From b9429ba3fb93894dc7b3ad7e9634924e5d43d9c9 Mon Sep 17 00:00:00 2001 From: Arun Raghavan Date: Tue, 1 Nov 2011 13:58:42 +0530 Subject: [PATCH] rtp: Set proper latency values on pulsesink/pulsesrc This sets a 50ms buffer-time on pulsesink and pulsesrc and a 25ms latency-time on pulsesink, which should decrease the overall latency of the audio pipeline (the current value being used is the default buffer-time of 200ms). The only concern here might be performance on lower-end hardware. On an AMD C-50 processor (dual-core 1GHz) based netbook, there was no real performance impact from this. --- gst/fsrtpconference/default-element-properties | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/gst/fsrtpconference/default-element-properties b/gst/fsrtpconference/default-element-properties index 80884c2..38ddf0a 100644 --- a/gst/fsrtpconference/default-element-properties +++ b/gst/fsrtpconference/default-element-properties @@ -47,3 +47,14 @@ ptime-multiple=20000000 [rtppcmapay] ptime-multiple=20000000 + +# Set appropriate buffer/latency parameters for voip. The key parameter is +# buffer-time, which determines the latency in the conventional sense (X us of +# buffering between client and playback/capture. We take a conservatively high +# value for these to lower CPU load on less powerful systems. +[pulsesink] +buffer-time=50000 +latency-time=25000 + +[pulsesrc] +buffer-time=50000 -- 1.7.7.1