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
This commit is contained in:
Guillaume Roguez
2017-06-22 12:12:09 -04:00
committed by Adrien Béraud
parent 8be8d9932c
commit 6d7c5f966e

View File

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