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• need help configuring ffmpeg to decode raw AAC with android ndk

  24 octobre 2016, par Matt Wolfe

  I’ve got an android app that gets raw AAC bytes from an external device and I want to decode that data but I can’t seem to get the decoder to work, yet ffmpeg seems to work fine for decoding an mp4 file that contains the same audio data (verified with isoviewer). Recently I was able to get this ffmpeg library on android to decode video frames from the same external device but audio won’t seem to work.

  Here is the ffmpeg output for the file with the same data :

  $ ffmpeg -i Video_2000-01-01_0411.mp4
  
  ffmpeg version 2.6.1 Copyright (c) 2000-2015 the FFmpeg developers
  
    built with Apple LLVM version 6.0 (clang-600.0.57) (based on LLVM 3.5svn)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/2.6.1 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libmp3lame --enable-libvo-aacenc --enable-libxvid --enable-vda
  
    libavutil      54. 20.100 / 54. 20.100
  
    libavcodec     56. 26.100 / 56. 26.100
  
    libavformat    56. 25.101 / 56. 25.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 11.102 /  5. 11.102
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'AXON_Flex_Video_2000-01-01_0411.mp4':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 1
  
      compatible_brands: isom3gp43gp5
  
    Duration: 00:00:15.73, start: 0.000000, bitrate: 1134 kb/s
  
      Stream #0:0(eng): Audio: aac (LC) (mp4a / 0x6134706D), 8000 Hz, mono, fltp, 40 kb/s (default)
  
      Metadata:
  
        handler_name    : soun
  
      Stream #0:1(eng): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 1087 kb/s, 29.32 fps, 26.58 tbr, 90k tbn, 1k tbc (default)
  
      Metadata:
  
        handler_name    : vide

  Here is my ndk code for setting up and decoding the audio :

  jint ffmpeg_init(JNIEnv * env, jobject this) {
  
      audioCodec = avcodec_find_decoder(AV_CODEC_ID_AAC);
  
      if (!audioCodec) {
  
          LOGE("audio codec %d not found", AV_CODEC_ID_AAC);
  
          return -1;
  
      }
  
  
  
      audioContext = avcodec_alloc_context3(audioCodec);
  
      if (!audioContext) {
  
          LOGE("Could not allocate codec context");
  
          return -1;
  
      }
  
  
  
       int openRet = avcodec_open2(audioContext, audioCodec, NULL);
  
          if (openRet < 0) {
  
            LOGE("Could not open codec, error:%d", openRet);
  
            return -1;
  
          }
  
  
  
      audioContext->sample_rate = 8000;
  
      audioContext->channel_layout = AV_CH_LAYOUT_MONO;
  
      audioContext->profile = FF_PROFILE_AAC_LOW;
  
      audioContext->bit_rate = 48 * 1024;
  
      audioContext->sample_fmt = AV_SAMPLE_FMT_FLTP;
  
  
  
    //  unsigned char extradata[] = {0x15, 0x88}; 
  
    //  audioContext->extradata = extradata;
  
    //  audioContext->extradata_size = sizeof(extradata);
  
      audioFrame = av_frame_alloc();
  
      if (!audioFrame) {
  
          LOGE("Could not create audio frame");
  
          return -1;
  
      }
  
  }
  
  
  
  
  
  jint ffmpeg_decodeAudio(JNIEnv *env, jobject this, jbyteArray aacData, jbyteArray output, int offset, int len) {
  
  
  
      LOGI("ffmpeg_decodeAudio()");
  
      char errbuf[128];
  
      AVPacket avpkt = {0};
  
      av_init_packet(&avpkt);
  
      LOGI("av_init_packet()");
  
      int error, got_frame;    
  
      uint8_t* buffer = (uint8_t *) (*env)->GetByteArrayElements(env, aacData,0);
  
      uint8_t* copy = av_malloc(len);  
  
      memcpy(copy, &buffer[offset], len);
  
      av_packet_from_data(&avpkt, copy, len);
  
  
  
  
  
      if ((error = avcodec_decode_audio4(audioContext, audioFrame, &got_frame, &avpkt)) < 0) {
  
          ffmpeg_log_error(error);
  
          av_free_packet(&avpkt);
  
          return error;
  
      }
  
      if (got_frame) {
  
          LOGE("Copying audioFrame->extended_data to output jbytearray, linesize[0]:%d", audioFrame->linesize[0]);
  
          (*env)->SetByteArrayRegion(env, output, 0, audioFrame->linesize[0],  *audioFrame->extended_data);
  
      }
  
  
  
      return 0;
  
  
  
  }

  As you can see I’ve got an init function that opens the decoder and creates the context, these things all work fine, without error. However when I call avcodec_decode_audio4 I get an error :

  FFMPEG error : -1094995529, Invalid data found when processing input

  I’ve tried all sorts of combinations of AVCodecContext properties. I’m not sure which I need to set for the decoder to do it’s job but from reading online I should just need to set the channel layout and the sample_rate (which I’ve tried by themself). I’ve also tried setting the extradata/extradata_size parameters to that which should match the video settings per : http://wiki.multimedia.cx/index.php?title=MPEG-4_Audio
  But no luck.

  Since the device we’re getting packets from sends aac data that have no sound at the beginning (but are valid packets), I’ve tried to just send those since they definitely should decode correctly.

  Here is an example of the initial audio packets that are of silence :

   010c9eb43f21f90fc87e46fff10a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5a5dffe214b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4b4bbd1c429696969696969696969696969696969696969696969696969696969696969696969696969696969696969696969696969696969696969696969696969696969696969696969696969696969696969696969696978

  Note that the data shown above is just a hex encoding of the data that I’m putting in AVPacket, and it was sent from an external device to the android application. My application doesn’t have direct access to the file though so I need to decode the raw frames/samples as I get them. When I look at the audio track data in isoviewer I can see that the audio track’s first sample is the same data as what I got from the device that contained that file (thus, the external device is just sending me the sample’s raw data). I believe this data can be derived from reading stsz (sample size) box starting at stco (chunk offset) boxes from the mdat box of the file.

  Also, isoviewer shows the esds box as having the following :

  ESDescriptor{esId=0, streamDependenceFlag=0, URLFlag=0, oCRstreamFlag=0, streamPriority=0, URLLength=0, URLString='null', remoteODFlag=0, dependsOnEsId=0, oCREsId=0, decoderConfigDescriptor=DecoderConfigDescriptor{objectTypeIndication=64, streamType=5, upStream=0, bufferSizeDB=513, maxBitRate=32000, avgBitRate=32000, decoderSpecificInfo=null, audioSpecificInfo=AudioSpecificConfig{configBytes=1588, audioObjectType=2 (AAC LC), samplingFrequencyIndex=11 (8000), samplingFrequency=0, channelConfiguration=1, syncExtensionType=0, frameLengthFlag=0, dependsOnCoreCoder=0, coreCoderDelay=0, extensionFlag=0, layerNr=0, numOfSubFrame=0, layer_length=0, aacSectionDataResilienceFlag=false, aacScalefactorDataResilienceFlag=false, aacSpectralDataResilienceFlag=false, extensionFlag3=0}, configDescriptorDeadBytes=, profileLevelIndicationDescriptors=[[]]}, slConfigDescriptor=SLConfigDescriptor{predefined=2}}

  And the binary is this :

  00 00 00 30 65 73 64 73 00 00 00 00 03 80 80 80
  
  1f 00 00 00 04 80 80 80 14 40 15 00 02 01 00 00
  
  7d 00 00 00 7d 00 05 80 80 80 02 15 88 06 01 02

• how to host live stream video

  21 août 2015, par Vhanjan

  Using jwplayer with html5 users can watch videos online..
  just put the source where the video file is located,, and the users can play the video that easy..

  i also tried to put a live stream source, as i remember it is rts ://sampledomain.com/video.mp4,
  then jwplayer easily play that mp4 file..

  my question is
  how can i host this kind of file "rts ://sampledomain.com/video.mp4" using my web camera or hypercam3 as my video recorder.. and send it to jwplayer so the users can play live stream in the browser using jwplayer..

  any hint are appreciated, but thank you very much for step by step tutorials..

• trying to figure out what's wrong in the code using ffmpeg.exe

  5 octobre 2013, par timmack

  I'm trying to upload videos and convert it to flv format as well as capturing image as thumbnails.I'm using visual web developer 2008 express edition.I've done so many research online and found something using ffmpeg.exe so I've downloaded an ffmpeg.exe and has tried so many codes but unfortunately I'm still unable to make it work.I want to save the converted video file to a certain folder in my server as well as the captured image as for its thumbnails after uploading the video.I'm not sure if the ffmpeg.exe has the problem or my codes but I suspect it's in my codes.I need somebody to help me find out what's something missing on my codes to make it work. Here's my codes below for your reference. Thanks

  protected void Button1_Click(object sender, EventArgs e)
  
      {
  
          //Convert the Video to flv format with FFMPEG
  
        string  ffmpegPath = Server.MapPath("~/ffmpeg/ffmpeg.exe");
  
        string tempLocation = Server.MapPath("~/tempVideos/");
  
        string mediaOutPath = Server.MapPath("~/Videos/");
  
        string thumbOutPath = Server.MapPath("~/Thumbs/");
  
        string currentFile = System.IO.Path.GetFileName(FileUpload1.PostedFile.FileName);
  
  
  
  
  
  
  
  //Upload the video     
  
  
  
  string vidPath = "Videos/" + outputfile;
  
  
  
  if (FileUpload1.PostedFile != null && FileUpload1.PostedFile.FileName != "")
  
  {
  
      try
  
  {
  
      // 10240 KB means 10MB, You can change the value based on your requirement
  
      if (FileUpload1.PostedFile.ContentLength > 1024000000)
  
      {
  
          Page.ClientScript.RegisterClientScriptBlock(typeof(Page), "Alert", "alert('File is too big.')", true);
  
      }
  
      else
  
      {
  
          //then save it to the Folder
  
  
  
  
  
          Convert(tempLocation + currentFile, mediaOutPath + currentFile, thumbOutPath + currentFile);
  
  
  
         FileUpload1.PostedFile.SaveAs(tempLocation + currentFile);
  
  
  
  
  
      }
  
           }
  
  catch (Exception ex)
  
  {
  
  Response.Write("Error: " + ex.Message);
  
  }
  
  }
  
  
  
  Page.ClientScript.RegisterClientScriptBlock(typeof(Page), "Alert", "alert('Video file has been uploaded!')", true);
  
  FlashVideo1.VideoURL = "tempVideos/" + currentFile;
  
  }
  
  
  
  protected void Convert(string fileIn, string fileOut, string thumbOut)
  
  {
  
  try
  
  {
  
  
  
      string mediaOutPath = Server.MapPath("~/Videos/");
  
      string thumbOutPath = Server.MapPath("~/Thumbs/");
  
      string inputfile = FileUpload1.PostedFile.FileName;
  
  
  
      string infile = Path.GetFileNameWithoutExtension(inputfile);
  
  
  
      fileIn = Path.GetFullPath( inputfile);
  
  
  
      fileOut = mediaOutPath + inputfile;
  
      thumbOut = inputfile;
  
  
  
  string ffmpegPath = Server.MapPath("~/ffmpeg/ffmpeg.exe");
  
  System.Diagnostics.Process proc = new System.Diagnostics.Process();
  
  proc.EnableRaisingEvents = true;
  
  proc.StartInfo.FileName = ffmpegPath;
  
  
  
  
  
  //converting video to flv format
  
  proc.StartInfo.Arguments = "-i " + fileIn + "-ar 22050 -ab 32 -f flv -s 320×240 -aspect 4:3 -y" + fileOut.Split('.')[0] + ".flv";
  
  proc.StartInfo.UseShellExecute = false;
  
  proc.StartInfo.CreateNoWindow = false;
  
  proc.StartInfo.RedirectStandardError = true;
  
  proc.StartInfo.RedirectStandardOutput = true;
  
  proc.Start();
  
  proc.WaitForExit();
  
  
  
  //capturing thumbnail image of the video 
  
  proc.StartInfo.Arguments = "-i \"{0}\" -an -y -s 320x240 -ss {1} -vframes 1 -f image2 \"{2}\"" + fileIn + thumbOut.Split('.')[0] + ".jpg";
  
  proc.StartInfo.UseShellExecute = false;
  
  proc.StartInfo.CreateNoWindow = false;
  
  proc.StartInfo.RedirectStandardError = true;
  
  proc.StartInfo.RedirectStandardOutput = true;
  
  proc.Start();
  
  proc.WaitForExit();
  
  proc.Close();
  
  }
  
  catch (Exception ex)
  
  {
  
  Response.Write("Error: " + ex.Message);
  
  }
  
  }