gst-rtsp video only works but video and audio with mpeg2 ts mux fails

[krakorar]

All:

The following line ("Video Only works" below) works well form me to capture video from my USB webcam, encode and stream over the network.  However, I also have a PCM microphone at ALSA hw:0,1 and I would like to use the MPEG2 transport muxer and RTP payloader to stream out video and audio but it fails ("Video and Audio muxed via MPEG2 transport fails" further below).  "test-launch" is actually gst-rtsp.  What am I doing wrong?  ;-)

Video Only works:

[root@am2mm examples]# ./test-launch --gst-debug=2 "( v4l2src device=/dev/video0 ! ffmpegcolorspace ! videoscale ! video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffenc_mpeg4 bitrate=6000000 ! rtpmp4vpay pt=96 name=pay0 )"** Message: listening on port 8554
** Message: added new client 0x18eb2360 ip 192.168.1.211:47809
** Message: attaching to context 0x18eac880
RTSP request message 0x1902de08
 request line:
   method: 'OPTIONS'
   uri:    'rtsp://192.168.1.178:8554'
   version: '1.0'
 headers:
   key: 'CSeq', value: '1'
   key: 'User-Agent', value: 'RealMedia Player Version 6.0.9.1235 (linux-2.0-libc6-i386-gcc2.95)'
   key: 'ClientChallenge', value: '9e26d33f2984236010ef6253fb1887f7'
   key: 'PlayerStarttime', value: '[28/03/2003:22:50:23 00:00]'
   key: 'CompanyID', value: 'KnKV4M4I/B2FjJ1TToLycw=='
   key: 'GUID', value: '00000000-0000-0000-0000-000000000000'
   key: 'RegionData', value: '0'
   key: 'ClientID', value: 'Linux_2.4_6.0.9.1235_play32_RN01_EN_586'
 body:
** Message: client 0x18eb2360: received a request
RTSP response message 0x7fff80ad7ad0
 status line:
   code:   '200'
   reason: 'OK'
   version: '1.0'
 headers:
   key: 'CSeq', value: '1'
   key: 'Public', value: 'OPTIONS, DESCRIBE, GET_PARAMETER, PAUSE, PLAY, SETUP, SET_PARAMETER, TEARDOWN'
   key: 'Server', value: 'GStreamer RTSP server'
 body: length 0
RTSP request message 0x1902de08
 request line:
   method: 'DESCRIBE'
   uri:    'rtsp://192.168.1.178:8554/test'
   version: '1.0'
 headers:
   key: 'CSeq', value: '2'
   key: 'Accept', value: 'application/sdp'
 body:
** Message: client 0x18eb2360: received a request
** Message: found media 0x18ffe300 for url abspath /test
** Message: enter mainloop
** Message: found stream 0 with payloader 0x190f6000
** Message: constructed media 0x19103010 for url /test
** Message: preparing media 0x19103010
** Message: live media 0x19103010
0:00:04.936055000  6397     0x18ea9010 WARN                     bin gstbin.c:2312:gst_bin_do_latency_func:<media-pipeline> failed to query latency
** Message: 0x19103010: got message type new-clock
0:00:06.102703000  6397     0x19142020 WARN           basetransform gstbasetransform.c:1049:gst_base_transform_acceptcaps:<videoscale0> transform could not transform video/x-raw-yuv, format=(fourcc)YUY2, framerate=(fraction)30/1, width=(int)640, height=(int)480 in anything we support
** Message: stream 0x1910d530 received caps 0x19147080, application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)MP4V-ES, profile-level-id=(string)1, config=(string)000001b001000001b58913000001000000012000c48d8800f514043c1463000001b24c61766335322e32302e30, ssrc=(guint)1892808546, payload=(int)96, clock-base=(guint)2225205298, seqnum-base=(guint)36350
** Message: 0x19103010: got message type async-done
** Message: object 0x19103010 is prerolled
RTSP response message 0x7fff80ad7a90
 status line:
   code:   '200'
   reason: 'OK'
   version: '1.0'
 headers:
   key: 'CSeq', value: '2'
   key: 'Content-Type', value: 'application/sdp'
   key: 'Content-Base', value: 'rtsp://192.168.1.178:8554/test/'
   key: 'Server', value: 'GStreamer RTSP server'
 body: length 382
00000000 (0x1914d000): 76 3d 30 0d 0a 6f 3d 2d 20 31 31 38 38 33 34 30  v=0..o=- 1188340
00000010 (0x1914d010): 36 35 36 31 38 30 38 38 33 20 31 20 49 4e 20 49  656180883 1 IN I
00000020 (0x1914d020): 50 34 20 31 32 37 2e 30 2e 30 2e 31 0d 0a 73 3d  P4 127.0.0.1..s=
00000030 (0x1914d030): 53 65 73 73 69 6f 6e 20 73 74 72 65 61 6d 65 64  Session streamed
00000040 (0x1914d040): 20 77 69 74 68 20 47 53 74 72 65 61 6d 65 72 0d   with GStreamer.
00000050 (0x1914d050): 0a 69 3d 72 74 73 70 2d 73 65 72 76 65 72 0d 0a  .i=rtsp-server..
00000060 (0x1914d060): 65 3d 4e 4f 4e 45 0d 0a 74 3d 30 20 30 0d 0a 61  e=NONE..t=0 0..a
00000070 (0x1914d070): 3d 74 6f 6f 6c 3a 47 53 74 72 65 61 6d 65 72 0d  =tool:GStreamer.
00000080 (0x1914d080): 0a 61 3d 74 79 70 65 3a 62 72 6f 61 64 63 61 73  .a=type:broadcas
00000090 (0x1914d090): 74 0d 0a 61 3d 72 61 6e 67 65 3a 6e 70 74 3d 6e  t..a=range:npt=n
000000a0 (0x1914d0a0): 6f 77 2d 0d 0a 6d 3d 76 69 64 65 6f 20 30 20 52  ow-..m=video 0 R
000000b0 (0x1914d0b0): 54 50 2f 41 56 50 20 39 36 0d 0a 63 3d 49 4e 20  TP/AVP 96..c=IN
000000c0 (0x1914d0c0): 49 50 34 20 31 32 37 2e 30 2e 30 2e 31 0d 0a 61  IP4 127.0.0.1..a
000000d0 (0x1914d0d0): 3d 72 74 70 6d 61 70 3a 39 36 20 4d 50 34 56 2d  =rtpmap:96 MP4V-
000000e0 (0x1914d0e0): 45 53 2f 39 30 30 30 30 0d 0a 61 3d 63 6f 6e 74  ES/90000..a=cont
000000f0 (0x1914d0f0): 72 6f 6c 3a 73 74 72 65 61 6d 3d 30 0d 0a 61 3d  rol:stream=0..a=
00000100 (0x1914d100): 66 6d 74 70 3a 39 36 20 70 72 6f 66 69 6c 65 2d  fmtp:96 profile-
00000110 (0x1914d110): 6c 65 76 65 6c 2d 69 64 3d 31 3b 63 6f 6e 66 69  level-id=1;confi
00000120 (0x1914d120): 67 3d 30 30 30 30 30 31 62 30 30 31 30 30 30 30  g=000001b0010000
00000130 (0x1914d130): 30 31 62 35 38 39 31 33 30 30 30 30 30 31 30 30  01b5891300000100
00000140 (0x1914d140): 30 30 30 30 30 31 32 30 30 30 63 34 38 64 38 38  0000012000c48d88
00000150 (0x1914d150): 30 30 66 35 31 34 30 34 33 63 31 34 36 33 30 30  00f514043c146300
00000160 (0x1914d160): 30 30 30 31 62 32 34 63 36 31 37 36 36 33 33 35  0001b24c61766335
00000170 (0x1914d170): 33 32 32 65 33 32 33 30 32 65 33 30 0d 0a        322e32302e30.. 
** Message: client 0x18eb2360: connection closed
** Message: finalize client 0x18eb2360
** Message: finalize media 0x19103010
** Message: stream 0x1910d530 received caps (nil), NULL
** Message: added new client 0x19125400 ip 192.168.1.211:48065
** Message: attaching to context 0x18eac880
RTSP request message 0x2aaaac005688
 request line:
   method: 'DESCRIBE'
   uri:    'rtsp://192.168.1.178:8554/test'
   version: '1.0'
 headers:
   key: 'CSeq', value: '1'
   key: 'Accept', value: 'application/sdp'
   key: 'User-Agent', value: 'MPlayer (LIVE555 Streaming Media v2009.07.27)'
 body:
** Message: client 0x19125400: received a request
** Message: found media 0x18ffe300 for url abspath /test
** Message: found stream 0 with payloader 0x190f6240
** Message: constructed media 0x191030b0 for url /test
** Message: preparing media 0x191030b0
** Message: live media 0x191030b0
0:00:06.570895000  6397     0x18ea9010 WARN                     bin gstbin.c:2312:gst_bin_do_latency_func:<media-pipeline> failed to query latency
** Message: 0x191030b0: got message type new-clock
0:00:07.736286000  6397     0x19153550 WARN           basetransform gstbasetransform.c:1049:gst_base_transform_acceptcaps:<videoscale1> transform could not transform video/x-raw-yuv, format=(fourcc)YUY2, framerate=(fraction)30/1, width=(int)640, height=(int)480 in anything we support
** Message: stream 0x2aaaac015f50 received caps 0x1938a440, application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)MP4V-ES, profile-level-id=(string)1, config=(string)000001b001000001b58913000001000000012000c48d8800f514043c1463000001b24c61766335322e32302e30, ssrc=(guint)3436990960, payload=(int)96, clock-base=(guint)2874306155, seqnum-base=(guint)15784
** Message: 0x191030b0: got message type async-done
** Message: object 0x191030b0 is prerolled
RTSP response message 0x7fff80ad7a90
 status line:
   code:   '200'
   reason: 'OK'
   version: '1.0'
 headers:
   key: 'CSeq', value: '1'
   key: 'Content-Type', value: 'application/sdp'
   key: 'Content-Base', value: 'rtsp://192.168.1.178:8554/test/'
   key: 'Server', value: 'GStreamer RTSP server'
 body: length 382
00000000 (0x191eedf0): 76 3d 30 0d 0a 6f 3d 2d 20 31 31 38 38 33 34 30  v=0..o=- 1188340
00000010 (0x191eee00): 36 35 36 31 38 30 38 38 33 20 31 20 49 4e 20 49  656180883 1 IN I
00000020 (0x191eee10): 50 34 20 31 32 37 2e 30 2e 30 2e 31 0d 0a 73 3d  P4 127.0.0.1..s=
00000030 (0x191eee20): 53 65 73 73 69 6f 6e 20 73 74 72 65 61 6d 65 64  Session streamed
00000040 (0x191eee30): 20 77 69 74 68 20 47 53 74 72 65 61 6d 65 72 0d   with GStreamer.
00000050 (0x191eee40): 0a 69 3d 72 74 73 70 2d 73 65 72 76 65 72 0d 0a  .i=rtsp-server..
00000060 (0x191eee50): 65 3d 4e 4f 4e 45 0d 0a 74 3d 30 20 30 0d 0a 61  e=NONE..t=0 0..a
00000070 (0x191eee60): 3d 74 6f 6f 6c 3a 47 53 74 72 65 61 6d 65 72 0d  =tool:GStreamer.
00000080 (0x191eee70): 0a 61 3d 74 79 70 65 3a 62 72 6f 61 64 63 61 73  .a=type:broadcas
00000090 (0x191eee80): 74 0d 0a 61 3d 72 61 6e 67 65 3a 6e 70 74 3d 6e  t..a=range:npt=n
000000a0 (0x191eee90): 6f 77 2d 0d 0a 6d 3d 76 69 64 65 6f 20 30 20 52  ow-..m=video 0 R
000000b0 (0x191eeea0): 54 50 2f 41 56 50 20 39 36 0d 0a 63 3d 49 4e 20  TP/AVP 96..c=IN
000000c0 (0x191eeeb0): 49 50 34 20 31 32 37 2e 30 2e 30 2e 31 0d 0a 61  IP4 127.0.0.1..a
000000d0 (0x191eeec0): 3d 72 74 70 6d 61 70 3a 39 36 20 4d 50 34 56 2d  =rtpmap:96 MP4V-
000000e0 (0x191eeed0): 45 53 2f 39 30 30 30 30 0d 0a 61 3d 63 6f 6e 74  ES/90000..a=cont
000000f0 (0x191eeee0): 72 6f 6c 3a 73 74 72 65 61 6d 3d 30 0d 0a 61 3d  rol:stream=0..a=
00000100 (0x191eeef0): 66 6d 74 70 3a 39 36 20 70 72 6f 66 69 6c 65 2d  fmtp:96 profile-
00000110 (0x191eef00): 6c 65 76 65 6c 2d 69 64 3d 31 3b 63 6f 6e 66 69  level-id=1;confi
00000120 (0x191eef10): 67 3d 30 30 30 30 30 31 62 30 30 31 30 30 30 30  g=000001b0010000
00000130 (0x191eef20): 30 31 62 35 38 39 31 33 30 30 30 30 30 31 30 30  01b5891300000100
00000140 (0x191eef30): 30 30 30 30 30 31 32 30 30 30 63 34 38 64 38 38  0000012000c48d88
00000150 (0x191eef40): 30 30 66 35 31 34 30 34 33 63 31 34 36 33 30 30  00f514043c146300
00000160 (0x191eef50): 30 30 30 31 62 32 34 63 36 31 37 36 36 33 33 35  0001b24c61766335
00000170 (0x191eef60): 33 32 32 65 33 32 33 30 32 65 33 30 0d 0a        322e32302e30.. 
RTSP request message 0x2aaaac005688
 request line:
   method: 'SETUP'
   uri:    'rtsp://192.168.1.178:8554/test/stream=0'
   version: '1.0'
 headers:
   key: 'CSeq', value: '2'
   key: 'Transport', value: 'RTP/AVP;unicast;client_port=34204-34205'
   key: 'User-Agent', value: 'MPlayer (LIVE555 Streaming Media v2009.07.27)'
 body:
** Message: client 0x19125400: received a request
** Message: reusing cached media 0x191030b0
** Message: manage new media 0x191030b0 in session 0x1906c3d0
RTSP response message 0x7fff80ad7a50
 status line:
   code:   '200'
   reason: 'OK'
   version: '1.0'
 headers:
   key: 'CSeq', value: '2'
   key: 'Transport', value: 'RTP/AVP;unicast;client_port=34204-34205;server_port=50644-50645;mode="PLAY"'
   key: 'Server', value: 'GStreamer RTSP server'
   key: 'Session', value: 'xbnktocikhmrcsny'
 body: length 0
RTSP request message 0x2aaaac005688
 request line:
   method: 'PLAY'
   uri:    'rtsp://192.168.1.178:8554/test/'
   version: '1.0'
 headers:
   key: 'CSeq', value: '3'
   key: 'Session', value: 'xbnktocikhmrcsny'
   key: 'Range', value: 'npt=0.000-'
   key: 'User-Agent', value: 'MPlayer (LIVE555 Streaming Media v2009.07.27)'
 body:
** Message: client 0x19125400: received a request
** Message: watching session 0x1912dd40
** Message: seeking to 0:00:00.000000000 - 99:99:99.999999999
** Message: done seeking 0
** Message: prerolled again
RTSP response message 0x7fff80ad7a10
 status line:
   code:   '200'
   reason: 'OK'
   version: '1.0'
 headers:
   key: 'CSeq', value: '3'
   key: 'RTP-Info', value: 'url=rtsp://192.168.1.178:8554/test/stream=0;seq=15811;rtptime=2874407267'
   key: 'Range', value: 'npt=now-'
   key: 'Server', value: 'GStreamer RTSP server'
   key: 'Session', value: 'xbnktocikhmrcsny'
 body: length 0
** Message: going to state PLAYING media 0x191030b0
** Message: adding 192.168.1.211:34204-34205
** Message: active 1 media 0x191030b0
** Message: state PLAYING media 0x191030b0
** Message: 0x2aaaac015f50: new source 0x19130cc0
** Message: structure: application/x-rtp-source-stats, ssrc=(guint)3763749761, internal=(boolean)false, validated=(boolean)false, received-bye=(boolean)false, is-csrc=(boolean)false, is-sender=(boolean)false, rtcp-from=(string)192.168.1.211:34205, have-rb=(boolean)false, rb-fractionlost=(guint)0, rb-packetslost=(int)0, rb-exthighestseq=(guint)0, rb-jitter=(guint)0, rb-lsr=(guint)0, rb-dlsr=(guint)0, rb-round-trip=(guint)0;
** Message: finding 192.168.1.211:34205
** Message: 0x2aaaac015f50: found transport 0x19054b90 for source  0x19130cc0
** Message: 0x2aaaac015f50: source 0x19130cc0 in transport 0x19054b90 is active
** Message: keep session 0x1912dd40 alive
** Message: 0x2aaaac015f50: new SDES 0x19130cc0

[root@am2mm examples]#


Video and Audio muxed via MPEG2 transport fails:

[root@am2mm examples]# ./test-launch --gst-debug=2 "( v4l2src device=/dev/video0 ! ffmpegcolorspace ! videoscale ! video/x-raw-yuv,width=640,height=480 ! ffenc_mpeg4 bitrate=6000000 ! ! queue ! mpegtsmux name=mux alsasrc device="hw:0" ! audio/x-raw-int ! lamemp3enc ! mux. rtpmp2tpay pt=96 name=pay0 )"
** Message: listening on port 8554
** Message: added new client 0x8aa8360 ip 192.168.1.211:19403
** Message: attaching to context 0x8aa2880
RTSP request message 0x8c23e48
 request line:
   method: 'OPTIONS'
   uri:    'rtsp://192.168.1.178:8554'
   version: '1.0'
 headers:
   key: 'CSeq', value: '1'
   key: 'User-Agent', value: 'RealMedia Player Version 6.0.9.1235 (linux-2.0-libc6-i386-gcc2.95)'
   key: 'ClientChallenge', value: '9e26d33f2984236010ef6253fb1887f7'
   key: 'PlayerStarttime', value: '[28/03/2003:22:50:23 00:00]'
   key: 'CompanyID', value: 'KnKV4M4I/B2FjJ1TToLycw=='
   key: 'GUID', value: '00000000-0000-0000-0000-000000000000'
   key: 'RegionData', value: '0'
   key: 'ClientID', value: 'Linux_2.4_6.0.9.1235_play32_RN01_EN_586'
 body:
** Message: client 0x8aa8360: received a request
RTSP response message 0x7fff2b639a70
 status line:
   code:   '200'
   reason: 'OK'
   version: '1.0'
 headers:
   key: 'CSeq', value: '1'
   key: 'Public', value: 'OPTIONS, DESCRIBE, GET_PARAMETER, PAUSE, PLAY, SETUP, SET_PARAMETER, TEARDOWN'
   key: 'Server', value: 'GStreamer RTSP server'
 body: length 0
RTSP request message 0x8c23e48
 request line:
   method: 'DESCRIBE'
   uri:    'rtsp://192.168.1.178:8554/test'
   version: '1.0'
 headers:
   key: 'CSeq', value: '2'
   key: 'Accept', value: 'application/sdp'
 body:
** Message: client 0x8aa8360: received a request
** Message: found media 0x8bf4300 for url abspath /test
0:00:04.444390000  6284      0x8a9f010 ERROR           GST_PIPELINE ./grammar.y:742:_gst_parse_yyparse: link without source element

** (lt-test-launch:6284): WARNING **: recoverable parsing error: link without source element
** Message: enter mainloop
** Message: found stream 0 with payloader 0x8d0a0a0
** Message: constructed media 0x8d1e800 for url /test
** Message: preparing media 0x8d1e800
** Message: live media 0x8d1e800
0:00:04.713394000  6284      0x8a9f010 WARN                     bin gstbin.c:2312:gst_bin_do_latency_func:<media-pipeline> failed to query latency
** Message: 0x8d1e800: got message type new-clock
0:00:05.879033000  6284      0x8d4ea10 WARN           basetransform gstbasetransform.c:1049:gst_base_transform_acceptcaps:<videoscale0> transform could not transform video/x-raw-yuv, format=(fourcc)YUY2, framerate=(fraction)100/1, width=(int)640, height=(int)480 in anything we support
0:00:05.932073000  6284      0x8d4a8d0 WARN               mpegtsmux mpegtsmux.c:884:mpegtsdemux_prepare_srcpad:<mux> New segment event was not handled
0:00:05.934343000  6284      0x8d476d0 WARN            baseaudiosrc gstbaseaudiosrc.c:810:gst_base_audio_src_create:<alsasrc0> create DISCONT of 51840 samples at sample 53376
0:00:05.934647000  6284      0x8d476d0 WARN            baseaudiosrc gstbaseaudiosrc.c:815:gst_base_audio_src_create:<alsasrc0> warning: Can't record audio fast enough
0:00:05.934676000  6284      0x8d476d0 WARN            baseaudiosrc gstbaseaudiosrc.c:815:gst_base_audio_src_create:<alsasrc0> warning: Dropped 51840 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.

** (lt-test-launch:6284): WARNING **: 0x8d1e800: got warning Can't record audio fast enough (gstbaseaudiosrc.c(815): gst_base_audio_src_create (): /GstPipeline:media-pipeline/GstBin:bin0/GstAlsaSrc:alsasrc0:
Dropped 51840 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.)
0:00:05.995885000  6284      0x8d4ea10 WARN                 basesrc gstbasesrc.c:2378:gst_base_src_loop:<v4l2src0> error: Internal data flow error.
0:00:05.995984000  6284      0x8d4ea10 WARN                 basesrc gstbasesrc.c:2378:gst_base_src_loop:<v4l2src0> error: streaming task paused, reason error (-5)

** (lt-test-launch:6284): WARNING **: 0x8d1e800: got error Internal data flow error. (gstbasesrc.c(2378): gst_base_src_loop (): /GstPipeline:media-pipeline/GstBin:bin0/GstV4l2Src:v4l2src0:
streaming task paused, reason error (-5))

[root@am2mm examples]#
[root@am2mm examples]#
[root@am2mm examples]# ./test-launch --gst-debug=2 "( v4l2src device=/dev/video0 ! ffmpegcolorspace ! videoscale ! video/x-raw-yuv,width=640,height=480 ! ffenc_mpeg4 bitrate=6000000 ! queue ! mpegtsmux name=mux alsasrc device="hw:0" ! audio/x-raw-int ! lamemp3enc ! queue ! mux. rtpmp2tpay pt=96 name=pay0 )"
** Message: listening on port 8554
** Message: added new client 0xb094360 ip 192.168.1.211:19915
** Message: attaching to context 0xb08e880
RTSP request message 0xb20fe58
 request line:
   method: 'OPTIONS'
   uri:    'rtsp://192.168.1.178:8554'
   version: '1.0'
 headers:
   key: 'CSeq', value: '1'
   key: 'User-Agent', value: 'RealMedia Player Version 6.0.9.1235 (linux-2.0-libc6-i386-gcc2.95)'
   key: 'ClientChallenge', value: '9e26d33f2984236010ef6253fb1887f7'
   key: 'PlayerStarttime', value: '[28/03/2003:22:50:23 00:00]'
   key: 'CompanyID', value: 'KnKV4M4I/B2FjJ1TToLycw=='
   key: 'GUID', value: '00000000-0000-0000-0000-000000000000'
   key: 'RegionData', value: '0'
   key: 'ClientID', value: 'Linux_2.4_6.0.9.1235_play32_RN01_EN_586'
 body:
** Message: client 0xb094360: received a request
RTSP response message 0x7fff800a74e0
 status line:
   code:   '200'
   reason: 'OK'
   version: '1.0'
 headers:
   key: 'CSeq', value: '1'
   key: 'Public', value: 'OPTIONS, DESCRIBE, GET_PARAMETER, PAUSE, PLAY, SETUP, SET_PARAMETER, TEARDOWN'
   key: 'Server', value: 'GStreamer RTSP server'
 body: length 0
RTSP request message 0xb20fe58
 request line:
   method: 'DESCRIBE'
   uri:    'rtsp://192.168.1.178:8554/test'
   version: '1.0'
 headers:
   key: 'CSeq', value: '2'
   key: 'Accept', value: 'application/sdp'
 body:
** Message: client 0xb094360: received a request
** Message: found media 0xb1e0300 for url abspath /test
** Message: found stream 0 with payloader 0xb2f60b0
** Message: constructed media 0xb309820 for url /test
** Message: preparing media 0xb309820
** Message: enter mainloop
** Message: live media 0xb309820
0:00:21.894683000  6321      0xb08b010 WARN                     bin gstbin.c:2312:gst_bin_do_latency_func:<media-pipeline> failed to query latency
** Message: 0xb309820: got message type new-clock
0:00:23.063460000  6321      0xb338ff0 WARN           basetransform gstbasetransform.c:1049:gst_base_transform_acceptcaps:<videoscale0> transform could not transform video/x-raw-yuv, format=(fourcc)YUY2, framerate=(fraction)100/1, width=(int)640, height=(int)480 in anything we support
0:00:23.116571000  6321 0x2aaaac000d70 WARN               mpegtsmux mpegtsmux.c:884:mpegtsdemux_prepare_srcpad:<mux> New segment event was not handled
0:00:23.173960000  6321      0xb338ff0 WARN                 basesrc gstbasesrc.c:2378:gst_base_src_loop:<v4l2src0> error: Internal data flow error.
0:00:23.174065000  6321      0xb338ff0 WARN                 basesrc gstbasesrc.c:2378:gst_base_src_loop:<v4l2src0> error: streaming task paused, reason error (-5)

** (lt-test-launch:6321): WARNING **: 0xb309820: got error Internal data flow error. (gstbasesrc.c(2378): gst_base_src_loop (): /GstPipeline:media-pipeline/GstBin:bin0/GstV4l2Src:v4l2src0:
streaming task paused, reason error (-5))

What am I doing wrong?

Thanks in advance ;-)!!!

Best Regards,

--
Rob Krakora
Senior Software Engineer
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
[phone] Ext. 206
[phone] Fax

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[G.B]

Any update on this? I am pretty much seeing the same thing that the mpegtsmux complains about "New seg event not handled" with gst-rtsp-server. But if I run the mpegtsmux pipeline with gst-launch and use filesink, the pipeline works well. Any ideas pls? Thanks