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• Intsallation of ffmpeg-php on Amazon Ec2 Linux AMI

  1er décembre 2015, par Tanny

  I am about two days into attempting to install FFMPEG-php with dependencies on an AWS EC2 instance running the Amazon Linux AMI. I’ve installed FFMPEG, and have read reportedly successful instructions on installing on Red Hat/Fedora. I have followed a number of tutorials and forum articles to do so, but have had no luck yet. As far as I can tell, the main problems are as followed :

  I have installed all the dependency for ffmpeg-php. I run the following command successfully.

  $wget http://downloads.sourceforge.net/project/ffmpeg-php/ffmpeg-php/0.6.0/ffmpeg-php-0.6.0.tbz2
  $tar xvfj ffmpeg-php-0.6.0.tbz2
  $phpize

  But when I run the following command it throw the error like below :

  $sudo ./configure

  configure : error : ffmpeg shared libraries not found. Make sure ffmpeg is compiled as shared libraries using the —enable-shared option}

  I have used enable shared option with shared enable option but it throw the same error.

  On to my question : Has anyone successfully installed FFMPEG-php on Amazon Linux ? Is there a fundamental incompatibility ? If anyone could share specific instructions on installing ffmpeg-php on amazon linux I would be greatly appreciative. Any other insights/experiences would also be appreciated.

• ffmpeg c++/cli wrapper for using in c# . AccessViolationException after call dll function by it's pointer

  25 juillet 2015, par skynet_v

  My target is to write a c++/cli wrap arount ffmpeg library, using by importing ffmpeg functions from dll-modules.
  Later I will use this interface in c#.
  This is my challenge, don’t ask me why))

  So i’ve implemented Wrap class, which is listed below :

  namespace FFMpegWrapLib 
  
  {
  
      public class Wrap
  
      {
  
      private:
  
  
  
      public:
  
          //wstring libavcodecDllName = "avcodec-56.dll";
  
          //wstring libavformatDllName = "avformat-56.dll";
  
          //wstring libswscaleDllName = "swscale-3.dll";
  
          //wstring libavutilDllName = "avutil-54.dll";
  
  
  
          HMODULE libavcodecDLL;
  
          HMODULE libavformatDLL; 
  
          HMODULE libswsscaleDLL;
  
          HMODULE libavutilDLL;
  
  
  
          AVFormatContext     **pFormatCtx = nullptr;
  
          AVCodecContext      *pCodecCtxOrig = nullptr;
  
          AVCodecContext      *pCodecCtx = nullptr;
  
          AVCodec             **pCodec = nullptr;
  
          AVFrame             **pFrame = nullptr;
  
          AVFrame             **pFrameRGB = nullptr;
  
          AVPacket            *packet = nullptr;
  
          int                 *frameFinished;
  
          int                 numBytes;
  
          uint8_t             *buffer = nullptr;
  
          struct SwsContext   *sws_ctx = nullptr;
  
  
  
          void                Init();
  
          void                AVRegisterAll();
  
          void                Release();
  
          bool                SaveFrame(const char *pFileName, AVFrame * frame, int w, int h);
  
          bool                GetStreamInfo();
  
          int                 FindVideoStream();
  
          bool                OpenInput(const char* file);
  
          AVCodec*            FindDecoder();
  
          AVCodecContext*     AllocContext3();
  
          bool                CopyContext();
  
          bool                OpenCodec2();
  
          AVFrame*            AllocFrame();
  
          int                 PictureGetSize();
  
          void*               Alloc(size_t size);
  
          int                 PictureFill(AVPicture *, const uint8_t *, enum AVPixelFormat, int, int);
  
          SwsContext*         GetSwsContext(int, int, enum AVPixelFormat, int, int, enum AVPixelFormat, int, SwsFilter *, SwsFilter *, const double *);
  
          int                 ReadFrame(AVFormatContext *s, AVPacket *pkt);
  
          int                 DecodeVideo2(AVCodecContext *avctx, AVFrame *picture, int *got_picture_ptr, const AVPacket *avpkt);
  
          int                 SwsScale(struct SwsContext *c, const uint8_t *const srcSlice[], const int srcStride[], int srcSliceY, int srcSliceH, uint8_t *const dst[], const int dstStride[]);
  
          void                PacketFree(AVPacket *pkt);
  
          void                BufferFree(void *ptr);
  
          void                FrameFree(AVFrame **frame);
  
          int                 CodecClose(AVCodecContext *);
  
          void                CloseInput(AVFormatContext **);
  
          bool                SeekFrame(AVFormatContext *s, int stream_index, int64_t timestamp, int flags);
  
  
  
          Wrap();
  
          ~Wrap();
  
  
  
          bool                GetVideoFrame(char* str_in_file, char* str_out_img, uint64_t time);
  
      };
  
  
  
      public ref class managedWrap
  
      {
  
      public:
  
  
  
          managedWrap(){}
  
          ~managedWrap(){ delete unmanagedWrap; }
  
  
  
          bool GetVideoFrameToFile(char* str_in_file, char* str_out_img, uint64_t time)
  
          {
  
              return unmanagedWrap->GetVideoFrame(str_in_file, str_out_img, time);
  
          }
  
  
  
          static Wrap* unmanagedWrap = new Wrap();
  
      };
  
  }

  So the imports to libavcodec and etc. are succesful.
  The problem is in AccessViolationException during calling dll func, for example, in OpenInput (i.e. av_open_input in native ffmpeg library)

  The OpenInput func code is below :

  bool FFMpegWrapLib::Wrap::OpenInput(const char* file)
  
  {
  
      typedef int avformat_open_input(AVFormatContext **, const char *, AVInputFormat *, AVDictionary **);
  
  
  
      avformat_open_input* pavformat_open_input = (avformat_open_input *)GetProcAddress(libavformatDLL, "avformat_open_input");
  
      if (pavformat_open_input == nullptr)
  
      {
  
          throw exception("Unable to find avformat_open_input function address in libavformat module");
  
          return false;
  
      }
  
  
  
      //pin_ptr<avformatcontext> pinFormatContext = &amp;(new interior_ptr<avformatcontext>(pCodecCtx));
  
      pFormatCtx = new AVFormatContext*;
  
      //*pFormatCtx = new AVFormatContext;
  
  
  
  
  
      int ret = pavformat_open_input(pFormatCtx, file, NULL, NULL); // here it fails
  
  
  
      return ret == 0;
  
  }
  
  </avformatcontext></avformatcontext>

  So the problem, i think, is that class-fields of Wrap class are in secure memory. And ffmpeg works with native memory, initialising pFormatCtx variable by it’s address.
  Can I avoid this, or it is impossible ?

• Mix video and audio to mp4 file with ffmpeg but audio does't keep step with video when playback

  28 juillet 2015, par dragonfly

  I managed to write a program to record video(h264/aac) on android with ffmpeg. The detail is as follows :

  1. Implement android.hardware.Camera.PreviewCallback to capture every frame from camera (yuv image) and send it to the ffmpeg in the jni layer.

     @Override
     
     public void onPreviewFrame(byte[] data, Camera camera) {
     
         // Log.d(TAG, "onPreviewFrame");
     
         if (mIsRecording) {
     
             // Log.d(TAG, "handlePreviewFrame");
     
             Parameters param = camera.getParameters();
     
             Size s = param.getPreviewSize();
     
             handlePreviewFrame(data, s.width, s.height, mBufSize);
     
         }
     
         camera.addCallbackBuffer(mPreviewBuffer);
     
     }
     
     
     
     
     
     private void handlePreviewFrame(byte[] data, int width, int height, int size) {
     
     
     
         if (mFormats == ImageFormat.NV21) {
     
                 //process the yuv data
     
     
     
         }
     
     
     
         synchronized (mMuxer) {
     
             //jni api
     
             mMuxer.putAudioVisualData(mYuvData, size, 0);
     
         }
     
     }

  2. Use android.media.AudioRecord to read the pcm data from the microphone, write pcm data to ffmpeg in the jni layer in a loop.

     while (this.isRecording) {
     
         int ret = audioRecord.read(tempBuffer, 0, 1024);
     
     
     
         if (ret == AudioRecord.ERROR_INVALID_OPERATION) {
     
             throw new IllegalStateException(
     
                     "read() returned AudioRecord.ERROR_INVALID_OPERATION");
     
         } else if (ret == AudioRecord.ERROR_BAD_VALUE) {
     
             throw new IllegalStateException("read() returned AudioRecord.ERROR_BAD_VALUE");
     
         } else if (ret == AudioRecord.ERROR_INVALID_OPERATION) {
     
             throw new IllegalStateException(
     
                     "read() returned AudioRecord.ERROR_INVALID_OPERATION");
     
         }
     
     
     
         // 处理数据
     
         handleAudioData(tempBuffer, ret);
     
     }
     
     
     
     private void handleAudioData(short[] data, int size)
     
     {
     
         // convert to byte[]
     
         //Log.d("VideoCaptureActivity", "handleAudioData");
     
         ByteBuffer buffer = ByteBuffer.allocate(data.length * 2);
     
         buffer.order(ByteOrder.LITTLE_ENDIAN);
     
         buffer.asShortBuffer().put(data);
     
         buffer.limit(size * 2);
     
         byte[] bytes = buffer.array();
     
         synchronized (muxing) {
     
         Log.d(TAG, "putAudio Data :" + size*2);
     
         muxing.putAudioVisualData(bytes, size * 2, 1);
     
         }
     
     }

  3. mix audio and video data in the jni layer. I refer to the example : https://ffmpeg.org/doxygen/trunk/muxing_8c-source.html

  The problem is that the example demonstrates audio and video encoding from some dummy source data generated on the fly. I need to encode audio from microphone and video from camera.

  I think the reason of my failure is that the pts in the expample is not applicable for my situation. my av function code is as follows :

  static int write_video_frame(AVFormatContext *oc, OutputStream *ost, char *data,
  
          int size) {
  
      int ret;
  
      AVCodecContext *c;
  
      int got_packet = 0;
  
  
  
      c = ost->st->codec;
  
  
  
      AVPacket pkt = { 0 };
  
      av_init_packet(&pkt);
  
  
  
      if (!video_st.hwcodec) {
  
          if (ost->zoom) {
  
              zoom(oc, ost, data);
  
          } else {
  
              avpicture_fill((AVPicture*) ost->frame, (const uint8_t *) data,
  
                      c->pix_fmt, c->width, c->height);
  
          }
  
          av_frame_make_writable(ost->frame);
  
          //ost->frame->pts = ost->next_pts++;
  
          ost->frame->pts = frame_count;
  
          /* encode the image */
  
          //ALOGI("avcodec_encode_video2 start");
  
          ret = avcodec_encode_video2(c, &pkt, ost->frame, &got_packet);
  
          //ALOGI("avcodec_encode_video2 end");
  
          if (ret < 0) {
  
              ALOGE("Error encoding video frame: %s", av_err2str(ret));
  
              return -1;
  
          }
  
      } else {
  
          if (size != 0) {
  
              pkt.data = (uint8_t *) data;
  
              pkt.size = size;
  
              pkt.pts = pkt.dts = ost->next_pts++;
  
              got_packet = 1;
  
          }
  
      }
  
  
  
      if (got_packet) {
  
          //ALOGI("video write_frame start");
  
          //pkt.pts = (int) timestamp;
  
          ret = write_frame(oc, &c->time_base, ost->st, &pkt);
  
          //ALOGI("video write_frame end");
  
          if (ret < 0) {
  
              ALOGE("Error while writing video frame: %s", av_err2str(ret));
  
              return -1;
  
          }
  
      }
  
      frame_count++;
  
      return 0;
  
  }
  
  
  
  
  
  
  
  
  
  
  
  static int write_audio_frame(AVFormatContext *oc, OutputStream *ost, char *data) {
  
      AVCodecContext *c;
  
      AVPacket pkt = { 0 }; // data and size must be 0;
  
      AVFrame *frame;
  
      int ret;
  
      int got_packet;
  
      int dst_nb_samples;
  
  
  
      av_init_packet(&pkt);
  
      c = ost->st->codec;
  
  
  
      if (audio_st.speex_echo_cancellation == 1
  
              && g_audio_echo_play_queue->start_flag == 1) {
  
          //ALOGI("encode_audio_handler in echo_cancel");
  
          QUEUE_ITEM* item = Get_Queue_Item(g_audio_echo_play_queue);
  
          if (item) {
  
              speex_dsp_echo_play_back((spx_int16_t *) item->data);
  
              //ALOGI("encode_audio_handler echo_play begin speex_echo_play_back");
  
              short *echo_processed = (short *) av_malloc(160 * sizeof(short));
  
              speex_dsp_echo_capture((spx_int16_t *) data, echo_processed);
  
              memcpy(data, (uint8_t *) echo_processed, 160);
  
              av_free(echo_processed);
  
              Free_Queue_Item(item, 1);
  
          }
  
      }
  
  
  
      frame = ost->tmp_frame;
  
      //update pts
  
      //frame->pts = ost->next_pts;
  
      //ost->next_pts += frame->nb_samples;
  
      if (frame) {
  
          /* convert samples from native format to destination codec format, using the resampler */
  
          /* compute destination number of samples */
  
          dst_nb_samples = av_rescale_rnd(
  
                  swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
  
                  c->sample_rate, c->sample_rate, AV_ROUND_UP);
  
  
  
          memcpy(frame->data[0], data, frame->nb_samples * 2);
  
          //frame->data[0] = data;
  
  
  
          /* when we pass a frame to the encoder, it may keep a reference to it
  
           * internally;
  
           * make sure we do not overwrite it here
  
           */
  
          ret = av_frame_make_writable(ost->frame);
  
          if (ret < 0) {
  
              ALOGE("write_audio_frame av_frame_make_writable ERROR %s",
  
                      av_err2str(ret));
  
              return -1;
  
          }
  
  
  
          /* convert to destination format */
  
          ret = swr_convert(ost->swr_ctx, ost->frame->data, dst_nb_samples,
  
                  (const uint8_t **) frame->data, frame->nb_samples);
  
  
  
          if (ret < 0) {
  
              ALOGI("Error while converting %s", av_err2str(ret));
  
              return -1;
  
          }
  
          frame = ost->frame;
  
          frame->pts = av_rescale_q(ost->samples_count,
  
                  (AVRational ) { 1, c->sample_rate }, c->time_base);
  
          ost->samples_count += dst_nb_samples;
  
      }
  
      ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
  
  
  
      if (ret < 0) {
  
          ALOGE("Error encoding audio frame: %s", av_err2str(ret));
  
          return -1;
  
      }
  
  
  
      if (got_packet) {
  
          //pkt.pts = (int) timestamp;
  
  
  
          ret = write_frame(oc, &c->time_base, ost->st, &pkt);
  
          if (ret < 0) {
  
              ALOGE("Error while writing audio frame: %s", av_err2str(ret));
  
              return -1;
  
          }
  
      }
  
      return (frame || got_packet) ? 0 : 1;
  
  }

  How do I deal with the pts of video and audio stream for my situation ? Who can give me some advice ?

  Can I ignore the pts provided by ffmpeg and calculate the pts in the java layer by myself and transmit it to ffmpeg ?