Commit: 5ac1b38b4b21959b552c0787e2f86a104624c615
Author: Sergey Sharybin
Date: Fri Feb 28 15:15:04 2014 +0600
https://developer.blender.org/rB5ac1b38b4b21959b552c0787e2f86a104624c615
Support planar sample formats in audio mixdown
Now libav-10 should work for output of ac3 container.
===================================================================
M intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
M intern/audaspace/ffmpeg/AUD_FFMPEGWriter.h
===================================================================
diff --git a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
index dcacf71..d30835d 100644
--- a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
+++ b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
@@ -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);
diff --git a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.h
b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.h
index 743d885..492aa35 100644
--- a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.h
+++ b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.h
@@ -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.
_______________________________________________
Bf-blender-cvs mailing list
[email protected]
http://lists.blender.org/mailman/listinfo/bf-blender-cvs
