static ALCenum 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_INVALID_VALUE;
}
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)
goto error;
data->ring = CreateRingBuffer(data->frameSize, (device->UpdateSize * data->sampleRateRatio) * device->NumUpdates);
if(data->ring == NULL)
goto error;
return ALC_NO_ERROR;
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_INVALID_VALUE;
}
static ALCenum pa_open_capture(ALCdevice *device, const ALCchar *deviceName)
{
ALuint frame_size;
pa_data *data;
PaError err;
if(!deviceName)
deviceName = pa_device;
else if(strcmp(deviceName, pa_device) != 0)
return ALC_INVALID_VALUE;
data = (pa_data*)calloc(1, sizeof(pa_data));
if(data == NULL)
return ALC_OUT_OF_MEMORY;
frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
data->ring = CreateRingBuffer(frame_size, device->UpdateSize*device->NumUpdates);
if(data->ring == NULL)
goto error;
data->params.device = -1;
if(!ConfigValueInt("port", "capture", &data->params.device) ||
data->params.device < 0)
data->params.device = Pa_GetDefaultOutputDevice();
data->params.suggestedLatency = 0.0f;
data->params.hostApiSpecificStreamInfo = NULL;
switch(device->FmtType)
{
case DevFmtByte:
data->params.sampleFormat = paInt8;
break;
case DevFmtUByte:
data->params.sampleFormat = paUInt8;
break;
case DevFmtShort:
data->params.sampleFormat = paInt16;
break;
case DevFmtInt:
data->params.sampleFormat = paInt32;
break;
case DevFmtFloat:
data->params.sampleFormat = paFloat32;
break;
case DevFmtUInt:
case DevFmtUShort:
ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType));
goto error;
}
data->params.channelCount = ChannelsFromDevFmt(device->FmtChans);
err = Pa_OpenStream(&data->stream, &data->params, NULL, device->Frequency,
paFramesPerBufferUnspecified, paNoFlag, pa_capture_cb, device);
if(err != paNoError)
{
ERR("Pa_OpenStream() returned an error: %s\n", Pa_GetErrorText(err));
goto error;
}
device->DeviceName = strdup(deviceName);
device->ExtraData = data;
return ALC_NO_ERROR;
error:
DestroyRingBuffer(data->ring);
free(data);
return ALC_INVALID_VALUE;
}
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)
{
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:
device->FmtChans = DevFmtMono;
break;
case 2:
device->FmtChans = DevFmtStereo;
break;
case 4:
device->FmtChans = DevFmtQuad;
break;
case 6:
device->FmtChans = DevFmtX51;
break;
case 7:
device->FmtChans = DevFmtX61;
break;
case 8:
device->FmtChans = DevFmtX71;
break;
default:
ERR("Unhandled channel count (%d), using Stereo\n", streamFormat.mChannelsPerFrame);
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 ALCboolean pulse_reset_playback(ALCdevice *device) //{{{
{
pulse_data *data = device->ExtraData;
pa_stream_flags_t flags = 0;
pa_channel_map chanmap;
pa_threaded_mainloop_lock(data->loop);
if(!(device->Flags&DEVICE_CHANNELS_REQUEST))
{
pa_operation *o;
o = pa_context_get_sink_info_by_name(data->context, data->device_name, sink_info_callback, device);
while(pa_operation_get_state(o) == PA_OPERATION_RUNNING)
pa_threaded_mainloop_wait(data->loop);
pa_operation_unref(o);
}
if(!(device->Flags&DEVICE_FREQUENCY_REQUEST))
flags |= PA_STREAM_FIX_RATE;
data->frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
data->attr.prebuf = -1;
data->attr.fragsize = -1;
data->attr.minreq = device->UpdateSize * data->frame_size;
data->attr.tlength = data->attr.minreq * maxu(device->NumUpdates, 2);
data->attr.maxlength = -1;
flags |= PA_STREAM_EARLY_REQUESTS;
flags |= PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE;
switch(device->FmtType)
{
case DevFmtByte:
device->FmtType = DevFmtUByte;
/* fall-through */
case DevFmtUByte:
data->spec.format = PA_SAMPLE_U8;
break;
case DevFmtUShort:
device->FmtType = DevFmtShort;
/* fall-through */
case DevFmtShort:
data->spec.format = PA_SAMPLE_S16NE;
break;
case DevFmtFloat:
data->spec.format = PA_SAMPLE_FLOAT32NE;
break;
}
data->spec.rate = device->Frequency;
data->spec.channels = ChannelsFromDevFmt(device->FmtChans);
if(pa_sample_spec_valid(&data->spec) == 0)
{
ERR("Invalid sample format\n");
pa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
if(!pa_channel_map_init_auto(&chanmap, data->spec.channels, PA_CHANNEL_MAP_WAVEEX))
{
ERR("Couldn't build map for channel count (%d)!\n", data->spec.channels);
pa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
SetDefaultWFXChannelOrder(device);
data->stream = connect_playback_stream(device, flags, &data->attr, &data->spec, &chanmap);
if(!data->stream)
{
pa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
pa_stream_set_state_callback(data->stream, stream_state_callback2, device);
data->spec = *(pa_stream_get_sample_spec(data->stream));
if(device->Frequency != data->spec.rate)
{
pa_operation *o;
if((device->Flags&DEVICE_FREQUENCY_REQUEST))
ERR("Failed to set frequency %dhz, got %dhz instead\n", device->Frequency, data->spec.rate);
device->Flags &= ~DEVICE_FREQUENCY_REQUEST;
/* Server updated our playback rate, so modify the buffer attribs
* accordingly. */
data->attr.minreq = (ALuint64)(data->attr.minreq/data->frame_size) *
data->spec.rate / device->Frequency * data->frame_size;
data->attr.tlength = data->attr.minreq * maxu(device->NumUpdates, 2);
o = pa_stream_set_buffer_attr(data->stream, &data->attr,
stream_success_callback, device);
while(pa_operation_get_state(o) == PA_OPERATION_RUNNING)
pa_threaded_mainloop_wait(data->loop);
pa_operation_unref(o);
device->Frequency = data->spec.rate;
}
stream_buffer_attr_callback(data->stream, device);
if(device->NumUpdates < 2)
{
pa_operation *o;
/* Server gave a comparatively large minreq, so modify the tlength. */
device->NumUpdates = 2;
data->attr.tlength = data->attr.minreq * device->NumUpdates;
o = pa_stream_set_buffer_attr(data->stream, &data->attr,
stream_success_callback, device);
while(pa_operation_get_state(o) == PA_OPERATION_RUNNING)
pa_threaded_mainloop_wait(data->loop);
pa_operation_unref(o);
}
#if PA_CHECK_VERSION(0,9,15)
if(pa_stream_set_buffer_attr_callback)
pa_stream_set_buffer_attr_callback(data->stream, stream_buffer_attr_callback, device);
#endif
pa_stream_set_moved_callback(data->stream, stream_device_callback, device);
pa_stream_set_write_callback(data->stream, stream_write_callback, device);
pa_stream_set_underflow_callback(data->stream, stream_signal_callback, device);
data->thread = StartThread(PulseProc, device);
if(!data->thread)
{
#if PA_CHECK_VERSION(0,9,15)
if(pa_stream_set_buffer_attr_callback)
pa_stream_set_buffer_attr_callback(data->stream, NULL, NULL);
#endif
pa_stream_set_moved_callback(data->stream, NULL, NULL);
pa_stream_set_write_callback(data->stream, NULL, NULL);
pa_stream_set_underflow_callback(data->stream, NULL, NULL);
pa_stream_disconnect(data->stream);
pa_stream_unref(data->stream);
data->stream = NULL;
pa_threaded_mainloop_unlock(data->loop);
return ALC_FALSE;
}
pa_threaded_mainloop_unlock(data->loop);
return ALC_TRUE;
} //}}}
static ALuint DSoundPlaybackProc(ALvoid *ptr)
{
ALCdevice *pDevice = (ALCdevice*)ptr;
DSoundPlaybackData *pData = (DSoundPlaybackData*)pDevice->ExtraData;
DSBCAPS DSBCaps;
DWORD LastCursor = 0;
DWORD PlayCursor;
VOID *WritePtr1, *WritePtr2;
DWORD WriteCnt1, WriteCnt2;
BOOL Playing = FALSE;
DWORD FrameSize;
DWORD FragSize;
DWORD avail;
HRESULT err;
SetRTPriority();
memset(&DSBCaps, 0, sizeof(DSBCaps));
DSBCaps.dwSize = sizeof(DSBCaps);
err = IDirectSoundBuffer_GetCaps(pData->DSsbuffer, &DSBCaps);
if(FAILED(err))
{
ERR("Failed to get buffer caps: 0x%lx\n", err);
aluHandleDisconnect(pDevice);
return 1;
}
FrameSize = FrameSizeFromDevFmt(pDevice->FmtChans, pDevice->FmtType);
FragSize = pDevice->UpdateSize * FrameSize;
IDirectSoundBuffer_GetCurrentPosition(pData->DSsbuffer, &LastCursor, NULL);
while(!pData->killNow)
{
// Get current play cursor
IDirectSoundBuffer_GetCurrentPosition(pData->DSsbuffer, &PlayCursor, NULL);
avail = (PlayCursor-LastCursor+DSBCaps.dwBufferBytes) % DSBCaps.dwBufferBytes;
if(avail < FragSize)
{
if(!Playing)
{
err = IDirectSoundBuffer_Play(pData->DSsbuffer, 0, 0, DSBPLAY_LOOPING);
if(FAILED(err))
{
ERR("Failed to play buffer: 0x%lx\n", err);
aluHandleDisconnect(pDevice);
return 1;
}
Playing = TRUE;
}
avail = WaitForSingleObjectEx(pData->hNotifyEvent, 2000, FALSE);
if(avail != WAIT_OBJECT_0)
ERR("WaitForSingleObjectEx error: 0x%lx\n", avail);
continue;
}
avail -= avail%FragSize;
// Lock output buffer
WriteCnt1 = 0;
WriteCnt2 = 0;
err = IDirectSoundBuffer_Lock(pData->DSsbuffer, LastCursor, avail, &WritePtr1, &WriteCnt1, &WritePtr2, &WriteCnt2, 0);
// If the buffer is lost, restore it and lock
if(err == DSERR_BUFFERLOST)
{
WARN("Buffer lost, restoring...\n");
err = IDirectSoundBuffer_Restore(pData->DSsbuffer);
if(SUCCEEDED(err))
{
Playing = FALSE;
LastCursor = 0;
err = IDirectSoundBuffer_Lock(pData->DSsbuffer, 0, DSBCaps.dwBufferBytes, &WritePtr1, &WriteCnt1, &WritePtr2, &WriteCnt2, 0);
}
}
// Successfully locked the output buffer
if(SUCCEEDED(err))
{
// If we have an active context, mix data directly into output buffer otherwise fill with silence
aluMixData(pDevice, WritePtr1, WriteCnt1/FrameSize);
aluMixData(pDevice, WritePtr2, WriteCnt2/FrameSize);
// Unlock output buffer only when successfully locked
IDirectSoundBuffer_Unlock(pData->DSsbuffer, WritePtr1, WriteCnt1, WritePtr2, WriteCnt2);
}
else
{
ERR("Buffer lock error: %#lx\n", err);
aluHandleDisconnect(pDevice);
return 1;
}
// Update old write cursor location
LastCursor += WriteCnt1+WriteCnt2;
LastCursor %= DSBCaps.dwBufferBytes;
}
return 0;
}
#endif
}
if(sampleRate != device->Frequency)
{
device->NumUpdates = (device->NumUpdates*sampleRate + (device->Frequency>>1)) /
device->Frequency;
device->NumUpdates = maxu(device->NumUpdates, 2);
device->Frequency = sampleRate;
}
device->FmtChans = DevFmtStereo;
device->FmtType = DevFmtShort;
SetDefaultWFXChannelOrder(device);
self->mFrameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
loc_bufq.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
loc_bufq.numBuffers = device->NumUpdates;
#ifdef SL_DATAFORMAT_PCM_EX
SLDataFormat_PCM_EX format_pcm;
format_pcm.formatType = SL_DATAFORMAT_PCM_EX;
format_pcm.numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
format_pcm.sampleRate = device->Frequency * 1000;
format_pcm.bitsPerSample = BytesFromDevFmt(device->FmtType) * 8;
format_pcm.containerSize = format_pcm.bitsPerSample;
format_pcm.channelMask = GetChannelMask(device->FmtChans);
format_pcm.endianness = IS_LITTLE_ENDIAN ? SL_BYTEORDER_LITTLEENDIAN :
SL_BYTEORDER_BIGENDIAN;
static ALCenum pulse_open_capture(ALCdevice *device, const ALCchar *device_name) //{{{
{
char *pulse_name = NULL;
pulse_data *data;
pa_stream_flags_t flags = 0;
pa_stream_state_t state;
pa_channel_map chanmap;
if(!allCaptureDevNameMap)
probe_devices(AL_TRUE);
if(!device_name)
device_name = pulse_device;
else if(strcmp(device_name, pulse_device) != 0)
{
ALuint i;
for(i = 0;i < numCaptureDevNames;i++)
{
if(strcmp(device_name, allCaptureDevNameMap[i].name) == 0)
{
pulse_name = allCaptureDevNameMap[i].device_name;
break;
}
}
if(i == numCaptureDevNames)
return ALC_INVALID_VALUE;
}
if(pulse_open(device, device_name) == ALC_FALSE)
return ALC_INVALID_VALUE;
data = device->ExtraData;
pa_threaded_mainloop_lock(data->loop);
data->samples = device->UpdateSize * device->NumUpdates;
data->frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
data->samples = maxu(data->samples, 100 * device->Frequency / 1000);
if(!(data->ring = CreateRingBuffer(data->frame_size, data->samples)))
{
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
data->attr.minreq = -1;
data->attr.prebuf = -1;
data->attr.maxlength = data->samples * data->frame_size;
data->attr.tlength = -1;
data->attr.fragsize = minu(data->samples, 50*device->Frequency/1000) *
data->frame_size;
data->spec.rate = device->Frequency;
data->spec.channels = ChannelsFromDevFmt(device->FmtChans);
switch(device->FmtType)
{
case DevFmtUByte:
data->spec.format = PA_SAMPLE_U8;
break;
case DevFmtShort:
data->spec.format = PA_SAMPLE_S16NE;
break;
case DevFmtFloat:
data->spec.format = PA_SAMPLE_FLOAT32NE;
break;
case DevFmtByte:
case DevFmtUShort:
ERR("Capture format type %#x capture not supported on PulseAudio\n", device->FmtType);
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
if(pa_sample_spec_valid(&data->spec) == 0)
{
ERR("Invalid sample format\n");
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
if(!pa_channel_map_init_auto(&chanmap, data->spec.channels, PA_CHANNEL_MAP_WAVEEX))
{
ERR("Couldn't build map for channel count (%d)!\n", data->spec.channels);
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
data->stream = pa_stream_new(data->context, "Capture Stream", &data->spec, &chanmap);
if(!data->stream)
{
ERR("pa_stream_new() failed: %s\n",
pa_strerror(pa_context_errno(data->context)));
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
pa_stream_set_state_callback(data->stream, stream_state_callback, data->loop);
flags |= PA_STREAM_START_CORKED|PA_STREAM_ADJUST_LATENCY;
if(pa_stream_connect_record(data->stream, pulse_name, &data->attr, flags) < 0)
{
ERR("Stream did not connect: %s\n",
pa_strerror(pa_context_errno(data->context)));
pa_stream_unref(data->stream);
data->stream = NULL;
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
while((state=pa_stream_get_state(data->stream)) != PA_STREAM_READY)
{
if(!PA_STREAM_IS_GOOD(state))
{
ERR("Stream did not get ready: %s\n",
pa_strerror(pa_context_errno(data->context)));
pa_stream_unref(data->stream);
data->stream = NULL;
pa_threaded_mainloop_unlock(data->loop);
goto fail;
}
pa_threaded_mainloop_wait(data->loop);
}
pa_stream_set_state_callback(data->stream, stream_state_callback2, device);
pa_threaded_mainloop_unlock(data->loop);
return ALC_NO_ERROR;
fail:
pulse_close(device);
return ALC_INVALID_VALUE;
} //}}}
static ALCboolean ALCsolarisBackend_reset(ALCsolarisBackend *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
audio_info_t info;
ALsizei frameSize;
ALsizei numChannels;
AUDIO_INITINFO(&info);
info.play.sample_rate = device->Frequency;
if(device->FmtChans != DevFmtMono)
device->FmtChans = DevFmtStereo;
numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
info.play.channels = numChannels;
switch(device->FmtType)
{
case DevFmtByte:
info.play.precision = 8;
info.play.encoding = AUDIO_ENCODING_LINEAR;
break;
case DevFmtUByte:
info.play.precision = 8;
info.play.encoding = AUDIO_ENCODING_LINEAR8;
break;
case DevFmtUShort:
case DevFmtInt:
case DevFmtUInt:
case DevFmtFloat:
device->FmtType = DevFmtShort;
/* fall-through */
case DevFmtShort:
info.play.precision = 16;
info.play.encoding = AUDIO_ENCODING_LINEAR;
break;
}
frameSize = numChannels * BytesFromDevFmt(device->FmtType);
info.play.buffer_size = device->UpdateSize*device->NumUpdates * frameSize;
if(ioctl(self->fd, AUDIO_SETINFO, &info) < 0)
{
ERR("ioctl failed: %s\n", strerror(errno));
return ALC_FALSE;
}
if(ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != (ALsizei)info.play.channels)
{
ERR("Failed to set %s, got %u channels instead\n", DevFmtChannelsString(device->FmtChans), info.play.channels);
return ALC_FALSE;
}
if(!((info.play.precision == 8 && info.play.encoding == AUDIO_ENCODING_LINEAR8 && device->FmtType == DevFmtUByte) ||
(info.play.precision == 8 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtByte) ||
(info.play.precision == 16 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtShort) ||
(info.play.precision == 32 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtInt)))
{
ERR("Could not set %s samples, got %d (0x%x)\n", DevFmtTypeString(device->FmtType),
info.play.precision, info.play.encoding);
return ALC_FALSE;
}
device->Frequency = info.play.sample_rate;
device->UpdateSize = (info.play.buffer_size/device->NumUpdates) + 1;
SetDefaultChannelOrder(device);
free(self->mix_data);
self->data_size = device->UpdateSize * FrameSizeFromDevFmt(
device->FmtChans, device->FmtType, device->AmbiOrder
);
self->mix_data = calloc(1, self->data_size);
return ALC_TRUE;
}
static ALuint WaveProc(ALvoid *ptr)
{
ALCdevice *Device = (ALCdevice*)ptr;
wave_data *data = (wave_data*)Device->ExtraData;
ALuint frameSize;
ALuint now, start;
ALuint64 avail, done;
size_t fs;
const ALuint restTime = (ALuint64)Device->UpdateSize * 1000 /
Device->Frequency / 2;
frameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
done = 0;
start = timeGetTime();
while(!data->killNow && Device->Connected)
{
now = timeGetTime();
avail = (ALuint64)(now-start) * Device->Frequency / 1000;
if(avail < done)
{
/* Timer wrapped (50 days???). Add the remainder of the cycle to
* the available count and reset the number of samples done */
avail += ((ALuint64)1<<32)*Device->Frequency/1000 - done;
done = 0;
}
if(avail-done < Device->UpdateSize)
{
Sleep(restTime);
continue;
}
while(avail-done >= Device->UpdateSize)
{
aluMixData(Device, data->buffer, Device->UpdateSize);
done += Device->UpdateSize;
if(!IS_LITTLE_ENDIAN)
{
ALuint bytesize = BytesFromDevFmt(Device->FmtType);
ALubyte *bytes = data->buffer;
ALuint i;
if(bytesize == 1)
{
for(i = 0;i < data->size;i++)
fputc(bytes[i], data->f);
}
else if(bytesize == 2)
{
for(i = 0;i < data->size;i++)
fputc(bytes[i^1], data->f);
}
else if(bytesize == 4)
{
for(i = 0;i < data->size;i++)
fputc(bytes[i^3], data->f);
}
}
else
{
fs = fwrite(data->buffer, frameSize, Device->UpdateSize,
data->f);
fs = fs;
}
if(ferror(data->f))
{
ERR("Error writing to file\n");
ALCdevice_Lock(Device);
aluHandleDisconnect(Device);
ALCdevice_Unlock(Device);
break;
}
}
}
return 0;
}
static ALCboolean opensl_start_playback(ALCdevice *Device)
{
osl_data *data = Device->ExtraData;
SLAndroidSimpleBufferQueueItf bufferQueue;
SLPlayItf player;
SLresult result;
ALuint i;
result = VCALL(data->bufferQueueObject,GetInterface)(SL_IID_BUFFERQUEUE, &bufferQueue);
PRINTERR(result, "bufferQueue->GetInterface");
if(SL_RESULT_SUCCESS == result)
{
result = VCALL(bufferQueue,RegisterCallback)(opensl_callback, Device);
PRINTERR(result, "bufferQueue->RegisterCallback");
}
if(SL_RESULT_SUCCESS == result)
{
data->frameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
data->bufferSize = Device->UpdateSize * data->frameSize;
data->buffer = calloc(Device->NumUpdates, data->bufferSize);
if(!data->buffer)
{
result = SL_RESULT_MEMORY_FAILURE;
PRINTERR(result, "calloc");
}
}
/* enqueue the first buffer to kick off the callbacks */
for(i = 0;i < Device->NumUpdates;i++)
{
if(SL_RESULT_SUCCESS == result)
{
ALvoid *buf = (ALbyte*)data->buffer + i*data->bufferSize;
result = VCALL(bufferQueue,Enqueue)(buf, data->bufferSize);
PRINTERR(result, "bufferQueue->Enqueue");
}
}
data->curBuffer = 0;
if(SL_RESULT_SUCCESS == result)
{
result = VCALL(data->bufferQueueObject,GetInterface)(SL_IID_PLAY, &player);
PRINTERR(result, "bufferQueue->GetInterface");
}
if(SL_RESULT_SUCCESS == result)
{
result = VCALL(player,SetPlayState)(SL_PLAYSTATE_PLAYING);
PRINTERR(result, "player->SetPlayState");
}
if(SL_RESULT_SUCCESS != result)
{
if(data->bufferQueueObject != NULL)
VCALL0(data->bufferQueueObject,Destroy)();
data->bufferQueueObject = NULL;
free(data->buffer);
data->buffer = NULL;
data->bufferSize = 0;
return ALC_FALSE;
}
return ALC_TRUE;
}
static int ALCsolarisBackend_mixerProc(void *ptr)
{
ALCsolarisBackend *self = ptr;
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
struct timeval timeout;
ALubyte *write_ptr;
ALint frame_size;
ALint to_write;
ssize_t wrote;
fd_set wfds;
int sret;
SetRTPriority();
althrd_setname(althrd_current(), MIXER_THREAD_NAME);
frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
ALCsolarisBackend_lock(self);
while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
ATOMIC_LOAD(&device->Connected, almemory_order_acquire) != DeviceConnect_Disconnected)
{
FD_ZERO(&wfds);
FD_SET(self->fd, &wfds);
timeout.tv_sec = 1;
timeout.tv_usec = 0;
ALCsolarisBackend_unlock(self);
sret = select(self->fd+1, NULL, &wfds, NULL, &timeout);
ALCsolarisBackend_lock(self);
if(sret < 0)
{
if(errno == EINTR)
continue;
ERR("select failed: %s\n", strerror(errno));
aluHandleDisconnect(device, "Failed to wait for playback buffer: %s", strerror(errno));
break;
}
else if(sret == 0)
{
WARN("select timeout\n");
continue;
}
write_ptr = self->mix_data;
to_write = self->data_size;
aluMixData(device, write_ptr, to_write/frame_size);
while(to_write > 0 && !ATOMIC_LOAD_SEQ(&self->killNow))
{
wrote = write(self->fd, write_ptr, to_write);
if(wrote < 0)
{
if(errno == EAGAIN || errno == EWOULDBLOCK || errno == EINTR)
continue;
ERR("write failed: %s\n", strerror(errno));
aluHandleDisconnect(device, "Failed to write playback samples: %s",
strerror(errno));
break;
}
to_write -= wrote;
write_ptr += wrote;
}
}
ALCsolarisBackend_unlock(self);
return 0;
}
static ALCboolean opensl_reset_playback(ALCdevice *Device)
{
osl_data *data = Device->ExtraData;
SLDataLocator_AndroidSimpleBufferQueue loc_bufq;
SLAndroidSimpleBufferQueueItf bufferQueue;
SLDataLocator_OutputMix loc_outmix;
SLDataFormat_PCM format_pcm;
SLDataSource audioSrc;
SLDataSink audioSnk;
SLPlayItf player;
SLInterfaceID id;
SLboolean req;
SLresult result;
ALuint i;
Device->UpdateSize = (ALuint64)Device->UpdateSize * 44100 / Device->Frequency;
Device->UpdateSize = Device->UpdateSize * Device->NumUpdates / 2;
Device->NumUpdates = 2;
Device->Frequency = 44100;
Device->FmtChans = DevFmtStereo;
Device->FmtType = DevFmtShort;
SetDefaultWFXChannelOrder(Device);
id = SL_IID_ANDROIDSIMPLEBUFFERQUEUE;
req = SL_BOOLEAN_TRUE;
loc_bufq.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
loc_bufq.numBuffers = Device->NumUpdates;
format_pcm.formatType = SL_DATAFORMAT_PCM;
format_pcm.numChannels = ChannelsFromDevFmt(Device->FmtChans);
format_pcm.samplesPerSec = Device->Frequency * 1000;
format_pcm.bitsPerSample = BytesFromDevFmt(Device->FmtType) * 8;
format_pcm.containerSize = format_pcm.bitsPerSample;
format_pcm.channelMask = GetChannelMask(Device->FmtChans);
format_pcm.endianness = SL_BYTEORDER_NATIVE;
audioSrc.pLocator = &loc_bufq;
audioSrc.pFormat = &format_pcm;
loc_outmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
loc_outmix.outputMix = data->outputMix;
audioSnk.pLocator = &loc_outmix;
audioSnk.pFormat = NULL;
result = SLEngineItf_CreateAudioPlayer(data->engine, &data->bufferQueueObject, &audioSrc, &audioSnk, 1, &id, &req);
PRINTERR(result, "engine->CreateAudioPlayer");
if(SL_RESULT_SUCCESS == result)
{
result = SLObjectItf_Realize(data->bufferQueueObject, SL_BOOLEAN_FALSE);
PRINTERR(result, "bufferQueue->Realize");
}
if(SL_RESULT_SUCCESS == result)
{
result = SLObjectItf_GetInterface(data->bufferQueueObject, SL_IID_BUFFERQUEUE, &bufferQueue);
PRINTERR(result, "bufferQueue->GetInterface");
}
if(SL_RESULT_SUCCESS == result)
{
result = (*bufferQueue)->RegisterCallback(bufferQueue, opensl_callback, Device);
PRINTERR(result, "bufferQueue->RegisterCallback");
}
if(SL_RESULT_SUCCESS == result)
{
data->frameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
data->bufferSize = Device->UpdateSize * data->frameSize;
data->buffer = calloc(1, data->bufferSize);
if(!data->buffer)
{
result = SL_RESULT_MEMORY_FAILURE;
PRINTERR(result, "calloc");
}
}
/* enqueue the first buffer to kick off the callbacks */
for(i = 0;i < Device->NumUpdates;i++)
{
if(SL_RESULT_SUCCESS == result)
{
result = (*bufferQueue)->Enqueue(bufferQueue, data->buffer, data->bufferSize);
PRINTERR(result, "bufferQueue->Enqueue");
}
}
if(SL_RESULT_SUCCESS == result)
{
result = SLObjectItf_GetInterface(data->bufferQueueObject, SL_IID_PLAY, &player);
PRINTERR(result, "bufferQueue->GetInterface");
}
if(SL_RESULT_SUCCESS == result)
{
result = SLPlayItf_SetPlayState(player, SL_PLAYSTATE_PLAYING);
PRINTERR(result, "player->SetPlayState");
}
if(SL_RESULT_SUCCESS != result)
{
if(data->bufferQueueObject != NULL)
SLObjectItf_Destroy(data->bufferQueueObject);
data->bufferQueueObject = NULL;
free(data->buffer);
data->buffer = NULL;
data->bufferSize = 0;
return ALC_FALSE;
}
return ALC_TRUE;
}
ALCboolean xaudio2_do_reset_playback(ALCdevice * device)
{
XAudio2Data* data = (XAudio2Data*)device->ExtraData;
HRESULT hr;
DWORD outputchannelMasks;
XAUDIO2_VOICE_DETAILS outputDetails;
WAVEFORMATEXTENSIBLE SourceType;
data->masterVoice->GetChannelMask(&outputchannelMasks);
data->masterVoice->GetVoiceDetails(&outputDetails);
if (!(device->Flags&DEVICE_FREQUENCY_REQUEST))
device->Frequency = outputDetails.InputSampleRate;
if (!(device->Flags&DEVICE_CHANNELS_REQUEST))
{
if (outputDetails.InputChannels == 1 && outputchannelMasks == MONO)
device->FmtChans = DevFmtMono;
else if (outputDetails.InputChannels == 2 && outputchannelMasks == STEREO)
device->FmtChans = DevFmtStereo;
else if (outputDetails.InputChannels == 4 && outputchannelMasks == QUAD)
device->FmtChans = DevFmtQuad;
else if (outputDetails.InputChannels == 6 && outputchannelMasks == X5DOT1)
device->FmtChans = DevFmtX51;
else if (outputDetails.InputChannels == 6 && outputchannelMasks == X5DOT1SIDE)
device->FmtChans = DevFmtX51Side;
else if (outputDetails.InputChannels == 7 && outputchannelMasks == X6DOT1)
device->FmtChans = DevFmtX61;
else if (outputDetails.InputChannels == 8 && outputchannelMasks == X7DOT1)
device->FmtChans = DevFmtX71;
else
ERR("Unhandled channel config: %d -- 0x%08lx\n", outputDetails.InputChannels, outputchannelMasks);
}
SetDefaultWFXChannelOrder(device);
switch (device->FmtChans)
{
case DevFmtMono:
SourceType.Format.nChannels = 1;
SourceType.dwChannelMask = MONO;
break;
case DevFmtStereo:
SourceType.Format.nChannels = 2;
SourceType.dwChannelMask = STEREO;
break;
case DevFmtQuad:
SourceType.Format.nChannels = 4;
SourceType.dwChannelMask = QUAD;
break;
case DevFmtX51:
SourceType.Format.nChannels = 6;
SourceType.dwChannelMask = X5DOT1;
break;
case DevFmtX51Side:
SourceType.Format.nChannels = 6;
SourceType.dwChannelMask = X5DOT1SIDE;
break;
case DevFmtX61:
SourceType.Format.nChannels = 7;
SourceType.dwChannelMask = X6DOT1;
break;
case DevFmtX71:
SourceType.Format.nChannels = 8;
SourceType.dwChannelMask = X7DOT1;
break;
}
switch (device->FmtType)
{
case DevFmtByte:
device->FmtType = DevFmtUByte;
/* fall-through */
case DevFmtUByte:
SourceType.Format.wBitsPerSample = 8;
SourceType.Samples.wValidBitsPerSample = 8;
SourceType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
break;
case DevFmtUShort:
device->FmtType = DevFmtShort;
/* fall-through */
case DevFmtShort:
SourceType.Format.wBitsPerSample = 16;
SourceType.Samples.wValidBitsPerSample = 16;
SourceType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
break;
case DevFmtFloat:
SourceType.Format.wBitsPerSample = 32;
SourceType.Samples.wValidBitsPerSample = 32;
SourceType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
break;
}
SourceType.Format.cbSize = (sizeof(SourceType) - sizeof(WAVEFORMATEX));
SourceType.Format.nSamplesPerSec = device->Frequency;
SourceType.Format.nBlockAlign = SourceType.Format.nChannels *
SourceType.Format.wBitsPerSample / 8;
SourceType.Format.nAvgBytesPerSec = SourceType.Format.nSamplesPerSec *
SourceType.Format.nBlockAlign;
SourceType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
/*-----------create source voice-------------*/
data->sourceVoiceCallback = new (std::nothrow) XAudio2SourceVoiceCallback(device);
if (data->sourceVoiceCallback == NULL)
{
ERR("create XAudio2SourceVoiceCallback() failed\n");
return ALC_FALSE;
}
data->sourceVoiceCallback->callbackEvent = CreateEventEx(NULL, NULL, 0, EVENT_ACCESS_MASK);
if (data->sourceVoiceCallback->callbackEvent == NULL)
{
ERR("create callback event failed\n");
return ALC_FALSE;
}
hr = data->xAudioObj->CreateSourceVoice(&data->sourceVoice, &SourceType.Format, 0,
XAUDIO2_DEFAULT_FREQ_RATIO, data->sourceVoiceCallback);
if (FAILED(hr))
{
ERR("CreateSourceVoice() failed: 0x%08lx\n", hr);
return ALC_FALSE;
}
/*---------create buffer--------------*/
//calculate frame size
data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
//start playing the audio engine
data->sourceVoice->Start();
//wait for the buffer to be created
WaitForSingleObjectEx(data->sourceVoiceCallback->callbackEvent, INFINITE, FALSE);
device->UpdateSize = data->bufferSize / data->frameSize;
device->NumUpdates = 1;
//set the running flag
InterlockedExchange(&data->running, TRUE);
return ALC_TRUE;
}
static ALCenum alsa_open_capture(ALCdevice *pDevice, const ALCchar *deviceName)
{
const char *driver = "default";
snd_pcm_hw_params_t *p;
snd_pcm_uframes_t bufferSizeInFrames;
snd_pcm_uframes_t periodSizeInFrames;
snd_pcm_format_t format;
ALuint frameSize;
alsa_data *data;
char str[128];
char *err;
int i;
ConfigValueStr("alsa", "capture", &driver);
if(!allCaptureDevNameMap)
allCaptureDevNameMap = probe_devices(SND_PCM_STREAM_CAPTURE, &numCaptureDevNames);
if(!deviceName)
deviceName = allCaptureDevNameMap[0].name;
else
{
size_t idx;
for(idx = 0;idx < numCaptureDevNames;idx++)
{
if(allCaptureDevNameMap[idx].name &&
strcmp(deviceName, allCaptureDevNameMap[idx].name) == 0)
{
if(idx > 0)
{
snprintf(str, sizeof(str), "%sCARD=%s,DEV=%d", capture_prefix,
allCaptureDevNameMap[idx].card, allCaptureDevNameMap[idx].dev);
driver = str;
}
break;
}
}
if(idx == numCaptureDevNames)
return ALC_INVALID_VALUE;
}
data = (alsa_data*)calloc(1, sizeof(alsa_data));
i = snd_pcm_open(&data->pcmHandle, driver, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
if(i < 0)
{
ERR("Could not open capture device '%s': %s\n", driver, snd_strerror(i));
free(data);
return ALC_INVALID_VALUE;
}
format = -1;
switch(pDevice->FmtType)
{
case DevFmtByte:
format = SND_PCM_FORMAT_S8;
break;
case DevFmtUByte:
format = SND_PCM_FORMAT_U8;
break;
case DevFmtShort:
format = SND_PCM_FORMAT_S16;
break;
case DevFmtUShort:
format = SND_PCM_FORMAT_U16;
break;
case DevFmtFloat:
format = SND_PCM_FORMAT_FLOAT;
break;
}
err = NULL;
bufferSizeInFrames = maxu(pDevice->UpdateSize*pDevice->NumUpdates,
100*pDevice->Frequency/1000);
periodSizeInFrames = minu(bufferSizeInFrames, 50*pDevice->Frequency/1000);
snd_pcm_hw_params_malloc(&p);
if((i=snd_pcm_hw_params_any(data->pcmHandle, p)) < 0)
err = "any";
/* set interleaved access */
if(i >= 0 && (i=snd_pcm_hw_params_set_access(data->pcmHandle, p, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
err = "set access";
/* set format (implicitly sets sample bits) */
if(i >= 0 && (i=snd_pcm_hw_params_set_format(data->pcmHandle, p, format)) < 0)
err = "set format";
/* set channels (implicitly sets frame bits) */
if(i >= 0 && (i=snd_pcm_hw_params_set_channels(data->pcmHandle, p, ChannelsFromDevFmt(pDevice->FmtChans))) < 0)
err = "set channels";
/* set rate (implicitly constrains period/buffer parameters) */
if(i >= 0 && (i=snd_pcm_hw_params_set_rate(data->pcmHandle, p, pDevice->Frequency, 0)) < 0)
err = "set rate near";
/* set buffer size in frame units (implicitly sets period size/bytes/time and buffer time/bytes) */
if(i >= 0 && (i=snd_pcm_hw_params_set_buffer_size_near(data->pcmHandle, p, &bufferSizeInFrames)) < 0)
err = "set buffer size near";
/* set buffer size in frame units (implicitly sets period size/bytes/time and buffer time/bytes) */
if(i >= 0 && (i=snd_pcm_hw_params_set_period_size_near(data->pcmHandle, p, &periodSizeInFrames, NULL)) < 0)
err = "set period size near";
/* install and prepare hardware configuration */
if(i >= 0 && (i=snd_pcm_hw_params(data->pcmHandle, p)) < 0)
err = "set params";
if(i < 0)
{
ERR("%s failed: %s\n", err, snd_strerror(i));
snd_pcm_hw_params_free(p);
goto error;
}
if((i=snd_pcm_hw_params_get_period_size(p, &bufferSizeInFrames, NULL)) < 0)
{
ERR("get size failed: %s\n", snd_strerror(i));
snd_pcm_hw_params_free(p);
goto error;
}
snd_pcm_hw_params_free(p);
frameSize = FrameSizeFromDevFmt(pDevice->FmtChans, pDevice->FmtType);
data->ring = CreateRingBuffer(frameSize, pDevice->UpdateSize*pDevice->NumUpdates);
if(!data->ring)
{
ERR("ring buffer create failed\n");
goto error;
}
data->size = snd_pcm_frames_to_bytes(data->pcmHandle, bufferSizeInFrames);
data->buffer = malloc(data->size);
if(!data->buffer)
{
ERR("buffer malloc failed\n");
goto error;
}
pDevice->szDeviceName = strdup(deviceName);
pDevice->ExtraData = data;
return ALC_NO_ERROR;
error:
free(data->buffer);
DestroyRingBuffer(data->ring);
snd_pcm_close(data->pcmHandle);
free(data);
pDevice->ExtraData = NULL;
return ALC_INVALID_VALUE;
}
static ALuint WaveProc(ALvoid *ptr)
{
ALCdevice *pDevice = (ALCdevice*)ptr;
wave_data *data = (wave_data*)pDevice->ExtraData;
ALuint frameSize;
ALuint now, start;
ALuint64 avail, done;
size_t fs;
union {
short s;
char b[sizeof(short)];
} uSB;
const ALuint restTime = (ALuint64)pDevice->UpdateSize * 1000 /
pDevice->Frequency / 2;
uSB.s = 1;
frameSize = FrameSizeFromDevFmt(pDevice->FmtChans, pDevice->FmtType);
done = 0;
start = timeGetTime();
while(!data->killNow && pDevice->Connected)
{
now = timeGetTime();
avail = (ALuint64)(now-start) * pDevice->Frequency / 1000;
if(avail < done)
{
/* Timer wrapped. Add the remainder of the cycle to the available
* count and reset the number of samples done */
avail += (ALuint64)0xFFFFFFFFu*pDevice->Frequency/1000 - done;
done = 0;
}
if(avail-done < pDevice->UpdateSize)
{
Sleep(restTime);
continue;
}
while(avail-done >= pDevice->UpdateSize)
{
aluMixData(pDevice, data->buffer, pDevice->UpdateSize);
done += pDevice->UpdateSize;
if(uSB.b[0] != 1)
{
ALuint bytesize = BytesFromDevFmt(pDevice->FmtType);
ALubyte *bytes = data->buffer;
ALuint i;
if(bytesize == 1)
{
for(i = 0;i < data->size;i++)
fputc(bytes[i], data->f);
}
else if(bytesize == 2)
{
for(i = 0;i < data->size;i++)
fputc(bytes[i^1], data->f);
}
else if(bytesize == 4)
{
for(i = 0;i < data->size;i++)
fputc(bytes[i^3], data->f);
}
}
else
fs = fwrite(data->buffer, frameSize, pDevice->UpdateSize,
data->f);
if(ferror(data->f))
{
AL_PRINT("Error writing to file\n");
aluHandleDisconnect(pDevice);
break;
}
}
}
return 0;
}
static void* thread_function(void* arg)
{
ALCdevice* device = (ALCdevice*)arg;
AndroidData* data = (AndroidData*)device->ExtraData;
JNIEnv* env;
(*javaVM)->AttachCurrentThread(javaVM, &env, NULL);
(*env)->PushLocalFrame(env, 2);
int sampleRateInHz = device->Frequency;
int channelConfig = ChannelsFromDevFmt(device->FmtChans) == 1 ? CHANNEL_CONFIGURATION_MONO : CHANNEL_CONFIGURATION_STEREO;
int audioFormat = BytesFromDevFmt(device->FmtType) == 1 ? ENCODING_PCM_8BIT : ENCODING_PCM_16BIT;
int bufferSizeInBytes = (*env)->CallStaticIntMethod(env, cAudioTrack,
mGetMinBufferSize, sampleRateInHz, channelConfig, audioFormat);
int bufferSizeInSamples = bufferSizeInBytes / FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
jobject track = (*env)->NewObject(env, cAudioTrack, mAudioTrack,
STREAM_MUSIC, sampleRateInHz, channelConfig, audioFormat, device->NumUpdates * bufferSizeInBytes, MODE_STREAM);
#ifdef HAVE_ANDROID_LOW_LATENCY
int started = 0;
size_t overallBytes = 0;
#else
(*env)->CallNonvirtualVoidMethod(env, track, cAudioTrack, mPlay);
#endif
jarray buffer = (*env)->NewByteArray(env, bufferSizeInBytes);
while (data->running)
{
void* pBuffer = (*env)->GetPrimitiveArrayCritical(env, buffer, NULL);
if (pBuffer)
{
aluMixData(device, pBuffer, bufferSizeInSamples);
(*env)->ReleasePrimitiveArrayCritical(env, buffer, pBuffer, 0);
#ifdef HAVE_ANDROID_LOW_LATENCY
if (bufferSizeInBytes >= 0)
{
if (started)
{
#endif
(*env)->CallNonvirtualIntMethod(env, track, cAudioTrack, mWrite, buffer, 0, bufferSizeInBytes);
#ifdef HAVE_ANDROID_LOW_LATENCY
}
else
{
overallBytes += (*env)->CallNonvirtualIntMethod(env, track, cAudioTrack, mWrite, buffer, 0, bufferSizeInBytes);
if (overallBytes >= (device->NumUpdates * bufferSizeInBytes))
{
(*env)->CallNonvirtualVoidMethod(env, track, cAudioTrack, mPlay);
started = 1;
}
}
}
#endif
}
else
{
AL_PRINT("Failed to get pointer to array bytes");
}
}
(*env)->CallNonvirtualVoidMethod(env, track, cAudioTrack, mStop);
(*env)->CallNonvirtualVoidMethod(env, track, cAudioTrack, mRelease);
(*env)->PopLocalFrame(env, NULL);
(*javaVM)->DetachCurrentThread(javaVM);
return NULL;
}
