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• playing video in android using javacv ffmpeg

  20 février 2017, par d91

  I’m trying to play a video stored in sd card using javacv. Following is my code

  public class MainActivity extends AppCompatActivity {
  
  
  
      @Override
  
      protected void onCreate(Bundle savedInstanceState) {
  
          super.onCreate(savedInstanceState);
  
          setContentView(R.layout.activity_main);
  
          playthevideo();
  
      }
  
  
  
      protected void playthevideo() {
  
  
  
         String imageInSD = "/storage/080A-0063/dama/" + "test3.mp4";
  
  
  
         FFmpegFrameGrabber grabber = new FFmpegFrameGrabber(imageInSD);
  
  
  
         AndroidFrameConverter converterToBitmap = new AndroidFrameConverter();
  
  
  
         OpenCVFrameConverter.ToIplImage converterToipi = new OpenCVFrameConverter.ToIplImage();
  
  
  
             try {
  
                 Log.d("Tag", "try");
  
                 grabber.start();
  
                 Log.d("Tag", "started");
  
  
  
                 int i = 0;
  
                 IplImage grabbedImage = null;
  
  
  
                 ImageView mimg = (ImageView) findViewById(R.id.a);
  
  
  
                 grabber.setFrameRate(grabber.getFrameRate());
  
                 ArrayList<bitmap> bitmapArray = new ArrayList<bitmap>();
  
                 while (((grabbedImage = converterToipi.convert(grabber.grabImage())) != null)) {
  
  
  
                     Log.d("Tag", String.valueOf(i));
  
  
  
                     int width = grabbedImage.width();
  
  
  
                     int height = grabbedImage.height();
  
  
  
                     if (grabbedImage == null) {
  
                         Log.d("Tag", "error");
  
                     }
  
  
  
                     IplImage container = IplImage.create(width, height, IPL_DEPTH_8U, 4);
  
  
  
                     cvCvtColor(grabbedImage, container, CV_BGR2RGBA);
  
  
  
                     Bitmap bitmapnew = Bitmap.createBitmap(width, height, Bitmap.Config.ARGB_8888);
  
  
  
                     bitmapnew.copyPixelsFromBuffer(container.getByteBuffer());
  
  
  
                     if (bitmapnew == null) {
  
                         Log.d("Tag", "bit error");
  
                     }
  
  
  
                     mimg.setImageBitmap(bitmapnew);
  
  
  
                     mimg.requestFocus();
  
  
  
                     i++;
  
                 }
  
  
  
                 Log.d("Tag", "go");
  
  
  
             }
  
             catch(Exception e) {
  
  
  
            }
  
       }
  
  }
  
  </bitmap></bitmap>

  just ignore the tags because those are only for my testing purposes..
  When I run this code the main activity layout is still loading while android monitor shows the value of "i" (which is current frame number) and suddenly after frame number 3671 code exits the while loop and the imageview shows a frame which is not the end frame of that video(it somewhere around staring of the video).
  I was unable to find a way to show the grabbed frames from ffmpegframegrabber so i decided to show the image in imageview in this way. Can anybody tell me why I’m getting this error or another error non path to play and show the video in android activity ?
  BTW javacv 1.3.1 is imported correctly into my android dev environment. Thanks.

• How to encode 3840 nb_samples to a codec that asks for 1024 using ffmpeg

  26 juillet 2018, par Gabulit

  FFmpeg has an example muxing code on https://ffmpeg.org/doxygen/4.0/muxing_8c-example.html

  This code generates frame by frame video and audio. What I am trying to do is to change

  ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
  
                                         c->sample_rate, nb_samples);

  to

  ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
  
                                         c->sample_rate, 3840);

  so that it generates 3840 samples per channel instead of 1024 samples which is the default for nb_samples (aac codec).

  I tried to combine code from https://ffmpeg.org/doxygen/4.0/transcode_aac_8c-example.html which has an example on buffering the frames.

  My resulting program crashes when generating audio samples after a couple of frames when assigning *q++ a new value at the first iteration :

  /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
  
   * 'nb_channels' channels. */
  
  static AVFrame *get_audio_frame(OutputStream *ost)
  
  {
  
      AVFrame *frame = ost->tmp_frame;
  
      int j, i, v;
  
      int16_t *q = (int16_t*)frame->data[0];
  
      /* check if we want to generate more frames */
  
      if (av_compare_ts(ost->next_pts, ost->enc->time_base,
  
                        STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
  
          return NULL;
  
      for (j = 0; j nb_samples; j++) {
  
          v = (int)(sin(ost->t) * 10000);
  
          for (i = 0; i < ost->enc->channels; i++)
  
              *q++ = v;
  
          ost->t     += ost->tincr;
  
          ost->tincr += ost->tincr2;
  
      }
  
      frame->pts = ost->next_pts;
  
      ost->next_pts  += frame->nb_samples;
  
      return frame;
  
  }

  Maybe I don’t get the logic behind encoding.

  Here is the full source that i’ve come up with :

  https://paste.ee/p/b07qf

  The reason i am trying to accomplish this task is that I have a capture card sdk that outputs 2 channel 16 bit raw pcm 48000Hz which has 3840 samples per channel and I am trying to encode its output to aac. So basically if I get the muxing example to work with 3840 nb_samples this will help me understand the concept.

  I have already looked at How to encode resampled PCM-audio to AAC using ffmpeg-API when input pcm samples count not equal 1024 but the example uses "encodeFrame", which the examples on ffmpeg documentation doesn’t use or I am mistaken.

  Any help is greatly appreciated.

• ffmpeg configuration difficulty with filter_complex and hls

  4 février 2020, par akc42

  I am trying to set up ffmpeg so that it will record from a microphone and encode the results at the same time into a .flac file for later syncing up with some video I will be making.

  The microphone is plugged into a raspberry pi (4B) and I am currently trying it with a blue yeti mic, but I think I can do the same with a focusrite scarlett 2i2 plugged in instead. However I was puzzling about how to start the server recording and decided I could do it from a web browser if I made a simple nodejs server that spawned ffmpeg as a child process.

  But then I was inspired by this sample ffmpeg command which displays (on my desktop with an graphical interface) a volume meter

  ffmpeg -hide_banner -i 'http://distribution.bbb3d.renderfarming.net/video/mp4/bbb_sunflower_1080p_30fps_normal.mp4' -filter_complex "showvolume=rate=25:f=0.95:o=v:m=p:dm=3:h=80:w=480:ds=log:s=2" -c:v libx264 -c:a aac -f mpegts - | ffplay -window_title "Peak Volume" -i -

  What if I could stream the video produced by the showvolume filter to the web browser that I am using to control the ffmpeg process (NOTE I don’t want to send the audio with this). So I tried to read up on hls (since the control device will be an ipad - in fact that is what I will record the video on), and came up with this command

  ffmpeg -hide_banner -f alsa -ac 2 -ar 48k -i hw:CARD=Microphone -filter_complex "asplit=2[main][vol],[vol]showvolume=rate=25:f=0.95:o=v:m=p:dm=3:h=80:w=480:ds=log:s=2[vid]" -map [main] -c:a:0 flac recordings/session_$(date +%a_%d_%b_%Y___%H_%M_%S).flac -map [vid] -preset veryfast -g 25 -an -sc_threshold 0 -c:v:1 libx264 -b:v:1 2000k -maxrate:v:1 2200k -bufsize:v:3000k -f hls -hls_time 4 -hls_flags independent_segments delete_segments -strftime 1 -hls_segment_filename recordings/volume-%Y%m%d-%s.ts recordings/volume.m3u8

  The problem is I am finding the documentation a bit opaque as to what happens once I have generated two streams - the main audio and a video stream, and this command throws both a warning and an error :-

  The warning is Guessed Channel Layout for Input Stream #0.0 : stereo

  and the error is

  [NULL @ 0x1baa130] Unable to find a suitable output format for 'hls'
  
  hls: Invalid argument

  What I am trying to do is set up stream labels [main] and [vol] as I split the incoming audio into two parts, then I pass [vol] through the "showvolume" filter and end up with stream [vid].

  I think I need to then use -map to specify encoding the [main] stream down to flac and writing it out to file (The file exists after I run the command although they have zero length), and use another -map to pass through to the -f hls section. But I think I have something wrong at this stage.

  Can someone help me get this command right.