[Libav-user] Could not find codec for mpeg4 on rtsp？

Tue Jul 5 07:27:59 CEST 2016

Hi all,
I use gstreamer make a mp4 file to a RTSP stream, and play it with ffplay.exe on windows.

But ffplay report that it could not find video codec for this stream.

Below is the debug log. What's wrong with it?

ffplay.exe -debug er rtsp://127.0.0.1:8554/test
ffplay version N-80906-gd5edb6c Copyright (c) 2003-2016 the FFmpeg developers
  built with gcc 5.4.0 (GCC)
  configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --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-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. 28.100 / 55. 28.100
  libavcodec     57. 48.101 / 57. 48.101
  libavformat    57. 41.100 / 57. 41.100
  libavdevice    57.  0.102 / 57.  0.102
  libavfilter     6. 47.100 /  6. 47.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
  libpostproc    54.  0.100 / 54.  0.100
[tcp @ 00000000000cc700] No default whitelist set sq=    0B f=0/0
[rtsp @ 00000000000ccb20] SDP:
v=0
o=- 14003480297848734151 1 IN IP4 127.0.0.1
s=Session streamed with GStreamer
i=rtsp-server
t=0 0
a=tool:GStreamer
a=type:broadcast
a=control:*
a=range:npt=0-308.731666666
m=video 0 RTP/AVP 96
c=IN IP4 0.0.0.0
b=AS:327
a=rtpmap:96 MPEG4-GENERIC/90000
a=framerate:30
a=fmtp:96 streamtype=4;profile-level-id=1;mode=generic;config=000001b001000001b58913000001000000012000c48d8800f50a041e1463000001b2476f6f676c65;sizelength=13;indexlength=3;indexdeltalength=3
a=control:stream=0
m=audio 0 RTP/AVP 97
c=IN IP4 0.0.0.0
b=AS:35
a=rtpmap:97 MP4A-LATM/22050
a=fmtp:97 cpresent=0;config=400027100000000000000000000000000000
a=control:stream=1

[rtsp @ 00000000000ccb20] video codec set to: (null)=    0B f=0/0
[rtsp @ 00000000000ccb20] audio codec set to: aac
[rtsp @ 00000000000ccb20] audio samplerate set to: 22050
[rtsp @ 00000000000ccb20] audio channels set to: 1
[rtp @ 00000000000ceee0] No default whitelist set sq=    0B f=0/0
[udp @ 000000000222edc0] No default whitelist set
[udp @ 000000000222edc0] end receive buffer size reported is 65536
[udp @ 0000000002240520] No default whitelist set
[udp @ 0000000002240520] end receive buffer size reported is 65536
[rtsp @ 00000000000ccb20] setting jitter buffer size to 500 f=0/0
[rtp @ 0000000002250b20] No default whitelist set
[udp @ 0000000002252d80] No default whitelist set
[udp @ 0000000002252d80] end receive buffer size reported is 65536
[udp @ 0000000002263020] No default whitelist set
[udp @ 0000000002263020] end receive buffer size reported is 65536
[rtsp @ 00000000000ccb20] setting jitter buffer size to 500
[rtsp @ 00000000000ccb20] hello state=0vq=    0KB sq=    0B f=0/0
[rtsp @ 00000000000ccb20] Non-increasing DTS in stream 0: packet 2 with DTS 0, packet 3 with DTS 0
[rtsp @ 00000000000ccb20] Non-increasing DTS in stream 0: packet 3 with DTS 0, packet 4 with DTS 0
[rtsp @ 00000000000ccb20] Non-increasing DTS in stream 0: packet 4 with DTS 0, packet 5 with DTS 0
[rtsp @ 00000000000ccb20] Non-increasing DTS in stream 0: packet 65 with DTS 180000, packet 66 with DTS 180000
[rtsp @ 00000000000ccb20] Non-increasing DTS in stream 0: packet 66 with DTS 180000, packet 67 with DTS 180000
[rtsp @ 00000000000ccb20] Non-increasing DTS in stream 0: packet 67 with DTS 180000, packet 68 with DTS 180000
[rtsp @ 00000000000ccb20] Non-increasing DTS in stream 0: packet 128 with DTS 360000, packet 129 with DTS 360000
[rtsp @ 00000000000ccb20] Non-increasing DTS in stream 0: packet 129 with DTS 360000, packet 130 with DTS 360000
[rtsp @ 00000000000ccb20] Non-increasing DTS in stream 0: packet 130 with DTS 360000, packet 131 with DTS 360000
[rtsp @ 00000000000ccb20] max_analyze_duration 5000000 reached at 5015510 microseconds st:1
[rtsp @ 00000000000ccb20] rfps: 29.916667 0.014984
    Last message repeated 1 times
[rtsp @ 00000000000ccb20] rfps: 30.000000 0.000001
[rtsp @ 00000000000ccb20] rfps: 60.000000 0.000002
[rtsp @ 00000000000ccb20] rfps: 120.000000 0.000010
[rtsp @ 00000000000ccb20] rfps: 240.000000 0.000040
[rtsp @ 00000000000ccb20] rfps: 29.970030 0.001923
    Last message repeated 1 times
[rtsp @ 00000000000ccb20] rfps: 59.940060 0.007691
    Last message repeated 1 times
[rtsp @ 00000000000ccb20] Setting avg frame rate based on r frame rate
[rtsp @ 00000000000ccb20] Could not find codec parameters for stream 0 (Video: none, 1 reference frame, none): unknown codec
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, rtsp, from 'rtsp://127.0.0.1:8554/test':
  Metadata::  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
    title           : Session streamed with GStreamer
    comment         : rtsp-server
  Duration: 00:05:08.73, start: 0.000000, bitrate: N/A
    Stream #0:0, 165, 1/90000: Video: none, 1 reference frame, none, 30 fps, 30 tbr, 90k tbn, 90k tbc
    Stream #0:1    nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0  , 110, 1/22050: Audio: aac (LC), 22050 Hz, mono, fltp
detected 4 logical cores
[ffplay_abuffer @ 00000000022c8600] Setting 'sample_rate' to value '22050'
[ffplay_abuffer @ 00000000022c8600] Setting 'sample_fmt' to value 'fltp'
[ffplay_abuffer @ 00000000022c8600] Setting 'channels' to value '1'
[ffplay_abuffer @ 00000000022c8600] Setting 'time_base' to value '1/22050'
[ffplay_abuffer @ 00000000022c8600] Setting 'channel_layout' to value '0x4'
[ffplay_abuffer @ 00000000022c8600] tb:1/22050 samplefmt:fltp samplerate:22050 chlayout:0x4
[ffplay_abuffersink @ 000000000229a940] auto-inserting filter 'auto-inserted resampler 0' between the filter 'ffplay_abuffer' and the filter 'ffplay_abuffersink'
[AVFilterGraph @ 00000000022d8260] query_formats: 2 queried, 0 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 000000000227bd80] [SWR @ 00000000030b6fc0] Using fltp internally between filters
[auto-inserted resampler 0 @ 000000000227bd80] ch:1 chl:mono fmt:fltp r:22050Hz -> ch:1 chl:mono fmt:s16 r:22050Hz
No codec could be found with id 0
Audio frame changed from rate:22050 ch:1 fmt:fltp layout:mono serial:-1 to rate:22050 ch:1 fmt:fltp layout:mono serial:1
[ffplay_abuffer @ 00000000022a1100] Setting 'sample_rate' to value '22050'
[ffplay_abuffer @ 00000000022a1100] Setting 'sample_fmt' to value 'fltp'
[ffplay_abuffer @ 00000000022a1100] Setting 'channels' to value '1'
[ffplay_abuffer @ 00000000022a1100] Setting 'time_base' to value '1/22050'
[ffplay_abuffer @ 00000000022a1100] Setting 'channel_layout' to value '0x4'
[ffplay_abuffer @ 00000000022a1100] tb:1/22050 samplefmt:fltp samplerate:22050 chlayout:0x4
[ffplay_abuffersink @ 000000000229a940] auto-inserting filter 'auto-inserted resampler 0' between the filter 'ffplay_abuffer' and the filter 'ffplay_abuffersink'
[AVFilterGraph @ 00000000022d7d40] query_formats: 2 queried, 0 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 00000000022b1a20] [SWR @ 00000000030b6f40] Using fltp internally between filters
[auto-inserted resampler 0 @ 00000000022b1a20] ch:1 chl:mono fmt:fltp r:22050Hz -> ch:1 chl:mono fmt:s16 r:22050Hz
   9.55 M-A:  0.000 fd=   0 aq=   31KB vq=    0KB sq=    0B f=0/0

zhangweili at ragile.com
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