_count; i++)
+            // align (32-bit)
-            get_bits(&s->gb, 8);
+            skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
 
-        if (s->crc_present && (s->downmix || s->dynrange))
+            aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);
+
+            if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {
+                av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",
+                       nsyncaux);
+                return AVERROR_INVALIDDATA;
+            }
+
+            if (get_bits1(&s->gb)) { // bAUXTimeStampFlag
+                avpriv_request_sample(s->avctx,
+                                      "Auxiliary Decode Time Stamp Flag");
+                // align (4-bit)
+                skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);
+                // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)
+                skip_bits_long(&s->gb, 44);
+            }
+
+            if ((s->core_downmix = get_bits1(&s->gb))) {
+                int am = get_bits(&s->gb, 3);
+                switch (am) {
+                case 0:
+                    s->core_downmix_amode = DCA_MONO;
+                    break;
+                case 1:
+                    s->core_downmix_amode = DCA_STEREO;
+                    break;
+                case 2:
+                    s->core_downmix_amode = DCA_STEREO_TOTAL;
+                    break;
+                case 3:
+                    s->core_downmix_amode = DCA_3F;
+                    break;
+                case 4:
+                    s->core_downmix_amode = DCA_2F1R;
+                    break;
+                case 5:
+                    s->core_downmix_amode = DCA_2F2R;
+                    break;
+                case 6:
+                    s->core_downmix_amode = DCA_3F1R;
+                    break;
+                default:
+                    av_log(s->avctx, AV_LOG_ERROR,
+                           "Invalid mode %d for embedded downmix coefficients\n",
+                           am);
+                    return AVERROR_INVALIDDATA;
+                }
+                for (out = 0; out < dca_channels[s->core_downmix_amode]; out++) {
+                    for (in = 0; in < s->prim_channels + !!s->lfe; in++) {
+                        uint16_t tmp = get_bits(&s->gb, 9);
+                        if ((tmp & 0xFF) > 241) {
+                            av_log(s->avctx, AV_LOG_ERROR,
+                                   "Invalid downmix coefficient code %"PRIu16"\n",
+                                   tmp);
+                            return AVERROR_INVALIDDATA;
+                        }
+                        s->core_downmix_codes[in][out] = tmp;
+                    }
+                }
+            }
+
+            align_get_bits(&s->gb); // byte align
+            skip_bits(&s->gb, 16);  // nAUXCRC16
+
+            // additional data (reserved, cf. ETSI TS 102 114 V1.4.1)
+            if ((reserved = (aux_data_end - get_bits_count(&s->gb))) < 0) {
+                 av_log(s->avctx, AV_LOG_ERROR,
+                        "Overread auxiliary data by %d bits\n", -reserved);
+                return AVERROR_INVALIDDATA;
+            } else if (reserved) {
+                avpriv_request_sample(s->avctx,
+                                      "Core auxiliary data reserved content");
+                skip_bits_long(&s->gb, reserved);
+            }
+        }
+
+        if (s->crc_present && s->dynrange)
             get_bits(&s->gb, 16);
     }
 
@@ -2119,6 +2165,55 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
     /* record number of core channels incase less than max channels are requested */
     num_core_channels = s->prim_channels;
 
+    if (s->prim_channels + !!s->lfe > 2 &&
+        avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
+            /* Stereo downmix coefficients
+             *
+             * The decoder can only downmix to 2-channel, so we need to ensure
+             * embedded downmix coefficients are actually targeting 2-channel.
+             */
+            if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||
+                                    s->core_downmix_amode == DCA_STEREO_TOTAL)) {
+                int sign, code;
+                for (i = 0; i < s->prim_channels + !!s->lfe; i++) {
+                    sign = s->core_downmix_codes[i][0] & 0x100 ? 1 : -1;
+                    code = s->core_downmix_codes[i][0] & 0x0FF;
+                    s->downmix_coef[i][0] = (!code ? 0.0f :
+                                             sign * dca_dmixtable[code - 1]);
+                    sign = s->core_downmix_codes[i][1] & 0x100 ? 1 : -1;
+                    code = s->core_downmix_codes[i][1] & 0x0FF;
+                    s->downmix_coef[i][1] = (!code ? 0.0f :
+                                             sign * dca_dmixtable[code - 1]);
+                }
+                s->output = s->core_downmix_amode;
+            } else {
+                int am = s->amode & DCA_CHANNEL_MASK;
+                if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {
+                    av_log(s->avctx, AV_LOG_ERROR,
+                           "Invalid channel mode %d\n", am);
+                    return AVERROR_INVALIDDATA;
+                }
+                if (s->prim_channels + !!s->lfe >
+                    FF_ARRAY_ELEMS(dca_default_coeffs[0])) {
+                    avpriv_request_sample(s->avctx, "Downmixing %d channels",
+                                          s->prim_channels + !!s->lfe);
+                    return AVERROR_PATCHWELCOME;
+                }
+                for (i = 0; i < s->prim_channels + !!s->lfe; i++) {
+                    s->downmix_coef[i][0] = dca_default_coeffs[am][i][0];
+                    s->downmix_coef[i][1] = dca_default_coeffs[am][i][1];
+                }
+            }
+            av_dlog(s->avctx, "Stereo downmix coeffs:\n");
+            for (i = 0; i < s->prim_channels + !!s->lfe; i++) {
+                av_dlog(s->avctx, "L, input channel %d = %f\n", i,
+                        s->downmix_coef[i][0]);
+                av_dlog(s->avctx, "R, input channel %d = %f\n", i,
+                        s->downmix_coef[i][1]);
+            }
+            av_dlog(s->avctx, "\n");
+    }
+
     if (s->ext_coding)
         s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
     else
@@ -2227,21 +2322,24 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
 
     full_channels = channels = s->prim_channels + !!s->lfe;
 
-    AV_NOWARN_DEPRECATED( request_channels = avctx->request_channels; );
-
     /* If we have XXCH then the channel layout is managed differently */
     /* note that XLL will also have another way to do things */
     if (!(s->core_ext_mask & DCA_EXT_XXCH)
-        || (s->core_ext_mask & DCA_EXT_XXCH && request_channels > 0
+        || (s->core_ext_mask & DCA_EXT_XXCH && avctx->request_channels > 0
-            && request_channels
+            && avctx->request_channels
             < num_core_channels + !!s->lfe + s->xxch_chset_nch[0]))
     { /* xxx should also do MA extensions */
         if (s->amode < 16) {
             avctx->channel_layout = dca_core_channel_layout[s->amode];
-
+#if FF_API_REQUEST_CHANNELS
-            if (s->xch_present && (!request_channels ||
+FF_DISABLE_DEPRECATION_WARNINGS
-                                   request_channels
+            if (s->xch_present && !s->xch_disable &&
-                                   > num_core_channels + !!s->lfe)) {
+                (!avctx->request_channels ||
+                 avctx->request_channels > num_core_channels + !!s->lfe)) {
+FF_ENABLE_DEPRECATION_WARNINGS
+#else
+            if (s->xch_present && !s->xch_disable) {
+#endif
                 avctx->channel_layout |= AV_CH_BACK_CENTER;
                 if (s->lfe) {
                     avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
@@ -2270,9 +2368,10 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
                 return AVERROR_INVALIDDATA;
             }
 
-            if (request_channels == 2 && s->prim_channels > 2) {
+            if (s->prim_channels + !!s->lfe > 2 &&
+                avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
                 channels = 2;
-                s->output = DCA_STEREO;
+                s->output = s->prim_channels == 2 ? s->amode : DCA_STEREO;
                 avctx->channel_layout = AV_CH_LAYOUT_STEREO;
             }
             else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {
@@ -2318,7 +2417,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
          * masks in some sense -- unfortunately some channels could overlap */
         if (av_popcount(channel_mask) != av_popcount(channel_layout)) {
             av_log(avctx, AV_LOG_DEBUG,
-                   "DTS-XXCH: Inconsistant avcodec/dts channel layouts\n");
+                   "DTS-XXCH: Inconsistent avcodec/dts channel layouts\n");
             return AVERROR_INVALIDDATA;
         }
 
@@ -2456,6 +2555,15 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
     for (i = 0; i < 2 * s->lfe * 4; i++)
         s->lfe_data[i] = s->lfe_data[i + lfe_samples];
 
+    /* AVMatrixEncoding
+     *
+     * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
+    ret = ff_side_data_update_matrix_encoding(frame,
+                                              (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?
+                                              AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);
+    if (ret < 0)
+        return ret;
+
     *got_frame_ptr = 1;
 
     return buf_size;
@@ -2486,11 +2594,15 @@ static av_cold int dca_decode_init(AVCodecContext *avctx)
     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
 
     /* allow downmixing to stereo */
-    AV_NOWARN_DEPRECATED( request_channels = avctx->request_channels; );
+#if FF_API_REQUEST_CHANNELS
-    if (avctx->channels > 0 && request_channels < avctx->channels &&
+FF_DISABLE_DEPRECATION_WARNINGS
-        request_channels == 2) {
+    if (avctx->request_channels == 2)
-        avctx->channels = request_channels;
+        avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;
-    }
+FF_ENABLE_DEPRECATION_WARNINGS
+#endif
+    if (avctx->channels > 2 &&
+        avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
+        avctx->channels = 2;
 
     return 0;
 }
@@ -2512,6 +2624,19 @@ static const AVProfile profiles[] = {
     { FF_PROFILE_UNKNOWN },
 };
 
+static const AVOption options[] = {
+    { "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM|AV_OPT_FLAG_AUDIO_PARAM },
+    { NULL },
+};
+
+static const AVClass dca_decoder_class = {
+    .class_name = "DCA decoder",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+    .category   = AV_CLASS_CATEGORY_DECODER,
+};
+
 AVCodec ff_dca_decoder = {
     .name            = "dca",
     .long_name       = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
@@ -2525,4 +2650,5 @@ AVCodec ff_dca_decoder = {
     .sample_fmts     = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                        AV_SAMPLE_FMT_NONE },
     .profiles        = NULL_IF_CONFIG_SMALL(profiles),
+    .priv_class      = &dca_decoder_class,
 };

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/dcadsp.c

@@ -24,25 +24,42 @@
 #include "libavutil/intreadwrite.h"
 #include "dcadsp.h"
 
-static void dca_lfe_fir_c(float *out, const float *in, const float *coefs,
+static void decode_hf_c(float dst[DCA_SUBBANDS][8],
-                          int decifactor, float scale)
+                        const int32_t vq_num[DCA_SUBBANDS],
+                        const int8_t hf_vq[1024][32], intptr_t vq_offset,
+                        int32_t scale[DCA_SUBBANDS][2],
+                        intptr_t start, intptr_t end)
 {
-    float *out2 = out + decifactor;
+    int i, l;
-    const float *cf0 = coefs;
+
-    const float *cf1 = coefs + 256;
+    for (l = start; l < end; l++) {
+        /* 1 vector -> 32 samples but we only need the 8 samples
+         * for this subsubframe. */
+        const int8_t *ptr = &hf_vq[vq_num[l]][vq_offset];
+        float fscale = scale[l][0] * (1 / 16.0);
+        for (i = 0; i < 8; i++)
+            dst[l][i] = ptr[i] * fscale;
+    }
+}
+
+static inline void
+dca_lfe_fir(float *out, const float *in, const float *coefs,
+            int decifactor)
+{
+    float *out2    = out + 2 * decifactor - 1;
+    int num_coeffs = 256 / decifactor;
     int j, k;
 
     /* One decimated sample generates 2*decifactor interpolated ones */
     for (k = 0; k < decifactor; k++) {
         float v0 = 0.0;
         float v1 = 0.0;
-        for (j = 0; j < 256 / decifactor; j++) {
+        for (j = 0; j < num_coeffs; j++, coefs++) {
-            float s = in[-j];
+            v0 += in[-j] * *coefs;
-            v0 += s * *cf0++;
+            v1 += in[j + 1 - num_coeffs] * *coefs;
-            v1 += s * *--cf1;
         }
-        *out++  = v0 * scale;
+        *out++  = v0;
-        *out2++ = v1 * scale;
+        *out2-- = v1;
     }
 }
 
@@ -74,9 +91,22 @@ static void dca_qmf_32_subbands(float samples_in[32][8], int sb_act,
     }
 }
 
+static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs)
+{
+    dca_lfe_fir(out, in, coefs, 32);
+}
+
+static void dca_lfe_fir1_c(float *out, const float *in, const float *coefs)
+{
+    dca_lfe_fir(out, in, coefs, 64);
+}
+
 av_cold void ff_dcadsp_init(DCADSPContext *s)
 {
-    s->lfe_fir = dca_lfe_fir_c;
+    s->lfe_fir[0] = dca_lfe_fir0_c;
+    s->lfe_fir[1] = dca_lfe_fir1_c;
     s->qmf_32_subbands = dca_qmf_32_subbands;
+    s->decode_hf = decode_hf_c;
     if (ARCH_ARM) ff_dcadsp_init_arm(s);
+    if (ARCH_X86) ff_dcadsp_init_x86(s);
 }

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/dcadsp.h

@@ -22,18 +22,25 @@
 #include "avfft.h"
 #include "synth_filter.h"
 
+#define DCA_SUBBANDS 64
+
 typedef struct DCADSPContext {
-    void (*lfe_fir)(float *out, const float *in, const float *coefs,
+    void (*lfe_fir[2])(float *out, const float *in, const float *coefs);
-                    int decifactor, float scale);
     void (*qmf_32_subbands)(float samples_in[32][8], int sb_act,
                             SynthFilterContext *synth, FFTContext *imdct,
                             float synth_buf_ptr[512],
                             int *synth_buf_offset, float synth_buf2[32],
                             const float window[512], float *samples_out,
                             float raXin[32], float scale);
+    void (*decode_hf)(float dst[DCA_SUBBANDS][8],
+                      const int32_t vq_num[DCA_SUBBANDS],
+                      const int8_t hf_vq[1024][32], intptr_t vq_offset,
+                      int32_t scale[DCA_SUBBANDS][2],
+                      intptr_t start, intptr_t end);
 } DCADSPContext;
 
 void ff_dcadsp_init(DCADSPContext *s);
 void ff_dcadsp_init_arm(DCADSPContext *s);
+void ff_dcadsp_init_x86(DCADSPContext *s);
 
 #endif /* AVCODEC_DCADSP_H */

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/dirac.c

@@ -30,6 +30,7 @@
 #include "dirac.h"
 #include "avcodec.h"
 #include "golomb.h"
+#include "internal.h"
 #include "mpeg12data.h"
 
 /* defaults for source parameters */
@@ -317,11 +318,10 @@ int avpriv_dirac_parse_sequence_header(AVCodecContext *avctx, GetBitContext *gb,
     if ((ret = parse_source_parameters(avctx, gb, source)))
         return ret;
 
-    if ((ret = av_image_check_size(source->width, source->height, 0, avctx)))
+    ret = ff_set_dimensions(avctx, source->width, source->height);
+    if (ret < 0)
         return ret;
 
-    avcodec_set_dimensions(avctx, source->width, source->height);
-
     /* [DIRAC_STD] picture_coding_mode shall be 0 for fields and 1 for frames
      * currently only used to signal field coding */
     picture_coding_mode = svq3_get_ue_golomb(gb);

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/dirac_dwt.c

@@ -536,6 +536,7 @@ int ff_spatial_idwt_init2(DWTContext *d, IDWTELEM *buffer, int width, int height
         d->vertical_compose_l0 = (void*)vertical_compose_fidelityiL0;
         d->vertical_compose_h0 = (void*)vertical_compose_fidelityiH0;
         d->horizontal_compose = horizontal_compose_fidelityi;
+        d->support = 0; // not really used
         break;
     case DWT_DIRAC_DAUB9_7:
         d->spatial_compose = spatial_compose_daub97i_dy;

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/diracdec.c

@@ -80,7 +80,7 @@
 #define DIVRNDUP(a, b) (((a) + (b) - 1) / (b))
 
 typedef struct {
-    AVFrame avframe;
+    AVFrame *avframe;
     int interpolated[3];    /* 1 if hpel[] is valid */
     uint8_t *hpel[3][4];
     uint8_t *hpel_base[3][4];
@@ -291,7 +291,7 @@ static DiracFrame *remove_frame(DiracFrame *framelist[], int picnum)
     int i, remove_idx = -1;
 
     for (i = 0; framelist[i]; i++)
-        if (framelist[i]->avframe.display_picture_number == picnum) {
+        if (framelist[i]->avframe->display_picture_number == picnum) {
             remove_pic = framelist[i];
             remove_idx = i;
         }
@@ -364,8 +364,8 @@ static void free_sequence_buffers(DiracContext *s)
     int i, j, k;
 
     for (i = 0; i < MAX_FRAMES; i++) {
-        if (s->all_frames[i].avframe.data[0]) {
+        if (s->all_frames[i].avframe->data[0]) {
-            av_frame_unref(&s->all_frames[i].avframe);
+            av_frame_unref(s->all_frames[i].avframe);
             memset(s->all_frames[i].interpolated, 0, sizeof(s->all_frames[i].interpolated));
         }
 
@@ -393,17 +393,23 @@ static void free_sequence_buffers(DiracContext *s)
 static av_cold int dirac_decode_init(AVCodecContext *avctx)
 {
     DiracContext *s = avctx->priv_data;
+    int i;
+
     s->avctx = avctx;
     s->frame_number = -1;
 
-    if (avctx->flags&CODEC_FLAG_EMU_EDGE) {
-        av_log(avctx, AV_LOG_ERROR, "Edge emulation not supported!\n");
-        return AVERROR_PATCHWELCOME;
-    }
-
     ff_dsputil_init(&s->dsp, avctx);
     ff_diracdsp_init(&s->diracdsp);
 
+    for (i = 0; i < MAX_FRAMES; i++) {
+        s->all_frames[i].avframe = av_frame_alloc();
+        if (!s->all_frames[i].avframe) {
+            while (i > 0)
+                av_frame_free(&s->all_frames[--i].avframe);
+            return AVERROR(ENOMEM);
+        }
+    }
+
     return 0;
 }
 
@@ -417,7 +423,13 @@ static void dirac_decode_flush(AVCodecContext *avctx)
 
 static av_cold int dirac_decode_end(AVCodecContext *avctx)
 {
+    DiracContext *s = avctx->priv_data;
+    int i;
+
     dirac_decode_flush(avctx);
+    for (i = 0; i < MAX_FRAMES; i++)
+        av_frame_free(&s->all_frames[i].avframe);
+
     return 0;
 }
 
@@ -1414,8 +1426,8 @@ static int mc_subpel(DiracContext *s, DiracBlock *block, const uint8_t *src[5],
         y + p->yblen > p->height+EDGE_WIDTH/2 ||
         x < 0 || y < 0) {
         for (i = 0; i < nplanes; i++) {
-            ff_emulated_edge_mc(s->edge_emu_buffer[i], p->stride,
+            ff_emulated_edge_mc(s->edge_emu_buffer[i], src[i],
-                                src[i], p->stride,
+                                p->stride, p->stride,
                                 p->xblen, p->yblen, x, y,
                                 p->width+EDGE_WIDTH/2, p->height+EDGE_WIDTH/2);
             src[i] = s->edge_emu_buffer[i];
@@ -1519,8 +1531,8 @@ static void interpolate_refplane(DiracContext *s, DiracFrame *ref, int plane, in
        just use 8 for everything for the moment */
     int i, edge = EDGE_WIDTH/2;
 
-    ref->hpel[plane][0] = ref->avframe.data[plane];
+    ref->hpel[plane][0] = ref->avframe->data[plane];
-    s->dsp.draw_edges(ref->hpel[plane][0], ref->avframe.linesize[plane], width, height, edge, edge, EDGE_TOP | EDGE_BOTTOM); /* EDGE_TOP | EDGE_BOTTOM values just copied to make it build, this needs to be ensured */
+    s->dsp.draw_edges(ref->hpel[plane][0], ref->avframe->linesize[plane], width, height, edge, edge, EDGE_TOP | EDGE_BOTTOM); /* EDGE_TOP | EDGE_BOTTOM values just copied to make it build, this needs to be ensured */
 
     /* no need for hpel if we only have fpel vectors */
     if (!s->mv_precision)
@@ -1528,18 +1540,18 @@ static void interpolate_refplane(DiracContext *s, DiracFrame *ref, int plane, in
 
     for (i = 1; i < 4; i++) {
         if (!ref->hpel_base[plane][i])
-            ref->hpel_base[plane][i] = av_malloc((height+2*edge) * ref->avframe.linesize[plane] + 32);
+            ref->hpel_base[plane][i] = av_malloc((height+2*edge) * ref->avframe->linesize[plane] + 32);
         /* we need to be 16-byte aligned even for chroma */
-        ref->hpel[plane][i] = ref->hpel_base[plane][i] + edge*ref->avframe.linesize[plane] + 16;
+        ref->hpel[plane][i] = ref->hpel_base[plane][i] + edge*ref->avframe->linesize[plane] + 16;
     }
 
     if (!ref->interpolated[plane]) {
         s->diracdsp.dirac_hpel_filter(ref->hpel[plane][1], ref->hpel[plane][2],
                                       ref->hpel[plane][3], ref->hpel[plane][0],
-                                      ref->avframe.linesize[plane], width, height);
+                                      ref->avframe->linesize[plane], width, height);
-        s->dsp.draw_edges(ref->hpel[plane][1], ref->avframe.linesize[plane], width, height, edge, edge, EDGE_TOP | EDGE_BOTTOM);
+        s->dsp.draw_edges(ref->hpel[plane][1], ref->avframe->linesize[plane], width, height, edge, edge, EDGE_TOP | EDGE_BOTTOM);
-        s->dsp.draw_edges(ref->hpel[plane][2], ref->avframe.linesize[plane], width, height, edge, edge, EDGE_TOP | EDGE_BOTTOM);
+        s->dsp.draw_edges(ref->hpel[plane][2], ref->avframe->linesize[plane], width, height, edge, edge, EDGE_TOP | EDGE_BOTTOM);
-        s->dsp.draw_edges(ref->hpel[plane][3], ref->avframe.linesize[plane], width, height, edge, edge, EDGE_TOP | EDGE_BOTTOM);
+        s->dsp.draw_edges(ref->hpel[plane][3], ref->avframe->linesize[plane], width, height, edge, edge, EDGE_TOP | EDGE_BOTTOM);
     }
     ref->interpolated[plane] = 1;
 }
@@ -1565,7 +1577,7 @@ static int dirac_decode_frame_internal(DiracContext *s)
 
     for (comp = 0; comp < 3; comp++) {
         Plane *p       = &s->plane[comp];
-        uint8_t *frame = s->current_picture->avframe.data[comp];
+        uint8_t *frame = s->current_picture->avframe->data[comp];
 
         /* FIXME: small resolutions */
         for (i = 0; i < 4; i++)
@@ -1629,6 +1641,29 @@ static int dirac_decode_frame_internal(DiracContext *s)
     return 0;
 }
 
+static int get_buffer_with_edge(AVCodecContext *avctx, AVFrame *f, int flags)
+{
+    int ret, i;
+    int chroma_x_shift, chroma_y_shift;
+    avcodec_get_chroma_sub_sample(avctx->pix_fmt, &chroma_x_shift, &chroma_y_shift);
+
+    f->width  = avctx->width  + 2 * EDGE_WIDTH;
+    f->height = avctx->height + 2 * EDGE_WIDTH + 2;
+    ret = ff_get_buffer(avctx, f, flags);
+    if (ret < 0)
+        return ret;
+
+    for (i = 0; f->data[i]; i++) {
+        int offset = (EDGE_WIDTH >> (i && i<3 ? chroma_y_shift : 0)) *
+                     f->linesize[i] + 32;
+        f->data[i] += offset;
+    }
+    f->width  = avctx->width;
+    f->height = avctx->height;
+
+    return 0;
+}
+
 /**
  * Dirac Specification ->
  * 11.1.1 Picture Header. picture_header()
@@ -1640,7 +1675,7 @@ static int dirac_decode_picture_header(DiracContext *s)
     GetBitContext *gb = &s->gb;
 
     /* [DIRAC_STD] 11.1.1 Picture Header. picture_header() PICTURE_NUM */
-    picnum = s->current_picture->avframe.display_picture_number = get_bits_long(gb, 32);
+    picnum = s->current_picture->avframe->display_picture_number = get_bits_long(gb, 32);
 
 
     av_log(s->avctx,AV_LOG_DEBUG,"PICTURE_NUM: %d\n",picnum);
@@ -1659,9 +1694,9 @@ static int dirac_decode_picture_header(DiracContext *s)
         /* Jordi: this is needed if the referenced picture hasn't yet arrived */
         for (j = 0; j < MAX_REFERENCE_FRAMES && refdist; j++)
             if (s->ref_frames[j]
-                && FFABS(s->ref_frames[j]->avframe.display_picture_number - refnum) < refdist) {
+                && FFABS(s->ref_frames[j]->avframe->display_picture_number - refnum) < refdist) {
                 s->ref_pics[i] = s->ref_frames[j];
-                refdist = FFABS(s->ref_frames[j]->avframe.display_picture_number - refnum);
+                refdist = FFABS(s->ref_frames[j]->avframe->display_picture_number - refnum);
             }
 
         if (!s->ref_pics[i] || refdist)
@@ -1670,22 +1705,21 @@ static int dirac_decode_picture_header(DiracContext *s)
         /* if there were no references at all, allocate one */
         if (!s->ref_pics[i])
             for (j = 0; j < MAX_FRAMES; j++)
-                if (!s->all_frames[j].avframe.data[0]) {
+                if (!s->all_frames[j].avframe->data[0]) {
                     s->ref_pics[i] = &s->all_frames[j];
-                    ff_get_buffer(s->avctx, &s->ref_pics[i]->avframe, AV_GET_BUFFER_FLAG_REF);
+                    get_buffer_with_edge(s->avctx, s->ref_pics[i]->avframe, AV_GET_BUFFER_FLAG_REF);
                     break;
                 }
     }
 
     /* retire the reference frames that are not used anymore */
-    AV_NOWARN_DEPRECATED(
+    if (s->current_picture->avframe->reference) {
-    if (s->current_picture->avframe.reference) {
         retire = picnum + dirac_get_se_golomb(gb);
         if (retire != picnum) {
             DiracFrame *retire_pic = remove_frame(s->ref_frames, retire);
 
             if (retire_pic)
-                retire_pic->avframe.reference &= DELAYED_PIC_REF;
+                retire_pic->avframe->reference &= DELAYED_PIC_REF;
             else
                 av_log(s->avctx, AV_LOG_DEBUG, "Frame to retire not found\n");
         }
@@ -1693,10 +1727,9 @@ static int dirac_decode_picture_header(DiracContext *s)
         /* if reference array is full, remove the oldest as per the spec */
         while (add_frame(s->ref_frames, MAX_REFERENCE_FRAMES, s->current_picture)) {
             av_log(s->avctx, AV_LOG_ERROR, "Reference frame overflow\n");
-            remove_frame(s->ref_frames, s->ref_frames[0]->avframe.display_picture_number)->avframe.reference &= DELAYED_PIC_REF;
+            remove_frame(s->ref_frames, s->ref_frames[0]->avframe->display_picture_number)->avframe->reference &= DELAYED_PIC_REF;
         }
     }
-    );
 
     if (s->num_refs) {
         if (dirac_unpack_prediction_parameters(s))  /* [DIRAC_STD] 11.2 Picture Prediction Data. picture_prediction() */
@@ -1719,7 +1752,7 @@ static int get_delayed_pic(DiracContext *s, AVFrame *picture, int *got_frame)
 
     /* find frame with lowest picture number */
     for (i = 1; s->delay_frames[i]; i++)
-        if (s->delay_frames[i]->avframe.display_picture_number < out->avframe.display_picture_number) {
+        if (s->delay_frames[i]->avframe->display_picture_number < out->avframe->display_picture_number) {
             out     = s->delay_frames[i];
             out_idx = i;
         }
@@ -1728,9 +1761,9 @@ static int get_delayed_pic(DiracContext *s, AVFrame *picture, int *got_frame)
         s->delay_frames[i] = s->delay_frames[i+1];
 
     if (out) {
-        AV_NOWARN_DEPRECATED( out->avframe.reference ^= DELAYED_PIC_REF; );
+        out->avframe->reference ^= DELAYED_PIC_REF;
         *got_frame = 1;
-        if((ret = av_frame_ref(picture, &out->avframe)) < 0)
+        if((ret = av_frame_ref(picture, out->avframe)) < 0)
             return ret;
     }
 
@@ -1792,14 +1825,14 @@ static int dirac_decode_data_unit(AVCodecContext *avctx, const uint8_t *buf, int
 
         /* find an unused frame */
         for (i = 0; i < MAX_FRAMES; i++)
-            if (s->all_frames[i].avframe.data[0] == NULL)
+            if (s->all_frames[i].avframe->data[0] == NULL)
                 pic = &s->all_frames[i];
         if (!pic) {
             av_log(avctx, AV_LOG_ERROR, "framelist full\n");
             return -1;
         }
 
-        avcodec_get_frame_defaults(&pic->avframe);
+        av_frame_unref(pic->avframe);
 
         /* [DIRAC_STD] Defined in 9.6.1 ... */
         tmp            =  parse_code & 0x03;                   /* [DIRAC_STD] num_refs()      */
@@ -1810,16 +1843,16 @@ static int dirac_decode_data_unit(AVCodecContext *avctx, const uint8_t *buf, int
         s->num_refs    = tmp;
         s->is_arith    = (parse_code & 0x48) == 0x08;          /* [DIRAC_STD] using_ac()      */
         s->low_delay   = (parse_code & 0x88) == 0x88;          /* [DIRAC_STD] is_low_delay()  */
-        AV_NOWARN_DEPRECATED( pic->avframe.reference = (parse_code & 0x0C) == 0x0C; );  /* [DIRAC_STD]  is_reference() */
+        pic->avframe->reference = (parse_code & 0x0C) == 0x0C;  /* [DIRAC_STD]  is_reference() */
-        pic->avframe.key_frame = s->num_refs == 0;             /* [DIRAC_STD] is_intra()      */
+        pic->avframe->key_frame = s->num_refs == 0;             /* [DIRAC_STD] is_intra()      */
-        pic->avframe.pict_type = s->num_refs + 1;              /* Definition of AVPictureType in avutil.h */
+        pic->avframe->pict_type = s->num_refs + 1;              /* Definition of AVPictureType in avutil.h */
 
-        if ((ret = ff_get_buffer(avctx, &pic->avframe, (parse_code & 0x0C) == 0x0C ? AV_GET_BUFFER_FLAG_REF : 0)) < 0)
+        if ((ret = get_buffer_with_edge(avctx, pic->avframe, (parse_code & 0x0C) == 0x0C ? AV_GET_BUFFER_FLAG_REF : 0)) < 0)
             return ret;
         s->current_picture = pic;
-        s->plane[0].stride = pic->avframe.linesize[0];
+        s->plane[0].stride = pic->avframe->linesize[0];
-        s->plane[1].stride = pic->avframe.linesize[1];
+        s->plane[1].stride = pic->avframe->linesize[1];
-        s->plane[2].stride = pic->avframe.linesize[2];
+        s->plane[2].stride = pic->avframe->linesize[2];
 
         /* [DIRAC_STD] 11.1 Picture parse. picture_parse() */
         if (dirac_decode_picture_header(s))
@@ -1835,7 +1868,7 @@ static int dirac_decode_data_unit(AVCodecContext *avctx, const uint8_t *buf, int
 static int dirac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame, AVPacket *pkt)
 {
     DiracContext *s     = avctx->priv_data;
-    DiracFrame *picture = data;
+    AVFrame *picture    = data;
     uint8_t *buf        = pkt->data;
     int buf_size        = pkt->size;
     int i, data_unit_size, buf_idx = 0;
@@ -1843,12 +1876,10 @@ static int dirac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame,
 
     /* release unused frames */
     for (i = 0; i < MAX_FRAMES; i++)
-        AV_NOWARN_DEPRECATED(
+        if (s->all_frames[i].avframe->data[0] && !s->all_frames[i].avframe->reference) {
-        if (s->all_frames[i].avframe.data[0] && !s->all_frames[i].avframe.reference) {
+            av_frame_unref(s->all_frames[i].avframe);
-            av_frame_unref(&s->all_frames[i].avframe);
             memset(s->all_frames[i].interpolated, 0, sizeof(s->all_frames[i].interpolated));
         }
-        );
 
     s->current_picture = NULL;
     *got_frame = 0;
@@ -1891,40 +1922,40 @@ static int dirac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame,
     if (!s->current_picture)
         return buf_size;
 
-    if (s->current_picture->avframe.display_picture_number > s->frame_number) {
+    if (s->current_picture->avframe->display_picture_number > s->frame_number) {
         DiracFrame *delayed_frame = remove_frame(s->delay_frames, s->frame_number);
 
-        AV_NOWARN_DEPRECATED( s->current_picture->avframe.reference |= DELAYED_PIC_REF; );
+        s->current_picture->avframe->reference |= DELAYED_PIC_REF;
 
         if (add_frame(s->delay_frames, MAX_DELAY, s->current_picture)) {
-            int min_num = s->delay_frames[0]->avframe.display_picture_number;
+            int min_num = s->delay_frames[0]->avframe->display_picture_number;
             /* Too many delayed frames, so we display the frame with the lowest pts */
             av_log(avctx, AV_LOG_ERROR, "Delay frame overflow\n");
             delayed_frame = s->delay_frames[0];
 
             for (i = 1; s->delay_frames[i]; i++)
-                if (s->delay_frames[i]->avframe.display_picture_number < min_num)
+                if (s->delay_frames[i]->avframe->display_picture_number < min_num)
-                    min_num = s->delay_frames[i]->avframe.display_picture_number;
+                    min_num = s->delay_frames[i]->avframe->display_picture_number;
 
             delayed_frame = remove_frame(s->delay_frames, min_num);
             add_frame(s->delay_frames, MAX_DELAY, s->current_picture);
         }
 
         if (delayed_frame) {
-            AV_NOWARN_DEPRECATED( delayed_frame->avframe.reference ^= DELAYED_PIC_REF; );
+            delayed_frame->avframe->reference ^= DELAYED_PIC_REF;
-            if((ret=av_frame_ref(data, &delayed_frame->avframe)) < 0)
+            if((ret=av_frame_ref(data, delayed_frame->avframe)) < 0)
                 return ret;
             *got_frame = 1;
         }
-    } else if (s->current_picture->avframe.display_picture_number == s->frame_number) {
+    } else if (s->current_picture->avframe->display_picture_number == s->frame_number) {
         /* The right frame at the right time :-) */
-        if((ret=av_frame_ref(data, &s->current_picture->avframe)) < 0)
+        if((ret=av_frame_ref(data, s->current_picture->avframe)) < 0)
             return ret;
         *got_frame = 1;
     }
 
     if (*got_frame)
-        s->frame_number = picture->avframe.display_picture_number + 1;
+        s->frame_number = picture->display_picture_number + 1;
 
     return buf_idx;
 }

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/eac3dec.c

@@ -346,23 +346,22 @@ int ff_eac3_parse_header(AC3DecodeContext *s)
     if (get_bits1(gbc)) {
         /* center and surround mix levels */
         if (s->channel_mode > AC3_CHMODE_STEREO) {
-            skip_bits(gbc, 2);  // skip preferred stereo downmix mode
+            s->preferred_downmix = get_bits(gbc, 2);
             if (s->channel_mode & 1) {
                 /* if three front channels exist */
-                skip_bits(gbc, 3); //skip Lt/Rt center mix level
+                s->center_mix_level_ltrt = get_bits(gbc, 3);
-                s->center_mix_level = get_bits(gbc, 3);
+                s->center_mix_level      = get_bits(gbc, 3);
             }
             if (s->channel_mode & 4) {
                 /* if a surround channel exists */
-                skip_bits(gbc, 3); //skip Lt/Rt surround mix level
+                s->surround_mix_level_ltrt = av_clip(get_bits(gbc, 3), 3, 7);
-                s->surround_mix_level = get_bits(gbc, 3);
+                s->surround_mix_level      = av_clip(get_bits(gbc, 3), 3, 7);
             }
         }
 
         /* lfe mix level */
-        if (s->lfe_on && get_bits1(gbc)) {
+        if (s->lfe_on && (s->lfe_mix_level_exists = get_bits1(gbc))) {
-            // TODO: use LFE mix level
+            s->lfe_mix_level = get_bits(gbc, 5);
-            skip_bits(gbc, 5); // skip LFE mix level code
         }
 
         /* info for mixing with other streams and substreams */
@@ -414,10 +413,11 @@ int ff_eac3_parse_header(AC3DecodeContext *s)
         s->bitstream_mode = get_bits(gbc, 3);
         skip_bits(gbc, 2); // skip copyright bit and original bitstream bit
         if (s->channel_mode == AC3_CHMODE_STEREO) {
-            skip_bits(gbc, 4); // skip Dolby surround and headphone mode
+            s->dolby_surround_mode  = get_bits(gbc, 2);
+            s->dolby_headphone_mode = get_bits(gbc, 2);
         }
         if (s->channel_mode >= AC3_CHMODE_2F2R) {
-            skip_bits(gbc, 2); // skip Dolby surround EX mode
+            s->dolby_surround_ex_mode = get_bits(gbc, 2);
         }
         for (i = 0; i < (s->channel_mode ? 1 : 2); i++) {
             if (get_bits1(gbc)) {

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/error_resilience.c

@@ -27,11 +27,13 @@
 
 #include <limits.h>
 
+#include "libavutil/internal.h"
 #include "avcodec.h"
 #include "error_resilience.h"
 #include "mpegvideo.h"
 #include "rectangle.h"
 #include "thread.h"
+#include "version.h"
 
 /**
  * @param stride the number of MVs to get to the next row
@@ -698,8 +700,8 @@ static int is_intra_more_likely(ERContext *s)
         return 0; // almost all MBs damaged -> use temporal prediction
 
     // prevent dsp.sad() check, that requires access to the image
-    if (CONFIG_MPEG_XVMC_DECODER    &&
+    if (CONFIG_XVMC    &&
-        s->avctx->xvmc_acceleration &&
+        s->avctx->hwaccel && s->avctx->hwaccel->decode_mb &&
         s->cur_pic->f.pict_type == AV_PICTURE_TYPE_I)
         return 1;
 
@@ -753,7 +755,7 @@ static int is_intra_more_likely(ERContext *s)
 
 void ff_er_frame_start(ERContext *s)
 {
-    if (!s->avctx->err_recognition)
+    if (!s->avctx->error_concealment)
         return;
 
     memset(s->error_status_table, ER_MB_ERROR | VP_START | ER_MB_END,
@@ -762,6 +764,17 @@ void ff_er_frame_start(ERContext *s)
     s->error_occurred = 0;
 }
 
+static int er_supported(ERContext *s)
+{
+    if(s->avctx->hwaccel && s->avctx->hwaccel->decode_slice           ||
+       s->avctx->codec->capabilities&CODEC_CAP_HWACCEL_VDPAU          ||
+       !s->cur_pic                                                    ||
+       s->cur_pic->field_picture
+    )
+        return 0;
+    return 1;
+}
+
 /**
  * Add a slice.
  * @param endx   x component of the last macroblock, can be -1
@@ -778,7 +791,7 @@ void ff_er_add_slice(ERContext *s, int startx, int starty,
     const int end_xy   = s->mb_index2xy[end_i];
     int mask           = -1;
 
-    if (s->avctx->hwaccel)
+    if (s->avctx->hwaccel && s->avctx->hwaccel->decode_slice)
         return;
 
     if (start_i > end_i || start_xy > end_xy) {
@@ -787,7 +800,7 @@ void ff_er_add_slice(ERContext *s, int startx, int starty,
         return;
     }
 
-    if (!s->avctx->err_recognition)
+    if (!s->avctx->error_concealment)
         return;
 
     mask &= ~VP_START;
@@ -828,7 +841,7 @@ void ff_er_add_slice(ERContext *s, int startx, int starty,
     s->error_status_table[start_xy] |= VP_START;
 
     if (start_xy > 0 && !(s->avctx->active_thread_type & FF_THREAD_SLICE) &&
-        s->avctx->skip_top * s->mb_width < start_i) {
+        er_supported(s) && s->avctx->skip_top * s->mb_width < start_i) {
         int prev_status = s->error_status_table[s->mb_index2xy[start_i - 1]];
 
         prev_status &= ~ VP_START;
@@ -851,11 +864,9 @@ void ff_er_frame_end(ERContext *s)
 
     /* We do not support ER of field pictures yet,
      * though it should not crash if enabled. */
-    if (!s->avctx->err_recognition || s->error_count == 0              ||
+    if (!s->avctx->error_concealment || s->error_count == 0            ||
         s->avctx->lowres                                               ||
-        s->avctx->hwaccel                                              ||
+        !er_supported(s)                                               ||
-        s->avctx->codec->capabilities&CODEC_CAP_HWACCEL_VDPAU          ||
-        !s->cur_pic || s->cur_pic->field_picture                               ||
         s->error_count == 3 * s->mb_width *
                           (s->avctx->skip_top + s->avctx->skip_bottom)) {
         return;
@@ -1173,8 +1184,8 @@ void ff_er_frame_end(ERContext *s)
     } else
         guess_mv(s);
 
-    /* the filters below are not XvMC compatible, skip them */
+    /* the filters below manipulate raw image, skip them */
-    if (CONFIG_MPEG_XVMC_DECODER && s->avctx->xvmc_acceleration)
+    if (CONFIG_XVMC && s->avctx->hwaccel && s->avctx->hwaccel->decode_mb)
         goto ec_clean;
     /* fill DC for inter blocks */
     for (mb_y = 0; mb_y < s->mb_height; mb_y++) {

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/frame_thread_encoder.c

@@ -126,6 +126,31 @@ int ff_frame_thread_encoder_init(AVCodecContext *avctx, AVDictionary *options){
        || !(avctx->codec->capabilities & CODEC_CAP_INTRA_ONLY))
         return 0;
 
+    if(   !avctx->thread_count
+       && avctx->codec_id == AV_CODEC_ID_MJPEG
+       && !(avctx->flags & CODEC_FLAG_QSCALE)) {
+        av_log(avctx, AV_LOG_DEBUG,
+               "Forcing thread count to 1 for MJPEG encoding, use -thread_type slice "
+               "or a constant quantizer if you want to use multiple cpu cores\n");
+        avctx->thread_count = 1;
+    }
+    if(   avctx->thread_count > 1
+       && avctx->codec_id == AV_CODEC_ID_MJPEG
+       && !(avctx->flags & CODEC_FLAG_QSCALE))
+        av_log(avctx, AV_LOG_WARNING,
+               "MJPEG CBR encoding works badly with frame multi-threading, consider "
+               "using -threads 1, -thread_type slice or a constant quantizer.\n");
+
+    if (avctx->codec_id == AV_CODEC_ID_HUFFYUV ||
+        avctx->codec_id == AV_CODEC_ID_FFVHUFF) {
+        // huffyuv doesnt support these with multiple frame threads currently
+        if (avctx->context_model > 0 || (avctx->flags & CODEC_FLAG_PASS1)) {
+            av_log(avctx, AV_LOG_WARNING,
+               "Forcing thread count to 1 for huffyuv encoding with first pass or context 1\n");
+            avctx->thread_count = 1;
+        }
+    }
+
     if(!avctx->thread_count) {
         avctx->thread_count = av_cpu_count();
         avctx->thread_count = FFMIN(avctx->thread_count, MAX_THREADS);

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/g726.c

@@ -97,6 +97,7 @@ typedef struct G726Context {
     int sez;            /**< estimated second order prediction */
     int y;              /**< quantizer scaling factor for the next iteration */
     int code_size;
+    int little_endian;  /**< little-endian bitstream as used in aiff and Sun AU */
 } G726Context;
 
 static const int quant_tbl16[] =                  /**< 16kbit/s 2bits per sample */
@@ -397,7 +398,7 @@ AVCodec ff_adpcm_g726_encoder = {
 };
 #endif
 
-#if CONFIG_ADPCM_G726_DECODER
+#if CONFIG_ADPCM_G726_DECODER || CONFIG_ADPCM_G726LE_DECODER
 static av_cold int g726_decode_init(AVCodecContext *avctx)
 {
     G726Context* c = avctx->priv_data;
@@ -409,6 +410,8 @@ static av_cold int g726_decode_init(AVCodecContext *avctx)
     avctx->channels       = 1;
     avctx->channel_layout = AV_CH_LAYOUT_MONO;
 
+    c->little_endian = !strcmp(avctx->codec->name, "g726le");
+
     c->code_size = avctx->bits_per_coded_sample;
     if (c->code_size < 2 || c->code_size > 5) {
         av_log(avctx, AV_LOG_ERROR, "Invalid number of bits %d\n", c->code_size);
@@ -443,7 +446,9 @@ static int g726_decode_frame(AVCodecContext *avctx, void *data,
     init_get_bits(&gb, buf, buf_size * 8);
 
     while (out_samples--)
-        *samples++ = g726_decode(c, get_bits(&gb, c->code_size));
+        *samples++ = g726_decode(c, c->little_endian ?
+                                    get_bits_le(&gb, c->code_size) :
+                                    get_bits(&gb, c->code_size));
 
     if (get_bits_left(&gb) > 0)
         av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
@@ -458,7 +463,9 @@ static void g726_decode_flush(AVCodecContext *avctx)
     G726Context *c = avctx->priv_data;
     g726_reset(c);
 }
+#endif
 
+#if CONFIG_ADPCM_G726_DECODER
 AVCodec ff_adpcm_g726_decoder = {
     .name           = "g726",
     .long_name      = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
@@ -471,3 +478,17 @@ AVCodec ff_adpcm_g726_decoder = {
     .capabilities   = CODEC_CAP_DR1,
 };
 #endif
+
+#if CONFIG_ADPCM_G726LE_DECODER
+AVCodec ff_adpcm_g726le_decoder = {
+    .name           = "g726le",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_ADPCM_G726LE,
+    .priv_data_size = sizeof(G726Context),
+    .init           = g726_decode_init,
+    .decode         = g726_decode_frame,
+    .flush          = g726_decode_flush,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("G.726 ADPCM little-endian"),
+};
+#endif

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/gsm.h

@@ -22,10 +22,24 @@
 #define AVCODEC_GSM_H
 
 /* bytes per block */
-#define GSM_BLOCK_SIZE    33
+#define GSM_BLOCK_SIZE     33
-#define GSM_MS_BLOCK_SIZE 65
+#define GSM_MS_BLOCK_SIZE  65
+#define MSN_MIN_BLOCK_SIZE 41
 
 /* samples per block */
 #define GSM_FRAME_SIZE 160
 
+enum GSMModes {
+    GSM_13000 = 0,
+    MSN_12400,
+    MSN_11800,
+    MSN_11200,
+    MSN_10600,
+    MSN_10000,
+    MSN_9400,
+    MSN_8800,
+    MSN_8200,
+    NUM_GSM_MODES
+};
+
 #endif /* AVCODEC_GSM_H */

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/gsm_parser.c

@@ -50,7 +50,8 @@ static int gsm_parse(AVCodecParserContext *s1, AVCodecContext *avctx,
             s->duration   = GSM_FRAME_SIZE;
             break;
         case AV_CODEC_ID_GSM_MS:
-            s->block_size = GSM_MS_BLOCK_SIZE;
+            s->block_size = avctx->block_align ? avctx->block_align
+                                               : GSM_MS_BLOCK_SIZE;
             s->duration   = GSM_FRAME_SIZE * 2;
             break;
         default:

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/gsmdec.c

@@ -34,14 +34,6 @@
 
 static av_cold int gsm_init(AVCodecContext *avctx)
 {
-    if (avctx->codec_tag == 0x0032 &&
-        avctx->bit_rate != 13000 &&
-        avctx->bit_rate != 17912 &&
-        avctx->bit_rate != 35824 &&
-        avctx->bit_rate != 71656) {
-        av_log(avctx, AV_LOG_ERROR, "Unsupported audio mode\n");
-        return AVERROR_PATCHWELCOME;
-    }
     avctx->channels       = 1;
     avctx->channel_layout = AV_CH_LAYOUT_MONO;
     if (!avctx->sample_rate)
@@ -55,8 +47,16 @@ static av_cold int gsm_init(AVCodecContext *avctx)
         break;
     case AV_CODEC_ID_GSM_MS:
         avctx->frame_size  = 2 * GSM_FRAME_SIZE;
-        avctx->block_align = GSM_MS_BLOCK_SIZE;
+        if (!avctx->block_align)
-    default: break;
+            avctx->block_align = GSM_MS_BLOCK_SIZE;
+        else
+            if (avctx->block_align < MSN_MIN_BLOCK_SIZE ||
+                avctx->block_align > GSM_MS_BLOCK_SIZE  ||
+                (avctx->block_align - MSN_MIN_BLOCK_SIZE) % 3) {
+                av_log(avctx, AV_LOG_ERROR, "Invalid block alignment %d\n",
+                       avctx->block_align);
+                return AVERROR_INVALIDDATA;
+            }
     }
 
     return 0;
@@ -88,12 +88,13 @@ static int gsm_decode_frame(AVCodecContext *avctx, void *data,
         init_get_bits(&gb, buf, buf_size * 8);
         if (get_bits(&gb, 4) != 0xd)
             av_log(avctx, AV_LOG_WARNING, "Missing GSM magic!\n");
-        res = gsm_decode_block(avctx, samples, &gb);
+        res = gsm_decode_block(avctx, samples, &gb, GSM_13000);
         if (res < 0)
             return res;
         break;
     case AV_CODEC_ID_GSM_MS:
-        res = ff_msgsm_decode_block(avctx, samples, buf);
+        res = ff_msgsm_decode_block(avctx, samples, buf,
+                                    (GSM_MS_BLOCK_SIZE - avctx->block_align) / 3);
         if (res < 0)
             return res;
     default: break;

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/gsmdec_data.c

@@ -92,3 +92,29 @@ const int16_t ff_gsm_dequant_tab[64][8] = {
     {-26879, -19199, -11520,  -3840,   3840,  11520,  19199,  26879},
     {-28671, -20479, -12288,  -4096,   4096,  12288,  20479,  28671}
 };
+
+static const int apcm_bits[11][13] = {
+    { 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1 },
+    { 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1 },
+    { 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1 },
+    { 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1 },
+    { 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1 },
+    { 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2 },
+    { 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2 },
+    { 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2 },
+    { 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2 },
+    { 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2 },
+    { 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3 }
+};
+
+const int* const ff_gsm_apcm_bits[][4] = {
+    { apcm_bits[10], apcm_bits[10], apcm_bits[10], apcm_bits[10] }, // 13000
+    { apcm_bits[10], apcm_bits[10], apcm_bits[10], apcm_bits[ 6] }, // 12400
+    { apcm_bits[10], apcm_bits[10], apcm_bits[ 7], apcm_bits[ 5] }, // 11800
+    { apcm_bits[10], apcm_bits[ 8], apcm_bits[ 5], apcm_bits[ 5] }, // 11200
+    { apcm_bits[ 9], apcm_bits[ 5], apcm_bits[ 5], apcm_bits[ 5] }, // 10600
+    { apcm_bits[ 5], apcm_bits[ 5], apcm_bits[ 5], apcm_bits[ 1] }, // 10000
+    { apcm_bits[ 5], apcm_bits[ 5], apcm_bits[ 2], apcm_bits[ 0] }, //  9400
+    { apcm_bits[ 5], apcm_bits[ 3], apcm_bits[ 0], apcm_bits[ 0] }, //  8800
+    { apcm_bits[ 4], apcm_bits[ 0], apcm_bits[ 0], apcm_bits[ 0] }, //  8200
+};

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/gsmdec_data.h

@@ -40,4 +40,6 @@ typedef struct GSMContext {
 extern const uint16_t ff_gsm_long_term_gain_tab[4];
 extern const int16_t ff_gsm_dequant_tab[64][8];
 
+extern const int* const ff_gsm_apcm_bits[][4];
+
 #endif /* AVCODEC_GSMDEC_DATA_H */

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/gsmdec_template.c

@@ -28,13 +28,22 @@
 #include "gsm.h"
 #include "gsmdec_data.h"
 
-static void apcm_dequant_add(GetBitContext *gb, int16_t *dst)
+static const int requant_tab[4][8] = {
+    { 0 },
+    { 0, 7 },
+    { 0, 2, 5, 7 },
+    { 0, 1, 2, 3, 4, 5, 6, 7 }
+};
+
+static void apcm_dequant_add(GetBitContext *gb, int16_t *dst, const int *frame_bits)
 {
-    int i;
+    int i, val;
     int maxidx = get_bits(gb, 6);
     const int16_t *tab = ff_gsm_dequant_tab[maxidx];
-    for (i = 0; i < 13; i++)
+    for (i = 0; i < 13; i++) {
-        dst[3*i] += tab[get_bits(gb, 3)];
+        val = get_bits(gb, frame_bits[i]);
+        dst[3*i] += tab[requant_tab[frame_bits[i]][val]];
+    }
 }
 
 static inline int gsm_mult(int a, int b)
@@ -118,7 +127,7 @@ static int postprocess(int16_t *data, int msr)
 }
 
 static int gsm_decode_block(AVCodecContext *avctx, int16_t *samples,
-                            GetBitContext *gb)
+                            GetBitContext *gb, int mode)
 {
     GSMContext *ctx = avctx->priv_data;
     int i;
@@ -139,7 +148,7 @@ static int gsm_decode_block(AVCodecContext *avctx, int16_t *samples,
         int offset   = get_bits(gb, 2);
         lag = av_clip(lag, 40, 120);
         long_term_synth(ref_dst, lag, gain_idx);
-        apcm_dequant_add(gb, ref_dst + offset);
+        apcm_dequant_add(gb, ref_dst + offset, ff_gsm_apcm_bits[mode][i]);
         ref_dst += 40;
     }
     memcpy(ctx->ref_buf, ctx->ref_buf + 160, 120 * sizeof(*ctx->ref_buf));

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/hpeldsp.c

@@ -56,6 +56,8 @@ av_cold void ff_hpeldsp_init(HpelDSPContext *c, int flags)
     hpel_funcs(avg, [3],  2);
     hpel_funcs(avg_no_rnd,, 16);
 
+    if (ARCH_AARCH64)
+        ff_hpeldsp_init_aarch64(c, flags);
     if (ARCH_ALPHA)
         ff_hpeldsp_init_alpha(c, flags);
     if (ARCH_ARM)

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/internal.h

@@ -27,6 +27,7 @@
 #include <stdint.h>
 
 #include "libavutil/buffer.h"
+#include "libavutil/channel_layout.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/pixfmt.h"
 #include "avcodec.h"
@@ -91,10 +92,18 @@ typedef struct AVCodecInternal {
      */
     int last_audio_frame;
 
-    AVFrame to_free;
+    AVFrame *to_free;
 
     FramePool *pool;
 
+    void *thread_ctx;
+
+    /**
+     * Current packet as passed into the decoder, to avoid having to pass the
+     * packet into every function.
+     */
+    AVPacket *pkt;
+
     /**
      * temporary buffer used for encoders to store their bitstream
      */
@@ -118,11 +127,10 @@ struct AVCodecDefault {
  * Return the hardware accelerated codec for codec codec_id and
  * pixel format pix_fmt.
  *
- * @param codec_id the codec to match
+ * @param avctx The codec context containing the codec_id and pixel format.
- * @param pix_fmt the pixel format to match
  * @return the hardware accelerated codec, or NULL if none was found.
  */
-AVHWAccel *ff_find_hwaccel(enum AVCodecID codec_id, enum AVPixelFormat pix_fmt);
+AVHWAccel *ff_find_hwaccel(AVCodecContext *avctx);
 
 /**
  * Return the index into tab at which {a,b} match elements {[0],[1]} of tab.
@@ -225,4 +233,16 @@ const uint8_t *avpriv_find_start_code(const uint8_t *p,
                                       const uint8_t *end,
                                       uint32_t *state);
 
+/**
+ * Check that the provided frame dimensions are valid and set them on the codec
+ * context.
+ */
+int ff_set_dimensions(AVCodecContext *s, int width, int height);
+
+/**
+ * Add or update AV_FRAME_DATA_MATRIXENCODING side data.
+ */
+int ff_side_data_update_matrix_encoding(AVFrame *frame,
+                                        enum AVMatrixEncoding matrix_encoding);
+
 #endif /* AVCODEC_INTERNAL_H */

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/lpc.c

@@ -20,7 +20,7 @@
  */
 
 #include "libavutil/common.h"
-#include "libavutil/lls.h"
+#include "libavutil/lls2.h"
 #include "libavutil/internal.h"
 
 #define LPC_USE_DOUBLE
@@ -209,7 +209,7 @@ int ff_lpc_calc_coefs(LPCContext *s,
     }
 
     if (lpc_type == FF_LPC_TYPE_CHOLESKY) {
-        LLSModel m[2];
+        LLSModel2 m[2];
         LOCAL_ALIGNED(32, double, var, [FFALIGN(MAX_LPC_ORDER+1,4)]);
         double av_uninit(weight);
         memset(var, 0, FFALIGN(MAX_LPC_ORDER+1,4)*sizeof(*var));
@@ -218,7 +218,7 @@ int ff_lpc_calc_coefs(LPCContext *s,
             m[0].coeff[max_order-1][j] = -lpc[max_order-1][j];
 
         for(; pass<lpc_passes; pass++){
-            avpriv_init_lls(&m[pass&1], max_order);
+            avpriv_init_lls2(&m[pass&1], max_order);
 
             weight=0;
             for(i=max_order; i<blocksize; i++){
@@ -239,7 +239,7 @@ int ff_lpc_calc_coefs(LPCContext *s,
 
                 m[pass&1].update_lls(&m[pass&1], var);
             }
-            avpriv_solve_lls(&m[pass&1], 0.001, 0);
+            avpriv_solve_lls2(&m[pass&1], 0.001, 0);
         }
 
         for(i=0; i<max_order; i++){

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/mpegaudioenc.c

@@ -1,788 +0,0 @@
-/*
- * The simplest mpeg audio layer 2 encoder
- * Copyright (c) 2000, 2001 Fabrice Bellard
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * The simplest mpeg audio layer 2 encoder.
- */
-
-#include "libavutil/channel_layout.h"
-
-#include "avcodec.h"
-#include "internal.h"
-#include "put_bits.h"
-
-#define FRAC_BITS   15   /* fractional bits for sb_samples and dct */
-#define WFRAC_BITS  14   /* fractional bits for window */
-
-#include "mpegaudio.h"
-#include "mpegaudiodsp.h"
-
-/* currently, cannot change these constants (need to modify
-   quantization stage) */
-#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
-
-#define SAMPLES_BUF_SIZE 4096
-
-typedef struct MpegAudioContext {
-    PutBitContext pb;
-    int nb_channels;
-    int lsf;           /* 1 if mpeg2 low bitrate selected */
-    int bitrate_index; /* bit rate */
-    int freq_index;
-    int frame_size; /* frame size, in bits, without padding */
-    /* padding computation */
-    int frame_frac, frame_frac_incr, do_padding;
-    short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
-    int samples_offset[MPA_MAX_CHANNELS];       /* offset in samples_buf */
-    int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
-    unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
-    /* code to group 3 scale factors */
-    unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
-    int sblimit; /* number of used subbands */
-    const unsigned char *alloc_table;
-} MpegAudioContext;
-
-/* define it to use floats in quantization (I don't like floats !) */
-#define USE_FLOATS
-
-#include "mpegaudiodata.h"
-#include "mpegaudiotab.h"
-
-static av_cold int MPA_encode_init(AVCodecContext *avctx)
-{
-    MpegAudioContext *s = avctx->priv_data;
-    int freq = avctx->sample_rate;
-    int bitrate = avctx->bit_rate;
-    int channels = avctx->channels;
-    int i, v, table;
-    float a;
-
-    if (channels <= 0 || channels > 2){
-        av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
-        return AVERROR(EINVAL);
-    }
-    bitrate = bitrate / 1000;
-    s->nb_channels = channels;
-    avctx->frame_size = MPA_FRAME_SIZE;
-    avctx->delay      = 512 - 32 + 1;
-
-    /* encoding freq */
-    s->lsf = 0;
-    for(i=0;i<3;i++) {
-        if (avpriv_mpa_freq_tab[i] == freq)
-            break;
-        if ((avpriv_mpa_freq_tab[i] / 2) == freq) {
-            s->lsf = 1;
-            break;
-        }
-    }
-    if (i == 3){
-        av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
-        return AVERROR(EINVAL);
-    }
-    s->freq_index = i;
-
-    /* encoding bitrate & frequency */
-    for(i=0;i<15;i++) {
-        if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
-            break;
-    }
-    if (i == 15){
-        av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
-        return AVERROR(EINVAL);
-    }
-    s->bitrate_index = i;
-
-    /* compute total header size & pad bit */
-
-    a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
-    s->frame_size = ((int)a) * 8;
-
-    /* frame fractional size to compute padding */
-    s->frame_frac = 0;
-    s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
-
-    /* select the right allocation table */
-    table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
-
-    /* number of used subbands */
-    s->sblimit = ff_mpa_sblimit_table[table];
-    s->alloc_table = ff_mpa_alloc_tables[table];
-
-    av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
-            bitrate, freq, s->frame_size, table, s->frame_frac_incr);
-
-    for(i=0;i<s->nb_channels;i++)
-        s->samples_offset[i] = 0;
-
-    for(i=0;i<257;i++) {
-        int v;
-        v = ff_mpa_enwindow[i];
-#if WFRAC_BITS != 16
-        v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
-#endif
-        filter_bank[i] = v;
-        if ((i & 63) != 0)
-            v = -v;
-        if (i != 0)
-            filter_bank[512 - i] = v;
-    }
-
-    for(i=0;i<64;i++) {
-        v = (int)(exp2((3 - i) / 3.0) * (1 << 20));
-        if (v <= 0)
-            v = 1;
-        scale_factor_table[i] = v;
-#ifdef USE_FLOATS
-        scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
-#else
-#define P 15
-        scale_factor_shift[i] = 21 - P - (i / 3);
-        scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0);
-#endif
-    }
-    for(i=0;i<128;i++) {
-        v = i - 64;
-        if (v <= -3)
-            v = 0;
-        else if (v < 0)
-            v = 1;
-        else if (v == 0)
-            v = 2;
-        else if (v < 3)
-            v = 3;
-        else
-            v = 4;
-        scale_diff_table[i] = v;
-    }
-
-    for(i=0;i<17;i++) {
-        v = ff_mpa_quant_bits[i];
-        if (v < 0)
-            v = -v;
-        else
-            v = v * 3;
-        total_quant_bits[i] = 12 * v;
-    }
-
-    return 0;
-}
-
-/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
-static void idct32(int *out, int *tab)
-{
-    int i, j;
-    int *t, *t1, xr;
-    const int *xp = costab32;
-
-    for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
-
-    t = tab + 30;
-    t1 = tab + 2;
-    do {
-        t[0] += t[-4];
-        t[1] += t[1 - 4];
-        t -= 4;
-    } while (t != t1);
-
-    t = tab + 28;
-    t1 = tab + 4;
-    do {
-        t[0] += t[-8];
-        t[1] += t[1-8];
-        t[2] += t[2-8];
-        t[3] += t[3-8];
-        t -= 8;
-    } while (t != t1);
-
-    t = tab;
-    t1 = tab + 32;
-    do {
-        t[ 3] = -t[ 3];
-        t[ 6] = -t[ 6];
-
-        t[11] = -t[11];
-        t[12] = -t[12];
-        t[13] = -t[13];
-        t[15] = -t[15];
-        t += 16;
-    } while (t != t1);
-
-
-    t = tab;
-    t1 = tab + 8;
-    do {
-        int x1, x2, x3, x4;
-
-        x3 = MUL(t[16], FIX(SQRT2*0.5));
-        x4 = t[0] - x3;
-        x3 = t[0] + x3;
-
-        x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
-        x1 = MUL((t[8] - x2), xp[0]);
-        x2 = MUL((t[8] + x2), xp[1]);
-
-        t[ 0] = x3 + x1;
-        t[ 8] = x4 - x2;
-        t[16] = x4 + x2;
-        t[24] = x3 - x1;
-        t++;
-    } while (t != t1);
-
-    xp += 2;
-    t = tab;
-    t1 = tab + 4;
-    do {
-        xr = MUL(t[28],xp[0]);
-        t[28] = (t[0] - xr);
-        t[0] = (t[0] + xr);
-
-        xr = MUL(t[4],xp[1]);
-        t[ 4] = (t[24] - xr);
-        t[24] = (t[24] + xr);
-
-        xr = MUL(t[20],xp[2]);
-        t[20] = (t[8] - xr);
-        t[ 8] = (t[8] + xr);
-
-        xr = MUL(t[12],xp[3]);
-        t[12] = (t[16] - xr);
-        t[16] = (t[16] + xr);
-        t++;
-    } while (t != t1);
-    xp += 4;
-
-    for (i = 0; i < 4; i++) {
-        xr = MUL(tab[30-i*4],xp[0]);
-        tab[30-i*4] = (tab[i*4] - xr);
-        tab[   i*4] = (tab[i*4] + xr);
-
-        xr = MUL(tab[ 2+i*4],xp[1]);
-        tab[ 2+i*4] = (tab[28-i*4] - xr);
-        tab[28-i*4] = (tab[28-i*4] + xr);
-
-        xr = MUL(tab[31-i*4],xp[0]);
-        tab[31-i*4] = (tab[1+i*4] - xr);
-        tab[ 1+i*4] = (tab[1+i*4] + xr);
-
-        xr = MUL(tab[ 3+i*4],xp[1]);
-        tab[ 3+i*4] = (tab[29-i*4] - xr);
-        tab[29-i*4] = (tab[29-i*4] + xr);
-
-        xp += 2;
-    }
-
-    t = tab + 30;
-    t1 = tab + 1;
-    do {
-        xr = MUL(t1[0], *xp);
-        t1[0] = (t[0] - xr);
-        t[0] = (t[0] + xr);
-        t -= 2;
-        t1 += 2;
-        xp++;
-    } while (t >= tab);
-
-    for(i=0;i<32;i++) {
-        out[i] = tab[bitinv32[i]];
-    }
-}
-
-#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
-
-static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
-{
-    short *p, *q;
-    int sum, offset, i, j;
-    int tmp[64];
-    int tmp1[32];
-    int *out;
-
-    offset = s->samples_offset[ch];
-    out = &s->sb_samples[ch][0][0][0];
-    for(j=0;j<36;j++) {
-        /* 32 samples at once */
-        for(i=0;i<32;i++) {
-            s->samples_buf[ch][offset + (31 - i)] = samples[0];
-            samples += incr;
-        }
-
-        /* filter */
-        p = s->samples_buf[ch] + offset;
-        q = filter_bank;
-        /* maxsum = 23169 */
-        for(i=0;i<64;i++) {
-            sum = p[0*64] * q[0*64];
-            sum += p[1*64] * q[1*64];
-            sum += p[2*64] * q[2*64];
-            sum += p[3*64] * q[3*64];
-            sum += p[4*64] * q[4*64];
-            sum += p[5*64] * q[5*64];
-            sum += p[6*64] * q[6*64];
-            sum += p[7*64] * q[7*64];
-            tmp[i] = sum;
-            p++;
-            q++;
-        }
-        tmp1[0] = tmp[16] >> WSHIFT;
-        for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
-        for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
-
-        idct32(out, tmp1);
-
-        /* advance of 32 samples */
-        offset -= 32;
-        out += 32;
-        /* handle the wrap around */
-        if (offset < 0) {
-            memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
-                    s->samples_buf[ch], (512 - 32) * 2);
-            offset = SAMPLES_BUF_SIZE - 512;
-        }
-    }
-    s->samples_offset[ch] = offset;
-}
-
-static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
-                                  unsigned char scale_factors[SBLIMIT][3],
-                                  int sb_samples[3][12][SBLIMIT],
-                                  int sblimit)
-{
-    int *p, vmax, v, n, i, j, k, code;
-    int index, d1, d2;
-    unsigned char *sf = &scale_factors[0][0];
-
-    for(j=0;j<sblimit;j++) {
-        for(i=0;i<3;i++) {
-            /* find the max absolute value */
-            p = &sb_samples[i][0][j];
-            vmax = abs(*p);
-            for(k=1;k<12;k++) {
-                p += SBLIMIT;
-                v = abs(*p);
-                if (v > vmax)
-                    vmax = v;
-            }
-            /* compute the scale factor index using log 2 computations */
-            if (vmax > 1) {
-                n = av_log2(vmax);
-                /* n is the position of the MSB of vmax. now
-                   use at most 2 compares to find the index */
-                index = (21 - n) * 3 - 3;
-                if (index >= 0) {
-                    while (vmax <= scale_factor_table[index+1])
-                        index++;
-                } else {
-                    index = 0; /* very unlikely case of overflow */
-                }
-            } else {
-                index = 62; /* value 63 is not allowed */
-            }
-
-            av_dlog(NULL, "%2d:%d in=%x %x %d\n",
-                    j, i, vmax, scale_factor_table[index], index);
-            /* store the scale factor */
-            av_assert2(index >=0 && index <= 63);
-            sf[i] = index;
-        }
-
-        /* compute the transmission factor : look if the scale factors
-           are close enough to each other */
-        d1 = scale_diff_table[sf[0] - sf[1] + 64];
-        d2 = scale_diff_table[sf[1] - sf[2] + 64];
-
-        /* handle the 25 cases */
-        switch(d1 * 5 + d2) {
-        case 0*5+0:
-        case 0*5+4:
-        case 3*5+4:
-        case 4*5+0:
-        case 4*5+4:
-            code = 0;
-            break;
-        case 0*5+1:
-        case 0*5+2:
-        case 4*5+1:
-        case 4*5+2:
-            code = 3;
-            sf[2] = sf[1];
-            break;
-        case 0*5+3:
-        case 4*5+3:
-            code = 3;
-            sf[1] = sf[2];
-            break;
-        case 1*5+0:
-        case 1*5+4:
-        case 2*5+4:
-            code = 1;
-            sf[1] = sf[0];
-            break;
-        case 1*5+1:
-        case 1*5+2:
-        case 2*5+0:
-        case 2*5+1:
-        case 2*5+2:
-            code = 2;
-            sf[1] = sf[2] = sf[0];
-            break;
-        case 2*5+3:
-        case 3*5+3:
-            code = 2;
-            sf[0] = sf[1] = sf[2];
-            break;
-        case 3*5+0:
-        case 3*5+1:
-        case 3*5+2:
-            code = 2;
-            sf[0] = sf[2] = sf[1];
-            break;
-        case 1*5+3:
-            code = 2;
-            if (sf[0] > sf[2])
-              sf[0] = sf[2];
-            sf[1] = sf[2] = sf[0];
-            break;
-        default:
-            av_assert2(0); //cannot happen
-            code = 0;           /* kill warning */
-        }
-
-        av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j,
-                sf[0], sf[1], sf[2], d1, d2, code);
-        scale_code[j] = code;
-        sf += 3;
-    }
-}
-
-/* The most important function : psycho acoustic module. In this
-   encoder there is basically none, so this is the worst you can do,
-   but also this is the simpler. */
-static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
-{
-    int i;
-
-    for(i=0;i<s->sblimit;i++) {
-        smr[i] = (int)(fixed_smr[i] * 10);
-    }
-}
-
-
-#define SB_NOTALLOCATED  0
-#define SB_ALLOCATED     1
-#define SB_NOMORE        2
-
-/* Try to maximize the smr while using a number of bits inferior to
-   the frame size. I tried to make the code simpler, faster and
-   smaller than other encoders :-) */
-static void compute_bit_allocation(MpegAudioContext *s,
-                                   short smr1[MPA_MAX_CHANNELS][SBLIMIT],
-                                   unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
-                                   int *padding)
-{
-    int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
-    int incr;
-    short smr[MPA_MAX_CHANNELS][SBLIMIT];
-    unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
-    const unsigned char *alloc;
-
-    memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
-    memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
-    memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
-
-    /* compute frame size and padding */
-    max_frame_size = s->frame_size;
-    s->frame_frac += s->frame_frac_incr;
-    if (s->frame_frac >= 65536) {
-        s->frame_frac -= 65536;
-        s->do_padding = 1;
-        max_frame_size += 8;
-    } else {
-        s->do_padding = 0;
-    }
-
-    /* compute the header + bit alloc size */
-    current_frame_size = 32;
-    alloc = s->alloc_table;
-    for(i=0;i<s->sblimit;i++) {
-        incr = alloc[0];
-        current_frame_size += incr * s->nb_channels;
-        alloc += 1 << incr;
-    }
-    for(;;) {
-        /* look for the subband with the largest signal to mask ratio */
-        max_sb = -1;
-        max_ch = -1;
-        max_smr = INT_MIN;
-        for(ch=0;ch<s->nb_channels;ch++) {
-            for(i=0;i<s->sblimit;i++) {
-                if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
-                    max_smr = smr[ch][i];
-                    max_sb = i;
-                    max_ch = ch;
-                }
-            }
-        }
-        if (max_sb < 0)
-            break;
-        av_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n",
-                current_frame_size, max_frame_size, max_sb, max_ch,
-                bit_alloc[max_ch][max_sb]);
-
-        /* find alloc table entry (XXX: not optimal, should use
-           pointer table) */
-        alloc = s->alloc_table;
-        for(i=0;i<max_sb;i++) {
-            alloc += 1 << alloc[0];
-        }
-
-        if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
-            /* nothing was coded for this band: add the necessary bits */
-            incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
-            incr += total_quant_bits[alloc[1]];
-        } else {
-            /* increments bit allocation */
-            b = bit_alloc[max_ch][max_sb];
-            incr = total_quant_bits[alloc[b + 1]] -
-                total_quant_bits[alloc[b]];
-        }
-
-        if (current_frame_size + incr <= max_frame_size) {
-            /* can increase size */
-            b = ++bit_alloc[max_ch][max_sb];
-            current_frame_size += incr;
-            /* decrease smr by the resolution we added */
-            smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
-            /* max allocation size reached ? */
-            if (b == ((1 << alloc[0]) - 1))
-                subband_status[max_ch][max_sb] = SB_NOMORE;
-            else
-                subband_status[max_ch][max_sb] = SB_ALLOCATED;
-        } else {
-            /* cannot increase the size of this subband */
-            subband_status[max_ch][max_sb] = SB_NOMORE;
-        }
-    }
-    *padding = max_frame_size - current_frame_size;
-    av_assert0(*padding >= 0);
-}
-
-/*
- * Output the mpeg audio layer 2 frame. Note how the code is small
- * compared to other encoders :-)
- */
-static void encode_frame(MpegAudioContext *s,
-                         unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
-                         int padding)
-{
-    int i, j, k, l, bit_alloc_bits, b, ch;
-    unsigned char *sf;
-    int q[3];
-    PutBitContext *p = &s->pb;
-
-    /* header */
-
-    put_bits(p, 12, 0xfff);
-    put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
-    put_bits(p, 2, 4-2);  /* layer 2 */
-    put_bits(p, 1, 1); /* no error protection */
-    put_bits(p, 4, s->bitrate_index);
-    put_bits(p, 2, s->freq_index);
-    put_bits(p, 1, s->do_padding); /* use padding */
-    put_bits(p, 1, 0);             /* private_bit */
-    put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
-    put_bits(p, 2, 0); /* mode_ext */
-    put_bits(p, 1, 0); /* no copyright */
-    put_bits(p, 1, 1); /* original */
-    put_bits(p, 2, 0); /* no emphasis */
-
-    /* bit allocation */
-    j = 0;
-    for(i=0;i<s->sblimit;i++) {
-        bit_alloc_bits = s->alloc_table[j];
-        for(ch=0;ch<s->nb_channels;ch++) {
-            put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
-        }
-        j += 1 << bit_alloc_bits;
-    }
-
-    /* scale codes */
-    for(i=0;i<s->sblimit;i++) {
-        for(ch=0;ch<s->nb_channels;ch++) {
-            if (bit_alloc[ch][i])
-                put_bits(p, 2, s->scale_code[ch][i]);
-        }
-    }
-
-    /* scale factors */
-    for(i=0;i<s->sblimit;i++) {
-        for(ch=0;ch<s->nb_channels;ch++) {
-            if (bit_alloc[ch][i]) {
-                sf = &s->scale_factors[ch][i][0];
-                switch(s->scale_code[ch][i]) {
-                case 0:
-                    put_bits(p, 6, sf[0]);
-                    put_bits(p, 6, sf[1]);
-                    put_bits(p, 6, sf[2]);
-                    break;
-                case 3:
-                case 1:
-                    put_bits(p, 6, sf[0]);
-                    put_bits(p, 6, sf[2]);
-                    break;
-                case 2:
-                    put_bits(p, 6, sf[0]);
-                    break;
-                }
-            }
-        }
-    }
-
-    /* quantization & write sub band samples */
-
-    for(k=0;k<3;k++) {
-        for(l=0;l<12;l+=3) {
-            j = 0;
-            for(i=0;i<s->sblimit;i++) {
-                bit_alloc_bits = s->alloc_table[j];
-                for(ch=0;ch<s->nb_channels;ch++) {
-                    b = bit_alloc[ch][i];
-                    if (b) {
-                        int qindex, steps, m, sample, bits;
-                        /* we encode 3 sub band samples of the same sub band at a time */
-                        qindex = s->alloc_table[j+b];
-                        steps = ff_mpa_quant_steps[qindex];
-                        for(m=0;m<3;m++) {
-                            sample = s->sb_samples[ch][k][l + m][i];
-                            /* divide by scale factor */
-#ifdef USE_FLOATS
-                            {
-                                float a;
-                                a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
-                                q[m] = (int)((a + 1.0) * steps * 0.5);
-                            }
-#else
-                            {
-                                int q1, e, shift, mult;
-                                e = s->scale_factors[ch][i][k];
-                                shift = scale_factor_shift[e];
-                                mult = scale_factor_mult[e];
-
-                                /* normalize to P bits */
-                                if (shift < 0)
-                                    q1 = sample << (-shift);
-                                else
-                                    q1 = sample >> shift;
-                                q1 = (q1 * mult) >> P;
-                                q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
-                            }
-#endif
-                            if (q[m] >= steps)
-                                q[m] = steps - 1;
-                            av_assert2(q[m] >= 0 && q[m] < steps);
-                        }
-                        bits = ff_mpa_quant_bits[qindex];
-                        if (bits < 0) {
-                            /* group the 3 values to save bits */
-                            put_bits(p, -bits,
-                                     q[0] + steps * (q[1] + steps * q[2]));
-                        } else {
-                            put_bits(p, bits, q[0]);
-                            put_bits(p, bits, q[1]);
-                            put_bits(p, bits, q[2]);
-                        }
-                    }
-                }
-                /* next subband in alloc table */
-                j += 1 << bit_alloc_bits;
-            }
-        }
-    }
-
-    /* padding */
-    for(i=0;i<padding;i++)
-        put_bits(p, 1, 0);
-
-    /* flush */
-    flush_put_bits(p);
-}
-
-static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
-                            const AVFrame *frame, int *got_packet_ptr)
-{
-    MpegAudioContext *s = avctx->priv_data;
-    const int16_t *samples = (const int16_t *)frame->data[0];
-    short smr[MPA_MAX_CHANNELS][SBLIMIT];
-    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
-    int padding, i, ret;
-
-    for(i=0;i<s->nb_channels;i++) {
-        filter(s, i, samples + i, s->nb_channels);
-    }
-
-    for(i=0;i<s->nb_channels;i++) {
-        compute_scale_factors(s->scale_code[i], s->scale_factors[i],
-                              s->sb_samples[i], s->sblimit);
-    }
-    for(i=0;i<s->nb_channels;i++) {
-        psycho_acoustic_model(s, smr[i]);
-    }
-    compute_bit_allocation(s, smr, bit_alloc, &padding);
-
-    if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0)
-        return ret;
-
-    init_put_bits(&s->pb, avpkt->data, avpkt->size);
-
-    encode_frame(s, bit_alloc, padding);
-
-    if (frame->pts != AV_NOPTS_VALUE)
-        avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
-
-    avpkt->size = put_bits_count(&s->pb) / 8;
-    *got_packet_ptr = 1;
-    return 0;
-}
-
-static const AVCodecDefault mp2_defaults[] = {
-    { (const uint8_t *) "b",    (const uint8_t *) "128k" },
-    { NULL },
-};
-
-AVCodec ff_mp2_encoder = {
-    .name                  = "mp2",
-    .long_name             = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
-    .type                  = AVMEDIA_TYPE_AUDIO,
-    .id                    = AV_CODEC_ID_MP2,
-    .priv_data_size        = sizeof(MpegAudioContext),
-    .init                  = MPA_encode_init,
-    .encode2               = MPA_encode_frame,
-    .sample_fmts           = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
-                                                            AV_SAMPLE_FMT_NONE },
-    .supported_samplerates = (const int[]){
-        44100, 48000,  32000, 22050, 24000, 16000, 0
-    },
-    .channel_layouts       = (const uint64_t[]){ AV_CH_LAYOUT_MONO,
-                                                 AV_CH_LAYOUT_STEREO,
-                                                 0 },
-    .defaults              = mp2_defaults,
-};

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/msgsmdec.c

@@ -26,13 +26,13 @@
 #include "gsmdec_template.c"
 
 int ff_msgsm_decode_block(AVCodecContext *avctx, int16_t *samples,
-                          const uint8_t *buf)
+                          const uint8_t *buf, int mode)
 {
     int res;
     GetBitContext gb;
     init_get_bits(&gb, buf, GSM_MS_BLOCK_SIZE * 8);
-    res = gsm_decode_block(avctx, samples, &gb);
+    res = gsm_decode_block(avctx, samples, &gb, mode);
     if (res < 0)
         return res;
-    return gsm_decode_block(avctx, samples + GSM_FRAME_SIZE, &gb);
+    return gsm_decode_block(avctx, samples + GSM_FRAME_SIZE, &gb, mode);
 }

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/msgsmdec.h

@@ -25,6 +25,6 @@
 #include "avcodec.h"
 
 int ff_msgsm_decode_block(AVCodecContext *avctx, int16_t *samples,
-                          const uint8_t *buf);
+                          const uint8_t *buf, int mode);
 
 #endif /* AVCODEC_MSGSMDEC_H */

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/old_codec_ids.h

@@ -38,7 +38,9 @@
     /* video codecs */
     CODEC_ID_MPEG1VIDEO,
     CODEC_ID_MPEG2VIDEO, ///< preferred ID for MPEG-1/2 video decoding
+#if FF_API_XVMC
     CODEC_ID_MPEG2VIDEO_XVMC,
+#endif
     CODEC_ID_H261,
     CODEC_ID_H263,
     CODEC_ID_RV10,

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/options.c

@@ -53,7 +53,7 @@ static void *codec_child_next(void *obj, void *prev)
 
 static const AVClass *codec_child_class_next(const AVClass *prev)
 {
-    const AVCodec *c = NULL;
+    AVCodec *c = NULL;
 
     /* find the codec that corresponds to prev */
     while (prev && (c = av_codec_next(c)))
@@ -61,7 +61,7 @@ static const AVClass *codec_child_class_next(const AVClass *prev)
             break;
 
     /* find next codec with priv options */
-    while ((c = av_codec_next(c)))
+    while (c = av_codec_next(c))
         if (c->priv_class)
             return c->priv_class;
     return NULL;
@@ -86,14 +86,6 @@ static const AVClass av_codec_context_class = {
     .get_category            = get_category,
 };
 
-#if FF_API_ALLOC_CONTEXT
-void avcodec_get_context_defaults2(AVCodecContext *s, enum AVMediaType codec_type){
-    AVCodec c= {0};
-    c.type= codec_type;
-    avcodec_get_context_defaults3(s, &c);
-}
-#endif
-
 int avcodec_get_context_defaults3(AVCodecContext *s, const AVCodec *codec)
 {
     int flags=0;
@@ -111,9 +103,7 @@ int avcodec_get_context_defaults3(AVCodecContext *s, const AVCodec *codec)
         flags= AV_OPT_FLAG_VIDEO_PARAM;
     else if(s->codec_type == AVMEDIA_TYPE_SUBTITLE)
         flags= AV_OPT_FLAG_SUBTITLE_PARAM;
-    AV_NOWARN_DEPRECATED(
     av_opt_set_defaults2(s, flags, flags);
-    );
 
     s->time_base           = (AVRational){0,1};
     s->get_buffer2         = avcodec_default_get_buffer2;
@@ -142,7 +132,7 @@ int avcodec_get_context_defaults3(AVCodecContext *s, const AVCodec *codec)
         int ret;
         const AVCodecDefault *d = codec->defaults;
         while (d->key) {
-            ret = av_opt_set(s, (const char *) d->key, (const char *) d->value, 0);
+            ret = av_opt_set(s, d->key, d->value, 0);
             av_assert0(ret >= 0);
             d++;
         }
@@ -164,26 +154,6 @@ AVCodecContext *avcodec_alloc_context3(const AVCodec *codec)
     return avctx;
 }
 
-#if FF_API_ALLOC_CONTEXT
-AVCodecContext *avcodec_alloc_context2(enum AVMediaType codec_type){
-    AVCodecContext *avctx= av_malloc(sizeof(AVCodecContext));
-
-    if(avctx==NULL) return NULL;
-
-    avcodec_get_context_defaults2(avctx, codec_type);
-
-    return avctx;
-}
-
-void avcodec_get_context_defaults(AVCodecContext *s){
-    avcodec_get_context_defaults2(s, AVMEDIA_TYPE_UNKNOWN);
-}
-
-AVCodecContext *avcodec_alloc_context(void){
-    return avcodec_alloc_context2(AVMEDIA_TYPE_UNKNOWN);
-}
-#endif
-
 int avcodec_copy_context(AVCodecContext *dest, const AVCodecContext *src)
 {
     if (avcodec_is_open(dest)) { // check that the dest context is uninitialized
@@ -203,7 +173,6 @@ int avcodec_copy_context(AVCodecContext *dest, const AVCodecContext *src)
     dest->codec           = NULL;
     dest->slice_offset    = NULL;
     dest->hwaccel         = NULL;
-    dest->thread_opaque   = NULL;
     dest->internal        = NULL;
 
     /* reallocate values that should be allocated separately */

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/rawdec.c

@@ -25,6 +25,7 @@
  */
 
 #include "avcodec.h"
+#include "internal.h"
 #include "raw.h"
 #include "libavutil/avassert.h"
 #include "libavutil/buffer.h"
@@ -40,6 +41,7 @@ typedef struct RawVideoContext {
     int flip;
     int is_2_4_bpp; // 2 or 4 bpp raw in avi/mov
     int is_yuv2;
+    int is_lt_16bpp; // 16bpp pixfmt and bits_per_coded_sample < 16
     int tff;
 } RawVideoContext;
 
@@ -107,7 +109,7 @@ static av_cold int raw_init_decoder(AVCodecContext *avctx)
     if (   avctx->codec_tag == MKTAG('r','a','w',' ')
         || avctx->codec_tag == MKTAG('N','O','1','6'))
         avctx->pix_fmt = avpriv_find_pix_fmt(pix_fmt_bps_mov,
-                                      avctx->bits_per_coded_sample);
+                                      avctx->bits_per_coded_sample & 0x1f);
     else if (avctx->codec_tag == MKTAG('W', 'R', 'A', 'W'))
         avctx->pix_fmt = avpriv_find_pix_fmt(pix_fmt_bps_avi,
                                       avctx->bits_per_coded_sample);
@@ -133,7 +135,7 @@ static av_cold int raw_init_decoder(AVCodecContext *avctx)
             memset(context->palette->data, 0, AVPALETTE_SIZE);
     }
 
-    if ((avctx->bits_per_coded_sample == 4 || avctx->bits_per_coded_sample == 2) &&
+    if (((avctx->bits_per_coded_sample & 0x1f) == 4 || (avctx->bits_per_coded_sample & 0x1f) == 2) &&
         avctx->pix_fmt == AV_PIX_FMT_PAL8 &&
        (!avctx->codec_tag || avctx->codec_tag == MKTAG('r','a','w',' '))) {
         context->is_2_4_bpp = 1;
@@ -141,6 +143,7 @@ static av_cold int raw_init_decoder(AVCodecContext *avctx)
                                                  FFALIGN(avctx->width, 16),
                                                  avctx->height);
     } else {
+        context->is_lt_16bpp = av_get_bits_per_pixel(desc) == 16 && avctx->bits_per_coded_sample && avctx->bits_per_coded_sample < 16;
         context->frame_size = avpicture_get_size(avctx->pix_fmt, avctx->width,
                                                  avctx->height);
     }
@@ -174,7 +177,7 @@ static int raw_decode(AVCodecContext *avctx, void *data, int *got_frame,
     int buf_size                   = avpkt->size;
     int linesize_align             = 4;
     int res, len;
-    int need_copy                  = !avpkt->buf || context->is_2_4_bpp || context->is_yuv2;
+    int need_copy                  = !avpkt->buf || context->is_2_4_bpp || context->is_yuv2 || context->is_lt_16bpp;
 
     AVFrame   *frame   = data;
     AVPicture *picture = data;
@@ -182,9 +185,9 @@ static int raw_decode(AVCodecContext *avctx, void *data, int *got_frame,
     frame->pict_type        = AV_PICTURE_TYPE_I;
     frame->key_frame        = 1;
     frame->reordered_opaque = avctx->reordered_opaque;
-    frame->pkt_pts          = avctx->pkt->pts;
+    frame->pkt_pts          = avctx->internal->pkt->pts;
-    av_frame_set_pkt_pos     (frame, avctx->pkt->pos);
+    av_frame_set_pkt_pos     (frame, avctx->internal->pkt->pos);
-    av_frame_set_pkt_duration(frame, avctx->pkt->duration);
+    av_frame_set_pkt_duration(frame, avctx->internal->pkt->duration);
 
     if (context->tff >= 0) {
         frame->interlaced_frame = 1;
@@ -206,14 +209,14 @@ static int raw_decode(AVCodecContext *avctx, void *data, int *got_frame,
         int i;
         uint8_t *dst = frame->buf[0]->data;
         buf_size = context->frame_size - AVPALETTE_SIZE;
-        if (avctx->bits_per_coded_sample == 4) {
+        if ((avctx->bits_per_coded_sample & 0x1f) == 4) {
             for (i = 0; 2 * i + 1 < buf_size && i<avpkt->size; i++) {
                 dst[2 * i + 0] = buf[i] >> 4;
                 dst[2 * i + 1] = buf[i] & 15;
             }
             linesize_align = 8;
         } else {
-            av_assert0(avctx->bits_per_coded_sample == 2);
+            av_assert0((avctx->bits_per_coded_sample & 0x1f) == 2);
             for (i = 0; 4 * i + 3 < buf_size && i<avpkt->size; i++) {
                 dst[4 * i + 0] = buf[i] >> 6;
                 dst[4 * i + 1] = buf[i] >> 4 & 3;
@@ -223,6 +226,17 @@ static int raw_decode(AVCodecContext *avctx, void *data, int *got_frame,
             linesize_align = 16;
         }
         buf = dst;
+    } else if (context->is_lt_16bpp) {
+        int i;
+        uint8_t *dst = frame->buf[0]->data;
+        if (desc->flags & AV_PIX_FMT_FLAG_BE) {
+            for (i = 0; i + 1 < buf_size; i += 2)
+                AV_WB16(dst + i, AV_RB16(buf + i) << (16 - avctx->bits_per_coded_sample));
+        } else {
+            for (i = 0; i + 1 < buf_size; i += 2)
+                AV_WL16(dst + i, AV_RL16(buf + i) << (16 - avctx->bits_per_coded_sample));
+        }
+        buf = dst;
     } else if (need_copy) {
         memcpy(frame->buf[0]->data, buf, buf_size);
         buf = frame->buf[0]->data;

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/tak_parser.c

@@ -97,7 +97,7 @@ static int tak_parse(AVCodecParserContext *s, AVCodecContext *avctx,
     }
 found:
 
-    if ((consumed && !buf_size && next == END_NOT_FOUND) ||
+    if (consumed && !buf_size && next == END_NOT_FOUND ||
         ff_combine_frame(pc, next, &buf, &buf_size) < 0) {
         *poutbuf      = NULL;
         *poutbuf_size = 0;

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/takdec.c

@@ -163,8 +163,8 @@ static void set_sample_rate_params(AVCodecContext *avctx)
     TAKDecContext *s  = avctx->priv_data;
     int shift         = 3 - (avctx->sample_rate / 11025);
     shift             = FFMAX(0, shift);
-    s->uval           = FFALIGN((avctx->sample_rate + 511) >> 9, 4) << shift;
+    s->uval           = FFALIGN(avctx->sample_rate + 511 >> 9, 4) << shift;
-    s->subframe_scale = FFALIGN((avctx->sample_rate + 511) >> 9, 4) << 1;
+    s->subframe_scale = FFALIGN(avctx->sample_rate + 511 >> 9, 4) << 1;
 }
 
 static av_cold int tak_decode_init(AVCodecContext *avctx)
@@ -190,7 +190,7 @@ static void decode_lpc(int32_t *coeffs, int mode, int length)
 
     if (mode == 1) {
         int a1 = *coeffs++;
-        for (i = 0; i < (length - 1) >> 1; i++) {
+        for (i = 0; i < length - 1 >> 1; i++) {
             *coeffs   += a1;
             coeffs[1] += *coeffs;
             a1         = coeffs[1];
@@ -204,7 +204,7 @@ static void decode_lpc(int32_t *coeffs, int mode, int length)
         coeffs[1] = a2;
         if (length > 2) {
             coeffs += 2;
-            for (i = 0; i < (length - 2) >> 1; i++) {
+            for (i = 0; i < length - 2 >> 1; i++) {
                 int a3    = *coeffs + a1;
                 int a4    = a3 + a2;
                 *coeffs   = a4;
@@ -436,8 +436,8 @@ static int decode_subframe(TAKDecContext *s, int32_t *decoded,
         int32_t *p2 = &tfilter[i - 1];
 
         for (j = 0; j < (i + 1) / 2; j++) {
-            x     = *p1 + ((s->predictors[i] * *p2 + 256) >> 9);
+            x     = *p1 + (s->predictors[i] * *p2 + 256 >> 9);
-            *p2  += (s->predictors[i] * *p1 + 256) >> 9;
+            *p2  += s->predictors[i] * *p1 + 256 >> 9;
             *p1++ = x;
             p2--;
         }
@@ -579,7 +579,7 @@ static int decorrelate(TAKDecContext *s, int c1, int c2, int length)
         for (i = 0; i < length; i++) {
             int32_t a = p1[i];
             int32_t b = p2[i];
-            b         = (dfactor * (b >> dshift) + 128) >> 8 << dshift;
+            b         = dfactor * (b >> dshift) + 128 >> 8 << dshift;
             p1[i]     = b - a;
         }
         break;
@@ -686,11 +686,12 @@ static int tak_decode_frame(AVCodecContext *avctx, void *data,
     if ((ret = ff_tak_decode_frame_header(avctx, gb, &s->ti, 0)) < 0)
         return ret;
 
-    if (avctx->err_recognition & AV_EF_CRCCHECK) {
+    if (avctx->err_recognition & (AV_EF_CRCCHECK|AV_EF_COMPLIANT)) {
         hsize = get_bits_count(gb) / 8;
         if (ff_tak_check_crc(pkt->data, hsize)) {
             av_log(avctx, AV_LOG_ERROR, "CRC error\n");
-            return AVERROR_INVALIDDATA;
+            if (avctx->err_recognition & AV_EF_EXPLODE)
+                return AVERROR_INVALIDDATA;
         }
     }
 
@@ -720,11 +721,9 @@ static int tak_decode_frame(AVCodecContext *avctx, void *data,
         return AVERROR_INVALIDDATA;
     }
 
-    if (s->ti.bps != avctx->bits_per_raw_sample) {
+    avctx->bits_per_raw_sample = s->ti.bps;
-        avctx->bits_per_raw_sample = s->ti.bps;
+    if ((ret = set_bps_params(avctx)) < 0)
-        if ((ret = set_bps_params(avctx)) < 0)
+        return ret;
-            return ret;
-    }
     if (s->ti.sample_rate != avctx->sample_rate) {
         avctx->sample_rate = s->ti.sample_rate;
         set_sample_rate_params(avctx);
@@ -767,7 +766,7 @@ static int tak_decode_frame(AVCodecContext *avctx, void *data,
     } else {
         if (s->ti.codec == TAK_CODEC_MONO_STEREO) {
             for (chan = 0; chan < avctx->channels; chan++)
-                if ((ret = decode_channel(s, chan)))
+                if (ret = decode_channel(s, chan))
                     return ret;
 
             if (avctx->channels == 2) {
@@ -777,7 +776,7 @@ static int tak_decode_frame(AVCodecContext *avctx, void *data,
                 }
 
                 s->dmode = get_bits(gb, 3);
-                if ((ret = decorrelate(s, 0, 1, s->nb_samples - 1)))
+                if (ret = decorrelate(s, 0, 1, s->nb_samples - 1))
                     return ret;
             }
         } else if (s->ti.codec == TAK_CODEC_MULTICHANNEL) {
@@ -825,18 +824,18 @@ static int tak_decode_frame(AVCodecContext *avctx, void *data,
 
             for (i = 0; i < chan; i++) {
                 if (s->mcdparams[i].present && s->mcdparams[i].index == 1)
-                    if ((ret = decode_channel(s, s->mcdparams[i].chan2)))
+                    if (ret = decode_channel(s, s->mcdparams[i].chan2))
                         return ret;
 
-                if ((ret = decode_channel(s, s->mcdparams[i].chan1)))
+                if (ret = decode_channel(s, s->mcdparams[i].chan1))
                     return ret;
 
                 if (s->mcdparams[i].present) {
                     s->dmode = mc_dmodes[s->mcdparams[i].index];
-                    if ((ret = decorrelate(s,
+                    if (ret = decorrelate(s,
                                           s->mcdparams[i].chan2,
                                           s->mcdparams[i].chan1,
-                                          s->nb_samples - 1)))
+                                          s->nb_samples - 1))
                         return ret;
                 }
             }
@@ -861,11 +860,12 @@ static int tak_decode_frame(AVCodecContext *avctx, void *data,
     else if (get_bits_left(gb) > 0)
         av_log(avctx, AV_LOG_DEBUG, "underread\n");
 
-    if (avctx->err_recognition & AV_EF_CRCCHECK) {
+    if (avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_COMPLIANT)) {
         if (ff_tak_check_crc(pkt->data + hsize,
                              get_bits_count(gb) / 8 - hsize)) {
             av_log(avctx, AV_LOG_ERROR, "CRC error\n");
-            return AVERROR_INVALIDDATA;
+            if (avctx->err_recognition & AV_EF_EXPLODE)
+                return AVERROR_INVALIDDATA;
         }
     }
 
@@ -894,7 +894,6 @@ static int tak_decode_frame(AVCodecContext *avctx, void *data,
                 samples[i] <<= 8;
         }
         break;
-    default: break;
     }
 
     *got_frame_ptr = 1;
@@ -932,6 +931,7 @@ static av_cold int tak_decode_close(AVCodecContext *avctx)
 
 AVCodec ff_tak_decoder = {
     .name             = "tak",
+    .long_name        = NULL_IF_CONFIG_SMALL("TAK (Tom's lossless Audio Kompressor)"),
     .type             = AVMEDIA_TYPE_AUDIO,
     .id               = AV_CODEC_ID_TAK,
     .priv_data_size   = sizeof(TAKDecContext),
@@ -941,7 +941,6 @@ AVCodec ff_tak_decoder = {
     .init_thread_copy = ONLY_IF_THREADS_ENABLED(init_thread_copy),
     .update_thread_context = ONLY_IF_THREADS_ENABLED(update_thread_context),
     .capabilities     = CODEC_CAP_DR1 | CODEC_CAP_FRAME_THREADS,
-    .long_name        = NULL_IF_CONFIG_SMALL("TAK (Tom's lossless Audio Kompressor)"),
     .sample_fmts      = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
                                                         AV_SAMPLE_FMT_S16P,
                                                         AV_SAMPLE_FMT_S32P,

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/truespeech.c

@@ -21,7 +21,6 @@
 
 #include "libavutil/channel_layout.h"
 #include "libavutil/intreadwrite.h"
-#include "libavutil/internal.h"
 #include "avcodec.h"
 #include "dsputil.h"
 #include "get_bits.h"

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/tta.c

@@ -29,8 +29,8 @@
 
 #define BITSTREAM_READER_LE
 #include <limits.h>
-#include "config.h"
 #include "ttadata.h"
+#include "ttadsp.h"
 #include "avcodec.h"
 #include "get_bits.h"
 #include "thread.h"
@@ -39,7 +39,6 @@
 #include "libavutil/crc.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/opt.h"
-#include "libavutil/internal.h"
 
 #define FORMAT_SIMPLE    1
 #define FORMAT_ENCRYPTED 2
@@ -58,39 +57,9 @@ typedef struct TTAContext {
     uint8_t crc_pass[8];
     uint8_t *pass;
     TTAChannel *ch_ctx;
+    TTADSPContext dsp;
 } TTAContext;
 
-static inline void ttafilter_process(TTAFilter *c, int32_t *in)
-{
-    register int32_t *dl = c->dl, *qm = c->qm, *dx = c->dx, sum = c->round;
-
-    if (c->error < 0) {
-        qm[0] -= dx[0]; qm[1] -= dx[1]; qm[2] -= dx[2]; qm[3] -= dx[3];
-        qm[4] -= dx[4]; qm[5] -= dx[5]; qm[6] -= dx[6]; qm[7] -= dx[7];
-    } else if (c->error > 0) {
-        qm[0] += dx[0]; qm[1] += dx[1]; qm[2] += dx[2]; qm[3] += dx[3];
-        qm[4] += dx[4]; qm[5] += dx[5]; qm[6] += dx[6]; qm[7] += dx[7];
-    }
-
-    sum += dl[0] * qm[0] + dl[1] * qm[1] + dl[2] * qm[2] + dl[3] * qm[3] +
-           dl[4] * qm[4] + dl[5] * qm[5] + dl[6] * qm[6] + dl[7] * qm[7];
-
-    dx[0] = dx[1]; dx[1] = dx[2]; dx[2] = dx[3]; dx[3] = dx[4];
-    dl[0] = dl[1]; dl[1] = dl[2]; dl[2] = dl[3]; dl[3] = dl[4];
-
-    dx[4] = ((dl[4] >> 30) | 1);
-    dx[5] = ((dl[5] >> 30) | 2) & ~1;
-    dx[6] = ((dl[6] >> 30) | 2) & ~1;
-    dx[7] = ((dl[7] >> 30) | 4) & ~3;
-
-    c->error = *in;
-    *in += (sum >> c->shift);
-
-    dl[4] = -dl[5]; dl[5] = -dl[6];
-    dl[6] = *in - dl[7]; dl[7] = *in;
-    dl[5] += dl[6]; dl[4] += dl[5];
-}
-
 static const int64_t tta_channel_layouts[7] = {
     AV_CH_LAYOUT_STEREO,
     AV_CH_LAYOUT_STEREO|AV_CH_LOW_FREQUENCY,
@@ -118,7 +87,7 @@ static int tta_check_crc(TTAContext *s, const uint8_t *buf, int buf_size)
 static uint64_t tta_check_crc64(uint8_t *pass)
 {
     uint64_t crc = UINT64_MAX, poly = 0x42F0E1EBA9EA3693U;
-    uint8_t *end = pass + strlen((const char *) pass);
+    uint8_t *end = pass + strlen((const char *)pass);
     int i;
 
     while (pass < end) {
@@ -236,6 +205,8 @@ static av_cold int tta_decode_init(AVCodecContext * avctx)
         return AVERROR_INVALIDDATA;
     }
 
+    ff_ttadsp_init(&s->dsp);
+
     return allocate_buffers(avctx);
 }
 
@@ -253,7 +224,8 @@ static int tta_decode_frame(AVCodecContext *avctx, void *data,
     int32_t *p;
 
     if (avctx->err_recognition & AV_EF_CRCCHECK) {
-        if (buf_size < 4 || tta_check_crc(s, buf, buf_size - 4))
+        if (buf_size < 4 ||
+            (tta_check_crc(s, buf, buf_size - 4) && avctx->err_recognition & AV_EF_EXPLODE))
             return AVERROR_INVALIDDATA;
     }
 
@@ -336,7 +308,8 @@ static int tta_decode_frame(AVCodecContext *avctx, void *data,
         *p = 1 + ((value >> 1) ^ ((value & 1) - 1));
 
         // run hybrid filter
-        ttafilter_process(filter, p);
+        s->dsp.ttafilter_process_dec(filter->qm, filter->dx, filter->dl, &filter->error, p,
+                                     filter->shift, filter->round);
 
         // fixed order prediction
 #define PRED(x, k) (int32_t)((((uint64_t)x << k) - x) >> k)
@@ -451,7 +424,6 @@ AVCodec ff_tta_decoder = {
     .init           = tta_decode_init,
     .close          = tta_decode_close,
     .decode         = tta_decode_frame,
-    .init_thread_copy = ONLY_IF_THREADS_ENABLED(init_thread_copy),
+    .capabilities   = CODEC_CAP_DR1,
-    .capabilities   = CODEC_CAP_DR1 | CODEC_CAP_FRAME_THREADS,
     .priv_class     = &tta_decoder_class,
 };

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/ttadsp.c

@@ -0,0 +1,57 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "ttadsp.h"
+
+static void ttafilter_process_dec_c(int32_t *qm, int32_t *dx, int32_t *dl,
+                                    int32_t *error, int32_t *in, int32_t shift,
+                                    int32_t round) {
+    if (*error < 0) {
+        qm[0] -= dx[0]; qm[1] -= dx[1]; qm[2] -= dx[2]; qm[3] -= dx[3];
+        qm[4] -= dx[4]; qm[5] -= dx[5]; qm[6] -= dx[6]; qm[7] -= dx[7];
+    } else if (*error > 0) {
+        qm[0] += dx[0]; qm[1] += dx[1]; qm[2] += dx[2]; qm[3] += dx[3];
+        qm[4] += dx[4]; qm[5] += dx[5]; qm[6] += dx[6]; qm[7] += dx[7];
+    }
+
+    round +=    dl[0] * qm[0] + dl[1] * qm[1] + dl[2] * qm[2] + dl[3] * qm[3] +
+                dl[4] * qm[4] + dl[5] * qm[5] + dl[6] * qm[6] + dl[7] * qm[7];
+
+    dx[0] = dx[1]; dx[1] = dx[2]; dx[2] = dx[3]; dx[3] = dx[4];
+    dl[0] = dl[1]; dl[1] = dl[2]; dl[2] = dl[3]; dl[3] = dl[4];
+
+    dx[4] = ((dl[4] >> 30) | 1);
+    dx[5] = ((dl[5] >> 30) | 2) & ~1;
+    dx[6] = ((dl[6] >> 30) | 2) & ~1;
+    dx[7] = ((dl[7] >> 30) | 4) & ~3;
+
+    *error = *in;
+    *in += (round >> shift);
+
+    dl[4] = -dl[5]; dl[5] = -dl[6];
+    dl[6] = *in - dl[7]; dl[7] = *in;
+    dl[5] += dl[6]; dl[4] += dl[5];
+}
+
+av_cold void ff_ttadsp_init(TTADSPContext *c)
+{
+    c->ttafilter_process_dec = ttafilter_process_dec_c;
+
+    if (ARCH_X86)
+        ff_ttadsp_init_x86(c);
+}

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/ttadsp.h

@@ -0,0 +1,34 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_TTADSP_H
+#define AVCODEC_TTADSP_H
+
+#include <stdint.h>
+#include "ttadata.h"
+
+typedef struct TTADSPContext {
+    void (*ttafilter_process_dec)(int32_t *qm, int32_t *dx, int32_t *dl,
+                                  int32_t *error, int32_t *in, int32_t shift,
+                                  int32_t round);
+} TTADSPContext;
+
+void ff_ttadsp_init(TTADSPContext *c);
+void ff_ttadsp_init_x86(TTADSPContext *c);
+
+#endif /* AVCODEC_TTADSP_H */

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/twinvq.c

@@ -216,7 +216,7 @@ static void dec_gain(TwinVQContext *tctx,
                      enum TwinVQFrameType ftype, float *out)
 {
     const TwinVQModeTab   *mtab =  tctx->mtab;
-    const TwinVQFrameData *bits = &tctx->bits;
+    const TwinVQFrameData *bits = &tctx->bits[tctx->cur_frame];
     int i, j;
     int sub        = mtab->fmode[ftype].sub;
     float step     = TWINVQ_AMP_MAX     / ((1 << TWINVQ_GAIN_BITS)     - 1);
@@ -377,11 +377,12 @@ static void imdct_and_window(TwinVQContext *tctx, enum TwinVQFrameType ftype,
 }
 
 static void imdct_output(TwinVQContext *tctx, enum TwinVQFrameType ftype,
-                         int wtype, float **out)
+                         int wtype, float **out, int offset)
 {
     const TwinVQModeTab *mtab = tctx->mtab;
     float *prev_buf           = tctx->prev_frame + tctx->last_block_pos[0];
     int size1, size2, i;
+    float *out1, *out2;
 
     for (i = 0; i < tctx->avctx->channels; i++)
         imdct_and_window(tctx, ftype, wtype,
@@ -395,15 +396,17 @@ static void imdct_output(TwinVQContext *tctx, enum TwinVQFrameType ftype,
     size2 = tctx->last_block_pos[0];
     size1 = mtab->size - size2;
 
-    memcpy(&out[0][0],     prev_buf,         size1 * sizeof(out[0][0]));
+    out1 = &out[0][0] + offset;
-    memcpy(&out[0][size1], tctx->curr_frame, size2 * sizeof(out[0][0]));
+    memcpy(out1,         prev_buf,         size1 * sizeof(*out1));
+    memcpy(out1 + size1, tctx->curr_frame, size2 * sizeof(*out1));
 
     if (tctx->avctx->channels == 2) {
-        memcpy(&out[1][0], &prev_buf[2 * mtab->size],
+        out2 = &out[1][0] + offset;
-               size1 * sizeof(out[1][0]));
+        memcpy(out2, &prev_buf[2 * mtab->size],
-        memcpy(&out[1][size1], &tctx->curr_frame[2 * mtab->size],
+               size1 * sizeof(*out2));
-               size2 * sizeof(out[1][0]));
+        memcpy(out2 + size1, &tctx->curr_frame[2 * mtab->size],
-        tctx->fdsp.butterflies_float(out[0], out[1], mtab->size);
+               size2 * sizeof(*out2));
+        tctx->fdsp.butterflies_float(out1, out2, mtab->size);
     }
 }
 
@@ -411,7 +414,7 @@ static void read_and_decode_spectrum(TwinVQContext *tctx, float *out,
                                      enum TwinVQFrameType ftype)
 {
     const TwinVQModeTab *mtab = tctx->mtab;
-    TwinVQFrameData *bits     = &tctx->bits;
+    TwinVQFrameData *bits     = &tctx->bits[tctx->cur_frame];
     int channels              = tctx->avctx->channels;
     int sub        = mtab->fmode[ftype].sub;
     int block_size = mtab->size / sub;
@@ -484,7 +487,7 @@ int ff_twinvq_decode_frame(AVCodecContext *avctx, void *data,
 
     /* get output buffer */
     if (tctx->discarded_packets >= 2) {
-        frame->nb_samples = mtab->size;
+        frame->nb_samples = mtab->size * tctx->frames_per_packet;
         if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
             return ret;
         out = (float **)frame->extended_data;
@@ -499,11 +502,17 @@ int ff_twinvq_decode_frame(AVCodecContext *avctx, void *data,
     if ((ret = tctx->read_bitstream(avctx, tctx, buf, buf_size)) < 0)
         return ret;
 
-    read_and_decode_spectrum(tctx, tctx->spectrum, tctx->bits.ftype);
+    for (tctx->cur_frame = 0; tctx->cur_frame < tctx->frames_per_packet;
+         tctx->cur_frame++) {
+        read_and_decode_spectrum(tctx, tctx->spectrum,
+                                 tctx->bits[tctx->cur_frame].ftype);
 
-    imdct_output(tctx, tctx->bits.ftype, tctx->bits.window_type, out);
+        imdct_output(tctx, tctx->bits[tctx->cur_frame].ftype,
+                     tctx->bits[tctx->cur_frame].window_type, out,
+                     tctx->cur_frame * mtab->size);
 
-    FFSWAP(float *, tctx->curr_frame, tctx->prev_frame);
+        FFSWAP(float *, tctx->curr_frame, tctx->prev_frame);
+    }
 
     if (tctx->discarded_packets < 2) {
         tctx->discarded_packets++;
@@ -513,6 +522,9 @@ int ff_twinvq_decode_frame(AVCodecContext *avctx, void *data,
 
     *got_frame_ptr = 1;
 
+    // VQF can deliver packets 1 byte greater than block align
+    if (buf_size == avctx->block_align + 1)
+        return buf_size;
     return avctx->block_align;
 }
 
@@ -693,7 +705,7 @@ static av_cold void init_bitstream_params(TwinVQContext *tctx)
             TWINVQ_WINDOW_TYPE_BITS +
             mtab->fmode[i].sub * (bse_bits[i] + n_ch * TWINVQ_SUB_GAIN_BITS);
 
-    if (tctx->codec == TWINVQ_CODEC_METASOUND) {
+    if (tctx->codec == TWINVQ_CODEC_METASOUND && !tctx->is_6kbps) {
         bsize_no_main_cb[1] += 2;
         bsize_no_main_cb[2] += 2;
     }
@@ -763,6 +775,20 @@ av_cold int ff_twinvq_decode_init(AVCodecContext *avctx)
     tctx->avctx       = avctx;
     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
 
+    if (!avctx->block_align) {
+        avctx->block_align = tctx->frame_size + 7 >> 3;
+    } else if (avctx->block_align * 8 < tctx->frame_size) {
+        av_log(avctx, AV_LOG_ERROR, "Block align is %d bits, expected %d\n",
+               avctx->block_align * 8, tctx->frame_size);
+        return AVERROR_INVALIDDATA;
+    }
+    tctx->frames_per_packet = avctx->block_align * 8 / tctx->frame_size;
+    if (tctx->frames_per_packet > TWINVQ_MAX_FRAMES_PER_PACKET) {
+        av_log(avctx, AV_LOG_ERROR, "Too many frames per packet (%d)\n",
+               tctx->frames_per_packet);
+        return AVERROR_INVALIDDATA;
+    }
+
     avpriv_float_dsp_init(&tctx->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
     if ((ret = init_mdct_win(tctx))) {
         av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/twinvq.h

@@ -58,6 +58,8 @@ enum TwinVQFrameType {
 #define TWINVQ_SUBBLOCKS_MAX     16
 #define TWINVQ_BARK_N_COEF_MAX   4
 
+#define TWINVQ_MAX_FRAMES_PER_PACKET 2
+
 /**
  * Parameters and tables that are different for each frame type
  */
@@ -139,6 +141,8 @@ typedef struct TwinVQContext {
 
     const TwinVQModeTab *mtab;
 
+    int is_6kbps;
+
     // history
     float lsp_hist[2][20];           ///< LSP coefficients of the last frame
     float bark_hist[3][2][40];       ///< BSE coefficients of last frame
@@ -162,7 +166,8 @@ typedef struct TwinVQContext {
     // scratch buffers
     float *tmp_buf;
 
-    TwinVQFrameData bits;
+    int frame_size, frames_per_packet, cur_frame;
+    TwinVQFrameData bits[TWINVQ_MAX_FRAMES_PER_PACKET];
 
     enum TwinVQCodec codec;
 

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/twinvqdec.c

@@ -251,7 +251,7 @@ static void read_cb_data(TwinVQContext *tctx, GetBitContext *gb,
 static int twinvq_read_bitstream(AVCodecContext *avctx, TwinVQContext *tctx,
                                  const uint8_t *buf, int buf_size)
 {
-    TwinVQFrameData     *bits = &tctx->bits;
+    TwinVQFrameData     *bits = &tctx->bits[0];
     const TwinVQModeTab *mtab = tctx->mtab;
     int channels              = tctx->avctx->channels;
     int sub;
@@ -268,7 +268,7 @@ static int twinvq_read_bitstream(AVCodecContext *avctx, TwinVQContext *tctx,
         return AVERROR_INVALIDDATA;
     }
 
-    bits->ftype = ff_twinvq_wtype_to_ftype_table[tctx->bits.window_type];
+    bits->ftype = ff_twinvq_wtype_to_ftype_table[tctx->bits[0].window_type];
 
     sub = mtab->fmode[bits->ftype].sub;
 
@@ -312,7 +312,7 @@ static int twinvq_read_bitstream(AVCodecContext *avctx, TwinVQContext *tctx,
         }
     }
 
-    return 0;
+    return (get_bits_count(&gb) + 7) / 8;
 }
 
 static av_cold int twinvq_decode_init(AVCodecContext *avctx)
@@ -396,13 +396,18 @@ static av_cold int twinvq_decode_init(AVCodecContext *avctx)
         return -1;
     }
 
-    avctx->block_align = (avctx->bit_rate * tctx->mtab->size
-                                          / avctx->sample_rate + 15) / 8;
-
     tctx->codec          = TWINVQ_CODEC_VQF;
     tctx->read_bitstream = twinvq_read_bitstream;
     tctx->dec_bark_env   = dec_bark_env;
     tctx->decode_ppc     = decode_ppc;
+    tctx->frame_size     = avctx->bit_rate * tctx->mtab->size
+                                           / avctx->sample_rate + 8;
+    tctx->is_6kbps       = 0;
+    if (avctx->block_align && avctx->block_align * 8 / tctx->frame_size > 1) {
+        av_log(avctx, AV_LOG_ERROR,
+               "VQF TwinVQ should have only one frame per packet\n");
+        return AVERROR_INVALIDDATA;
+    }
 
     return ff_twinvq_decode_init(avctx);
 }

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/version.h

@@ -27,14 +27,14 @@
  */
 
 #ifdef __FRAMEWORK__
-#include "avutil.h"
+#include "version.h"
 #else
-#include "libavutil/avutil.h"
+#include "libavutil/version.h"
 #endif
 
 #define LIBAVCODEC_VERSION_MAJOR 55
-#define LIBAVCODEC_VERSION_MINOR  33
+#define LIBAVCODEC_VERSION_MINOR  52
-#define LIBAVCODEC_VERSION_MICRO 101
+#define LIBAVCODEC_VERSION_MICRO 102
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
                                                LIBAVCODEC_VERSION_MINOR, \
@@ -55,19 +55,9 @@
 #ifndef FF_API_REQUEST_CHANNELS
 #define FF_API_REQUEST_CHANNELS (LIBAVCODEC_VERSION_MAJOR < 56)
 #endif
-#ifndef FF_API_ALLOC_CONTEXT
-#define FF_API_ALLOC_CONTEXT    (LIBAVCODEC_VERSION_MAJOR < 55)
-#endif
-#ifndef FF_API_AVCODEC_OPEN
-#define FF_API_AVCODEC_OPEN     (LIBAVCODEC_VERSION_MAJOR < 55)
-#endif
 #ifndef FF_API_OLD_DECODE_AUDIO
 #define FF_API_OLD_DECODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 56)
 #endif
-#ifndef FF_API_OLD_TIMECODE
-#define FF_API_OLD_TIMECODE (LIBAVCODEC_VERSION_MAJOR < 55)
-#endif
-
 #ifndef FF_API_OLD_ENCODE_AUDIO
 #define FF_API_OLD_ENCODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 56)
 #endif
@@ -77,8 +67,11 @@
 #ifndef FF_API_CODEC_ID
 #define FF_API_CODEC_ID          (LIBAVCODEC_VERSION_MAJOR < 56)
 #endif
+#ifndef FF_API_AUDIO_CONVERT
+#define FF_API_AUDIO_CONVERT     (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
 #ifndef FF_API_AVCODEC_RESAMPLE
-#define FF_API_AVCODEC_RESAMPLE  (LIBAVCODEC_VERSION_MAJOR < 56)
+#define FF_API_AVCODEC_RESAMPLE  FF_API_AUDIO_CONVERT
 #endif
 #ifndef FF_API_DEINTERLACE
 #define FF_API_DEINTERLACE       (LIBAVCODEC_VERSION_MAJOR < 56)
@@ -104,5 +97,53 @@
 #ifndef FF_API_VOXWARE
 #define FF_API_VOXWARE           (LIBAVCODEC_VERSION_MAJOR < 56)
 #endif
+#ifndef FF_API_SET_DIMENSIONS
+#define FF_API_SET_DIMENSIONS    (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_DEBUG_MV
+#define FF_API_DEBUG_MV          (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_AC_VLC
+#define FF_API_AC_VLC            (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_OLD_MSMPEG4
+#define FF_API_OLD_MSMPEG4       (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_ASPECT_EXTENDED
+#define FF_API_ASPECT_EXTENDED   (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_THREAD_OPAQUE
+#define FF_API_THREAD_OPAQUE     (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_CODEC_PKT
+#define FF_API_CODEC_PKT         (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_ARCH_ALPHA
+#define FF_API_ARCH_ALPHA        (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_XVMC
+#define FF_API_XVMC              (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_ERROR_RATE
+#define FF_API_ERROR_RATE        (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_QSCALE_TYPE
+#define FF_API_QSCALE_TYPE       (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_MB_TYPE
+#define FF_API_MB_TYPE           (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_MAX_BFRAMES
+#define FF_API_MAX_BFRAMES       (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_FAST_MALLOC
+#define FF_API_FAST_MALLOC       (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_NEG_LINESIZES
+#define FF_API_NEG_LINESIZES     (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
+#ifndef FF_API_EMU_EDGE
+#define FF_API_EMU_EDGE          (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
 
 #endif /* AVCODEC_VERSION_H */

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/videodsp.h

@@ -30,8 +30,8 @@
 #include <stdint.h>
 
 #define EMULATED_EDGE(depth) \
-void ff_emulated_edge_mc_ ## depth(uint8_t *dst, ptrdiff_t dst_stride, \
+void ff_emulated_edge_mc_ ## depth(uint8_t *dst, const uint8_t *src, \
-                                   const uint8_t *src, ptrdiff_t src_stride, \
+                                   ptrdiff_t dst_stride, ptrdiff_t src_stride, \
                                    int block_w, int block_h,\
                                    int src_x, int src_y, int w, int h);
 
@@ -47,8 +47,10 @@ typedef struct VideoDSPContext {
      * @param dst_stride number of bytes between 2 vertically adjacent samples
      *                   in destination buffer
      * @param src source buffer
-     * @param src_stride number of bytes between 2 vertically adjacent samples
+     * @param dst_linesize number of bytes between 2 vertically adjacent
-     *                   in source buffer
+     *                     samples in the destination buffer
+     * @param src_linesize number of bytes between 2 vertically adjacent
+     *                     samples in both the source buffer
      * @param block_w width of block
      * @param block_h height of block
      * @param src_x x coordinate of the top left sample of the block in the
@@ -58,8 +60,9 @@ typedef struct VideoDSPContext {
      * @param w width of the source buffer
      * @param h height of the source buffer
      */
-    void (*emulated_edge_mc)(uint8_t *dst, ptrdiff_t dst_stride,
+    void (*emulated_edge_mc)(uint8_t *dst, const uint8_t *src,
-                             const uint8_t *src, ptrdiff_t src_stride,
+                             ptrdiff_t dst_linesize,
+                             ptrdiff_t src_linesize,
                              int block_w, int block_h,
                              int src_x, int src_y, int w, int h);
 

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/videodsp_template.c

@@ -20,8 +20,9 @@
  */
 
 #include "bit_depth_template.c"
-void FUNC(ff_emulated_edge_mc)(uint8_t *buf, ptrdiff_t buf_stride,
+void FUNC(ff_emulated_edge_mc)(uint8_t *buf, const uint8_t *src,
-                               const uint8_t *src, ptrdiff_t src_stride,
+                               ptrdiff_t buf_linesize,
+                               ptrdiff_t src_linesize,
                                int block_w, int block_h,
                                int src_x, int src_y, int w, int h)
 {
@@ -32,12 +33,12 @@ void FUNC(ff_emulated_edge_mc)(uint8_t *buf, ptrdiff_t buf_stride,
         return;
 
     if (src_y >= h) {
-        src -= src_y * src_stride;
+        src -= src_y * src_linesize;
-        src += (h - 1) * src_stride;
+        src += (h - 1) * src_linesize;
         src_y = h - 1;
     } else if (src_y <= -block_h) {
-        src -= src_y * src_stride;
+        src -= src_y * src_linesize;
-        src += (1 - block_h) * src_stride;
+        src += (1 - block_h) * src_linesize;
         src_y = 1 - block_h;
     }
     if (src_x >= w) {
@@ -56,30 +57,30 @@ void FUNC(ff_emulated_edge_mc)(uint8_t *buf, ptrdiff_t buf_stride,
     av_assert2(start_x < end_x && block_w);
 
     w    = end_x - start_x;
-    src += start_y * src_stride + start_x * sizeof(pixel);
+    src += start_y * src_linesize + start_x * sizeof(pixel);
     buf += start_x * sizeof(pixel);
 
     // top
     for (y = 0; y < start_y; y++) {
         memcpy(buf, src, w * sizeof(pixel));
-        buf += buf_stride;
+        buf += buf_linesize;
     }
 
     // copy existing part
     for (; y < end_y; y++) {
         memcpy(buf, src, w * sizeof(pixel));
-        src += src_stride;
+        src += src_linesize;
-        buf += buf_stride;
+        buf += buf_linesize;
     }
 
     // bottom
-    src -= src_stride;
+    src -= src_linesize;
     for (; y < block_h; y++) {
         memcpy(buf, src, w * sizeof(pixel));
-        buf += buf_stride;
+        buf += buf_linesize;
     }
 
-    buf -= block_h * buf_stride + start_x * sizeof(pixel);
+    buf -= block_h * buf_linesize + start_x * sizeof(pixel);
     while (block_h--) {
         pixel *bufp = (pixel *) buf;
 
@@ -92,6 +93,6 @@ void FUNC(ff_emulated_edge_mc)(uint8_t *buf, ptrdiff_t buf_stride,
         for (x = end_x; x < block_w; x++) {
             bufp[x] = bufp[end_x - 1];
         }
-        buf += buf_stride;
+        buf += buf_linesize;
     }
 }

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/wmadec.c

@@ -100,6 +100,9 @@ static av_cold int wma_decode_init(AVCodecContext * avctx)
         }
     }
 
+    for (i=0; i<MAX_CHANNELS; i++)
+        s->max_exponent[i] = 1.0;
+
     if(ff_wma_init(avctx, flags2)<0)
         return -1;
 
@@ -948,6 +951,7 @@ static av_cold void flush(AVCodecContext *avctx)
 #if CONFIG_WMAV1_DECODER
 AVCodec ff_wmav1_decoder = {
     .name           = "wmav1",
+    .long_name      = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_WMAV1,
     .priv_data_size = sizeof(WMACodecContext),
@@ -956,7 +960,6 @@ AVCodec ff_wmav1_decoder = {
     .decode         = wma_decode_superframe,
     .flush          = flush,
     .capabilities   = CODEC_CAP_DR1,
-    .long_name      = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                       AV_SAMPLE_FMT_NONE },
 };
@@ -964,6 +967,7 @@ AVCodec ff_wmav1_decoder = {
 #if CONFIG_WMAV2_DECODER
 AVCodec ff_wmav2_decoder = {
     .name           = "wmav2",
+    .long_name      = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_WMAV2,
     .priv_data_size = sizeof(WMACodecContext),
@@ -972,7 +976,6 @@ AVCodec ff_wmav2_decoder = {
     .decode         = wma_decode_superframe,
     .flush          = flush,
     .capabilities   = CODEC_CAP_DR1,
-    .long_name      = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                       AV_SAMPLE_FMT_NONE },
 };

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/wmalosslessdec.c

@@ -24,7 +24,6 @@
 
 #include "libavutil/attributes.h"
 #include "libavutil/avassert.h"
-#include "libavutil/internal.h"
 
 #include "avcodec.h"
 #include "internal.h"
@@ -128,8 +127,8 @@ typedef struct WmallDecodeCtx {
 
     int8_t  mclms_order;
     int8_t  mclms_scaling;
-    int16_t mclms_coeffs[128];
+    int16_t mclms_coeffs[WMALL_MAX_CHANNELS * WMALL_MAX_CHANNELS * 32];
-    int16_t mclms_coeffs_cur[4];
+    int16_t mclms_coeffs_cur[WMALL_MAX_CHANNELS * WMALL_MAX_CHANNELS];
     int16_t mclms_prevvalues[WMALL_MAX_CHANNELS * 2 * 32];
     int16_t mclms_updates[WMALL_MAX_CHANNELS * 2 * 32];
     int     mclms_recent;
@@ -349,11 +348,11 @@ static int decode_tilehdr(WmallDecodeCtx *s)
             if (num_samples[c] == min_channel_len) {
                 if (fixed_channel_layout || channels_for_cur_subframe == 1 ||
                    (min_channel_len == s->samples_per_frame - s->min_samples_per_subframe)) {
-                    contains_subframe[c] = in_use = 1;
+                    contains_subframe[c] = 1;
                 } else {
-                    if (get_bits1(&s->gb))
+                    contains_subframe[c] = get_bits1(&s->gb);
-                        contains_subframe[c] = in_use = 1;
                 }
+                in_use |= contains_subframe[c];
             } else
                 contains_subframe[c] = 0;
         }
@@ -653,10 +652,10 @@ static void mclms_update(WmallDecodeCtx *s, int icoef, int *pred)
     if (s->mclms_recent == 0) {
         memcpy(&s->mclms_prevvalues[order * num_channels],
                s->mclms_prevvalues,
-               2 * order * num_channels);
+               sizeof(int16_t) * order * num_channels);
         memcpy(&s->mclms_updates[order * num_channels],
                s->mclms_updates,
-               2 * order * num_channels);
+               sizeof(int16_t) * order * num_channels);
         s->mclms_recent = num_channels * order;
     }
 }
@@ -677,7 +676,7 @@ static void mclms_predict(WmallDecodeCtx *s, int icoef, int *pred)
         for (i = 0; i < ich; i++)
             pred[ich] += s->channel_residues[i][icoef] *
                          s->mclms_coeffs_cur[i + num_channels * ich];
-        pred[ich] += 1 << (s->mclms_scaling - 1);
+        pred[ich] += 1 << s->mclms_scaling - 1;
         pred[ich] >>= s->mclms_scaling;
         s->channel_residues[ich][icoef] += pred[ich];
     }
@@ -709,7 +708,7 @@ static void lms_update(WmallDecodeCtx *s, int ich, int ilms,
 {
     int icoef;
     int recent = s->cdlms[ich][ilms].recent;
-    int range  = 1 << (s->bits_per_sample - 1);
+    int range  = 1 << s->bits_per_sample - 1;
 
     if (residue < 0) {
         for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++)
@@ -1039,9 +1038,10 @@ static int decode_frame(WmallDecodeCtx *s)
         len = get_bits(gb, s->log2_frame_size);
 
     /* decode tile information */
-    if (decode_tilehdr(s)) {
+    if ((ret = decode_tilehdr(s))) {
         s->packet_loss = 1;
-        return 0;
+        av_frame_unref(s->frame);
+        return ret;
     }
 
     /* read drc info */
@@ -1076,8 +1076,11 @@ static int decode_frame(WmallDecodeCtx *s)
 
     /* decode all subframes */
     while (!s->parsed_all_subframes) {
+        int decoded_samples = s->channel[0].decoded_samples;
         if (decode_subframe(s) < 0) {
             s->packet_loss = 1;
+            if (s->frame->nb_samples)
+                s->frame->nb_samples = decoded_samples;
             return 0;
         }
     }
@@ -1183,6 +1186,8 @@ static int decode_packet(AVCodecContext *avctx, void *data, int *got_frame_ptr,
     if (s->packet_done || s->packet_loss) {
         s->packet_done = 0;
 
+        if (!buf_size)
+            return 0;
         /* sanity check for the buffer length */
         if (buf_size < avctx->block_align) {
             av_log(avctx, AV_LOG_ERROR, "buf size %d invalid\n", buf_size);
@@ -1305,6 +1310,7 @@ static av_cold int decode_close(AVCodecContext *avctx)
 
 AVCodec ff_wmalossless_decoder = {
     .name           = "wmalossless",
+    .long_name      = NULL_IF_CONFIG_SMALL("Windows Media Audio Lossless"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_WMALOSSLESS,
     .priv_data_size = sizeof(WmallDecodeCtx),
@@ -1313,7 +1319,6 @@ AVCodec ff_wmalossless_decoder = {
     .decode         = decode_packet,
     .flush          = flush,
     .capabilities   = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1 | CODEC_CAP_DELAY,
-    .long_name      = NULL_IF_CONFIG_SMALL("Windows Media Audio Lossless"),
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
                                                       AV_SAMPLE_FMT_S32P,
                                                       AV_SAMPLE_FMT_NONE },

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/wmaprodec.c

@@ -89,7 +89,6 @@
 #include "libavutil/float_dsp.h"
 #include "libavutil/intfloat.h"
 #include "libavutil/intreadwrite.h"
-#include "libavutil/internal.h"
 #include "avcodec.h"
 #include "internal.h"
 #include "get_bits.h"
@@ -682,7 +681,7 @@ static void decode_decorrelation_matrix(WMAProDecodeCtx *s,
 /**
  *@brief Decode channel transformation parameters
  *@param s codec context
- *@return 0 in case of success, < 0 in case of bitstream errors
+ *@return >= 0 in case of success, < 0 in case of bitstream errors
  */
 static int decode_channel_transform(WMAProDecodeCtx* s)
 {
@@ -1649,6 +1648,7 @@ static void flush(AVCodecContext *avctx)
  */
 AVCodec ff_wmapro_decoder = {
     .name           = "wmapro",
+    .long_name      = NULL_IF_CONFIG_SMALL("Windows Media Audio 9 Professional"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_WMAPRO,
     .priv_data_size = sizeof(WMAProDecodeCtx),
@@ -1657,7 +1657,6 @@ AVCodec ff_wmapro_decoder = {
     .decode         = decode_packet,
     .capabilities   = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
     .flush          = flush,
-    .long_name      = NULL_IF_CONFIG_SMALL("Windows Media Audio 9 Professional"),
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                       AV_SAMPLE_FMT_NONE },
 };

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/wmavoice.c

@@ -30,7 +30,6 @@
 #include "libavutil/channel_layout.h"
 #include "libavutil/float_dsp.h"
 #include "libavutil/mem.h"
-#include "libavutil/internal.h"
 #include "avcodec.h"
 #include "internal.h"
 #include "get_bits.h"
@@ -2063,6 +2062,7 @@ static av_cold void wmavoice_flush(AVCodecContext *ctx)
 
 AVCodec ff_wmavoice_decoder = {
     .name             = "wmavoice",
+    .long_name        = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
     .type             = AVMEDIA_TYPE_AUDIO,
     .id               = AV_CODEC_ID_WMAVOICE,
     .priv_data_size   = sizeof(WMAVoiceContext),
@@ -2072,5 +2072,4 @@ AVCodec ff_wmavoice_decoder = {
     .decode           = wmavoice_decode_packet,
     .capabilities     = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
     .flush            = wmavoice_flush,
-    .long_name        = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
 };

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/x86/dcadsp_init.c

@@ -0,0 +1,108 @@
+/*
+ * Copyright (c) 2012-2014 Christophe Gisquet <christophe.gisquet@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/attributes.h"
+#include "libavutil/cpu.h"
+#include "libavutil/x86/cpu.h"
+#include "libavcodec/dcadsp.h"
+
+void ff_decode_hf_sse(float dst[DCA_SUBBANDS][8], const int vq_num[DCA_SUBBANDS],
+                      const int8_t hf_vq[1024][32], intptr_t vq_offset,
+                      int scale[DCA_SUBBANDS][2], intptr_t start, intptr_t end);
+void ff_decode_hf_sse2(float dst[DCA_SUBBANDS][8], const int vq_num[DCA_SUBBANDS],
+                       const int8_t hf_vq[1024][32], intptr_t vq_offset,
+                       int scale[DCA_SUBBANDS][2], intptr_t start, intptr_t end);
+void ff_decode_hf_sse4(float dst[DCA_SUBBANDS][8], const int vq_num[DCA_SUBBANDS],
+                       const int8_t hf_vq[1024][32], intptr_t vq_offset,
+                       int scale[DCA_SUBBANDS][2], intptr_t start, intptr_t end);
+void ff_dca_lfe_fir0_sse(float *out, const float *in, const float *coefs);
+void ff_dca_lfe_fir1_sse(float *out, const float *in, const float *coefs);
+
+av_cold void ff_dcadsp_init_x86(DCADSPContext *s)
+{
+    int cpu_flags = av_get_cpu_flags();
+
+    if (EXTERNAL_SSE(cpu_flags)) {
+#if ARCH_X86_32
+        s->decode_hf = ff_decode_hf_sse;
+#endif
+        s->lfe_fir[0]        = ff_dca_lfe_fir0_sse;
+        s->lfe_fir[1]        = ff_dca_lfe_fir1_sse;
+    }
+
+    if (EXTERNAL_SSE2(cpu_flags)) {
+        s->decode_hf = ff_decode_hf_sse2;
+    }
+
+    if (EXTERNAL_SSE4(cpu_flags)) {
+        s->decode_hf = ff_decode_hf_sse4;
+    }
+}
+
+
+#define SYNTH_FILTER_FUNC(opt)                                                 \
+void ff_synth_filter_inner_##opt(float *synth_buf_ptr, float synth_buf2[32],   \
+                                 const float window[512],                      \
+                                 float out[32], intptr_t offset, float scale); \
+static void synth_filter_##opt(FFTContext *imdct,                              \
+                              float *synth_buf_ptr, int *synth_buf_offset,     \
+                              float synth_buf2[32], const float window[512],   \
+                              float out[32], const float in[32], float scale)  \
+{                                                                              \
+    float *synth_buf= synth_buf_ptr + *synth_buf_offset;                       \
+                                                                               \
+    imdct->imdct_half(imdct, synth_buf, in);                                   \
+                                                                               \
+    ff_synth_filter_inner_##opt(synth_buf, synth_buf2, window,                 \
+                                out, *synth_buf_offset, scale);                \
+                                                                               \
+    *synth_buf_offset = (*synth_buf_offset - 32) & 511;                        \
+}                                                                              \
+
+#if HAVE_YASM
+#if ARCH_X86_32
+SYNTH_FILTER_FUNC(sse)
+#endif
+SYNTH_FILTER_FUNC(sse2)
+SYNTH_FILTER_FUNC(avx)
+SYNTH_FILTER_FUNC(fma3)
+#endif /* HAVE_YASM */
+
+av_cold void ff_synth_filter_init_x86(SynthFilterContext *s)
+{
+#if HAVE_YASM
+    int cpu_flags = av_get_cpu_flags();
+
+#if ARCH_X86_32
+    if (EXTERNAL_SSE(cpu_flags)) {
+        s->synth_filter_float = synth_filter_sse;
+    }
+#endif
+    if (EXTERNAL_SSE2(cpu_flags)) {
+        s->synth_filter_float = synth_filter_sse2;
+    }
+    if (EXTERNAL_AVX(cpu_flags)) {
+        s->synth_filter_float = synth_filter_avx;
+    }
+    if (EXTERNAL_FMA3(cpu_flags)) {
+        s->synth_filter_float = synth_filter_fma3;
+    }
+#endif /* HAVE_YASM */
+}

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/x86/hpeldsp_init.c

@@ -22,7 +22,7 @@
  * MMX optimization by Nick Kurshev <nickols_k@mail.ru>
  */
 
-#include "config.h"
+#include "libavutil/attributes.h"
 #include "libavutil/cpu.h"
 #include "libavutil/x86/asm.h"
 #include "libavutil/x86/cpu.h"
@@ -252,19 +252,19 @@ static void hpeldsp_init_sse2(HpelDSPContext *c, int flags, int cpu_flags)
 #endif /* HAVE_SSE2_EXTERNAL */
 }
 
-void ff_hpeldsp_init_x86(HpelDSPContext *c, int flags)
+av_cold void ff_hpeldsp_init_x86(HpelDSPContext *c, int flags)
 {
     int cpu_flags = av_get_cpu_flags();
 
     if (INLINE_MMX(cpu_flags))
         hpeldsp_init_mmx(c, flags, cpu_flags);
 
-    if (EXTERNAL_MMXEXT(cpu_flags))
-        hpeldsp_init_mmxext(c, flags, cpu_flags);
-
     if (EXTERNAL_AMD3DNOW(cpu_flags))
         hpeldsp_init_3dnow(c, flags, cpu_flags);
 
+    if (EXTERNAL_MMXEXT(cpu_flags))
+        hpeldsp_init_mmxext(c, flags, cpu_flags);
+
     if (EXTERNAL_SSE2(cpu_flags))
         hpeldsp_init_sse2(c, flags, cpu_flags);
 }

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/x86/mpegaudiodsp.c

@@ -30,7 +30,9 @@
 static void imdct36_blocks_ ## CPU(float *out, float *buf, float *in, int count, int switch_point, int block_type);\
 void ff_imdct36_float_ ## CPU(float *out, float *buf, float *in, float *win);
 
+#if ARCH_X86_32
 DECL(sse)
+#endif
 DECL(sse2)
 DECL(sse3)
 DECL(ssse3)
@@ -223,7 +225,9 @@ static void imdct36_blocks_ ## CPU1(float *out, float *buf, float *in,      \
 }
 
 #if HAVE_SSE
+#if ARCH_X86_32
 DECL_IMDCT_BLOCKS(sse,sse)
+#endif
 DECL_IMDCT_BLOCKS(sse2,sse)
 DECL_IMDCT_BLOCKS(sse3,sse)
 DECL_IMDCT_BLOCKS(ssse3,sse)
@@ -235,9 +239,7 @@ DECL_IMDCT_BLOCKS(avx,avx)
 
 av_cold void ff_mpadsp_init_x86(MPADSPContext *s)
 {
-#if HAVE_SSE2_INLINE || HAVE_YASM
     int cpu_flags = av_get_cpu_flags();
-#endif
 
     int i, j;
     for (j = 0; j < 4; j++) {
@@ -254,15 +256,17 @@ av_cold void ff_mpadsp_init_x86(MPADSPContext *s)
     }
 
 #if HAVE_SSE2_INLINE
-    if (cpu_flags & AV_CPU_FLAG_SSE2) {
+    if (INLINE_SSE2(cpu_flags)) {
         s->apply_window_float = apply_window_mp3;
     }
 #endif /* HAVE_SSE2_INLINE */
 
 #if HAVE_YASM
+#if ARCH_X86_32
     if (EXTERNAL_SSE(cpu_flags)) {
         s->imdct36_blocks_float = imdct36_blocks_sse;
     }
+#endif
     if (EXTERNAL_SSE2(cpu_flags)) {
         s->imdct36_blocks_float = imdct36_blocks_sse2;
     }

Frameworks/FFMPEG/ffmpeg-minimal/libavcodec/x86/ttadsp_init.c

@@ -0,0 +1,42 @@
+/*
+ * Copyright (c) 2014 James Almer
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavcodec/ttadsp.h"
+#include "libavutil/x86/cpu.h"
+#include "config.h"
+
+void ff_ttafilter_process_dec_ssse3(int32_t *qm, int32_t *dx, int32_t *dl,
+                                    int32_t *error, int32_t *in, int32_t shift,
+                                    int32_t round);
+void ff_ttafilter_process_dec_sse4(int32_t *qm, int32_t *dx, int32_t *dl,
+                                   int32_t *error, int32_t *in, int32_t shift,
+                                   int32_t round);
+
+av_cold void ff_ttadsp_init_x86(TTADSPContext *c)
+{
+#if HAVE_YASM
+    int cpu_flags = av_get_cpu_flags();
+
+    if (EXTERNAL_SSSE3(cpu_flags))
+        c->ttafilter_process_dec = ff_ttafilter_process_dec_ssse3;
+    if (EXTERNAL_SSE4(cpu_flags))
+        c->ttafilter_process_dec = ff_ttafilter_process_dec_sse4;
+#endif
+}

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/aacdec.c

@@ -20,7 +20,6 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include "config.h"
 #include "libavutil/intreadwrite.h"
 #include "avformat.h"
 #include "internal.h"
@@ -91,7 +90,7 @@ static int adts_aac_read_header(AVFormatContext *s)
 
 AVInputFormat ff_aac_demuxer = {
     .name           = "aac",
-    .long_name      = NULL_IF_CONFIG_SMALL("raw ADTS AAC (Advanced Audio Coding)"),
+    .long_name      = "raw ADTS AAC (Advanced Audio Coding)",
     .read_probe     = adts_aac_probe,
     .read_header    = adts_aac_read_header,
     .read_packet    = ff_raw_read_partial_packet,

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/allformats.c

@@ -19,8 +19,8 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include "avformat.h"
 #include "config.h"
+#include "avformat.h"
 #include "rtp.h"
 #include "rdt.h"
 #include "url.h"
@@ -46,8 +46,7 @@
     {                                                                   \
         extern URLProtocol ff_##x##_protocol;                           \
         if (CONFIG_##X##_PROTOCOL)                                      \
-            ffurl_register_protocol(&ff_##x##_protocol,                 \
+            ffurl_register_protocol(&ff_##x##_protocol);                \
-                                    sizeof(ff_##x##_protocol));         \
     }
 
 void av_register_all(void)
@@ -136,8 +135,10 @@ void av_register_all(void)
     REGISTER_MUXDEMUX(H261,             h261);
     REGISTER_MUXDEMUX(H263,             h263);
     REGISTER_MUXDEMUX(H264,             h264);
-    REGISTER_DEMUXER (H265,             h265);
+    REGISTER_MUXER   (HDS,              hds);
+    REGISTER_MUXDEMUX(HEVC,             hevc);
     REGISTER_MUXDEMUX(HLS,              hls);
+    REGISTER_DEMUXER (HNM,              hnm);
     REGISTER_MUXDEMUX(ICO,              ico);
     REGISTER_DEMUXER (IDCIN,            idcin);
     REGISTER_DEMUXER (IDF,              idf);
@@ -205,6 +206,7 @@ void av_register_all(void)
     REGISTER_DEMUXER (NUV,              nuv);
     REGISTER_MUXDEMUX(OGG,              ogg);
     REGISTER_MUXDEMUX(OMA,              oma);
+    REGISTER_MUXER   (OPUS,             opus);
     REGISTER_DEMUXER (PAF,              paf);
     REGISTER_MUXDEMUX(PCM_ALAW,         pcm_alaw);
     REGISTER_MUXDEMUX(PCM_MULAW,        pcm_mulaw);
@@ -248,9 +250,10 @@ void av_register_all(void)
     REGISTER_MUXDEMUX(SAP,              sap);
     REGISTER_DEMUXER (SBG,              sbg);
     REGISTER_DEMUXER (SDP,              sdp);
+    REGISTER_DEMUXER (SDR2,             sdr2);
 #if CONFIG_RTPDEC
-    av_register_rtp_dynamic_payload_handlers();
+    ff_register_rtp_dynamic_payload_handlers();
-    av_register_rdt_dynamic_payload_handlers();
+    ff_register_rdt_dynamic_payload_handlers();
 #endif
     REGISTER_DEMUXER (SEGAFILM,         segafilm);
     REGISTER_MUXER   (SEGMENT,          segment);
@@ -264,6 +267,7 @@ void av_register_all(void)
     REGISTER_DEMUXER (SOL,              sol);
     REGISTER_MUXDEMUX(SOX,              sox);
     REGISTER_MUXDEMUX(SPDIF,            spdif);
+    REGISTER_MUXER   (SPEEX,            speex);
     REGISTER_MUXDEMUX(SRT,              srt);
     REGISTER_DEMUXER (STR,              str);
     REGISTER_DEMUXER (SUBVIEWER1,       subviewer1);
@@ -282,6 +286,7 @@ void av_register_all(void)
     REGISTER_DEMUXER (TTA,              tta);
     REGISTER_DEMUXER (TXD,              txd);
     REGISTER_DEMUXER (TTY,              tty);
+    REGISTER_MUXER   (UNCODEDFRAMECRC,  uncodedframecrc);
     REGISTER_MUXDEMUX(VC1,              vc1);
     REGISTER_MUXDEMUX(VC1T,             vc1t);
     REGISTER_DEMUXER (VIVO,             vivo);
@@ -334,6 +339,7 @@ void av_register_all(void)
     REGISTER_PROTOCOL(RTP,              rtp);
     REGISTER_PROTOCOL(SCTP,             sctp);
     REGISTER_PROTOCOL(SRTP,             srtp);
+    REGISTER_PROTOCOL(SUBFILE,          subfile);
     REGISTER_PROTOCOL(TCP,              tcp);
     REGISTER_PROTOCOL(TLS,              tls);
     REGISTER_PROTOCOL(UDP,              udp);

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/ape.c

@@ -22,7 +22,6 @@
 
 #include <stdio.h>
 
-#include "config.h"
 #include "libavutil/intreadwrite.h"
 #include "avformat.h"
 #include "internal.h"
@@ -87,10 +86,14 @@ typedef struct {
 
 static int ape_probe(AVProbeData * p)
 {
-    if (p->buf[0] == 'M' && p->buf[1] == 'A' && p->buf[2] == 'C' && p->buf[3] == ' ')
+    int version = AV_RL16(p->buf+4);
-        return AVPROBE_SCORE_MAX;
+    if (AV_RL32(p->buf) != MKTAG('M', 'A', 'C', ' '))
+        return 0;
 
-    return 0;
+    if (version < APE_MIN_VERSION || version > APE_MAX_VERSION)
+        return AVPROBE_SCORE_MAX/4;
+
+    return AVPROBE_SCORE_MAX;
 }
 
 static void ape_dumpinfo(AVFormatContext * s, APEContext * ape_ctx)
@@ -279,18 +282,20 @@ static int ape_read_header(AVFormatContext * s)
         ape->totalsamples += ape->blocksperframe * (ape->totalframes - 1);
 
     if (ape->seektablelength > 0) {
-        ape->seektable = av_malloc(ape->seektablelength);
+        ape->seektable = av_mallocz(ape->seektablelength);
         if (!ape->seektable)
             return AVERROR(ENOMEM);
         for (i = 0; i < ape->seektablelength / sizeof(uint32_t) && !pb->eof_reached; i++)
             ape->seektable[i] = avio_rl32(pb);
         if (ape->fileversion < 3810) {
-            ape->bittable = av_malloc(ape->totalframes);
+            ape->bittable = av_mallocz(ape->totalframes);
             if (!ape->bittable)
                 return AVERROR(ENOMEM);
             for (i = 0; i < ape->totalframes && !pb->eof_reached; i++)
                 ape->bittable[i] = avio_r8(pb);
         }
+        if (pb->eof_reached)
+            av_log(s, AV_LOG_WARNING, "File truncated\n");
     }
 
     ape->frames[0].pos     = ape->firstframe;
@@ -355,8 +360,8 @@ static int ape_read_header(AVFormatContext * s)
     st->duration  = total_blocks;
     avpriv_set_pts_info(st, 64, 1, ape->samplerate);
 
-    st->codec->extradata = av_malloc(APE_EXTRADATA_SIZE);
+    if (ff_alloc_extradata(st->codec, APE_EXTRADATA_SIZE))
-    st->codec->extradata_size = APE_EXTRADATA_SIZE;
+        return AVERROR(ENOMEM);
     AV_WL16(st->codec->extradata + 0, ape->fileversion);
     AV_WL16(st->codec->extradata + 2, ape->compressiontype);
     AV_WL16(st->codec->extradata + 4, ape->formatflags);
@@ -412,8 +417,10 @@ static int ape_read_packet(AVFormatContext * s, AVPacket * pkt)
     AV_WL32(pkt->data    , nblocks);
     AV_WL32(pkt->data + 4, ape->frames[ape->currentframe].skip);
     ret = avio_read(s->pb, pkt->data + extra_size, ape->frames[ape->currentframe].size);
-    if (ret < 0)
+    if (ret < 0) {
+        av_free_packet(pkt);
         return ret;
+    }
 
     pkt->pts = ape->frames[ape->currentframe].pts;
     pkt->stream_index = 0;
@@ -454,7 +461,7 @@ static int ape_read_seek(AVFormatContext *s, int stream_index, int64_t timestamp
 
 AVInputFormat ff_ape_demuxer = {
     .name           = "ape",
-    .long_name      = NULL_IF_CONFIG_SMALL("Monkey's Audio"),
+    .long_name      = "Monkey's Audio",
     .priv_data_size = sizeof(APEContext),
     .read_probe     = ape_probe,
     .read_header    = ape_read_header,

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/apetag.c

@@ -62,15 +62,15 @@ static int ape_tag_read_field(AVFormatContext *s)
         if (!st)
             return AVERROR(ENOMEM);
 
-        size -= avio_get_str(pb, size, (char *) filename, sizeof(filename));
+        size -= avio_get_str(pb, size, (char *)filename, sizeof(filename));
         if (size <= 0) {
             av_log(s, AV_LOG_WARNING, "Skipping binary tag '%s'.\n", key);
             return 0;
         }
 
-        av_dict_set(&st->metadata, (const char *) key, (const char *) filename, 0);
+        av_dict_set(&st->metadata, (const char *)key, (const char *)filename, 0);
 
-        if ((id = ff_guess_image2_codec((const char *) filename)) != AV_CODEC_ID_NONE) {
+        if ((id = ff_guess_image2_codec((const char *)filename)) != AV_CODEC_ID_NONE) {
             AVPacket pkt;
             int ret;
 
@@ -88,14 +88,8 @@ static int ape_tag_read_field(AVFormatContext *s)
             st->attached_pic.stream_index = st->index;
             st->attached_pic.flags       |= AV_PKT_FLAG_KEY;
         } else {
-            st->codec->extradata = av_malloc(size + FF_INPUT_BUFFER_PADDING_SIZE);
+            if (ff_get_extradata(st->codec, s->pb, size) < 0)
-            if (!st->codec->extradata)
                 return AVERROR(ENOMEM);
-            if (avio_read(pb, st->codec->extradata, size) != size) {
-                av_freep(&st->codec->extradata);
-                return AVERROR(EIO);
-            }
-            st->codec->extradata_size = size;
             st->codec->codec_type = AVMEDIA_TYPE_ATTACHMENT;
         }
     } else {
@@ -108,7 +102,7 @@ static int ape_tag_read_field(AVFormatContext *s)
             return c;
         }
         value[c] = 0;
-        av_dict_set(&s->metadata, (const char *) key, (const char *) value, AV_DICT_DONT_STRDUP_VAL);
+        av_dict_set(&s->metadata, (const char *)key, (const char *)value, AV_DICT_DONT_STRDUP_VAL);
     }
     return 0;
 }
@@ -128,7 +122,7 @@ int64_t ff_ape_parse_tag(AVFormatContext *s)
     avio_seek(pb, file_size - APE_TAG_FOOTER_BYTES, SEEK_SET);
 
     avio_read(pb, buf, 8);     /* APETAGEX */
-    if (strncmp((const char *) buf, APE_TAG_PREAMBLE, 8)) {
+    if (strncmp((const char *)buf, APE_TAG_PREAMBLE, 8)) {
         return 0;
     }
 
@@ -194,7 +188,7 @@ int ff_ape_write_tag(AVFormatContext *s)
     while ((e = av_dict_get(s->metadata, "", e, AV_DICT_IGNORE_SUFFIX))) {
         int val_len;
 
-        if (!string_is_ascii((const unsigned char *) e->key)) {
+        if (!string_is_ascii((const uint8_t *)e->key)) {
             av_log(s, AV_LOG_WARNING, "Non ASCII keys are not allowed\n");
             continue;
         }
@@ -203,7 +197,7 @@ int ff_ape_write_tag(AVFormatContext *s)
         avio_wl32(dyn_bc, val_len);            // value length
         avio_wl32(dyn_bc, 0);                  // item flags
         avio_put_str(dyn_bc, e->key);          // key
-        avio_write(dyn_bc, (const unsigned char *) e->value, val_len); // value
+        avio_write(dyn_bc, (const unsigned char *)e->value, val_len); // value
         count++;
     }
     if (!count)
@@ -215,7 +209,7 @@ int ff_ape_write_tag(AVFormatContext *s)
     size += 20;
 
     // header
-    avio_write(s->pb, (const unsigned char *) "APETAGEX", 8);   // id
+    avio_write(s->pb, (const unsigned char *)"APETAGEX", 8);   // id
     avio_wl32(s->pb, APE_TAG_VERSION);  // version
     avio_wl32(s->pb, size);
     avio_wl32(s->pb, count);
@@ -223,7 +217,7 @@ int ff_ape_write_tag(AVFormatContext *s)
     avio_write(s->pb, dyn_buf, size - 20);
 
     // footer
-    avio_write(s->pb, (const unsigned char *) "APETAGEX", 8);   // id
+    avio_write(s->pb, (const unsigned char *)"APETAGEX", 8);   // id
     avio_wl32(s->pb, APE_TAG_VERSION);  // version
     avio_wl32(s->pb, size);             // size
     avio_wl32(s->pb, count);            // tag count

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/asf.h

@@ -43,6 +43,7 @@ typedef struct ASFStream {
     int timestamp;
     int64_t duration;
     int skip_to_key;
+    int pkt_clean;
 
     int ds_span;                /* descrambling  */
     int ds_packet_size;
@@ -188,6 +189,4 @@ extern const AVMetadataConv ff_asf_metadata_conv[];
 
 extern AVInputFormat ff_asf_demuxer;
 
-void ff_put_guid(AVIOContext *s, const ff_asf_guid *g);
-
 #endif /* AVFORMAT_ASF_H */

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/asfdec.c

@@ -1120,8 +1120,7 @@ static int asf_parse_packet(AVFormatContext *s, AVIOContext *pb, AVPacket *pkt)
         if (url_feof(pb))
             return AVERROR_EOF;
 
-        if (asf->packet_size_left < FRAME_HEADER_SIZE ||
+        if (asf->packet_size_left < FRAME_HEADER_SIZE) {
-            asf->packet_segments < 1) {
             int ret = asf->packet_size_left + asf->packet_padsize;
 
             assert(ret >= 0);
@@ -1136,7 +1135,7 @@ static int asf_parse_packet(AVFormatContext *s, AVIOContext *pb, AVPacket *pkt)
         }
         if (asf->packet_time_start == 0) {
             if (asf_read_frame_header(s, pb) < 0) {
-                asf->packet_segments = 0;
+                asf->packet_time_start = asf->packet_segments = 0;
                 continue;
             }
             if (asf->stream_index < 0 ||
@@ -1158,6 +1157,16 @@ static int asf_parse_packet(AVFormatContext *s, AVIOContext *pb, AVPacket *pkt)
         asf_st = asf->asf_st;
         av_assert0(asf_st);
 
+        if (!asf_st->frag_offset && asf->packet_frag_offset) {
+            av_dlog(s, "skipping asf data pkt with fragment offset for "
+                    "stream:%d, expected:%d but got %d from pkt)\n",
+                    asf->stream_index, asf_st->frag_offset,
+                    asf->packet_frag_offset);
+            avio_skip(pb, asf->packet_frag_size);
+            asf->packet_size_left -= asf->packet_frag_size;
+            continue;
+        }
+
         if (asf->packet_replic_size == 1) {
             // frag_offset is here used as the beginning timestamp
             asf->packet_frag_timestamp = asf->packet_time_start;
@@ -1190,6 +1199,7 @@ static int asf_parse_packet(AVFormatContext *s, AVIOContext *pb, AVPacket *pkt)
             asf_st->pkt.dts          = asf->packet_frag_timestamp - asf->hdr.preroll;
             asf_st->pkt.stream_index = asf->stream_index;
             asf_st->pkt.pos          = asf_st->packet_pos = asf->packet_pos;
+            asf_st->pkt_clean        = 0;
 
             if (asf_st->pkt.data && asf_st->palette_changed) {
                 uint8_t *pal;
@@ -1230,6 +1240,11 @@ static int asf_parse_packet(AVFormatContext *s, AVIOContext *pb, AVPacket *pkt)
             continue;
         }
 
+        if (asf->packet_frag_offset != asf_st->frag_offset && !asf_st->pkt_clean) {
+            memset(asf_st->pkt.data + asf_st->frag_offset, 0, asf_st->pkt.size - asf_st->frag_offset);
+            asf_st->pkt_clean = 1;
+        }
+
         ret = avio_read(pb, asf_st->pkt.data + asf->packet_frag_offset,
                         asf->packet_frag_size);
         if (ret != asf->packet_frag_size) {
@@ -1347,7 +1362,6 @@ static void asf_reset_header(AVFormatContext *s)
     int i;
 
     asf->packet_size_left      = 0;
-    asf->packet_segments       = 0;
     asf->packet_flags          = 0;
     asf->packet_property       = 0;
     asf->packet_timestamp      = 0;
@@ -1450,30 +1464,30 @@ static int64_t asf_read_pts(AVFormatContext *s, int stream_index,
     return pts;
 }
 
-static void asf_build_simple_index(AVFormatContext *s, int stream_index)
+static int asf_build_simple_index(AVFormatContext *s, int stream_index)
 {
     ff_asf_guid g;
     ASFContext *asf     = s->priv_data;
     int64_t current_pos = avio_tell(s->pb);
+    int ret = 0;
 
-    if(avio_seek(s->pb, asf->data_object_offset + asf->data_object_size, SEEK_SET) < 0) {
+    if((ret = avio_seek(s->pb, asf->data_object_offset + asf->data_object_size, SEEK_SET)) < 0) {
-        asf->index_read= -1;
+        return ret;
-        return;
     }
 
-    ff_get_guid(s->pb, &g);
+    if ((ret = ff_get_guid(s->pb, &g)) < 0)
+        goto end;
 
     /* the data object can be followed by other top-level objects,
      * skip them until the simple index object is reached */
     while (ff_guidcmp(&g, &ff_asf_simple_index_header)) {
         int64_t gsize = avio_rl64(s->pb);
         if (gsize < 24 || url_feof(s->pb)) {
-            avio_seek(s->pb, current_pos, SEEK_SET);
+            goto end;
-            asf->index_read= -1;
-            return;
         }
         avio_skip(s->pb, gsize - 24);
-        ff_get_guid(s->pb, &g);
+        if ((ret = ff_get_guid(s->pb, &g)) < 0)
+            goto end;
     }
 
     {
@@ -1481,7 +1495,8 @@ static void asf_build_simple_index(AVFormatContext *s, int stream_index)
         int pct, ict;
         int i;
         int64_t av_unused gsize = avio_rl64(s->pb);
-        ff_get_guid(s->pb, &g);
+        if ((ret = ff_get_guid(s->pb, &g)) < 0)
+            goto end;
         itime = avio_rl64(s->pb);
         pct   = avio_rl32(s->pb);
         ict   = avio_rl32(s->pb);
@@ -1504,7 +1519,12 @@ static void asf_build_simple_index(AVFormatContext *s, int stream_index)
         }
         asf->index_read = ict > 1;
     }
+end:
+//     if (url_feof(s->pb)) {
+//         ret = 0;
+//     }
     avio_seek(s->pb, current_pos, SEEK_SET);
+    return ret;
 }
 
 static int asf_read_seek(AVFormatContext *s, int stream_index,
@@ -1512,6 +1532,7 @@ static int asf_read_seek(AVFormatContext *s, int stream_index,
 {
     ASFContext *asf = s->priv_data;
     AVStream *st    = s->streams[stream_index];
+    int ret = 0;
 
     if (s->packet_size <= 0)
         return -1;
@@ -1525,10 +1546,20 @@ static int asf_read_seek(AVFormatContext *s, int stream_index,
             return ret;
     }
 
-    if (!asf->index_read)
+    /* explicitly handle the case of seeking to 0 */
-        asf_build_simple_index(s, stream_index);
+    if (!pts) {
+        asf_reset_header(s);
+        avio_seek(s->pb, s->data_offset, SEEK_SET);
+        return 0;
+    }
 
-    if ((asf->index_read > 0 && st->index_entries)) {
+    if (!asf->index_read) {
+        ret = asf_build_simple_index(s, stream_index);
+        if (ret < 0)
+            asf->index_read = -1;
+    }
+
+    if (asf->index_read > 0 && st->index_entries) {
         int index = av_index_search_timestamp(st, pts, flags);
         if (index >= 0) {
             /* find the position */

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/astdec.c

@@ -19,23 +19,23 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include "config.h"
-
 #include "libavutil/channel_layout.h"
 #include "libavutil/intreadwrite.h"
-#include "libavutil/internal.h"
 #include "avformat.h"
 #include "internal.h"
 #include "ast.h"
 
 static int ast_probe(AVProbeData *p)
 {
-    if (AV_RL32(p->buf) == MKTAG('S','T','R','M') &&
+    if (AV_RL32(p->buf) != MKTAG('S','T','R','M'))
-        AV_RB16(p->buf + 10) &&
+        return 0;
-        AV_RB16(p->buf + 12) &&
+
-        AV_RB32(p->buf + 16))
+    if (!AV_RB16(p->buf + 10) ||
-        return AVPROBE_SCORE_MAX / 3 * 2;
+        !AV_RB16(p->buf + 12) || AV_RB16(p->buf + 12) > 256 ||
-    return 0;
+        !AV_RB32(p->buf + 16) || AV_RB32(p->buf + 16) > 8*48000)
+        return AVPROBE_SCORE_MAX / 8;
+
+    return AVPROBE_SCORE_MAX / 3 * 2;
 }
 
 static int ast_read_header(AVFormatContext *s)
@@ -112,7 +112,7 @@ static int ast_read_packet(AVFormatContext *s, AVPacket *pkt)
 
 AVInputFormat ff_ast_demuxer = {
     .name           = "ast",
-    .long_name      = NULL_IF_CONFIG_SMALL("AST (Audio Stream)"),
+    .long_name      = "AST (Audio Stream)",
     .read_probe     = ast_probe,
     .read_header    = ast_read_header,
     .read_packet    = ast_read_packet,

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/avformat.h

@@ -173,6 +173,58 @@
  *
  * @defgroup lavf_encoding Muxing
  * @{
+ * Muxers take encoded data in the form of @ref AVPacket "AVPackets" and write
+ * it into files or other output bytestreams in the specified container format.
+ *
+ * The main API functions for muxing are avformat_write_header() for writing the
+ * file header, av_write_frame() / av_interleaved_write_frame() for writing the
+ * packets and av_write_trailer() for finalizing the file.
+ *
+ * At the beginning of the muxing process, the caller must first call
+ * avformat_alloc_context() to create a muxing context. The caller then sets up
+ * the muxer by filling the various fields in this context:
+ *
+ * - The @ref AVFormatContext.oformat "oformat" field must be set to select the
+ *   muxer that will be used.
+ * - Unless the format is of the AVFMT_NOFILE type, the @ref AVFormatContext.pb
+ *   "pb" field must be set to an opened IO context, either returned from
+ *   avio_open2() or a custom one.
+ * - Unless the format is of the AVFMT_NOSTREAMS type, at least one stream must
+ *   be created with the avformat_new_stream() function. The caller should fill
+ *   the @ref AVStream.codec "stream codec context" information, such as the
+ *   codec @ref AVCodecContext.codec_type "type", @ref AVCodecContext.codec_id
+ *   "id" and other parameters (e.g. width / height, the pixel or sample format,
+ *   etc.) as known. The @ref AVCodecContext.time_base "codec timebase" should
+ *   be set to the timebase that the caller desires to use for this stream (note
+ *   that the timebase actually used by the muxer can be different, as will be
+ *   described later).
+ * - The caller may fill in additional information, such as @ref
+ *   AVFormatContext.metadata "global" or @ref AVStream.metadata "per-stream"
+ *   metadata, @ref AVFormatContext.chapters "chapters", @ref
+ *   AVFormatContext.programs "programs", etc. as described in the
+ *   AVFormatContext documentation. Whether such information will actually be
+ *   stored in the output depends on what the container format and the muxer
+ *   support.
+ *
+ * When the muxing context is fully set up, the caller must call
+ * avformat_write_header() to initialize the muxer internals and write the file
+ * header. Whether anything actually is written to the IO context at this step
+ * depends on the muxer, but this function must always be called. Any muxer
+ * private options must be passed in the options parameter to this function.
+ *
+ * The data is then sent to the muxer by repeatedly calling av_write_frame() or
+ * av_interleaved_write_frame() (consult those functions' documentation for
+ * discussion on the difference between them; only one of them may be used with
+ * a single muxing context, they should not be mixed). Do note that the timing
+ * information on the packets sent to the muxer must be in the corresponding
+ * AVStream's timebase. That timebase is set by the muxer (in the
+ * avformat_write_header() step) and may be different from the timebase the
+ * caller set on the codec context.
+ *
+ * Once all the data has been written, the caller must call av_write_trailer()
+ * to flush any buffered packets and finalize the output file, then close the IO
+ * context (if any) and finally free the muxing context with
+ * avformat_free_context().
  * @}
  *
  * @defgroup lavf_io I/O Read/Write
@@ -217,6 +269,7 @@
 
 struct AVFormatContext;
 
+struct AVDeviceInfoList;
 
 /**
  * @defgroup metadata_api Public Metadata API
@@ -298,6 +351,7 @@ struct AVFormatContext;
  * Allocate and read the payload of a packet and initialize its
  * fields with default values.
  *
+ * @param s    associated IO context
  * @param pkt packet
  * @param size desired payload size
  * @return >0 (read size) if OK, AVERROR_xxx otherwise
@@ -313,6 +367,7 @@ int av_get_packet(AVIOContext *s, AVPacket *pkt, int size);
  * when there is no reasonable way to know (an upper bound of)
  * the final size.
  *
+ * @param s    associated IO context
  * @param pkt packet
  * @param size amount of data to read
  * @return >0 (read size) if OK, AVERROR_xxx otherwise, previous data
@@ -346,6 +401,8 @@ typedef struct AVProbeData {
 } AVProbeData;
 
 #define AVPROBE_SCORE_RETRY (AVPROBE_SCORE_MAX/4)
+#define AVPROBE_SCORE_STREAM_RETRY (AVPROBE_SCORE_MAX/4-1)
+
 #define AVPROBE_SCORE_EXTENSION  50 ///< score for file extension
 #define AVPROBE_SCORE_MAX       100 ///< maximum score
 
@@ -461,6 +518,27 @@ typedef struct AVOutputFormat {
 
     void (*get_output_timestamp)(struct AVFormatContext *s, int stream,
                                  int64_t *dts, int64_t *wall);
+    /**
+     * Allows sending messages from application to device.
+     */
+    int (*control_message)(struct AVFormatContext *s, int type,
+                           void *data, size_t data_size);
+
+    /**
+     * Write an uncoded AVFrame.
+     *
+     * See av_write_uncoded_frame() for details.
+     *
+     * The library will free *frame afterwards, but the muxer can prevent it
+     * by setting the pointer to NULL.
+     */
+    int (*write_uncoded_frame)(struct AVFormatContext *, int stream_index,
+                               AVFrame **frame, unsigned flags);
+    /**
+     * Returns device list with it properties.
+     * @see avdevice_list_devices() for more details.
+     */
+    int (*get_device_list)(struct AVFormatContext *s, struct AVDeviceInfoList *device_list);
 } AVOutputFormat;
 /**
  * @}
@@ -589,6 +667,12 @@ typedef struct AVInputFormat {
      * Active streams are all streams that have AVStream.discard < AVDISCARD_ALL.
      */
     int (*read_seek2)(struct AVFormatContext *s, int stream_index, int64_t min_ts, int64_t ts, int64_t max_ts, int flags);
+
+    /**
+     * Returns device list with it properties.
+     * @see avdevice_list_devices() for more details.
+     */
+    int (*get_device_list)(struct AVFormatContext *s, struct AVDeviceInfoList *device_list);
 } AVInputFormat;
 /**
  * @}
@@ -764,6 +848,7 @@ typedef struct AVStream {
         int64_t last_dts;
         int64_t duration_gcd;
         int duration_count;
+        int64_t rfps_duration_sum;
         double (*duration_error)[2][MAX_STD_TIMEBASES];
         int64_t codec_info_duration;
         int64_t codec_info_duration_fields;
@@ -783,6 +868,11 @@ typedef struct AVStream {
 
     int pts_wrap_bits; /**< number of bits in pts (used for wrapping control) */
 
+#if FF_API_REFERENCE_DTS
+    /* a hack to keep ABI compatibility for ffmpeg and other applications, which accesses parser even
+     * though it should not */
+    int64_t do_not_use;
+#endif
     // Timestamp generation support:
     /**
      * Timestamp corresponding to the last dts sync point.
@@ -791,7 +881,6 @@ typedef struct AVStream {
      * a DTS is received from the underlying container. Otherwise set to
      * AV_NOPTS_VALUE by default.
      */
-    int64_t reference_dts;
     int64_t first_dts;
     int64_t cur_dts;
     int64_t last_IP_pts;
@@ -896,6 +985,24 @@ typedef struct AVStream {
      */
     int pts_wrap_behavior;
 
+    /**
+     * Internal data to prevent doing update_initial_durations() twice
+     */
+    int update_initial_durations_done;
+
+    /**
+     * Internal data to generate dts from pts
+     */
+    int64_t pts_reorder_error[MAX_REORDER_DELAY+1];
+    uint8_t pts_reorder_error_count[MAX_REORDER_DELAY+1];
+
+    /**
+     * Internal data to analyze DTS and detect faulty mpeg streams
+     */
+    int64_t last_dts_for_order_check;
+    uint8_t dts_ordered;
+    uint8_t dts_misordered;
+
 } AVStream;
 
 AVRational av_stream_get_r_frame_rate(const AVStream *s);
@@ -946,6 +1053,13 @@ typedef struct AVChapter {
 } AVChapter;
 
 
+/**
+ * Callback used by devices to communicate with application.
+ */
+typedef int (*av_format_control_message)(struct AVFormatContext *s, int type,
+                                         void *data, size_t data_size);
+
+
 /**
  * The duration of a video can be estimated through various ways, and this enum can be used
  * to know how the duration was estimated.
@@ -956,6 +1070,8 @@ enum AVDurationEstimationMethod {
     AVFMT_DURATION_FROM_BITRATE ///< Duration estimated from bitrate (less accurate)
 };
 
+typedef struct AVFormatInternal AVFormatInternal;
+
 /**
  * Format I/O context.
  * New fields can be added to the end with minor version bumps.
@@ -966,32 +1082,41 @@ enum AVDurationEstimationMethod {
  */
 typedef struct AVFormatContext {
     /**
-     * A class for logging and AVOptions. Set by avformat_alloc_context().
+     * A class for logging and @ref avoptions. Set by avformat_alloc_context().
      * Exports (de)muxer private options if they exist.
      */
     const AVClass *av_class;
 
     /**
-     * Can only be iformat or oformat, not both at the same time.
+     * The input container format.
      *
-     * decoding: set by avformat_open_input().
+     * Demuxing only, set by avformat_open_input().
-     * encoding: set by the user.
      */
     struct AVInputFormat *iformat;
+
+    /**
+     * The output container format.
+     *
+     * Muxing only, must be set by the caller before avformat_write_header().
+     */
     struct AVOutputFormat *oformat;
 
     /**
      * Format private data. This is an AVOptions-enabled struct
      * if and only if iformat/oformat.priv_class is not NULL.
+     *
+     * - muxing: set by avformat_write_header()
+     * - demuxing: set by avformat_open_input()
      */
     void *priv_data;
 
     /**
      * I/O context.
      *
-     * decoding: either set by the user before avformat_open_input() (then
+     * - demuxing: either set by the user before avformat_open_input() (then
-     * the user must close it manually) or set by avformat_open_input().
+     *             the user must close it manually) or set by avformat_open_input().
-     * encoding: set by the user.
+     * - muxing: set by the user before avformat_write_header(). The caller must
+     *           take care of closing / freeing the IO context.
      *
      * Do NOT set this field if AVFMT_NOFILE flag is set in
      * iformat/oformat.flags. In such a case, the (de)muxer will handle
@@ -1002,37 +1127,54 @@ typedef struct AVFormatContext {
     /* stream info */
     int ctx_flags; /**< Format-specific flags, see AVFMTCTX_xx */
 
+    /**
+     * Number of elements in AVFormatContext.streams.
+     *
+     * Set by avformat_new_stream(), must not be modified by any other code.
+     */
+    unsigned int nb_streams;
     /**
      * A list of all streams in the file. New streams are created with
      * avformat_new_stream().
      *
-     * decoding: streams are created by libavformat in avformat_open_input().
+     * - demuxing: streams are created by libavformat in avformat_open_input().
-     * If AVFMTCTX_NOHEADER is set in ctx_flags, then new streams may also
+     *             If AVFMTCTX_NOHEADER is set in ctx_flags, then new streams may also
-     * appear in av_read_frame().
+     *             appear in av_read_frame().
-     * encoding: streams are created by the user before avformat_write_header().
+     * - muxing: streams are created by the user before avformat_write_header().
+     *
+     * Freed by libavformat in avformat_free_context().
      */
-    unsigned int nb_streams;
     AVStream **streams;
 
-    char filename[1024]; /**< input or output filename */
+    /**
+     * input or output filename
+     *
+     * - demuxing: set by avformat_open_input()
+     * - muxing: may be set by the caller before avformat_write_header()
+     */
+    char filename[1024];
 
     /**
-     * Decoding: position of the first frame of the component, in
+     * Position of the first frame of the component, in
      * AV_TIME_BASE fractional seconds. NEVER set this value directly:
      * It is deduced from the AVStream values.
+     *
+     * Demuxing only, set by libavformat.
      */
     int64_t start_time;
 
     /**
-     * Decoding: duration of the stream, in AV_TIME_BASE fractional
+     * Duration of the stream, in AV_TIME_BASE fractional
      * seconds. Only set this value if you know none of the individual stream
      * durations and also do not set any of them. This is deduced from the
      * AVStream values if not set.
+     *
+     * Demuxing only, set by libavformat.
      */
     int64_t duration;
 
     /**
-     * Decoding: total stream bitrate in bit/s, 0 if not
+     * Total stream bitrate in bit/s, 0 if not
      * available. Never set it directly if the file_size and the
      * duration are known as FFmpeg can compute it automatically.
      */
@@ -1058,13 +1200,16 @@ typedef struct AVFormatContext {
 #define AVFMT_FLAG_KEEP_SIDE_DATA 0x40000 ///< Don't merge side data but keep it separate.
 
     /**
-     * decoding: size of data to probe; encoding: unused.
+     * Maximum size of the data read from input for determining
+     * the input container format.
+     * Demuxing only, set by the caller before avformat_open_input().
      */
     unsigned int probesize;
 
     /**
-     * decoding: maximum time (in AV_TIME_BASE units) during which the input should
+     * Maximum duration (in AV_TIME_BASE units) of the data read
-     * be analyzed in avformat_find_stream_info().
+     * from input in avformat_find_stream_info().
+     * Demuxing only, set by the caller before avformat_find_stream_info().
      */
     int max_analyze_duration;
 
@@ -1099,8 +1244,8 @@ typedef struct AVFormatContext {
      * accurate seeking (depends on demuxer).
      * Demuxers for which a full in-memory index is mandatory will ignore
      * this.
-     * muxing  : unused
+     * - muxing: unused
-     * demuxing: set by user
+     * - demuxing: set by user
      */
     unsigned int max_index_size;
 
@@ -1118,41 +1263,49 @@ typedef struct AVFormatContext {
      * in the trailer.  To write chapters in the trailer, nb_chapters
      * must be zero when write_header is called and non-zero when
      * write_trailer is called.
-     * muxing  : set by user
+     * - muxing: set by user
-     * demuxing: set by libavformat
+     * - demuxing: set by libavformat
      */
     unsigned int nb_chapters;
     AVChapter **chapters;
 
+    /**
+     * Metadata that applies to the whole file.
+     *
+     * - demuxing: set by libavformat in avformat_open_input()
+     * - muxing: may be set by the caller before avformat_write_header()
+     *
+     * Freed by libavformat in avformat_free_context().
+     */
     AVDictionary *metadata;
 
     /**
      * Start time of the stream in real world time, in microseconds
-     * since the unix epoch (00:00 1st January 1970). That is, pts=0
+     * since the Unix epoch (00:00 1st January 1970). That is, pts=0 in the
-     * in the stream was captured at this real world time.
+     * stream was captured at this real world time.
-     * - encoding: Set by user.
+     * Muxing only, set by the caller before avformat_write_header().
-     * - decoding: Unused.
      */
     int64_t start_time_realtime;
 
     /**
-     * decoding: number of frames used to probe fps
+     * The number of frames used for determining the framerate in
+     * avformat_find_stream_info().
+     * Demuxing only, set by the caller before avformat_find_stream_info().
      */
     int fps_probe_size;
 
     /**
      * Error recognition; higher values will detect more errors but may
      * misdetect some more or less valid parts as errors.
-     * - encoding: unused
+     * Demuxing only, set by the caller before avformat_open_input().
-     * - decoding: Set by user.
      */
     int error_recognition;
 
     /**
      * Custom interrupt callbacks for the I/O layer.
      *
-     * decoding: set by the user before avformat_open_input().
+     * demuxing: set by the user before avformat_open_input().
-     * encoding: set by the user before avformat_write_header()
+     * muxing: set by the user before avformat_write_header()
      * (mainly useful for AVFMT_NOFILE formats). The callback
      * should also be passed to avio_open2() if it's used to
      * open the file.
@@ -1165,6 +1318,24 @@ typedef struct AVFormatContext {
     int debug;
 #define FF_FDEBUG_TS        0x0001
 
+    /**
+     * Maximum buffering duration for interleaving.
+     *
+     * To ensure all the streams are interleaved correctly,
+     * av_interleaved_write_frame() will wait until it has at least one packet
+     * for each stream before actually writing any packets to the output file.
+     * When some streams are "sparse" (i.e. there are large gaps between
+     * successive packets), this can result in excessive buffering.
+     *
+     * This field specifies the maximum difference between the timestamps of the
+     * first and the last packet in the muxing queue, above which libavformat
+     * will output a packet regardless of whether it has queued a packet for all
+     * the streams.
+     *
+     * Muxing only, set by the caller before avformat_write_header().
+     */
+    int64_t max_interleave_delta;
+
     /**
      * Transport stream id.
      * This will be moved into demuxer private options. Thus no API/ABI compatibility
@@ -1239,14 +1410,14 @@ typedef struct AVFormatContext {
     /**
      * Correct single timestamp overflows
      * - encoding: unused
-     * - decoding: Set by user via AVOPtions (NO direct access)
+     * - decoding: Set by user via AVOptions (NO direct access)
      */
     unsigned int correct_ts_overflow;
 
     /**
      * Force seeking to any (also non key) frames.
      * - encoding: unused
-     * - decoding: Set by user via AVOPtions (NO direct access)
+     * - decoding: Set by user via AVOptions (NO direct access)
      */
     int seek2any;
 
@@ -1316,6 +1487,12 @@ typedef struct AVFormatContext {
      */
     AVRational offset_timebase;
 
+    /**
+     * An opaque field for libavformat internal usage.
+     * Must not be accessed in any way by callers.
+     */
+    AVFormatInternal *internal;
+
     /**
      * IO repositioned flag.
      * This is set by avformat when the underlaying IO context read pointer
@@ -1323,9 +1500,70 @@ typedef struct AVFormatContext {
      * Demuxers can use the flag to detect such changes.
      */
     int io_repositioned;
+
+    /**
+     * Forced video codec.
+     * This allows forcing a specific decoder, even when there are multiple with
+     * the same codec_id.
+     * Demuxing: Set by user via av_format_set_video_codec (NO direct access).
+     */
+    AVCodec *video_codec;
+
+    /**
+     * Forced audio codec.
+     * This allows forcing a specific decoder, even when there are multiple with
+     * the same codec_id.
+     * Demuxing: Set by user via av_format_set_audio_codec (NO direct access).
+     */
+    AVCodec *audio_codec;
+
+    /**
+     * Forced subtitle codec.
+     * This allows forcing a specific decoder, even when there are multiple with
+     * the same codec_id.
+     * Demuxing: Set by user via av_format_set_subtitle_codec (NO direct access).
+     */
+    AVCodec *subtitle_codec;
+
+    /**
+     * Number of bytes to be written as padding in a metadata header.
+     * Demuxing: Unused.
+     * Muxing: Set by user via av_format_set_metadata_header_padding.
+     */
+    int metadata_header_padding;
+
+    /**
+     * User data.
+     * This is a place for some private data of the user.
+     * Mostly usable with control_message_cb or any future callbacks in device's context.
+     */
+    void *opaque;
+
+    /**
+     * Callback used by devices to communicate with application.
+     */
+    av_format_control_message control_message_cb;
+
+    /**
+     * Output timestamp offset, in microseconds.
+     * Muxing: set by user via AVOptions (NO direct access)
+     */
+    int64_t output_ts_offset;
 } AVFormatContext;
 
 int av_format_get_probe_score(const AVFormatContext *s);
+AVCodec * av_format_get_video_codec(const AVFormatContext *s);
+void      av_format_set_video_codec(AVFormatContext *s, AVCodec *c);
+AVCodec * av_format_get_audio_codec(const AVFormatContext *s);
+void      av_format_set_audio_codec(AVFormatContext *s, AVCodec *c);
+AVCodec * av_format_get_subtitle_codec(const AVFormatContext *s);
+void      av_format_set_subtitle_codec(AVFormatContext *s, AVCodec *c);
+int       av_format_get_metadata_header_padding(const AVFormatContext *s);
+void      av_format_set_metadata_header_padding(AVFormatContext *s, int c);
+void *    av_format_get_opaque(const AVFormatContext *s);
+void      av_format_set_opaque(AVFormatContext *s, void *opaque);
+av_format_control_message av_format_get_control_message_cb(const AVFormatContext *s);
+void      av_format_set_control_message_cb(AVFormatContext *s, av_format_control_message callback);
 
 /**
  * Returns the method used to set ctx->duration.
@@ -1439,6 +1677,7 @@ const AVClass *avformat_get_class(void);
  * User is required to call avcodec_close() and avformat_free_context() to
  * clean up the allocation by avformat_new_stream().
  *
+ * @param s media file handle
  * @param c If non-NULL, the AVCodecContext corresponding to the new stream
  * will be initialized to use this codec. This is needed for e.g. codec-specific
  * defaults to be set, so codec should be provided if it is known.
@@ -1496,6 +1735,7 @@ AVInputFormat *av_find_input_format(const char *short_name);
 /**
  * Guess the file format.
  *
+ * @param pd        data to be probed
  * @param is_opened Whether the file is already opened; determines whether
  *                  demuxers with or without AVFMT_NOFILE are probed.
  */
@@ -1504,6 +1744,7 @@ AVInputFormat *av_probe_input_format(AVProbeData *pd, int is_opened);
 /**
  * Guess the file format.
  *
+ * @param pd        data to be probed
  * @param is_opened Whether the file is already opened; determines whether
  *                  demuxers with or without AVFMT_NOFILE are probed.
  * @param score_max A probe score larger that this is required to accept a
@@ -1657,7 +1898,7 @@ int av_find_best_stream(AVFormatContext *ic,
                         enum AVMediaType type,
                         int wanted_stream_nb,
                         int related_stream,
-                        const AVCodec **decoder_ret,
+                        AVCodec **decoder_ret,
                         int flags);
 
 #if FF_API_READ_PACKET
@@ -1707,6 +1948,8 @@ int av_read_frame(AVFormatContext *s, AVPacket *pkt);
 /**
  * Seek to the keyframe at timestamp.
  * 'timestamp' in 'stream_index'.
+ *
+ * @param s media file handle
  * @param stream_index If stream_index is (-1), a default
  * stream is selected, and timestamp is automatically converted
  * from AV_TIME_BASE units to the stream specific time_base.
@@ -1734,6 +1977,7 @@ int av_seek_frame(AVFormatContext *s, int stream_index, int64_t timestamp,
  * keyframes (this may not be supported by all demuxers).
  * If flags contain AVSEEK_FLAG_BACKWARD, it is ignored.
  *
+ * @param s media file handle
  * @param stream_index index of the stream which is used as time base reference
  * @param min_ts smallest acceptable timestamp
  * @param ts target timestamp
@@ -1833,49 +2077,108 @@ int avformat_write_header(AVFormatContext *s, AVDictionary **options);
 /**
  * Write a packet to an output media file.
  *
- * The packet shall contain one audio or video frame.
+ * This function passes the packet directly to the muxer, without any buffering
- * The packet must be correctly interleaved according to the container
+ * or reordering. The caller is responsible for correctly interleaving the
- * specification, if not then av_interleaved_write_frame must be used.
+ * packets if the format requires it. Callers that want libavformat to handle
+ * the interleaving should call av_interleaved_write_frame() instead of this
+ * function.
  *
  * @param s media file handle
- * @param pkt The packet, which contains the stream_index, buf/buf_size,
+ * @param pkt The packet containing the data to be written. Note that unlike
- *            dts/pts, ...
+ *            av_interleaved_write_frame(), this function does not take
- *            This can be NULL (at any time, not just at the end), in
+ *            ownership of the packet passed to it (though some muxers may make
- *            order to immediately flush data buffered within the muxer,
+ *            an internal reference to the input packet).
- *            for muxers that buffer up data internally before writing it
+ *            <br>
- *            to the output.
+ *            This parameter can be NULL (at any time, not just at the end), in
+ *            order to immediately flush data buffered within the muxer, for
+ *            muxers that buffer up data internally before writing it to the
+ *            output.
+ *            <br>
+ *            Packet's @ref AVPacket.stream_index "stream_index" field must be
+ *            set to the index of the corresponding stream in @ref
+ *            AVFormatContext.streams "s->streams". It is very strongly
+ *            recommended that timing information (@ref AVPacket.pts "pts", @ref
+ *            AVPacket.dts "dts", @ref AVPacket.duration "duration") is set to
+ *            correct values.
  * @return < 0 on error, = 0 if OK, 1 if flushed and there is no more data to flush
+ *
+ * @see av_interleaved_write_frame()
  */
 int av_write_frame(AVFormatContext *s, AVPacket *pkt);
 
 /**
  * Write a packet to an output media file ensuring correct interleaving.
  *
- * The packet must contain one audio or video frame.
+ * This function will buffer the packets internally as needed to make sure the
- * If the packets are already correctly interleaved, the application should
+ * packets in the output file are properly interleaved in the order of
- * call av_write_frame() instead as it is slightly faster. It is also important
+ * increasing dts. Callers doing their own interleaving should call
- * to keep in mind that completely non-interleaved input will need huge amounts
+ * av_write_frame() instead of this function.
- * of memory to interleave with this, so it is preferable to interleave at the
- * demuxer level.
  *
  * @param s media file handle
- * @param pkt The packet containing the data to be written. pkt->buf must be set
+ * @param pkt The packet containing the data to be written.
- * to a valid AVBufferRef describing the packet data. Libavformat takes
+ *            <br>
- * ownership of this reference and will unref it when it sees fit. The caller
+ *            If the packet is reference-counted, this function will take
- * must not access the data through this reference after this function returns.
+ *            ownership of this reference and unreference it later when it sees
- * This can be NULL (at any time, not just at the end), to flush the
+ *            fit.
- * interleaving queues.
+ *            The caller must not access the data through this reference after
- * Packet's @ref AVPacket.stream_index "stream_index" field must be set to the
+ *            this function returns. If the packet is not reference-counted,
- * index of the corresponding stream in @ref AVFormatContext.streams
+ *            libavformat will make a copy.
- * "s.streams".
+ *            <br>
- * It is very strongly recommended that timing information (@ref AVPacket.pts
+ *            This parameter can be NULL (at any time, not just at the end), to
- * "pts", @ref AVPacket.dts "dts" @ref AVPacket.duration "duration") is set to
+ *            flush the interleaving queues.
- * correct values.
+ *            <br>
+ *            Packet's @ref AVPacket.stream_index "stream_index" field must be
+ *            set to the index of the corresponding stream in @ref
+ *            AVFormatContext.streams "s->streams". It is very strongly
+ *            recommended that timing information (@ref AVPacket.pts "pts", @ref
+ *            AVPacket.dts "dts", @ref AVPacket.duration "duration") is set to
+ *            correct values.
  *
- * @return 0 on success, a negative AVERROR on error.
+ * @return 0 on success, a negative AVERROR on error. Libavformat will always
+ *         take care of freeing the packet, even if this function fails.
+ *
+ * @see av_write_frame(), AVFormatContext.max_interleave_delta
  */
 int av_interleaved_write_frame(AVFormatContext *s, AVPacket *pkt);
 
+/**
+ * Write a uncoded frame to an output media file.
+ *
+ * The frame must be correctly interleaved according to the container
+ * specification; if not, then av_interleaved_write_frame() must be used.
+ *
+ * See av_interleaved_write_frame() for details.
+ */
+int av_write_uncoded_frame(AVFormatContext *s, int stream_index,
+                           AVFrame *frame);
+
+/**
+ * Write a uncoded frame to an output media file.
+ *
+ * If the muxer supports it, this function allows to write an AVFrame
+ * structure directly, without encoding it into a packet.
+ * It is mostly useful for devices and similar special muxers that use raw
+ * video or PCM data and will not serialize it into a byte stream.
+ *
+ * To test whether it is possible to use it with a given muxer and stream,
+ * use av_write_uncoded_frame_query().
+ *
+ * The caller gives up ownership of the frame and must not access it
+ * afterwards.
+ *
+ * @return  >=0 for success, a negative code on error
+ */
+int av_interleaved_write_uncoded_frame(AVFormatContext *s, int stream_index,
+                                       AVFrame *frame);
+
+/**
+ * Test whether a muxer supports uncoded frame.
+ *
+ * @return  >=0 if an uncoded frame can be written to that muxer and stream,
+ *          <0 if not
+ */
+int av_write_uncoded_frame_query(AVFormatContext *s, int stream_index);
+
 /**
  * Write the stream trailer to an output media file and free the
  * file private data.
@@ -1999,6 +2302,7 @@ void av_pkt_dump_log2(void *avcl, int level, AVPacket *pkt, int dump_payload,
  *
  * @param tags list of supported codec_id-codec_tag pairs, as stored
  * in AVInputFormat.codec_tag and AVOutputFormat.codec_tag
+ * @param tag  codec tag to match to a codec ID
  */
 enum AVCodecID av_codec_get_id(const struct AVCodecTag * const *tags, unsigned int tag);
 
@@ -2008,6 +2312,7 @@ enum AVCodecID av_codec_get_id(const struct AVCodecTag * const *tags, unsigned i
  *
  * @param tags list of supported codec_id-codec_tag pairs, as stored
  * in AVInputFormat.codec_tag and AVOutputFormat.codec_tag
+ * @param id   codec ID to match to a codec tag
  */
 unsigned int av_codec_get_tag(const struct AVCodecTag * const *tags, enum AVCodecID id);
 
@@ -2027,6 +2332,9 @@ int av_find_default_stream_index(AVFormatContext *s);
 
 /**
  * Get the index for a specific timestamp.
+ *
+ * @param st        stream that the timestamp belongs to
+ * @param timestamp timestamp to retrieve the index for
  * @param flags if AVSEEK_FLAG_BACKWARD then the returned index will correspond
  *                 to the timestamp which is <= the requested one, if backward
  *                 is 0, then it will be >=
@@ -2123,6 +2431,7 @@ int av_sdp_create(AVFormatContext *ac[], int n_files, char *buf, int size);
  * Return a positive value if the given filename has one of the given
  * extensions, 0 otherwise.
  *
+ * @param filename   file name to check against the given extensions
  * @param extensions a comma-separated list of filename extensions
  */
 int av_match_ext(const char *filename, const char *extensions);
@@ -2130,6 +2439,8 @@ int av_match_ext(const char *filename, const char *extensions);
 /**
  * Test if the given container can store a codec.
  *
+ * @param ofmt           container to check for compatibility
+ * @param codec_id       codec to potentially store in container
  * @param std_compliance standards compliance level, one of FF_COMPLIANCE_*
  *
  * @return 1 if codec with ID codec_id can be stored in ofmt, 0 if it cannot.
@@ -2157,6 +2468,14 @@ const struct AVCodecTag *avformat_get_riff_video_tags(void);
  * @return the table mapping RIFF FourCCs for audio to AVCodecID.
  */
 const struct AVCodecTag *avformat_get_riff_audio_tags(void);
+/**
+ * @return the table mapping MOV FourCCs for video to libavcodec AVCodecID.
+ */
+const struct AVCodecTag *avformat_get_mov_video_tags(void);
+/**
+ * @return the table mapping MOV FourCCs for audio to AVCodecID.
+ */
+const struct AVCodecTag *avformat_get_mov_audio_tags(void);
 
 /**
  * @}

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/avidec.c

@@ -19,6 +19,8 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include <stdint.h>
+
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/bswap.h"
@@ -111,8 +113,8 @@ static int avi_load_index(AVFormatContext *s);
 static int guess_ni_flag(AVFormatContext *s);
 
 #define print_tag(str, tag, size)                        \
-    av_dlog(NULL, "%s: tag=%c%c%c%c size=0x%x\n",        \
+    av_dlog(NULL, "pos:%"PRIX64" %s: tag=%c%c%c%c size=0x%x\n", \
-            str, tag & 0xff,                             \
+            avio_tell(pb), str, tag & 0xff,              \
             (tag >> 8) & 0xff,                           \
             (tag >> 16) & 0xff,                          \
             (tag >> 24) & 0xff,                          \
@@ -135,10 +137,10 @@ static int get_riff(AVFormatContext *s, AVIOContext *pb)
     int i;
 
     /* check RIFF header */
-    avio_read(pb, (unsigned char *)header, 4);
+    avio_read(pb, header, 4);
     avi->riff_end  = avio_rl32(pb); /* RIFF chunk size */
     avi->riff_end += avio_tell(pb); /* RIFF chunk end */
-    avio_read(pb, (unsigned char *)header + 4, 4);
+    avio_read(pb, header + 4, 4);
 
     for (i = 0; avi_headers[i][0]; i++)
         if (!memcmp(header, avi_headers[i], 8))
@@ -303,7 +305,7 @@ static int avi_read_tag(AVFormatContext *s, AVStream *st, uint32_t tag,
     value = av_malloc(size + 1);
     if (!value)
         return AVERROR(ENOMEM);
-    avio_read(pb, (unsigned char *)value, size);
+    avio_read(pb, value, size);
     value[size] = 0;
 
     AV_WL32(key, tag);
@@ -348,7 +350,8 @@ static void avi_read_nikon(AVFormatContext *s, uint64_t end)
                 uint16_t size    = avio_rl16(s->pb);
                 const char *name = NULL;
                 char buffer[64]  = { 0 };
-                size -= avio_read(s->pb, (unsigned char *)buffer,
+                size = FFMIN(size, tag_end - avio_tell(s->pb));
+                size -= avio_read(s->pb, buffer,
                                   FFMIN(size, sizeof(buffer) - 1));
                 switch (tag) {
                 case 0x03:
@@ -444,7 +447,7 @@ static int avi_read_header(AVFormatContext *s)
             size += (size & 1);
             size -= avio_read(pb, date, FFMIN(size, sizeof(date) - 1));
             avio_skip(pb, size);
-            avi_metadata_creation_time(&s->metadata, (char *)date);
+            avi_metadata_creation_time(&s->metadata, date);
             break;
         }
         case MKTAG('d', 'm', 'l', 'h'):
@@ -522,7 +525,8 @@ static int avi_read_header(AVFormatContext *s)
                     avi->dv_demux = avpriv_dv_init_demux(s);
                     if (!avi->dv_demux)
                         goto fail;
-                }
+                } else
+                    goto fail;
                 s->streams[0]->priv_data = ast;
                 avio_skip(pb, 3 * 4);
                 ast->scale = avio_rl32(pb);
@@ -622,6 +626,10 @@ static int avi_read_header(AVFormatContext *s)
                 if (cur_pos < list_end)
                     size = FFMIN(size, list_end - cur_pos);
                 st = s->streams[stream_index];
+                if (st->codec->codec_type != AVMEDIA_TYPE_UNKNOWN) {
+                    avio_skip(pb, size);
+                    break;
+                }
                 switch (codec_type) {
                 case AVMEDIA_TYPE_VIDEO:
                     if (amv_file_format) {
@@ -647,15 +655,8 @@ static int avi_read_header(AVFormatContext *s)
                             st->codec->extradata_size = esize - 10 * 4;
                         } else
                             st->codec->extradata_size =  size - 10 * 4;
-                        st->codec->extradata      = av_malloc(st->codec->extradata_size +
+                        if (ff_get_extradata(st->codec, pb, st->codec->extradata_size) < 0)
-                                                              FF_INPUT_BUFFER_PADDING_SIZE);
-                        if (!st->codec->extradata) {
-                            st->codec->extradata_size = 0;
                             return AVERROR(ENOMEM);
-                        }
-                        avio_read(pb,
-                                  st->codec->extradata,
-                                  st->codec->extradata_size);
                     }
 
                     // FIXME: check if the encoder really did this correctly
@@ -688,6 +689,8 @@ static int avi_read_header(AVFormatContext *s)
                     /* This is needed to get the pict type which is necessary
                      * for generating correct pts. */
                     st->need_parsing = AVSTREAM_PARSE_HEADERS;
+                    if (st->codec->codec_tag == MKTAG('V', 'S', 'S', 'H'))
+                        st->need_parsing = AVSTREAM_PARSE_FULL;
 
                     if (st->codec->codec_tag == 0 && st->codec->height > 0 &&
                         st->codec->extradata_size < 1U << 30) {
@@ -747,9 +750,9 @@ static int avi_read_header(AVFormatContext *s)
                         av_log(s, AV_LOG_DEBUG, "overriding invalid dshow_block_align of %d\n", ast->dshow_block_align);
                         ast->dshow_block_align = 0;
                     }
-                    if ((st->codec->codec_id == AV_CODEC_ID_AAC && ast->dshow_block_align == 1024 && ast->sample_size == 1024) ||
+                    if (st->codec->codec_id == AV_CODEC_ID_AAC && ast->dshow_block_align == 1024 && ast->sample_size == 1024 ||
-                       (st->codec->codec_id == AV_CODEC_ID_AAC && ast->dshow_block_align == 4096 && ast->sample_size == 4096) ||
+                       st->codec->codec_id == AV_CODEC_ID_AAC && ast->dshow_block_align == 4096 && ast->sample_size == 4096 ||
-                       (st->codec->codec_id == AV_CODEC_ID_MP3 && ast->dshow_block_align == 1152 && ast->sample_size == 1152)) {
+                       st->codec->codec_id == AV_CODEC_ID_MP3 && ast->dshow_block_align == 1152 && ast->sample_size == 1152) {
                         av_log(s, AV_LOG_DEBUG, "overriding sample_size\n");
                         ast->sample_size = 0;
                     }
@@ -780,13 +783,8 @@ static int avi_read_header(AVFormatContext *s)
                 st = s->streams[stream_index];
 
                 if (size<(1<<30)) {
-                    st->codec->extradata_size= size;
+                    if (ff_get_extradata(st->codec, pb, size) < 0)
-                    st->codec->extradata= av_mallocz(st->codec->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
-                    if (!st->codec->extradata) {
-                        st->codec->extradata_size= 0;
                         return AVERROR(ENOMEM);
-                    }
-                    avio_read(pb, st->codec->extradata, st->codec->extradata_size);
                 }
 
                 if (st->codec->extradata_size & 1) //FIXME check if the encoder really did this correctly
@@ -906,12 +904,14 @@ fail:
 static int read_gab2_sub(AVStream *st, AVPacket *pkt)
 {
     if (pkt->size >= 7 &&
-        !strcmp((char *)pkt->data, "GAB2") && AV_RL16(pkt->data + 5) == 2) {
+        pkt->size < INT_MAX - AVPROBE_PADDING_SIZE &&
+        !strcmp(pkt->data, "GAB2") && AV_RL16(pkt->data + 5) == 2) {
         uint8_t desc[256];
         int score      = AVPROBE_SCORE_EXTENSION, ret;
         AVIStream *ast = st->priv_data;
         AVInputFormat *sub_demuxer;
         AVRational time_base;
+        int size;
         AVIOContext *pb = avio_alloc_context(pkt->data + 7,
                                              pkt->size - 7,
                                              0, NULL, NULL, NULL, NULL);
@@ -921,17 +921,23 @@ static int read_gab2_sub(AVStream *st, AVPacket *pkt)
         if (desc_len > pb->buf_end - pb->buf_ptr)
             goto error;
 
-        ret = avio_get_str16le(pb, desc_len, (char *)desc, sizeof(desc));
+        ret = avio_get_str16le(pb, desc_len, desc, sizeof(desc));
         avio_skip(pb, desc_len - ret);
         if (*desc)
-            av_dict_set(&st->metadata, "title", (const char *)desc, 0);
+            av_dict_set(&st->metadata, "title", desc, 0);
 
         avio_rl16(pb);   /* flags? */
         avio_rl32(pb);   /* data size */
 
-        pd = (AVProbeData) { .buf      = pb->buf_ptr,
+        size = pb->buf_end - pb->buf_ptr;
-                             .buf_size = pb->buf_end - pb->buf_ptr };
+        pd = (AVProbeData) { .buf      = av_mallocz(size + AVPROBE_PADDING_SIZE),
-        if (!(sub_demuxer = av_probe_input_format2(&pd, 1, &score)))
+                             .buf_size = size };
+        if (!pd.buf)
+            goto error;
+        memcpy(pd.buf, pb->buf_ptr, size);
+        sub_demuxer = av_probe_input_format2(&pd, 1, &score);
+        av_freep(&pd.buf);
+        if (!sub_demuxer)
             goto error;
 
         if (!(ast->sub_ctx = avformat_alloc_context()))
@@ -1044,7 +1050,7 @@ start_sync:
             goto start_sync;
         }
 
-        n = get_stream_idx(d);
+        n = avi->dv_demux ? 0 : get_stream_idx(d);
 
         if (!((i - avi->last_pkt_pos) & 1) &&
             get_stream_idx(d + 1) < s->nb_streams)
@@ -1161,6 +1167,8 @@ static int avi_read_packet(AVFormatContext *s, AVPacket *pkt)
         int size = avpriv_dv_get_packet(avi->dv_demux, pkt);
         if (size >= 0)
             return size;
+        else
+            goto resync;
     }
 
     if (avi->non_interleaved) {
@@ -1311,10 +1319,9 @@ FF_ENABLE_DEPRECATION_WARNINGS
                     size);
             pkt->stream_index = avi->stream_index;
 
-            if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
+            if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO && st->index_entries) {
                 AVIndexEntry *e;
                 int index;
-                av_assert0(st->index_entries);
 
                 index = av_index_search_timestamp(st, ast->frame_offset, 0);
                 e     = &st->index_entries[index];
@@ -1584,6 +1591,12 @@ static int avi_read_seek(AVFormatContext *s, int stream_index,
     int64_t pos, pos_min;
     AVIStream *ast;
 
+    /* Does not matter which stream is requested dv in avi has the
+     * stream information in the first video stream.
+     */
+    if (avi->dv_demux)
+        stream_index = 0;
+
     if (!avi->index_loaded) {
         /* we only load the index on demand */
         avi_load_index(s);
@@ -1616,7 +1629,6 @@ static int avi_read_seek(AVFormatContext *s, int stream_index,
         /* One and only one real stream for DV in AVI, and it has video  */
         /* offsets. Calling with other stream indexes should have failed */
         /* the av_index_search_timestamp call above.                     */
-        av_assert0(stream_index == 0);
 
         if (avio_seek(s->pb, pos, SEEK_SET) < 0)
             return -1;

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/avio.c

@@ -42,8 +42,10 @@ URLProtocol *ffurl_protocol_next(URLProtocol *prev)
 static const char *urlcontext_to_name(void *ptr)
 {
     URLContext *h = (URLContext *)ptr;
-    if(h->prot) return h->prot->name;
+    if (h->prot)
-    else        return "NULL";
+        return h->prot->name;
+    else
+        return "NULL";
 }
 
 static void *urlcontext_child_next(void *obj, void *prev)
@@ -64,53 +66,48 @@ static const AVClass *urlcontext_child_class_next(const AVClass *prev)
             break;
 
     /* find next protocol with priv options */
-    while ((p = ffurl_protocol_next(p)))
+    while (p = ffurl_protocol_next(p))
         if (p->priv_data_class)
             return p->priv_data_class;
     return NULL;
-
 }
 
-static const AVOption options[] = {{NULL}};
+static const AVOption options[] = { { NULL } };
 const AVClass ffurl_context_class = {
-    .class_name     = "URLContext",
+    .class_name       = "URLContext",
-    .item_name      = urlcontext_to_name,
+    .item_name        = urlcontext_to_name,
-    .option         = options,
+    .option           = options,
-    .version        = LIBAVUTIL_VERSION_INT,
+    .version          = LIBAVUTIL_VERSION_INT,
-    .child_next     = urlcontext_child_next,
+    .child_next       = urlcontext_child_next,
     .child_class_next = urlcontext_child_class_next,
 };
 /*@}*/
 
-
 const char *avio_enum_protocols(void **opaque, int output)
 {
     URLProtocol *p;
     *opaque = ffurl_protocol_next(*opaque);
-    if (!(p = *opaque)) return NULL;
+    if (!(p = *opaque))
+        return NULL;
     if ((output && p->url_write) || (!output && p->url_read))
         return p->name;
     return avio_enum_protocols(opaque, output);
 }
 
-int ffurl_register_protocol(URLProtocol *protocol, int size)
+int ffurl_register_protocol(URLProtocol *protocol)
 {
     URLProtocol **p;
-    if (size < sizeof(URLProtocol)) {
-        URLProtocol* temp = av_mallocz(sizeof(URLProtocol));
-        memcpy(temp, protocol, size);
-        protocol = temp;
-    }
     p = &first_protocol;
-    while (*p != NULL) p = &(*p)->next;
+    while (*p != NULL)
-    *p = protocol;
+        p = &(*p)->next;
+    *p             = protocol;
     protocol->next = NULL;
     return 0;
 }
 
-static int url_alloc_for_protocol (URLContext **puc, struct URLProtocol *up,
+static int url_alloc_for_protocol(URLContext **puc, struct URLProtocol *up,
-                                   const char *filename, int flags,
+                                  const char *filename, int flags,
-                                   const AVIOInterruptCB *int_cb)
+                                  const AVIOInterruptCB *int_cb)
 {
     URLContext *uc;
     int err;
@@ -135,18 +132,22 @@ static int url_alloc_for_protocol (URLContext **puc, struct URLProtocol *up,
         goto fail;
     }
     uc->av_class = &ffurl_context_class;
-    uc->filename = (char *) &uc[1];
+    uc->filename = (char *)&uc[1];
     strcpy(uc->filename, filename);
-    uc->prot = up;
+    uc->prot            = up;
-    uc->flags = flags;
+    uc->flags           = flags;
-    uc->is_streamed = 0; /* default = not streamed */
+    uc->is_streamed     = 0; /* default = not streamed */
     uc->max_packet_size = 0; /* default: stream file */
     if (up->priv_data_size) {
         uc->priv_data = av_mallocz(up->priv_data_size);
+        if (!uc->priv_data) {
+            err = AVERROR(ENOMEM);
+            goto fail;
+        }
         if (up->priv_data_class) {
             int proto_len= strlen(up->name);
             char *start = strchr(uc->filename, ',');
-            *(const AVClass**)uc->priv_data = up->priv_data_class;
+            *(const AVClass **)uc->priv_data = up->priv_data_class;
             av_opt_set_defaults(uc->priv_data);
             if(!strncmp(up->name, uc->filename, proto_len) && uc->filename + proto_len == start){
                 int ret= 0;
@@ -178,8 +179,11 @@ static int url_alloc_for_protocol (URLContext **puc, struct URLProtocol *up,
 
     *puc = uc;
     return 0;
- fail:
+fail:
     *puc = NULL;
+    if (uc)
+        av_freep(&uc->priv_data);
+    av_freep(&uc);
 #if CONFIG_NETWORK
     if (up->flags & URL_PROTOCOL_FLAG_NETWORK)
         ff_network_close();
@@ -187,19 +191,22 @@ static int url_alloc_for_protocol (URLContext **puc, struct URLProtocol *up,
     return err;
 }
 
-int ffurl_connect(URLContext* uc, AVDictionary **options)
+int ffurl_connect(URLContext *uc, AVDictionary **options)
 {
     int err =
-        uc->prot->url_open2 ? uc->prot->url_open2(uc, uc->filename, uc->flags, options) :
+        uc->prot->url_open2 ? uc->prot->url_open2(uc,
+                                                  uc->filename,
+                                                  uc->flags,
+                                                  options) :
         uc->prot->url_open(uc, uc->filename, uc->flags);
     if (err)
         return err;
     uc->is_connected = 1;
-    //We must be careful here as ffurl_seek() could be slow, for example for http
+    /* We must be careful here as ffurl_seek() could be slow,
-    if(   (uc->flags & AVIO_FLAG_WRITE)
+     * for example for http */
-       || !strcmp(uc->prot->name, "file"))
+    if ((uc->flags & AVIO_FLAG_WRITE) || !strcmp(uc->prot->name, "file"))
-        if(!uc->is_streamed && ffurl_seek(uc, 0, SEEK_SET) < 0)
+        if (!uc->is_streamed && ffurl_seek(uc, 0, SEEK_SET) < 0)
-            uc->is_streamed= 1;
+            uc->is_streamed = 1;
     return 0;
 }
 
@@ -208,24 +215,19 @@ int ffurl_connect(URLContext* uc, AVDictionary **options)
     "ABCDEFGHIJKLMNOPQRSTUVWXYZ"                \
     "0123456789+-."
 
-int ffurl_alloc(URLContext **puc, const char *filename, int flags,
+static struct URLProtocol *url_find_protocol(const char *filename)
-                const AVIOInterruptCB *int_cb)
 {
     URLProtocol *up = NULL;
     char proto_str[128], proto_nested[128], *ptr;
     size_t proto_len = strspn(filename, URL_SCHEME_CHARS);
 
-    if (!first_protocol) {
+    if (filename[proto_len] != ':' &&
-        av_log(NULL, AV_LOG_WARNING, "No URL Protocols are registered. "
+        (filename[proto_len] != ',' || !strchr(filename + proto_len + 1, ':')) ||
-                                     "Missing call to av_register_all()?\n");
-    }
-
-    if ((filename[proto_len] != ':' &&
-        (filename[proto_len] != ',' || !strchr(filename + proto_len + 1, ':'))) ||
         is_dos_path(filename))
         strcpy(proto_str, "file");
     else
-        av_strlcpy(proto_str, filename, FFMIN(proto_len+1, sizeof(proto_str)));
+        av_strlcpy(proto_str, filename,
+                   FFMIN(proto_len + 1, sizeof(proto_str)));
 
     if ((ptr = strchr(proto_str, ',')))
         *ptr = '\0';
@@ -233,15 +235,33 @@ int ffurl_alloc(URLContext **puc, const char *filename, int flags,
     if ((ptr = strchr(proto_nested, '+')))
         *ptr = '\0';
 
-    while ((up = ffurl_protocol_next(up))) {
+    while (up = ffurl_protocol_next(up)) {
         if (!strcmp(proto_str, up->name))
-            return url_alloc_for_protocol (puc, up, filename, flags, int_cb);
+            break;
         if (up->flags & URL_PROTOCOL_FLAG_NESTED_SCHEME &&
             !strcmp(proto_nested, up->name))
-            return url_alloc_for_protocol (puc, up, filename, flags, int_cb);
+            break;
     }
+
+    return up;
+}
+
+int ffurl_alloc(URLContext **puc, const char *filename, int flags,
+                const AVIOInterruptCB *int_cb)
+{
+    URLProtocol *p = NULL;
+
+    if (!first_protocol) {
+        av_log(NULL, AV_LOG_WARNING, "No URL Protocols are registered. "
+                                     "Missing call to av_register_all()?\n");
+    }
+
+    p = url_find_protocol(filename);
+    if (p)
+       return url_alloc_for_protocol(puc, p, filename, flags, int_cb);
+
     *puc = NULL;
-    if (!strcmp("https", proto_str))
+    if (av_strstart("https:", filename, NULL))
         av_log(NULL, AV_LOG_WARNING, "https protocol not found, recompile with openssl or gnutls enabled.\n");
     return AVERROR_PROTOCOL_NOT_FOUND;
 }
@@ -250,7 +270,7 @@ int ffurl_open(URLContext **puc, const char *filename, int flags,
                const AVIOInterruptCB *int_cb, AVDictionary **options)
 {
     int ret = ffurl_alloc(puc, filename, flags, int_cb);
-    if (ret)
+    if (ret < 0)
         return ret;
     if (options && (*puc)->prot->priv_data_class &&
         (ret = av_opt_set_dict((*puc)->priv_data, options)) < 0)
@@ -264,8 +284,11 @@ fail:
     return ret;
 }
 
-static inline int retry_transfer_wrapper(URLContext *h, unsigned char *buf, int size, int size_min,
+static inline int retry_transfer_wrapper(URLContext *h, uint8_t *buf,
-                                         int (*transfer_func)(URLContext *h, unsigned char *buf, int size))
+                                         int size, int size_min,
+                                         int (*transfer_func)(URLContext *h,
+                                                              uint8_t *buf,
+                                                              int size))
 {
     int ret, len;
     int fast_retries = 5;
@@ -275,7 +298,7 @@ static inline int retry_transfer_wrapper(URLContext *h, unsigned char *buf, int
     while (len < size_min) {
         if (ff_check_interrupt(&h->interrupt_callback))
             return AVERROR_EXIT;
-        ret = transfer_func(h, buf+len, size-len);
+        ret = transfer_func(h, buf + len, size - len);
         if (ret == AVERROR(EINTR))
             continue;
         if (h->flags & AVIO_FLAG_NONBLOCK)
@@ -296,7 +319,7 @@ static inline int retry_transfer_wrapper(URLContext *h, unsigned char *buf, int
         } else if (ret < 1)
             return (ret < 0 && ret != AVERROR_EOF) ? ret : len;
         if (ret)
-           fast_retries = FFMAX(fast_retries, 2);
+            fast_retries = FFMAX(fast_retries, 2);
         len += ret;
     }
     return len;
@@ -341,7 +364,8 @@ int ffurl_closep(URLContext **hh)
 {
     URLContext *h= *hh;
     int ret = 0;
-    if (!h) return 0; /* can happen when ffurl_open fails */
+    if (!h)
+        return 0;     /* can happen when ffurl_open fails */
 
     if (h->is_connected && h->prot->url_close)
         ret = h->prot->url_close(h);
@@ -364,11 +388,18 @@ int ffurl_close(URLContext *h)
 }
 
 
+const char *avio_find_protocol_name(const char *url)
+{
+    URLProtocol *p = url_find_protocol(url);
+
+    return p ? p->name : NULL;
+}
+
 int avio_check(const char *url, int flags)
 {
     URLContext *h;
     int ret = ffurl_alloc(&h, url, flags, NULL);
-    if (ret)
+    if (ret < 0)
         return ret;
 
     if (h->prot->url_check) {
@@ -387,8 +418,8 @@ int64_t ffurl_size(URLContext *h)
 {
     int64_t pos, size;
 
-    size= ffurl_seek(h, 0, AVSEEK_SIZE);
+    size = ffurl_seek(h, 0, AVSEEK_SIZE);
-    if(size<0){
+    if (size < 0) {
         pos = ffurl_seek(h, 0, SEEK_CUR);
         if ((size = ffurl_seek(h, -1, SEEK_END)) < 0)
             return size;

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/avio.h

@@ -156,6 +156,15 @@ typedef struct AVIOContext {
 
 /* unbuffered I/O */
 
+/**
+ * Return the name of the protocol that will handle the passed URL.
+ *
+ * NULL is returned if no protocol could be found for the given URL.
+ *
+ * @return Name of the protocol or NULL.
+ */
+const char *avio_find_protocol_name(const char *url);
+
 /**
  * Return AVIO_FLAG_* access flags corresponding to the access permissions
  * of the resource in url, or a negative value corresponding to an
@@ -372,9 +381,10 @@ int avio_get_str16be(AVIOContext *pb, int maxlen, char *buf, int buflen);
  *
  * @param s Used to return the pointer to the created AVIOContext.
  * In case of failure the pointed to value is set to NULL.
+ * @param url resource to access
  * @param flags flags which control how the resource indicated by url
  * is to be opened
- * @return 0 in case of success, a negative value corresponding to an
+ * @return >= 0 in case of success, a negative value corresponding to an
  * AVERROR code in case of failure
  */
 int avio_open(AVIOContext **s, const char *url, int flags);
@@ -387,13 +397,14 @@ int avio_open(AVIOContext **s, const char *url, int flags);
  *
  * @param s Used to return the pointer to the created AVIOContext.
  * In case of failure the pointed to value is set to NULL.
+ * @param url resource to access
  * @param flags flags which control how the resource indicated by url
  * is to be opened
  * @param int_cb an interrupt callback to be used at the protocols level
  * @param options  A dictionary filled with protocol-private options. On return
  * this parameter will be destroyed and replaced with a dict containing options
  * that were not found. May be NULL.
- * @return 0 in case of success, a negative value corresponding to an
+ * @return >= 0 in case of success, a negative value corresponding to an
  * AVERROR code in case of failure
  */
 int avio_open2(AVIOContext **s, const char *url, int flags,
@@ -460,6 +471,8 @@ const char *avio_enum_protocols(void **opaque, int output);
 /**
  * Pause and resume playing - only meaningful if using a network streaming
  * protocol (e.g. MMS).
+ *
+ * @param h     IO context from which to call the read_pause function pointer
  * @param pause 1 for pause, 0 for resume
  */
 int     avio_pause(AVIOContext *h, int pause);
@@ -467,6 +480,8 @@ int     avio_pause(AVIOContext *h, int pause);
 /**
  * Seek to a given timestamp relative to some component stream.
  * Only meaningful if using a network streaming protocol (e.g. MMS.).
+ *
+ * @param h IO context from which to call the seek function pointers
  * @param stream_index The stream index that the timestamp is relative to.
  *        If stream_index is (-1) the timestamp should be in AV_TIME_BASE
  *        units from the beginning of the presentation.

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/aviobuf.c

@@ -201,12 +201,14 @@ int64_t avio_seek(AVIOContext *s, int64_t offset, int whence)
     int64_t offset1;
     int64_t pos;
     int force = whence & AVSEEK_FORCE;
+    int buffer_size;
     whence &= ~AVSEEK_FORCE;
 
     if(!s)
         return AVERROR(EINVAL);
 
-    pos = s->pos - (s->write_flag ? 0 : (s->buf_end - s->buffer));
+    buffer_size = s->buf_end - s->buffer;
+    pos = s->pos - (s->write_flag ? 0 : buffer_size);
 
     if (whence != SEEK_CUR && whence != SEEK_SET)
         return AVERROR(EINVAL);
@@ -219,7 +221,7 @@ int64_t avio_seek(AVIOContext *s, int64_t offset, int whence)
     }
     offset1 = offset - pos;
     if (!s->must_flush && (!s->direct || !s->seek) &&
-        offset1 >= 0 && offset1 <= (s->buf_end - s->buffer)) {
+        offset1 >= 0 && offset1 <= buffer_size) {
         /* can do the seek inside the buffer */
         s->buf_ptr = s->buffer + offset1;
     } else if ((!s->seekable ||
@@ -232,9 +234,20 @@ int64_t avio_seek(AVIOContext *s, int64_t offset, int whence)
         if (s->eof_reached)
             return AVERROR_EOF;
         s->buf_ptr = s->buf_end + offset - s->pos;
-    } else {
+    } else if(!s->write_flag && offset1 < 0 && -offset1 < buffer_size>>1 && s->seek && offset > 0) {
         int64_t res;
 
+        pos -= FFMIN(buffer_size>>1, pos);
+        if ((res = s->seek(s->opaque, pos, SEEK_SET)) < 0)
+            return res;
+        s->buf_end =
+        s->buf_ptr = s->buffer;
+        s->pos = pos;
+        s->eof_reached = 0;
+        fill_buffer(s);
+        return avio_seek(s, offset, SEEK_SET | force);
+    } else {
+        int64_t res;
         if (s->write_flag) {
             flush_buffer(s);
             s->must_flush = 1;
@@ -317,17 +330,24 @@ int avio_put_str(AVIOContext *s, const char *str)
 
 int avio_put_str16le(AVIOContext *s, const char *str)
 {
-    const uint8_t *q = (const uint8_t *) str;
+    const uint8_t *q = str;
     int ret = 0;
+    int err = 0;
 
     while (*q) {
         uint32_t ch;
         uint16_t tmp;
 
-        GET_UTF8(ch, *q++, break;)
+        GET_UTF8(ch, *q++, goto invalid;)
         PUT_UTF16(ch, tmp, avio_wl16(s, tmp); ret += 2;)
+        continue;
+invalid:
+        av_log(s, AV_LOG_ERROR, "Invaid UTF8 sequence in avio_put_str16le\n");
+        err = AVERROR(EINVAL);
     }
     avio_wl16(s, 0);
+    if (err)
+        return err;
     ret += 2;
     return ret;
 }
@@ -891,7 +911,7 @@ int avio_printf(AVIOContext *s, const char *fmt, ...)
     va_start(ap, fmt);
     ret = vsnprintf(buf, sizeof(buf), fmt, ap);
     va_end(ap);
-    avio_write(s, (const unsigned char *) buf, strlen(buf));
+    avio_write(s, buf, strlen(buf));
     return ret;
 }
 
@@ -1029,14 +1049,20 @@ int ffio_open_dyn_packet_buf(AVIOContext **s, int max_packet_size)
 
 int avio_close_dyn_buf(AVIOContext *s, uint8_t **pbuffer)
 {
-    DynBuffer *d = s->opaque;
+    DynBuffer *d;
     int size;
     static const char padbuf[FF_INPUT_BUFFER_PADDING_SIZE] = {0};
     int padding = 0;
 
+    if (!s) {
+        *pbuffer = NULL;
+        return 0;
+    }
+    d = s->opaque;
+
     /* don't attempt to pad fixed-size packet buffers */
     if (!s->max_packet_size) {
-        avio_write(s, (const unsigned char *) padbuf, sizeof(padbuf));
+        avio_write(s, padbuf, sizeof(padbuf));
         padding = FF_INPUT_BUFFER_PADDING_SIZE;
     }
 

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/dtsdec.c

@@ -19,7 +19,6 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include "config.h"
 #include "libavcodec/bytestream.h"
 #include "avformat.h"
 #include "rawdec.h"
@@ -73,7 +72,7 @@ static int dts_probe(AVProbeData *p)
 
 AVInputFormat ff_dts_demuxer = {
     .name           = "dts",
-    .long_name      = NULL_IF_CONFIG_SMALL("raw DTS"),
+    .long_name      = "raw DTS",
     .read_probe     = dts_probe,
     .read_header    = ff_raw_audio_read_header,
     .read_packet    = ff_raw_read_partial_packet,

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/dtshddec.c

@@ -19,7 +19,6 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include "config.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/dict.h"
 #include "avformat.h"
@@ -91,7 +90,7 @@ static int dtshd_read_header(AVFormatContext *s)
             value = av_malloc(chunk_size);
             if (!value)
                 goto skip;
-            avio_read(pb, (unsigned char *) value, chunk_size);
+            avio_read(pb, (unsigned char *)value, chunk_size);
             value[chunk_size - 1] = 0;
             av_dict_set(&s->metadata, "fileinfo", value,
                         AV_DICT_DONT_STRDUP_VAL);
@@ -129,7 +128,7 @@ static int raw_read_packet(AVFormatContext *s, AVPacket *pkt)
 
 AVInputFormat ff_dtshd_demuxer = {
     .name           = "dtshd",
-    .long_name      = NULL_IF_CONFIG_SMALL("raw DTS-HD"),
+    .long_name      = "raw DTS-HD",
     .priv_data_size = sizeof(DTSHDDemuxContext),
     .read_probe     = dtshd_probe,
     .read_header    = dtshd_read_header,

Frameworks/FFMPEG/ffmpeg-minimal/libavformat/id3v2.c

@@ -354,7 +354,7 @@ static void read_comm(AVFormatContext *s, AVIOContext *pb, int taglen,
  * Parse GEOB tag into a ID3v2ExtraMetaGEOB struct.
  */
 static void read_geobtag(AVFormatContext *s, AVIOContext *pb, int taglen,
-                         char *tag, ID3v2ExtraMeta **extra_meta)
+                         char *tag, ID3v2ExtraMeta **extra_meta, int isv34)
 {
     ID3v2ExtraMetaGEOB *geob_data = NULL;
     ID3v2ExtraMeta *new_extra     = NULL;
@@ -486,7 +486,7 @@ static void free_apic(void *obj)
 }
 
 static void read_apic(AVFormatContext *s, AVIOContext *pb, int taglen,
-                      char *tag, ID3v2ExtraMeta **extra_meta)
+                      char *tag, ID3v2ExtraMeta **extra_meta, int isv34)
 {
     int enc, pic_type;
     char mimetype[64];
@@ -508,7 +508,12 @@ static void read_apic(AVFormatContext *s, AVIOContext *pb, int taglen,
     taglen--;
 
     /* mimetype */
+    if (isv34) {
     taglen -= avio_get_str(pb, taglen, mimetype, sizeof(mimetype));
+    } else {
+        avio_read(pb, mimetype, 3);
+        mimetype[3] = 0;
+    }
     while (mime->id != AV_CODEC_ID_NONE) {
         if (!av_strncasecmp(mime->str, mimetype, sizeof(mimetype))) {
             id = mime->id;
@@ -559,7 +564,7 @@ fail:
     avio_seek(pb, end, SEEK_SET);
 }
 
-static void read_chapter(AVFormatContext *s, AVIOContext *pb, int len, char *ttag, ID3v2ExtraMeta **extra_meta)
+static void read_chapter(AVFormatContext *s, AVIOContext *pb, int len, char *ttag, ID3v2ExtraMeta **extra_meta, int isv34)
 {
     AVRational time_base = {1, 1000};
     uint32_t start, end;
@@ -568,6 +573,14 @@ static void read_chapter(AVFormatContext *s, AVIOContext *pb, int len, char *tta
     int taglen;
     char tag[5];
 
+    if (!s) {
+        /* We should probably just put the chapter data to extra_meta here
+         * and do the AVFormatContext-needing part in a separate
+         * ff_id3v2_parse_apic()-like function. */
+        av_log(NULL, AV_LOG_DEBUG, "No AVFormatContext, skipped ID3 chapter data\n");
+        return;
+    }
+
     if (decode_str(s, pb, 0, &dst, &len) < 0)
         return;
     if (len < 16)
@@ -585,15 +598,14 @@ static void read_chapter(AVFormatContext *s, AVIOContext *pb, int len, char *tta
 
     len -= 16;
     while (len > 10) {
-        avio_read(pb, (unsigned char *) tag, 4);
+        if (avio_read(pb, tag, 4) < 4)
+            goto end;
         tag[4] = 0;
         taglen = avio_rb32(pb);
         avio_skip(pb, 2);
         len -= 10;
-        if (taglen < 0 || taglen > len) {
+        if (taglen < 0 || taglen > len)
-            av_free(dst);
+            goto end;
-            return;
-        }
         if (tag[0] == 'T')
             read_ttag(s, pb, taglen, &chapter->metadata, tag);
         else
@@ -603,14 +615,60 @@ static void read_chapter(AVFormatContext *s, AVIOContext *pb, int len, char *tta
 
     ff_metadata_conv(&chapter->metadata, NULL, ff_id3v2_34_metadata_conv);
     ff_metadata_conv(&chapter->metadata, NULL, ff_id3v2_4_metadata_conv);
+end:
     av_free(dst);
 }
 
+static void free_priv(void *obj)
+{
+    ID3v2ExtraMetaPRIV *priv = obj;
+    av_freep(&priv->owner);
+    av_freep(&priv->data);
+    av_freep(&priv);
+}
+
+static void read_priv(AVFormatContext *s, AVIOContext *pb, int taglen,
+                      char *tag, ID3v2ExtraMeta **extra_meta, int isv34)
+{
+    ID3v2ExtraMeta *meta;
+    ID3v2ExtraMetaPRIV *priv;
+
+    meta = av_mallocz(sizeof(*meta));
+    priv = av_mallocz(sizeof(*priv));
+
+    if (!meta || !priv)
+        goto fail;
+
+    if (decode_str(s, pb, ID3v2_ENCODING_ISO8859, &priv->owner, &taglen) < 0)
+        goto fail;
+
+    priv->data = av_malloc(taglen);
+    if (!priv->data)
+        goto fail;
+
+    priv->datasize = taglen;
+
+    if (avio_read(pb, priv->data, priv->datasize) != priv->datasize)
+        goto fail;
+
+    meta->tag   = "PRIV";
+    meta->data  = priv;
+    meta->next  = *extra_meta;
+    *extra_meta = meta;
+
+    return;
+
+fail:
+    if (priv)
+        free_priv(priv);
+    av_freep(&meta);
+}
+
 typedef struct ID3v2EMFunc {
     const char *tag3;
     const char *tag4;
     void (*read)(AVFormatContext *, AVIOContext *, int, char *,
-                 ID3v2ExtraMeta **);
+                 ID3v2ExtraMeta **, int isv34);
     void (*free)(void *obj);
 } ID3v2EMFunc;
 
@@ -618,6 +676,7 @@ static const ID3v2EMFunc id3v2_extra_meta_funcs[] = {
     { "GEO", "GEOB", read_geobtag, free_geobtag },
     { "PIC", "APIC", read_apic,    free_apic    },
     { "CHAP","CHAP", read_chapter, NULL         },
+    { "PRIV","PRIV", read_priv,    free_priv    },
     { NULL }
 };
 
@@ -640,16 +699,17 @@ static const ID3v2EMFunc *get_extra_meta_func(const char *tag, int isv34)
     return NULL;
 }
 
-static void id3v2_parse(AVFormatContext *s, int len, uint8_t version,
+static void id3v2_parse(AVIOContext *pb, AVDictionary **metadata,
+                        AVFormatContext *s, int len, uint8_t version,
                         uint8_t flags, ID3v2ExtraMeta **extra_meta)
 {
     int isv34, unsync;
     unsigned tlen;
     char tag[5];
-    int64_t next, end = avio_tell(s->pb) + len;
+    int64_t next, end = avio_tell(pb) + len;
     int taghdrlen;
     const char *reason = NULL;
-    AVIOContext pb;
+    AVIOContext pb_local;
     AVIOContext *pbx;
     unsigned char *buffer = NULL;
     int buffer_size       = 0;
@@ -683,7 +743,7 @@ static void id3v2_parse(AVFormatContext *s, int len, uint8_t version,
     unsync = flags & 0x80;
 
     if (isv34 && flags & 0x40) { /* Extended header present, just skip over it */
-        int extlen = get_size(s->pb, 4);
+        int extlen = get_size(pb, 4);
         if (version == 4)
             /* In v2.4 the length includes the length field we just read. */
             extlen -= 4;
@@ -692,7 +752,7 @@ static void id3v2_parse(AVFormatContext *s, int len, uint8_t version,
             reason = "invalid extended header length";
             goto error;
         }
-        avio_skip(s->pb, extlen);
+        avio_skip(pb, extlen);
         len -= extlen + 4;
         if (len < 0) {
             reason = "extended header too long.";
@@ -708,18 +768,20 @@ static void id3v2_parse(AVFormatContext *s, int len, uint8_t version,
         unsigned long dlen;
 
         if (isv34) {
-            avio_read(s->pb, (unsigned char *) tag, 4);
+            if (avio_read(pb, tag, 4) < 4)
+                break;
             tag[4] = 0;
             if (version == 3) {
-                tlen = avio_rb32(s->pb);
+                tlen = avio_rb32(pb);
             } else
-                tlen = get_size(s->pb, 4);
+                tlen = get_size(pb, 4);
-            tflags  = avio_rb16(s->pb);
+            tflags  = avio_rb16(pb);
             tunsync = tflags & ID3v2_FLAG_UNSYNCH;
         } else {
-            avio_read(s->pb, (unsigned char *) tag, 3);
+            if (avio_read(pb, tag, 3) < 3)
+                break;
             tag[3] = 0;
-            tlen   = avio_rb24(s->pb);
+            tlen   = avio_rb24(pb);
         }
         if (tlen > (1<<28))
             break;
@@ -728,7 +790,7 @@ static void id3v2_parse(AVFormatContext *s, int len, uint8_t version,
         if (len < 0)
             break;
 
-        next = avio_tell(s->pb) + tlen;
+        next = avio_tell(pb) + tlen;
 
         if (!tlen) {
             if (tag[0])
@@ -740,7 +802,7 @@ static void id3v2_parse(AVFormatContext *s, int len, uint8_t version,
         if (tflags & ID3v2_FLAG_DATALEN) {
             if (tlen < 4)
                 break;
-            dlen = avio_rb32(s->pb);
+            dlen = avio_rb32(pb);
             tlen -= 4;
         } else
             dlen = tlen;
@@ -759,38 +821,38 @@ static void id3v2_parse(AVFormatContext *s, int len, uint8_t version,
                 type = "encrypted and compressed";
 
             av_log(s, AV_LOG_WARNING, "Skipping %s ID3v2 frame %s.\n", type, tag);
-            avio_skip(s->pb, tlen);
+            avio_skip(pb, tlen);
         /* check for text tag or supported special meta tag */
-        } else if (tag[0] == 'T' || memcmp(tag, "COMM", 4) == 0 ||
+        } else if (tag[0] == 'T' ||
                    (extra_meta &&
                     (extra_func = get_extra_meta_func(tag, isv34)))) {
-            pbx = s->pb;
+            pbx = pb;
 
             if (unsync || tunsync || tcomp) {
-                av_fast_malloc(&buffer, (unsigned int *) &buffer_size, tlen);
+                av_fast_malloc(&buffer, &buffer_size, tlen);
                 if (!buffer) {
                     av_log(s, AV_LOG_ERROR, "Failed to alloc %d bytes\n", tlen);
                     goto seek;
                 }
             }
             if (unsync || tunsync) {
-                int64_t end = avio_tell(s->pb) + tlen;
+                int64_t end = avio_tell(pb) + tlen;
                 uint8_t *b;
 
                 b = buffer;
-                while (avio_tell(s->pb) < end && b - buffer < tlen && !s->pb->eof_reached) {
+                while (avio_tell(pb) < end && b - buffer < tlen && !pb->eof_reached) {
-                    *b++ = avio_r8(s->pb);
+                    *b++ = avio_r8(pb);
-                    if (*(b - 1) == 0xff && avio_tell(s->pb) < end - 1 &&
+                    if (*(b - 1) == 0xff && avio_tell(pb) < end - 1 &&
                         b - buffer < tlen &&
-                        !s->pb->eof_reached ) {
+                        !pb->eof_reached ) {
-                        uint8_t val = avio_r8(s->pb);
+                        uint8_t val = avio_r8(pb);
-                        *b++ = val ? val : avio_r8(s->pb);
+                        *b++ = val ? val : avio_r8(pb);
                     }
                 }
-                ffio_init_context(&pb, buffer, b - buffer, 0, NULL, NULL, NULL,
+                ffio_init_context(&pb_local, buffer, b - buffer, 0, NULL, NULL, NULL,
                                   NULL);
                 tlen = b - buffer;
-                pbx  = &pb; // read from sync buffer
+                pbx  = &pb_local; // read from sync buffer
             }
 
 #if CONFIG_ZLIB
@@ -806,7 +868,7 @@ static void id3v2_parse(AVFormatContext *s, int len, uint8_t version,
                     }
 
                     if (!(unsync || tunsync)) {
-                        err = avio_read(s->pb, buffer, tlen);
+                        err = avio_read(pb, buffer, tlen);
                         if (err < 0) {
                             av_log(s, AV_LOG_ERROR, "Failed to read compressed tag\n");
                             goto seek;
@@ -819,28 +881,26 @@ static void id3v2_parse(AVFormatContext *s, int len, uint8_t version,
                         av_log(s, AV_LOG_ERROR, "Failed to uncompress tag: %d\n", err);
                         goto seek;
                     }
-                    ffio_init_context(&pb, uncompressed_buffer, dlen, 0, NULL, NULL, NULL, NULL);
+                    ffio_init_context(&pb_local, uncompressed_buffer, dlen, 0, NULL, NULL, NULL, NULL);
                     tlen = dlen;
-                    pbx = &pb; // read from sync buffer
+                    pbx = &pb_local; // read from sync buffer
                 }
 #endif
             if (tag[0] == 'T')
                 /* parse text tag */
-                read_ttag(s, pbx, tlen, &s->metadata, tag);
+                read_ttag(s, pbx, tlen, metadata, tag);
-            else if (memcmp(tag, "COMM", 4) == 0)
-                read_comm(s, pbx, tlen, &s->metadata);
             else
                 /* parse special meta tag */
-                extra_func->read(s, pbx, tlen, tag, extra_meta);
+                extra_func->read(s, pbx, tlen, tag, extra_meta, isv34);
         } else if (!tag[0]) {
             if (tag[1])
                 av_log(s, AV_LOG_WARNING, "invalid frame id, assuming padding\n");
-            avio_skip(s->pb, tlen);
+            avio_skip(pb, tlen);
             break;
         }
         /* Skip to end of tag */
 seek:
-        avio_seek(s->pb, next, SEEK_SET);
+        avio_seek(pb, next, SEEK_SET);
     }
 
     /* Footer preset, always 10 bytes, skip over it */
@@ -851,14 +911,15 @@ error:
     if (reason)
         av_log(s, AV_LOG_INFO, "ID3v2.%d tag skipped, cannot handle %s\n",
                version, reason);
-    avio_seek(s->pb, end, SEEK_SET);
+    avio_seek(pb, end, SEEK_SET);
     av_free(buffer);
     av_free(uncompressed_buffer);
     return;
 }
 
-void ff_id3v2_read(AVFormatContext *s, const char *magic,
+static void id3v2_read_internal(AVIOContext *pb, AVDictionary **metadata,
-                   ID3v2ExtraMeta **extra_meta)
+                                AVFormatContext *s, const char *magic,
+                                ID3v2ExtraMeta **extra_meta)
 {
     int len, ret;
     uint8_t buf[ID3v2_HEADER_SIZE];
@@ -867,10 +928,10 @@ void ff_id3v2_read(AVFormatContext *s, const char *magic,
 
     do {
         /* save the current offset in case there's nothing to read/skip */
-        off = avio_tell(s->pb);
+        off = avio_tell(pb);
-        ret = avio_read(s->pb, buf, ID3v2_HEADER_SIZE);
+        ret = avio_read(pb, buf, ID3v2_HEADER_SIZE);
         if (ret != ID3v2_HEADER_SIZE) {
-            avio_seek(s->pb, off, SEEK_SET);
+            avio_seek(pb, off, SEEK_SET);
             break;
         }
         found_header = ff_id3v2_match(buf, magic);
@@ -880,15 +941,27 @@ void ff_id3v2_read(AVFormatContext *s, const char *magic,
                   ((buf[7] & 0x7f) << 14) |
                   ((buf[8] & 0x7f) << 7) |
                    (buf[9] & 0x7f);
-            id3v2_parse(s, len, buf[3], buf[5], extra_meta);
+            id3v2_parse(pb, metadata, s, len, buf[3], buf[5], extra_meta);
         } else {
-            avio_seek(s->pb, off, SEEK_SET);
+            avio_seek(pb, off, SEEK_SET);
         }
     } while (found_header);
-    ff_metadata_conv(&s->metadata, NULL, ff_id3v2_34_metadata_conv);
+    ff_metadata_conv(metadata, NULL, ff_id3v2_34_metadata_conv);
-    ff_metadata_conv(&s->metadata, NULL, id3v2_2_metadata_conv);
+    ff_metadata_conv(metadata, NULL, id3v2_2_metadata_conv);
-    ff_metadata_conv(&s->metadata, NULL, ff_id3v2_4_metadata_conv);
+    ff_metadata_conv(metadata, NULL, ff_id3v2_4_metadata_conv);
-    merge_date(&s->metadata);
+    merge_date(metadata);
+}
+
+void ff_id3v2_read_dict(AVIOContext *pb, AVDictionary **metadata,
+                        const char *magic, ID3v2ExtraMeta **extra_meta)
+{
+    id3v2_read_internal(pb, metadata, NULL, magic, extra_meta);
+}
+
+void ff_id3v2_read(AVFormatContext *s, const char *magic,
+                   ID3v2ExtraMeta **extra_meta)
+{
+    id3v2_read_internal(s->pb, &s->metadata, s, magic, extra_meta);
 }
 
 void ff_id3v2_free_extra_meta(ID3v2ExtraMeta **extra_meta)