OSStatus AUEffectBase::ChangeStreamFormat( AudioUnitScope inScope,
AudioUnitElement inElement,
const CAStreamBasicDescription & inPrevFormat,
const CAStreamBasicDescription & inNewFormat)
{
OSStatus result = AUBase::ChangeStreamFormat(inScope, inElement, inPrevFormat, inNewFormat);
if (result == noErr)
{
// for the moment this only dependency we know about
// where a parameter's range may change is with the sample rate
// and effects are only publishing parameters in the global scope!
if (GetParamHasSampleRateDependency() && fnotequal(inPrevFormat.mSampleRate, inNewFormat.mSampleRate))
PropertyChanged(kAudioUnitProperty_ParameterList, kAudioUnitScope_Global, 0);
}
return result;
}
OSStatus AudioOutputVolumeUp(AudioDeviceID device, UInt32 *level) {
Float32 right, left;
OSStatus err = AudioOutputGetVolume(device, &left, &right);
if (noErr == err) {
Float32 max = left > right ? left : right;
UInt32 lvl = __AudioOutputVolumeGetLevel(max);
if (kAudioOutputVolumeMaxLevel == lvl) {
/* If not max level */
if (fnotequal(max, 1)) {
err = _AudioOutputSetVolume(device, left, right, 1);
}
} else {
lvl++;
check(lvl <= kAudioOutputVolumeMaxLevel);
err = _AudioOutputSetVolume(device, left, right, kAudioOutputVolumeLevels[lvl]);
}
if (level) *level = lvl;
}
return err;
}
bool operator<(const AudioStreamBasicDescription& x, const AudioStreamBasicDescription& y)
{
bool theAnswer = false;
bool isDone = false;
// note that if either side is 0, that field is skipped
// format ID is the first order sort
if((!isDone) && ((x.mFormatID != 0) && (y.mFormatID != 0)))
{
if(x.mFormatID != y.mFormatID)
{
// formats are sorted numerically except that linear
// PCM is always first
if(x.mFormatID == kAudioFormatLinearPCM)
{
theAnswer = true;
}
else if(y.mFormatID == kAudioFormatLinearPCM)
{
theAnswer = false;
}
else
{
theAnswer = x.mFormatID < y.mFormatID;
}
isDone = true;
}
}
// mixable is always better than non-mixable for linear PCM and should be the second order sort item
if((!isDone) && ((x.mFormatID == kAudioFormatLinearPCM) && (y.mFormatID == kAudioFormatLinearPCM)))
{
if(((x.mFormatFlags & kIsNonMixableFlag) == 0) && ((y.mFormatFlags & kIsNonMixableFlag) != 0))
{
theAnswer = true;
isDone = true;
}
else if(((x.mFormatFlags & kIsNonMixableFlag) != 0) && ((y.mFormatFlags & kIsNonMixableFlag) == 0))
{
theAnswer = false;
isDone = true;
}
}
// floating point vs integer for linear PCM only
if((!isDone) && ((x.mFormatID == kAudioFormatLinearPCM) && (y.mFormatID == kAudioFormatLinearPCM)))
{
if((x.mFormatFlags & kAudioFormatFlagIsFloat) != (y.mFormatFlags & kAudioFormatFlagIsFloat))
{
// floating point is better than integer
theAnswer = y.mFormatFlags & kAudioFormatFlagIsFloat;
isDone = true;
}
}
// bit depth
if((!isDone) && ((x.mBitsPerChannel != 0) && (y.mBitsPerChannel != 0)))
{
if(x.mBitsPerChannel != y.mBitsPerChannel)
{
// deeper bit depths are higher quality
theAnswer = x.mBitsPerChannel < y.mBitsPerChannel;
isDone = true;
}
}
// sample rate
if((!isDone) && fnonzero(x.mSampleRate) && fnonzero(y.mSampleRate))
{
if(fnotequal(x.mSampleRate, y.mSampleRate))
{
// higher sample rates are higher quality
theAnswer = x.mSampleRate < y.mSampleRate;
isDone = true;
}
}
// number of channels
if((!isDone) && ((x.mChannelsPerFrame != 0) && (y.mChannelsPerFrame != 0)))
{
if(x.mChannelsPerFrame != y.mChannelsPerFrame)
{
// more channels is higher quality
theAnswer = x.mChannelsPerFrame < y.mChannelsPerFrame;
isDone = true;
}
}
return theAnswer;
}
// _______________________________________________________________________________________
//
// called to create the file -- or update its format/channel layout/properties based on an encoder
// setting change
void CAAudioFile::FileFormatChanged(const FSRef *parentDir, CFStringRef filename, AudioFileTypeID filetype)
{
LOG_FUNCTION("CAAudioFile::FileFormatChanged", "%p", this);
XThrowIf(mMode != kPreparingToCreate && mMode != kPreparingToWrite, kExtAudioFileError_InvalidOperationOrder, "new file not prepared");
UInt32 propertySize;
OSStatus err;
AudioStreamBasicDescription saveFileDataFormat = mFileDataFormat;
#if VERBOSE_CONVERTER
mFileDataFormat.PrintFormat(stdout, "", "Specified file data format");
#endif
// Find out the actual format the converter will produce. This is necessary in
// case the bitrate has forced a lower sample rate, which needs to be set correctly
// in the stream description passed to AudioFileCreate.
if (mConverter != NULL) {
propertySize = sizeof(AudioStreamBasicDescription);
Float64 origSampleRate = mFileDataFormat.mSampleRate;
XThrowIfError(AudioConverterGetProperty(mConverter, kAudioConverterCurrentOutputStreamDescription, &propertySize, &mFileDataFormat), "get audio converter's output stream description");
// do the same for the channel layout being output by the converter
#if VERBOSE_CONVERTER
mFileDataFormat.PrintFormat(stdout, "", "Converter output");
#endif
if (fiszero(mFileDataFormat.mSampleRate))
mFileDataFormat.mSampleRate = origSampleRate;
err = AudioConverterGetPropertyInfo(mConverter, kAudioConverterOutputChannelLayout, &propertySize, NULL);
if (err == noErr && propertySize > 0) {
AudioChannelLayout *layout = static_cast<AudioChannelLayout *>(malloc(propertySize));
err = AudioConverterGetProperty(mConverter, kAudioConverterOutputChannelLayout, &propertySize, layout);
if (err) {
free(layout);
XThrow(err, "couldn't get audio converter's output channel layout");
}
mFileChannelLayout = layout;
#if VERBOSE_CHANNELMAP
printf("got new file's channel layout from converter: %s\n", CAChannelLayouts::ConstantToString(mFileChannelLayout.Tag()));
#endif
free(layout);
}
}
// create the output file
if (mMode == kPreparingToCreate) {
CAStreamBasicDescription newFileDataFormat = mFileDataFormat;
if (fiszero(newFileDataFormat.mSampleRate))
newFileDataFormat.mSampleRate = 44100; // just make something up for now
#if VERBOSE_CONVERTER
newFileDataFormat.PrintFormat(stdout, "", "Applied to new file");
#endif
XThrowIfError(AudioFileCreate(parentDir, filename, filetype, &newFileDataFormat, 0, &mFSRef, &mAudioFile), "create audio file");
mMode = kPreparingToWrite;
mOwnOpenFile = true;
} else if (saveFileDataFormat != mFileDataFormat || fnotequal(saveFileDataFormat.mSampleRate, mFileDataFormat.mSampleRate)) {
// second check must be explicit since operator== on ASBD treats SR of zero as "don't care"
if (fiszero(mFileDataFormat.mSampleRate))
mFileDataFormat.mSampleRate = mClientDataFormat.mSampleRate;
#if VERBOSE_CONVERTER
mFileDataFormat.PrintFormat(stdout, "", "Applied to new file");
#endif
XThrowIf(fiszero(mFileDataFormat.mSampleRate), kExtAudioFileError_InvalidDataFormat, "file's sample rate is 0");
XThrowIfError(AudioFileSetProperty(mAudioFile, kAudioFilePropertyDataFormat, sizeof(AudioStreamBasicDescription), &mFileDataFormat), "couldn't update file's data format");
}
UInt32 deferSizeUpdates = 1;
err = AudioFileSetProperty(mAudioFile, kAudioFilePropertyDeferSizeUpdates, sizeof(UInt32), &deferSizeUpdates);
if (mConverter != NULL) {
// encoder
// get the magic cookie, if any, from the converter
delete[] mMagicCookie; mMagicCookie = NULL;
mMagicCookieSize = 0;
err = AudioConverterGetPropertyInfo(mConverter, kAudioConverterCompressionMagicCookie, &propertySize, NULL);
// we can get a noErr result and also a propertySize == 0
// -- if the file format does support magic cookies, but this file doesn't have one.
if (err == noErr && propertySize > 0) {
mMagicCookie = new Byte[propertySize];
XThrowIfError(AudioConverterGetProperty(mConverter, kAudioConverterCompressionMagicCookie, &propertySize, mMagicCookie), "get audio converter's magic cookie");
mMagicCookieSize = propertySize; // the converter lies and tell us the wrong size
// now set the magic cookie on the output file
UInt32 willEatTheCookie = false;
// the converter wants to give us one; will the file take it?
err = AudioFileGetPropertyInfo(mAudioFile, kAudioFilePropertyMagicCookieData,
NULL, &willEatTheCookie);
if (err == noErr && willEatTheCookie) {
#if VERBOSE_CONVERTER
printf("Setting cookie on encoded file\n");
#endif
XThrowIfError(AudioFileSetProperty(mAudioFile, kAudioFilePropertyMagicCookieData, mMagicCookieSize, mMagicCookie), "set audio file's magic cookie");
}
}
// get maximum packet size
propertySize = sizeof(UInt32);
XThrowIfError(AudioConverterGetProperty(mConverter, kAudioConverterPropertyMaximumOutputPacketSize, &propertySize, &mFileMaxPacketSize), "get audio converter's maximum output packet size");
AllocateBuffers(true /* okToFail */);
} else {
InitFileMaxPacketSize();
}
if (mFileChannelLayout.IsValid() && mFileChannelLayout.NumberChannels() > 2) {
// don't bother tagging mono/stereo files
UInt32 isWritable;
err = AudioFileGetPropertyInfo(mAudioFile, kAudioFilePropertyChannelLayout, NULL, &isWritable);
if (!err && isWritable) {
#if VERBOSE_CHANNELMAP
printf("writing file's channel layout: %s\n", CAChannelLayouts::ConstantToString(mFileChannelLayout.Tag()));
#endif
err = AudioFileSetProperty(mAudioFile, kAudioFilePropertyChannelLayout,
mFileChannelLayout.Size(), &mFileChannelLayout.Layout());
if (err)
CAXException::Warning("could not set the file's channel layout", err);
} else {
#if VERBOSE_CHANNELMAP
printf("file won't accept a channel layout (write)\n");
#endif
}
}
UpdateClientMaxPacketSize(); // also sets mFrame0Offset
mPacketMark = 0;
mFrameMark = 0;
}
