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Replace Core Audio output with Core Media runtime 

The output now uses AVSampleBufferAudioRenderer to play all formats, and
uses that to resample. It also supports Spatial Audio on macOS 12.0 or
newer. Note that there are some outstanding bugs with Spatial Audio
support. Namely that it appears to be limited to only 192 kHz at mono or
stereo, or 352800 Hz at surround configurations. This breaks DSD64
playback at stereo formats, as well as possibly other things. This is
entirely an Apple bug. I have reported it to Apple with reference code
FB10441301 for reference, in case anyone else wants to complain that it
isn't fixed.

Signed-off-by: Christopher Snowhill <kode54@gmail.com>
lastfm
Christopher Snowhill 2022-06-23 23:17:31 -07:00
parent 2c2a058126
commit dccb7f8b47
37 changed files with 1118 additions and 4007 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

3
.gitmodules vendored
View File

@ -19,6 +19,3 @@
[submodule "Frameworks/libsidplayfp/sidplayfp"]
path = Frameworks/libsidplayfp/sidplayfp
url = https://github.com/kode54/libsidplayfp.git
[submodule "Audio/ThirdParty/r8brain-free-src"]
path = Audio/ThirdParty/r8brain-free-src
url = https://github.com/kode54/r8brain-free-src

1
Audio/AudioPlayer.h
View File

@ -116,6 +116,7 @@ using std::atomic_bool;
- (void)setShouldContinue:(BOOL)s;
//- (BufferChain *)bufferChain;
- (void)launchOutputThread;
- (BOOL)selectNextBuffer;
- (void)endOfInputPlayed;
- (void)reportPlayCount;
- (void)sendDelegateMethod:(SEL)selector withVoid:(void *)obj waitUntilDone:(BOOL)wait;

64
Audio/AudioPlayer.m
View File

@ -83,7 +83,7 @@
bufferChain = [[BufferChain alloc] initWithController:self];
[self notifyStreamChanged:userInfo];
while(![bufferChain open:url withOutputFormat:[output format] withOutputConfig:[output config] withUserInfo:userInfo withRGInfo:rgi]) {
while(![bufferChain open:url withUserInfo:userInfo withRGInfo:rgi]) {
bufferChain = nil;
[self requestNextStream:userInfo];
@ -369,7 +369,7 @@
}
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] withOutputFormat:[output format] withOutputConfig:[output config] withUserInfo:nextStreamUserInfo withRGInfo:nextStreamRGInfo]) {
if([lastChain setTrack:nextStream] && [newChain openWithInput:[lastChain inputNode] withUserInfo:nextStreamUserInfo withRGInfo:nextStreamRGInfo]) {
[newChain setStreamURL:nextStream];
[self addChainToQueue:newChain];
@ -384,7 +384,7 @@
lastChain = nil;
while(shouldContinue && ![newChain open:nextStream withOutputFormat:[output format] withOutputConfig:[output config] withUserInfo:nextStreamUserInfo withRGInfo:nextStreamRGInfo]) {
while(shouldContinue && ![newChain open:nextStream withUserInfo:nextStreamUserInfo withRGInfo:nextStreamRGInfo]) {
if(nextStream == nil) {
newChain = nil;
atomic_fetch_sub(&refCount, 1);
@ -423,35 +423,47 @@
}
}
- (BOOL)selectNextBuffer {
BOOL signalStopped = NO;
do {
@synchronized(chainQueue) {
endOfInputReached = NO;
if([chainQueue count] <= 0) {
// End of playlist
signalStopped = YES;
break;
}
bufferChain = nil;
bufferChain = [chainQueue objectAtIndex:0];
[chainQueue removeObjectAtIndex:0];
DLog(@"New!!! %@ %@", bufferChain, [[bufferChain inputNode] decoder]);
[semaphore signal];
}
} while(0);
if(signalStopped) {
[self stop];
bufferChain = nil;
[self notifyPlaybackStopped:nil];
return NO;
}
return YES;
}
- (void)endOfInputPlayed {
// Once we get here:
// - the buffer chain for the next playlist entry (started in endOfInputReached) have been working for some time
// already, so that there is some decoded and converted data to play
// - the buffer chain for the next entry is the first item in chainQueue
@synchronized(chainQueue) {
endOfInputReached = NO;
if([chainQueue count] <= 0) {
// End of playlist
[self stop];
bufferChain = nil;
[self notifyPlaybackStopped:nil];
return;
}
bufferChain = nil;
bufferChain = [chainQueue objectAtIndex:0];
[chainQueue removeObjectAtIndex:0];
DLog(@"New!!! %@ %@", bufferChain, [[bufferChain inputNode] decoder]);
[semaphore signal];
}
[self notifyStreamChanged:[bufferChain userInfo]];
[output setEndOfStream:NO];
}

6
Audio/Chain/BufferChain.h
View File

@ -28,15 +28,13 @@
- (id)initWithController:(id)c;
- (void)buildChain;
- (BOOL)open:(NSURL *)url withOutputFormat:(AudioStreamBasicDescription)outputFormat withOutputConfig:(uint32_t)outputConfig withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi;
- (BOOL)open:(NSURL *)url withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi;
// Used when changing tracks to reuse the same decoder
- (BOOL)openWithInput:(InputNode *)i withOutputFormat:(AudioStreamBasicDescription)outputFormat withOutputConfig:(uint32_t)outputConfig withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi;
- (BOOL)openWithInput:(InputNode *)i withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi;
// Used when resetting the decoder on seek
- (BOOL)openWithDecoder:(id<CogDecoder>)decoder
withOutputFormat:(AudioStreamBasicDescription)outputFormat
withOutputConfig:(uint32_t)outputConfig
withUserInfo:(id)userInfo
withRGInfo:(NSDictionary *)rgi;

30
Audio/Chain/BufferChain.m
View File

@ -40,7 +40,7 @@
finalNode = converterNode;
}
- (BOOL)open:(NSURL *)url withOutputFormat:(AudioStreamBasicDescription)outputFormat withOutputConfig:(uint32_t)outputConfig withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi {
- (BOOL)open:(NSURL *)url withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi {
[self setStreamURL:url];
[self setUserInfo:userInfo];
@ -66,13 +66,7 @@
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;
outputConfig = inputChannelConfig;
if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig outputFormat:outputFormat outputConfig:outputConfig isLossless:[[properties valueForKey:@"encoding"] isEqualToString:@"lossless"]])
if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig isLossless:[[properties valueForKey:@"encoding"] isEqualToString:@"lossless"]])
return NO;
[self setRGInfo:rgi];
@ -82,7 +76,7 @@
return YES;
}
- (BOOL)openWithInput:(InputNode *)i withOutputFormat:(AudioStreamBasicDescription)outputFormat withOutputConfig:(uint32_t)outputConfig withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi {
- (BOOL)openWithInput:(InputNode *)i withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi {
DLog(@"New buffer chain!");
[self setUserInfo:userInfo];
[self buildChain];
@ -97,14 +91,8 @@
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;
outputConfig = inputChannelConfig;
DLog(@"Input Properties: %@", properties);
if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig outputFormat:outputFormat outputConfig:outputConfig isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
return NO;
[self setRGInfo:rgi];
@ -113,8 +101,6 @@
}
- (BOOL)openWithDecoder:(id<CogDecoder>)decoder
withOutputFormat:(AudioStreamBasicDescription)outputFormat
withOutputConfig:(uint32_t)outputConfig
withUserInfo:(id)userInfo
withRGInfo:(NSDictionary *)rgi;
{
@ -134,13 +120,7 @@
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;
outputConfig = inputChannelConfig;
if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig outputFormat:outputFormat outputConfig:outputConfig isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
return NO;
[self setRGInfo:rgi];

29
Audio/Chain/ConverterNode.h
View File

@ -14,15 +14,11 @@
#import "Node.h"
#import "HeadphoneFilter.h"
#define DSD_DECIMATE 1
@interface ConverterNode : Node {
NSDictionary *rgInfo;
void *_r8bstate;
void *inputBuffer;
size_t inputBufferSize;
size_t inpSize, inpOffset;
@ -31,29 +27,12 @@
BOOL convertEntered;
BOOL paused;
BOOL skipResampler;
unsigned int PRIME_LEN_;
unsigned int N_samples_to_add_;
unsigned int N_samples_to_drop_;
unsigned int is_preextrapolated_;
unsigned int is_postextrapolated_;
int latencyEaten;
int latencyEatenPost;
double sampleRatio;
float volumeScale;
void *floatBuffer;
size_t floatBufferSize;
size_t floatSize, floatOffset;
void *extrapolateBuffer;
size_t extrapolateBufferSize;
#if DSD_DECIMATE
void **dsd2pcm;
size_t dsd2pcmCount;
@ -65,10 +44,8 @@
AudioStreamBasicDescription inputFormat;
AudioStreamBasicDescription floatFormat;
AudioStreamBasicDescription dmFloatFormat; // downmixed/upmixed float format
AudioStreamBasicDescription outputFormat;
uint32_t inputChannelConfig;
uint32_t outputChannelConfig;
BOOL streamFormatChanged;
AudioStreamBasicDescription newInputFormat;
@ -77,15 +54,13 @@
AudioChunk *lastChunkIn;
void *hdcd_decoder;
HeadphoneFilter *hFilter;
}
@property AudioStreamBasicDescription inputFormat;
- (id)initWithController:(id)c previous:(id)p;
- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inputFormat withInputConfig:(uint32_t)inputConfig outputFormat:(AudioStreamBasicDescription)outputFormat outputConfig:(uint32_t)outputConfig isLossless:(BOOL)lossless;
- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inputFormat withInputConfig:(uint32_t)inputConfig isLossless:(BOOL)lossless;
- (void)cleanUp;
- (void)process;
@ -93,8 +68,6 @@
- (void)setRGInfo:(NSDictionary *)rgi;
- (void)setOutputFormat:(AudioStreamBasicDescription)format outputConfig:(uint32_t)outputConfig;
- (void)inputFormatDidChange:(AudioStreamBasicDescription)format inputConfig:(uint32_t)inputConfig;
- (void)refreshVolumeScaling;

252
Audio/Chain/ConverterNode.mm
View File

@ -6,6 +6,9 @@
// Copyright 2005 __MyCompanyName__. All rights reserved.
//
#import <Accelerate/Accelerate.h>
#import <Foundation/Foundation.h>
#import "ConverterNode.h"
#import "BufferChain.h"
@ -13,17 +16,12 @@
#import "Logging.h"
#import "lpc.h"
#import "util.h"
#import "hdcd_decode2.h"
#ifdef _DEBUG
#import "BadSampleCleaner.h"
#endif
#import "r8bstate.h"
#if !DSD_DECIMATE
#include "dsd2float.h"
#endif
@ -56,7 +54,6 @@ static void *kConverterNodeContext = &kConverterNodeContext;
if(self) {
rgInfo = nil;
_r8bstate = 0;
inputBuffer = NULL;
inputBufferSize = 0;
floatBuffer = NULL;
@ -66,11 +63,6 @@ static void *kConverterNodeContext = &kConverterNodeContext;
convertEntered = NO;
paused = NO;
skipResampler = YES;
extrapolateBuffer = NULL;
extrapolateBufferSize = 0;
#if DSD_DECIMATE
dsd2pcm = NULL;
dsd2pcmCount = 0;
@ -458,7 +450,7 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
continue;
} else if(streamFormatChanged) {
[self cleanUp];
[self setupWithInputFormat:newInputFormat withInputConfig:newInputChannelConfig outputFormat:outputFormat outputConfig:outputChannelConfig isLossless:rememberedLossless];
[self setupWithInputFormat:newInputFormat withInputConfig:newInputChannelConfig isLossless:rememberedLossless];
continue;
} else
break;
@ -471,7 +463,6 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
UInt32 ioNumberPackets;
int amountReadFromFC;
int amountRead = 0;
int amountToIgnorePostExtrapolated = 0;
if(stopping)
return 0;
@ -496,13 +487,6 @@ tryagain:
// Approximately the most we want on input
ioNumberPackets = CHUNK_SIZE;
if(!skipResampler && ioNumberPackets < PRIME_LEN_)
ioNumberPackets = PRIME_LEN_;
// We want to upscale this count if the ratio is below zero
if(sampleRatio < 1.0) {
ioNumberPackets = ((uint32_t)(ioNumberPackets / sampleRatio) + 15) & ~15;
}
#if DSD_DECIMATE
const size_t sizeScale = 3;
@ -555,41 +539,8 @@ tryagain:
}
}
// Pad end of track with input format silence
if(stopping || paused || streamFormatChanged || [self shouldContinue] == NO || [self endOfStream] == YES) {
if(!skipResampler && !is_postextrapolated_) {
#if DSD_DECIMATE
if(dsd2pcm) {
uint32_t amountToSkip = dsd2pcmLatency * inputFormat.mBytesPerPacket;
memset(((uint8_t *)inputBuffer) + bytesReadFromInput, 0x55, amountToSkip);
bytesReadFromInput += amountToSkip;
}
#endif
is_postextrapolated_ = 1;
#if DSD_DECIMATE
} else if(!is_postextrapolated_ && dsd2pcm) {
is_postextrapolated_ = 3;
#endif
}
}
BOOL isBigEndian = !!(inputFormat.mFormatFlags & kAudioFormatFlagIsBigEndian);
if(!bytesReadFromInput && streamFormatChanged && !skipResampler && is_postextrapolated_ < 2) {
AudioChunk *chunk = lastChunkIn;
lastChunkIn = nil;
AudioStreamBasicDescription inf = [chunk format];
size_t frameCount = [chunk frameCount];
size_t bytesRead = frameCount * inf.mBytesPerPacket;
if(frameCount) {
amountToIgnorePostExtrapolated = (int)frameCount;
NSData *samples = [chunk removeSamples:frameCount];
memcpy(inputBuffer, [samples bytes], bytesRead);
}
bytesReadFromInput += bytesRead;
}
if(!bytesReadFromInput) {
convertEntered = NO;
return amountRead;
@ -701,82 +652,9 @@ tryagain:
#endif
}
// Extrapolate start
if(!skipResampler && !is_preextrapolated_) {
size_t samples_in_buffer = bytesReadFromInput / floatFormat.mBytesPerPacket;
size_t prime = min(samples_in_buffer, PRIME_LEN_);
size_t _N_samples_to_add_ = N_samples_to_add_;
#if DSD_DECIMATE
if(dsd2pcm) _N_samples_to_add_ += dsd2pcmLatency;
#endif
size_t newSize = _N_samples_to_add_ * floatFormat.mBytesPerPacket;
newSize += bytesReadFromInput;
if(newSize > inputBufferSize) {
inputBuffer = realloc(inputBuffer, inputBufferSize = newSize * 3);
}
size_t bytesToSkip = 0;
#if DSD_DECIMATE
if(dsd2pcm) {
bytesToSkip = dsd2pcmLatency * floatFormat.mBytesPerPacket;
if(bytesReadFromInput >= bytesToSkip) {
bytesReadFromInput -= bytesToSkip;
} else {
bytesToSkip = 0;
}
}
#endif
memmove(((uint8_t *)inputBuffer) + N_samples_to_add_ * floatFormat.mBytesPerPacket, ((uint8_t *)inputBuffer) + bytesToSkip, bytesReadFromInput);
lpc_extrapolate_bkwd((float *)(((uint8_t *)inputBuffer) + _N_samples_to_add_ * floatFormat.mBytesPerPacket), samples_in_buffer, prime, floatFormat.mChannelsPerFrame, LPC_ORDER, _N_samples_to_add_, &extrapolateBuffer, &extrapolateBufferSize);
#ifdef _DEBUG
[BadSampleCleaner cleanSamples:(float *)inputBuffer
amount:_N_samples_to_add_ * floatFormat.mChannelsPerFrame
location:@"pre-extrapolated data"];
#endif
bytesReadFromInput += _N_samples_to_add_ * floatFormat.mBytesPerPacket;
latencyEaten = N_samples_to_drop_;
#if DSD_DECIMATE
if(dsd2pcm) latencyEaten += (int)ceil(dsd2pcmLatency * sampleRatio);
#endif
is_preextrapolated_ = 2;
#if DSD_DECIMATE
} else if(dsd2pcm && !is_preextrapolated_) {
latencyEaten = dsd2pcmLatency;
is_preextrapolated_ = 3;
#endif
}
if(is_postextrapolated_ == 1) {
size_t samples_in_buffer = bytesReadFromInput / floatFormat.mBytesPerPacket;
size_t prime = min(samples_in_buffer, PRIME_LEN_);
size_t newSize = bytesReadFromInput;
newSize += N_samples_to_add_ * floatFormat.mBytesPerPacket;
if(newSize > inputBufferSize) {
inputBuffer = realloc(inputBuffer, inputBufferSize = newSize * 3);
}
lpc_extrapolate_fwd((float *)inputBuffer, samples_in_buffer, prime, floatFormat.mChannelsPerFrame, LPC_ORDER, N_samples_to_add_, &extrapolateBuffer, &extrapolateBufferSize);
#ifdef _DEBUG
[BadSampleCleaner cleanSamples:(float *)(inputBuffer) + samples_in_buffer * floatFormat.mChannelsPerFrame
amount:N_samples_to_add_ * floatFormat.mChannelsPerFrame
location:@"post-extrapolated data"];
#endif
bytesReadFromInput += N_samples_to_add_ * floatFormat.mBytesPerPacket;
latencyEatenPost = N_samples_to_drop_;
is_postextrapolated_ = 2;
} else if(is_postextrapolated_ == 3) { // No need to skip the end
latencyEatenPost = 0;
}
// Input now contains bytesReadFromInput worth of floats, in the input sample rate
inpSize = bytesReadFromInput;
inpOffset = amountToIgnorePostExtrapolated * floatFormat.mBytesPerPacket;
inpOffset = 0;
}
if(inpOffset != inpSize && floatOffset == floatSize) {
@ -788,7 +666,6 @@ tryagain:
ioNumberPackets = (UInt32)inputSamples;
ioNumberPackets = (UInt32)ceil((float)ioNumberPackets * sampleRatio);
ioNumberPackets = (ioNumberPackets + 255) & ~255;
size_t newSize = ioNumberPackets * floatFormat.mBytesPerPacket;
@ -805,64 +682,12 @@ tryagain:
size_t inputDone = 0;
size_t outputDone = 0;
if(!skipResampler) {
// This is needed at least for the flush, or else there won't be room for it
ioNumberPackets += ((r8bstate *)_r8bstate)->latency();
#ifdef _DEBUG
[BadSampleCleaner cleanSamples:(float *)(((uint8_t *)inputBuffer) + inpOffset)
amount:inputSamples * floatFormat.mChannelsPerFrame
location:@"resampler input"];
#endif
outputDone = ((r8bstate *)_r8bstate)->resample((float *)(((uint8_t *)inputBuffer) + inpOffset), inputSamples, &inputDone, (float *)floatBuffer, ioNumberPackets);
#ifdef _DEBUG
[BadSampleCleaner cleanSamples:(float *)floatBuffer
amount:outputDone * floatFormat.mChannelsPerFrame
location:@"resampler output"];
#endif
if(latencyEatenPost) {
// Post file flush
size_t odone = 0;
do {
odone = ((r8bstate *)_r8bstate)->flush((float *)(((uint8_t *)floatBuffer) + outputDone * floatFormat.mBytesPerPacket), ioNumberPackets - outputDone);
#ifdef _DEBUG
[BadSampleCleaner cleanSamples:(float *)(floatBuffer + outputDone * floatFormat.mBytesPerPacket)
amount:odone * floatFormat.mChannelsPerFrame
location:@"resampler flushed output"];
#endif
outputDone += odone;
} while(odone > 0);
}
} else {
memcpy(floatBuffer, (((uint8_t *)inputBuffer) + inpOffset), inputSamples * floatFormat.mBytesPerPacket);
inputDone = inputSamples;
outputDone = inputSamples;
}
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, ((uint8_t *)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;
}
amountReadFromFC = (int)(outputDone * floatFormat.mBytesPerPacket);
scale_by_volume((float *)floatBuffer, amountReadFromFC / sizeof(float), volumeScale
@ -882,7 +707,7 @@ tryagain:
if(ioNumberPackets > (floatSize - floatOffset))
ioNumberPackets = (UInt32)(floatSize - floatOffset);
ioNumberPackets -= ioNumberPackets % outputFormat.mBytesPerPacket;
ioNumberPackets -= ioNumberPackets % dmFloatFormat.mBytesPerPacket;
memcpy(((uint8_t *)dest) + amountRead, ((uint8_t *)floatBuffer) + floatOffset, ioNumberPackets);
@ -953,16 +778,11 @@ static float db_to_scale(float db) {
volumeScale = scale;
}
- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inf withInputConfig:(uint32_t)inputConfig outputFormat:(AudioStreamBasicDescription)outf outputConfig:(uint32_t)outputConfig isLossless:(BOOL)lossless {
- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inf withInputConfig:(uint32_t)inputConfig isLossless:(BOOL)lossless {
// Make the converter
inputFormat = inf;
outputFormat = outf;
inputChannelConfig = inputConfig;
outputChannelConfig = outputConfig;
nodeFormat = outputFormat;
nodeChannelConfig = outputChannelConfig;
rememberedLossless = lossless;
@ -1010,38 +830,12 @@ static float db_to_scale(float db) {
// This is a post resampler, post-down/upmix format
dmFloatFormat = floatFormat;
dmFloatFormat.mSampleRate = outputFormat.mSampleRate;
dmFloatFormat.mChannelsPerFrame = outputFormat.mChannelsPerFrame;
dmFloatFormat.mBytesPerFrame = (32 / 8) * dmFloatFormat.mChannelsPerFrame;
dmFloatFormat.mBytesPerPacket = dmFloatFormat.mBytesPerFrame * floatFormat.mFramesPerPacket;
skipResampler = outputFormat.mSampleRate == floatFormat.mSampleRate;
sampleRatio = (double)outputFormat.mSampleRate / (double)floatFormat.mSampleRate;
if(!skipResampler) {
const int channelCount = floatFormat.mChannelsPerFrame;
_r8bstate = (void *)(new r8bstate(channelCount, 1024, floatFormat.mSampleRate, outputFormat.mSampleRate));
PRIME_LEN_ = max(floatFormat.mSampleRate / 20, 1024u);
PRIME_LEN_ = min(PRIME_LEN_, 16384u);
PRIME_LEN_ = max(PRIME_LEN_, 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_ = 0;
is_postextrapolated_ = 0;
}
latencyEaten = 0;
latencyEatenPost = 0;
nodeFormat = dmFloatFormat;
nodeChannelConfig = inputChannelConfig;
PrintStreamDesc(&inf);
PrintStreamDesc(&outf);
PrintStreamDesc(&nodeFormat);
[self refreshVolumeScaling];
@ -1063,12 +857,6 @@ static float db_to_scale(float db) {
[self cleanUp];
}
- (void)setOutputFormat:(AudioStreamBasicDescription)format outputConfig:(uint32_t)outputConfig {
DLog(@"SETTING OUTPUT FORMAT!");
outputFormat = format;
outputChannelConfig = outputConfig;
}
- (void)inputFormatDidChange:(AudioStreamBasicDescription)format inputConfig:(uint32_t)inputConfig {
DLog(@"FORMAT CHANGED");
paused = YES;
@ -1076,7 +864,7 @@ static float db_to_scale(float db) {
usleep(500);
}
[self cleanUp];
[self setupWithInputFormat:format withInputConfig:inputConfig outputFormat:outputFormat outputConfig:outputChannelConfig isLossless:rememberedLossless];
[self setupWithInputFormat:format withInputConfig:inputConfig isLossless:rememberedLossless];
}
- (void)setRGInfo:(NSDictionary *)rgi {
@ -1090,17 +878,10 @@ static float db_to_scale(float db) {
while(convertEntered) {
usleep(500);
}
if(hFilter) {
hFilter = nil;
}
if(hdcd_decoder) {
free(hdcd_decoder);
hdcd_decoder = NULL;
}
if(_r8bstate) {
delete(r8bstate *)_r8bstate;
_r8bstate = NULL;
}
#if DSD_DECIMATE
if(dsd2pcm && dsd2pcmCount) {
for(size_t i = 0; i < dsd2pcmCount; ++i) {
@ -1111,11 +892,6 @@ static float db_to_scale(float db) {
dsd2pcm = NULL;
}
#endif
if(extrapolateBuffer) {
free(extrapolateBuffer);
extrapolateBuffer = NULL;
extrapolateBufferSize = 0;
}
if(floatBuffer) {
free(floatBuffer);
floatBuffer = NULL;

4
Audio/Chain/Downmix.h
View File

@ -9,11 +9,7 @@
#import <CoreAudio/CoreAudio.h>
#import <Foundation/Foundation.h>
#import "HeadphoneFilter.h"
@interface DownmixProcessor : NSObject {
HeadphoneFilter *hFilter;
AudioStreamBasicDescription inputFormat;
AudioStreamBasicDescription outputFormat;

80
Audio/Chain/Downmix.m
View File

@ -275,92 +275,12 @@ static void *kDownmixProcessorContext = &kDownmixProcessorContext;
inConfig = iConfig;
outConfig = oConfig;
[self setupVirt];
[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.headphoneVirtualization" options:0 context:kDownmixProcessorContext];
[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.hrirPath" options:0 context:kDownmixProcessorContext];
}
return self;
}
- (void)dealloc {
[[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.headphoneVirtualization" context:kDownmixProcessorContext];
[[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.hrirPath" context:kDownmixProcessorContext];
}
- (void)setupVirt {
@synchronized(hFilter) {
hFilter = nil;
}
BOOL hVirt = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] boolForKey:@"headphoneVirtualization"];
if(hVirt &&
outputFormat.mChannelsPerFrame >= 2 &&
(outConfig & AudioConfigStereo) == AudioConfigStereo &&
inputFormat.mChannelsPerFrame >= 1 &&
(inConfig & (AudioConfig7Point1 | AudioChannelBackCenter)) != 0) {
NSString *userPreset = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] stringForKey:@"hrirPath"];
NSURL *presetUrl = nil;
if(userPreset && ![userPreset isEqualToString:@""]) {
presetUrl = [NSURL fileURLWithPath:userPreset];
if(![HeadphoneFilter validateImpulseFile:presetUrl])
presetUrl = nil;
}
if(!presetUrl) {
presetUrl = [[NSBundle mainBundle] URLForResource:@"gsx" withExtension:@"wv"];
if(![HeadphoneFilter validateImpulseFile:presetUrl])
presetUrl = nil;
}
if(presetUrl) {
@synchronized(hFilter) {
hFilter = [[HeadphoneFilter alloc] initWithImpulseFile:presetUrl forSampleRate:outputFormat.mSampleRate withInputChannels:inputFormat.mChannelsPerFrame withConfig:inConfig];
}
}
}
}
- (void)observeValueForKeyPath:(NSString *)keyPath
ofObject:(id)object
change:(NSDictionary *)change
context:(void *)context {
if(context == kDownmixProcessorContext) {
DLog(@"SOMETHING CHANGED!");
if([keyPath isEqualToString:@"values.headphoneVirtualization"] ||
[keyPath isEqualToString:@"values.hrirPath"]) {
// Reset the converter, without rebuffering
[self setupVirt];
}
} else {
[super observeValueForKeyPath:keyPath ofObject:object change:change context:context];
}
}
- (void)process:(const void *)inBuffer frameCount:(size_t)frames output:(void *)outBuffer {
@synchronized(hFilter) {
if(hFilter) {
uint32_t outChannels = outputFormat.mChannelsPerFrame;
if(outChannels > 2) {
float tempBuffer[frames * 2];
[hFilter process:(const float *)inBuffer sampleCount:frames toBuffer:&tempBuffer[0]];
cblas_scopy((int)frames, tempBuffer, 2, (float *)outBuffer, outChannels);
cblas_scopy((int)frames, tempBuffer + 1, 2, ((float *)outBuffer) + 1, outChannels);
for(size_t i = 2; i < outChannels; ++i) {
vDSP_vclr(((float *)outBuffer) + i, outChannels, (int)frames);
}
} else {
[hFilter process:(const float *)inBuffer sampleCount:frames toBuffer:(float *)outBuffer];
}
return;
}
}
if(inputFormat.mChannelsPerFrame > 2 && outConfig == AudioConfigStereo) {
downmix_to_stereo((const float *)inBuffer, inputFormat.mChannelsPerFrame, inConfig, (float *)outBuffer, frames);
} else if(inputFormat.mChannelsPerFrame > 1 && outConfig == AudioConfigMono) {

47
Audio/Chain/HeadphoneFilter.h
View File

@ -1,47 +0,0 @@
//
// HeadphoneFilter.h
// CogAudio Framework
//
// Created by Christopher Snowhill on 1/24/22.
//
#ifndef HeadphoneFilter_h
#define HeadphoneFilter_h
#import <Accelerate/Accelerate.h>
#import <Cocoa/Cocoa.h>
@interface HeadphoneFilter : NSObject {
vDSP_DFT_Setup dftSetupF;
vDSP_DFT_Setup dftSetupB;
size_t fftSize;
size_t fftSizeOver2;
size_t bufferSize;
size_t paddedBufferSize;
size_t channelCount;
DSPSplitComplex signal_fft;
DSPSplitComplex input_filtered_signal_per_channel[2];
DSPSplitComplex input_filtered_signal_totals[2];
DSPSplitComplex *impulse_responses;
float **prevInputs;
float *left_result;
float *right_result;
float *paddedSignal;
}
+ (BOOL)validateImpulseFile:(NSURL *)url;
- (id)initWithImpulseFile:(NSURL *)url forSampleRate:(double)sampleRate withInputChannels:(size_t)channels withConfig:(uint32_t)config;
- (void)process:(const float *)inBuffer sampleCount:(size_t)count toBuffer:(float *)outBuffer;
- (void)reset;
@end
#endif /* HeadphoneFilter_h */

678
Audio/Chain/HeadphoneFilter.mm
View File

@ -1,678 +0,0 @@
//
// HeadphoneFilter.m
// CogAudio Framework
//
// Created by Christopher Snowhill on 1/24/22.
//
#import "HeadphoneFilter.h"
#import "AudioChunk.h"
#import "AudioDecoder.h"
#import "AudioSource.h"
#import <stdlib.h>
#import "r8bstate.h"
#import "lpc.h"
#import "util.h"
@interface impulseCacheObject : NSObject {
}
@property NSURL *URL;
@property int sampleCount;
@property int channelCount;
@property double sampleRate;
@property double targetSampleRate;
@property NSData *data;
@end
@implementation impulseCacheObject
@synthesize URL;
@synthesize sampleCount;
@synthesize channelCount;
@synthesize sampleRate;
@synthesize targetSampleRate;
@synthesize data;
@end
@interface impulseCache : NSObject {
}
@property NSMutableArray<impulseCacheObject *> *cacheObjects;
+ (impulseCache *)sharedController;
- (const float *)getImpulse:(NSURL *)url sampleCount:(int *)sampleCount channelCount:(int *)channelCount sampleRate:(double)sampleRate;
@end
// Apparently _mm_malloc is Intel-only on newer macOS targets, so use supported posix_memalign
static void *_memalign_malloc(size_t size, size_t align) {
void *ret = NULL;
if(posix_memalign(&ret, align, size) != 0) {
return NULL;
}
return ret;
}
@implementation impulseCache
static impulseCache *_sharedController = nil;
+ (impulseCache *)sharedController {
@synchronized(self) {
if(!_sharedController) {
_sharedController = [[impulseCache alloc] init];
}
}
return _sharedController;
}
- (id)init {
self = [super init];
if(self) {
self.cacheObjects = [[NSMutableArray alloc] init];
}
return self;
}
- (impulseCacheObject *)addImpulse:(NSURL *)url sampleCount:(int)sampleCount channelCount:(int)channelCount originalSampleRate:(double)originalSampleRate targetSampleRate:(double)targetSampleRate impulseBuffer:(const float *)impulseBuffer {
impulseCacheObject *obj = [[impulseCacheObject alloc] init];
obj.URL = url;
obj.sampleCount = sampleCount;
obj.channelCount = channelCount;
obj.sampleRate = originalSampleRate;
obj.targetSampleRate = targetSampleRate;
obj.data = [NSData dataWithBytes:impulseBuffer length:(sampleCount * channelCount * sizeof(float))];
@synchronized(self.cacheObjects) {
[self.cacheObjects addObject:obj];
}
return obj;
}
- (const float *)getImpulse:(NSURL *)url sampleCount:(int *)retSampleCount channelCount:(int *)retImpulseChannels sampleRate:(double)sampleRate {
BOOL impulseFound = NO;
const float *impulseData = NULL;
double sampleRateOfSource = 0;
int sampleCount = 0;
int impulseChannels = 0;
impulseCacheObject *cacheObject = nil;
@synchronized(self.cacheObjects) {
for(impulseCacheObject *obj in self.cacheObjects) {
if([obj.URL isEqualTo:url] &&
obj.targetSampleRate == sampleRate) {
*retSampleCount = obj.sampleCount;
*retImpulseChannels = obj.channelCount;
return (const float *)[obj.data bytes];
}
}
for(impulseCacheObject *obj in self.cacheObjects) {
if([obj.URL isEqualTo:url] &&
obj.sampleRate == obj.targetSampleRate) {
impulseData = (const float *)[obj.data bytes];
sampleCount = obj.sampleCount;
impulseChannels = obj.channelCount;
sampleRateOfSource = obj.sampleRate;
impulseFound = YES;
break;
}
}
}
if(!impulseFound) {
id<CogSource> source = [AudioSource audioSourceForURL:url];
if(!source)
return NULL;
if(![source open:url])
return NULL;
id<CogDecoder> decoder = [AudioDecoder audioDecoderForSource:source];
if(decoder == nil) {
[source close];
source = nil;
return NULL;
}
if(![decoder open:source]) {
decoder = nil;
[source close];
source = nil;
return NULL;
}
NSDictionary *properties = [decoder properties];
sampleRateOfSource = [[properties objectForKey:@"sampleRate"] floatValue];
sampleCount = [[properties objectForKey:@"totalFrames"] intValue];
impulseChannels = [[properties objectForKey:@"channels"] intValue];
if([[properties objectForKey:@"floatingPoint"] boolValue] != YES ||
[[properties objectForKey:@"bitsPerSample"] intValue] != 32 ||
!([[properties objectForKey:@"endian"] isEqualToString:@"host"] ||
[[properties objectForKey:@"endian"] isEqualToString:@"little"]) ||
(impulseChannels != 14 && impulseChannels != 7)) {
[decoder close];
decoder = nil;
[source close];
source = nil;
return NULL;
}
float *impulseBuffer = (float *)_memalign_malloc(sampleCount * sizeof(float) * impulseChannels, 16);
if(!impulseBuffer) {
[decoder close];
decoder = nil;
[source close];
source = nil;
return NULL;
}
if([decoder readAudio:impulseBuffer frames:sampleCount] != sampleCount) {
free(impulseBuffer);
[decoder close];
decoder = nil;
[source close];
source = nil;
return NULL;
}
[decoder close];
decoder = nil;
[source close];
source = nil;
cacheObject = [self addImpulse:url sampleCount:sampleCount channelCount:impulseChannels originalSampleRate:sampleRateOfSource targetSampleRate:sampleRateOfSource impulseBuffer:impulseBuffer];
free(impulseBuffer);
impulseData = (const float *)[cacheObject.data bytes];
}
if(sampleRateOfSource != sampleRate) {
double sampleRatio = sampleRate / sampleRateOfSource;
int resampledCount = (int)ceil((double)sampleCount * sampleRatio);
r8bstate *_r8bstate = new r8bstate(impulseChannels, 1024, sampleRateOfSource, sampleRate);
unsigned long PRIME_LEN_ = MAX(sampleRateOfSource / 20, 1024u);
PRIME_LEN_ = MIN(PRIME_LEN_, 16384u);
PRIME_LEN_ = MAX(PRIME_LEN_, 2 * LPC_ORDER + 1);
unsigned int N_samples_to_add_ = sampleRateOfSource;
unsigned int N_samples_to_drop_ = sampleRate;
samples_len(&N_samples_to_add_, &N_samples_to_drop_, 20, 8192u);
int resamplerLatencyIn = (int)N_samples_to_add_;
int resamplerLatencyOut = (int)N_samples_to_drop_;
float *tempImpulse = (float *)_memalign_malloc((sampleCount + resamplerLatencyIn * 2 + 1024) * sizeof(float) * impulseChannels, 16);
if(!tempImpulse) {
return nil;
}
resampledCount += resamplerLatencyOut * 2 + 1024;
float *resampledImpulse = (float *)_memalign_malloc(resampledCount * sizeof(float) * impulseChannels, 16);
if(!resampledImpulse) {
free(tempImpulse);
return nil;
}
size_t prime = MIN(sampleCount, PRIME_LEN_);
void *extrapolate_buffer = NULL;
size_t extrapolate_buffer_size = 0;
memcpy(tempImpulse + resamplerLatencyIn * impulseChannels, impulseData, sampleCount * sizeof(float) * impulseChannels);
lpc_extrapolate_bkwd(tempImpulse + N_samples_to_add_ * impulseChannels, sampleCount, prime, impulseChannels, LPC_ORDER, N_samples_to_add_, &extrapolate_buffer, &extrapolate_buffer_size);
lpc_extrapolate_fwd(tempImpulse + N_samples_to_add_ * impulseChannels, sampleCount, prime, impulseChannels, LPC_ORDER, N_samples_to_add_, &extrapolate_buffer, &extrapolate_buffer_size);
free(extrapolate_buffer);
size_t inputDone = 0;
size_t outputDone = 0;
outputDone = _r8bstate->resample(tempImpulse, sampleCount + N_samples_to_add_ * 2, &inputDone, resampledImpulse, resampledCount);
free(tempImpulse);
if(outputDone < resampledCount) {
outputDone += _r8bstate->flush(resampledImpulse + outputDone * impulseChannels, resampledCount - outputDone);
}
delete _r8bstate;
outputDone -= N_samples_to_drop_ * 2;
// Do this instead of the memmove
float *resampledImpulseData = resampledImpulse + N_samples_to_drop_ * impulseChannels;
/*memmove(resampledImpulse, resampledImpulse + N_samples_to_drop_ * impulseChannels, outputDone * sizeof(float) * impulseChannels);*/
sampleCount = (int)outputDone;
// Normalize resampled impulse by sample ratio
float fSampleRatio = (float)sampleRatio;
vDSP_vsdiv(resampledImpulseData, 1, &fSampleRatio, resampledImpulseData, 1, sampleCount * impulseChannels);
cacheObject = [self addImpulse:url sampleCount:sampleCount channelCount:impulseChannels originalSampleRate:sampleRateOfSource targetSampleRate:sampleRate impulseBuffer:resampledImpulseData];
free(resampledImpulse);
impulseData = (const float *)[cacheObject.data bytes];
}
*retSampleCount = sampleCount;
*retImpulseChannels = impulseChannels;
return impulseData;
}
@end
@implementation HeadphoneFilter
enum {
speaker_is_back_center = -1,
speaker_not_present = -2,
};
static const uint32_t max_speaker_index = 10;
static const int8_t speakers_to_hesuvi_7[11][2] = {
// front left
{ 0, 1 },
// front right
{ 1, 0 },
// front center
{ 6, 6 },
// lfe
{ 6, 6 },
// back left
{ 4, 5 },
// back right
{ 5, 4 },
// front center left
{ speaker_not_present, speaker_not_present },
// front center right
{ speaker_not_present, speaker_not_present },
// back center
{ speaker_is_back_center, speaker_is_back_center },
// side left
{ 2, 3 },
// side right
{ 3, 2 }
};
static const int8_t speakers_to_hesuvi_14[11][2] = {
// front left
{ 0, 1 },
// front right
{ 8, 7 },
// front center
{ 6, 13 },
// lfe
{ 6, 13 },
// back left
{ 4, 5 },
// back right
{ 12, 11 },
// front center left
{ speaker_not_present, speaker_not_present },
// front center right
{ speaker_not_present, speaker_not_present },
// back center
{ speaker_is_back_center, speaker_is_back_center },
// side left
{ 2, 3 },
// side right
{ 10, 9 }
};
+ (BOOL)validateImpulseFile:(NSURL *)url {
id<CogSource> source = [AudioSource audioSourceForURL:url];
if(!source)
return NO;
if(![source open:url])
return NO;
id<CogDecoder> decoder = [AudioDecoder audioDecoderForSource:source];
if(decoder == nil) {
[source close];
source = nil;
return NO;
}
if(![decoder open:source]) {
decoder = nil;
[source close];
source = nil;
return NO;
}
NSDictionary *properties = [decoder properties];
[decoder close];
decoder = nil;
[source close];
source = nil;
int impulseChannels = [[properties objectForKey:@"channels"] intValue];
if([[properties objectForKey:@"floatingPoint"] boolValue] != YES ||
[[properties objectForKey:@"bitsPerSample"] intValue] != 32 ||
!([[properties objectForKey:@"endian"] isEqualToString:@"host"] ||
[[properties objectForKey:@"endian"] isEqualToString:@"little"]) ||
(impulseChannels != 14 && impulseChannels != 7))
return NO;
return YES;
}
- (id)initWithImpulseFile:(NSURL *)url forSampleRate:(double)sampleRate withInputChannels:(size_t)channels withConfig:(uint32_t)config {
self = [super init];
if(self) {
int sampleCount = 0;
int impulseChannels = 0;
const float *impulseBuffer = [[impulseCache sharedController] getImpulse:url sampleCount:&sampleCount channelCount:&impulseChannels sampleRate:sampleRate];
if(!impulseBuffer) {
return nil;
}
channelCount = channels;
bufferSize = 512;
fftSize = sampleCount + bufferSize;
int pow = 1;
while(fftSize > 2) {
pow++;
fftSize /= 2;
}
fftSize = 2 << pow;
float *deinterleavedImpulseBuffer = (float *)_memalign_malloc(fftSize * sizeof(float) * impulseChannels, 16);
if(!deinterleavedImpulseBuffer) {
return nil;
}
for(size_t i = 0; i < impulseChannels; ++i) {
cblas_scopy(sampleCount, impulseBuffer + i, impulseChannels, deinterleavedImpulseBuffer + i * fftSize, 1);
vDSP_vclr(deinterleavedImpulseBuffer + i * fftSize + sampleCount, 1, fftSize - sampleCount);
}
paddedBufferSize = fftSize;
fftSizeOver2 = (fftSize + 1) / 2;
const size_t fftSizeOver2Plus1 = fftSizeOver2 + 1; // DFT float overwrites plus one, double doesn't
dftSetupF = vDSP_DFT_zrop_CreateSetup(nil, fftSize, vDSP_DFT_FORWARD);
dftSetupB = vDSP_DFT_zrop_CreateSetup(nil, fftSize, vDSP_DFT_INVERSE);
if(!dftSetupF || !dftSetupB) {
free(deinterleavedImpulseBuffer);
return nil;
}
paddedSignal = (float *)_memalign_malloc(sizeof(float) * paddedBufferSize, 16);
if(!paddedSignal) {
free(deinterleavedImpulseBuffer);
return nil;
}
signal_fft.realp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
signal_fft.imagp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
if(!signal_fft.realp || !signal_fft.imagp) {
free(deinterleavedImpulseBuffer);
return nil;
}
input_filtered_signal_per_channel[0].realp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
input_filtered_signal_per_channel[0].imagp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
if(!input_filtered_signal_per_channel[0].realp ||
!input_filtered_signal_per_channel[0].imagp) {
free(deinterleavedImpulseBuffer);
return nil;
}
input_filtered_signal_per_channel[1].realp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
input_filtered_signal_per_channel[1].imagp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
if(!input_filtered_signal_per_channel[1].realp ||
!input_filtered_signal_per_channel[1].imagp) {
free(deinterleavedImpulseBuffer);
return nil;
}
input_filtered_signal_totals[0].realp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
input_filtered_signal_totals[0].imagp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
if(!input_filtered_signal_totals[0].realp ||
!input_filtered_signal_totals[0].imagp) {
free(deinterleavedImpulseBuffer);
return nil;
}
input_filtered_signal_totals[1].realp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
input_filtered_signal_totals[1].imagp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
if(!input_filtered_signal_totals[1].realp ||
!input_filtered_signal_totals[1].imagp) {
free(deinterleavedImpulseBuffer);
return nil;
}
impulse_responses = (DSPSplitComplex *)calloc(sizeof(DSPSplitComplex), channels * 2);
if(!impulse_responses) {
free(deinterleavedImpulseBuffer);
return nil;
}
for(size_t i = 0; i < channels; ++i) {
impulse_responses[i * 2 + 0].realp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
impulse_responses[i * 2 + 0].imagp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
impulse_responses[i * 2 + 1].realp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
impulse_responses[i * 2 + 1].imagp = (float *)_memalign_malloc(sizeof(float) * fftSizeOver2Plus1, 16);
if(!impulse_responses[i * 2 + 0].realp || !impulse_responses[i * 2 + 0].imagp ||
!impulse_responses[i * 2 + 1].realp || !impulse_responses[i * 2 + 1].imagp) {
free(deinterleavedImpulseBuffer);
return nil;
}
uint32_t channelFlag = [AudioChunk extractChannelFlag:(uint32_t)i fromConfig:config];
uint32_t channelIndex = [AudioChunk findChannelIndex:channelFlag];
int leftInChannel = speaker_not_present;
int rightInChannel = speaker_not_present;
if(impulseChannels == 7) {
if(channelIndex <= max_speaker_index) {
leftInChannel = speakers_to_hesuvi_7[channelIndex][0];
rightInChannel = speakers_to_hesuvi_7[channelIndex][1];
}
} else {
if(channelIndex <= max_speaker_index) {
leftInChannel = speakers_to_hesuvi_14[channelIndex][0];
rightInChannel = speakers_to_hesuvi_14[channelIndex][1];
}
}
if(leftInChannel == speaker_is_back_center || rightInChannel == speaker_is_back_center) {
float *temp;
if(impulseChannels == 7) {
temp = (float *)malloc(sizeof(float) * fftSize);
if(!temp) {
free(deinterleavedImpulseBuffer);
return nil;
}
cblas_scopy((int)fftSize, deinterleavedImpulseBuffer + 4 * fftSize, 1, temp, 1);
vDSP_vadd(temp, 1, deinterleavedImpulseBuffer + 5 * fftSize, 1, temp, 1, fftSize);
vDSP_ctoz((DSPComplex *)temp, 2, &impulse_responses[i * 2 + 0], 1, fftSizeOver2);
vDSP_ctoz((DSPComplex *)temp, 2, &impulse_responses[i * 2 + 1], 1, fftSizeOver2);
} else {
temp = (float *)malloc(sizeof(float) * fftSize * 2);
if(!temp) {
free(deinterleavedImpulseBuffer);
return nil;
}
cblas_scopy((int)fftSize, deinterleavedImpulseBuffer + 4 * fftSize, 1, temp, 1);
vDSP_vadd(temp, 1, deinterleavedImpulseBuffer + 12 * fftSize, 1, temp, 1, fftSize);
cblas_scopy((int)fftSize, deinterleavedImpulseBuffer + 5 * fftSize, 1, temp + fftSize, 1);
vDSP_vadd(temp + fftSize, 1, deinterleavedImpulseBuffer + 11 * fftSize, 1, temp + fftSize, 1, fftSize);
vDSP_ctoz((DSPComplex *)temp, 2, &impulse_responses[i * 2 + 0], 1, fftSizeOver2);
vDSP_ctoz((DSPComplex *)(temp + fftSize), 2, &impulse_responses[i * 2 + 1], 1, fftSizeOver2);
}
free(temp);
} else if(leftInChannel == speaker_not_present || rightInChannel == speaker_not_present) {
vDSP_ctoz((DSPComplex *)(deinterleavedImpulseBuffer + impulseChannels * fftSize), 2, &impulse_responses[i * 2 + 0], 1, fftSizeOver2);
vDSP_ctoz((DSPComplex *)(deinterleavedImpulseBuffer + impulseChannels * fftSize), 2, &impulse_responses[i * 2 + 1], 1, fftSizeOver2);
} else {
vDSP_ctoz((DSPComplex *)(deinterleavedImpulseBuffer + leftInChannel * fftSize), 2, &impulse_responses[i * 2 + 0], 1, fftSizeOver2);
vDSP_ctoz((DSPComplex *)(deinterleavedImpulseBuffer + rightInChannel * fftSize), 2, &impulse_responses[i * 2 + 1], 1, fftSizeOver2);
}
vDSP_DFT_Execute(dftSetupF, impulse_responses[i * 2 + 0].realp, impulse_responses[i * 2 + 0].imagp, impulse_responses[i * 2 + 0].realp, impulse_responses[i * 2 + 0].imagp);
vDSP_DFT_Execute(dftSetupF, impulse_responses[i * 2 + 1].realp, impulse_responses[i * 2 + 1].imagp, impulse_responses[i * 2 + 1].realp, impulse_responses[i * 2 + 1].imagp);
}
free(deinterleavedImpulseBuffer);
left_result = (float *)_memalign_malloc(sizeof(float) * fftSize, 16);
right_result = (float *)_memalign_malloc(sizeof(float) * fftSize, 16);
if(!left_result || !right_result)
return nil;
prevInputs = (float **)calloc(channels, sizeof(float *));
if(!prevInputs)
return nil;
for(size_t i = 0; i < channels; ++i) {
prevInputs[i] = (float *)_memalign_malloc(sizeof(float) * fftSize, 16);
if(!prevInputs[i])
return nil;
vDSP_vclr(prevInputs[i], 1, fftSize);
}
}
return self;
}
- (void)dealloc {
if(dftSetupF) vDSP_DFT_DestroySetup(dftSetupF);
if(dftSetupB) vDSP_DFT_DestroySetup(dftSetupB);
free(paddedSignal);
free(signal_fft.realp);
free(signal_fft.imagp);
free(input_filtered_signal_per_channel[0].realp);
free(input_filtered_signal_per_channel[0].imagp);
free(input_filtered_signal_per_channel[1].realp);
free(input_filtered_signal_per_channel[1].imagp);
free(input_filtered_signal_totals[0].realp);
free(input_filtered_signal_totals[0].imagp);
free(input_filtered_signal_totals[1].realp);
free(input_filtered_signal_totals[1].imagp);
if(impulse_responses) {
for(size_t i = 0; i < channelCount * 2; ++i) {
free(impulse_responses[i].realp);
free(impulse_responses[i].imagp);
}
free(impulse_responses);
}
free(left_result);
free(right_result);
if(prevInputs) {
for(size_t i = 0; i < channelCount; ++i) {
free(prevInputs[i]);
}
free(prevInputs);
}
}
- (void)process:(const float *)inBuffer sampleCount:(size_t)count toBuffer:(float *)outBuffer {
const float scale = 1.0 / (4.0 * (float)fftSize);
while(count > 0) {
const size_t countToDo = (count > bufferSize) ? bufferSize : count;
const size_t prevToDo = fftSize - countToDo;
vDSP_vclr(input_filtered_signal_totals[0].realp, 1, fftSizeOver2);
vDSP_vclr(input_filtered_signal_totals[0].imagp, 1, fftSizeOver2);
vDSP_vclr(input_filtered_signal_totals[1].realp, 1, fftSizeOver2);
vDSP_vclr(input_filtered_signal_totals[1].imagp, 1, fftSizeOver2);
for(size_t i = 0; i < channelCount; ++i) {
cblas_scopy((int)prevToDo, prevInputs[i] + countToDo, 1, paddedSignal, 1);
cblas_scopy((int)countToDo, inBuffer + i, (int)channelCount, paddedSignal + prevToDo, 1);
cblas_scopy((int)fftSize, paddedSignal, 1, prevInputs[i], 1);
vDSP_ctoz((DSPComplex *)paddedSignal, 2, &signal_fft, 1, fftSizeOver2);
vDSP_DFT_Execute(dftSetupF, signal_fft.realp, signal_fft.imagp, signal_fft.realp, signal_fft.imagp);
// One channel forward, then multiply and back twice
float preserveIRNyq = impulse_responses[i * 2 + 0].imagp[0];
float preserveSigNyq = signal_fft.imagp[0];
impulse_responses[i * 2 + 0].imagp[0] = 0;
signal_fft.imagp[0] = 0;
vDSP_zvmul(&signal_fft, 1, &impulse_responses[i * 2 + 0], 1, &input_filtered_signal_per_channel[0], 1, fftSizeOver2, 1);
input_filtered_signal_per_channel[0].imagp[0] = preserveIRNyq * preserveSigNyq;
impulse_responses[i * 2 + 0].imagp[0] = preserveIRNyq;
preserveIRNyq = impulse_responses[i * 2 + 1].imagp[0];
impulse_responses[i * 2 + 1].imagp[0] = 0;
vDSP_zvmul(&signal_fft, 1, &impulse_responses[i * 2 + 1], 1, &input_filtered_signal_per_channel[1], 1, fftSizeOver2, 1);
input_filtered_signal_per_channel[1].imagp[0] = preserveIRNyq * preserveSigNyq;
impulse_responses[i * 2 + 1].imagp[0] = preserveIRNyq;
vDSP_zvadd(&input_filtered_signal_totals[0], 1, &input_filtered_signal_per_channel[0], 1, &input_filtered_signal_totals[0], 1, fftSizeOver2);
vDSP_zvadd(&input_filtered_signal_totals[1], 1, &input_filtered_signal_per_channel[1], 1, &input_filtered_signal_totals[1], 1, fftSizeOver2);
}
vDSP_DFT_Execute(dftSetupB, input_filtered_signal_totals[0].realp, input_filtered_signal_totals[0].imagp, input_filtered_signal_totals[0].realp, input_filtered_signal_totals[0].imagp);
vDSP_DFT_Execute(dftSetupB, input_filtered_signal_totals[1].realp, input_filtered_signal_totals[1].imagp, input_filtered_signal_totals[1].realp, input_filtered_signal_totals[1].imagp);
vDSP_ztoc(&input_filtered_signal_totals[0], 1, (DSPComplex *)left_result, 2, fftSizeOver2);
vDSP_ztoc(&input_filtered_signal_totals[1], 1, (DSPComplex *)right_result, 2, fftSizeOver2);
float *left_ptr = left_result + prevToDo;
float *right_ptr = right_result + prevToDo;
vDSP_vsmul(left_ptr, 1, &scale, left_ptr, 1, countToDo);
vDSP_vsmul(right_ptr, 1, &scale, right_ptr, 1, countToDo);
cblas_scopy((int)countToDo, left_ptr, 1, outBuffer + 0, 2);
cblas_scopy((int)countToDo, right_ptr, 1, outBuffer + 1, 2);
inBuffer += countToDo * channelCount;
outBuffer += countToDo * 2;
count -= countToDo;
}
}
- (void)reset {
for(size_t i = 0; i < channelCount; ++i) {
vDSP_vclr(prevInputs[i], 1, fftSize);
}
}
@end

45
Audio/Chain/HeadphoneFilterPFFFT.h
View File

@ -1,45 +0,0 @@
//
// HeadphoneFilter.h
// CogAudio Framework
//
// Created by Christopher Snowhill on 1/24/22.
//
#ifndef HeadphoneFilter_h
#define HeadphoneFilter_h
#import <Cocoa/Cocoa.h>
#import "pffft.h"
@interface HeadphoneFilter : NSObject {
PFFFT_Setup *fftSetup;
size_t fftSize;
size_t bufferSize;
size_t paddedBufferSize;
size_t channelCount;
float *workBuffer;
float **impulse_responses;
float **prevInputs;
float *left_result;
float *right_result;
float *paddedSignal;
}
+ (BOOL)validateImpulseFile:(NSURL *)url;
- (id)initWithImpulseFile:(NSURL *)url forSampleRate:(double)sampleRate withInputChannels:(size_t)channels withConfig:(uint32_t)config;
- (void)process:(const float *)inBuffer sampleCount:(size_t)count toBuffer:(float *)outBuffer;
- (void)reset;
@end
#endif /* HeadphoneFilter_h */

447
Audio/Chain/HeadphoneFilterPFFFT.mm
View File

@ -1,447 +0,0 @@
//
// HeadphoneFilter.m
// CogAudio Framework
//
// Created by Christopher Snowhill on 1/24/22.
//
#import "AudioChunk.h"
#import "AudioDecoder.h"
#import "AudioSource.h"
#import "HeadphoneFilter.h"
#import <stdlib.h>
#import "r8bstate.h"
#import "lpc.h"
#import "util.h"
#import "pffft_double.h"
@implementation HeadphoneFilter
enum {
speaker_is_back_center = -1,
speaker_not_present = -2,
};
static const uint32_t max_speaker_index = 10;
static const int8_t speakers_to_hesuvi_7[11][2] = {
// front left
{ 0, 1 },
// front right
{ 1, 0 },
// front center
{ 6, 6 },
// lfe
{ 6, 6 },
// back left
{ 4, 5 },
// back right
{ 5, 4 },
// front center left
{ speaker_not_present, speaker_not_present },
// front center right
{ speaker_not_present, speaker_not_present },
// back center
{ speaker_is_back_center, speaker_is_back_center },
// side left
{ 2, 3 },
// side right
{ 3, 2 }
};
static const int8_t speakers_to_hesuvi_14[11][2] = {
// front left
{ 0, 1 },
// front right
{ 8, 7 },
// front center
{ 6, 13 },
// lfe
{ 6, 13 },
// back left
{ 4, 5 },
// back right
{ 12, 11 },
// front center left
{ speaker_not_present, speaker_not_present },
// front center right
{ speaker_not_present, speaker_not_present },
// back center
{ speaker_is_back_center, speaker_is_back_center },
// side left
{ 2, 3 },
// side right
{ 10, 9 }
};
+ (BOOL)validateImpulseFile:(NSURL *)url {
id<CogSource> source = [AudioSource audioSourceForURL:url];
if(!source)
return NO;
if(![source open:url])
return NO;
id<CogDecoder> decoder = [AudioDecoder audioDecoderForSource:source];
if(decoder == nil) {
[source close];
source = nil;
return NO;
}
if(![decoder open:source]) {
decoder = nil;
[source close];
source = nil;
return NO;
}
NSDictionary *properties = [decoder properties];
[decoder close];
decoder = nil;
[source close];
source = nil;
int impulseChannels = [[properties objectForKey:@"channels"] intValue];
if([[properties objectForKey:@"floatingPoint"] boolValue] != YES ||
[[properties objectForKey:@"bitsPerSample"] intValue] != 32 ||
!([[properties objectForKey:@"endian"] isEqualToString:@"host"] ||
[[properties objectForKey:@"endian"] isEqualToString:@"little"]) ||
(impulseChannels != 14 && impulseChannels != 7))
return NO;
return YES;
}
- (id)initWithImpulseFile:(NSURL *)url forSampleRate:(double)sampleRate withInputChannels:(size_t)channels withConfig:(uint32_t)config {
self = [super init];
if(self) {
id<CogSource> source = [AudioSource audioSourceForURL:url];
if(!source)
return nil;
if(![source open:url])
return nil;
id<CogDecoder> decoder = [AudioDecoder audioDecoderForSource:source];
if(decoder == nil) {
[source close];
source = nil;
return nil;
}
if(![decoder open:source]) {
decoder = nil;
[source close];
source = nil;
return nil;
}
NSDictionary *properties = [decoder properties];
double sampleRateOfSource = [[properties objectForKey:@"sampleRate"] doubleValue];
int sampleCount = [[properties objectForKey:@"totalFrames"] intValue];
int impulseChannels = [[properties objectForKey:@"channels"] intValue];
if([[properties objectForKey:@"floatingPoint"] boolValue] != YES ||
[[properties objectForKey:@"bitsPerSample"] intValue] != 32 ||
!([[properties objectForKey:@"endian"] isEqualToString:@"host"] ||
[[properties objectForKey:@"endian"] isEqualToString:@"little"]) ||
(impulseChannels != 14 && impulseChannels != 7)) {
[decoder close];
decoder = nil;
[source close];
source = nil;
return nil;
}
float *impulseBuffer = (float *)pffft_aligned_malloc(sampleCount * sizeof(float) * impulseChannels);
if(!impulseBuffer) {
[decoder close];
decoder = nil;
[source close];
source = nil;
return nil;
}
if([decoder readAudio:impulseBuffer frames:sampleCount] != sampleCount) {
pffft_aligned_free(impulseBuffer);
[decoder close];
decoder = nil;
[source close];
source = nil;
return nil;
}
[decoder close];
decoder = nil;
[source close];
source = nil;
if(sampleRateOfSource != sampleRate) {
double sampleRatio = sampleRate / sampleRateOfSource;
int resampledCount = (int)ceil((double)sampleCount * sampleRatio);
r8bstate *_r8bstate = new r8bstate(impulseChannels, 1024, sampleRateOfSource, sampleRate);
unsigned long PRIME_LEN_ = MAX(sampleRateOfSource / 20, 1024u);
PRIME_LEN_ = MIN(PRIME_LEN_, 16384u);
PRIME_LEN_ = MAX(PRIME_LEN_, 2 * LPC_ORDER + 1);
unsigned int N_samples_to_add_ = sampleRateOfSource;
unsigned int N_samples_to_drop_ = sampleRate;
samples_len(&N_samples_to_add_, &N_samples_to_drop_, 20, 8192u);
int resamplerLatencyIn = (int)N_samples_to_add_;
int resamplerLatencyOut = (int)N_samples_to_drop_;
float *tempImpulse = (float *)pffft_aligned_malloc((sampleCount + resamplerLatencyIn * 2 + 1024) * sizeof(float) * impulseChannels);
if(!tempImpulse) {
pffft_aligned_free(impulseBuffer);
return nil;
}
resampledCount += resamplerLatencyOut * 2 + 1024;
float *resampledImpulse = (float *)pffft_aligned_malloc(resampledCount * sizeof(float) * impulseChannels);
if(!resampledImpulse) {
pffft_aligned_free(impulseBuffer);
pffft_aligned_free(tempImpulse);
return nil;
}
size_t prime = MIN(sampleCount, PRIME_LEN_);
void *extrapolate_buffer = NULL;
size_t extrapolate_buffer_size = 0;
memcpy(tempImpulse + resamplerLatencyIn * impulseChannels, impulseBuffer, sampleCount * sizeof(float) * impulseChannels);
lpc_extrapolate_bkwd(tempImpulse + N_samples_to_add_ * impulseChannels, sampleCount, prime, impulseChannels, LPC_ORDER, N_samples_to_add_, &extrapolate_buffer, &extrapolate_buffer_size);
lpc_extrapolate_fwd(tempImpulse + N_samples_to_add_ * impulseChannels, sampleCount, prime, impulseChannels, LPC_ORDER, N_samples_to_add_, &extrapolate_buffer, &extrapolate_buffer_size);
free(extrapolate_buffer);
size_t inputDone = 0;
size_t outputDone = 0;
outputDone = _r8bstate->resample(tempImpulse, sampleCount + N_samples_to_add_ * 2, &inputDone, resampledImpulse, resampledCount);
if(outputDone < resampledCount) {
outputDone += _r8bstate->flush(resampledImpulse + outputDone * impulseChannels, resampledCount - outputDone);
}
delete _r8bstate;
outputDone -= N_samples_to_drop_ * 2;
memmove(resampledImpulse, resampledImpulse + N_samples_to_drop_ * impulseChannels, outputDone * sizeof(float) * impulseChannels);
pffft_aligned_free(tempImpulse);
pffft_aligned_free(impulseBuffer);
impulseBuffer = resampledImpulse;
sampleCount = (int)outputDone;
// Normalize resampled impulse by sample ratio
float fSampleRatio = (float)sampleRatio;
vDSP_vsdiv(impulseBuffer, 1, &fSampleRatio, impulseBuffer, 1, sampleCount * impulseChannels);
}
channelCount = channels;
bufferSize = 512;
fftSize = sampleCount + bufferSize;
fftSize = (size_t)pffftd_next_power_of_two((int)fftSize);
float *deinterleavedImpulseBuffer = (float *)pffft_aligned_malloc(fftSize * sizeof(float) * impulseChannels);
if(!deinterleavedImpulseBuffer) {
pffft_aligned_free(impulseBuffer);
return nil;
}
for(size_t i = 0; i < impulseChannels; ++i) {
cblas_scopy(sampleCount, impulseBuffer + i, impulseChannels, deinterleavedImpulseBuffer + i * fftSize, 1);
vDSP_vclr(deinterleavedImpulseBuffer + i * fftSize + sampleCount, 1, fftSize - sampleCount);
}
pffft_aligned_free(impulseBuffer);
paddedBufferSize = fftSize;
fftSetup = pffft_new_setup((int)fftSize, PFFFT_REAL);
if(!fftSetup) {
pffft_aligned_free(deinterleavedImpulseBuffer);
return nil;
}
workBuffer = (float *)pffft_aligned_malloc(sizeof(float) * fftSize);
if(!workBuffer) {
pffft_aligned_free(deinterleavedImpulseBuffer);
return nil;
}
paddedSignal = (float *)pffft_aligned_malloc(sizeof(float) * paddedBufferSize);
if(!paddedSignal) {
pffft_aligned_free(deinterleavedImpulseBuffer);
return nil;
}
impulse_responses = (float **)calloc(sizeof(float *), channels * 2);
if(!impulse_responses) {
pffft_aligned_free(deinterleavedImpulseBuffer);
return nil;
}
for(size_t i = 0; i < channels; ++i) {
impulse_responses[i * 2 + 0] = (float *)pffft_aligned_malloc(sizeof(float) * fftSize * 2);
impulse_responses[i * 2 + 1] = (float *)pffft_aligned_malloc(sizeof(float) * fftSize * 2);
if(!impulse_responses[i * 2 + 0] || !impulse_responses[i * 2 + 1]) {
pffft_aligned_free(deinterleavedImpulseBuffer);
return nil;
}
uint32_t channelFlag = [AudioChunk extractChannelFlag:(uint32_t)i fromConfig:config];
uint32_t channelIndex = [AudioChunk findChannelIndex:channelFlag];
int leftInChannel = speaker_not_present;
int rightInChannel = speaker_not_present;
if(impulseChannels == 7) {
if(channelIndex <= max_speaker_index) {
leftInChannel = speakers_to_hesuvi_7[channelIndex][0];
rightInChannel = speakers_to_hesuvi_7[channelIndex][1];
}
} else {
if(channelIndex <= max_speaker_index) {
leftInChannel = speakers_to_hesuvi_14[channelIndex][0];
rightInChannel = speakers_to_hesuvi_14[channelIndex][1];
}
}
if(leftInChannel == speaker_is_back_center || rightInChannel == speaker_is_back_center) {
if(impulseChannels == 7) {
cblas_scopy((int)fftSize, deinterleavedImpulseBuffer + 4 * fftSize, 1, impulse_responses[i * 2 + 0], 1);
vDSP_vadd(impulse_responses[i * 2 + 0], 1, deinterleavedImpulseBuffer + 5 * fftSize, 1, impulse_responses[i * 2 + 0], 1, fftSize);
cblas_scopy((int)fftSize, impulse_responses[i * 2 + 0], 1, impulse_responses[i * 2 + 1], 1);
} else {
cblas_scopy((int)fftSize, deinterleavedImpulseBuffer + 4 * fftSize, 1, impulse_responses[i * 2 + 0], 1);
vDSP_vadd(impulse_responses[i * 2 + 0], 1, deinterleavedImpulseBuffer + 12 * fftSize, 1, impulse_responses[i * 2 + 0], 1, fftSize);
cblas_scopy((int)fftSize, deinterleavedImpulseBuffer + 5 * fftSize, 1, impulse_responses[i * 2 + 1], 1);
vDSP_vadd(impulse_responses[i * 2 + 1], 1, deinterleavedImpulseBuffer + 11 * fftSize, 1, impulse_responses[i * 2 + 1], 1, fftSize);
}
} else if(leftInChannel == speaker_not_present || rightInChannel == speaker_not_present) {
vDSP_vclr(impulse_responses[i * 2 + 0], 1, fftSize);
vDSP_vclr(impulse_responses[i * 2 + 1], 1, fftSize);
} else {
cblas_scopy((int)fftSize, deinterleavedImpulseBuffer + leftInChannel * fftSize, 1, impulse_responses[i * 2 + 0], 1);
cblas_scopy((int)fftSize, deinterleavedImpulseBuffer + rightInChannel * fftSize, 1, impulse_responses[i * 2 + 1], 1);
}
pffft_transform(fftSetup, impulse_responses[i * 2 + 0], impulse_responses[i * 2 + 0], workBuffer, PFFFT_FORWARD);
pffft_transform(fftSetup, impulse_responses[i * 2 + 1], impulse_responses[i * 2 + 1], workBuffer, PFFFT_FORWARD);
}
pffft_aligned_free(deinterleavedImpulseBuffer);
left_result = (float *)pffft_aligned_malloc(sizeof(float) * fftSize);
right_result = (float *)pffft_aligned_malloc(sizeof(float) * fftSize);
if(!left_result || !right_result)
return nil;
prevInputs = (float **)calloc(sizeof(float *), channels);
if(!prevInputs) {
return nil;
}
for(size_t i = 0; i < channels; ++i) {
prevInputs[i] = (float *)pffft_aligned_malloc(sizeof(float) * fftSize);
if(!prevInputs[i]) {
return nil;
}
vDSP_vclr(prevInputs[i], 1, fftSize);
}
}
return self;
}
- (void)dealloc {
if(fftSetup) pffft_destroy_setup(fftSetup);
pffft_aligned_free(workBuffer);
pffft_aligned_free(paddedSignal);
if(impulse_responses) {
for(size_t i = 0; i < channelCount * 2; ++i) {
pffft_aligned_free(impulse_responses[i]);
}
free(impulse_responses);
}
if(prevInputs) {
for(size_t i = 0; i < channelCount; ++i) {
pffft_aligned_free(prevInputs[i]);
}
free(prevInputs);
}
pffft_aligned_free(left_result);
pffft_aligned_free(right_result);
}
- (void)process:(const float *)inBuffer sampleCount:(size_t)count toBuffer:(float *)outBuffer {
const float scale = 1.0 / ((float)fftSize);
while(count > 0) {
const size_t countToDo = (count > bufferSize) ? bufferSize : count;
const size_t outOffset = fftSize - countToDo;
vDSP_vclr(left_result, 1, fftSize);
vDSP_vclr(right_result, 1, fftSize);
for(size_t i = 0; i < channelCount; ++i) {
cblas_scopy((int)outOffset, prevInputs[i] + countToDo, 1, paddedSignal, 1);
cblas_scopy((int)countToDo, inBuffer + i, (int)channelCount, paddedSignal + outOffset, 1);
cblas_scopy((int)fftSize, paddedSignal, 1, prevInputs[i], 1);
pffft_transform(fftSetup, paddedSignal, paddedSignal, workBuffer, PFFFT_FORWARD);
pffft_zconvolve_accumulate(fftSetup, paddedSignal, impulse_responses[i * 2 + 0], left_result, 1.0);
pffft_zconvolve_accumulate(fftSetup, paddedSignal, impulse_responses[i * 2 + 1], right_result, 1.0);
}
pffft_transform(fftSetup, left_result, left_result, workBuffer, PFFFT_BACKWARD);
pffft_transform(fftSetup, right_result, right_result, workBuffer, PFFFT_BACKWARD);
vDSP_vsmul(left_result + outOffset, 1, &scale, left_result + outOffset, 1, countToDo);
vDSP_vsmul(right_result + outOffset, 1, &scale, right_result + outOffset, 1, countToDo);
cblas_scopy((int)countToDo, left_result + outOffset, 1, outBuffer + 0, 2);
cblas_scopy((int)countToDo, right_result + outOffset, 1, outBuffer + 1, 2);
inBuffer += countToDo * channelCount;
outBuffer += countToDo * 2;
count -= countToDo;
}
}
- (void)reset {
for(size_t i = 0; i < channelCount; ++i) {
vDSP_vclr(prevInputs[i], 1, fftSize);
}
}
@end

2
Audio/Chain/Node.m
View File

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

5
Audio/Chain/OutputNode.h
View File

@ -13,7 +13,7 @@
#import <CoreAudio/AudioHardware.h>
#import "Node.h"
#import "OutputCoreAudio.h"
#import "OutputAVFoundation.h"
@interface OutputNode : Node {
AudioStreamBasicDescription format;
@ -21,7 +21,7 @@
double amountPlayed;
double amountPlayedInterval;
OutputCoreAudio *output;
OutputAVFoundation *output;
BOOL paused;
BOOL started;
@ -39,6 +39,7 @@
- (void)resetAmountPlayed;
- (void)resetAmountPlayedInterval;
- (BOOL)selectNextBuffer;
- (void)endOfInputPlayed;
- (BOOL)chainQueueHasTracks;

10
Audio/Chain/OutputNode.m
View File

@ -9,7 +9,7 @@
#import "OutputNode.h"
#import "AudioPlayer.h"
#import "BufferChain.h"
#import "OutputCoreAudio.h"
#import "OutputAVFoundation.h"
#import "Logging.h"
@ -23,7 +23,7 @@
started = NO;
intervalReported = NO;
output = [[OutputCoreAudio alloc] initWithController:self];
output = [[OutputAVFoundation alloc] initWithController:self];
[output setup];
}
@ -68,6 +68,10 @@
intervalReported = NO;
}
- (BOOL)selectNextBuffer {
return [controller selectNextBuffer];
}
- (void)endOfInputPlayed {
if(!intervalReported) {
intervalReported = YES;
@ -137,8 +141,6 @@
format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket;
channelConfig = config;
[converter setOutputFormat:format
outputConfig:channelConfig];
[converter inputFormatDidChange:[bufferChain inputFormat] inputConfig:[bufferChain inputConfig]];
}
}

226
Audio/CogAudio.xcodeproj/project.pbxproj
View File

@ -25,8 +25,8 @@
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 /* OutputCoreAudio.h in Headers */ = {isa = PBXBuildFile; fileRef = 17D21C9C0B8BE4BA00D1EBDE /* OutputCoreAudio.h */; settings = {ATTRIBUTES = (Public, ); }; };
17D21CC60B8BE4BA00D1EBDE /* OutputCoreAudio.m in Sources */ = {isa = PBXBuildFile; fileRef = 17D21C9D0B8BE4BA00D1EBDE /* OutputCoreAudio.m */; };
17D21CC50B8BE4BA00D1EBDE /* OutputAVFoundation.h in Headers */ = {isa = PBXBuildFile; fileRef = 17D21C9C0B8BE4BA00D1EBDE /* OutputAVFoundation.h */; settings = {ATTRIBUTES = (Public, ); }; };
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8E8D3D2D0CBAEE6E00135C1B /* AudioContainer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = AudioContainer.h; sourceTree = "<group>"; };
@ -258,6 +189,7 @@
8328995A27CB51C900D7F028 /* Security.framework in Frameworks */,
83725A9127AA16D50003F694 /* AVFoundation.framework in Frameworks */,
8DC2EF570486A6940098B216 /* Cocoa.framework in Frameworks */,
8350416D28646149006B32CC /* CoreMedia.framework in Frameworks */,
83725A9027AA16C90003F694 /* Accelerate.framework in Frameworks */,
17D21DAD0B8BE76800D1EBDE /* AudioToolbox.framework in Frameworks */,
17D21DAE0B8BE76800D1EBDE /* AudioUnit.framework in Frameworks */,
@ -370,22 +302,20 @@
children = (
834FD4EA27AF8F380063BC83 /* AudioChunk.h */,
834FD4EC27AF91220063BC83 /* AudioChunk.m */,
834FD4EE27AF93680063BC83 /* ChunkList.h */,
834FD4EF27AF93680063BC83 /* ChunkList.m */,
834FD4F227AFA2150063BC83 /* Downmix.h */,
834FD4F327AFA2150063BC83 /* Downmix.m */,
17D21C760B8BE4BA00D1EBDE /* BufferChain.h */,
17D21C770B8BE4BA00D1EBDE /* BufferChain.m */,
834FD4EE27AF93680063BC83 /* ChunkList.h */,
834FD4EF27AF93680063BC83 /* ChunkList.m */,
8EC1225D0B993BD500C5B3AD /* ConverterNode.h */,
8EC1225E0B993BD500C5B3AD /* ConverterNode.mm */,
83504163286447DA006B32CC /* Downmix.h */,
83504164286447DA006B32CC /* Downmix.m */,
17D21C7A0B8BE4BA00D1EBDE /* InputNode.h */,
17D21C7B0B8BE4BA00D1EBDE /* InputNode.m */,
17D21C7C0B8BE4BA00D1EBDE /* Node.h */,
17D21C7D0B8BE4BA00D1EBDE /* Node.m */,
17D21C7E0B8BE4BA00D1EBDE /* OutputNode.h */,
17D21C7F0B8BE4BA00D1EBDE /* OutputNode.m */,
835EDD7C279FE307001EDCCE /* HeadphoneFilter.h */,
835EDD7A279FE23A001EDCCE /* HeadphoneFilter.mm */,
);
path = Chain;
sourceTree = "<group>";
@ -393,8 +323,8 @@
17D21C9B0B8BE4BA00D1EBDE /* Output */ = {
isa = PBXGroup;
children = (
17D21C9C0B8BE4BA00D1EBDE /* OutputCoreAudio.h */,
17D21C9D0B8BE4BA00D1EBDE /* OutputCoreAudio.m */,
17D21C9C0B8BE4BA00D1EBDE /* OutputAVFoundation.h */,
17D21C9D0B8BE4BA00D1EBDE /* OutputAVFoundation.m */,
);
path = Output;
sourceTree = "<group>";
@ -402,10 +332,8 @@
17D21CD80B8BE5B400D1EBDE /* ThirdParty */ = {
isa = PBXGroup;
children = (
83F18ADE27D1E8EF00385946 /* r8brain-free-src */,
8377C64A27B8C51500E8BC0F /* deadbeef */,
835C88AE279811A500E28EAE /* hdcd */,
835C88A22797D4D400E28EAE /* lvqcl */,
17D21DC40B8BE79700D1EBDE /* CoreAudioUtils */,
);
path = ThirdParty;
@ -449,26 +377,6 @@
name = "Other Sources";
sourceTree = "<group>";
};
835C88A22797D4D400E28EAE /* lvqcl */ = {
isa = PBXGroup;
children = (
835C88A32797D4D400E28EAE /* License */,
835C88A62797D4D400E28EAE /* lpc.c */,
835C88A72797D4D400E28EAE /* lpc.h */,
835C88AC2797DA5800E28EAE /* util.h */,
);
path = lvqcl;
sourceTree = "<group>";
};
835C88A32797D4D400E28EAE /* License */ = {
isa = PBXGroup;
children = (
835C88A42797D4D400E28EAE /* LICENSE.LGPL */,
835C88A52797D4D400E28EAE /* License.txt */,
);
path = License;
sourceTree = "<group>";
};
835C88AE279811A500E28EAE /* hdcd */ = {
isa = PBXGroup;
children = (
@ -481,6 +389,7 @@
83725A8F27AA16C90003F694 /* Frameworks */ = {
isa = PBXGroup;
children = (
8350416C28646149006B32CC /* CoreMedia.framework */,
8328995927CB51C900D7F028 /* Security.framework */,
);
name = Frameworks;
@ -504,58 +413,6 @@
path = Visualization;
sourceTree = "<group>";
};
83B69B6A2845DF6500D2435A /* pffft_double */ = {
isa = PBXGroup;
children = (
83B69B6B2845DF6500D2435A /* pffft_double.h */,
83B69B6C2845DF6500D2435A /* simd */,
83B69B732845DF6500D2435A /* pffft_priv_impl.h */,
83B69B742845DF6500D2435A /* pffft_double.c */,
);
path = pffft_double;
sourceTree = "<group>";
};
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isa = PBXGroup;
children = (
83B69B6D2845DF6500D2435A /* pf_neon_double_from_avx.h */,
83B69B6E2845DF6500D2435A /* pf_double.h */,
83B69B6F2845DF6500D2435A /* pf_neon_double.h */,
83B69B702845DF6500D2435A /* pf_sse2_double.h */,
83B69B712845DF6500D2435A /* pf_avx_double.h */,
83B69B722845DF6500D2435A /* pf_scalar_double.h */,
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path = simd;
sourceTree = "<group>";
};
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isa = PBXGroup;
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8363BABD284E428F00E5C9DD /* pffft.cpp */,
83B69B6A2845DF6500D2435A /* pffft_double */,
83F18ADF27D1E8EF00385946 /* CDSPHBDownsampler.h */,
83F18AF827D1E8EF00385946 /* CDSPSincFilterGen.h */,
83F18AF927D1E8EF00385946 /* r8butil.h */,
83F18AFB27D1E8EF00385946 /* LICENSE */,
83F18AFC27D1E8EF00385946 /* r8bbase.h */,
83F18AFD27D1E8EF00385946 /* CDSPFIRFilter.h */,
83F18B0827D1E8EF00385946 /* CDSPProcessor.h */,
83F18B0927D1E8EF00385946 /* README.md */,
83F18B0A27D1E8EF00385946 /* fft4g.h */,
83F18B0B27D1E8EF00385946 /* CDSPRealFFT.h */,
83F18B0C27D1E8EF00385946 /* CDSPFracInterpolator.h */,
83F18B1727D1E8EF00385946 /* CDSPBlockConvolver.h */,
83F18B1827D1E8EF00385946 /* CDSPHBUpsampler.inc */,
83F18B1927D1E8EF00385946 /* r8bconf.h */,
83F18B1A27D1E8EF00385946 /* r8bbase.cpp */,
83F18B1B27D1E8EF00385946 /* pffft.h */,
83F18B1C27D1E8EF00385946 /* CDSPResampler.h */,
83F18B1D27D1E8EF00385946 /* CDSPHBUpsampler.h */,
83F18B5527D1F5E900385946 /* r8bstate.h */,
);
path = "r8brain-free-src";
sourceTree = "<group>";
};
/* End PBXGroup section */
/* Begin PBXHeadersBuildPhase section */
@ -563,64 +420,38 @@
isa = PBXHeadersBuildPhase;
buildActionMask = 2147483647;
files = (
83B69B7C2845DF6500D2435A /* pffft_priv_impl.h in Headers */,
17D21CA10B8BE4BA00D1EBDE /* BufferChain.h in Headers */,
17D21CA50B8BE4BA00D1EBDE /* InputNode.h in Headers */,
17D21CA70B8BE4BA00D1EBDE /* Node.h in Headers */,
83B69B7B2845DF6500D2435A /* pf_scalar_double.h in Headers */,
83F18B3427D1E8EF00385946 /* r8butil.h in Headers */,
8399CF2C27B5D1D5008751F1 /* NSDictionary+Merge.h in Headers */,
17D21CA90B8BE4BA00D1EBDE /* OutputNode.h in Headers */,
83B69B792845DF6500D2435A /* pf_sse2_double.h in Headers */,
83F18B4527D1E8EF00385946 /* CDSPRealFFT.h in Headers */,
83F18B3827D1E8EF00385946 /* CDSPFIRFilter.h in Headers */,
8328995427CB511000D7F028 /* RedundantPlaylistDataStore.h in Headers */,
17D21CC50B8BE4BA00D1EBDE /* OutputCoreAudio.h in Headers */,
834FD4F427AFA2150063BC83 /* Downmix.h in Headers */,
17D21CC50B8BE4BA00D1EBDE /* OutputAVFoundation.h in Headers */,
83504165286447DA006B32CC /* Downmix.h in Headers */,
17D21CC70B8BE4BA00D1EBDE /* Status.h in Headers */,
835C88AB2797D4D400E28EAE /* lpc.h in Headers */,
17D21CF30B8BE5EF00D1EBDE /* Semaphore.h in Headers */,
17D21DC70B8BE79700D1EBDE /* CoreAudioUtils.h in Headers */,
83F18B3327D1E8EF00385946 /* CDSPSincFilterGen.h in Headers */,
17D21EBD0B8BF44000D1EBDE /* AudioPlayer.h in Headers */,
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17A2D3C50B8D1D37000778C4 /* AudioDecoder.h in Headers */,
8347C7412796C58800FA8A7D /* NSFileHandle+CreateFile.h in Headers */,
17C940230B900909008627D6 /* AudioMetadataReader.h in Headers */,
839065F32853338700636FBB /* dsd2float.h in Headers */,
17B619300B909BC300BC003F /* AudioPropertiesReader.h in Headers */,
83F18B5427D1E8F000385946 /* CDSPHBUpsampler.h in Headers */,
835EDD7D279FE307001EDCCE /* HeadphoneFilter.h in Headers */,
839366671815923C006DD712 /* CogPluginMulti.h in Headers */,
83F18B5327D1E8F000385946 /* CDSPResampler.h in Headers */,
17ADB13C0B97926D00257CA2 /* AudioSource.h in Headers */,
835C88B1279811A500E28EAE /* hdcd_decode2.h in Headers */,
8EC1225F0B993BD500C5B3AD /* ConverterNode.h in Headers */,
8384912718080FF100E7332D /* Logging.h in Headers */,
8377C64E27B8C54400E8BC0F /* fft.h in Headers */,
83F18B5027D1E8F000385946 /* r8bconf.h in Headers */,
835FAC5E27BCA14D00BA8562 /* BadSampleCleaner.h in Headers */,
8E8D3D2F0CBAEE6E00135C1B /* AudioContainer.h in Headers */,
83F18B3727D1E8EF00385946 /* r8bbase.h in Headers */,
B0575F2D0D687A0800411D77 /* Helper.h in Headers */,
835C88AD2797DA5800E28EAE /* util.h in Headers */,
07DB5F3E0ED353A900C2E3EF /* AudioMetadataWriter.h in Headers */,
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@ -685,10 +516,6 @@
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buildActionMask = 2147483647;
files = (
83F18B3627D1E8EF00385946 /* LICENSE in Resources */,
835C88A92797D4D400E28EAE /* License.txt in Resources */,
835C88A82797D4D400E28EAE /* LICENSE.LGPL in Resources */,
83F18B4327D1E8EF00385946 /* README.md in Resources */,
);
runOnlyForDeploymentPostprocessing = 0;
};
@ -699,28 +526,23 @@
isa = PBXSourcesBuildPhase;
buildActionMask = 2147483647;
files = (
83D40B572852CB3B003BB85C /* pffft_double.c in Sources */,
835EDD7B279FE23A001EDCCE /* HeadphoneFilter.mm in Sources */,
17D21CA20B8BE4BA00D1EBDE /* BufferChain.m in Sources */,
17D21CA60B8BE4BA00D1EBDE /* InputNode.m in Sources */,
83504166286447DA006B32CC /* Downmix.m in Sources */,
8399CF2D27B5D1D5008751F1 /* NSDictionary+Merge.m in Sources */,
17D21CA80B8BE4BA00D1EBDE /* Node.m in Sources */,
17D21CAA0B8BE4BA00D1EBDE /* OutputNode.m in Sources */,
8377C65327B8CAD100E8BC0F /* VisualizationController.m in Sources */,
834FD4F527AFA2150063BC83 /* Downmix.m in Sources */,
17D21CC60B8BE4BA00D1EBDE /* OutputCoreAudio.m in Sources */,
17D21CC60B8BE4BA00D1EBDE /* OutputAVFoundation.m in Sources */,
835C88B2279811A500E28EAE /* hdcd_decode2.c in Sources */,
835FAC5F27BCA14D00BA8562 /* BadSampleCleaner.m in Sources */,
834FD4ED27AF91220063BC83 /* AudioChunk.m in Sources */,
83F18B5127D1E8F000385946 /* r8bbase.cpp in Sources */,
17D21CF40B8BE5EF00D1EBDE /* Semaphore.m in Sources */,
8347C7422796C58800FA8A7D /* NSFileHandle+CreateFile.m in Sources */,
17D21DC80B8BE79700D1EBDE /* CoreAudioUtils.m in Sources */,
8328995327CB511000D7F028 /* RedundantPlaylistDataStore.m in Sources */,
8377C64C27B8C51500E8BC0F /* fft_accelerate.c in Sources */,
8363BABE284E428F00E5C9DD /* pffft.cpp in Sources */,
839366681815923C006DD712 /* CogPluginMulti.m in Sources */,
835C88AA2797D4D400E28EAE /* lpc.c in Sources */,
17D21EBE0B8BF44000D1EBDE /* AudioPlayer.m in Sources */,
17F94DD60B8D0F7000A34E87 /* PluginController.mm in Sources */,
17A2D3C60B8D1D37000778C4 /* AudioDecoder.m in Sources */,
@ -756,8 +578,6 @@
GCC_PREFIX_HEADER = CogAudio_Prefix.pch;
GCC_PREPROCESSOR_DEFINITIONS = (
"DEBUG=1",
"R8B_EXTFFT=1",
"R8B_PFFFT_DOUBLE=1",
);
INFOPLIST_FILE = Info.plist;
INSTALL_PATH = "@executable_path/../Frameworks";
@ -786,8 +606,6 @@
GCC_PRECOMPILE_PREFIX_HEADER = YES;
GCC_PREFIX_HEADER = CogAudio_Prefix.pch;
GCC_PREPROCESSOR_DEFINITIONS = (
"R8B_EXTFFT=1",
"R8B_PFFFT_DOUBLE=1",
);
INFOPLIST_FILE = Info.plist;
INSTALL_PATH = "@executable_path/../Frameworks";

43
Audio/Output/OutputCoreAudio.h → Audio/Output/OutputAVFoundation.h
View File

@ -1,9 +1,9 @@
//
// OutputCoreAudio.h
// OutputAVFoundation.h
// Cog
//
// Created by Vincent Spader on 8/2/05.
// Copyright 2005 Vincent Spader. All rights reserved.
// Created by Christopher Snowhill on 6/23/22.
// Copyright 2022 Christopher Snowhill. All rights reserved.
//
#import <AssertMacros.h>
@ -26,8 +26,6 @@ using std::atomic_long;
#import "VisualizationController.h"
#import "Semaphore.h"
//#define OUTPUT_LOG
#ifdef OUTPUT_LOG
#import <stdio.h>
@ -35,28 +33,25 @@ using std::atomic_long;
@class OutputNode;
@interface OutputCoreAudio : NSObject {
@interface OutputAVFoundation : NSObject {
OutputNode *outputController;
Semaphore *writeSemaphore;
Semaphore *readSemaphore;
BOOL stopInvoked;
BOOL running;
BOOL stopping;
BOOL stopped;
BOOL started;
BOOL paused;
BOOL stopNext;
BOOL restarted;
BOOL eqEnabled;
BOOL eqInitialized;
BOOL dontRemix;
BOOL streamFormatStarted;
atomic_long bytesRendered;
atomic_long bytesHdcdSustained;
double secondsHdcdSustained;
BOOL defaultdevicelistenerapplied;
BOOL currentdevicelistenerapplied;
@ -67,10 +62,7 @@ using std::atomic_long;
float volume;
float eqPreamp;
AVAudioFormat *_deviceFormat;
AudioDeviceID outputDeviceID;
AudioStreamBasicDescription deviceFormat; // info about the default device
AudioStreamBasicDescription streamFormat; // stream format last seen in render callback
AudioStreamBasicDescription visFormat; // Mono format for vis
@ -78,18 +70,31 @@ using std::atomic_long;
uint32_t deviceChannelConfig;
uint32_t streamChannelConfig;
AUAudioUnit *_au;
AVSampleBufferAudioRenderer *audioRenderer;
AVSampleBufferRenderSynchronizer *renderSynchronizer;
CMAudioFormatDescriptionRef audioFormatDescription;
AudioChannelLayoutTag streamTag;
id currentPtsObserver;
NSLock *currentPtsLock;
CMTime currentPts, lastPts;
double secondsLatency;
CMTime outputPts, trackPts, lastCheckpointPts;
AudioTimeStamp timeStamp;
size_t _bufferSize;
AudioUnit _eq;
DownmixProcessor *downmixer;
DownmixProcessor *downmixerForVis;
VisualizationController *visController;
os_workgroup_t wg;
os_workgroup_join_token_s wgToken;
float inputBuffer[2048 * 32]; // 2048 samples times maximum supported channel count
float eqBuffer[2048 * 32];
#ifdef OUTPUT_LOG
FILE *_logFile;

935
Audio/Output/OutputAVFoundation.m Normal file
View File

@ -0,0 +1,935 @@
//
// OutputAVFoundation.m
// Cog
//
// Created by Christopher Snowhill on 6/23/22.
// Copyright 2022 Christopher Snowhill. All rights reserved.
//
#import "OutputAVFoundation.h"
#import "OutputNode.h"
#ifdef _DEBUG
#import "BadSampleCleaner.h"
#endif
#import "Logging.h"
#import <Accelerate/Accelerate.h>
extern void scale_by_volume(float *buffer, size_t count, float volume);
static NSString *CogPlaybackDidBeginNotficiation = @"CogPlaybackDidBeginNotficiation";
@implementation OutputAVFoundation
static void *kOutputAVFoundationContext = &kOutputAVFoundationContext;
static void fillBuffers(AudioBufferList *ioData, const float *inbuffer, size_t count, size_t offset) {
const size_t channels = ioData->mNumberBuffers;
for(int i = 0; i < channels; ++i) {
const size_t maxCount = (ioData->mBuffers[i].mDataByteSize / sizeof(float)) - offset;
float *output = ((float *)ioData->mBuffers[i].mData) + offset;
const float *input = inbuffer + i;
cblas_scopy((int)((count > maxCount) ? maxCount : count), input, (int)channels, output, 1);
ioData->mBuffers[i].mNumberChannels = 1;
}
}
static void clearBuffers(AudioBufferList *ioData, size_t count, size_t offset) {
for(int i = 0; i < ioData->mNumberBuffers; ++i) {
memset(ioData->mBuffers[i].mData + offset * sizeof(float), 0, count * sizeof(float));
ioData->mBuffers[i].mNumberChannels = 1;
}
}
static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) {
if(inNumberFrames > 1024 || !inRefCon) {
clearBuffers(ioData, inNumberFrames, 0);
return 0;
}
OutputAVFoundation *_self = (__bridge OutputAVFoundation *)inRefCon;
fillBuffers(ioData, &_self->inputBuffer[0], inNumberFrames, 0);
return 0;
}
- (int)renderInput {
int amountToRead, amountRead = 0;
amountToRead = 1024;
int visTabulated = 0;
float visAudio[amountToRead]; // Chunk size
if(stopping == YES || [outputController shouldContinue] == NO) {
// Chain is dead, fill out the serial number pointer forever with silence
stopping = YES;
return 0;
}
AudioChunk *chunk = [outputController readChunk:1024];
int frameCount = (int)[chunk frameCount];
AudioStreamBasicDescription format = [chunk format];
uint32_t config = [chunk channelConfig];
double chunkDuration = 0;
if(frameCount) {
// 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
BOOL formatClipped = NO;
BOOL isSurround = format.mChannelsPerFrame > 2;
const double maxSampleRate = isSurround ? 352800.0 : 192000.0;
if(format.mSampleRate > maxSampleRate) {
format.mSampleRate = maxSampleRate;
formatClipped = YES;
}
if(!streamFormatStarted || config != streamChannelConfig || memcmp(&streamFormat, &format, sizeof(format)) != 0) {
if(formatClipped) {
ALog(@"Sample rate clipped to no more than %f Hz!", maxSampleRate);
}
streamFormat = format;
streamChannelConfig = config;
streamFormatStarted = YES;
[self updateStreamFormat];
downmixerForVis = [[DownmixProcessor alloc] initWithInputFormat:format inputConfig:config andOutputFormat:visFormat outputConfig:AudioConfigMono];
}
chunkDuration = [chunk duration];
NSData *samples = [chunk removeSamples:frameCount];
#ifdef _DEBUG
[BadSampleCleaner cleanSamples:(float *)[samples bytes]
amount:frameCount * format.mChannelsPerFrame
location:@"pre downmix"];
#endif
[downmixerForVis process:[samples bytes]
frameCount:frameCount
output:visAudio];
visTabulated += frameCount;
cblas_scopy((int)(frameCount * streamFormat.mChannelsPerFrame), (const float *)[samples bytes], 1, &inputBuffer[0], 1);
amountRead = frameCount;
} else {
return 0;
}
float volumeScale = 1.0;
double sustained;
@synchronized(self) {
sustained = secondsHdcdSustained;
}
if(sustained > 0) {
if(sustained < amountRead) {
@synchronized(self) {
secondsHdcdSustained = 0;
}
} else {
@synchronized(self) {
secondsHdcdSustained -= chunkDuration;
}
volumeScale = 0.5;
}
}
if(eqEnabled) {
volumeScale *= eqPreamp;
}
scale_by_volume(&inputBuffer[0], amountRead * streamFormat.mChannelsPerFrame, volumeScale);
[visController postSampleRate:streamFormat.mSampleRate];
[visController postVisPCM:visAudio amount:visTabulated];
return amountRead;
}
- (id)initWithController:(OutputNode *)c {
self = [super init];
if(self) {
outputController = c;
volume = 1.0;
outputDeviceID = -1;
secondsHdcdSustained = 0;
#ifdef OUTPUT_LOG
_logFile = fopen("/tmp/CogAudioLog.raw", "wb");
#endif
}
return self;
}
static OSStatus
default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const AudioObjectPropertyAddress *inAddresses, void *inUserData) {
OutputAVFoundation *this = (__bridge OutputAVFoundation *)inUserData;
return [this setOutputDeviceByID:-1];
}
static OSStatus
current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, const AudioObjectPropertyAddress *inAddresses, void *inUserData) {
OutputAVFoundation *this = (__bridge OutputAVFoundation *)inUserData;
for(UInt32 i = 0; i < inNumberAddresses; ++i) {
switch(inAddresses[i].mSelector) {
case kAudioDevicePropertyDeviceIsAlive:
return [this setOutputDeviceByID:-1];
case kAudioDevicePropertyNominalSampleRate:
case kAudioDevicePropertyStreamFormat:
this->outputdevicechanged = YES;
return noErr;
}
}
return noErr;
}
- (void)observeValueForKeyPath:(NSString *)keyPath ofObject:(id)object change:(NSDictionary *)change context:(void *)context {
if(context != kOutputAVFoundationContext) {
[super observeValueForKeyPath:keyPath ofObject:object change:change context:context];
return;
}
if([keyPath isEqualToString:@"values.outputDevice"]) {
NSDictionary *device = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] objectForKey:@"outputDevice"];
[self setOutputDeviceWithDeviceDict:device];
} else if([keyPath isEqualToString:@"values.GraphicEQenable"]) {
BOOL enabled = [[[[NSUserDefaultsController sharedUserDefaultsController] defaults] objectForKey:@"GraphicEQenable"] boolValue];
[self setEqualizerEnabled:enabled];
} else if([keyPath isEqualToString:@"values.eqPreamp"]) {
float preamp = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] floatForKey:@"eqPreamp"];
eqPreamp = pow(10.0, preamp / 20.0);
} else if([keyPath isEqualToString:@"values.dontRemix"]) {
dontRemix = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] boolForKey:@"dontRemix"];
}
}
- (BOOL)signalEndOfStream:(double)latency {
BOOL ret = [outputController selectNextBuffer];
dispatch_after(dispatch_time(DISPATCH_TIME_NOW, (int64_t)(NSEC_PER_SEC * latency)), dispatch_get_main_queue(), ^{
[self->outputController endOfInputPlayed];
[self->outputController resetAmountPlayed];
self->lastCheckpointPts = self->trackPts;
self->trackPts = kCMTimeZero;
});
return ret;
}
- (void)threadEntry:(id)arg {
running = YES;
started = NO;
secondsLatency = 1.0;
while(!stopping) {
if([outputController shouldReset]) {
[outputController setShouldReset:NO];
[self removeSynchronizerBlock];
[renderSynchronizer setRate:0];
[audioRenderer stopRequestingMediaData];
[audioRenderer flush];
currentPts = kCMTimeZero;
lastPts = kCMTimeZero;
outputPts = kCMTimeZero;
trackPts = kCMTimeZero;
lastCheckpointPts = kCMTimeZero;
secondsLatency = 1.0;
started = NO;
[self synchronizerBlock];
}
if(stopping)
break;
while(!stopping && [audioRenderer isReadyForMoreMediaData]) {
CMSampleBufferRef bufferRef = [self makeSampleBuffer];
if(bufferRef) {
CMTime chunkDuration = CMSampleBufferGetDuration(bufferRef);
outputPts = CMTimeAdd(outputPts, chunkDuration);
trackPts = CMTimeAdd(trackPts, chunkDuration);
[audioRenderer enqueueSampleBuffer:bufferRef];
} else {
if([outputController endOfStream] == YES && ![self signalEndOfStream:secondsLatency]) {
// Wait for output to catch up
CMTime currentTime;
[currentPtsLock lock];
currentTime = currentPts;
[currentPtsLock unlock];
CMTime timeToWait = CMTimeSubtract(outputPts, currentTime);
double secondsToWait = CMTimeGetSeconds(timeToWait);
if(secondsToWait > 0) {
usleep(secondsToWait * 1000000.0);
}
stopping = YES;
}
break;
}
}
if(!paused && !started) {
[self resume];
}
if([outputController shouldContinue] == NO) {
break;
}
usleep(5000);
}
stopped = YES;
[self stop];
}
- (OSStatus)setOutputDeviceByID:(AudioDeviceID)deviceID {
OSStatus err;
BOOL defaultDevice = NO;
AudioObjectPropertyAddress theAddress = {
.mSelector = kAudioHardwarePropertyDefaultOutputDevice,
.mScope = kAudioObjectPropertyScopeGlobal,
.mElement = kAudioObjectPropertyElementMaster
};
if(deviceID == -1) {
defaultDevice = YES;
UInt32 size = sizeof(AudioDeviceID);
err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &theAddress, 0, NULL, &size, &deviceID);
if(err != noErr) {
DLog(@"THERE'S NO DEFAULT OUTPUT DEVICE");
return err;
}
}
if(audioRenderer) {
if(defaultdevicelistenerapplied && !defaultDevice) {
/* Already set above
* theAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice; */
AudioObjectRemovePropertyListener(kAudioObjectSystemObject, &theAddress, default_device_changed, (__bridge void *_Nullable)(self));
defaultdevicelistenerapplied = NO;
}
outputdevicechanged = NO;
if(outputDeviceID != deviceID) {
if(currentdevicelistenerapplied) {
if(devicealivelistenerapplied) {
theAddress.mSelector = kAudioDevicePropertyDeviceIsAlive;
AudioObjectRemovePropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
devicealivelistenerapplied = NO;
}
theAddress.mSelector = kAudioDevicePropertyStreamFormat;
AudioObjectRemovePropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
theAddress.mSelector = kAudioDevicePropertyNominalSampleRate;
AudioObjectRemovePropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
currentdevicelistenerapplied = NO;
}
DLog(@"Device: %i\n", deviceID);
outputDeviceID = deviceID;
CFStringRef deviceUID = NULL;
theAddress.mSelector = kAudioDevicePropertyDeviceUID;
UInt32 size = sizeof(deviceUID);
err = AudioObjectGetPropertyData(outputDeviceID, &theAddress, 0, NULL, &size, &deviceUID);
if(err != noErr) {
DLog(@"Unable to get UID of device");
return err;
}
[audioRenderer setAudioOutputDeviceUniqueID:(__bridge NSString *)deviceUID];
CFRelease(deviceUID);
outputdevicechanged = YES;
}
if(!currentdevicelistenerapplied) {
if(!devicealivelistenerapplied && !defaultDevice) {
theAddress.mSelector = kAudioDevicePropertyDeviceIsAlive;
AudioObjectAddPropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
devicealivelistenerapplied = YES;
}
theAddress.mSelector = kAudioDevicePropertyStreamFormat;
AudioObjectAddPropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
theAddress.mSelector = kAudioDevicePropertyNominalSampleRate;
AudioObjectAddPropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
currentdevicelistenerapplied = YES;
}
if(!defaultdevicelistenerapplied && defaultDevice) {
theAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
AudioObjectAddPropertyListener(kAudioObjectSystemObject, &theAddress, default_device_changed, (__bridge void *_Nullable)(self));
defaultdevicelistenerapplied = YES;
}
} else {
err = noErr;
}
if(err != noErr) {
DLog(@"No output device with ID %d could be found.", deviceID);
return err;
}
return err;
}
- (BOOL)setOutputDeviceWithDeviceDict:(NSDictionary *)deviceDict {
NSNumber *deviceIDNum = [deviceDict objectForKey:@"deviceID"];
AudioDeviceID outputDeviceID = [deviceIDNum unsignedIntValue] ?: -1;
__block OSStatus err = [self setOutputDeviceByID:outputDeviceID];
if(err != noErr) {
// Try matching by name.
NSString *userDeviceName = deviceDict[@"name"];
[self enumerateAudioOutputsUsingBlock:
^(NSString *deviceName, AudioDeviceID deviceID, AudioDeviceID systemDefaultID, BOOL *stop) {
if([deviceName isEqualToString:userDeviceName]) {
err = [self setOutputDeviceByID:deviceID];
#if 0
// Disable. Would cause loop by triggering `-observeValueForKeyPath:ofObject:change:context:` above.
// Update `outputDevice`, in case the ID has changed.
NSDictionary *deviceInfo = @{
@"name": deviceName,
@"deviceID": @(deviceID),
};
[[NSUserDefaults standardUserDefaults] setObject:deviceInfo forKey:@"outputDevice"];
#endif
DLog(@"Found output device: \"%@\" (%d).", deviceName, deviceID);
*stop = YES;
}
}];
}
if(err != noErr) {
ALog(@"No output device could be found, your random error code is %d. Have a nice day!", err);
return NO;
}
return YES;
}
// The following is largely a copy pasta of -awakeFromNib from "OutputsArrayController.m".
// TODO: Share the code. (How to do this across xcodeproj?)
- (void)enumerateAudioOutputsUsingBlock:(void(NS_NOESCAPE ^ _Nonnull)(NSString *deviceName, AudioDeviceID deviceID, AudioDeviceID systemDefaultID, BOOL *stop))block {
UInt32 propsize;
AudioObjectPropertyAddress theAddress = {
.mSelector = kAudioHardwarePropertyDevices,
.mScope = kAudioObjectPropertyScopeGlobal,
.mElement = kAudioObjectPropertyElementMaster
};
__Verify_noErr(AudioObjectGetPropertyDataSize(kAudioObjectSystemObject, &theAddress, 0, NULL, &propsize));
UInt32 nDevices = propsize / (UInt32)sizeof(AudioDeviceID);
AudioDeviceID *devids = malloc(propsize);
__Verify_noErr(AudioObjectGetPropertyData(kAudioObjectSystemObject, &theAddress, 0, NULL, &propsize, devids));
theAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
AudioDeviceID systemDefault;
propsize = sizeof(systemDefault);
__Verify_noErr(AudioObjectGetPropertyData(kAudioObjectSystemObject, &theAddress, 0, NULL, &propsize, &systemDefault));
theAddress.mScope = kAudioDevicePropertyScopeOutput;
for(UInt32 i = 0; i < nDevices; ++i) {
UInt32 isAlive = 0;
propsize = sizeof(isAlive);
theAddress.mSelector = kAudioDevicePropertyDeviceIsAlive;
__Verify_noErr(AudioObjectGetPropertyData(devids[i], &theAddress, 0, NULL, &propsize, &isAlive));
if(!isAlive) continue;
CFStringRef name = NULL;
propsize = sizeof(name);
theAddress.mSelector = kAudioDevicePropertyDeviceNameCFString;
__Verify_noErr(AudioObjectGetPropertyData(devids[i], &theAddress, 0, NULL, &propsize, &name));
propsize = 0;
theAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
__Verify_noErr(AudioObjectGetPropertyDataSize(devids[i], &theAddress, 0, NULL, &propsize));
if(propsize < sizeof(UInt32)) {
if(name) CFRelease(name);
continue;
}
AudioBufferList *bufferList = (AudioBufferList *)malloc(propsize);
__Verify_noErr(AudioObjectGetPropertyData(devids[i], &theAddress, 0, NULL, &propsize, bufferList));
UInt32 bufferCount = bufferList->mNumberBuffers;
free(bufferList);
if(!bufferCount) {
if(name) CFRelease(name);
continue;
}
BOOL stop = NO;
NSString *deviceName = name ? [NSString stringWithString:(__bridge NSString *)name] : [NSString stringWithFormat:@"Unknown device %u", (unsigned int)devids[i]];
block(deviceName,
devids[i],
systemDefault,
&stop);
CFRelease(name);
if(stop) {
break;
}
}
free(devids);
}
- (void)updateStreamFormat {
visFormat = streamFormat;
visFormat.mChannelsPerFrame = 1;
visFormat.mBytesPerFrame = visFormat.mChannelsPerFrame * (visFormat.mBitsPerChannel / 8);
visFormat.mBytesPerPacket = visFormat.mBytesPerFrame * visFormat.mFramesPerPacket;
/* Set the channel layout for the audio queue */
AudioChannelLayoutTag tag = 0;
AudioChannelLayout layout = { 0 };
switch(streamFormat.mChannelsPerFrame) {
case 1:
tag = kAudioChannelLayoutTag_Mono;
deviceChannelConfig = AudioConfigMono;
break;
case 2:
tag = kAudioChannelLayoutTag_Stereo;
deviceChannelConfig = AudioConfigStereo;
break;
case 3:
tag = kAudioChannelLayoutTag_WAVE_3_0;
deviceChannelConfig = AudioConfig3Point0;
break;
case 4:
tag = kAudioChannelLayoutTag_WAVE_4_0_A;
deviceChannelConfig = AudioConfig4Point0;
break;
case 5:
tag = kAudioChannelLayoutTag_WAVE_5_0_A;
deviceChannelConfig = AudioConfig5Point0;
break;
case 6:
tag = kAudioChannelLayoutTag_WAVE_5_1_A;
deviceChannelConfig = AudioConfig5Point1;
break;
case 7:
tag = kAudioChannelLayoutTag_WAVE_6_1;
deviceChannelConfig = AudioConfig6Point1;
break;
case 8:
tag = kAudioChannelLayoutTag_WAVE_7_1;
deviceChannelConfig = AudioConfig7Point1;
break;
default:
tag = 0;
break;
}
streamTag = tag;
if(tag) {
layout.mChannelLayoutTag = tag;
}
if(CMAudioFormatDescriptionCreate(kCFAllocatorDefault, &streamFormat, sizeof(layout), &layout, 0, NULL, NULL, &audioFormatDescription) != noErr) {
return;
}
AudioStreamBasicDescription asbd = streamFormat;
asbd.mFormatFlags &= ~kAudioFormatFlagIsPacked;
UInt32 maximumFrames = 1024;
AudioUnitSetProperty(_eq, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &maximumFrames, sizeof(maximumFrames));
AudioUnitSetProperty(_eq, kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 0, &asbd, sizeof(asbd));
AudioUnitSetProperty(_eq, kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output, 0, &asbd, sizeof(asbd));
AudioUnitReset(_eq, kAudioUnitScope_Input, 0);
AudioUnitReset(_eq, kAudioUnitScope_Output, 0);
AudioUnitReset(_eq, kAudioUnitScope_Global, 0);
eqEnabled = [[[[NSUserDefaultsController sharedUserDefaultsController] defaults] objectForKey:@"GraphicEQenable"] boolValue];
}
- (CMSampleBufferRef)makeSampleBuffer {
CMBlockBufferRef blockBuffer = nil;
int samplesRendered = [self makeBlockBuffer:&blockBuffer];
if(!samplesRendered) return nil;
CMSampleBufferRef sampleBuffer = nil;
OSStatus err = CMAudioSampleBufferCreateReadyWithPacketDescriptions(kCFAllocatorDefault, blockBuffer, audioFormatDescription, samplesRendered, outputPts, nil, &sampleBuffer);
if(err != noErr) {
return nil;
}
return sampleBuffer;
}
- (int)makeBlockBuffer:(CMBlockBufferRef *)blockBufferOut {
OSStatus status;
CMBlockBufferRef blockListBuffer = nil;
status = CMBlockBufferCreateEmpty(kCFAllocatorDefault, 0, 0, &blockListBuffer);
if(status != noErr || !blockListBuffer) return 0;
int samplesRendered = [self renderInput];
if(samplesRendered) {
if(eqEnabled) {
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[1024 * i];
ioData->mBuffers[i].mDataByteSize = 1024 * 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[1024 * i], 1, &inputBuffer[i], channels);
}
}
}
CMBlockBufferRef blockBuffer = nil;
size_t dataByteSize = samplesRendered * sizeof(float) * streamFormat.mChannelsPerFrame;
status = CMBlockBufferCreateWithMemoryBlock(kCFAllocatorDefault, nil, dataByteSize, kCFAllocatorDefault, nil, 0, dataByteSize, kCMBlockBufferAssureMemoryNowFlag, &blockBuffer);
if(status != noErr || !blockBuffer) {
return 0;
}
status = CMBlockBufferReplaceDataBytes(&inputBuffer[0], blockBuffer, 0, dataByteSize);
if(status != noErr) {
return 0;
}
status = CMBlockBufferAppendBufferReference(blockListBuffer, blockBuffer, 0, CMBlockBufferGetDataLength(blockBuffer), 0);
if(status != noErr) {
return 0;
}
}
*blockBufferOut = blockListBuffer;
return samplesRendered;
}
- (BOOL)setup {
if(audioRenderer || renderSynchronizer)
[self stop];
@synchronized(self) {
stopInvoked = NO;
running = NO;
stopping = NO;
stopped = NO;
paused = NO;
outputDeviceID = -1;
restarted = NO;
downmixerForVis = nil;
AudioComponentDescription desc;
NSError *err;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_HALOutput;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
audioRenderer = [[AVSampleBufferAudioRenderer alloc] init];
renderSynchronizer = [[AVSampleBufferRenderSynchronizer alloc] init];
if(audioRenderer == nil || renderSynchronizer == nil)
return NO;
if(@available(macOS 12.0, *)) {
[audioRenderer setAllowedAudioSpatializationFormats:AVAudioSpatializationFormatMonoStereoAndMultichannel];
}
// Setup the output device before mucking with settings
NSDictionary *device = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] objectForKey:@"outputDevice"];
if(device) {
BOOL ok = [self setOutputDeviceWithDeviceDict:device];
if(!ok) {
// Ruh roh.
[self setOutputDeviceWithDeviceDict:nil];
[[[NSUserDefaultsController sharedUserDefaultsController] defaults] removeObjectForKey:@"outputDevice"];
}
} else {
[self setOutputDeviceWithDeviceDict:nil];
}
AudioComponent comp = NULL;
desc.componentType = kAudioUnitType_Effect;
desc.componentSubType = kAudioUnitSubType_GraphicEQ;
comp = AudioComponentFindNext(comp, &desc);
if(!comp)
return NO;
OSStatus _err = AudioComponentInstanceNew(comp, &_eq);
if(err)
return NO;
UInt32 value;
UInt32 size = sizeof(value);
value = CHUNK_SIZE;
AudioUnitSetProperty(_eq, kAudioUnitProperty_MaximumFramesPerSlice,
kAudioUnitScope_Global, 0, &value, size);
value = 127;
AudioUnitSetProperty(_eq, kAudioUnitProperty_RenderQuality,
kAudioUnitScope_Global, 0, &value, size);
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProcRefCon = (__bridge void *)self;
callbackStruct.inputProc = eqRenderCallback;
AudioUnitSetProperty(_eq, kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input, 0, &callbackStruct, sizeof(callbackStruct));
AudioUnitReset(_eq, kAudioUnitScope_Input, 0);
AudioUnitReset(_eq, kAudioUnitScope_Output, 0);
AudioUnitReset(_eq, kAudioUnitScope_Global, 0);
_err = AudioUnitInitialize(_eq);
if(_err)
return NO;
eqInitialized = YES;
[self setEqualizerEnabled:[[[[NSUserDefaultsController sharedUserDefaultsController] defaults] objectForKey:@"GraphicEQenable"] boolValue]];
[outputController beginEqualizer:_eq];
visController = [VisualizationController sharedController];
[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.outputDevice" options:0 context:kOutputAVFoundationContext];
[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.GraphicEQenable" options:0 context:kOutputAVFoundationContext];
[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.eqPreamp" options:(NSKeyValueObservingOptionInitial | NSKeyValueObservingOptionNew) context:kOutputAVFoundationContext];
[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.dontRemix" options:(NSKeyValueObservingOptionInitial | NSKeyValueObservingOptionNew) context:kOutputAVFoundationContext];
observersapplied = YES;
[renderSynchronizer addRenderer:audioRenderer];
currentPts = kCMTimeZero;
lastPts = kCMTimeZero;
outputPts = kCMTimeZero;
trackPts = kCMTimeZero;
lastCheckpointPts = kCMTimeZero;
bzero(&timeStamp, sizeof(timeStamp));
timeStamp.mFlags = kAudioTimeStampSampleTimeValid;
[self synchronizerBlock];
[audioRenderer setVolume:volume];
return (err == nil);
}
}
- (void)synchronizerBlock {
__block NSLock *lock = [[NSLock alloc] init];
currentPtsLock = lock;
CMTime interval = CMTimeMakeWithSeconds(1.0 / 60.0, 1000000000);
CMTime *lastPts = &self->lastPts;
CMTime *outputPts = &self->outputPts;
OutputNode *outputController = self->outputController;
double *latencySecondsOut = &self->secondsLatency;
VisualizationController *visController = self->visController;
currentPtsObserver = [renderSynchronizer addPeriodicTimeObserverForInterval:interval
queue:NULL
usingBlock:^(CMTime time) {
[lock lock];
self->currentPts = time;
[lock unlock];
CMTime timeToAdd = CMTimeSubtract(time, *lastPts);
self->lastPts = time;
double timeAdded = CMTimeGetSeconds(timeToAdd);
if(timeAdded > 0) {
[outputController incrementAmountPlayed:timeAdded];
}
CMTime latencyTime = CMTimeSubtract(*outputPts, time);
double latencySeconds = CMTimeGetSeconds(latencyTime);
if(latencySeconds < 0)
latencySeconds = 0;
*latencySecondsOut = latencySeconds;
[visController postLatency:latencySeconds];
}];
}
- (void)removeSynchronizerBlock {
[renderSynchronizer removeTimeObserver:currentPtsObserver];
}
- (void)setVolume:(double)v {
volume = v * 0.01f;
if(audioRenderer) {
[audioRenderer setVolume:volume];
}
}
- (void)setEqualizerEnabled:(BOOL)enabled {
if(enabled && !eqEnabled) {
if(_eq) {
AudioUnitReset(_eq, kAudioUnitScope_Input, 0);
AudioUnitReset(_eq, kAudioUnitScope_Output, 0);
AudioUnitReset(_eq, kAudioUnitScope_Global, 0);
}
}
eqEnabled = enabled;
}
- (void)start {
[self threadEntry:nil];
}
- (void)stop {
@synchronized(self) {
if(stopInvoked) return;
stopInvoked = YES;
if(observersapplied) {
[[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.outputDevice" context:kOutputAVFoundationContext];
[[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.GraphicEQenable" context:kOutputAVFoundationContext];
[[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.eqPreamp" context:kOutputAVFoundationContext];
[[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.dontRemix" context:kOutputAVFoundationContext];
observersapplied = NO;
}
stopping = YES;
paused = NO;
if(defaultdevicelistenerapplied || currentdevicelistenerapplied || devicealivelistenerapplied) {
AudioObjectPropertyAddress theAddress = {
.mScope = kAudioObjectPropertyScopeGlobal,
.mElement = kAudioObjectPropertyElementMaster
};
if(defaultdevicelistenerapplied) {
theAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
AudioObjectRemovePropertyListener(kAudioObjectSystemObject, &theAddress, default_device_changed, (__bridge void *_Nullable)(self));
defaultdevicelistenerapplied = NO;
}
if(devicealivelistenerapplied) {
theAddress.mSelector = kAudioDevicePropertyDeviceIsAlive;
AudioObjectRemovePropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
devicealivelistenerapplied = NO;
}
if(currentdevicelistenerapplied) {
theAddress.mSelector = kAudioDevicePropertyStreamFormat;
AudioObjectRemovePropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
theAddress.mSelector = kAudioDevicePropertyNominalSampleRate;
AudioObjectRemovePropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
currentdevicelistenerapplied = NO;
}
}
if(renderSynchronizer || audioRenderer) {
if(renderSynchronizer) {
[self removeSynchronizerBlock];
}
if(audioRenderer) {
[audioRenderer stopRequestingMediaData];
[audioRenderer flush];
}
renderSynchronizer = nil;
audioRenderer = nil;
}
if(running) {
while(!stopped) {
stopping = YES;
usleep(5000);
}
}
if(_eq) {
[outputController endEqualizer:_eq];
if(eqInitialized) {
AudioUnitUninitialize(_eq);
eqInitialized = NO;
}
AudioComponentInstanceDispose(_eq);
_eq = NULL;
}
if(downmixerForVis) {
downmixerForVis = nil;
}
#ifdef OUTPUT_LOG
if(_logFile) {
fclose(_logFile);
_logFile = NULL;
}
#endif
outputController = nil;
visController = nil;
}
}
- (void)dealloc {
[self stop];
}
- (void)pause {
paused = YES;
if(started)
[renderSynchronizer setRate:0];
}
- (void)resume {
[renderSynchronizer setRate:1.0 time:currentPts];
paused = NO;
started = YES;
}
- (void)sustainHDCD {
@synchronized(self) {
secondsHdcdSustained = 10.0;
}
}
@end

986
Audio/Output/OutputCoreAudio.m
View File

@ -1,986 +0,0 @@
//
// OutputCoreAudio.m
// Cog
//
// Created by Vincent Spader on 8/2/05.
// Copyright 2005 Vincent Spader. All rights reserved.
//
#import "OutputCoreAudio.h"
#import "OutputNode.h"
#ifdef _DEBUG
#import "BadSampleCleaner.h"
#endif
#import "Logging.h"
#import <Accelerate/Accelerate.h>
extern void scale_by_volume(float *buffer, size_t count, float volume);
static NSString *CogPlaybackDidBeginNotficiation = @"CogPlaybackDidBeginNotficiation";
@implementation OutputCoreAudio
static void *kOutputCoreAudioContext = &kOutputCoreAudioContext;
static void fillBuffers(AudioBufferList *ioData, float *inbuffer, size_t count, size_t offset) {
const size_t channels = ioData->mNumberBuffers;
for(int i = 0; i < channels; ++i) {
const size_t maxCount = (ioData->mBuffers[i].mDataByteSize / sizeof(float)) - offset;
float *output = ((float *)ioData->mBuffers[i].mData) + offset;
const float *input = inbuffer + i;
cblas_scopy((int)((count > maxCount) ? maxCount : count), input, (int)channels, output, 1);
ioData->mBuffers[i].mNumberChannels = 1;
}
}
static void clearBuffers(AudioBufferList *ioData, size_t count, size_t offset) {
for(int i = 0; i < ioData->mNumberBuffers; ++i) {
memset(ioData->mBuffers[i].mData + offset * sizeof(float), 0, count * sizeof(float));
ioData->mBuffers[i].mNumberChannels = 1;
}
}
static void scaleBuffersByVolume(AudioBufferList *ioData, float volume) {
if(volume != 1.0) {
for(int i = 0; i < ioData->mNumberBuffers; ++i) {
scale_by_volume((float *)ioData->mBuffers[i].mData, ioData->mBuffers[i].mDataByteSize / sizeof(float), volume);
}
}
}
static OSStatus renderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) {
@autoreleasepool {
if(!inRefCon) return 0;
OutputCoreAudio *_self = (__bridge OutputCoreAudio *)inRefCon;
const int channels = _self->deviceFormat.mChannelsPerFrame;
const int bytesPerPacket = channels * sizeof(float);
size_t amountToRead, amountRead = 0;
amountToRead = inNumberFrames * bytesPerPacket;
int visTabulated = 0;
float visAudio[inNumberFrames]; // Chunk size
if(_self->stopping == YES || [_self->outputController shouldContinue] == NO) {
// Chain is dead, fill out the serial number pointer forever with silence
clearBuffers(ioData, amountToRead / bytesPerPacket, 0);
atomic_fetch_add(&_self->bytesRendered, amountToRead);
_self->stopping = YES;
return 0;
}
if([[_self->outputController buffer] isEmpty] && ![_self->outputController chainQueueHasTracks]) {
// Hit end of last track, pad with silence until queue event stops us
clearBuffers(ioData, amountToRead / bytesPerPacket, 0);
atomic_fetch_add(&_self->bytesRendered, amountToRead);
return 0;
}
AudioChunk *chunk = [[_self->outputController buffer] removeSamples:(amountToRead / bytesPerPacket)];
size_t frameCount = [chunk frameCount];
AudioStreamBasicDescription format = [chunk format];
uint32_t config = [chunk channelConfig];
if(frameCount) {
if(!_self->streamFormatStarted || config != _self->streamChannelConfig || memcmp(&_self->streamFormat, &format, sizeof(format)) != 0) {
_self->streamFormat = format;
_self->streamChannelConfig = config;
_self->streamFormatStarted = YES;
_self->downmixer = [[DownmixProcessor alloc] initWithInputFormat:format inputConfig:config andOutputFormat:_self->deviceFormat outputConfig:_self->deviceChannelConfig];
_self->downmixerForVis = [[DownmixProcessor alloc] initWithInputFormat:format inputConfig:config andOutputFormat:_self->visFormat outputConfig:AudioConfigMono];
}
double chunkDuration = [chunk duration];
NSData *samples = [chunk removeSamples:frameCount];
#ifdef _DEBUG
[BadSampleCleaner cleanSamples:(float *)[samples bytes]
amount:frameCount * format.mChannelsPerFrame
location:@"pre downmix"];
#endif
float downmixedData[frameCount * channels];
[_self->downmixer process:[samples bytes] frameCount:frameCount output:downmixedData];
#ifdef _DEBUG
[BadSampleCleaner cleanSamples:downmixedData
amount:frameCount * channels
location:@"post downmix"];
#endif
[_self->downmixerForVis process:[samples bytes] frameCount:frameCount output:visAudio];
visTabulated += frameCount;
fillBuffers(ioData, downmixedData, frameCount, 0);
amountRead = frameCount * bytesPerPacket;
[_self->outputController incrementAmountPlayed:chunkDuration];
atomic_fetch_add(&_self->bytesRendered, amountRead);
[_self->writeSemaphore signal];
}
// Try repeatedly! Buffer wraps can cause a slight data shortage, as can
// unexpected track changes.
while((amountRead < amountToRead) && [_self->outputController shouldContinue] == YES) {
chunk = [[_self->outputController buffer] removeSamples:((amountToRead - amountRead) / bytesPerPacket)];
frameCount = [chunk frameCount];
format = [chunk format];
config = [chunk channelConfig];
if(frameCount) {
if(!_self->streamFormatStarted || config != _self->streamChannelConfig || memcmp(&_self->streamFormat, &format, sizeof(format)) != 0) {
_self->streamFormat = format;
_self->streamFormatStarted = YES;
_self->downmixer = [[DownmixProcessor alloc] initWithInputFormat:format inputConfig:config andOutputFormat:_self->deviceFormat outputConfig:_self->deviceChannelConfig];
_self->downmixerForVis = [[DownmixProcessor alloc] initWithInputFormat:format inputConfig:config andOutputFormat:_self->visFormat outputConfig:AudioConfigMono];
}
atomic_fetch_add(&_self->bytesRendered, frameCount * bytesPerPacket);
double chunkDuration = [chunk duration];
NSData *samples = [chunk removeSamples:frameCount];
#ifdef _DEBUG
[BadSampleCleaner cleanSamples:(float *)[samples bytes]
amount:frameCount * format.mChannelsPerFrame
location:@"pre downmix"];
#endif
float downmixedData[frameCount * channels];
[_self->downmixer process:[samples bytes] frameCount:frameCount output:downmixedData];
fillBuffers(ioData, downmixedData, frameCount, amountRead / bytesPerPacket);
#ifdef _DEBUG
[BadSampleCleaner cleanSamples:downmixedData
amount:frameCount * channels
location:@"post downmix"];
#endif
[_self->downmixerForVis process:[samples bytes] frameCount:frameCount output:visAudio + visTabulated];
visTabulated += frameCount;
[_self->outputController incrementAmountPlayed:chunkDuration];
amountRead += frameCount * bytesPerPacket;
[_self->writeSemaphore signal];
} else {
[_self->readSemaphore timedWait:500];
}
}
float volumeScale = 1.0;
long sustained = atomic_load_explicit(&_self->bytesHdcdSustained, memory_order_relaxed);
if(sustained) {
if(sustained < amountRead) {
atomic_store(&_self->bytesHdcdSustained, 0);
} else {
atomic_fetch_sub(&_self->bytesHdcdSustained, amountRead);
}
volumeScale = 0.5;
}
if(_self->eqEnabled) {
volumeScale *= _self->eqPreamp;
}
scaleBuffersByVolume(ioData, _self->volume * volumeScale);
if(amountRead < amountToRead) {
// Either underrun, or no data at all. Caller output tends to just
// buffer loop if it doesn't get anything, so always produce a full
// buffer, and silence anything we couldn't supply.
clearBuffers(ioData, (amountToRead - amountRead) / bytesPerPacket, amountRead / bytesPerPacket);
}
[_self->visController postSampleRate:_self->deviceFormat.mSampleRate];
[_self->visController postVisPCM:visAudio amount:visTabulated];
return 0;
}
};
- (id)initWithController:(OutputNode *)c {
self = [super init];
if(self) {
outputController = c;
volume = 1.0;
outputDeviceID = -1;
atomic_init(&bytesRendered, 0);
atomic_init(&bytesHdcdSustained, 0);
writeSemaphore = [[Semaphore alloc] init];
readSemaphore = [[Semaphore alloc] init];
#ifdef OUTPUT_LOG
_logFile = fopen("/tmp/CogAudioLog.raw", "wb");
#endif
}
return self;
}
static OSStatus
default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const AudioObjectPropertyAddress *inAddresses, void *inUserData) {
OutputCoreAudio *this = (__bridge OutputCoreAudio *)inUserData;
return [this setOutputDeviceByID:-1];
}
static OSStatus
current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, const AudioObjectPropertyAddress *inAddresses, void *inUserData) {
OutputCoreAudio *this = (__bridge OutputCoreAudio *)inUserData;
for(UInt32 i = 0; i < inNumberAddresses; ++i) {
switch(inAddresses[i].mSelector) {
case kAudioDevicePropertyDeviceIsAlive:
return [this setOutputDeviceByID:-1];
case kAudioDevicePropertyNominalSampleRate:
case kAudioDevicePropertyStreamFormat:
this->outputdevicechanged = YES;
return noErr;
}
}
return noErr;
}
- (void)observeValueForKeyPath:(NSString *)keyPath ofObject:(id)object change:(NSDictionary *)change context:(void *)context {
if(context != kOutputCoreAudioContext) {
[super observeValueForKeyPath:keyPath ofObject:object change:change context:context];
return;
}
if([keyPath isEqualToString:@"values.outputDevice"]) {
NSDictionary *device = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] objectForKey:@"outputDevice"];
[self setOutputDeviceWithDeviceDict:device];
} else if([keyPath isEqualToString:@"values.GraphicEQenable"]) {
BOOL enabled = [[[[NSUserDefaultsController sharedUserDefaultsController] defaults] objectForKey:@"GraphicEQenable"] boolValue];
[self setEqualizerEnabled:enabled];
} else if([keyPath isEqualToString:@"values.eqPreamp"]) {
float preamp = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] floatForKey:@"eqPreamp"];
eqPreamp = pow(10.0, preamp / 20.0);
}
}
- (void)signalEndOfStream {
[outputController resetAmountPlayed];
[outputController endOfInputPlayed];
}
- (void)threadEntry:(id)arg {
running = YES;
started = NO;
stopNext = NO;
atomic_store(&bytesRendered, 0);
NSMutableArray *delayedEvents = [[NSMutableArray alloc] init];
BOOL delayedEventsPopped = YES;
while(!stopping) {
if(outputdevicechanged) {
[self resetIfOutputChanged];
outputdevicechanged = NO;
}
if(restarted) break;
if([outputController shouldReset]) {
@autoreleasepool {
[[outputController buffer] reset];
}
[outputController setShouldReset:NO];
[delayedEvents removeAllObjects];
delayedEventsPopped = YES;
}
while([delayedEvents count]) {
size_t localBytesRendered = atomic_load_explicit(&bytesRendered, memory_order_relaxed);
double secondsRendered = (double)localBytesRendered / (double)(deviceFormat.mBytesPerPacket * deviceFormat.mSampleRate);
if(secondsRendered >= [[delayedEvents objectAtIndex:0] doubleValue]) {
if([outputController chainQueueHasTracks])
delayedEventsPopped = YES;
[self signalEndOfStream];
[delayedEvents removeObjectAtIndex:0];
} else
break;
}
if(stopping)
break;
size_t frameCount = 0;
if(![[outputController buffer] isFull]) {
@autoreleasepool {
AudioChunk *chunk = [outputController readChunk:512];
frameCount = [chunk frameCount];
if(frameCount) {
[[outputController buffer] addChunk:chunk];
}
}
}
if(frameCount) {
[readSemaphore signal];
continue;
} else if([outputController shouldContinue] == NO)
break;
else if([[outputController buffer] isFull]) {
if(!started) {
started = YES;
if(!paused) {
NSError *err;
[_au startHardwareAndReturnError:&err];
}
}
} else {
// End of input possibly reached
if(delayedEventsPopped && [outputController endOfStream] == YES) {
double secondsBuffered = [[outputController buffer] listDuration];
size_t _bytesRendered = atomic_load_explicit(&bytesRendered, memory_order_relaxed);
secondsBuffered += (double)_bytesRendered / (double)(deviceFormat.mBytesPerPacket * deviceFormat.mSampleRate);
if([outputController chainQueueHasTracks]) {
if(secondsBuffered <= 0.005)
secondsBuffered = 0.0;
else
secondsBuffered -= 0.005;
} else {
stopNext = YES;
break;
}
[delayedEvents addObject:@(secondsBuffered)];
delayedEventsPopped = NO;
if(!started) {
started = YES;
if(!paused) {
NSError *err;
[_au startHardwareAndReturnError:&err];
}
}
}
}
[readSemaphore signal];
[writeSemaphore timedWait:5000];
}
stopped = YES;
[self stop];
}
- (OSStatus)setOutputDeviceByID:(AudioDeviceID)deviceID {
OSStatus err;
BOOL defaultDevice = NO;
AudioObjectPropertyAddress theAddress = {
.mSelector = kAudioHardwarePropertyDefaultOutputDevice,
.mScope = kAudioObjectPropertyScopeGlobal,
.mElement = kAudioObjectPropertyElementMaster
};
if(deviceID == -1) {
defaultDevice = YES;
UInt32 size = sizeof(AudioDeviceID);
err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &theAddress, 0, NULL, &size, &deviceID);
if(err != noErr) {
DLog(@"THERE'S NO DEFAULT OUTPUT DEVICE");
return err;
}
}
if(_au) {
if(defaultdevicelistenerapplied && !defaultDevice) {
/* Already set above
* theAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice; */
AudioObjectRemovePropertyListener(kAudioObjectSystemObject, &theAddress, default_device_changed, (__bridge void *_Nullable)(self));
defaultdevicelistenerapplied = NO;
}
outputdevicechanged = NO;
if(outputDeviceID != deviceID) {
if(currentdevicelistenerapplied) {
if(devicealivelistenerapplied) {
theAddress.mSelector = kAudioDevicePropertyDeviceIsAlive;
AudioObjectRemovePropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
devicealivelistenerapplied = NO;
}
theAddress.mSelector = kAudioDevicePropertyStreamFormat;
AudioObjectRemovePropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
theAddress.mSelector = kAudioDevicePropertyNominalSampleRate;
AudioObjectRemovePropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
currentdevicelistenerapplied = NO;
}
DLog(@"Device: %i\n", deviceID);
outputDeviceID = deviceID;
NSError *nserr;
[_au setDeviceID:outputDeviceID error:&nserr];
if(nserr != nil) {
return (OSErr)[nserr code];
}
outputdevicechanged = YES;
}
if(!currentdevicelistenerapplied) {
if(!devicealivelistenerapplied && !defaultDevice) {
theAddress.mSelector = kAudioDevicePropertyDeviceIsAlive;
AudioObjectAddPropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
devicealivelistenerapplied = YES;
}
theAddress.mSelector = kAudioDevicePropertyStreamFormat;
AudioObjectAddPropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
theAddress.mSelector = kAudioDevicePropertyNominalSampleRate;
AudioObjectAddPropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
currentdevicelistenerapplied = YES;
}
if(!defaultdevicelistenerapplied && defaultDevice) {
theAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
AudioObjectAddPropertyListener(kAudioObjectSystemObject, &theAddress, default_device_changed, (__bridge void *_Nullable)(self));
defaultdevicelistenerapplied = YES;
}
} else {
err = noErr;
}
if(err != noErr) {
DLog(@"No output device with ID %d could be found.", deviceID);
return err;
}
return err;
}
- (BOOL)setOutputDeviceWithDeviceDict:(NSDictionary *)deviceDict {
NSNumber *deviceIDNum = [deviceDict objectForKey:@"deviceID"];
AudioDeviceID outputDeviceID = [deviceIDNum unsignedIntValue] ?: -1;
__block OSStatus err = [self setOutputDeviceByID:outputDeviceID];
if(err != noErr) {
// Try matching by name.
NSString *userDeviceName = deviceDict[@"name"];
[self enumerateAudioOutputsUsingBlock:
^(NSString *deviceName, AudioDeviceID deviceID, AudioDeviceID systemDefaultID, BOOL *stop) {
if([deviceName isEqualToString:userDeviceName]) {
err = [self setOutputDeviceByID:deviceID];
#if 0
// Disable. Would cause loop by triggering `-observeValueForKeyPath:ofObject:change:context:` above.
// Update `outputDevice`, in case the ID has changed.
NSDictionary *deviceInfo = @{
@"name": deviceName,
@"deviceID": @(deviceID),
};
[[NSUserDefaults standardUserDefaults] setObject:deviceInfo forKey:@"outputDevice"];
#endif
DLog(@"Found output device: \"%@\" (%d).", deviceName, deviceID);
*stop = YES;
}
}];
}
if(err != noErr) {
ALog(@"No output device could be found, your random error code is %d. Have a nice day!", err);
return NO;
}
return YES;
}
// The following is largely a copy pasta of -awakeFromNib from "OutputsArrayController.m".
// TODO: Share the code. (How to do this across xcodeproj?)
- (void)enumerateAudioOutputsUsingBlock:(void(NS_NOESCAPE ^ _Nonnull)(NSString *deviceName, AudioDeviceID deviceID, AudioDeviceID systemDefaultID, BOOL *stop))block {
UInt32 propsize;
AudioObjectPropertyAddress theAddress = {
.mSelector = kAudioHardwarePropertyDevices,
.mScope = kAudioObjectPropertyScopeGlobal,
.mElement = kAudioObjectPropertyElementMaster
};
__Verify_noErr(AudioObjectGetPropertyDataSize(kAudioObjectSystemObject, &theAddress, 0, NULL, &propsize));
UInt32 nDevices = propsize / (UInt32)sizeof(AudioDeviceID);
AudioDeviceID *devids = malloc(propsize);
__Verify_noErr(AudioObjectGetPropertyData(kAudioObjectSystemObject, &theAddress, 0, NULL, &propsize, devids));
theAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
AudioDeviceID systemDefault;
propsize = sizeof(systemDefault);
__Verify_noErr(AudioObjectGetPropertyData(kAudioObjectSystemObject, &theAddress, 0, NULL, &propsize, &systemDefault));
theAddress.mScope = kAudioDevicePropertyScopeOutput;
for(UInt32 i = 0; i < nDevices; ++i) {
UInt32 isAlive = 0;
propsize = sizeof(isAlive);
theAddress.mSelector = kAudioDevicePropertyDeviceIsAlive;
__Verify_noErr(AudioObjectGetPropertyData(devids[i], &theAddress, 0, NULL, &propsize, &isAlive));
if(!isAlive) continue;
CFStringRef name = NULL;
propsize = sizeof(name);
theAddress.mSelector = kAudioDevicePropertyDeviceNameCFString;
__Verify_noErr(AudioObjectGetPropertyData(devids[i], &theAddress, 0, NULL, &propsize, &name));
propsize = 0;
theAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
__Verify_noErr(AudioObjectGetPropertyDataSize(devids[i], &theAddress, 0, NULL, &propsize));
if(propsize < sizeof(UInt32)) {
if(name) CFRelease(name);
continue;
}
AudioBufferList *bufferList = (AudioBufferList *)malloc(propsize);
__Verify_noErr(AudioObjectGetPropertyData(devids[i], &theAddress, 0, NULL, &propsize, bufferList));
UInt32 bufferCount = bufferList->mNumberBuffers;
free(bufferList);
if(!bufferCount) {
if(name) CFRelease(name);
continue;
}
BOOL stop = NO;
NSString *deviceName = name ? [NSString stringWithString:(__bridge NSString *)name] : [NSString stringWithFormat:@"Unknown device %u", (unsigned int)devids[i]];
block(deviceName,
devids[i],
systemDefault,
&stop);
CFRelease(name);
if(stop) {
break;
}
}
free(devids);
}
- (void)resetIfOutputChanged {
AVAudioFormat *format = _au.outputBusses[0].format;
if(!restarted && (!_deviceFormat || ![_deviceFormat isEqual:format])) {
[outputController restartPlaybackAtCurrentPosition];
restarted = YES;
}
}
- (BOOL)updateDeviceFormat {
AVAudioFormat *format = _au.outputBusses[0].format;
if(!_deviceFormat || ![_deviceFormat isEqual:format]) {
NSError *err;
AVAudioFormat *renderFormat;
[outputController incrementAmountPlayed:[[outputController buffer] listDuration]];
@autoreleasepool {
[[outputController buffer] reset];
}
_deviceFormat = format;
deviceFormat = *(format.streamDescription);
/// Seems some 3rd party devices return incorrect stuff...or I just don't like noninterleaved data.
deviceFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved;
// deviceFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsFloat;
// deviceFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
// We don't want more than 8 channels
if(deviceFormat.mChannelsPerFrame > 8) {
deviceFormat.mChannelsPerFrame = 8;
}
deviceFormat.mBytesPerFrame = deviceFormat.mChannelsPerFrame * (deviceFormat.mBitsPerChannel / 8);
deviceFormat.mBytesPerPacket = deviceFormat.mBytesPerFrame * deviceFormat.mFramesPerPacket;
visFormat = deviceFormat;
visFormat.mChannelsPerFrame = 1;
visFormat.mBytesPerFrame = visFormat.mChannelsPerFrame * (visFormat.mBitsPerChannel / 8);
visFormat.mBytesPerPacket = visFormat.mBytesPerFrame * visFormat.mFramesPerPacket;
/* Set the channel layout for the audio queue */
AudioChannelLayoutTag tag = 0;
switch(deviceFormat.mChannelsPerFrame) {
case 1:
tag = kAudioChannelLayoutTag_Mono;
deviceChannelConfig = AudioConfigMono;
break;
case 2:
tag = kAudioChannelLayoutTag_Stereo;
deviceChannelConfig = AudioConfigStereo;
break;
case 3:
tag = kAudioChannelLayoutTag_DVD_4;
deviceChannelConfig = AudioConfig3Point0;
break;
case 4:
tag = kAudioChannelLayoutTag_Quadraphonic;
deviceChannelConfig = AudioConfig4Point0;
break;
case 5:
tag = kAudioChannelLayoutTag_MPEG_5_0_A;
deviceChannelConfig = AudioConfig5Point0;
break;
case 6:
tag = kAudioChannelLayoutTag_MPEG_5_1_A;
deviceChannelConfig = AudioConfig5Point1;
break;
case 7:
tag = kAudioChannelLayoutTag_MPEG_6_1_A;
deviceChannelConfig = AudioConfig6Point1;
break;
case 8:
tag = kAudioChannelLayoutTag_MPEG_7_1_A;
deviceChannelConfig = AudioConfig7Point1;
break;
}
renderFormat = [[AVAudioFormat alloc] initWithStreamDescription:&deviceFormat channelLayout:[[AVAudioChannelLayout alloc] initWithLayoutTag:tag]];
[_au.inputBusses[0] setFormat:renderFormat error:&err];
if(err != nil)
return NO;
[outputController setFormat:&deviceFormat channelConfig:deviceChannelConfig];
AudioStreamBasicDescription asbd = deviceFormat;
asbd.mFormatFlags &= ~kAudioFormatFlagIsPacked;
AudioUnitSetProperty(_eq, kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 0, &asbd, sizeof(asbd));
AudioUnitSetProperty(_eq, kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output, 0, &asbd, sizeof(asbd));
AudioUnitReset(_eq, kAudioUnitScope_Input, 0);
AudioUnitReset(_eq, kAudioUnitScope_Output, 0);
AudioUnitReset(_eq, kAudioUnitScope_Global, 0);
eqEnabled = [[[[NSUserDefaultsController sharedUserDefaultsController] defaults] objectForKey:@"GraphicEQenable"] boolValue];
}
return YES;
}
- (void)audioOutputBlock {
__block AudioUnit eq = _eq;
__block AudioStreamBasicDescription *format = &deviceFormat;
__block BOOL *eqEnabled = &self->eqEnabled;
__block void *refCon = (__bridge void *)self;
#ifdef OUTPUT_LOG
__block FILE *logFile = _logFile;
#endif
_au.outputProvider = ^AUAudioUnitStatus(AudioUnitRenderActionFlags *_Nonnull actionFlags, const AudioTimeStamp *_Nonnull timestamp, AUAudioFrameCount frameCount, NSInteger inputBusNumber, AudioBufferList *_Nonnull inputData) {
// This expects multiple buffers, so:
if(!frameCount) return 0;
int i;
const int channels = format->mChannelsPerFrame;
if(!channels) return 0;
const int channelsminusone = channels - 1;
float buffers[frameCount * channels];
uint8_t bufferlistbuffer[sizeof(AudioBufferList) + sizeof(AudioBuffer) * channelsminusone];
AudioBufferList *ioData = (AudioBufferList *)(&bufferlistbuffer[0]);
ioData->mNumberBuffers = channels;
memset(buffers, 0, sizeof(buffers));
for(i = 0; i < channels; ++i) {
ioData->mBuffers[i].mNumberChannels = 1;
ioData->mBuffers[i].mData = buffers + frameCount * i;
ioData->mBuffers[i].mDataByteSize = frameCount * sizeof(float);
}
OSStatus ret;
if(*eqEnabled)
ret = AudioUnitRender(eq, actionFlags, timestamp, (UInt32)inputBusNumber, frameCount, ioData);
else
ret = renderCallback(refCon, actionFlags, timestamp, (UInt32)inputBusNumber, frameCount, ioData);
if(ret)
return ret;
for(i = 0; i < channels; ++i) {
float *outBuffer = ((float *)inputData->mBuffers[0].mData) + i;
const float *inBuffer = ((float *)ioData->mBuffers[i].mData);
const int frameCount = ioData->mBuffers[i].mDataByteSize / sizeof(float);
cblas_scopy(frameCount, inBuffer, 1, outBuffer, channels);
}
#ifdef OUTPUT_LOG
if(logFile) {
fwrite(inputData->mBuffers[0].mData, 1, inputData->mBuffers[0].mDataByteSize, logFile);
}
// memset(inputData->mBuffers[0].mData, 0, inputData->mBuffers[0].mDataByteSize);
#endif
inputData->mBuffers[0].mNumberChannels = channels;
#ifdef _DEBUG
[BadSampleCleaner cleanSamples:(float *)inputData->mBuffers[0].mData
amount:inputData->mBuffers[0].mDataByteSize / sizeof(float)
location:@"final output"];
#endif
return 0;
};
}
- (BOOL)setup {
if(_au)
[self stop];
@synchronized(self) {
stopInvoked = NO;
running = NO;
stopping = NO;
stopped = NO;
paused = NO;
stopNext = NO;
outputDeviceID = -1;
restarted = NO;
downmixer = nil;
downmixerForVis = nil;
AudioComponentDescription desc;
NSError *err;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_HALOutput;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
_au = [[AUAudioUnit alloc] initWithComponentDescription:desc error:&err];
if(err != nil)
return NO;
// Setup the output device before mucking with settings
NSDictionary *device = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] objectForKey:@"outputDevice"];
if(device) {
BOOL ok = [self setOutputDeviceWithDeviceDict:device];
if(!ok) {
// Ruh roh.
[self setOutputDeviceWithDeviceDict:nil];
[[[NSUserDefaultsController sharedUserDefaultsController] defaults] removeObjectForKey:@"outputDevice"];
}
} else {
[self setOutputDeviceWithDeviceDict:nil];
}
_deviceFormat = nil;
AudioComponent comp = NULL;
desc.componentType = kAudioUnitType_Effect;
desc.componentSubType = kAudioUnitSubType_GraphicEQ;
comp = AudioComponentFindNext(comp, &desc);
if(!comp)
return NO;
OSStatus _err = AudioComponentInstanceNew(comp, &_eq);
if(err)
return NO;
[self updateDeviceFormat];
[self audioOutputBlock];
[_au setMaximumFramesToRender:512];
UInt32 value;
UInt32 size = sizeof(value);
value = CHUNK_SIZE;
AudioUnitSetProperty(_eq, kAudioUnitProperty_MaximumFramesPerSlice,
kAudioUnitScope_Global, 0, &value, size);
value = 127;
AudioUnitSetProperty(_eq, kAudioUnitProperty_RenderQuality,
kAudioUnitScope_Global, 0, &value, size);
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProcRefCon = (__bridge void *)self;
callbackStruct.inputProc = renderCallback;
AudioUnitSetProperty(_eq, kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input, 0, &callbackStruct, sizeof(callbackStruct));
AudioUnitReset(_eq, kAudioUnitScope_Input, 0);
AudioUnitReset(_eq, kAudioUnitScope_Output, 0);
AudioUnitReset(_eq, kAudioUnitScope_Global, 0);
_err = AudioUnitInitialize(_eq);
if(_err)
return NO;
eqInitialized = YES;
[self setEqualizerEnabled:[[[[NSUserDefaultsController sharedUserDefaultsController] defaults] objectForKey:@"GraphicEQenable"] boolValue]];
[outputController beginEqualizer:_eq];
[_au allocateRenderResourcesAndReturnError:&err];
visController = [VisualizationController sharedController];
[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.outputDevice" options:0 context:kOutputCoreAudioContext];
[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.GraphicEQenable" options:0 context:kOutputCoreAudioContext];
[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.eqPreamp" options:(NSKeyValueObservingOptionInitial | NSKeyValueObservingOptionNew) context:kOutputCoreAudioContext];
observersapplied = YES;
return (err == nil);
}
}
- (void)setVolume:(double)v {
volume = v * 0.01f;
}
- (void)setEqualizerEnabled:(BOOL)enabled {
if(enabled && !eqEnabled) {
if(_eq) {
AudioUnitReset(_eq, kAudioUnitScope_Input, 0);
AudioUnitReset(_eq, kAudioUnitScope_Output, 0);
AudioUnitReset(_eq, kAudioUnitScope_Global, 0);
}
}
eqEnabled = enabled;
}
- (void)start {
[self threadEntry:nil];
}
- (void)stop {
@synchronized(self) {
if(stopInvoked) return;
stopInvoked = YES;
if(observersapplied) {
[[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.outputDevice" context:kOutputCoreAudioContext];
[[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.GraphicEQenable" context:kOutputCoreAudioContext];
[[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.eqPreamp" context:kOutputCoreAudioContext];
observersapplied = NO;
}
if(stopNext && !paused) {
if(!started) {
// This happens if playback is started on a very short file, and the queue is empty or at the end of the playlist
started = YES;
NSError *err;
[_au startHardwareAndReturnError:&err];
}
while(![[outputController buffer] isEmpty]) {
[writeSemaphore signal];
[readSemaphore signal];
usleep(500);
}
}
if(stopNext) {
stopNext = NO;
[self signalEndOfStream];
}
stopping = YES;
paused = NO;
[writeSemaphore signal];
[readSemaphore signal];
if(defaultdevicelistenerapplied || currentdevicelistenerapplied || devicealivelistenerapplied) {
AudioObjectPropertyAddress theAddress = {
.mScope = kAudioObjectPropertyScopeGlobal,
.mElement = kAudioObjectPropertyElementMaster
};
if(defaultdevicelistenerapplied) {
theAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
AudioObjectRemovePropertyListener(kAudioObjectSystemObject, &theAddress, default_device_changed, (__bridge void *_Nullable)(self));
defaultdevicelistenerapplied = NO;
}
if(devicealivelistenerapplied) {
theAddress.mSelector = kAudioDevicePropertyDeviceIsAlive;
AudioObjectRemovePropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
devicealivelistenerapplied = NO;
}
if(currentdevicelistenerapplied) {
theAddress.mSelector = kAudioDevicePropertyStreamFormat;
AudioObjectRemovePropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
theAddress.mSelector = kAudioDevicePropertyNominalSampleRate;
AudioObjectRemovePropertyListener(outputDeviceID, &theAddress, current_device_listener, (__bridge void *_Nullable)(self));
currentdevicelistenerapplied = NO;
}
}
if(_au) {
if(started)
[_au stopHardware];
_au = nil;
}
if(running) {
while(!stopped) {
stopping = YES;
[readSemaphore signal];
[writeSemaphore timedWait:5000];
}
}
if(_eq) {
[outputController endEqualizer:_eq];
if(eqInitialized) {
AudioUnitUninitialize(_eq);
eqInitialized = NO;
}
AudioComponentInstanceDispose(_eq);
_eq = NULL;
}
if(downmixer) {
downmixer = nil;
}
if(downmixerForVis) {
downmixerForVis = nil;
}
#ifdef OUTPUT_LOG
if(_logFile) {
fclose(_logFile);
_logFile = NULL;
}
#endif
outputController = nil;
visController = nil;
}
}
- (void)dealloc {
[self stop];
}
- (void)pause {
paused = YES;
if(started)
[_au stopHardware];
}
- (void)resume {
NSError *err;
[_au startHardwareAndReturnError:&err];
paused = NO;
started = YES;
}
- (void)sustainHDCD {
atomic_store(&bytesHdcdSustained, deviceFormat.mSampleRate * 10 * sizeof(float) * 2);
}
@end

504
Audio/ThirdParty/lvqcl/License/LICENSE.LGPL vendored
View File

@ -1,504 +0,0 @@
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
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That's all there is to it!

39
Audio/ThirdParty/lvqcl/License/License.txt vendored
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@ -1,39 +0,0 @@
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

193
Audio/ThirdParty/lvqcl/lpc.c vendored
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@ -1,193 +0,0 @@
/*
* 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];
}
}
}

48
Audio/ThirdParty/lvqcl/lpc.h vendored
View File

@ -1,48 +0,0 @@
#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;
#ifdef __cplusplus
extern "C" {
#endif
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);
#ifdef __cplusplus
}
#endif
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

55
Audio/ThirdParty/lvqcl/util.h vendored
View File

@ -1,55 +0,0 @@
/* 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 || *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

1
Audio/ThirdParty/r8brain-free-src vendored

@ -1 +0,0 @@
Subproject commit afd61e7ed76d86a9bc6cb91fd0a9f305f853fe38

164
Audio/ThirdParty/r8bstate.h vendored
View File

@ -1,164 +0,0 @@
//
// r8bstate.h
// CogAudio Framework
//
// Created by Christopher Snowhill on 3/3/22.
//
#ifndef r8bstate_h
#define r8bstate_h
#include <Accelerate/Accelerate.h>
#include "r8bbase.h"
#include "CDSPResampler.h"
struct r8bstate {
int channelCount;
int bufferCapacity;
size_t remainder;
uint64_t inProcessed;
uint64_t outProcessed;
double sampleRatio;
r8b::CFixedBuffer<double> InBuf;
r8b::CFixedBuffer<double> *OutBufs;
r8b::CDSPResampler24 **Resamps;
r8bstate(int _channelCount, int _bufferCapacity, double srcRate, double dstRate)
: channelCount(_channelCount), bufferCapacity(_bufferCapacity), inProcessed(0), outProcessed(0), remainder(0) {
InBuf.alloc(bufferCapacity);
OutBufs = new r8b::CFixedBuffer<double>[channelCount];
Resamps = new r8b::CDSPResampler24 *[channelCount];
for(int i = 0; i < channelCount; ++i) {
Resamps[i] = new r8b::CDSPResampler24(srcRate, dstRate, bufferCapacity);
}
sampleRatio = dstRate / srcRate;
}
~r8bstate() {
delete[] OutBufs;
for(int i = 0; i < channelCount; ++i) {
delete Resamps[i];
}
delete[] Resamps;
}
double latency() {
return ((double)inProcessed * sampleRatio) - (double)outProcessed;
}
int resample(const float *input, size_t inCount, size_t *inDone, float *output, size_t outMax) {
int ret = 0;
int i;
if(inDone) *inDone = 0;
while(remainder > 0) {
size_t blockCount = remainder;
if(blockCount > outMax)
blockCount = outMax;
for(i = 0; i < channelCount; ++i) {
vDSP_vdpsp(&OutBufs[i][0], 1, output + i, channelCount, blockCount);
}
remainder -= blockCount;
if(remainder > 0) {
for(i = 0; i < channelCount; ++i) {
memmove(&OutBufs[i][0], &OutBufs[i][blockCount], remainder * sizeof(double));
}
}
output += channelCount * blockCount;
outProcessed += blockCount;
outMax -= blockCount;
ret += blockCount;
if(!outMax)
return ret;
}
while(inCount > 0) {
size_t blockCount = inCount;
if(blockCount > bufferCapacity)
blockCount = bufferCapacity;
int outputDone = 0;
for(i = 0; i < channelCount; ++i) {
double *outputPointer;
vDSP_vspdp(input + i, channelCount, &InBuf[0], 1, blockCount);
outputDone = Resamps[i]->process(InBuf, (int)blockCount, outputPointer);
if(outputDone) {
if(outputDone > outMax) {
vDSP_vdpsp(outputPointer, 1, output + i, channelCount, outMax);
remainder = outputDone - outMax;
OutBufs[i].alloc((int)remainder);
memcpy(&OutBufs[i][0], outputPointer + outMax, remainder);
} else {
vDSP_vdpsp(outputPointer, 1, output + i, channelCount, outputDone);
}
}
}
size_t outputActual = outputDone - remainder;
input += channelCount * blockCount;
output += channelCount * outputActual;
inCount -= blockCount;
if(inDone) *inDone += blockCount;
inProcessed += blockCount;
outProcessed += outputActual;
outMax -= outputActual;
ret += outputActual;
if(remainder)
break;
}
return ret;
}
int flush(float *output, size_t outMax) {
int ret = 0;
int i;
if(remainder > 0) {
size_t blockCount = remainder;
if(blockCount > outMax)
blockCount = outMax;
for(i = 0; i < channelCount; ++i) {
vDSP_vdpsp(&OutBufs[i][0], 1, output + i, channelCount, blockCount);
}
remainder -= blockCount;
if(remainder > 0) {
for(i = 0; i < channelCount; ++i) {
memmove(&OutBufs[i][0], &OutBufs[i][blockCount], remainder * sizeof(double));
}
}
output += channelCount * blockCount;
outProcessed += blockCount;
outMax -= blockCount;
ret += blockCount;
if(!outMax)
return ret;
}
uint64_t outputWanted = ceil(inProcessed * sampleRatio);
memset(&InBuf[0], 0, sizeof(double) * bufferCapacity);
while(outProcessed < outputWanted) {
int outputDone = 0;
for(int i = 0; i < channelCount; ++i) {
double *outputPointer;
outputDone = Resamps[i]->process(InBuf, bufferCapacity, outputPointer);
if(outputDone) {
if(outputDone > (outputWanted - outProcessed))
outputDone = (int)(outputWanted - outProcessed);
if(outputDone > outMax) {
vDSP_vdpsp(outputPointer, 1, output + i, channelCount, outMax);
remainder = outputDone - outMax;
OutBufs[i].alloc((int)remainder);
memcpy(&OutBufs[i][0], outputPointer + outMax, remainder);
} else {
vDSP_vdpsp(outputPointer, 1, output + i, channelCount, outputDone);
}
}
}
size_t outputActual = outputDone - remainder;
outProcessed += outputActual;
output += channelCount * outputActual;
outMax -= outputActual;
ret += outputActual;
if(remainder)
break;
}
return ret;
}
};
#endif /* r8bstate_h */

6
Audio/Visualization/VisualizationController.h
View File

@ -11,11 +11,15 @@ NS_ASSUME_NONNULL_BEGIN
@interface VisualizationController : NSObject {
double sampleRate;
float visAudio[4096];
double latency;
float *visAudio;
int visAudioCursor, visAudioSize;
}
+ (VisualizationController *)sharedController;
- (void)postLatency:(double)latency;
- (void)postSampleRate:(double)sampleRate;
- (void)postVisPCM:(const float *)inPCM amount:(int)amount;
- (double)readSampleRate;

64
Audio/Visualization/VisualizationController.m
View File

@ -26,7 +26,8 @@ static VisualizationController *_sharedController = nil;
- (id)init {
self = [super init];
if(self) {
vDSP_vclr(visAudio, 1, 4096);
visAudio = NULL;
latency = 0;
}
return self;
}
@ -37,22 +38,42 @@ static VisualizationController *_sharedController = nil;
- (void)postSampleRate:(double)sampleRate {
@synchronized(self) {
self->sampleRate = sampleRate;
if(self->sampleRate != sampleRate) {
self->sampleRate = sampleRate;
int visAudioSize = (int)(sampleRate * 30.0);
void *visAudio = realloc(self->visAudio, visAudioSize * sizeof(float));
if(visAudio && visAudioSize) {
if(visAudioSize > self->visAudioSize) {
bzero(((float *)visAudio) + self->visAudioSize, sizeof(float) * (visAudioSize - self->visAudioSize));
}
self->visAudio = visAudio;
self->visAudioSize = visAudioSize;
visAudioCursor %= visAudioSize;
}
}
}
}
- (void)postVisPCM:(const float *)inPCM amount:(int)amount {
int skipAmount = 0;
if(amount > 4096) {
skipAmount = amount - 4096;
amount = 4096;
}
@synchronized(self) {
cblas_scopy(4096 - amount, visAudio + amount, 1, visAudio, 1);
cblas_scopy(amount, inPCM + skipAmount, 1, visAudio + 4096 - amount, 1);
int samplesRead = 0;
while(amount > 0) {
int amountToCopy = (int)(visAudioSize - visAudioCursor);
if(amountToCopy > amount) amountToCopy = amount;
cblas_scopy(amountToCopy, inPCM + samplesRead, 1, visAudio + visAudioCursor, 1);
visAudioCursor = visAudioCursor + amountToCopy;
if(visAudioCursor >= visAudioSize) visAudioCursor -= visAudioSize;
amount -= amountToCopy;
samplesRead += amountToCopy;
}
}
}
- (void)postLatency:(double)latency {
self->latency = latency;
assert(latency < 30.0);
}
- (double)readSampleRate {
@synchronized(self) {
return sampleRate;
@ -61,9 +82,30 @@ static VisualizationController *_sharedController = nil;
- (void)copyVisPCM:(float *)outPCM visFFT:(float *)outFFT {
@synchronized(self) {
cblas_scopy(4096, visAudio, 1, outPCM, 1);
fft_calculate(visAudio, outFFT, 2048);
if(!sampleRate) {
bzero(outPCM, 4096 * sizeof(float));
bzero(outFFT, 2048 * sizeof(float));
return;
}
int latencySamples = (int)(sampleRate * latency);
int readCursor = visAudioCursor - latencySamples;
int samples = 4096;
int samplesRead = 0;
if(readCursor < 0)
readCursor += visAudioSize;
else if(readCursor >= visAudioSize)
readCursor -= visAudioSize;
while(samples > 0) {
int samplesToRead = (int)(visAudioSize - readCursor);
if(samplesToRead > samples) samplesToRead = samples;
cblas_scopy(samplesToRead, visAudio + readCursor, 1, outPCM + samplesRead, 1);
samplesRead += samplesToRead;
readCursor += samplesToRead;
samples -= samplesToRead;
if(readCursor >= visAudioSize) readCursor -= visAudioSize;
}
}
fft_calculate(outPCM, outFFT, 2048);
}
@end

4
Cog.xcodeproj/project.pbxproj
View File

@ -126,7 +126,6 @@
8355D6B8180613FB00D05687 /* Security.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 8355D6B7180613FB00D05687 /* Security.framework */; };
8359009D17FF06570060F3ED /* ArchiveSource.bundle in CopyFiles */ = {isa = PBXBuildFile; fileRef = 8359FF3117FEF35D0060F3ED /* ArchiveSource.bundle */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
835A8FD327957310005B3C39 /* json.c in Sources */ = {isa = PBXBuildFile; fileRef = 835A8FC627957310005B3C39 /* json.c */; };
835EDD8A27A000E8001EDCCE /* gsx.wv in Resources */ = {isa = PBXBuildFile; fileRef = 835EDD8027A000E8001EDCCE /* gsx.wv */; };
835F00BB279BD1CD00055FCF /* SecondsFormatter.m in Sources */ = {isa = PBXBuildFile; fileRef = 835F00B8279BD1CD00055FCF /* SecondsFormatter.m */; };
835FAC7927BCDF2A00BA8562 /* AVIFDecoder.m in Sources */ = {isa = PBXBuildFile; fileRef = 835FAC7527BCDF2A00BA8562 /* AVIFDecoder.m */; };
835FAC7E27BCDF5B00BA8562 /* libaom.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 835FAC7C27BCDF5B00BA8562 /* libaom.a */; };
@ -948,7 +947,6 @@
835C888B22CC1881001B4B3F /* en */ = {isa = PBXFileReference; lastKnownFileType = text.html; name = en; path = en.lproj/Credits.html; sourceTree = "<group>"; };
835C888C22CC1882001B4B3F /* en */ = {isa = PBXFileReference; lastKnownFileType = text.plist.strings; name = en; path = en.lproj/InfoPlist.strings; sourceTree = "<group>"; };
835C888D22CC1882001B4B3F /* en */ = {isa = PBXFileReference; lastKnownFileType = text.plist.strings; name = en; path = en.lproj/Localizable.strings; sourceTree = "<group>"; };
835EDD8027A000E8001EDCCE /* gsx.wv */ = {isa = PBXFileReference; lastKnownFileType = file; path = gsx.wv; sourceTree = "<group>"; };
835F00B4279BD1CD00055FCF /* SecondsFormatter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = SecondsFormatter.h; sourceTree = "<group>"; };
835F00B8279BD1CD00055FCF /* SecondsFormatter.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; path = SecondsFormatter.m; sourceTree = "<group>"; };
835FAC6727BCDF2A00BA8562 /* aomdx.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = aomdx.h; sourceTree = "<group>"; };
@ -1545,7 +1543,6 @@
8E7575D909F31E930080F1EE /* Localizable.strings */,
832923AE279FAC400048201E /* Cog.q1.json */,
831B99BE27C23E88005A969B /* Cog.sdef */,
835EDD8027A000E8001EDCCE /* gsx.wv */,
830C379F27B95E3000E02BB0 /* Equalizer.xib */,
17D1B0D00F6320EA00694C57 /* InfoInspector.xib */,
17342A980D5FD20B00E8D854 /* MainMenu.xib */,
@ -2506,7 +2503,6 @@
17818A950C0B27AC001C4916 /* aiff.icns in Resources */,
8384916818083EAB00E7332D /* shuffleOffTemplate.pdf in Resources */,
836F462128207F43005B9B87 /* Plaque.png in Resources */,
835EDD8A27A000E8001EDCCE /* gsx.wv in Resources */,
17818A960C0B27AC001C4916 /* ape.icns in Resources */,
17818A970C0B27AC001C4916 /* m3u.icns in Resources */,
17818A980C0B27AC001C4916 /* mpc.icns in Resources */,

73
Preferences/Preferences/Base.lproj/Preferences.xib
View File

@ -207,11 +207,11 @@
</connections>
</customObject>
<customView id="58" userLabel="OutputView">
<rect key="frame" x="0.0" y="0.0" width="530" height="156"/>
<rect key="frame" x="0.0" y="0.0" width="530" height="108"/>
<autoresizingMask key="autoresizingMask"/>
<subviews>
<popUpButton verticalHuggingPriority="750" fixedFrame="YES" translatesAutoresizingMaskIntoConstraints="NO" id="60">
<rect key="frame" x="144" y="49" width="251" height="26"/>
<rect key="frame" x="144" y="43" width="251" height="26"/>
<autoresizingMask key="autoresizingMask"/>
<popUpButtonCell key="cell" type="push" title="Item1" bezelStyle="rounded" alignment="left" lineBreakMode="truncatingTail" state="on" borderStyle="borderAndBezel" inset="2" arrowPosition="arrowAtCenter" preferredEdge="maxY" selectedItem="62" id="210">
<behavior key="behavior" lightByBackground="YES" lightByGray="YES"/>
@ -232,7 +232,7 @@
</connections>
</popUpButton>
<textField horizontalHuggingPriority="251" verticalHuggingPriority="750" fixedFrame="YES" allowsCharacterPickerTouchBarItem="YES" translatesAutoresizingMaskIntoConstraints="NO" id="zkP-2E-1Kc">
<rect key="frame" x="17" y="22" width="123" height="17"/>
<rect key="frame" x="17" y="20" width="123" height="17"/>
<autoresizingMask key="autoresizingMask" flexibleMaxX="YES" flexibleMinY="YES"/>
<textFieldCell key="cell" scrollable="YES" lineBreakMode="clipping" sendsActionOnEndEditing="YES" alignment="right" title="Volume Level:" id="wK4-EF-8Wa">
<font key="font" metaFont="system"/>
@ -241,7 +241,7 @@
</textFieldCell>
</textField>
<popUpButton verticalHuggingPriority="750" fixedFrame="YES" translatesAutoresizingMaskIntoConstraints="NO" id="2v7-Ef-ekr">
<rect key="frame" x="144" y="16" width="251" height="26"/>
<rect key="frame" x="144" y="14" width="251" height="26"/>
<autoresizingMask key="autoresizingMask" flexibleMaxX="YES" flexibleMinY="YES"/>
<popUpButtonCell key="cell" type="push" bezelStyle="rounded" alignment="left" lineBreakMode="truncatingTail" borderStyle="borderAndBezel" imageScaling="proportionallyDown" inset="2" id="vmS-eb-zen">
<behavior key="behavior" lightByBackground="YES" lightByGray="YES"/>
@ -262,7 +262,7 @@
</connections>
</popUpButton>
<textField verticalHuggingPriority="750" horizontalCompressionResistancePriority="250" fixedFrame="YES" allowsCharacterPickerTouchBarItem="YES" translatesAutoresizingMaskIntoConstraints="NO" id="65">
<rect key="frame" x="17" y="55" width="123" height="17"/>
<rect key="frame" x="17" y="49" width="123" height="17"/>
<autoresizingMask key="autoresizingMask"/>
<textFieldCell key="cell" sendsActionOnEndEditing="YES" alignment="right" title="Output Device: " id="211">
<font key="font" metaFont="system"/>
@ -271,7 +271,7 @@
</textFieldCell>
</textField>
<button verticalHuggingPriority="750" fixedFrame="YES" translatesAutoresizingMaskIntoConstraints="NO" id="DhK-tx-xFv">
<rect key="frame" x="18" y="119" width="227" height="18"/>
<rect key="frame" x="18" y="71" width="227" height="18"/>
<autoresizingMask key="autoresizingMask" flexibleMaxX="YES" flexibleMinY="YES"/>
<buttonCell key="cell" type="check" title="Limit volume control to 100%" bezelStyle="regularSquare" imagePosition="left" state="on" inset="2" id="ds2-aw-ebU">
<behavior key="behavior" changeContents="YES" doesNotDimImage="YES" lightByContents="YES"/>
@ -281,67 +281,8 @@
<binding destination="52" name="value" keyPath="values.volumeLimit" id="7Sl-LJ-ljd"/>
</connections>
</button>
<button verticalHuggingPriority="750" fixedFrame="YES" translatesAutoresizingMaskIntoConstraints="NO" id="1aX-Fo-arv">
<rect key="frame" x="18" y="90" width="20" height="18"/>
<autoresizingMask key="autoresizingMask" flexibleMaxX="YES" flexibleMinY="YES"/>
<buttonCell key="cell" type="check" bezelStyle="regularSquare" imagePosition="left" state="on" inset="2" id="EJb-yF-qRd">
<behavior key="behavior" changeContents="YES" doesNotDimImage="YES" lightByContents="YES"/>
<font key="font" metaFont="system"/>
</buttonCell>
<connections>
<binding destination="52" name="value" keyPath="values.headphoneVirtualization" id="1Mx-dJ-ySQ"/>
</connections>
</button>
<button verticalHuggingPriority="750" fixedFrame="YES" translatesAutoresizingMaskIntoConstraints="NO" id="4qt-T1-7jG" userLabel="Push Button - Select a SoundFont">
<rect key="frame" x="36" y="82" width="73" height="32"/>
<autoresizingMask key="autoresizingMask" flexibleMaxX="YES" flexibleMinY="YES"/>
<buttonCell key="cell" type="push" title="Select" bezelStyle="rounded" alignment="center" borderStyle="border" imageScaling="proportionallyDown" inset="2" id="Otm-xP-EEU" userLabel="Button Cell - Select a SoundFont">
<behavior key="behavior" pushIn="YES" lightByBackground="YES" lightByGray="YES"/>
<font key="font" metaFont="system"/>
</buttonCell>
<connections>
<action selector="setHrir:" target="57" id="3Rw-rb-bBp"/>
</connections>
</button>
<button verticalHuggingPriority="750" fixedFrame="YES" translatesAutoresizingMaskIntoConstraints="NO" id="kOF-Po-oeH" userLabel="Push Button - Select a SoundFont">
<rect key="frame" x="102" y="82" width="43" height="32"/>
<autoresizingMask key="autoresizingMask" flexibleMaxX="YES" flexibleMinY="YES"/>
<buttonCell key="cell" type="push" title="X" bezelStyle="rounded" alignment="center" borderStyle="border" imageScaling="proportionallyDown" inset="2" id="eOk-lh-q1L" userLabel="Button Cell - Select a SoundFont">
<behavior key="behavior" pushIn="YES" lightByBackground="YES" lightByGray="YES"/>
<font key="font" metaFont="system"/>
</buttonCell>
<connections>
<action selector="clearHrir:" target="57" id="tLQ-zH-90Z"/>
</connections>
</button>
<textField horizontalHuggingPriority="251" verticalHuggingPriority="750" fixedFrame="YES" allowsCharacterPickerTouchBarItem="YES" translatesAutoresizingMaskIntoConstraints="NO" id="iZm-aN-M1J" userLabel="Text Field - Selected:">
<rect key="frame" x="145" y="91" width="45" height="17"/>
<autoresizingMask key="autoresizingMask" flexibleMaxX="YES" flexibleMinY="YES"/>
<textFieldCell key="cell" scrollable="YES" lineBreakMode="clipping" sendsActionOnEndEditing="YES" title="HRIR:" id="1Bl-uL-Xda" userLabel="Text Field Cell - Selected">
<font key="font" metaFont="system"/>
<color key="textColor" name="controlTextColor" catalog="System" colorSpace="catalog"/>
<color key="backgroundColor" name="controlColor" catalog="System" colorSpace="catalog"/>
</textFieldCell>
</textField>
<textField horizontalHuggingPriority="251" verticalHuggingPriority="750" fixedFrame="YES" allowsCharacterPickerTouchBarItem="YES" translatesAutoresizingMaskIntoConstraints="NO" id="bsA-1J-aVt">
<rect key="frame" x="187" y="91" width="275" height="17"/>
<autoresizingMask key="autoresizingMask" flexibleMaxX="YES" flexibleMinY="YES"/>
<textFieldCell key="cell" scrollable="YES" lineBreakMode="clipping" sendsActionOnEndEditing="YES" title="Label" id="5vB-dF-Jvk">
<font key="font" metaFont="system"/>
<color key="textColor" name="controlTextColor" catalog="System" colorSpace="catalog"/>
<color key="backgroundColor" name="controlColor" catalog="System" colorSpace="catalog"/>
<connections>
<binding destination="52" name="value" keyPath="values.hrirPath" id="vtm-Ze-9v6">
<dictionary key="options">
<string key="NSNullPlaceholder">Built-in</string>
<string key="NSValueTransformerName">PathToFileTransformer</string>
</dictionary>
</binding>
</connections>
</textFieldCell>
</textField>
</subviews>
<point key="canvasLocation" x="-151" y="303"/>
<point key="canvasLocation" x="-151" y="319"/>
</customView>
<customObject id="i5B-ga-Atm" userLabel="MIDIPane" customClass="MIDIPane">
<connections>

2
Preferences/Preferences/OutputPane.h
View File

@ -15,7 +15,5 @@
}
- (IBAction)takeDeviceID:(id)sender;
- (IBAction)setHrir:(id)sender;
- (IBAction)clearHrir:(id)sender;
@end

31
Preferences/Preferences/OutputPane.m
View File

@ -7,7 +7,6 @@
//
#import "OutputPane.h"
#import "HeadphoneFilter.h"
@implementation OutputPane
@ -26,34 +25,4 @@
[[NSUserDefaults standardUserDefaults] setObject:device forKey:@"outputDevice"];
}
- (IBAction)setHrir:(id)sender {
NSArray *fileTypes = @[@"wav", @"wv"];
NSOpenPanel *panel = [NSOpenPanel openPanel];
[panel setAllowsMultipleSelection:NO];
[panel setCanChooseDirectories:NO];
[panel setCanChooseFiles:YES];
[panel setFloatingPanel:YES];
[panel setAllowedFileTypes:fileTypes];
NSString *oldPath = [[NSUserDefaults standardUserDefaults] stringForKey:@"hrirPath"];
if(oldPath != nil)
[panel setDirectoryURL:[NSURL fileURLWithPath:oldPath]];
NSInteger result = [panel runModal];
if(result == NSModalResponseOK) {
NSString *path = [[panel URL] path];
if([NSClassFromString(@"HeadphoneFilter") validateImpulseFile:[NSURL fileURLWithPath:path]])
[[NSUserDefaults standardUserDefaults] setValue:[[panel URL] path] forKey:@"hrirPath"];
else {
NSAlert *alert = [[NSAlert alloc] init];
[alert setMessageText:@"Invalid impulse"];
[alert setInformativeText:@"The selected file does not conform to the HeSuVi HRIR specification."];
[alert addButtonWithTitle:@"Ok"];
[alert runModal];
}
}
}
- (IBAction)clearHrir:(id)sender {
[[NSUserDefaults standardUserDefaults] setValue:@"" forKey:@"hrirPath"];
}
@end

3
Preferences/Preferences/Preferences.xcodeproj/project.pbxproj
View File

@ -108,7 +108,6 @@
8347435F20E6D5A000063D45 /* es */ = {isa = PBXFileReference; lastKnownFileType = text.plist.strings; name = es; path = es.lproj/Preferences.strings; sourceTree = "<group>"; };
835C888922CC1880001B4B3F /* en */ = {isa = PBXFileReference; lastKnownFileType = text.plist.strings; name = en; path = en.lproj/InfoPlist.strings; sourceTree = "<group>"; };
835C888A22CC1880001B4B3F /* en */ = {isa = PBXFileReference; lastKnownFileType = text.plist.strings; name = en; path = en.lproj/Localizable.strings; sourceTree = "<group>"; };
8363BABF284E450E00E5C9DD /* pffft.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = pffft.h; path = "../../Audio/ThirdParty/r8brain-free-src/pffft.h"; sourceTree = "<group>"; };
83651DA327322C8700A2C097 /* MIDIFlavorBehaviorArrayController.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; path = MIDIFlavorBehaviorArrayController.m; sourceTree = "<group>"; };
83651DA427322C8700A2C097 /* MIDIFlavorBehaviorArrayController.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = MIDIFlavorBehaviorArrayController.h; sourceTree = "<group>"; };
8372053518E3DEAF007EFAD4 /* ResamplerBehaviorArrayController.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ResamplerBehaviorArrayController.h; sourceTree = "<group>"; };
@ -127,7 +126,6 @@
83B06728180D85B8008E3612 /* MIDIPane.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; path = MIDIPane.m; sourceTree = "<group>"; };
83B0672A180D8B39008E3612 /* midi.png */ = {isa = PBXFileReference; lastKnownFileType = image.png; name = midi.png; path = Icons/midi.png; sourceTree = "<group>"; };
83BC5AB320E4C90F00631CD4 /* Base */ = {isa = PBXFileReference; lastKnownFileType = file.xib; name = Base; path = Base.lproj/Preferences.xib; sourceTree = "<group>"; };
83D34B7D27A10FA700784D34 /* HeadphoneFilter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = HeadphoneFilter.h; path = ../../Audio/Chain/HeadphoneFilter.h; sourceTree = "<group>"; };
83EF495D17FBC96A00642E3C /* VolumeBehaviorArrayController.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = VolumeBehaviorArrayController.h; sourceTree = "<group>"; };
83EF495E17FBC96A00642E3C /* VolumeBehaviorArrayController.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; path = VolumeBehaviorArrayController.m; sourceTree = "<group>"; };
83F27E651810DD3A00CEF538 /* appearance@2x.png */ = {isa = PBXFileReference; lastKnownFileType = image.png; name = "appearance@2x.png"; path = "Icons/appearance@2x.png"; sourceTree = "<group>"; };
@ -208,7 +206,6 @@
8307D30828604ECF000FF8EB /* PlaylistController.h */,
83F27E711810E41A00CEF538 /* Transformers */,
8384913618081ECB00E7332D /* Logging.h */,
83D34B7D27A10FA700784D34 /* HeadphoneFilter.h */,
17D503410ABDB1660022D1E8 /* Custom */,
17D5033F0ABDB1570022D1E8 /* Panes */,
17D1B3F60F6349CE00694C57 /* PreferencePanePlugin.h */,

44
README.md
View File

@ -74,47 +74,3 @@ git config core.hooksPath .githooks
## Mini window
![Mini window](https://github.com/losnoco/Cog/blob/main/.github/images/MiniWindow.png)
ADDENDUM - 2022-01-25
I've added an HRIR convolver, based on the HeSuVi HRIR format. A preset is
bundled with the app, but external presets may be selected, in either RIFF
WAVE, or Wavpack format, 7 or 14 channel, 32 bit floating point. A bad or
missing preset will fall back on the built-in preset. The settings dialog
won't let you assign a bad preset, either.
The channel layout of the HeSuVi HRIR is as follows:
![HeSuVi HRIR channel layout](https://github.com/losnoco/Cog/blob/main/.github/images/HeSuVi-channel-map.png)
I altered this image from the original on the HeSuVi HRIR developer readme,
as their image incorrectly swapped the left/right output order for the right
input channels, which are actually supposed to be right output first, then
left output. Otherwise, if observed as the original image, all output comes
from either the center or the left side.
The layout for 7 channel no-reverb presets is the same, except it only
includes the first 7 channels of the layout, and the opposite speakers
simply mirror their respective input channels, and the center input is
used for both left and right outputs.
[Here are some preset HRIR files](https://cogcdn.cog.losno.co/HeSuVi-hrir-basic.7z).
The original HeSuVi project is located [here](https://sourceforge.net/projects/hesuvi/).
ADDENDUM - 2022-02-01
I've replaced the resampler with libsoxr, with no quality control settings exposed.
ADDENDUM - 2022-03-03
I've replaced the resampler yet again. This time, with [R8Brain](https://github.com/avaneev/r8brain-free-src), using the 24 bit preset, since the player doesn't go over the depth of 24 bit integer or 32 bit floating point, and outputs 32 bit floating point. It still processes the resampler in 64 bits, though.
ADDENDUM - 2022-03-04
It has come to my attention that my recent commits may have assumed that Aleksey Vaneev, and possibly any other Russian or Belarusian people are in support of the war in Ukraine. I did not intend to make that assumption. I know that a lot of people are either against it, have family and/or friends affected by it, or in some unfortunate cases, have the wool pulled over their eyes by the local propaganda machine. The only thing I knew at all, was comments to the effect of him being anti-vaccines, which is a bit of a sore subject with me. I do not know anyone personally who has died from COVID-19, but I do support using the greatest caution to avoid it, including vaccines, and masks, and social distancing. I will maintain my fork of the library in case anything happens to upstream, and keep an open mind for now, I hope. Hope this long winded mess finds my users well, and doesn't further piss off anyone.

1
Visualization/SpectrumViewSK.m
View File

@ -326,6 +326,7 @@ extern NSString *CogPlaybackDidStopNotficiation;
- (void)playbackDidStop:(NSNotification *)notification {
stopped = YES;
paused = NO;
bandsReset = NO;
[self updateVisListening];
[self repaint];
}

BIN
gsx.wv
View File

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