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AAudio: Migrate over to the new backend callback system.

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This commit is contained in:
David Reid
2021-01-31 10:17:06 +10:00
parent 3391a4826f
commit ddb8f845f7
1 changed files with 274 additions and 162 deletions
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miniaudio.h
+274 -162
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@@ -5556,6 +5556,58 @@ callback.
MA_API ma_result ma_device_handle_backend_data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount);
/*
Calculates an appropriate buffer size from a descriptor, native sample rate and performance profile.
This function is used by backends for helping determine an appropriately sized buffer to use with
the device depending on the values of `periodSizeInFrames` and `periodSizeInMilliseconds` in the
`pDescriptor` object. Since buffer size calculations based on time depends on the sample rate, a
best guess at the device's native sample rate is also required which is where `nativeSampleRate`
comes in. In addition, the performance profile is also needed for cases where both the period size
in frames and milliseconds are both zero.
Parameters
----------
pDescriptor (in)
A pointer to device descriptor whose `periodSizeInFrames` and `periodSizeInMilliseconds` members
will be used for the calculation of the buffer size.
nativeSampleRate (in)
The device's native sample rate. This is only ever used when the `periodSizeInFrames` member of
`pDescriptor` is zero. In this case, `periodSizeInMilliseconds` will be used instead, in which
case a sample rate is required to convert to a size in frames.
performanceProfile (in)
When both the `periodSizeInFrames` and `periodSizeInMilliseconds` members of `pDescriptor` are
zero, miniaudio will fall back to a buffer size based on the performance profile. The profile
to use for this calculation is determine by this parameter.
Return Value
------------
The calculated buffer size in frames.
Thread Safety
-------------
This is safe so long as nothing modifies `pDescriptor` at the same time. However, this function
should only ever be called from within the backend's device initialization routine and therefore
shouldn't have any multithreading concerns.
Callback Safety
---------------
This is safe to call within the data callback, but there is no reason to ever do this.
Remarks
-------
If `nativeSampleRate` is zero, this function will fall back to `pDescriptor->sampleRate`. If that
is also zero, `MA_DEFAULT_SAMPLE_RATE` will be used instead.
*/
MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_descriptor(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile);
/*
@@ -29537,11 +29589,7 @@ static ma_aaudio_data_callback_result_t ma_stream_data_callback_capture__aaudio(
ma_device* pDevice = (ma_device*)pUserData;
MA_ASSERT(pDevice != NULL);
if (pDevice->type == ma_device_type_duplex) {
ma_device__handle_duplex_callback_capture(pDevice, frameCount, pAudioData, &pDevice->aaudio.duplexRB);
} else {
ma_device__send_frames_to_client(pDevice, frameCount, pAudioData); /* Send directly to the client. */
}
ma_device_handle_backend_data_callback(pDevice, NULL, pAudioData, frameCount);
(void)pStream;
return MA_AAUDIO_CALLBACK_RESULT_CONTINUE;
@@ -29552,24 +29600,20 @@ static ma_aaudio_data_callback_result_t ma_stream_data_callback_playback__aaudio
ma_device* pDevice = (ma_device*)pUserData;
MA_ASSERT(pDevice != NULL);
if (pDevice->type == ma_device_type_duplex) {
ma_device__handle_duplex_callback_playback(pDevice, frameCount, pAudioData, &pDevice->aaudio.duplexRB);
} else {
ma_device__read_frames_from_client(pDevice, frameCount, pAudioData); /* Read directly from the client. */
}
ma_device_handle_backend_data_callback(pDevice, pAudioData, NULL, frameCount);
(void)pStream;
return MA_AAUDIO_CALLBACK_RESULT_CONTINUE;
}
static ma_result ma_open_stream__aaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, const ma_device_config* pConfig, const ma_device* pDevice, ma_AAudioStream** ppStream)
static ma_result ma_create_and_configure_AAudioStreamBuilder__aaudio(ma_context* pContext, const ma_device_id* pDeviceID, ma_device_type deviceType, ma_share_mode shareMode, const ma_device_descriptor* pDescriptor, const ma_device_config* pConfig, ma_device* pDevice, ma_AAudioStreamBuilder** ppBuilder)
{
ma_AAudioStreamBuilder* pBuilder;
ma_aaudio_result_t resultAA;
ma_uint32 bufferCapacityInFrames;
MA_ASSERT(deviceType != ma_device_type_duplex); /* This function should not be called for a full-duplex device type. */
*ppStream = NULL;
/* Safety. */
*ppBuilder = NULL;
resultAA = ((MA_PFN_AAudio_createStreamBuilder)pContext->aaudio.AAudio_createStreamBuilder)(&pBuilder);
if (resultAA != MA_AAUDIO_OK) {
@@ -29583,36 +29627,42 @@ static ma_result ma_open_stream__aaudio(ma_context* pContext, ma_device_type dev
((MA_PFN_AAudioStreamBuilder_setDirection)pContext->aaudio.AAudioStreamBuilder_setDirection)(pBuilder, (deviceType == ma_device_type_playback) ? MA_AAUDIO_DIRECTION_OUTPUT : MA_AAUDIO_DIRECTION_INPUT);
((MA_PFN_AAudioStreamBuilder_setSharingMode)pContext->aaudio.AAudioStreamBuilder_setSharingMode)(pBuilder, (shareMode == ma_share_mode_shared) ? MA_AAUDIO_SHARING_MODE_SHARED : MA_AAUDIO_SHARING_MODE_EXCLUSIVE);
if (pConfig != NULL) {
ma_uint32 bufferCapacityInFrames;
if (pDevice == NULL || !pDevice->usingDefaultSampleRate) {
((MA_PFN_AAudioStreamBuilder_setSampleRate)pContext->aaudio.AAudioStreamBuilder_setSampleRate)(pBuilder, pConfig->sampleRate);
/* If we have a device descriptor make sure we configure the stream builder to take our requested parameters. */
if (pDescriptor != NULL) {
MA_ASSERT(pConfig != NULL); /* We must have a device config if we also have a descriptor. The config is required for AAudio specific configuration options. */
if (pDescriptor->sampleRate != 0) {
((MA_PFN_AAudioStreamBuilder_setSampleRate)pContext->aaudio.AAudioStreamBuilder_setSampleRate)(pBuilder, pDescriptor->sampleRate);
}
if (deviceType == ma_device_type_capture) {
if (pDevice == NULL || !pDevice->capture.usingDefaultChannels) {
((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pConfig->capture.channels);
if (pDescriptor->channels != 0) {
((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pDescriptor->channels);
}
if (pDevice == NULL || !pDevice->capture.usingDefaultFormat) {
((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pConfig->capture.format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT);
if (pDescriptor->format != ma_format_unknown) {
((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pDescriptor->format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT);
}
} else {
if (pDevice == NULL || !pDevice->playback.usingDefaultChannels) {
((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pConfig->playback.channels);
if (pDescriptor->channels != 0) {
((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pDescriptor->channels);
}
if (pDevice == NULL || !pDevice->playback.usingDefaultFormat) {
((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pConfig->playback.format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT);
if (pDescriptor->format != ma_format_unknown) {
((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pDescriptor->format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT);
}
}
bufferCapacityInFrames = pConfig->periodSizeInFrames * pConfig->periods;
if (bufferCapacityInFrames == 0) {
bufferCapacityInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->periodSizeInMilliseconds, pConfig->sampleRate) * pConfig->periods;
}
/*
AAudio is annoying when it comes to it's buffer calculation stuff because it doesn't let you
retrieve the actual sample rate until after you've opened the stream. But you need to configure
the buffer capacity before you open the stream... :/
To solve, we're just going to assume MA_DEFAULT_SAMPLE_RATE (48000) and move on.
*/
bufferCapacityInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, pDescriptor->sampleRate, pConfig->performanceProfile) * pDescriptor->periodCount;
((MA_PFN_AAudioStreamBuilder_setBufferCapacityInFrames)pContext->aaudio.AAudioStreamBuilder_setBufferCapacityInFrames)(pBuilder, bufferCapacityInFrames);
((MA_PFN_AAudioStreamBuilder_setFramesPerDataCallback)pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback)(pBuilder, bufferCapacityInFrames / pConfig->periods);
((MA_PFN_AAudioStreamBuilder_setFramesPerDataCallback)pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback)(pBuilder, bufferCapacityInFrames / pDescriptor->periodCount);
if (deviceType == ma_device_type_capture) {
if (pConfig->aaudio.inputPreset != ma_aaudio_input_preset_default && pContext->aaudio.AAudioStreamBuilder_setInputPreset != NULL) {
@@ -29634,21 +29684,61 @@ static ma_result ma_open_stream__aaudio(ma_context* pContext, ma_device_type dev
/* Not sure how this affects things, but since there's a mapping between miniaudio's performance profiles and AAudio's performance modes, let go ahead and set it. */
((MA_PFN_AAudioStreamBuilder_setPerformanceMode)pContext->aaudio.AAudioStreamBuilder_setPerformanceMode)(pBuilder, (pConfig->performanceProfile == ma_performance_profile_low_latency) ? MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY : MA_AAUDIO_PERFORMANCE_MODE_NONE);
/* We need to set an error callback to detect device changes. */
if (pDevice != NULL) { /* <-- pDevice should never be null if pDescriptor is not null, which is always the case if we hit this branch. Check anyway for safety. */
((MA_PFN_AAudioStreamBuilder_setErrorCallback)pContext->aaudio.AAudioStreamBuilder_setErrorCallback)(pBuilder, ma_stream_error_callback__aaudio, (void*)pDevice);
}
}
((MA_PFN_AAudioStreamBuilder_setErrorCallback)pContext->aaudio.AAudioStreamBuilder_setErrorCallback)(pBuilder, ma_stream_error_callback__aaudio, (void*)pDevice);
*ppBuilder = pBuilder;
resultAA = ((MA_PFN_AAudioStreamBuilder_openStream)pContext->aaudio.AAudioStreamBuilder_openStream)(pBuilder, ppStream);
if (resultAA != MA_AAUDIO_OK) {
*ppStream = NULL;
((MA_PFN_AAudioStreamBuilder_delete)pContext->aaudio.AAudioStreamBuilder_delete)(pBuilder);
return ma_result_from_aaudio(resultAA);
}
((MA_PFN_AAudioStreamBuilder_delete)pContext->aaudio.AAudioStreamBuilder_delete)(pBuilder);
return MA_SUCCESS;
}
static ma_result ma_open_stream_and_close_builder__aaudio(ma_context* pContext, ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream** ppStream)
{
ma_result result;
result = ma_result_from_aaudio(((MA_PFN_AAudioStreamBuilder_openStream)pContext->aaudio.AAudioStreamBuilder_openStream)(pBuilder, ppStream));
((MA_PFN_AAudioStreamBuilder_delete)pContext->aaudio.AAudioStreamBuilder_delete)(pBuilder);
return result;
}
static ma_result ma_open_stream_basic__aaudio(ma_context* pContext, const ma_device_id* pDeviceID, ma_device_type deviceType, ma_share_mode shareMode, ma_AAudioStream** ppStream)
{
ma_result result;
ma_AAudioStreamBuilder* pBuilder;
*ppStream = NULL;
result = ma_create_and_configure_AAudioStreamBuilder__aaudio(pContext, pDeviceID, deviceType, shareMode, NULL, NULL, NULL, &pBuilder);
if (result != MA_SUCCESS) {
return result;
}
return ma_open_stream_and_close_builder__aaudio(pContext, pBuilder, ppStream);
}
static ma_result ma_open_stream__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_type deviceType, const ma_device_descriptor* pDescriptor, ma_AAudioStream** ppStream)
{
ma_result result;
ma_AAudioStreamBuilder* pBuilder;
MA_ASSERT(pConfig != NULL);
MA_ASSERT(pConfig->deviceType != ma_device_type_duplex); /* This function should not be called for a full-duplex device type. */
*ppStream = NULL;
result = ma_create_and_configure_AAudioStreamBuilder__aaudio(pDevice->pContext, pDescriptor->pDeviceID, deviceType, pDescriptor->shareMode, pDescriptor, pConfig, pDevice, &pBuilder);
if (result != MA_SUCCESS) {
return result;
}
return ma_open_stream_and_close_builder__aaudio(pDevice->pContext, pBuilder, ppStream);
}
static ma_result ma_close_stream__aaudio(ma_context* pContext, ma_AAudioStream* pStream)
{
return ma_result_from_aaudio(((MA_PFN_AAudioStream_close)pContext->aaudio.AAudioStream_close)(pStream));
@@ -29658,7 +29748,7 @@ static ma_bool32 ma_has_default_device__aaudio(ma_context* pContext, ma_device_t
{
/* The only way to know this is to try creating a stream. */
ma_AAudioStream* pStream;
ma_result result = ma_open_stream__aaudio(pContext, deviceType, NULL, ma_share_mode_shared, NULL, NULL, &pStream);
ma_result result = ma_open_stream_basic__aaudio(pContext, NULL, deviceType, ma_share_mode_shared, &pStream);
if (result != MA_SUCCESS) {
return MA_FALSE;
}
@@ -29719,18 +29809,33 @@ static ma_result ma_context_enumerate_devices__aaudio(ma_context* pContext, ma_e
return MA_SUCCESS;
}
static ma_result ma_context_get_device_info__aaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo)
static void ma_context_add_native_data_format_from_AAudioStream_ex__aaudio(ma_context* pContext, ma_AAudioStream* pStream, ma_format format, ma_uint32 flags, ma_device_info* pDeviceInfo)
{
MA_ASSERT(pContext != NULL);
MA_ASSERT(pStream != NULL);
MA_ASSERT(pDeviceInfo != NULL);
pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format;
pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = ((MA_PFN_AAudioStream_getChannelCount)pContext->aaudio.AAudioStream_getChannelCount)(pStream);
pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = ((MA_PFN_AAudioStream_getSampleRate)pContext->aaudio.AAudioStream_getSampleRate)(pStream);
pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = flags;
pDeviceInfo->nativeDataFormatCount += 1;
}
static void ma_context_add_native_data_format_from_AAudioStream__aaudio(ma_context* pContext, ma_AAudioStream* pStream, ma_uint32 flags, ma_device_info* pDeviceInfo)
{
/* AAudio supports s16 and f32. */
ma_context_add_native_data_format_from_AAudioStream_ex__aaudio(pContext, pStream, ma_format_f32, flags, pDeviceInfo);
ma_context_add_native_data_format_from_AAudioStream_ex__aaudio(pContext, pStream, ma_format_s16, flags, pDeviceInfo);
}
static ma_result ma_context_get_device_info__aaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
{
ma_AAudioStream* pStream;
ma_result result;
MA_ASSERT(pContext != NULL);
/* No exclusive mode with AAudio. */
if (shareMode == ma_share_mode_exclusive) {
return MA_SHARE_MODE_NOT_SUPPORTED;
}
/* ID */
if (pDeviceID != NULL) {
pDeviceInfo->id.aaudio = pDeviceID->aaudio;
@@ -29746,31 +29851,24 @@ static ma_result ma_context_get_device_info__aaudio(ma_context* pContext, ma_dev
}
pDeviceInfo->nativeDataFormatCount = 0;
/* We'll need to open the device to get accurate sample rate and channel count information. */
result = ma_open_stream__aaudio(pContext, deviceType, pDeviceID, shareMode, NULL, NULL, &pStream);
result = ma_open_stream_basic__aaudio(pContext, pDeviceID, deviceType, ma_share_mode_shared, &pStream);
if (result != MA_SUCCESS) {
return result;
}
pDeviceInfo->minChannels = ((MA_PFN_AAudioStream_getChannelCount)pContext->aaudio.AAudioStream_getChannelCount)(pStream);
pDeviceInfo->maxChannels = pDeviceInfo->minChannels;
pDeviceInfo->minSampleRate = ((MA_PFN_AAudioStream_getSampleRate)pContext->aaudio.AAudioStream_getSampleRate)(pStream);
pDeviceInfo->maxSampleRate = pDeviceInfo->minSampleRate;
ma_context_add_native_data_format_from_AAudioStream__aaudio(pContext, pStream, 0, pDeviceInfo);
ma_close_stream__aaudio(pContext, pStream);
pStream = NULL;
/* AAudio supports s16 and f32. */
pDeviceInfo->formatCount = 2;
pDeviceInfo->formats[0] = ma_format_s16;
pDeviceInfo->formats[1] = ma_format_f32;
return MA_SUCCESS;
}
static void ma_device_uninit__aaudio(ma_device* pDevice)
static ma_result ma_device_uninit__aaudio(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
@@ -29784,12 +29882,57 @@ static void ma_device_uninit__aaudio(ma_device* pDevice)
pDevice->aaudio.pStreamPlayback = NULL;
}
if (pDevice->type == ma_device_type_duplex) {
ma_pcm_rb_uninit(&pDevice->aaudio.duplexRB);
}
return MA_SUCCESS;
}
static ma_result ma_device_init__aaudio(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice)
static ma_result ma_device_init_by_type__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_type deviceType, ma_device_descriptor* pDescriptor, ma_AAudioStream** ppStream)
{
ma_result result;
int32_t bufferCapacityInFrames;
int32_t framesPerDataCallback;
ma_AAudioStream* pStream;
MA_ASSERT(pDevice != NULL);
MA_ASSERT(pConfig != NULL);
MA_ASSERT(pDescriptor != NULL);
*ppStream = NULL; /* Safety. */
/* First step is to open the stream. From there we'll be able to extract the internal configuration. */
result = ma_open_stream__aaudio(pDevice, pConfig, deviceType, pDescriptor, &pStream);
if (result != MA_SUCCESS) {
return result; /* Failed to open the AAudio stream. */
}
/* Now extract the internal configuration. */
pDescriptor->format = (((MA_PFN_AAudioStream_getFormat)pDevice->pContext->aaudio.AAudioStream_getFormat)(pStream) == MA_AAUDIO_FORMAT_PCM_I16) ? ma_format_s16 : ma_format_f32;
pDescriptor->channels = ((MA_PFN_AAudioStream_getChannelCount)pDevice->pContext->aaudio.AAudioStream_getChannelCount)(pStream);
pDescriptor->sampleRate = ((MA_PFN_AAudioStream_getSampleRate)pDevice->pContext->aaudio.AAudioStream_getSampleRate)(pStream);
/* For the channel map we need to be sure we don't overflow any buffers. */
if (pDescriptor->channels <= MA_MAX_CHANNELS) {
ma_get_standard_channel_map(ma_standard_channel_map_default, pDescriptor->channels, pDescriptor->channelMap); /* <-- Cannot find info on channel order, so assuming a default. */
} else {
ma_channel_map_init_blank(MA_MAX_CHANNELS, pDescriptor->channelMap); /* Too many channels. Use a blank channel map. */
}
bufferCapacityInFrames = ((MA_PFN_AAudioStream_getBufferCapacityInFrames)pDevice->pContext->aaudio.AAudioStream_getBufferCapacityInFrames)(pStream);
framesPerDataCallback = ((MA_PFN_AAudioStream_getFramesPerDataCallback)pDevice->pContext->aaudio.AAudioStream_getFramesPerDataCallback)(pStream);
if (framesPerDataCallback > 0) {
pDescriptor->periodSizeInFrames = framesPerDataCallback;
pDescriptor->periodCount = bufferCapacityInFrames / framesPerDataCallback;
} else {
pDescriptor->periodSizeInFrames = bufferCapacityInFrames;
pDescriptor->periodCount = 1;
}
*ppStream = pStream;
return MA_SUCCESS;
}
static ma_result ma_device_init__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
ma_result result;
@@ -29800,86 +29943,22 @@ static ma_result ma_device_init__aaudio(ma_context* pContext, const ma_device_co
}
/* No exclusive mode with AAudio. */
if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) ||
((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) {
if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
return MA_SHARE_MODE_NOT_SUPPORTED;
}
/* We first need to try opening the stream. */
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
int32_t bufferCapacityInFrames;
int32_t framesPerDataCallback;
result = ma_open_stream__aaudio(pContext, ma_device_type_capture, pConfig->capture.pDeviceID, pConfig->capture.shareMode, pConfig, pDevice, (ma_AAudioStream**)&pDevice->aaudio.pStreamCapture);
result = ma_device_init_by_type__aaudio(pDevice, pConfig, ma_device_type_capture, pDescriptorCapture, (ma_AAudioStream**)&pDevice->aaudio.pStreamCapture);
if (result != MA_SUCCESS) {
return result; /* Failed to open the AAudio stream. */
}
pDevice->capture.internalFormat = (((MA_PFN_AAudioStream_getFormat)pContext->aaudio.AAudioStream_getFormat)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture) == MA_AAUDIO_FORMAT_PCM_I16) ? ma_format_s16 : ma_format_f32;
pDevice->capture.internalChannels = ((MA_PFN_AAudioStream_getChannelCount)pContext->aaudio.AAudioStream_getChannelCount)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
pDevice->capture.internalSampleRate = ((MA_PFN_AAudioStream_getSampleRate)pContext->aaudio.AAudioStream_getSampleRate)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
ma_get_standard_channel_map(ma_standard_channel_map_default, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); /* <-- Cannot find info on channel order, so assuming a default. */
bufferCapacityInFrames = ((MA_PFN_AAudioStream_getBufferCapacityInFrames)pContext->aaudio.AAudioStream_getBufferCapacityInFrames)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
framesPerDataCallback = ((MA_PFN_AAudioStream_getFramesPerDataCallback)pContext->aaudio.AAudioStream_getFramesPerDataCallback)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
if (framesPerDataCallback > 0) {
pDevice->capture.internalPeriodSizeInFrames = framesPerDataCallback;
pDevice->capture.internalPeriods = bufferCapacityInFrames / framesPerDataCallback;
} else {
pDevice->capture.internalPeriodSizeInFrames = bufferCapacityInFrames;
pDevice->capture.internalPeriods = 1;
return result;
}
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
int32_t bufferCapacityInFrames;
int32_t framesPerDataCallback;
result = ma_open_stream__aaudio(pContext, ma_device_type_playback, pConfig->playback.pDeviceID, pConfig->playback.shareMode, pConfig, pDevice, (ma_AAudioStream**)&pDevice->aaudio.pStreamPlayback);
result = ma_device_init_by_type__aaudio(pDevice, pConfig, ma_device_type_playback, pDescriptorPlayback, (ma_AAudioStream**)&pDevice->aaudio.pStreamPlayback);
if (result != MA_SUCCESS) {
return result; /* Failed to open the AAudio stream. */
}
pDevice->playback.internalFormat = (((MA_PFN_AAudioStream_getFormat)pContext->aaudio.AAudioStream_getFormat)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback) == MA_AAUDIO_FORMAT_PCM_I16) ? ma_format_s16 : ma_format_f32;
pDevice->playback.internalChannels = ((MA_PFN_AAudioStream_getChannelCount)pContext->aaudio.AAudioStream_getChannelCount)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback);
pDevice->playback.internalSampleRate = ((MA_PFN_AAudioStream_getSampleRate)pContext->aaudio.AAudioStream_getSampleRate)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback);
ma_get_standard_channel_map(ma_standard_channel_map_default, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); /* <-- Cannot find info on channel order, so assuming a default. */
bufferCapacityInFrames = ((MA_PFN_AAudioStream_getBufferCapacityInFrames)pContext->aaudio.AAudioStream_getBufferCapacityInFrames)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback);
framesPerDataCallback = ((MA_PFN_AAudioStream_getFramesPerDataCallback)pContext->aaudio.AAudioStream_getFramesPerDataCallback)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback);
if (framesPerDataCallback > 0) {
pDevice->playback.internalPeriodSizeInFrames = framesPerDataCallback;
pDevice->playback.internalPeriods = bufferCapacityInFrames / framesPerDataCallback;
} else {
pDevice->playback.internalPeriodSizeInFrames = bufferCapacityInFrames;
pDevice->playback.internalPeriods = 1;
}
}
if (pConfig->deviceType == ma_device_type_duplex) {
ma_uint32 rbSizeInFrames = (ma_uint32)ma_calculate_frame_count_after_resampling(pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalPeriodSizeInFrames) * pDevice->capture.internalPeriods;
ma_result result = ma_pcm_rb_init(pDevice->capture.format, pDevice->capture.channels, rbSizeInFrames, NULL, &pDevice->pContext->allocationCallbacks, &pDevice->aaudio.duplexRB);
if (result != MA_SUCCESS) {
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback);
}
return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[AAudio] Failed to initialize ring buffer.", result);
}
/* We need a period to act as a buffer for cases where the playback and capture device's end up desyncing. */
{
ma_uint32 marginSizeInFrames = rbSizeInFrames / pDevice->capture.internalPeriods;
void* pMarginData;
ma_pcm_rb_acquire_write(&pDevice->aaudio.duplexRB, &marginSizeInFrames, &pMarginData);
{
MA_ZERO_MEMORY(pMarginData, marginSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels));
}
ma_pcm_rb_commit_write(&pDevice->aaudio.duplexRB, marginSizeInFrames, pMarginData);
return result;
}
}
@@ -30020,7 +30099,7 @@ static ma_result ma_context_uninit__aaudio(ma_context* pContext)
return MA_SUCCESS;
}
static ma_result ma_context_init__aaudio(const ma_context_config* pConfig, ma_context* pContext)
static ma_result ma_context_init__aaudio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
{
size_t i;
const char* libNames[] = {
@@ -30067,15 +30146,18 @@ static ma_result ma_context_init__aaudio(const ma_context_config* pConfig, ma_co
pContext->aaudio.AAudioStream_requestStart = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_requestStart");
pContext->aaudio.AAudioStream_requestStop = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_requestStop");
pContext->isBackendAsynchronous = MA_TRUE;
pContext->onUninit = ma_context_uninit__aaudio;
pContext->onEnumDevices = ma_context_enumerate_devices__aaudio;
pContext->onGetDeviceInfo = ma_context_get_device_info__aaudio;
pContext->onDeviceInit = ma_device_init__aaudio;
pContext->onDeviceUninit = ma_device_uninit__aaudio;
pContext->onDeviceStart = ma_device_start__aaudio;
pContext->onDeviceStop = ma_device_stop__aaudio;
pCallbacks->onContextInit = ma_context_init__aaudio;
pCallbacks->onContextUninit = ma_context_uninit__aaudio;
pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__aaudio;
pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__aaudio;
pCallbacks->onDeviceInit = ma_device_init__aaudio;
pCallbacks->onDeviceUninit = ma_device_uninit__aaudio;
pCallbacks->onDeviceStart = ma_device_start__aaudio;
pCallbacks->onDeviceStop = ma_device_stop__aaudio;
pCallbacks->onDeviceRead = NULL; /* Not used because AAudio is asynchronous. */
pCallbacks->onDeviceWrite = NULL; /* Not used because AAudio is asynchronous. */
pCallbacks->onDeviceAudioThread = NULL; /* Not used because AAudio is asynchronous. */
(void)pConfig;
return MA_SUCCESS;
@@ -30760,7 +30842,6 @@ static ma_result ma_device_init__opensl(ma_device* pDevice, const ma_device_conf
#ifdef MA_ANDROID
SLDataLocator_AndroidSimpleBufferQueue queue;
SLresult resultSL;
ma_uint32 periodSizeInFrames;
size_t bufferSizeInBytes;
SLInterfaceID itfIDs1[1];
const SLboolean itfIDsRequired1[] = {SL_BOOLEAN_TRUE};
@@ -30875,22 +30956,8 @@ static ma_result ma_device_init__opensl(ma_device* pDevice, const ma_device_conf
ma_deconstruct_SLDataFormat_PCM__opensl(&pcm, &pDescriptorCapture->format, &pDescriptorCapture->channels, &pDescriptorCapture->sampleRate, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap));
/* Buffer. */
if (pDescriptorCapture->periodSizeInFrames == 0) {
if (pDescriptorCapture->periodSizeInMilliseconds == 0) {
if (pConfig->performanceProfile == ma_performance_profile_low_latency) {
periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY, pDescriptorCapture->sampleRate);
} else {
periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE, pDescriptorCapture->sampleRate);
}
} else {
periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptorCapture->periodSizeInMilliseconds, pDescriptorCapture->sampleRate);
}
} else {
periodSizeInFrames = pDescriptorCapture->periodSizeInFrames;
}
pDescriptorCapture->periodSizeInFrames = periodSizeInFrames;
pDevice->opensl.currentBufferIndexCapture = 0;
pDescriptorCapture->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorCapture, pDescriptorCapture->sampleRate);
pDevice->opensl.currentBufferIndexCapture = 0;
bufferSizeInBytes = pDescriptorCapture->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) * pDescriptorCapture->periodCount;
pDevice->opensl.pBufferCapture = (ma_uint8*)ma__calloc_from_callbacks(bufferSizeInBytes, &pDevice->pContext->allocationCallbacks);
@@ -32425,6 +32492,9 @@ MA_API ma_result ma_context_init(const ma_backend backends[], ma_uint32 backendC
for (iBackend = 0; iBackend < backendsToIterateCount; ++iBackend) {
ma_backend backend = pBackendsToIterate[iBackend];
/* Make sure all callbacks are reset so we don't accidentally drag in any from previously failed initialization attempts. */
MA_ZERO_OBJECT(&pContext->callbacks);
/*
I've had a subtle bug where some state is set by the backend's ma_context_init__*() function, but then later failed because
a setting in the context that was set in the prior failed attempt was left unchanged in the next attempt which resulted in
@@ -32481,6 +32551,12 @@ MA_API ma_result ma_context_init(const ma_backend backends[], ma_uint32 backendC
pContext->callbacks.onContextInit = ma_context_init__jack;
} break;
#endif
#ifdef MA_HAS_AAUDIO
case ma_backend_aaudio:
{
pContext->callbacks.onContextInit = ma_context_init__aaudio;
} break;
#endif
#ifdef MA_HAS_OPENSL
case ma_backend_opensl:
{
@@ -32584,8 +32660,7 @@ MA_API ma_result ma_context_init(const ma_backend backends[], ma_uint32 backendC
#ifdef MA_HAS_AAUDIO
case ma_backend_aaudio:
{
ma_post_log_message(pContext, NULL, MA_LOG_LEVEL_VERBOSE, "Attempting to initialize AAudio backend...");
result = ma_context_init__aaudio(pConfig, pContext);
/*result = ma_context_init__aaudio(pConfig, pContext);*/
} break;
#endif
#ifdef MA_HAS_OPENSL
@@ -33729,6 +33804,42 @@ MA_API ma_result ma_device_handle_backend_data_callback(ma_device* pDevice, void
return MA_SUCCESS;
}
MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_descriptor(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
{
if (pDescriptor == NULL) {
return 0;
}
/*
We must have a non-0 native sample rate, but some backends don't allow retrieval of this at the
time when the size of the buffer needs to be determined. In this case we need to just take a best
guess and move on. We'll try using the sample rate in pDescriptor first. If that's not set we'll
just fall back to MA_DEFAULT_SAMPLE_RATE.
*/
if (nativeSampleRate == 0) {
nativeSampleRate = pDescriptor->sampleRate;
}
if (nativeSampleRate == 0) {
nativeSampleRate = MA_DEFAULT_SAMPLE_RATE;
}
MA_ASSERT(nativeSampleRate != 0);
if (pDescriptor->periodSizeInFrames == 0) {
if (pDescriptor->periodSizeInMilliseconds == 0) {
if (performanceProfile == ma_performance_profile_low_latency) {
return ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY, nativeSampleRate);
} else {
return ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE, nativeSampleRate);
}
} else {
return ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptor->periodSizeInMilliseconds, nativeSampleRate);
}
} else {
return pDescriptor->periodSizeInFrames;
}
}
#endif /* MA_NO_DEVICE_IO */
@@ -64667,10 +64778,11 @@ REVISION HISTORY
v0.10.32 - TBD
- WASAPI: Fix a deadlock in exclusive mode.
- PulseAudio: Yet another refactor, this time to remove the dependency on `pa_threaded_mainloop`.
- Internal refactoring to migrate to the new backend callback system for the following backends:
- Internal refactoring to migrate over to the new backend callback system for the following backends:
- PulseAudio
- ALSA
- Core Audio
- AAudio
- OpenSL|ES
- Update to latest version of c89atomic.
- Fix a bug where thread handles are not being freed.