[Libav-user] libavcodec vorbis encoding

[Александр Рухлов]

Thanks for the help
I tried to write in normal way to the container
FLAC to Ogg, MP2 to Wav - works without problems, Vorbis to Ogg - writes
garbage
in the container the distorted data are located
https://dl.dropbox.com/u/2114502/flac.ogg
https://dl.dropbox.com/u/2114502/vorbis.ogg
maybe this codec has any specific parameters ???

static void audio_encode_example(const char *filename)
{
    AVFormatContext* out_format_context;
    AVStream * audio_stream ;

    AVCodec *codec;
    AVCodecContext *c= NULL;
    AVFrame *frame;
    AVPacket pkt;
    int i, j, k, ret, got_output;
    int buffer_size;
    FILE *f;
    uint16_t *samples;
    float t, tincr;

    printf("Encode audio file %s\n", filename);

    out_format_context = avformat_alloc_context();
    out_format_context->oformat = av_guess_format(NULL, filename, NULL);

    if (out_format_context->oformat == NULL){
        fprintf(stderr, "Could not guess output format\n");
        exit(1);
    }
    audio_stream = av_new_stream(out_format_context, 0);

    if (!audio_stream) {
        fprintf(stderr, "Could not alloc audio stream!");
        exit(1);
    }

    /* find encoder */
    //codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
    codec = avcodec_find_encoder(AV_CODEC_ID_VORBIS);
    //codec = avcodec_find_encoder(AV_CODEC_ID_FLAC);

    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }

    //c = avcodec_alloc_context3(codec);
    c=audio_stream->codec;

    c->bit_rate = 64000;

    c->sample_fmt = *codec->sample_fmts;

    c->sample_rate    = select_sample_rate(codec);
    c->channel_layout = select_channel_layout(codec);
    c->channels       =
av_get_channel_layout_nb_channels(c->channel_layout);

    if(out_format_context->oformat->flags & AVFMT_GLOBALHEADER)
        c->flags |= CODEC_FLAG_GLOBAL_HEADER;

    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }
    if(avio_open(&out_format_context->pb, filename, AVIO_FLAG_WRITE)<0){
            fprintf(stderr, "Error occurred when opening output file");
            exit(1);
        }

    if (avformat_write_header(out_format_context, NULL) < 0) {
            fprintf(stderr, "Error occurred when opening output file");
            exit(1);
    }

    frame = avcodec_alloc_frame();
    if (!frame) {
        fprintf(stderr, "Could not allocate audio frame\n");
        exit(1);
    }

    frame->nb_samples     = c->frame_size;
    frame->format         = c->sample_fmt;
    frame->channel_layout = c->channel_layout;

    buffer_size = av_samples_get_buffer_size(NULL, c->channels,
c->frame_size,
                                             c->sample_fmt, 0);
    samples = (uint16_t*)av_malloc(buffer_size);
    if (!samples) {
        fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
                buffer_size);
        exit(1);
    }

    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
                                   (const uint8_t*)samples, buffer_size, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not setup audio frame\n");
        exit(1);
    }
    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
    for(i=0;i<200;i++) {
        av_init_packet(&pkt);
        pkt.data = NULL;
        pkt.size = 0;

        for (j = 0; j < c->frame_size; j++) {
            samples[2*j] = (int)(sin(t) * 10000);

            for (k = 1; k < c->channels; k++)
                samples[2*j + k] = samples[2*j];
            t += tincr;
        }

        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding audio frame\n");
            exit(1);
        }
        if (got_output) {
            pkt.pts=pkt.dts=AV_NOPTS_VALUE;
            //pkt.pts=pkt.dts=time;
            int ret = av_interleaved_write_frame(out_format_context, &pkt);
            if (ret < 0) {
                fprintf(stderr,"Error while write audio");
            }
            //fwrite(pkt.data, 1, pkt.size, f);
            av_free_packet(&pkt);
        }
    }

    for (got_output = 1; got_output; i++) {
        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
        if (ret < 0) {
            fprintf(stderr, "Error encoding frame\n");
            exit(1);
        }

        if (got_output) {
            pkt.pts=pkt.dts=AV_NOPTS_VALUE;
            int ret = av_interleaved_write_frame(out_format_context, &pkt);
            if (ret < 0) {
                fprintf(stderr,"Error while write audio");
            }
            //fwrite(pkt.data, 1, pkt.size, f);
            av_free_packet(&pkt);
        }
    }
    av_write_trailer(out_format_context);
    avio_close(out_format_context->pb);

    av_freep(&samples);
    av_freep(&frame);
    avcodec_close(c);
    av_free(c);
}

int main(int argc, char **argv)
{
    //avcodec_register_all();
    av_register_all();
    //audio_encode_example("test.wav");
    audio_encode_example("test.ogg");
    printf("Encoding is completed...\n");
    getch();
    return 0;
}

2013/1/10 Carl Eugen Hoyos <cehoyos at ag.or.at>

> Александр Рухлов <A.Rukhlov at ...> writes:
>
> > Signal of 440 Hz it is coded and written to the container.
> > AAC, MP2, AC3 works without problems.BUT categorically
> > Vorbis doesn't want to workfunction avcodec_encode_audio2()
> > works without mistakes and even something returns, but
> > these data are damaged
>
> Is it possible that you are trying to write
> a raw vorbis file (as opposed to muxing
> vorbis in a container format)?
> I don't think raw vorbis is defined.
> (And generally, you should never "fwrite()"
> what the encoder returns, but feed a muxer,
> this can be the adts, mp2 or ac3 muxer if
> you want raw files.)
>
> Carl Eugen
>
> _______________________________________________
> Libav-user mailing list
> Libav-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/libav-user
>
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