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fix: FFmpeg backend 7.0 compile

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This commit is contained in:
shenleban tongying 2024-05-21 11:54:42 -04:00
parent 1c372370fb
commit e2e1ca2d49
1 changed files with 37 additions and 1 deletions
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38
src/ffmpegaudio.cc
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@ -179,6 +179,31 @@ bool DecoderContext::openCodec( QString & errorString )
return false;
}
// 61 = FFmpeg 7.0 -> https://github.com/FFmpeg/FFmpeg/blob/release/7.0/libavcodec/version_major.h
#if LIBAVCODEC_VERSION_MAJOR >= 61
gdDebug( "Codec open: %s: channels: %d, rate: %d, format: %s\n",
codec_->long_name,
codecContext_->ch_layout.nb_channels,
codecContext_->sample_rate,
av_get_sample_fmt_name( codecContext_->sample_fmt ) );
if ( !av_channel_layout_check( &codecContext_->ch_layout ) ) {
av_channel_layout_default( &codecContext_->ch_layout, codecContext_->ch_layout.nb_channels );
}
if ( swr_alloc_set_opts2( &swr_,
&codecContext_->ch_layout,
AV_SAMPLE_FMT_S16,
44100,
&codecContext_->ch_layout,
codecContext_->sample_fmt,
codecContext_->sample_rate,
0,
nullptr ) != 0
) {
qDebug() << "swr_alloc_set_opts2 failed.";
}
#else
gdDebug( "Codec open: %s: channels: %d, rate: %d, format: %s\n",
codec_->long_name,
codecContext_->channels,
@ -200,6 +225,8 @@ bool DecoderContext::openCodec( QString & errorString )
codecContext_->sample_rate,
0,
nullptr );
#endif
if ( !swr_ || swr_init( swr_ ) < 0 ) {
av_log( nullptr, AV_LOG_ERROR, "Cannot create sample rate converter \n" );
@ -268,8 +295,11 @@ bool DecoderContext::openOutputDevice( QString & errorString )
errorString += QStringLiteral( "Failed to create audioOutput." );
return false;
}
#if LIBAVCODEC_VERSION_MAJOR >= 61
audioOutput->setAudioFormat( 44100, codecContext_->ch_layout.nb_channels );
#else
audioOutput->setAudioFormat( 44100, codecContext_->channels );
#endif
return true;
}
@ -329,7 +359,13 @@ void DecoderContext::stop()
bool DecoderContext::normalizeAudio( AVFrame * frame, vector< uint8_t > & samples )
{
auto dst_freq = 44100;
#if LIBAVCODEC_VERSION_MAJOR >= 61
auto dst_channels = codecContext_->ch_layout.nb_channels;
#else
auto dst_channels = codecContext_->channels;
#endif
int out_count = (int64_t)frame->nb_samples * dst_freq / frame->sample_rate + 256;
int out_size = av_samples_get_buffer_size( nullptr, dst_channels, out_count, AV_SAMPLE_FMT_S16, 1 );
samples.resize( out_size );