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[Visualization] Resample all visualizer audio 

Visualizer audio is now resampled to 44100 Hz, for consistency across
the system.

Signed-off-by: Christopher Snowhill <kode54@gmail.com>
lastfm
Christopher Snowhill 2022-06-24 03:43:50 -07:00
parent 2e45deb8d3
commit 3a9db65b53
3 changed files with 75 additions and 12 deletions
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6
Audio/Output/OutputAVFoundation.h
View File

@ -36,9 +36,12 @@ using std::atomic_long;
@interface OutputAVFoundation : NSObject { @interface OutputAVFoundation : NSObject {
OutputNode *outputController; OutputNode *outputController;
BOOL r8bFlushing, r8bFlushed, r8bDone; BOOL r8bDone;
void *r8bstate, *r8bold; void *r8bstate, *r8bold;
void *r8bvis;
double lastVisRate;
BOOL stopInvoked; BOOL stopInvoked;
BOOL running; BOOL running;
BOOL stopping; BOOL stopping;
@ -46,6 +49,7 @@ using std::atomic_long;
BOOL started; BOOL started;
BOOL paused; BOOL paused;
BOOL restarted; BOOL restarted;
BOOL stopFlush;
BOOL eqEnabled; BOOL eqEnabled;
BOOL eqInitialized; BOOL eqInitialized;

74
Audio/Output/OutputAVFoundation.m
View File

@ -63,7 +63,6 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
amountToRead = 1024; amountToRead = 1024;
int visTabulated = 0;
float visAudio[amountToRead]; // Chunk size float visAudio[amountToRead]; // Chunk size
if(stopping == YES || [outputController shouldContinue] == NO) { if(stopping == YES || [outputController shouldContinue] == NO) {
@ -145,8 +144,65 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
[downmixerForVis process:outputPtr [downmixerForVis process:outputPtr
frameCount:frameCount frameCount:frameCount
output:visAudio]; output:&visAudio[0]];
visTabulated += frameCount;
[visController postSampleRate:44100.0];
float visTemp[8192];
if(newFormat.mSampleRate != 44100.0) {
if(newFormat.mSampleRate != lastVisRate) {
if(r8bvis) {
for(;;) {
int samplesFlushed;
[currentPtsLock lock];
samplesFlushed = (int)r8bstate_flush(r8bvis, &visTemp[0], 8192);
[currentPtsLock unlock];
if(samplesFlushed) {
[visController postVisPCM:visTemp amount:samplesFlushed];
} else {
break;
}
}
[currentPtsLock lock];
r8bstate_delete(r8bvis);
r8bvis = NULL;
[currentPtsLock unlock];
}
lastVisRate = newFormat.mSampleRate;
[currentPtsLock lock];
r8bvis = r8bstate_new(1, 1024, lastVisRate, 44100.0);
[currentPtsLock unlock];
}
if(r8bvis) {
size_t inDone = 0;
int samplesProcessed;
[currentPtsLock lock];
samplesProcessed = (int)r8bstate_resample(r8bvis, &visAudio[0], frameCount, &inDone, &visTemp[0], 8192);
[currentPtsLock unlock];
if(samplesProcessed) {
[visController postVisPCM:&visTemp[0] amount:samplesProcessed];
}
}
} else if(r8bvis) {
for(;;) {
int samplesFlushed;
[currentPtsLock lock];
samplesFlushed = (int)r8bstate_flush(r8bvis, &visTemp[0], 8192);
[currentPtsLock unlock];
if(samplesFlushed) {
[visController postVisPCM:visTemp amount:samplesFlushed];
} else {
break;
}
}
[currentPtsLock lock];
r8bstate_delete(r8bvis);
r8bvis = NULL;
[currentPtsLock unlock];
[visController postVisPCM:&visAudio[0] amount:frameCount];
} else {
[visController postVisPCM:&visAudio[0] amount:frameCount];
}
cblas_scopy((int)(frameCount * newFormat.mChannelsPerFrame), outputPtr, 1, &inputBuffer[0], 1); cblas_scopy((int)(frameCount * newFormat.mChannelsPerFrame), outputPtr, 1, &inputBuffer[0], 1);
amountRead = frameCount; amountRead = frameCount;
@ -178,9 +234,6 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
scale_by_volume(&inputBuffer[0], amountRead * newFormat.mChannelsPerFrame, volumeScale); scale_by_volume(&inputBuffer[0], amountRead * newFormat.mChannelsPerFrame, volumeScale);
[visController postSampleRate:newFormat.mSampleRate];
[visController postVisPCM:visAudio amount:visTabulated];
return amountRead; return amountRead;
} }
@ -750,12 +803,13 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
downmixerForVis = nil; downmixerForVis = nil;
r8bFlushing = NO;
r8bFlushed = NO;
r8bDone = NO; r8bDone = NO;
r8bstate = NULL; r8bstate = NULL;
r8bold = NULL; r8bold = NULL;
r8bvis = NULL;
lastVisRate = 44100.0;
AudioComponentDescription desc; AudioComponentDescription desc;
NSError *err; NSError *err;
@ -999,6 +1053,10 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
r8bstate_delete(r8bold); r8bstate_delete(r8bold);
r8bold = NULL; r8bold = NULL;
} }
if(r8bvis) {
r8bstate_delete(r8bvis);
r8bvis = NULL;
}
} }
} }

7
Audio/ThirdParty/r8bstate.hpp vendored
View File

@ -21,11 +21,12 @@ struct r8bstate {
uint64_t inProcessed; uint64_t inProcessed;
uint64_t outProcessed; uint64_t outProcessed;
double sampleRatio; double sampleRatio;
double dstRate;
r8b::CFixedBuffer<double> InBuf; r8b::CFixedBuffer<double> InBuf;
r8b::CFixedBuffer<double> *OutBufs; r8b::CFixedBuffer<double> *OutBufs;
r8b::CDSPResampler24 **Resamps; r8b::CDSPResampler24 **Resamps;
r8bstate(int _channelCount, int _bufferCapacity, double srcRate, double dstRate) r8bstate(int _channelCount, int _bufferCapacity, double srcRate, double _dstRate)
: channelCount(_channelCount), bufferCapacity(_bufferCapacity), inProcessed(0), outProcessed(0), remainder(0) { : channelCount(_channelCount), bufferCapacity(_bufferCapacity), inProcessed(0), outProcessed(0), remainder(0), dstRate(_dstRate) {
InBuf.alloc(bufferCapacity); InBuf.alloc(bufferCapacity);
OutBufs = new r8b::CFixedBuffer<double>[channelCount]; OutBufs = new r8b::CFixedBuffer<double>[channelCount];
Resamps = new r8b::CDSPResampler24 *[channelCount]; Resamps = new r8b::CDSPResampler24 *[channelCount];
@ -44,7 +45,7 @@ struct r8bstate {
} }
double latency() { double latency() {
return ((double)inProcessed * sampleRatio) - (double)outProcessed; return (((double)inProcessed * sampleRatio) - (double)outProcessed) / dstRate;
} }
int resample(const float *input, size_t inCount, size_t *inDone, float *output, size_t outMax) { int resample(const float *input, size_t inCount, size_t *inDone, float *output, size_t outMax) {