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• 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


  


• Add metadata while converting MKV to MP3 with FFMPEG

  17 août 2020, par Elderhoard

  I am trying to convert MKV files to MP3, while adding metadata and album artwork through a batch file. I am generating a PNG through FFMPEG, then converting to MP3 while adding metadata, and finally adding the album artwork i got initially.

  


  I have tried adding the metadata while converting to MP3 and while adding the artwork to no avail. I read something about it flushing the buffer too quickly but thought I might be able to get around it by adding it while converting it.

  


  Individually, all parts work, but I can't get it to add the title and artist to the metadata, or at least in a place where VLC can read it. any suggestions ?

  


  @echo off
::Extracts a PNG thumbnail 
for %%A in ("*.mkv") do (ffmpeg -ss 30 -i "%%A" -qscale:v 4 -frames:v 1 "%%~nA.png")

::Convert from MKV to MP3 and adds title and artist based on file name delimited by "-" eg Metallica - Enter Sandman.mkv
SETLOCAL ENABLEDELAYEDEXPANSION
for %%A in ("*.mkv") do (
    set filename=%%~nA
    set artist=
    set song=
    echo "!filename!"

    for /F "tokens=1,2 delims=-" %%G in ("!filename!") do (
        set artist="%%G"
        set song="%%H"
        echo !artist!
        echo !song!
    )

    echo !song! by !artist!

    ffmpeg -i "%%A" -b:a 192K -id3v2_version 4 -write_id3v2 1 -metadata title="%song%" -metadata artist="%artist%" -flush_packets 0 -vn "%%~nA.mp3"
)

::Add Artwork to MP3
for %%A in ("*.mp3") do (ffmpeg -i "%%A" -i "%%~nA.png" -map 0:0 -map 1:0 -c copy -id3v2_version 3 "UPDATED%%~nA.mp3")


  


• Trouble with converting webm into mp3 with pydub in python

  15 août 2020, par rc_marty

  so basically I want to convert song what I downloaded from youtube in webm and convert to into mp3

  


  when I wanted export song just with song.export("neco.mp3") it didn't work too

  


  I have in workfolder ffmpeg.exe and ffprobe.exe

  


  here is the code

  


  from pydub import AudioSegment

song = AudioSegment.from_file(downloaded.webm,"webm")
print("Loaded")
song.export("neco.mp3", format="mp3", bitrate="320k")
print("Converted and saved")


  


  here is the output of the console

  


  Loaded&#xA;Traceback (most recent call last):&#xA;  File "e:/martan/projekty/Python/programek na pisnicky/songDownloader.py", line 188, in <module>&#xA;    song.export("neco.mp3", format="mp3", bitrate="320k")&#xA;  File "C:\Users\BIBRAIN\AppData\Local\Programs\Python\Python38\lib\site-packages\pydub\audio_segment.py", line 911, in export&#xA;    raise CouldntEncodeError(&#xA;pydub.exceptions.CouldntEncodeError: Encoding failed. ffmpeg/avlib returned error code: 1&#xA;&#xA;Command:[&#x27;ffmpeg&#x27;, &#x27;-y&#x27;, &#x27;-f&#x27;, &#x27;wav&#x27;, &#x27;-i&#x27;, &#x27;C:\\Users\\BIBRAIN\\AppData\\Local\\Temp\\tmpo20ooz_z&#x27;, &#x27;-b:a&#x27;, &#x27;320k&#x27;, &#x27;-f&#x27;, &#x27;mp3&#x27;, &#x27;C:\\Users\\BIBRAIN\\AppData\\Local\\Temp\\tmpiqpl57g7&#x27;]&#xA;&#xA;Output from ffmpeg/avlib:&#xA;&#xA;ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers&#xA;  built with gcc 10.2.1 (GCC) 20200726&#xA;  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libgsm --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf&#xA;  libavutil      56. 51.100 / 56. 51.100&#xA;  libavcodec     58. 91.100 / 58. 91.100&#xA;  libavformat    58. 45.100 / 58. 45.100&#xA;  libavdevice    58. 10.100 / 58. 10.100&#xA;  libavfilter     7. 85.100 /  7. 85.100&#xA;  libswscale      5.  7.100 /  5.  7.100&#xA;  libswresample   3.  7.100 /  3.  7.100&#xA;  libpostproc    55.  7.100 / 55.  7.100&#xA;Guessed Channel Layout for Input Stream #0.0 : stereo&#xA;Input #0, wav, from &#x27;C:\Users\BIBRAIN\AppData\Local\Temp\tmpo20ooz_z&#x27;:&#xA;  Duration: 00:03:54.71, bitrate: 3072 kb/s&#xA;    Stream #0:0: Audio: pcm_s32le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s32, 3072 kb/s&#xA;Stream mapping:&#xA;  Stream #0:0 -> #0:0 (pcm_s32le (native) -> mp3 (mp3_mf))&#xA;Press [q] to stop, [?] for help&#xA;[mp3_mf @ 00000000004686c0] could not find any MFT for the given media type&#xA;[mp3_mf @ 00000000004686c0] could not create MFT&#xA;Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height&#xA;Conversion failed!&#xA;</module>

  


  I think it is something with codec but I have no idea what to do