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

webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h

@@ -309,6 +309,19 @@ int WebRtcNetEQ_RecInSyncRTP(void* inst,
                              WebRtcNetEQ_RTPInfo* rtp_info,
                              uint32_t receive_timestamp);
 
+/*
+ * Set a minimum latency for the jitter buffer. The overall delay is the max of
+ * |minimum_delay_ms| and the latency that is internally computed based on the
+ * inter-arrival times.
+ */
+int WebRtcNetEQ_SetMinimumDelay(void *inst, int minimum_delay_ms);
+
+/*
+ * Get the least required delay in milliseconds given inter-arrival times
+ * and playout mode.
+ */
+int WebRtcNetEQ_GetRequiredDelayMs(const void* inst);
+
 #ifdef __cplusplus
 }
 #endif