[gst-devel] Problem with clockrates in rtp stream

[Morris Ford]

I got this resolved by changing the audio to aac. Here is the pipeline with
audi and video desktop capture. This streams successfully to my android
phone.

./test-launch --gst-debug-level=0 "( ximagesrc use-damage=false endx=640
endy=640 startx=100 starty=100 ! video/x-raw-rgb,framerate=10/1 ! queue
max-size-buffers=5 ! ffmpegcolorspace ! videoscale !
video/x-raw-yuv,width=640,height=640 ! ffenc_mpeg4 ! queue ! rtpmp4vpay
name=pay0 pt=96  pulsesrc
device=alsa_output.pci-0000_00_1b.0.analog-stereo.monitor ! audioconvert !
ffenc_aac ! queue ! rtpmp4apay name=pay1 pt=97 )"

Thanks for the help.
Morris


On Wed, May 5, 2010 at 2:27 PM, Morris Ford <morrishford at gmail.com> wrote:

> bump
>
>
> On Wed, May 5, 2010 at 10:18 AM, Morris Ford <morrishford at gmail.com>wrote:
>
>> I changed the pipeline to
>>
>> ./test-launch --gst-debug-level=0 "( ximagesrc use-damage=false endx=640
>> endy=640 startx=100 starty=100 ! video/x-raw-rgb,framerate=10/1 ! queue
>> max-size-buffers=5 ! ffmpegcolorspace ! videoscale !
>> video/x-raw-yuv,width=640,height=640 ! ffenc_mpeg4 ! queue ! rtpmp4vpay
>> name=pay0 pt=96  audiotestsrc is-live=true ! audio/x-raw-int ! faac !
>> audio/mpeg,mpegversion=4 ! queue ! rtpmp4apay name=pay1 pt=97 )"
>>
>> There is no audio in the stream. I am using Quicktime on the client side
>> so I can easily examine the stream type and it shows no audio stream.
>>
>>
>> On Wed, May 5, 2010 at 9:44 AM, Wim Taymans <wim.taymans at gmail.com>wrote:
>>
>>> On Wed, 2010-05-05 at 09:28 -0400, Morris Ford wrote:
>>> > I am trying to stream an mp4v / mp4a rtp stream. The pipeline is at
>>> > the moment:
>>> >
>>> > ./test-launch --gst-debug-level=0 "( ximagesrc use-damage=false
>>> > endx=640 endy=640 startx=100 starty=100 !
>>> > video/x-raw-rgb,framerate=10/1 ! queue max-size-buffers=5 !
>>> > ffmpegcolorspace ! videoscale ! video/x-raw-yuv,width=640,height=640 !
>>> > ffenc_mpeg4 ! queue ! rtpmp4vpay name=pay0 pt=96 ! rtpmux name=muxer
>>> > name=pay0 pt=96  audiotestsrc is-live=true !
>>> > audio/x-raw-int,rate=(int)90000 ! faac ! audio/mpegversion=4 ! queue !
>>> > rtpmp4apay name=pay1 pt=97 ! muxer. muxer. )"
>>> >
>>>
>>> Don't use rtpmux here, it's not needed and it confuses the server
>>> because it expects elements with names pay%d to be unlinked elements.
>>>
>>> Wim
>>>
>>> > The problem I am having is that the audio part is missing. If I remove
>>> > the caps after the faac plugin, the audio is in the stream. When I
>>> > look at the startup log I see a clock-rate of 44100 for the mp4a
>>> > stream and a clock rate of 90000 for the mp4v stream. I am pretty sure
>>> > this is what is wrong. If I stream with the caps for mpegversion=4
>>> > removed I see audio and video clockrates both = 90000.
>>> >
>>> > How do I get the clockrates to match up?
>>> >
>>> > I tried to increase the audio rate to 90000 but it increased to 88200.
>>> > I tried to reduce the video clock rate but I could not get it to
>>> > change.
>>> >
>>> ------------------------------------------------------------------------------
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>>> > gstreamer-devel at lists.sourceforge.net
>>> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>>
>>>
>>>
>>>
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>>
>>
>
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