Recherche avancée

Sur d’autres sites (9541)

• Decode audio using libavcodec and play using libAO ?

  21 mars 2012, par Ashika Umanga Umagiliya

  I use following code snippet to decode audio files (tested with MP3,WAV,WMV).

  But when it plays the audio , it just gives static sounds and crashes time to time.
  Any hints on what i am doing wrong here ?

  #include 
  
  #include 
  
  #include 
  
  #include 
  
  
  
  
  
  extern "C" {
  
  #include "libavutil/mathematics.h"
  
  #include "libavformat/avformat.h"
  
  #include "libswscale/swscale.h"
  
  #include <ao></ao>ao.h>
  
  
  
  }
  
  
  
  void die(const char *msg)
  
  {
  
      fprintf(stderr,"%s\n",msg);
  
      exit(1);
  
  }
  
  
  
  int main(int argc, char **argv)
  
  {
  
  
  
      const char* input_filename=argv[1];
  
  
  
      //avcodec_register_all();
  
      av_register_all();
  
      //av_ini
  
  
  
      AVFormatContext* container=avformat_alloc_context();
  
      if(avformat_open_input(&amp;container,input_filename,NULL,NULL)&lt;0){
  
          die("Could not open file");
  
      }
  
  
  
      if(av_find_stream_info(container)&lt;0){
  
          die("Could not find file info");
  
      }
  
      av_dump_format(container,0,input_filename,false);
  
  
  
      int stream_id=-1;
  
      int i;
  
      for(i=0;inb_streams;i++){
  
          if(container->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO){
  
              stream_id=i;
  
              break;
  
          }
  
      }
  
      if(stream_id==-1){
  
          die("Could not find Audio Stream");
  
      }
  
  
  
      AVDictionary *metadata=container->metadata;
  
  
  
      AVCodecContext *ctx=container->streams[stream_id]->codec;
  
      AVCodec *codec=avcodec_find_decoder(ctx->codec_id);
  
  
  
      if(codec==NULL){
  
          die("cannot find codec!");
  
      }
  
  
  
      if(avcodec_open(ctx,codec)&lt;0){
  
          die("Codec cannot be found");
  
      }
  
  
  
      //ctx=avcodec_alloc_context3(codec);
  
  
  
      //initialize AO lib
  
      ao_initialize();
  
  
  
      int driver=ao_default_driver_id();
  
  
  
      ao_sample_format sformat;
  
      sformat.bits=16;
  
      sformat.channels=2;
  
      sformat.rate=44100;
  
      sformat.byte_format=AO_FMT_NATIVE;
  
      sformat.matrix=0;
  
  
  
      ao_device *adevice=ao_open_live(driver,&amp;sformat,NULL);
  
      //end of init AO LIB
  
  
  
      AVPacket packet;
  
      av_init_packet(&amp;packet);
  
  
  
      AVFrame *frame=avcodec_alloc_frame();
  
  
  
      int buffer_size=AVCODEC_MAX_AUDIO_FRAME_SIZE;
  
      uint8_t buffer[buffer_size];
  
      packet.data=buffer;
  
      packet.size =buffer_size;
  
  
  
  
  
  
  
      int len;
  
      int frameFinished=0;
  
      while(av_read_frame(container,&amp;packet)>=0)
  
      {
  
  
  
          if(packet.stream_index==stream_id){
  
              //printf("Audio Frame read  \n");
  
              int len=avcodec_decode_audio4(ctx,frame,&amp;frameFinished,&amp;packet);
  
              //frame->
  
              if(frameFinished){
  
                  //printf("Finished reading Frame %d %d\n",packet.size,len);
  
                  ao_play(adevice, (char*)frame->data, len);
  
              }
  
  
  
          }
  
  
  
  
  
      }
  
  
  
      av_close_input_file(container);
  
      ao_shutdown();
  
      return 0;
  
  }

• Getting accurate time from FFMPeg with Objective C (Audio Queue Services)

  2 avril 2012, par Winston

  My iPhone app plays an audio file using FFMPeg.

  I'm getting the elapsed time (to show to user) from the playing audio (in minutes and seconds after converting from microseconds, given by FFMPeg) like so :

  AudioTimeStamp currentTimeStamp;
  
  AudioQueueGetCurrentTime (audioQueue, NULL, &currentTimeStamp, NULL);
  
  
  
  getFFMPEGtime = currentTimeStamp.mSampleTime/self.basicAudioDescription.mSampleRate;
  
  
  
  self.currentAudioTime = [NSString stringWithFormat: @"%02d:%02d",
  
                              (int) getFFMPEGtime / (int)60000000,
  
                              (int) ((getFFMPEGtime % 60000000)/1000000)];

  Everything works fine, but when I scrub back or forward to play another portion of the song, the elapsed time will go back to zero, no matter the current position. The timer will always zero out.

  I know I'm suposed to do some math to keep track of the old time and the new time, maybe constructing another clock or so, perhaps implementing another callback function, etc... I'm not sure what way I should go.

  My questions are :

  1) What's the best approach to keep track of the elapsed time when going back/forward in a song, avoiding the clock to always going back to zero ?

  2) Should I look deeply into FFMPeg functions or should I stick with Objective-C and Cocoa Touch for solving this problem ?

  Please, I need some advices/ideas from experienced programmers. I'm stuck. Thanks beforehand !

• Stream H264 To Android Using FFMPEG

  26 février 2013, par GroovyDotCom

  I'm trying to stream a .ts file containing H.264 and AAC as an RTP stream to an Android device.

  I tried :

  .\ffmpeg -fflags +genpts -re -i 1.ts -vcodec copy -an -f rtp rtp ://127.0.0.1:10
  000 -vn -acodec copy -f rtp rtp ://127.0.0.1:20000 -newaudio

  FFMPEG displays what should be in your SDP file and I copied this into an SDP file and tried playing from VLC and FFPLAY. VLC plays audio but just gives errors re : bad NAL unit types for video. FFPLAY doesn't play anything.

  My best guess if that the FFMPEG H.264 RTP implementation is broken or at least it doesn't work in video passthru mode (i.e. using the -vcodec copy).

  I need a fix for FFMPEG or an alternate simple open-source solution. I don't want to install FFMPEG in my Android client.

  thanks.