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
This commit is contained in:
Sergey Sharybin
2014-02-28 14:44:01 +06:00
parent 8fa15d1050
commit e82b5f61fe

View File

@@ -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