Recherche avancée

Sur d’autres sites (9597)

• libswresample : swr_convert() not producing enough samples

  20 septembre 2016, par Tsherr

  I’m trying to use ffmpeg/libswresample to resample streaming audio in my c++ application. Changing the sample width works well and the result sounds as one would expect ; however, when changing the sample rate the result is somewhat crackly. I am unsure if it is due to incorrect usage of the libswresample library, or if I’m misunderstanding the resampling theory.

  Here is my resampling process, simplified for demonstration’s sake :

  //Externally supplied data
  
  const uint8_t* in_samples //contains the audio data to be resampled
  
  int in_num_samples = 256
  
  
  
  //Set up resampling context
  
  SwrContext *swr = swr_alloc();
  
  av_opt_set_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_STEREO, 0);
  
  av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
  
  av_opt_set_int(swr, "in_sample_rate", 44100, 0);
  
  av_opt_set_int(swr, "out_sample_rate", 22050, 0);
  
  av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLT, 0);
  
  av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_FLT, 0);
  
  swr_init(swr);
  
  
  
  //Perform the resampe
  
  uint8_t* out_samples;
  
  int out_num_samples = av_rescale_rnd(swr_get_delay(swr, in_samplerate) + in_num_samples, out_samplerate, in_samplerate, AV_ROUND_UP);
  
  av_samples_alloc(&out_samples, NULL, out_num_channels, out_num_samples, AV_SAMPLE_FMT_FLT, 0);
  
  out_num_samples = swr_convert(swr, &out_samples, out_num_samples, &in_samples, in_num_samples);
  
  av_freep(&out_samples);
  
  swr_free(&swr);

  I suspect that the reason the resampled audio does not sound right is because swr_convert() returns 112, where I expect it to return 128 (the number of samples of the resampled audio) :
  Downsampling 256 samples from a samplerate of 44100 to a samplerate of 22050 should yield 128 samples, yet swr_convert() is producing 112 samples. When expressed in terms of audio duration this is also puzzling. 256 samples at 44100 = 5.8 ms, but 112 samples at 22050 = 5.07 ms. Shouldn’t the downsampling process not alter the duration of the resampled audio ?

  I have also stepped through an example provided with ffmpeg, in which swr_convert() also returns a smaller number than I would expect. So, I suspect that the problem is not due to a bug in libswresample but rather my own lack of understanding.

• ffmpeg Error initializing output stream 0:0 — Error while opening encoder for output stream

  23 juin 2018, par gufidown

  I am a newbie in terms of video processing in general and ffmpeg in particular. So this might be a real beginner’s question.

  I am trying to extract frames from an avi. This is the command :

  ffmpeg.exe -i 123.avi -vf select='gt(scene\,0.4)',scale=1920:-1,tile=6x3 -frames:v 1 456.jpg

  And this is the console output :

  ffmpeg version 4.0 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7.3.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
  
    libavutil      56. 14.100 / 56. 14.100
  
    libavcodec     58. 18.100 / 58. 18.100
  
    libavformat    58. 12.100 / 58. 12.100
  
    libavdevice    58.  3.100 / 58.  3.100
  
    libavfilter     7. 16.100 /  7. 16.100
  
    libswscale      5.  1.100 /  5.  1.100
  
    libswresample   3.  1.100 /  3.  1.100
  
    libpostproc    55.  1.100 / 55.  1.100
  
  Input #0, avi, from 'C:\work\exports\123.avi':
  
    Duration: 00:02:12.43, start: 0.000000, bitrate: 4455 kb/s
  
      Stream #0:0: Video: rawvideo, pal8, 164x485, 4459 kb/s, 7 fps, 7 tbr, 7 tbn, 7 tbc
  
      Metadata:
  
        title           : FileAVI write  
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (rawvideo (native) -> mjpeg (native))
  
  Press [q] to stop, [?] for help
  
  [swscaler @ 000000eb4f415ec0] deprecated pixel format used, make sure you did set range correctly
  
  Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
  
  Conversion failed!

  Do I need a particular encoder when working with avi files ? Which one and how do I specify it ?

• gstreamer h264 multicast missing SPS/PPS information

  6 mars 2015, par RBI

  I have two video streaming units capable of streaming live video inputs :

  • AXIS Q7424-R Video Encoder
  • EPIPHAN VGADVI Broadcaster 99460 -

  I am using gstreamer to view these streams on client terminals running linux. I am interested in the h264, rtp multicast streams (which both units support).

  I can stream the Epiphan video using the following gstreamer pipeline :

  gst-launch-0.10 udpsrc multicast-group=ADDRESS port=PORT caps='application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264' ! rtph264depay ! ffdec_h254 ! autovideosink

  However, this pipeline does not work for the Axis unit as I get the following error repeatedly :

  ffmpeg:0:: non-existing PPS referenced
  
  ffmpeg:0:: non-existing PPS 0 referenced
  
  ffmpeg:0:: decode_slice_header error
  
  ffmpeg:0:: no frame!
  
  ffdec_h264: decoding error (len:-1, have_data: 0)

  I have read that this error means that the ffmpeg decoder is missing the SPS/PPS information provided by a keyframe. The axis unit has a GOV parameter which is the interval at which i-frames are sent ; it is set to 32.

  Note that I can view both units’ rtp streams in unicast with the following :

  gst-launch-0.10 rtspsrc location=rtsp://ADDRESS:PORT/... latency=100 ! rtph264depay ! ffdec_h264 ! autovideosink

  Since unicast works and the unicast and multicast pipelines are the same (except for the source), my guess is either :

  • My udpsrc caps are simply incorrect for the axis stream (and I don’t really know where/how to verify it)

  • or, the axis multicast format/encoding is different and requires a modification to the pipeline (I find this unlikely since unicast is working and I don’t understand why the encoding would change between unicast/multicast).

  Any suggestions are appreciated as I am limited by my knowledge of gstreamer and media formats in terms of what to try next.