Implement capture support for the CoreAudio backend
This commit is contained in:
Chris Robinson
parent
ea83608ee4
commit
3f0214ed6b
425
Alc/coreaudio.c
425
Alc/coreaudio.c
@ -31,25 +31,107 @@
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#include <CoreServices/CoreServices.h>
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#include <unistd.h>
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#include <AudioUnit/AudioUnit.h>
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#include <AudioToolbox/AudioToolbox.h>
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/* toggle verbose tty output among CoreAudio code */
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#define CA_VERBOSE 1
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typedef struct {
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AudioUnit OutputUnit;
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ALuint FrameSize;
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AudioUnit audioUnit;
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ALuint frameSize;
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ALdouble sampleRateRatio; // Ratio of hardware sample rate / requested sample rate
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AudioStreamBasicDescription format; // This is the OpenAL format as a CoreAudio ASBD
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AudioConverterRef audioConverter; // Sample rate converter if needed
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AudioBufferList *bufferList; // Buffer for data coming from the input device
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ALCvoid *resampleBuffer; // Buffer for returned RingBuffer data when resampling
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RingBuffer *ring;
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} ca_data;
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static const ALCchar ca_device[] = "CoreAudio Default";
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static int ca_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp,
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UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData)
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static void destroy_buffer_list(AudioBufferList* list)
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{
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if(list)
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{
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for(UInt32 i = 0;i < list->mNumberBuffers;i++)
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free(list->mBuffers[i].mData);
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free(list);
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}
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}
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static AudioBufferList* allocate_buffer_list(UInt32 channelCount, UInt32 byteSize)
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{
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AudioBufferList *list;
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list = calloc(1, sizeof(AudioBufferList) + sizeof(AudioBuffer));
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if(list)
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{
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list->mNumberBuffers = 1;
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list->mBuffers[0].mNumberChannels = channelCount;
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list->mBuffers[0].mDataByteSize = byteSize;
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list->mBuffers[0].mData = malloc(byteSize);
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if(list->mBuffers[0].mData == NULL)
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{
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free(list);
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list = NULL;
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}
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}
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return list;
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}
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static OSStatus ca_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp,
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UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData)
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{
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ALCdevice *device = (ALCdevice*)inRefCon;
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ca_data *data = (ca_data*)device->ExtraData;
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aluMixData(device, ioData->mBuffers[0].mData,
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ioData->mBuffers[0].mDataByteSize / data->FrameSize);
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ioData->mBuffers[0].mDataByteSize / data->frameSize);
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return noErr;
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}
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static OSStatus ca_capture_conversion_callback(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets,
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AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void* inUserData)
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{
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ALCdevice *device = (ALCdevice*)inUserData;
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ca_data *data = (ca_data*)device->ExtraData;
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// Read from the ring buffer and store temporarily in a large buffer
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ReadRingBuffer(data->ring, data->resampleBuffer, (ALsizei)(*ioNumberDataPackets));
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// Set the input data
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ioData->mNumberBuffers = 1;
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ioData->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
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ioData->mBuffers[0].mData = data->resampleBuffer;
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ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * data->format.mBytesPerFrame;
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return noErr;
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}
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static OSStatus ca_capture_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags,
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const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber,
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UInt32 inNumberFrames, AudioBufferList *ioData)
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{
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ALCdevice *device = (ALCdevice*)inRefCon;
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ca_data *data = (ca_data*)device->ExtraData;
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AudioUnitRenderActionFlags flags = 0;
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OSStatus err;
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// fill the bufferList with data from the input device
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err = AudioUnitRender(data->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, data->bufferList);
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if(err != noErr)
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{
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AL_PRINT("AudioUnitRender error: %d\n", err);
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return err;
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}
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WriteRingBuffer(data->ring, data->bufferList->mBuffers[0].mData, inNumberFrames);
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return noErr;
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}
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@ -83,7 +165,7 @@ static ALCboolean ca_open_playback(ALCdevice *device, const ALCchar *deviceName)
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data = calloc(1, sizeof(*data));
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device->ExtraData = data;
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err = OpenAComponent(comp, &data->OutputUnit);
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err = OpenAComponent(comp, &data->audioUnit);
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if(err != noErr)
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{
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AL_PRINT("OpenAComponent failed\n");
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@ -99,7 +181,7 @@ static void ca_close_playback(ALCdevice *device)
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{
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ca_data *data = (ca_data*)device->ExtraData;
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CloseComponent(data->OutputUnit);
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CloseComponent(data->audioUnit);
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free(data);
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device->ExtraData = NULL;
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@ -114,14 +196,14 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
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UInt32 size;
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/* init and start the default audio unit... */
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err = AudioUnitInitialize(data->OutputUnit);
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err = AudioUnitInitialize(data->audioUnit);
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if(err != noErr)
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{
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AL_PRINT("AudioUnitInitialize failed\n");
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return ALC_FALSE;
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}
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err = AudioOutputUnitStart(data->OutputUnit);
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err = AudioOutputUnitStart(data->audioUnit);
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if(err != noErr)
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{
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AL_PRINT("AudioOutputUnitStart failed\n");
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@ -130,7 +212,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
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/* retrieve default output unit's properties (output side) */
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size = sizeof(AudioStreamBasicDescription);
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err = AudioUnitGetProperty(data->OutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
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err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
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if(err != noErr || size != sizeof(AudioStreamBasicDescription))
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{
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AL_PRINT("AudioUnitGetProperty failed\n");
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@ -148,7 +230,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
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#endif
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/* set default output unit's input side to match output side */
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err = AudioUnitSetProperty(data->OutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
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err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
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if(err != noErr)
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{
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AL_PRINT("AudioUnitSetProperty failed\n");
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@ -262,7 +344,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
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kAudioFormatFlagsNativeEndian |
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kLinearPCMFormatFlagIsPacked;
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err = AudioUnitSetProperty(data->OutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
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err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
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if(err != noErr)
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{
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AL_PRINT("AudioUnitSetProperty failed\n");
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@ -270,11 +352,11 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
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}
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/* setup callback */
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data->FrameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
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data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
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input.inputProc = ca_callback;
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input.inputProcRefCon = device;
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err = AudioUnitSetProperty(data->OutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
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err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
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if(err != noErr)
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{
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AL_PRINT("AudioUnitSetProperty failed\n");
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@ -289,17 +371,313 @@ static void ca_stop_playback(ALCdevice *device)
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ca_data *data = (ca_data*)device->ExtraData;
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OSStatus err;
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AudioOutputUnitStop(data->OutputUnit);
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err = AudioUnitUninitialize(data->OutputUnit);
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AudioOutputUnitStop(data->audioUnit);
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err = AudioUnitUninitialize(data->audioUnit);
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if(err != noErr)
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AL_PRINT("-- AudioUnitUninitialize failed.\n");
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}
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static ALCboolean ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
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{
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AudioStreamBasicDescription requestedFormat; // The application requested format
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AudioStreamBasicDescription hardwareFormat; // The hardware format
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AudioStreamBasicDescription outputFormat; // The AudioUnit output format
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AURenderCallbackStruct input;
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ComponentDescription desc;
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AudioDeviceID inputDevice;
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UInt32 outputFrameCount;
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UInt32 propertySize;
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UInt32 enableIO;
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Component comp;
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ca_data *data;
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OSStatus err;
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desc.componentType = kAudioUnitType_Output;
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desc.componentSubType = kAudioUnitSubType_HALOutput;
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desc.componentManufacturer = kAudioUnitManufacturer_Apple;
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desc.componentFlags = 0;
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desc.componentFlagsMask = 0;
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// Search for component with given description
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comp = FindNextComponent(NULL, &desc);
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if(comp == NULL)
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{
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AL_PRINT("FindNextComponent failed\n");
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return ALC_FALSE;
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}
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data = calloc(1, sizeof(*data));
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device->ExtraData = data;
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// Open the component
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err = OpenAComponent(comp, &data->audioUnit);
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if(err != noErr)
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{
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AL_PRINT("OpenAComponent failed\n");
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goto error;
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}
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// Turn off AudioUnit output
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enableIO = 0;
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err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
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if(err != noErr)
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{
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AL_PRINT("AudioUnitSetProperty failed\n");
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goto error;
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}
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// Turn on AudioUnit input
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enableIO = 1;
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err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
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if(err != noErr)
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{
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AL_PRINT("AudioUnitSetProperty failed\n");
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goto error;
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}
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// Get the default input device
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propertySize = sizeof(AudioDeviceID);
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err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propertySize, &inputDevice);
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if(err != noErr)
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{
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AL_PRINT("AudioHardwareGetProperty failed\n");
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goto error;
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}
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if(inputDevice == kAudioDeviceUnknown)
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{
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AL_PRINT("No input device found\n");
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goto error;
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}
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// Track the input device
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err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
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if(err != noErr)
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{
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AL_PRINT("AudioUnitSetProperty failed\n");
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goto error;
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}
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// set capture callback
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input.inputProc = ca_capture_callback;
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input.inputProcRefCon = device;
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err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
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if(err != noErr)
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{
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AL_PRINT("AudioUnitSetProperty failed\n");
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goto error;
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}
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// Initialize the device
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err = AudioUnitInitialize(data->audioUnit);
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if(err != noErr)
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{
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AL_PRINT("AudioUnitInitialize failed\n");
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goto error;
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}
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// Get the hardware format
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propertySize = sizeof(AudioStreamBasicDescription);
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err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
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if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription))
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{
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AL_PRINT("AudioUnitGetProperty failed\n");
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goto error;
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}
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// Set up the requested format description
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switch(device->FmtType)
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{
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case DevFmtUByte:
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requestedFormat.mBitsPerChannel = 8;
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requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
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break;
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case DevFmtShort:
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requestedFormat.mBitsPerChannel = 16;
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requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
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break;
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case DevFmtFloat:
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requestedFormat.mBitsPerChannel = 32;
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requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
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break;
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case DevFmtByte:
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case DevFmtUShort:
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AL_PRINT("%s samples not supported\n", DevFmtTypeString(device->FmtType));
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goto error;
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}
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switch(device->FmtChans)
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{
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case DevFmtMono:
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requestedFormat.mChannelsPerFrame = 1;
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break;
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case DevFmtStereo:
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requestedFormat.mChannelsPerFrame = 2;
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break;
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case DevFmtQuad:
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case DevFmtX51:
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case DevFmtX51Side:
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case DevFmtX61:
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case DevFmtX71:
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AL_PRINT("%s not supported\n", DevFmtChannelsString(device->FmtChans));
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goto error;
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}
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requestedFormat.mBytesPerFrame = requestedFormat.mChannelsPerFrame * requestedFormat.mBitsPerChannel / 8;
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requestedFormat.mBytesPerPacket = requestedFormat.mBytesPerFrame;
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requestedFormat.mSampleRate = device->Frequency;
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requestedFormat.mFormatID = kAudioFormatLinearPCM;
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requestedFormat.mReserved = 0;
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requestedFormat.mFramesPerPacket = 1;
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// save requested format description for later use
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data->format = requestedFormat;
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data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
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// Use intermediate format for sample rate conversion (outputFormat)
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// Set sample rate to the same as hardware for resampling later
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outputFormat = requestedFormat;
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outputFormat.mSampleRate = hardwareFormat.mSampleRate;
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// Determine sample rate ratio for resampling
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data->sampleRateRatio = outputFormat.mSampleRate / device->Frequency;
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// The output format should be the requested format, but using the hardware sample rate
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// This is because the AudioUnit will automatically scale other properties, except for sample rate
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err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
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if(err != noErr)
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{
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AL_PRINT("AudioUnitSetProperty failed\n");
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goto error;
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}
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// Set the AudioUnit output format frame count
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outputFrameCount = device->UpdateSize * data->sampleRateRatio;
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err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount));
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if(err != noErr)
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{
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AL_PRINT("AudioUnitSetProperty failed: %d\n", err);
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goto error;
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}
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// Set up sample converter
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err = AudioConverterNew(&outputFormat, &requestedFormat, &data->audioConverter);
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if(err != noErr)
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{
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AL_PRINT("AudioConverterNew failed: %d\n", err);
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goto error;
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}
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// Create a buffer for use in the resample callback
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data->resampleBuffer = malloc(device->UpdateSize * data->frameSize * data->sampleRateRatio);
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// Allocate buffer for the AudioUnit output
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data->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * data->frameSize * data->sampleRateRatio);
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if(data->bufferList == NULL)
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{
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alcSetError(device, ALC_OUT_OF_MEMORY);
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goto error;
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}
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data->ring = CreateRingBuffer(data->frameSize, (device->UpdateSize * data->sampleRateRatio) * device->NumUpdates);
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if(data->ring == NULL)
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{
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alcSetError(device, ALC_OUT_OF_MEMORY);
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goto error;
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}
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return ALC_TRUE;
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error:
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DestroyRingBuffer(data->ring);
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free(data->resampleBuffer);
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destroy_buffer_list(data->bufferList);
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if(data->audioConverter)
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AudioConverterDispose(data->audioConverter);
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if(data->audioUnit)
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CloseComponent(data->audioUnit);
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free(data);
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device->ExtraData = NULL;
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return ALC_FALSE;
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(void)device;
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(void)deviceName;
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}
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static void ca_close_capture(ALCdevice *device)
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{
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ca_data *data = (ca_data*)device->ExtraData;
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DestroyRingBuffer(data->ring);
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free(data->resampleBuffer);
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destroy_buffer_list(data->bufferList);
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AudioConverterDispose(data->audioConverter);
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CloseComponent(data->audioUnit);
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free(data);
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device->ExtraData = NULL;
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}
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static void ca_start_capture(ALCdevice *device)
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{
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ca_data *data = (ca_data*)device->ExtraData;
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OSStatus err = AudioOutputUnitStart(data->audioUnit);
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if(err != noErr)
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AL_PRINT("AudioOutputUnitStart failed\n");
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}
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static void ca_stop_capture(ALCdevice *device)
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{
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ca_data *data = (ca_data*)device->ExtraData;
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OSStatus err = AudioOutputUnitStop(data->audioUnit);
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if(err != noErr)
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AL_PRINT("AudioOutputUnitStop failed\n");
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}
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|
||||
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;
|
||||
}
|
||||
}
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user