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• ffmpeg frame number or timecode overlay error ; need to reset frame count

  14 juillet 2015, par Samuel Mehr

  I’ve got timecode or frame number overlays as per other posts here, but I’m converting videos from AVCHD 59.94fps to H.264 30fps, and the timecode/frameN overlay is keeping the original values. So as I play the new video, the frames are labeled 1,3,5,7, ... but the video plays at normal speed. Is there an option to reset the frame count to the output video’s framerate ?

  thanks !

  ffmpeg -i in.mov -c:v libx264 -r 30 -s 1280x720 -b:v 5400 -vf "drawtext=fontsize=24:fontfile=FONTHERE.otf: timecode='00\:00\:00\:00': r=30: x=(w-tw)/2: y=h-(2*lh): fontcolor=white: box=1: boxcolor=0x00000099" testtest.mp4

• Rotate video adding black bars with ffmpeg

  13 février 2016, par MrMoog

  I have a lot of mp4 full hd videos made with a smartphone that looks rotated 90°.

  These files have proper rotation metadata and they plays correctly on VLC or other desktop mediaplayers, but I want to display them on my dvd player that does not handles mp4 metadata.

  So I tried to transpose with ffmpeg but I obtain 1080x1920 files, resolution obviously not recognized by tv/dvd-player.

  How can I transpose, resize and add black side (left/right) bars of the correct dimensions to get "correct" 1920x1080 videos ?

• Decode AAC to PCM with ffmpeg on android

  26 novembre 2012, par JeffG

  I have built ffmpeg 0.8.12 (love) with the android NDK (r8c) on ubuntu.
  I then use the generated library in another android application through JNI.

  Essentially what I want to do is pass a byte stream from java to my c jni function and use ffmpeg to decode it into a PCM audio buffer which will then be passed back to java to be played using Android's AudioTrack. I can successfully pass the buffer through to jni (have checked the values) and ffmpeg seems to initialise correctly, but when it tries to decode the first frame, it throws an error in the aac_decode_frame_int method in aacdec.c "channel element 0.0 is not allocated". The aac file plays fine and is valid.

  Here is my jni code to do the decoding

  jint Java_com_example_testffmpeg_MainActivity_decodeAacBytes(JNIEnv * env,
  
          jobject this, jbyteArray input, jint numBytes) {
  
  
  
      //copy bytes from java
  
      jbyte* bufferPtr = (*env)->GetByteArrayElements(env, input, NULL);
  
      uint8_t inputBytes[numBytes + FF_INPUT_BUFFER_PADDING_SIZE];
  
      memset(inputBytes, 0, numBytes + FF_INPUT_BUFFER_PADDING_SIZE);
  
      memcpy(inputBytes, bufferPtr, numBytes);
  
      (*env)->ReleaseByteArrayElements(env, input, bufferPtr, 0);
  
  
  
      av_register_all();
  
  
  
      AVCodec *codec = avcodec_find_decoder(CODEC_ID_AAC);
  
  
  
      if (codec == NULL) {
  
          LOGE("Cant find AAC codec\n");
  
          return 0;
  
      }
  
      LOGI("AAC codec found\n");
  
  
  
      AVCodecContext *avCtx = avcodec_alloc_context();
  
  
  
      if (avCtx == NULL) {
  
          LOGE("Could not allocate codec context\n");
  
          return 0;
  
      }
  
      LOGI("codec context allocated\n");
  
  
  
      if (avcodec_open2(avCtx, codec, NULL) < 0) {
  
          LOGE("Could not open codec\n");
  
          return 0;
  
      }
  
      LOGI("AAC codec opened");
  
  
  
      //the input buffer
  
      AVPacket avPacket;
  
      av_init_packet(&avPacket);
  
  
  
      LOGI("AVPacket initialised\n");
  
  
  
      avPacket.size = numBytes; //input buffer size
  
      avPacket.data = inputBytes; // the input buffer
  
  
  
      int outSize;
  
      int len;
  
      uint8_t *outbuf = malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
  
  
  
      while (avPacket.size > 0) {
  
          outSize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
  
          len = avcodec_decode_audio3(avCtx, (short *) outbuf, &outSize,
  
                  &avPacket);
  
  
  
          if (len < 0) {
  
              LOGE("Error while decoding\n");
  
              return 0;
  
          }
  
  
  
          if (outSize > 0) {
  
              LOGI("Decoded some stuff\n");
  
          }
  
  
  
          avPacket.size -= len;
  
          avPacket.data += len;
  
      }
  
  
  
      LOGI("Freeing memory\n");
  
  
  
      av_free_packet(&avPacket);
  
      avcodec_close(avCtx);
  
      av_free(avCtx);
  
  
  
      return 0;
  
  }

  The problem occurs in the call to avcodec_decode_audio3, when the decoding first occurs. I have stepped through the ffmpeg code, but can't find the problem. Any help would be greatly appreciated !