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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 ?

• Revision 00d54aa331 : First pass clean up. One of a series of changes to clean up two pass allocation

  9 mai 2014, par Paul Wilkins

  Changed Paths :
   Modify /vp9/encoder/vp9_firstpass.c

  
  
  First pass clean up.

  One of a series of changes to clean up two pass
  allocation as precursor to support for multiple arf
  or boosted frames per GF/ARF group.

  This change pulls out the calculation of the total bits
  allocated to a GF/ARF group into a function, to aid
  readability and reduce the line count for define_gf_group().

  This change should have no material impact on output.

  Change-Id : I716fba08e26f9ddde3257e7d9b188453791883a3

• Size of files was increased after splitting by Ffmpeg

  29 avril 2014, par user3584205

  I use this code to split file to multiple parts :

  @echo off
  
  for %%i in (*.mp4) do ( 
  
    ffmpeg -i "%%~i" -vcodec copy -acodec copy -ss 00:00:00 -t 00:00:05 "D:\Ebook\%%~ni_1.mp4"
  
    ffmpeg -i "%%~i" -vcodec copy -acodec copy -ss 00:00:05 -t 00:00:10 "D:\Ebook\%%~ni_2.mp4"
  
    ffmpeg -i "%%~i" -vcodec copy -acodec copy -ss 00:00:10 "D:\Ebook\%%~ni_3.mp4"
  
    )
  
  pause

  It worked but I have a problem. It is total size of parts is larger than original file.

  Original : 700 MB and after splitting :

  Part 1: 225
  
  Part 1: 500
  
  Part 2: 250

  Why ? And how to keep same quality and size of files after splitting ? Thank you very much !