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• How to adjust audio volume with AVFilter API from FFmpeg in Linux ?

  20 décembre 2024, par wangt13

  I am working on an embedded Linux system (kernel-5.10.24), and developing an audio player by using FFmpeg and ALSA libraries.

  


  Now the basic functions are done, and I can play MP3 file by using AVCodec and SWresample interfaces of FFmpeg.

  


  int decode_play(void)
{
    .....
    while (1) {
        if (av_read_frame(pfmtctx, packet) < 0) {
            avcodec_flush_buffers(pcodectx);
            printf("Got end of media, breaking\n");
            break;
        }
        if (packet->stream_index != stream) {
            continue;
        }
        int res = avcodec_send_packet(pcodectx, packet);
        if (res < 0) {
            printf("Error in decoding audio frame.\n");
            break;
        }

        while (res >= 0) {
            res = avcodec_receive_frame(pcodectx, pframe);
            if (res == AVERROR(EAGAIN)) {
                break;
            } else if (res == AVERROR_EOF) {
                break;
            } else if (res >= 0) {
                int outsamples = swr_convert(swrctx, &buf, pcodectx->frame_size, (const uint8_t **)pframe->data, pframe->nb_samples);
                snd_pcm_write(pcm_handle, buf, outsamples);
            }
        }
    }
}


  


  Now, I want to add volume changing to this audio player.
  
I found that there is AVfilter functions in FFmpeg, which can be used to adjust audio's volume. How can I use it in current design ?
I am not sure what filters should I use, and where should I plugin the filters ?
Should I put the filters before swr_convert or after the swr_convert ?

  


  With Erdal's help, I changed the player as follows,

  


  int decode_play(void)
{
    /* Init filter graph as filter_audio.c does */
    init_filter_graph(&graph, &src, &sink);

    while (1) {
        if (av_read_frame(pfmtctx, packet) < 0) {
            avcodec_flush_buffers(pcodectx);
            printf("Got end of media, breaking\n");
            break;
        }
        if (packet->stream_index != stream) {
            continue;
        }
        int res = avcodec_send_packet(pcodectx, packet);
        if (res < 0) {
            printf("Error in decoding audio frame.\n");
            break;
        }

        while (res >= 0) {
            res = avcodec_receive_frame(pcodectx, pframe);
            if (res == AVERROR(EAGAIN)) {
                break;
            } else if (res == AVERROR_EOF) {
                break;
            } else if (res >= 0) {
                int ret = av_buffersrc_add_frame(src, pframe);
                if (ret < 0) {
                    continue;
                }
                while ((ret = av_buffersink_get_frame(sink, filt_frame)) >= 0) {
                    int nb_samples = av_rescale_rnd(filt_frame->nb_samples, RESAMPLE_SAMPLE_RATE, filt_frame->sample_rate, AV_ROUND_UP);
                    printf("nb_samples: %d, f.nb_sample: %d, f.sr: %d\n", nb_samples, filt_frame->nb_samples, filt_frame->sample_rate);

                    int outs = snd_pcm_writei(playback_handle, filt_frame->extended_data[0], nb_samples);
                    if (outs < 0) {
                        snd_pcm_prepare(playback_handle);
                    }
                }
            }
        }
    }
}


  


  The original audio in format of 44100 Hz, stereo, fltp. And I chose to use aformat to do resampling instead of swresample as follows.

  


      snprintf(options_str, sizeof(options_str),
            "sample_fmts=%s:sample_rates=%d:channel_layouts=stereo",
            av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 32000);
    if (avfilter_init_str(aformat_ctx, options_str) < 0) {
        printf("error init aformat filter");
        return -1;
    }


  


  I succeeded playing the resampled audio with Avfilter.

  


• avcodec/svq1dec : Add assert to ensure "stages >= 0"

  3 février 2015, par Michael Niedermayer

  avcodec/svq1dec : Add assert to ensure "stages >= 0"
  This is currently always true, the assert protects against
  
  future changes to the code breaking this assumtation
  Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
  

  • [DH] libavcodec/svq1dec.c

• lavfi : Move new field to the end of AVFilterLink

  29 juin 2016, par Timo Rothenpieler

  lavfi : Move new field to the end of AVFilterLink
  Even though this is not part of the public API, some external
  
  applications access fields after it, thus breaking after updating from
  
  ffmpeg 3.0 or earlier.
  
  Since it is not public, it can be freely moved to the end to avoid
  
  that problem in the future.
  Reviewed-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavfilter/avfilter.h