Using gstreamer-0.10 with Wowza
Hi
I've begun experimenting using Wowza together with the gstreamer media framework.
Here's the cmd line i'm using:
It results in creating a 320x240 video test pattern encoded using x264.
Since there's no RTCP enabled with such command line, i put the systemclock sync parameter in the Application.xml config file.
Server-side (on the local host), here's the myStream.sdp file:
I've begun experimenting using Wowza together with the gstreamer media framework.
Here's the cmd line i'm using:
gst-launch-0.10 -vvv videotestsrc ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! udpsink port=5000 host=127.0.0.1 sync=false
It results in creating a 320x240 video test pattern encoded using x264.
Since there's no RTCP enabled with such command line, i put the systemclock sync parameter in the Application.xml config file.
Server-side (on the local host), here's the myStream.sdp file:
v=0
o=- 1188340656180883 1 IN IP4 127.0.0.1
s=Session streamed by GStreamer
i=server.sh
t=0 0
a=tool:GStreamer
a=type:broadcast
m=video 5000 RTP/AVP 96
c=IN IP4 127.0.0.1
a=rtpmap:96 H264/90000
Here's the debug log (relevant parts):
DEBUG server comment - cmd: play
INFO server comment - MediaStreamMediaCasterPlay: startPlay
INFO server comment - RTPMediaCaster.create
INFO server comment - RTPMediaCaster.init
DEBUG server comment - cmd: setBufferTime
INFO server comment - RTPMediaCaster.Reconnector: start
INFO server comment - RTPSessionDescriptionDataProviderBasic.getStreamInfo: /usr/local/WowzaMediaServerPro/content/myStream.sdp
DEBUG server comment - sdp: v=0
DEBUG server comment - sdp: o=- 1188340656180883 1 IN IP4 127.0.0.1
DEBUG server comment - sdp: s=Session streamed by GStreamer
DEBUG server comment - sdp: i=server.sh
DEBUG server comment - sdp: t=0 0
DEBUG server comment - sdp: a=tool:GStreamer
DEBUG server comment - sdp: a=type:broadcast
DEBUG server comment - sdp: m=video 5000 RTP/AVP 96
DEBUG stream setbuffertime myStream.sdp -
DEBUG server comment - sdp: c=IN IP4 127.0.0.1
DEBUG server comment - sdp: a=rtpmap:96 H264/90000
DEBUG server comment - onFlushNotifyClients: false
DEBUG server comment - flushInterval: 0
DEBUG server comment - verboseDebug: false
INFO stream create - -
DEBUG server comment - RTPDePacketizerRFC3984H264.init
INFO stream publish myStream.sdp -
INFO server comment - UDPTransport.bind: /127.0.0.1:5000
DEBUG server comment - config: session: setReuseAddress: from:false to:true
DEBUG server comment - config: session: setReceiveBufferSize: from:55296 to:65000
DEBUG server comment - config: session: setSendBufferSize: from:55296 to:55296
DEBUG server comment - config: session: setTrafficClass: from:0 to:0
INFO server comment - UDPTransport.bind: /127.0.0.1:5001
DEBUG server comment - config: session: setReuseAddress: from:false to:true
DEBUG server comment - config: session: setReceiveBufferSize: from:55296 to:65000
DEBUG server comment - config: session: setSendBufferSize: from:55296 to:55296
DEBUG server comment - config: session: setTrafficClass: from:0 to:0
INFO server comment - RTPMediaCaster.Reconnector: stop
DEBUG server comment - rtp[video:1024] {80 60 11 af 25 e9 6f d6 e3 2f 49 73 1c 89 30 00 }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:1
INFO server comment - UDPTransport.firstPacket: /127.0.0.1:5000
DEBUG server comment - rtp[video:286] {80 60 11 b0 25 e9 6f d6 e3 2f 49 73 1c 49 c7 bf }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:2
DEBUG server comment - rtp[video:1024] {80 60 11 b1 25 e9 7b 8f e3 2f 49 73 1c 89 10 00 }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:3
DEBUG server comment - rtp[video:990] {80 60 11 b2 25 e9 7b 8f e3 2f 49 73 1c 49 de 0a }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:4
DEBUG server comment - rtp[video:1024] {80 60 11 b3 25 e9 87 46 e3 2f 49 73 1c 89 30 00 }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:5
DEBUG server comment - rtp[video:148] {80 60 11 b4 25 e9 87 46 e3 2f 49 73 1c 49 49 9f }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:6
DEBUG server comment - rtp[video:1024] {80 60 11 b5 25 e9 92 fe e3 2f 49 73 1c 89 30 00 }
DEBUG server comment - myStream.sdp: 17:36:15: waitforend: dropped:7
DEBUG server comment - rtp[video:177] {80 60 11 b6 25 e9 92 fe e3 2f 49 73 1c 49 7a 2f }
Of course this won't be an easy task, yet i'm wondering what this log mean ? Why are all frames dropped ?
Cheers
Florent
Hi guys,
Right now I have working gstreamer with Wowza successfully with H264 and AAC for flash clients and Iphone clients.
The following pipelines are working nice for me.
You can also run both together at the same time:
This is the sdp file that I'm using for this pipelines:
It's very important to notice that every different compilation of gstreamer has very different caps even for the same version. I mean, if you are using a different x264 library you will get different caps on your gstreamer pipeline outoput. This means that you need to update your sdp file with the caps from your gstreamer output. MAKE SURE YOU USE THE -v OPTION ON YOUR PIPELINES TO GET THESE CAPS.
""NOTE: I''VE REMOVED SOME OUTPUT BECAUSE THERE IS A TEXT LIMITATION.""
For example, the caps for the previous pipelines are (it's likely that the following caps and my previous sdp file doesn't match because it's from a different box):
The last (bolth) lines of each media type (audio and video) are the final caps that you need to include on the sdp file.
Finally notice that on relation to the video caps for x264 (concretely the sprop-parameter-sets) have on the command line caps different backslash that should be removed on the sdp file. For example from the previous caps:
sprop-parameter-sets=(string)\"Z0LAFrkQFAe2AiAAAAMAIAAAAwPR4sXU\\,aM4XIA\\=\\=\"
should end on the sdp file like:
Z0LAFrkQFAe2AiAAAAMAIAAAAwPR4sXU,aM4XIA==
I hope this would help someone.
Thank you.
Right now I have working gstreamer with Wowza successfully with H264 and AAC for flash clients and Iphone clients.
The following pipelines are working nice for me.
gst-launch-0.10 -v videotestsrc ! "video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480, framerate=(fraction)15/1" ! queue ! x264enc bitrate=500 cabac=false pass=qual quantizer=27 subme=4 threads=0 bframes=0 dct8x8=false ! queue ! rtph264pay ! queue ! udpsink host=127.0.0.1 port=50000 gst-launch-0.10 -v audiotestsrc ! "audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)2" ! queue ! tee name=audio audio. ! queue ! faac ! queue ! rtpmp4gpay ! queue ! udpsink host=127.0.0.1 port=50002
You can also run both together at the same time:
gst-launch-0.10 -v videotestsrc ! "video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480, framerate=(fraction)15/1" ! queue ! x264enc bitrate=500 cabac=false pass=qual quantizer=27 subme=4 threads=0 bframes=0 dct8x8=false ! queue ! rtph264pay ! queue ! udpsink host=127.0.0.1 port=50000 audiotestsrc ! "audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)2" ! queue ! tee name=audio audio. ! queue ! faac ! queue ! rtpmp4gpay ! queue ! udpsink host=127.0.0.1 port=50002
This is the sdp file that I'm using for this pipelines:
v=0 o=- 1208520720 2590316915 IN IP4 127.0.0.1 s=MY FIRST PIPELINE i=N/A c=IN IP4 127.0.0.1 t=0 0 m=video 50000 RTP/AVP 96 a=rtpmap:96 H264/90000 a=fmtp:96 media=video; clock-rate=90000; encoding-name=H264; profile-level-id=4d4015; sprop-parameter-sets=Z01AFbkQOhH3EYCIAAADAAgAAAMA9HixdQ==,aO48gA== a=control:trackID=1 m=audio 50002 RTP/AVP 96 a=rtpmap:96 MPEG4-GENERIC/48000/2 a=fmtp:96 media=audio; clock-rate=48000; encoding-name=MPEG4-GENERIC; encoding-params=2; streamtype=5; profile-level-id=1; mode=AAC-hbr; config=0990; sizeLength=13; indexLength=3; indexDeltaLength=3; payload=96 a=control:trackID=2
It's very important to notice that every different compilation of gstreamer has very different caps even for the same version. I mean, if you are using a different x264 library you will get different caps on your gstreamer pipeline outoput. This means that you need to update your sdp file with the caps from your gstreamer output. MAKE SURE YOU USE THE -v OPTION ON YOUR PIPELINES TO GET THESE CAPS.
""NOTE: I''VE REMOVED SOME OUTPUT BECAUSE THERE IS A TEXT LIMITATION.""
For example, the caps for the previous pipelines are (it's likely that the following caps and my previous sdp file doesn't match because it's from a different box):
Setting pipeline to PAUSED ... /GstPipeline:pipeline0/GstVideoTestSrc:videotestsrc0.GstPad:src: caps = video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480, framerate=(fraction)15/1 /GstPipeline:pipeline0/GstAudioTestSrc:audiotestsrc0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)2 Pipeline is PREROLLING ... .................. /GstPipeline:pipeline0/GstQueue:queue3.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)2 /GstPipeline:pipeline0/GstTee:audio.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)2 /GstPipeline:pipeline0/GstTee:audio.GstPad:src0: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)2 /GstPipeline:pipeline0/GstTee:audio.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)2 /GstPipeline:pipeline0/GstQueue:queue4.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)2 /GstPipeline:pipeline0/GstQueue:queue4.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)2 /GstPipeline:pipeline0/GstFaac:faac0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)2 /GstPipeline:pipeline0/GstFaac:faac0.GstPad:src: caps = audio/mpeg, mpegversion=(int)4, channels=(int)2, rate=(int)48000, codec_data=(buffer)0990 /GstPipeline:pipeline0/GstQueue:queue5.GstPad:sink: caps = audio/mpeg, mpegversion=(int)4, channels=(int)2, rate=(int)48000, codec_data=(buffer)0990 /GstPipeline:pipeline0/GstQueue:queue5.GstPad:src: caps = audio/mpeg, mpegversion=(int)4, channels=(int)2, rate=(int)48000, codec_data=(buffer)0990 /GstPipeline:pipeline0/GstRtpMP4GPay:rtpmp4gpay0.GstPad:src: caps = application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)2, streamtype=(string)5, profile-level-id=(string)1, mode=(string)AAC-hbr, config=(string)0990, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, payload=(int)96, ssrc=(guint)3072717366, clock-base=(guint)2502145401, seqnum-base=(guint)57789 /GstPipeline:pipeline0/GstRtpMP4GPay:rtpmp4gpay0.GstPad:sink: caps = audio/mpeg, mpegversion=(int)4, channels=(int)2, rate=(int)48000, codec_data=(buffer)0990 /GstPipeline:pipeline0/GstQueue:queue6.GstPad:sink: caps = application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)2, streamtype=(string)5, profile-level-id=(string)1, mode=(string)AAC-hbr, config=(string)0990, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, payload=(int)96, ssrc=(guint)3072717366, clock-base=(guint)2502145401, seqnum-base=(guint)57789 /GstPipeline:pipeline0/GstQueue:queue6.GstPad:src: caps = application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)2, streamtype=(string)5, profile-level-id=(string)1, mode=(string)AAC-hbr, config=(string)0990, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, payload=(int)96, ssrc=(guint)3072717366, clock-base=(guint)2502145401, seqnum-base=(guint)57789 [B]/GstPipeline:pipeline0/GstUDPSink:udpsink1.GstPad:sink: caps = application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)2, streamtype=(string)5, profile-level-id=(string)1, mode=(string)AAC-hbr, config=(string)0990, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, payload=(int)96, ssrc=(guint)3072717366, clock-base=(guint)2502145401, seqnum-base=(guint)57789[/B] /GstPipeline:pipeline0/GstQueue:queue1.GstPad:sink: caps = video/x-h264, width=(int)640, height=(int)480, framerate=(fraction)15/1, codec_data=(buffer)0142c016ffe100186742c016b9101407b60220000003002000000303d1e2c5d401000468ce1720 /GstPipeline:pipeline0/GstQueue:queue1.GstPad:src: caps = video/x-h264, width=(int)640, height=(int)480, framerate=(fraction)15/1, codec_data=(buffer)0142c016ffe100186742c016b9101407b60220000003002000000303d1e2c5d401000468ce1720 /GstPipeline:pipeline0/GstRtpH264Pay:rtph264pay0.GstPad:src: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, profile-level-id=(string)42c016, sprop-parameter-sets=(string)\"Z0LAFrkQFAe2AiAAAAMAIAAAAwPR4sXU\\,aM4XIA\\=\\=\", payload=(int)96, ssrc=(guint)1090311487, clock-base=(guint)3529879298, seqnum-base=(guint)18909 /GstPipeline:pipeline0/GstRtpH264Pay:rtph264pay0.GstPad:sink: caps = video/x-h264, width=(int)640, height=(int)480, framerate=(fraction)15/1, codec_data=(buffer)0142c016ffe100186742c016b9101407b60220000003002000000303d1e2c5d401000468ce1720 /GstPipeline:pipeline0/GstQueue:queue2.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, profile-level-id=(string)42c016, sprop-parameter-sets=(string)\"Z0LAFrkQFAe2AiAAAAMAIAAAAwPR4sXU\\,aM4XIA\\=\\=\", payload=(int)96, ssrc=(guint)1090311487, clock-base=(guint)3529879298, seqnum-base=(guint)18909 /GstPipeline:pipeline0/GstQueue:queue2.GstPad:src: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, profile-level-id=(string)42c016, sprop-parameter-sets=(string)\"Z0LAFrkQFAe2AiAAAAMAIAAAAwPR4sXU\\,aM4XIA\\=\\=\", payload=(int)96, ssrc=(guint)1090311487, clock-base=(guint)3529879298, seqnum-base=(guint)18909 [B]/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, profile-level-id=(string)42c016, sprop-parameter-sets=(string)\"Z0LAFrkQFAe2AiAAAAMAIAAAAwPR4sXU\\,aM4XIA\\=\\=\", payload=(int)96, ssrc=(guint)1090311487, clock-base=(guint)3529879298, seqnum-base=(guint)18909 [/B]Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock
The last (bolth) lines of each media type (audio and video) are the final caps that you need to include on the sdp file.
Finally notice that on relation to the video caps for x264 (concretely the sprop-parameter-sets) have on the command line caps different backslash that should be removed on the sdp file. For example from the previous caps:
sprop-parameter-sets=(string)\"Z0LAFrkQFAe2AiAAAAMAIAAAAwPR4sXU\\,aM4XIA\\=\\=\"
should end on the sdp file like:
Z0LAFrkQFAe2AiAAAAMAIAAAAwPR4sXU,aM4XIA==
I hope this would help someone.
Thank you.
Hello,
long time since last post but I hope someone can help. My problem is to stream video from Gstreamer to Wowza, where Gstreamer is running on ARM architecture on Pandaboard computer with Ubuntu 11.10. My actual achievement is RTP streaming from Gstreamer to VLC then from VLC to Wowza and it works so I can see the video in flash sample application.
I've tried cbao.rambo's solution but the only I can have is to see video in VLC (when I run it with SDP file), when I'm trying stream to Wowza directly, it doesn't see anything. These are my Gstreamer output and SDP file:
I used the same parameters as cbao.rambo:
Gstreamer output:
SDP:
My connections looks like that:
Gstreamer [Pandaboard] -> VLC [PC] -> Wowza [PC same as VLC] - WORKS!
Gstreamer [Pandaboard] -> Wowza [PC] - does not work
I removed slashes '\' in sprop-parameter-sets. Either they were or not it didn't make any difference. I don't no what can be wrong. I have the newest version of Gstreamer on my Pandaboard.
Any ideas?
Thanks!
long time since last post but I hope someone can help. My problem is to stream video from Gstreamer to Wowza, where Gstreamer is running on ARM architecture on Pandaboard computer with Ubuntu 11.10. My actual achievement is RTP streaming from Gstreamer to VLC then from VLC to Wowza and it works so I can see the video in flash sample application.
I've tried cbao.rambo's solution but the only I can have is to see video in VLC (when I run it with SDP file), when I'm trying stream to Wowza directly, it doesn't see anything. These are my Gstreamer output and SDP file:
I used the same parameters as cbao.rambo:
gst-launch-0.10 -v videotestsrc ! "video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480, framerate=(fraction)15/1" ! queue ! x264enc bitrate=500 cabac=false pass=qual quantizer=27 subme=4 threads=0 bframes=0 dct8x8=false ! queue ! rtph264pay ! queue ! udpsink host=192.168.3.206 port=5000
Gstreamer output:
Setting pipeline to PAUSED ... /GstPipeline:pipeline0/GstVideoTestSrc:videotestsrc0.GstPad:src: caps = video/x-raw-yuv, format=(fourcc)I420, framerate=(fraction)15/1, width=(int)640, height=(int)480, color-matrix=(string)sdtv, chroma-site=(string)mpeg2 Pipeline is PREROLLING ... /GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = video/x-raw-yuv, format=(fourcc)I420, framerate=(fraction)15/1, width=(int)640, height=(int)480, color-matrix=(string)sdtv, chroma-site=(string)mpeg2 /GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps = video/x-raw-yuv, format=(fourcc)I420, framerate=(fraction)15/1, width=(int)640, height=(int)480, color-matrix=(string)sdtv, chroma-site=(string)mpeg2 /GstPipeline:pipeline0/GstQueue:queue0.GstPad:sink: caps = video/x-raw-yuv, format=(fourcc)I420, framerate=(fraction)15/1, width=(int)640, height=(int)480, color-matrix=(string)sdtv, chroma-site=(string)mpeg2 /GstPipeline:pipeline0/GstQueue:queue0.GstPad:src: caps = video/x-raw-yuv, format=(fourcc)I420, framerate=(fraction)15/1, width=(int)640, height=(int)480, color-matrix=(string)sdtv, chroma-site=(string)mpeg2 /GstPipeline:pipeline0/GstX264Enc:x264enc0.GstPad:src: caps = video/x-h264, width=(int)640, height=(int)480, framerate=(fraction)15/1, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)014d4016ffe10017674d4016d900a03db0110000030001773594000f162e4801000468cbc5c8, stream-format=(string)avc, alignment=(string)au /GstPipeline:pipeline0/GstX264Enc:x264enc0.GstPad:sink: caps = video/x-raw-yuv, format=(fourcc)I420, framerate=(fraction)15/1, width=(int)640, height=(int)480, color-matrix=(string)sdtv, chroma-site=(string)mpeg2 /GstPipeline:pipeline0/GstQueue:queue1.GstPad:sink: caps = video/x-h264, width=(int)640, height=(int)480, framerate=(fraction)15/1, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)014d4016ffe10017674d4016d900a03db0110000030001773594000f162e4801000468cbc5c8, stream-format=(string)avc, alignment=(string)au /GstPipeline:pipeline0/GstQueue:queue1.GstPad:src: caps = video/x-h264, width=(int)640, height=(int)480, framerate=(fraction)15/1, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)014d4016ffe10017674d4016d900a03db0110000030001773594000f162e4801000468cbc5c8, stream-format=(string)avc, alignment=(string)au /GstPipeline:pipeline0/GstRtpH264Pay:rtph264pay0.GstPad:src: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, sprop-parameter-sets=(string)\"Z01AFtkAoD2wEQAAAwABdzWUAA8WLkg\\=\\,aMvFyA\\=\\=\", payload=(int)96, ssrc=(uint)3975972560, clock-base=(uint)4006607711, seqnum-base=(uint)19939 /GstPipeline:pipeline0/GstRtpH264Pay:rtph264pay0.GstPad:sink: caps = video/x-h264, width=(int)640, height=(int)480, framerate=(fraction)15/1, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)014d4016ffe10017674d4016d900a03db0110000030001773594000f162e4801000468cbc5c8, stream-format=(string)avc, alignment=(string)au /GstPipeline:pipeline0/GstRtpH264Pay:rtph264pay0: timestamp = 4006607711 /GstPipeline:pipeline0/GstRtpH264Pay:rtph264pay0: seqnum = 19939 /GstPipeline:pipeline0/GstQueue:queue2.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, sprop-parameter-sets=(string)\"Z01AFtkAoD2wEQAAAwABdzWUAA8WLkg\\=\\,aMvFyA\\=\\=\", payload=(int)96, ssrc=(uint)3975972560, clock-base=(uint)4006607711, seqnum-base=(uint)19939 /GstPipeline:pipeline0/GstQueue:queue2.GstPad:src: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, sprop-parameter-sets=(string)\"Z01AFtkAoD2wEQAAAwABdzWUAA8WLkg\\=\\,aMvFyA\\=\\=\", payload=(int)96, ssrc=(uint)3975972560, clock-base=(uint)4006607711, seqnum-base=(uint)19939 /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, sprop-parameter-sets=(string)\"Z01AFtkAoD2wEQAAAwABdzWUAA8WLkg\\=\\,aMvFyA\\=\\=\", payload=(int)96, ssrc=(uint)3975972560, clock-base=(uint)4006607711, seqnum-base=(uint)19939 Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock
SDP:
v=0 o=- 14818005123115786627 14818005123115786627 IN IP4 192.168.3.206 s=Pipeline1 i=N/A c=IN IP4 192.168.3.206 t=0 0 m=video 5000 RTP/AVP 96 a=rtpmap:96 H264/90000 a=fmtp:96 media=video; clock-rate=90000; encoding-name=H264; sprop-parameter-sets=Z01AFtkAoD2wEQAAAwABdzWUAA8WLkg=,aMvFyA==; payload=96; ssrc=3975972560; clock-base=4006607711; seqnum-base=19939; a=control:trackID=1
My connections looks like that:
Gstreamer [Pandaboard] -> VLC [PC] -> Wowza [PC same as VLC] - WORKS!
Gstreamer [Pandaboard] -> Wowza [PC] - does not work
I removed slashes '\' in sprop-parameter-sets. Either they were or not it didn't make any difference. I don't no what can be wrong. I have the newest version of Gstreamer on my Pandaboard.
Any ideas?
Thanks!
ALARM CALLED OFF! Everything is working!
silly me.. it was just Windows Firewall blocking connections to Wowza from Pandaboard.. I was confused because it didn't block VLC..
silly me.. it was just Windows Firewall blocking connections to Wowza from Pandaboard.. I was confused because it didn't block VLC..
You might want to explain what your gst-launch command line is !
I have personally not had much luck with gstreamer. Other then what I have suggested I do not know any magic bullets.
Charlie
Charlie
These type of issues are usually due to av sync issues. Take a look at this post to see if any of the suggestions here help:
http://www.wowza.com/forums/showthread.php?t=5355
Charlie
http://www.wowza.com/forums/showthread.php?t=5355
Charlie
Hi Charlie,
I've been trying the different options of that post but without success.
Any other ideas?
It's really important for us get this working with gstreamer.
Thank you very much in advance.
I've been trying the different options of that post but without success.
Any other ideas?
It's really important for us get this working with gstreamer.
Thank you very much in advance.
Hi,
i have a similar problem using gstreamer to send an rtp stream to wowza.
The player starts playing for 1s then it stops for about 40s then starts playing again.
When i make two sdp files one for audio and one for video i can play them, using the same gstream-cmd and everything is fine. But i would like to be able to use just one flash-player :)
It would be nice if we would have an example of an gstreamer-cmd working with wowza, in the forum i only find stuff not working.
gst-cmd:
gst-log:
sdp-file:
other information:
GStreamer Core Library version 0.10.22
Wowza Media Server Pro10 1.7.2 build12107
Played with setting in post
http://www.wowza.com/forums/showthread.php?t=5355
<AVSyncMethod>systemclock</AVSyncMethod>
<MaxRTCPWaitTime>12000</MaxRTCPWaitTime>
<Name>sortPackets</Name>
<Name>sortBufferSize</Name>
any idea?
i have a similar problem using gstreamer to send an rtp stream to wowza.
The player starts playing for 1s then it stops for about 40s then starts playing again.
When i make two sdp files one for audio and one for video i can play them, using the same gstream-cmd and everything is fine. But i would like to be able to use just one flash-player :)
It would be nice if we would have an example of an gstreamer-cmd working with wowza, in the forum i only find stuff not working.
gst-cmd:
gst-launch-0.10 -vvv videotestsrc ! capsfilter caps="video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480, framerate=(fraction)30/1" ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! udpsink port=6000 host=127.0.0.1 sync=false audiotestsrc ! "audio/x-raw-int, rate=(int)48000" ! faac ! rtpmp4gpay ! udpsink host=127.0.0.1 port=6002
gst-log:
/GstPipeline:pipeline0/GstRtpH264Pay:rtph264pay0.GstPad:src: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, profile-level-id=(string)4d401e, sprop-parameter-sets=(string)Z01AHpJUBQHtgIgAAAMACAAAAwHkeLF1, payload=(int)96, ssrc=(guint)2198359392, clock-base=(guint)614858601, seqnum-base=(guint)21776 /GstPipeline:pipeline0/GstUDPSink:udpsink1.GstPad:sink: caps = application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)1, streamtype=(string)5, profile-level-id=(string)1, mode=(string)AAC-hbr, config=(string)0988, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, payload=(int)96, ssrc=(guint)2805765163, clock-base=(guint)4260627807, seqnum-base=(guint)57770
sdp-file:
v=0 o=- 116704633 116704633 IN IP4 127.0.0.1 s=GStreamer t=0 0 c=IN IP4 127.0.0.1 m=video 6000 RTP/AVP 96 a=rtpmap:96 H264/90000 a=control:trackID=1 m=audio 6002 RTP/AVP 96 a=rtpmap:96 MPEG4-GENERIC/48000/2 a=fmtp:96 media=audio; clock-rate=48000; encoding-name=MPEG4-GENERIC; encoding-params=1; streamtype=5; profile-level-id=1; mode=AAC-hbr; config=0b88; sizeLength=13; indexLength=3; indexDeltaLength=3; payload=96; ssrc=2805765163; clock-base=4260627807; seqnum-base=57770 a=control:trackID=2
other information:
GStreamer Core Library version 0.10.22
Wowza Media Server Pro10 1.7.2 build12107
Played with setting in post
http://www.wowza.com/forums/showthread.php?t=5355
<AVSyncMethod>systemclock</AVSyncMethod>
<MaxRTCPWaitTime>12000</MaxRTCPWaitTime>
<Name>sortPackets</Name>
<Name>sortBufferSize</Name>
any idea?
Great!
Charlie
Charlie
Hi guys,
I'm trying to stream a DVD from my Play Station 2 to my TV capture card using GStreamer until Wowza and from there to my Flowplayer client.
I have it working with audio or video but not with both at the same time. My problem when I'm trying both is that the video get delayed and plays very slow. Like on slow motion.
I'm using the last version of GStreamer 0.10.24 (compiled by myselft) and Wowza 2 preview 6 (which I think is the latest version).
This is my SDP file:
v=0
o=- 1208520720 2590316915 IN IP4 83.244.217.196
s=Gstreamer
i=N/A
c=IN IP4 83.234.38.8
t=0 0
m=video 10004 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 media=video; clock-rate=90000; encoding-name=H264; profile-level-id=4d401e; sprop-parameter-sets=Z01AHpJUBQHtgIgAAAMACAAAAwGUeLF1,aO48gA==; payload=96; ssrc=2977118887; clock-base=179044607; seqnum-bas\
e=51513;
a=control:trackID=1
m=audio 10002 RTP/AVP 96
a=rtpmap:96 MPEG4-GENERIC/48000/2
a=fmtp:96 media=audio; clock-rate=16000; encoding-name=MPEG4-GENERIC; encoding-params=1; streamtype=5; profile-level-id=1; mode=AAC-hbr; config=0c08; objectType=64; sizeLength=13; indexLength=3; indexDeltaLength\
=3; payload=96; ssrc=2977118887; clock-base=179044607; seqnum-base=51513;
a=control:trackID=2
And this is my Gstreamer pipeline:
gst-launch-0.10 -v v4l2src device=/dev/video0 ! "video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480, framerate=(fraction)25/1" ! queue ! ffdeinterlace ! queue ! x264enc bitrate=200 byte-stream=true ! queue ! rtph264pay ! queue ! udpsink host=89.234.38.8 port=10004 alsasrc ! "audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)16000, channels=(int)1" ! queue ! volume volume=10 ! queue ! faac ! queue ! rtpmp4gpay ! queue ! udpsink host=89.234.38.8 port=10002
Btw, I've started the stream with the DEBUG level on my wowza server and I don't see any errors at all.
Any help or thoughts?
PD: I forgot to mention that I have the RTP/AVSyncMethod attribute to systemclock
Thank you very much.
I'm trying to stream a DVD from my Play Station 2 to my TV capture card using GStreamer until Wowza and from there to my Flowplayer client.
I have it working with audio or video but not with both at the same time. My problem when I'm trying both is that the video get delayed and plays very slow. Like on slow motion.
I'm using the last version of GStreamer 0.10.24 (compiled by myselft) and Wowza 2 preview 6 (which I think is the latest version).
This is my SDP file:
v=0
o=- 1208520720 2590316915 IN IP4 83.244.217.196
s=Gstreamer
i=N/A
c=IN IP4 83.234.38.8
t=0 0
m=video 10004 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 media=video; clock-rate=90000; encoding-name=H264; profile-level-id=4d401e; sprop-parameter-sets=Z01AHpJUBQHtgIgAAAMACAAAAwGUeLF1,aO48gA==; payload=96; ssrc=2977118887; clock-base=179044607; seqnum-bas\
e=51513;
a=control:trackID=1
m=audio 10002 RTP/AVP 96
a=rtpmap:96 MPEG4-GENERIC/48000/2
a=fmtp:96 media=audio; clock-rate=16000; encoding-name=MPEG4-GENERIC; encoding-params=1; streamtype=5; profile-level-id=1; mode=AAC-hbr; config=0c08; objectType=64; sizeLength=13; indexLength=3; indexDeltaLength\
=3; payload=96; ssrc=2977118887; clock-base=179044607; seqnum-base=51513;
a=control:trackID=2
And this is my Gstreamer pipeline:
gst-launch-0.10 -v v4l2src device=/dev/video0 ! "video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480, framerate=(fraction)25/1" ! queue ! ffdeinterlace ! queue ! x264enc bitrate=200 byte-stream=true ! queue ! rtph264pay ! queue ! udpsink host=89.234.38.8 port=10004 alsasrc ! "audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)16000, channels=(int)1" ! queue ! volume volume=10 ! queue ! faac ! queue ! rtpmp4gpay ! queue ! udpsink host=89.234.38.8 port=10002
Btw, I've started the stream with the DEBUG level on my wowza server and I don't see any errors at all.
Any help or thoughts?
PD: I forgot to mention that I have the RTP/AVSyncMethod attribute to systemclock
Thank you very much.
Hi guys,
I think I found my problem. Just in case someone else need this info.
The only problem was on the sdp file. I had the following line:
a=rtpmap:96 MPEG4-GENERIC/48000/2
However, If you have a look to my pipelines I was using an audio stream of 16000 Hz.
I adjusted the stream and the sdp files for a 48000 Hz audio stream and everything is working perfectly now.
I guess that changing that line to 16000 Hz instead of 48000 Hz should work as well.
Thx.
I think I found my problem. Just in case someone else need this info.
The only problem was on the sdp file. I had the following line:
a=rtpmap:96 MPEG4-GENERIC/48000/2
However, If you have a look to my pipelines I was using an audio stream of 16000 Hz.
I adjusted the stream and the sdp files for a 48000 Hz audio stream and everything is working perfectly now.
I guess that changing that line to 16000 Hz instead of 48000 Hz should work as well.
Thx.
Looks quite ok to me. Are you sure you are waiting long enough ? What do the wowza debug messages tell ?
Also, try setting the RTP/AVSyncMethod to systemclock in [install-dir]/conf/[application]/Application.xml.
Charlie
Charlie
Setting the RTP/AVSyncMethod attribute to systemclock made it!
Thank you Charlie and Florent.
Thank you Charlie and Florent.
1) why do i need to restart the server after stopping and restarting the (publisher) streaming process ? Is there a workaround ? If i do not, the flash player only shows a black screen
It is because when the encoder starts/stop it changes the time basis which screws up Wowza Pro and Flash. There are two methods to work around this. Take a look at the sdpFileCheckFreqency and streamTimeout properties as discussed in the Native Real-time Transport Protocol (RTP) Streaming section of the User's Guide. You can setup Wowza Pro to monitor the SDP file for date or size changes ( sdpFileCheckFreqency) so the stream is restarted when the SDP file changes. You can instruct Wowza Pro monitor the stream for long breaks ( streamTimeout) and restart the stream if needed. The streamTimeout is tricky to get right since if it is too short it will restart the stream when it first tries to connect if it can't find a key frame quickly enough.
2) audio plays back faster
3) my audio tests samples (voice/claps) seem to have a direct effect on drifting speed ! If i speak longer, the desync comes faster !
This is a tough one. I just don't have the resources to dig into gstreamer and debug something like this. I know there are plenty of RTP based encoders where there is no issue with AV sync. So it is most likely not a Wowza Pro issue. There may be rounding issues in GStreamer. I have seen this problem in other encoders where they don't deal with timecodes properly and they calculate time differentials which insert small rounding errors that add up over time.
Charlie
I've found a gstreamer's pipeline and a SDP that work with Wowza, sending live video and audio to Wowza.
However, I'm only able to send audio or video separately, but not together.
Here is my last SDP file (caps are copied form gstreamer output).
If i remove video or audio from SDP, the other (video or audio, depending) is played alright:
v=0
o=- 14818005123115786627 14818005123115786627 IN IP4 localhost.localdomain
s=Gstreamer
i=N/A
c=IN IP4 237.1.1.3
t=0 0
m=video 6070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 media=video; clock-rate=90000; encoding-name=H264; profile-level-id=4d4015; sprop-parameter-sets=Z01AFZJUDwX3/gAwACIgAAADACAAAAZR4sXU; payload=96; ssrc=3120057827; clock-base=1065471660; seqnum-base=39429;
a=control:trackID=1
m=audio 6060 RTP/AVP 96
a=rtpmap:96 MPEG4-GENERIC/48000/2
a=fmtp:96 media=audio; clock-rate=44100; encoding-name=MPEG4-GENERIC; encoding-params=1; streamtype=5; profile-level-id=1; mode=AAC-hbr; config=0a08; objectType=64; sizeLength=13; indexLength=3; indexDeltaLength=3; payload=96; ssrc=4278936313; clock-base=160119643; seqnum-base=26806
a=control:trackID=2
However, I'm only able to send audio or video separately, but not together.
Here is my last SDP file (caps are copied form gstreamer output).
If i remove video or audio from SDP, the other (video or audio, depending) is played alright:
v=0
o=- 14818005123115786627 14818005123115786627 IN IP4 localhost.localdomain
s=Gstreamer
i=N/A
c=IN IP4 237.1.1.3
t=0 0
m=video 6070 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 media=video; clock-rate=90000; encoding-name=H264; profile-level-id=4d4015; sprop-parameter-sets=Z01AFZJUDwX3/gAwACIgAAADACAAAAZR4sXU; payload=96; ssrc=3120057827; clock-base=1065471660; seqnum-base=39429;
a=control:trackID=1
m=audio 6060 RTP/AVP 96
a=rtpmap:96 MPEG4-GENERIC/48000/2
a=fmtp:96 media=audio; clock-rate=44100; encoding-name=MPEG4-GENERIC; encoding-params=1; streamtype=5; profile-level-id=1; mode=AAC-hbr; config=0a08; objectType=64; sizeLength=13; indexLength=3; indexDeltaLength=3; payload=96; ssrc=4278936313; clock-base=160119643; seqnum-base=26806
a=control:trackID=2
How long does it take for it to go out of sync? RTCP is used to synchronize the video and audio but is only needed to initially sync the stream. MPEG-TS synchronizes the audio and video in the stream (all streams run on a 90KHz clock and are pre-synchronized). Not sure why your stream is going out of sync.
Charlie
Charlie
Well, it's hard to say: it's perceivable after 20 minutes or so.
1) why do i need to restart the server after stopping and restarting the (publisher) streaming process ? Is there a workaround ? If i do not, the flash player only shows a black screen
2) audio plays back faster
3) my audio tests samples (voice/claps) seem to have a direct effect on drifting speed ! If i speak longer, the desync comes faster ! :mad:
Streamer: gstreamer-0.10.21 (cvs)
Wowza: 1.5.3 build7825
Client: Flash Player 10 (Vista)
1) why do i need to restart the server after stopping and restarting the (publisher) streaming process ? Is there a workaround ? If i do not, the flash player only shows a black screen
2) audio plays back faster
3) my audio tests samples (voice/claps) seem to have a direct effect on drifting speed ! If i speak longer, the desync comes faster ! :mad:
Streamer: gstreamer-0.10.21 (cvs)
Wowza: 1.5.3 build7825
Client: Flash Player 10 (Vista)
Upgrade the most recent patch:
http://www.wowza.com/devbuild.html
I believe it only works with the most recent version of GStreamer.
Charlie
http://www.wowza.com/devbuild.html
I believe it only works with the most recent version of GStreamer.
Charlie
Hi Charlie,
I am now looking for a way to send MP3 or AAC data with gstreamer as well:
AAC:
gst-launch audiotestsrc ! faac ! rtpmp4apay ! udpsink host=$server port=$audio_port
Here is the sdp audio part i've been using
m=audio 10002 RTP/AVP 96
a=rtpmap:96 mpeg4-generic/48000/2
a=control:trackID=2
a=fmtp:96 streamtype=5; profile-level-id=255; mode=AAC-hbr; config=1190; objectType=64; sizeLength=13; indexLength=3; indexDeltaLength=3
The server raises exceptions:
ERROR server comment - addDataA[this.size:513 this.dataA.length:513 this.startDataLoc:512 this.dataLoc:512 data.length:303 offset:0 size:303 missing:1 ]: java.lang.ArrayIndexOutOfBoundsException
java.lang.ArrayIndexOutOfBoundsException
at com.wowza.wms.amf.AMFPacket.addData(Unknown Source)
at com.wowza.wms.stream.live.LiveReceiver.addAudioData(Unknown Source)
at com.wowza.wms.stream.live.MediaStreamLive.addAudioData(Unknown Source)
at com.wowza.wms.rtp.depacketizer.RTPPacket.write(Unknown Source)
at com.wowza.wms.rtp.depacketizer.RTPPacket.write(Unknown Source)
at com.wowza.wms.rtp.depacketizer.RTPDePacketizerMPEG4AAC.handleRTPPacket(Unknown Source)
Given gstreamer's detailed output (below) how should the SDP file be ?
caps = application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)MP4A-LATM, cpresent=(string)0, config=(string)40001320, payload=(int)96, ssrc=(guint)3653097729, clock-base=(guint)2206318833, seqnum-base=(guint)48474
MP3:
gst-launch audiotestsrc ! lame ! rtpmpapay ! udpsink host=$server port=$audio_port
I do not know how to tell the server in the sdp file (if supported) that audio is MP3. Do you have sample SDP files for MP3 rtp streams ?
THanks
I am now looking for a way to send MP3 or AAC data with gstreamer as well:
AAC:
gst-launch audiotestsrc ! faac ! rtpmp4apay ! udpsink host=$server port=$audio_port
Here is the sdp audio part i've been using
m=audio 10002 RTP/AVP 96
a=rtpmap:96 mpeg4-generic/48000/2
a=control:trackID=2
a=fmtp:96 streamtype=5; profile-level-id=255; mode=AAC-hbr; config=1190; objectType=64; sizeLength=13; indexLength=3; indexDeltaLength=3
The server raises exceptions:
ERROR server comment - addDataA[this.size:513 this.dataA.length:513 this.startDataLoc:512 this.dataLoc:512 data.length:303 offset:0 size:303 missing:1 ]: java.lang.ArrayIndexOutOfBoundsException
java.lang.ArrayIndexOutOfBoundsException
at com.wowza.wms.amf.AMFPacket.addData(Unknown Source)
at com.wowza.wms.stream.live.LiveReceiver.addAudioData(Unknown Source)
at com.wowza.wms.stream.live.MediaStreamLive.addAudioData(Unknown Source)
at com.wowza.wms.rtp.depacketizer.RTPPacket.write(Unknown Source)
at com.wowza.wms.rtp.depacketizer.RTPPacket.write(Unknown Source)
at com.wowza.wms.rtp.depacketizer.RTPDePacketizerMPEG4AAC.handleRTPPacket(Unknown Source)
Given gstreamer's detailed output (below) how should the SDP file be ?
caps = application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)MP4A-LATM, cpresent=(string)0, config=(string)40001320, payload=(int)96, ssrc=(guint)3653097729, clock-base=(guint)2206318833, seqnum-base=(guint)48474
MP3:
gst-launch audiotestsrc ! lame ! rtpmpapay ! udpsink host=$server port=$audio_port
I do not know how to tell the server in the sdp file (if supported) that audio is MP3. Do you have sample SDP files for MP3 rtp streams ?
THanks
Here's what i'm trying:
m=audio 10002 RTP/AVP 96
a=rtpmap:96 mpa-robust/90000
And the corresponding gst caps:
/GstPipeline:pipeline0/GstRtpMPAPay:rtpmpapay0.GstPad:src: caps = application/x-rtp, media=(string)audio, clock-rate=(int)90000, encoding-name=(string)MPA, payload=(int)96, ssrc=(guint)1315959326, clock-base=(guint)2051349402, seqnum-base=(guint)41023
/GstPipeline:pipeline0/GstRtpMPAPay:rtpmpapay0.GstPad:sink: caps = audio/mpeg, mpegversion=(int)1, mpegaudioversion=(int)1, layer=(int)3, channels=(int)2, rate=(int)48000
m=audio 10002 RTP/AVP 96
a=rtpmap:96 mpa-robust/90000
And the corresponding gst caps:
/GstPipeline:pipeline0/GstRtpMPAPay:rtpmpapay0.GstPad:src: caps = application/x-rtp, media=(string)audio, clock-rate=(int)90000, encoding-name=(string)MPA, payload=(int)96, ssrc=(guint)1315959326, clock-base=(guint)2051349402, seqnum-base=(guint)41023
/GstPipeline:pipeline0/GstRtpMPAPay:rtpmpapay0.GstPad:sink: caps = audio/mpeg, mpegversion=(int)1, mpegaudioversion=(int)1, layer=(int)3, channels=(int)2, rate=(int)48000
From
http://www.rfc-editor.org/rfc/rfc3016.txt:
5.4 SDP usage of MPEG-4 Audio
The MIME media type audio/MP4A-LATM string is mapped to fields in the
Session Description Protocol (SDP), RFC 2327, as follows:
o The MIME type (audio) goes in SDP "m=" as the media name.
o The MIME subtype (MP4A-LATM) goes in SDP "a=rtpmap" as the
encoding name.
o The required parameter "rate" goes in "a=rtpmap" as the clock
rate.
o The optional parameter "ptime" goes in SDP "a=ptime" attribute.
o The optional parameter "profile-level-id" goes in the "a=fmtp"
line to indicate the coder capability. The "object" parameter
goes in the "a=fmtp" attribute. The payload-format-specific
parameters
Here's what i am trying for AAC:
m=audio 10002 RTP/AVP 96
a=rtpmap:96 MP4A-LATM/48000/2
Where gst caps are:
caps = application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)MP4A-LATM, cpresent=(string)0, config=(string)40001320, payload=(int)96, ssrc=(guint)3006373333, clock-base=(guint)1009694188, seqnum-base=(guint)27071
Problem is, if i do this the video doesn't play back anymore ! (Black screen). No errors though...
5.4 SDP usage of MPEG-4 Audio
The MIME media type audio/MP4A-LATM string is mapped to fields in the
Session Description Protocol (SDP), RFC 2327, as follows:
o The MIME type (audio) goes in SDP "m=" as the media name.
o The MIME subtype (MP4A-LATM) goes in SDP "a=rtpmap" as the
encoding name.
o The required parameter "rate" goes in "a=rtpmap" as the clock
rate.
o The optional parameter "ptime" goes in SDP "a=ptime" attribute.
o The optional parameter "profile-level-id" goes in the "a=fmtp"
line to indicate the coder capability. The "object" parameter
goes in the "a=fmtp" attribute. The payload-format-specific
parameters
Here's what i am trying for AAC:
m=audio 10002 RTP/AVP 96
a=rtpmap:96 MP4A-LATM/48000/2
Where gst caps are:
caps = application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)MP4A-LATM, cpresent=(string)0, config=(string)40001320, payload=(int)96, ssrc=(guint)3006373333, clock-base=(guint)1009694188, seqnum-base=(guint)27071
Problem is, if i do this the video doesn't play back anymore ! (Black screen). No errors though...
Here is my latest sdp audio section test:
m=audio 10002 RTP/AVP 96
a=rtpmap:96 MP4-LATM/48000/2
a=fmtp:96 profile-level-id=1; bitrate=128000; cpresent=0;config=40001320
Here are the default faac parameters:
outputformat : Format of output frames
flags: accès en lecture, accès en écriture
Enum "GstFaacOutputFormat" Default: 1, "ADTS headers" Current: 0, "Raw AAC"
(0): Raw AAC - OUTPUTFORMAT_RAW
(1): ADTS headers - OUTPUTFORMAT_ADTS
bitrate : Bitrate in bits/sec
flags: accès en lecture, accès en écriture
Integer. Range: 8000 - 320000 Default: 128000 Current: 128000
profile : MPEG/AAC encoding profile
flags: accès en lecture, accès en écriture
Enum "GstFaacProfile" Default: 1, "Main profile" Current: 1, "Main profile"
(1): Main profile - MAIN
(2): Low complexity profile - LC
(3): Scalable sampling rate profile - SSR
(4): Long term prediction profile - LTP
tns : Use temporal noise shaping
flags: accès en lecture, accès en écriture
Boolean. Default: false Current: false
midside : Allow mid/side encoding
flags: accès en lecture, accès en écriture
Boolean. Default: true Current: true
shortctl : Block type encorcing
flags: accès en lecture, accès en écriture
Enum "GstFaacShortCtl" Default: 1, "No short blocks" Current: 0, "Normal block type"
(0): Normal block type - SHORTCTL_NORMAL
(1): No short blocks - SHORTCTL_NOSHORT
(2): No long blocks - SHORTCTL_NOLONG
It seems to me everything should be alright here (according to RFC). Does anyone see any obvious mistake ?
m=audio 10002 RTP/AVP 96
a=rtpmap:96 MP4-LATM/48000/2
a=fmtp:96 profile-level-id=1; bitrate=128000; cpresent=0;config=40001320
Here are the default faac parameters:
outputformat : Format of output frames
flags: accès en lecture, accès en écriture
Enum "GstFaacOutputFormat" Default: 1, "ADTS headers" Current: 0, "Raw AAC"
(0): Raw AAC - OUTPUTFORMAT_RAW
(1): ADTS headers - OUTPUTFORMAT_ADTS
bitrate : Bitrate in bits/sec
flags: accès en lecture, accès en écriture
Integer. Range: 8000 - 320000 Default: 128000 Current: 128000
profile : MPEG/AAC encoding profile
flags: accès en lecture, accès en écriture
Enum "GstFaacProfile" Default: 1, "Main profile" Current: 1, "Main profile"
(1): Main profile - MAIN
(2): Low complexity profile - LC
(3): Scalable sampling rate profile - SSR
(4): Long term prediction profile - LTP
tns : Use temporal noise shaping
flags: accès en lecture, accès en écriture
Boolean. Default: false Current: false
midside : Allow mid/side encoding
flags: accès en lecture, accès en écriture
Boolean. Default: true Current: true
shortctl : Block type encorcing
flags: accès en lecture, accès en écriture
Enum "GstFaacShortCtl" Default: 1, "No short blocks" Current: 0, "Normal block type"
(0): Normal block type - SHORTCTL_NORMAL
(1): No short blocks - SHORTCTL_NOSHORT
(2): No long blocks - SHORTCTL_NOLONG
It seems to me everything should be alright here (according to RFC). Does anyone see any obvious mistake ?
Hi
Just found the solution: you have to use the rtpmp4gpay plugin which reports the appropriate parameters to fill in the sdp file; and it works !!!
However, if the flash player is left open for some time, audio and video are desynced.
Why is that ?
Is there a way to avoid it (e.g. using RTSP/RTCP/mpegts) ?
Just found the solution: you have to use the rtpmp4gpay plugin which reports the appropriate parameters to fill in the sdp file; and it works !!!
However, if the flash player is left open for some time, audio and video are desynced.
Why is that ?
Is there a way to avoid it (e.g. using RTSP/RTCP/mpegts) ?
