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Sur d’autres sites (5876)

  • FFMPEG problems with real-time buffer

    27 juin 2024, par Charles Kiel

    I'm trying to use FFMPEG (Windows) to encode a stream from a video capture card via dshow and send to a RTMP server. THis is my command line ;

    



        ffmpeg -f dshow -i video="AVerMedia BDA Analog Capture Secondary":audio="Microphone (6- C-Media USB Audi" -vf scale=1280:720 -vcodec libx264 -r 30 -rtbufsize 702000k -acodec mp3 -ac 2 -ar 44100 -ab 128k -pix_fmt yuv420p -tune zerolatency -preset ultrafast -f flv "rtmp://xxx.xxx.xxx.xxx/stream/key"        ffmpeg version N-86950-g1bef008 Copyright (c) 2000-2017 the FFmpeg developers
    built with gcc 7.1.0 (GCC)
    configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable
    -libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspe
    ex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
    libavutil      55. 70.100 / 55. 70.100
    libavcodec     57.102.100 / 57.102.100
    libavformat    57. 76.100 / 57. 76.100
    libavdevice    57.  7.100 / 57.  7.100
    libavfilter     6. 98.100 /  6. 98.100
    libswscale      4.  7.102 /  4.  7.102
    libswresample   2.  8.100 /  2.  8.100
    libpostproc    54.  6.100 / 54.  6.100
    Guessed Channel Layout for Input Stream #0.1 : stereo
    Input #0, dshow, from 'video=AVerMedia BDA Analog Capture Secondary:audio=Microphone (6- C-Media USB Audi':
    Duration: N/A, start: 2035.202000, bitrate: N/A
    Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 720x480, 29.97 fps, 29.97 tbr, 10000k tbn, 10000k tbc
    Stream #0:1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
    [dshow @ 00000000005f90e0] real-time buffer [AVerMedia BDA Analog Capture Secondary] [video input] too full or near too full (68% of size: 3041280 [rtbufsize parameter])! frame dropped!
    [dshow @ 00000000005f90e0] real-time buffer [AVerMedia BDA Analog Capture Secondary] [video input] too full or near too full (90% of size: 3041280 [rtbufsize parameter])! frame dropped!
    [dshow @ 00000000005f90e0] real-time buffer [AVerMedia BDA Analog Capture Secondary] [video input] too full or near too full (113% of size: 3041280 [rtbufsize parameter])! frame dropped!
    Last message repeated 46 times
    Stream mapping:
    Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
    Stream #0:1 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
    Press [q] to stop, [?] for help
    [dshow @ 00000000005f90e0] real-time buffer [AVerMedia BDA Analog Capture Secondary] [video input] too full or near too full (113% of size: 3041280 [rtbufsize parameter])! frame dropped!
    [libx264 @ 0000000005b16640] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
    [libx264 @ 0000000005b16640] profile Constrained Baseline, level 3.1
    [libx264 @ 0000000005b16640] 264 - core 152 r2851 ba24899 - H.264/MPEG-4 AVC codec - Copyleft 2003-2017 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=2
    1,11 fast_pskip=1 chroma_qp_offset=0 threads=11 lookahead_threads=11 sliced_threads=1 slices=11 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ra
    tio=1.40 aq=0
    [dshow @ 00000000005f90e0] real-time buffer [AVerMedia BDA Analog Capture Secondary] [video input] too full or near too full (113% of size: 3041280 [rtbufsize parameter])! frame dropped!
    Past duration 0.999992 too large


    



    The buffer too full message are non-stop. I can use Open Broadcast Software (OBS) and stream with no problem (I'm pretty sure it also uses ffmpeg), so I'm doing something wrong.

    


  • ffmpeg removing silence makes mp3 longer ?

    13 août 2017, par pocketg99

    I’ve been using the following command to attempt to remove silent segments from an mp3 file

    ffmpeg -i "podcasts/audio1.mp3" -af silenceremove=1:0:-50dB "/tmp/pod-sil.mp3"

    For some reason the resulting mp3 is twice as log as the input mp3. It is not half as fast. There does not appear to be any duplicated audio. There is some silence, but not an hour’s worth. For a given portion of the input file, you can find the same thing in the output file by going to twice the timestamp of the input file.

    The files are long so I have not yet listened to them all the way through. I really have no idea where the extra length is coming from, the files seem normal.

    Here is the full output from ffmpeg

    ffmpeg version 2.8.11-0ubuntu0.16.04.1 Copyright (c) 2000-2017 the FFmpeg developers
    
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
    
  configuration: --prefix=/usr --extra-version=0ubuntu0.16.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
    
  libavutil      54. 31.100 / 54. 31.100
    
  libavcodec     56. 60.100 / 56. 60.100
    
  libavformat    56. 40.101 / 56. 40.101
    
  libavdevice    56.  4.100 / 56.  4.100
    
  libavfilter     5. 40.101 /  5. 40.101
    
  libavresample   2.  1.  0 /  2.  1.  0
    
  libswscale      3.  1.101 /  3.  1.101
    
  libswresample   1.  2.101 /  1.  2.101
    
  libpostproc    53.  3.100 / 53.  3.100
    
[mp3 @ 0x21880e0] Skipping 0 bytes of junk at 0. 
    
[mp3 @ 0x21880e0] Estimating duration from bitrate, this may be inaccurate
    
Input #0, mp3, from 'podcasts/audio1.mp3':
    
  Duration: 01:00:00.20, start: 0.000000, bitrate: 320 kb/s
    
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
    
File '/tmp/pod-sil.mp3' already exists. Overwrite ? [y/N] y
    
Output #0, mp3, to '/tmp/pod-sil.mp3':
    
  Metadata:
    
    TSSE            : Lavf56.40.101
    
    Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, fltp
    
    Metadata:
    
      encoder         : Lavc56.60.100 libmp3lame
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (mp3 (native) -> mp3 (libmp3lame))
    
Press [q] to stop, [?] for help
    
[libmp3lame @ 0x21999e0] Trying to remove 1152 samples, but the queue is empty
    
size=   56253kB time=01:00:00.16 bitrate= 128.0kbits/s    
    
video:0kB audio:56253kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000439%

  • dfa : Disallow odd width/height and add proper bounds check for DDS1 chunks

    11 août 2017, par Diego Biurrun
    dfa : Disallow odd width/height and add proper bounds check for DDS1 chunks

    DDS1 chunks are decoded in 2x2 blocks, odd chunk width or height is not
    
allowed in that case. Also ensure that the decode buffer is big enough
    
for all blocks being processed.
    

    Bug-Id : CVE-2017-9992
    
CC : libav-stable@libav.org
    • [DBH] libavcodec/dfa.c

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