[Libav-user] Encoding live audiopackets

[Ruurd Adema]

I finally found the problem and solved it.

Paul, your tip was very helpfull, thanks for that!

Initially I was writing to less samples to the encoder (960). To solve that I added an audio_fifo system, but I didn’t correct the amount of samples in the buffer frame, so the encoder was still processing too less samples.

Changing the sample amount of the frame made it work.

Ruurd

> On 15 Jul 2015, at 20:06, Ruurd Adema <ruurdadema at me.com> wrote:
> 
> Yes, I tried that, I used an audio_fifo for that. Unfortunately makes no difference:
> 
> // This part works well
> if (CODEC_TYPE == AV_CODEC_ID_PCM_S16LE)
> {
>     audiopacket->GetPacketTime(&audio_pts, audio_stream->time_base.den);
>     
>     pkt.pts          = audio_pts;
>     pkt.dts          = pkt.pts; 
>     pkt.flags       |= AV_PKT_FLAG_KEY;                 
>     pkt.stream_index = audio_stream->index;
>     pkt.data         = (uint8_t *)audiopacket_data;
>     pkt.size         = audiopacket_size;
> 
>     av_interleaved_write_frame(output_fmt_ctx, &pkt);
> } 
> // This part doesn't work
> else if (CODEC_TYPE == AV_CODEC_ID_AAC)
> {
>     frame                 = av_frame_alloc();
>     frame->format         = audio_stream->codec->sample_fmt;
>     frame->channel_layout = audio_stream->codec->channel_layout;
>     frame->sample_rate    = audio_stream->codec->sample_rate;
>     frame->nb_samples     = audiopacket_sample_count;
> 
>     requested_size = av_samples_get_buffer_size(NULL, audio_stream->codec->channels, audio_stream->codec->frame_size, audio_stream->codec->sample_fmt, 1);
> 
>     result = av_audio_fifo_write(audio_fifo, &audiopacket_data, audiopacket_sample_count);
> 
>     audiopacket->GetPacketTime(&audio_pts, audio_stream->time_base.den);
>     
>     frame->pts = audio_pts;
> 
>     frame_buf = av_malloc(requested_size);
> 
>     // Check if there are enough samples to feed the encoder
>     if (av_audio_fifo_size(audio_fifo) >= audio_stream->codec->frame_size)
>     {
>         result = av_audio_fifo_read(audio_fifo, &frame_buf, audio_stream->codec->frame_size);
> 
>         if (avcodec_fill_audio_frame(frame, audiopacket_channel_count, audio_stream->codec->sample_fmt, (const uint8_t *)frame_buf, requested_size, 1) < 0)
>         {
>             fprintf(stderr, "[ERROR] Filling audioframe failed!\n");
>             exit(-1);
>         }
>         
>         if (avcodec_encode_audio2(audio_stream->codec, &pkt, frame, &got_packet) != 0)
>         {
>             fprintf(stderr, "[ERROR] Encoding audio failed\n");
>         }
> 
>         if (got_packet) 
>         {
>             pkt.stream_index = audio_stream->index;
>             pkt.flags       |= AV_PKT_FLAG_KEY; 
> 
>             av_interleaved_write_frame(output_fmt_ctx, &pkt);
>         }
>     }
>     free(frame_buf);
>     av_frame_free(&frame);   
> }
> av_free_packet(&pkt);
> 
> Thank, Ruurd
> 
>> On 14 Jul 2015, at 21:16, Paul B Mahol <onemda at gmail.com <mailto:onemda at gmail.com>> wrote:
>> 
>> 
>> Dana 14. 7. 2015. 20:49 osoba "Ruurd Adema" <ruurdadema at me.com <mailto:ruurdadema at me.com>> napisala je:
>> >
>> > I'm trying to write live incoming audiopackets into a mov file with AAC encoding using the FFmpeg api.
>> >
>> > When using no encoding (AV_CODEC_ID_PCM_S16LE) it works well, when using AAC encoding (AV_CODEC_ID_AAC) it fails. The resulting audiofile plays too fast and sounds distorted.
>> >
>> > I’m new to the FFmpeg api, (and quite a beginner in programming anyway), so big chance I forgot something or doing something wrong. Is there anyone willing to help me with this one?
>> >
>> > audiopacket_sample_count  = audiopacket->GetSampleFrameCount();
>> > audiopacket_channel_count = decklink_config()->audio_channel_count;
>> > audiopacket_size          = audiopacket_sample_count * (decklink_config()->audio_sampletype/8) * audiopacket_channel_count;
>> >
>> > audiopacket->GetBytes(&audiopacket_data);
>> >
>> > av_init_packet(&pkt);    
>> >
>> > if (AUDIO_TYPE == AV_CODEC_ID_PCM_S16LE)
>> > {
>> >     audiopacket->GetPacketTime(&audio_pts, audio_stream->time_base.den);
>> >
>> >     pkt.pts          = audio_pts;
>> >     pkt.dts          = pkt.pts; 
>> >     pkt.flags       |= AV_PKT_FLAG_KEY;                 
>> >     pkt.stream_index = audio_stream->index;
>> >     pkt.data         = (uint8_t *)audiopacket_data;
>> >     pkt.size         = audiopacket_size;
>> >
>> >     av_interleaved_write_frame(output_fmt_ctx, &pkt);
>> > } 
>> > else if (AUDIO_TYPE == AV_CODEC_ID_AAC)
>> > {
>> >     frame = av_frame_alloc();
>> >     frame->format = audio_stream->codec->sample_fmt;
>> >     frame->channel_layout = audio_stream->codec->channel_layout;
>> >     frame->sample_rate = audio_stream->codec->sample_rate;
>> >     frame->nb_samples = audiopacket_sample_count;
>> >
>> >     audiopacket->GetPacketTime(&audio_pts, audio_stream->time_base.den);
>> >
>> >     frame->pts = audio_pts;
>> >
>> >     if (avcodec_fill_audio_frame(frame, audiopacket_channel_count, audio_stream->codec->sample_fmt, (const uint8_t *)audiopacket_data, audiopacket_size, 0) < 0)
>> >     {
>> >         fprintf(stderr, "[ERROR] Filling audioframe failed!\n");
>> >         exit(-1);
>> >     }
>> >
>> >     if (avcodec_encode_audio2(audio_stream->codec, &pkt, frame, &got_packet) != 0)
>> >     {
>> >         fprintf(stderr, "[ERROR] Encoding audio failed\n");
>> >     }
>> >
>> >     if (got_packet) 
>> >     {
>> >         pkt.stream_index = audio_stream->index;
>> >         pkt.flags       |= AV_PKT_FLAG_KEY; 
>> >
>> >         av_interleaved_write_frame(output_fmt_ctx, &pkt);
>> >     }
>> >     av_frame_free(&frame); 
>> > }
>> > av_free_packet(&pkt);
>> >
>> >
>> >
>> > _______________________________________________
>> > Libav-user mailing list
>> > Libav-user at ffmpeg.org <mailto:Libav-user at ffmpeg.org>
>> > http://ffmpeg.org/mailman/listinfo/libav-user <http://ffmpeg.org/mailman/listinfo/libav-user>
>> >
>> 
>> Do you send exact same number of samples that aac encoder request? You need to buffer samples....
>> _______________________________________________
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> 
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