diff --git a/mini_al.h b/mini_al.h index c0845593..68f0b476 100644 --- a/mini_al.h +++ b/mini_al.h @@ -12861,9 +12861,9 @@ 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) { @@ -14040,27 +14040,28 @@ 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 +14186,11 @@ 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 +14198,27 @@ 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; + //bestFormat = origFormat; + 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 +14293,7 @@ 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. @@ -14292,7 +14305,7 @@ mal_result mal_device_init__coreaudio(mal_context* pContext, mal_device_type dev } } - actualBufferSizeInFrames = actualBufferSizeInFrames / pDevice->periods; + actualBufferSizeInFrames = mal_next_power_of_2(actualBufferSizeInFrames / pDevice->periods); result = mal_set_AudioObject_buffer_size_in_frames(pContext, deviceObjectID, deviceType, &actualBufferSizeInFrames); if (result != MAL_SUCCESS) { return result; @@ -14310,10 +14323,16 @@ 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 +14369,7 @@ 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) {