commit e46c8d387587ba148e229a7bb18f1cc0708a2a87
author turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Wed May 22 20:39:43 2013 +0000
committer turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Wed May 22 20:39:43 2013 +0000
tree 571dcf36f899b7c7da4515f33048c59a38dfc089
parent 561990fd73786354a1cdc6587029f333d79eb2a7

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/automode.c

diff --git a/webrtc/modules/audio_coding/neteq/automode.c b/webrtc/modules/audio_coding/neteq/automode.c
index edee98e..ea6fa8d 100644
--- a/webrtc/modules/audio_coding/neteq/automode.c
+++ b/webrtc/modules/audio_coding/neteq/automode.c
@@ -216,6 +216,14 @@
             streamingMode);
         if (tempvar > 0)
         {
+            int high_lim_delay;
+            /* Convert the minimum delay from milliseconds to packets in Q8.
+             * |fsHz| is sampling rate in Hertz, and |inst->packetSpeechLenSamp|
+             * is the number of samples per packet (according to the last
+             * decoding).
+             */
+            int32_t minimum_delay_q8 = ((inst->minimum_delay_ms *
+                (fsHz / 1000)) << 8) / inst->packetSpeechLenSamp;
             inst->optBufLevel = tempvar;
 
             if (streamingMode != 0)
@@ -224,6 +232,13 @@
                     inst->maxCSumIatQ8);
             }
 
+            /* The required delay. */
+            inst->required_delay_q8 = inst->optBufLevel;
+
+            // Maintain the target delay.
+            inst->optBufLevel = WEBRTC_SPL_MAX(inst->optBufLevel,
+                                               minimum_delay_q8);
+
             /*********/
             /* Limit */
             /*********/
@@ -238,8 +253,12 @@
             maxBufLen = WEBRTC_SPL_LSHIFT_W32(maxBufLen, 8); /* shift to Q8 */
 
             /* Enforce upper limit; 75% of maxBufLen */
-            inst->optBufLevel = WEBRTC_SPL_MIN( inst->optBufLevel,
-                (maxBufLen >> 1) + (maxBufLen >> 2) ); /* 1/2 + 1/4 = 75% */
+            /* 1/2 + 1/4 = 75% */
+            high_lim_delay = (maxBufLen >> 1) + (maxBufLen >> 2);
+            inst->optBufLevel = WEBRTC_SPL_MIN(inst->optBufLevel,
+                                               high_lim_delay);
+            inst->required_delay_q8 = WEBRTC_SPL_MIN(inst->required_delay_q8,
+                                                     high_lim_delay);
         }
         else
         {
@@ -700,6 +719,7 @@
      */
     inst->optBufLevel = WEBRTC_SPL_MIN(4,
         (maxBufLenPackets >> 1) + (maxBufLenPackets >> 1)); /* 75% of maxBufLenPackets */
+    inst->required_delay_q8 = inst->optBufLevel;
     inst->levelFiltFact = 253;
 
     /*