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Core Audio: Fix some bugs with capture mode.

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This commit is contained in:
David Reid
2020-11-05 20:19:25 +10:00
parent bda207bc87
commit 11811d29f3
1 changed files with 34 additions and 9 deletions
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miniaudio.h
+34 -9
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@@ -3821,7 +3821,8 @@ struct ma_device
ma_uint32 deviceObjectIDCapture;
/*AudioUnit*/ ma_ptr audioUnitPlayback;
/*AudioUnit*/ ma_ptr audioUnitCapture;
/*AudioBufferList**/ ma_ptr pAudioBufferList; /* Only used for input devices. */
/*AudioBufferList**/ ma_ptr pAudioBufferList; /* Only used for input devices. */
ma_uint32 audioBufferSizeInBytes; /* Only used for input devices. The size in bytes of each buffer in pAudioBufferList. */
ma_event stopEvent;
ma_uint32 originalPeriodSizeInFrames;
ma_uint32 originalPeriodSizeInMilliseconds;
@@ -24269,6 +24270,7 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla
ma_device* pDevice = (ma_device*)pUserData;
AudioBufferList* pRenderedBufferList;
ma_stream_layout layout;
ma_uint32 iBuffer;
OSStatus status;
MA_ASSERT(pDevice != NULL);
@@ -24285,6 +24287,18 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla
#if defined(MA_DEBUG_OUTPUT)
printf("INFO: Input Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", busNumber, frameCount, pRenderedBufferList->mNumberBuffers);
#endif
/*
When you call AudioUnitRender(), Core Audio tries to be helpful by setting the mDataByteSize to the number of bytes
that were actually rendered. The problem with this is that the next call can fail with -50 due to the size no longer
being set to the capacity of the buffer, but instead the size in bytes of the previous render. This will cause a
problem when a future call to this callback specifies a larger number of frames.
To work around this we need to explicitly set the size of each buffer to their respective size in bytes.
*/
for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) {
pRenderedBufferList->mBuffers[iBuffer].mDataByteSize = pDevice->coreaudio.audioBufferSizeInBytes;
}
status = ((ma_AudioUnitRender_proc)pDevice->pContext->coreaudio.AudioUnitRender)((AudioUnit)pDevice->coreaudio.audioUnitCapture, pActionFlags, pTimeStamp, busNumber, frameCount, pRenderedBufferList);
if (status != noErr) {
@@ -24295,7 +24309,6 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla
}
if (layout == ma_stream_layout_interleaved) {
UInt32 iBuffer;
for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) {
if (pRenderedBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->capture.internalChannels) {
if (pDevice->type == ma_device_type_duplex) {
@@ -24347,7 +24360,6 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla
*/
if ((pRenderedBufferList->mNumberBuffers % pDevice->capture.internalChannels) == 0) {
ma_uint8 tempBuffer[4096];
UInt32 iBuffer;
for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; iBuffer += pDevice->capture.internalChannels) {
ma_uint32 framesRemaining = frameCount;
while (framesRemaining > 0) {
@@ -24825,6 +24837,7 @@ typedef struct
AudioComponent component;
AudioUnit audioUnit;
AudioBufferList* pAudioBufferList; /* Only used for input devices. */
ma_uint32 audioBufferSizeInBytes;
ma_format formatOut;
ma_uint32 channelsOut;
ma_uint32 sampleRateOut;
@@ -24841,6 +24854,7 @@ static ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_dev
UInt32 enableIOFlag;
AudioStreamBasicDescription bestFormat;
ma_uint32 actualPeriodSizeInFrames;
ma_uint32 actualPeriodSizeInFramesSize = sizeof(actualPeriodSizeInFrames);
AURenderCallbackStruct callbackInfo;
#if defined(MA_APPLE_DESKTOP)
AudioObjectID deviceObjectID;
@@ -25087,11 +25101,16 @@ static ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_dev
if (result != MA_SUCCESS) {
return result;
}
pData->periodSizeInFramesOut = actualPeriodSizeInFrames;
#else
actualPeriodSizeInFrames = 2048;
pData->periodSizeInFramesOut = actualPeriodSizeInFrames;
/*
I don't know how to configure buffer sizes on iOS so for now we're not allowing it to be configured. Instead we're
just going to set it to the value of kAudioUnitProperty_MaximumFramesPerSlice.
*/
status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &actualPeriodSizeInFrames, &actualPeriodSizeInFramesSize);
if (status != noErr) {
((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
return ma_result_from_OSStatus(status);
}
#endif
@@ -25120,6 +25139,8 @@ static ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_dev
}
}
pData->periodSizeInFramesOut = actualPeriodSizeInFrames;
/* We need a buffer list if this is an input device. We render into this in the input callback. */
if (deviceType == ma_device_type_capture) {
ma_bool32 isInterleaved = (bestFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved) == 0;
@@ -25144,16 +25165,18 @@ static ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_dev
}
if (isInterleaved) {
pData->audioBufferSizeInBytes = actualPeriodSizeInFrames * ma_get_bytes_per_frame(pData->formatOut, pData->channelsOut);
pBufferList->mNumberBuffers = 1;
pBufferList->mBuffers[0].mNumberChannels = pData->channelsOut;
pBufferList->mBuffers[0].mDataByteSize = actualPeriodSizeInFrames * ma_get_bytes_per_frame(pData->formatOut, pData->channelsOut);
pBufferList->mBuffers[0].mDataByteSize = pData->audioBufferSizeInBytes;
pBufferList->mBuffers[0].mData = (ma_uint8*)pBufferList + sizeof(AudioBufferList);
} else {
ma_uint32 iBuffer;
pData->audioBufferSizeInBytes = actualPeriodSizeInFrames * ma_get_bytes_per_sample(pData->formatOut);
pBufferList->mNumberBuffers = pData->channelsOut;
for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; ++iBuffer) {
pBufferList->mBuffers[iBuffer].mNumberChannels = 1;
pBufferList->mBuffers[iBuffer].mDataByteSize = actualPeriodSizeInFrames * ma_get_bytes_per_sample(pData->formatOut);
pBufferList->mBuffers[iBuffer].mDataByteSize = pData->audioBufferSizeInBytes;
pBufferList->mBuffers[iBuffer].mData = (ma_uint8*)pBufferList + ((sizeof(AudioBufferList) - sizeof(AudioBuffer)) + (sizeof(AudioBuffer) * pData->channelsOut)) + (actualPeriodSizeInFrames * ma_get_bytes_per_sample(pData->formatOut) * iBuffer);
}
}
@@ -25278,6 +25301,7 @@ static ma_result ma_device_reinit_internal__coreaudio(ma_device* pDevice, ma_dev
#endif
pDevice->coreaudio.audioUnitCapture = (ma_ptr)data.audioUnit;
pDevice->coreaudio.pAudioBufferList = (ma_ptr)data.pAudioBufferList;
pDevice->coreaudio.audioBufferSizeInBytes = data.audioBufferSizeInBytes;
pDevice->capture.internalFormat = data.formatOut;
pDevice->capture.internalChannels = data.channelsOut;
@@ -25354,6 +25378,7 @@ static ma_result ma_device_init__coreaudio(ma_context* pContext, const ma_device
#endif
pDevice->coreaudio.audioUnitCapture = (ma_ptr)data.audioUnit;
pDevice->coreaudio.pAudioBufferList = (ma_ptr)data.pAudioBufferList;
pDevice->coreaudio.audioBufferSizeInBytes = data.audioBufferSizeInBytes;
pDevice->capture.internalFormat = data.formatOut;
pDevice->capture.internalChannels = data.channelsOut;