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ffmpeg convert the the audio to raw format

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This commit is contained in:
Xiao YiFang 2022-11-15 19:54:34 +08:00
parent edea394892
commit 12acff095c
2 changed files with 70 additions and 9 deletions
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2
audiooutput.cpp
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@ -23,7 +23,7 @@ static QAudioFormat format( int sampleRate, int channelCount )
QAudioFormat out; QAudioFormat out;
out.setSampleRate( sampleRate ); out.setSampleRate( sampleRate );
out.setChannelCount( 2 ); out.setChannelCount( channelCount );
#if QT_VERSION < QT_VERSION_CHECK( 6, 0, 0 ) #if QT_VERSION < QT_VERSION_CHECK( 6, 0, 0 )
out.setByteOrder( QAudioFormat::LittleEndian ); out.setByteOrder( QAudioFormat::LittleEndian );
out.setCodec( QLatin1String( "audio/pcm" ) ); out.setCodec( QLatin1String( "audio/pcm" ) );

77
ffmpegaudio.cc
View file

@ -241,7 +241,7 @@ bool DecoderContext::openCodec( QString & errorString )
{ {
swr_ = swr_alloc_set_opts( NULL, swr_ = swr_alloc_set_opts( NULL,
av_get_default_channel_layout(2), codecContext_->channel_layout,
AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16,
codecContext_->sample_rate, codecContext_->sample_rate,
codecContext_->channel_layout, codecContext_->channel_layout,
@ -383,14 +383,75 @@ bool DecoderContext::play( QString & errorString )
bool DecoderContext::normalizeAudio( AVFrame * frame, vector<uint8_t > & samples ) bool DecoderContext::normalizeAudio( AVFrame * frame, vector<uint8_t > & samples )
{ {
int lineSize = 0; int lineSize = 0;
// int dataSize = av_samples_get_buffer_size( &lineSize, codecContext_->channels, int dataSize = av_samples_get_buffer_size( &lineSize, codecContext_->channels,
// frame->nb_samples, codecContext_->sample_fmt, 1 ); frame->nb_samples, codecContext_->sample_fmt, 1 );
int dataSize = frame->nb_samples * 2 * 2; // Portions from: https://code.google.com/p/lavfilters/source/browse/decoder/LAVAudio/LAVAudio.cpp
samples.resize( dataSize ); // But this one use 8, 16, 32 bits integer, respectively.
uint8_t *data[2] = { 0 }; switch ( codecContext_->sample_fmt )
data[0] = &samples.front(); //输出格式为AV_SAMPLE_FMT_S16(packet类型),所以转换后的LR两通道都存在data[0]中 {
case AV_SAMPLE_FMT_U8:
case AV_SAMPLE_FMT_S16:
{
samples.resize( dataSize );
memcpy( &samples.front(), frame->data[0], lineSize );
}
break;
// Planar
case AV_SAMPLE_FMT_U8P:
{
samples.resize( dataSize );
swr_convert( swr_, data, frame->nb_samples, (const uint8_t**)frame->data, frame->nb_samples ); uint8_t * out = ( uint8_t * )&samples.front();
for ( int i = 0; i < frame->nb_samples; i++ )
{
for ( int ch = 0; ch < codecContext_->channels; ch++ )
{
*out++ = ( ( uint8_t * )frame->extended_data[ch] )[i];
}
}
}
break;
case AV_SAMPLE_FMT_S16P:
{
samples.resize( dataSize );
int16_t * out = ( int16_t * )&samples.front();
for ( int i = 0; i < frame->nb_samples; i++ )
{
for ( int ch = 0; ch < codecContext_->channels; ch++ )
{
*out++ = ( ( int16_t * )frame->extended_data[ch] )[i];
}
}
}
break;
case AV_SAMPLE_FMT_S32:
/* Pass through */
case AV_SAMPLE_FMT_S32P:
/* Pass through */
case AV_SAMPLE_FMT_FLT:
/* Pass through */
case AV_SAMPLE_FMT_FLTP:
/* Pass through */
{
samples.resize( dataSize / 2 );
uint8_t *out = ( uint8_t * )&samples.front();
swr_convert( swr_, &out, frame->nb_samples, (const uint8_t**)frame->extended_data, frame->nb_samples );
}
break;
case AV_SAMPLE_FMT_DBL:
case AV_SAMPLE_FMT_DBLP:
{
samples.resize( dataSize / 4 );
uint8_t *out = ( uint8_t * )&samples.front();
swr_convert( swr_, &out, frame->nb_samples, (const uint8_t**)frame->extended_data, frame->nb_samples );
}
break;
default:
return false;
}
return true; return true;
} }