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• Can all Android devices play local 60fps mp4 files ?

  3 juin 2017, par Ken York

  We have an android app with 60fps mp4 files in the raw directory.

  The mp4s are encoded H.264 baseline 60fps

  They work well and look good on all our test devices using VideoView.

  We are targeting API16(4.1) and above.

  Is it guaranteed that 60fps mp4 files will always play on any API16+ device?

  If not, would there be an error msg, crash, or just play at 30fps ?

  Thanks all!

• compress video like facebook using ffmpeg

  3 juin 2017, par Mahesh Prajapati

  i have already used many of the commands but it will reduced size when i will reduce the height width but facebook reduce the size without reducing height width so how can it possible using ffmpeg if not exactly result but want similar result.

  hear is the some sample example i have done with height width decries.

  $cmd3="$ffmpeg -i Wildlife.wmv  -r 30 -s 400x224 -c:v libx264 -y Wildlife_30fps_400x224_testing.mp4";
  

  but i dont want to change height width i want to reduce size as well as convert all the extension only in .mp4 format.

• vlc in android , i need compile ffmpeg from source file not from git

  3 juin 2017, par Eva Zana

  i have the rules.mak file from ffmpeg in vlc android. i modified in libav file now i want thev changes to impact.the problem is that the files is compiled from git and not from my changes

  ifdef USE_FFMPEG
  FFMPEG_HASH=0768aaec1d683226e613e692080a588359c31334
  FFMPEG_SNAPURL := http://git.videolan.org/?p=ffmpeg.git;a=snapshot;h=$(FFMPEG_HASH);sf=tgz
  FFMPEG_GITURL := http://git.videolan.org/git/ffmpeg.git
  else
  FFMPEG_HASH=6ac0e7818399a57e4684202bac79f35b3561ad1e
  FFMPEG_SNAPURL := http://git.libav.org/?p=libav.git;a=snapshot;h=$(FFMPEG_HASH);sf=tgz
  FFMPEG_GITURL := git://git.libav.org/libav.git
  endif
  
  FFMPEG_BASENAME := $(subst .,_,$(subst \,_,$(subst /,_,$(FFMPEG_HASH))))
  
  
  
  
  # Build
  PKGS += ffmpeg
  ifeq ($(call need_pkg,"libavcodec >= 55.0.0 libavformat >= 53.21.0 libswscale"),)
  PKGS_FOUND += ffmpeg
  endif
  
  FFMPEGCONF += --nm="$(NM)" --ar="$(AR)"
  
  $(TARBALLS)/ffmpeg-$(FFMPEG_BASENAME).tar.xz:
      $(call download_git,$(FFMPEG_GITURL),,$(FFMPEG_HASH))
  
  .sum-ffmpeg: $(TARBALLS)/ffmpeg-$(FFMPEG_BASENAME).tar.xz
      $(call check_githash,$(FFMPEG_HASH))
      touch $@
  
  ffmpeg: ffmpeg-$(FFMPEG_BASENAME).tar.xz .sum-ffmpeg
      rm -Rf $@ $@-$(FFMPEG_BASENAME)
      mkdir -p $@-$(FFMPEG_BASENAME)
      tar xvJf "$<" --strip-components=1 -C $@-$(FFMPEG_BASENAME)
      $(MOVE)
  
  .ffmpeg: ffmpeg
      cd $< && $(HOSTVARS) ./configure \
          --extra-ldflags="$(LDFLAGS)" $(FFMPEGCONF) \
          --prefix="$(PREFIX)" --enable-static --disable-shared
      cd $< && $(MAKE) install-libs install-headers
      touch $@
  

  I need to chage that instead git from the url git that it will take from a source file i do i do that?

• Audio recorded with MediaRecorder on Chrome missing duration

  3 juin 2017, par suppp111

  I am recording audio (oga/vorbis) files with MediaRecorder. When I record these file through Chrome I get problems: I cannot edit the files on ffmpeg and when I try to play them on Firefox it says they are corrupt (they do play fine on Chrome though).

  Looking at their metadata on ffmpeg I get this:

  Input #0, matroska,webm, from '91.oga':
    Metadata:
      encoder         : Chrome
    Duration: N/A, start: 0.000000, bitrate: N/A
      Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp (default)
  [STREAM]
  index=0
  codec_name=opus
  codec_long_name=Opus (Opus Interactive Audio Codec)
  profile=unknown
  codec_type=audio
  codec_time_base=1/48000
  codec_tag_string=[0][0][0][0]
  codec_tag=0x0000
  sample_fmt=fltp
  sample_rate=48000
  channels=1
  channel_layout=mono
  bits_per_sample=0
  id=N/A
  r_frame_rate=0/0
  avg_frame_rate=0/0
  time_base=1/1000
  start_pts=0
  start_time=0.000000
  duration_ts=N/A
  duration=N/A
  bit_rate=N/A
  max_bit_rate=N/A
  bits_per_raw_sample=N/A
  nb_frames=N/A
  nb_read_frames=N/A
  nb_read_packets=N/A
  DISPOSITION:default=1
  DISPOSITION:dub=0
  DISPOSITION:original=0
  DISPOSITION:comment=0
  DISPOSITION:lyrics=0
  DISPOSITION:karaoke=0
  DISPOSITION:forced=0
  DISPOSITION:hearing_impaired=0
  DISPOSITION:visual_impaired=0
  DISPOSITION:clean_effects=0
  DISPOSITION:attached_pic=0
  TAG:language=eng
  [/STREAM]
  [FORMAT]
  filename=91.oga
  nb_streams=1
  nb_programs=0
  format_name=matroska,webm
  format_long_name=Matroska / WebM
  start_time=0.000000
  duration=N/A
  size=7195
  bit_rate=N/A
  probe_score=100
  TAG:encoder=Chrome
  

  As you can see there are problems with the duration. I have looked at posts like this: How can I add predefined length to audio recorded from MediaRecorder in Chrome?

  But even trying that, I got errors when trying to chop and merge files.For example when running:

  ffmpeg -f concat  -i 89_inputs.txt -c copy final.oga
  

  I get a lot of this:

  [oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57612, current: 1980; changing to 57613. This may result in incorrect timestamps in the output file.
  [oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57613, current: 2041; changing to 57614. This may result in incorrect timestamps in the output file.
  DTS -442721849179034176, next:42521 st:0 invalid dropping
  PTS -442721849179034176, next:42521 invalid dropping st:0
  [oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57614, current: 2041; changing to 57615. This may result in incorrect timestamps in the output file.
  [oga @ 00000000006789c0] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
  DTS -442721849179031296, next:42521 st:0 invalid dropping
  PTS -442721849179031296, next:42521 invalid dropping st:0
  

  Does anyone know what we need to do to audio files recorded from Chrome for them to be useful? Or is there a problem with my setup?

  Recorder js:

  if (navigator.getUserMedia) {
    console.log('getUserMedia supported.');
  
    var constraints = { audio: true };
    var chunks = [];
  
    var onSuccess = function(stream) {
      var mediaRecorder = new MediaRecorder(stream);
  
      record.onclick = function() {
        mediaRecorder.start();
        console.log(mediaRecorder.state);
        console.log("recorder started");
        record.style.background = "red";
  
        stop.disabled = false;
        record.disabled = true;
  
        var aud = document.getElementById("audioClip");
        start = aud.currentTime;
      }
  
      stop.onclick = function() {
        console.log(mediaRecorder.state);
        console.log("Recording request sent.");
        mediaRecorder.stop();
      }
  
      mediaRecorder.onstop = function(e) {
        console.log("data available after MediaRecorder.stop() called.");
  
        var audio = document.createElement('audio');
        audio.setAttribute('controls', '');
        audio.setAttribute('id', 'audioClip');
  
        audio.controls = true;
        var blob = new Blob(chunks, { 'type' : 'audio/ogg; codecs="vorbis"' });
        chunks = [];
        var audioURL = window.URL.createObjectURL(blob);
        audio.src = audioURL;
  
        sendRecToPost(blob);   // this just send the audio blob to the server by post
        console.log("recorder stopped");
  
      }
  

• avformat_open_input fails only with a custom IO context

  2 juin 2017, par Tim

  Running into an odd issue with avformat_open_input, it is failing with:

  Invalid data found when processing input

  But this only happens when I attempt to read the file using a custom AVIOContext.

  My custom code is as follows (error checking omitted for clarity):

  auto fmtCtx = avformat_alloc_context();
  auto ioBufferSize = 32768;
  auto ioBuffer = (unsigned char *)av_malloc(ioBufferSize);
  auto ioCtx = avio_alloc_context(ioBuffer,
                                  ioBufferSize,
                                  0,
                                  reinterpret_cast(this),
                                  &imageIORead,
                                  NULL,
                                  &imageIOSeek));
  
  fmtCtx -> pb = ioCtx;
  fmtCtx -> flags |= AVFMT_FLAG_CUSTOM_IO;
  
  int err = avformat_open_input(&fmtCtx, NULL, NULL, NULL);
  

  imageIOSeek is never called, but properly handles the whence parameter including the AVSEEK_SIZE option. My file data is already loaded in memory, so imageIORead is trivial (returning 0 at EOF):

  int imageIORead(void *opaque, uint8_t *buf, int buf_size) {
      Image *d = (Image *)buf;
      int rc = std::min(buf_size, static_cast(d->data.size() - d->pos));
  
      memcpy(buf, d->data.data() + d->pos, rc);
      d->pos += rc;
      return rc;
  }
  

  The data being read is loaded from a file on disk:

  /tmp/25.jpeg

  The following code is able to open and extract the image correctly:

  auto fmtCtx = avformat_alloc_context();
  int err = avformat_open_input(&fmtCtx, "/tmp/25.jpeg", NULL, NULL);
  

  The project is using a minified version of libavformat including only the formats we need. I don't believe this is the cause of the problem since the file can be open and handled properly when the path is specified. I haven't seen any configure options specifically targeting support for custom IO contexts.

  This is the image in question: 25.jpeg