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Updated mini_al to latest dev version

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Corrects issue with audio on RPI
This commit is contained in:
Ray 2018-07-11 16:39:26 +02:00
parent 414bb6018b
commit 43178f3488
1 changed files with 101 additions and 48 deletions
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149
src/external/mini_al.h vendored
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@ -4263,7 +4263,7 @@ static MAL_INLINE void mal_device__send_frames_to_client(mal_device* pDevice, ma
pDevice->_dspFrames = (const mal_uint8*)pSamples;
mal_uint8 chunkBuffer[4096];
mal_uint32 chunkFrameCount = sizeof(chunkBuffer) / mal_get_bytes_per_sample(pDevice->format) / pDevice->channels;
mal_uint32 chunkFrameCount = sizeof(chunkBuffer) / mal_get_bytes_per_frame(pDevice->format, pDevice->channels);
for (;;) {
mal_uint32 framesJustRead = (mal_uint32)mal_dsp_read(&pDevice->dsp, chunkFrameCount, chunkBuffer, pDevice->dsp.pUserData);
@ -11337,7 +11337,6 @@ done:
return result;
}
mal_result mal_context_init__pulse(mal_context* pContext)
{
mal_assert(pContext != NULL);
@ -11489,6 +11488,36 @@ mal_result mal_context_init__pulse(mal_context* pContext)
pContext->onEnumDevices = mal_context_enumerate_devices__pulse;
pContext->onGetDeviceInfo = mal_context_get_device_info__pulse;
// Although we have found the libpulse library, it doesn't necessarily mean PulseAudio is useable. We need to initialize
// and connect a dummy PulseAudio context to test PulseAudio's usability.
mal_pa_mainloop* pMainLoop = ((mal_pa_mainloop_new_proc)pContext->pulse.pa_mainloop_new)();
if (pMainLoop == NULL) {
return MAL_NO_BACKEND;
}
mal_pa_mainloop_api* pAPI = ((mal_pa_mainloop_get_api_proc)pContext->pulse.pa_mainloop_get_api)(pMainLoop);
if (pAPI == NULL) {
((mal_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop);
return MAL_NO_BACKEND;
}
mal_pa_context* pPulseContext = ((mal_pa_context_new_proc)pContext->pulse.pa_context_new)(pAPI, pContext->config.pulse.pApplicationName);
if (pPulseContext == NULL) {
((mal_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop);
return MAL_NO_BACKEND;
}
int error = ((mal_pa_context_connect_proc)pContext->pulse.pa_context_connect)(pPulseContext, pContext->config.pulse.pServerName, 0, NULL);
if (error != MAL_PA_OK) {
((mal_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext);
((mal_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop);
return MAL_NO_BACKEND;
}
((mal_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)(pPulseContext);
((mal_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext);
((mal_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop);
return MAL_SUCCESS;
}
@ -12147,14 +12176,11 @@ typedef const char* (* mal_jack_port_name_proc) (const mal_
typedef void* (* mal_jack_port_get_buffer_proc) (mal_jack_port_t* port, mal_jack_nframes_t nframes);
typedef void (* mal_jack_free_proc) (void* ptr);
mal_result mal_context_open_client__jack(mal_context* pContext, mal_device_type type, const mal_device_id* pDeviceID, mal_jack_client_t** ppClient)
mal_result mal_context_open_client__jack(mal_context* pContext, mal_jack_client_t** ppClient)
{
mal_assert(pContext != NULL);
mal_assert(ppClient != NULL);
(void)type;
(void)pDeviceID;
if (ppClient) {
*ppClient = NULL;
}
@ -12167,7 +12193,7 @@ mal_result mal_context_open_client__jack(mal_context* pContext, mal_device_type
mal_jack_status_t status;
mal_jack_client_t* pClient = ((mal_jack_client_open_proc)pContext->jack.jack_client_open)(clientName, (pContext->config.jack.tryStartServer) ? 0 : mal_JackNoStartServer, &status, NULL);
if (pClient == NULL) {
return mal_context_post_error(pContext, NULL, MAL_LOG_LEVEL_ERROR, "[JACK] Failed to open client.", MAL_FAILED_TO_OPEN_BACKEND_DEVICE);
return MAL_FAILED_TO_OPEN_BACKEND_DEVICE;
}
if (ppClient) {
@ -12237,9 +12263,9 @@ mal_result mal_context_get_device_info__jack(mal_context* pContext, mal_device_t
// The channel count and sample rate can only be determined by opening the device.
mal_jack_client_t* pClient;
mal_result result = mal_context_open_client__jack(pContext, deviceType, pDeviceID, &pClient);
mal_result result = mal_context_open_client__jack(pContext, &pClient);
if (result != MAL_SUCCESS) {
return result;
return mal_context_post_error(pContext, NULL, MAL_LOG_LEVEL_ERROR, "[JACK] Failed to open client.", MAL_FAILED_TO_OPEN_BACKEND_DEVICE);
}
pDeviceInfo->minSampleRate = ((mal_jack_get_sample_rate_proc)pContext->jack.jack_get_sample_rate)((mal_jack_client_t*)pClient);
@ -12349,6 +12375,16 @@ mal_result mal_context_init__jack(mal_context* pContext)
pContext->onEnumDevices = mal_context_enumerate_devices__jack;
pContext->onGetDeviceInfo = mal_context_get_device_info__jack;
// Getting here means the JACK library is installed, but it doesn't necessarily mean it's usable. We need to quickly test this by connecting
// a temporary client.
mal_jack_client_t* pDummyClient;
mal_result result = mal_context_open_client__jack(pContext, &pDummyClient);
if (result != MAL_SUCCESS) {
return MAL_NO_BACKEND;
}
((mal_jack_client_close_proc)pContext->jack.jack_client_close)((mal_jack_client_t*)pDummyClient);
return MAL_SUCCESS;
}
@ -12462,9 +12498,9 @@ mal_result mal_device_init__jack(mal_context* pContext, mal_device_type type, ma
// Open the client.
mal_result result = mal_context_open_client__jack(pContext, type, pDeviceID, (mal_jack_client_t**)&pDevice->jack.pClient);
mal_result result = mal_context_open_client__jack(pContext, (mal_jack_client_t**)&pDevice->jack.pClient);
if (result != MAL_SUCCESS) {
return result;
return mal_post_error(pDevice, MAL_LOG_LEVEL_ERROR, "[JACK] Failed to open client.", MAL_FAILED_TO_OPEN_BACKEND_DEVICE);
}
// Callbacks.
@ -12861,9 +12897,9 @@ mal_result mal_format_from_AudioStreamBasicDescription(const AudioStreamBasicDes
}
// We are not currently supporting non-interleaved formats (this will be added in a future version of mini_al).
if ((pDescription->mFormatFlags & kAudioFormatFlagIsNonInterleaved) != 0) {
return MAL_FORMAT_NOT_SUPPORTED;
}
//if ((pDescription->mFormatFlags & kAudioFormatFlagIsNonInterleaved) != 0) {
// return MAL_FORMAT_NOT_SUPPORTED;
//}
if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsFloat) != 0) {
if (pDescription->mBitsPerChannel == 32) {
@ -13395,14 +13431,13 @@ mal_result mal_set_AudioObject_buffer_size_in_frames(mal_context* pContext, Audi
propAddress.mScope = (deviceType == mal_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
propAddress.mElement = kAudioObjectPropertyElementMaster;
OSStatus status = ((mal_AudioObjectSetPropertyData_proc)pContext->coreaudio.AudioObjectSetPropertyData)(deviceObjectID, &propAddress, 0, NULL, sizeof(chosenBufferSizeInFrames), &chosenBufferSizeInFrames);
((mal_AudioObjectSetPropertyData_proc)pContext->coreaudio.AudioObjectSetPropertyData)(deviceObjectID, &propAddress, 0, NULL, sizeof(chosenBufferSizeInFrames), &chosenBufferSizeInFrames);
// Get the actual size of the buffer.
UInt32 dataSize = sizeof(*pBufferSizeInOut);
OSStatus status = ((mal_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, &chosenBufferSizeInFrames);
if (status != noErr) {
// Getting here means we were unable to set the buffer size. In this case just use whatever is currently selected.
UInt32 dataSize = sizeof(*pBufferSizeInOut);
OSStatus status = ((mal_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pBufferSizeInOut);
if (status != noErr) {
return mal_result_from_OSStatus(status);
}
return mal_result_from_OSStatus(status);
}
*pBufferSizeInOut = chosenBufferSizeInFrames;
@ -13903,7 +13938,7 @@ mal_result mal_context_init__coreaudio(mal_context* pContext)
// It looks like Apple has moved some APIs from AudioUnit into AudioToolbox on more recent versions of macOS. They are still
// defined in AudioUnit, but just in case they decided to remove them from there entirely I'm going to do implement a fallback.
// defined in AudioUnit, but just in case they decide to remove them from there entirely I'm going to implement a fallback.
// The way it'll work is that it'll first try AudioUnit, and if the required symbols are not present there we'll fall back to
// AudioToolbox.
pContext->coreaudio.hAudioUnit = mal_dlopen("AudioUnit.framework/AudioUnit");
@ -14040,27 +14075,28 @@ OSStatus mal_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFlags* pA
mal_device* pDevice = (mal_device*)pUserData;
mal_assert(pDevice != NULL);
// I'm not going to trust the input frame count. I'm instead going to base this off the size of the first buffer.
UInt32 actualFrameCount = ((AudioBufferList*)pDevice->coreaudio.pAudioBufferList)->mBuffers[0].mDataByteSize / mal_get_bytes_per_sample(pDevice->internalFormat) / ((AudioBufferList*)pDevice->coreaudio.pAudioBufferList)->mBuffers[0].mNumberChannels;
if (actualFrameCount == 0) {
return noErr;
}
OSStatus status = ((mal_AudioUnitRender_proc)pDevice->pContext->coreaudio.AudioUnitRender)((AudioUnit)pDevice->coreaudio.audioUnit, pActionFlags, pTimeStamp, busNumber, actualFrameCount, (AudioBufferList*)pDevice->coreaudio.pAudioBufferList);
if (status != noErr) {
return status;
}
AudioBufferList* pRenderedBufferList = (AudioBufferList*)pDevice->coreaudio.pAudioBufferList;
mal_assert(pRenderedBufferList);
#if defined(MAL_DEBUG_OUTPUT)
printf("INFO: Input Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", busNumber, frameCount, pRenderedBufferList->mNumberBuffers);
#endif
OSStatus status = ((mal_AudioUnitRender_proc)pDevice->pContext->coreaudio.AudioUnitRender)((AudioUnit)pDevice->coreaudio.audioUnit, pActionFlags, pTimeStamp, busNumber, frameCount, pRenderedBufferList);
if (status != noErr) {
#if defined(MAL_DEBUG_OUTPUT)
printf(" ERROR: AudioUnitRender() failed with %d\n", status);
#endif
return status;
}
// For now we can assume everything is interleaved.
for (UInt32 iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) {
if (pRenderedBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->internalChannels) {
mal_uint32 frameCountForThisBuffer = pRenderedBufferList->mBuffers[iBuffer].mDataByteSize / mal_get_bytes_per_frame(pDevice->internalFormat, pDevice->internalChannels);
if (frameCountForThisBuffer > 0) {
mal_device__send_frames_to_client(pDevice, frameCountForThisBuffer, pRenderedBufferList->mBuffers[iBuffer].mData);
}
mal_device__send_frames_to_client(pDevice, frameCount, pRenderedBufferList->mBuffers[iBuffer].mData);
#if defined(MAL_DEBUG_OUTPUT)
printf(" mDataByteSize=%d\n", pRenderedBufferList->mBuffers[iBuffer].mDataByteSize);
#endif
} else {
// This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's
// not interleaved, in which case we can't handle right now since mini_al does not yet support non-interleaved streams.
@ -14185,11 +14221,11 @@ mal_result mal_device_init__coreaudio(mal_context* pContext, mal_device_type dev
// for the sample data format. If the sample data format is not supported by mini_al it must be ignored completely.
//
// On mobile platforms this is a bit different. We just force the use of whatever the audio unit's current format is set to.
AudioStreamBasicDescription bestFormat;
{
AudioUnitScope formatScope = (deviceType == mal_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output;
AudioUnitElement formatElement = (deviceType == mal_device_type_playback) ? MAL_COREAUDIO_OUTPUT_BUS : MAL_COREAUDIO_INPUT_BUS;
AudioStreamBasicDescription bestFormat;
#if defined(MAL_APPLE_DESKTOP)
result = mal_device_find_best_format__coreaudio(pDevice, &bestFormat);
if (result != MAL_SUCCESS) {
@ -14197,15 +14233,26 @@ mal_result mal_device_init__coreaudio(mal_context* pContext, mal_device_type dev
return result;
}
// From what I can see, Apple's documentation implies that we should keep the sample rate consistent.
AudioStreamBasicDescription origFormat;
UInt32 origFormatSize = sizeof(origFormat);
if (deviceType == mal_device_type_playback) {
status = ((mal_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)((AudioUnit)pDevice->coreaudio.audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, MAL_COREAUDIO_OUTPUT_BUS, &origFormat, &origFormatSize);
} else {
status = ((mal_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)((AudioUnit)pDevice->coreaudio.audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, MAL_COREAUDIO_INPUT_BUS, &origFormat, &origFormatSize);
}
if (status != noErr) {
((mal_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnit);
return result;
}
bestFormat.mSampleRate = origFormat.mSampleRate;
status = ((mal_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)((AudioUnit)pDevice->coreaudio.audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, sizeof(bestFormat));
if (status != noErr) {
// We failed to set the format, so fall back to the current format of the audio unit.
UInt32 propSize = sizeof(bestFormat);
status = ((mal_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)((AudioUnit)pDevice->coreaudio.audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, &propSize);
if (status != noErr) {
((mal_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnit);
return mal_result_from_OSStatus(status);
}
bestFormat = origFormat;
}
#else
UInt32 propSize = sizeof(bestFormat);
@ -14280,7 +14327,7 @@ mal_result mal_device_init__coreaudio(mal_context* pContext, mal_device_type dev
if (deviceType == mal_device_type_playback) {
fDeviceType = 1.0f;
} else {
fDeviceType = 1.0f;
fDeviceType = 6.0f;
}
// Backend tax. Need to fiddle with this.
@ -14310,10 +14357,16 @@ mal_result mal_device_init__coreaudio(mal_context* pContext, mal_device_type dev
// Note how inFramesToProcess is smaller than mMaxFramesPerSlice. To fix, we need to set kAudioUnitProperty_MaximumFramesPerSlice to that
// of the size of our buffer, or do it the other way around and set our buffer size to the kAudioUnitProperty_MaximumFramesPerSlice.
{
AudioUnitScope propScope = (deviceType == mal_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output;
/*AudioUnitScope propScope = (deviceType == mal_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output;
AudioUnitElement propBus = (deviceType == mal_device_type_playback) ? MAL_COREAUDIO_OUTPUT_BUS : MAL_COREAUDIO_INPUT_BUS;
status = ((mal_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)((AudioUnit)pDevice->coreaudio.audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, propScope, propBus, &actualBufferSizeInFrames, sizeof(actualBufferSizeInFrames));
if (status != noErr) {
((mal_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnit);
return mal_result_from_OSStatus(status);
}*/
status = ((mal_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)((AudioUnit)pDevice->coreaudio.audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &actualBufferSizeInFrames, sizeof(actualBufferSizeInFrames));
if (status != noErr) {
((mal_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnit);
return mal_result_from_OSStatus(status);
@ -14350,7 +14403,7 @@ mal_result mal_device_init__coreaudio(mal_context* pContext, mal_device_type dev
// We need a buffer list if this is an input device. We render into this in the input callback.
if (deviceType == mal_device_type_capture) {
mal_bool32 isInterleaved = MAL_TRUE; // TODO: Add support for non-interleaved streams.
mal_bool32 isInterleaved = (bestFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved) == 0;
size_t allocationSize = sizeof(AudioBufferList) - sizeof(AudioBuffer); // Subtract sizeof(AudioBuffer) because that part is dynamically sized.
if (isInterleaved) {
@ -23547,7 +23600,7 @@ mal_uint64 mal_src_read_deinterleaved__sinc(mal_src* pSRC, mal_uint64 frameCount
if (framesReadFromClient != 0) {
pSRC->sinc.inputFrameCount += framesReadFromClient;
} else {
// We couldn't get anything more from the client. If not more output samples can be computed from the available input samples
// We couldn't get anything more from the client. If no more output samples can be computed from the available input samples
// we need to return.
if (((pSRC->sinc.timeIn - pSRC->sinc.inputFrameCount) * inverseFactor) < 1) {
break;