// _______________________________________________________________________________________
//
void CAAudioFile::UpdateClientMaxPacketSize()
{
LOG_FUNCTION("CAAudioFile::UpdateClientMaxPacketSize", "%p", this);
mFrame0Offset = 0;
if (mConverter != NULL) {
AudioConverterPropertyID property = (mMode == kReading) ?
kAudioConverterPropertyMaximumOutputPacketSize :
kAudioConverterPropertyMaximumInputPacketSize;
UInt32 propertySize = sizeof(UInt32);
XThrowIfError(AudioConverterGetProperty(mConverter, property, &propertySize, &mClientMaxPacketSize),
"get audio converter's maximum packet size");
if (mFileDataFormat.mBitsPerChannel == 0) {
AudioConverterPrimeInfo primeInfo;
propertySize = sizeof(primeInfo);
OSStatus err = AudioConverterGetProperty(mConverter, kAudioConverterPrimeInfo, &propertySize, &primeInfo);
if (err == noErr)
mFrame0Offset = primeInfo.leadingFrames;
#if VERBOSE_CONVERTER
printf("kAudioConverterPrimeInfo: err = %ld, leadingFrames = %ld\n", err, mFrame0Offset);
#endif
}
} else {
mClientMaxPacketSize = mFileMaxPacketSize;
}
}
static int ffat_update_ctx(AVCodecContext *avctx)
{
ATDecodeContext *at = avctx->priv_data;
AudioStreamBasicDescription format;
UInt32 size = sizeof(format);
if (!AudioConverterGetProperty(at->converter,
kAudioConverterCurrentInputStreamDescription,
&size, &format)) {
if (format.mSampleRate)
avctx->sample_rate = format.mSampleRate;
avctx->channels = format.mChannelsPerFrame;
avctx->channel_layout = av_get_default_channel_layout(avctx->channels);
avctx->frame_size = format.mFramesPerPacket;
}
if (!AudioConverterGetProperty(at->converter,
kAudioConverterCurrentOutputStreamDescription,
&size, &format)) {
format.mSampleRate = avctx->sample_rate;
format.mChannelsPerFrame = avctx->channels;
AudioConverterSetProperty(at->converter,
kAudioConverterCurrentOutputStreamDescription,
size, &format);
}
if (!AudioConverterGetPropertyInfo(at->converter, kAudioConverterOutputChannelLayout,
&size, NULL) && size) {
AudioChannelLayout *layout = av_malloc(size);
uint64_t layout_mask = 0;
int i;
if (!layout)
return AVERROR(ENOMEM);
AudioConverterGetProperty(at->converter, kAudioConverterOutputChannelLayout,
&size, layout);
if (!(layout = ffat_convert_layout(layout, &size)))
return AVERROR(ENOMEM);
for (i = 0; i < layout->mNumberChannelDescriptions; i++) {
int id = ffat_get_channel_id(layout->mChannelDescriptions[i].mChannelLabel);
if (id < 0)
goto done;
if (layout_mask & (1 << id))
goto done;
layout_mask |= 1 << id;
layout->mChannelDescriptions[i].mChannelFlags = i; // Abusing flags as index
}
avctx->channel_layout = layout_mask;
qsort(layout->mChannelDescriptions, layout->mNumberChannelDescriptions,
sizeof(AudioChannelDescription), &ffat_compare_channel_descriptions);
for (i = 0; i < layout->mNumberChannelDescriptions; i++)
at->channel_map[i] = layout->mChannelDescriptions[i].mChannelFlags;
done:
av_free(layout);
}
if (!avctx->frame_size)
avctx->frame_size = 2048;
return 0;
}
// Some audio formats have a magic cookie associated with them which is required to decompress audio data
// When converting audio, a magic cookie may be returned by the Audio Converter so that it may be stored along with
// the output data -- This is done so that it may then be passed back to the Audio Converter at a later time as required
static void WriteCookie(AudioConverterRef converter, AudioFileID destinationFileID)
{
// grab the cookie from the converter and write it to the destinateion file
UInt32 cookieSize = 0;
OSStatus error = AudioConverterGetPropertyInfo(converter, kAudioConverterCompressionMagicCookie, &cookieSize, NULL);
// if there is an error here, then the format doesn't have a cookie - this is perfectly fine as some formats do not
if (noErr == error && 0 != cookieSize) {
char* cookie = new char [cookieSize];
error = AudioConverterGetProperty(converter, kAudioConverterCompressionMagicCookie, &cookieSize, cookie);
if (noErr == error) {
error = AudioFileSetProperty(destinationFileID, kAudioFilePropertyMagicCookieData, cookieSize, cookie);
if (noErr == error) {
printf("Writing magic cookie to destination file: %ld\n", cookieSize);
} else {
printf("Even though some formats have cookies, some files don't take them and that's OK\n");
}
} else {
printf("Could not Get kAudioConverterCompressionMagicCookie from Audio Converter!\n");
}
delete [] cookie;
}
}
void
AACEncode::setBitrate(int bitrate)
{
if(m_bitrate != bitrate) {
m_converterMutex.lock();
UInt32 br = bitrate;
AudioConverterDispose(m_audioConverter);
const OSType subtype = kAudioFormatMPEG4AAC;
AudioClassDescription requestedCodecs[2] = {
{
kAudioEncoderComponentType,
subtype,
kAppleSoftwareAudioCodecManufacturer
},
{
kAudioEncoderComponentType,
subtype,
kAppleHardwareAudioCodecManufacturer
}
};
AudioConverterNewSpecific(&m_in, &m_out, 2,requestedCodecs, &m_audioConverter);
OSStatus result = AudioConverterSetProperty(m_audioConverter, kAudioConverterEncodeBitRate, sizeof(br), &br);
UInt32 propSize = sizeof(br);
if(result == noErr) {
AudioConverterGetProperty(m_audioConverter, kAudioConverterEncodeBitRate, &propSize, &br);
m_bitrate = br;
}
m_converterMutex.unlock();
}
}
// _______________________________________________________________________________________
//
void CAAudioFile::FlushEncoder()
{
if (mConverter != NULL) {
mFinishingEncoding = true;
WritePacketsFromCallback(WriteInputProc, this);
mFinishingEncoding = false;
// get priming info from converter, set it on the file
if (mFileDataFormat.mBitsPerChannel == 0) {
UInt32 propertySize;
OSStatus err;
AudioConverterPrimeInfo primeInfo;
propertySize = sizeof(primeInfo);
err = AudioConverterGetProperty(mConverter, kAudioConverterPrimeInfo, &propertySize, &primeInfo);
if (err == noErr) {
AudioFilePacketTableInfo pti;
propertySize = sizeof(pti);
err = AudioFileGetProperty(mAudioFile, kAudioFilePropertyPacketTableInfo, &propertySize, &pti);
if (err == noErr) {
//printf("old packet table info: %qd valid, %ld priming, %ld remainder\n", pti.mNumberValidFrames, pti.mPrimingFrames, pti.mRemainderFrames);
UInt64 totalFrames = pti.mNumberValidFrames + pti.mPrimingFrames + pti.mRemainderFrames;
pti.mPrimingFrames = primeInfo.leadingFrames;
pti.mRemainderFrames = primeInfo.trailingFrames;
pti.mNumberValidFrames = totalFrames - pti.mPrimingFrames - pti.mRemainderFrames;
//printf("new packet table info: %qd valid, %ld priming, %ld remainder\n", pti.mNumberValidFrames, pti.mPrimingFrames, pti.mRemainderFrames);
XThrowIfError(AudioFileSetProperty(mAudioFile, kAudioFilePropertyPacketTableInfo, sizeof(pti), &pti), "couldn't set packet table info on audio file");
}
}
}
}
}
// _______________________________________________________________________________________
//
CFArrayRef CAAudioFile::GetConverterConfig()
{
CFArrayRef plist;
UInt32 propertySize = sizeof(plist);
XThrowIfError(AudioConverterGetProperty(mConverter, kAudioConverterPropertySettings, &propertySize, &plist), "get converter property settings");
return plist;
}
// setup conversion from Squeak to device frame format, or vice-versa.
// requires: stereo for output, stereo or mono for input.
//
static int Stream_setFormat(Stream *s, int frameCount, int sampleRate, int stereo)
{
int nChannels= 1 + stereo;
AudioStreamBasicDescription imgFmt, devFmt;
UInt32 sz= sizeof(devFmt);
if (0 == s->direction) nChannels= 2; // insist
if (checkError(AudioDeviceGetProperty(s->id, 0, s->direction,
kAudioDevicePropertyStreamFormat,
&sz, &devFmt),
"GetProperty", "StreamFormat"))
return 0;
debugf("stream %p[%d] device format:\n", s, s->direction); dumpFormat(&devFmt);
imgFmt.mSampleRate = sampleRate;
imgFmt.mFormatID = kAudioFormatLinearPCM;
#if defined(WORDS_BIGENDIAN)
imgFmt.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsBigEndian;
#else
imgFmt.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
#endif
imgFmt.mBytesPerPacket = SqueakFrameSize / (3 - nChannels);
imgFmt.mFramesPerPacket = 1;
imgFmt.mBytesPerFrame = SqueakFrameSize / (3 - nChannels);
imgFmt.mChannelsPerFrame = nChannels;
imgFmt.mBitsPerChannel = 16;
debugf("stream %p[%d] image format:\n", s, s->direction); dumpFormat(&imgFmt);
if (s->direction) // input
{
if (checkError(AudioConverterNew(&devFmt, &imgFmt, &s->converter), "AudioConverter", "New"))
return 0;
sz= sizeof(s->cvtBufSize);
s->cvtBufSize= 512 * devFmt.mBytesPerFrame;
if (checkError(AudioConverterGetProperty(s->converter, kAudioConverterPropertyCalculateOutputBufferSize,
&sz, &s->cvtBufSize),
"GetProperty", "OutputBufferSize"))
return 0;
}
else // output
{
if (checkError(AudioConverterNew(&imgFmt, &devFmt, &s->converter), "AudioConverter", "New"))
return 0;
}
s->channels= nChannels;
s->sampleRate= sampleRate;
s->imgBufSize= SqueakFrameSize * nChannels * frameCount;
frameCount= max(frameCount, 512 * sampleRate / devFmt.mSampleRate);
s->buffer= Buffer_new((s->direction ? DeviceFrameSize : SqueakFrameSize) * nChannels * frameCount * 2);
debugf("stream %p[%d] sound buffer size %d/%d (%d)\n", s, s->direction, s->imgBufSize, s->buffer->size, frameCount);
return 1;
}
UInt32 AudioConverterX::getEncodeBitRate()
{
UInt32 value;
UInt32 size = sizeof value;
CHECKCA(AudioConverterGetProperty(m_converter.get(),
kAudioConverterEncodeBitRate, &size, &value));
return value;
}
UInt32 AudioConverterX::getSampleRateConverterComplexity()
{
UInt32 value;
UInt32 size = sizeof value;
CHECKCA(AudioConverterGetProperty(m_converter.get(),
kAudioConverterSampleRateConverterComplexity, &size, &value));
return value;
}
UInt32 AudioConverterX::getCodecQuality()
{
UInt32 result;
UInt32 size = sizeof result;
CHECKCA(AudioConverterGetProperty(m_converter.get(),
kAudioConverterCodecQuality, &size, &result));
return result;
}
UInt32 AudioConverterX::getMaximumOutputPacketSize()
{
UInt32 result;
UInt32 size = sizeof result;
CHECKCA(AudioConverterGetProperty(m_converter.get(),
kAudioConverterPropertyMaximumOutputPacketSize,
&size, &result));
return result;
}
UInt32 AudioConverterX::getBitRateControlMode()
{
UInt32 result;
UInt32 size = sizeof result;
CHECKCA(AudioConverterGetProperty(m_converter.get(),
kAudioCodecPropertyBitRateControlMode,
&size, &result));
return result;
}
UInt32 AudioConverterX::getSoundQualityForVBR()
{
UInt32 result;
UInt32 size = sizeof result;
CHECKCA(AudioConverterGetProperty(m_converter.get(),
kAudioCodecPropertySoundQualityForVBR,
&size, &result));
return result;
}
UInt32 AudioConverterX::getPrimeMethod()
{
UInt32 value;
UInt32 size = sizeof value;
CHECKCA(AudioConverterGetProperty(m_converter.get(),
kAudioConverterPrimeMethod,
&size, &value));
return value;
}
AudioStreamBasicDescription AudioConverterX::getOutputStreamDescription()
{
AudioStreamBasicDescription result;
UInt32 size = sizeof(result);
CHECKCA(AudioConverterGetProperty(m_converter.get(),
kAudioConverterCurrentOutputStreamDescription,
&size, &result));
return result;
}
AudioConverterPrimeInfo AudioConverterX::getPrimeInfo()
{
AudioConverterPrimeInfo result;
UInt32 size = sizeof(AudioConverterPrimeInfo);
CHECKCA(AudioConverterGetProperty(m_converter.get(),
kAudioConverterPrimeInfo,
&size, &result));
return result;
}
std::vector<AudioValueRange> AudioConverterX::getApplicableEncodeBitRates()
{
UInt32 size;
Boolean writable;
CHECKCA(AudioConverterGetPropertyInfo(m_converter.get(),
kAudioConverterApplicableEncodeBitRates, &size, &writable));
std::vector<AudioValueRange> vec(size / sizeof(AudioValueRange));
CHECKCA(AudioConverterGetProperty(m_converter.get(),
kAudioConverterApplicableEncodeBitRates, &size, vec.data()));
return vec;
}
std::vector<uint8_t> AudioConverterX::getCompressionMagicCookie()
{
UInt32 size;
Boolean writable;
CHECKCA(AudioConverterGetPropertyInfo(m_converter.get(),
kAudioConverterCompressionMagicCookie, &size, &writable));
std::vector<uint8_t> vec(size / sizeof(uint8_t));
CHECKCA(AudioConverterGetProperty(m_converter.get(),
kAudioConverterCompressionMagicCookie, &size, vec.data()));
return vec;
}
std::shared_ptr<AudioChannelLayout> AudioConverterX::getOutputChannelLayout()
{
UInt32 size;
Boolean writable;
CHECKCA(AudioConverterGetPropertyInfo(m_converter.get(),
kAudioConverterOutputChannelLayout, &size, &writable));
std::shared_ptr<AudioChannelLayout> acl(
static_cast<AudioChannelLayout*>(std::malloc(size)),
std::free);
CHECKCA(AudioConverterGetProperty(m_converter.get(),
kAudioConverterOutputChannelLayout, &size, acl.get()));
return acl;
}
static void ffat_update_ctx(AVCodecContext *avctx)
{
ATDecodeContext *at = avctx->priv_data;
UInt32 size = sizeof(unsigned);
AudioConverterPrimeInfo prime_info;
AudioStreamBasicDescription out_format;
AudioConverterGetProperty(at->converter,
kAudioConverterPropertyMaximumOutputPacketSize,
&size, &at->pkt_size);
if (at->pkt_size <= 0)
at->pkt_size = 1024 * 50;
size = sizeof(prime_info);
if (!AudioConverterGetProperty(at->converter,
kAudioConverterPrimeInfo,
&size, &prime_info)) {
avctx->initial_padding = prime_info.leadingFrames;
}
size = sizeof(out_format);
if (!AudioConverterGetProperty(at->converter,
kAudioConverterCurrentOutputStreamDescription,
&size, &out_format)) {
if (out_format.mFramesPerPacket)
avctx->frame_size = out_format.mFramesPerPacket;
if (out_format.mBytesPerPacket && avctx->codec_id == AV_CODEC_ID_ILBC)
avctx->block_align = out_format.mBytesPerPacket;
}
at->frame_size = avctx->frame_size;
if (avctx->codec_id == AV_CODEC_ID_PCM_MULAW ||
avctx->codec_id == AV_CODEC_ID_PCM_ALAW) {
at->pkt_size *= 1024;
avctx->frame_size *= 1024;
}
}
// Sets the packet table containing information about the number of valid frames in a file and where they begin and end
// for the file types that support this information.
// Calling this function makes sure we write out the priming and remainder details to the destination file
static void WritePacketTableInfo(AudioConverterRef converter, AudioFileID destinationFileID)
{
UInt32 isWritable;
UInt32 dataSize;
OSStatus error = AudioFileGetPropertyInfo(destinationFileID, kAudioFilePropertyPacketTableInfo, &dataSize, &isWritable);
if (noErr == error && isWritable) {
AudioConverterPrimeInfo primeInfo;
dataSize = sizeof(primeInfo);
// retrieve the leadingFrames and trailingFrames information from the converter,
error = AudioConverterGetProperty(converter, kAudioConverterPrimeInfo, &dataSize, &primeInfo);
if (noErr == error) {
// we have some priming information to write out to the destination file
/* The total number of packets in the file times the frames per packet (or counting each packet's
frames individually for a variable frames per packet format) minus mPrimingFrames, minus
mRemainderFrames, should equal mNumberValidFrames.
*/
AudioFilePacketTableInfo pti;
dataSize = sizeof(pti);
error = AudioFileGetProperty(destinationFileID, kAudioFilePropertyPacketTableInfo, &dataSize, &pti);
if (noErr == error) {
// there's priming to write out to the file
UInt64 totalFrames = pti.mNumberValidFrames + pti.mPrimingFrames + pti.mRemainderFrames; // get the total number of frames from the output file
printf("Total number of frames from output file: %lld\n", totalFrames);
pti.mPrimingFrames = primeInfo.leadingFrames;
pti.mRemainderFrames = primeInfo.trailingFrames;
pti.mNumberValidFrames = totalFrames - pti.mPrimingFrames - pti.mRemainderFrames;
error = AudioFileSetProperty(destinationFileID, kAudioFilePropertyPacketTableInfo, sizeof(pti), &pti);
if (noErr == error) {
printf("Writing packet table information to destination file: %ld\n", sizeof(pti));
printf(" Total valid frames: %lld\n", pti.mNumberValidFrames);
printf(" Priming frames: %ld\n", pti.mPrimingFrames);
printf(" Remainder frames: %ld\n", pti.mRemainderFrames);
} else {
printf("Some audio files can't contain packet table information and that's OK\n");
}
} else {
printf("Getting kAudioFilePropertyPacketTableInfo error: %ld\n", error);
}
} else {
printf("No kAudioConverterPrimeInfo available and that's OK\n");
}
} else {
printf("GetPropertyInfo for kAudioFilePropertyPacketTableInfo error: %ld, isWritable: %ld\n", error, isWritable);
}
}
// Write output channel layout to destination file
static void WriteDestinationChannelLayout(AudioConverterRef converter, AudioFileID sourceFileID, AudioFileID destinationFileID)
{
UInt32 layoutSize = 0;
bool layoutFromConverter = true;
OSStatus error = AudioConverterGetPropertyInfo(converter, kAudioConverterOutputChannelLayout, &layoutSize, NULL);
// if the Audio Converter doesn't have a layout see if the input file does
if (error || 0 == layoutSize) {
error = AudioFileGetPropertyInfo(sourceFileID, kAudioFilePropertyChannelLayout, &layoutSize, NULL);
layoutFromConverter = false;
}
if (noErr == error && 0 != layoutSize) {
char* layout = new char[layoutSize];
if (layoutFromConverter) {
error = AudioConverterGetProperty(converter, kAudioConverterOutputChannelLayout, &layoutSize, layout);
if (error) printf("Could not Get kAudioConverterOutputChannelLayout from Audio Converter!\n");
} else {
error = AudioFileGetProperty(sourceFileID, kAudioFilePropertyChannelLayout, &layoutSize, layout);
if (error) printf("Could not Get kAudioFilePropertyChannelLayout from source file!\n");
}
if (noErr == error) {
error = AudioFileSetProperty(destinationFileID, kAudioFilePropertyChannelLayout, layoutSize, layout);
if (noErr == error) {
printf("Writing channel layout to destination file: %ld\n", layoutSize);
} else {
printf("Even though some formats have layouts, some files don't take them and that's OK\n");
}
}
delete [] layout;
}
}
/***********************************************************************
* hb_work_encCoreAudio_init
***********************************************************************
*
**********************************************************************/
int encCoreAudioInit( hb_work_object_t * w, hb_job_t * job, enum AAC_MODE mode )
{
hb_work_private_t * pv = calloc( 1, sizeof( hb_work_private_t ) );
hb_audio_t * audio = w->audio;
AudioStreamBasicDescription input, output;
UInt32 tmp, tmpsiz = sizeof( tmp );
OSStatus err;
w->private_data = pv;
pv->job = job;
// pass the number of channels used into the private work data
pv->nchannels = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT( audio->config.out.mixdown );
bzero( &input, sizeof( AudioStreamBasicDescription ) );
input.mSampleRate = ( Float64 ) audio->config.out.samplerate;
input.mFormatID = kAudioFormatLinearPCM;
input.mFormatFlags = kLinearPCMFormatFlagIsFloat | kAudioFormatFlagsNativeEndian;
input.mBytesPerPacket = 4 * pv->nchannels;
input.mFramesPerPacket = 1;
input.mBytesPerFrame = input.mBytesPerPacket * input.mFramesPerPacket;
input.mChannelsPerFrame = pv->nchannels;
input.mBitsPerChannel = 32;
bzero( &output, sizeof( AudioStreamBasicDescription ) );
switch ( mode )
{
case AAC_MODE_HE:
output.mFormatID = kAudioFormatMPEG4AAC_HE;
break;
case AAC_MODE_LC:
default:
output.mFormatID = kAudioFormatMPEG4AAC;
break;
}
output.mSampleRate = ( Float64 ) audio->config.out.samplerate;
output.mChannelsPerFrame = pv->nchannels;
// let CoreAudio decide the rest...
// initialise encoder
err = AudioConverterNew( &input, &output, &pv->converter );
if( err != noErr)
{
// Retry without the samplerate
bzero( &output, sizeof( AudioStreamBasicDescription ) );
switch ( mode )
{
case AAC_MODE_HE:
output.mFormatID = kAudioFormatMPEG4AAC_HE;
break;
case AAC_MODE_LC:
default:
output.mFormatID = kAudioFormatMPEG4AAC;
break;
}
output.mChannelsPerFrame = pv->nchannels;
err = AudioConverterNew( &input, &output, &pv->converter );
if( err != noErr)
{
hb_log( "Error creating an AudioConverter err=%"PRId64" %"PRIu64, (int64_t)err, (uint64_t)output.mBytesPerFrame );
*job->die = 1;
return 0;
}
}
if( ( audio->config.out.mixdown == HB_AMIXDOWN_6CH ) && ( audio->config.in.codec == HB_ACODEC_AC3) )
{
SInt32 channelMap[6] = { 2, 1, 3, 4, 5, 0 };
AudioConverterSetProperty( pv->converter, kAudioConverterChannelMap,
sizeof( channelMap ), channelMap );
}
// set encoder quality to maximum
tmp = kAudioConverterQuality_Max;
AudioConverterSetProperty( pv->converter, kAudioConverterCodecQuality,
sizeof( tmp ), &tmp );
// set encoder bitrate control mode to constrained variable
tmp = kAudioCodecBitRateControlMode_VariableConstrained;
AudioConverterSetProperty( pv->converter, kAudioCodecPropertyBitRateControlMode,
sizeof( tmp ), &tmp );
// get available bitrates
AudioValueRange *bitrates;
ssize_t bitrateCounts;
err = AudioConverterGetPropertyInfo( pv->converter, kAudioConverterApplicableEncodeBitRates,
&tmpsiz, NULL);
bitrates = malloc( tmpsiz );
err = AudioConverterGetProperty( pv->converter, kAudioConverterApplicableEncodeBitRates,
&tmpsiz, bitrates);
bitrateCounts = tmpsiz / sizeof( AudioValueRange );
// set bitrate
tmp = audio->config.out.bitrate * 1000;
if( tmp < bitrates[0].mMinimum )
tmp = bitrates[0].mMinimum;
if( tmp > bitrates[bitrateCounts-1].mMinimum )
tmp = bitrates[bitrateCounts-1].mMinimum;
free( bitrates );
if( tmp != audio->config.out.bitrate * 1000 )
hb_log( "encca_aac: sanitizing track %d audio bitrate %d to %"PRIu32"",
audio->config.out.track, audio->config.out.bitrate, tmp/1000 );
AudioConverterSetProperty( pv->converter, kAudioConverterEncodeBitRate,
sizeof( tmp ), &tmp );
// get real input
tmpsiz = sizeof( input );
AudioConverterGetProperty( pv->converter,
kAudioConverterCurrentInputStreamDescription,
&tmpsiz, &input );
// get real output
tmpsiz = sizeof( output );
AudioConverterGetProperty( pv->converter,
kAudioConverterCurrentOutputStreamDescription,
&tmpsiz, &output );
// set sizes
pv->isamplesiz = input.mBytesPerPacket;
pv->isamples = output.mFramesPerPacket;
pv->osamplerate = output.mSampleRate;
// get maximum output size
AudioConverterGetProperty( pv->converter,
kAudioConverterPropertyMaximumOutputPacketSize,
&tmpsiz, &tmp );
pv->omaxpacket = tmp;
// get magic cookie (elementary stream descriptor)
tmp = HB_CONFIG_MAX_SIZE;
AudioConverterGetProperty( pv->converter,
kAudioConverterCompressionMagicCookie,
&tmp, w->config->aac.bytes );
// CoreAudio returns a complete ESDS, but we only need
// the DecoderSpecific info.
UInt8* buffer = NULL;
ReadESDSDescExt(w->config->aac.bytes, &buffer, &tmpsiz, 0);
w->config->aac.length = tmpsiz;
memmove( w->config->aac.bytes, buffer,
w->config->aac.length );
pv->list = hb_list_init();
pv->buf = NULL;
return 0;
}
OSStatus DoConvertFile(CFURLRef sourceURL, CFURLRef destinationURL, OSType outputFormat, Float64 outputSampleRate)
{
ExtAudioFileRef sourceFile = 0;
ExtAudioFileRef destinationFile = 0;
Boolean canResumeFromInterruption = true; // we can continue unless told otherwise
OSStatus error = noErr;
// in this sample we should never be on the main thread here
assert(![NSThread isMainThread]);
// transition thread state to kStateRunning before continuing
ThreadStateSetRunning();
printf("DoConvertFile\n");
try {
CAStreamBasicDescription srcFormat, dstFormat;
// open the source file
XThrowIfError(ExtAudioFileOpenURL(sourceURL, &sourceFile), "ExtAudioFileOpenURL failed");
// get the source data format
UInt32 size = sizeof(srcFormat);
XThrowIfError(ExtAudioFileGetProperty(sourceFile, kExtAudioFileProperty_FileDataFormat, &size, &srcFormat), "couldn't get source data format");
printf("\nSource file format: "); srcFormat.Print();
// setup the output file format
dstFormat.mSampleRate = (outputSampleRate == 0 ? srcFormat.mSampleRate : outputSampleRate); // set sample rate
if (outputFormat == kAudioFormatLinearPCM) {
// if PCM was selected as the destination format, create a 16-bit int PCM file format description
dstFormat.mFormatID = outputFormat;
dstFormat.mChannelsPerFrame = srcFormat.NumberChannels();
dstFormat.mBitsPerChannel = 16;
dstFormat.mBytesPerPacket = dstFormat.mBytesPerFrame = 2 * dstFormat.mChannelsPerFrame;
dstFormat.mFramesPerPacket = 1;
dstFormat.mFormatFlags = kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsSignedInteger; // little-endian
} else {
// compressed format - need to set at least format, sample rate and channel fields for kAudioFormatProperty_FormatInfo
dstFormat.mFormatID = outputFormat;
dstFormat.mChannelsPerFrame = (outputFormat == kAudioFormatiLBC ? 1 : srcFormat.NumberChannels()); // for iLBC num channels must be 1
// use AudioFormat API to fill out the rest of the description
size = sizeof(dstFormat);
XThrowIfError(AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &size, &dstFormat), "couldn't create destination data format");
}
printf("\nDestination file format: "); dstFormat.Print();
// create the destination file
XThrowIfError(ExtAudioFileCreateWithURL(destinationURL, kAudioFileCAFType, &dstFormat, NULL, kAudioFileFlags_EraseFile, &destinationFile), "ExtAudioFileCreateWithURL failed!");
// set the client format - The format must be linear PCM (kAudioFormatLinearPCM)
// You must set this in order to encode or decode a non-PCM file data format
// You may set this on PCM files to specify the data format used in your calls to read/write
CAStreamBasicDescription clientFormat;
if (outputFormat == kAudioFormatLinearPCM) {
clientFormat = dstFormat;
} else {
clientFormat.SetCanonical(srcFormat.NumberChannels(), true);
clientFormat.mSampleRate = srcFormat.mSampleRate;
}
printf("\nClient data format: "); clientFormat.Print();
printf("\n");
size = sizeof(clientFormat);
XThrowIfError(ExtAudioFileSetProperty(sourceFile, kExtAudioFileProperty_ClientDataFormat, size, &clientFormat), "couldn't set source client format");
size = sizeof(clientFormat);
XThrowIfError(ExtAudioFileSetProperty(destinationFile, kExtAudioFileProperty_ClientDataFormat, size, &clientFormat), "couldn't set destination client format");
// can the audio converter (which in this case is owned by an ExtAudioFile object) resume conversion after an interruption?
AudioConverterRef audioConverter;
size = sizeof(audioConverter);
XThrowIfError(ExtAudioFileGetProperty(destinationFile, kExtAudioFileProperty_AudioConverter, &size, &audioConverter), "Couldn't get Audio Converter!");
// this property may be queried at any time after construction of the audio converter (which in this case is owned by an ExtAudioFile object)
// after setting the output format -- there's no clear reason to prefer construction time, interruption time, or potential resumption time but we prefer
// construction time since it means less code to execute during or after interruption time
UInt32 canResume = 0;
size = sizeof(canResume);
error = AudioConverterGetProperty(audioConverter, kAudioConverterPropertyCanResumeFromInterruption, &size, &canResume);
if (noErr == error) {
// we recieved a valid return value from the GetProperty call
// if the property's value is 1, then the codec CAN resume work following an interruption
// if the property's value is 0, then interruptions destroy the codec's state and we're done
if (0 == canResume) canResumeFromInterruption = false;
printf("Audio Converter %s continue after interruption!\n", (canResumeFromInterruption == 0 ? "CANNOT" : "CAN"));
} else {
// if the property is unimplemented (kAudioConverterErr_PropertyNotSupported, or paramErr returned in the case of PCM),
// then the codec being used is not a hardware codec so we're not concerned about codec state
// we are always going to be able to resume conversion after an interruption
if (kAudioConverterErr_PropertyNotSupported == error) {
printf("kAudioConverterPropertyCanResumeFromInterruption property not supported!\n");
} else {
printf("AudioConverterGetProperty kAudioConverterPropertyCanResumeFromInterruption result %ld\n", error);
}
error = noErr;
}
// set up buffers
UInt32 bufferByteSize = 32768;
char srcBuffer[bufferByteSize];
// keep track of the source file offset so we know where to reset the source for
// reading if interrupted and input was not consumed by the audio converter
SInt64 sourceFrameOffset = 0;
//***** do the read and write - the conversion is done on and by the write call *****//
printf("Converting...\n");
while (1) {
AudioBufferList fillBufList;
fillBufList.mNumberBuffers = 1;
fillBufList.mBuffers[0].mNumberChannels = clientFormat.NumberChannels();
fillBufList.mBuffers[0].mDataByteSize = bufferByteSize;
fillBufList.mBuffers[0].mData = srcBuffer;
// client format is always linear PCM - so here we determine how many frames of lpcm
// we can read/write given our buffer size
UInt32 numFrames;
if (clientFormat.mBytesPerFrame > 0) // rids bogus analyzer div by zero warning mBytesPerFrame can't be 0 and is protected by an Assert
numFrames = clientFormat.BytesToFrames(bufferByteSize); // (bufferByteSize / clientFormat.mBytesPerFrame);
XThrowIfError(ExtAudioFileRead(sourceFile, &numFrames, &fillBufList), "ExtAudioFileRead failed!");
if (!numFrames) {
// this is our termination condition
error = noErr;
break;
}
sourceFrameOffset += numFrames;
// this will block if we're interrupted
Boolean wasInterrupted = ThreadStatePausedCheck();
if ((error || wasInterrupted) && (false == canResumeFromInterruption)) {
// this is our interruption termination condition
// an interruption has occured but the audio converter cannot continue
error = kMyAudioConverterErr_CannotResumeFromInterruptionError;
break;
}
error = ExtAudioFileWrite(destinationFile, numFrames, &fillBufList);
// if interrupted in the process of the write call, we must handle the errors appropriately
if (error) {
if (kExtAudioFileError_CodecUnavailableInputConsumed == error) {
printf("ExtAudioFileWrite kExtAudioFileError_CodecUnavailableInputConsumed error %ld\n", error);
/*
Returned when ExtAudioFileWrite was interrupted. You must stop calling
ExtAudioFileWrite. If the underlying audio converter can resume after an
interruption (see kAudioConverterPropertyCanResumeFromInterruption), you must
wait for an EndInterruption notification from AudioSession, then activate the session
before resuming. In this situation, the buffer you provided to ExtAudioFileWrite was successfully
consumed and you may proceed to the next buffer
*/
} else if (kExtAudioFileError_CodecUnavailableInputNotConsumed == error) {
printf("ExtAudioFileWrite kExtAudioFileError_CodecUnavailableInputNotConsumed error %ld\n", error);
/*
Returned when ExtAudioFileWrite was interrupted. You must stop calling
ExtAudioFileWrite. If the underlying audio converter can resume after an
interruption (see kAudioConverterPropertyCanResumeFromInterruption), you must
wait for an EndInterruption notification from AudioSession, then activate the session
before resuming. In this situation, the buffer you provided to ExtAudioFileWrite was not
successfully consumed and you must try to write it again
*/
// seek back to last offset before last read so we can try again after the interruption
sourceFrameOffset -= numFrames;
XThrowIfError(ExtAudioFileSeek(sourceFile, sourceFrameOffset), "ExtAudioFileSeek failed!");
} else {
XThrowIfError(error, "ExtAudioFileWrite error!");
}
} // if
} // while
}
catch (CAXException e) {
char buf[256];
fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf));
error = e.mError;
}
// close
if (destinationFile) ExtAudioFileDispose(destinationFile);
if (sourceFile) ExtAudioFileDispose(sourceFile);
// transition thread state to kStateDone before continuing
ThreadStateSetDone();
return error;
}
OSStatus DoConvertFile(CFURLRef sourceURL, CFURLRef destinationURL, OSType outputFormat, Float64 outputSampleRate, UInt32 outputBitRate)
{
AudioFileID sourceFileID = 0;
AudioFileID destinationFileID = 0;
AudioConverterRef converter = NULL;
Boolean canResumeFromInterruption = true; // we can continue unless told otherwise
CAStreamBasicDescription srcFormat, dstFormat;
AudioFileIO afio = {};
char *outputBuffer = NULL;
AudioStreamPacketDescription *outputPacketDescriptions = NULL;
OSStatus error = noErr;
// in this sample we should never be on the main thread here
assert(![NSThread isMainThread]);
// transition thread state to kStateRunning before continuing
printf("\nDoConvertFile\n");
try {
// get the source file
XThrowIfError(AudioFileOpenURL(sourceURL, kAudioFileReadPermission, 0, &sourceFileID), "AudioFileOpenURL failed");
// get the source data format
UInt32 size = sizeof(srcFormat);
XThrowIfError(AudioFileGetProperty(sourceFileID, kAudioFilePropertyDataFormat, &size, &srcFormat), "couldn't get source data format");
// setup the output file format
dstFormat.mSampleRate = (outputSampleRate == 0 ? srcFormat.mSampleRate : outputSampleRate); // set sample rate
if (outputFormat == kAudioFormatLinearPCM) {
// if the output format is PC create a 16-bit int PCM file format description as an example
dstFormat.mFormatID = outputFormat;
dstFormat.mChannelsPerFrame = srcFormat.NumberChannels();
dstFormat.mBitsPerChannel = 16;
dstFormat.mBytesPerPacket = dstFormat.mBytesPerFrame = 2 * dstFormat.mChannelsPerFrame;
dstFormat.mFramesPerPacket = 1;
dstFormat.mFormatFlags = kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsSignedInteger; // little-endian
} else {
// compressed format - need to set at least format, sample rate and channel fields for kAudioFormatProperty_FormatInfo
dstFormat.mFormatID = outputFormat;
dstFormat.mChannelsPerFrame = (outputFormat == kAudioFormatiLBC ? 1 : srcFormat.NumberChannels()); // for iLBC num channels must be 1
// use AudioFormat API to fill out the rest of the description
size = sizeof(dstFormat);
XThrowIfError(AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &size, &dstFormat), "couldn't create destination data format");
}
printf("Source File format: "); srcFormat.Print();
printf("Destination format: "); dstFormat.Print();
// create the AudioConverter
XThrowIfError(AudioConverterNew(&srcFormat, &dstFormat, &converter), "AudioConverterNew failed!");
// if the source has a cookie, get it and set it on the Audio Converter
ReadCookie(sourceFileID, converter);
// get the actual formats back from the Audio Converter
size = sizeof(srcFormat);
XThrowIfError(AudioConverterGetProperty(converter, kAudioConverterCurrentInputStreamDescription, &size, &srcFormat), "AudioConverterGetProperty kAudioConverterCurrentInputStreamDescription failed!");
size = sizeof(dstFormat);
XThrowIfError(AudioConverterGetProperty(converter, kAudioConverterCurrentOutputStreamDescription, &size, &dstFormat), "AudioConverterGetProperty kAudioConverterCurrentOutputStreamDescription failed!");
printf("Formats returned from AudioConverter:\n");
printf(" Source format: "); srcFormat.Print();
printf(" Destination File format: "); dstFormat.Print();
// if encoding to AAC set the bitrate
// kAudioConverterEncodeBitRate is a UInt32 value containing the number of bits per second to aim for when encoding data
// when you explicitly set the bit rate and the sample rate, this tells the encoder to stick with both bit rate and sample rate
// but there are combinations (also depending on the number of channels) which will not be allowed
// if you do not explicitly set a bit rate the encoder will pick the correct value for you depending on samplerate and number of channels
// bit rate also scales with the number of channels, therefore one bit rate per sample rate can be used for mono cases
// and if you have stereo or more, you can multiply that number by the number of channels.
if (outputBitRate == 0) {
outputBitRate = 192000; // 192kbs
}
if (dstFormat.mFormatID == kAudioFormatMPEG4AAC) {
UInt32 propSize = sizeof(outputBitRate);
// set the bit rate depending on the samplerate chosen
XThrowIfError(AudioConverterSetProperty(converter, kAudioConverterEncodeBitRate, propSize, &outputBitRate),
"AudioConverterSetProperty kAudioConverterEncodeBitRate failed!");
// get it back and print it out
AudioConverterGetProperty(converter, kAudioConverterEncodeBitRate, &propSize, &outputBitRate);
printf ("AAC Encode Bitrate: %u\n", (unsigned int)outputBitRate);
}
// can the Audio Converter resume conversion after an interruption?
// this property may be queried at any time after construction of the Audio Converter after setting its output format
// there's no clear reason to prefer construction time, interruption time, or potential resumption time but we prefer
// construction time since it means less code to execute during or after interruption time
UInt32 canResume = 0;
size = sizeof(canResume);
error = AudioConverterGetProperty(converter, kAudioConverterPropertyCanResumeFromInterruption, &size, &canResume);
if (noErr == error) {
// we recieved a valid return value from the GetProperty call
// if the property's value is 1, then the codec CAN resume work following an interruption
// if the property's value is 0, then interruptions destroy the codec's state and we're done
if (0 == canResume) canResumeFromInterruption = false;
printf("Audio Converter %s continue after interruption!\n", (canResumeFromInterruption == 0 ? "CANNOT" : "CAN"));
} else {
// if the property is unimplemented (kAudioConverterErr_PropertyNotSupported, or paramErr returned in the case of PCM),
// then the codec being used is not a hardware codec so we're not concerned about codec state
// we are always going to be able to resume conversion after an interruption
if (kAudioConverterErr_PropertyNotSupported == error) {
printf("kAudioConverterPropertyCanResumeFromInterruption property not supported - see comments in source for more info.\n");
} else {
printf("AudioConverterGetProperty kAudioConverterPropertyCanResumeFromInterruption result %ld, paramErr is OK if PCM\n", error);
}
error = noErr;
}
// create the destination file
XThrowIfError(AudioFileCreateWithURL(destinationURL, kAudioFileCAFType, &dstFormat, kAudioFileFlags_EraseFile, &destinationFileID), "AudioFileCreateWithURL failed!");
// set up source buffers and data proc info struct
afio.srcFileID = sourceFileID;
afio.srcBufferSize = 32768;
afio.srcBuffer = new char [afio.srcBufferSize];
afio.srcFilePos = 0;
afio.srcFormat = srcFormat;
if (srcFormat.mBytesPerPacket == 0) {
// if the source format is VBR, we need to get the maximum packet size
// use kAudioFilePropertyPacketSizeUpperBound which returns the theoretical maximum packet size
// in the file (without actually scanning the whole file to find the largest packet,
// as may happen with kAudioFilePropertyMaximumPacketSize)
size = sizeof(afio.srcSizePerPacket);
XThrowIfError(AudioFileGetProperty(sourceFileID, kAudioFilePropertyPacketSizeUpperBound, &size, &afio.srcSizePerPacket), "AudioFileGetProperty kAudioFilePropertyPacketSizeUpperBound failed!");
// how many packets can we read for our buffer size?
afio.numPacketsPerRead = afio.srcBufferSize / afio.srcSizePerPacket;
// allocate memory for the PacketDescription structures describing the layout of each packet
afio.packetDescriptions = new AudioStreamPacketDescription [afio.numPacketsPerRead];
} else {
// CBR source format
afio.srcSizePerPacket = srcFormat.mBytesPerPacket;
afio.numPacketsPerRead = afio.srcBufferSize / afio.srcSizePerPacket;
afio.packetDescriptions = NULL;
}
// set up output buffers
UInt32 outputSizePerPacket = dstFormat.mBytesPerPacket; // this will be non-zero if the format is CBR
UInt32 theOutputBufSize = 32768;
outputBuffer = new char[theOutputBufSize];
if (outputSizePerPacket == 0) {
// if the destination format is VBR, we need to get max size per packet from the converter
size = sizeof(outputSizePerPacket);
XThrowIfError(AudioConverterGetProperty(converter, kAudioConverterPropertyMaximumOutputPacketSize, &size, &outputSizePerPacket), "AudioConverterGetProperty kAudioConverterPropertyMaximumOutputPacketSize failed!");
// allocate memory for the PacketDescription structures describing the layout of each packet
outputPacketDescriptions = new AudioStreamPacketDescription [theOutputBufSize / outputSizePerPacket];
}
UInt32 numOutputPackets = theOutputBufSize / outputSizePerPacket;
// if the destination format has a cookie, get it and set it on the output file
WriteCookie(converter, destinationFileID);
// write destination channel layout
if (srcFormat.mChannelsPerFrame > 2) {
WriteDestinationChannelLayout(converter, sourceFileID, destinationFileID);
}
UInt64 totalOutputFrames = 0; // used for debgging printf
SInt64 outputFilePos = 0;
// loop to convert data
printf("Converting...\n");
while (1) {
// set up output buffer list
AudioBufferList fillBufList;
fillBufList.mNumberBuffers = 1;
fillBufList.mBuffers[0].mNumberChannels = dstFormat.mChannelsPerFrame;
fillBufList.mBuffers[0].mDataByteSize = theOutputBufSize;
fillBufList.mBuffers[0].mData = outputBuffer;
// this will block if we're interrupted
Boolean wasInterrupted = NO;
if ((error || wasInterrupted) && (false == canResumeFromInterruption)) {
// this is our interruption termination condition
// an interruption has occured but the Audio Converter cannot continue
error = kMyAudioConverterErr_CannotResumeFromInterruptionError;
break;
}
// convert data
UInt32 ioOutputDataPackets = numOutputPackets;
printf("AudioConverterFillComplexBuffer...\n");
error = AudioConverterFillComplexBuffer(converter, EncoderDataProc, &afio, &ioOutputDataPackets, &fillBufList, outputPacketDescriptions);
// if interrupted in the process of the conversion call, we must handle the error appropriately
if (error) {
if (kAudioConverterErr_HardwareInUse == error) {
printf("Audio Converter returned kAudioConverterErr_HardwareInUse!\n");
} else {
XThrowIfError(error, "AudioConverterFillComplexBuffer error!");
}
} else {
if (ioOutputDataPackets == 0) {
// this is the EOF conditon
error = noErr;
break;
}
}
if (noErr == error) {
// write to output file
UInt32 inNumBytes = fillBufList.mBuffers[0].mDataByteSize;
XThrowIfError(AudioFileWritePackets(destinationFileID, false, inNumBytes, outputPacketDescriptions, outputFilePos, &ioOutputDataPackets, outputBuffer), "AudioFileWritePackets failed!");
printf("Convert Output: Write %lu packets at position %lld, size: %ld\n", ioOutputDataPackets, outputFilePos, inNumBytes);
// advance output file packet position
outputFilePos += ioOutputDataPackets;
if (dstFormat.mFramesPerPacket) {
// the format has constant frames per packet
totalOutputFrames += (ioOutputDataPackets * dstFormat.mFramesPerPacket);
} else if (outputPacketDescriptions != NULL) {
// variable frames per packet require doing this for each packet (adding up the number of sample frames of data in each packet)
for (UInt32 i = 0; i < ioOutputDataPackets; ++i)
totalOutputFrames += outputPacketDescriptions[i].mVariableFramesInPacket;
}
}
} // while
if (noErr == error) {
// write out any of the leading and trailing frames for compressed formats only
if (dstFormat.mBitsPerChannel == 0) {
// our output frame count should jive with
printf("Total number of output frames counted: %lld\n", totalOutputFrames);
WritePacketTableInfo(converter, destinationFileID);
}
// write the cookie again - sometimes codecs will update cookies at the end of a conversion
WriteCookie(converter, destinationFileID);
}
}
catch (CAXException e) {
char buf[256];
fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf));
error = e.mError;
}
// cleanup
if (converter) AudioConverterDispose(converter);
if (destinationFileID) AudioFileClose(destinationFileID);
if (sourceFileID) AudioFileClose(sourceFileID);
if (afio.srcBuffer) delete [] afio.srcBuffer;
if (afio.packetDescriptions) delete [] afio.packetDescriptions;
if (outputBuffer) delete [] outputBuffer;
if (outputPacketDescriptions) delete [] outputPacketDescriptions;
return error;
}
JNIEXPORT jint JNICALL Java_com_apple_audio_toolbox_AudioConverter_AudioConverterGetProperty
(JNIEnv *, jclass, jint inAudioConverter, jint inPropertyID, jint ioPropertyDataSize, jint outPropertyData)
{
return (jint)AudioConverterGetProperty((AudioConverterRef)inAudioConverter, (AudioConverterPropertyID)inPropertyID, (UInt32 *)ioPropertyDataSize, (void *)outPropertyData);
}
static GVBool gviHardwareInitPlayback(GVIDevice * device)
{
GVIHardwareData * data = (GVIHardwareData *)device->m_data;
UInt32 size;
OSStatus result;
UInt32 primeMethod;
SInt32 channelMap[100];
int i;
// create the array of sources
data->m_playbackSources = gviNewSourceList();
if(!data->m_playbackSources)
return GVFalse;
// get the playback format
size = sizeof(AudioStreamBasicDescription);
result = AudioDeviceGetProperty(device->m_deviceID, 0, false, kAudioDevicePropertyStreamFormat, &size, &data->m_playbackStreamDescriptor);
if(result != noErr)
{
gviFreeSourceList(data->m_playbackSources);
return GVFalse;
}
// create a converter from the GV format to the playback format
result = AudioConverterNew(&GVIVoiceFormat, &data->m_playbackStreamDescriptor, &data->m_playbackConverter);
if(result != noErr)
{
gviFreeSourceList(data->m_playbackSources);
return GVFalse;
}
// set it to do no priming
primeMethod = kConverterPrimeMethod_None;
result = AudioConverterSetProperty(data->m_playbackConverter, kAudioConverterPrimeMethod, sizeof(UInt32), &primeMethod);
if(result != noErr)
{
AudioConverterDispose(data->m_playbackConverter);
gviFreeSourceList(data->m_playbackSources);
return GVFalse;
}
// setup the converter to map the input channel to all output channels
result = AudioConverterGetPropertyInfo(data->m_playbackConverter, kAudioConverterChannelMap, &size, NULL);
if(result == noErr)
{
result = AudioConverterGetProperty(data->m_playbackConverter, kAudioConverterChannelMap, &size, channelMap);
if(result == noErr)
{
for(i = 0 ; i < (size / sizeof(SInt32)) ; i++)
channelMap[i] = 0;
AudioConverterSetProperty(data->m_playbackConverter, kAudioConverterChannelMap, size, channelMap);
}
}
// allocate the playback buffer
data->m_playbackBuffer = (GVSample *)gsimalloc(GVIBytesPerFrame);
if(!data->m_playbackBuffer)
{
AudioConverterDispose(data->m_playbackConverter);
gviFreeSourceList(data->m_playbackSources);
return GVFalse;
}
// add property listener
AudioDeviceAddPropertyListener(device->m_deviceID, 0, false, kAudioDevicePropertyDeviceIsAlive, gviPropertyListener, device);
#if GVI_VOLUME_IN_SOFTWARE
// init volume
data->m_playbackVolume = (GVScalar)1.0;
#endif
return GVTrue;
}
AACEncode::AACEncode(int frequencyInHz, int channelCount, int bitrate)
: m_sentConfig(false), m_bitrate(bitrate)
{
OSStatus result = 0;
AudioStreamBasicDescription in = {0}, out = {0};
// passing anything except 48000, 44100, and 22050 for mSampleRate results in "!dat"
// OSStatus when querying for kAudioConverterPropertyMaximumOutputPacketSize property
// below
in.mSampleRate = frequencyInHz;
// passing anything except 2 for mChannelsPerFrame results in "!dat" OSStatus when
// querying for kAudioConverterPropertyMaximumOutputPacketSize property below
in.mChannelsPerFrame = channelCount;
in.mBitsPerChannel = 16;
in.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
in.mFormatID = kAudioFormatLinearPCM;
in.mFramesPerPacket = 1;
in.mBytesPerFrame = in.mBitsPerChannel * in.mChannelsPerFrame / 8;
in.mBytesPerPacket = in.mFramesPerPacket*in.mBytesPerFrame;
m_in = in;
out.mFormatID = kAudioFormatMPEG4AAC;
out.mFormatFlags = 0;
out.mFramesPerPacket = kSamplesPerFrame;
out.mSampleRate = frequencyInHz;
out.mChannelsPerFrame = channelCount;
m_out = out;
UInt32 outputBitrate = bitrate;
UInt32 propSize = sizeof(outputBitrate);
UInt32 outputPacketSize = 0;
const OSType subtype = kAudioFormatMPEG4AAC;
AudioClassDescription requestedCodecs[2] = {
{
kAudioEncoderComponentType,
subtype,
kAppleSoftwareAudioCodecManufacturer
},
{
kAudioEncoderComponentType,
subtype,
kAppleHardwareAudioCodecManufacturer
}
};
result = AudioConverterNewSpecific(&in, &out, 2, requestedCodecs, &m_audioConverter);
if(result == noErr) {
result = AudioConverterSetProperty(m_audioConverter, kAudioConverterEncodeBitRate, propSize, &outputBitrate);
}
if(result == noErr) {
result = AudioConverterGetProperty(m_audioConverter, kAudioConverterPropertyMaximumOutputPacketSize, &propSize, &outputPacketSize);
}
if(result == noErr) {
m_outputPacketMaxSize = outputPacketSize;
m_bytesPerSample = 2 * channelCount;
uint8_t sampleRateIndex = 0;
switch(frequencyInHz) {
case 96000:
sampleRateIndex = 0;
break;
case 88200:
sampleRateIndex = 1;
break;
case 64000:
sampleRateIndex = 2;
break;
case 48000:
sampleRateIndex = 3;
break;
case 44100:
sampleRateIndex = 4;
break;
case 32000:
sampleRateIndex = 5;
break;
case 24000:
sampleRateIndex = 6;
break;
case 22050:
sampleRateIndex = 7;
break;
case 16000:
sampleRateIndex = 8;
break;
case 12000:
sampleRateIndex = 9;
break;
case 11025:
sampleRateIndex = 10;
break;
case 8000:
sampleRateIndex = 11;
break;
case 7350:
sampleRateIndex = 12;
break;
default:
sampleRateIndex = 15;
}
makeAsc(sampleRateIndex, uint8_t(channelCount));
} else {
DLog("Error setting up audio encoder %x", (int)result);
}
}
// soundsource overrides
Result SoundSourceCoreAudio::tryOpen(const AudioSourceConfig& audioSrcCfg) {
const QString fileName(getLocalFileName());
//Open the audio file.
OSStatus err;
/** This code blocks works with OS X 10.5+ only. DO NOT DELETE IT for now. */
CFStringRef urlStr = CFStringCreateWithCharacters(0,
reinterpret_cast<const UniChar *>(fileName.unicode()),
fileName.size());
CFURLRef urlRef = CFURLCreateWithFileSystemPath(NULL, urlStr,
kCFURLPOSIXPathStyle, false);
err = ExtAudioFileOpenURL(urlRef, &m_audioFile);
CFRelease(urlStr);
CFRelease(urlRef);
/** TODO: Use FSRef for compatibility with 10.4 Tiger.
Note that ExtAudioFileOpen() is deprecated above Tiger, so we must maintain
both code paths if someone finishes this part of the code.
FSRef fsRef;
CFURLGetFSRef(reinterpret_cast<CFURLRef>(url.get()), &fsRef);
err = ExtAudioFileOpen(&fsRef, &m_audioFile);
*/
if (err != noErr) {
qDebug() << "SSCA: Error opening file " << fileName;
return ERR;
}
// get the input file format
UInt32 inputFormatSize = sizeof(m_inputFormat);
err = ExtAudioFileGetProperty(m_audioFile,
kExtAudioFileProperty_FileDataFormat, &inputFormatSize,
&m_inputFormat);
if (err != noErr) {
qDebug() << "SSCA: Error getting file format (" << fileName << ")";
return ERR;
}
m_bFileIsMp3 = m_inputFormat.mFormatID == kAudioFormatMPEGLayer3;
// create the output format
const UInt32 numChannels =
(kChannelCountZero < audioSrcCfg.channelCountHint) ? audioSrcCfg.channelCountHint : 2;
m_outputFormat = CAStreamBasicDescription(m_inputFormat.mSampleRate,
numChannels, CAStreamBasicDescription::kPCMFormatFloat32, true);
// set the client format
err = ExtAudioFileSetProperty(m_audioFile,
kExtAudioFileProperty_ClientDataFormat, sizeof(m_outputFormat),
&m_outputFormat);
if (err != noErr) {
qDebug() << "SSCA: Error setting file property";
return ERR;
}
//get the total length in frames of the audio file - copypasta: http://discussions.apple.com/thread.jspa?threadID=2364583&tstart=47
SInt64 totalFrameCount;
UInt32 totalFrameCountSize = sizeof(totalFrameCount);
err = ExtAudioFileGetProperty(m_audioFile,
kExtAudioFileProperty_FileLengthFrames, &totalFrameCountSize,
&totalFrameCount);
if (err != noErr) {
qDebug() << "SSCA: Error getting number of frames";
return ERR;
}
//
// WORKAROUND for bug in ExtFileAudio
//
AudioConverterRef acRef;
UInt32 acrsize = sizeof(AudioConverterRef);
err = ExtAudioFileGetProperty(m_audioFile,
kExtAudioFileProperty_AudioConverter, &acrsize, &acRef);
//_ThrowExceptionIfErr(@"kExtAudioFileProperty_AudioConverter", err);
AudioConverterPrimeInfo primeInfo;
UInt32 piSize = sizeof(AudioConverterPrimeInfo);
memset(&primeInfo, 0, piSize);
err = AudioConverterGetProperty(acRef, kAudioConverterPrimeInfo, &piSize,
&primeInfo);
if (err != kAudioConverterErr_PropertyNotSupported) { // Only if decompressing
//_ThrowExceptionIfErr(@"kAudioConverterPrimeInfo", err);
m_headerFrames = primeInfo.leadingFrames;
} else {
m_headerFrames = 0;
}
setChannelCount(m_outputFormat.NumberChannels());
setFrameRate(m_inputFormat.mSampleRate);
// NOTE(uklotzde): This is what I found when migrating
// the code from SoundSource (sample-oriented) to the new
// AudioSource (frame-oriented) API. It is not documented
// when m_headerFrames > 0 and what the consequences are.
setFrameCount(totalFrameCount/* - m_headerFrames*/);
//Seek to position 0, which forces us to skip over all the header frames.
//This makes sure we're ready to just let the Analyser rip and it'll
//get the number of samples it expects (ie. no header frames).
seekSampleFrame(0);
return OK;
}
/***********************************************************************
* hb_work_encCoreAudio_init
***********************************************************************
*
**********************************************************************/
int encCoreAudioInit(hb_work_object_t *w, hb_job_t *job, enum AAC_MODE mode)
{
hb_work_private_t *pv = calloc(1, sizeof(hb_work_private_t));
hb_audio_t *audio = w->audio;
AudioStreamBasicDescription input, output;
UInt32 tmp, tmpsiz = sizeof(tmp);
OSStatus err;
w->private_data = pv;
pv->job = job;
// pass the number of channels used into the private work data
pv->nchannels =
hb_mixdown_get_discrete_channel_count(audio->config.out.mixdown);
bzero(&input, sizeof(AudioStreamBasicDescription));
input.mSampleRate = (Float64)audio->config.out.samplerate;
input.mFormatID = kAudioFormatLinearPCM;
input.mFormatFlags = (kLinearPCMFormatFlagIsFloat|kAudioFormatFlagsNativeEndian);
input.mBytesPerPacket = 4 * pv->nchannels;
input.mFramesPerPacket = 1;
input.mBytesPerFrame = input.mBytesPerPacket * input.mFramesPerPacket;
input.mChannelsPerFrame = pv->nchannels;
input.mBitsPerChannel = 32;
bzero(&output, sizeof(AudioStreamBasicDescription));
switch (mode)
{
case AAC_MODE_HE:
output.mFormatID = kAudioFormatMPEG4AAC_HE;
break;
case AAC_MODE_LC:
default:
output.mFormatID = kAudioFormatMPEG4AAC;
break;
}
output.mSampleRate = (Float64)audio->config.out.samplerate;
output.mChannelsPerFrame = pv->nchannels;
// let CoreAudio decide the rest
// initialise encoder
err = AudioConverterNew(&input, &output, &pv->converter);
if (err != noErr)
{
// Retry without the samplerate
bzero(&output, sizeof(AudioStreamBasicDescription));
switch (mode)
{
case AAC_MODE_HE:
output.mFormatID = kAudioFormatMPEG4AAC_HE;
break;
case AAC_MODE_LC:
default:
output.mFormatID = kAudioFormatMPEG4AAC;
break;
}
output.mChannelsPerFrame = pv->nchannels;
err = AudioConverterNew(&input, &output, &pv->converter);
if (err != noErr)
{
hb_log("Error creating an AudioConverter err=%"PRId64" output.mBytesPerFrame=%"PRIu64"",
(int64_t)err, (uint64_t)output.mBytesPerFrame);
*job->done_error = HB_ERROR_UNKNOWN;
*job->die = 1;
return -1;
}
}
// set encoder quality to maximum
tmp = kAudioConverterQuality_Max;
AudioConverterSetProperty(pv->converter, kAudioConverterCodecQuality,
sizeof(tmp), &tmp);
if (audio->config.out.bitrate > 0)
{
// set encoder bitrate control mode to constrained variable
tmp = kAudioCodecBitRateControlMode_VariableConstrained;
AudioConverterSetProperty(pv->converter,
kAudioCodecPropertyBitRateControlMode,
sizeof(tmp), &tmp);
// get available bitrates
AudioValueRange *bitrates;
ssize_t bitrateCounts;
err = AudioConverterGetPropertyInfo(pv->converter,
kAudioConverterApplicableEncodeBitRates,
&tmpsiz, NULL);
bitrates = malloc(tmpsiz);
err = AudioConverterGetProperty(pv->converter,
kAudioConverterApplicableEncodeBitRates,
&tmpsiz, bitrates);
bitrateCounts = tmpsiz / sizeof(AudioValueRange);
// set bitrate
tmp = audio->config.out.bitrate * 1000;
if (tmp < bitrates[0].mMinimum)
tmp = bitrates[0].mMinimum;
if (tmp > bitrates[bitrateCounts-1].mMinimum)
tmp = bitrates[bitrateCounts-1].mMinimum;
free(bitrates);
if (tmp != audio->config.out.bitrate * 1000)
{
hb_log("encCoreAudioInit: sanitizing track %d audio bitrate %d to %"PRIu32"",
audio->config.out.track, audio->config.out.bitrate, tmp / 1000);
}
AudioConverterSetProperty(pv->converter,
kAudioConverterEncodeBitRate,
sizeof(tmp), &tmp);
}
else if (audio->config.out.quality >= 0)
{
if (mode != AAC_MODE_LC)
{
hb_error("encCoreAudioInit: internal error, VBR set but not applicable");
return 1;
}
// set encoder bitrate control mode to variable
tmp = kAudioCodecBitRateControlMode_Variable;
AudioConverterSetProperty(pv->converter,
kAudioCodecPropertyBitRateControlMode,
sizeof(tmp), &tmp);
// set quality
tmp = audio->config.out.quality;
AudioConverterSetProperty(pv->converter,
kAudioCodecPropertySoundQualityForVBR,
sizeof(tmp), &tmp);
}
else
{
hb_error("encCoreAudioInit: internal error, bitrate/quality not set");
return 1;
}
// get real input
tmpsiz = sizeof(input);
AudioConverterGetProperty(pv->converter,
kAudioConverterCurrentInputStreamDescription,
&tmpsiz, &input);
// get real output
tmpsiz = sizeof(output);
AudioConverterGetProperty(pv->converter,
kAudioConverterCurrentOutputStreamDescription,
&tmpsiz, &output);
// set sizes
pv->isamplesiz = input.mBytesPerPacket;
pv->isamples = output.mFramesPerPacket;
pv->osamplerate = output.mSampleRate;
audio->config.out.samples_per_frame = pv->isamples;
// channel remapping
pv->remap = hb_audio_remap_init(AV_SAMPLE_FMT_FLT, &hb_aac_chan_map,
audio->config.in.channel_map);
if (pv->remap == NULL)
{
hb_error("encCoreAudioInit: hb_audio_remap_init() failed");
}
uint64_t layout = hb_ff_mixdown_xlat(audio->config.out.mixdown, NULL);
hb_audio_remap_set_channel_layout(pv->remap, layout);
// get maximum output size
AudioConverterGetProperty(pv->converter,
kAudioConverterPropertyMaximumOutputPacketSize,
&tmpsiz, &tmp);
pv->omaxpacket = tmp;
// get magic cookie (elementary stream descriptor)
tmp = HB_CONFIG_MAX_SIZE;
AudioConverterGetProperty(pv->converter,
kAudioConverterCompressionMagicCookie,
&tmp, w->config->extradata.bytes);
// CoreAudio returns a complete ESDS, but we only need
// the DecoderSpecific info.
UInt8* buffer = NULL;
ReadESDSDescExt(w->config->extradata.bytes, &buffer, &tmpsiz, 0);
w->config->extradata.length = tmpsiz;
memmove(w->config->extradata.bytes, buffer, w->config->extradata.length);
free(buffer);
pv->list = hb_list_init();
pv->buf = NULL;
return 0;
}
