[Libav-user] FFmpeg transcoded sound (AAC) stops after half video time

[Jan Drabner]

Am 8/13/2015 um 9:47 AM schrieb Jan Drabner:
> (This is the same post I made on Stackoverflow: 
> http://stackoverflow.com/questions/31968745/ffmpeg-transcoded-sound-aac-stops-after-half-video-time 
> )
>
> I have a strange problem in my C/C++ FFmpeg transcoder, which takes an 
> input MP4 (varying input codecs) and produces and output MP4 (x264, 
> baseline & AAC LC @44100 sample rate with libfdk_aac):
>
> The resulting mp4 video has fine images (x264) and the audio (AAC LC) 
> works fine as well, but is only played until exactly the half of the 
> video.
> The audio is not slowed down, not stretched and doesn't stutter. It 
> just stops right in the middle of the video.
>
> One hint may be that the input file has a sample rate of 22050 and 
> 44100/22050 is 0.5, but I really don't get why this would make the 
> sound just stop. I'd expect such an error leading to sound being at 
> the wrong speed. Everything works just fine if I don't try to enforce 
> 44100 and instead just use the incoming sample_rate.
>
> Another guess would be that the pts calculation doesn't work. But the 
> audio sounds just fine (until it stops) and I do exactly the same for 
> the video part, where it works flawlessly. "Exactly", as in the same 
> code, but "audio"-variables replaced with "video"-variables.
>
> FFmpeg reports no errors during the whole process. I also flush the 
> decoders/encoders/interleaved_writing after all the package reading 
> from the input is done. It works well for the video so I doubt there 
> is much wrong with my general approach.
>
> Here are the functions of my code (stripped off the error handling & 
> other class stuff):
>
> AudioCodecContext Setup
>
> |outContext->_audioCodec 
> =avcodec_find_encoder(outContext->_audioTargetCodecID);outContext->_audioStream 
> =avformat_new_stream(outContext->_formatContext,outContext->_audioCodec);outContext->_audioCodecContext 
> =outContext->_audioStream->codec;outContext->_audioCodecContext->channels 
> =2;outContext->_audioCodecContext->channel_layout 
> =av_get_default_channel_layout(2);outContext->_audioCodecContext->sample_rate 
> =44100;outContext->_audioCodecContext->sample_fmt 
> =outContext->_audioCodec->sample_fmts[0];outContext->_audioCodecContext->bit_rate 
> =128000;outContext->_audioCodecContext->strict_std_compliance 
> =FF_COMPLIANCE_EXPERIMENTAL;outContext->_audioCodecContext->time_base 
> =(AVRational){1,outContext->_audioCodecContext->sample_rate};outContext->_audioStream->time_base 
> =(AVRational){1,outContext->_audioCodecContext->sample_rate};intretVal 
> =avcodec_open2(outContext->_audioCodecContext,outContext->_audioCodec,NULL); 
> |
>
> Resampler Setup
>
> |outContext->_audioResamplerContext 
> =swr_alloc_set_opts(NULL,outContext->_audioCodecContext->channel_layout,outContext->_audioCodecContext->sample_fmt,outContext->_audioCodecContext->sample_rate,_inputContext._audioCodecContext->channel_layout,_inputContext._audioCodecContext->sample_fmt,_inputContext._audioCodecContext->sample_rate,0,NULL);intretVal 
> =swr_init(outContext->_audioResamplerContext); |
>
> Decoding
>
> |decodedBytes 
> =avcodec_decode_audio4(_inputContext._audioCodecContext,_inputContext._audioTempFrame,&p_gotAudioFrame,&_inputContext._currentPacket); 
> |
>
> Converting (only if decoding produced a frame, of course)
>
> |intretVal 
> =swr_convert(outContext->_audioResamplerContext,outContext->_audioConvertedFrame->data,outContext->_audioConvertedFrame->nb_samples,(constuint8_t**)_inputContext._audioTempFrame->data,_inputContext._audioTempFrame->nb_samples); 
> |
>
> Encoding (only if decoding produced a frame, of course)
>
> |outContext->_audioConvertedFrame->pts 
> =av_frame_get_best_effort_timestamp(_inputContext._audioTempFrame);// 
> Init the new 
> packetav_init_packet(&outContext->_audioPacket);outContext->_audioPacket.data 
> =NULL;outContext->_audioPacket.size =0;// EncodeintretVal 
> =avcodec_encode_audio2(outContext->_audioCodecContext,&outContext->_audioPacket,outContext->_audioConvertedFrame,&p_gotPacket);// 
> Set pts/dts time stamps for writing 
> interleavedav_packet_rescale_ts(&outContext->_audioPacket,outContext->_audioCodecContext->time_base,outContext->_audioStream->time_base);outContext->_audioPacket.stream_index 
> =outContext->_audioStream->index; |
>
> Writing (only if encoding produced a packet, of course)
>
> |intretVal 
> =av_interleaved_write_frame(outContext->_formatContext,&outContext->_audioPacket); 
> |
>
> I am quite out of ideas about what would cause such a behaviour.
>
>
Obviously, I meant that 22050/44100 is 0.5 ;)
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