[Speex-dev] Speex and RTP
david.hogan at freshtel.net
Sun Feb 4 15:25:37 PST 2007
One thing I would note is that speex is designed for 8kHz, 16kHz or
32kHz. 160 samples is equal to 20ms of 8kHz audio. Have you tried
resampling from 11.025kHz to 8kHz and then using the speex 8kHz mode?
(nb, or narrow band). Or, if you want to preserve the higher quality of
your 11.025kHz sample rate, resample to 16kHz and use the wideband speex
I believe there is new resampling functionality in the speex svn head,
although I haven't tested it yet. You might also want to check out
'Secret Rabbit Code' for your resampling.
Hope this helps.
From: speex-dev-bounces at xiph.org
[mailto:speex-dev-bounces at xiph.org] On Behalf Of Randy Schefiele
Sent: Saturday, 3 February 2007 7:12 AM
To: speex-dev at xiph.org
Subject: [Speex-dev] Speex and RTP
I am currently developing a RTSP/RTP/SDP solution to stream
Speex encoded data. Using my current source, I have successfully
streamed u-law and PCM encoded audio but have been unsuccessful thus far
Because of some constraints of my system, I am encoding audio at
11.025kHz. I am still using the 160 samples per frame which makes my
frame size 28 bytes. I have successfully written the SDP needed to
describe the Speex stream but I haven't found a player that will play
the stream. I could be doing something wrong, but does anyone know of
any players that can play RTP Speex streams so I can be sure that my
solution is incorrect?
Also, I have looked over the draft RFC of the Speex RTP payload
and I was curious to know if, because I am not using a standard sample
rate, my solution will work. In the draft RFC (and the Speex manual for
that matter) it always refers to Speex frames containing 20ms of audio.
In my case, a Speex frame actually contains roughly 14.5 seconds of
audio. Am I going to need to hack the Speex encoder to make my frames
be 20ms in order for the RTP to successfully play?
Any advice/help anyone can give would be greatly appreciated.
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