Update of /cvsroot/audacity/audacity-src/src/effects/audiounits
In directory
23jxhf1.ch3.sourceforge.com:/tmp/cvs-serv21252/src/effects/audiounits
Modified Files:
AudioUnitEffect.cpp AudioUnitEffect.h
Log Message:
Allow more AudioUnit effects like Guitar Rig to work by using a single
component.
Remove unnecessary buffer copies.
Index: AudioUnitEffect.h
===================================================================
RCS file:
/cvsroot/audacity/audacity-src/src/effects/audiounits/AudioUnitEffect.h,v
retrieving revision 1.5
retrieving revision 1.6
diff -u -d -r1.5 -r1.6
--- AudioUnitEffect.h 20 May 2009 06:22:49 -0000 1.5
+++ AudioUnitEffect.h 26 Jul 2009 09:16:09 -0000 1.6
@@ -53,7 +53,8 @@
bool DoRender(AudioUnit unit, int numChannels,
float *leftBuffer, float *rightBuffer,
int len, int unitBlockSize,
- AudioTimeStamp *timeStamp);
+ AudioTimeStamp *timeStamp,
+ AudioBufferList *bufferList);
bool CopyParameters(AudioUnit srcUnit, AudioUnit dstUnit);
Index: AudioUnitEffect.cpp
===================================================================
RCS file:
/cvsroot/audacity/audacity-src/src/effects/audiounits/AudioUnitEffect.cpp,v
retrieving revision 1.15
retrieving revision 1.16
diff -u -d -r1.15 -r1.16
--- AudioUnitEffect.cpp 20 May 2009 06:22:49 -0000 1.15
+++ AudioUnitEffect.cpp 26 Jul 2009 09:16:09 -0000 1.16
@@ -96,17 +96,16 @@
mComponent(component)
{
SetEffectFlags(PLUGIN_EFFECT | PROCESS_EFFECT);
- OSErr result;
mUnit = NULL;
- result = OpenAComponent(mComponent, &mUnit);
- if (result != 0)
- return;
+ OpenAComponent(mComponent, &mUnit);
}
AudioUnitEffect::~AudioUnitEffect()
{
- CloseComponent(mUnit);
+ if (mUnit) {
+ CloseComponent(mUnit);
+ }
}
wxString AudioUnitEffect::GetEffectName()
@@ -142,7 +141,9 @@
bool AudioUnitEffect::Init()
{
- ComponentResult auResult;
+ if (!mUnit) {
+ return false;
+ }
mSupportsMono = SetRateAndChannels(mUnit, 1, mProjectRate);
mSupportsStereo = SetRateAndChannels(mUnit, 2, mProjectRate);
@@ -158,12 +159,6 @@
}
}
- auResult = AudioUnitInitialize(mUnit);
- if (auResult != 0) {
- printf("Unable to initialize\n");
- return false;
- }
-
return true;
}
@@ -235,7 +230,6 @@
void AudioUnitEffect::End()
{
- AudioUnitUninitialize(mUnit);
}
bool AudioUnitEffect::SetRateAndChannels(AudioUnit unit,
@@ -256,7 +250,7 @@
streamFormat.mSampleRate = sampleRate;
streamFormat.mFormatID = kAudioFormatLinearPCM;
streamFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked |
- kAudioFormatFlagIsNonInterleaved;
+ kAudioFormatFlagIsNonInterleaved;
streamFormat.mBitsPerChannel = 32;
streamFormat.mChannelsPerFrame = numChannels;
streamFormat.mFramesPerPacket = 1;
@@ -355,35 +349,23 @@
sampleCount rstart,
sampleCount len)
{
- int numChannels = (right != NULL? 2: 1);
+ int numChannels = (right != NULL ? 2 : 1);
Float64 sampleRate = left->GetRate();
- AudioUnit trackUnit;
+ AudioUnit trackUnit = mUnit;
AURenderCallbackStruct callbackStruct;
AudioTimeStamp timeStamp;
+ AudioBufferList *bufferList = NULL;
ComponentResult auResult;
int waveTrackBlockSize;
- UInt32 size, unitBlockSize;
- float *leftBuffer, *rightBuffer;
-
- // Audio Units cannot have their rate and number of channels set
- // after they've been initialized. So, we open a new audio unit
- // for each track (or pair of tracks) and copy the parameters
- // over. The only area where this might sometimes present a
- // problem is when the parameter is tied to the sample rate, and
- // the sample rate for a track is different than the project sample
- // rate. The user can always work around this by making the track
- // and project sample rates match temporarily, though.
-
- auResult = OpenAComponent(mComponent, &trackUnit);
- if (auResult != 0) {
- printf("Couldn't open audio unit\n");
- return false;
- }
+ UInt32 size;
+ UInt32 unitBlockSize;
+ float *leftBuffer;
+ float *rightBuffer;
+ bool success = true;
if (!SetRateAndChannels(trackUnit, numChannels, sampleRate)) {
printf("Unable to setup audio unit for channels=%d rate=%.1f\n",
numChannels, sampleRate);
- CloseComponent(trackUnit);
return false;
}
@@ -395,6 +377,7 @@
0,
&unitBlockSize,
&size);
+
if (unitBlockSize == 0 || auResult != 0) {
printf("Warning: didn't get audio unit's MaximumFramesPerSlice\n");
printf("Trying to set MaximumFramesPerSlice to 512\n");
@@ -412,13 +395,6 @@
auResult = AudioUnitInitialize(trackUnit);
if (auResult != 0) {
printf("Couldn't initialize audio unit\n");
- CloseComponent(trackUnit);
- return false;
- }
-
- if (!CopyParameters(mUnit, trackUnit)) {
- AudioUnitUninitialize(trackUnit);
- CloseComponent(trackUnit);
return false;
}
@@ -426,7 +402,6 @@
if (auResult != 0) {
printf("Reset failed.\n");
AudioUnitUninitialize(trackUnit);
- CloseComponent(trackUnit);
return false;
}
@@ -441,7 +416,6 @@
if (auResult != 0) {
printf("Setting input render callback failed.\n");
AudioUnitUninitialize(trackUnit);
- CloseComponent(trackUnit);
return false;
}
@@ -452,130 +426,136 @@
waveTrackBlockSize = left->GetMaxBlockSize() * 2;
- leftBuffer = rightBuffer = NULL;
- mLeftBufferForCallback = mRightBufferForCallback = NULL;
+ leftBuffer = NULL;
+ rightBuffer = NULL;
+ mLeftBufferForCallback = NULL;
+ mRightBufferForCallback = NULL;
if (left) {
leftBuffer = new float[waveTrackBlockSize];
- mLeftBufferForCallback = new float[unitBlockSize];
}
+
if (right) {
rightBuffer = new float[waveTrackBlockSize];
- mRightBufferForCallback = new float[unitBlockSize];
}
+ bufferList = (AudioBufferList *)malloc(sizeof(UInt32) +
+ numChannels * sizeof(AudioBuffer));
+ bufferList->mNumberBuffers = numChannels;
+
sampleCount originalLen = len;
sampleCount ls = lstart;
sampleCount rs = rstart;
while (len) {
int block = waveTrackBlockSize;
- if (block > len)
+ if (block > len) {
block = len;
+ }
- if (left)
+ if (left) {
left->Get((samplePtr)leftBuffer, floatSample, ls, block);
- if (right)
+ }
+
+ if (right) {
right->Get((samplePtr)rightBuffer, floatSample, rs, block);
+ }
- if (!DoRender(trackUnit, numChannels, leftBuffer, rightBuffer,
- block, unitBlockSize, &timeStamp)) {
- if (leftBuffer) {
- delete[] leftBuffer;
- delete[] mLeftBufferForCallback;
- }
- if (rightBuffer) {
- delete[] rightBuffer;
- delete[] mRightBufferForCallback;
- }
- AudioUnitUninitialize(trackUnit);
- CloseComponent(trackUnit);
- return false;
+ success = DoRender(trackUnit, numChannels, leftBuffer, rightBuffer,
+ block, unitBlockSize, &timeStamp, bufferList);
+ if (!success) {
+ break;
}
- if (left)
+ if (left) {
left->Set((samplePtr)leftBuffer, floatSample, ls, block);
- if (right)
+ }
+
+ if (right) {
right->Set((samplePtr)rightBuffer, floatSample, rs, block);
+ }
len -= block;
ls += block;
rs += block;
if (left && right) {
- if (TrackGroupProgress(count, (ls-lstart)/(double)originalLen))
- return false;
+ if (TrackGroupProgress(count, (ls-lstart)/(double)originalLen)) {
+ success = false;
+ break;
+ }
}
else {
- if (TrackProgress(count, (ls-lstart)/(double)originalLen))
- return false;
+ if (TrackProgress(count, (ls-lstart)/(double)originalLen)) {
+ success = false;
+ break;
+ }
}
}
+ if (bufferList) {
+ free(bufferList);
+ }
+
if (leftBuffer) {
delete[] leftBuffer;
- delete[] mLeftBufferForCallback;
}
+
if (rightBuffer) {
delete[] rightBuffer;
- delete[] mRightBufferForCallback;
}
+
AudioUnitUninitialize(trackUnit);
- CloseComponent(trackUnit);
- return true;
+ return success;
}
bool AudioUnitEffect::DoRender(AudioUnit unit,
int numChannels,
- float *leftBuffer, float *rightBuffer,
+ float *leftBuffer,
+ float *rightBuffer,
int len,
int unitBlockSize,
- AudioTimeStamp *timeStamp)
+ AudioTimeStamp *timeStamp,
+ AudioBufferList *bufferList)
{
- AudioBufferList *bufferList;
AudioUnitRenderActionFlags flags;
ComponentResult auResult;
- int i, j, block;
+ int block;
- bufferList = (AudioBufferList *)malloc(sizeof(UInt32) +
- numChannels * sizeof(AudioBuffer));
- bufferList->mNumberBuffers = numChannels;
-
- i = 0;
- while(i < len) {
+ for (int i = 0; i < len; i += block) {
block = unitBlockSize;
- if (i + block > len)
+ if (i + block > len) {
block = len - i;
+ }
- if (leftBuffer)
- memcpy(mLeftBufferForCallback, &leftBuffer[i], block*sizeof(float));
- if (rightBuffer)
- memcpy(mRightBufferForCallback, &rightBuffer[i], block*sizeof(float));
-
- flags = 0;
- for(j=0; j<numChannels; j++) {
+ for (int j = 0; j<numChannels; j++) {
bufferList->mBuffers[j].mNumberChannels = 1;
bufferList->mBuffers[j].mData = NULL;
bufferList->mBuffers[j].mDataByteSize = sizeof(float) * block;
}
+
+ if (leftBuffer) {
+ mLeftBufferForCallback = &leftBuffer[i];
+ bufferList->mBuffers[0].mData = mLeftBufferForCallback;
+ }
+
+ if (rightBuffer) {
+ mRightBufferForCallback = &rightBuffer[i];
+ bufferList->mBuffers[1].mData = mRightBufferForCallback;
+ }
+
+ flags = 0;
+
auResult = AudioUnitRender(unit, &flags, timeStamp,
0, block, bufferList);
if (auResult != 0) {
printf("Render failed: %d %4.4s\n", (int)auResult, (char *)&auResult);
- free(bufferList);
return false;
}
- if (leftBuffer)
- memcpy(&leftBuffer[i], bufferList->mBuffers[0].mData,
block*sizeof(float));
- if (rightBuffer)
- memcpy(&rightBuffer[i], bufferList->mBuffers[1].mData,
block*sizeof(float));
-
timeStamp->mSampleTime += block;
- i += block;
}
- free(bufferList);
return true;
}
@@ -590,10 +570,13 @@
{
AudioUnitEffect *This = (AudioUnitEffect *)inRefCon;
- if (This->mLeftBufferForCallback)
+ if (This->mLeftBufferForCallback) {
ioData->mBuffers[0].mData = This->mLeftBufferForCallback;
- if (This->mRightBufferForCallback)
+ }
+
+ if (This->mRightBufferForCallback) {
ioData->mBuffers[1].mData = This->mRightBufferForCallback;
+ }
return 0;
}
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