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• Revert "swscale : disable ARM code until its build failure with clang/iphone is fixed"

  12 janvier 2014, par Michael Niedermayer

  Revert "swscale : disable ARM code until its build failure with clang/iphone is fixed"
  This reverts commit c8c7736c1025bcf5bb27e104a0d0eae749408739.
  

  • [DH] libswscale/arm/Makefile
  • [DH] libswscale/swscale_unscaled.c

• Iphone Streaming and playing Audio Problem

  20 août 2011, par KayKay

  I am trying to make an app that plays audio stream using ffmpeg, libmms.
  I can open mms server, get stream, and decode audio frame to raw frame using suitable codec.
  However I don't know how to do next.
  I think I must use AudioToolbox/AudioToolbox.h and make audioqueue.
  but however when I give audioqueuebuffer decode buffer's memory and play, Only plays the white noise.
  Here is my code.

  What am i missing ?
  Any comment and hint is very appreciated.
  Thanks very much.

  while(av_read_frame(pFormatCtx, &pkt)>=0)
  
  {
  
      int pkt_decoded_len = 0;
  
      int frame_decoded_len;
  
      int decode_buff_remain=AVCODEC_MAX_AUDIO_FRAME_SIZE * 5;
  
      if(pkt.stream_index==audiostream)
  
      {
  
          frame_decoded_len=decode_buff_remain;
  
          int16_t *decode_buff_ptr = decode_buffer;
  
          int decoded_tot_len=0;
  
          pkt_decoded_len = avcodec_decode_audio2(pCodecCtx, decode_buff_ptr, &frame_decoded_len,
  
                                                  pkt.data, pkt.size);
  
          if (pkt_decoded_len <0) break;
  
          AudioQueueAllocateBuffer(audioQueue, kBufferSize, &buffers[i]);
  
          AQOutputCallback(self, audioQueue, buffers[i], pkt_decoded_len);
  
  
  
          if(i == 1){
  
              AudioQueueSetParameter(audioQueue, kAudioQueueParam_Volume, 1.0);
  
              AudioQueueStart(audioQueue, NULL);
  
          }
  
          i++;
  
      }
  
  }
  
  
  
  
  
  void AQOutputCallback(void *inData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer, int copySize)
  
  {
  
      mmsDemoViewController *staticApp = (mmsDemoViewController *)inData;
  
      [staticApp handleBufferCompleteForQueue:inAQ buffer:inBuffer size:copySize];
  
  }
  
  
  
  - (void)handleBufferCompleteForQueue:(AudioQueueRef)inAQ
  
                            buffer:(AudioQueueBufferRef)inBuffer
  
                              size:(int)copySize
  
  {
  
      inBuffer->mAudioDataByteSize = inBuffer->mAudioDataBytesCapacity;
  
      memcpy((char*)inBuffer->mAudioData, (const char*)decode_buffer, copySize);
  
  
  
      AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL);
  
  }

• iPhone ffmpeg dev using libav to decode from AMR to ACC codec

  10 octobre 2011, par VictorT

  It seems to be that, the AMR support of AudioQueue has been disappeared since iOS 4.3 was released. I can't use audio frame received from RSTP server with old way :

  audioFormat.mFormatID = kAudioFormatAMR; 
  
  int err = AudioQueueNewOutput(&audioFormat, MyAudioQueueOutputCallback, self, NULL, kCFRunLoopCommonModes, 0, &audioQueue);

  As a result I received an error in last string.

  Maybe someone know how to decode AMR AVPacket into raw buffer and encode it with AAC or MP3 using LIBAV ?

  I've tried to use

  avcodec_decode_audio3

  It works and I can get raw buffer but when I'm trying to encode it with

  avcodec_encode_audio

  I get 0 as result

  This is my method to encode buffer :

  - (AVPacket) encodeRawFrame:(const short *) in_buffer withSize:(unsigned int) in_buf_byte_size
  
  {
  
      AVPacket res;
  
      AVCodec *codec;
  
      AVCodecContext *c= NULL;
  
      int count, out_size, outbuf_size, frame_byte_size;
  
      uint8_t *outbuf;
  
  
  
      avcodec_init();
  
      avcodec_register_all();
  
  
  
      printf("Audio encoding\n");
  
  
  
      codec = avcodec_find_encoder(CODEC_ID_AAC);
  
      if (!codec) {
  
          fprintf(stderr, "codec not found\n");
  
          return res;
  
      }
  
  
  
      c= avcodec_alloc_context();
  
  
  
      c->bit_rate = 64000;
  
      c->sample_rate = 24000;
  
      c->channels = 2;
  
  
  
      if (avcodec_open(c, codec) < 0) 
  
      {
  
          fprintf(stderr, "could not open codec\n");
  
      }
  
      else
  
      {
  
          frame_byte_size=c->frame_size*2*2;
  
          count = in_buf_byte_size/frame_byte_size;
  
  
  
          fprintf(stderr, "Number of frames: %d\n", count);
  
  
  
          outbuf_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
  
          outbuf = (uint8_t*) malloc(outbuf_size);
  
  
  
          out_size = avcodec_encode_audio(c, outbuf, outbuf_size, &in_buffer[frame_byte_size*i]);
  
          if(out_size >= 0)
  
          {
  
              res.size = outbuf_size;
  
              res.data = malloc(outbuf_size);                
  
          }
  
  
  
          free(outbuf);
  
      }
  
  
  
  
  
      avcodec_close(c);
  
      av_free(c);
  
      return res;
  
  }

  After encoding "out_size" is always 0 and result buffer is empty.

  Thanks.