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Autres articles (39)

  • Personnaliser les catégories

    21 juin 2013, par

    Formulaire de création d’une catégorie
    Pour ceux qui connaissent bien SPIP, une catégorie peut être assimilée à une rubrique.
    Dans le cas d’un document de type catégorie, les champs proposés par défaut sont : Texte
    On peut modifier ce formulaire dans la partie :
    Administration > Configuration des masques de formulaire.
    Dans le cas d’un document de type média, les champs non affichés par défaut sont : Descriptif rapide
    Par ailleurs, c’est dans cette partie configuration qu’on peut indiquer le (...)

  • Ajouter notes et légendes aux images

    7 février 2011, par

    Pour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
    Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
    Modification lors de l’ajout d’un média
    Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)

  • Contribute to translation

    13 avril 2011

    You can help us to improve the language used in the software interface to make MediaSPIP more accessible and user-friendly. You can also translate the interface into any language that allows it to spread to new linguistic communities.
    To do this, we use the translation interface of SPIP where the all the language modules of MediaSPIP are available. Just subscribe to the mailing list and request further informantion on translation.
    MediaSPIP is currently available in French and English (...)

Sur d’autres sites (7562)

  • Fixing A/V sync issues with RTP/RTCP sent from mediasoup

    10 décembre 2017, par artushin

    A little background : I’m attempting to record a webrtc call being made through the mediasoup v2 SFU. I’m using mediasoup’s room.createRtpStreamer() method to generate a stream which mirrors RTP/RTCP to ffmpeg. Two streamers are created for audio and video within 30ms of each other and begin broadcasting. FFmpeg then spins up and starts accepting. Pretty sure RTCP is working since ffmpeg is always starting with a keyframe despite being started after the streamer begins broadcasting.

    The problem is that I encounter audio/video desynchronization with seemingly random offsets. My current theory is that this offset is based on how old the last keyframe is that RTCP requests to start the stream. See below for ffmpeg configuration and output but my question is : what ffmpeg arguments can I use to adjust the video frame timestamps to match the audio frame timestamps or vice versa ? I’ve messed around with -map 0:0,0:1 -map 0:1,0:1 but it doesn’t seem to do what I’m looking for.

    ffmpeg flags :

    '-y',
    '-loglevel',
    'debug',
    '-dump',
    '-protocol_whitelist',
    'file,crypto,udp,rtp,data',
    '-analyzeduration',
    '20M',
    '-probesize',
    '20M',
    '-i',
    `data:text/plain;base64,${sdp.toString('base64')}`,
    '-fflags',
    '+genpts',
    '-vcodec',
    'copy',
    '-acodec',
    'aac',
    '-bsf:v',
    'h264_mp4toannexb',
    '-start_number',
    '0',
    '-hls_list_size',
    '2147480000',
    '-hls_wrap',
    '0',
    '-hls_time',
    '10',

    SDP used for input (template) :

    v=0
    o=- 0 0 IN IP4 <%=ip %>
    s=title
    c=IN IP4 <%=ip %>
    m=audio <%=audioPort %> RTP/AVPF <%=audioPayload %>
    a=sendrecv
    a=rtcp-mux
    a=rtpmap:<%=audioPayload %> opus/48000/2
    a=fmtp:<%=audioPayload %> minptime=10; useinbandfec=1
    m=video <%=videoPort %> RTP/AVPF <%=videoPayload %>
    a=sendrecv
    a=rtcp-mux
    a=rtpmap:<%=videoPayload %> H264/90000
    a=rtcp-fb:<%=videoPayload %> ccm fir
    a=rtcp-fb:<%=videoPayload %> nack
    a=rtcp-fb:<%=videoPayload %> nack pli
    a=rtcp-fb:<%=videoPayload %> goog-remb
    a=rtcp-fb:<%=videoPayload %> transport-cc
    a=fmtp:<%=videoPayload %> level-asymmetry-allowed=1;packetization-mode=1

    ffmpeg output - garbled with some timestamps

    1512775954585 - stderr: ffmpeg version 3.4 Copyright (c) 2000-2017 the FFmpeg developers
     built with Apple LLVM version 9.0.0 (clang-900.0.38)
     configuration: --prefix=/usr/local/Cellar/ffmpeg/3.4 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-gpl --enable-ffplay --enable-libmp3lame --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma
    1512775954587 - stderr:   libavutil      55. 78.100 / 55. 78.100
     libavcodec     57.107.100 / 57.107.100
     libavformat    57. 83.100 / 57. 83.100
     libavdevice    57. 10.100 / 57. 10.100
    1512775954587 - stderr:   libavfilter     6.107.100 /  6.107.100
     libavresample   3.  7.  0 /  3.  7.  0
     libswscale      4.  8.100 /  4.  8.100
     libswresample   2.  9.100 /  2.  9.100
     libpostproc    54.  7.100 / 54.  7.100
    Splitting the commandline.
    Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
    Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
    Reading option '-dump' ... matched as option 'dump' (dump each input packet) with argument '1'.
    Reading option '-protocol_whitelist' ...1512775954589 - stderr:  matched as AVOption 'protocol_whitelist' with argument 'file,crypto,udp,rtp,data'.
    Reading option '-analyzeduration' ...1512775954590 - stderr:  matched as AVOption 'analyzeduration' with argument '20M'.
    Reading option '-probesize' ...1512775954590 - stderr:  matched as AVOption 'probesize' with argument '20M'.
    Reading option '-i' ... matched as input url with argument 'data:text/plain;base64,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'.
    Reading option '-fflags' ...1512775954591 - stderr:  matched as AVOption 'fflags' with argument '+genpts'.
    Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'copy'.
    Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'aac'.
    Reading option '-vsync' ... matched as option 'vsync' (video sync method) with argument '0'.
    Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:0,0:1'.
    Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:1,0:1'.
    Reading option '-bsf:v' ... matched as option 'bsf' (A comma-separated list of bitstream filters) with argument 'h264_mp4toannexb'.
    Reading option '-start_number' ...1512775954591 - stderr:  matched as AVOption 'start_number' with argument '0'.
    Reading option '-hls_list_size' ...1512775954591 - stderr:  matched as AVOption 'hls_list_size' with argument '2147480000'.
    Reading option '-hls_wrap' ... matched as AVOption 'hls_wrap' with argument '0'.
    Reading option '-hls_time' ...1512775954591 - stderr:  matched as AVOption 'hls_time' with argument '10'.
    Reading option '/tmp/archive/e0c92fa0-dad5-11e7-8687-090dda95b1a4_10e1c990-dc70-11e7-888d-9f39ca0c79bc/1512775954465.m3u8' ... matched as output url.
    Finished splitting the commandline.
    Parsing a group of options: global .
    Applying option y (overwrite output files) with argument 1.
    1512775954592 - stderr: Applying option loglevel (set logging level) with argument debug.
    Applying option dump (dump each input packet) with argument 1.
    Applying option vsync (video sync method) with argument 0.
    Successfully parsed a group of options.
    Parsing a group of options: input url data:text/plain;base64,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.
    Successfully parsed a group of options.
    Opening an input file: data:text/plain;base64,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.
    1512775954592 - stderr: [NULL @ 0x7f81fd000000] Opening 'data:text/plain;base64,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' for reading
    1512775954593 - stderr: [data @ 0x7f81fca001a0] Content-type: text/plain
    {"level":"info","time":"Dec 8, 2017 11:32 PM","message":"ffmpeg started"}
    1512775954595 - stderr: [sdp @ 0x7f81fd000000] Format sdp probed with size=2048 and score=50
    1512775954598 - stderr: [sdp @ 0x7f81fd000000] audio codec set to: opus
    [sdp @ 0x7f81fd000000] audio samplerate set to: 48000
    [sdp @ 0x7f81fd000000] audio channels set to: 2
    1512775954639 - stderr: [sdp @ 0x7f81fd000000] video codec set to: h264
    [sdp @ 0x7f81fd000000] RTP Packetization Mode: 1
    [sdp @ 0x7f81fd000000] RTP Profile IDC: 42 Profile IOP: e0 Level: 1f
    [udp @ 0x7f81fcb007e0] end receive buffer size reported is 65536
    [udp @ 0x7f81fbe00180] end receive buffer size reported is 65536
    [sdp @ 0x7f81fd000000] setting jitter buffer size to 500
    [udp @ 0x7f81fbe00680] end receive buffer size reported is 65536
    [udp @ 0x7f81fbe00740] end receive buffer size reported is 65536
    [sdp @ 0x7f81fd000000] setting jitter buffer size to 500
    [sdp @ 0x7f81fd000000] Before avformat_find_stream_info() pos: 479 bytes read:479 seeks:0 nb_streams:2
    {"level":"info","time":"Dec 8, 2017 11:32 PM","message":"new active speaker","activePeer":"9f05d96a-9641-4c63-8f0e-486b98e48eb5"}
    1512775954773 - stderr: [AVBSFContext @ 0x7f81fc8018e0] nal_unit_type: 7, nal_ref_idc: 3
    [AVBSFContext @ 0x7f81fc8018e0] nal_unit_type: 8, nal_ref_idc: 3
    [AVBSFContext @ 0x7f81fc8018e0] nal_unit_type: 5, nal_ref_idc: 3
    1512775954774 - stderr: [h264 @ 0x7f8200000c00] nal_unit_type: 7, nal_ref_idc: 3
    1512775954774 - stderr: [h264 @ 0x7f8200000c00] nal_unit_type: 8, nal_ref_idc: 3
    1512775954774 - stderr: [h264 @ 0x7f8200000c00] nal_unit_type: 5, nal_ref_idc: 3
    1512775954774 - stderr: [h264 @ 0x7f8200000c00] Reinit context to 640x480, pix_fmt: yuv420p
    1512775954801 - stderr: [h264 @ 0x7f8200000c00] nal_unit_type: 1, nal_ref_idc: 3
    1512775954939 - stderr: [h264 @ 0x7f8200000c00] nal_unit_type: 7, nal_ref_idc: 3
    1512775954939 - stderr: [h264 @ 0x7f8200000c00] nal_unit_type: 8, nal_ref_idc: 3
    1512775954940 - stderr: [h264 @ 0x7f8200000c00] nal_unit_type: 5, nal_ref_idc: 3
    1512775955003 - stderr: [h264 @ 0x7f8200000c00] nal_unit_type: 1, nal_ref_idc: 3
    1512775955999 - stderr:     Last message repeated 3 times
    [sdp @ 0x7f81fd000000] All info found
    1512775955999 - stderr: [sdp @ 0x7f81fd000000] rfps: 29.750000 0.019566
    [sdp @ 0x7f81fd000000] rfps: 29.833333 0.015263
    [sdp @ 0x7f81fd000000] rfps: 29.916667 0.011503
    1512775955999 - stderr: [sdp @ 0x7f81fd000000] rfps: 30.000000 0.008285
    [sdp @ 0x7f81fd000000] rfps: 31.000000 0.011990
       Last message repeated 1 times
    [sdp @ 0x7f81fd000000] rfps: 29.970030 0.009380
    [sdp @ 0x7f81fd000000] After avformat_find_stream_info() pos: 479 bytes read:479 seeks:0 frames:98
    1512775956000 - stderr: Input #0, sdp, from 'data:text/plain;base64,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':
    1512775956000 - stderr:   Metadata:
       title           : e0c92fa0-dad5-11e7-8687-090dda95b1a4 fooboar
     Duration: N/A, start: 0.000000, bitrate: N/A
       Stream #0:0, 70, 1/48000: Audio: opus, 48000 Hz, stereo, fltp1512775956000 - stderr:
       Stream #0:1, 28, 1/90000: Video: h264 (Constrained Baseline), 1 reference frame, yuv420p(progressive, left), 640x480, 0/1, 30 tbr, 90k tbn, 180k tbc
    1512775956000 - stderr: Successfully opened the file.
    Parsing a group of options: output url /tmp/archive/e0c92fa0-dad5-11e7-8687-090dda95b1a4_10e1c990-dc70-11e7-888d-9f39ca0c79bc/1512775954465.m3u8.
    Applying option vcodec (force video codec ('copy' to copy stream)) with argument copy.
    Applying option acodec (force audio codec ('copy' to copy stream)) with argument aac.
    Applying option map (set input stream mapping) with argument 0:0,0:1.
    Applying option map (set input stream mapping) with argument 0:1,0:1.
    Applying option bsf:v (A comma-separated list of bitstream filters) with argument h264_mp4toannexb.
    Successfully parsed a group of options.
    1512775956000 - stderr: Opening an output file: /tmp/archive/e0c92fa0-dad5-11e7-8687-090dda95b1a4_10e1c990-dc70-11e7-888d-9f39ca0c79bc/1512775954465.m3u8.
    1512775956000 - stderr: Successfully opened the file.
    1512775956001 - stderr: [AVBSFContext @ 0x7f81fcb020a0] The input looks like it is Annex B already
    Stream mapping:
    1512775956001 - stderr:   Stream #0:0 -> #0:0 [sync #0:1] (opus (native) -> aac (native))
     Stream #0:1 -> #0:1 (copy)
    Press [q] to stop, [?] for help
    cur_dts is invalid (this is harmless if it occurs once at the start per stream)
    stream #0:
     keyframe=1
     duration=0.000
     dts=0.000  pts=0.000
     size=82
    1512775956001 - stderr: [SWR @ 0x7f820001e600] Using fltp internally between filters
    1512775956002 - stderr: detected 8 logical cores
    1512775956003 - stderr: [graph_0_in_0_0 @ 0x7f81fc90dfc0] Setting 'time_base' to value '1/48000'
    [graph_0_in_0_0 @ 0x7f81fc90dfc0] Setting 'sample_rate' to value '48000'
    1512775956003 - stderr: [graph_0_in_0_0 @ 0x7f81fc90dfc0] Setting 'sample_fmt' to value 'fltp'
    [graph_0_in_0_0 @ 0x7f81fc90dfc0] Setting 'channel_layout' to value '0x3'
    [graph_0_in_0_0 @ 0x7f81fc90dfc0] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x3
    [format_out_0_0 @ 0x7f81fc914da0] Setting 'sample_fmts' to value 'fltp'
    [format_out_0_0 @ 0x7f81fc914da0] Setting 'sample_rates' to value '96000|88200|64000|48000|44100|32000|24000|22050|16000|12000|11025|8000|7350'
    1512775956004 - stderr: [AVFilterGraph @ 0x7f81fbd02200] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed
    1512775956007 - stderr: [hls @ 0x7f81fd80f000] Opening '/tmp/archive/e0c92fa0-dad5-11e7-8687-090dda95b1a4_10e1c990-dc70-11e7-888d-9f39ca0c79bc/15127759544650.ts' for writing
    [file @ 0x7f81fbf01ee0] Setting default whitelist 'file,crypto'
    1512775956007 - stderr: [mpegts @ 0x7f81fd877800] muxrate VBR, pcr every 9000 pkts, sdt every 2147483647, pat/pmt every 2147483647 pkts
    Output #0, hls, to '/tmp/archive/e0c92fa0-dad5-11e7-8687-090dda95b1a4_10e1c990-dc70-11e7-888d-9f39ca0c79bc/1512775954465.m3u8':
     Metadata:
       title           : e0c92fa0-dad5-11e7-8687-090dda95b1a4 fooboar
       encoder         : Lavf57.83.100
    1512775956007 - stderr:     Stream #0:0, 0, 1/90000: Audio: aac (LC), 48000 Hz, stereo, fltp, delay 1024, 128 kb/s
       Metadata:
         encoder         : Lavc57.107.100 aac
       Stream #0:1, 0, 1/90000: Video: h264 (Constrained Baseline), 1 reference frame, yuv420p(progressive, left), 640x480 (0x0), 0/1, q=2-31, 30 tbr, 90k tbn, 90k tbc
    cur_dts is invalid (this is harmless if it occurs once at the start per stream)
    1512775956007 - stderr:     Last message repeated 1 times
    stream #0:
     keyframe=1
     duration=0.000
     dts=0.020  pts=0.020
     size=79
    1512775956007 - stderr: cur_dts is invalid (this is harmless if it occurs once at the start per stream)
       Last message repeated 1 times
    stream #0:
     keyframe=1
     duration=0.000
    1512775956007 - stderr:   dts=0.040  pts=0.040
     size=75
    1512775956014 - stderr: cur_dts is invalid (this is harmless if it occurs once at the start per stream)
    stream #0:
     keyframe=1
     duration=0.000
     dts=0.0601512775956014 - stderr:   pts=0.060
     size=81
    1512775956017 - stderr: cur_dts is invalid (this is harmless if it occurs once at the start per stream)
    stream #0:
     keyframe=1
     duration=0.000
     dts=0.080  pts=0.080
     size=76
    1512775956022 - stderr: cur_dts is invalid (this is harmless if it occurs once at the start per stream)
    stream #0:
     keyframe=1
     duration=0.000
     dts=0.1001512775956022 - stderr:   pts=0.100
     size=79
    1512775956023 - stderr: cur_dts is invalid (this is harmless if it occurs once at the start per stream)
    stream #0:
     keyframe=1
    1512775956023 - stderr:   duration=0.000
     dts=0.120  pts=0.120
     size=95
    1512775956024 - stderr: cur_dts is invalid (this is harmless if it occurs once at the start per stream)
    1512775956024 - stderr: stream #0:
     keyframe=1
     duration=0.000
     dts=0.140  pts=0.140
     size=93
    1512775956025 - stderr: cur_dts is invalid (this is harmless if it occurs once at the start per stream)
    stream #0:
     keyframe=1
    1512775956025 - stderr:   duration=0.000
     dts=0.160  pts=0.160
     size=94
    1512775956026 - stderr: cur_dts is invalid (this is harmless if it occurs once at the start per stream)
    stream #1:
     keyframe=1
    1512775956026 - stderr:   duration=0.000
     dts=N/A  pts=N/A
     size=992
    [hls @ 0x7f81fd80f000] Timestamps are unset in a packet for stream 1. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
    1512775956026 - stderr: stream #0:
     keyframe=1
     duration=0.000
    1512775956026 - stderr:   dts=0.180  pts=0.180
     size=108
    1512775956027 - stderr: stream #1:
     keyframe=0
     duration=0.000
    1512775956027 - stderr:   dts=0.002  pts=0.002
     size=3047
    [hls @ 0x7f81fd80f000] pkt->duration = 0, maybe the hls segment duration will not precise
    stream #0:
     keyframe=1
     duration=0.000
     dts=0.200  pts=0.200
     size=89
    1512775956060 - stderr: stream #0:
     keyframe=1
     duration=0.000
     dts=0.220  pts=0.220
     size=73
    stream #0:
     keyframe=1
     duration=0.000
     dts=0.240  pts=0.240
     size=78
    stream #0:
     keyframe=1
     duration=0.000
     dts=0.260  pts=0.260

    Notice how the first frame for stream #1 (video) starts after a number of audio frames ? Specifically, it starts at stream #0 dts/pts 0.18. In this situation, the a/v sync issue is pretty much unnoticeable, but with a bunch of repros, I’ve determined that the a/v sync offset is always the duration of however long audio frames were sent before the first video frame (sometimes seconds). I’m consistently starting the RTP streams only tens of ms apart, so I can’t control for this variance on the input side.

    After the initial audio frames come in, the first video frame has a dts/pts around 0. What ffmpeg setting would I use to adjust the timestamps accordingly ? I don’t care about losing the starting audio that doesn’t have video, so any solution that would adjust the timestamps works.

  • Merge commit ’7e18a727d2c2a19f22fcf68875d1b05fd2eafcef’

    18 juillet 2014, par Michael Niedermayer
    Merge commit ’7e18a727d2c2a19f22fcf68875d1b05fd2eafcef’
    

    * commit ’7e18a727d2c2a19f22fcf68875d1b05fd2eafcef’ :
    arm : cosmetics : Consistently use lowercase for shift operators

    Merged-by : Michael Niedermayer <michaelni@gmx.at>

    • [DH] libavcodec/arm/mlpdsp_armv5te.S
    • [DH] libavcodec/arm/synth_filter_vfp.S
  • Fixing "RTP : dropping old packet received too late" in FFMPEG

    25 juin 2014, par user985030

    I have been using FFMPEG 0.6 for years with no problems and recently ported much of my code to 2.2 ; however, there is still a problem that I cannot resolve after fixing many of the deprecated functions. I am generating simulated video and then using RTSP to unicast this generated stream. The problem is that when I change the height and width of my video data, I basically recreate a new stream to send the subscribed client. The algorithm I used to do this in 0.6 worked like a charm, never had any problems. Now that I have upgraded, I get "RTP : dropping old packet received too late" as soon as I change my frame size. I think I have been dropping packets all along, but the new code is causing connection issues for me. The packets being dropped in the past were negligible and I really didn’t care if I missed them as long as the stream eventually corrected itself. Is there a flag that I can set to not drop these packets ? Or at least recover more quickly ? I believe that this has something to do with receiving packets out of order. There is a section of code that does a diff in FFMPEG in the rtpdec.c file in the function rtp_parts_one_packet. The only reference I found similar to my issue is here :

    http://en.it-usenet.org/thread/16949/6708/#post6707. Any tips would be greatly appreciated. In the meantime, I am just going to patch the FFMPEG code to not do the following check :

    if (diff &lt; 0) {

    /* Packet older than the previously emitted one, drop */

    av_log(s->st ? s->st->codec : NULL,        AV_LOG_WARNING,

    "RTP: dropping old packet received too late\n");

    return -1;

    }

    By commenting out the above code, I am able to run my streaming application like I used to, but I have a feeling that I am not doing something correctly but I am not sure what it is. Thanks for any advice !