[Libav-user] aac encoder in real time scenario

Hi all,

Good moring,

I've seen that it's necessary to show the init methods, so here you have:

----------------------------------------8<-------------------------------------------

int audio_avcodec_init_encode(struct audio_avcodec_encode_state *aavces,
int bit_rate, int sample_rate, int channels){

    int enabled=0;
    avcodec_register_all();

    aavces->c= NULL;

    /* find the encoder */
    aavces->codec = avcodec_find_encoder(CODEC_ID_AAC);  //AQUÍ STRING
*codec, ara AAC default
    if (!aavces->codec) {
        fprintf(stderr, "\n[avcodec - audio - encode] Codec not found");
        //exit(1);
        return enabled;
    }else enabled = 1;

    aavces->c= avcodec_alloc_context();

    /* put sample parameters */
    aavces->c->bit_rate = bit_rate;//64000;
    aavces->c->sample_fmt = AV_SAMPLE_FMT_S16;
    //aavces->c->channel_layout = AV_CH_LAYOUT_STEREO;

    aavces->c->sample_rate = sample_rate;//48000;   //TODO: get it from
dp_map
    aavces->c->channels = channels;//2;                //TODO
    aavces->c->profile = FF_PROFILE_AAC_MAIN;//FF_PROFILE_AAC_LOW;
    //aavces->c->time_base = (AVRational){1, sample_rate};
    aavces->c->time_base.num = 1;
    aavces->c->time_base.den = sample_rate;
    aavces->c->codec_type = AVMEDIA_TYPE_AUDIO;

    /* open it */
    if (avcodec_open(aavces->c, aavces->codec) < 0) {
        fprintf(stderr, "\n[avcodec - audio - encode] Could not open
codec");
        //exit(1);
        return enabled;
    }else enabled = 1;

    /* the codec gives us the frame size, in samples */
    //aavces->frame_size = aavces->c->frame_size;
    //aavces->samples = malloc(aavces->frame_size * 2 *
aavces->c->channels);

    aavces->outbuf_size = 1024;//FF_MIN_BUFFER_SIZE * 10;
    aavces->outbuf = (uint8_t *)av_malloc(aavces->outbuf_size);

    aavces->fifo_buf =
av_fifo_alloc(2*MAX_AUDIO_PACKET_SIZE);//FF_MIN_BUFFER_SIZE);
    aavces->fifo_outbuf = (uint8_t *)av_malloc(MAX_AUDIO_PACKET_SIZE);

    if (!(aavces->outbuf == NULL))enabled = 1;

    printf("\n[avcodec - audio - encode] Enabled!",enabled);

    return enabled;

}

------------------------------->8------------------------------------------------------

Anyone can help me, please?

Hope not being a concept problem...

Thanks,
--------------------
  Gerard C.L.
--------------------

2013/3/14 Gerard C.L. <gerardcl at gmail.com>

> Hi all,
>
> I'm developing an AAC encoder in a real time environment.
>
> The scene is:
> - Capture format -> PCM: 48kHz, stereo, 16b/sample.  at 25fps  -> so, per
> frame, 7680Bytes have to be encoded.
>
> The first problem become when I realised that the encoder works on fixed
> chunk sizes (in this case, for the audio configuration, the size is
> 4096Bytes per chunk). So, working like a file encoder, I was only encoding
> 4096bytes of the 7680 per frame.
> The solution was implementing FIFOs, using the av_fifo_.. methods. So now,
> I can hear the entire captured sound per frame, but I hear some garbage and
> I don't know if it's because of the encoder or how I work with the fifo or
> if I have conceptual errors in my mind. To note that I'm playing the sound
> after saving it to a file, could it be also the problem?
>
> I'm copying the piece of code I've implemented right now, I'd love if some
> one gets the error... I'm so noob...
>
>
> -----------------------------------8<------------------------------------------------------------------
> int audio_avcodec_encode(struct audio_avcodec_encode_state *aavces,
> unsigned char *inbuf, unsigned char *outbuf, int inbufsize) {
>     AVPacket pkt;
>     int frameBytes;
>     int outsize = 0;
>     int packetSize = 0;
>     int ret;
>     int nfifoBytes;
>     int encBytes = 0;
>     int sizeTmp = 0;
>
>     frameBytes = aavces->c->frame_size * aavces->c->channels * 2;
>     av_fifo_realloc2(aavces->fifo_buf,av_fifo_size(aavces->fifo_buf) +
> inbufsize);
>
>     // Put the raw audio samples into the FIFO.
>     ret = av_fifo_generic_write(aavces->fifo_buf, /*(int8_t*)*/inbuf,
> inbufsize, NULL );
>
>     printf("\n[avcodec encode] raw buffer intput size: %d ; fifo size:
> %d",inbufsize, ret);
>
>     //encoding each frameByte block
>     while ((ret = av_fifo_size(aavces->fifo_buf)) >= frameBytes) {
>         ret = av_fifo_generic_read(aavces->fifo_buf,
> aavces->fifo_outbuf,frameBytes, NULL );
>
>         av_init_packet(&pkt);
>
>         pkt.size = avcodec_encode_audio(aavces->c,
> aavces->outbuf,aavces->outbuf_size, (int16_t*) aavces->fifo_outbuf);
>
>         if (pkt.size < 0) {
>             printf("FFmpeg : ERROR - Can't encode audio frame.");
>         }
>         // Rescale from the codec time_base to the AVStream time_base.
>         if (aavces->c->coded_frame && aavces->c->coded_frame->pts !=
> (int64_t) (AV_NOPTS_VALUE ))
>             pkt.pts =
> av_rescale_q(aavces->c->coded_frame->pts,aavces->c->time_base,
> aavces->c->time_base);
>
>         printf("\nFFmpeg : (%d) Writing audio frame with PTS:
> %lld.",aavces->c->frame_number, pkt.pts);
>         printf("\n[avcodec - audio - encode] Encoder returned %d bytes of
> data",pkt.size);
>
>         pkt.data = aavces->outbuf;
>         pkt.flags |= AV_PKT_FLAG_KEY;
>
>         memcpy(outbuf, pkt.data, pkt.size);
>     }
>
>     // any bytes left in audio FIFO to encode?
>     nfifoBytes = av_fifo_size(aavces->fifo_buf);
>
>     printf("\n[avcodec encode] raw buffer intput size: %d", nfifoBytes);
>
>     if (nfifoBytes > 0) {
>         memset(aavces->fifo_outbuf, 0, frameBytes);
>         if (aavces->c->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME) {
>             int nFrameSizeTmp = aavces->c->frame_size;
>             if (aavces->c->frame_size != 1 &&
> (aavces->c->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME))
>                 aavces->c->frame_size = nfifoBytes / (aavces->c->channels
> * 2);
>
>             if (av_fifo_generic_read(aavces->fifo_buf,
> aavces->fifo_outbuf,nfifoBytes, NULL ) == 0) {
>                 if (aavces->c->frame_size != 1)
>                     encBytes = avcodec_encode_audio(aavces->c,
> aavces->outbuf,aavces->outbuf_size,(int16_t*) aavces->fifo_outbuf);
>                 else
>                     encBytes = avcodec_encode_audio(aavces->c,
> aavces->outbuf,nfifoBytes, (int16_t*) aavces->fifo_outbuf);
>             }
>             aavces->c->frame_size = nFrameSizeTmp;// restore the native
> frame size
>         } else
>             printf("\n[audio encoder] codec does not support small
> frames");
>     }
>
>     // Now flush the encoder.
>     if (encBytes <= 0){
>         encBytes = avcodec_encode_audio(aavces->c,
> aavces->outbuf,aavces->outbuf_size, NULL );
>         printf("\nFFmpeg : flushing the encoder");
>     }
>     if (encBytes < 0) {
>         printf("\nFFmpeg : ERROR - Can't encode LAST audio frame.");
>     }
>     av_init_packet(&pkt);
>
>     sizeTmp = pkt.size;
>
>     pkt.size = encBytes;
>     pkt.data = aavces->outbuf;
>     pkt.flags |= AV_PKT_FLAG_KEY;
>
>     // Rescale from the codec time_base to the AVStream time_base.
>     if (aavces->c->coded_frame && aavces->c->coded_frame->pts != (int64_t)
> (AV_NOPTS_VALUE ))
>         pkt.pts =
> av_rescale_q(aavces->c->coded_frame->pts,aavces->c->time_base,
> aavces->c->time_base);
>
>     printf("\nFFmpeg : (%d) Writing audio frame with PTS:
> %lld.",aavces->c->frame_number, pkt.pts);
>     printf("\n[avcodec - audio - encode] Encoder returned %d bytes of
> data\n",pkt.size);
>
>     memcpy(outbuf + sizeTmp, pkt.data, pkt.size);
>
>     outsize = sizeTmp + pkt.size;
>
>     return outsize;
> }
>
> -------------------------------------------------->8-------------------------------------------------
>
>
> Then, I'm saving outbuf with outsize per frame encoded.
>
> Any idea of what I'm doing wrong?
>
> Thanks in advance!
> --------------------
>   Gerard C.L.
> --------------------
>
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[Libav-user] aac encoder in real time scenario

遇到和这个人一毛一样的问题声音中存在噪声其他的还好

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遇到和这个人一毛一样的问题声音中存在噪声其他的还好