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• 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 to publish selfmade stream with ffmpeg and c++ to rtmp server ?

  25 octobre 2013, par Alexandr R

  Have a nice day to you, people !

  I am writing an application for Windows that will capture the screen and send the stream to Wowza server by rtmp (for broadcasting). My application use ffmpeg and Qt.
  I capture the screen with WinApi, convert a buffer to YUV444(because it's simplest) and encode frame as described at the file decoding_encoding.c (from FFmpeg examples) :

  ///////////////////////////
  
  //Encoder initialization
  
  ///////////////////////////
  
  avcodec_register_all();
  
  codec=avcodec_find_encoder(AV_CODEC_ID_H264);
  
  c = avcodec_alloc_context3(codec);
  
  c->width=scr_width;
  
  c->height=scr_height;
  
  c->bit_rate = 400000;
  
  int base_num=1;
  
  int base_den=1;//for one frame per second
  
  c->time_base= (AVRational){base_num,base_den};
  
  c->gop_size = 10;
  
  c->max_b_frames=1;
  
  c->pix_fmt = AV_PIX_FMT_YUV444P;
  
  av_opt_set(c->priv_data, "preset", "slow", 0);
  
  
  
  frame = avcodec_alloc_frame();
  
  frame->format = c->pix_fmt;
  
  frame->width  = c->width;
  
  frame->height = c->height;
  
  
  
  for(int counter=0;counter<10;counter++)
  
  {
  
  ///////////////////////////
  
  //Capturing Screen
  
  ///////////////////////////
  
      GetCapScr(shotbuf,scr_width,scr_height);//result: shotbuf is filled by screendata from HBITMAP
  
  ///////////////////////////
  
  //Convert buffer to YUV444 (standard formula)
  
  //It's handmade function because of problems with prepare buffer to swscale from HBITMAP
  
  ///////////////////////////
  
      RGBtoYUV(shotbuf,frame->linesize,frame->data,scr_width,scr_height);//result in frame->data
  
  ///////////////////////////
  
  //Encode Screenshot
  
  ///////////////////////////
  
      av_init_packet(&pkt);
  
      pkt.data = NULL;    // packet data will be allocated by the encoder
  
      pkt.size = 0;
  
      frame->pts = counter;
  
      avcodec_encode_video2(c, &pkt, frame, &got_output);
  
      if (got_output) 
  
      {
  
          //I think that  sending packet by rtmp  must be here!
  
          av_free_packet(&pkt);             
  
  
  
      }
  
  
  
  }
  
  // Get the delayed frames
  
  for (int got_output = 1,i=0; got_output; i++)
  
  {
  
      ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
  
      if (ret < 0)
  
          {
  
              fprintf(stderr, "Error encoding frame\n");
  
              exit(1);
  
          }
  
          if (got_output)
  
          {
  
          //I think that  sending packet by rtmp  must be here!
  
          av_free_packet(&pkt);      
  
          }
  
  }
  
  
  
  ///////////////////////////
  
  //Deinitialize encoder
  
  ///////////////////////////
  
  avcodec_close(c);
  
  av_free(c);
  
  av_freep(&frame->data[0]);
  
  avcodec_free_frame(&frame);

  I need to send video stream generated by this code to RTMP server.
  In other words, I need c++/c analog for this command :

  ffmpeg -re -i "sample.h264" -f flv rtmp://sample.url.com/screen/test_stream

  It's useful, but I don't want to save stream to file, I want to use ffmpeg libraries for realtime encoding screen capture and sending encoded frames to RTMP server inside my own application.
  Please give me a little example how to initialize AVFormatContext properly and to send my encoded video AVPackets to server.

  Thanks.

• Encoding H.264 CBR videos with FFmpeg

  27 août 2015, par Cornstalks

  I’m trying to encode a video with ffmpeg into H.264 (via the libx264 library) with a constant bit rate. I know, I know, VBR is often preferred, but for this specific job I’m required to use CBR (just as long as it’s so many kilobytes per second ; it doesn’t have to be an exact kilobytes per frame, afaik). My sample video I’m using to test is from here : http://a1408.g.akamai.net/5/1408/1388/2005110403/1a1a1ad948be278cff2d96046ad90768d848b41947aa1986/sample_iTunes.mov.zip (it comes from http://support.apple.com/kb/HT1425)

  I can get a constant bit rate when encoding the video with MPEG-4 Video (using the commands ffmpeg -i sample_iTunes.mov -b 819968 -minrate 819968 -maxrate 819968 out.mov), and the bit rate is as expected. Reading the video’s specs via the QuickTime Inspector, it’s got a data rate of 844.94 kbit/s. Cool.

  However, when I change the codec to libx264, it seems to completely ignore my bitrate requests ! The command I’m trying is "ffmpeg -i sample_iTunes.mov -vcodec libx264 -vpre medium -b 819968 -vb 819968 -minrate 819968 -maxrate 819968 -bufsize 400000 test.mov". But when I check the video’s specs via the QuickTime Inspector, it’s got a data rate of 254.74 kbit/s. WTF ? That’s not even close !

  I’ve tried changing so many parameters and adding tons of different things, and I’ve spent 2 days googling this, but I can’t seem to get it to work. If I encode the video with the MainConcept H.264 encoder, I can get a constant bitrate, but I need this to work with ffmpeg.

  If someone can help me figure out how to do CBR H.264 encoding with FFmpeg, I will love you forever !