API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay.
TEST=unit-test, manual, trybots.
R=henrik.lundin@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1384005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq/automode.h b/webrtc/modules/audio_coding/neteq/automode.h
index 5996a51..49878c0 100644
--- a/webrtc/modules/audio_coding/neteq/automode.h
+++ b/webrtc/modules/audio_coding/neteq/automode.h
@@ -89,6 +89,12 @@
reached 0 */
int16_t extraDelayMs; /* extra delay for sync with video */
+ int minimum_delay_ms; /* Desired delay, NetEq maintains this amount of
+ delay unless jitter statistics suggests a higher value. */
+ int required_delay_q8; /* Smallest delay required. This is computed
+ according to inter-arrival time and playout mode. It has the same unit
+ as |optBufLevel|. */
+
/* Peak-detection */
/* vector with the latest peak periods (peak spacing in samples) */
uint32_t peakPeriodSamp[NUM_PEAKS];