Modify to notify MediaQualityStatus checking media direction

1) Add statement to check the direction in Rtp, Rtcp, Jitter and packet loss notification
2) Modify to reset the statistics status when the direction changed in the MediQualityAnalzyer
3) Modify to check rtcp enabled to reset the inactivity counter for rtcp

Bug: 272068717
Test: atest ImsMediaNativeTests, Verified Voice Call hold/resume case in live network. Verified VZW VoWIFI 10.3 which includes hold/resume operation.
Change-Id: I61b0b9cce09139e0a9599c4ca4a07e3899b62276

3 files changed, 59 insertions, 16 deletions

diff --git a/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/audio/AudioSession.cpp b/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/audio/AudioSession.cpp
index 31d44ea7..f872d730 100644
--- a/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/audio/AudioSession.cpp
+++ b/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/audio/AudioSession.cpp
@@ -125,10 +125,8 @@ ImsMediaResult AudioSession::startGraph(RtpConfig* config)
 
     IMLOGI1("[startGraph] state[%d]", getState());
 
-    if (mMediaQualityAnalyzer != nullptr &&
-            !mMediaQualityAnalyzer->isSameConfig(reinterpret_cast<AudioConfig*>(config)))
+    if (mMediaQualityAnalyzer != nullptr)
     {
-        mMediaQualityAnalyzer->stop();
         mMediaQualityAnalyzer->setConfig(reinterpret_cast<AudioConfig*>(config));
         mMediaQualityAnalyzer->start();
     }
diff --git a/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/audio/MediaQualityAnalyzer.cpp b/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/audio/MediaQualityAnalyzer.cpp
index 491d9172..4661ee0e 100644
--- a/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/audio/MediaQualityAnalyzer.cpp
+++ b/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/audio/MediaQualityAnalyzer.cpp
@@ -26,6 +26,8 @@
 #include <algorithm>
 #include <numeric>
 
+using namespace android::telephony::imsmedia;
+
 #define DEFAULT_PARAM                            (-1)
 #define DEFAULT_INACTIVITY_TIME_FOR_CALL_QUALITY (4)
 #define CALL_QUALITY_MONITORING_TIME             (5)
@@ -34,12 +36,17 @@
 #define TIMER_INTERVAL                           (1000)   // 1 sec
 #define STOP_TIMEOUT                             (1000)   // 1 sec
 #define MESSAGE_PROCESSING_INTERVAL              (20000)  // 20 msec
+#define MEDIA_DIRECTION_CONTAINS_RECEIVE(a)            \
+    ((a) == RtpConfig::MEDIA_DIRECTION_SEND_RECEIVE || \
+            (a) == RtpConfig::MEDIA_DIRECTION_RECEIVE_ONLY)
 
 MediaQualityAnalyzer::MediaQualityAnalyzer()
 {
     mTimeStarted = 0;
     mCodecType = 0;
     mCodecAttribute = 0;
+    mIsRxRtpEnabled = false;
+    mIsRtcpEnabled = false;
     mCallback = nullptr;
     std::unique_ptr<RtcpXrEncoder> analyzer(new RtcpXrEncoder());
     mRtcpXrEncoder = std::move(analyzer);
@@ -67,10 +74,17 @@ MediaQualityAnalyzer::~MediaQualityAnalyzer()
 
 void MediaQualityAnalyzer::setConfig(AudioConfig* config)
 {
+    if (!isSameConfig(config))
+    {
+        reset();
+    }
+
+    mIsRxRtpEnabled = MEDIA_DIRECTION_CONTAINS_RECEIVE(config->getMediaDirection());
     mCodecType = config->getCodecType();
     mCodecAttribute = config->getEvsParams().getEvsBandwidth();
 
-    IMLOGD2("[setCodecType] type[%d], bandwidth[%d]", mCodecType, mCodecAttribute);
+    mCallQuality.setCodecType(convertAudioCodecType(
+            mCodecType, ImsMediaAudioUtil::FindMaxEvsBandwidthFromRange(mCodecAttribute)));
 
     if (mCodecType == AudioConfig::CODEC_AMR)
     {
@@ -80,6 +94,20 @@ void MediaQualityAnalyzer::setConfig(AudioConfig* config)
     {
         mRtcpXrEncoder->setSamplingRate(16);
     }
+
+    // Enable RTCP if both interval and direction is valid
+    bool isRtcpEnabled = (config->getRtcpConfig().getIntervalSec() > 0 &&
+            config->getMediaDirection() != RtpConfig::MEDIA_DIRECTION_NO_FLOW);
+
+    if (mIsRtcpEnabled != isRtcpEnabled)
+    {
+        mIsRtcpEnabled = isRtcpEnabled;
+        mCountRtcpInactivity = 0;
+        mNumRtcpPacketReceived = 0;
+    }
+
+    IMLOGI4("[setConfig] codec type[%d], bandwidth[%d], rxRtp[%d], rtcp[%d]", mCodecType,
+            mCodecAttribute, mIsRxRtpEnabled, mIsRtcpEnabled);
 }
 
 void MediaQualityAnalyzer::setCallback(BaseSessionCallback* callback)
@@ -112,16 +140,18 @@ void MediaQualityAnalyzer::setMediaQualityThreshold(const MediaQualityThreshold&
 bool MediaQualityAnalyzer::isSameConfig(AudioConfig* config)
 {
     return (mCodecType == config->getCodecType() &&
-            mCodecAttribute == config->getEvsParams().getEvsBandwidth());
+            mCodecAttribute == config->getEvsParams().getEvsBandwidth() &&
+            mIsRxRtpEnabled == MEDIA_DIRECTION_CONTAINS_RECEIVE(config->getMediaDirection()));
 }
 
 void MediaQualityAnalyzer::start()
 {
-    IMLOGD0("[start]");
-    mCallQuality.setCodecType(convertAudioCodecType(
-            mCodecType, ImsMediaAudioUtil::FindMaxEvsBandwidthFromRange(mCodecAttribute)));
-    mTimeStarted = ImsMediaTimer::GetTimeInMilliSeconds();
-    StartThread();
+    if (IsThreadStopped())
+    {
+        IMLOGD0("[start]");
+        mTimeStarted = ImsMediaTimer::GetTimeInMilliSeconds();
+        StartThread();
+    }
 }
 
 void MediaQualityAnalyzer::stop()
@@ -392,7 +422,7 @@ void MediaQualityAnalyzer::processData(const int32_t timeCount)
 void MediaQualityAnalyzer::processMediaQuality()
 {
     // media quality rtp inactivity
-    if (mNumRxPacket == 0)
+    if (mNumRxPacket == 0 && mIsRxRtpEnabled)
     {
         mCountRtpInactivity += 1000;
     }
@@ -404,7 +434,7 @@ void MediaQualityAnalyzer::processMediaQuality()
     }
 
     // media quality rtcp inactivity
-    if (mNumRtcpPacketReceived == 0)
+    if (mNumRtcpPacketReceived == 0 && mIsRtcpEnabled)
     {
         mCountRtcpInactivity += 1000;
     }
@@ -468,7 +498,7 @@ void MediaQualityAnalyzer::processMediaQuality()
     bool shouldNotify = false;
 
     // check jitter notification
-    if (!mJitterThreshold.empty())
+    if (!mJitterThreshold.empty() && mIsRxRtpEnabled)
     {
         if (mJitterChecker.checkNotifiable(mJitterThreshold, mQualityStatus.getRtpJitterMillis()))
         {
@@ -477,7 +507,7 @@ void MediaQualityAnalyzer::processMediaQuality()
     }
 
     // check packet loss notification
-    if (!mPacketLossThreshold.empty())
+    if (!mPacketLossThreshold.empty() && mIsRxRtpEnabled)
     {
         if (mPacketLossChecker.checkNotifiable(
                     mPacketLossThreshold, mQualityStatus.getRtpPacketLossRate()))
@@ -500,7 +530,7 @@ void MediaQualityAnalyzer::processMediaQuality()
         return;
     }
 
-    if (!mCurrentRtpInactivityTimes.empty())
+    if (!mCurrentRtpInactivityTimes.empty() && mIsRxRtpEnabled)
     {
         std::vector<int32_t>::iterator rtpIter = std::find_if(mCurrentRtpInactivityTimes.begin(),
                 mCurrentRtpInactivityTimes.end(),
@@ -518,7 +548,7 @@ void MediaQualityAnalyzer::processMediaQuality()
         }
     }
 
-    if (mRtcpInactivityTime != 0 && mCountRtcpInactivity == mRtcpInactivityTime)
+    if (mRtcpInactivityTime != 0 && mCountRtcpInactivity == mRtcpInactivityTime && mIsRtcpEnabled)
     {
         notifyMediaQualityStatus();
         mCountRtcpInactivity = 0;
@@ -754,6 +784,17 @@ void MediaQualityAnalyzer::reset()
     mNumRxPacket = 0;
     mNumLostPacket = 0;
     mJitterRxPacket = 0.0;
+
+    // rtp and rtcp inactivity
+    mCountRtpInactivity = 0;
+    mCountRtcpInactivity = 0;
+    mNumRtcpPacketReceived = 0;
+
+    // reset the status
+    mQualityStatus = MediaQualityStatus();
+
+    mPacketLossChecker.initialize(mRtpHysteresisTime);
+    mJitterChecker.initialize(mRtpHysteresisTime);
 }
 
 void MediaQualityAnalyzer::clearPacketList(std::list<RtpPacket*>& list, const int32_t seq)
diff --git a/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/include/audio/MediaQualityAnalyzer.h b/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/include/audio/MediaQualityAnalyzer.h
index f83ea088..feff4892 100644
--- a/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/include/audio/MediaQualityAnalyzer.h
+++ b/service/src/com/android/telephony/imsmedia/lib/libimsmedia/core/include/audio/MediaQualityAnalyzer.h
@@ -256,6 +256,10 @@ protected:
     int32_t mCodecType;
     /** The codec attribute of the audio session, it could be bandwidth in evs codec */
     int32_t mCodecAttribute;
+    /** Whether RTP is activated for the receiver or not */
+    bool mIsRxRtpEnabled;
+    /** Whether RTCP is activated for both sender and receiver */
+    bool mIsRtcpEnabled;
     /** The begin of the rx rtp packet sequence number for Rtcp-Xr report */
     int32_t mBeginSeq;
     /** The end of the rx rtp packet sequence number for Rtcp-Xr report */