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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
  • Writing numpy arrays using cv2 VideoWriter

    8 juillet 2015, par JustInTime

    I have a problem with writing a toy example video using opencv2.3.1 VideoWriter, here is how I do it :

    writer = cv2.VideoWriter('test1.avi',cv.CV_FOURCC('P','I','M','1'),25,(640,480))
    for i in range(1000):
       x = np.random.randint(10,size=(480,640)).astype('uint8')
       writer.write(x)
    #del writer (with or without tested)

    I tried every possible combination resulting with a 0 bytes file if the extension was mpg, and 5.5kb if it was avi. I should say that some pointed out that I should build the ffmpeg library from source and not apt-get it. Well I did that on a fresh machine based on the help of this site http://vinayhacks.blogspot.com/2011/11/installing-opencv-231-with-ffmpeg-on-64.html. which also presented an error while compiling opencv(the error was related to ffmpeg). Now I am really out of ideas, How to generate a video using OPENCV ?

    Thanks in advance