bool audio::orchestra::api::Core::open(uint32_t _device,
audio::orchestra::mode _mode,
uint32_t _channels,
uint32_t _firstChannel,
uint32_t _sampleRate,
audio::format _format,
uint32_t *_bufferSize,
const audio::orchestra::StreamOptions& _options) {
// Get device ID
uint32_t nDevices = getDeviceCount();
if (nDevices == 0) {
// This should not happen because a check is made before this function is called.
ATA_ERROR("no devices found!");
return false;
}
if (_device >= nDevices) {
// This should not happen because a check is made before this function is called.
ATA_ERROR("device ID is invalid!");
return false;
}
AudioDeviceID deviceList[ nDevices/2 ];
uint32_t dataSize = sizeof(AudioDeviceID) * nDevices/2;
AudioObjectPropertyAddress property = {
kAudioHardwarePropertyDevices,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster
};
OSStatus result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
&property,
0,
nullptr,
&dataSize,
(void *) &deviceList);
if (result != noErr) {
ATA_ERROR("OS-X system error getting device IDs.");
return false;
}
AudioDeviceID id = deviceList[ _device/2 ];
// Setup for stream mode.
bool isInput = false;
if (_mode == audio::orchestra::mode_input) {
isInput = true;
property.mScope = kAudioDevicePropertyScopeInput;
} else {
property.mScope = kAudioDevicePropertyScopeOutput;
}
// Get the stream "configuration".
AudioBufferList *bufferList = nil;
dataSize = 0;
property.mSelector = kAudioDevicePropertyStreamConfiguration;
result = AudioObjectGetPropertyDataSize(id, &property, 0, nullptr, &dataSize);
if ( result != noErr
|| dataSize == 0) {
ATA_ERROR("system error (" << getErrorCode(result) << ") getting stream configuration info for device (" << _device << ").");
return false;
}
// Allocate the AudioBufferList.
bufferList = (AudioBufferList *) malloc(dataSize);
if (bufferList == nullptr) {
ATA_ERROR("memory error allocating AudioBufferList.");
return false;
}
result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, bufferList);
if ( result != noErr
|| dataSize == 0) {
ATA_ERROR("system error (" << getErrorCode(result) << ") getting stream configuration for device (" << _device << ").");
return false;
}
// Search for one or more streams that contain the desired number of
// channels. CoreAudio devices can have an arbitrary number of
// streams and each stream can have an arbitrary number of channels.
// For each stream, a single buffer of interleaved samples is
// provided. orchestra prefers the use of one stream of interleaved
// data or multiple consecutive single-channel streams. However, we
// now support multiple consecutive multi-channel streams of
// interleaved data as well.
uint32_t iStream, offsetCounter = _firstChannel;
uint32_t nStreams = bufferList->mNumberBuffers;
bool monoMode = false;
bool foundStream = false;
// First check that the device supports the requested number of
// channels.
uint32_t deviceChannels = 0;
for (iStream=0; iStream<nStreams; iStream++) {
deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
}
if (deviceChannels < (_channels + _firstChannel)) {
free(bufferList);
ATA_ERROR("the device (" << _device << ") does not support the requested channel count.");
return false;
}
// Look for a single stream meeting our needs.
uint32_t firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
for (iStream=0; iStream<nStreams; iStream++) {
streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
if (streamChannels >= _channels + offsetCounter) {
firstStream = iStream;
channelOffset = offsetCounter;
foundStream = true;
break;
}
if (streamChannels > offsetCounter) {
break;
}
offsetCounter -= streamChannels;
}
// If we didn't find a single stream above, then we should be able
// to meet the channel specification with multiple streams.
if (foundStream == false) {
monoMode = true;
offsetCounter = _firstChannel;
for (iStream=0; iStream<nStreams; iStream++) {
streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
if (streamChannels > offsetCounter) {
break;
}
offsetCounter -= streamChannels;
}
firstStream = iStream;
channelOffset = offsetCounter;
int32_t channelCounter = _channels + offsetCounter - streamChannels;
if (streamChannels > 1) {
monoMode = false;
}
while (channelCounter > 0) {
streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
if (streamChannels > 1) {
monoMode = false;
}
channelCounter -= streamChannels;
streamCount++;
}
}
free(bufferList);
// Determine the buffer size.
AudioValueRange bufferRange;
dataSize = sizeof(AudioValueRange);
property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, &bufferRange);
if (result != noErr) {
ATA_ERROR("system error (" << getErrorCode(result) << ") getting buffer size range for device (" << _device << ").");
return false;
}
if (bufferRange.mMinimum > *_bufferSize) {
*_bufferSize = (uint64_t) bufferRange.mMinimum;
} else if (bufferRange.mMaximum < *_bufferSize) {
*_bufferSize = (uint64_t) bufferRange.mMaximum;
}
if (_options.flags.m_minimizeLatency == true) {
*_bufferSize = (uint64_t) bufferRange.mMinimum;
}
// Set the buffer size. For multiple streams, I'm assuming we only
// need to make this setting for the master channel.
uint32_t theSize = (uint32_t) *_bufferSize;
dataSize = sizeof(uint32_t);
property.mSelector = kAudioDevicePropertyBufferFrameSize;
result = AudioObjectSetPropertyData(id, &property, 0, nullptr, dataSize, &theSize);
if (result != noErr) {
ATA_ERROR("system error (" << getErrorCode(result) << ") setting the buffer size for device (" << _device << ").");
return false;
}
// If attempting to setup a duplex stream, the bufferSize parameter
// MUST be the same in both directions!
*_bufferSize = theSize;
if ( m_mode == audio::orchestra::mode_output
&& _mode == audio::orchestra::mode_input
&& *_bufferSize != m_bufferSize) {
ATA_ERROR("system error setting buffer size for duplex stream on device (" << _device << ").");
return false;
}
m_bufferSize = *_bufferSize;
m_nBuffers = 1;
// Check and if necessary, change the sample rate for the device.
double nominalRate;
dataSize = sizeof(double);
property.mSelector = kAudioDevicePropertyNominalSampleRate;
result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, &nominalRate);
if (result != noErr) {
ATA_ERROR("system error (" << getErrorCode(result) << ") getting current sample rate.");
return false;
}
// Only change the sample rate if off by more than 1 Hz.
if (fabs(nominalRate - (double)_sampleRate) > 1.0) {
// Set a property listener for the sample rate change
double reportedRate = 0.0;
AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
result = AudioObjectAddPropertyListener(id, &tmp, &rateListener, (void *) &reportedRate);
if (result != noErr) {
ATA_ERROR("system error (" << getErrorCode(result) << ") setting sample rate property listener for device (" << _device << ").");
return false;
}
nominalRate = (double) _sampleRate;
result = AudioObjectSetPropertyData(id, &property, 0, nullptr, dataSize, &nominalRate);
if (result != noErr) {
ATA_ERROR("system error (" << getErrorCode(result) << ") setting sample rate for device (" << _device << ").");
return false;
}
// Now wait until the reported nominal rate is what we just set.
uint32_t microCounter = 0;
while (reportedRate != nominalRate) {
microCounter += 5000;
if (microCounter > 5000000) {
break;
}
std::this_thread::sleep_for(std::chrono::milliseconds(5));
}
// Remove the property listener.
AudioObjectRemovePropertyListener(id, &tmp, &rateListener, (void *) &reportedRate);
if (microCounter > 5000000) {
ATA_ERROR("timeout waiting for sample rate update for device (" << _device << ").");
return false;
}
}
// Now set the stream format for all streams. Also, check the
// physical format of the device and change that if necessary.
AudioStreamBasicDescription description;
dataSize = sizeof(AudioStreamBasicDescription);
property.mSelector = kAudioStreamPropertyVirtualFormat;
result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, &description);
if (result != noErr) {
ATA_ERROR("system error (" << getErrorCode(result) << ") getting stream format for device (" << _device << ").");
return false;
}
// Set the sample rate and data format id. However, only make the
// change if the sample rate is not within 1.0 of the desired
// rate and the format is not linear pcm.
bool updateFormat = false;
if (fabs(description.mSampleRate - (double)_sampleRate) > 1.0) {
description.mSampleRate = (double) _sampleRate;
updateFormat = true;
}
if (description.mFormatID != kAudioFormatLinearPCM) {
description.mFormatID = kAudioFormatLinearPCM;
updateFormat = true;
}
if (updateFormat) {
result = AudioObjectSetPropertyData(id, &property, 0, nullptr, dataSize, &description);
if (result != noErr) {
ATA_ERROR("system error (" << getErrorCode(result) << ") setting sample rate or data format for device (" << _device << ").");
return false;
}
}
// Now check the physical format.
property.mSelector = kAudioStreamPropertyPhysicalFormat;
result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, &description);
if (result != noErr) {
ATA_ERROR("system error (" << getErrorCode(result) << ") getting stream physical format for device (" << _device << ").");
return false;
}
//std::cout << "Current physical stream format:" << std::endl;
//std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;
//std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
//std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;
//std::cout << " sample rate = " << description.mSampleRate << std::endl;
if ( description.mFormatID != kAudioFormatLinearPCM
|| description.mBitsPerChannel < 16) {
description.mFormatID = kAudioFormatLinearPCM;
//description.mSampleRate = (double) sampleRate;
AudioStreamBasicDescription testDescription = description;
uint32_t formatFlags;
// We'll try higher bit rates first and then work our way down.
std::vector< std::pair<uint32_t, uint32_t> > physicalFormats;
formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
physicalFormats.push_back(std::pair<float, uint32_t>(32, formatFlags));
formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
physicalFormats.push_back(std::pair<float, uint32_t>(32, formatFlags));
physicalFormats.push_back(std::pair<float, uint32_t>(24, formatFlags)); // 24-bit packed
formatFlags &= ~(kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh);
physicalFormats.push_back(std::pair<float, uint32_t>(24.2, formatFlags)); // 24-bit in 4 bytes, aligned low
formatFlags |= kAudioFormatFlagIsAlignedHigh;
physicalFormats.push_back(std::pair<float, uint32_t>(24.4, formatFlags)); // 24-bit in 4 bytes, aligned high
formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
physicalFormats.push_back(std::pair<float, uint32_t>(16, formatFlags));
physicalFormats.push_back(std::pair<float, uint32_t>(8, formatFlags));
bool setPhysicalFormat = false;
for(uint32_t i=0; i<physicalFormats.size(); i++) {
testDescription = description;
testDescription.mBitsPerChannel = (uint32_t) physicalFormats[i].first;
testDescription.mFormatFlags = physicalFormats[i].second;
if ( (24 == (uint32_t)physicalFormats[i].first)
&& ~(physicalFormats[i].second & kAudioFormatFlagIsPacked)) {
testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;
} else {
testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
}
testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
result = AudioObjectSetPropertyData(id, &property, 0, nullptr, dataSize, &testDescription);
if (result == noErr) {
setPhysicalFormat = true;
//std::cout << "Updated physical stream format:" << std::endl;
//std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
//std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
//std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
//std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;
break;
}
}
if (!setPhysicalFormat) {
ATA_ERROR("system error (" << getErrorCode(result) << ") setting physical data format for device (" << _device << ").");
return false;
}
} // done setting virtual/physical formats.
// Get the stream / device latency.
uint32_t latency;
dataSize = sizeof(uint32_t);
property.mSelector = kAudioDevicePropertyLatency;
if (AudioObjectHasProperty(id, &property) == true) {
result = AudioObjectGetPropertyData(id, &property, 0, nullptr, &dataSize, &latency);
if (result == kAudioHardwareNoError) {
m_latency[ _mode ] = latency;
} else {
ATA_ERROR("system error (" << getErrorCode(result) << ") getting device latency for device (" << _device << ").");
return false;
}
}
// Byte-swapping: According to AudioHardware.h, the stream data will
// always be presented in native-endian format, so we should never
// need to byte swap.
m_doByteSwap[modeToIdTable(_mode)] = false;
// From the CoreAudio documentation, PCM data must be supplied as
// 32-bit floats.
m_userFormat = _format;
m_deviceFormat[modeToIdTable(_mode)] = audio::format_float;
if (streamCount == 1) {
m_nDeviceChannels[modeToIdTable(_mode)] = description.mChannelsPerFrame;
} else {
// multiple streams
m_nDeviceChannels[modeToIdTable(_mode)] = _channels;
}
m_nUserChannels[modeToIdTable(_mode)] = _channels;
m_channelOffset[modeToIdTable(_mode)] = channelOffset; // offset within a CoreAudio stream
m_deviceInterleaved[modeToIdTable(_mode)] = true;
if (monoMode == true) {
m_deviceInterleaved[modeToIdTable(_mode)] = false;
}
// Set flags for buffer conversion.
m_doConvertBuffer[modeToIdTable(_mode)] = false;
if (m_userFormat != m_deviceFormat[modeToIdTable(_mode)]) {
m_doConvertBuffer[modeToIdTable(_mode)] = true;
}
if (m_nUserChannels[modeToIdTable(_mode)] < m_nDeviceChannels[modeToIdTable(_mode)]) {
m_doConvertBuffer[modeToIdTable(_mode)] = true;
}
if (streamCount == 1) {
if ( m_nUserChannels[modeToIdTable(_mode)] > 1
&& m_deviceInterleaved[modeToIdTable(_mode)] == false) {
m_doConvertBuffer[modeToIdTable(_mode)] = true;
}
} else if (monoMode) {
m_doConvertBuffer[modeToIdTable(_mode)] = true;
}
m_private->iStream[modeToIdTable(_mode)] = firstStream;
m_private->nStreams[modeToIdTable(_mode)] = streamCount;
m_private->id[modeToIdTable(_mode)] = id;
// Allocate necessary internal buffers.
uint64_t bufferBytes;
bufferBytes = m_nUserChannels[modeToIdTable(_mode)] * *_bufferSize * audio::getFormatBytes(m_userFormat);
// m_userBuffer[modeToIdTable(_mode)] = (char *) calloc(bufferBytes, 1);
m_userBuffer[modeToIdTable(_mode)].resize(bufferBytes, 0);
if (m_userBuffer[modeToIdTable(_mode)].size() == 0) {
ATA_ERROR("error allocating user buffer memory.");
goto error;
}
// If possible, we will make use of the CoreAudio stream buffers as
// "device buffers". However, we can't do this if using multiple
// streams.
if ( m_doConvertBuffer[modeToIdTable(_mode)]
&& m_private->nStreams[modeToIdTable(_mode)] > 1) {
bool makeBuffer = true;
bufferBytes = m_nDeviceChannels[modeToIdTable(_mode)] * audio::getFormatBytes(m_deviceFormat[modeToIdTable(_mode)]);
if (_mode == audio::orchestra::mode_input) {
if ( m_mode == audio::orchestra::mode_output
&& m_deviceBuffer) {
uint64_t bytesOut = m_nDeviceChannels[0] * audio::getFormatBytes(m_deviceFormat[0]);
if (bufferBytes <= bytesOut) {
makeBuffer = false;
}
}
}
if (makeBuffer) {
bufferBytes *= *_bufferSize;
if (m_deviceBuffer) {
free(m_deviceBuffer);
m_deviceBuffer = nullptr;
}
m_deviceBuffer = (char *) calloc(bufferBytes, 1);
if (m_deviceBuffer == nullptr) {
ATA_ERROR("error allocating device buffer memory.");
goto error;
}
}
}
m_sampleRate = _sampleRate;
m_device[modeToIdTable(_mode)] = _device;
m_state = audio::orchestra::state::stopped;
ATA_VERBOSE("Set state as stopped");
// Setup the buffer conversion information structure.
if (m_doConvertBuffer[modeToIdTable(_mode)]) {
if (streamCount > 1) {
setConvertInfo(_mode, 0);
} else {
setConvertInfo(_mode, channelOffset);
}
}
if ( _mode == audio::orchestra::mode_input
&& m_mode == audio::orchestra::mode_output
&& m_device[0] == _device) {
// Only one callback procedure per device.
m_mode = audio::orchestra::mode_duplex;
} else {
#if defined(MAC_OS_X_VERSION_10_5) && (MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5)
result = AudioDeviceCreateIOProcID(id, &audio::orchestra::api::Core::callbackEvent, this, &m_private->procId[modeToIdTable(_mode)]);
#else
// deprecated in favor of AudioDeviceCreateIOProcID()
result = AudioDeviceAddIOProc(id, &audio::orchestra::api::Core::callbackEvent, this);
#endif
if (result != noErr) {
ATA_ERROR("system error setting callback for device (" << _device << ").");
goto error;
}
if ( m_mode == audio::orchestra::mode_output
&& _mode == audio::orchestra::mode_input) {
m_mode = audio::orchestra::mode_duplex;
} else {
m_mode = _mode;
}
}
// Setup the device property listener for over/underload.
property.mSelector = kAudioDeviceProcessorOverload;
result = AudioObjectAddPropertyListener(id, &property, &audio::orchestra::api::Core::xrunListener, this);
return true;
error:
m_userBuffer[0].clear();
m_userBuffer[1].clear();
if (m_deviceBuffer) {
free(m_deviceBuffer);
m_deviceBuffer = 0;
}
m_state = audio::orchestra::state::closed;
ATA_VERBOSE("Set state as closed");
return false;
}
static inline gboolean
_io_proc_spdif_start (GstCoreAudio * core_audio)
{
OSErr status;
GST_DEBUG_OBJECT (core_audio,
"osx ring buffer start ioproc ID: %p device_id %lu",
core_audio->procID, (gulong) core_audio->device_id);
if (!core_audio->io_proc_active) {
/* Add IOProc callback */
status = AudioDeviceCreateIOProcID (core_audio->device_id,
(AudioDeviceIOProc) _io_proc_spdif,
(void *) core_audio, &core_audio->procID);
if (status != noErr) {
GST_ERROR_OBJECT (core_audio->osxbuf,
":AudioDeviceCreateIOProcID failed: %d", (int) status);
return FALSE;
}
core_audio->io_proc_active = TRUE;
}
core_audio->io_proc_needs_deactivation = FALSE;
/* Start device */
status = AudioDeviceStart (core_audio->device_id, core_audio->procID);
if (status != noErr) {
GST_ERROR_OBJECT (core_audio->osxbuf,
"AudioDeviceStart failed: %d", (int) status);
return FALSE;
}
return TRUE;
}
AudioDeviceIOProcID CAHALAudioDevice::CreateIOProcID(AudioDeviceIOProc inIOProc, void* inClientData)
{
AudioDeviceIOProcID theAnswer = NULL;
OSStatus theError = AudioDeviceCreateIOProcID(mObjectID, inIOProc, inClientData, &theAnswer);
ThrowIfError(theError, CAException(theError), "CAHALAudioDevice::CreateIOProcID: got an error creating the IOProc ID");
return theAnswer;
}
static au_instance_t *audiounits_create_recorder(SEXP source, float rate, int chs, int flags) {
UInt32 propsize=0;
OSStatus err;
au_instance_t *ap = (au_instance_t*) calloc(sizeof(au_instance_t), 1);
ap->source = source;
ap->sample_rate = rate;
ap->done = NO;
ap->position = 0;
ap->length = LENGTH(source);
ap->stereo = (chs == 2) ? YES : NO;
propsize = sizeof(ap->inDev);
err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propsize, &ap->inDev);
if (err) {
free(ap);
Rf_error("unable to find default audio input (%08x)", err);
}
propsize = sizeof(ap->fmtIn);
err = AudioDeviceGetProperty(ap->inDev, 0, YES, kAudioDevicePropertyStreamFormat, &propsize, &ap->fmtIn);
if (err) {
free(ap);
Rf_error("unable to retrieve audio input format (%08x)", err);
}
/* Rprintf(" recording format: %f, chs: %d, fpp: %d, bpp: %d, bpf: %d, flags: %x\n", ap->fmtIn.mSampleRate, ap->fmtIn.mChannelsPerFrame, ap->fmtIn.mFramesPerPacket, ap->fmtIn.mBytesPerPacket, ap->fmtIn.mBytesPerFrame, ap->fmtIn.mFormatFlags); */
ap->srFrac = 1.0;
if (ap->fmtIn.mSampleRate != ap->sample_rate) ap->srFrac = ap->sample_rate / ap->fmtIn.mSampleRate;
ap->srRun = 0.0;
#if defined(MAC_OS_X_VERSION_10_5) && (MAC_OS_X_VERSION_MIN_REQUIRED>=MAC_OS_X_VERSION_10_5)
err = AudioDeviceCreateIOProcID(ap->inDev, inputRenderProc, ap, &ap->inIOProcID );
#else
err = AudioDeviceAddIOProc(ap->inDev, inputRenderProc, ap);
#endif
if (err) {
free(ap);
Rf_error("unable to register recording callback (%08x)", err);
}
R_PreserveObject(ap->source);
Rf_setAttrib(ap->source, Rf_install("rate"), Rf_ScalarInteger(rate)); /* we adjust the rate */
Rf_setAttrib(ap->source, Rf_install("bits"), Rf_ScalarInteger(16)); /* we say it's 16 because we don't know - float is always 32-bit */
Rf_setAttrib(ap->source, Rf_install("class"), Rf_mkString("audioSample"));
if (ap->stereo) {
SEXP dim = Rf_allocVector(INTSXP, 2);
INTEGER(dim)[0] = 2;
INTEGER(dim)[1] = LENGTH(ap->source) / 2;
Rf_setAttrib(ap->source, R_DimSymbol, dim);
}
return ap;
}
void AudioTee::start() {
if (mInputDevice.mID == kAudioDeviceUnknown || mOutputDevice.mID == kAudioDeviceUnknown) return;
if (mInputDevice.mFormat.mSampleRate != mOutputDevice.mFormat.mSampleRate) {
printf("Error in AudioTee::Start() - sample rate mismatch: %f / %f\n", mInputDevice.mFormat.mSampleRate, mOutputDevice.mFormat.mSampleRate);
return;
}
mWorkBuf = new Byte[mInputDevice.mBufferSizeFrames * mInputDevice.mFormat.mBytesPerFrame];
memset(mWorkBuf, 0, mInputDevice.mBufferSizeFrames * mInputDevice.mFormat.mBytesPerFrame);
UInt32 framesInHistoryBuffer = NextPowerOfTwo(mInputDevice.mFormat.mSampleRate * mSecondsInHistoryBuffer);
mHistoryBufferMaxByteSize = mInputDevice.mFormat.mBytesPerFrame * framesInHistoryBuffer;
mHistBuf = new CARingBuffer();
mHistBuf->Allocate(2, mInputDevice.mFormat.mBytesPerFrame, framesInHistoryBuffer);
printf("Initializing history buffer with byte capacity %u — %f seconds at %f kHz", mHistoryBufferMaxByteSize, (mHistoryBufferMaxByteSize / mInputDevice.mFormat.mSampleRate / (4 * 2)), mInputDevice.mFormat.mSampleRate);
printf("Initializing work buffer with mBufferSizeFrames:%u and mBytesPerFrame %u\n", mInputDevice.mBufferSizeFrames, mInputDevice.mFormat.mBytesPerFrame);
mInputIOProcID = NULL;
AudioDeviceCreateIOProcID(mInputDevice.mID, InputIOProc, this, &mInputIOProcID);
AudioDeviceStart(mInputDevice.mID, mInputIOProcID);
mOutputIOProc = OutputIOProc;
mOutputIOProcID = NULL;
AudioDeviceCreateIOProcID(mOutputDevice.mID, mOutputIOProc, this, &mOutputIOProcID);
AudioDeviceStart(mOutputDevice.mID, mOutputIOProcID);
}
bool CCoreAudioDevice::AddIOProc(AudioDeviceIOProc ioProc, void* pCallbackData)
{
// Allow only one IOProc at a time
if (!m_DeviceId || m_IoProc)
return false;
OSStatus ret = AudioDeviceCreateIOProcID(m_DeviceId, ioProc, pCallbackData, &m_IoProc);
if (ret)
{
CLog::Log(LOGERROR, "CCoreAudioDevice::AddIOProc: "
"Unable to add IOProc. Error = %s", GetError(ret).c_str());
m_IoProc = NULL;
return false;
}
Start();
return true;
}
bool CCoreAudioDevice::AddIOProc()
{
// Allow only one IOProc at a time
if (!m_DeviceId || m_IoProc)
return false;
OSStatus ret = AudioDeviceCreateIOProcID(m_DeviceId, DirectRenderCallback, this, &m_IoProc);
if (ret)
{
CLog::Log(LOGERROR, "CCoreAudioDevice::AddIOProc: "
"Unable to add IOProc. Error = %s", GetError(ret).c_str());
m_IoProc = NULL;
return false;
}
Start();
CLog::Log(LOGDEBUG, "CCoreAudioDevice::AddIOProc: "
"IOProc %p set for device 0x%04x", m_IoProc, (uint)m_DeviceId);
return true;
}
bool start (AudioIODeviceCallback* cb)
{
if (! started)
{
callback = nullptr;
if (deviceID != 0)
{
#if MAC_OS_X_VERSION_MIN_REQUIRED < MAC_OS_X_VERSION_10_5
if (OK (AudioDeviceAddIOProc (deviceID, audioIOProc, this)))
#else
if (OK (AudioDeviceCreateIOProcID (deviceID, audioIOProc, this, &audioProcID)))
#endif
{
if (OK (AudioDeviceStart (deviceID, audioIOProc)))
{
started = true;
}
else
{
#if MAC_OS_X_VERSION_MIN_REQUIRED < MAC_OS_X_VERSION_10_5
OK (AudioDeviceRemoveIOProc (deviceID, audioIOProc));
#else
OK (AudioDeviceDestroyIOProcID (deviceID, audioProcID));
audioProcID = 0;
#endif
}
}
}
}
if (started)
{
const ScopedLock sl (callbackLock);
callback = cb;
}
return started && (inputDevice == nullptr || inputDevice->start (cb));
}
int macosx_audio_open(audio_desc_t ad, audio_format* ifmt, audio_format *ofmt)
{
OSStatus err = noErr;
UInt32 propertySize;
Boolean writable;
obtained_ = false;
add = ad;
//dev[0] = devices[ad];
UNUSED(ofmt);
// Get the default input device ID.
err = AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDefaultInputDevice, &propertySize, &writable);
if (err != noErr) {
return 0;
}
err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propertySize, &(devices[ad].inputDeviceID_));
if (err != noErr) {
debug_msg("error kAudioHardwarePropertyDefaultInputDevice");
return 0;
}
if (devices[ad].inputDeviceID_ == kAudioDeviceUnknown) {
debug_msg("error kAudioDeviceUnknown");
return 0;
}
// Get the input stream description.
err = AudioDeviceGetPropertyInfo(devices[ad].inputDeviceID_, 0, true, kAudioDevicePropertyStreamFormat, &propertySize, &writable);
if (err != noErr) {
debug_msg("error AudioDeviceGetPropertyInfo");
return 0;
}
err = AudioDeviceGetProperty(devices[ad].inputDeviceID_, 0, true, kAudioDevicePropertyStreamFormat, &propertySize, &(devices[ad].inputStreamBasicDescription_));
//printf("inputStreamBasicDescription_.mBytesPerFrame %d\n", devices[add].inputStreamBasicDescription_);
if (err != noErr) {
debug_msg("error AudioDeviceGetProperty");
return 0;
}
// nastavime maly endian
devices[ad].inputStreamBasicDescription_.mFormatFlags &= (kAudioFormatFlagIsBigEndian & 0);
if (writable) {
err = AudioDeviceSetProperty(devices[ad].inputDeviceID_, NULL, 0, true, kAudioDevicePropertyStreamFormat, sizeof(AudioStreamBasicDescription), &(devices[ad].inputStreamBasicDescription_));
if (err != noErr) printf("err: AudioDeviceSetProperty: kAudioDevicePropertyStreamFormat\n");
}
/* set the buffer size of the device */
/*
int bufferByteSize = 8192;
propertySize = sizeof(bufferByteSize);
err = AudioDeviceSetProperty(devices[ad].inputDeviceID_, NULL, 0, true, kAudioDevicePropertyBufferSize, propertySize, &bufferByteSize);
if (err != noErr) debug_msg("err: Set kAudioDevicePropertyBufferSize to %d\n", bufferByteSize);
else debug_msg("sucessfully set kAudioDevicePropertyBufferSize to %d\n", bufferByteSize);
*/
// Set the device sample rate -- a temporary fix for the G5's
// built-in audio and possibly other audio devices.
Boolean IsInput = 0;
int inChannel = 0;
Float64 theAnswer = 44100;
UInt32 theSize = sizeof(theAnswer);
err = AudioDeviceSetProperty(devices[ad].inputDeviceID_, NULL, inChannel, IsInput,
kAudioDevicePropertyNominalSampleRate, theSize, &theAnswer);
if (err != noErr) {
debug_msg("error AudioDeviceSetProperty\n");
return 0;
}
debug_msg("Sample rate, %f\n", theAnswer);
#if defined(MAC_OS_X_VERSION_10_5) && (MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5)
err = AudioDeviceCreateIOProcID(devices[ad].inputDeviceID_, audioIOProc, (void*)NULL, &devices[ad].inputDeviceProcID_);
if (err != noErr) {
debug_msg("error AudioDeviceCreateIOProcID, %s\n", GetMacOSStatusCommentString(err));
return 0;
}
err = OpenADefaultComponent(kAudioUnitType_Output, kAudioUnitSubType_DefaultOutput, &(devices[ad].outputUnit_));
// The HAL AU maybe a better way to in the future...
//err = OpenADefaultComponent(kAudioUnitType_Output, kAudioUnitSubType_HALOutput, &(devices[ad].outputUnit_));
if (err != noErr) {
debug_msg("error OpenADefaultComponent\n");
return 0;
}
#else
// Register the AudioDeviceIOProc.
err = AudioDeviceAddIOProc(devices[ad].inputDeviceID_, audioIOProc, NULL);
if (err != noErr) {
debug_msg("error AudioDeviceAddIOProc\n");
return 0;
}
err = OpenDefaultAudioOutput(&(devices[ad].outputUnit_));
if (err != noErr) {
debug_msg("error OpenDefaultAudioOutput\n");
return 0;
}
#endif
// Register a callback function to provide output data to the unit.
devices[ad].input.inputProc = outputRenderer;
devices[ad].input.inputProcRefCon = 0;
/* These would be needed if HAL used
* UInt32 enableIO =1;
err = AudioUnitSetProperty(devices[ad].outputUnit_, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, (const void*)&enableIO, sizeof(UInt32));
enableIO=0;
err = AudioUnitSetProperty(devices[ad].outputUnit_, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, (const void*)&enableIO, sizeof(UInt32));
if (err != noErr) {
debug_msg("error AudioUnitSetProperty EnableIO with error %ld: %s\n", err, GetMacOSStatusErrorString(err));
return 0;
}*/
#if defined(MAC_OS_X_VERSION_10_5) && (MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5)
err = AudioUnitSetProperty(devices[ad].outputUnit_, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, 0, &(devices[ad].input), sizeof(AURenderCallbackStruct));
#else
err = AudioUnitSetProperty(devices[ad].outputUnit_, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &(devices[ad].input), sizeof(AURenderCallbackStruct));
#endif
if (err != noErr) {
debug_msg("error AudioUnitSetProperty1 with error %ld: %s\n", err, GetMacOSStatusErrorString(err));
return 0;
}
// Define the Mash stream description. Mash puts 20ms of data into each read
// and write call. 20ms at 8000Hz equals 160 samples. Each sample is a u_char,
// so that's 160 bytes. Mash uses 8-bit mu-law internally, so we need to convert
// to 16-bit linear before using the audio data.
devices[ad].mashStreamBasicDescription_.mSampleRate = 8000.0;
//devices[ad].mashStreamBasicDescription_.mSampleRate = ifmt->sample_rate;
devices[ad].mashStreamBasicDescription_.mFormatID = kAudioFormatLinearPCM;
#ifdef WORDS_BIGENDIAN
devices[ad].mashStreamBasicDescription_.mFormatFlags =kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsBigEndian |kLinearPCMFormatFlagIsPacked;
#else
devices[ad].mashStreamBasicDescription_.mFormatFlags =kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
#endif
devices[ad].mashStreamBasicDescription_.mBytesPerPacket = 2;
devices[ad].mashStreamBasicDescription_.mFramesPerPacket = 1;
devices[ad].mashStreamBasicDescription_.mBytesPerFrame = 2;
devices[ad].mashStreamBasicDescription_.mChannelsPerFrame = 1;
devices[ad].mashStreamBasicDescription_.mBitsPerChannel = 16;
// Inform the default output unit of our source format.
err = AudioUnitSetProperty(devices[ad].outputUnit_, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &(devices[ad].mashStreamBasicDescription_), sizeof(AudioStreamBasicDescription));
if (err != noErr) {
debug_msg("error AudioUnitSetProperty2");
printf("error setting output unit source format\n");
return 0;
}
// check the stream format
err = AudioUnitGetPropertyInfo(devices[ad].outputUnit_, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &propertySize, &writable);
if (err != noErr) debug_msg("err getting propert info for kAudioUnitProperty_StreamFormat\n");
err = AudioUnitGetProperty(devices[ad].outputUnit_, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamdesc_, &propertySize);
if (err != noErr) debug_msg("err getting values for kAudioUnitProperty_StreamFormat\n");
char name[128];
audio_format_name(ifmt, name, 128);
debug_msg("Requested ifmt %s\n",name);
debug_msg("ifmt bytes pre block: %d\n",ifmt->bytes_per_block);
// handle the requested format
if (ifmt->encoding != DEV_S16) {
audio_format_change_encoding(ifmt, DEV_S16);
debug_msg("Requested ifmt changed to %s\n",name);
debug_msg("ifmt bytes pre block: %d\n",ifmt->bytes_per_block);
}
audio_format_name(ofmt, name, 128);
debug_msg("Requested ofmt %s\n",name);
debug_msg("ofmt bytes pre block: %d\n",ofmt->bytes_per_block);
// Allocate the read buffer and Z delay line.
//readBufferSize_ = 8192;
readBufferSize_ = ifmt->bytes_per_block * ringBufferFactor_;
//readBufferSize_ = 320;
//printf("readBufferSize_ %d\n", readBufferSize_);
readBuffer_ = malloc(sizeof(u_char)*readBufferSize_);
bzero(readBuffer_, readBufferSize_ * sizeof(u_char));
//memset(readBuffer_, PCMU_AUDIO_ZERO, readBufferSize_);
//inputReadIndex_ = -1;
inputReadIndex_ = 0; inputWriteIndex_ = 0;
zLine_ = malloc(sizeof(double)*DECIM441_LENGTH / 80);
availableInput_ = 0;
// Allocate the write buffer.
//writeBufferSize_ = 8000;
writeBufferSize_ = ofmt->bytes_per_block * ringBufferFactor_;
writeBuffer_ = malloc(sizeof(SInt16)*writeBufferSize_);
bzero(writeBuffer_, writeBufferSize_ * sizeof(SInt16));
outputReadIndex_ = 0; outputWriteIndex_ = 0;
//outputWriteIndex_ = -1;
// Start audio processing.
err = AudioUnitInitialize(devices[ad].outputUnit_);
if (err != noErr) {
debug_msg("error AudioUnitInitialize\n");
return 0;
}
err = AudioDeviceStart(devices[ad].inputDeviceID_, audioIOProc);
if (err != noErr) {
fprintf(stderr, "Input device error: AudioDeviceStart\n");
return 0;
}
err = AudioOutputUnitStart(devices[ad].outputUnit_);
if (err != noErr) {
fprintf(stderr, "Output device error: AudioOutputUnitStart\n");
return 0;
}
// Inform the default output unit of our source format.
/*
err = AudioUnitSetProperty(devices[ad].outputUnit_, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &(devices[ad].mashStreamBasicDescription_), sizeof(AudioStreamBasicDescription));
if (err != noErr) {
debug_msg("error AudioUnitSetProperty3");
return 0;
}
*/
return 1;
};
// ----------------------------------------------------------------------------
void AudioCoreDriver::initialize(PlayerLibSidplay* player, int sampleRate, int bitsPerSample)
// ----------------------------------------------------------------------------
{
if (mInstanceId != 0)
return;
mPlayer = player;
mNumSamplesInBuffer = 512;
mIsPlaying = false;
mIsPlayingPreRenderedBuffer = false;
mBufferUnderrunDetected = false;
mBufferUnderrunCount = 0;
mSpectrumTemporalSmoothing = 0.5f;
mPreRenderedBuffer = NULL;
mPreRenderedBufferSampleCount = 0;
mPreRenderedBufferPlaybackPosition = 0;
if (!mIsInitialized)
{
OSStatus err;
//get default output device
UInt32 propertySize = sizeof(mDeviceID);
AudioObjectPropertyAddress prop1 = {kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster};
err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &prop1, 0, NULL, &propertySize, &mDeviceID);
if (err != kAudioHardwareNoError) {
printf("AudioObjectGetPropertyData(kAudioHardwarePropertyDefaultOutputDevice) failed\n");
return;
}
if (mDeviceID == kAudioDeviceUnknown)
return;
err = queryStreamFormat(mDeviceID, mStreamFormat);
if (err != kAudioHardwareNoError) {
printf("queryStreamFormat failed\n");
return;
}
//add property listeners
#if USE_NEW_API
AudioObjectPropertyAddress prop5 = {
kAudioDevicePropertyStreamFormat, //TODO this is deprecated, how to get this notification?
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster
};
err = AudioObjectAddPropertyListener(mDeviceID, &prop5, streamFormatChanged, (void*)this);
if (err != kAudioHardwareNoError) {
printf("AudioObjectAddPropertyListener(streamFormatChanged) failed\n");
return;
}
AudioObjectPropertyAddress prop6 = {
kAudioDeviceProcessorOverload,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster
};
err = AudioObjectAddPropertyListener(mDeviceID, &prop6, overloadDetected, (void*)this);
if (err != kAudioHardwareNoError) {
printf("AudioObjectAddPropertyListener(overloadDetected) failed\n");
return;
}
AudioObjectPropertyAddress prop7 = {
kAudioHardwarePropertyDefaultOutputDevice,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster
};
err = AudioObjectAddPropertyListener(mDeviceID, &prop7, deviceChanged, (void*)this);
if (err != kAudioHardwareNoError) {
printf("AudioObjectAddPropertyListener(deviceChanged) failed\n");
return;
}
#else
if (AudioDeviceAddPropertyListener(mDeviceID, 0, false, kAudioDevicePropertyStreamFormat, streamFormatChanged, (void*) this) != kAudioHardwareNoError)
return;
if (AudioDeviceAddPropertyListener(mDeviceID, 0, false, kAudioDeviceProcessorOverload, overloadDetected, (void*) this) != kAudioHardwareNoError)
return;
if (AudioHardwareAddPropertyListener(kAudioHardwarePropertyDefaultOutputDevice, deviceChanged, (void*) this) != kAudioHardwareNoError)
return;
#endif
mSampleBuffer1 = new short[mNumSamplesInBuffer];
memset(mSampleBuffer1, 0, sizeof(short) * mNumSamplesInBuffer);
mSampleBuffer2 = new short[mNumSamplesInBuffer];
memset(mSampleBuffer2, 0, sizeof(short) * mNumSamplesInBuffer);
mSpectrumBuffer = new float[mNumSamplesInBuffer/2];
memset(mSpectrumBuffer, 0, sizeof(float) * (mNumSamplesInBuffer/2));
mSampleBuffer = mSampleBuffer1;
mRetSampleBuffer = mSampleBuffer2;
int bufferByteSize = mNumSamplesInBuffer * mStreamFormat.mChannelsPerFrame * sizeof(float);
propertySize = sizeof(bufferByteSize);
#if USE_NEW_API
AudioObjectPropertyAddress prop8 = {
kAudioDevicePropertyBufferSize,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster
};
err = AudioObjectSetPropertyData(mDeviceID, &prop8, 0, NULL, propertySize, &bufferByteSize);
if (err != kAudioHardwareNoError) {
printf("AudioObjectSetPropertyData(kAudioDevicePropertyBufferSize) failed\n");
return;
}
#else
if (AudioDeviceSetProperty(mDeviceID, NULL, 0, false, kAudioDevicePropertyBufferSize, propertySize, &bufferByteSize) != kAudioHardwareNoError)
return;
#endif
mScaleFactor = sBitScaleFactor;
mPreRenderedBufferScaleFactor = sBitScaleFactor;
if (AudioDeviceCreateIOProcID(mDeviceID, emulationPlaybackProc, (void*) this, &mEmulationPlaybackProcID) != kAudioHardwareNoError)
{
delete[] mSampleBuffer1;
mSampleBuffer1 = NULL;
delete[] mSampleBuffer2;
mSampleBuffer2 = NULL;
mSampleBuffer = NULL;
mRetSampleBuffer = NULL;
delete[] mSpectrumBuffer;
mSpectrumBuffer = NULL;
return;
}
if (AudioDeviceCreateIOProcID(mDeviceID, preRenderedBufferPlaybackProc, (void*) this, &mPreRenderedBufferPlaybackProcID) != kAudioHardwareNoError)
{
delete[] mSampleBuffer1;
mSampleBuffer1 = NULL;
delete[] mSampleBuffer2;
mSampleBuffer2 = NULL;
mSampleBuffer = NULL;
mRetSampleBuffer = NULL;
delete[] mSpectrumBuffer;
mSpectrumBuffer = NULL;
return;
}
}
mVolume = 1.0f;
mIsInitialized = true;
}
// Initialize the BlockSound class
BlockSound::BlockSound() {
sample_size = 0;
#ifdef __APPLE__
remaining = 0;
UInt32 size = sizeof(device);
if (AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
&size, (void *)&device) != noErr) return;
size = sizeof(format);
if (AudioDeviceGetProperty(device, 0, false, kAudioDevicePropertyStreamFormat,
&size, &format) != noErr) return;
// Set up a format we like...
format.mSampleRate = 44100.0; // 44.1kHz
format.mChannelsPerFrame = 2; // stereo
if (AudioDeviceSetProperty(device, NULL, 0, false,
kAudioDevicePropertyStreamFormat,
sizeof(format), &format) != noErr) return;
// Check we got linear pcm - what to do if we did not ???
if (format.mFormatID != kAudioFormatLinearPCM) return;
// Attach the callback and start the device
# if MAC_OS_X_VERSION_MAX_ALLOWED >= MAC_OS_X_VERSION_10_5
if (AudioDeviceCreateIOProcID(device, audio_cb, (void *)this, &audio_proc_id) != noErr) return;
AudioDeviceStart(device, audio_proc_id);
# else
if (AudioDeviceAddIOProc(device, audio_cb, (void *)this) != noErr) return;
AudioDeviceStart(device, audio_cb);
# endif
sample_size = (int)format.mSampleRate;
#elif defined(WIN32)
WAVEFORMATEX format;
memset(&format, 0, sizeof(format));
format.cbSize = sizeof(format);
format.wFormatTag = WAVE_FORMAT_PCM;
format.nChannels = 2;
format.nSamplesPerSec = 44100;
format.nAvgBytesPerSec = 44100 * 4;
format.nBlockAlign = 4;
format.wBitsPerSample = 16;
data_handle = GlobalAlloc(GMEM_MOVEABLE | GMEM_SHARE, format.nSamplesPerSec * 4);
if (!data_handle) return;
data_ptr = (LPSTR)GlobalLock(data_handle);
header_handle = GlobalAlloc(GMEM_MOVEABLE | GMEM_SHARE, sizeof(WAVEHDR));
if (!header_handle) return;
header_ptr = (WAVEHDR *)GlobalLock(header_handle);
header_ptr->lpData = data_ptr;
header_ptr->dwFlags = 0;
header_ptr->dwLoops = 0;
if (waveOutOpen(&device, WAVE_MAPPER, &format, 0, 0, WAVE_ALLOWSYNC)
!= MMSYSERR_NOERROR) return;
sample_size = format.nSamplesPerSec;
#else
# ifdef HAVE_ALSA_ASOUNDLIB_H
handle = NULL;
if (snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0) >= 0) {
// Initialize PCM sound stuff...
snd_pcm_hw_params_t *params;
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_hw_params_any(handle, params);
snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16);
snd_pcm_hw_params_set_channels(handle, params, 2);
unsigned rate = 44100;
int dir;
snd_pcm_hw_params_set_rate_near(handle, params, &rate, &dir);
snd_pcm_uframes_t period = (int)rate;
snd_pcm_hw_params_set_period_size_near(handle, params, &period, &dir);
sample_size = rate;
if (snd_pcm_hw_params(handle, params) < 0) {
sample_size = 0;
snd_pcm_close(handle);
handle = NULL;
}
}
# endif // HAVE_ALSA_ASOUNDLIB_H
#endif // __APPLE__
if (sample_size) {
// Make an explosion sound by passing white noise through a low pass
// filter with a decreasing frequency...
sample_data = new short[2 * sample_size];
short *sample_ptr = sample_data;
int max_sample = 2 * sample_size - 2;
*sample_ptr++ = 0;
*sample_ptr++ = 0;
for (int j = max_sample; j > 0; j --, sample_ptr ++) {
float freq = (float)j / (float)max_sample;
float volume = 32767.0 * (0.5 * sqrt(freq) + 0.5);
float sample = 0.0001 * ((rand() % 20001) - 10000);
*sample_ptr = (int)(volume * freq * sample +
(1.0 - freq) * sample_ptr[-2]);
}
}
}
static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
void *drv_opaque)
{
OSStatus status;
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
int err;
const char *typ = "playback";
AudioValueRange frameRange;
Audiodev *dev = drv_opaque;
AudiodevCoreaudioPerDirectionOptions *cpdo = dev->u.coreaudio.out;
int frames;
/* create mutex */
err = pthread_mutex_init(&core->mutex, NULL);
if (err) {
dolog("Could not create mutex\nReason: %s\n", strerror (err));
return -1;
}
audio_pcm_init_info (&hw->info, as);
status = coreaudio_get_voice(&core->outputDeviceID);
if (status != kAudioHardwareNoError) {
coreaudio_logerr2 (status, typ,
"Could not get default output Device\n");
return -1;
}
if (core->outputDeviceID == kAudioDeviceUnknown) {
dolog ("Could not initialize %s - Unknown Audiodevice\n", typ);
return -1;
}
/* get minimum and maximum buffer frame sizes */
status = coreaudio_get_framesizerange(core->outputDeviceID,
&frameRange);
if (status != kAudioHardwareNoError) {
coreaudio_logerr2 (status, typ,
"Could not get device buffer frame range\n");
return -1;
}
frames = audio_buffer_frames(
qapi_AudiodevCoreaudioPerDirectionOptions_base(cpdo), as, 11610);
if (frameRange.mMinimum > frames) {
core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMinimum;
dolog ("warning: Upsizing Buffer Frames to %f\n", frameRange.mMinimum);
} else if (frameRange.mMaximum < frames) {
core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMaximum;
dolog ("warning: Downsizing Buffer Frames to %f\n", frameRange.mMaximum);
}
else {
core->audioDevicePropertyBufferFrameSize = frames;
}
/* set Buffer Frame Size */
status = coreaudio_set_framesize(core->outputDeviceID,
&core->audioDevicePropertyBufferFrameSize);
if (status != kAudioHardwareNoError) {
coreaudio_logerr2 (status, typ,
"Could not set device buffer frame size %" PRIu32 "\n",
(uint32_t)core->audioDevicePropertyBufferFrameSize);
return -1;
}
/* get Buffer Frame Size */
status = coreaudio_get_framesize(core->outputDeviceID,
&core->audioDevicePropertyBufferFrameSize);
if (status != kAudioHardwareNoError) {
coreaudio_logerr2 (status, typ,
"Could not get device buffer frame size\n");
return -1;
}
hw->samples = (cpdo->has_buffer_count ? cpdo->buffer_count : 4) *
core->audioDevicePropertyBufferFrameSize;
/* get StreamFormat */
status = coreaudio_get_streamformat(core->outputDeviceID,
&core->outputStreamBasicDescription);
if (status != kAudioHardwareNoError) {
coreaudio_logerr2 (status, typ,
"Could not get Device Stream properties\n");
core->outputDeviceID = kAudioDeviceUnknown;
return -1;
}
/* set Samplerate */
core->outputStreamBasicDescription.mSampleRate = (Float64) as->freq;
status = coreaudio_set_streamformat(core->outputDeviceID,
&core->outputStreamBasicDescription);
if (status != kAudioHardwareNoError) {
coreaudio_logerr2 (status, typ, "Could not set samplerate %d\n",
as->freq);
core->outputDeviceID = kAudioDeviceUnknown;
return -1;
}
/* set Callback */
core->ioprocid = NULL;
status = AudioDeviceCreateIOProcID(core->outputDeviceID,
audioDeviceIOProc,
hw,
&core->ioprocid);
if (status != kAudioHardwareNoError || core->ioprocid == NULL) {
coreaudio_logerr2 (status, typ, "Could not set IOProc\n");
core->outputDeviceID = kAudioDeviceUnknown;
return -1;
}
/* start Playback */
if (!isPlaying(core->outputDeviceID)) {
status = AudioDeviceStart(core->outputDeviceID, core->ioprocid);
if (status != kAudioHardwareNoError) {
coreaudio_logerr2 (status, typ, "Could not start playback\n");
AudioDeviceDestroyIOProcID(core->outputDeviceID, core->ioprocid);
core->outputDeviceID = kAudioDeviceUnknown;
return -1;
}
}
return 0;
}
/*****************************************************************************
* Setup a encoded digital stream (SPDIF)
*****************************************************************************/
static int OpenSPDIF(void)
{
OSStatus err = noErr;
UInt32 i_param_size, b_mix = 0;
Boolean b_writeable = 0;
AudioStreamID *p_streams = NULL;
int i, i_streams = 0;
AudioObjectPropertyAddress property_address;
/* Start doing the SPDIF setup process. */
ao->b_digital = 1;
/* Hog the device. */
ao->i_hog_pid = getpid() ;
err = SetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertyHogMode,
sizeof(ao->i_hog_pid), &ao->i_hog_pid);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err);
ao->i_hog_pid = -1;
goto err_out;
}
/* Set mixable to false if we are allowed to. */
err = IsAudioPropertySettable(ao->i_selected_dev,
kAudioDevicePropertySupportsMixing,
&b_writeable);
err = GetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertySupportsMixing,
sizeof(UInt32), &b_mix);
if (err != noErr && b_writeable)
{
b_mix = 0;
err = SetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertySupportsMixing,
sizeof(UInt32), &b_mix);
ao->b_changed_mixing = 1;
}
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
goto err_out;
}
/* Get a list of all the streams on this device. */
i_param_size = GetAudioPropertyArray(ao->i_selected_dev,
kAudioDevicePropertyStreams,
kAudioDevicePropertyScopeOutput,
(void **)&p_streams);
if (!i_param_size) {
ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams.\n");
goto err_out;
}
i_streams = i_param_size / sizeof(AudioStreamID);
ao_msg(MSGT_AO, MSGL_V, "current device stream number: %d\n", i_streams);
for (i = 0; i < i_streams && ao->i_stream_index < 0; ++i)
{
/* Find a stream with a cac3 stream. */
AudioStreamBasicDescription *p_format_list = NULL;
int i_formats = 0, j = 0, b_digital = 0;
i_param_size = GetGlobalAudioPropertyArray(p_streams[i],
kAudioStreamPropertyPhysicalFormats,
(void **)&p_format_list);
if (!i_param_size) {
ao_msg(MSGT_AO, MSGL_WARN,
"Could not get number of stream formats.\n");
continue;
}
i_formats = i_param_size / sizeof(AudioStreamBasicDescription);
/* Check if one of the supported formats is a digital format. */
for (j = 0; j < i_formats; ++j)
{
if (p_format_list[j].mFormatID == 'IAC3' ||
p_format_list[j].mFormatID == kAudioFormat60958AC3)
{
b_digital = 1;
break;
}
}
if (b_digital)
{
/* If this stream supports a digital (cac3) format, then set it. */
int i_requested_rate_format = -1;
int i_current_rate_format = -1;
int i_backup_rate_format = -1;
ao->i_stream_id = p_streams[i];
ao->i_stream_index = i;
if (ao->b_revert == 0)
{
/* Retrieve the original format of this stream first if not done so already. */
err = GetAudioProperty(ao->i_stream_id,
kAudioStreamPropertyPhysicalFormat,
sizeof(ao->sfmt_revert), &ao->sfmt_revert);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN,
"Could not retrieve the original stream format: [%4.4s]\n",
(char *)&err);
if (p_format_list) free(p_format_list);
continue;
}
ao->b_revert = 1;
}
for (j = 0; j < i_formats; ++j)
if (p_format_list[j].mFormatID == 'IAC3' ||
p_format_list[j].mFormatID == kAudioFormat60958AC3)
{
if (p_format_list[j].mSampleRate == ao->stream_format.mSampleRate)
{
i_requested_rate_format = j;
break;
}
if (p_format_list[j].mSampleRate == ao->sfmt_revert.mSampleRate)
i_current_rate_format = j;
else if (i_backup_rate_format < 0 || p_format_list[j].mSampleRate > p_format_list[i_backup_rate_format].mSampleRate)
i_backup_rate_format = j;
}
if (i_requested_rate_format >= 0) /* We prefer to output at the samplerate of the original audio. */
ao->stream_format = p_format_list[i_requested_rate_format];
else if (i_current_rate_format >= 0) /* If not possible, we will try to use the current samplerate of the device. */
ao->stream_format = p_format_list[i_current_rate_format];
else ao->stream_format = p_format_list[i_backup_rate_format]; /* And if we have to, any digital format will be just fine (highest rate possible). */
}
if (p_format_list) free(p_format_list);
}
if (p_streams) free(p_streams);
if (ao->i_stream_index < 0)
{
ao_msg(MSGT_AO, MSGL_WARN,
"Cannot find any digital output stream format when OpenSPDIF().\n");
goto err_out;
}
print_format(MSGL_V, "original stream format:", &ao->sfmt_revert);
if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
goto err_out;
property_address.mSelector = kAudioDevicePropertyDeviceHasChanged;
property_address.mScope = kAudioObjectPropertyScopeGlobal;
property_address.mElement = kAudioObjectPropertyElementMaster;
err = AudioObjectAddPropertyListener(ao->i_selected_dev,
&property_address,
DeviceListener,
NULL);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddPropertyListener for kAudioDevicePropertyDeviceHasChanged failed: [%4.4s]\n", (char *)&err);
/* FIXME: If output stream is not native byte-order, we need change endian somewhere. */
/* Although there's no such case reported. */
#if HAVE_BIGENDIAN
if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian))
#else
if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
#endif
ao_msg(MSGT_AO, MSGL_WARN,
"Output stream has non-native byte order, digital output may fail.\n");
/* For ac3/dts, just use packet size 6144 bytes as chunk size. */
ao->chunk_size = ao->stream_format.mBytesPerPacket;
ao_data.samplerate = ao->stream_format.mSampleRate;
ao_data.channels = ao->stream_format.mChannelsPerFrame;
ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket);
ao_data.outburst = ao->chunk_size;
ao_data.buffersize = ao_data.bps;
ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
ao->buffer_len = ao->num_chunks * ao->chunk_size;
ao->buffer = av_fifo_alloc(ao->buffer_len);
ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
/* Create IOProc callback. */
err = AudioDeviceCreateIOProcID(ao->i_selected_dev,
(AudioDeviceIOProc)RenderCallbackSPDIF,
(void *)ao,
&ao->renderCallback);
if (err != noErr || ao->renderCallback == NULL)
{
ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err);
goto err_out1;
}
reset();
return CONTROL_TRUE;
err_out1:
if (ao->b_revert)
AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
err_out:
if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
{
int b_mix = 1;
err = SetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertySupportsMixing,
sizeof(int), &b_mix);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n",
(char *)&err);
}
if (ao->i_hog_pid == getpid())
{
ao->i_hog_pid = -1;
err = SetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertyHogMode,
sizeof(ao->i_hog_pid), &ao->i_hog_pid);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n",
(char *)&err);
}
av_fifo_free(ao->buffer);
free(ao);
ao = NULL;
return CONTROL_FALSE;
}
/*
* Initialize CoreAudio and return an opaque pointer
* for keeping track of our audio device
*
*/
void *audio_init(void) {
OSStatus s;
AudioDeviceID adev_id;
AudioValueRange avr;
UInt32 framesize = 512;
UInt32 sz;
AudioStreamBasicDescription fmt;
CoreAudioDevice *device;
DSFYDEBUG("Initializing CoreAudio\n");
sz = sizeof (adev_id);
s = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, &sz, &adev_id);
if(s != 0) {
fprintf(stderr, "AudioHardwareGetProperty() failed with error %d\n", s);
return NULL;
}
else if (adev_id == kAudioDeviceUnknown) {
fprintf(stderr, "AudioHardwareGetProperty() returned device kAudioDeviceUnknown\n");
return NULL;
}
/* Find out the bounds of buffer frame size */
sz = sizeof(avr);
if ((s = AudioDeviceGetProperty(adev_id, 0, false,
kAudioDevicePropertyBufferFrameSizeRange, &sz, &avr))) {
printf("AudioDeviceGetProperty() failed with error %d\n", s);
return NULL;
}
/* Keep the requested number of frames within bounds */
if (framesize < avr.mMinimum)
framesize = avr.mMinimum;
else if (framesize > avr.mMaximum)
framesize = avr.mMaximum;
/* Set buffer frame size for device */
sz = sizeof (framesize);
s = AudioDeviceSetProperty (adev_id, 0, 0, false,
kAudioDevicePropertyBufferFrameSize, sz, &framesize);
if (s != kAudioHardwareNoError) {
fprintf(stderr, "AudioDeviceSetProperty() failed with error %d\n", s);
return NULL;
}
/* Get current audio format */
sz = sizeof (fmt);
if (AudioDeviceGetProperty
(adev_id, 0, false, kAudioDevicePropertyStreamFormat,
&sz, &fmt)) {
DSFYDEBUG ("AudioDeviceGetProperty() failed\n");
return NULL;
}
/* Setup audio format */
fmt.mFormatID = kAudioFormatLinearPCM;
fmt.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked;
fmt.mSampleRate = 44100;
fmt.mChannelsPerFrame = 2;
fmt.mBytesPerFrame = sizeof(Float32) * fmt.mChannelsPerFrame;
fmt.mFramesPerPacket = 1;
fmt.mBytesPerPacket = fmt.mFramesPerPacket * fmt.mBytesPerFrame;
fmt.mReserved = 0;
/* Update audio format */
sz = sizeof (fmt);
if (AudioDeviceSetProperty
(adev_id, NULL, 0, false,
kAudioDevicePropertyStreamFormat, sz, &fmt)) {
DSFYDEBUG ("AudioDeviceSetProperty() failed\n");
return NULL;
}
DSFYDEBUG ("kAudioDevicePropertyStreamFormat: mSampleRate %f\n",
fmt.mSampleRate);
DSFYDEBUG
("kAudioDevicePropertyStreamFormat: mFormatFlags 0x%08x (IsSignedInteger:%s, isFloat:%s, isBigEndian:%s, kLinearPCMFormatFlagIsNonInterleaved:%s, kAudioFormatFlagIsPacked:%s)\n",
fmt.mFormatFlags,
fmt.mFormatFlags & kLinearPCMFormatFlagIsSignedInteger ?
"yes" : "no",
fmt.mFormatFlags & kLinearPCMFormatFlagIsFloat ? "yes" :
"no",
fmt.mFormatFlags & kLinearPCMFormatFlagIsBigEndian ? "yes" :
"no",
fmt.mFormatFlags & kLinearPCMFormatFlagIsNonInterleaved ?
"yes" : "no",
fmt.mFormatFlags & kAudioFormatFlagIsPacked ? "yes" : "no");
DSFYDEBUG ("kAudioDevicePropertyStreamFormat: mBitsPerChannel %u\n",
fmt.mBitsPerChannel);
DSFYDEBUG
("kAudioDevicePropertyStreamFormat: mChannelsPerFrame %u\n",
fmt.mChannelsPerFrame);
DSFYDEBUG ("kAudioDevicePropertyStreamFormat: mFramesPerPacket %u\n",
fmt.mFramesPerPacket);
DSFYDEBUG ("kAudioDevicePropertyStreamFormat: mBytesPerFrame %u\n",
fmt.mBytesPerFrame);
DSFYDEBUG ("kAudioDevicePropertyStreamFormat: mBytesPerPacket %u\n",
fmt.mBytesPerPacket);
device = (CoreAudioDevice *)malloc(sizeof(CoreAudioDevice));
if(!device)
return NULL;
device->adev_id = adev_id;
device->playing = 0;
device->buflen = 0;
device->bufsize = sizeof(short) * 32768;
if (AudioDeviceCreateIOProcID
(device->adev_id, audio_callback, device, &device->proc_id)) {
DSFYDEBUG ("AudioDeviceCreateIOProcID() returned FAIL!\n");
free(device);
return NULL;
}
device->buf = (short *)malloc(device->bufsize);
pthread_mutex_init(&device->mutex, NULL);
pthread_cond_init(&device->event, NULL);
pthread_cond_init(&device->hold, NULL);
DSFYDEBUG("Done initializing CoreAudio\n");
return device;
}
int
audio_output_open(void)
{
UInt32 size;
/* Obtain the audio device ID */
size = sizeof adid;
if (AudioHardwareGetProperty(
kAudioHardwarePropertyDefaultOutputDevice,
&size, &adid) != 0 || adid == kAudioDeviceUnknown)
goto error;
/* Adjust the stream format */
size = sizeof afmt;
if (AudioDeviceGetProperty(adid, 0, false,
kAudioDevicePropertyStreamFormat, &size, &afmt) != 0 ||
afmt.mFormatID != kAudioFormatLinearPCM)
goto error;
/* To be set on the first run */
afmt.mSampleRate = 0;
afmt.mChannelsPerFrame = 0;
afmt.mBytesPerFrame = afmt.mChannelsPerFrame * sizeof (float);
afmt.mBytesPerPacket = afmt.mBytesPerFrame * afmt.mFramesPerPacket;
/* Decide what buffer size to use */
size = sizeof abuflen;
abuflen = 32768;
AudioDeviceSetProperty(adid, 0, 0, false,
kAudioDevicePropertyBufferSize, size, &abuflen);
if (AudioDeviceGetProperty(adid, 0, false,
kAudioDevicePropertyBufferSize, &size, &abuflen) != 0)
goto error;
#ifdef BUILD_VOLUME
/* Store the audio channels */
size = sizeof achans;
AudioDeviceGetProperty(adid, 0, false,
kAudioDevicePropertyPreferredChannelsForStereo, &size, &achans);
#endif /* BUILD_VOLUME */
/* The buffer size reported is in floats */
abuflen /= sizeof(float);
abufnew = g_malloc(abuflen * sizeof(int16_t));
abufcur = g_malloc(abuflen * sizeof(int16_t));
/* Locking down the buffer length */
abuflock = g_mutex_new();
abufdrained = g_cond_new();
/* Add our own I/O handling routine */
#ifdef MAC_OS_X_VERSION_10_5
if (AudioDeviceCreateIOProcID(adid, audio_output_ioproc, NULL,
&aprocid) != 0)
#else /* !MAC_OS_X_VERSION_10_5 */
if (AudioDeviceAddIOProc(adid, audio_output_ioproc, NULL) != 0)
#endif /* MAC_OS_X_VERSION_10_5 */
goto error;
return (0);
error:
g_printerr("%s\n", _("Cannot open the audio device."));
return (-1);
}
/*****************************************************************************
* Setup a encoded digital stream (SPDIF)
*****************************************************************************/
int CoreAudioAUHAL::OpenSPDIF(struct CoreAudioDeviceParameters *deviceParameters, const CStdString& strName, int channels, float sampleRate, int bitsPerSample, int packetSize)
{
OSStatus err = noErr;
UInt32 i_param_size = 0, b_mix = 0;
Boolean b_writeable = false;
AudioStreamID *p_streams = NULL;
int i = 0, i_streams = 0;
// We're digital.
s_lastPlayWasSpdif = true;
/* Start doing the SPDIF setup proces */
//deviceParameters->b_digital = true;
deviceParameters->b_changed_mixing = false;
/* Hog the device */
i_param_size = sizeof(deviceParameters->i_hog_pid);
deviceParameters->i_hog_pid = getpid();
err = AudioDeviceSetProperty(deviceParameters->device_id, 0, 0, FALSE,
kAudioDevicePropertyHogMode, i_param_size, &deviceParameters->i_hog_pid);
if( err != noErr )
{
CLog::Log(LOGERROR, "Failed to set hogmode: [%4.4s]", (char *)&err );
return false;
}
/* Set mixable to false if we are allowed to */
err = AudioDeviceGetPropertyInfo(deviceParameters->device_id, 0, FALSE, kAudioDevicePropertySupportsMixing,
&i_param_size, &b_writeable );
err = AudioDeviceGetProperty(deviceParameters->device_id, 0, FALSE, kAudioDevicePropertySupportsMixing,
&i_param_size, &b_mix );
if( !err && b_writeable )
{
b_mix = 0;
err = AudioDeviceSetProperty( deviceParameters->device_id, 0, 0, FALSE,
kAudioDevicePropertySupportsMixing, i_param_size, &b_mix );
deviceParameters->b_changed_mixing = true;
}
if( err != noErr )
{
CLog::Log(LOGERROR, "Failed to set mixmode: [%4.4s]", (char *)&err );
return false;
}
/* Get a list of all the streams on this device */
err = AudioDeviceGetPropertyInfo(deviceParameters->device_id, 0, FALSE,
kAudioDevicePropertyStreams,
&i_param_size, NULL );
if( err != noErr )
{
CLog::Log(LOGERROR, "Could not get number of streams: [%4.4s]", (char *)&err );
return false;
}
i_streams = i_param_size / sizeof( AudioStreamID );
p_streams = (AudioStreamID *)malloc( i_param_size );
if( p_streams == NULL )
return false;
err = AudioDeviceGetProperty(deviceParameters->device_id, 0, FALSE,
kAudioDevicePropertyStreams,
&i_param_size, p_streams );
if( err != noErr )
{
CLog::Log(LOGERROR, "Could not get number of streams: [%4.4s]", (char *)&err );
free( p_streams );
return false;
}
for( i = 0; i < i_streams && deviceParameters->i_stream_index < 0 ; i++ )
{
/* Find a stream with a cac3 stream */
AudioStreamBasicDescription *p_format_list = NULL;
int i_formats = 0, j = 0;
bool b_digital = false;
/* Retrieve all the stream formats supported by each output stream */
err = AudioStreamGetPropertyInfo( p_streams[i], 0,
kAudioStreamPropertyPhysicalFormats,
&i_param_size, NULL );
if( err != noErr )
{
CLog::Log(LOGERROR, "Could not get number of streamformats: [%4.4s]", (char *)&err );
continue;
}
i_formats = i_param_size / sizeof( AudioStreamBasicDescription );
p_format_list = (AudioStreamBasicDescription *)malloc( i_param_size );
if( p_format_list == NULL )
continue;
err = AudioStreamGetProperty( p_streams[i], 0,
kAudioStreamPropertyPhysicalFormats,
&i_param_size, p_format_list );
if( err != noErr )
{
CLog::Log(LOGERROR, "Could not get the list of streamformats: [%4.4s]", (char *)&err );
free( p_format_list );
continue;
}
/* Check if one of the supported formats is a digital format */
for( j = 0; j < i_formats; j++ )
{
if( p_format_list[j].mFormatID == 'IAC3' ||
p_format_list[j].mFormatID == kAudioFormat60958AC3 )
{
b_digital = true;
break;
}
}
if( b_digital )
{
/* if this stream supports a digital (cac3) format, then go set it. */
int i_requested_rate_format = -1;
int i_current_rate_format = -1;
int i_backup_rate_format = -1;
deviceParameters->i_stream_id = p_streams[i];
deviceParameters->i_stream_index = i;
if(deviceParameters->b_revert == false )
{
/* Retrieve the original format of this stream first if not done so already */
i_param_size = sizeof(deviceParameters->sfmt_revert);
err = AudioStreamGetProperty(deviceParameters->i_stream_id, 0,
kAudioStreamPropertyPhysicalFormat,
&i_param_size,
&deviceParameters->sfmt_revert );
if( err != noErr )
{
CLog::Log(LOGERROR, "Could not retrieve the original streamformat: [%4.4s]", (char *)&err );
//continue;
}
else deviceParameters->b_revert = true;
}
for( j = 0; j < i_formats; j++ )
{
if( p_format_list[j].mFormatID == 'IAC3' ||
p_format_list[j].mFormatID == kAudioFormat60958AC3 )
{
if( p_format_list[j].mSampleRate == sampleRate)
{
i_requested_rate_format = j;
break;
}
else if( p_format_list[j].mSampleRate == deviceParameters->sfmt_revert.mSampleRate )
{
i_current_rate_format = j;
}
else
{
if( i_backup_rate_format < 0 || p_format_list[j].mSampleRate > p_format_list[i_backup_rate_format].mSampleRate )
i_backup_rate_format = j;
}
}
}
if( i_requested_rate_format >= 0 ) /* We prefer to output at the samplerate of the original audio */
deviceParameters->stream_format = p_format_list[i_requested_rate_format];
else if( i_current_rate_format >= 0 ) /* If not possible, we will try to use the current samplerate of the device */
deviceParameters->stream_format = p_format_list[i_current_rate_format];
else deviceParameters->stream_format = p_format_list[i_backup_rate_format]; /* And if we have to, any digital format will be just fine (highest rate possible) */
}
free( p_format_list );
}
free( p_streams );
CLog::Log(LOGINFO, STREAM_FORMAT_MSG("original stream format: ", deviceParameters->sfmt_revert ) );
if( !AudioStreamChangeFormat(deviceParameters, deviceParameters->i_stream_id, deviceParameters->stream_format))
return false;
// Get device hardware buffer size
uint32_t audioDeviceLatency, audioStreamLatency, audioDeviceBufferFrameSize, audioDeviceSafetyOffset;
deviceParameters->hardwareFrameLatency = 0;
i_param_size = sizeof(uint32_t);
err = AudioDeviceGetProperty(deviceParameters->device_id,
0, false,
kAudioDevicePropertyLatency,
&i_param_size,
&audioDeviceLatency);
if (err == noErr) deviceParameters->hardwareFrameLatency += audioDeviceLatency;
err = AudioDeviceGetProperty(deviceParameters->device_id,
0, false,
kAudioDevicePropertyBufferFrameSize,
&i_param_size,
&audioDeviceBufferFrameSize);
if (err == noErr) deviceParameters->hardwareFrameLatency += audioDeviceBufferFrameSize;
err = AudioDeviceGetProperty(deviceParameters->device_id,
0, false,
kAudioDevicePropertySafetyOffset,
&i_param_size,
&audioDeviceSafetyOffset);
if (err == noErr) deviceParameters->hardwareFrameLatency += audioDeviceSafetyOffset;
err = AudioStreamGetProperty(deviceParameters->i_stream_id,
0,
kAudioStreamPropertyLatency,
&i_param_size,
&audioStreamLatency);
if (err == noErr) deviceParameters->hardwareFrameLatency += audioStreamLatency;
CLog::Log(LOGINFO, "Hardware latency: %i frames (%.2f msec @ %.0fHz)", deviceParameters->hardwareFrameLatency,
(float)deviceParameters->hardwareFrameLatency / deviceParameters->stream_format.mSampleRate * 1000,
deviceParameters->stream_format.mSampleRate);
// initialise the CoreAudio sink buffer
uint32_t framecount = 1;
while(framecount <= deviceParameters->stream_format.mSampleRate) // ensure power of 2
{
framecount <<= 1;
}
#warning free
deviceParameters->outputBuffer = (PaUtilRingBuffer *)malloc(sizeof(PaUtilRingBuffer));
deviceParameters->outputBufferData = calloc(1, framecount * channels * bitsPerSample/8); // use uncompressed size if encoding ac3
PaUtil_InitializeRingBuffer(deviceParameters->outputBuffer, channels * bitsPerSample/8, framecount, deviceParameters->outputBufferData);
/* Add IOProc callback */
err = AudioDeviceCreateIOProcID(deviceParameters->device_id,
(AudioDeviceIOProc)RenderCallbackSPDIF,
deviceParameters,
&deviceParameters->sInputIOProcID);
if( err != noErr )
{
CLog::Log(LOGERROR, "AudioDeviceAddIOProcID failed: [%4.4s]", (char *)&err );
return false;
}
/* Start device */
err = AudioDeviceStart(deviceParameters->device_id, (AudioDeviceIOProc)RenderCallbackSPDIF );
if( err != noErr )
{
CLog::Log(LOGERROR, "AudioDeviceStart failed: [%4.4s]", (char *)&err );
err = AudioDeviceDestroyIOProcID(deviceParameters->device_id,
(AudioDeviceIOProc)RenderCallbackSPDIF);
if( err != noErr )
{
CLog::Log(LOGERROR, "AudioDeviceRemoveIOProc failed: [%4.4s]", (char *)&err );
}
return false;
}
return true;
}
