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• Correct command to transmit audio to ip camera using ffmpeg ?

  4 novembre 2016, par the_naive

  So I found some hints in this discussion on the correct command to transmit audio to Axis IP camera through using ffmpeg in windows, but still I have not managed to successfully transmit audio to the camera.

  The command I’m using is the following :

  ffmpeg -v debug -y -re -f dshow -i "audio=Microphone (2- High Definition Audio Device)" -c:a pcm_mulaw -ac 1 -ar 16000 -b:a 128k -f flv http://oper
  
  ator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi -multiple_requests 1 -reconnect_at_eof 1 -reconnect_streamed 1 -content_type "audio/basic" -report

  The ouput I get following this command is the following :

  ffmpeg started on 2016-11-04 at 17:32:13
  
  Report written to "ffmpeg-20161104-173213.log"
  
  Command line:
  
  ffmpeg -v debug -y -re -f dshow -i "audio=Microphone (2- High Definition Audio Device)" -c:a pcm_mulaw -ac 1 -ar 16000 -b:a 128k -f flv http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi -content_type audio/basic -multiple_requests 1 -reconnect 1 -reconnect_at_eof 1 -reconnect_streamed 1 -report
  
  ffmpeg version N-82225-gb4e9252 Copyright (c) 2000-2016 the FFmpeg developers
  
    built with gcc 5.4.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-libebur128 --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-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --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-decklink --enable-zlib
  
    libavutil      55. 35.100 / 55. 35.100
  
    libavcodec     57. 66.101 / 57. 66.101
  
    libavformat    57. 57.100 / 57. 57.100
  
    libavdevice    57.  2.100 / 57.  2.100
  
    libavfilter     6. 66.100 /  6. 66.100
  
    libswscale      4.  3.100 /  4.  3.100
  
    libswresample   2.  4.100 /  2.  4.100
  
    libpostproc    54.  2.100 / 54.  2.100
  
  Splitting the commandline.
  
  Reading option '-v' ... matched as option 'v' (set logging level) with argument 'debug'.
  
  Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
  
  Reading option '-re' ... matched as option 're' (read input at native frame rate) with argument '1'.
  
  Reading option '-f' ... matched as option 'f' (force format) with argument 'dshow'.
  
  Reading option '-i' ... matched as input file with argument 'audio=Microphone (2- High Definition Audio Device)'.
  
  Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'pcm_mulaw'.
  
  Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '1'.
  
  Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '16000'.
  
  Reading option '-b:a' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '128k'.
  
  Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
  
  Reading option 'http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi' ... matched as output file.
  
  Reading option '-content_type' ... matched as AVOption 'content_type' with argument 'audio/basic'.
  
  Reading option '-multiple_requests' ... matched as AVOption 'multiple_requests' with argument '1'.
  
  Reading option '-reconnect' ... matched as AVOption 'reconnect' with argument '1'.
  
  Reading option '-reconnect_at_eof' ... matched as AVOption 'reconnect_at_eof' with argument '1'.
  
  Reading option '-reconnect_streamed' ... matched as AVOption 'reconnect_streamed' with argument '1'.
  
  Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
  
  Trailing options were found on the commandline.
  
  Finished splitting the commandline.
  
  Parsing a group of options: global .
  
  Applying option v (set logging level) with argument debug.
  
  Applying option y (overwrite output files) with argument 1.
  
  Applying option report (generate a report) with argument 1.
  
  Successfully parsed a group of options.
  
  Parsing a group of options: input file audio=Microphone (2- High Definition Audio Device).
  
  Applying option re (read input at native frame rate) with argument 1.
  
  Applying option f (force format) with argument dshow.
  
  Successfully parsed a group of options.
  
  Opening an input file: audio=Microphone (2- High Definition Audio Device).
  
  [dshow @ 00000000000279e0] Selecting pin Capture on audio only
  
  dshow passing through packet of type audio size    88200 timestamp 310221040000 orig timestamp 310221040000 graph timestamp 310226130000 diff 5090000 Microphone (2- High Definition Audio Device)
  
  [dshow @ 00000000000279e0] All info found
  
  Guessed Channel Layout for Input Stream #0.0 : stereo
  
  Input #0, dshow, from 'audio=Microphone (2- High Definition Audio Device)':
  
    Duration: N/A, start: 31022.104000, bitrate: 1411 kb/s
  
      Stream #0:0, 1, 1/10000000: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
  
  Successfully opened the file.
  
  Parsing a group of options: output file http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi.
  
  Applying option c:a (codec name) with argument pcm_mulaw.
  
  Applying option ac (set number of audio channels) with argument 1.
  
  Applying option ar (set audio sampling rate (in Hz)) with argument 16000.
  
  Applying option b:a (video bitrate (please use -b:v)) with argument 128k.
  
  Applying option f (force format) with argument flv.
  
  Successfully parsed a group of options.
  
  Opening an output file: http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi.
  
  [http @ 0000000001c94040] Setting default whitelist 'http,https,tls,rtp,tcp,udp,crypto,httpproxy'
  
  [http @ 0000000001c94040] request: POST /axis-cgi/audio/transmit.cgi HTTP/1.1
  
  
  
  Transfer-Encoding: chunked
  
  
  
  User-Agent: Lavf/57.57.100
  
  
  
  Accept: */*
  
  
  
  Expect: 100-continue
  
  
  
  Connection: close
  
  
  
  Host: 10.10.210.2
  
  
  
  Icy-MetaData: 1
  
  
  
  
  
  
  
  
  
  [http @ 0000000001c94040] request: POST /axis-cgi/audio/transmit.cgi HTTP/1.1
  
  
  
  Transfer-Encoding: chunked
  
  
  
  User-Agent: Lavf/57.57.100
  
  
  
  Accept: */*
  
  
  
  Connection: close
  
  
  
  Host: 10.10.210.2
  
  
  
  Icy-MetaData: 1
  
  
  
  Authorization: Digest username="operator", realm="AXIS_ACCC8E027F47", nonce="0EcsO3xABQA=ab5efc4740a6c625ecf6a6729d0d67d2b62b615a", uri="/axis-cgi/audio/transmit.cgi", response="4bd3a627b20d6bcaba9e2f595ef6cd2a", algorithm="MD5", qop="auth", cnonce="6a579dd6664b57eb", nc=00000001
  
  
  
  
  
  
  
  
  
  Successfully opened the file.
  
  detected 8 logical cores
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'time_base' to value '1/44100'
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'sample_rate' to value '44100'
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'sample_fmt' to value 's16'
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'channel_layout' to value '0x3'
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
  
  [audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'sample_fmts' to value 's16'
  
  [audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'sample_rates' to value '16000'
  
  [audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'channel_layouts' to value '0x4'
  
  [audio format for output stream 0:0 @ 0000000001c9fa20] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
  
  [AVFilterGraph @ 000000000002ab20] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
  
  [auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] Using s16p internally between filters
  
  [auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] Matrix coefficients:
  
  [auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] FC: FL:0.500000 FR:0.500000 
  
  [auto-inserted resampler 0 @ 0000000001ca4060] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:1 chl:mono fmt:s16 r:16000Hz
  
  Output #0, flv, to 'http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi':
  
    Metadata:
  
      encoder         : Lavf57.57.100
  
      Stream #0:0, 0, 1/1000: Audio: pcm_mulaw ([8][0][0][0] / 0x0008), 16000 Hz, mono, s16, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc57.66.101 pcm_mulaw
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_mulaw (native))
  
  Press [q] to stop, [?] for help
  
  cur_dts is invalid (this is harmless if it occurs once at the start per stream)
  
  av_interleaved_write_frame(): Unknown error
  
  No more output streams to write to, finishing.
  
  Error writing trailer of http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi: Error number -10053 occurredsize=       8kB time=00:00:00.49 bitrate= 131.2kbits/s speed=79.6x    
  
  video:0kB audio:8kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.492485%
  
  Input file #0 (audio=Microphone (2- High Definition Audio Device)):
  
    Input stream #0:0 (audio): 1 packets read (88200 bytes); 1 frames decoded (22050 samples); 
  
    Total: 1 packets (88200 bytes) demuxed
  
  Output file #0 (http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi):
  
    Output stream #0:0 (audio): 1 frames encoded (7984 samples); 1 packets muxed (7984 bytes); 
  
    Total: 1 packets (7984 bytes) muxed
  
  1 frames successfully decoded, 0 decoding errors
  
  [AVIOContext @ 0000000001c9e4c0] Statistics: 0 seeks, 2 writeouts
  
  dshow passing through packet of type audio size    12152 timestamp 310226130000 orig timestamp 310226130000 graph timestamp 310226820000 diff 690000 Microphone (2- High Definition Audio Device)
  
  Conversion failed!

  For some reason, despite setting multiple_requests, reconnect_eof, reconnect_streamed all to true, connection becomes closed.

  Could you please tell me what I’m doing wrong ?

• IE11 not playing mp4 file

  11 juin 2014, par John Qualis

  I am using ffmpeg to convert a freely available public RTSP stream to a mp4 file. I can play the file quite well in Chrome using a standard HTML5 video client on a windows 7 machine but not in IE11. Any ideas why the mp4 will not play in IE11 or WMP ?

  ffmpeg -i rtsp://dmzosx001.dpa.act.gov.au/medium -acodec copy 
  
  -vcodec copy -f mp4 -movflags frag_keyframe+empty_moov 
  
  -min_frag_duration 1000 -reset_timestamps 1 -vsync 1 
  
  -flags global_header -bsf:v dump_extra -y output.mp4

• Maintaining exact aspect ratio when scaling videos using ffmpeg

  31 mai 2012, par Skkard

  I have a mkv video, which is a mix of multiple resolution recordings, e.g. I have the first few seconds of widescreen 16:9 (1024x576) resolution, and the rest of the video if 4:3 (768x576) resolution. I want to scale this video down 3 times, while copying all the other attributes (audio codec, subtitles etc.). I use ffmpeg -i  -vf scale=iw/2:-1 -acodec copy . Also, VLC detects it's resolution as 720x576.

  The problem is that after the scaling, the resolution constantly becomes 4:3 (360x288). How can I maintain the dynamic aspect ratio of the input video file i.e. the 16:9 parts to scale to 16:9, while the 4:3 parts scale to 4:3 ?

  update

  The player size actually changes, atleast in mplayer, when the resolution is switched. I figured out the main problem. It seems each frame is tagged with a Sample Aspect Ratio (SAR), so when the player plays it, it can find the display aspect ratio. This SAR value isn't getting copied over when encoding to MKV. When encoding to MPG, it does get copied over and I get an exact copy, with the player switching sizes, but not with MKV.

  Output of ffprobe -show_streams filename :

  ffprobe version 0.10.3 Copyright (c) 2007-2012 the FFmpeg developers
  
    built on May  9 2012 17:51:07 with gcc 4.7.0 20120505 (prerelease)
  
    configuration: --prefix=/usr --enable-libmp3lame --enable-libvorbis --enable-libxvid --enable-libx264 --enable-libvpx --enable-libtheora --enable-libgsm --enable-libspeex --enable-postproc --enable-shared --enable-x11grab --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libschroedinger --enable-libopenjpeg --enable-librtmp --enable-libpulse --enable-gpl --enable-version3 --enable-runtime-cpudetect --disable-debug --disable-static
  
    libavutil      51. 35.100 / 51. 35.100
  
    libavcodec     53. 61.100 / 53. 61.100
  
    libavformat    53. 32.100 / 53. 32.100
  
    libavdevice    53.  4.100 / 53.  4.100
  
    libavfilter     2. 61.100 /  2. 61.100
  
    libswscale      2.  1.100 /  2.  1.100
  
    libswresample   0.  6.100 /  0.  6.100
  
    libpostproc    52.  0.100 / 52.  0.100
  
  Input #0, matroska,webm, from 'sample.mkv':
  
    Metadata:
  
      title           : Pan prstenu. Dve veze
  
    Duration: 00:00:29.80, start: 0.000000, bitrate: 3124 kb/s
  
      Stream #0:0(eng): Video: mpeg2video (Main), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 15000 kb/s, 25 fps, 25 tbr, 1k tbn, 50 tbc (default)
  
      Stream #0:1(cze): Audio: mp2, 48000 Hz, stereo, s16, 128 kb/s (default)
  
  [STREAM]
  
  index=0
  
  codec_name=mpeg2video
  
  codec_long_name=MPEG-2 video
  
  codec_type=video
  
  codec_time_base=1/50
  
  codec_tag_string=[0][0][0][0]
  
  codec_tag=0x0000
  
  width=720
  
  height=576
  
  has_b_frames=1
  
  sample_aspect_ratio=64:45
  
  display_aspect_ratio=16:9
  
  pix_fmt=yuv420p
  
  level=8
  
  timecode=16:35:19:10
  
  id=N/A
  
  r_frame_rate=25/1
  
  avg_frame_rate=25/1
  
  time_base=1/1000
  
  start_time=0.000000
  
  duration=N/A
  
  nb_frames=N/A
  
  TAG:language=eng
  
  [/STREAM]
  
  [STREAM]
  
  index=1
  
  codec_name=mp2
  
  codec_long_name=MP2 (MPEG audio layer 2)
  
  codec_type=audio
  
  codec_time_base=1/48000
  
  codec_tag_string=[0][0][0][0]
  
  codec_tag=0x0000
  
  sample_fmt=s16
  
  sample_rate=48000
  
  channels=2
  
  bits_per_sample=0
  
  id=N/A
  
  r_frame_rate=0/0
  
  avg_frame_rate=125/3
  
  time_base=1/1000
  
  start_time=0.000000
  
  duration=N/A
  
  nb_frames=N/A
  
  TAG:language=cze
  
  [/STREAM]