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;
}
bool CAUOutputDevice::GetPreferredChannelLayout(CCoreAudioChannelLayout& layout)
{
if (!m_DeviceId)
return false;
AudioObjectPropertyAddress propertyAddress;
propertyAddress.mScope = kAudioDevicePropertyScopeOutput;
propertyAddress.mElement = 0;
propertyAddress.mSelector = kAudioDevicePropertyPreferredChannelLayout;
if (!AudioObjectHasProperty(m_DeviceId, &propertyAddress))
return false;
UInt32 propertySize = 0;
OSStatus ret = AudioObjectGetPropertyDataSize(m_DeviceId, &propertyAddress, 0, NULL, &propertySize);
if (ret != noErr)
CLog::Log(LOGERROR, "CAUOutputDevice::GetPreferredChannelLayout: "
"Unable to retrieve preferred channel layout size. Error = %s", GetError(ret).c_str());
void *pBuf = malloc(propertySize);
ret = AudioObjectGetPropertyData(m_DeviceId, &propertyAddress, 0, NULL, &propertySize, pBuf);
if (ret != noErr)
CLog::Log(LOGERROR, "CAUOutputDevice::GetPreferredChannelLayout: "
"Unable to retrieve preferred channel layout. Error = %s", GetError(ret).c_str());
else
{
// Copy the result into the caller's instance
layout.CopyLayout(*((AudioChannelLayout*)pBuf));
}
free(pBuf);
return (ret == noErr);
}
static inline gboolean
_audio_device_set_mixing (AudioDeviceID device_id, gboolean enable_mix)
{
OSStatus status = noErr;
UInt32 propertySize = 0, can_mix = enable_mix;
Boolean writable = FALSE;
gboolean res = FALSE;
AudioObjectPropertyAddress audioDeviceSupportsMixingAddress = {
kAudioDevicePropertySupportsMixing,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster
};
if (AudioObjectHasProperty (device_id, &audioDeviceSupportsMixingAddress)) {
/* Set mixable to false if we are allowed to */
status = AudioObjectIsPropertySettable (device_id,
&audioDeviceSupportsMixingAddress, &writable);
if (status) {
GST_DEBUG ("AudioObjectIsPropertySettable: %d", (int) status);
}
status = AudioObjectGetPropertyDataSize (device_id,
&audioDeviceSupportsMixingAddress, 0, NULL, &propertySize);
if (status) {
GST_DEBUG ("AudioObjectGetPropertyDataSize: %d", (int) status);
}
status = AudioObjectGetPropertyData (device_id,
&audioDeviceSupportsMixingAddress, 0, NULL, &propertySize, &can_mix);
if (status) {
GST_DEBUG ("AudioObjectGetPropertyData: %d", (int) status);
}
if (status == noErr && writable) {
can_mix = enable_mix;
status = AudioObjectSetPropertyData (device_id,
&audioDeviceSupportsMixingAddress, 0, NULL, propertySize, &can_mix);
res = TRUE;
}
if (status != noErr) {
GST_ERROR ("failed to set mixmode: %d", (int) status);
}
} else {
GST_DEBUG ("property not found, mixing coudln't be changed");
}
return res;
}
static BOOL DeviceHasMute(AudioDeviceID deviceID, Boolean isInput)
{
Boolean writable = false;
OSStatus err = noErr;
AudioObjectPropertyAddress propertyAddress;
propertyAddress.mSelector = kAudioDevicePropertyMute;
propertyAddress.mScope = isInput ? kAudioDevicePropertyScopeInput : kAudioDevicePropertyScopeOutput;
propertyAddress.mElement = 0;
if (AudioObjectHasProperty(deviceID, &propertyAddress))
{
/* check if we can set it */
err = AudioObjectIsPropertySettable(deviceID, &propertyAddress, &writable);
if (err == noErr)
return writable;
}
return FALSE;
}
bool CCoreAudioDevice::GetMixingSupport()
{
if (!m_DeviceId)
return false;
UInt32 size;
UInt32 mix = 0;
Boolean writable = false;
AudioObjectPropertyAddress propertyAddress;
propertyAddress.mScope = kAudioDevicePropertyScopeOutput;
propertyAddress.mElement = 0;
propertyAddress.mSelector = kAudioDevicePropertySupportsMixing;
if( AudioObjectHasProperty( m_DeviceId, &propertyAddress ) )
{
OSStatus ret = AudioObjectIsPropertySettable(m_DeviceId, &propertyAddress, &writable);
if (ret)
{
CLog::Log(LOGERROR, "CCoreAudioDevice::SupportsMixing: "
"Unable to get propertyinfo mixing support. Error = %s", GetError(ret).c_str());
writable = false;
}
if (writable)
{
size = sizeof(mix);
ret = AudioObjectGetPropertyData(m_DeviceId, &propertyAddress, 0, NULL, &size, &mix);
if (ret != noErr)
mix = 0;
}
}
CLog::Log(LOGDEBUG, "CCoreAudioDevice::SupportsMixing: "
"Device mixing support : %s.", mix ? "'Yes'" : "'No'");
return (mix > 0);
}
static OSStatus ca_change_mixing(struct ao *ao, AudioDeviceID device,
uint32_t val, bool *changed) {
*changed = false;
AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
.mSelector = kAudioDevicePropertySupportsMixing,
.mScope = kAudioObjectPropertyScopeGlobal,
.mElement = kAudioObjectPropertyElementMaster,
};
if (AudioObjectHasProperty(device, &p_addr)) {
OSStatus err;
Boolean writeable = 0;
err = CA_SETTABLE(device, kAudioDevicePropertySupportsMixing,
&writeable);
if (!CHECK_CA_WARN("can't tell if mixing property is settable")) {
return err;
}
if (!writeable)
return noErr;
err = CA_SET(device, kAudioDevicePropertySupportsMixing, &val);
if (err != noErr)
return err;
if (!CHECK_CA_WARN("can't set mix mode")) {
return err;
}
*changed = true;
}
return noErr;
}
bool CAHALAudioObject::HasProperty(AudioObjectPropertyAddress& inAddress) const
{
return AudioObjectHasProperty(mObjectID, &inAddress);
}
int main (int argc, const char * argv[])
{
OSStatus result = noErr;
Float32 theVolume = 0.0;
UInt32 theMute = 0;
Boolean doSetMute = false, doPrintDeviceList = true;
const char* theNewDefaultDeviceString = NULL;
Boolean didFindNewDevice = false;
CFStringRef theNewDefaultDeviceName = 0;
for (int i = 1; i < argc; ++i) {
const char *arg = argv[i];
if (arg[0] != '-') {
usage();
} else {
arg += 1;
if (arg[0] == 'v' || !strcmp(arg, "-volume")) {
if (++i == argc)
MissingArgument();
arg = argv[i];
sscanf(arg, "%f", &theVolume);
} else if (arg[0] == 'm' || !strcmp(arg, "-mute")) {
if (++i == argc)
MissingArgument();
arg = argv[i];
doSetMute = true;
sscanf(arg, "%d", &theMute);
} else if (arg[0] == 'd' || !strcmp(arg, "-device")) {
if (++i == argc)
MissingArgument();
theNewDefaultDeviceString = argv[i];
theNewDefaultDeviceName = CFStringCreateWithCString(kCFAllocatorDefault, theNewDefaultDeviceString, CFStringGetSystemEncoding());
} else if (arg[0] == 'a' || !strcmp(arg, "-all")) {
doPrintDeviceList = true;
} else if (arg[0] == 'h' || !strcmp(arg, "-help")) {
usage();
} else {
fprintf(stderr, "unknown argument: %s\n\n", arg - 1);
usage();
}
}
}
UInt32 thePropSize;
AudioDeviceID *theDeviceList = NULL;
UInt32 theNumDevices = 0;
if (doPrintDeviceList || theNewDefaultDeviceString)
{
// get the device list
AudioObjectPropertyAddress thePropertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
result = AudioObjectGetPropertyDataSize(kAudioObjectSystemObject, &thePropertyAddress, 0, NULL, &thePropSize);
if (result) { printf("Error in AudioObjectGetPropertyDataSize: %d\n", result); goto end; }
// Find out how many devices are on the system
theNumDevices = thePropSize / sizeof(AudioDeviceID);
theDeviceList = (AudioDeviceID*)calloc(theNumDevices, sizeof(AudioDeviceID));
result = AudioObjectGetPropertyData(kAudioObjectSystemObject, &thePropertyAddress, 0, NULL, &thePropSize, theDeviceList);
if (result) { printf("Error in AudioObjectGetPropertyData: %d\n", result); goto end; }
CFStringRef theDeviceName;
for (UInt32 i=0; i < theNumDevices; i++)
{
// get the device name
thePropSize = sizeof(CFStringRef);
thePropertyAddress.mSelector = kAudioObjectPropertyName;
thePropertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
thePropertyAddress.mElement = kAudioObjectPropertyElementMaster;
result = AudioObjectGetPropertyData(theDeviceList[i], &thePropertyAddress, 0, NULL, &thePropSize, &theDeviceName);
if (result) { printf("Error in AudioObjectGetPropertyData: %d\n", result); goto end; }
if (doPrintDeviceList)
CFShow(theDeviceName);
if (theNewDefaultDeviceString)
{
if (CFStringCompare(theDeviceName, theNewDefaultDeviceName, 0) == kCFCompareEqualTo)
{
// we found the device, now it as the default output device
thePropertyAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
thePropertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
thePropertyAddress.mElement = kAudioObjectPropertyElementMaster;
result = AudioObjectSetPropertyData(kAudioObjectSystemObject, &thePropertyAddress, 0, NULL, sizeof(AudioDeviceID), &theDeviceList[i]);
if (result) { printf("Error in AudioObjectSetPropertyData: kAudioHardwarePropertyDefaultOutputDevice: %d\n", result); goto end; }
else
{
didFindNewDevice = true;
printf("\tSuccessfully changed default output device to %s\n", theNewDefaultDeviceString);
if (!doPrintDeviceList) break;
}
}
}
CFRelease(theDeviceName);
}
if (theNewDefaultDeviceString)
{
if (!didFindNewDevice)
printf("Error! Could not find specified device %s. Using current default output device\n", theNewDefaultDeviceString);
CFRelease(theNewDefaultDeviceName);
}
}
if (doSetMute || theVolume > 0.)
{
AudioDeviceID theDefaultOutputDeviceID;
thePropSize = sizeof(theDefaultOutputDeviceID);
CFStringRef theDefaultOutputDeviceName;
AudioObjectPropertyAddress thePropertyAddress = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
// get the ID of the default output device
result = AudioObjectGetPropertyData(kAudioObjectSystemObject, &thePropertyAddress, 0, NULL, &thePropSize, &theDefaultOutputDeviceID);
if (result) { printf("Error in AudioObjectGetPropertyData: %d\n", result); goto end; }
thePropSize = sizeof(CFStringRef);
thePropertyAddress.mSelector = kAudioObjectPropertyName;
thePropertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
thePropertyAddress.mElement = kAudioObjectPropertyElementMaster;
// get the name of the default output device
result = AudioObjectGetPropertyData(theDefaultOutputDeviceID, &thePropertyAddress, 0, NULL, &thePropSize, &theDefaultOutputDeviceName);
if (result) { printf("Error in AudioObjectGetPropertyData: %d\n", result); goto end; }
const char* theDefaultOutputDeviceString = CFStringGetCStringPtr(theDefaultOutputDeviceName, CFStringGetSystemEncoding());
if (doSetMute)
{
thePropSize = sizeof(theMute);
thePropertyAddress.mSelector = kAudioDevicePropertyMute;
thePropertyAddress.mScope = kAudioDevicePropertyScopeOutput;
thePropertyAddress.mElement = kAudioObjectPropertyElementMaster;
// check if the device supports setting mute
if (AudioObjectHasProperty(theDefaultOutputDeviceID, &thePropertyAddress))
{
printf("\tSetting %s mute %s\n", theDefaultOutputDeviceString, (theMute) ? "on" : "off");
// if so then set it
result = AudioObjectSetPropertyData(theDefaultOutputDeviceID, &thePropertyAddress, 0, NULL, thePropSize, &theMute);
if (result) { printf("Error in AudioObjectSetPropertyData: %d\n", result); goto end; }
}
else
{
// if the device does not support master mute control, do nothing
printf("Error: the default output device does not support mute control\n");
}
}
if (theVolume > 0.)
{
thePropSize = sizeof(theVolume);
thePropertyAddress.mSelector = kAudioDevicePropertyVolumeScalar;
thePropertyAddress.mScope = kAudioDevicePropertyScopeOutput;
thePropertyAddress.mElement = kAudioObjectPropertyElementMaster;
// see if the device supports volume control, if so, then set the user specified volume
if (AudioObjectHasProperty(theDefaultOutputDeviceID, &thePropertyAddress))
{
printf("\tSetting %s volume to %g\n", theDefaultOutputDeviceString, theVolume);
result = AudioObjectSetPropertyData(theDefaultOutputDeviceID, &thePropertyAddress, 0, NULL, thePropSize, &theVolume);
if (result) { printf("Error in AudioObjectSetPropertyData: %d\n", result); goto end; }
}
else
{
// if the device does not support master volume control, do nothing
printf("Error: the default output device does not support volume control\n");
}
}
CFRelease(theDefaultOutputDeviceName);
}
end:
if (theDeviceList)
free(theDeviceList);
return result;
}
