void WaveFileWriter::AddStereoSamplesBE(const short* sample_data, u32 count, int sample_rate)
{
if (!file)
PanicAlertT("WaveFileWriter - file not open.");
if (count > BUFFER_SIZE * 2)
PanicAlert("WaveFileWriter - buffer too small (count = %u).", count);
if (skip_silence)
{
bool all_zero = true;
for (u32 i = 0; i < count * 2; i++)
{
if (sample_data[i])
all_zero = false;
}
if (all_zero)
return;
}
for (u32 i = 0; i < count; i++)
{
// Flip the audio channels from RL to LR
conv_buffer[2 * i] = Common::swap16((u16)sample_data[2 * i + 1]);
conv_buffer[2 * i + 1] = Common::swap16((u16)sample_data[2 * i]);
}
CheckSampleRate(sample_rate);
file.WriteBytes(conv_buffer.data(), count * 4);
audio_size += count * 4;
}
bool ParseCmdLine ( int argc, const char* argv[] ) {
int i=1;
while ( i< argc ) {
if ( argv[i][0] == '-'
&& argv[i][1] == 's' )
{
if ( ++i < argc ) {
gSampleRate = atoi(argv[i++]);
if (!CheckSampleRate ()) return false;
} else {
return false;
}
} else
if ( argv[i][0] == '-'
&& argv[i][1] == 'r' )
{
if ( ++i < argc ) {
gDataRateK = atoi(argv[i++]);
} else {
return false;
}
} else if ( argv[i][0] == '-'
&& argv[i][1] == 'f' )
{
if ( ++i < argc ) {
gChannelFreq = atoi(argv[i++]);
} else {
return false;
}
} else if ( argv[i][0] == '-'
&& argv[i][1] == 'o')
{
if ( ++i < argc ) {
gDisplayMode = atoi(argv[i++]);
} else return false;
if ( gDisplayMode > 1 ) return false;
} else if ( argv[i][0] == '-'
&& argv[i][1] == 'h')
{
return false;
}
else {
return false;
}
}
return true;
}
USBAudioClockSource* USBAudioDevice::FindClockSource(int freq)
{
USBAudioControlInterface * iface = m_acInterfaceList.First();
while(iface)
{
USBAudioClockSource* clockSource = iface->m_clockSourceList.First();
while(clockSource)
{
if(CheckSampleRate(clockSource, freq))
return clockSource;
clockSource = iface->m_clockSourceList.Next(clockSource);
}
iface = m_acInterfaceList.Next(iface);
}
return NULL;
}
void WaveFileWriter::AddStereoSamples(const short* sample_data, u32 count, int sample_rate)
{
if (!file)
PanicAlertT("WaveFileWriter - file not open.");
if (skip_silence)
{
bool all_zero = true;
for (u32 i = 0; i < count * 2; i++)
{
if (sample_data[i])
all_zero = false;
}
if (all_zero)
return;
}
CheckSampleRate(sample_rate);
file.WriteBytes(sample_data, count * 4);
audio_size += count * 4;
}
bool ExportFFmpeg::InitCodecs(AudacityProject *project)
{
AVCodec * codec = NULL;
// Configure the audio stream's codec context.
mEncAudioCodecCtx = mEncAudioStream->codec;
FFmpegLibsInst->avcodec_get_context_defaults(mEncAudioCodecCtx);
mEncAudioCodecCtx->codec_id = ExportFFmpegOptions::fmts[mSubFormat].codecid;
mEncAudioCodecCtx->codec_type = CODEC_TYPE_AUDIO;
mEncAudioCodecCtx->codec_tag = FFmpegLibsInst->av_codec_get_tag((const AVCodecTag **)mEncFormatCtx->oformat->codec_tag,mEncAudioCodecCtx->codec_id);
mSampleRate = (int)project->GetRate();
mEncAudioCodecCtx->global_quality = -99999; //quality mode is off by default;
// Each export type has its own settings
switch (mSubFormat)
{
case FMT_M4A:
mEncAudioCodecCtx->bit_rate = 98000;
mEncAudioCodecCtx->bit_rate *= mChannels;
mEncAudioCodecCtx->profile = FF_PROFILE_AAC_LOW;
mEncAudioCodecCtx->cutoff = 0;
mEncAudioCodecCtx->global_quality = gPrefs->Read(wxT("/FileFormats/AACQuality"),-99999);
if (!CheckSampleRate(mSampleRate,
ExportFFmpegOptions::iAACSampleRates[0],
ExportFFmpegOptions::iAACSampleRates[11],
&ExportFFmpegOptions::iAACSampleRates[0]))
{
mSampleRate = AskResample(mEncAudioCodecCtx->bit_rate,mSampleRate,
ExportFFmpegOptions::iAACSampleRates[0],
ExportFFmpegOptions::iAACSampleRates[11],
&ExportFFmpegOptions::iAACSampleRates[0]);
}
break;
case FMT_AC3:
mEncAudioCodecCtx->bit_rate = gPrefs->Read(wxT("/FileFormats/AC3BitRate"), 192000);
if (!CheckSampleRate(mSampleRate,ExportFFmpegAC3Options::iAC3SampleRates[0], ExportFFmpegAC3Options::iAC3SampleRates[2], &ExportFFmpegAC3Options::iAC3SampleRates[0]))
mSampleRate = AskResample(mEncAudioCodecCtx->bit_rate,mSampleRate, ExportFFmpegAC3Options::iAC3SampleRates[0], ExportFFmpegAC3Options::iAC3SampleRates[2], &ExportFFmpegAC3Options::iAC3SampleRates[0]);
break;
case FMT_AMRNB:
mSampleRate = 8000;
mEncAudioCodecCtx->bit_rate = gPrefs->Read(wxT("/FileFormats/AMRNBBitRate"), 12200);
break;
#if FFMPEG_STABLE
case FMT_AMRWB:
mSampleRate = 16000;
mEncAudioCodecCtx->bit_rate = gPrefs->Read(wxT("/FileFormats/AMRWBBitRate"), 23850);
break;
#endif
case FMT_WMA2:
mEncAudioCodecCtx->bit_rate = gPrefs->Read(wxT("/FileFormats/WMABitRate"), 198000);
if (!CheckSampleRate(mSampleRate,ExportFFmpegWMAOptions::iWMASampleRates[0], ExportFFmpegWMAOptions::iWMASampleRates[4], &ExportFFmpegWMAOptions::iWMASampleRates[0]))
mSampleRate = AskResample(mEncAudioCodecCtx->bit_rate,mSampleRate, ExportFFmpegWMAOptions::iWMASampleRates[0], ExportFFmpegWMAOptions::iWMASampleRates[4], &ExportFFmpegWMAOptions::iWMASampleRates[0]);
mEncAudioCodecCtx->flags2 |= CODEC_FLAG2_BIT_RESERVOIR | 0x0004;
break;
case FMT_OTHER:
strncpy(mEncAudioStream->language,gPrefs->Read(wxT("/FileFormats/FFmpegLanguage"),wxT("")).mb_str(wxConvUTF8),4);
mEncAudioCodecCtx->sample_rate = gPrefs->Read(wxT("/FileFormats/FFmpegSampleRate"),(long)0);
if (mEncAudioCodecCtx->sample_rate != 0) mSampleRate = mEncAudioCodecCtx->sample_rate;
mEncAudioCodecCtx->bit_rate = gPrefs->Read(wxT("/FileFormats/FFmpegBitRate"), (long)0);
strncpy((char *)&mEncAudioCodecCtx->codec_tag,gPrefs->Read(wxT("/FileFormats/FFmpegTag"),wxT("")).mb_str(wxConvUTF8),4);
mEncAudioCodecCtx->global_quality = gPrefs->Read(wxT("/FileFormats/FFmpegQuality"),(long)-99999);
mEncAudioCodecCtx->cutoff = gPrefs->Read(wxT("/FileFormats/FFmpegCutOff"),(long)0);
mEncAudioCodecCtx->flags2 = 0;
if (gPrefs->Read(wxT("/FileFormats/FFmpegBitReservoir"),true)) mEncAudioCodecCtx->flags2 |= CODEC_FLAG2_BIT_RESERVOIR;
if (gPrefs->Read(wxT("/FileFormats/FFmpegVariableBlockLen"),true)) mEncAudioCodecCtx->flags2 |= 0x0004; //WMA only?
mEncAudioCodecCtx->use_lpc = gPrefs->Read(wxT("/FileFormats/FFmpegUseLPC"),true);
mEncAudioCodecCtx->compression_level = gPrefs->Read(wxT("/FileFormats/FFmpegCompLevel"),-1);
mEncAudioCodecCtx->frame_size = gPrefs->Read(wxT("/FileFormats/FFmpegFrameSize"),(long)0);
mEncAudioCodecCtx->lpc_coeff_precision = gPrefs->Read(wxT("/FileFormats/FFmpegLPCCoefPrec"),(long)0);
mEncAudioCodecCtx->min_prediction_order = gPrefs->Read(wxT("/FileFormats/FFmpegMinPredOrder"),(long)-1);
mEncAudioCodecCtx->max_prediction_order = gPrefs->Read(wxT("/FileFormats/FFmpegMaxPredOrder"),(long)-1);
mEncAudioCodecCtx->min_partition_order = gPrefs->Read(wxT("/FileFormats/FFmpegMinPartOrder"),(long)-1);
mEncAudioCodecCtx->max_partition_order = gPrefs->Read(wxT("/FileFormats/FFmpegMaxPartOrder"),(long)-1);
mEncAudioCodecCtx->prediction_order_method = gPrefs->Read(wxT("/FileFormats/FFmpegPredOrderMethod"),(long)0);
mEncFormatCtx->mux_rate = gPrefs->Read(wxT("/FileFormats/FFmpegMuxRate"),(long)0);
mEncFormatCtx->packet_size = gPrefs->Read(wxT("/FileFormats/FFmpegPacketSize"),(long)0);
mEncAudioCodecCtx->codec_id = (CodecID)gPrefs->Read(wxT("/FileFormats/FFmpegCodec"), mEncFormatDesc->audio_codec);
break;
default:
return false;
}
// This happens if user refused to resample the project
if (mSampleRate == 0) return false;
if (mEncAudioCodecCtx->global_quality >= 0)
{
/* I'm not sure this is required, regardless of FFmpeg version
#if FFMPEG_STABLE
mEncAudioCodecCtx->bit_rate = 0;
#endif
*/
mEncAudioCodecCtx->flags |= CODEC_FLAG_QSCALE;
}
else mEncAudioCodecCtx->global_quality = -99999;
this->mEncAudioStream->quality = mEncAudioCodecCtx->global_quality = mEncAudioCodecCtx->global_quality * FF_QP2LAMBDA;
mEncAudioCodecCtx->sample_rate = mSampleRate;
mEncAudioCodecCtx->channels = mChannels;
mEncAudioCodecCtx->time_base.num = 1;
mEncAudioCodecCtx->time_base.den = mEncAudioCodecCtx->sample_rate;
mEncAudioCodecCtx->sample_fmt = SAMPLE_FMT_S16;
//mEncAudioCodecCtx->strict_std_compliance = FF_COMPLIANCE_STRICT;
// Is the required audio codec compiled into libavcodec?
if ((codec = FFmpegLibsInst->avcodec_find_encoder(mEncAudioCodecCtx->codec_id)) == NULL)
{
wxLogMessage(wxT("FFmpeg : ERROR - Can't find audio codec 0x%x."),mEncAudioCodecCtx->codec_id);
wxMessageBox(wxString::Format(_("FFmpeg cannot find audio codec 0x%x.\nSupport for this codec is probably not compiled in."),mEncAudioCodecCtx->codec_id));
return false;
}
if (mEncFormatCtx->oformat->flags & AVFMT_GLOBALHEADER)
{
mEncAudioCodecCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;
mEncFormatCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
// Open the codec.
if (FFmpegLibsInst->avcodec_open(mEncAudioCodecCtx, codec) < 0 || mEncAudioCodecCtx->frame_size == 0)
{
wxLogMessage(wxT("FFmpeg : ERROR - Can't open audio codec 0x%x."),mEncAudioCodecCtx->codec_id);
return false;
}
wxLogMessage(wxT("FFmpeg : Audio Output Codec Frame Size: %d samples."), mEncAudioCodecCtx->frame_size);
if ((mEncAudioCodecCtx->codec_id >= CODEC_ID_PCM_S16LE) && (mEncAudioCodecCtx->codec_id <= CODEC_ID_PCM_DVD))
{
mEncAudioEncodedBufSiz = FF_MIN_BUFFER_SIZE;
}
// Allocate a buffer for the encoder to store encoded audio frames into.
if ((mEncAudioEncodedBuf = (uint8_t*)FFmpegLibsInst->av_malloc(mEncAudioEncodedBufSiz)) == NULL)
{
wxLogMessage(wxT("FFmpeg : ERROR - Can't allocate buffer to hold encoded audio."));
return false;
}
// The encoder may require a minimum number of raw audio samples for each encoding but we can't
// guarantee we'll get this minimum each time an audio frame is decoded from the input file so
// we use a FIFO to store up incoming raw samples until we have enough for one call to the codec.
#if FFMPEG_STABLE
FFmpegLibsInst->av_fifo_init(&mEncAudioFifo, 1024);
#else
mEncAudioFifo = FFmpegLibsInst->av_fifo_alloc(1024);
#endif
// Allocate a buffer to read OUT of the FIFO into. The FIFO maintains its own buffer internally.
if ((mEncAudioFifoOutBuf = (uint8_t*)FFmpegLibsInst->av_malloc(2*MAX_AUDIO_PACKET_SIZE)) == NULL)
{
wxLogMessage(wxT("FFmpeg : ERROR - Can't allocate buffer to read into from audio FIFO."));
return false;
}
return true;
}
bool ExportFFmpeg::InitCodecs(AudacityProject *project)
{
AVCodec *codec = NULL;
AVDictionary *options = NULL;
// Configure the audio stream's codec context.
mEncAudioCodecCtx = mEncAudioStream->codec;
mEncAudioCodecCtx->codec_id = ExportFFmpegOptions::fmts[mSubFormat].codecid;
mEncAudioCodecCtx->codec_type = AVMEDIA_TYPE_AUDIO;
mEncAudioCodecCtx->codec_tag = av_codec_get_tag((const AVCodecTag **)mEncFormatCtx->oformat->codec_tag,mEncAudioCodecCtx->codec_id);
mSampleRate = (int)project->GetRate();
mEncAudioCodecCtx->global_quality = -99999; //quality mode is off by default;
// Each export type has its own settings
switch (mSubFormat)
{
case FMT_M4A:
mEncAudioCodecCtx->bit_rate = 98000;
mEncAudioCodecCtx->bit_rate *= mChannels;
mEncAudioCodecCtx->profile = FF_PROFILE_AAC_LOW;
mEncAudioCodecCtx->cutoff = 0;
mEncAudioCodecCtx->global_quality = gPrefs->Read(wxT("/FileFormats/AACQuality"),-99999);
if (!CheckSampleRate(mSampleRate,
ExportFFmpegOptions::iAACSampleRates[0],
ExportFFmpegOptions::iAACSampleRates[11],
&ExportFFmpegOptions::iAACSampleRates[0]))
{
mSampleRate = AskResample(mEncAudioCodecCtx->bit_rate,mSampleRate,
ExportFFmpegOptions::iAACSampleRates[0],
ExportFFmpegOptions::iAACSampleRates[11],
&ExportFFmpegOptions::iAACSampleRates[0]);
}
break;
case FMT_AC3:
mEncAudioCodecCtx->bit_rate = gPrefs->Read(wxT("/FileFormats/AC3BitRate"), 192000);
if (!CheckSampleRate(mSampleRate,ExportFFmpegAC3Options::iAC3SampleRates[0], ExportFFmpegAC3Options::iAC3SampleRates[2], &ExportFFmpegAC3Options::iAC3SampleRates[0]))
mSampleRate = AskResample(mEncAudioCodecCtx->bit_rate,mSampleRate, ExportFFmpegAC3Options::iAC3SampleRates[0], ExportFFmpegAC3Options::iAC3SampleRates[2], &ExportFFmpegAC3Options::iAC3SampleRates[0]);
break;
case FMT_AMRNB:
mSampleRate = 8000;
mEncAudioCodecCtx->bit_rate = gPrefs->Read(wxT("/FileFormats/AMRNBBitRate"), 12200);
break;
case FMT_WMA2:
mEncAudioCodecCtx->bit_rate = gPrefs->Read(wxT("/FileFormats/WMABitRate"), 198000);
if (!CheckSampleRate(mSampleRate,ExportFFmpegWMAOptions::iWMASampleRates[0], ExportFFmpegWMAOptions::iWMASampleRates[4], &ExportFFmpegWMAOptions::iWMASampleRates[0]))
mSampleRate = AskResample(mEncAudioCodecCtx->bit_rate,mSampleRate, ExportFFmpegWMAOptions::iWMASampleRates[0], ExportFFmpegWMAOptions::iWMASampleRates[4], &ExportFFmpegWMAOptions::iWMASampleRates[0]);
break;
case FMT_OTHER:
av_dict_set(&mEncAudioStream->metadata, "language", gPrefs->Read(wxT("/FileFormats/FFmpegLanguage"),wxT("")).ToUTF8(), 0);
mEncAudioCodecCtx->sample_rate = gPrefs->Read(wxT("/FileFormats/FFmpegSampleRate"),(long)0);
if (mEncAudioCodecCtx->sample_rate != 0) mSampleRate = mEncAudioCodecCtx->sample_rate;
mEncAudioCodecCtx->bit_rate = gPrefs->Read(wxT("/FileFormats/FFmpegBitRate"), (long)0);
strncpy((char *)&mEncAudioCodecCtx->codec_tag,gPrefs->Read(wxT("/FileFormats/FFmpegTag"),wxT("")).mb_str(wxConvUTF8),4);
mEncAudioCodecCtx->global_quality = gPrefs->Read(wxT("/FileFormats/FFmpegQuality"),(long)-99999);
mEncAudioCodecCtx->cutoff = gPrefs->Read(wxT("/FileFormats/FFmpegCutOff"),(long)0);
mEncAudioCodecCtx->flags2 = 0;
if (gPrefs->Read(wxT("/FileFormats/FFmpegBitReservoir"),true))
av_dict_set(&options, "reservoir", "1", 0);
if (gPrefs->Read(wxT("/FileFormats/FFmpegVariableBlockLen"),true)) mEncAudioCodecCtx->flags2 |= 0x0004; //WMA only?
mEncAudioCodecCtx->compression_level = gPrefs->Read(wxT("/FileFormats/FFmpegCompLevel"),-1);
mEncAudioCodecCtx->frame_size = gPrefs->Read(wxT("/FileFormats/FFmpegFrameSize"),(long)0);
//FIXME The list of supported options for the seleced encoder should be extracted instead of a few hardcoded
set_dict_int(&options, "lpc_coeff_precision", gPrefs->Read(wxT("/FileFormats/FFmpegLPCCoefPrec"),(long)0));
set_dict_int(&options, "min_prediction_order", gPrefs->Read(wxT("/FileFormats/FFmpegMinPredOrder"),(long)-1));
set_dict_int(&options, "max_prediction_order", gPrefs->Read(wxT("/FileFormats/FFmpegMaxPredOrder"),(long)-1));
set_dict_int(&options, "min_partition_order", gPrefs->Read(wxT("/FileFormats/FFmpegMinPartOrder"),(long)-1));
set_dict_int(&options, "max_partition_order", gPrefs->Read(wxT("/FileFormats/FFmpegMaxPartOrder"),(long)-1));
set_dict_int(&options, "prediction_order_method", gPrefs->Read(wxT("/FileFormats/FFmpegPredOrderMethod"),(long)0));
set_dict_int(&options, "muxrate", gPrefs->Read(wxT("/FileFormats/FFmpegMuxRate"),(long)0));
mEncFormatCtx->packet_size = gPrefs->Read(wxT("/FileFormats/FFmpegPacketSize"),(long)0);
codec = avcodec_find_encoder_by_name(gPrefs->Read(wxT("/FileFormats/FFmpegCodec")).ToUTF8());
if (!codec)
mEncAudioCodecCtx->codec_id = mEncFormatDesc->audio_codec;
break;
default:
return false;
}
// This happens if user refused to resample the project
if (mSampleRate == 0) return false;
if (mEncAudioCodecCtx->global_quality >= 0)
{
mEncAudioCodecCtx->flags |= CODEC_FLAG_QSCALE;
}
else mEncAudioCodecCtx->global_quality = 0;
mEncAudioCodecCtx->global_quality = mEncAudioCodecCtx->global_quality * FF_QP2LAMBDA;
mEncAudioCodecCtx->sample_rate = mSampleRate;
mEncAudioCodecCtx->channels = mChannels;
mEncAudioCodecCtx->time_base.num = 1;
mEncAudioCodecCtx->time_base.den = mEncAudioCodecCtx->sample_rate;
mEncAudioCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
mEncAudioCodecCtx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
if (mEncAudioCodecCtx->codec_id == AV_CODEC_ID_AC3)
{
// As of Jan 4, 2011, the default AC3 encoder only accept SAMPLE_FMT_FLT samples.
// But, currently, Audacity only supports SAMPLE_FMT_S16. So, for now, look for the
// "older" AC3 codec. this is not a proper solution, but will suffice until other
// encoders no longer support SAMPLE_FMT_S16.
codec = avcodec_find_encoder_by_name("ac3_fixed");
}
if (!codec)
{
codec = avcodec_find_encoder(mEncAudioCodecCtx->codec_id);
}
// Is the required audio codec compiled into libavcodec?
if (codec == NULL)
{
wxMessageBox(wxString::Format(_("FFmpeg cannot find audio codec 0x%x.\nSupport for this codec is probably not compiled in."), (unsigned int) mEncAudioCodecCtx->codec_id),
_("FFmpeg Error"), wxOK|wxCENTER|wxICON_EXCLAMATION);
return false;
}
if (codec->sample_fmts) {
for (int i=0; codec->sample_fmts[i] != AV_SAMPLE_FMT_NONE; i++) {
enum AVSampleFormat fmt = codec->sample_fmts[i];
if ( fmt == AV_SAMPLE_FMT_U8
|| fmt == AV_SAMPLE_FMT_U8P
|| fmt == AV_SAMPLE_FMT_S16
|| fmt == AV_SAMPLE_FMT_S16P
|| fmt == AV_SAMPLE_FMT_S32
|| fmt == AV_SAMPLE_FMT_S32P
|| fmt == AV_SAMPLE_FMT_FLT
|| fmt == AV_SAMPLE_FMT_FLTP) {
mEncAudioCodecCtx->sample_fmt = fmt;
}
if ( fmt == AV_SAMPLE_FMT_S16
|| fmt == AV_SAMPLE_FMT_S16P)
break;
}
}
if (mEncFormatCtx->oformat->flags & AVFMT_GLOBALHEADER)
{
mEncAudioCodecCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;
mEncFormatCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
// Open the codec.
if (avcodec_open2(mEncAudioCodecCtx, codec, &options) < 0)
{
wxMessageBox(wxString::Format(_("FFmpeg : ERROR - Can't open audio codec 0x%x."),mEncAudioCodecCtx->codec_id),
_("FFmpeg Error"), wxOK|wxCENTER|wxICON_EXCLAMATION);
return false;
}
default_frame_size = mEncAudioCodecCtx->frame_size;
if (default_frame_size == 0)
default_frame_size = 1024; // arbitrary non zero value;
wxLogDebug(wxT("FFmpeg : Audio Output Codec Frame Size: %d samples."), mEncAudioCodecCtx->frame_size);
// The encoder may require a minimum number of raw audio samples for each encoding but we can't
// guarantee we'll get this minimum each time an audio frame is decoded from the input file so
// we use a FIFO to store up incoming raw samples until we have enough for one call to the codec.
mEncAudioFifo = av_fifo_alloc(1024);
mEncAudioFifoOutBufSiz = 2*MAX_AUDIO_PACKET_SIZE;
// Allocate a buffer to read OUT of the FIFO into. The FIFO maintains its own buffer internally.
if ((mEncAudioFifoOutBuf = (uint8_t*)av_malloc(mEncAudioFifoOutBufSiz)) == NULL)
{
wxMessageBox(wxString::Format(_("FFmpeg : ERROR - Can't allocate buffer to read into from audio FIFO.")),
_("FFmpeg Error"), wxOK|wxCENTER|wxICON_EXCLAMATION);
return false;
}
return true;
}
