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Dependencies: Update bundled libraries

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libvgm: 7cad78367fa35c3f7b3ae16a296d31063cd3a7e4
flac: v1.5.0
ogg: v1.3.6
opus: v1.5.2-153-g7aa5be98
opusfile: v0.12-51-g24d6e75

Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This commit is contained in:
Christopher Snowhill 2025-07-15 07:03:18 -07:00
parent aebc52880b
commit f05aada52d
33 changed files with 905 additions and 115 deletions
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2
.gitignore vendored
View file

@ -36,7 +36,7 @@ Xcode-config/SENTRY_SETTINGS.xcconfig
/ThirdParty/ffmpeg/lib/libavformat.61.dylib
/ThirdParty/ffmpeg/lib/libavutil.59.dylib
/ThirdParty/ffmpeg/lib/libswresample.5.dylib
/ThirdParty/flac/lib/libFLAC.12.dylib
/ThirdParty/flac/lib/libFLAC.14.dylib
/ThirdParty/libid3tag/lib/libid3tag.a
/ThirdParty/libmad/lib/libmad.a
/ThirdParty/libopenmpt/lib/libopenmpt.a

8
Cog.xcodeproj/project.pbxproj
View file

@ -129,7 +129,7 @@
836EF0C827BB91E600BF35B2 /* libogg.0.dylib in CopyFiles */ = {isa = PBXBuildFile; fileRef = 836EF0C427BB919300BF35B2 /* libogg.0.dylib */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
836EF0C927BB91E900BF35B2 /* libvorbisfile.3.dylib in CopyFiles */ = {isa = PBXBuildFile; fileRef = 836EF0C627BB91AB00BF35B2 /* libvorbisfile.3.dylib */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
836EF0CA27BB91EB00BF35B2 /* libvorbis.0.dylib in CopyFiles */ = {isa = PBXBuildFile; fileRef = 836EF0C527BB91AB00BF35B2 /* libvorbis.0.dylib */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
836EF0CB27BB91EE00BF35B2 /* libFLAC.12.dylib in CopyFiles */ = {isa = PBXBuildFile; fileRef = 836EF0C727BB91BC00BF35B2 /* libFLAC.12.dylib */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
836EF0CB27BB91EE00BF35B2 /* libFLAC.14.dylib in CopyFiles */ = {isa = PBXBuildFile; fileRef = 836EF0C727BB91BC00BF35B2 /* libFLAC.14.dylib */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
836EF0E027BB98A800BF35B2 /* libopus.0.dylib in CopyFiles */ = {isa = PBXBuildFile; fileRef = 836EF0CC27BB94E100BF35B2 /* libopus.0.dylib */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
836EF0E127BB98AB00BF35B2 /* libopusfile.0.dylib in CopyFiles */ = {isa = PBXBuildFile; fileRef = 836EF0CD27BB94F100BF35B2 /* libopusfile.0.dylib */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
836F5BF91A357A01002730CC /* sidplay.bundle in CopyFiles */ = {isa = PBXBuildFile; fileRef = 8314D6411A354DFF00EEE8E6 /* sidplay.bundle */; settings = {ATTRIBUTES = (CodeSignOnCopy, RemoveHeadersOnCopy, ); }; };
@ -721,7 +721,7 @@
836DF617298F6F1700CD0580 /* libsoxr.0.dylib in CopyFiles */,
836EF0E127BB98AB00BF35B2 /* libopusfile.0.dylib in CopyFiles */,
836EF0E027BB98A800BF35B2 /* libopus.0.dylib in CopyFiles */,
836EF0CB27BB91EE00BF35B2 /* libFLAC.12.dylib in CopyFiles */,
836EF0CB27BB91EE00BF35B2 /* libFLAC.14.dylib in CopyFiles */,
836EF0CA27BB91EB00BF35B2 /* libvorbis.0.dylib in CopyFiles */,
83256B69286661FC0036D9C0 /* libmpg123.0.dylib in CopyFiles */,
836EF0C927BB91E900BF35B2 /* libvorbisfile.3.dylib in CopyFiles */,
@ -951,7 +951,7 @@
836EF0C427BB919300BF35B2 /* libogg.0.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libogg.0.dylib; path = ThirdParty/ogg/lib/libogg.0.dylib; sourceTree = "<group>"; };
836EF0C527BB91AB00BF35B2 /* libvorbis.0.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libvorbis.0.dylib; path = ThirdParty/vorbis/lib/libvorbis.0.dylib; sourceTree = "<group>"; };
836EF0C627BB91AB00BF35B2 /* libvorbisfile.3.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libvorbisfile.3.dylib; path = ThirdParty/vorbis/lib/libvorbisfile.3.dylib; sourceTree = "<group>"; };
836EF0C727BB91BC00BF35B2 /* libFLAC.12.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libFLAC.12.dylib; path = ThirdParty/flac/lib/libFLAC.12.dylib; sourceTree = "<group>"; };
836EF0C727BB91BC00BF35B2 /* libFLAC.14.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libFLAC.14.dylib; path = ThirdParty/flac/lib/libFLAC.14.dylib; sourceTree = "<group>"; };
836EF0CC27BB94E100BF35B2 /* libopus.0.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libopus.0.dylib; path = ThirdParty/opus/lib/libopus.0.dylib; sourceTree = "<group>"; };
836EF0CD27BB94F100BF35B2 /* libopusfile.0.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libopusfile.0.dylib; path = ThirdParty/opusfile/lib/libopusfile.0.dylib; sourceTree = "<group>"; };
836F6B2518BDB80D0095E648 /* vgmstream.xcodeproj */ = {isa = PBXFileReference; lastKnownFileType = "wrapper.pb-project"; name = vgmstream.xcodeproj; path = Plugins/vgmstream/vgmstream.xcodeproj; sourceTree = "<group>"; };
@ -1154,7 +1154,7 @@
835FAC7D27BCDF5B00BA8562 /* libavif.a */,
836EF0CD27BB94F100BF35B2 /* libopusfile.0.dylib */,
836EF0CC27BB94E100BF35B2 /* libopus.0.dylib */,
836EF0C727BB91BC00BF35B2 /* libFLAC.12.dylib */,
836EF0C727BB91BC00BF35B2 /* libFLAC.14.dylib */,
836EF0C527BB91AB00BF35B2 /* libvorbis.0.dylib */,
836EF0C627BB91AB00BF35B2 /* libvorbisfile.3.dylib */,
836EF0C427BB919300BF35B2 /* libogg.0.dylib */,

24
Frameworks/vgmstream/libvgmstream.xcodeproj/project.pbxproj
View file

@ -62,7 +62,7 @@
8306B0EB20984590000302D4 /* wave_segmented.c in Sources */ = {isa = PBXBuildFile; fileRef = 8306B0D02098458F000302D4 /* wave_segmented.c */; };
8306B0EC20984590000302D4 /* pcm_sre.c in Sources */ = {isa = PBXBuildFile; fileRef = 8306B0D12098458F000302D4 /* pcm_sre.c */; };
8306B0ED20984590000302D4 /* txtp.c in Sources */ = {isa = PBXBuildFile; fileRef = 8306B0D22098458F000302D4 /* txtp.c */; };
8306B0EE20984590000302D4 /* smc_smh.c in Sources */ = {isa = PBXBuildFile; fileRef = 8306B0D32098458F000302D4 /* smc_smh.c */; };
8306B0EE20984590000302D4 /* smh_smc.c in Sources */ = {isa = PBXBuildFile; fileRef = 8306B0D32098458F000302D4 /* smh_smc.c */; };
8306B0EF20984590000302D4 /* ubi_bao.c in Sources */ = {isa = PBXBuildFile; fileRef = 8306B0D420984590000302D4 /* ubi_bao.c */; };
8306B0F020984590000302D4 /* atsl.c in Sources */ = {isa = PBXBuildFile; fileRef = 8306B0D520984590000302D4 /* atsl.c */; };
8306B0F120984590000302D4 /* ppst.c in Sources */ = {isa = PBXBuildFile; fileRef = 8306B0D620984590000302D4 /* ppst.c */; };
@ -154,6 +154,8 @@
832BF82B21E0514B006F50F1 /* xopus.c in Sources */ = {isa = PBXBuildFile; fileRef = 832BF81921E0514A006F50F1 /* xopus.c */; };
832BF82C21E0514B006F50F1 /* hca_keys_awb.h in Headers */ = {isa = PBXBuildFile; fileRef = 832BF81A21E0514A006F50F1 /* hca_keys_awb.h */; };
832BF82D21E0514B006F50F1 /* nus3audio.c in Sources */ = {isa = PBXBuildFile; fileRef = 832BF81B21E0514B006F50F1 /* nus3audio.c */; };
832F9EA62E29047F00D08728 /* srcd.c in Sources */ = {isa = PBXBuildFile; fileRef = 832F9EA52E29047F00D08728 /* srcd.c */; };
832F9EA82E2905D500D08728 /* mhwk.c in Sources */ = {isa = PBXBuildFile; fileRef = 832F9EA72E2905D500D08728 /* mhwk.c */; };
832FC36C278FA4CB0056A860 /* ubi_ckd_cwav_streamfile.h in Headers */ = {isa = PBXBuildFile; fileRef = 832FC367278FA4CB0056A860 /* ubi_ckd_cwav_streamfile.h */; };
832FC36D278FA4CB0056A860 /* ubi_ckd_cwav.c in Sources */ = {isa = PBXBuildFile; fileRef = 832FC36B278FA4CB0056A860 /* ubi_ckd_cwav.c */; };
832FC36F278FAE3E0056A860 /* encrypted_mc161_streamfile.h in Headers */ = {isa = PBXBuildFile; fileRef = 832FC36E278FAE3E0056A860 /* encrypted_mc161_streamfile.h */; };
@ -515,7 +517,7 @@
836F6F9418BDC2190095E648 /* iivb.c in Sources */ = {isa = PBXBuildFile; fileRef = 836F6E5818BDC2180095E648 /* iivb.c */; };
836F6F9518BDC2190095E648 /* kraw.c in Sources */ = {isa = PBXBuildFile; fileRef = 836F6E5918BDC2180095E648 /* kraw.c */; };
836F6F9618BDC2190095E648 /* lsf.c in Sources */ = {isa = PBXBuildFile; fileRef = 836F6E5A18BDC2180095E648 /* lsf.c */; };
836F6F9818BDC2190095E648 /* mattel_hyperscan.c in Sources */ = {isa = PBXBuildFile; fileRef = 836F6E5C18BDC2180095E648 /* mattel_hyperscan.c */; };
836F6F9818BDC2190095E648 /* kvag.c in Sources */ = {isa = PBXBuildFile; fileRef = 836F6E5C18BDC2180095E648 /* kvag.c */; };
836F6F9918BDC2190095E648 /* maxis_xa.c in Sources */ = {isa = PBXBuildFile; fileRef = 836F6E5D18BDC2180095E648 /* maxis_xa.c */; };
836F6F9A18BDC2190095E648 /* meta.h in Headers */ = {isa = PBXBuildFile; fileRef = 836F6E5E18BDC2180095E648 /* meta.h */; };
836F6F9C18BDC2190095E648 /* mp4.c in Sources */ = {isa = PBXBuildFile; fileRef = 836F6E6018BDC2180095E648 /* mp4.c */; };
@ -1020,7 +1022,7 @@
8306B0D02098458F000302D4 /* wave_segmented.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = wave_segmented.c; sourceTree = "<group>"; };
8306B0D12098458F000302D4 /* pcm_sre.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = pcm_sre.c; sourceTree = "<group>"; };
8306B0D22098458F000302D4 /* txtp.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = txtp.c; sourceTree = "<group>"; };
8306B0D32098458F000302D4 /* smc_smh.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = smc_smh.c; sourceTree = "<group>"; };
8306B0D32098458F000302D4 /* smh_smc.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = smh_smc.c; sourceTree = "<group>"; };
8306B0D420984590000302D4 /* ubi_bao.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = ubi_bao.c; sourceTree = "<group>"; };
8306B0D520984590000302D4 /* atsl.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = atsl.c; sourceTree = "<group>"; };
8306B0D620984590000302D4 /* ppst.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = ppst.c; sourceTree = "<group>"; };
@ -1112,6 +1114,8 @@
832BF81921E0514A006F50F1 /* xopus.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = xopus.c; sourceTree = "<group>"; };
832BF81A21E0514A006F50F1 /* hca_keys_awb.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = hca_keys_awb.h; sourceTree = "<group>"; };
832BF81B21E0514B006F50F1 /* nus3audio.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = nus3audio.c; sourceTree = "<group>"; };
832F9EA52E29047F00D08728 /* srcd.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; path = srcd.c; sourceTree = "<group>"; };
832F9EA72E2905D500D08728 /* mhwk.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; path = mhwk.c; sourceTree = "<group>"; };
832FC367278FA4CB0056A860 /* ubi_ckd_cwav_streamfile.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ubi_ckd_cwav_streamfile.h; sourceTree = "<group>"; };
832FC36B278FA4CB0056A860 /* ubi_ckd_cwav.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = ubi_ckd_cwav.c; sourceTree = "<group>"; };
832FC36E278FAE3E0056A860 /* encrypted_mc161_streamfile.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = encrypted_mc161_streamfile.h; sourceTree = "<group>"; };
@ -1473,7 +1477,7 @@
836F6E5818BDC2180095E648 /* iivb.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = iivb.c; sourceTree = "<group>"; };
836F6E5918BDC2180095E648 /* kraw.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = kraw.c; sourceTree = "<group>"; };
836F6E5A18BDC2180095E648 /* lsf.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = lsf.c; sourceTree = "<group>"; };
836F6E5C18BDC2180095E648 /* mattel_hyperscan.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = mattel_hyperscan.c; sourceTree = "<group>"; };
836F6E5C18BDC2180095E648 /* kvag.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = kvag.c; sourceTree = "<group>"; };
836F6E5D18BDC2180095E648 /* maxis_xa.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = maxis_xa.c; sourceTree = "<group>"; };
836F6E5E18BDC2180095E648 /* meta.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = meta.h; sourceTree = "<group>"; };
836F6E6018BDC2180095E648 /* mp4.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = mp4.c; sourceTree = "<group>"; };
@ -2498,6 +2502,7 @@
83D20074248DDB760048BD24 /* ktsr.c */,
834F7CFE2C7085EA003AC386 /* ktsr_streamfile.h */,
830EBE122004656E0023AA10 /* ktss.c */,
836F6E5C18BDC2180095E648 /* kvag.c */,
8373342423F60CDB00DE14DC /* kwb.c */,
835B9B8C2730BF2C00F87EE3 /* lopu_fb.c */,
836F6E9918BDC2180095E648 /* lp_ap_lep.c */,
@ -2509,11 +2514,11 @@
8373341F23F60CDB00DE14DC /* lrmd_streamfile.h */,
836F6E5A18BDC2180095E648 /* lsf.c */,
83EDE5D61A70951A005F5D84 /* madp.c */,
836F6E5C18BDC2180095E648 /* mattel_hyperscan.c */,
836F6E5D18BDC2180095E648 /* maxis_xa.c */,
836F6E9D18BDC2180095E648 /* mcg.c */,
83709E011ECBC1A4005C03D3 /* mcss.c */,
836F6E5E18BDC2180095E648 /* meta.h */,
832F9EA72E2905D500D08728 /* mhwk.c */,
834FE0DE215C79EB000A5D3D /* mib_mih.c */,
836F6EAC18BDC2180095E648 /* mic_koei.c */,
83B73C4A2D8FF19800A57F08 /* mio.c */,
@ -2660,7 +2665,7 @@
83CBF5422D46339200AA2D75 /* skex.c */,
836F6EBB18BDC2180095E648 /* sl3.c */,
836F6EF218BDC2190095E648 /* sli.c */,
8306B0D32098458F000302D4 /* smc_smh.c */,
8306B0D32098458F000302D4 /* smh_smc.c */,
837CEAEB23487F2B00E62A4A /* smk.c */,
83F0AA5C21E2028B004BBC04 /* smp.c */,
836F6EBC18BDC2180095E648 /* smpl.c */,
@ -2679,6 +2684,7 @@
83A21F84201D8981000F04B9 /* sqex_sead.c */,
837CEAF023487F2C00E62A4A /* sqex_streamfile.h */,
83C0C75C2AA435C60056AFD8 /* squeak.c */,
832F9EA52E29047F00D08728 /* srcd.c */,
834FE0D6215C79E9000A5D3D /* sscf.c */,
8396BE792935FC2F00CD0580 /* sscf_encrypted.h */,
8396BE782935FC2F00CD0580 /* sscf_encrypted.c */,
@ -3245,7 +3251,7 @@
83A21F8D201D8982000F04B9 /* sqex_sead.c in Sources */,
834F7ED32C70A786003AC386 /* mixing.c in Sources */,
83EED5D3203A8BC7008BEB45 /* ea_swvr.c in Sources */,
8306B0EE20984590000302D4 /* smc_smh.c in Sources */,
8306B0EE20984590000302D4 /* smh_smc.c in Sources */,
834F7DAB2C7093EA003AC386 /* asf_decoder.c in Sources */,
834F7E0C2C7093EA003AC386 /* ws_decoder.c in Sources */,
834F7EDB2C70A786003AC386 /* streamfile_api.c in Sources */,
@ -3553,7 +3559,7 @@
834F7E962C70A45C003AC386 /* txtp_parser.c in Sources */,
834FE0EA215C79ED000A5D3D /* aif_asobo.c in Sources */,
836F700418BDC2190095E648 /* vas_kceo.c in Sources */,
836F6F9818BDC2190095E648 /* mattel_hyperscan.c in Sources */,
836F6F9818BDC2190095E648 /* kvag.c in Sources */,
8306B0BA20984552000302D4 /* blocked_wsi.c in Sources */,
834F7E802C709EA0003AC386 /* adp_ongakukan.c in Sources */,
83BAFB6C19F45EB3005DAB60 /* bfstm.c in Sources */,
@ -3667,6 +3673,7 @@
834F7DB62C7093EA003AC386 /* compresswave_lib.c in Sources */,
834F7E042C7093EA003AC386 /* vorbis_custom_utils_awc.c in Sources */,
8306B0EC20984590000302D4 /* pcm_sre.c in Sources */,
832F9EA62E29047F00D08728 /* srcd.c in Sources */,
836F6FC818BDC2190095E648 /* pos.c in Sources */,
836F6F8918BDC2190095E648 /* gca.c in Sources */,
836F6FA418BDC2190095E648 /* nds_hwas.c in Sources */,
@ -3805,6 +3812,7 @@
836F6FA918BDC2190095E648 /* ngc_adpdtk.c in Sources */,
836F6FDC18BDC2190095E648 /* iab.c in Sources */,
83C7282122BC893D00678B4A /* msf_konami.c in Sources */,
832F9EA82E2905D500D08728 /* mhwk.c in Sources */,
833E82F72A2858EF00CD0580 /* reader.c in Sources */,
83B69B222845A26600D2435A /* bw_mp3_riff.c in Sources */,
834F7E192C709A1D003AC386 /* ea_schl_abk.c in Sources */,

4
Info.plist.template
View file

@ -656,6 +656,7 @@
<string>mds</string>
<string>mdsp</string>
<string>med</string>
<string>mhwk</string>
<string>mjb</string>
<string>mi4</string>
<string>mib</string>
@ -829,7 +830,7 @@
<string>skx</string>
<string>slb</string>
<string>sli</string>
<string>smc</string>
<string>smh</string>
<string>smk</string>
<string>smp</string>
<string>smv</string>
@ -851,6 +852,7 @@
<string>spsd</string>
<string>spw</string>
<string>srsa</string>
<string>srcd</string>
<string>ss2</string>
<string>ssd</string>
<string>ssf</string>

8
Plugins/Flac/Flac.xcodeproj/project.pbxproj
View file

@ -10,7 +10,7 @@
17C93F080B8FF67A008627D6 /* FlacDecoder.m in Sources */ = {isa = PBXBuildFile; fileRef = 17C93F040B8FF67A008627D6 /* FlacDecoder.m */; };
8301C147287805F500651A6E /* NSDictionary+Merge.m in Sources */ = {isa = PBXBuildFile; fileRef = 8301C146287805F500651A6E /* NSDictionary+Merge.m */; };
834D18FE2D7F949900F13B89 /* CogAudio.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 834D18FD2D7F949900F13B89 /* CogAudio.framework */; };
836EF0DA27BB970B00BF35B2 /* libFLAC.12.dylib in Frameworks */ = {isa = PBXBuildFile; fileRef = 836EF0D927BB970B00BF35B2 /* libFLAC.12.dylib */; };
836EF0DA27BB970B00BF35B2 /* libFLAC.14.dylib in Frameworks */ = {isa = PBXBuildFile; fileRef = 836EF0D927BB970B00BF35B2 /* libFLAC.14.dylib */; };
83AA660B27B7DAE40098D4B8 /* cuesheet.m in Sources */ = {isa = PBXBuildFile; fileRef = 83AA660A27B7DAE40098D4B8 /* cuesheet.m */; };
8D5B49B4048680CD000E48DA /* Cocoa.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 1058C7ADFEA557BF11CA2CBB /* Cocoa.framework */; };
/* End PBXBuildFile section */
@ -40,7 +40,7 @@
834A42B4287AF7AA00EB9D9B /* AudioChunk.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = AudioChunk.h; path = ../../Audio/Chain/AudioChunk.h; sourceTree = "<group>"; };
834D18FD2D7F949900F13B89 /* CogAudio.framework */ = {isa = PBXFileReference; explicitFileType = wrapper.framework; path = CogAudio.framework; sourceTree = BUILT_PRODUCTS_DIR; };
8356BD1927B3CCBB0074E50C /* HTTPSource.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = HTTPSource.h; path = ../HTTPSource/HTTPSource.h; sourceTree = "<group>"; };
836EF0D927BB970B00BF35B2 /* libFLAC.12.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libFLAC.12.dylib; path = ../../ThirdParty/flac/lib/libFLAC.12.dylib; sourceTree = "<group>"; };
836EF0D927BB970B00BF35B2 /* libFLAC.14.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libFLAC.14.dylib; path = ../../ThirdParty/flac/lib/libFLAC.14.dylib; sourceTree = "<group>"; };
83747C4A2862DCF40021245F /* Shared.xcconfig */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text.xcconfig; path = Shared.xcconfig; sourceTree = "<group>"; };
8384912D180816C900E7332D /* Logging.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = Logging.h; path = ../../Utils/Logging.h; sourceTree = "<group>"; };
83AA660A27B7DAE40098D4B8 /* cuesheet.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; path = cuesheet.m; sourceTree = "<group>"; };
@ -56,7 +56,7 @@
buildActionMask = 2147483647;
files = (
834D18FE2D7F949900F13B89 /* CogAudio.framework in Frameworks */,
836EF0DA27BB970B00BF35B2 /* libFLAC.12.dylib in Frameworks */,
836EF0DA27BB970B00BF35B2 /* libFLAC.14.dylib in Frameworks */,
8D5B49B4048680CD000E48DA /* Cocoa.framework in Frameworks */,
);
runOnlyForDeploymentPostprocessing = 0;
@ -115,7 +115,7 @@
1058C7ACFEA557BF11CA2CBB /* Linked Frameworks */ = {
isa = PBXGroup;
children = (
836EF0D927BB970B00BF35B2 /* libFLAC.12.dylib */,
836EF0D927BB970B00BF35B2 /* libFLAC.14.dylib */,
1058C7ADFEA557BF11CA2CBB /* Cocoa.framework */,
);
name = "Linked Frameworks";

9
Plugins/Flac/FlacDecoder.m
View file

@ -337,6 +337,8 @@ void ErrorCallback(const FLAC__StreamDecoder *decoder, FLAC__StreamDecoderErrorS
FLAC__StreamDecoderInitStatus ret;
if(isOggFlac) {
FLAC__stream_decoder_set_decode_chained_stream(decoder, true);
ret = FLAC__stream_decoder_init_ogg_stream(decoder,
ReadCallback,
([source seekable] ? SeekCallback : NULL),
@ -384,8 +386,13 @@ void ErrorCallback(const FLAC__StreamDecoder *decoder, FLAC__StreamDecoderErrorS
AudioChunk *chunk = nil;
while (blockBufferFrames <= 0) {
if(FLAC__stream_decoder_get_state(decoder) == FLAC__STREAM_DECODER_END_OF_STREAM) {
FLAC__StreamDecoderState state = FLAC__stream_decoder_get_state(decoder);
if(state == FLAC__STREAM_DECODER_END_OF_STREAM) {
return nil;
} else if(state == FLAC__STREAM_DECODER_END_OF_LINK) {
if(!FLAC__stream_decoder_finish_link(decoder)) {
return nil;
}
}
if(!FLAC__stream_decoder_process_single(decoder)) {

8
Plugins/Opus/OpusPlugin.xcodeproj/project.pbxproj
View file

@ -7,7 +7,7 @@
objects = {
/* Begin PBXBuildFile section */
8301C14928780C1A00651A6E /* libFLAC.12.dylib in Frameworks */ = {isa = PBXBuildFile; fileRef = 8301C14828780C1A00651A6E /* libFLAC.12.dylib */; };
8301C14928780C1A00651A6E /* libFLAC.14.dylib in Frameworks */ = {isa = PBXBuildFile; fileRef = 8301C14828780C1A00651A6E /* libFLAC.14.dylib */; };
83186316285CEC91001422CC /* NSDictionary+Merge.m in Sources */ = {isa = PBXBuildFile; fileRef = 83186315285CEC91001422CC /* NSDictionary+Merge.m */; };
834D19102D7F956400F13B89 /* CogAudio.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 834D190F2D7F956400F13B89 /* CogAudio.framework */; };
836EF0CF27BB952F00BF35B2 /* libopusfile.0.dylib in Frameworks */ = {isa = PBXBuildFile; fileRef = 836EF0CE27BB952F00BF35B2 /* libopusfile.0.dylib */; };
@ -29,7 +29,7 @@
/* End PBXCopyFilesBuildPhase section */
/* Begin PBXFileReference section */
8301C14828780C1A00651A6E /* libFLAC.12.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libFLAC.12.dylib; path = ../../ThirdParty/flac/lib/libFLAC.12.dylib; sourceTree = "<group>"; };
8301C14828780C1A00651A6E /* libFLAC.14.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libFLAC.14.dylib; path = ../../ThirdParty/flac/lib/libFLAC.14.dylib; sourceTree = "<group>"; };
83186314285CEC91001422CC /* NSDictionary+Merge.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = "NSDictionary+Merge.h"; path = "../../../Utils/NSDictionary+Merge.h"; sourceTree = "<group>"; };
83186315285CEC91001422CC /* NSDictionary+Merge.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; name = "NSDictionary+Merge.m"; path = "../../../Utils/NSDictionary+Merge.m"; sourceTree = "<group>"; };
833F68411CDBCABC00AFB9F0 /* es */ = {isa = PBXFileReference; lastKnownFileType = text.plist.strings; name = es; path = es.lproj/InfoPlist.strings; sourceTree = "<group>"; };
@ -61,7 +61,7 @@
files = (
834D19102D7F956400F13B89 /* CogAudio.framework in Frameworks */,
8375B04017FFEA400092A79F /* Cocoa.framework in Frameworks */,
8301C14928780C1A00651A6E /* libFLAC.12.dylib in Frameworks */,
8301C14928780C1A00651A6E /* libFLAC.14.dylib in Frameworks */,
836EF0CF27BB952F00BF35B2 /* libopusfile.0.dylib in Frameworks */,
);
runOnlyForDeploymentPostprocessing = 0;
@ -100,7 +100,7 @@
isa = PBXGroup;
children = (
834D190F2D7F956400F13B89 /* CogAudio.framework */,
8301C14828780C1A00651A6E /* libFLAC.12.dylib */,
8301C14828780C1A00651A6E /* libFLAC.14.dylib */,
836EF0CE27BB952F00BF35B2 /* libopusfile.0.dylib */,
8375B03F17FFEA400092A79F /* Cocoa.framework */,
8375B04117FFEA400092A79F /* Other Frameworks */,

8
Plugins/Vorbis/VorbisPlugin.xcodeproj/project.pbxproj
View file

@ -8,7 +8,7 @@
/* Begin PBXBuildFile section */
17C93D360B8FDA66008627D6 /* VorbisDecoder.m in Sources */ = {isa = PBXBuildFile; fileRef = 17C93D340B8FDA66008627D6 /* VorbisDecoder.m */; };
8301C14B287810F300651A6E /* libFLAC.12.dylib in Frameworks */ = {isa = PBXBuildFile; fileRef = 8301C14A287810F300651A6E /* libFLAC.12.dylib */; };
8301C14B287810F300651A6E /* libFLAC.14.dylib in Frameworks */ = {isa = PBXBuildFile; fileRef = 8301C14A287810F300651A6E /* libFLAC.14.dylib */; };
83186313285CEBD2001422CC /* NSDictionary+Merge.m in Sources */ = {isa = PBXBuildFile; fileRef = 83186312285CEBD2001422CC /* NSDictionary+Merge.m */; };
834D19172D7F95A600F13B89 /* CogAudio.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 834D19162D7F95A600F13B89 /* CogAudio.framework */; };
836EF0D627BB969D00BF35B2 /* libvorbisfile.3.dylib in Frameworks */ = {isa = PBXBuildFile; fileRef = 836EF0D427BB969D00BF35B2 /* libvorbisfile.3.dylib */; };
@ -26,7 +26,7 @@
17C93D330B8FDA66008627D6 /* VorbisDecoder.h */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.h; path = VorbisDecoder.h; sourceTree = "<group>"; };
17C93D340B8FDA66008627D6 /* VorbisDecoder.m */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.objc; path = VorbisDecoder.m; sourceTree = "<group>"; };
32DBCF630370AF2F00C91783 /* VorbisPlugin_Prefix.pch */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = VorbisPlugin_Prefix.pch; sourceTree = "<group>"; };
8301C14A287810F300651A6E /* libFLAC.12.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libFLAC.12.dylib; path = ../../ThirdParty/flac/lib/libFLAC.12.dylib; sourceTree = "<group>"; };
8301C14A287810F300651A6E /* libFLAC.14.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libFLAC.14.dylib; path = ../../ThirdParty/flac/lib/libFLAC.14.dylib; sourceTree = "<group>"; };
83186311285CEBD2001422CC /* NSDictionary+Merge.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = "NSDictionary+Merge.h"; path = "../../Utils/NSDictionary+Merge.h"; sourceTree = "<group>"; };
83186312285CEBD2001422CC /* NSDictionary+Merge.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; name = "NSDictionary+Merge.m"; path = "../../Utils/NSDictionary+Merge.m"; sourceTree = "<group>"; };
834A42AB287AF0B000EB9D9B /* AudioChunk.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = AudioChunk.h; path = ../../Audio/Chain/AudioChunk.h; sourceTree = "<group>"; };
@ -51,7 +51,7 @@
buildActionMask = 2147483647;
files = (
834D19172D7F95A600F13B89 /* CogAudio.framework in Frameworks */,
8301C14B287810F300651A6E /* libFLAC.12.dylib in Frameworks */,
8301C14B287810F300651A6E /* libFLAC.14.dylib in Frameworks */,
8D5B49B4048680CD000E48DA /* Cocoa.framework in Frameworks */,
836EF0DF27BB987000BF35B2 /* libvorbis.0.dylib in Frameworks */,
836EF0D627BB969D00BF35B2 /* libvorbisfile.3.dylib in Frameworks */,
@ -111,7 +111,7 @@
1058C7ACFEA557BF11CA2CBB /* Linked Frameworks */ = {
isa = PBXGroup;
children = (
8301C14A287810F300651A6E /* libFLAC.12.dylib */,
8301C14A287810F300651A6E /* libFLAC.14.dylib */,
836EF0DE27BB987000BF35B2 /* libvorbis.0.dylib */,
836EF0D427BB969D00BF35B2 /* libvorbisfile.3.dylib */,
1058C7ADFEA557BF11CA2CBB /* Cocoa.framework */,

17
ThirdParty/flac/README.md vendored
View file

@ -1,18 +1,21 @@
Build with CMake, using the following options:
```
cmake -B build-x86 -DCMAKE_OSX_ARCHITECTURES="x86_64" -DCMAKE_OSX_DEPLOYMENT_TARGET="10.13" -DBUILD_SHARED_LIBS=ON
cmake -B build-arm -DCMAKE_OSX_ARCHITECTURES="arm64" -DCMAKE_OSX_DEPLOYMENT_TARGET="11.0" -DBUILD_SHARED_LIBS=ON
cmake -B build-x86 -DCMAKE_LIBRARY_PATH=/usr/local/lib -DCMAKE_OSX_ARCHITECTURES="x86_64" \
-DCMAKE_OSX_DEPLOYMENT_TARGET="10.13" -DCMAKE_BUILD_TYPE=RelWithDebInfo -DBUILD_SHARED_LIBS=ON
cmake -B build-arm -DCMAKE_OSX_ARCHITECTURES="arm64" -DCMAKE_OSX_DEPLOYMENT_TARGET="11.0" \
-DCMAKE_BUILD_TYPE=RelWithDebInfo -DBUILD_SHARED_LIBS=ON
```
For the debug overlay, BUILD_TYPE was replaced with `Debug`.
And some minor tweaks with `install_name_tool -id` to make sure that the
resulting libFLAC.12.dylib referred to itself with @rpath and not full
paths of the build directory, and imported libogg.0.dylib with an @rpath.
And some minor tweaks with `install_name_tool -change` to fix up the
resulting libFLAC.14.dylib so that it imports libogg.0.dylib using an
@rpath path instead of an absolute path to Homebrew's install.
Version 1.4.3 was used from the official source code download:
Version 1.5.0 was used from the official source code download:
https://downloads.xiph.org/releases/flac/flac-1.4.3.tar.xz
https://downloads.xiph.org/releases/flac/flac-1.5.0.tar.xz
x86_64 and arm64 were built separately, to allow for intrinsic functions
to be used for x86_64.

2
ThirdParty/flac/include/FLAC/FLAC_assert.h vendored
View file

@ -1,6 +1,6 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2001-2009 Josh Coalson
* Copyright (C) 2011-2023 Xiph.Org Foundation
* Copyright (C) 2011-2025 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions

52
ThirdParty/flac/include/FLAC/all.h vendored
View file

@ -1,6 +1,6 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2023 Xiph.Org Foundation
* Copyright (C) 2011-2025 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@ -436,6 +436,56 @@
*
*/
/** \defgroup porting_1_4_3_to_1_5_0 Porting from FLAC 1.4.3 to 1.5.0
* \ingroup porting
*
* \brief
* This module describes porting from FLAC 1.4.3 to FLAC 1.5.0.
*
* \section porting_1_4_3_to_1_5_0_summary Summary
*
* Between FLAC 1.4.3 and FLAC 1.5.0, there have been changes to
* existing C functions and enums:
* - the functions FLAC__metadata_get_streaminfo,
* FLAC__metadata_get_tags and FLAC__metadata_get_cuesheet can now
* read from Ogg FLAC files
* - when encoding Ogg FLAC files, the 'samples' argument of the
* write callback no longer always returns 0.
* - two error statuses have been added to the
* FLAC__StreamDecoderErrorStatus enum
* - one status is added to FLAC__StreamDecoderState, which is only
* used when the new decoding of chained streams functionality is
* enabled
* - one status is added to FLAC__StreamDecoderReadStatus, which is
* only used when the new decoding of chained streams functionality
* is enabled
* - the function
* FLAC__metadata_chain_write_with_callbacks_and_tempfile can now
* be used regardless of whether a tempfile is actually needed,
* which is useful when writing to a new file is desired anyway
*
* Furthermore, there have been the following additions:
* - the functions FLAC__stream_decoder_set_decode_chained_stream,
* FLAC__stream_decoder_get_decode_chained_stream,
* FLAC__stream_decoder_finish_link,
* FLAC__stream_decoder_skip_single_link,
* FLAC__stream_decoder_process_until_end_of_link and
* FLAC__stream_decoder_get_link_lengths have been added
* to support decoding of chained streams
* - the function FLAC__metadata_chain_write_new_file has been added,
* which is useful to combine copying of a file with changing its
* metadata
* - the function FLAC__stream_decoder_find_total_samples was added,
* which seeks to the end of a file to find the total number of
* samples
* - the function FLAC__stream_encoder_set_num_threads and
* FLAC__stream_encoder_get_num_threads have been added, which
* can be used to enable multithreading in the encoder
*
* For the C++ API, changes are the same, but in C++ nomenclature
*
*/
/** \defgroup flac FLAC C API
*
* The FLAC C API is the interface to libFLAC, a set of structures

2
ThirdParty/flac/include/FLAC/callback.h vendored
View file

@ -1,6 +1,6 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2004-2009 Josh Coalson
* Copyright (C) 2011-2023 Xiph.Org Foundation
* Copyright (C) 2011-2025 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions

6
ThirdParty/flac/include/FLAC/export.h vendored
View file

@ -1,6 +1,6 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2023 Xiph.Org Foundation
* Copyright (C) 2011-2025 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@ -95,9 +95,9 @@
/** These \#defines will mirror the libtool-based library version number, see
* http://www.gnu.org/software/libtool/manual/libtool.html#Libtool-versioning
*/
#define FLAC_API_VERSION_CURRENT 13
#define FLAC_API_VERSION_CURRENT 14
#define FLAC_API_VERSION_REVISION 0 /**< see above */
#define FLAC_API_VERSION_AGE 1 /**< see above */
#define FLAC_API_VERSION_AGE 0 /**< see above */
#ifdef __cplusplus
extern "C" {

2
ThirdParty/flac/include/FLAC/format.h vendored
View file

@ -1,6 +1,6 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2023 Xiph.Org Foundation
* Copyright (C) 2011-2025 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions

72
ThirdParty/flac/include/FLAC/metadata.h vendored
View file

@ -1,6 +1,6 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2001-2009 Josh Coalson
* Copyright (C) 2011-2023 Xiph.Org Foundation
* Copyright (C) 2011-2025 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@ -62,9 +62,9 @@
* This module provides functions for creating and manipulating FLAC
* metadata blocks in memory, and three progressively more powerful
* interfaces for traversing and editing metadata in native FLAC files.
* Note that currently only the Chain interface (level 2) supports Ogg
* FLAC files, and it is read-only i.e. no writing back changed
* metadata to file.
* Note that currently only the Chain interface (level 2) and some
* functions in level 0 supports Ogg FLAC files, and they are
* read-only i.e. no writing back changed metadata to file.
*
* There are three metadata interfaces of increasing complexity:
*
@ -144,10 +144,11 @@ extern "C" {
* \{
*/
/** Read the STREAMINFO metadata block of the given FLAC file. This function
* will try to skip any ID3v2 tag at the head of the file.
/** Read the STREAMINFO metadata block of the given FLAC or Ogg FLAC
* file. This function will try to skip any ID3v2 tag at the head
* of the file.
*
* \param filename The path to the FLAC file to read.
* \param filename The path to the FLAC or Ogg FLAC file to read.
* \param streaminfo A pointer to space for the STREAMINFO block. Since
* FLAC__StreamMetadata is a simple structure with no
* memory allocation involved, you pass the address of
@ -163,10 +164,11 @@ extern "C" {
*/
FLAC_API FLAC__bool FLAC__metadata_get_streaminfo(const char *filename, FLAC__StreamMetadata *streaminfo);
/** Read the VORBIS_COMMENT metadata block of the given FLAC file. This
* function will try to skip any ID3v2 tag at the head of the file.
/** Read the VORBIS_COMMENT metadata block of the given FLAC or Ogg FLAC
* file. This function will try to skip any ID3v2 tag at the head
* of the file.
*
* \param filename The path to the FLAC file to read.
* \param filename The path to the FLAC or Ogg FLAC file to read.
* \param tags The address where the returned pointer will be
* stored. The \a tags object must be deleted by
* the caller using FLAC__metadata_object_delete().
@ -182,10 +184,11 @@ FLAC_API FLAC__bool FLAC__metadata_get_streaminfo(const char *filename, FLAC__St
*/
FLAC_API FLAC__bool FLAC__metadata_get_tags(const char *filename, FLAC__StreamMetadata **tags);
/** Read the CUESHEET metadata block of the given FLAC file. This
* function will try to skip any ID3v2 tag at the head of the file.
/** Read the CUESHEET metadata block of the given FLAC or Ogg FLAC
* file. This function will try to skip any ID3v2 tag at the head
* of the file.
*
* \param filename The path to the FLAC file to read.
* \param filename The path to the FLAC or Ogg FLAC file to read.
* \param cuesheet The address where the returned pointer will be
* stored. The \a cuesheet object must be deleted by
* the caller using FLAC__metadata_object_delete().
@ -207,7 +210,8 @@ FLAC_API FLAC__bool FLAC__metadata_get_cuesheet(const char *filename, FLAC__Stre
* function takes a number of parameters that act as constraints to
* the search. The PICTURE block with the largest area matching all
* the constraints will be returned, or \a *picture will be set to
* \c NULL if there was no such block.
* \c NULL if there was no such block. This function does not
* currently support reading from Ogg FLAC files.
*
* \param filename The path to the FLAC file to read.
* \param picture The address where the returned pointer will be
@ -784,9 +788,6 @@ typedef enum {
/**< FLAC__metadata_chain_write_with_callbacks() was called when the
* chain write requires a tempfile; use
* FLAC__metadata_chain_write_with_callbacks_and_tempfile() instead.
* Or, FLAC__metadata_chain_write_with_callbacks_and_tempfile() was
* called when the chain write does not require a tempfile; use
* FLAC__metadata_chain_write_with_callbacks() instead.
* Always check FLAC__metadata_chain_check_if_tempfile_needed()
* before writing via callbacks. */
@ -923,11 +924,11 @@ FLAC_API FLAC__bool FLAC__metadata_chain_read_ogg_with_callbacks(FLAC__Metadata_
* edited metadata back to the FLAC file does not require rewriting the
* entire file. If rewriting is required, then a temporary workfile is
* required. When writing metadata using callbacks, you must check
* this function to know whether to call
* FLAC__metadata_chain_write_with_callbacks() or
* FLAC__metadata_chain_write_with_callbacks_and_tempfile(). When
* writing with FLAC__metadata_chain_write(), the temporary file is
* handled internally.
* this function to know whether
* FLAC__metadata_chain_write_with_callbacks() can be used or
* FLAC__metadata_chain_write_with_callbacks_and_tempfile() is
* necessary. When writing with FLAC__metadata_chain_write(), the
* temporary file is handled internally.
*
* \param chain A pointer to an existing chain.
* \param use_padding
@ -990,6 +991,28 @@ FLAC_API FLAC__bool FLAC__metadata_chain_check_if_tempfile_needed(FLAC__Metadata
*/
FLAC_API FLAC__bool FLAC__metadata_chain_write(FLAC__Metadata_Chain *chain, FLAC__bool use_padding, FLAC__bool preserve_file_stats);
/** Write all metadata out to a new FLAC file.
*
* This function works similar to FLAC__metadata_chain_write(), but is
* useful if writing to a new file is desired. This is more efficient
* than copying the file before changing it.
*
* For this write function to be used, the chain must have been read with
* FLAC__metadata_chain_read()/FLAC__metadata_chain_read_ogg(), not
* FLAC__metadata_chain_read_with_callbacks()/FLAC__metadata_chain_read_ogg_with_callbacks().
* See also FLAC__metadata_chain_write_with_callbacks_and_tempfile()
*
* \param chain A pointer to an existing chain.
* \param use_padding See FLAC__metadata_chain_write()
* \param filename The filename of the new file.
* \assert
* \code chain != NULL \endcode
* \retval FLAC__bool
* \c true if the write succeeded, else \c false. On failure,
* check the status with FLAC__metadata_chain_status().
*/
FLAC_API FLAC__bool FLAC__metadata_chain_write_new_file(FLAC__Metadata_Chain *chain, const char *filename, FLAC__bool use_padding);
/** Write all metadata out to a FLAC stream via callbacks.
*
* (See FLAC__metadata_chain_write() for the details on how padding is
@ -1031,7 +1054,8 @@ FLAC_API FLAC__bool FLAC__metadata_chain_write_with_callbacks(FLAC__Metadata_Cha
* FLAC file to edit, and a temporary handle to which the new FLAC
* file will be written. It is the caller's job to move this temporary
* FLAC file on top of the original FLAC file to complete the metadata
* edit.
* edit. This version of the write-with-callbacks function can also be
* used if writing to a new file is desired anyway.
*
* The \a handle must be open for reading and be seekable. The
* equivalent minimum stdio fopen() file mode is \c "r" (or \c "rb"
@ -1046,8 +1070,6 @@ FLAC_API FLAC__bool FLAC__metadata_chain_write_with_callbacks(FLAC__Metadata_Cha
* For this write function to be used, the chain must have been read with
* FLAC__metadata_chain_read_with_callbacks()/FLAC__metadata_chain_read_ogg_with_callbacks(),
* not FLAC__metadata_chain_read()/FLAC__metadata_chain_read_ogg().
* Also, FLAC__metadata_chain_check_if_tempfile_needed() must have returned
* \c true.
*
* \param chain A pointer to an existing chain.
* \param use_padding See FLAC__metadata_chain_write()

2
ThirdParty/flac/include/FLAC/ordinals.h vendored
View file

@ -1,6 +1,6 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2023 Xiph.Org Foundation
* Copyright (C) 2011-2025 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions

224
ThirdParty/flac/include/FLAC/stream_decoder.h vendored
View file

@ -1,6 +1,6 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2023 Xiph.Org Foundation
* Copyright (C) 2011-2025 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@ -235,12 +235,20 @@ typedef enum {
* state and can no longer be used.
*/
FLAC__STREAM_DECODER_UNINITIALIZED
FLAC__STREAM_DECODER_UNINITIALIZED,
/**< The decoder is in the uninitialized state; one of the
* FLAC__stream_decoder_init_*() functions must be called before samples
* can be processed.
*/
FLAC__STREAM_DECODER_END_OF_LINK
/**< The decoder has reached the end of an Ogg FLAC chain link and a new
* link follows; FLAC__stream_decoder_finish_link() has to be called to
* progress. This state is only returned when decoding of chained
* streams is enabled with
* FLAC__stream_decoder_set_decode_chained_stream()
*/
} FLAC__StreamDecoderState;
/** Maps a FLAC__StreamDecoderState to a C string.
@ -307,9 +315,14 @@ typedef enum {
* of \c 0.
*/
FLAC__STREAM_DECODER_READ_STATUS_ABORT
FLAC__STREAM_DECODER_READ_STATUS_ABORT,
/**< An unrecoverable error occurred. The decoder will return from the process call. */
FLAC__STREAM_DECODER_READ_STATUS_END_OF_LINK
/**< The read was attempted while at the end of the link. The semantics
* are the same as for FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM.
*/
} FLAC__StreamDecoderReadStatus;
/** Maps a FLAC__StreamDecoderReadStatus to a C string.
@ -442,9 +455,21 @@ typedef enum {
FLAC__STREAM_DECODER_ERROR_STATUS_UNPARSEABLE_STREAM,
/**< The decoder encountered reserved fields in use in the stream. */
FLAC__STREAM_DECODER_ERROR_STATUS_BAD_METADATA
FLAC__STREAM_DECODER_ERROR_STATUS_BAD_METADATA,
/**< The decoder encountered a corrupted metadata block. */
FLAC__STREAM_DECODER_ERROR_STATUS_OUT_OF_BOUNDS,
/**< The decoder encountered a otherwise valid frame in which
* the decoded samples exceeded the range offered by the stated
* bit depth. */
FLAC__STREAM_DECODER_ERROR_STATUS_MISSING_FRAME
/**< Two adjacent frames had frame numbers increasing by more than
* 1 or sample numbers increasing by more than the blocksize,
* indicating that one or more frame/frames was missing between
* them. The decoder will sent out one or more ´fake' constant
* subframes to fill up the gap. */
} FLAC__StreamDecoderErrorStatus;
/** Maps a FLAC__StreamDecoderErrorStatus to a C string.
@ -782,6 +807,33 @@ FLAC_API void FLAC__stream_decoder_delete(FLAC__StreamDecoder *decoder);
*/
FLAC_API FLAC__bool FLAC__stream_decoder_set_ogg_serial_number(FLAC__StreamDecoder *decoder, long serial_number);
/** Set the "decode chained stream" flag. If set to \c false, the
* decoder will stop decoding when it encounters the end-of-stream
* of the first link in the chain. If set to \c true, decoding will
* continue: at the start of each link, FLAC__stream_decoder_get_state
* will return FLAC__STREAM_DECODER_SEARCH_FOR_METADATA.
*
* This function was made to decode chained Ogg streams, but it should
* be possible to chain regular FLAC files through the read callback
* by using FLAC__STREAM_DECODER_READ_STATUS_END_OF_LINK
* appropriately.
*
* Note that when this flag is set to true, the serial number set
* with FLAC__stream_decoder_set_ogg_serial_number is ignored.
*
* \note
* This function has no effect with native FLAC decoding.
*
* \default \c false
* \param decoder A decoder instance to set.
* \param allow Whether to allow chained streams.
* \assert
* \code decoder != NULL \endcode
* \retval FLAC__bool
* \c false if the decoder is already initialized, else \c true.
*/
FLAC_API FLAC__bool FLAC__stream_decoder_set_decode_chained_stream(FLAC__StreamDecoder* decoder, FLAC__bool value);
/** Set the "MD5 signature checking" flag. If \c true, the decoder will
* compute the MD5 signature of the unencoded audio data while decoding
* and compare it to the signature from the STREAMINFO block, if it
@ -814,7 +866,7 @@ FLAC_API FLAC__bool FLAC__stream_decoder_set_md5_checking(FLAC__StreamDecoder *d
* \code decoder != NULL \endcode
* \a type is valid
* \retval FLAC__bool
* \c false if the decoder is already initialized, else \c true.
* \c false if type is invalid, else \c true.
*/
FLAC_API FLAC__bool FLAC__stream_decoder_set_metadata_respond(FLAC__StreamDecoder *decoder, FLAC__MetadataType type);
@ -829,7 +881,7 @@ FLAC_API FLAC__bool FLAC__stream_decoder_set_metadata_respond(FLAC__StreamDecode
* \code decoder != NULL \endcode
* \code id != NULL \endcode
* \retval FLAC__bool
* \c false if the decoder is already initialized, else \c true.
* \c false when memory allocation fails, else \c true.
*/
FLAC_API FLAC__bool FLAC__stream_decoder_set_metadata_respond_application(FLAC__StreamDecoder *decoder, const FLAC__byte id[4]);
@ -841,7 +893,7 @@ FLAC_API FLAC__bool FLAC__stream_decoder_set_metadata_respond_application(FLAC__
* \assert
* \code decoder != NULL \endcode
* \retval FLAC__bool
* \c false if the decoder is already initialized, else \c true.
* \c always \c true.
*/
FLAC_API FLAC__bool FLAC__stream_decoder_set_metadata_respond_all(FLAC__StreamDecoder *decoder);
@ -855,7 +907,7 @@ FLAC_API FLAC__bool FLAC__stream_decoder_set_metadata_respond_all(FLAC__StreamDe
* \code decoder != NULL \endcode
* \a type is valid
* \retval FLAC__bool
* \c false if the decoder is already initialized, else \c true.
* \c false if type is invalid, else \c true.
*/
FLAC_API FLAC__bool FLAC__stream_decoder_set_metadata_ignore(FLAC__StreamDecoder *decoder, FLAC__MetadataType type);
@ -870,7 +922,7 @@ FLAC_API FLAC__bool FLAC__stream_decoder_set_metadata_ignore(FLAC__StreamDecoder
* \code decoder != NULL \endcode
* \code id != NULL \endcode
* \retval FLAC__bool
* \c false if the decoder is already initialized, else \c true.
* \c false if memory allocation fails, else \c true.
*/
FLAC_API FLAC__bool FLAC__stream_decoder_set_metadata_ignore_application(FLAC__StreamDecoder *decoder, const FLAC__byte id[4]);
@ -882,7 +934,7 @@ FLAC_API FLAC__bool FLAC__stream_decoder_set_metadata_ignore_application(FLAC__S
* \assert
* \code decoder != NULL \endcode
* \retval FLAC__bool
* \c false if the decoder is already initialized, else \c true.
* \c always \c true.
*/
FLAC_API FLAC__bool FLAC__stream_decoder_set_metadata_ignore_all(FLAC__StreamDecoder *decoder);
@ -906,6 +958,17 @@ FLAC_API FLAC__StreamDecoderState FLAC__stream_decoder_get_state(const FLAC__Str
*/
FLAC_API const char *FLAC__stream_decoder_get_resolved_state_string(const FLAC__StreamDecoder *decoder);
/** Get the "decode chained stream" flag as described in
* \code FLAC__stream_decoder_set_decode_chained_stream \endcode.
*
* \param decoder A decoder instance to query.
* \assert
* \code decoder != NULL \endcode
* \retval FLAC__bool
* See above.
*/
FLAC_API FLAC__bool FLAC__stream_decoder_get_decode_chained_stream(const FLAC__StreamDecoder* decoder);
/** Get the "MD5 signature checking" flag.
* This is the value of the setting, not whether or not the decoder is
* currently checking the MD5 (remember, it can be turned off automatically
@ -932,6 +995,32 @@ FLAC_API FLAC__bool FLAC__stream_decoder_get_md5_checking(const FLAC__StreamDeco
*/
FLAC_API FLAC__uint64 FLAC__stream_decoder_get_total_samples(const FLAC__StreamDecoder *decoder);
/** Seek to the end of the file being decoded to find the total number
* of samples. This will return a number of samples even if
* FLAC__stream_decoder_get_total_samples() returns 0. It can also
* be used to find the total number of samples of a chained stream,
* as it returns the total number of samples in all chain links
* combined. See FLAC__stream_decoder_set_decode_chained_stream()
*
* For this function to work, the stream must be seekable. Also, as
* seeking can fail, this function returns 0 when it was unable to
* find the total number of samples. Use
* FLAC__stream_decoder_get_state() in that case to find out whether
* the decoder is still valid.
*
* The state in which the decoder is left after calling this function
* is undefined and might change in the future. Use
* FLAC__stream_decoder_reset() to return the decoder to a defined
* state.
*
* \param decoder A decoder instance to query.
* \assert
* \code decoder != NULL \endcode
* \retval uint32_t
* See above.
*/
FLAC_API FLAC__uint64 FLAC__stream_decoder_find_total_samples(FLAC__StreamDecoder *decoder);
/** Get the current number of channels in the stream being decoded.
* Will only be valid after decoding has started and will contain the
* value from the most recently decoded frame header.
@ -1023,6 +1112,43 @@ FLAC_API FLAC__bool FLAC__stream_decoder_get_decode_position(const FLAC__StreamD
*/
FLAC_API const void *FLAC__stream_decoder_get_client_data(FLAC__StreamDecoder *decoder);
#define FLAC__STREAM_DECODER_GET_LINK_LENGTHS_INVALID -1
#define FLAC__STREAM_DECODER_GET_LINK_LENGTHS_NOT_INDEXED -2
#define FLAC__STREAM_DECODER_GET_LINK_LENGTHS_MEMORY_ALLOCATION_ERROR -3
/** Get the link lengths in samples in a chained stream
*
* After either processing the whole file or using
* FLAC__stream_decoder_find_total_samples, this function
* returns an array with link lengths in samples. If it fails,
* it returns a negative number as error code, currently either
* FLAC__STREAM_DECODER_GET_LINK_LENGTHS_INVALID if the current
* decoder is not in a valid state or not processing a chained
* stream, FLAC__STREAM_DECODER_GET_LINK_LENGTHS_NOT_INDEXED if
* the stream hasn't been indexed yet, and
* FLAC__STREAM_DECODER_GET_LINK_LENGTHS_MEMORY_ALLOCATION_ERROR
* if allocating memory failed.
*
* If the function succeeds, the return value is the number of
* links. The link_lengths parameter is a FLAC__uint64 pointer
* which is allocated by the call, and must be freed by the user.
* If a null pointer is passed, the function only returns the
* number of links, not their lengths. The length of the returned
* array is equal to the number of links and thus to the return
* value of the function.
*
* \param decoder A decoder instance to query.
* \param link_lengths Address at which to return the link lengths
* array.
* \assert
* \code decoder != NULL \endcode
* \retval int32_t
* \c the number of links if successful, zero or a negative
* number if unsuccessful
*/
FLAC_API int32_t FLAC__stream_decoder_get_link_lengths(FLAC__StreamDecoder *decoder, FLAC__uint64 **link_lengths);
/** Initialize the decoder instance to decode native FLAC streams.
*
* This flavor of initialization sets up the decoder to decode from a
@ -1394,6 +1520,23 @@ FLAC_API FLAC__StreamDecoderInitStatus FLAC__stream_decoder_init_ogg_file(
*/
FLAC_API FLAC__bool FLAC__stream_decoder_finish(FLAC__StreamDecoder *decoder);
/** Finish the decoding process of the current link.
* Checks MD5 for current link and start processing of the next link. This
* function should only be used when the decoder state is
* FLAC__STREAM_DECODER_END_OF_LINK. After calling this function, the state
* is set to FLAC__STREAM_DECODER_SEARCH_FOR_METADATA.
*
* \param decoder An uninitialized decoder instance.
* \assert
* \code decoder != NULL \endcode
* \retval FLAC__bool
* \c false if MD5 checking is on AND a STREAMINFO block was available
* AND the MD5 signature in the STREAMINFO block was non-zero AND the
* signature does not match the one computed by the decoder; else
* \c true.
*/
FLAC_API FLAC__bool FLAC__stream_decoder_finish_link(FLAC__StreamDecoder *decoder);
/** Flush the stream input.
* The decoder's input buffer will be cleared and the state set to
* \c FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC. This will also turn
@ -1495,6 +1638,31 @@ FLAC_API FLAC__bool FLAC__stream_decoder_process_single(FLAC__StreamDecoder *dec
*/
FLAC_API FLAC__bool FLAC__stream_decoder_process_until_end_of_metadata(FLAC__StreamDecoder *decoder);
/** Decode until the end of the Ogg chain link.
* This version instructs the decoder to decode from the current
* position and continue until the end of link or until the
* callbacks return a fatal error. This function should not be
* used without enabling decoding of chained streams with
* FLAC__stream_decoder_set_decode_chained_stream()
*
* To start processing the next link after calling this function,
* call FLAC__stream_decoder_finish_link()
*
* As the decoder needs more input it will call the read callback.
* As each metadata block and frame is decoded, the metadata or write
* callback will be called with the decoded metadata or frame.
*
* \param decoder An initialized decoder instance.
* \assert
* \code decoder != NULL \endcode
* \retval FLAC__bool
* \c false if any fatal read, write, or memory allocation error
* occurred (meaning decoding must stop), else \c true; for more
* information about the decoder, check the decoder state with
* FLAC__stream_decoder_get_state().
*/
FLAC_API FLAC__bool FLAC__stream_decoder_process_until_end_of_link(FLAC__StreamDecoder *decoder);
/** Decode until the end of the stream.
* This version instructs the decoder to decode from the current position
* and continue until the end of stream (the read callback returns
@ -1557,6 +1725,34 @@ FLAC_API FLAC__bool FLAC__stream_decoder_process_until_end_of_stream(FLAC__Strea
*/
FLAC_API FLAC__bool FLAC__stream_decoder_skip_single_frame(FLAC__StreamDecoder *decoder);
/** Skip one Ogg chain link.
* This version instructs the decoder to 'skip' the current ogg chain
* link. This function should not be used without enabling decoding
* of chained streams with
* FLAC__stream_decoder_set_decode_chained_stream()
*
* When the location ot the next link is already known (for example,
* because there has already been a seek in the stream) this function
* will seek to the end of the link and start processing, but when it
* is not, it will do various seeks (when possible) to find the end
* of the link. If seeking is not possible (or the tell and length
* callback do not work) it will simply read forward instead of
* seeking, much like a muted (but faster) version of
* FLAC__stream_decoder_process_until_end_of_link().
*
* \param decoder An initialized decoder instance not in a metadata
* state.
* \assert
* \code decoder != NULL \endcode
* \retval FLAC__bool
* \c false if any fatal read, write, or memory allocation error
* occurred (meaning decoding must stop), or when decoding a format
* that does not support chaining, else \c true; for more
* information about the decoder, check the decoder state with
* FLAC__stream_decoder_get_state().
*/
FLAC_API FLAC__bool FLAC__stream_decoder_skip_single_link(FLAC__StreamDecoder *decoder);
/** Flush the input and seek to an absolute sample.
* Decoding will resume at the given sample. Note that because of
* this, the next write callback may contain a partial block. The
@ -1566,6 +1762,14 @@ FLAC_API FLAC__bool FLAC__stream_decoder_skip_single_frame(FLAC__StreamDecoder *
* with FLAC__stream_decoder_flush() or reset with
* FLAC__stream_decoder_reset() before decoding can continue.
*
* When seeking in a chained stream with decoding of such streams
* enabled with FLAC__stream_decoder_set_decode_chained_stream(),
* this function seeks in the whole stream, over all links. When
* a seek to another link is performed, the decoder will also
* return metadata blocks of that link. If this is not desired,
* use FLAC__stream_decoder_set_metadata_ignore_all() before
* seeking.
*
* \param decoder A decoder instance.
* \param sample The target sample number to seek to.
* \assert

79
ThirdParty/flac/include/FLAC/stream_encoder.h vendored
View file

@ -1,6 +1,6 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2023 Xiph.Org Foundation
* Copyright (C) 2011-2025 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@ -288,6 +288,11 @@ typedef enum {
extern FLAC_API const char * const FLAC__StreamEncoderStateString[];
#define FLAC__STREAM_ENCODER_SET_NUM_THREADS_OK 0
#define FLAC__STREAM_ENCODER_SET_NUM_THREADS_NOT_COMPILED_WITH_MULTITHREADING_ENABLED 1
#define FLAC__STREAM_ENCODER_SET_NUM_THREADS_ALREADY_INITIALIZED 2
#define FLAC__STREAM_ENCODER_SET_NUM_THREADS_TOO_MANY_THREADS 3
/** Possible return values for the FLAC__stream_encoder_init_*() functions.
*/
typedef enum {
@ -533,11 +538,17 @@ typedef FLAC__StreamEncoderReadStatus (*FLAC__StreamEncoderReadCallback)(const F
* callback is being called to write metadata.
*
* \note
* Unlike when writing to native FLAC, when writing to Ogg FLAC the
* write callback will be called twice when writing each audio
* frame; once for the page header, and once for the page body.
* When writing the page header, the \a samples argument to the
* write callback will be \c 0.
* Unlike when writing to native FLAC, when writing to Ogg FLAC the write
* callback will be called at least twice when writing each audio frame; once
* for the page header and once for the page body, possibly repeating this
* pair of calls several times in a batch with the same value of
* \a current_frame. When writing the page header, as well as in all but the
* first page body write of the batch, the \a samples argument to the write
* callback will be \c 0. For the write callback call containing the first
* page body, the \a samples argument is the number of samples contained in
* all newly added complete packets (not pages). This means that in case a
* packet is split over two pages, they are counted in the samples argument
* of the page on which the packet is completed.
*
* \note In general, FLAC__StreamEncoder functions which change the
* state should not be called on the \a encoder while in the callback.
@ -1102,6 +1113,52 @@ FLAC_API FLAC__bool FLAC__stream_encoder_set_min_residual_partition_order(FLAC__
*/
FLAC_API FLAC__bool FLAC__stream_encoder_set_max_residual_partition_order(FLAC__StreamEncoder *encoder, uint32_t value);
/** Set the maximum number of threads to use during encoding.
* Set to a value different than 1 to enable multithreaded encoding.
*
* Note that enabling multithreading encoding (i.e., setting a value
* different than 1) results in the behaviour of
* FLAC__stream_encoder_finish(), FLAC__stream_encoder_process(),
* FLAC__stream_encoder_process_interleaved() subtly changing.
* For example, calling one of the process functions with enough
* samples to fill a block might not always result in a call to
* the write callback with a frame coding these samples. Instead,
* blocks with samples are distributed among worker threads,
* meaning that the first few calls to those functions will
* return very quickly, while later calls might block if all
* threads are occupied. Also, certain calls to the process
* functions will results in several calls to the write callback,
* while subsequent calls might again return very quickly with no
* calls to the write callback.
*
* Also, a call to FLAC__stream_encoder_finish() blocks while
* waiting for all threads to finish, and therefore might take much
* longer than when not multithreading and result in multiple calls
* to the write callback.
*
* Calls to the write callback are guaranteed to be in the correct
* order.
*
* Currently, passing a value of 0 is synonymous with a value of 1,
* but this might change in the future.
*
* \default \c 1
* \param encoder An encoder instance to set.
* \param value See above.
* \assert
* \code encoder != NULL \endcode
* \retval uint32_t
* - \c FLAC__STREAM_ENCODER_SET_NUM_THREADS_OK if the number of threads was set correctly,
* - \c FLAC__STREAM_ENCODER_SET_NUM_THREADS_NOT_COMPILED_WITH_MULTITHREADING_ENABLED when
* multithreading was not enabled at compilation,
* - \c FLAC__STREAM_ENCODER_SET_NUM_THREADS_ALREADY_INITIALIZED when the encoder was
* already initialized,
* - \c FLAC__STREAM_ENCODER_SET_NUM_THREADS_TOO_MANY_THREADS when
* the number of threads was larger than the maximum allowed number of threads (currently
* 128).
*/
FLAC_API uint32_t FLAC__stream_encoder_set_num_threads(FLAC__StreamEncoder *encoder, uint32_t value);
/** Deprecated. Setting this value has no effect.
*
* \default \c 0
@ -1435,6 +1492,16 @@ FLAC_API uint32_t FLAC__stream_encoder_get_min_residual_partition_order(const FL
*/
FLAC_API uint32_t FLAC__stream_encoder_get_max_residual_partition_order(const FLAC__StreamEncoder *encoder);
/** Get maximum number of threads setting.
*
* \param encoder An encoder instance to query.
* \assert
* \code encoder != NULL \endcode
* \retval uint32_t
* See FLAC__stream_encoder_set_num_threads().
*/
FLAC_API uint32_t FLAC__stream_encoder_get_num_threads(const FLAC__StreamEncoder *encoder);
/** Get the Rice parameter search distance setting.
*
* \param encoder An encoder instance to query.

BIN
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11
ThirdParty/libvgm/README.md vendored
View file

@ -2,12 +2,17 @@ This is verbatim from the following repository:
https://github.com/ValleyBell/libvgm.git
Built on an M1 Mac mini, using CMake from Homebrew, with the following
Built on an M4 Mac mini, using CMake from Homebrew, with the following
options:
```
cmake .. -DCMAKE_OSX_ARCHITECTURES="x86_64;arm64" -DCMAKE_OSX_DEPLOYMENT_TARGET="10.13" \
-DBUILD_LIBAUDIO=NO -DBUILD_PLAYER=NO -DBUILD_VGM2WAV=NO
-DCMAKE_BUILD_TYPE=RelWithDebInfo -DBUILD_LIBAUDIO=NO -DBUILD_PLAYER=NO -DBUILD_VGM2WAV=NO
```
As of this edit, commit: 82ba45d3906a0b54b6de2555468dd9e9598f617d
And the debug overlays were made with the above, except for:
```
-DCMAKE_BUILD_TYPE=Debug
```
As of this edit, commit: 7cad78367fa35c3f7b3ae16a296d31063cd3a7e4

1
ThirdParty/libvgm/include/libvgm/emu/EmuCores.h vendored
View file

@ -17,5 +17,6 @@
#define FCC_CTR_ 0x43545200 // superctr
#define FCC_LAOO 0x4C414F4F // laoo
#define FCC_RN22 0x524E3232 // Mao757 (RN22)
#define FCC_EITO 0x4549544F // eito
#endif // __EMUCORES_H__

4
ThirdParty/libvgm/include/libvgm/emu/SoundDevs.h vendored
View file

@ -44,6 +44,6 @@
#define DEVID_GA20 0x28
#define DEVID_MIKEY 0x29
#define DEVID_K007232 0x2A
#define DEVID_K005289 0x2B
#define DEVID_K005289 0x2B
#define DEVID_MSM5205 0x2C // variants: MSM5205, MSM6585
#endif // __SOUNDDEVS_H__

5
ThirdParty/libvgm/include/libvgm/player/vgmplayer.hpp vendored
View file

@ -248,6 +248,7 @@ protected:
void Cmd_GGStereo(void); // command 4F - set GameGear Stereo mask
void Cmd_SN76489(void); // command 50 - SN76489 register write
void Cmd_Reg8_Data8(void); // command 51/54/55/5A..5D - Register, Data (8-bit)
void Cmd_MSM5205_Reg(void); // command 32 - MSM5205 register write (4-bit offset, 4-bit data)
void Cmd_CPort_Reg8_Data8(void); // command 52/53/56..59/5E/5F - Port (in command byte), Register, Data (8-bit)
void Cmd_Port_Reg8_Data8(void); // command D0..D2 - Port, Register, Data (8-bit)
void Cmd_Ofs8_Data8(void); // command B3/B5..BB/BE/BF - Offset (8-bit), Data (8-bit)
@ -281,8 +282,8 @@ protected:
enum
{
_HDR_BUF_SIZE = 0x100,
_OPT_DEV_COUNT = 0x2c,
_CHIP_COUNT = 0x2c,
_OPT_DEV_COUNT = 0x2d,
_CHIP_COUNT = 0x2d,
_PCM_BANK_COUNT = 0x40
};

8
ThirdParty/ogg/README.md vendored
View file

@ -1,10 +1,10 @@
Build with CMake, using the following options:
```
cmake .. -DCMAKE_OSX_ARCHITECTURES="x86_64;arm64" -DCMAKE_OSX_DEPLOYMENT_TARGET="10.12" -DBUILD_SHARED_LIBS=ON
cmake .. -DCMAKE_OSX_ARCHITECTURES="x86_64;arm64" -DCMAKE_OSX_DEPLOYMENT_TARGET="10.12" \
-DCMAKE_BUILD_TYPE=RelWithDebInfo -DBUILD_SHARED_LIBS=ON
```
And some minor tweaks with `install_name_tool -id` to make sure that the
resulting libogg.0.dylib referred to itself with @rpath.
And for the debug overlay, replacing the BUILD_TYPE with `Debug`.
Version 1.3.5 was used.
Version 1.3.6 was used.

9
ThirdParty/opus/README.md vendored
View file

@ -1,13 +1,12 @@
Build with CMake, using the following options:
```
cmake .. -DCMAKE_OSX_ARCHITECTURES="x86_64;arm64" -DCMAKE_OSX_DEPLOYMENT_TARGET="10.12" -DBUILD_SHARED_LIBS=ON
cmake .. -DCMAKE_OSX_ARCHITECTURES="x86_64;arm64" -DCMAKE_OSX_DEPLOYMENT_TARGET="10.13" \
-DCMAKE_BUILD_TYPE=RelWithDebInfo -DBUILD_SHARED_LIBS=ON
```
And some minor tweaks with `install_name_tool -id` to make sure that the
resulting libopus.0.dylib referred to itself with @rpath and not full
paths of the build directory.
The debug overlay was built the same, except with BUILD_TYPE set to `Debug`.
Version v1.3.1-101-g66d060c7 was used from the following repository:
Version v1.5.2-153-g7aa5be98 was used from the following repository:
https://github.com/xiph/opus.git

199
ThirdParty/opus/include/opus.h vendored
View file

@ -103,7 +103,7 @@ extern "C" {
* @endcode
*
* where opus_encoder_get_size() returns the required size for the encoder state. Note that
* future versions of this code may change the size, so no assuptions should be made about it.
* future versions of this code may change the size, so no assumptions should be made about it.
*
* The encoder state is always continuous in memory and only a shallow copy is sufficient
* to copy it (e.g. memcpy())
@ -198,7 +198,7 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
* @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
* @param [in] application <tt>int</tt>: Coding mode (one of @ref OPUS_APPLICATION_VOIP, @ref OPUS_APPLICATION_AUDIO, or @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
* @param [out] error <tt>int*</tt>: @ref opus_errorcodes
* @note Regardless of the sampling rate and number channels selected, the Opus encoder
* can switch to a lower audio bandwidth or number of channels if the bitrate
@ -222,7 +222,7 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
* @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
* @param [in] application <tt>int</tt>: Coding mode (one of OPUS_APPLICATION_VOIP, OPUS_APPLICATION_AUDIO, or OPUS_APPLICATION_RESTRICTED_LOWDELAY)
* @retval #OPUS_OK Success or @ref opus_errorcodes
*/
OPUS_EXPORT int opus_encoder_init(
@ -268,6 +268,42 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes an Opus frame.
* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
* @param [in] pcm <tt>opus_int32*</tt>: Input signal (interleaved if 2 channels) representing (or slightly exceeding) 24-bit values. length is frame_size*channels*sizeof(opus_int32)
* @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
* input signal.
* This must be an Opus frame size for
* the encoder's sampling rate.
* For example, at 48 kHz the permitted
* values are 120, 240, 480, 960, 1920,
* and 2880.
* Passing in a duration of less than
* 10 ms (480 samples at 48 kHz) will
* prevent the encoder from using the LPC
* or hybrid modes.
* @param [out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode24(
OpusEncoder *st,
const opus_int32 *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes an Opus frame from floating point input.
* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
* @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
@ -357,7 +393,7 @@ OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NON
* error = opus_decoder_init(dec, Fs, channels);
* @endcode
* where opus_decoder_get_size() returns the required size for the decoder state. Note that
* future versions of this code may change the size, so no assuptions should be made about it.
* future versions of this code may change the size, so no assumptions should be made about it.
*
* The decoder state is always continuous in memory and only a shallow copy is sufficient
* to copy it (e.g. memcpy())
@ -398,6 +434,21 @@ OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NON
*/
typedef struct OpusDecoder OpusDecoder;
/** Opus DRED decoder.
* This contains the complete state of an Opus DRED decoder.
* It is position independent and can be freely copied.
* @see opus_dred_decoder_create,opus_dred_decoder_init
*/
typedef struct OpusDREDDecoder OpusDREDDecoder;
/** Opus DRED state.
* This contains the complete state of an Opus DRED packet.
* It is position independent and can be freely copied.
* @see opus_dred_create,opus_dred_init
*/
typedef struct OpusDRED OpusDRED;
/** Gets the size of an <code>OpusDecoder</code> structure.
* @param [in] channels <tt>int</tt>: Number of channels.
* This must be 1 or 2.
@ -468,6 +519,31 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode an Opus packet.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
* @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
* @param [out] pcm <tt>opus_int32*</tt>: Output signal (interleaved if 2 channels) representing (or slightly exceeding) 24-bit values. length
* is frame_size*channels*sizeof(opus_int32)
* @param [in] frame_size Number of samples per channel of available space in \a pcm.
* If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
* not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
* then frame_size needs to be exactly the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
* FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
* @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
* decoded. If no such data is available, the frame is decoded as if it were lost.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode24(
OpusDecoder *st,
const unsigned char *data,
opus_int32 len,
opus_int32 *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode an Opus packet with floating point output.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
@ -511,6 +587,113 @@ OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NON
*/
OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
/** Gets the size of an <code>OpusDREDDecoder</code> structure.
* @returns The size in bytes.
*/
OPUS_EXPORT int opus_dred_decoder_get_size(void);
/** Allocates and initializes an OpusDREDDecoder state.
* @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
*/
OPUS_EXPORT OpusDREDDecoder *opus_dred_decoder_create(int *error);
/** Initializes an <code>OpusDREDDecoder</code> state.
* @param[in] dec <tt>OpusDREDDecoder*</tt>: State to be initialized.
*/
OPUS_EXPORT int opus_dred_decoder_init(OpusDREDDecoder *dec);
/** Frees an <code>OpusDREDDecoder</code> allocated by opus_dred_decoder_create().
* @param[in] dec <tt>OpusDREDDecoder*</tt>: State to be freed.
*/
OPUS_EXPORT void opus_dred_decoder_destroy(OpusDREDDecoder *dec);
/** Perform a CTL function on an Opus DRED decoder.
*
* Generally the request and subsequent arguments are generated
* by a convenience macro.
* @param dred_dec <tt>OpusDREDDecoder*</tt>: DRED Decoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls or
* @ref opus_decoderctls.
* @see opus_genericctls
* @see opus_decoderctls
*/
OPUS_EXPORT int opus_dred_decoder_ctl(OpusDREDDecoder *dred_dec, int request, ...);
/** Gets the size of an <code>OpusDRED</code> structure.
* @returns The size in bytes.
*/
OPUS_EXPORT int opus_dred_get_size(void);
/** Allocates and initializes a DRED state.
* @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
*/
OPUS_EXPORT OpusDRED *opus_dred_alloc(int *error);
/** Frees an <code>OpusDRED</code> allocated by opus_dred_create().
* @param[in] dec <tt>OpusDRED*</tt>: State to be freed.
*/
OPUS_EXPORT void opus_dred_free(OpusDRED *dec);
/** Decode an Opus DRED packet.
* @param [in] dred_dec <tt>OpusDRED*</tt>: DRED Decoder state
* @param [in] dred <tt>OpusDRED*</tt>: DRED state
* @param [in] data <tt>char*</tt>: Input payload
* @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
* @param [in] max_dred_samples <tt>opus_int32</tt>: Maximum number of DRED samples that may be needed (if available in the packet).
* @param [in] sampling_rate <tt>opus_int32</tt>: Sampling rate used for max_dred_samples argument. Needs not match the actual sampling rate of the decoder.
* @param [out] dred_end <tt>opus_int32*</tt>: Number of non-encoded (silence) samples between the DRED timestamp and the last DRED sample.
* @param [in] defer_processing <tt>int</tt>: Flag (0 or 1). If set to one, the CPU-intensive part of the DRED decoding is deferred until opus_dred_process() is called.
* @returns Offset (positive) of the first decoded DRED samples, zero if no DRED is present, or @ref opus_errorcodes
*/
OPUS_EXPORT int opus_dred_parse(OpusDREDDecoder *dred_dec, OpusDRED *dred, const unsigned char *data, opus_int32 len, opus_int32 max_dred_samples, opus_int32 sampling_rate, int *dred_end, int defer_processing) OPUS_ARG_NONNULL(1);
/** Finish decoding an Opus DRED packet. The function only needs to be called if opus_dred_parse() was called with defer_processing=1.
* The source and destination will often be the same DRED state.
* @param [in] dred_dec <tt>OpusDRED*</tt>: DRED Decoder state
* @param [in] src <tt>OpusDRED*</tt>: Source DRED state to start the processing from.
* @param [out] dst <tt>OpusDRED*</tt>: Destination DRED state to store the updated state after processing.
* @returns @ref opus_errorcodes
*/
OPUS_EXPORT int opus_dred_process(OpusDREDDecoder *dred_dec, const OpusDRED *src, OpusDRED *dst);
/** Decode audio from an Opus DRED packet with 16-bit output.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state
* @param [in] dred <tt>OpusDRED*</tt>: DRED state
* @param [in] dred_offset <tt>opus_int32</tt>: position of the redundancy to decode (in samples before the beginning of the real audio data in the packet).
* @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
* is frame_size*channels*sizeof(opus_int16)
* @param [in] frame_size Number of samples per channel to decode in \a pcm.
* frame_size <b>must</b> be a multiple of 2.5 ms.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_EXPORT int opus_decoder_dred_decode(OpusDecoder *st, const OpusDRED *dred, opus_int32 dred_offset, opus_int16 *pcm, opus_int32 frame_size);
/** Decode audio from an Opus DRED packet with 24-bit output.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state
* @param [in] dred <tt>OpusDRED*</tt>: DRED state
* @param [in] dred_offset <tt>opus_int32</tt>: position of the redundancy to decode (in samples before the beginning of the real audio data in the packet).
* @param [out] pcm <tt>opus_int32*</tt>: Output signal (interleaved if 2 channels). length
* is frame_size*channels*sizeof(opus_int16)
* @param [in] frame_size Number of samples per channel to decode in \a pcm.
* frame_size <b>must</b> be a multiple of 2.5 ms.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_EXPORT int opus_decoder_dred_decode24(OpusDecoder *st, const OpusDRED *dred, opus_int32 dred_offset, opus_int32 *pcm, opus_int32 frame_size);
/** Decode audio from an Opus DRED packet with floating point output.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state
* @param [in] dred <tt>OpusDRED*</tt>: DRED state
* @param [in] dred_offset <tt>opus_int32</tt>: position of the redundancy to decode (in samples before the beginning of the real audio data in the packet).
* @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
* is frame_size*channels*sizeof(float)
* @param [in] frame_size Number of samples per channel to decode in \a pcm.
* frame_size <b>must</b> be a multiple of 2.5 ms.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_EXPORT int opus_decoder_dred_decode_float(OpusDecoder *st, const OpusDRED *dred, opus_int32 dred_offset, float *pcm, opus_int32 frame_size);
/** Parse an opus packet into one or more frames.
* Opus_decode will perform this operation internally so most applications do
* not need to use this function.
@ -583,6 +766,14 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
/** Checks whether an Opus packet has LBRR.
* @param [in] packet <tt>char*</tt>: Opus packet
* @param [in] len <tt>opus_int32</tt>: Length of packet
* @returns 1 is LBRR is present, 0 otherwise
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_has_lbrr(const unsigned char packet[], opus_int32 len);
/** Gets the number of samples of an Opus packet.
* @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
* @param [in] packet <tt>char*</tt>: Opus packet

41
ThirdParty/opus/include/opus_custom.h vendored
View file

@ -104,7 +104,8 @@ typedef struct OpusCustomDecoder OpusCustomDecoder;
/** The mode contains all the information necessary to create an
encoder. Both the encoder and decoder need to be initialized
with exactly the same mode, otherwise the output will be
corrupted.
corrupted. The mode MUST NOT BE DESTROYED until the encoders and
decoders that use it are destroyed as well.
@brief Mode configuration
*/
typedef struct OpusCustomMode OpusCustomMode;
@ -229,6 +230,27 @@ OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode(
int maxCompressedBytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes a frame of audio.
* @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
* @param [in] pcm <tt>opus_int32*</tt>: PCM audio in signed 32-bit format (native endian) representing (or slightly exceeding) 24-bit values.
* There must be exactly frame_size samples per channel.
* @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
* @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
* @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
* (can change from one frame to another)
* @return Number of bytes written to "compressed".
* If negative, an error has occurred (see error codes). It is IMPORTANT that
* the length returned be somehow transmitted to the decoder. Otherwise, no
* decoding is possible.
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode24(
OpusCustomEncoder *st,
const opus_int32 *pcm,
int frame_size,
unsigned char *compressed,
int maxCompressedBytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on an Opus custom encoder.
*
* Generally the request and subsequent arguments are generated
@ -325,6 +347,23 @@ OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode(
int frame_size
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode an opus custom frame
* @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
* @param [in] len <tt>int</tt>: Number of bytes in payload
* @param [out] pcm <tt>opus_int32*</tt>: Output signal (interleaved if 2 channels) representing (or slightly exceeding) 24-bit values. length
* is frame_size*channels*sizeof(opus_int32)
* @param [in] frame_size Number of samples per channel of available space in *pcm.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode24(
OpusCustomDecoder *st,
const unsigned char *data,
int len,
opus_int32 *pcm,
int frame_size
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on an Opus custom decoder.
*
* Generally the request and subsequent arguments are generated

35
ThirdParty/opus/include/opus_defines.h vendored
View file

@ -169,15 +169,32 @@ extern "C" {
#define OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 4046
#define OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST 4047
#define OPUS_GET_IN_DTX_REQUEST 4049
#define OPUS_SET_DRED_DURATION_REQUEST 4050
#define OPUS_GET_DRED_DURATION_REQUEST 4051
#define OPUS_SET_DNN_BLOB_REQUEST 4052
/*#define OPUS_GET_DNN_BLOB_REQUEST 4053 */
/** Defines for the presence of extended APIs. */
#define OPUS_HAVE_OPUS_PROJECTION_H
/* Macros to trigger compilation errors when the wrong types are provided to a CTL */
#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x))
#ifdef DISABLE_PTR_CHECK
/* Disable checks to prevent ubsan from complaining about NULL checks
in test_opus_api. */
#define __opus_check_int_ptr(ptr) (ptr)
#define __opus_check_uint_ptr(ptr) (ptr)
#define __opus_check_uint8_ptr(ptr) (ptr)
#define __opus_check_val16_ptr(ptr) (ptr)
#define __opus_check_void_ptr(ptr) (ptr)
#else
#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr)))
#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr)))
#define __opus_check_uint8_ptr(ptr) ((ptr) + ((ptr) - (opus_uint8*)(ptr)))
#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr)))
#define __opus_check_void_ptr(x) ((void)((void *)0 == (x)), (x))
#endif
/** @endcond */
/** @defgroup opus_ctlvalues Pre-defined values for CTL interface
@ -482,7 +499,8 @@ extern "C" {
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Disable inband FEC (default).</dd>
* <dt>1</dt><dd>Enable inband FEC.</dd>
* <dt>1</dt><dd>Inband FEC enabled. If the packet loss rate is sufficiently high, Opus will automatically switch to SILK even at high rates to enable use of that FEC.</dd>
* <dt>2</dt><dd>Inband FEC enabled, but does not necessarily switch to SILK if we have music.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x)
@ -491,7 +509,8 @@ extern "C" {
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Inband FEC disabled (default).</dd>
* <dt>1</dt><dd>Inband FEC enabled.</dd>
* <dt>1</dt><dd>Inband FEC enabled. If the packet loss rate is sufficiently high, Opus will automatically switch to SILK even at high rates to enable use of that FEC.</dd>
* <dt>2</dt><dd>Inband FEC enabled, but does not necessarily switch to SILK if we have music.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x)
@ -618,6 +637,18 @@ extern "C" {
* @hideinitializer */
#define OPUS_GET_PREDICTION_DISABLED(x) OPUS_GET_PREDICTION_DISABLED_REQUEST, __opus_check_int_ptr(x)
/** If non-zero, enables Deep Redundancy (DRED) and use the specified maximum number of 10-ms redundant frames
* @hideinitializer */
#define OPUS_SET_DRED_DURATION(x) OPUS_SET_DRED_DURATION_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured Deep Redundancy (DRED) maximum number of frames.
* @hideinitializer */
#define OPUS_GET_DRED_DURATION(x) OPUS_GET_DRED_DURATION_REQUEST, __opus_check_int_ptr(x)
/** Provide external DNN weights from binary object (only when explicitly built without the weights)
* @hideinitializer */
#define OPUS_SET_DNN_BLOB(data, len) OPUS_SET_DNN_BLOB_REQUEST, __opus_check_void_ptr(data), __opus_check_int(len)
/**@}*/
/** @defgroup opus_genericctls Generic CTLs

80
ThirdParty/opus/include/opus_multistream.h vendored
View file

@ -143,7 +143,7 @@ extern "C" {
* <a href="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">Vorbis
* channel ordering</a>. A decoder may wish to apply an additional permutation
* to the mapping the encoder used to achieve a different output channel
* order (e.g. for outputing in WAV order).
* order (e.g. for outputting in WAV order).
*
* Each multistream packet contains an Opus packet for each stream, and all of
* the Opus packets in a single multistream packet must have the same
@ -382,6 +382,44 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode(
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes a multistream Opus frame.
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
* @param[in] pcm <tt>const opus_int32*</tt>: The input signal as interleaved
* samples representing (or slightly exceeding) 24-bit values.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode24(
OpusMSEncoder *st,
const opus_int32 *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes a multistream Opus frame from floating point input.
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
* @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
@ -510,7 +548,7 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_crea
int *error
) OPUS_ARG_NONNULL(5);
/** Intialize a previously allocated decoder state object.
/** Initialize a previously allocated decoder state object.
* The memory pointed to by \a st must be at least the size returned by
* opus_multistream_encoder_get_size().
* This is intended for applications which use their own allocator instead of
@ -591,6 +629,44 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode(
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode a multistream Opus packet.
* @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int32*</tt>: Output signal, with interleaved
* samples representing (or slightly exceeding) 24-bit values.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode24(
OpusMSDecoder *st,
const unsigned char *data,
opus_int32 len,
opus_int32 *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode a multistream Opus packet with floating point output.
* @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.

77
ThirdParty/opus/include/opus_projection.h vendored
View file

@ -260,6 +260,44 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode(
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes a projection Opus frame.
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
* @param[in] pcm <tt>const opus_int32*</tt>: The input signal as interleaved
* samples representing (or slightly exceeding) 24-bit values.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode24(
OpusProjectionEncoder *st,
const opus_int32 *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes a projection Opus frame from floating point input.
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
@ -405,7 +443,7 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionDecoder *opus_projection_decod
) OPUS_ARG_NONNULL(5);
/** Intialize a previously allocated projection decoder state object.
/** Initialize a previously allocated projection decoder state object.
* The memory pointed to by \a st must be at least the size returned by
* opus_projection_decoder_get_size().
* This is intended for applications which use their own allocator instead of
@ -493,6 +531,43 @@ OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode(
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode a projection Opus packet.
* @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int32*</tt>: Output signal, with interleaved
* samples representing (or slightly exceeding) 24-bit values.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode24(
OpusProjectionDecoder *st,
const unsigned char *data,
opus_int32 len,
opus_int32 *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode a projection Opus packet with floating point output.
* @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.

19
ThirdParty/opusfile/README.md vendored
View file

@ -1,13 +1,22 @@
Build with CMake, using the following options:
```
cmake .. -DCMAKE_OSX_ARCHITECTURES="x86_64;arm64" -DCMAKE_OSX_DEPLOYMENT_TARGET="10.12" -DBUILD_SHARED_LIBS=ON
env PATH=/usr/local/bin:$PATH cmake -B build-x86 -DCMAKE_LIBRARY_PATH=/usr/local/lib \
-DCMAKE_OSX_ARCHITECTURES="x86_64" -DCMAKE_OSX_DEPLOYMENT_TARGET="10.13" \
-DCMAKE_BUILD_TYPE=RelWithDebInfo -DBUILD_SHARED_LIBS=ON -DOP_DISABLE_HTTP=ON -DOP_DISABLE_DOCS=ON
cmake build-arm -DCMAKE_OSX_ARCHITECTURES="arm64" -DCMAKE_OSX_DEPLOYMENT_TARGET="11.0" \
-DCMAKE_BUILD_TYPE=RelWithDebInfo -DBUILD_SHARED_LIBS=ON -DOP_DISABLE_HTTP=ON \
-DOP_DISABLE_DOCS=ON
```
And some minor tweaks with `install_name_tool -id` to make sure that the
resulting libopusfile.0.dylib referred to itself with @rpath and not full
paths of the build directory.
The debug overlay was built the same, except with BUILD_TYPE set to `Debug`.
Version v0.12-38-gcf218fb was used from the following repository:
The CMakeCache.txt also needed hacking to replace the libopus with our own build for compatible version
numbers in the import.
Finally, `install_name_tool -change` was needed to change the import path for libogg. The version
numbers were already compatible.
Version v0.12-51-g24d6e75 was used from the following repository:
https://github.com/xiph/opusfile.git

2
ThirdParty/opusfile/include/opusfile.h vendored
View file

@ -1816,7 +1816,7 @@ void op_set_decode_callback(OggOpusFile *_of,
#define OP_TRACK_GAIN (3008)
/**Gain offset type that indicates that the provided offset should be used as
the gain directly, without applying any the header or track gains.*/
the gain directly, without applying any of the header or track gains.*/
#define OP_ABSOLUTE_GAIN (3009)
/**Sets the gain to be used for decoded output.