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• why the output of mp3 decode sounds so delayed ?(with ffmpeg mp3lame lib)

  11 mars 2014, par user3401739

  i'm recording sound and encoding to mp3 with ffmpeg lib. then decode the mp3 data right away, play the decode data, but it's sounds so delayed.
  here are the codes :
  the function encode first parameter accepts the raw pcm data, len = 44100.

  encode parameters :

  cntx_->channels = 1;
  
  cntx_->sample_rate = 44100;
  
  cntx_->sample_fmt = 6;
  
  cntx_->channel_layout =  AV_CH_LAYOUT_MONO;
  
  cntx_->bit_rate = 8000;
  
  err_ = avcodec_open2(cntx_, codec_, NULL);
  
  
  
  vector<unsigned char="char">       encode(unsigned char* encode_data, unsigned int len)
  
  {
  
      vector<unsigned char="char"> ret;
  
      AVPacket avpkt;
  
      av_init_packet(&amp;avpkt);
  
  
  
      unsigned int len_encoded = 0; 
  
      int data_left = len / 2;
  
      int miss_c = 0, i = 0;
  
      while (data_left > 0)
  
      {
  
          int sz = data_left > cntx_->frame_size ? cntx_->frame_size : data_left;
  
          mp3_frame_->nb_samples = sz;
  
          mp3_frame_->format = cntx_->sample_fmt;
  
          mp3_frame_->channel_layout = cntx_->channel_layout;
  
  
  
          int needed_size = av_samples_get_buffer_size(NULL, 1,
  
              mp3_frame_->nb_samples, cntx_->sample_fmt, 1);
  
  
  
          int r = avcodec_fill_audio_frame(mp3_frame_, 1, cntx_->sample_fmt, encode_data + len_encoded, needed_size, 0);
  
  
  
          int gotted = -1;
  
  
  
          r = avcodec_encode_audio2(cntx_, &amp;avpkt, mp3_frame_, &amp;gotted);
  
          if (gotted){
  
              i++;
  
              ret.insert(ret.end(), avpkt.data, avpkt.data + avpkt.size);
  
          }
  
          else if (gotted == 0){
  
              miss_c++;
  
          }
  
          len_encoded += needed_size;
  
          data_left -= sz;
  
          av_free_packet(&amp;avpkt);
  
      }
  
      return ret;
  
  }
  
  
  
  std::vector<unsigned char="char">  decode(unsigned char* data, unsigned int len)
  
  {
  
      std::vector<unsigned char="char"> ret;
  
  
  
      AVPacket avpkt;
  
      av_init_packet(&amp;avpkt);
  
      avpkt.data = data;
  
      avpkt.size = len;
  
  
  
      AVFrame* pframe = av_frame_alloc();
  
      while (avpkt.size > 0){
  
          int goted = -1;av_frame_unref(pframe);
  
          int used = avcodec_decode_audio4(cntx_, pframe, &amp;goted, &amp;avpkt);
  
          if (goted){
  
              ret.insert(ret.end(), pframe->data[0], pframe->data[0] + pframe->linesize[0]);
  
              avpkt.data += used;
  
              avpkt.size -= used;
  
              avpkt.dts = avpkt.pts = AV_NOPTS_VALUE; 
  
          }
  
          else if (goted == 0){
  
              avpkt.data += used;
  
              avpkt.size -= used;
  
              avpkt.dts = avpkt.pts = AV_NOPTS_VALUE; 
  
          }
  
          else if(goted &lt; 0){
  
              break;
  
          }
  
      }
  
      av_frame_free(&amp;pframe);
  
      return ret;
  
  }
  
  </unsigned></unsigned></unsigned></unsigned>

  Suppose it's the 100th call to encode(data, len), this "frame" would appear in 150th or later in the decode call, the latency is not acceptable. It seems the mp3lame encoder would keep the sample data for later use, but not my desire.
  I don't know what is going wrong. Thank you for any information.

  today i debug the code again and post some detail :

  encode : each pcm sample frame len = 23040 ,which is 10 times of mp3 frame size, each time call encode only output 9 frames, this output cause decode output 20736 samples, 1 frame(2304 bytes) is lost, and the sound is noisy.

  if the mp3 or mp2 encode is not suitable for real time voice transfer, which encoder should i choose ?

• avcodec : Add explicit capability flag for encoder flushing

  10 avril 2020, par Philip Langdale

  avcodec : Add explicit capability flag for encoder flushing
  Previously, there was no way to flush an encoder such that after
  
  draining, the encoder could be used again. We generally suggested
  
  that clients teardown and replace the encoder instance in these
  
  situations. However, for at least some hardware encoders, the cost of
  
  this tear down/replace cycle is very high, which can get in the way of
  
  some use-cases - for example : segmented encoding with nvenc.
  To help address that use case, we added support for calling
  
  avcodec_flush_buffers() to nvenc and things worked in practice,
  
  although it was not clearly documented as to whether this should work
  
  or not. There was only one previous example of an encoder implementing
  
  the flush callback (audiotoolboxenc) and it's unclear if that was
  
  intentional or not. However, it was clear that calling
  
  avocdec_flush_buffers() on any other encoder would leave the encoder in
  
  an undefined state, and that's not great.
  As part of cleaning this up, this change introduces a formal capability
  
  flag for encoders that support flushing and ensures a flush call is a
  
  no-op for any other encoder. This allows client code to check if it is
  
  meaningful to call flush on an encoder before actually doing it.
  I have not attempted to separate the steps taken inside
  
  avcodec_flush_buffers() because it's not doing anything that's wrong
  
  for an encoder. But I did add a sanity check to reject attempts to
  
  flush a frame threaded encoder because I couldn't wrap my head around
  
  whether that code path was actually safe or not. As this combination
  
  doesn't exist today, we'll deal with it if it ever comes up.
  

  • [DH] doc/APIchanges
  • [DH] libavcodec/audiotoolboxenc.c
  • [DH] libavcodec/avcodec.h
  • [DH] libavcodec/decode.c
  • [DH] libavcodec/nvenc_h264.c
  • [DH] libavcodec/nvenc_hevc.c
  • [DH] libavcodec/version.h

• Building x264 with YASM : failing the ASM check

  11 janvier 2020, par radiofreemyourenji

  My question up front is, "I have new yasm, I think x264 is supposed to be cool with that, why is x264 not cool with that ?"

  For reasons, I am building a CentOS docker image (based on centos:latest) that contains a from-scratch ffmpeg build, following the guide here. It’s a good guide, it’s worked for me before, so I was feeling good about it.

  Today I’m hitting a choke point on the libx264 build point : specifically, I say

  PKG_CONFIG_PATH="/tmp/ffmpeg_build/lib/pkgconfig" \
  
    ./configure \
  
    --prefix="/tmp/ffmpeg_build" \
  
    --bindir="/tmp/bin" \
  
    --enable-static

  And I get a reply back

  Found no assembler

  Minimum version is nasm-2.13

  If you really want to compile without asm, configure with —disable-asm.

  That’s unexpected. I have yasm, which I understand to be 1) there to do the things nasm does but better, and 2) to be the daisy-fresh most modern version given that I pulled it from its repo about an hour ago, and built it about fifty-nine minutes ago. For what it’s worth, nasm is on the box too since the instructions request it, but it’s below their stated version (i.e. it’s "NASM version 2.10.07 compiled on Jun 9 2014")

  So it seems like yasm is not being found. There’s another StackExchange question that mentions that problem, which came out to a pathing issue. So, I added yasm to my path like so :

  PATH=/tmp/ffmpeg_sources/yasm:$PATH \
  
    PKG_CONFIG_PATH="/tmp/ffmpeg_build/lib/pkgconfig" \
  
    ./configure
  
    ...etc

  That still gave the Found-no-assembler problem. As a last, confused resort, I told the script explicitly what I wanted to use for the variable $AS, because based on my quick look into configure, that looked like where yasm/nasm was meant to go. The command becomes :

  AS=`which yasm`
  
    PKG_CONFIG_PATH="/tmp/ffmpeg_build/lib/pkgconfig" \
  
    ./configure
  
    ...etc

  That at least gave

  Found yasm 1.3.0

  Minimum version is nasm-2.13

  If you really want to compile without asm, configure with —disable-asm.

  Looking in the config.log I see the following :

  checking whether /tmp/bin/yasm supports vmovdqa32 [eax]{k1}{z}, zmm0... no
  
  Failed commandline was:
  
  --------------------------------------------------
  
  /tmp/bin/yasm conftest.asm  -I. -I$(SRCPATH) -DARCH_X86_64=1 -I$(SRCPATH)/common/x86/ -f elf64  -o conftest.o
  
  conftest.asm:1: error: instruction expected after label
  
  conftest.asm:1: warning: ignoring unrecognized character `{'
  
  conftest.asm:1: warning: ignoring unrecognized character `}'
  
  conftest.asm:1: warning: ignoring unrecognized character `{'
  
  conftest.asm:1: warning: ignoring unrecognized character `}'
  
  --------------------------------------------------
  
  Failed program was:
  
  --------------------------------------------------
  
  vmovdqa32 [eax]{k1}{z}, zmm0
  
  --------------------------------------------------

  So : what’s the deal here ? Is my assumption that yasm drop-in replaces nasm bad ? Is yasm good for this purpose, but I’m not providing the right information to ./configure ? Are my instructions for building ffmpeg for CentOS simply out of date with respect to this prerequisite and I should just try harder to get a modern nasm ?