[Libav-user] Encoding live audiopackets

[Paul B Mahol]

Dana 14. 7. 2015. 20:49 osoba "Ruurd Adema" <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);
>
>
>
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>

Do you send exact same number of samples that aac encoder request? You need
to buffer samples....
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