Commit: e82b5f61fee64a02fd71be605fea75dad4f7ce22
Author: Sergey Sharybin
Date: Fri Feb 28 14:44:01 2014 +0600
https://developer.blender.org/rBe82b5f61fee64a02fd71be605fea75dad4f7ce22
Attempt to fix mixdown when building without sndfile
This case was handled specially in writeffmpeg.c and seems it makes
audio export happy in all cases now.
TODO: libav-10 doesn't work with AC3 codec yet because this bloody
library ONLY supports FLTP format and FFmpeg ONLY supports FLT.
This is not fun guy, it really isn't! Where's your conscience??
CC: nexyon
===================================================================
M intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
===================================================================
diff --git a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
index 859227a..dcacf71 100644
--- a/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
+++ b/intern/audaspace/ffmpeg/AUD_FFMPEGWriter.cpp
@@ -175,8 +175,10 @@ AUD_FFMPEGWriter::AUD_FFMPEGWriter(std::string filename,
AUD_DeviceSpecs specs,
m_output_buffer.resize(FF_MIN_BUFFER_SIZE);
int samplesize = AUD_MAX(AUD_SAMPLE_SIZE(m_specs),
AUD_DEVICE_SAMPLE_SIZE(m_specs));
- if(m_codecCtx->frame_size <= 1)
- m_input_size = 0;
+ if(m_codecCtx->frame_size <= 1) {
+ m_input_size = FF_MIN_BUFFER_SIZE * 8 /
m_codecCtx->bits_per_coded_sample / m_codecCtx->channels;
+ m_input_buffer.resize(m_input_size *
samplesize);
+ }
else
{
m_input_buffer.resize(m_codecCtx->frame_size *
samplesize);
@@ -190,7 +192,7 @@ AUD_FFMPEGWriter::AUD_FFMPEGWriter(std::string filename,
AUD_DeviceSpecs specs,
avcodec_get_frame_defaults(m_frame);
m_frame->linesize[0] = m_input_size * samplesize;
m_frame->format = m_codecCtx->sample_fmt;
- m_frame->nb_samples = m_codecCtx->frame_size;
+ m_frame->nb_samples = m_input_size;
# ifdef FFMPEG_HAVE_AVFRAME_SAMPLE_RATE
m_frame->sample_rate = m_codecCtx->sample_rate;
# endif
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