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• Make Qt Player codec independent

  16 mars 2016, par Tejas Virpariya

  I develop Qt application which can play more then one video file using bellow code.

  QMediaPlayer *player;
  
  QString fileName = "C:/username/test.h264";
  
  player->setmedia(QUrl::fromLocalFile(fileName));

  In starting I cannot play all types of video file, so I install codec on my system, now when my player start codec decoder start, and my CPU usage reach at high.(Show the bellow Image)

  

  You can see in above image right side bottom corner LAW(Red label) which saw external decoder started.

  Now, I want to make my Qt Player codec independent, means I know my player have to play only .h264 file, so I will use only h264 decoder and no need of audio so I will not use audio decoder.

  As per my knowledge, QMediaPlayer start decoder when it come in picture, correct me if i am wrong. So What can I do to stop external decoder and decode frame internally and play successfully ?

  EDIT : code for audio decode using FFmpeg

  FFmpegAudio.pro

  TARGET = fooAudioFFMPEG
  
  QT       += core gui qml quick widgets
  
  TEMPLATE = app
  
  SOURCES += main.cpp \
  
      mainwindow.cpp
  
  HEADERS += mainwindow.h \
  
      wrapper.h
  
  FORMS += mainwindow.ui
  
  QMAKE_CXXFLAGS += -D__STDC_CONSTANT_MACROS
  
  
  
  LIBS += -pthread
  
  LIBS += -L/usr/local/lib
  
  LIBS += -lavdevice
  
  LIBS += -lavfilter
  
  LIBS += -lpostproc
  
  LIBS += -lavformat
  
  LIBS += -lavcodec
  
  LIBS += -ldl
  
  LIBS += -lXfixes
  
  LIBS += -lXext
  
  LIBS += -lX11
  
  LIBS += -lasound
  
  LIBS += -lSDL
  
  LIBS += -lx264
  
  LIBS += -lvpx
  
  LIBS += -lvorbisenc
  
  LIBS += -lvorbis
  
  LIBS += -logg
  
  LIBS += -lopencore-amrwb
  
  LIBS += -lopencore-amrnb
  
  LIBS += -lmp3lame
  
  LIBS += -lfaac
  
  LIBS += -lz
  
  LIBS += -lrt
  
  LIBS += -lswscale
  
  LIBS += -lavutil
  
  LIBS += -lm

  mainwindow.h

  #ifndef MAINWINDOW_H
  
  #define MAINWINDOW_H
  
  
  
  #include <qmainwindow>
  
  
  
  namespace Ui {
  
      class MainWindow;
  
  }
  
  
  
  class MainWindow : public QMainWindow {
  
      Q_OBJECT
  
  public:
  
      MainWindow(QWidget *parent = 0);
  
      ~MainWindow();
  
  
  
  protected:
  
      void changeEvent(QEvent *e);
  
  
  
  private:
  
      Ui::MainWindow *ui;
  
  
  
  private slots:
  
      void on_pushButton_clicked();
  
  };
  
  
  
  #endif // MAINWINDOW_H
  
  </qmainwindow>

  wrapper.h

  #ifndef WRAPPER_H_
  
  #define WRAPPER_H_
  
  
  
  #include 
  
  
  
  #include <libavutil></libavutil>opt.h>
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  #include <libavutil></libavutil>channel_layout.h>
  
  #include <libavutil></libavutil>common.h>
  
  #include <libavutil></libavutil>imgutils.h>
  
  #include <libavutil></libavutil>mathematics.h>
  
  #include <libavutil></libavutil>samplefmt.h>
  
  
  
  #define INBUF_SIZE 4096
  
  #define AUDIO_INBUF_SIZE 20480
  
  #define AUDIO_REFILL_THRESH 4096
  
  
  
  
  
  
  
  /* check that a given sample format is supported by the encoder */
  
  static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
  
  {
  
      const enum AVSampleFormat *p = codec->sample_fmts;
  
  
  
      while (*p != AV_SAMPLE_FMT_NONE) {
  
          if (*p == sample_fmt)
  
              return 1;
  
          p++;
  
      }
  
      return 0;
  
  }
  
  
  
  /* just pick the highest supported samplerate */
  
  static int select_sample_rate(AVCodec *codec)
  
  {
  
      const int *p;
  
      int best_samplerate = 0;
  
  
  
      if (!codec->supported_samplerates)
  
          return 44100;
  
  
  
      p = codec->supported_samplerates;
  
      while (*p) {
  
          best_samplerate = FFMAX(*p, best_samplerate);
  
          p++;
  
      }
  
      return best_samplerate;
  
  }
  
  
  
  /* select layout with the highest channel count */
  
  static int select_channel_layout(AVCodec *codec)
  
  {
  
      const uint64_t *p;
  
      uint64_t best_ch_layout = 0;
  
      int best_nb_channells   = 0;
  
  
  
      if (!codec->channel_layouts)
  
          return AV_CH_LAYOUT_STEREO;
  
  
  
      p = codec->channel_layouts;
  
      while (*p) {
  
          int nb_channels = av_get_channel_layout_nb_channels(*p);
  
  
  
          if (nb_channels > best_nb_channells) {
  
              best_ch_layout    = *p;
  
              best_nb_channells = nb_channels;
  
          }
  
          p++;
  
      }
  
      return best_ch_layout;
  
  }
  
  
  
  /*
  
   * Audio encoding example
  
   */
  
  static void audio_encode_example(const char *filename)
  
  {
  
      AVCodec *codec;
  
      AVCodecContext *c= NULL;
  
      AVFrame *frame;
  
      AVPacket pkt;
  
      int i, j, k, ret, got_output;
  
      int buffer_size;
  
      FILE *f;
  
      uint16_t *samples;
  
      float t, tincr;
  
  
  
      printf("Encode audio file %s\n", filename);
  
  
  
      /* find the MP2 encoder */
  
      codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
  
      if (!codec) {
  
          fprintf(stderr, "Codec not found\n");
  
          exit(1);
  
      }
  
  
  
      c = avcodec_alloc_context3(codec);
  
      if (!c) {
  
          fprintf(stderr, "Could not allocate audio codec context\n");
  
          exit(1);
  
      }
  
  
  
      /* put sample parameters */
  
      c->bit_rate = 64000;
  
  
  
      /* check that the encoder supports s16 pcm input */
  
      c->sample_fmt = AV_SAMPLE_FMT_S16;
  
      if (!check_sample_fmt(codec, c->sample_fmt)) {
  
          fprintf(stderr, "Encoder does not support sample format %s",
  
                  av_get_sample_fmt_name(c->sample_fmt));
  
          exit(1);
  
      }
  
  
  
      /* select other audio parameters supported by the encoder */
  
      c->sample_rate    = select_sample_rate(codec);
  
      c->channel_layout = select_channel_layout(codec);
  
      c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
  
  
  
      /* open it */
  
      if (avcodec_open2(c, codec, NULL) &lt; 0) {
  
          fprintf(stderr, "Could not open codec\n");
  
          exit(1);
  
      }
  
  
  
      f = fopen(filename, "wb");
  
      if (!f) {
  
          fprintf(stderr, "Could not open %s\n", filename);
  
          exit(1);
  
      }
  
  
  
      /* frame containing input raw audio */
  
      frame = avcodec_alloc_frame();
  
      if (!frame) {
  
          fprintf(stderr, "Could not allocate audio frame\n");
  
          exit(1);
  
      }
  
  
  
      frame->nb_samples     = c->frame_size;
  
      frame->format         = c->sample_fmt;
  
      frame->channel_layout = c->channel_layout;
  
  
  
      /* the codec gives us the frame size, in samples,
  
       * we calculate the size of the samples buffer in bytes */
  
      buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
  
                                               c->sample_fmt, 0);
  
      samples = (uint16_t *)av_malloc(buffer_size);
  
      if (!samples) {
  
          fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
  
                  buffer_size);
  
          exit(1);
  
      }
  
      /* setup the data pointers in the AVFrame */
  
      ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
  
                                     (const uint8_t*)samples, buffer_size, 0);
  
      if (ret &lt; 0) {
  
          fprintf(stderr, "Could not setup audio frame\n");
  
          exit(1);
  
      }
  
  
  
      /* encode a single tone sound */
  
      t = 0;
  
      tincr = 2 * M_PI * 440.0 / c->sample_rate;
  
      for(i=0;i&lt;200;i++) {
  
          av_init_packet(&amp;pkt);
  
          pkt.data = NULL; // packet data will be allocated by the encoder
  
          pkt.size = 0;
  
  
  
          for (j = 0; j &lt; c->frame_size; j++) {
  
              samples[2*j] = (int)(sin(t) * 10000);
  
  
  
              for (k = 1; k &lt; c->channels; k++)
  
                  samples[2*j + k] = samples[2*j];
  
              t += tincr;
  
          }
  
          /* encode the samples */
  
          ret = avcodec_encode_audio2(c, &amp;pkt, frame, &amp;got_output);
  
          if (ret &lt; 0) {
  
              fprintf(stderr, "Error encoding audio frame\n");
  
              exit(1);
  
          }
  
          if (got_output) {
  
              fwrite(pkt.data, 1, pkt.size, f);
  
              av_free_packet(&amp;pkt);
  
          }
  
      }
  
  
  
      /* get the delayed frames */
  
      for (got_output = 1; got_output; i++) {
  
          ret = avcodec_encode_audio2(c, &amp;pkt, NULL, &amp;got_output);
  
          if (ret &lt; 0) {
  
              fprintf(stderr, "Error encoding frame\n");
  
              exit(1);
  
          }
  
  
  
          if (got_output) {
  
              fwrite(pkt.data, 1, pkt.size, f);
  
              av_free_packet(&amp;pkt);
  
          }
  
      }
  
      fclose(f);
  
  
  
      av_freep(&amp;samples);
  
      avcodec_free_frame(&amp;frame);
  
      avcodec_close(c);
  
      av_free(c);
  
  }
  
  
  
  /*
  
   * Audio decoding.
  
   */
  
  static void audio_decode_example(const char *outfilename, const char *filename)
  
  {
  
      AVCodec *codec;
  
      AVCodecContext *c= NULL;
  
      int len;
  
      FILE *f, *outfile;
  
      uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
  
      AVPacket avpkt;
  
      AVFrame *decoded_frame = NULL;
  
  
  
      av_init_packet(&amp;avpkt);
  
  
  
      printf("Decode audio file %s to %s\n", filename, outfilename);
  
  
  
      /* find the mpeg audio decoder */
  
      codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
  
      if (!codec) {
  
          fprintf(stderr, "Codec not found\n");
  
          exit(1);
  
      }
  
  
  
      c = avcodec_alloc_context3(codec);
  
      if (!c) {
  
          fprintf(stderr, "Could not allocate audio codec context\n");
  
          exit(1);
  
      }
  
  
  
      /* open it */
  
      if (avcodec_open2(c, codec, NULL) &lt; 0) {
  
          fprintf(stderr, "Could not open codec\n");
  
          exit(1);
  
      }
  
  
  
      f = fopen(filename, "rb");
  
      if (!f) {
  
          fprintf(stderr, "Could not open %s\n", filename);
  
          exit(1);
  
      }
  
      outfile = fopen(outfilename, "wb");
  
      if (!outfile) {
  
          av_free(c);
  
          exit(1);
  
      }
  
  
  
      /* decode until eof */
  
      avpkt.data = inbuf;
  
      avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
  
  
  
      while (avpkt.size > 0) {
  
          int got_frame = 0;
  
  
  
          if (!decoded_frame) {
  
              if (!(decoded_frame = avcodec_alloc_frame())) {
  
                  fprintf(stderr, "Could not allocate audio frame\n");
  
                  exit(1);
  
              }
  
          } else
  
              avcodec_get_frame_defaults(decoded_frame);
  
  
  
          len = avcodec_decode_audio4(c, decoded_frame, &amp;got_frame, &amp;avpkt);
  
          if (len &lt; 0) {
  
              fprintf(stderr, "Error while decoding\n");
  
              exit(1);
  
          }
  
          if (got_frame) {
  
              /* if a frame has been decoded, output it */
  
              int data_size = av_samples_get_buffer_size(NULL, c->channels,
  
                                                         decoded_frame->nb_samples,
  
                                                         c->sample_fmt, 1);
  
              fwrite(decoded_frame->data[0], 1, data_size, outfile);
  
          }
  
          avpkt.size -= len;
  
          avpkt.data += len;
  
          avpkt.dts =
  
          avpkt.pts = AV_NOPTS_VALUE;
  
          if (avpkt.size &lt; AUDIO_REFILL_THRESH) {
  
              /* Refill the input buffer, to avoid trying to decode
  
               * incomplete frames. Instead of this, one could also use
  
               * a parser, or use a proper container format through
  
               * libavformat. */
  
              memmove(inbuf, avpkt.data, avpkt.size);
  
              avpkt.data = inbuf;
  
              len = fread(avpkt.data + avpkt.size, 1,
  
                          AUDIO_INBUF_SIZE - avpkt.size, f);
  
              if (len > 0)
  
                  avpkt.size += len;
  
          }
  
      }
  
  
  
      fclose(outfile);
  
      fclose(f);
  
  
  
      avcodec_close(c);
  
      av_free(c);
  
      avcodec_free_frame(&amp;decoded_frame);
  
  }
  
  
  
  /*
  
   * Main WRAPPER function
  
   */
  
  void service(){
  
  
  
  
  
      /* register all the codecs */
  
      avcodec_register_all();
  
  
  
  
  
      audio_encode_example("test.mp2");
  
      audio_decode_example("test.sw", "test.mp2");
  
  
  
  }
  
  
  
  #endif 

  main.cpp

  #include <qapplication>
  
  #include "mainwindow.h"
  
  
  
  extern "C"{
  
      #include "wrapper.h"
  
  }
  
  
  
  int main(int argc, char *argv[])
  
  {
  
      service(); //calling the function service inside the wrapper
  
  
  
      QApplication a(argc, argv);
  
      MainWindow w;
  
      w.show();
  
      return a.exec();
  
  }
  
  </qapplication>

  mainwindow.cpp

  #include "mainwindow.h"
  
  #include "ui_mainwindow.h"
  
  
  
  MainWindow::MainWindow(QWidget *parent) :
  
      QMainWindow(parent),
  
      ui(new Ui::MainWindow)
  
  {
  
      ui->setupUi(this);
  
  }
  
  
  
  MainWindow::~MainWindow()
  
  {
  
      delete ui;
  
  }
  
  
  
  void MainWindow::changeEvent(QEvent *e)
  
  {
  
      QMainWindow::changeEvent(e);
  
      switch (e->type()) {
  
      case QEvent::LanguageChange:
  
          ui->retranslateUi(this);
  
          break;
  
      default:
  
          break;
  
      }
  
  }
  
  
  
  void MainWindow::on_pushButton_clicked()
  
  {
  
          this->close();
  
  }

  mainwindow.ui
  //Nothing important

  Thanks.

• Webm video playing video & audio in Movies & TV app but will only play video in windows media player and html ?

  24 décembre 2019, par Kate LeVering

  I have created a webm video file with transparency with ffmpeg out of a series of png files. Then I add the audio track again in ffmpeg (I have tried both the opus and vorbis codecs). When I play it in the Movies & TV app it plays just fine (audio and video). In windows media player only the video plays. In html (inside a video tag) the video will play if it is set to ’muted’ but if it is not muted it doesn’t play.

  I am not sure what is going on. Does anyone have any insights. Do I need to run the audio from a seperate file ?

  Thanks, Kate

• wowza + live + ffmpeg + hls player, how to create the playlist.m3u8 ?

  9 mai 2018, par Ziv Barber

  I’m trying to setup a wowza live test server and then I can play hls from my mobile app. It do work without any problem for vod. I can play it in my app. I can also see the .m3p8 file if I enter this uri in the browser.
  I tried to do the same in live mode (my goal is to test some streaming parameters for live streaming). I tried to use ffmpeg to create the live stream :

  ffmpeg -re -i "myInputTestVideo.mp4" -vcodec libx264 -vb 150000 -g 60 -vprofile baseline -level 2.1 -acodec aac -ab 64000 -ar 48000 -ac 2 -vbsf h264_mp4toannexb -strict experimental -f mpegts udp ://127.0.0.1:10000

  I created a "source file" and connected it to the "Incoming Streams".
  I can see in my application’s Monitoring / Network tab that it do getting the data from ffmpeg.

  My problem is how to get the playlist.m3p8 file so I can play it from inside my app (hls based) ?

  Again, for now I need a way to test playing with the streaming settings and in real live I’ll have a real live streaming source.