Recherche avancée

Sur d’autres sites (10153)

• Availability of WebM (VP8) Video Hardware IP Designs

  10 janvier 2011, par noreply@blogger.com (John Luther)

  Hello from the frigid city of Oulu, in the far north of Finland. Our WebM hardware development team, formerly part of On2 Technologies, is now up-to-speed and working hard on a number of video efforts for WebM.
  

  • VP8 (the video codec used in WebM) hardware decoder IP is available from Google for semiconductor companies who want to support high-quality WebM playback in their chipsets.
  • The Oulu team will release the first VP8 video hardware encoder IP in the first quarter of 2011. We have the IP running in an FPGA environment, and rigorous testing is underway. Once all features have been tested and implemented, the encoder will be launched as well.

  WebM video hardware IPs are implemented and delivered as RTL (VHDL/Verilog) source code, which is a register-level hardware description language for creating digital circuit designs. The code is based on the Hantro brand video IP from On2, which has been successfully deployed by numerous chipset companies around the world. Our designs support VP8 up to 1080p resolution and can run 30 or 60fps, depending on the foundry process and hardware clock frequency.

  The WebM/VP8 hardware decoder implementation has already been licensed to over twenty partners and is proven in silicon. We expect the first commercial chips to integrate our VP8 decoder IP to be available in the first quarter of 2011. For example, Chinese semiconductor maker Rockchip last week demonstrated full WebM hardware playback on their new RK29xx series processor at CES in Las Vegas (video below).
  

  
  Note : To view the video in WebM format, ensure that you’ve enrolled in the YouTube HTML5 trial and are using a WebM-compatible browser. You can also view the video on YouTube.

  Hardware implementations of the VP8 encoder also bring exciting possibilities for WebM in portable devices. Not only can hardware-accelerated devices play high-quality WebM content, but hardware encoding also enables high-resolution, real-time video communications apps on the same devices. For example, when VP8 video encoding is fully off-loaded to a hardware accelerator, you can run 720p or even 1080p video conferencing at full framerate on a portable device with minimal battery use.

  The WebM hardware video IP team will be focusing on further developing the VP8 hardware designs while also helping our semiconductor partners to implement WebM video compression in their chipsets. If you have any questions, please visit our Hardware page.

  Happy New Year to the WebM community !

  Jani Huoponen, Product Manager
  Aki Kuusela, Engineering Manager

  

• Android AudioRecord to FFMPEG encode native AAC

  8 mars 2013, par Curtis Kiu

  I am doing video chatting in android and i would like to port ffmpeg to stream rtsp or rtmp but now i have a try in RTSP first.
  Somehow the problem now is av_write_frame or av_interleaved_write_frame is fail to work or just crash.
  Maybe...
  AudioRecord Sample format is not equals to FFMPEG setting
  Frame receive is not equals

  So code... AudioRecorder
  http://pastebin.com/iWtB3Jhy
  package com.curtis.broadcaster.Publisher ;

  import android.app.Activity;
  
  import android.graphics.Bitmap;
  
  import android.media.AudioFormat;
  
  import android.media.AudioRecord;
  
  import android.media.AudioRecord.OnRecordPositionUpdateListener;
  
  import android.media.MediaRecorder;
  
  import android.os.Bundle;
  
  import android.util.Log;
  
  
  
  public class Publisher extends Activity {
  
      private int mAudioBufferSize;
  
      private int mAudioBufferSampleSize;
  
      private AudioRecord mAudioRecord;
  
      private boolean inRecordMode = false;
  
      private short[] audioBuffer;
  
      private String Tag = "Publisher/Publisher.java";
  
  
  
      public void onCreate(Bundle savedInstanceState) {
  
          Log.i(Tag, "|| onCreate()");
  
          super.onCreate(savedInstanceState);
  
          initAudioRecord();
  
          Log.i(Tag, "-- End onCreate()");
  
      }
  
  
  
      @Override
  
      public void onResume() {
  
          Log.i(Tag, "|| onResume()");
  
          super.onResume();
  
          inRecordMode = true;
  
          Thread t = new Thread(new Runnable() {
  
  
  
              public void run() {
  
                  Log.i(Tag, "|| Run Threat t");
  
                  getSamples();
  
                  Log.i(Tag, "-- End Threat t");
  
              }
  
          });
  
          t.start();
  
          Log.i(Tag, "-- End onResume()");
  
      }
  
  
  
      protected void onPause() {
  
          Log.i(Tag, "|| Run onPause()");
  
          inRecordMode = false;
  
          super.onPause();
  
          Log.i(Tag, "-- End onPause()");
  
      }
  
  
  
      @Override
  
      protected void onDestroy() {
  
          Log.i(Tag, "|| Run onDestroy()");
  
          if (mAudioRecord != null) {
  
              mAudioRecord.release();
  
              Log.i(Tag + " onDestroy", "mAudioRecord.release()");
  
          }
  
          jniStopAll();
  
          super.onDestroy();
  
          android.os.Process.killProcess(android.os.Process.myPid());
  
          Log.i(Tag, "-- End onDestroy()");
  
      }
  
  
  
      public OnRecordPositionUpdateListener mListener = new OnRecordPositionUpdateListener() {
  
  
  
          public void onPeriodicNotification(AudioRecord recorder) {
  
              Log.i(Tag + " mListener(onPeriodicNotification)", "time is "
  
                      + System.currentTimeMillis());
  
              jniSetAudioSample(audioBuffer);
  
          //  audioBuffer = new short[mAudioBufferSampleSize];
  
          }
  
  
  
          public void onMarkerReached(AudioRecord recorder) {
  
              Log.i(Tag + " mListener(onMarkerReached)",
  
                      "time is " + System.currentTimeMillis());
  
              inRecordMode = false;
  
              recorder.stop();
  
              Log.i(Tag, "recorder.stop()");
  
          }
  
      };
  
  
  
      private void initAudioRecord() {
  
          try {
  
              jniCheck();
  
              int sampleRate = 44100;
  
              int channelConfig = AudioFormat.CHANNEL_IN_MONO;
  
              int audioFormat = AudioFormat.ENCODING_PCM_16BIT;
  
              mAudioBufferSize = 2 * AudioRecord.getMinBufferSize(sampleRate,
  
                      channelConfig, audioFormat);
  
              mAudioBufferSampleSize = mAudioBufferSize / 2;
  
              Log.i(Tag, "Buffer Size " + mAudioBufferSize);
  
              Log.i(Tag, "new AudioRecord begin");
  
  
  
              mAudioRecord = new AudioRecord(MediaRecorder.AudioSource.MIC,
  
                      sampleRate, channelConfig, audioFormat, mAudioBufferSize);
  
              Log.i(Tag, "new AudioRecord end");
  
  
  
              jniInitFFMpeg();
  
          } catch (IllegalArgumentException e) {
  
              Log.i(Tag, "initAudioRecord go Errors");
  
              e.printStackTrace();
  
          }
  
  
  
          // mAudioRecord.setNotificationMarkerPosition(10000);
  
          mAudioRecord.setPositionNotificationPeriod(1024);
  
          mAudioRecord.setRecordPositionUpdateListener(mListener);
  
  
  
          int audioRecordState = mAudioRecord.getState();
  
          if (audioRecordState != AudioRecord.STATE_INITIALIZED) {
  
              finish();
  
          }
  
  
  
      }
  
  
  
      private void getSamples() {
  
          Log.i(Tag, "|| getSamples()");
  
          if (mAudioRecord == null)
  
              return;
  
  
  
          audioBuffer = new short[mAudioBufferSampleSize];
  
          mAudioRecord.startRecording();
  
          int audioRecordingState = mAudioRecord.getRecordingState();
  
          if (audioRecordingState != AudioRecord.RECORDSTATE_RECORDING) {
  
              finish();
  
          }
  
          while (inRecordMode) {
  
              int samplesRead = mAudioRecord.read(audioBuffer, 0,
  
                      mAudioBufferSampleSize);
  
              Log.i(Tag, "getSamples >>SamplesRead : " + samplesRead);
  
          }
  
          mAudioRecord.stop();
  
          Log.i(Tag, "mAudioRecord.stop()");
  
      }
  
  
  
      private native void jniCheck();
  
  
  
      private native void jniInitFFMpeg();
  
  
  
      private native void jniSetAudioSample(short[] audioBuffer);
  
  
  
      private native void jniStopAll();
  
  
  
      static {
  
          System.loadLibrary("ffmpeg");
  
          System.loadLibrary("testerv4");
  
  
  
      }
  
  
  
  }

  FFMPEG JNI http://pastebin.com/hgPva35b

  #include 
  
  #include <android></android>log.h>
  
  #include <android></android>bitmap.h>
  
  
  
  #include 
  
  #include 
  
  #include 
  
  #include 
  
  #include <sys></sys>time.h>
  
  #include "libavformat/rtsp.h"
  
  
  
  #include <libavutil></libavutil>mathematics.h>
  
  #include <libavformat></libavformat>avformat.h>
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  #include <libswscale></libswscale>swscale.h>
  
  
  
  #undef exit
  
  /* Log System */
  
  #define  LOG_TAG    "FFMPEGSample - v4a"
  
  #define DEBUG_TAG   "FFMPEG-AUDIO PART"
  
  #define  LOGI(...)  __android_log_print(ANDROID_LOG_INFO,LOG_TAG,__VA_ARGS__)
  
  #define  LOGE(...)  __android_log_print(ANDROID_LOG_ERROR,LOG_TAG,__VA_ARGS__)
  
  
  
  /* 5 seconds stream duration */
  
  #define STREAM_DURATION   5.0
  
  #define STREAM_FRAME_RATE 25 /* 25 images/s */
  
  #define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
  
  #define STREAM_PIX_FMT      PIX_FMT_YUV420P /* default pix_fmt */
  
  #define VIDEO_CODEC_ID      CODEC_ID_FLV1
  
  #define AUDIO_CODEC_ID      CODEC_ID_AAC
  
  
  
  static int sws_flags = SWS_BICUBIC;
  
  int mode = 1; //1 = only audio, 2 = only video, 3 = both video and audio
  
  
  
  AVFormatContext *avForCtx;
  
  //AVFormatContext *oc;
  
  AVStream *audio_st, *video_st;
  
  double audio_pts, video_pts;
  
  int frameCount, audioFrameCount, start;
  
  char *url;
  
  
  
  /*Audio Declare*/
  
  float t, tincr, tincr2;
  
  int16_t *samples;
  
  uint8_t *audio_outbuf;
  
  int audio_outbuf_size;
  
  int audio_input_frame_size;
  
  
  
  AVFormatContext *createAVFormatContext();
  
  AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id);
  
  void open_video(AVFormatContext *oc, AVStream *st);
  
  void open_audio(AVFormatContext *oc, AVStream *st);
  
  AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id);
  
  void write_audio_frame(AVFormatContext *oc, AVStream *st);
  
  void write_video_frame(AVFormatContext *oc, AVStream *st);
  
  void init();
  
  void setAudioSample(unsigned char *inSample[]);
  
  void stopAll();
  
  
  
  /*/////////////////////////////////JNI Bridge////////////////////////////////////// */
  
  void Java_com_curtis_broadcaster_Publisher_Publisher_jniCheck(JNIEnv* env,
  
          jobject this) {
  
      LOGI("-@ JNI work fine @-");
  
  }
  
  void Java_com_curtis_broadcaster_Publisher_Publisher_jniInitFFMpeg(JNIEnv* env,
  
          jobject this) {
  
      LOGI("-@ Init Encorder @-");
  
  
  
      /* initialize libavcodec, and register all codecs and formats */
  
      avcodec_init();
  
      avcodec_register_all();
  
      av_register_all();
  
      avformat_network_init(); //ERROR
  
  
  
  
  
      /* allocate the output media context */
  
      avForCtx = createAVFormatContext();
  
      frameCount = 1;
  
      audioFrameCount = 1;
  
      start = 0;
  
  
  
      /* add the audio and video streams using the default format codecs
  
       and initialize the codecs */
  
      video_st = NULL;
  
      audio_st = NULL;
  
      if (mode == 1 || mode == 3) {
  
          audio_st = add_audio_stream(avForCtx, AUDIO_CODEC_ID);
  
          LOGI("(Init Encorder) - addAudioStream");
  
      }
  
      if (mode == 2 || mode == 3) {
  
          video_st = add_video_stream(avForCtx, VIDEO_CODEC_ID);
  
          LOGI("(Init Encorder) - addVideoStream");
  
  
  
      }
  
  
  
      //  av_dump_format(avForCtx, 0, "rtsp://192.168.1.104/live/live", 1);
  
      LOGI("(Init Encorder) - Waiting to call open_*");
  
  
  
      if (audio_st) {
  
          open_audio(avForCtx, audio_st);
  
          LOGI("(Init Encorder) - open_audio");
  
      }
  
  
  
      if (video_st) {
  
          open_video(avForCtx, video_st);
  
          LOGI("(Init Encorder) - open_video");
  
      }
  
  
  
      av_write_header(avForCtx);
  
      LOGI("-@ Finish Init Encorder @-");
  
  
  
  }
  
  
  
  void Java_com_curtis_broadcaster_Publisher_Publisher_jniSetAudioSample(
  
          JNIEnv* env, jobject this, unsigned char *inSample[]) {
  
      if (audio_st) {
  
          LOGI("-@ Start setAudioSample @-");
  
          samples = (int16_t *) inSample;
  
  
  
          write_audio_frame(avForCtx, audio_st);
  
          LOGI("-@ Finish setAudioSample @-");
  
      }
  
  }
  
  
  
  void Java_com_curtis_broadcaster_Publisher_Publisher_jniStopAll(JNIEnv* env,
  
          jobject this) {
  
      LOGI("-@ Stopping All @-");
  
      //close_audio(avForCtx, audio_st);
  
      //close_video(avForCtx, video_st);
  
      LOGI("-@ Stopped All @-");
  
  }
  
  /*/////////////////////////////END JNI Bridge////////////////////////////////////// */
  
  
  
  /* New Added Coding */
  
  AVFormatContext *createAVFormatContext() {
  
      LOGI("-@OPEN - createAVFormatContext@-");
  
  
  
      AVFormatContext *ctx = avformat_alloc_context();
  
      //  ctx->oformat = av_guess_format("flv", "rtmp://192.168.1.104/live/live",
  
      //      NULL);
  
      //  ctx->oformat = av_guess_format("flv", NULL, NULL);
  
  
  
      //if (!av_guess_format("flv", NULL, NULL)) {
  
  
  
      //LOGI("-flv Can not Guess Format-");
  
      //}
  
  
  
      ctx->oformat = av_guess_format("rtsp", NULL, NULL);
  
  
  
      if (!av_guess_format("rtsp", NULL, NULL)) {
  
  
  
          LOGI("-flv Can not Guess Format-");
  
      }
  
  
  
      /*
  
       LOGI("%d",avformat_alloc_output_context2(&amp;ctx, ctx->oformat, "flv",
  
       "rtmp://192.168.1.104/live/live"));
  
       if (!ctx) {
  
       LOGI("-@avformat_alloc_output_context2 fail@-");
  
       }*/
  
      //   LOGI("flv %d",avformat_alloc_output_context2(&amp;ctx, ctx->oformat, "flv",
  
      //   "rtmp://192.168.1.104/live/live"));
  
      //   LOGI("rtmp %d",avformat_alloc_output_context2(&amp;ctx, ctx->oformat, "rtmp",
  
      //   "rtmp://192.168.1.104/live/live"));
  
      //   LOGI("mpeg4 %d",avformat_alloc_output_context2(&amp;ctx, ctx->oformat, "mpeg4",
  
      //   "rtmp://192.168.1.104/live/live"));
  
      //   LOGI("NULL %d",avformat_alloc_output_context2(&amp;ctx, ctx->oformat, NULL,
  
      //   "rtmp://192.168.1.104/live/live"));
  
      avformat_alloc_output_context2(&amp;ctx, ctx->oformat, "sdp",
  
              "rtsp://192.168.1.104:1935/live/live");
  
  
  
      if (!ctx) {
  
          LOGI("-@avformat_alloc_output_context2 fail@-");
  
      }
  
  
  
      LOGI("-@CLOSE - createAVFormatContext@-");
  
  
  
      return ctx;
  
  }
  
  
  
  /**************************************************************/
  
  /* audio output */
  
  
  
  /*
  
   * add an audio output stream
  
   */
  
  AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id) {
  
      LOGI("-@OPEN - add_audio_stream@-");
  
  
  
      AVCodecContext *c;
  
      AVStream *st = avformat_new_stream(oc, avcodec_find_encoder(codec_id));
  
  
  
      if (!st) {
  
          LOGI("-@add_audio_stream - Could not alloc stream@-");
  
          exit(1);
  
      }
  
      st->id = 1;
  
  
  
      c = st->codec;
  
      c->codec_id = AUDIO_CODEC_ID;
  
      c->codec_type = AVMEDIA_TYPE_AUDIO;
  
  
  
      /* put sample parameters */
  
      c->sample_fmt = AV_SAMPLE_FMT_FLT;
  
      //c->sample_fmt = AV_SAMPLE_FMT_S16;
  
      c->bit_rate = 100000;
  
      c->sample_rate = 44100;
  
      c->channels = 1;
  
  
  
      // some formats want stream headers to be separate
  
      if (oc->oformat->flags &amp; AVFMT_GLOBALHEADER)
  
          c->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
      LOGI("-@Close - add_audio_stream@-");
  
  
  
      return st;
  
  }
  
  
  
  void open_audio(AVFormatContext *oc, AVStream *st) {
  
      LOGI("@- open_audio -@");
  
  
  
      AVCodecContext *c;
  
      AVCodec *codec;
  
  
  
      c = st->codec;
  
      c->strict_std_compliance = -2;
  
      /* find the audio encoder */
  
      codec = avcodec_find_encoder(c->codec_id);
  
      if (!codec) {
  
          LOGI("@- open_audio E:codec not found-@");
  
          exit(1);
  
      }
  
  
  
      /* open it */
  
      if (avcodec_open(c, codec) &lt; 0) {
  
          LOGI("%d",avcodec_open(c, codec));
  
          LOGI("@- open_audio E:could not open codec-@");
  
          exit(1);
  
      }
  
  
  
      /* init signal generator */
  
      t = 0;
  
      tincr = 2 * M_PI * 110.0 / c->sample_rate;
  
      /* increment frequency by 110 Hz per second */
  
      tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
  
  
  
      audio_outbuf_size = 10000;
  
      audio_outbuf = av_malloc(audio_outbuf_size);
  
  
  
      /* ugly hack for PCM codecs (will be removed ASAP with new PCM
  
       support to compute the input frame size in samples */
  
      if (c->frame_size &lt;= 1) {
  
          audio_input_frame_size = audio_outbuf_size / c->channels;
  
          switch (st->codec->codec_id) {
  
          case CODEC_ID_PCM_S16LE:
  
          case CODEC_ID_PCM_S16BE:
  
          case CODEC_ID_PCM_U16LE:
  
          case CODEC_ID_PCM_U16BE:
  
              audio_input_frame_size >>= 1;
  
              break;
  
          default:
  
              break;
  
          }
  
      } else {
  
          audio_input_frame_size = c->frame_size;
  
      }
  
      LOGI("audio_input_frame_size : %d",audio_input_frame_size);
  
      samples = av_malloc(audio_input_frame_size * 2 * c->channels);
  
      LOGI("@- Close open_audio -@");
  
  
  
  }
  
  
  
  /* prepare a 16 bit dummy audio frame of &#39;frame_size&#39; samples and
  
   &#39;nb_channels&#39; channels */
  
  void get_audio_frame(int16_t *samples, int frame_size, int nb_channels) {
  
      LOGI("@- get_audio_frame -@");
  
  
  
      int j, i, v;
  
      int16_t *q;
  
  
  
      q = samples;
  
      for (j = 0; j &lt; frame_size; j++) {
  
          v = (int) (sin(t) * 10000);
  
          for (i = 0; i &lt; nb_channels; i++)
  
              *q++ = v;
  
          t += tincr;
  
          tincr += tincr2;
  
          LOGI("@- audio_frame Looping -@");
  
      }
  
      LOGI("@- CLOSE get_audio_frame -@");
  
  
  
  }
  
  
  
  void write_audio_frame(AVFormatContext *oc, AVStream *st) {
  
      LOGI("@- write_audio_frame -@");
  
  
  
      AVCodecContext *c;
  
      AVPacket pkt;
  
      av_init_packet(&amp;pkt);
  
  
  
      c = st->codec;
  
  
  
      //get_audio_frame(samples, audio_input_frame_size, c->channels);
  
      LOGI("@- write_audio_frame : got frame from get_audio_frame -@");
  
  
  
      pkt.size
  
              = avcodec_encode_audio(c, audio_outbuf, audio_outbuf_size, samples);
  
      LOGI("%d",pkt.size);
  
  
  
      if (c->coded_frame &amp;&amp; c->coded_frame->pts != AV_NOPTS_VALUE)
  
          pkt.pts
  
                  = av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
  
      LOGI("%d",pkt.pts);
  
  
  
      pkt.flags |= AV_PKT_FLAG_KEY;
  
      pkt.stream_index = st->index;
  
      pkt.data = audio_outbuf;
  
      LOGI("Finish PKT");
  
  
  
      /* write the compressed frame in the media file */
  
      //  if (av_interleaved_write_frame(oc, &amp;pkt) != 0) {
  
      //  LOGI("@- write_audio_frame E:Error while writing audio frame -@");
  
      //  exit(1);
  
      //  }
  
  
  
      if (av_interleaved_write_frame(oc, &amp;pkt) != 0) {
  
          LOGI("Error while writing audio frame %d\n", audioFrameCount);
  
      } else {
  
          LOGI("Writing Audio Frame %d", audioFrameCount);
  
      }
  
  
  
      LOGI("@- CLOSE write_audio_frame -@");
  
      audioFrameCount++;
  
      av_free_packet(&amp;pkt);
  
  }
  
  
  
  void close_audio(AVFormatContext *oc, AVStream *st) {
  
      avcodec_close(st->codec);
  
  
  
      av_free(samples);
  
      av_free(audio_outbuf);
  
  }
  
  
  
  /**************************************************************/
  
  /* video output */
  
  
  
  AVFrame *picture, *tmp_picture;
  
  uint8_t *video_outbuf;
  
  int frame_count, video_outbuf_size;
  
  
  
  /* add a video output stream */
  
  AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id) {
  
      AVCodecContext *c;
  
      AVStream *st;
  
      AVCodec *codec;
  
  
  
      st = avformat_new_stream(oc, NULL);
  
      if (!st) {
  
          fprintf(stderr, "Could not alloc stream\n");
  
          exit(1);
  
      }
  
  
  
      c = st->codec;
  
  
  
      /* find the video encoder */
  
      codec = avcodec_find_encoder(codec_id);
  
      if (!codec) {
  
          fprintf(stderr, "codec not found\n");
  
          exit(1);
  
      }
  
      avcodec_get_context_defaults3(c, codec);
  
  
  
      c->codec_id = codec_id;
  
  
  
      /* put sample parameters */
  
      c->bit_rate = 400000;
  
      /* resolution must be a multiple of two */
  
      c->width = 352;
  
      c->height = 288;
  
      /* time base: this is the fundamental unit of time (in seconds) in terms
  
       of which frame timestamps are represented. for fixed-fps content,
  
       timebase should be 1/framerate and timestamp increments should be
  
       identically 1. */
  
      c->time_base.den = STREAM_FRAME_RATE;
  
      c->time_base.num = 1;
  
      c->gop_size = 12; /* emit one intra frame every twelve frames at most */
  
      c->pix_fmt = STREAM_PIX_FMT;
  
      if (c->codec_id == CODEC_ID_MPEG2VIDEO) {
  
          /* just for testing, we also add B frames */
  
          c->max_b_frames = 2;
  
      }
  
      if (c->codec_id == CODEC_ID_MPEG1VIDEO) {
  
          /* Needed to avoid using macroblocks in which some coeffs overflow.
  
           This does not happen with normal video, it just happens here as
  
           the motion of the chroma plane does not match the luma plane. */
  
          c->mb_decision = 2;
  
      }
  
      // some formats want stream headers to be separate
  
      if (oc->oformat->flags &amp; AVFMT_GLOBALHEADER)
  
          c->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
  
  
      return st;
  
  }
  
  
  
  AVFrame *alloc_picture(enum PixelFormat pix_fmt, int width, int height) {
  
      AVFrame * picture;
  
      uint8_t *picture_buf;
  
      int size;
  
  
  
      picture = avcodec_alloc_frame();
  
      if (!picture)
  
          return NULL;
  
      size = avpicture_get_size(pix_fmt, width, height);
  
      picture_buf = av_malloc(size);
  
      if (!picture_buf) {
  
          av_free(picture);
  
          return NULL;
  
      }
  
      avpicture_fill((AVPicture *) picture, picture_buf, pix_fmt, width, height);
  
      return picture;
  
  }
  
  
  
  void open_video(AVFormatContext *oc, AVStream *st) {
  
      AVCodec *codec;
  
      AVCodecContext *c;
  
  
  
      c = st->codec;
  
  
  
      /* find the video encoder */
  
      codec = avcodec_find_encoder(c->codec_id);
  
      if (!codec) {
  
          fprintf(stderr, "codec not found\n");
  
          exit(1);
  
      }
  
  
  
      /* open the codec */
  
      if (avcodec_open(c, codec) &lt; 0) {
  
          fprintf(stderr, "could not open codec\n");
  
          exit(1);
  
      }
  
  
  
      video_outbuf = NULL;
  
      if (!(oc->oformat->flags &amp; AVFMT_RAWPICTURE)) {
  
          /* allocate output buffer */
  
          /* XXX: API change will be done */
  
          /* buffers passed into lav* can be allocated any way you prefer,
  
           as long as they&#39;re aligned enough for the architecture, and
  
           they&#39;re freed appropriately (such as using av_free for buffers
  
           allocated with av_malloc) */
  
          video_outbuf_size = 200000;
  
          video_outbuf = av_malloc(video_outbuf_size);
  
      }
  
  
  
      /* allocate the encoded raw picture */
  
      picture = alloc_picture(c->pix_fmt, c->width, c->height);
  
      if (!picture) {
  
          fprintf(stderr, "Could not allocate picture\n");
  
          exit(1);
  
      }
  
  
  
      /* if the output format is not YUV420P, then a temporary YUV420P
  
       picture is needed too. It is then converted to the required
  
       output format */
  
      tmp_picture = NULL;
  
      if (c->pix_fmt != PIX_FMT_YUV420P) {
  
          tmp_picture = alloc_picture(PIX_FMT_YUV420P, c->width, c->height);
  
          if (!tmp_picture) {
  
              fprintf(stderr, "Could not allocate temporary picture\n");
  
              exit(1);
  
          }
  
      }
  
  }
  
  
  
  /* prepare a dummy image */
  
  void fill_yuv_image(AVFrame *pict, int frame_index, int width, int height) {
  
      int x, y, i;
  
  
  
      i = frame_index;
  
  
  
      /* Y */
  
      for (y = 0; y &lt; height; y++) {
  
          for (x = 0; x &lt; width; x++) {
  
              pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
  
          }
  
      }
  
  
  
      /* Cb and Cr */
  
      for (y = 0; y &lt; height / 2; y++) {
  
          for (x = 0; x &lt; width / 2; x++) {
  
              pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
  
              pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
  
          }
  
      }
  
  }
  
  
  
  void write_video_frame(AVFormatContext *oc, AVStream *st) {
  
      int out_size, ret;
  
      AVCodecContext *c;
  
      struct SwsContext *img_convert_ctx;
  
  
  
      c = st->codec;
  
  
  
      if (frame_count >= STREAM_NB_FRAMES) {
  
          /* no more frame to compress. The codec has a latency of a few
  
           frames if using B frames, so we get the last frames by
  
           passing the same picture again */
  
      } else {
  
          if (c->pix_fmt != PIX_FMT_YUV420P) {
  
              /* as we only generate a YUV420P picture, we must convert it
  
               to the codec pixel format if needed */
  
              if (img_convert_ctx == NULL) {
  
                  img_convert_ctx = sws_getContext(c->width, c->height,
  
                          PIX_FMT_YUV420P, c->width, c->height, c->pix_fmt,
  
                          sws_flags, NULL, NULL, NULL);
  
                  if (img_convert_ctx == NULL) {
  
                      fprintf(stderr,
  
                              "Cannot initialize the conversion context\n");
  
                      exit(1);
  
                  }
  
              }
  
              fill_yuv_image(tmp_picture, frame_count, c->width, c->height);
  
              sws_scale(img_convert_ctx, tmp_picture->data,
  
                      tmp_picture->linesize, 0, c->height, picture->data,
  
                      picture->linesize);
  
          } else {
  
              fill_yuv_image(picture, frame_count, c->width, c->height);
  
          }
  
      }
  
  
  
      if (oc->oformat->flags &amp; AVFMT_RAWPICTURE) {
  
          /* raw video case. The API will change slightly in the near
  
           future for that. */
  
          AVPacket pkt;
  
          av_init_packet(&amp;pkt);
  
  
  
          pkt.flags |= AV_PKT_FLAG_KEY;
  
          pkt.stream_index = st->index;
  
          pkt.data = (uint8_t *) picture;
  
          pkt.size = sizeof(AVPicture);
  
  
  
          ret = av_interleaved_write_frame(oc, &amp;pkt);
  
      } else {
  
          /* encode the image */
  
          out_size = avcodec_encode_video(c, video_outbuf, video_outbuf_size,
  
                  picture);
  
          /* if zero size, it means the image was buffered */
  
          if (out_size > 0) {
  
              AVPacket pkt;
  
              av_init_packet(&amp;pkt);
  
  
  
              if (c->coded_frame->pts != AV_NOPTS_VALUE)
  
                  pkt.pts = av_rescale_q(c->coded_frame->pts, c->time_base,
  
                          st->time_base);
  
              if (c->coded_frame->key_frame)
  
                  pkt.flags |= AV_PKT_FLAG_KEY;
  
              pkt.stream_index = st->index;
  
              pkt.data = video_outbuf;
  
              pkt.size = out_size;
  
  
  
              /* write the compressed frame in the media file */
  
              ret = av_interleaved_write_frame(oc, &amp;pkt);
  
          } else {
  
              ret = 0;
  
          }
  
      }
  
      if (ret != 0) {
  
          fprintf(stderr, "Error while writing video frame\n");
  
          exit(1);
  
      }
  
      frame_count++;
  
  }
  
  
  
  void close_video(AVFormatContext *oc, AVStream *st) {
  
      avcodec_close(st->codec);
  
      av_free(picture->data[0]);
  
      av_free(picture);
  
      if (tmp_picture) {
  
          av_free(tmp_picture->data[0]);
  
          av_free(tmp_picture);
  
      }
  
      av_free(video_outbuf);
  
  }

  Android Manifest has been set and init everything.
  Please give me some ideas..
  Some log message to yours http://pastebin.com/uPD5LyH2

• how in batch mode convert video files to webm

  18 mars 2012, par Tanya

  I have alot of MOV files from the camera.
  I want to convert all of them to WebM with same resolution and FPS.

  I know, that FFMPEG can be used for that, but not sure, how to use it to process files in batch mode.

  Need help, how to do it.