commit d821a09b4e919341a89bc55a96204e5bf898824b
author dhavalc <dhavalc@google.com> Wed Jul 05 07:49:28 2023 +0000
committer Dhavalkumar Chaudhary <dhavalc@google.com> Wed Jul 05 13:29:57 2023 +0000
tree dcea6a086a146851ea8653415df89290dafbe948
parent 6013a77b88da7602333a903c323e330afbf4ab87

Added changes to handle ssrc change in case of Handover on jitterBuffer side

- RtpStack will send indication to jitter buffer for SRRC change,
This request should not send to AoC for processing as this will have
null dataBuffer, Data size as SSRC value and aoc will send error response as invalid payload as below.

- [RTP_EP] RX payload is too large (1735700372, 320 max)

Bug: 289974090

Test: Verified using peer side VoLTE -> VoWiFi and VoWiFi -> VoLTE handover.
Change-Id: Ib2068479860203673594b2e66d09725a7b42ea64

service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/audio/AudioJitterBuffer.cpp

diff --git a/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/audio/AudioJitterBuffer.cpp b/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/audio/AudioJitterBuffer.cpp
index eb31738..44a464d 100644
--- a/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/audio/AudioJitterBuffer.cpp
+++ b/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/audio/AudioJitterBuffer.cpp
@@ -149,10 +149,7 @@
         mJitterAnalyzer.Reset();
         mJitterAnalyzer.SetMinMaxJitterBufferSize(mMinJitterBufferSize, mMaxJitterBufferSize);
 
-        if (!mWaiting)  // add indication to check the ssrc changed in Get()
-        {
-            mDataQueue.Add(&currEntry);
-        }
+        mDataQueue.Add(&currEntry);
 
         IMLOGI1("[Add] ssrc[%u]", mSsrc);
         return;
@@ -292,13 +289,12 @@
     bool bForceToPlay = false;
     mCheckUpdateJitterPacketCnt++;
 
-    if (!mWaiting && mDataQueue.Get(&pEntry) &&
-            pEntry->subtype == MEDIASUBTYPE_REFRESHED)  // ssrc changed
+    if (mDataQueue.Get(&pEntry) && pEntry->subtype == MEDIASUBTYPE_REFRESHED)  // ssrc changed
     {
         Reset();
         mDataQueue.Delete();  // delete indication frame of ssrc
 
-        if (mDataQueue.Get(&pEntry))  // get next frame
+        if (!mWaiting && mDataQueue.Get(&pEntry))  // get next frame
         {
             mCurrPlayingTS = pEntry->nTimestamp;  // play directly
             mWaiting = false;