diff options
Diffstat (limited to 'Alc/backends/coreaudio.c')
| -rw-r--r-- | Alc/backends/coreaudio.c | 719 |
1 files changed, 719 insertions, 0 deletions
diff --git a/Alc/backends/coreaudio.c b/Alc/backends/coreaudio.c new file mode 100644 index 00000000..c84be846 --- /dev/null +++ b/Alc/backends/coreaudio.c @@ -0,0 +1,719 @@ +/** + * OpenAL cross platform audio library + * Copyright (C) 1999-2007 by authors. + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + * Or go to http://www.gnu.org/copyleft/lgpl.html + */ + +#include "config.h" + +#include <stdio.h> +#include <stdlib.h> +#include <string.h> + +#include "alMain.h" +#include "AL/al.h" +#include "AL/alc.h" + +#include <CoreServices/CoreServices.h> +#include <unistd.h> +#include <AudioUnit/AudioUnit.h> +#include <AudioToolbox/AudioToolbox.h> + + +typedef struct { + 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 void destroy_buffer_list(AudioBufferList* list) +{ + if(list) + { + UInt32 i; + for(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); + + 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) + { + ERR("AudioUnitRender error: %d\n", err); + return err; + } + + WriteRingBuffer(data->ring, data->bufferList->mBuffers[0].mData, inNumberFrames); + + return noErr; +} + +static ALCboolean ca_open_playback(ALCdevice *device, const ALCchar *deviceName) +{ + ComponentDescription desc; + Component comp; + ca_data *data; + OSStatus err; + + if(!deviceName) + deviceName = ca_device; + else if(strcmp(deviceName, ca_device) != 0) + return ALC_FALSE; + + /* open the default output unit */ + desc.componentType = kAudioUnitType_Output; + desc.componentSubType = kAudioUnitSubType_DefaultOutput; + desc.componentManufacturer = kAudioUnitManufacturer_Apple; + desc.componentFlags = 0; + desc.componentFlagsMask = 0; + + comp = FindNextComponent(NULL, &desc); + if(comp == NULL) + { + ERR("FindNextComponent failed\n"); + return ALC_FALSE; + } + + data = calloc(1, sizeof(*data)); + device->ExtraData = data; + + err = OpenAComponent(comp, &data->audioUnit); + if(err != noErr) + { + ERR("OpenAComponent failed\n"); + free(data); + device->ExtraData = NULL; + return ALC_FALSE; + } + + return ALC_TRUE; +} + +static void ca_close_playback(ALCdevice *device) +{ + ca_data *data = (ca_data*)device->ExtraData; + + CloseComponent(data->audioUnit); + + free(data); + device->ExtraData = NULL; +} + +static ALCboolean ca_reset_playback(ALCdevice *device) +{ + ca_data *data = (ca_data*)device->ExtraData; + AudioStreamBasicDescription streamFormat; + AURenderCallbackStruct input; + OSStatus err; + UInt32 size; + + /* init and start the default audio unit... */ + err = AudioUnitInitialize(data->audioUnit); + if(err != noErr) + { + ERR("AudioUnitInitialize failed\n"); + return ALC_FALSE; + } + + err = AudioOutputUnitStart(data->audioUnit); + if(err != noErr) + { + ERR("AudioOutputUnitStart failed\n"); + return ALC_FALSE; + } + + /* retrieve default output unit's properties (output side) */ + size = sizeof(AudioStreamBasicDescription); + err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size); + if(err != noErr || size != sizeof(AudioStreamBasicDescription)) + { + ERR("AudioUnitGetProperty failed\n"); + return ALC_FALSE; + } + +#if 0 + TRACE("Output streamFormat of default output unit -\n"); + TRACE(" streamFormat.mFramesPerPacket = %d\n", streamFormat.mFramesPerPacket); + TRACE(" streamFormat.mChannelsPerFrame = %d\n", streamFormat.mChannelsPerFrame); + TRACE(" streamFormat.mBitsPerChannel = %d\n", streamFormat.mBitsPerChannel); + TRACE(" streamFormat.mBytesPerPacket = %d\n", streamFormat.mBytesPerPacket); + TRACE(" streamFormat.mBytesPerFrame = %d\n", streamFormat.mBytesPerFrame); + TRACE(" streamFormat.mSampleRate = %5.0f\n", streamFormat.mSampleRate); +#endif + + /* set default output unit's input side to match output side */ + err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size); + if(err != noErr) + { + ERR("AudioUnitSetProperty failed\n"); + return ALC_FALSE; + } + + if(device->Frequency != streamFormat.mSampleRate) + { + if((device->Flags&DEVICE_FREQUENCY_REQUEST)) + ERR("CoreAudio does not support changing sample rates (wanted %dhz, got %dhz)\n", device->Frequency, streamFormat.mSampleRate); + device->Flags &= ~DEVICE_FREQUENCY_REQUEST; + + device->UpdateSize = (ALuint)((ALuint64)device->UpdateSize * + streamFormat.mSampleRate / + device->Frequency); + device->Frequency = streamFormat.mSampleRate; + } + + /* FIXME: How to tell what channels are what in the output device, and how + * to specify what we're giving? eg, 6.0 vs 5.1 */ + switch(streamFormat.mChannelsPerFrame) + { + case 1: + if((device->Flags&DEVICE_CHANNELS_REQUEST) && + device->FmtChans != DevFmtMono) + { + ERR("Failed to set %s, got Mono instead\n", DevFmtChannelsString(device->FmtChans)); + device->Flags &= ~DEVICE_CHANNELS_REQUEST; + } + device->FmtChans = DevFmtMono; + break; + case 2: + if((device->Flags&DEVICE_CHANNELS_REQUEST) && + device->FmtChans != DevFmtStereo) + { + ERR("Failed to set %s, got Stereo instead\n", DevFmtChannelsString(device->FmtChans)); + device->Flags &= ~DEVICE_CHANNELS_REQUEST; + } + device->FmtChans = DevFmtStereo; + break; + case 4: + if((device->Flags&DEVICE_CHANNELS_REQUEST) && + device->FmtChans != DevFmtQuad) + { + ERR("Failed to set %s, got Quad instead\n", DevFmtChannelsString(device->FmtChans)); + device->Flags &= ~DEVICE_CHANNELS_REQUEST; + } + device->FmtChans = DevFmtQuad; + break; + case 6: + if((device->Flags&DEVICE_CHANNELS_REQUEST) && + device->FmtChans != DevFmtX51) + { + ERR("Failed to set %s, got 5.1 Surround instead\n", DevFmtChannelsString(device->FmtChans)); + device->Flags &= ~DEVICE_CHANNELS_REQUEST; + } + device->FmtChans = DevFmtX51; + break; + case 7: + if((device->Flags&DEVICE_CHANNELS_REQUEST) && + device->FmtChans != DevFmtX61) + { + ERR("Failed to set %s, got 6.1 Surround instead\n", DevFmtChannelsString(device->FmtChans)); + device->Flags &= ~DEVICE_CHANNELS_REQUEST; + } + device->FmtChans = DevFmtX61; + break; + case 8: + if((device->Flags&DEVICE_CHANNELS_REQUEST) && + device->FmtChans != DevFmtX71) + { + ERR("Failed to set %s, got 7.1 Surround instead\n", DevFmtChannelsString(device->FmtChans)); + device->Flags &= ~DEVICE_CHANNELS_REQUEST; + } + device->FmtChans = DevFmtX71; + break; + default: + ERR("Unhandled channel count (%d), using Stereo\n", streamFormat.mChannelsPerFrame); + device->Flags &= ~DEVICE_CHANNELS_REQUEST; + device->FmtChans = DevFmtStereo; + streamFormat.mChannelsPerFrame = 2; + break; + } + SetDefaultWFXChannelOrder(device); + + /* use channel count and sample rate from the default output unit's current + * parameters, but reset everything else */ + streamFormat.mFramesPerPacket = 1; + switch(device->FmtType) + { + case DevFmtUByte: + device->FmtType = DevFmtByte; + /* fall-through */ + case DevFmtByte: + streamFormat.mBitsPerChannel = 8; + streamFormat.mBytesPerPacket = streamFormat.mChannelsPerFrame; + streamFormat.mBytesPerFrame = streamFormat.mChannelsPerFrame; + break; + case DevFmtUShort: + case DevFmtFloat: + device->FmtType = DevFmtShort; + /* fall-through */ + case DevFmtShort: + streamFormat.mBitsPerChannel = 16; + streamFormat.mBytesPerPacket = 2 * streamFormat.mChannelsPerFrame; + streamFormat.mBytesPerFrame = 2 * streamFormat.mChannelsPerFrame; + break; + } + streamFormat.mFormatID = kAudioFormatLinearPCM; + streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | + kAudioFormatFlagsNativeEndian | + kLinearPCMFormatFlagIsPacked; + + err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription)); + if(err != noErr) + { + ERR("AudioUnitSetProperty failed\n"); + return ALC_FALSE; + } + + /* setup callback */ + data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType); + input.inputProc = ca_callback; + input.inputProcRefCon = device; + + err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct)); + if(err != noErr) + { + ERR("AudioUnitSetProperty failed\n"); + return ALC_FALSE; + } + + return ALC_TRUE; +} + +static void ca_stop_playback(ALCdevice *device) +{ + ca_data *data = (ca_data*)device->ExtraData; + OSStatus err; + + AudioOutputUnitStop(data->audioUnit); + err = AudioUnitUninitialize(data->audioUnit); + if(err != noErr) + ERR("-- 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) + { + ERR("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) + { + ERR("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) + { + ERR("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) + { + ERR("AudioUnitSetProperty failed\n"); + goto error; + } + + // Get the default input device + propertySize = sizeof(AudioDeviceID); + err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propertySize, &inputDevice); + if(err != noErr) + { + ERR("AudioHardwareGetProperty failed\n"); + goto error; + } + + if(inputDevice == kAudioDeviceUnknown) + { + ERR("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) + { + ERR("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) + { + ERR("AudioUnitSetProperty failed\n"); + goto error; + } + + // Initialize the device + err = AudioUnitInitialize(data->audioUnit); + if(err != noErr) + { + ERR("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)) + { + ERR("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: + ERR("%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: + ERR("%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) + { + ERR("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) + { + ERR("AudioUnitSetProperty failed: %d\n", err); + goto error; + } + + // Set up sample converter + err = AudioConverterNew(&outputFormat, &requestedFormat, &data->audioConverter); + if(err != noErr) + { + ERR("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; +} + +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) + ERR("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) + ERR("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) + { + ERR("AudioConverterFillComplexBuffer error: %d\n", err); + alcSetError(device, ALC_INVALID_VALUE); + } + } + else + alcSetError(device, ALC_INVALID_VALUE); +} + +static const BackendFuncs ca_funcs = { + ca_open_playback, + ca_close_playback, + ca_reset_playback, + ca_stop_playback, + ca_open_capture, + ca_close_capture, + ca_start_capture, + ca_stop_capture, + ca_capture_samples, + ca_available_samples +}; + +ALCboolean alc_ca_init(BackendFuncs *func_list) +{ + *func_list = ca_funcs; + return ALC_TRUE; +} + +void alc_ca_deinit(void) +{ +} + +void alc_ca_probe(enum DevProbe type) +{ + switch(type) + { + case DEVICE_PROBE: + AppendDeviceList(ca_device); + break; + case ALL_DEVICE_PROBE: + AppendAllDeviceList(ca_device); + break; + case CAPTURE_DEVICE_PROBE: + AppendCaptureDeviceList(ca_device); + break; + } +} |