Adds APIs for reporting pacer queuing delay.

BUG=2775
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8959005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5621 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/pacing/include/paced_sender.h

@@ -49,6 +49,7 @@ class PacedSender : public Module {
                                   bool retransmission) = 0;
     // Called when it's a good time to send a padding data.
     virtual int TimeToSendPadding(int bytes) = 0;
+
    protected:
     virtual ~Callback() {}
   };

webrtc/modules/pacing/paced_sender_unittest.cc

@@ -421,7 +421,8 @@ TEST_F(PacedSenderTest, Pause) {
   EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
   EXPECT_EQ(0, send_bucket_->Process());
 
-  EXPECT_CALL(callback_, TimeToSendPacket(_, _, second_capture_time_ms, false))
+  EXPECT_CALL(
+      callback_, TimeToSendPacket(_, _, second_capture_time_ms, false))
       .Times(1)
       .WillRepeatedly(Return(true));
   EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());

webrtc/video_engine/include/vie_rtp_rtcp.h

@@ -405,6 +405,13 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
       const int video_channel,
       ReceiveBandwidthEstimatorStats* output) const { return -1; }
 
+  // This function gets the PacedSender queuing delay for the last sent frame.
+  // TODO(jiayl): remove the default impl when libjingle is updated.
+  virtual int GetPacerQueuingDelayMs(
+      const int video_channel, int* delay_ms) const {
+    return -1;
+  }
+
   // This function enables capturing of RTP packets to a binary file on a
   // specific channel and for a given direction. The file can later be
   // replayed using e.g. RTP Tools rtpplay since the binary file format is

webrtc/video_engine/vie_encoder.cc

@@ -734,6 +734,10 @@ int32_t ViEEncoder::SendCodecStatistics(
   return 0;
 }
 
+int32_t ViEEncoder::PacerQueuingDelayMs() const {
+  return paced_sender_->QueueInMs();
+}
+
 int32_t ViEEncoder::EstimatedSendBandwidth(
     uint32_t* available_bandwidth) const {
   WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",

webrtc/video_engine/vie_encoder.h

@@ -109,6 +109,7 @@ class ViEEncoder
   int32_t SendCodecStatistics(uint32_t* num_key_frames,
                               uint32_t* num_delta_frames);
 
+  int PacerQueuingDelayMs() const;
   int32_t EstimatedSendBandwidth(
         uint32_t* available_bandwidth) const;
 

webrtc/video_engine/vie_rtp_rtcp_impl.cc

@@ -1040,6 +1040,25 @@ int ViERTP_RTCPImpl::GetReceiveBandwidthEstimatorStats(
   return 0;
 }
 
+int ViERTP_RTCPImpl::GetPacerQueuingDelayMs(
+    const int video_channel, int* delay_ms) const {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
+               ViEId(shared_data_->instance_id(), video_channel),
+               "%s(channel: %d)", __FUNCTION__, video_channel);
+  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+  if (!vie_encoder) {
+    WEBRTC_TRACE(kTraceError, kTraceVideo,
+                 ViEId(shared_data_->instance_id(), video_channel),
+                 "%s: Could not get encoder for channel %d", __FUNCTION__,
+                 video_channel);
+    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+    return -1;
+  }
+  *delay_ms = vie_encoder->PacerQueuingDelayMs();
+  return 0;
+}
+
 int ViERTP_RTCPImpl::StartRTPDump(const int video_channel,
                                   const char file_nameUTF8[1024],
                                   RTPDirections direction) {

webrtc/video_engine/vie_rtp_rtcp_impl.h

@@ -116,6 +116,8 @@ class ViERTP_RTCPImpl
       unsigned int* estimated_bandwidth) const;
   virtual int GetReceiveBandwidthEstimatorStats(
       const int video_channel, ReceiveBandwidthEstimatorStats* output) const;
+  virtual int GetPacerQueuingDelayMs(const int video_channel,
+                                     int* delay_ms) const;
   virtual int StartRTPDump(const int video_channel,
                            const char file_nameUTF8[1024],
                            RTPDirections direction);