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From: | Michel de Boer |
Subject: | Re: [Ccrtp-devel] problem while reducing the seding rate |
Date: | Thu, 12 May 2005 20:33:54 +0200 |
User-agent: | Mozilla Thunderbird 1.0 (X11/20041206) |
Why in the first place do you want to transmit the data at a different speed? What codec do you use for the audio? I have only experience with G.711 and GSM encoded audiostreams. These codecs have a constant bit rate, eg. G.711 has 160 bytes/20ms, GSM has 33 bytes/20ms So I have to make sure to send the audio streams with these speeds. I just set the correct payload type with setPayloadFormat and it all works fine as long as I deliver the data at the correct rate to ccRTP. I am not sure how RTP works with variable bit rate codecs. Dinil Divakaran wrote:
For the time being I fixed sending 2833 events as follows: rtp_session->setExpireTimeout(duration of event) This way the packets stay within the oldness check in ccRTP.But, we can not give a static argument to setExpireTimeout since the number of packets change depending on the data that has to be transmitted. Hence, if the value set by setExpireTimeout is okay for a 1 MB data, it need not be useful for sending data larger than 1 MB. This happens since the packets do not stay within the oldness check as the number of packets increase.David Sugar wrote:Oh, this is about sending 2833 events, not receiving...sorry :). Hmm, let me think about this one further!
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