Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1802993002

Cr-Commit-Position: refs/heads/master@{#12015}

webrtc/modules/rtp_rtcp/include/rtp_receiver.h

@@ -45,7 +45,6 @@ class RtpReceiver {
   // Creates an audio-enabled RTP receiver.
   static RtpReceiver* CreateAudioReceiver(
       Clock* clock,
-      RtpAudioFeedback* incoming_audio_feedback,
       RtpData* incoming_payload_callback,
       RtpFeedback* incoming_messages_callback,
       RTPPayloadRegistry* rtp_payload_registry);

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h

@@ -217,16 +217,6 @@ class RtpFeedback {
   virtual void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) = 0;
 };
 
-class RtpAudioFeedback {
- public:
-  virtual void OnPlayTelephoneEvent(const uint8_t event,
-                                    const uint16_t lengthMs,
-                                    const uint8_t volume) = 0;
-
- protected:
-  virtual ~RtpAudioFeedback() {}
-};
-
 class RtcpIntraFrameObserver {
  public:
   virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
@@ -357,16 +347,6 @@ class NullRtpData : public RtpData {
   }
 };
 
-// Null object version of RtpAudioFeedback.
-class NullRtpAudioFeedback : public RtpAudioFeedback {
- public:
-  virtual ~NullRtpAudioFeedback() {}
-
-  void OnPlayTelephoneEvent(const uint8_t event,
-                            const uint16_t lengthMs,
-                            const uint8_t volume) override {}
-};
-
 // Statistics about packet loss for a single directional connection. All values
 // are totals since the connection initiated.
 struct RtpPacketLossStats {

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc

@@ -20,13 +20,11 @@
 
 namespace webrtc {
 RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
-    RtpData* data_callback,
+    RtpData* data_callback) {
-    RtpAudioFeedback* incoming_messages_callback) {
+  return new RTPReceiverAudio(data_callback);
-  return new RTPReceiverAudio(data_callback, incoming_messages_callback);
 }
 
-RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback,
+RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
-                                   RtpAudioFeedback* incoming_messages_callback)
     : RTPReceiverStrategy(data_callback),
       TelephoneEventHandler(),
       last_received_frequency_(8000),
@@ -40,8 +38,7 @@ RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback,
       g722_payload_type_(-1),
       last_received_g722_(false),
       num_energy_(0),
-      current_remote_energy_(),
+      current_remote_energy_() {
-      cb_audio_feedback_(incoming_messages_callback) {
   last_payload_.Audio.channels = 1;
   memset(current_remote_energy_, 0, sizeof(current_remote_energy_));
 }

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h

@@ -28,8 +28,7 @@ class CriticalSectionWrapper;
 class RTPReceiverAudio : public RTPReceiverStrategy,
                          public TelephoneEventHandler {
  public:
-  RTPReceiverAudio(RtpData* data_callback,
+  explicit RTPReceiverAudio(RtpData* data_callback);
-                   RtpAudioFeedback* incoming_messages_callback);
   virtual ~RTPReceiverAudio() {}
 
   // The following three methods implement the TelephoneEventHandler interface.
@@ -119,8 +118,6 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
 
   uint8_t num_energy_;
   uint8_t current_remote_energy_[kRtpCsrcSize];
-
-  RtpAudioFeedback* cb_audio_feedback_;
 };
 }  // namespace webrtc
 

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc

@@ -34,33 +34,26 @@ RtpReceiver* RtpReceiver::CreateVideoReceiver(
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
-      clock, NullObjectRtpAudioFeedback(), incoming_messages_callback,
+      clock, incoming_messages_callback, rtp_payload_registry,
-      rtp_payload_registry,
       RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
 }
 
 RtpReceiver* RtpReceiver::CreateAudioReceiver(
     Clock* clock,
-    RtpAudioFeedback* incoming_audio_feedback,
     RtpData* incoming_payload_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
-  if (!incoming_audio_feedback)
-    incoming_audio_feedback = NullObjectRtpAudioFeedback();
   if (!incoming_payload_callback)
     incoming_payload_callback = NullObjectRtpData();
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
-      clock, incoming_audio_feedback, incoming_messages_callback,
+      clock, incoming_messages_callback, rtp_payload_registry,
-      rtp_payload_registry,
+      RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback));
-      RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback,
-                                               incoming_audio_feedback));
 }
 
 RtpReceiverImpl::RtpReceiverImpl(
     Clock* clock,
-    RtpAudioFeedback* incoming_audio_messages_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry,
     RTPReceiverStrategy* rtp_media_receiver)
@@ -79,7 +72,6 @@ RtpReceiverImpl::RtpReceiverImpl(
       last_received_frame_time_ms_(-1),
       last_received_sequence_number_(0),
       nack_method_(kNackOff) {
-  assert(incoming_audio_messages_callback);
   assert(incoming_messages_callback);
 
   memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h

@@ -26,7 +26,6 @@ class RtpReceiverImpl : public RtpReceiver {
   // want callbacks to do nothing). This class takes ownership of the media
   // receiver but nothing else.
   RtpReceiverImpl(Clock* clock,
-                  RtpAudioFeedback* incoming_audio_messages_callback,
                   RtpFeedback* incoming_messages_callback,
                   RTPPayloadRegistry* rtp_payload_registry,
                   RTPReceiverStrategy* rtp_media_receiver);

webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h

@@ -27,9 +27,7 @@ class TelephoneEventHandler;
 class RTPReceiverStrategy {
  public:
   static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback);
-  static RTPReceiverStrategy* CreateAudioStrategy(
+  static RTPReceiverStrategy* CreateAudioStrategy(RtpData* data_callback);
-      RtpData* data_callback,
-      RtpAudioFeedback* incoming_messages_callback);
 
   virtual ~RTPReceiverStrategy() {}
 

webrtc/modules/rtp_rtcp/source/rtp_utility.cc

@@ -27,11 +27,6 @@ RtpFeedback* NullObjectRtpFeedback() {
   return &null_rtp_feedback;
 }
 
-RtpAudioFeedback* NullObjectRtpAudioFeedback() {
-  static NullRtpAudioFeedback null_rtp_audio_feedback;
-  return &null_rtp_audio_feedback;
-}
-
 ReceiveStatistics* NullObjectReceiveStatistics() {
   static NullReceiveStatistics null_receive_statistics;
   return &null_receive_statistics;

webrtc/modules/rtp_rtcp/source/rtp_utility.h

@@ -26,7 +26,6 @@ const uint8_t kRtpMarkerBitMask = 0x80;
 
 RtpData* NullObjectRtpData();
 RtpFeedback* NullObjectRtpFeedback();
-RtpAudioFeedback* NullObjectRtpAudioFeedback();
 ReceiveStatistics* NullObjectReceiveStatistics();
 
 namespace RtpUtility {

webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc

@@ -97,7 +97,7 @@ class RtpRtcpAPITest : public ::testing::Test {
     rtp_payload_registry_.reset(new RTPPayloadRegistry(
             RTPPayloadStrategy::CreateStrategy(true)));
     rtp_receiver_.reset(RtpReceiver::CreateAudioReceiver(
-        &fake_clock_, NULL, NULL, NULL, rtp_payload_registry_.get()));
+        &fake_clock_, NULL, NULL, rtp_payload_registry_.get()));
   }
 
   rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

@@ -86,7 +86,6 @@ class RtpRtcpAudioTest : public ::testing::Test {
   ~RtpRtcpAudioTest() {}
 
   void SetUp() override {
-    audioFeedback = new NullRtpAudioFeedback();
     data_receiver1 = new VerifyingAudioReceiver();
     data_receiver2 = new VerifyingAudioReceiver();
     rtp_callback = new RTPCallback();
@@ -109,16 +108,14 @@ class RtpRtcpAudioTest : public ::testing::Test {
 
     module1 = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
-        &fake_clock, audioFeedback, data_receiver1, NULL,
+        &fake_clock, data_receiver1, NULL, rtp_payload_registry1_.get()));
-        rtp_payload_registry1_.get()));
 
     configuration.receive_statistics = receive_statistics2_.get();
     configuration.outgoing_transport = transport2;
 
     module2 = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
-        &fake_clock, audioFeedback, data_receiver2, NULL,
+        &fake_clock, data_receiver2, NULL, rtp_payload_registry2_.get()));
-        rtp_payload_registry2_.get()));
 
     transport1->SetSendModule(module2, rtp_payload_registry2_.get(),
                               rtp_receiver2_.get(), receive_statistics2_.get());
@@ -131,7 +128,6 @@ class RtpRtcpAudioTest : public ::testing::Test {
     delete module2;
     delete transport1;
     delete transport2;
-    delete audioFeedback;
     delete data_receiver1;
     delete data_receiver2;
     delete rtp_callback;
@@ -149,7 +145,6 @@ class RtpRtcpAudioTest : public ::testing::Test {
   VerifyingAudioReceiver* data_receiver2;
   LoopBackTransport* transport1;
   LoopBackTransport* transport2;
-  NullRtpAudioFeedback* audioFeedback;
   RTPCallback* rtp_callback;
   uint32_t test_ssrc;
   uint32_t test_timestamp;

webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc

@@ -101,7 +101,7 @@ class RtpRtcpRtcpTest : public ::testing::Test {
     rtp_feedback1_.reset(new TestRtpFeedback(module1));
 
     rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
-        &fake_clock, NULL, receiver, rtp_feedback1_.get(),
+        &fake_clock, receiver, rtp_feedback1_.get(),
         rtp_payload_registry1_.get()));
 
     configuration.receive_statistics = receive_statistics2_.get();
@@ -113,7 +113,7 @@ class RtpRtcpRtcpTest : public ::testing::Test {
     rtp_feedback2_.reset(new TestRtpFeedback(module2));
 
     rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
-        &fake_clock, NULL, receiver, rtp_feedback2_.get(),
+        &fake_clock, receiver, rtp_feedback2_.get(),
         rtp_payload_registry2_.get()));
 
     transport1->SetSendModule(module2, rtp_payload_registry2_.get(),

webrtc/voice_engine/channel.cc

@@ -722,7 +722,6 @@ Channel::Channel(int32_t channelId,
           ReceiveStatistics::Create(Clock::GetRealTimeClock())),
       rtp_receiver_(
           RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
-                                           nullptr,
                                            this,
                                            this,
                                            rtp_payload_registry_.get())),