Introduced osg::AudioStream class to help manage audio streams coming in from movie reading plugins
This commit is contained in:
@@ -56,7 +56,7 @@ void FFmpegDecoderAudio::open(AVStream * const stream)
|
||||
|
||||
m_frequency = m_context->sample_rate;
|
||||
m_nb_channels = m_context->channels;
|
||||
m_sample_format = FFmpegSampleFormat(m_context->sample_fmt);
|
||||
m_sample_format = osg::AudioStream::SampleFormat(m_context->sample_fmt);
|
||||
|
||||
// Check stream sanity
|
||||
if (m_context->codec_id == CODEC_ID_NONE)
|
||||
@@ -199,23 +199,23 @@ void FFmpegDecoderAudio::adjustBufferEndTps(const size_t buffer_size)
|
||||
|
||||
switch (sampleFormat())
|
||||
{
|
||||
case SAMPLE_FORMAT_U8:
|
||||
case osg::AudioStream::SAMPLE_FORMAT_U8:
|
||||
sample_size *= 1;
|
||||
break;
|
||||
|
||||
case SAMPLE_FORMAT_S16:
|
||||
case osg::AudioStream::SAMPLE_FORMAT_S16:
|
||||
sample_size *= 2;
|
||||
break;
|
||||
|
||||
case SAMPLE_FORMAT_S24:
|
||||
case osg::AudioStream::SAMPLE_FORMAT_S24:
|
||||
sample_size *= 3;
|
||||
break;
|
||||
|
||||
case SAMPLE_FORMAT_S32:
|
||||
case osg::AudioStream::SAMPLE_FORMAT_S32:
|
||||
sample_size *= 4;
|
||||
break;
|
||||
|
||||
case SAMPLE_FORMAT_F32:
|
||||
case osg::AudioStream::SAMPLE_FORMAT_F32:
|
||||
sample_size *= 4;
|
||||
break;
|
||||
|
||||
|
||||
Reference in New Issue
Block a user