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https://github.com/xiaoyifang/goldendict-ng.git
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Use libswresample to convert 32-bit and float audio into s16
Since libao+pulseaudio cannot play 32-bit or flt/fltp/dbl/dblp audio, the following audio formats are passed through libswresample to convert into AV_SAMPLE_FMT_S16, which is accepted by libao: * AV_SAMPLE_FMT_S32 * AV_SAMPLE_FMT_S32P * AV_SAMPLE_FMT_FLT * AV_SAMPLE_FMT_FLTP * AV_SAMPLE_FMT_DBL * AV_SAMPLE_FMT_DBLP This fixes issue #949 and issue #1014. Now FFmpeg+libao internal player can play with pulseaudio backend enabled in /etc/libao.conf . Signed-off-by: hrimfaxi <outmatch@gmail.com>
This commit is contained in:
hrimfaxi
parent
ec40c1dcfd
commit
2d2db3b208
111
ffmpegaudio.cc
111
ffmpegaudio.cc
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@ -19,6 +19,7 @@ extern "C" {
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#include <libavcodec/avcodec.h>
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#include <libavformat/avformat.h>
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#include <libavutil/avutil.h>
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#include "libswresample/swresample.h"
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}
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#include <QString>
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@ -100,6 +101,8 @@ struct DecoderContext
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ao_device * aoDevice_;
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bool avformatOpened_;
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SwrContext *swr_;
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DecoderContext( QByteArray const & audioData, QAtomicInt & isCancelled );
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~DecoderContext();
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@ -122,7 +125,8 @@ DecoderContext::DecoderContext( QByteArray const & audioData, QAtomicInt & isCan
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avioContext_( NULL ),
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audioStream_( NULL ),
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aoDevice_( NULL ),
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avformatOpened_( false )
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avformatOpened_( false ),
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swr_( NULL )
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{
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}
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@ -243,11 +247,36 @@ bool DecoderContext::openCodec( QString & errorString )
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av_log( NULL, AV_LOG_INFO, "Codec open: %s: channels: %d, rate: %d, format: %s\n", codec_->long_name,
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codecContext_->channels, codecContext_->sample_rate, av_get_sample_fmt_name( codecContext_->sample_fmt ) );
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if ( codecContext_->sample_fmt == AV_SAMPLE_FMT_S32 ||
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codecContext_->sample_fmt == AV_SAMPLE_FMT_S32P ||
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codecContext_->sample_fmt == AV_SAMPLE_FMT_FLT ||
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codecContext_->sample_fmt == AV_SAMPLE_FMT_FLTP ||
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codecContext_->sample_fmt == AV_SAMPLE_FMT_DBL ||
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codecContext_->sample_fmt == AV_SAMPLE_FMT_DBLP )
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{
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swr_ = swr_alloc_set_opts( NULL,
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codecContext_->channel_layout,
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AV_SAMPLE_FMT_S16,
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codecContext_->sample_rate,
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codecContext_->channel_layout,
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codecContext_->sample_fmt,
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codecContext_->sample_rate,
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0,
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NULL );
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swr_init( swr_ );
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}
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return true;
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}
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void DecoderContext::closeCodec()
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{
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if ( swr_ )
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{
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swr_free( &swr_ );
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}
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if ( !formatContext_ )
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{
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if ( avioContext_ )
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@ -306,11 +335,12 @@ bool DecoderContext::openOutputDevice( QString & errorString )
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}
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ao_sample_format aoSampleFormat;
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memset (&aoSampleFormat, 0, sizeof(aoSampleFormat) );
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aoSampleFormat.channels = codecContext_->channels;
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aoSampleFormat.rate = codecContext_->sample_rate;
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aoSampleFormat.byte_format = AO_FMT_NATIVE;
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aoSampleFormat.matrix = 0;
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aoSampleFormat.bits = qMin( 32, av_get_bytes_per_sample( codecContext_->sample_fmt ) << 3 );
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aoSampleFormat.bits = qMin( 16, av_get_bytes_per_sample( codecContext_->sample_fmt ) << 3 );
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if ( aoSampleFormat.bits == 0 )
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{
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@ -484,34 +514,11 @@ bool DecoderContext::normalizeAudio( AVFrame * frame, vector<char> & samples )
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{
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case AV_SAMPLE_FMT_U8:
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case AV_SAMPLE_FMT_S16:
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case AV_SAMPLE_FMT_S32:
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{
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samples.resize( dataSize );
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memcpy( &samples.front(), frame->data[0], lineSize );
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}
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break;
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case AV_SAMPLE_FMT_FLT:
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{
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samples.resize( dataSize );
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int32_t * out = ( int32_t * )&samples.front();
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for ( int i = 0; i < dataSize; i += sizeof( float ) )
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{
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*out++ = toInt32( *( float * )frame->data[i] );
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}
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}
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break;
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case AV_SAMPLE_FMT_DBL:
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{
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samples.resize( dataSize / 2 );
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int32_t * out = ( int32_t * )&samples.front();
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for ( int i = 0; i < dataSize; i += sizeof( double ) )
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{
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*out++ = toInt32( *( double * )frame->data[i] );
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}
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}
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break;
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// Planar
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case AV_SAMPLE_FMT_U8P:
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{
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@ -541,48 +548,28 @@ bool DecoderContext::normalizeAudio( AVFrame * frame, vector<char> & samples )
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}
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}
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break;
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case AV_SAMPLE_FMT_S32:
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/* Pass through */
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case AV_SAMPLE_FMT_S32P:
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{
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samples.resize( dataSize );
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int32_t * out = ( int32_t * )&samples.front();
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for ( int i = 0; i < frame->nb_samples; i++ )
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{
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for ( int ch = 0; ch < codecContext_->channels; ch++ )
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{
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*out++ = ( ( int32_t * )frame->extended_data[ch] )[i];
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}
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}
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}
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break;
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/* Pass through */
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case AV_SAMPLE_FMT_FLT:
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/* Pass through */
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case AV_SAMPLE_FMT_FLTP:
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{
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samples.resize( dataSize );
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float ** data = ( float ** )frame->extended_data;
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int32_t * out = ( int32_t * )&samples.front();
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for ( int i = 0; i < frame->nb_samples; i++ )
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{
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for ( int ch = 0; ch < codecContext_->channels; ch++ )
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{
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*out++ = toInt32( data[ch][i] );
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}
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}
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}
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break;
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case AV_SAMPLE_FMT_DBLP:
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/* Pass through */
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{
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samples.resize( dataSize / 2 );
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double ** data = ( double ** )frame->extended_data;
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int32_t * out = ( int32_t * )&samples.front();
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for ( int i = 0; i < frame->nb_samples; i++ )
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{
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for ( int ch = 0; ch < codecContext_->channels; ch++ )
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{
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*out++ = toInt32( data[ch][i] );
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}
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}
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uint8_t *out = ( uint8_t * )&samples.front();
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swr_convert( swr_, &out, frame->nb_samples, (const uint8_t**)frame->extended_data, frame->nb_samples );
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}
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break;
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case AV_SAMPLE_FMT_DBL:
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case AV_SAMPLE_FMT_DBLP:
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{
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samples.resize( dataSize / 4 );
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uint8_t *out = ( uint8_t * )&samples.front();
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swr_convert( swr_, &out, frame->nb_samples, (const uint8_t**)frame->extended_data, frame->nb_samples );
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}
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break;
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default:
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@ -108,6 +108,7 @@ win32 {
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-logg
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!CONFIG( no_ffmpeg_player ) {
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LIBS += -lao \
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-lswresample-gd \
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-lavutil-gd \
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-lavformat-gd \
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-lavcodec-gd
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@ -156,7 +157,8 @@ unix:!mac {
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PKGCONFIG += ao \
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libavutil \
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libavformat \
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libavcodec
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libavcodec \
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libswresample \
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}
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arm {
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LIBS += -liconv
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@ -210,6 +212,7 @@ mac {
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-llzo2
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!CONFIG( no_ffmpeg_player ) {
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LIBS += -lao \
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-lswresample-gd \
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-lavutil-gd \
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-lavformat-gd \
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-lavcodec-gd
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