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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.

  


• Revision 3826 : Pouvoir mettre le volume en cookie afin de pouvoir le réutiliser de pages ...

  20 août 2010, par kent1 — Log

  Pouvoir mettre le volume en cookie afin de pouvoir le réutiliser de pages en pages

• How to change the .ts file names in side the m3u8 file

  13 août 2020, par Isuru Bandara

  Recently I am using FFmpeg for a project. I used FFmpeg to convert mp3 bitrates and other stuff. Those are working perfectly. But now I want to add mp3%2Fmusic%2F[outputN.ts] ?alt=media for every outputN.ts texts inside the m3u8 file. This my m3u8 file data structure,

  


  ---

  


  #EXTM3U

  


  #EXT-X-VERSION:3

  


  #EXT-X-TARGETDURATION:2

  


  #EXT-X-MEDIA-SEQUENCE:0

  


  #EXTINF:2.005333,

  


  'output000.ts'

  


  #EXTINF:2.005333,

  


  'output001.ts'

  


  ---

  


  For Example, Now I want to apply mp3%2Fmusic%2F and ?alt=media for every text (output000.ts,output001.ts and so on) inside this m3u8 file using windows cmd. For instance,

  


  ---

  


  #EXTINF:2.005333,

  


  mp3%2Fmusic%2Foutput000.ts ?alt=media

  


  #EXTINF:2.005333,

  


  mp3%2Fmusic%2Foutput001.ts ?alt=media

  


  ---