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Revert to previous low latency output system

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This reverts usage of the AVFoundation output to use
the previous lower latency CoreAudio output, and
paves the way for a change I am cooking up soon.

Fixes several issues with playback and seeking latency.

Signed-off-by: Christopher Snowhill <kode54@gmail.com>
This commit is contained in:
Christopher Snowhill 2023-10-02 07:04:26 -07:00
parent ca1f9381b5
commit 7ac32284ff
No known key found for this signature in database
16 changed files with 1184 additions and 226 deletions
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6
Audio/AudioPlayer.m
View file

@ -91,7 +91,7 @@
bufferChain = [[BufferChain alloc] initWithController:self];
[self notifyStreamChanged:userInfo];
while(![bufferChain open:url withUserInfo:userInfo withRGInfo:rgi]) {
while(![bufferChain open:url withOutputFormat:[output format] withUserInfo:userInfo withRGInfo:rgi]) {
bufferChain = nil;
[self requestNextStream:userInfo];
@ -401,7 +401,7 @@
}
if(pathsEqual || ([[nextStream scheme] isEqualToString:[[lastChain streamURL] scheme]] && (([nextStream host] == nil && [[lastChain streamURL] host] == nil) || [[nextStream host] isEqualToString:[[lastChain streamURL] host]]) && [[nextStream path] isEqualToString:[[lastChain streamURL] path]])) {
if([lastChain setTrack:nextStream] && [newChain openWithInput:[lastChain inputNode] withUserInfo:nextStreamUserInfo withRGInfo:nextStreamRGInfo]) {
if([lastChain setTrack:nextStream] && [newChain openWithInput:[lastChain inputNode] withOutputFormat:[output format] withUserInfo:nextStreamUserInfo withRGInfo:nextStreamRGInfo]) {
[newChain setStreamURL:nextStream];
@synchronized(chainQueue) {
@ -420,7 +420,7 @@
NSURL *url = nextStream;
while(shouldContinue && ![newChain open:url withUserInfo:nextStreamUserInfo withRGInfo:nextStreamRGInfo]) {
while(shouldContinue && ![newChain open:url withOutputFormat:[output format] withUserInfo:nextStreamUserInfo withRGInfo:nextStreamRGInfo]) {
if(nextStream == nil) {
newChain = nil;
if(output)

5
Audio/Chain/BufferChain.h
View file

@ -28,13 +28,14 @@
- (id)initWithController:(id)c;
- (void)buildChain;
- (BOOL)open:(NSURL *)url withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi;
- (BOOL)open:(NSURL *)url withOutputFormat:(AudioStreamBasicDescription)outputFormat withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi;
// Used when changing tracks to reuse the same decoder
- (BOOL)openWithInput:(InputNode *)i withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi;
- (BOOL)openWithInput:(InputNode *)i withOutputFormat:(AudioStreamBasicDescription)outputFormat withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi;
// Used when resetting the decoder on seek
- (BOOL)openWithDecoder:(id<CogDecoder>)decoder
withOutputFormat:(AudioStreamBasicDescription)outputFormat
withUserInfo:(id)userInfo
withRGInfo:(NSDictionary *)rgi;

23
Audio/Chain/BufferChain.m
View file

@ -40,7 +40,7 @@
finalNode = converterNode;
}
- (BOOL)open:(NSURL *)url withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi {
- (BOOL)open:(NSURL *)url withOutputFormat:(AudioStreamBasicDescription)outputFormat withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi {
[self setStreamURL:url];
[self setUserInfo:userInfo];
@ -66,7 +66,11 @@
if([properties valueForKey:@"channelConfig"])
inputChannelConfig = [[properties valueForKey:@"channelConfig"] unsignedIntValue];
if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig isLossless:[[properties valueForKey:@"encoding"] isEqualToString:@"lossless"]])
outputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
outputFormat.mBytesPerFrame = ((outputFormat.mBitsPerChannel + 7) / 8) * outputFormat.mChannelsPerFrame;
outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame * outputFormat.mFramesPerPacket;
if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig outputFormat:outputFormat isLossless:[[properties valueForKey:@"encoding"] isEqualToString:@"lossless"]])
return NO;
[self setRGInfo:rgi];
@ -76,7 +80,7 @@
return YES;
}
- (BOOL)openWithInput:(InputNode *)i withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi {
- (BOOL)openWithInput:(InputNode *)i withOutputFormat:(AudioStreamBasicDescription)outputFormat withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi {
DLog(@"New buffer chain!");
[self setUserInfo:userInfo];
[self buildChain];
@ -91,8 +95,12 @@
if([properties valueForKey:@"channelConfig"])
inputChannelConfig = [[properties valueForKey:@"channelConfig"] unsignedIntValue];
outputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
outputFormat.mBytesPerFrame = ((outputFormat.mBitsPerChannel + 7) / 8) * outputFormat.mChannelsPerFrame;
outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame * outputFormat.mFramesPerPacket;
DLog(@"Input Properties: %@", properties);
if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig outputFormat:outputFormat isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
return NO;
[self setRGInfo:rgi];
@ -101,6 +109,7 @@
}
- (BOOL)openWithDecoder:(id<CogDecoder>)decoder
withOutputFormat:(AudioStreamBasicDescription)outputFormat
withUserInfo:(id)userInfo
withRGInfo:(NSDictionary *)rgi;
{
@ -120,7 +129,11 @@
if([properties valueForKey:@"channelConfig"])
inputChannelConfig = [[properties valueForKey:@"channelConfig"] unsignedIntValue];
if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
outputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
outputFormat.mBytesPerFrame = ((outputFormat.mBitsPerChannel + 7) / 8) * outputFormat.mChannelsPerFrame;
outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame * outputFormat.mFramesPerPacket;
if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig outputFormat:outputFormat isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
return NO;
[self setRGInfo:rgi];

29
Audio/Chain/ConverterNode.h
View file

@ -12,11 +12,15 @@
#import <AudioUnit/AudioUnit.h>
#import <CoreAudio/AudioHardware.h>
#import <soxr.h>
#import "Node.h"
@interface ConverterNode : Node {
NSDictionary *rgInfo;
soxr_t soxr;
void *inputBuffer;
size_t inputBufferSize;
size_t inpSize, inpOffset;
@ -25,12 +29,33 @@
BOOL convertEntered;
BOOL paused;
BOOL skipResampler;
unsigned int PRIME_LEN_;
unsigned int N_samples_to_add_;
unsigned int N_samples_to_drop_;
BOOL is_preextrapolated_;
int is_postextrapolated_;
int latencyEaten;
int latencyEatenPost;
double sampleRatio;
float volumeScale;
void *floatBuffer;
size_t floatBufferSize;
void *extrapolateBuffer;
size_t extrapolateBufferSize;
BOOL rememberedLossless;
AudioStreamBasicDescription inputFormat;
AudioStreamBasicDescription floatFormat;
AudioStreamBasicDescription outputFormat;
uint32_t inputChannelConfig;
@ -43,7 +68,7 @@
- (id)initWithController:(id)c previous:(id)p;
- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inputFormat withInputConfig:(uint32_t)inputConfig isLossless:(BOOL)lossless;
- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inputFormat withInputConfig:(uint32_t)inputConfig outputFormat:(AudioStreamBasicDescription)outputFormat isLossless:(BOOL)lossless;
- (void)cleanUp;
- (void)process;
@ -51,6 +76,8 @@
- (void)setRGInfo:(NSDictionary *)rgi;
- (void)setOutputFormat:(AudioStreamBasicDescription)outputFormat;
- (void)inputFormatDidChange:(AudioStreamBasicDescription)format inputConfig:(uint32_t)inputConfig;
- (void)refreshVolumeScaling;

187
Audio/Chain/ConverterNode.m
View file

@ -16,6 +16,9 @@
#import "Logging.h"
#import "lpc.h"
#import "util.h"
#ifdef _DEBUG
#import "BadSampleCleaner.h"
#endif
@ -48,13 +51,21 @@ static void *kConverterNodeContext = &kConverterNodeContext;
if(self) {
rgInfo = nil;
soxr = 0;
inputBuffer = NULL;
inputBufferSize = 0;
floatBuffer = NULL;
floatBufferSize = 0;
stopping = NO;
convertEntered = NO;
paused = NO;
skipResampler = YES;
extrapolateBuffer = NULL;
extrapolateBufferSize = 0;
[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.volumeScaling" options:0 context:kConverterNodeContext];
}
@ -102,8 +113,10 @@ void scale_by_volume(float *buffer, size_t count, float volume) {
}
}
if(streamFormatChanged) {
[self cleanUp];
[self setupWithInputFormat:newInputFormat withInputConfig:newInputChannelConfig isLossless:rememberedLossless];
@autoreleasepool {
[self cleanUp];
[self setupWithInputFormat:newInputFormat withInputConfig:newInputChannelConfig outputFormat:self->outputFormat isLossless:rememberedLossless];
}
}
}
}
@ -173,6 +186,57 @@ void scale_by_volume(float *buffer, size_t count, float volume) {
return nil;
}
if(stopping || paused || streamFormatChanged || [self shouldContinue] == NO || [self endOfStream] == YES) {
if(!skipResampler) {
if(!is_postextrapolated_) {
is_postextrapolated_ = 1;
}
} else {
is_postextrapolated_ = 3;
}
}
// Extrapolate start
if(!skipResampler && !is_preextrapolated_) {
size_t inputSamples = bytesReadFromInput / floatFormat.mBytesPerPacket;
size_t prime = MIN(inputSamples, PRIME_LEN_);
size_t _N_samples_to_add_ = N_samples_to_add_;
size_t newSize = _N_samples_to_add_ * floatFormat.mBytesPerPacket;
newSize += bytesReadFromInput;
if(newSize > inputBufferSize) {
inputBuffer = realloc(inputBuffer, inputBufferSize = newSize * 3);
}
memmove(inputBuffer + _N_samples_to_add_ * floatFormat.mBytesPerPacket, inputBuffer, bytesReadFromInput);
lpc_extrapolate_bkwd(inputBuffer + _N_samples_to_add_ * floatFormat.mBytesPerPacket, inputSamples, prime, floatFormat.mChannelsPerFrame, LPC_ORDER, _N_samples_to_add_, &extrapolateBuffer, &extrapolateBufferSize);
bytesReadFromInput += _N_samples_to_add_ * floatFormat.mBytesPerPacket;
latencyEaten = N_samples_to_drop_;
is_preextrapolated_ = YES;
}
if(is_postextrapolated_ == 1) {
size_t inputSamples = bytesReadFromInput / floatFormat.mBytesPerPacket;
size_t prime = MIN(inputSamples, PRIME_LEN_);
size_t _N_samples_to_add_ = N_samples_to_add_;
size_t newSize = bytesReadFromInput;
newSize += _N_samples_to_add_ * floatFormat.mBytesPerPacket;
if(newSize > inputBufferSize) {
inputBuffer = realloc(inputBuffer, inputBufferSize = newSize * 3);
}
lpc_extrapolate_fwd(inputBuffer, inputSamples, prime, floatFormat.mChannelsPerFrame, LPC_ORDER, _N_samples_to_add_, &extrapolateBuffer, &extrapolateBufferSize);
bytesReadFromInput += _N_samples_to_add_ * floatFormat.mBytesPerPacket;
latencyEatenPost = N_samples_to_drop_;
is_postextrapolated_ = 2;
} else if(is_postextrapolated_ == 3) {
latencyEatenPost = 0;
}
// Input now contains bytesReadFromInput worth of floats, in the input sample rate
inpSize = bytesReadFromInput;
inpOffset = 0;
@ -182,16 +246,77 @@ void scale_by_volume(float *buffer, size_t count, float volume) {
ioNumberPackets -= ioNumberPackets % floatFormat.mBytesPerPacket;
if(ioNumberPackets) {
size_t inputSamples = ioNumberPackets / floatFormat.mBytesPerPacket;
ioNumberPackets = (UInt32)inputSamples;
ioNumberPackets = (UInt32)ceil((float)ioNumberPackets * sampleRatio);
ioNumberPackets = (ioNumberPackets + 255) & ~255;
size_t newSize = ioNumberPackets * floatFormat.mBytesPerPacket;
if(!floatBuffer || floatBufferSize < newSize) {
floatBuffer = realloc(floatBuffer, floatBufferSize = newSize * 3);
}
if(stopping) {
convertEntered = NO;
return nil;
}
size_t inputDone = 0;
size_t outputDone = 0;
if(!skipResampler) {
ioNumberPackets += soxr_delay(soxr);
soxr_process(soxr, (float *)(((uint8_t *)inputBuffer) + inpOffset), inputSamples, &inputDone, floatBuffer, ioNumberPackets, &outputDone);
if(latencyEatenPost) {
// Post file or format change flush
size_t idone = 0, odone = 0;
do {
soxr_process(soxr, NULL, 0, &idone, floatBuffer + outputDone * floatFormat.mBytesPerPacket, ioNumberPackets - outputDone, &odone);
outputDone += odone;
} while(odone > 0);
}
} else {
memcpy(floatBuffer, (((uint8_t *)inputBuffer) + inpOffset), inputSamples * floatFormat.mBytesPerPacket);
inputDone = inputSamples;
outputDone = inputSamples;
}
inpOffset += inputDone * floatFormat.mBytesPerPacket;
if(latencyEaten) {
if(outputDone > latencyEaten) {
outputDone -= latencyEaten;
memmove(floatBuffer, floatBuffer + latencyEaten * floatFormat.mBytesPerPacket, outputDone * floatFormat.mBytesPerPacket);
latencyEaten = 0;
} else {
latencyEaten -= outputDone;
outputDone = 0;
}
}
if(latencyEatenPost) {
if(outputDone > latencyEatenPost) {
outputDone -= latencyEatenPost;
} else {
outputDone = 0;
}
latencyEatenPost = 0;
}
ioNumberPackets = (UInt32)outputDone * floatFormat.mBytesPerPacket;
}
if(ioNumberPackets) {
AudioChunk *chunk = [[AudioChunk alloc] init];
[chunk setFormat:nodeFormat];
if(nodeChannelConfig) {
[chunk setChannelConfig:nodeChannelConfig];
}
scale_by_volume((float *)(((uint8_t *)inputBuffer) + inpOffset), ioNumberPackets / sizeof(float), volumeScale);
[chunk assignSamples:(((uint8_t *)inputBuffer) + inpOffset) frameCount:ioNumberPackets / floatFormat.mBytesPerPacket];
inpOffset += ioNumberPackets;
scale_by_volume(floatBuffer, ioNumberPackets / sizeof(float), volumeScale);
[chunk assignSamples:floatBuffer frameCount:ioNumberPackets / floatFormat.mBytesPerPacket];
convertEntered = NO;
return chunk;
}
@ -260,9 +385,10 @@ static float db_to_scale(float db) {
volumeScale = scale;
}
- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inf withInputConfig:(uint32_t)inputConfig isLossless:(BOOL)lossless {
- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inf withInputConfig:(uint32_t)inputConfig outputFormat:(AudioStreamBasicDescription)outf isLossless:(BOOL)lossless {
// Make the converter
inputFormat = inf;
outputFormat = outf;
inputChannelConfig = inputConfig;
@ -289,11 +415,40 @@ static float db_to_scale(float db) {
inpOffset = 0;
inpSize = 0;
// This is a post resampler, post-down/upmix format
// This is a post resampler format
nodeFormat = floatFormat;
nodeFormat.mSampleRate = outputFormat.mSampleRate;
nodeChannelConfig = inputChannelConfig;
sampleRatio = (double)outputFormat.mSampleRate / (double)floatFormat.mSampleRate;
skipResampler = fabs(sampleRatio - 1.0) < 1e-7;
if(!skipResampler) {
soxr_quality_spec_t q_spec = soxr_quality_spec(SOXR_HQ, 0);
soxr_io_spec_t io_spec = soxr_io_spec(SOXR_FLOAT32_I, SOXR_FLOAT32_I);
soxr_runtime_spec_t runtime_spec = soxr_runtime_spec(0);
soxr_error_t error;
soxr = soxr_create(floatFormat.mSampleRate, outputFormat.mSampleRate, floatFormat.mChannelsPerFrame, &error, &io_spec, &q_spec, &runtime_spec);
if(error)
return NO;
PRIME_LEN_ = MAX(floatFormat.mSampleRate / 20, 1024u);
PRIME_LEN_ = MIN(PRIME_LEN_, 16384u);
PRIME_LEN_ = MAX(PRIME_LEN_, (unsigned int)(2 * LPC_ORDER + 1));
N_samples_to_add_ = floatFormat.mSampleRate;
N_samples_to_drop_ = outputFormat.mSampleRate;
samples_len(&N_samples_to_add_, &N_samples_to_drop_, 20, 8192u);
is_preextrapolated_ = NO;
is_postextrapolated_ = 0;
}
latencyEaten = 0;
latencyEatenPost = 0;
PrintStreamDesc(&inf);
PrintStreamDesc(&nodeFormat);
@ -324,7 +479,7 @@ static float db_to_scale(float db) {
usleep(500);
}
[self cleanUp];
[self setupWithInputFormat:format withInputConfig:inputConfig isLossless:rememberedLossless];
[self setupWithInputFormat:format withInputConfig:inputConfig outputFormat:self->outputFormat isLossless:rememberedLossless];
}
- (void)setRGInfo:(NSDictionary *)rgi {
@ -338,6 +493,20 @@ static float db_to_scale(float db) {
while(convertEntered) {
usleep(500);
}
if(soxr) {
soxr_delete(soxr);
soxr = NULL;
}
if(extrapolateBuffer) {
free(extrapolateBuffer);
extrapolateBuffer = NULL;
extrapolateBufferSize = 0;
}
if(floatBuffer) {
free(floatBuffer);
floatBuffer = NULL;
floatBufferSize = 0;
}
if(inputBuffer) {
free(inputBuffer);
inputBuffer = NULL;

2
Audio/Chain/Node.m
View file

@ -11,7 +11,7 @@
#import "BufferChain.h"
#import "Logging.h"
#import "OutputAVFoundation.h"
#import "OutputCoreAudio.h"
#import <pthread.h>

4
Audio/Chain/OutputNode.h
View file

@ -13,7 +13,7 @@
#import <CoreAudio/AudioHardware.h>
#import "Node.h"
#import "OutputAVFoundation.h"
#import "OutputCoreAudio.h"
@interface OutputNode : Node {
AudioStreamBasicDescription format;
@ -21,7 +21,7 @@
double amountPlayed;
double amountPlayedInterval;
OutputAVFoundation *output;
OutputCoreAudio *output;
BOOL paused;
BOOL started;

5
Audio/Chain/OutputNode.m
View file

@ -9,7 +9,7 @@
#import "OutputNode.h"
#import "AudioPlayer.h"
#import "BufferChain.h"
#import "OutputAVFoundation.h"
#import "OutputCoreAudio.h"
#import "Logging.h"
@ -23,7 +23,7 @@
started = NO;
intervalReported = NO;
output = [[OutputAVFoundation alloc] initWithController:self];
output = [[OutputCoreAudio alloc] initWithController:self];
[output setup];
}
@ -149,6 +149,7 @@
format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket;
channelConfig = config;
[converter setOutputFormat:format];
[converter inputFormatDidChange:[bufferChain inputFormat] inputConfig:[bufferChain inputConfig]];
}
}

52
Audio/CogAudio.xcodeproj/project.pbxproj
View file

@ -25,8 +25,6 @@
17D21CA80B8BE4BA00D1EBDE /* Node.m in Sources */ = {isa = PBXBuildFile; fileRef = 17D21C7D0B8BE4BA00D1EBDE /* Node.m */; };
17D21CA90B8BE4BA00D1EBDE /* OutputNode.h in Headers */ = {isa = PBXBuildFile; fileRef = 17D21C7E0B8BE4BA00D1EBDE /* OutputNode.h */; settings = {ATTRIBUTES = (Public, ); }; };
17D21CAA0B8BE4BA00D1EBDE /* OutputNode.m in Sources */ = {isa = PBXBuildFile; fileRef = 17D21C7F0B8BE4BA00D1EBDE /* OutputNode.m */; };
17D21CC50B8BE4BA00D1EBDE /* OutputAVFoundation.h in Headers */ = {isa = PBXBuildFile; fileRef = 17D21C9C0B8BE4BA00D1EBDE /* OutputAVFoundation.h */; settings = {ATTRIBUTES = (Public, ); }; };
17D21CC60B8BE4BA00D1EBDE /* OutputAVFoundation.m in Sources */ = {isa = PBXBuildFile; fileRef = 17D21C9D0B8BE4BA00D1EBDE /* OutputAVFoundation.m */; };
17D21CC70B8BE4BA00D1EBDE /* Status.h in Headers */ = {isa = PBXBuildFile; fileRef = 17D21C9E0B8BE4BA00D1EBDE /* Status.h */; settings = {ATTRIBUTES = (Public, ); }; };
17D21CF30B8BE5EF00D1EBDE /* CogSemaphore.h in Headers */ = {isa = PBXBuildFile; fileRef = 17D21CF10B8BE5EF00D1EBDE /* CogSemaphore.h */; settings = {ATTRIBUTES = (Public, ); }; };
17D21CF40B8BE5EF00D1EBDE /* CogSemaphore.m in Sources */ = {isa = PBXBuildFile; fileRef = 17D21CF20B8BE5EF00D1EBDE /* CogSemaphore.m */; };
@ -66,6 +64,13 @@
8350416D28646149006B32CC /* CoreMedia.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 8350416C28646149006B32CC /* CoreMedia.framework */; };
835C88B1279811A500E28EAE /* hdcd_decode2.h in Headers */ = {isa = PBXBuildFile; fileRef = 835C88AF279811A500E28EAE /* hdcd_decode2.h */; };
835C88B2279811A500E28EAE /* hdcd_decode2.c in Sources */ = {isa = PBXBuildFile; fileRef = 835C88B0279811A500E28EAE /* hdcd_decode2.c */; };
835DD2672ACAF1D90057E319 /* OutputCoreAudio.m in Sources */ = {isa = PBXBuildFile; fileRef = 835DD2652ACAF1D90057E319 /* OutputCoreAudio.m */; };
835DD2682ACAF1D90057E319 /* OutputCoreAudio.h in Headers */ = {isa = PBXBuildFile; fileRef = 835DD2662ACAF1D90057E319 /* OutputCoreAudio.h */; };
835DD2702ACAF5AD0057E319 /* LICENSE.LGPL in Resources */ = {isa = PBXBuildFile; fileRef = 835DD26B2ACAF5AD0057E319 /* LICENSE.LGPL */; };
835DD2712ACAF5AD0057E319 /* License.txt in Resources */ = {isa = PBXBuildFile; fileRef = 835DD26C2ACAF5AD0057E319 /* License.txt */; };
835DD2722ACAF5AD0057E319 /* lpc.h in Headers */ = {isa = PBXBuildFile; fileRef = 835DD26D2ACAF5AD0057E319 /* lpc.h */; };
835DD2732ACAF5AD0057E319 /* util.h in Headers */ = {isa = PBXBuildFile; fileRef = 835DD26E2ACAF5AD0057E319 /* util.h */; };
835DD2742ACAF5AD0057E319 /* lpc.c in Sources */ = {isa = PBXBuildFile; fileRef = 835DD26F2ACAF5AD0057E319 /* lpc.c */; };
835FAC5E27BCA14D00BA8562 /* BadSampleCleaner.h in Headers */ = {isa = PBXBuildFile; fileRef = 835FAC5C27BCA14D00BA8562 /* BadSampleCleaner.h */; };
835FAC5F27BCA14D00BA8562 /* BadSampleCleaner.m in Sources */ = {isa = PBXBuildFile; fileRef = 835FAC5D27BCA14D00BA8562 /* BadSampleCleaner.m */; };
836DF618298F6F5F00CD0580 /* libsoxr.0.dylib in Frameworks */ = {isa = PBXBuildFile; fileRef = 836DF615298F6E8900CD0580 /* libsoxr.0.dylib */; };
@ -132,8 +137,6 @@
17D21C7D0B8BE4BA00D1EBDE /* Node.m */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.objc; path = Node.m; sourceTree = "<group>"; };
17D21C7E0B8BE4BA00D1EBDE /* OutputNode.h */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.h; path = OutputNode.h; sourceTree = "<group>"; };
17D21C7F0B8BE4BA00D1EBDE /* OutputNode.m */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.objc; path = OutputNode.m; sourceTree = "<group>"; };
17D21C9C0B8BE4BA00D1EBDE /* OutputAVFoundation.h */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.h; path = OutputAVFoundation.h; sourceTree = "<group>"; };
17D21C9D0B8BE4BA00D1EBDE /* OutputAVFoundation.m */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.objc; path = OutputAVFoundation.m; sourceTree = "<group>"; };
17D21C9E0B8BE4BA00D1EBDE /* Status.h */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.h; path = Status.h; sourceTree = "<group>"; };
17D21CF10B8BE5EF00D1EBDE /* CogSemaphore.h */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.h; path = CogSemaphore.h; sourceTree = "<group>"; };
17D21CF20B8BE5EF00D1EBDE /* CogSemaphore.m */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.objc; path = CogSemaphore.m; sourceTree = "<group>"; };
@ -174,6 +177,13 @@
8350416C28646149006B32CC /* CoreMedia.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = CoreMedia.framework; path = System/Library/Frameworks/CoreMedia.framework; sourceTree = SDKROOT; };
835C88AF279811A500E28EAE /* hdcd_decode2.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = hdcd_decode2.h; sourceTree = "<group>"; };
835C88B0279811A500E28EAE /* hdcd_decode2.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = hdcd_decode2.c; sourceTree = "<group>"; };
835DD2652ACAF1D90057E319 /* OutputCoreAudio.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; path = OutputCoreAudio.m; sourceTree = "<group>"; };
835DD2662ACAF1D90057E319 /* OutputCoreAudio.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = OutputCoreAudio.h; sourceTree = "<group>"; };
835DD26B2ACAF5AD0057E319 /* LICENSE.LGPL */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = LICENSE.LGPL; sourceTree = "<group>"; };
835DD26C2ACAF5AD0057E319 /* License.txt */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = License.txt; sourceTree = "<group>"; };
835DD26D2ACAF5AD0057E319 /* lpc.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = lpc.h; sourceTree = "<group>"; };
835DD26E2ACAF5AD0057E319 /* util.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = util.h; sourceTree = "<group>"; };
835DD26F2ACAF5AD0057E319 /* lpc.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = lpc.c; sourceTree = "<group>"; };
835FAC5C27BCA14D00BA8562 /* BadSampleCleaner.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = BadSampleCleaner.h; path = Utils/BadSampleCleaner.h; sourceTree = SOURCE_ROOT; };
835FAC5D27BCA14D00BA8562 /* BadSampleCleaner.m */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.objc; name = BadSampleCleaner.m; path = Utils/BadSampleCleaner.m; sourceTree = SOURCE_ROOT; };
836DF615298F6E8900CD0580 /* libsoxr.0.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libsoxr.0.dylib; path = ../ThirdParty/soxr/lib/libsoxr.0.dylib; sourceTree = "<group>"; };
@ -354,12 +364,12 @@
17D21C9B0B8BE4BA00D1EBDE /* Output */ = {
isa = PBXGroup;
children = (
835DD2662ACAF1D90057E319 /* OutputCoreAudio.h */,
835DD2652ACAF1D90057E319 /* OutputCoreAudio.m */,
834A41AD287ABD6F00EB9D9B /* FSurroundFilter.h */,
834A41AE287ABD6F00EB9D9B /* FSurroundFilter.mm */,
839E56EB2879515D00DFB5F4 /* HeadphoneFilter.h */,
839E56EC2879515D00DFB5F4 /* HeadphoneFilter.mm */,
17D21C9C0B8BE4BA00D1EBDE /* OutputAVFoundation.h */,
17D21C9D0B8BE4BA00D1EBDE /* OutputAVFoundation.m */,
);
path = Output;
sourceTree = "<group>";
@ -367,6 +377,7 @@
17D21CD80B8BE5B400D1EBDE /* ThirdParty */ = {
isa = PBXGroup;
children = (
835DD2692ACAF5AD0057E319 /* lvqcl */,
834A41A4287A90AB00EB9D9B /* fsurround */,
839E56E02879450300DFB5F4 /* hrtf */,
831A50152865A8800049CFE4 /* rsstate.cpp */,
@ -437,6 +448,26 @@
path = hdcd;
sourceTree = "<group>";
};
835DD2692ACAF5AD0057E319 /* lvqcl */ = {
isa = PBXGroup;
children = (
835DD26A2ACAF5AD0057E319 /* License */,
835DD26D2ACAF5AD0057E319 /* lpc.h */,
835DD26E2ACAF5AD0057E319 /* util.h */,
835DD26F2ACAF5AD0057E319 /* lpc.c */,
);
path = lvqcl;
sourceTree = "<group>";
};
835DD26A2ACAF5AD0057E319 /* License */ = {
isa = PBXGroup;
children = (
835DD26B2ACAF5AD0057E319 /* LICENSE.LGPL */,
835DD26C2ACAF5AD0057E319 /* License.txt */,
);
path = License;
sourceTree = "<group>";
};
83725A8F27AA16C90003F694 /* Frameworks */ = {
isa = PBXGroup;
children = (
@ -488,18 +519,20 @@
834A41AC287A90AB00EB9D9B /* channelmaps.h in Headers */,
17D21CA50B8BE4BA00D1EBDE /* InputNode.h in Headers */,
834A41A9287A90AB00EB9D9B /* freesurround_decoder.h in Headers */,
835DD2732ACAF5AD0057E319 /* util.h in Headers */,
17D21CA70B8BE4BA00D1EBDE /* Node.h in Headers */,
8399CF2C27B5D1D5008751F1 /* NSDictionary+Merge.h in Headers */,
17D21CA90B8BE4BA00D1EBDE /* OutputNode.h in Headers */,
8328995427CB511000D7F028 /* RedundantPlaylistDataStore.h in Headers */,
17D21CC50B8BE4BA00D1EBDE /* OutputAVFoundation.h in Headers */,
83504165286447DA006B32CC /* Downmix.h in Headers */,
839E56E52879450300DFB5F4 /* HrtfData.h in Headers */,
83B74281289E027F005AAC28 /* CogAudio-Bridging-Header.h in Headers */,
17D21CC70B8BE4BA00D1EBDE /* Status.h in Headers */,
17D21CF30B8BE5EF00D1EBDE /* CogSemaphore.h in Headers */,
835DD2682ACAF1D90057E319 /* OutputCoreAudio.h in Headers */,
839E56E62879450300DFB5F4 /* Endianness.h in Headers */,
17D21DC70B8BE79700D1EBDE /* CoreAudioUtils.h in Headers */,
835DD2722ACAF5AD0057E319 /* lpc.h in Headers */,
839E56ED2879515D00DFB5F4 /* HeadphoneFilter.h in Headers */,
17D21EBD0B8BF44000D1EBDE /* AudioPlayer.h in Headers */,
831A50182865A8B30049CFE4 /* rsstate.h in Headers */,
@ -589,6 +622,8 @@
isa = PBXResourcesBuildPhase;
buildActionMask = 2147483647;
files = (
835DD2712ACAF5AD0057E319 /* License.txt in Resources */,
835DD2702ACAF5AD0057E319 /* LICENSE.LGPL in Resources */,
);
runOnlyForDeploymentPostprocessing = 0;
};
@ -608,7 +643,6 @@
831A50162865A8800049CFE4 /* rsstate.cpp in Sources */,
17D21CA80B8BE4BA00D1EBDE /* Node.m in Sources */,
17D21CAA0B8BE4BA00D1EBDE /* OutputNode.m in Sources */,
17D21CC60B8BE4BA00D1EBDE /* OutputAVFoundation.m in Sources */,
835C88B2279811A500E28EAE /* hdcd_decode2.c in Sources */,
835FAC5F27BCA14D00BA8562 /* BadSampleCleaner.m in Sources */,
834FD4ED27AF91220063BC83 /* AudioChunk.m in Sources */,
@ -630,8 +664,10 @@
834FD4F127AF93680063BC83 /* ChunkList.m in Sources */,
834A41B0287ABD6F00EB9D9B /* FSurroundFilter.mm in Sources */,
8EC122600B993BD500C5B3AD /* ConverterNode.m in Sources */,
835DD2672ACAF1D90057E319 /* OutputCoreAudio.m in Sources */,
8E8D3D300CBAEE6E00135C1B /* AudioContainer.m in Sources */,
B0575F300D687A4000411D77 /* Helper.m in Sources */,
835DD2742ACAF5AD0057E319 /* lpc.c in Sources */,
834A41AA287A90AB00EB9D9B /* freesurround_decoder.cpp in Sources */,
07DB5F3F0ED353A900C2E3EF /* AudioMetadataWriter.m in Sources */,
);

4
Audio/Output/OutputCoreAudio.h
View file

@ -47,8 +47,6 @@ using std::atomic_long;
NSLock *outputLock;
BOOL rsDone;
void *rsstate, *rsold;
double secondsLatency;
double visPushed;
@ -126,7 +124,7 @@ using std::atomic_long;
AudioChunk *chunkRemain;
int resamplerRemain, visResamplerRemain;
int visResamplerRemain;
BOOL resetStreamFormat;

262
Audio/Output/OutputCoreAudio.m
View file

@ -72,27 +72,12 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
AudioStreamBasicDescription format;
uint32_t config;
if([outputController peekFormat:&format channelConfig:&config]) {
// XXX ERROR with AirPods - Can't go higher than CD*8 surround - 192k stereo
// Emits to console: [AUScotty] Initialize: invalid FFT size 16384
// DSD256 stereo emits: [AUScotty] Initialize: invalid FFT size 65536
AudioStreamBasicDescription origFormat;
uint32_t origConfig = config;
memcpy(&origFormat, &format, sizeof(origFormat));
BOOL formatClipped = NO;
const double targetSampleRate = deviceFormat.mSampleRate;
double srcRate = format.mSampleRate;
double dstRate = targetSampleRate;
UInt32 srcChannels = format.mChannelsPerFrame;
UInt32 dstChannels = deviceFormat.mChannelsPerFrame;
if(fabs(format.mSampleRate - targetSampleRate) > 1e-5) {
format.mSampleRate = targetSampleRate;
formatClipped = YES;
if(fabs(srcRate - lastClippedSampleRate) > 1e-5) {
lastClippedSampleRate = srcRate;
streamFormatStarted = NO;
}
}
if(srcChannels != dstChannels) {
format.mChannelsPerFrame = dstChannels;
format.mBytesPerFrame = ((format.mBitsPerChannel + 7) / 8) * dstChannels;
@ -103,19 +88,6 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
downmixer = nil;
}
if(!streamFormatStarted || config != realStreamChannelConfig || memcmp(&newFormat, &format, sizeof(format)) != 0) {
[outputLock lock];
if(formatClipped) {
ALog(@"Format clipped to %f Hz", targetSampleRate);
if(rsstate) {
rsold = rsstate;
rsstate = NULL;
}
rsstate = rsstate_new(srcChannels, srcRate, dstRate);
} else if(rsstate) {
rsold = rsstate;
rsstate = NULL;
}
[outputLock unlock];
newFormat = format;
newChannelConfig = config;
streamFormatStarted = YES;
@ -126,13 +98,6 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
visFormat.mBytesPerPacket = visFormat.mBytesPerFrame * visFormat.mFramesPerPacket;
downmixerForVis = [[DownmixProcessor alloc] initWithInputFormat:origFormat inputConfig:origConfig andOutputFormat:visFormat outputConfig:AudioConfigMono];
[outputLock lock];
if(!rsold) {
realStreamFormat = format;
realStreamChannelConfig = config;
streamFormatChanged = YES;
}
[outputLock unlock];
}
}
@ -164,35 +129,6 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
location:@"pre downmix"];
#endif
const float *outputPtr = (const float *)[samples bytes];
[outputLock lock];
if(rsstate) {
if(frameCount + resamplerRemain > 8192) {
int leftoverFrameCount = frameCount + resamplerRemain - 8192;
int frameOffset = frameCount - leftoverFrameCount;
chunkRemain = chunk;
[chunk assignSamples:&outputPtr[frameOffset * format.mChannelsPerFrame] frameCount:leftoverFrameCount];
frameCount = frameOffset;
}
cblas_scopy(frameCount * format.mChannelsPerFrame, outputPtr, 1, &rsInBuffer[resamplerRemain * format.mChannelsPerFrame], 1);
outputPtr = &rsInBuffer[0];
frameCount += resamplerRemain;
resamplerRemain = 0;
size_t inDone = 0;
size_t framesDone = rsstate_resample(rsstate, outputPtr, frameCount, &inDone, &rsTempBuffer[0], amountToRead);
resamplerRemain = (int)(frameCount - inDone);
if(resamplerRemain && inDone) {
memmove(&rsInBuffer[0], &rsInBuffer[inDone * format.mChannelsPerFrame], resamplerRemain * format.mBytesPerPacket);
}
if(!framesDone) {
[outputLock unlock];
return 0;
}
frameCount = (int)framesDone;
outputPtr = &rsTempBuffer[0];
chunkDuration = frameCount / newFormat.mSampleRate;
}
[outputLock unlock];
[downmixerForVis process:outputPtr
frameCount:frameCount
output:&visAudio[0]];
@ -403,14 +339,6 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
visPushed = 0.0;
started = NO;
restarted = NO;
if(rsstate) {
rsstate_delete(rsstate);
rsstate = NULL;
}
if(rsold) {
rsstate_delete(rsold);
rsold = NULL;
}
if(rsvis) {
rsstate_delete(rsvis);
rsvis = NULL;
@ -665,6 +593,9 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
_deviceFormat = format;
deviceFormat = *(format.streamDescription);
// Force format for HRTF
deviceFormat.mSampleRate = 96000.0;
/// Seems some 3rd party devices return incorrect stuff...or I just don't like noninterleaved data.
deviceFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved;
// deviceFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsFloat;
@ -892,127 +823,89 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
int samplesRenderedTotal = 0;
for(size_t i = 0; i < 2;) {
int samplesRendered;
int samplesRendered = inputRendered;
if(i == 0) {
[outputLock lock];
if(!rsold) {
[outputLock unlock];
++i;
continue;
samplePtr = &inputBuffer[0];
if(samplesRendered || fsurround) {
[outputLock lock];
if(fsurround) {
int countToProcess = samplesRendered;
if(countToProcess < 4096) {
bzero(samplePtr + countToProcess * 2, (4096 - countToProcess) * 2 * sizeof(float));
countToProcess = 4096;
}
samplesRendered = rsstate_flush(rsold, &rsTempBuffer[0], 4096);
if(samplesRendered < 4096) {
rsstate_delete(rsold);
rsold = NULL;
[outputLock unlock];
rsDone = YES;
++i;
continue;
[fsurround process:samplePtr output:&fsurroundBuffer[0] count:countToProcess];
samplePtr = &fsurroundBuffer[0];
if(resetStreamFormat || samplesRendered < 4096) {
bzero(&fsurroundBuffer[4096 * 6], 4096 * 2 * sizeof(float));
[fsurround process:&fsurroundBuffer[4096 * 6] output:&fsurroundBuffer[4096 * 6] count:4096];
samplesRendered += 2048;
}
[outputLock unlock];
samplePtr = &rsTempBuffer[0];
} else {
samplesRendered = inputRendered;
samplePtr = &inputBuffer[0];
if(rsDone) {
rsDone = NO;
realStreamFormat = newFormat;
realStreamChannelConfig = newChannelConfig;
[self updateStreamFormat];
if(!FSurroundDelayRemoved) {
FSurroundDelayRemoved = YES;
if(samplesRendered > 2048) {
samplePtr += 2048 * 6;
samplesRendered -= 2048;
}
}
}
[outputLock unlock];
if(samplesRendered || fsurround) {
[outputLock lock];
if(fsurround) {
int countToProcess = samplesRendered;
if(countToProcess < 4096) {
bzero(samplePtr + countToProcess * 2, (4096 - countToProcess) * 2 * sizeof(float));
countToProcess = 4096;
if(!samplesRendered) {
return 0;
}
[outputLock lock];
if(hrtf) {
[hrtf process:samplePtr sampleCount:samplesRendered toBuffer:&hrtfBuffer[0]];
samplePtr = &hrtfBuffer[0];
}
[outputLock unlock];
if(eqEnabled && eqInitialized) {
const int channels = streamFormat.mChannelsPerFrame;
if(channels > 0) {
const size_t channelsminusone = channels - 1;
uint8_t tempBuffer[sizeof(AudioBufferList) + sizeof(AudioBuffer) * channelsminusone];
AudioBufferList *ioData = (AudioBufferList *)&tempBuffer[0];
ioData->mNumberBuffers = channels;
for(size_t i = 0; i < channels; ++i) {
ioData->mBuffers[i].mData = &eqBuffer[4096 * i];
ioData->mBuffers[i].mDataByteSize = samplesRendered * sizeof(float);
ioData->mBuffers[i].mNumberChannels = 1;
}
[fsurround process:samplePtr output:&fsurroundBuffer[0] count:countToProcess];
samplePtr = &fsurroundBuffer[0];
if(resetStreamFormat || samplesRendered < 4096) {
bzero(&fsurroundBuffer[4096 * 6], 4096 * 2 * sizeof(float));
[fsurround process:&fsurroundBuffer[4096 * 6] output:&fsurroundBuffer[4096 * 6] count:4096];
samplesRendered += 2048;
status = AudioUnitRender(_eq, NULL, &timeStamp, 0, samplesRendered, ioData);
if(status != noErr) {
return 0;
}
if(!FSurroundDelayRemoved) {
FSurroundDelayRemoved = YES;
if(samplesRendered > 2048) {
samplePtr += 2048 * 6;
samplesRendered -= 2048;
}
}
}
[outputLock unlock];
if(!samplesRendered) {
break;
}
[outputLock lock];
if(hrtf) {
[hrtf process:samplePtr sampleCount:samplesRendered toBuffer:&hrtfBuffer[0]];
samplePtr = &hrtfBuffer[0];
}
[outputLock unlock];
if(eqEnabled && eqInitialized) {
const int channels = streamFormat.mChannelsPerFrame;
if(channels > 0) {
const size_t channelsminusone = channels - 1;
uint8_t tempBuffer[sizeof(AudioBufferList) + sizeof(AudioBuffer) * channelsminusone];
AudioBufferList *ioData = (AudioBufferList *)&tempBuffer[0];
ioData->mNumberBuffers = channels;
for(size_t i = 0; i < channels; ++i) {
ioData->mBuffers[i].mData = &eqBuffer[4096 * i];
ioData->mBuffers[i].mDataByteSize = samplesRendered * sizeof(float);
ioData->mBuffers[i].mNumberChannels = 1;
}
status = AudioUnitRender(_eq, NULL, &timeStamp, 0, samplesRendered, ioData);
if(status != noErr) {
return 0;
}
timeStamp.mSampleTime += ((double)samplesRendered) / streamFormat.mSampleRate;
for(int i = 0; i < channels; ++i) {
cblas_scopy(samplesRendered, &eqBuffer[4096 * i], 1, samplePtr + i, channels);
}
timeStamp.mSampleTime += ((double)samplesRendered) / streamFormat.mSampleRate;
for(int i = 0; i < channels; ++i) {
cblas_scopy(samplesRendered, &eqBuffer[4096 * i], 1, samplePtr + i, channels);
}
}
}
if(downmixer) {
[downmixer process:samplePtr frameCount:samplesRendered output:&downmixBuffer[0]];
samplePtr = &downmixBuffer[0];
}
if(downmixer) {
[downmixer process:samplePtr frameCount:samplesRendered output:&downmixBuffer[0]];
samplePtr = &downmixBuffer[0];
}
#ifdef OUTPUT_LOG
size_t dataByteSize = samplesRendered * sizeof(float) * deviceFormat.mChannelsPerFrame;
size_t dataByteSize = samplesRendered * sizeof(float) * deviceFormat.mChannelsPerFrame;
fwrite(samplePtr, 1, dataByteSize, _logFile);
fwrite(samplePtr, 1, dataByteSize, _logFile);
#endif
}
if(i == 0) {
samplesRenderedTotal += samplesRendered;
if(samplesRendered) {
break;
}
++i;
} else {
inputBufferLastTime = 0;
samplesRenderedTotal += samplesRendered;
++i;
}
}
inputBufferLastTime = 0;
samplesRenderedTotal += samplesRendered;
return samplesRenderedTotal;
}
@ -1093,10 +986,6 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
downmixer = nil;
downmixerForVis = nil;
rsDone = NO;
rsstate = NULL;
rsold = NULL;
lastClippedSampleRate = 0.0;
rsvis = NULL;
@ -1106,7 +995,6 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
chunkRemain = nil;
resamplerRemain = 0;
visResamplerRemain = 0;
secondsLatency = 0;
@ -1187,7 +1075,9 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
[self audioOutputBlock];
[_au allocateRenderResourcesAndReturnError:&err];
[self updateDeviceFormat];
visController = [VisualizationController sharedController];
[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.outputDevice" options:0 context:kOutputCoreAudioContext];
@ -1326,14 +1216,6 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
#endif
outputController = nil;
visController = nil;
if(rsstate) {
rsstate_delete(rsstate);
rsstate = NULL;
}
if(rsold) {
rsstate_delete(rsold);
rsold = NULL;
}
if(rsvis) {
rsstate_delete(rsvis);
rsvis = NULL;

504
Audio/ThirdParty/lvqcl/License/LICENSE.LGPL vendored Normal file
View file

@ -0,0 +1,504 @@
GNU LESSER GENERAL PUBLIC LICENSE
Version 2.1, February 1999
Copyright (C) 1991, 1999 Free Software Foundation, Inc.
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
[This is the first released version of the Lesser GPL. It also counts
as the successor of the GNU Library Public License, version 2, hence
the version number 2.1.]
Preamble
The licenses for most software are designed to take away your
freedom to share and change it. By contrast, the GNU General Public
Licenses are intended to guarantee your freedom to share and change
free software--to make sure the software is free for all its users.
This license, the Lesser General Public License, applies to some
specially designated software packages--typically libraries--of the
Free Software Foundation and other authors who decide to use it. You
can use it too, but we suggest you first think carefully about whether
this license or the ordinary General Public License is the better
strategy to use in any particular case, based on the explanations below.
When we speak of free software, we are referring to freedom of use,
not price. Our General Public Licenses are designed to make sure that
you have the freedom to distribute copies of free software (and charge
for this service if you wish); that you receive source code or can get
it if you want it; that you can change the software and use pieces of
it in new free programs; and that you are informed that you can do
these things.
To protect your rights, we need to make restrictions that forbid
distributors to deny you these rights or to ask you to surrender these
rights. These restrictions translate to certain responsibilities for
you if you distribute copies of the library or if you modify it.
For example, if you distribute copies of the library, whether gratis
or for a fee, you must give the recipients all the rights that we gave
you. You must make sure that they, too, receive or can get the source
code. If you link other code with the library, you must provide
complete object files to the recipients, so that they can relink them
with the library after making changes to the library and recompiling
it. And you must show them these terms so they know their rights.
We protect your rights with a two-step method: (1) we copyright the
library, and (2) we offer you this license, which gives you legal
permission to copy, distribute and/or modify the library.
To protect each distributor, we want to make it very clear that
there is no warranty for the free library. Also, if the library is
modified by someone else and passed on, the recipients should know
that what they have is not the original version, so that the original
author's reputation will not be affected by problems that might be
introduced by others.
Finally, software patents pose a constant threat to the existence of
any free program. We wish to make sure that a company cannot
effectively restrict the users of a free program by obtaining a
restrictive license from a patent holder. Therefore, we insist that
any patent license obtained for a version of the library must be
consistent with the full freedom of use specified in this license.
Most GNU software, including some libraries, is covered by the
ordinary GNU General Public License. This license, the GNU Lesser
General Public License, applies to certain designated libraries, and
is quite different from the ordinary General Public License. We use
this license for certain libraries in order to permit linking those
libraries into non-free programs.
When a program is linked with a library, whether statically or using
a shared library, the combination of the two is legally speaking a
combined work, a derivative of the original library. The ordinary
General Public License therefore permits such linking only if the
entire combination fits its criteria of freedom. The Lesser General
Public License permits more lax criteria for linking other code with
the library.
We call this license the "Lesser" General Public License because it
does Less to protect the user's freedom than the ordinary General
Public License. It also provides other free software developers Less
of an advantage over competing non-free programs. These disadvantages
are the reason we use the ordinary General Public License for many
libraries. However, the Lesser license provides advantages in certain
special circumstances.
For example, on rare occasions, there may be a special need to
encourage the widest possible use of a certain library, so that it becomes
a de-facto standard. To achieve this, non-free programs must be
allowed to use the library. A more frequent case is that a free
library does the same job as widely used non-free libraries. In this
case, there is little to gain by limiting the free library to free
software only, so we use the Lesser General Public License.
In other cases, permission to use a particular library in non-free
programs enables a greater number of people to use a large body of
free software. For example, permission to use the GNU C Library in
non-free programs enables many more people to use the whole GNU
operating system, as well as its variant, the GNU/Linux operating
system.
Although the Lesser General Public License is Less protective of the
users' freedom, it does ensure that the user of a program that is
linked with the Library has the freedom and the wherewithal to run
that program using a modified version of the Library.
The precise terms and conditions for copying, distribution and
modification follow. Pay close attention to the difference between a
"work based on the library" and a "work that uses the library". The
former contains code derived from the library, whereas the latter must
be combined with the library in order to run.
GNU LESSER GENERAL PUBLIC LICENSE
TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
0. This License Agreement applies to any software library or other
program which contains a notice placed by the copyright holder or
other authorized party saying it may be distributed under the terms of
this Lesser General Public License (also called "this License").
Each licensee is addressed as "you".
A "library" means a collection of software functions and/or data
prepared so as to be conveniently linked with application programs
(which use some of those functions and data) to form executables.
The "Library", below, refers to any such software library or work
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That's all there is to it!

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SoX resampler plugin for foobar2000 audio player
Copyright (C) lvqcl
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
This software uses code of SoX licensed under the terms of LGPLv2.1
Copyright (C) robs@users.sourceforge.net
Copyright (C) Chris Bagwell and SoX contributors
Copyright (C) Reuben Thomas
This software uses code of FFmpeg licensed under the terms of LGPLv2.1
Copyright (C) Fabrice Bellard
Copyright (C) Michael Niedermayer <michaelni@gmx.at>
Copyright (C) Alex Converse <alex dot converse at gmail dot com>
Copyright (C) Loren Merritt
Copyright (C) Vitor Sessak
Copyright (C) x264 project
This software uses code of General Purpose FFT Package
Copyright (C) Takuya OOURA
This software uses code of foobar2000 1.4 SDK
Copyright (C) 2001-2018, Peter Pawlowski

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/*
* Copyright (c) 2013, 2018 lvqcl
*
* Permission to use, copy, modify, and distribute this software for any
* purpose with or without fee is hereby granted, provided that the above
* copyright notice and this permission notice appear in all copies.
*
* THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
* WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
* ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
* ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
* OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#include <memory.h>
#include <stdlib.h>
#include <stdbool.h>
#include "lpc.h"
static void apply_window(float *const data, const size_t data_len) {
#if 0
if (0) // subtract the mean
{
double mean = 0;
for(int i = 0; i < (int)data_len; i++)
mean += data[i];
mean /= data_len;
for(int i = 0; i < (int)data_len; i++)
data[i] -= (float)mean;
}
#endif
if(1) // Welch window
{
const float n2 = (data_len + 1) / 2.0f;
for(int i = 0; i < (int)data_len; i++) {
float k = (i + 1 - n2) / n2;
data[data_len - 1 - i] *= 1.0f - k * k;
}
}
}
static float vorbis_lpc_from_data(float *data, float *lpci, int n, int m, double *aut, double *lpc) {
double error;
double epsilon;
int i, j;
/* autocorrelation, p+1 lag coefficients */
j = m + 1;
while(j--) {
double d = 0; /* double needed for accumulator depth */
for(i = j; i < n; i++) d += (double)data[i] * data[i - j];
aut[j] = d;
}
/* Generate lpc coefficients from autocorr values */
/* set our noise floor to about -100dB */
error = aut[0] * (1. + 1e-10);
epsilon = 1e-9 * aut[0] + 1e-10;
for(i = 0; i < m; i++) {
double r = -aut[i + 1];
if(error < epsilon) {
memset(lpc + i, 0, (m - i) * sizeof(*lpc));
goto done;
}
/* Sum up this iteration's reflection coefficient; note that in
Vorbis we don't save it. If anyone wants to recycle this code
and needs reflection coefficients, save the results of 'r' from
each iteration. */
for(j = 0; j < i; j++) r -= lpc[j] * aut[i - j];
r /= error;
/* Update LPC coefficients and total error */
lpc[i] = r;
for(j = 0; j < i / 2; j++) {
double tmp = lpc[j];
lpc[j] += r * lpc[i - 1 - j];
lpc[i - 1 - j] += r * tmp;
}
if(i & 1) lpc[j] += lpc[j] * r;
error *= 1. - r * r;
}
done:
/* slightly damp the filter */
{
double g = .99;
double damp = g;
for(j = 0; j < m; j++) {
lpc[j] *= damp;
damp *= g;
}
}
for(j = 0; j < m; j++) lpci[j] = (float)lpc[j];
/* we need the error value to know how big an impulse to hit the
filter with later */
return error;
}
static void vorbis_lpc_predict(float *coeff, float *prime, int m, float *data, long n, float *work) {
/* in: coeff[0...m-1] LPC coefficients
prime[0...m-1] initial values (allocated size of n+m-1)
out: data[0...n-1] data samples */
long i, j, o, p;
float y;
if(!prime)
for(i = 0; i < m; i++)
work[i] = 0.f;
else
for(i = 0; i < m; i++)
work[i] = prime[i];
for(i = 0; i < n; i++) {
y = 0;
o = i;
p = m;
for(j = 0; j < m; j++)
y -= work[o++] * coeff[--p];
data[i] = work[o] = y;
}
}
void lpc_extrapolate2(float *const data, const size_t data_len, const int nch, const int lpc_order, const size_t extra_bkwd, const size_t extra_fwd, void **extrapolate_buffer, size_t *extrapolate_buffer_size) {
const size_t tdata_size = sizeof(float) * (extra_bkwd + data_len + extra_fwd);
const size_t aut_size = sizeof(double) * (lpc_order + 1);
const size_t lpc_size = sizeof(double) * lpc_order;
const size_t lpci_size = sizeof(float) * lpc_order;
const size_t work_size = sizeof(float) * (extra_bkwd + lpc_order + extra_fwd);
const size_t new_size = tdata_size + aut_size + lpc_size + lpci_size + work_size;
if(new_size > *extrapolate_buffer_size) {
*extrapolate_buffer = realloc(*extrapolate_buffer, new_size);
*extrapolate_buffer_size = new_size;
}
float *tdata = (float *)(*extrapolate_buffer); // for 1 channel only
double *aut = (double *)(*extrapolate_buffer + tdata_size);
double *lpc = (double *)(*extrapolate_buffer + tdata_size + aut_size);
float *lpci = (float *)(*extrapolate_buffer + tdata_size + aut_size + lpc_size);
float *work = (float *)(*extrapolate_buffer + tdata_size + aut_size + lpc_size + lpci_size);
for(int c = 0; c < nch; c++) {
if(extra_bkwd) {
for(int i = 0; i < (int)data_len; i++)
tdata[data_len - 1 - i] = data[i * nch + c];
} else {
for(int i = 0; i < (int)data_len; i++)
tdata[i] = data[i * nch + c];
}
apply_window(tdata, data_len);
vorbis_lpc_from_data(tdata, lpci, (int)data_len, lpc_order, aut, lpc);
// restore after apply_window
if(extra_bkwd) {
for(int i = 0; i < (int)data_len; i++)
tdata[data_len - 1 - i] = data[i * nch + c];
} else {
for(int i = 0; i < (int)data_len; i++)
tdata[i] = data[i * nch + c];
}
vorbis_lpc_predict(lpci, tdata + data_len - lpc_order, lpc_order, tdata + data_len, extra_fwd + extra_bkwd, work);
if(extra_bkwd) {
for(int i = 0; i < extra_bkwd; i++)
data[(-i - 1) * nch + c] = tdata[data_len + i];
} else {
for(int i = 0; i < extra_fwd; i++)
data[(i + data_len) * nch + c] = tdata[data_len + i];
}
}
}

40
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#ifndef MY_LPC_H
#define MY_LPC_H
/* data - beginning of the data
* data_len - length of data (in samples) that are base for extrapolation
* nch - number of (interleaved) channels
* lpc_order - LPC order
* extra_bkwd - number of samples to pre-extrapolate
* extra_fwd - number of samples to post-extrapolate
*
* D = data; N = num_channels; LEN = data_len*N; EX = extra*N
*
* memory layout when invdir == false:
*
* [||||||||||||||||||||||||||||||||][||||||||||||||||||||][
* ^ D[0] ^ D[LEN] ^ D[LEN+EX]
*
* memory layout when invdir == true:
* ][||||||||||||||||||||][||||||||||||||||||||||||||||||||][
* ^ D[0] ^ D[LEN]
* ^ D[-1*N-EX] ^ D[-1*N]
*
*/
static const size_t LPC_ORDER = 32;
void lpc_extrapolate2(float * const data, const size_t data_len, const int nch, const int lpc_order, const size_t extra_bkwd, const size_t extra_fwd, void ** extrapolate_buffer, size_t * extrapolate_buffer_size);
static inline void lpc_extrapolate_bkwd(float * const data, const size_t data_len, const size_t prime_len, const int nch, const int lpc_order, const size_t extra_bkwd, void ** extrapolate_buffer, size_t * extrapolate_buffer_size)
{
(void)data_len;
lpc_extrapolate2(data, prime_len, nch, lpc_order, extra_bkwd, 0, extrapolate_buffer, extrapolate_buffer_size);
}
static inline void lpc_extrapolate_fwd(float * const data, const size_t data_len, const size_t prime_len, const int nch, const int lpc_order, const size_t extra_fwd, void ** extrapolate_buffer, size_t * extrapolate_buffer_size)
{
lpc_extrapolate2(data + (data_len - prime_len)*nch, prime_len, nch, lpc_order, 0, extra_fwd, extrapolate_buffer, extrapolate_buffer_size);
}
#endif

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/* Copyright (c) 2018 lvqcl. All rights reserved.
*
* This library is free software; you can redistribute it and/or modify it
* under the terms of the GNU Lesser General Public License as published by
* the Free Software Foundation; either version 2.1 of the License, or (at
* your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser
* General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with this library; if not, write to the Free Software Foundation,
* Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef UTIL_H_
#define UTIL_H_
#ifndef min
#define min(a,b) (((a)<(b))?(a):(b))
#endif
#ifndef max
#define max(a,b) (((a)>(b))?(a):(b))
#endif
static inline unsigned local_gcd(unsigned a, unsigned b)
{
if (a == 0 || b == 0) return 0;
unsigned c = a % b;
while (c != 0) { a = b; b = c; c = a % b; }
return b;
}
/*
In: *r1 and *r2: samplerates;
Out: *r1 and *r2: numbers of samples;
multiply r1 and r2 by n so that durations is 1/N th of second;
limit n so that r1 and r2 aren't bigger than M
*/
static void samples_len(unsigned* r1, unsigned* r2, unsigned N, unsigned M) // example: r1 = 44100, r2 = 48000, N = 20, M = 8192
{
if (r1 == 0 || r2 == 0) return;
unsigned v = local_gcd(*r1, *r2); // v = 300
*r1 /= v; *r2 /= v; // r1 = 147; r2 = 160 == 1/300th of second
unsigned n = (v + N-1) / N; // n = 300/20 = 15 times
unsigned z = max(*r1, *r2); // z = 160
if (z*n > M) n = M / z; // 160*15 = 2400 < 8192;; if M == 1024: n = 1024/160 = 6; 160*6 = 960
if (n < 1) n = 1;
*r1 *= n; *r2 *= n; // r1 = 147*15 = 2205 samples, r2 = 160*15 = 2400 samples
}
#endif