Support planar sample formats in audio mixdown

Now libav-10 should work for output of ac3 container.
This commit is contained in:
Sergey Sharybin 2014-02-28 15:15:04 +06:00
parent 1e05956a0b
commit 5ac1b38b4b
Notes: blender-bot 2023-02-14 11:08:38 +01:00
Referenced by issue #38768, new "audio" button in 2.70 release does not 'mixdown' audio
2 changed files with 36 additions and 2 deletions

View File

@ -169,6 +169,19 @@ AUD_FFMPEGWriter::AUD_FFMPEGWriter(std::string filename, AUD_DeviceSpecs specs,
if(!codec)
AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
if(codec->sample_fmts) {
// Check if the prefered sample format for this codec is supported.
const enum AVSampleFormat *p = codec->sample_fmts;
for(; *p != -1; p++) {
if(*p == m_stream->codec->sample_fmt)
break;
}
if(*p == -1) {
// Sample format incompatible with codec. Defaulting to a format known to work.
m_stream->codec->sample_fmt = codec->sample_fmts[0];
}
}
if(avcodec_open2(m_codecCtx, codec, NULL))
AUD_THROW(AUD_ERROR_FFMPEG, codec_error);
@ -199,8 +212,11 @@ AUD_FFMPEGWriter::AUD_FFMPEGWriter(std::string filename, AUD_DeviceSpecs specs,
# ifdef FFMPEG_HAVE_FRAME_CHANNEL_LAYOUT
m_frame->channel_layout = m_codecCtx->channel_layout;
# endif
m_audio_sample_size = av_get_bytes_per_sample(m_codecCtx->sample_fmt);
m_sample_size = av_get_bytes_per_sample(m_codecCtx->sample_fmt);
m_frame_pts = 0;
m_deinterleave = av_sample_fmt_is_planar(m_codecCtx->sample_fmt);
if(m_deinterleave)
m_deinterleave_buffer.resize(m_input_size * m_codecCtx->channels * m_sample_size);
#endif
try
@ -284,6 +300,17 @@ void AUD_FFMPEGWriter::encode(sample_t* data)
m_frame->channel_layout = m_codecCtx->channel_layout;
#endif
if(m_deinterleave) {
for(int channel = 0; channel < m_codecCtx->channels; channel++) {
for(int i = 0; i < m_frame->nb_samples; i++) {
memcpy(reinterpret_cast<uint8_t*>(m_deinterleave_buffer.getBuffer()) + (i + channel * m_frame->nb_samples) * m_sample_size,
reinterpret_cast<uint8_t*>(data) + (m_codecCtx->channels * i + channel) * m_sample_size, m_sample_size);
}
}
data = m_deinterleave_buffer.getBuffer();
}
avcodec_fill_audio_frame(m_frame, m_codecCtx->channels, m_codecCtx->sample_fmt, reinterpret_cast<uint8_t*>(data),
m_frame->nb_samples * av_get_bytes_per_sample(m_codecCtx->sample_fmt) * m_codecCtx->channels, 1);

View File

@ -90,7 +90,14 @@ private:
/**
* Number of bytes per sample.
*/
int m_audio_sample_size;
int m_sample_size;
/**
* Need to de-interleave audio for planar sample formats.
*/
bool m_deinterleave;
AUD_Buffer m_deinterleave_buffer;
/**
* The input buffer for the format converted data before encoding.