Add API to retreive last received RTP timestamp to VoiceEngine.

BUG=

Review URL: https://webrtc-codereview.appspot.com/969016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3271 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/voice_engine/channel.cc

@@ -799,6 +799,8 @@ Channel::OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
                  rtpHeader->header.payloadType,
                  rtpHeader->type.Audio.channel);
 
+    _lastRemoteTimeStamp = rtpHeader->header.timestamp;
+
     if (!_playing)
     {
         // Avoid inserting into NetEQ when we are not playing. Count the
@@ -1152,6 +1154,7 @@ Channel::Channel(const WebRtc_Word32 channelId,
     _insertExtraRTPPacket(false),
     _extraMarkerBit(false),
     _lastLocalTimeStamp(0),
+    _lastRemoteTimeStamp(0),
     _lastPayloadType(0),
     _includeAudioLevelIndication(false),
     _rtpPacketTimedOut(false),

webrtc/voice_engine/channel.h

@@ -359,6 +359,7 @@ public:
     int InsertExtraRTPPacket(unsigned char payloadType, bool markerBit,
                              const char* payloadData,
                              unsigned short payloadSize);
+    uint32_t LastRemoteTimeStamp() { return _lastRemoteTimeStamp; }
 
 public:
     // From AudioPacketizationCallback in the ACM
@@ -635,6 +636,7 @@ private:
     bool _insertExtraRTPPacket;
     bool _extraMarkerBit;
     WebRtc_UWord32 _lastLocalTimeStamp;
+    uint32_t _lastRemoteTimeStamp;
     WebRtc_Word8 _lastPayloadType;
     bool _includeAudioLevelIndication;
     // VoENetwork

webrtc/voice_engine/include/voe_rtp_rtcp.h

@@ -245,6 +245,10 @@ public:
         int channel, unsigned char payloadType, bool markerBit,
         const char* payloadData, unsigned short payloadSize) = 0;
 
+    // Gets the timestamp of the last RTP packet received by |channel|.
+    virtual int GetLastRemoteTimeStamp(int channel,
+                                       uint32_t* lastRemoteTimeStamp) = 0;
+
 protected:
     VoERTP_RTCP() {}
     virtual ~VoERTP_RTCP() {}

webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc

@@ -48,3 +48,9 @@ TEST_F(RtpRtcpBeforeStreamingTest, GetLocalSsrcObeysSetLocalSsrc) {
   EXPECT_EQ(0, voe_rtp_rtcp_->GetLocalSSRC(channel_, result));
   EXPECT_EQ(1234u, result);
 }
+
+TEST_F(RtpRtcpBeforeStreamingTest, GetLastRemoteTimeStamp) {
+  uint32_t timestamp;
+  EXPECT_EQ(0, voe_rtp_rtcp_->GetLastRemoteTimeStamp(channel_, &timestamp));
+  EXPECT_EQ(0u, timestamp);
+}

webrtc/voice_engine/voe_rtp_rtcp_impl.cc

@@ -662,6 +662,27 @@ int VoERTP_RTCPImpl::InsertExtraRTPPacket(int channel,
                                             payloadSize);
 }
 
+int VoERTP_RTCPImpl::GetLastRemoteTimeStamp(int channel,
+                                            uint32_t* timestamp) {
+    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+                 "GetLastRemoteTimeStamp(channel=%d, timestamp=?)", channel);
+    if (!_shared->statistics().Initialized())
+    {
+        _shared->SetLastError(VE_NOT_INITED, kTraceError);
+        return -1;
+    }
+    voe::ScopedChannel sc(_shared->channel_manager(), channel);
+    voe::Channel* channelPtr = sc.ChannelPtr();
+    if (channelPtr == NULL)
+    {
+        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+            "GetLastRemoteTimeStamp() failed to locate channel");
+        return -1;
+    }
+    *timestamp = channelPtr->LastRemoteTimeStamp();
+    return 0;
+}
+
 #endif  // #ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
 
 }  // namespace webrtc

webrtc/voice_engine/voe_rtp_rtcp_impl.h

@@ -111,7 +111,8 @@ public:
                                      bool markerBit,
                                      const char* payloadData,
                                      unsigned short payloadSize);
-
+    virtual int GetLastRemoteTimeStamp(int channel,
+                                       uint32_t* lastRemoteTimeStamp);
 protected:
     VoERTP_RTCPImpl(voe::SharedData* shared);
     virtual ~VoERTP_RTCPImpl();