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
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