diff options
| -rw-r--r-- | Alc/coreaudio.c | 425 |
1 files changed, 402 insertions, 23 deletions
diff --git a/Alc/coreaudio.c b/Alc/coreaudio.c index e32e15fa..eeaf8019 100644 --- a/Alc/coreaudio.c +++ b/Alc/coreaudio.c @@ -31,25 +31,107 @@ #include <CoreServices/CoreServices.h> #include <unistd.h> #include <AudioUnit/AudioUnit.h> +#include <AudioToolbox/AudioToolbox.h> /* toggle verbose tty output among CoreAudio code */ #define CA_VERBOSE 1 typedef struct { - AudioUnit OutputUnit; - ALuint FrameSize; + AudioUnit audioUnit; + + ALuint frameSize; + ALdouble sampleRateRatio; // Ratio of hardware sample rate / requested sample rate + AudioStreamBasicDescription format; // This is the OpenAL format as a CoreAudio ASBD + + AudioConverterRef audioConverter; // Sample rate converter if needed + AudioBufferList *bufferList; // Buffer for data coming from the input device + ALCvoid *resampleBuffer; // Buffer for returned RingBuffer data when resampling + + RingBuffer *ring; } ca_data; static const ALCchar ca_device[] = "CoreAudio Default"; -static int ca_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, - UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) + +static void destroy_buffer_list(AudioBufferList* list) +{ + if(list) + { + for(UInt32 i = 0;i < list->mNumberBuffers;i++) + free(list->mBuffers[i].mData); + free(list); + } +} + +static AudioBufferList* allocate_buffer_list(UInt32 channelCount, UInt32 byteSize) +{ + AudioBufferList *list; + + list = calloc(1, sizeof(AudioBufferList) + sizeof(AudioBuffer)); + if(list) + { + list->mNumberBuffers = 1; + + list->mBuffers[0].mNumberChannels = channelCount; + list->mBuffers[0].mDataByteSize = byteSize; + list->mBuffers[0].mData = malloc(byteSize); + if(list->mBuffers[0].mData == NULL) + { + free(list); + list = NULL; + } + } + return list; +} + +static OSStatus ca_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, + UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) { ALCdevice *device = (ALCdevice*)inRefCon; ca_data *data = (ca_data*)device->ExtraData; aluMixData(device, ioData->mBuffers[0].mData, - ioData->mBuffers[0].mDataByteSize / data->FrameSize); + ioData->mBuffers[0].mDataByteSize / data->frameSize); + + return noErr; +} + +static OSStatus ca_capture_conversion_callback(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets, + AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void* inUserData) +{ + ALCdevice *device = (ALCdevice*)inUserData; + ca_data *data = (ca_data*)device->ExtraData; + + // Read from the ring buffer and store temporarily in a large buffer + ReadRingBuffer(data->ring, data->resampleBuffer, (ALsizei)(*ioNumberDataPackets)); + + // Set the input data + ioData->mNumberBuffers = 1; + ioData->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame; + ioData->mBuffers[0].mData = data->resampleBuffer; + ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * data->format.mBytesPerFrame; + + return noErr; +} + +static OSStatus ca_capture_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, + UInt32 inNumberFrames, AudioBufferList *ioData) +{ + ALCdevice *device = (ALCdevice*)inRefCon; + ca_data *data = (ca_data*)device->ExtraData; + AudioUnitRenderActionFlags flags = 0; + OSStatus err; + + // fill the bufferList with data from the input device + err = AudioUnitRender(data->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, data->bufferList); + if(err != noErr) + { + AL_PRINT("AudioUnitRender error: %d\n", err); + return err; + } + + WriteRingBuffer(data->ring, data->bufferList->mBuffers[0].mData, inNumberFrames); return noErr; } @@ -83,7 +165,7 @@ static ALCboolean ca_open_playback(ALCdevice *device, const ALCchar *deviceName) data = calloc(1, sizeof(*data)); device->ExtraData = data; - err = OpenAComponent(comp, &data->OutputUnit); + err = OpenAComponent(comp, &data->audioUnit); if(err != noErr) { AL_PRINT("OpenAComponent failed\n"); @@ -99,7 +181,7 @@ static void ca_close_playback(ALCdevice *device) { ca_data *data = (ca_data*)device->ExtraData; - CloseComponent(data->OutputUnit); + CloseComponent(data->audioUnit); free(data); device->ExtraData = NULL; @@ -114,14 +196,14 @@ static ALCboolean ca_reset_playback(ALCdevice *device) UInt32 size; /* init and start the default audio unit... */ - err = AudioUnitInitialize(data->OutputUnit); + err = AudioUnitInitialize(data->audioUnit); if(err != noErr) { AL_PRINT("AudioUnitInitialize failed\n"); return ALC_FALSE; } - err = AudioOutputUnitStart(data->OutputUnit); + err = AudioOutputUnitStart(data->audioUnit); if(err != noErr) { AL_PRINT("AudioOutputUnitStart failed\n"); @@ -130,7 +212,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device) /* retrieve default output unit's properties (output side) */ size = sizeof(AudioStreamBasicDescription); - err = AudioUnitGetProperty(data->OutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size); + err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size); if(err != noErr || size != sizeof(AudioStreamBasicDescription)) { AL_PRINT("AudioUnitGetProperty failed\n"); @@ -148,7 +230,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device) #endif /* set default output unit's input side to match output side */ - err = AudioUnitSetProperty(data->OutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size); + err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size); if(err != noErr) { AL_PRINT("AudioUnitSetProperty failed\n"); @@ -262,7 +344,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device) kAudioFormatFlagsNativeEndian | kLinearPCMFormatFlagIsPacked; - err = AudioUnitSetProperty(data->OutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription)); + err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription)); if(err != noErr) { AL_PRINT("AudioUnitSetProperty failed\n"); @@ -270,11 +352,11 @@ static ALCboolean ca_reset_playback(ALCdevice *device) } /* setup callback */ - data->FrameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType); + data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType); input.inputProc = ca_callback; input.inputProcRefCon = device; - err = AudioUnitSetProperty(data->OutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct)); + err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct)); if(err != noErr) { AL_PRINT("AudioUnitSetProperty failed\n"); @@ -289,17 +371,313 @@ static void ca_stop_playback(ALCdevice *device) ca_data *data = (ca_data*)device->ExtraData; OSStatus err; - AudioOutputUnitStop(data->OutputUnit); - err = AudioUnitUninitialize(data->OutputUnit); + AudioOutputUnitStop(data->audioUnit); + err = AudioUnitUninitialize(data->audioUnit); if(err != noErr) AL_PRINT("-- AudioUnitUninitialize failed.\n"); } static ALCboolean ca_open_capture(ALCdevice *device, const ALCchar *deviceName) { + AudioStreamBasicDescription requestedFormat; // The application requested format + AudioStreamBasicDescription hardwareFormat; // The hardware format + AudioStreamBasicDescription outputFormat; // The AudioUnit output format + AURenderCallbackStruct input; + ComponentDescription desc; + AudioDeviceID inputDevice; + UInt32 outputFrameCount; + UInt32 propertySize; + UInt32 enableIO; + Component comp; + ca_data *data; + OSStatus err; + + desc.componentType = kAudioUnitType_Output; + desc.componentSubType = kAudioUnitSubType_HALOutput; + desc.componentManufacturer = kAudioUnitManufacturer_Apple; + desc.componentFlags = 0; + desc.componentFlagsMask = 0; + + // Search for component with given description + comp = FindNextComponent(NULL, &desc); + if(comp == NULL) + { + AL_PRINT("FindNextComponent failed\n"); + return ALC_FALSE; + } + + data = calloc(1, sizeof(*data)); + device->ExtraData = data; + + // Open the component + err = OpenAComponent(comp, &data->audioUnit); + if(err != noErr) + { + AL_PRINT("OpenAComponent failed\n"); + goto error; + } + + // Turn off AudioUnit output + enableIO = 0; + err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint)); + if(err != noErr) + { + AL_PRINT("AudioUnitSetProperty failed\n"); + goto error; + } + + // Turn on AudioUnit input + enableIO = 1; + err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint)); + if(err != noErr) + { + AL_PRINT("AudioUnitSetProperty failed\n"); + goto error; + } + + // Get the default input device + propertySize = sizeof(AudioDeviceID); + err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propertySize, &inputDevice); + if(err != noErr) + { + AL_PRINT("AudioHardwareGetProperty failed\n"); + goto error; + } + + if(inputDevice == kAudioDeviceUnknown) + { + AL_PRINT("No input device found\n"); + goto error; + } + + // Track the input device + err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID)); + if(err != noErr) + { + AL_PRINT("AudioUnitSetProperty failed\n"); + goto error; + } + + // set capture callback + input.inputProc = ca_capture_callback; + input.inputProcRefCon = device; + + err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct)); + if(err != noErr) + { + AL_PRINT("AudioUnitSetProperty failed\n"); + goto error; + } + + // Initialize the device + err = AudioUnitInitialize(data->audioUnit); + if(err != noErr) + { + AL_PRINT("AudioUnitInitialize failed\n"); + goto error; + } + + // Get the hardware format + propertySize = sizeof(AudioStreamBasicDescription); + err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize); + if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription)) + { + AL_PRINT("AudioUnitGetProperty failed\n"); + goto error; + } + + // Set up the requested format description + switch(device->FmtType) + { + case DevFmtUByte: + requestedFormat.mBitsPerChannel = 8; + requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked; + break; + case DevFmtShort: + requestedFormat.mBitsPerChannel = 16; + requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked; + break; + case DevFmtFloat: + requestedFormat.mBitsPerChannel = 32; + requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked; + break; + case DevFmtByte: + case DevFmtUShort: + AL_PRINT("%s samples not supported\n", DevFmtTypeString(device->FmtType)); + goto error; + } + + switch(device->FmtChans) + { + case DevFmtMono: + requestedFormat.mChannelsPerFrame = 1; + break; + case DevFmtStereo: + requestedFormat.mChannelsPerFrame = 2; + break; + + case DevFmtQuad: + case DevFmtX51: + case DevFmtX51Side: + case DevFmtX61: + case DevFmtX71: + AL_PRINT("%s not supported\n", DevFmtChannelsString(device->FmtChans)); + goto error; + } + + requestedFormat.mBytesPerFrame = requestedFormat.mChannelsPerFrame * requestedFormat.mBitsPerChannel / 8; + requestedFormat.mBytesPerPacket = requestedFormat.mBytesPerFrame; + requestedFormat.mSampleRate = device->Frequency; + requestedFormat.mFormatID = kAudioFormatLinearPCM; + requestedFormat.mReserved = 0; + requestedFormat.mFramesPerPacket = 1; + + // save requested format description for later use + data->format = requestedFormat; + data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType); + + // Use intermediate format for sample rate conversion (outputFormat) + // Set sample rate to the same as hardware for resampling later + outputFormat = requestedFormat; + outputFormat.mSampleRate = hardwareFormat.mSampleRate; + + // Determine sample rate ratio for resampling + data->sampleRateRatio = outputFormat.mSampleRate / device->Frequency; + + // The output format should be the requested format, but using the hardware sample rate + // This is because the AudioUnit will automatically scale other properties, except for sample rate + err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat)); + if(err != noErr) + { + AL_PRINT("AudioUnitSetProperty failed\n"); + goto error; + } + + // Set the AudioUnit output format frame count + outputFrameCount = device->UpdateSize * data->sampleRateRatio; + err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount)); + if(err != noErr) + { + AL_PRINT("AudioUnitSetProperty failed: %d\n", err); + goto error; + } + + // Set up sample converter + err = AudioConverterNew(&outputFormat, &requestedFormat, &data->audioConverter); + if(err != noErr) + { + AL_PRINT("AudioConverterNew failed: %d\n", err); + goto error; + } + + // Create a buffer for use in the resample callback + data->resampleBuffer = malloc(device->UpdateSize * data->frameSize * data->sampleRateRatio); + + // Allocate buffer for the AudioUnit output + data->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * data->frameSize * data->sampleRateRatio); + if(data->bufferList == NULL) + { + alcSetError(device, ALC_OUT_OF_MEMORY); + goto error; + } + + data->ring = CreateRingBuffer(data->frameSize, (device->UpdateSize * data->sampleRateRatio) * device->NumUpdates); + if(data->ring == NULL) + { + alcSetError(device, ALC_OUT_OF_MEMORY); + goto error; + } + + return ALC_TRUE; + +error: + DestroyRingBuffer(data->ring); + free(data->resampleBuffer); + destroy_buffer_list(data->bufferList); + + if(data->audioConverter) + AudioConverterDispose(data->audioConverter); + if(data->audioUnit) + CloseComponent(data->audioUnit); + + free(data); + device->ExtraData = NULL; + return ALC_FALSE; - (void)device; - (void)deviceName; +} + +static void ca_close_capture(ALCdevice *device) +{ + ca_data *data = (ca_data*)device->ExtraData; + + DestroyRingBuffer(data->ring); + free(data->resampleBuffer); + destroy_buffer_list(data->bufferList); + + AudioConverterDispose(data->audioConverter); + CloseComponent(data->audioUnit); + + free(data); + device->ExtraData = NULL; +} + +static void ca_start_capture(ALCdevice *device) +{ + ca_data *data = (ca_data*)device->ExtraData; + OSStatus err = AudioOutputUnitStart(data->audioUnit); + if(err != noErr) + AL_PRINT("AudioOutputUnitStart failed\n"); +} + +static void ca_stop_capture(ALCdevice *device) +{ + ca_data *data = (ca_data*)device->ExtraData; + OSStatus err = AudioOutputUnitStop(data->audioUnit); + if(err != noErr) + AL_PRINT("AudioOutputUnitStop failed\n"); +} + +static ALCuint ca_available_samples(ALCdevice *device) +{ + ca_data *data = device->ExtraData; + return RingBufferSize(data->ring) / data->sampleRateRatio; +} + +static void ca_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples) +{ + ca_data *data = (ca_data*)device->ExtraData; + + if(samples <= ca_available_samples(device)) + { + AudioBufferList *list; + UInt32 frameCount; + OSStatus err; + + // If no samples are requested, just return + if(samples == 0) + return; + + // Allocate a temporary AudioBufferList to use as the return resamples data + list = alloca(sizeof(AudioBufferList) + sizeof(AudioBuffer)); + + // Point the resampling buffer to the capture buffer + list->mNumberBuffers = 1; + list->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame; + list->mBuffers[0].mDataByteSize = samples * data->frameSize; + list->mBuffers[0].mData = buffer; + + // Resample into another AudioBufferList + frameCount = samples; + err = AudioConverterFillComplexBuffer(data->audioConverter, ca_capture_conversion_callback, device, + &frameCount, list, NULL); + if(err != noErr) + { + AL_PRINT("AudioConverterFillComplexBuffer error: %d\n", err); + alcSetError(device, ALC_INVALID_VALUE); + } + } + else + alcSetError(device, ALC_INVALID_VALUE); } static const BackendFuncs ca_funcs = { @@ -308,11 +686,11 @@ static const BackendFuncs ca_funcs = { ca_reset_playback, ca_stop_playback, ca_open_capture, - NULL, - NULL, - NULL, - NULL, - NULL + ca_close_capture, + ca_start_capture, + ca_stop_capture, + ca_capture_samples, + ca_available_samples }; void alc_ca_init(BackendFuncs *func_list) @@ -335,6 +713,7 @@ void alc_ca_probe(enum DevProbe type) AppendAllDeviceList(ca_device); break; case CAPTURE_DEVICE_PROBE: + AppendCaptureDeviceList(ca_device); break; } } |