Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6649 4adac7df-926f-26a2-2b94-8c16560cd09d

6 files changed, 19 insertions, 43 deletions

diff --git a/modules/rtp_rtcp/interface/rtp_rtcp.h b/modules/rtp_rtcp/interface/rtp_rtcp.h
index 235ca849..34f4d3f1 100644
--- a/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -68,6 +68,7 @@ class RtpRtcp : public Module {
     RemoteBitrateEstimator* remote_bitrate_estimator;
     PacedSender* paced_sender;
     BitrateStatisticsObserver* send_bitrate_observer;
+    FrameCountObserver* send_frame_count_observer;
   };
 
   /*
@@ -340,10 +341,6 @@ class RtpRtcp : public Module {
 
     virtual int TimeToSendPadding(int bytes) = 0;
 
-    virtual void RegisterSendFrameCountObserver(
-        FrameCountObserver* observer) = 0;
-    virtual FrameCountObserver* GetSendFrameCountObserver() const = 0;
-
     virtual bool GetSendSideDelay(int* avg_send_delay_ms,
                                   int* max_send_delay_ms) const = 0;
 
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 855d51b7..a7f81c61 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -38,7 +38,8 @@ RtpRtcp::Configuration::Configuration()
       audio_messages(NullObjectRtpAudioFeedback()),
       remote_bitrate_estimator(NULL),
       paced_sender(NULL),
-      send_bitrate_observer(NULL) {
+      send_bitrate_observer(NULL),
+      send_frame_count_observer(NULL) {
 }
 
 RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
@@ -62,7 +63,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
                   configuration.outgoing_transport,
                   configuration.audio_messages,
                   configuration.paced_sender,
-                  configuration.send_bitrate_observer),
+                  configuration.send_bitrate_observer,
+                  configuration.send_frame_count_observer),
       rtcp_sender_(configuration.id,
                    configuration.audio,
                    configuration.clock,
@@ -1350,15 +1352,6 @@ StreamDataCountersCallback*
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 
-void ModuleRtpRtcpImpl::RegisterSendFrameCountObserver(
-    FrameCountObserver* observer) {
-  rtp_sender_.RegisterFrameCountObserver(observer);
-}
-
-FrameCountObserver* ModuleRtpRtcpImpl::GetSendFrameCountObserver() const {
-  return rtp_sender_.GetFrameCountObserver();
-}
-
 bool ModuleRtpRtcpImpl::IsDefaultModule() const {
   CriticalSectionScoped cs(critical_section_module_ptrs_.get());
   return !child_modules_.empty();
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index b65131fb..37a0fa46 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -373,10 +373,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
 
   void OnRequestSendReport();
 
-  virtual void RegisterSendFrameCountObserver(
-      FrameCountObserver* observer) OVERRIDE;
-  virtual FrameCountObserver* GetSendFrameCountObserver() const OVERRIDE;
-
  protected:
   void RegisterChildModule(RtpRtcp* module);
 
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 858fc42a..c98d4981 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -46,7 +46,8 @@ RTPSender::RTPSender(const int32_t id,
                      Transport* transport,
                      RtpAudioFeedback* audio_feedback,
                      PacedSender* paced_sender,
-                     BitrateStatisticsObserver* bitrate_callback)
+                     BitrateStatisticsObserver* bitrate_callback,
+                     FrameCountObserver* frame_count_observer)
     : clock_(clock),
       bitrate_sent_(clock, this),
       id_(id),
@@ -71,9 +72,9 @@ RTPSender::RTPSender(const int32_t id,
       packet_history_(clock),
       // Statistics
       statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      frame_count_observer_(NULL),
       rtp_stats_callback_(NULL),
       bitrate_callback_(bitrate_callback),
+      frame_count_observer_(frame_count_observer),
       // RTP variables
       start_timestamp_forced_(false),
       start_timestamp_(0),
@@ -1664,16 +1665,6 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, uint16_t* length,
   *length += 2;
 }
 
-void RTPSender::RegisterFrameCountObserver(FrameCountObserver* observer) {
-  CriticalSectionScoped cs(statistics_crit_.get());
-  frame_count_observer_ = observer;
-}
-
-FrameCountObserver* RTPSender::GetFrameCountObserver() const {
-  CriticalSectionScoped cs(statistics_crit_.get());
-  return frame_count_observer_;
-}
-
 void RTPSender::RegisterRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   CriticalSectionScoped cs(statistics_crit_.get());
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 0cc35cf4..265e69c3 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -70,7 +70,8 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
   RTPSender(const int32_t id, const bool audio, Clock *clock,
             Transport *transport, RtpAudioFeedback *audio_feedback,
             PacedSender *paced_sender,
-            BitrateStatisticsObserver* bitrate_callback);
+            BitrateStatisticsObserver* bitrate_callback,
+            FrameCountObserver* frame_count_observer);
   virtual ~RTPSender();
 
   void ProcessBitrate();
@@ -265,9 +266,6 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
   int32_t SetFecParameters(const FecProtectionParams *delta_params,
                            const FecProtectionParams *key_params);
 
-  virtual void RegisterFrameCountObserver(FrameCountObserver* observer);
-  virtual FrameCountObserver* GetFrameCountObserver() const;
-
   int SendPadData(int payload_type, uint32_t timestamp, int64_t capture_time_ms,
                   int32_t bytes, StorageType store,
                   bool force_full_size_packets, bool only_pad_after_markerbit);
@@ -373,11 +371,11 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
   scoped_ptr<CriticalSectionWrapper> statistics_crit_;
   SendDelayMap send_delays_ GUARDED_BY(statistics_crit_);
   std::map<FrameType, uint32_t> frame_counts_ GUARDED_BY(statistics_crit_);
-  FrameCountObserver* frame_count_observer_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
   BitrateStatisticsObserver* const bitrate_callback_;
+  FrameCountObserver* const frame_count_observer_;
 
   // RTP variables
   bool start_timestamp_forced_ GUARDED_BY(send_critsect_);
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index e08aa202..42ee023e 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -94,7 +94,7 @@ class RtpSenderTest : public ::testing::Test {
 
   virtual void SetUp() {
     rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
-                                    &mock_paced_sender_, NULL));
+                                    &mock_paced_sender_, NULL, NULL));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 
@@ -672,7 +672,7 @@ TEST_F(RtpSenderTest, SendPadding) {
 TEST_F(RtpSenderTest, SendRedundantPayloads) {
   MockTransport transport;
   rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport, NULL,
-                                  &mock_paced_sender_, NULL));
+                                  &mock_paced_sender_, NULL, NULL));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   // Make all packets go through the pacer.
   EXPECT_CALL(mock_paced_sender_,
@@ -817,6 +817,9 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
     uint32_t delta_frames_;
   } callback;
 
+  rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
+                                  &mock_paced_sender_, NULL, &callback));
+
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
@@ -825,8 +828,6 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
-  rtp_sender_->RegisterFrameCountObserver(&callback);
-
   ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
                                              4321, payload, sizeof(payload),
                                              NULL));
@@ -845,7 +846,7 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
   EXPECT_EQ(1U, callback.key_frames_);
   EXPECT_EQ(1U, callback.delta_frames_);
 
-  rtp_sender_->RegisterFrameCountObserver(NULL);
+  rtp_sender_.reset();
 }
 
 TEST_F(RtpSenderTest, BitrateCallbacks) {
@@ -866,7 +867,7 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
     BitrateStatistics bitrate_;
   } callback;
   rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
-                                  &mock_paced_sender_, &callback));
+                                  &mock_paced_sender_, &callback, NULL));
 
   // Simulate kNumPackets sent with kPacketInterval ms intervals.
   const uint32_t kNumPackets = 15;
@@ -922,7 +923,7 @@ class RtpSenderAudioTest : public RtpSenderTest {
   virtual void SetUp() {
     payload_ = kAudioPayload;
     rtp_sender_.reset(new RTPSender(0, true, &fake_clock_, &transport_, NULL,
-                                    &mock_paced_sender_, NULL));
+                                    &mock_paced_sender_, NULL, NULL));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 };