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/webrtc_neteq.c

diff --git a/webrtc/modules/audio_coding/neteq/webrtc_neteq.c b/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
index 31940c8..8347925 100644
--- a/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+++ b/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
@@ -437,6 +437,7 @@
     NetEqMainInst->MCUinst.first_packet = 1;
     NetEqMainInst->MCUinst.one_desc = 0;
     NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs = 0;
+    NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.minimum_delay_ms = 0;
     NetEqMainInst->MCUinst.NoOfExpandCalls = 0;
     NetEqMainInst->MCUinst.fs = fs;
 
@@ -529,6 +530,19 @@
     return (0);
 }
 
+int WebRtcNetEQ_SetMinimumDelay(void *inst, int minimum_delay_ms) {
+  MainInst_t *NetEqMainInst = (MainInst_t*) inst;
+  if (NetEqMainInst == NULL)
+    return -1;
+  if (minimum_delay_ms < 0 || minimum_delay_ms > 10000) {
+      NetEqMainInst->ErrorCode = -FAULTY_DELAYVALUE;
+      return -1;
+  }
+  NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.minimum_delay_ms =
+      minimum_delay_ms;
+  return 0;
+}
+
 int WebRtcNetEQ_SetPlayoutMode(void *inst, enum WebRtcNetEQPlayoutMode playoutMode)
 {
     MainInst_t *NetEqMainInst = (MainInst_t*) inst;
@@ -1213,7 +1227,7 @@
     /* Get optimal buffer size */
     /***************************/
 
-    if (NetEqMainInst->MCUinst.fs != 0 && NetEqMainInst->MCUinst.fs <= WEBRTC_SPL_WORD16_MAX)
+    if (NetEqMainInst->MCUinst.fs != 0)
     {
         /* preferredBufferSize = Bopt * packSizeSamples / (fs/1000) */
         stats->preferredBufferSize
@@ -1693,3 +1707,25 @@
   }
   return SYNC_PAYLOAD_LEN_BYTES;
 }
+
+int WebRtcNetEQ_GetRequiredDelayMs(const void* inst) {
+  const MainInst_t* NetEqMainInst = (MainInst_t*)inst;
+  const AutomodeInst_t* auto_mode = (NetEqMainInst == NULL) ? NULL :
+      &NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst;
+
+  /* Instance sanity */
+  if (NetEqMainInst == NULL || auto_mode == NULL)
+    return 0;
+
+  if (NetEqMainInst->MCUinst.fs == 0)
+    return 0;  // Sampling rate not initialized.
+
+  /* |required_delay_q8| has the unit of packets in Q8 domain, therefore,
+   * the corresponding delay is
+   * required_delay_ms = (1000 * required_delay_q8 * samples_per_packet /
+   *     sample_rate_hz) / 256;
+   */
+  return (auto_mode->required_delay_q8 *
+      ((auto_mode->packetSpeechLenSamp * 1000) / NetEqMainInst->MCUinst.fs) +
+      128) >> 8;
+}