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media: reduce RTP jitter limits
Increasing jitter is good for SIP but not RTP stream as the peceptual consequences are worst. Limit to 50ms and 1500 packets the RTP buffer seems to be reasonable in our context. Change-Id: I0bebd27c74981d3e411ca57af820529b1a946363
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committed by
Adrien Béraud

parent
8be8d9932c
commit
6d7c5f966e
@ -44,9 +44,9 @@
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namespace ring {
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// maximum number of packets the jitter buffer can queue
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const unsigned jitterBufferMaxSize_ {5000};
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const unsigned jitterBufferMaxSize_ {1500};
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// maximum time a packet can be queued
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const constexpr auto jitterBufferMaxDelay_ = std::chrono::milliseconds(1500);
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const constexpr auto jitterBufferMaxDelay_ = std::chrono::milliseconds(50);
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MediaDecoder::MediaDecoder() :
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inputCtx_(avformat_alloc_context()),
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