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• ffmpeg encoded mp4 with faststart not playing in firefox [duplicate]

  12 mars 2018, par adam smith

  This question already has an answer here :

  • Which pixel format for web mp4 video ?
    
    1 answer
    

  i am having problems getting mp4 files encoded by ffmpeg to play using firefox without it downloading the entire thing first ; while chrome plays immediately. I am using ffmpeg to encode the mp4 and applying ’-movflags faststart’ to make sure the moov atom is at the start, and i have used Atomic Parsely to check the output structure.

  I have checked that the server is using byte range etc, all good and other files are working well.

  I have compared the Atomic Parsely return from a file encoded using flipfactory that does playback in firefox immediately ; its moov atom contains a iod atom that is not present in the ffmpeg encoding.

  example of ffmpeg command :

  -i source_file_path -c:v libx264 -vf scale=1024 :-1 -preset veryfast -b:v 1000k -ac 2 -bsf:a aac_adtstoasc -b:a 128k -movflags faststart -f mp4 proxy_output_path

  example of ffmpeg output :

  Atom ftyp @ 0 of size : 32, ends @ 32
  Atom moov @ 32 of size : 6879133, ends @ 6879165
  Atom mvhd @ 40 of size : 108, ends @ 148
  Atom trak @ 148 of size : 3567272, ends @ 3567420
  Atom tkhd @ 156 of size : 92, ends @ 248
  Atom edts @ 248 of size : 36, ends @ 284
  Atom elst @ 256 of size : 28, ends @ 284
  Atom tref @ 284 of size : 20, ends @ 304
  Atom tmcd @ 292 of size : 12, ends @ 304
   
  Atom mdia @ 304 of size : 3567116, ends @ 3567420
  Atom mdhd @ 312 of size : 32, ends @ 344
  Atom hdlr @ 344 of size : 45, ends @ 389
  Atom minf @ 389 of size : 3567031, ends @ 3567420
  Atom vmhd @ 397 of size : 20, ends @ 417
  Atom dinf @ 417 of size : 36, ends @ 453
  Atom dref @ 425 of size : 28, ends @ 453
  Atom stbl @ 453 of size : 3566967, ends @ 3567420
  Atom stsd @ 461 of size : 151, ends @ 612
  Atom avc1 @ 477 of size : 135, ends @ 612
  Atom avcC @ 563 of size : 49, ends @ 612
  Atom stts @ 612 of size : 24, ends @ 636
  Atom stss @ 636 of size : 5736, ends @ 6372
  Atom ctts @ 6372 of size : 1734376, ends @ 1740748
  Atom stsc @ 1740748 of size : 40, ends @ 1740788
  Atom stsz @ 1740788 of size : 913320, ends @ 2654108
  Atom stco @ 2654108 of size : 913312, ends @ 3567420

  example of flipfactory output :

  Atom ftyp @ 0 of size : 24, ends @ 24
  Atom moov @ 24 of size : 561374, ends @ 561398
  Atom mvhd @ 32 of size : 108, ends @ 140
  Atom iods @ 140 of size : 33, ends @ 173
  Atom trak @ 173 of size : 268275, ends @ 268448
  Atom tkhd @ 181 of size : 92, ends @ 273
  Atom mdia @ 273 of size : 268175, ends @ 268448
  Atom mdhd @ 281 of size : 32, ends @ 313
  Atom hdlr @ 313 of size : 68, ends @ 381
  Atom minf @ 381 of size : 268067, ends @ 268448
  Atom smhd @ 389 of size : 16, ends @ 405
  Atom dinf @ 405 of size : 36, ends @ 441
  Atom dref @ 413 of size : 28, ends @ 441
  Atom stbl @ 441 of size : 268007, ends @ 268448
  Atom stsd @ 449 of size : 91, ends @ 540
  Atom mp4a @ 465 of size : 75, ends @ 540
  Atom esds @ 501 of size : 39, ends @ 540
  Atom stts @ 540 of size : 24, ends @ 564
  Atom stsc @ 564 of size : 22324, ends @ 22888
  Atom stsz @ 22888 of size : 234212, ends @ 257100
  Atom stco @ 257100 of size : 11348, ends @ 268448

  i am not sure that adding the iod atom will resolve the issue, but thats all i can see that is different. I cannot however find anything in the ffmpeg documentation that seems to help.

  any recommendations for resolving using ffmpeg would be great.

• Dash.js not playing mpd files made with ffmpeg

  31 décembre 2022, par Macster

  I'm using ffmpeg to create chunks and manifest of a webm file which I want to live stream with Dash.js. Unfortunately Dash.js won't play the mpd file, no matter which way I create the chunks and manifest. However, the sample mpd URL from Dash.js is working.

  


  Commands

  


  ffmpeg -re -r 25 -i Dash/strm.webm
-map 0:v:0
-pix_fmt yuv420p
-c:v libvpx
-s 640x480 -keyint_min 60 -g 60 -speed 6 -tile-columns 4 -frame-parallel 1 -threads 8 -static-thresh 0 -max-intra-rate 300 -deadline realtime -lag-in-frames 0 -error-resilient 1
-b:v 3000k
-f webm_chunk
-header "Dash/glass_360.hdr"
-chunk_start_index 1 Dash/glass_360_%d.chk
-map 0:a:0
-c:a libvorbis
-b:a 128k -ar 44100
-f webm_chunk
-audio_chunk_duration 2000
-header Dash/glass_171.hdr
-chunk_start_index 1 Dash/glass_171_%d.chk


//Manifest
ffmpeg
-f webm_dash_manifest -live 1
-i Dash/glass_360.hdr
-f webm_dash_manifest -live 1
-i Dash/glass_171.hdr
-c copy
-map 0 -map 1
-f webm_dash_manifest -live 1
-adaptation_sets "id=0,streams=0 id=1,streams=1"
-chunk_start_index 1
-chunk_duration_ms 2000
-time_shift_buffer_depth 7200
-minimum_update_period 7200 Dash/glass_video_manifest.mpd


  


  Manifest output

  


  ffmpeg version git-2020-05-27-8b5ffae Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200523
  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 --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
  libavutil      56. 49.100 / 56. 49.100
  libavcodec     58. 87.101 / 58. 87.101
  libavformat    58. 43.100 / 58. 43.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 83.100 /  7. 83.100
  libswscale      5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
Input #0, webm_dash_manifest, from 'Dash/glass_360.hdr':
  Metadata:
    ENCODER         : Lavf58.43.100
  Duration: N/A, bitrate: N/A
    Stream #0:0(eng): Video: vp8, yuv420p, 640x480, SAR 1:1 DAR 4:3, 1k tbr, 1k tbn, 1k tbc (default)
    Metadata:
      ALPHA_MODE      : 1
      ENCODER         : Lavc58.87.101 libvpx
      webm_dash_manifest_file_name: glass_360.hdr
      webm_dash_manifest_track_number: 1
Input #1, webm_dash_manifest, from 'Dash/glass_171.hdr':
  Metadata:
    ENCODER         : Lavf58.43.100
  Duration: N/A, bitrate: N/A
    Stream #1:0(eng): Audio: vorbis, 44100 Hz, mono, fltp (default)
    Metadata:
      ENCODER         : Lavc58.87.101 libvorbis
      webm_dash_manifest_file_name: glass_171.hdr
      webm_dash_manifest_track_number: 1
Output #0, webm_dash_manifest, to 'Dash/glass_video_manifest.mpd':
  Metadata:
    encoder         : Lavf58.43.100
    Stream #0:0(eng): Video: vp8, yuv420p, 640x480 [SAR 1:1 DAR 4:3], q=2-31, 1k tbr, 1k tbn, 1k tbc (default)
    Metadata:
      ALPHA_MODE      : 1
      ENCODER         : Lavc58.87.101 libvpx
      webm_dash_manifest_file_name: glass_360.hdr
      webm_dash_manifest_track_number: 1
    Stream #0:1(eng): Audio: vorbis, 44100 Hz, mono, fltp (default)
    Metadata:
      ENCODER         : Lavc58.87.101 libvorbis
      webm_dash_manifest_file_name: glass_171.hdr
      webm_dash_manifest_track_number: 1
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #1:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame=    0 fps=0.0 q=-1.0 Lsize=       1kB time=00:00:00.00 bitrate=N/A speed=   0x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:4kB muxing overhead: unknown


  


  Manifest file
  (glass_video_manifest.mpd)
  
I tried to delete the ContetntComponent like suggested in other questions, but it didn't work.

  


  &lt;?xml version="1.0" encoding="UTF-8"?>&#xA;&#xA;<period start="PT0S">&#xA;<adaptationset mimetype="video/webm" codecs="vp8" lang="eng" bitstreamswitching="true" subsegmentalignment="true" subsegmentstartswithsap="1">&#xA;<contentcomponent type="video"></contentcomponent>&#xA;<segmenttemplate timescale="1000" duration="2000" media="glass_$RepresentationID$_$Number$.chk" startnumber="1" initialization="glass_$RepresentationID$.hdr"></segmenttemplate>&#xA;<representation bandwidth="1000000" width="640" height="480" codecs="vp8" mimetype="video/webm" startswithsap="1"></representation>&#xA;</adaptationset>&#xA;<adaptationset mimetype="audio/webm" codecs="vorbis" lang="eng" bitstreamswitching="true" subsegmentalignment="true" subsegmentstartswithsap="1">&#xA;<contentcomponent type="audio"></contentcomponent>&#xA;<segmenttemplate timescale="1000" duration="2000" media="glass_$RepresentationID$_$Number$.chk" startnumber="1" initialization="glass_$RepresentationID$.hdr"></segmenttemplate>&#xA;<representation bandwidth="128000" audiosamplingrate="44100" codecs="vorbis" mimetype="audio/webm" startswithsap="1"></representation>&#xA;</adaptationset>&#xA;</period>&#xA;&#xA;

  


  Dash.js-Player

  


  <code class="echappe-js">&lt;script&gt;&amp;#xA;&amp;#xA;(function(){&amp;#xA;  // var url = &quot;https://dash.akamaized.net/envivio/EnvivioDash3/manifest.mpd&quot;;&amp;#xA;  var url = &quot;http://localhost:8081/videos/Dash/glass_live_manifest.mpd&quot;;&amp;#xA;  var player = dashjs.MediaPlayer().create();&amp;#xA;                &amp;#xA;  // config&amp;#xA;  targetLatency = 2.0;        // Lowering this value will lower latency but may decrease the player&amp;#x27;s ability to build a stable buffer.&amp;#xA;  minDrift = 0.05;            // Minimum latency deviation allowed before activating catch-up mechanism.&amp;#xA;  catchupPlaybackRate = 0.5;  // Maximum catch-up rate, as a percentage, for low latency live streams.&amp;#xA;  stableBuffer = 2;           // The time that the internal buffer target will be set to post startup/seeks (NOT top quality).&amp;#xA;  bufferAtTopQuality = 2;     // The time that the internal buffer target will be set to once playing the top quality.&amp;#xA;&amp;#xA;   player.updateSettings({&amp;#xA;      &amp;#x27;streaming&amp;#x27;: {&amp;#xA;          &amp;#x27;liveDelay&amp;#x27;: 2,&amp;#xA;          &amp;#x27;liveCatchUpMinDrift&amp;#x27;: 0.05,&amp;#xA;          &amp;#x27;liveCatchUpPlaybackRate&amp;#x27;: 0.5,&amp;#xA;          &amp;#x27;stableBufferTime&amp;#x27;: 2,&amp;#xA;          &amp;#x27;bufferTimeAtTopQuality&amp;#x27;: 2,&amp;#xA;          &amp;#x27;bufferTimeAtTopQualityLongForm&amp;#x27;: 2,&amp;#xA;          &amp;#x27;bufferToKeep&amp;#x27;: 2,&amp;#xA;          &amp;#x27;bufferAheadToKeep&amp;#x27;: 2,&amp;#xA;          &amp;#x27;lowLatencyEnabled&amp;#x27;: true,&amp;#xA;          &amp;#x27;fastSwitchEnabled&amp;#x27;: true,&amp;#xA;          &amp;#x27;abr&amp;#x27;: {&amp;#xA;               &amp;#x27;limitBitrateByPortal&amp;#x27;: true&amp;#xA;          },&amp;#xA;       }&amp;#xA;   });&amp;#xA;&amp;#xA;   console.log(player.getSettings());&amp;#xA;&amp;#xA;   setInterval(() =&gt; {&amp;#xA;      console.log(&amp;#x27;Live latency= &amp;#x27;, player.getCurrentLiveLatency());&amp;#xA;      console.log(&amp;#x27;Buffer length= &amp;#x27;, player.getBufferLength(&amp;#x27;video&amp;#x27;));&amp;#xA;     }, 3000);&amp;#xA;&amp;#xA;     player.initialize(document.querySelector(&quot;#videoPlayer&quot;), url, true);&amp;#xA;&amp;#xA;   })();&amp;#xA;&lt;/script&gt;&#xA;

  


  Chrome

  


  {debug: {…}, streaming: {…}}
dash.all.min.js:2 XHR finished loading: GET "http://localhost:8081/videos/Dash/glass_live_manifest.mpd".
load @ dash.all.min.js:2
C @ dash.all.min.js:2
load @ dash.all.min.js:2
load @ dash.all.min.js:2
load @ dash.all.min.js:2
load @ dash.all.min.js:2
se @ dash.all.min.js:2
te @ dash.all.min.js:2
initialize @ dash.all.min.js:2
(anonymous) @ Dash:92
(anonymous) @ Dash:94
DevTools failed to load SourceMap: Could not parse content for http://localhost:8081/js/dash.all.min.js.map: Cannot read property 'length' of undefined
Dash:88 Live latency=  NaN
Dash:89 Buffer length=  NaN
Dash:88 Live latency=  NaN
Dash:89 Buffer length=  NaN
Dash:88 Live latency=  NaN
Dash:89 Buffer length=  NaN
Dash:88 Live latency=  NaN
Dash:89 Buffer length=  NaN
Dash:88 Live latency=  NaN
Dash:89 Buffer length=  NaN


  


  UPDATE

  


  Well, it seems like the problem in general was, that the mpd's wouldn't play from that /dash folder. So i took a look into the code and found a bad routing. Anyways, the mpd would't start with the given command I used, probably becasue it creates a dynamic manifest, like @Markus Schumann says. So I'm going with a new one which seems to be working for now, but not very well.

  


  ffmpeg -y -re -i strm.webm 
-c:v libx264 -x264opts "keyint=24:min-keyint=24:no-scenecut" 
-r 24 -c:a aac -b:a 128k -bf 1 -b_strategy 0 -sc_threshold 0 -pix_fmt yuv420p 
-map 0:v:0 -map 0:a:0 -map 0:v:0 -map 0:a:0 -map 0:v:0 -map 0:a:0 -b:v:0 250k  
-filter:v:0 "scale=-2:240" -profile:v:0 baseline -b:v:1 750k  
-filter:v:1 "scale=-2:480" -profile:v:1 main -b:v:2 1500k 
-filter:v:2 "scale=-2:720" -profile:v:2 high 
-use_timeline 1 -use_template 1 -window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a" 
-f dash glass_video_manifest.mpd


  


• Playing Mp3 file using FFmpeg on Android

  2 avril 2017, par satyres

  This question has been asked a lot but no code has worked for me . i’ve been able to play a file decoded with ffmpeg on Android but it’s noisy and glitchy.
  i’ve found code in book called "linux sound programming" using latest ffmpeg version to decode an mp3 file.
  the code tries to decode an mp3 file to pcm and then put it in a file called output.
  what i want to do is to get the byte decoded on the fly and send them to AudioTrack in Java.

   void JNICALL Java_com_example_home_hellondk_MainActivity_loadFile
  
          (JNIEnv* env, jobject obj,jstring file,jbyteArray array)
  
  {
  
  
  
   jboolean isfilenameCopy;
  
           const char * filename = (*env)->GetStringUTFChars(env, file,
  
                   &isfilenameCopy);
  
  jclass cls = (*env)->GetObjectClass(env, obj);
  
           jmethodID play = (*env)->GetMethodID(env, cls, "playSound", "([BI)V");
  
      AVCodec *codec;
  
      AVCodecContext *c= NULL;
  
      int len;
  
      FILE *f, *outfile;
  
      uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
  
      AVPacket avpkt;
  
      AVFrame *decoded_frame = NULL;
  
  AVFormatContext* container=NULL;
  
      av_init_packet(&avpkt);
  
   int num_streams = 0;
  
      int sample_size = 0;
  
      printf("Decode audio file %s \n", filename);
  
  LOGE("Decode audio file %s\n", filename);
  
      /* find the MPEG audio decoder */
  
    /*  codec = avcodec_find_decoder(AV_CODEC_ID_MP3);
  
      if (!codec) {
  
          fprintf(stderr, "Codec not found\n");
  
          LOGE("Codec not found\n");
  
          exit(1);
  
      }*/
  
  int lError;
  
           if ((lError = avformat_open_input(&container, filename, NULL, NULL))
  
                   != 0) {
  
               LOGE("Error open source file: %d", lError);
  
               exit(1);
  
           }
  
           if ((lError =  avformat_find_stream_info(container,NULL)) < 0) {
  
               LOGE("Error find stream information: %d", lError);
  
               exit(1);
  
           }
  
           LOGE("Stage 1.5");
  
          LOGE("audio format: %s", container->iformat->name);
  
         LOGE("audio bitrate: %llu", container->bit_rate);
  
  
  
      int stream_id = -1;
  
          // To find the first audio stream. This process may not be necessary
  
          // if you can gurarantee that the container contains only the desired
  
          // audio stream
  
             LOGE("nb_streams: %d", container->nb_streams);
  
          int i;
  
          for (i = 0; i < container->nb_streams; i++) {
  
              if (container->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
  
                  stream_id = i;
  
                   LOGE("stream_id: %d", stream_id);
  
                  break;
  
              }
  
          }
  
  
  
          AVCodecContext* codec_context = container->streams[stream_id]->codec;
  
          codec = avcodec_find_decoder(codec_context->codec_id);
  
           LOGE("stream_id: %d", stream_id);
  
          LOGE("codec %s", codec->name);
  
          if (!codec) {
  
              fprintf(stderr, "codec not found\n");
  
              exit(1);
  
          }
  
  
  
      c = avcodec_alloc_context3(codec);
  
      if (!c) {
  
          fprintf(stderr, "Could not allocate audio codec context\n");
  
            LOGE("Could not allocate audio codec context\n");
  
          exit(1);
  
      }
  
  
  
      /* open it */
  
      if (avcodec_open2(c, codec, NULL) < 0) {
  
          fprintf(stderr, "Could not open codec\n");
  
           LOGE("Could not open codec\n");
  
          exit(1);
  
      }
  
  
  
      f = fopen(filename, "rb");
  
      if (!f) {
  
          fprintf(stderr, "Could not open %s\n", filename);
  
          LOGE("Could not open %s\n",filename);
  
          exit(1);
  
      }
  
      const char *outfilename;
  
   outfile = fopen(outfilename, "wb");
  
      if (!outfile) {
  
          av_free(c);
  
          exit(1);
  
      }
  
  
  
  
  
          avpkt.data = inbuf;
  
          avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
  
          LOGE("Stage 5");
  
      /* decode until eof */
  
  
  
       while (avpkt.size > 0) {
  
             int got_frame = 0;
  
             if (!decoded_frame) {
  
                 if (!(decoded_frame = av_frame_alloc())) {
  
                     fprintf(stderr, "out of memory\n");
  
                     exit(1);
  
                 }
  
                 } else {
  
                             av_frame_unref(decoded_frame);
  
                         }
  
                         printf("Stream idx %d\n", avpkt.stream_index);
  
                         len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
  
                         if (len < 0) {
  
                             fprintf(stderr, "Error while decoding\n");
  
                             exit(1);
  
                         }
  
                         if (got_frame) {
  
                             printf("Decoded frame nb_samples %d, format %d\n",
  
                                    decoded_frame->nb_samples,
  
                                    decoded_frame->format);
  
                             if (decoded_frame->data[1] != NULL)
  
                                 printf("Data[1] not null\n");
  
                             else
  
                                 printf("Data[1] is null\n");
  
                             /* if a frame has been decoded, output it */
  
                             int data_size = av_samples_get_buffer_size(NULL, c->channels,
  
                                                                        decoded_frame->nb_samples,
  
                                                                        c->sample_fmt, 1);
  
                             // first time: count the number of  planar streams
  
                             if (num_streams == 0) {
  
                                 while (num_streams < AV_NUM_DATA_POINTERS &&
  
                                        decoded_frame->data[num_streams] != NULL)
  
                                     num_streams++;
  
                                 printf("Number of streams %d\n", num_streams);
  
                             }
  
                             // first time: set sample_size from 0 to e.g 2 for 16-bit data
  
                             if (sample_size == 0) {
  
                                 sample_size =
  
                                     data_size / (num_streams * decoded_frame->nb_samples);
  
                             }
  
                             int m, n;
  
  
  
                             for (n = 0; n < decoded_frame->nb_samples; n++) {
  
                                 // interleave the samples from the planar streams
  
                                 for (m = 0; m < num_streams; m++) {
  
                                    fwrite(&decoded_frame->data[m][n*sample_size],
  
                                            1, sample_size, outfile);
  
  
  
                                 }
  
                             }
  
  
  
                  /*            jbyte *bytes = (*env)->GetByteArrayElements(env, array, NULL);
  
                              memcpy(bytes, decoded_frame->data[1], data_size);
  
                              (*env)->ReleaseByteArrayElements(env, array, bytes, 0);
  
                             (*env)->CallVoidMethod(env, obj, play, array, data_size);
  
  
  
  */
  
                         }
  
                         avpkt.size -= len;
  
                         avpkt.data += len;
  
                         if (avpkt.size < AUDIO_REFILL_THRESH) {
  
                             /* Refill the input buffer, to avoid trying to decode
  
                              * incomplete frames. Instead of this, one could also use
  
                              * a parser, or use a proper container format through
  
                                * libavformat. */
  
                                          memmove(inbuf, avpkt.data, avpkt.size);
  
                                          avpkt.data = inbuf;
  
                                          len = fread(avpkt.data + avpkt.size, 1,
  
                                                      AUDIO_INBUF_SIZE - avpkt.size, f);
  
                                          if (len > 0)
  
                                              avpkt.size += len;
  
                                      }
  
                                  }
  
  
  
      fclose(f);
  
  
  
      avcodec_free_context(&c);
  
  av_frame_free(&decoded_frame);
  
  
  
  }

  the decoded bytes are in this section
  fwrite(&decoded_frame->data[m][n*sample_size],                                           1, sample_size, outfile);

  the code that let you send bytes to java is this :

    jbyte *bytes = (*env)->GetByteArrayElements(env, array, NULL);
  
    memcpy(bytes, decoded_frame->data[0], data_size);
  
   (*env)->ReleaseByteArrayElements(env, array, bytes, 0);
  
   (*env)->CallVoidMethod(env, obj, play, array, data_size);

  i’ve been working on it now for more than a week and nothing worked for me.

  Thank you in advance for your help