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• ffmpeg is buffering 10 seconds of data every 10 seconds, but I need 1 second of data every second

  30 juillet 2014, par user306517

  I am trying to render an URL, and when I use ffmpeg to do it, it is giving me 10 seconds of data, every 10 seconds. I need to get the data in a way that I am getting 1 second of data every second.

  it looks like the URL i am trying to read from is a "m3u8" url, which contains a bunch of m3u8 files, which contain .ts files (transport stream).

  any help here would be appreciated.

  per suggestion - my ffmpeg command looks like this :

  ffmpeg -y -i ".....m3u8" -ar 32000 -ac 1 -acodec pcm_s16le /tmp/foo.wav

• FFMPEG amix filter causes main audio stream to cut out

  21 janvier 2019, par user3600107

  So I have a video called 1.mkv and would like to mix in a variety of different audio clips at certain points. To do this I’m using the -filter_complex option. However, I’m running into some problems because when ffmpeg tries to mix in the first audio stream the audio works for a short while when the clip is playing and then all audio cuts out. I’m running ffmpeg version 2.8.15-0(which is up to date with my distro). Another "weird" thing about the video output is that in xplayer the video will freeze after the audio cuts out, and will work if you skip far enough ahead in the video(not sure if this is helpful but it might give some extra clues).

  Full command :

  ffmpeg -i "1.mkv" -i "5.wav" -i "2.wav" -i "3.wav" -i "6.wav" -i "7.wav" -i "4.wav" -i "9.wav" -i "8.wav" -i "10.wav" -filter_complex "[0:0]setdar=4/3[v0];
  
  
  
  [2:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,adelay=20000|20000,volume=0.5[ad2];
  
  
  
  [4:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,adelay=900000|900000,volume=0.5[ad4];
  
  
  
  [3:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,adelay=300000|300000,volume=0.5[ad3];
  
  
  
  [1:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,adelay=720000|720000,volume=0.5[ad1];
  
  
  
  [7:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,adelay=1140000|1140000,volume=0.5[ad7];
  
  
  
  [9:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,adelay=1260000|1260000,volume=0.5[ad9];
  
  
  
  [8:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,adelay=1020000|1020000,volume=0.5[ad8];
  
  
  
  [5:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,adelay=960000|960000,volume=0.5[ad5];
  
  
  
  [6:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,adelay=480000|480000,volume=0.5[ad6];
  
  [0:1]volume=1[ad0];
  
  [ad0][ad2][ad4][ad3][ad1][ad7][ad9][ad8][ad5][ad6]amix=inputs=10:duration=first:dropout_transition=0,dynaudnorm[a0]" -map "[v0]" -map "[a0]" -c:v libx264 -ar 44100  -c:a libmp3lame -preset ultrafast -crf 17 -b:v 1M out2.flv

  partial command

  ffmpeg -i "1.mkv" -i "2.wav" -filter_complex "[0:0]setdar=4/3[v0];
  
  [1:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,adelay=20000|20000,volume=0.5[ad2];
  
  [0:1]volume=1[ad0];
  
  [ad0][ad2]amix=inputs=2:duration=first:dropout_transition=0,dynaudnorm[a0]" -map "[v0]" -map "[a0]" -c:v libx264 -ar 44100  -c:a libmp3lame -preset ultrafast -crf 17 -b:v 1M out2.flv

• Reading mp3 file using ffmpeg caues memory leaks, even after freeing it in main

  12 août 2020, par leonardltk1

  i am continuously reading mp3 files and processing them, but the memory keeps getting build up even though i freed it.

  


  At the bottom read_audio_mp3(), they are already freeing some variable.
why do i still face a memory build up and how do i deal with it ?

  


  following this code : https://rodic.fr/blog/libavcodec-tutorial-decode-audio-file/, i read mp3 using this function

  


      int read_audio_mp3(string filePath_str, const int sample_rate, 
      double** output_buffer, int &AUDIO_DURATION){
      const char* path = filePath_str.c_str();

      /* Reads the file header and stores information about the file format. */
        AVFormatContext* format = avformat_alloc_context();
        if (avformat_open_input(&format, path, NULL, NULL) != 0) {
            fprintf(stderr, "Could not open file '%s'\n", path);
            return -1;
        }

      /* Check out the stream information in the file. */
        if (avformat_find_stream_info(format, NULL) < 0) {
            fprintf(stderr, "Could not retrieve stream info from file '%s'\n", path);
            return -1;
        }

      /* find an audio stream. */
        int stream_index =- 1;
        for (unsigned i=0; inb_streams; i++) {
          if (format->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
            stream_index = i;
            break;
          }
        }
        if (stream_index == -1) {
            fprintf(stderr, "Could not retrieve audio stream from file '%s'\n", path);
            return -1;
        }
        AVStream* stream = format->streams[stream_index];

      // find & open codec
        AVCodecContext* codec = stream->codec;
        if (avcodec_open2(codec, avcodec_find_decoder(codec->codec_id), NULL) < 0) {
            fprintf(stderr, "Failed to open decoder for stream #%u in file '%s'\n", stream_index, path);
            return -1;
        }

      // prepare resampler
        struct SwrContext* swr = swr_alloc();
        av_opt_set_int(swr, "in_channel_count",  codec->channels, 0);
        av_opt_set_int(swr, "out_channel_count", 1, 0);
        av_opt_set_int(swr, "in_channel_layout",  codec->channel_layout, 0);
        av_opt_set_int(swr, "out_channel_layout", AV_CH_LAYOUT_MONO, 0);
        av_opt_set_int(swr, "in_sample_rate", codec->sample_rate, 0);
        av_opt_set_int(swr, "out_sample_rate", sample_rate, 0);
        av_opt_set_sample_fmt(swr, "in_sample_fmt",  codec->sample_fmt, 0);
        av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_DBL,  0);
        swr_init(swr);
        if (!swr_is_initialized(swr)) {
            fprintf(stderr, "Resampler has not been properly initialized\n");
            return -1;
        }

      /* Allocate an audio frame. */
        AVPacket packet;
        av_init_packet(&packet);
        AVFrame* frame = av_frame_alloc();
        if (!frame) {
          fprintf(stderr, "Error allocating the frame\n");
          return -1;
        }

      // iterate through frames
        *output_buffer = NULL;
        AUDIO_DURATION = 0;
        while (av_read_frame(format, &packet) >= 0) {
          // decode one frame
            int gotFrame;
            if (avcodec_decode_audio4(codec, frame, &gotFrame, &packet) < 0) {
              // free packet
                av_free_packet(&packet);
                break;
            }
            if (!gotFrame) {
              // free packet
                av_free_packet(&packet);
                continue;
            }
          // resample frames
            double* buffer;
            av_samples_alloc((uint8_t**) &buffer, NULL, 1, frame->nb_samples, AV_SAMPLE_FMT_DBL, 0);
            int frame_count = swr_convert(swr, (uint8_t**) &buffer, frame->nb_samples, (const uint8_t**) frame->data, frame->nb_samples);
          // append resampled frames to output_buffer
            *output_buffer = (double*) realloc(*output_buffer,
             (AUDIO_DURATION + frame->nb_samples) * sizeof(double));
            memcpy(*output_buffer + AUDIO_DURATION, buffer, frame_count * sizeof(double));
            AUDIO_DURATION += frame_count;
          // free buffer & packet
            av_free_packet(&packet);
            av_free( buffer );
        }

      // clean up
        av_frame_free(&frame);
        swr_free(&swr);
        avcodec_close(codec);
        avformat_free_context(format);

      return 0;
    }


  


  Main Script : MemoryLeak.cpp

  


      // imports&#xA;      #include <fstream>&#xA;      #include &#xA;      #include &#xA;      #include &#xA;      #include &#xA;      #include <iostream>&#xA;      #include <sstream>&#xA;      #include <vector>&#xA;      #include <sys></sys>time.h> &#xA;      extern "C"&#xA;      {&#xA;      #include <libavutil></libavutil>opt.h>&#xA;      #include <libavcodec></libavcodec>avcodec.h>&#xA;      #include <libavformat></libavformat>avformat.h>&#xA;      #include <libswresample></libswresample>swresample.h>&#xA;      }&#xA;      using namespace std;&#xA;&#xA;    int main (int argc, char ** argv) {&#xA;      string wavpath = argv[1];&#xA;      printf("wavpath=%s\n", wavpath.c_str());&#xA;&#xA;      printf("\n==== Params =====\n");&#xA;      // Init&#xA;        int AUDIO_DURATION;&#xA;        int sample_rate = 8000;&#xA;        av_register_all();&#xA;&#xA;      printf("\n==== Reading MP3 =====\n");&#xA;        while (true) {&#xA;            // Read mp3&#xA;              double* buffer;&#xA;              if (read_audio_mp3(wavpath, sample_rate, &amp;buffer, AUDIO_DURATION) != 0) {&#xA;                printf("Cannot read %s\n", wavpath.c_str());&#xA;                continue;&#xA;              }&#xA;&#xA;            /* &#xA;              Process the buffer for down stream tasks.&#xA;            */&#xA;&#xA;            // Freeing the buffer&#xA;            free(buffer);&#xA;        }&#xA;&#xA;      return 0 ;&#xA;    }&#xA;</vector></sstream></iostream></fstream>

  


  Compiling

  


      g++ -o ./MemoryLeak.out -Ofast -Wall -Wextra \
        -std=c++11 "./MemoryLeak.cpp" \
        -lavformat -lavcodec -lavutil -lswresample


  


  Running, by right my input an argument wav.scp that reads text file of all the mp3s.
But for easy to replicate purpose, i only read 1 file song.mp3 in and i keep re-reading it

  


  ./MemoryLeak.out song.mp3


  


  Why do i know i have memory leaks ?

  


  

  1. I was running up 32 jobs in parallel for 14 million files, and when i wake up in the morning, they were abruptly killed.

  


  2. I run htop and i monitor the progress when i re-run it, and i saw that the VIRT & RES & Mem are continuously increasing.

  


  


  


  Edit 1 :
My setup :

  


  


  ffmpeg version 2.8.15-0ubuntu0.16.04.1
built with gcc 5.4.0