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Autres articles (58)

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    This page lists some websites based on MediaSPIP.

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    9 novembre 2010, par

    MediaSPIP Core fournit par défaut trois pages différentes de configuration (ces pages utilisent le plugin de configuration CFG pour fonctionner) : une page spécifique à la configuration générale du squelettes ; une page spécifique à la configuration de la page d’accueil du site ; une page spécifique à la configuration des secteurs ;
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Sur d’autres sites (10298)

  • Evolution #3928 : les emails sont publics dans le privé

    28 juin 2019, par jluc -

    À défaut de ne pas diffuser la liste des auteurs avec leurs mails dans le core, on le propose dans le plugin autorite :
    cf https://zone.spip.net/trac/spip-zone/changeset/115819/spip-zone

  • ffmpeg rotate and resize

    30 mai 2019, par user1321964

    I’m trying to resize a movie and to rotate it counter-clockwise using ffmpeg.
    I can achieve one or the other ok individually

    ffmpeg -i MVI_8189.MOV -vf "transpose=2" -ss 00:00:12 -t 00:00:05 rotate.MOV

    OR

    ffmpeg -i MVI_8189.MOV -s 720x406 -ss 00:00:12 -t 00:00:05 resize.MOV

    but if I try to do both, either-

    ffmpeg -i MVI_8189.MOV -s 720x406 -ss 00:00:12 -t 00:00:05 resize.MOV

    ffmpeg -i resize.MOV -vf "transpose=2" toutp.MOV

    OR

    ffmpeg -i MVI_8189.MOV -vf "transpose=2" -ss 00:00:12 -t 00:00:05 rotate.MOV

    ffmpeg -i rotate.MOV -s 406x720 toutp.MOV

    the final movie does not play, only audio, no video ; neither previewing in Finder, nor in Quicktime. More seriously once I have attempted to play it, I can’t then play any other movie whatsoever (only audio) - quicktime won’t quit, constantly need to force-quit it. The only way to then restore seems to be reboot OS-X

    This though does work -

    ffmpeg -i MVI_8189.MOV -vf "transpose=2" -ss 00:00:12 -t 00:00:05 rotate.MOV

    ffmpeg -i rotate.MOV -s 720x406 toutp.MOV

    but doesn’t do what I want - it stretches the movie in the wrong direction

    Any ideas ?

    ffmpeg version 1.0.git Copyright (c) 2000-2012 the FFmpeg developers
    built on Dec 24 2012 18:43:25 with llvm-gcc 4.2.1 (LLVM build 2336.1.00)

    MAC OSX 10.7.5

    ==========
    output from

    ffmpeg -i MVI_8189.MOV -vf "transpose=2" -ss 00:00:12 -t 00:00:05 rotate.MOV
    ffmpeg -i rotate.MOV -s 406x720  toutp.MOV



    ffmpeg version 1.0.git Copyright (c) 2000-2012 the FFmpeg developers
     built on Dec 24 2012 18:43:25 with llvm-gcc 4.2.1 (LLVM build 2336.1.00)
     configuration: --prefix=/usr/local --enable-shared --disable-mmx --enable-libmp3lame --enable-gpl --enable-zlib --enable-libvorbis --enable-libfaac --enable-nonfree --enable-libx264 --enable-pthreads
     libavutil      52. 12.100 / 52. 12.100
     libavcodec     54. 81.100 / 54. 81.100
     libavformat    54. 49.102 / 54. 49.102
     libavdevice    54.  3.102 / 54.  3.102
     libavfilter     3. 29.101 /  3. 29.101
     libswscale      2.  1.103 /  2.  1.103
     libswresample   0. 17.102 /  0. 17.102
     libpostproc    52.  2.100 / 52.  2.100
    Guessed Channel Layout for  Input Stream #0.1 : stereo
    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'MVI_8189.MOV':
     Metadata:
       major_brand     : qt  
       minor_version   : 537331968
       compatible_brands: qt  CAEP
       creation_time   : 2014-08-17 23:13:43
     Duration: 00:01:32.40, start: 0.000000, bitrate: 44677 kb/s
       Stream #0:0(eng): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuvj420p, 1920x1080, 43132 kb/s, 25 fps, 25 tbr, 25k tbn, 50k tbc
       Metadata:
         creation_time   : 2014-08-17 23:13:43
       Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, s16, 1536 kb/s
       Metadata:
         creation_time   : 2014-08-17 23:13:43
    [libx264 @ 0x7fc67c043600] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX
    [libx264 @ 0x7fc67c043600] profile High, level 4.0
    [libx264 @ 0x7fc67c043600] 264 - core 128 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
    Output #0, mov, to 'rotate.MOV':
     Metadata:
       major_brand     : qt  
       minor_version   : 537331968
       compatible_brands: qt  CAEP
       encoder         : Lavf54.49.102
       Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), yuvj420p, 1080x1920, q=-1--1, 12800 tbn, 25 tbc
       Metadata:
         creation_time   : 2014-08-17 23:13:43
       Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, s16, 128 kb/s
       Metadata:
         creation_time   : 2014-08-17 23:13:43
    Stream mapping:
     Stream #0:0 -> #0:0 (h264 -> libx264)
     Stream #0:1 -> #0:1 (pcm_s16le -> libfaac)
    Press [q] to stop, [?] for help
    frame=  125 fps=5.7 q=-2.0 Lsize=    6394kB time=00:00:05.00 bitrate=10470.5kbits/s    /s    
    video:6322kB audio:67kB subtitle:0 global headers:0kB muxing overhead 0.079207%
    [libx264 @ 0x7fc67c043600] frame I:1     Avg QP:25.87  size:113032
    [libx264 @ 0x7fc67c043600] frame P:65    Avg QP:25.78  size: 64251
    [libx264 @ 0x7fc67c043600] frame B:59    Avg QP:26.73  size: 37017
    [libx264 @ 0x7fc67c043600] consecutive B-frames:  5.6% 94.4%  0.0%  0.0%
    [libx264 @ 0x7fc67c043600] mb I  I16..4: 10.7% 79.9%  9.3%
    [libx264 @ 0x7fc67c043600] mb P  I16..4:  3.6% 12.0%  0.3%  P16..4: 52.0% 19.1% 11.3%  0.0%  0.0%    skip: 1.7%
    [libx264 @ 0x7fc67c043600] mb B  I16..4:  0.9%  1.7%  0.0%  B16..8: 45.4%  5.1%  0.8%  direct:24.2%  skip:21.9%  L0:50.1% L1:44.5% BI: 5.3%
    [libx264 @ 0x7fc67c043600] 8x8 transform intra:74.5% inter:79.5%
    [libx264 @ 0x7fc67c043600] coded y,uvDC,uvAC intra: 59.4% 94.3% 79.5% inter: 38.2% 80.5% 30.1%
    [libx264 @ 0x7fc67c043600] i16 v,h,dc,p: 30% 21% 22% 28%
    [libx264 @ 0x7fc67c043600] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 18% 14% 24%  6%  8%  9%  7%  8%  7%
    [libx264 @ 0x7fc67c043600] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 28% 18% 17%  6%  7%  8%  6%  6%  4%
    [libx264 @ 0x7fc67c043600] i8c dc,h,v,p: 60% 14% 18%  8%
    [libx264 @ 0x7fc67c043600] Weighted P-Frames: Y:0.0% UV:0.0%
    [libx264 @ 0x7fc67c043600] ref P L0: 43.9% 12.6% 26.8% 16.7%
    [libx264 @ 0x7fc67c043600] ref B L0: 68.6% 31.4%
    [libx264 @ 0x7fc67c043600] kb/s:10357.28
    ffmpeg version 1.0.git Copyright (c) 2000-2012 the FFmpeg developers
     built on Dec 24 2012 18:43:25 with llvm-gcc 4.2.1 (LLVM build 2336.1.00)
     configuration: --prefix=/usr/local --enable-shared --disable-mmx --enable-libmp3lame --enable-gpl --enable-zlib --enable-libvorbis --enable-libfaac --enable-nonfree --enable-libx264 --enable-pthreads
     libavutil      52. 12.100 / 52. 12.100
     libavcodec     54. 81.100 / 54. 81.100
     libavformat    54. 49.102 / 54. 49.102
     libavdevice    54.  3.102 / 54.  3.102
     libavfilter     3. 29.101 /  3. 29.101
     libswscale      2.  1.103 /  2.  1.103
     libswresample   0. 17.102 /  0. 17.102
     libpostproc    52.  2.100 / 52.  2.100
    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'rotate.MOV':
     Metadata:
       major_brand     : qt  
       minor_version   : 512
       compatible_brands: qt  
       encoder         : Lavf54.49.102
     Duration: 00:00:05.01, start: 0.010667, bitrate: 10446 kb/s
       Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuvj420p, 1080x1920, 10358 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc
       Metadata:
         handler_name    : DataHandler
       Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 109 kb/s
       Metadata:
         handler_name    : DataHandler
    [libx264 @ 0x7fcf1c05ba00] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX
    [libx264 @ 0x7fcf1c05ba00] profile High, level 3.0
    [libx264 @ 0x7fcf1c05ba00] 264 - core 128 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
    Output #0, mov, to 'toutp.MOV':
     Metadata:
       major_brand     : qt  
       minor_version   : 512
       compatible_brands: qt  
       encoder         : Lavf54.49.102
       Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), yuvj420p, 406x720, q=-1--1, 12800 tbn, 25 tbc
       Metadata:
         handler_name    : DataHandler
       Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, s16, 128 kb/s
       Metadata:
         handler_name    : DataHandler
    Stream mapping:
     Stream #0:0 -> #0:0 (h264 -> libx264)
     Stream #0:1 -> #0:1 (aac -> libfaac)
    Press [q] to stop, [?] for help
    frame=  125 fps= 32 q=-2.0 Lsize=     469kB time=00:00:05.01 bitrate= 766.2kbits/s    
    video:396kB audio:68kB subtitle:0 global headers:0kB muxing overhead 1.088481%
    [libx264 @ 0x7fcf1c05ba00] frame I:1     Avg QP:24.20  size: 31395
    [libx264 @ 0x7fcf1c05ba00] frame P:42    Avg QP:24.51  size:  7137
    [libx264 @ 0x7fcf1c05ba00] frame B:82    Avg QP:27.71  size:   893
    [libx264 @ 0x7fcf1c05ba00] consecutive B-frames:  0.8% 16.0% 57.6% 25.6%
    [libx264 @ 0x7fcf1c05ba00] mb I  I16..4:  7.0% 62.2% 30.8%
    [libx264 @ 0x7fcf1c05ba00] mb P  I16..4:  0.2%  0.8%  0.1%  P16..4: 52.8% 18.7% 12.5%  0.0%  0.0%    skip:14.9%
    [libx264 @ 0x7fcf1c05ba00] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8: 30.0%  0.8%  0.1%  direct: 0.9%  skip:68.2%  L0:36.0% L1:62.1% BI: 2.0%
    [libx264 @ 0x7fcf1c05ba00] 8x8 transform intra:66.4% inter:78.3%
    [libx264 @ 0x7fcf1c05ba00] coded y,uvDC,uvAC intra: 79.3% 93.6% 84.9% inter: 13.6% 32.2% 10.4%
    [libx264 @ 0x7fcf1c05ba00] i16 v,h,dc,p: 50% 19%  7% 24%
    [libx264 @ 0x7fcf1c05ba00] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 20% 20% 11%  5%  8% 10%  9%  7%  8%
    [libx264 @ 0x7fcf1c05ba00] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 27% 22%  8%  5%  8%  8% 10%  5%  5%
    [libx264 @ 0x7fcf1c05ba00] i8c dc,h,v,p: 42% 22% 24% 13%
    [libx264 @ 0x7fcf1c05ba00] Weighted P-Frames: Y:0.0% UV:0.0%
    [libx264 @ 0x7fcf1c05ba00] ref P L0: 48.7% 16.3% 25.6%  9.4%
    [libx264 @ 0x7fcf1c05ba00] ref B L0: 73.7% 21.3%  5.0%
    [libx264 @ 0x7fcf1c05ba00] ref B L1: 90.7%  9.3%
    [libx264 @ 0x7fcf1c05ba00] kb/s:647.04
  • ffmpeg : Trying to access Ebur128Context->integrated_loudness but unsuccessful

    12 avril 2019, par Sourabh Jain

    [FFMPEG] Trying to access Ebur128Context->integrated_loudness but unsuccessful

    I am trying to run ebur128Filter on audio file . similar to be doing
    [http://ffmpeg.org/doxygen/2.6/f__ebur128_8c_source.html#l00135]

    ffmpeg -i sample.wav -filter_complex ebur128=peak=true -f null -

    result of which is :

    [Parsed_ebur128_0 @ 0x7f9d38403ec0] Summary:

    Integrated loudness:
    I: -15.5 LUFS
    Threshold: -25.6 LUFS

    Loudness range:
    LRA: 1.5 LU
    Threshold: -35.5 LUFS
    LRA low: -16.3 LUFS
    LRA high: -14.8 LUFS

    True peak:
    Peak: -0.4 dBFS
    /*
    * Copyright (c) 2010 Nicolas George
    * Copyright (c) 2011 Stefano Sabatini
    * Copyright (c) 2012 Clément Bœsch
    *
    * Permission is hereby granted, free of charge, to any person obtaining a copy
    * of this software and associated documentation files (the "Software"), to deal
    * in the Software without restriction, including without limitation the rights
    * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
    * copies of the Software, and to permit persons to whom the Software is
    * furnished to do so, subject to the following conditions:
    *
    * The above copyright notice and this permission notice shall be included in
    * all copies or substantial portions of the Software.
    *
    * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
    * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
    * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
    * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
    * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
    * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
    * THE SOFTWARE.
    */

    /**
    * @file
    * API example for audio decoding and filtering
    * @example filtering_audio.c
    */

    #include

    #include <libavcodec></libavcodec>avcodec.h>
    #include <libavformat></libavformat>avformat.h>
    #include <libavfilter></libavfilter>buffersink.h>
    #include <libavfilter></libavfilter>buffersrc.h>
    #include <libavutil></libavutil>opt.h>

    #define MAX_CHANNELS 63



    static const char *filter_descr = "ebur128=peak=true";

    static AVFormatContext *fmt_ctx;
    static AVCodecContext *dec_ctx;
    AVFilterContext *buffersink_ctx;
    AVFilterContext *buffersrc_ctx;
    AVFilterGraph *filter_graph;
    static int audio_stream_index = -1;

    struct rect { int x, y, w, h; };


    struct hist_entry {
       int count;                      ///&lt; how many times the corresponding value occurred
       double energy;                  ///&lt; E = 10^((L + 0.691) / 10)
       double loudness;                ///&lt; L = -0.691 + 10 * log10(E)
    };


    struct integrator {
       double *cache[MAX_CHANNELS];    ///&lt; window of filtered samples (N ms)
       int cache_pos;                  ///&lt; focus on the last added bin in the cache array
       double sum[MAX_CHANNELS];       ///&lt; sum of the last N ms filtered samples (cache content)
       int filled;                     ///&lt; 1 if the cache is completely filled, 0 otherwise
       double rel_threshold;           ///&lt; relative threshold
       double sum_kept_powers;         ///&lt; sum of the powers (weighted sums) above absolute threshold
       int nb_kept_powers;             ///&lt; number of sum above absolute threshold
       struct hist_entry *histogram;   ///&lt; histogram of the powers, used to compute LRA and I
    };

    typedef struct EBUR128Context {
       const AVClass *class;           ///&lt; AVClass context for log and options purpose

       /* peak metering */
       int peak_mode;                  ///&lt; enabled peak modes
       double *true_peaks;             ///&lt; true peaks per channel
       double *sample_peaks;           ///&lt; sample peaks per channel
       double *true_peaks_per_frame;   ///&lt; true peaks in a frame per channel
    #if CONFIG_SWRESAMPLE
       SwrContext *swr_ctx;            ///&lt; over-sampling context for true peak metering
       double *swr_buf;                ///&lt; resampled audio data for true peak metering
       int swr_linesize;
    #endif

       /* video  */
       int do_video;                   ///&lt; 1 if video output enabled, 0 otherwise
       int w, h;                       ///&lt; size of the video output
       struct rect text;               ///&lt; rectangle for the LU legend on the left
       struct rect graph;              ///&lt; rectangle for the main graph in the center
       struct rect gauge;              ///&lt; rectangle for the gauge on the right
       AVFrame *outpicref;             ///&lt; output picture reference, updated regularly
       int meter;                      ///&lt; select a EBU mode between +9 and +18
       int scale_range;                ///&lt; the range of LU values according to the meter
       int y_zero_lu;                  ///&lt; the y value (pixel position) for 0 LU
       int y_opt_max;                  ///&lt; the y value (pixel position) for 1 LU
       int y_opt_min;                  ///&lt; the y value (pixel position) for -1 LU
       int *y_line_ref;                ///&lt; y reference values for drawing the LU lines in the graph and the gauge

       /* audio */
       int nb_channels;                ///&lt; number of channels in the input
       double *ch_weighting;           ///&lt; channel weighting mapping
       int sample_count;               ///&lt; sample count used for refresh frequency, reset at refresh

       /* Filter caches.
        * The mult by 3 in the following is for X[i], X[i-1] and X[i-2] */
       double x[MAX_CHANNELS * 3];     ///&lt; 3 input samples cache for each channel
       double y[MAX_CHANNELS * 3];     ///&lt; 3 pre-filter samples cache for each channel
       double z[MAX_CHANNELS * 3];     ///&lt; 3 RLB-filter samples cache for each channel

    #define I400_BINS  (48000 * 4 / 10)
    #define I3000_BINS (48000 * 3)
       struct integrator i400;         ///&lt; 400ms integrator, used for Momentary loudness  (M), and Integrated loudness (I)
       struct integrator i3000;        ///&lt;    3s integrator, used for Short term loudness (S), and Loudness Range      (LRA)

       /* I and LRA specific */
       double integrated_loudness;     ///&lt; integrated loudness in LUFS (I)
       double loudness_range;          ///&lt; loudness range in LU (LRA)
       double lra_low, lra_high;       ///&lt; low and high LRA values

       /* misc */
       int loglevel;                   ///&lt; log level for frame logging
       int metadata;                   ///&lt; whether or not to inject loudness results in frames
       int dual_mono;                  ///&lt; whether or not to treat single channel input files as dual-mono
       double pan_law;                 ///&lt; pan law value used to calculate dual-mono measurements
       int target;                     ///&lt; target level in LUFS used to set relative zero LU in visualization
       int gauge_type;                 ///&lt; whether gauge shows momentary or short
       int scale;                      ///&lt; display scale type of statistics
    } EBUR128Context;

    void dump_ebur128_context(void *priv);

    static int open_input_file(const char *filename)
    {
       int ret;
       AVCodec *dec;

       if ((ret = avformat_open_input(&amp;fmt_ctx, filename, NULL, NULL)) &lt; 0) {
           av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
           return ret;
       }

       if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) &lt; 0) {
           av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
           return ret;
       }

       /* select the audio stream */
       ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &amp;dec, 0);
       if (ret &lt; 0) {
           av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
           return ret;
       }
       audio_stream_index = ret;

       /* create decoding context */
       dec_ctx = avcodec_alloc_context3(dec);
       if (!dec_ctx)
           return AVERROR(ENOMEM);
       avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);

       /* init the audio decoder */
       if ((ret = avcodec_open2(dec_ctx, dec, NULL)) &lt; 0) {
           av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
           return ret;
       }

       return 0;
    }

    static int init_filters(const char *filters_descr)
    {
       char args[512];
       int ret = 0;
       const AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
       const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
       AVFilterInOut *outputs = avfilter_inout_alloc();
       AVFilterInOut *inputs  = avfilter_inout_alloc();
       static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
       static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
       static const int out_sample_rates[] = { 8000, -1 };
       const AVFilterLink *outlink;
       AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;

       filter_graph = avfilter_graph_alloc();
       if (!outputs || !inputs || !filter_graph) {
           ret = AVERROR(ENOMEM);
           goto end;
       }

       /* buffer audio source: the decoded frames from the decoder will be inserted here. */
       if (!dec_ctx->channel_layout)
           dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
       snprintf(args, sizeof(args),
               "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
                time_base.num, time_base.den, dec_ctx->sample_rate,
                av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
       ret = avfilter_graph_create_filter(&amp;buffersrc_ctx, abuffersrc, "in",
                                          args, NULL, filter_graph);
       if (ret &lt; 0) {
           av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
           goto end;
       }

       /* buffer audio sink: to terminate the filter chain. */
       ret = avfilter_graph_create_filter(&amp;buffersink_ctx, abuffersink, "out",
                                          NULL, NULL, filter_graph);
       if (ret &lt; 0) {
           av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
           goto end;
       }

       ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
                                 AV_OPT_SEARCH_CHILDREN);
       if (ret &lt; 0) {
           av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
           goto end;
       }

       ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
                                 AV_OPT_SEARCH_CHILDREN);
       if (ret &lt; 0) {
           av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
           goto end;
       }

       ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
                                 AV_OPT_SEARCH_CHILDREN);
       if (ret &lt; 0) {
           av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
           goto end;
       }

       /*
        * Set the endpoints for the filter graph. The filter_graph will
        * be linked to the graph described by filters_descr.
        */

       /*
        * The buffer source output must be connected to the input pad of
        * the first filter described by filters_descr; since the first
        * filter input label is not specified, it is set to "in" by
        * default.
        */
       outputs->name       = av_strdup("in");
       outputs->filter_ctx = buffersrc_ctx;
       outputs->pad_idx    = 0;
       outputs->next       = NULL;

       /*
        * The buffer sink input must be connected to the output pad of
        * the last filter described by filters_descr; since the last
        * filter output label is not specified, it is set to "out" by
        * default.
        */
       inputs->name       = av_strdup("out");
       inputs->filter_ctx = buffersink_ctx;
       inputs->pad_idx    = 0;
       inputs->next       = NULL;

       if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
                                           &amp;inputs, &amp;outputs, NULL)) &lt; 0)
           goto end;

       if ((ret = avfilter_graph_config(filter_graph, NULL)) &lt; 0)
           goto end;

       /* Print summary of the sink buffer
        * Note: args buffer is reused to store channel layout string */
       outlink = buffersink_ctx->inputs[0];
       av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
       av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
              (int)outlink->sample_rate,
              (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
              args);

    end:
       avfilter_inout_free(&amp;inputs);
       avfilter_inout_free(&amp;outputs);

       return ret;
    }

    static void print_frame(const AVFrame *frame)
    {
    //    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
    //    const uint16_t *p     = (uint16_t*)frame->data[0];
    //    const uint16_t *p_end = p + n;
    //
    //    while (p &lt; p_end) {
    //        fputc(*p    &amp; 0xff, stdout);
    //        fputc(*p>>8 &amp; 0xff, stdout);
    //        p++;
    //    }
    //    fflush(stdout);
    }

    int main(int argc, char **argv)
    {
       av_log_set_level(AV_LOG_DEBUG);
       int ret;
       AVPacket packet;
       AVFrame *frame = av_frame_alloc();
       AVFrame *filt_frame = av_frame_alloc();

       if (!frame || !filt_frame) {
           perror("Could not allocate frame");
           exit(1);
       }


       if ((ret = open_input_file(argv[1])) &lt; 0)
           goto end;
       if ((ret = init_filters(filter_descr)) &lt; 0)
           goto end;

       /* read all packets */
       while (1) {
           if ((ret = av_read_frame(fmt_ctx, &amp;packet)) &lt; 0)
               break;

           if (packet.stream_index == audio_stream_index) {
               ret = avcodec_send_packet(dec_ctx, &amp;packet);
               if (ret &lt; 0) {
                   av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
                   break;
               }

               while (ret >= 0) {
                   ret = avcodec_receive_frame(dec_ctx, frame);
                   if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
                       break;
                   } else if (ret &lt; 0) {
                       av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
                       goto end;
                   }

                   if (ret >= 0) {
                       /* push the audio data from decoded frame into the filtergraph */
                       if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) &lt; 0) {
                           av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
                           break;
                       }

                       /* pull filtered audio from the filtergraph */
                       while (1) {
                           ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
                           if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                               break;
                           if (ret &lt; 0)
                               goto end;
                           print_frame(filt_frame);
                           av_frame_unref(filt_frame);
                       }
                       av_frame_unref(frame);
                   }
               }
           }
           av_packet_unref(&amp;packet);
       }
       if(filter_graph->nb_filters){
       av_log(filter_graph, AV_LOG_INFO, "hello : %d \n",
                   filter_graph->nb_filters);
       int i;
       for (int i = 0; i &lt; filter_graph->nb_filters; i++){
           av_log(filter_graph, AV_LOG_INFO, "name : %s \n",
                           filter_graph->filters[i]->name);
       }
       }

       av_log(filter_graph, AV_LOG_INFO, "name : %s \n",
                               filter_graph->filters[2]->name);
       void* priv = filter_graph->filters[2]->priv;

       dump_ebur128_context(&amp;priv);

    end:


       avfilter_graph_free(&amp;filter_graph);
       avcodec_free_context(&amp;dec_ctx);
       avformat_close_input(&amp;fmt_ctx);
       av_frame_free(&amp;frame);
       av_frame_free(&amp;filt_frame);

       if (ret &lt; 0 &amp;&amp; ret != AVERROR_EOF) {
           fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
           exit(1);
       }

       exit(0);
    }

    void dump_ebur128_context(void *priv){
       EBUR128Context *ebur128 = priv;

       av_log(ebur128, AV_LOG_INFO, "integrated_loudness : %5.1f \n",
                               ebur128->integrated_loudness);
       av_log(ebur128, AV_LOG_INFO, "lra_low : %5.1f \n",
                                   ebur128->lra_low);
       av_log(ebur128, AV_LOG_INFO, "lra_high : %5.1f \n",
                                   ebur128->lra_high);


    }
    program fails while accessing integrated loudness in dump_ebur128_context.

    can someone guide me about , how I should proceed in here.