Recherche avancée

Sur d’autres sites (4833)

• ffmpeg ndk multiple definition libavcodec.a(golomb.o)

  4 décembre 2014, par WLGfx

  I’m trying to link ffmpeg as static libraries with android NDK but I’m getting ’multiple definition’ error’ errors as below. I’ve also included my build script which runs through everything just fine but when I come to using the libraries in Eclipse with the ADT plugin, I can’t get anywhere.

  From this it looks like it wants something to do with VLC. I don’t want anything to do with VLC, just ffmpeg for video streaming. Everything works fine with shared libraries, but I’m after a very tiny player because I’m restricted to space on the device.

  EDIT : Also ’log2_tab_tab.o’ has multiple definitions.

  error: jni/libs\libavcodec.a(golomb.o): multiple definition of 'ff_golomb_vlc_len'  Ffplayer            C/C++ Problem
  
  error: jni/libs\libavcodec.a(golomb.o): multiple definition of 'ff_interleaved_dirac_golomb_vlc_code'   Ffplayer            C/C++ Problem
  
  error: jni/libs\libavcodec.a(golomb.o): multiple definition of 'ff_interleaved_golomb_vlc_len'  Ffplayer            C/C++ Problem
  
  error: jni/libs\libavcodec.a(golomb.o): multiple definition of 'ff_interleaved_se_golomb_vlc_code'  Ffplayer            C/C++ Problem
  
  error: jni/libs\libavcodec.a(golomb.o): multiple definition of 'ff_interleaved_ue_golomb_vlc_code'  Ffplayer            C/C++ Problem
  
  error: jni/libs\libavcodec.a(golomb.o): multiple definition of 'ff_se_golomb_vlc_code'  Ffplayer            C/C++ Problem
  
  error: jni/libs\libavcodec.a(golomb.o): multiple definition of 'ff_ue_golomb_len'   Ffplayer            C/C++ Problem
  
  error: jni/libs\libavcodec.a(golomb.o): multiple definition of 'ff_ue_golomb_vlc_code'  Ffplayer            C/C++ Problem
  
  error: jni/libs\libavcodec.a(log2_tab.o): multiple definition of 'ff_log2_tab'  Ffplayer            C/C++ Problem
  
  error: jni/libs\libavformat.a(log2_tab.o): multiple definition of 'ff_log2_tab' Ffplayer            C/C++ Problem
  
  jni/libs\libavformat.a(golomb_tab.o): previous definition here  Ffplayer            C/C++ Problem
  
  jni/libs\libavutil.a(log2_tab.o): previous definition here  Ffplayer            C/C++ Problem
  
  make.exe: *** [obj/local/armeabi-v7a-hard/libffplayer.so] Error 1   Ffplayer            C/C++ Problem

  Using the latest branch of ffmpeg (2.4.3) my build script for Android (using toolchain 8 because it’s old hardware I’m working with) and wanting the NEON hardware support :

  export ANDROID_NDK=/home/carl/dev/ndk
  
  export TOOLCHAIN=/home/carl/temp/ffmpeg
  
  export SYSROOT=$TOOLCHAIN/sysroot/
  
  $ANDROID_NDK/build/tools/make-standalone-toolchain.sh \
  
      --platform=android-8 --install-dir=$TOOLCHAIN
  
  
  
  export PATH=$TOOLCHAIN/bin:$PATH
  
  export CC=arm-linux-androideabi-gcc
  
  export LD=arm-linux-androideabi-ld
  
  export AR=arm-linux-androideabi-ar
  
  
  
  CFLAGS="-O3 -Wall -mthumb -pipe -fpic -fasm \
  
    -finline-limit=300 -ffast-math \
  
    -fstrict-aliasing -Werror=strict-aliasing \
  
    -fmodulo-sched -fmodulo-sched-allow-regmoves \
  
    -Werror=implicit-function-declaration \
  
    -Wno-psabi -Wa,--noexecstack"
  
  #  -D__ARM_ARCH_5__ -D__ARM_ARCH_5E__ \
  
  #  -D__ARM_ARCH_5T__ -D__ARM_ARCH_5TE__ \
  
  #  -DANDROID -DNDEBUG"
  
  
  
  EXTRA_CFLAGS="-march=armv7-a -mfpu=neon \
  
                -mfloat-abi=softfp -mvectorize-with-neon-quad \
  
                -DHAVE_ISNAN -DHAVE_ISINF
  
                -std=c99"
  
  EXTRA_LDFLAGS="-Wl,--fix-cortex-a8"
  
  
  
  FFMPEG_FLAGS="--prefix=/home/dev/ffmpeg/build \
  
    --target-os=linux \
  
    --arch=arm \
  
    --enable-cross-compile \
  
    --cross-prefix=arm-linux-androideabi- \
  
    --enable-shared \
  
    --enable-static \
  
    --enable-small \
  
    --disable-symver \
  
    --disable-doc \
  
    --disable-ffplay \
  
    --disable-ffmpeg \
  
    --disable-ffprobe \
  
    --disable-ffserver \
  
    --disable-avdevice \
  
    --disable-avfilter \
  
    --disable-encoders  \
  
    --disable-muxers \
  
    --disable-demuxers \
  
    --disable-filters \
  
    --disable-devices \
  
    --disable-decoders \
  
    --enable-decoder=mjpeg \
  
    --enable-decoder=mp1 \
  
    --enable-decoder=mp2 \
  
    --enable-decoder=mp3 \
  
    --enable-decoder=mpeg1_vdpau \
  
    --enable-decoder=mpeg1video \
  
    --enable-decoder=mpeg2video \
  
    --enable-decoder=mpeg4 \
  
    --enable-decoder=mpeg4_vdpau \
  
    --enable-decoder=mpegvideo \
  
    --enable-decoder=mpeg_xvmc \
  
    --enable-decoder=h261 \
  
    --enable-decoder=h263 \
  
    --enable-decoder=h263i \
  
    --enable-decoder=h263p \
  
    --enable-hwaccel=h263_vaapi \
  
    --enable-hwaccel=h263_vdpau \
  
    --enable-hwaccel=mpeg1_vdpau \
  
    --enable-hwaccel=mpeg1_xvmc \
  
    --enable-hwaccel=mpeg2_dxva2 \
  
    --enable-hwaccel=mpeg2_vaapi \
  
    --enable-hwaccel=mpeg2_vdpau \
  
    --enable-hwaccel=mpeg2_xvmc \
  
    --enable-hwaccel=mpeg4_vaapi \
  
    --enable-hwaccel=mpeg4_vdpau \
  
    --enable-demuxer=aac \
  
    --enable-demuxer=ac3 \
  
    --enable-demuxer=h261 \
  
    --enable-demuxer=h263 \
  
    --enable-demuxer=pcm_s16be \
  
    --enable-demuxer=pcm_s16le \
  
    --enable-demuxer=pcm_s8 \
  
    --enable-demuxer=mpegps \
  
    --enable-demuxer=mpegts \
  
    --enable-demuxer=mpegtsraw \
  
    --enable-demuxer=mpegvideo \
  
    --enable-demuxer=rtp \
  
    --enable-demuxer=rtsp \
  
    --enable-parser=aac \
  
    --enable-parser=mpegvideo \
  
    --enable-parser=ac3 \
  
    --enable-parser=h261 \
  
    --enable-parser=h263 \
  
    --enable-parser=mjpeg \
  
    --enable-parser=mpeg4video \
  
    --enable-parser=mpegaudio \
  
    --enable-protocol=rtp \
  
    --enable-protocol=file \
  
    --enable-protocol=ftp \
  
    --enable-protocol=tcp \
  
    --enable-protocol=http \
  
    --enable-protocol=udp \
  
    --enable-protocol=pipe \
  
    --enable-protocol=unix \
  
    --enable-network \
  
    --disable-swscale  \
  
    --enable-asm \
  
    --enable-memalign-hack \
  
    --disable-golomb \
  
    --enable-stripping \
  
    --enable-pthreads \
  
    --disable-symver \
  
    --enable-version3"
  
  
  
  ./configure $FFMPEG_FLAGS --extra-cflags="$CFLAGS $EXTRA_CFLAGS" \
  
    --extra-ldflags="$EXTRA_LDFLAGS"
  
  
  
  make clean
  
  echo "Project now cleaned"
  
  make -j4
  
  
  
  echo "Stripping multiple references from libraries"
  
  arm-linux-androideabi-ar d libavcodec.a log2_tab.o
  
  arm-linux-androideabi-ar d libavutil.a log2_tab.o
  
  
  
  echo "Done..."

  And this is the Android.mk file which works fine.

  LOCAL_PATH := $(call my-dir)
  
  
  
  include $(CLEAR_VARS)
  
  LOCAL_MODULE    := avutil
  
  LOCAL_SRC_FILES := libs\libavutil.a
  
  include $(PREBUILT_STATIC_LIBRARY)
  
  
  
  include $(CLEAR_VARS)
  
  LOCAL_MODULE    := avformat
  
  LOCAL_SRC_FILES := libs\libavformat.a
  
  include $(PREBUILT_STATIC_LIBRARY)
  
  
  
  include $(CLEAR_VARS)
  
  LOCAL_MODULE    := avcodec
  
  LOCAL_SRC_FILES := libs\libavcodec.a
  
  include $(PREBUILT_STATIC_LIBRARY)
  
  
  
  include $(CLEAR_VARS)
  
  
  
  LOCAL_MODULE    := ffplayer
  
  LOCAL_SRC_FILES := ffplayer.cpp
  
  
  
  LOCAL_C_INCLUDES := C:\DEV\ffmpeg\
  
  
  
  LOCAL_LDLIBS    += -llog -ljnigraphics -lGLESv2 -ldl
  
  LOCAL_LDLIBS    += -lstdc++ -lc
  
  LOCAL_LDLIBS    += -lz -lm
  
  
  
  LOCAL_WHOLE_STATIC_LIBRARIES += libavutil libavformat libavcodec
  
  
  
  include $(BUILD_SHARED_LIBRARY)

  If anybody can spot what’s wrong with this it would be so appreciated.

• 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


  


• How to convert H264 RTP stream from PCAP to a playable video file

  7 novembre 2017, par yoosha

  I have captured stream of H264 in PCAP files and trying to create media files from the data. The container is not important (avi,mp4,mkv,…).
  When I’m using videosnarf or rtpbreak (combined with python code that adds 00 00 00 01 before each packet) and then ffmpeg, the result is OK only if the input frame rate is constant (or near constant). However, when the input is vfr, the result plays too fast (and on same rare cases too slow).
  For example :

  videosnarf -i captured.pcap –c
  ffmpeg -i H264-media-1.264 output.avi

  After doing some investigation of the issue I believe now that since the videosnarf (and rtpbreak) are removing the RTP header from the packets, the timestamp is lost and ffmpeg is referring to the input data as cbr.

  1. I would like to know if there is a way to pass (on a separate file ?)
     the timestamps vector or any other information to ffmpeg so the
     result will be created correctly ?
  2. Is there any other way I can take the data out of the PCAP file and play it or convert it and then play it ?
  3. Since all work is done in Python, any suggestion of libraries/modules that can help with the work (even if requires some codding) is welcome as well.

  Note : All work is done offline, no limitations on the output. It can be cbr/vbr, any playable container and transcoding. The only "limitation" I have : it should all run on linux…

  Thanks
  Y

  Some additional information :
  Since the nothing provides the FFMPEG with the timestamp data, i decided to try a different approach : skip videosnarf and use Python code to pipe the packets directly to ffmpeg (using the "-f -i -" options) but then it refuses to accept it unless I provide an SDP file...
  How do I provide the SDP file ? is it an additional input file ? ("-i config.sdp")

  The following code is an unsuccessful try doing the above :

  import time  
  
  import sys  
  
  import shutil  
  
  import subprocess  
  
  import os  
  
  import dpkt  
  
  
  
  if len(sys.argv) < 2:  
  
      print "argument required!"  
  
      print "txpcap <pcap file="file">"  
  
      sys.exit(2)  
  
  pcap_full_path = sys.argv[1]  
  
  
  
  ffmp_cmd = ['ffmpeg','-loglevel','debug','-y','-i','109c.sdp','-f','rtp','-i','-','-na','-vcodec','copy','p.mp4']  
  
  
  
  ffmpeg_proc = subprocess.Popen(ffmp_cmd,stdout = subprocess.PIPE,stdin = subprocess.PIPE)  
  
  
  
  with open(pcap_full_path, "rb") as pcap_file:  
  
      pcapReader = dpkt.pcap.Reader(pcap_file)  
  
      for ts, data in pcapReader:  
  
          if len(data) &lt; 49:  
  
              continue  
  
          ffmpeg_proc.stdin.write(data[42:])
  
  
  
  sout, err = ffmpeg_proc.communicate()  
  
  print "stdout ---------------------------------------"  
  
  print sout  
  
  print "stderr ---------------------------------------"  
  
  print err  
  
  </pcap>

  In general this will pipe the packets from the PCAP file to the following command :

  ffmpeg -loglevel debug -y -i 109c.sdp -f rtp -i - -na -vcodec copy p.mp4

  SDP file : [RTP includes dynamic payload type # 109, H264]

  v=0
  o=- 0 0 IN IP4 ::1
  s=No Name
  c=IN IP4 ::1
  t=0 0
  a=tool:libavformat 53.32.100
  m=video 0 RTP/AVP 109
  a=rtpmap:109 H264/90000
  a=fmtp:109
  packetization-mode=1 ;profile-level-id=64000c ;sprop-parameter-sets=Z2QADKwkpAeCP6wEQAAAAwBAAAAFI8UKkg==,aMvMsiw= ;
  b=AS:200

  Results :

  ffmpeg version 0.10.2 Copyright (c) 2000-2012 the FFmpeg developers
  built on Mar 20 2012 04:34:50 with gcc 4.4.6 20110731 (Red Hat
  4.4.6-3) configuration : —prefix=/usr —libdir=/usr/lib64 —shlibdir=/usr/lib64 —mandir=/usr/share/man —enable-shared —enable-runtime-cpudetect —enable-gpl —enable-version3 —enable-postproc —enable-avfilter —enable-pthreads —enable-x11grab —enable-vdpau —disable-avisynth —enable-frei0r —enable-libopencv —enable-libdc1394 —enable-libdirac —enable-libgsm —enable-libmp3lame —enable-libnut —enable-libopencore-amrnb —enable-libopencore-amrwb —enable-libopenjpeg —enable-librtmp —enable-libschroedinger —enable-libspeex —enable-libtheora —enable-libvorbis —enable-libvpx —enable-libx264 —enable-libxavs —enable-libxvid —extra-cflags=’-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector —param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC’ —disable-stripping libavutil 51. 35.100 / 51. 35.100 libavcodec 53. 61.100 / 53. 61.100 libavformat 53. 32.100
  / 53. 32.100 libavdevice 53. 4.100 / 53. 4.100
  libavfilter 2. 61.100 / 2. 61.100 libswscale 2. 1.100
  / 2. 1.100 libswresample 0. 6.100 / 0. 6.100
  libpostproc 52. 0.100 / 52. 0.100 [sdp @ 0x15c0c00] Format sdp
  probed with size=2048 and score=50 [sdp @ 0x15c0c00] video codec set
  to : h264 [NULL @ 0x15c7240] RTP Packetization Mode : 1 [NULL @
  0x15c7240] RTP Profile IDC : 64 Profile IOP : 0 Level : c [NULL @
  0x15c7240] Extradata set to 0x15c78e0 (size : 36) !error,_recognition
  separate : 1 ; 1 [h264 @ 0x15c7240] error,_recognition combined : 1 ;
  10001 [sdp @ 0x15c0c00] decoding for stream 0 failed [sdp @
  0x15c0c00] Could not find codec parameters (Video : h264) [sdp @
  0x15c0c00] Estimating duration from bitrate, this may be inaccurate
  109c.sdp : could not find codec parameters Traceback (most recent
  call last) : File "./ffpipe.py", line 26, in
  ffmpeg_proc.stdin.write(data[42 :]) IOError : [Errno 32] Broken pipe

  (forgive the mass above, the editor keep on complaining about code that is not indented OK ??)

  I’m working on this issue for days... any help/suggestion/hint will be appreciated.