Recherche avancée

Sur d’autres sites (11772)

• Use Windows ffmpeg to record audio output without using the StereoMix

  5 janvier 2019, par DevtelSoftware

  I am looking for a way to record the audio output (speakers) using Windows ffmpeg.
  I need to do this WITHOUT installing any extra dshow filters and without having the StereoMix input enabled (since this is not available on many computers).

  I have read in the ffmpeg documentation that the -map would allow redirecting an audio output so that ffmpeg sees it as an audio input but I can’t find any example of how to do that.

  In Linux I managed to do it like this :

  ffmpeg -f pulse -ac 2 -ar 44100 -i alsa_output.pci-0000_00_1f.4.analog-stereo.monitor -f pulse -ac 2 -ar 44100 -i alsa_input.pci-0000_00_1f.4.analog-stereo -filter_complex amix=inputs=2 test.mp4

  However I can’t find a similar way to do it in Windows and MacOSX.

  So in short, is it possible with the Windows ffmpeg to record audio from the speakers without extra dshow filters (out-of-the-box) ? Same question goes for MacOSX.

  Thanks !

• How to stream H.264 bitstream to browser

  21 janvier 2019, par BobtheMagicMoose

  This is a followup to https://raspberrypi.stackexchange.com/questions/93254/stream-usb-webcam-with-audio?noredirect=1#comment150507_93254

  I, like many other brave tinkerers before me, thought it would be a simple task to take an old USB camera (c920) can pair it with a raspberry pi to make a network streaming device (e.g., baby monitor). As those that have gone before me, I have now realized (after two days of tearing my hair out), that this is an extremely complicated task.

  Problem statement : I have a raspberry pi zero and a c920 webcam. I want to use the H.264 bitstream from the webcam and serve it on the pi without transcoding it (the feeble processor would really struggle). I want to combine the video stream with its audio and send it over to a browser (phone, tablet, pc - something HTML5 without NAPI).

  My current strategy is to do the following :

  ffmpeg -re -f s16le -i /dev/zero -f v4l2 -thread_queue_size 512 -codec:v h264 -s 1920x1080 -i /dev/video0 -codec:v copy -acodec aac -ab 128k -g 50 http://localhost:8090/camera.ffm (this is with dummy audio - I figured I would add audio later)

  Followed by sudo ffserver -d -f /etc/ffserver.conf to received the feed and broadcast it as a stream. This is the ffserver.conf file :

  `HTTPPort 8090
  
  HTTPBindAddress 0.0.0.0
  
  MaxHTTPConnections 2000
  
  MaxClients 1000
  
  MaxBandwidth 100000
  
  CustomLog -
  
  <feed>
  
    File /tmp/streamwebm.ffm
  
    FileMaxSize 50M
  
    ACL allow localhost
  
    ACL allow 128.199.149.46
  
    #ACL allow 127.0.0.1
  
    ACL allow 192.168.0.0 192.168.0.255
  
  </feed>
  
  <stream stream="stream">
  
  Format webm
  
  
  
  # Video Settings
  
  VideoFrameRate 30
  
  VideoSize 1920x1080
  
  
  
  # Audio settings
  
  AudioCodec libvorbis
  
  AudioSampleRate 48000
  
  AVOptionAudio flags +global_header
  
  
  
  MaxTime 0
  
  AVOptionVideo me_range 16
  
  AVOptionVideo qdiff 4
  
  AVOptionVideo qmin 4
  
  AVOptionVideo qmax 40
  
  #AVOptionVideo good
  
  AVOptionVideo flags +global_header
  
  
  
  # Streaming settings
  
  PreRoll 10
  
  StartSendOnKey
  
  
  
  Metadata author "author"
  
  Metadata copyright "copyright"
  
  Metadata title "Web app name"
  
  Metadata comment "comment"
  
  </stream>

  My basic html is<video>  <source src="http://localhost:8090/stream"> </source></video>

  The stream however, doesn’t work (the browser won’t connect) and I get the following :
  

  And the browser on the client says (failed) NET::ERR_CONNECTION_REFUSED

  Thoughts :
   Begin stream simple mp4 with ffserver explains that ffserver can’t stream .mp4 because of headers or something. This is why I am using webm (which doesn’t support h.264 I believe and is causing the really slow performance converting to vp9). I’m not concerned about CPU usage at the moment, just want to get an image to appear on the browser !

  • I hear one issue deals with ’chunking’ - that the camera h.264 is a bitstream but h.264 streams for html5 should be chunked. Not sure how that would work.

  • I have tried VLC for some things (RTP) but haven’t have success.

  • Most resources (SE and other sites) are from 2010-2015 and it seems as thought v4l2 and other things have developed since then.

  • As my problem is most likely general ignorance of the subject matter, I would appreciate any answers that provide some general understanding as to the theory behind different techniques. I know this makes the question more of a call for opinion and less appropriate for SE, but I’m fixing to throw my computer out the window (you know the feeling).

  Thank you !

• MP3 files created using FFmpeg are not starting playback in browser immediately. Is there any major difference between FFmpeg and AVCONV ?

  23 janvier 2019, par AR5

  I am working on a website that streams music. We recently changed server from Debian (with avconv) to a Centos7 (with FFmpeg) server.
  The mp3 files created on Debian server start playback on browser (I have tested Chrome and Firefox) start almost at the same time they start loading into the browser (I used Network tab on Developer Tools to monitor this)

  Now after the switch to Centos/FFmpeg server, the files being created on this new server are displaying a strange behavior. They only start playback after about 1MB is loaded into the browser.

  I have used identical settings for converting original file into MP3 in both AVCONV and FFmpeg but the files created using FFmpeg are showing this issue. Is there something that might be causing such an issue ? Are there differences in terms of audio conversion between AVCONV and FFmpeg ?

  I have already tried

  I first found that the files created on old server (Debian/Avconv) were VBR (variable bitrate) and the ones created on new server were CBR (constant bitrate), so I tried switching to VBR but the issue still persisted.

  I checked the mp3 files using MediaInfo app and there seems to be no difference between the files.

  I also checked if both files were being served as 206 Partial Content and they both are indeed.

  I am trying to create mp3 files using FFmpeg that work exactly like the ones created before using avconv

  I am trying to make the streaming site work on the new server but the mp3 files created using FFmpeg are not playing back correctly as compared to the ones created on the old server. I am trying to figure out what I might be doing wrong ? or if there is a difference between avconv and FFmpeg that is causing this issue.

  I am really stuck on this issue, any help will be really appreciated.

  ---

  Edit

  I don’t have access to old server anymore so I couldn’t retrieve the log output of avconv. The command that I was using was as follows :

  avconv -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"

  Here is the command and log output from new server :

  ffmpeg -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"
  
  ffmpeg version 2.8.15 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-28)
  
    configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
  
    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 @ 0xd60be0] Skipping 0 bytes of junk at 240044.
  
  Input #0, mp3, from '/test/Track 01.mp3':
  
    Metadata:
  
      album           : Future Hndrxx Presents: The WIZRD
  
      artist          : Future
  
      genre           : Hip-Hop
  
      title           : Never Stop
  
      track           : 1
  
      lyrics-eng      : rgf.is
  
      WEB SITE        : rgf.is
  
      TAGGINGTIME     : rgf.is
  
      WEB             : rgf.is
  
      date            : 2019
  
      encoder         : Lavf56.40.101
  
    Duration: 00:04:51.40, start: 0.025056, bitrate: 121 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 114 kb/s
  
      Metadata:
  
        encoder         : Lavc56.60
  
      Stream #0:1: Video: png, rgb24(pc), 333x333 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
  [mp3 @ 0xd66ec0] Frame rate very high for a muxer not efficiently supporting it.
  
  Please consider specifying a lower framerate, a different muxer or -vsync 2
  
  Output #0, mp3, to '/test/Track 01 (converted).mp3':
  
    Metadata:
  
      TALB            : Future Hndrxx Presents: The WIZRD
  
      TPE1            : Future
  
      TCON            : Hip-Hop
  
      TIT2            : Never Stop
  
      TRCK            : 1
  
      lyrics-eng      : rgf.is
  
      WEB SITE        : rgf.is
  
      TAGGINGTIME     : rgf.is
  
      WEB             : rgf.is
  
      TDRC            : 2019
  
      TSSE            : Lavf56.40.101
  
      Stream #0:0: Video: png, rgb24, 333x333 [SAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
        encoder         : Lavc56.60.100 png
  
      Stream #0:1: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc56.60.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (png (native) -> png (native))
  
    Stream #0:0 -> #0:1 (mp3 (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  [libmp3lame @ 0xd9b0c0] Trying to remove 1152 samples, but the queue is emptys/s
  
  frame=    1 fps=0.1 q=-0.0 Lsize=    4788kB time=00:04:51.39 bitrate= 134.6kbits/s
  
  video:234kB audio:4553kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.014809%

  Samples of MP3 files

  I have uploaded samples of mp3 files created using both avconv and FFmpeg. Please find these here : https://drive.google.com/drive/folders/1gRTmMM2iSK0VWQ4Zaf_iBNQe5laFJl08?usp=sharing