Have AudioSendStream register CNG payload types with the RtpRtcpModule.

webrtc/audio/audio_send_stream_unittest.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder_factory.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h"
 #include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
 #include "webrtc/voice_engine/transmit_mixer.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
@@ skipping 172 matching lines @@
   MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
   RtpTransportControllerSendInterface* transport() { return &fake_transport_; }
   BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
   rtc::TaskQueue* worker_queue() { return &worker_queue_; }
   RtcEventLog* event_log() { return &event_log_; }
   MockVoiceEngine* voice_engine() { return &voice_engine_; }
 
   void SetupDefaultChannelProxy(bool audio_bwe_enabled) {
     using testing::StrEq;
     channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
+    EXPECT_CALL(*channel_proxy_, GetRtpRtcp(_, _))
+        .WillRepeatedly(Invoke(
+            [this](RtpRtcp** rtp_rtcp_module, RtpReceiver** rtp_receiver) {
+              *rtp_rtcp_module = &this->rtp_rtcp_;
+              *rtp_receiver = nullptr;  // Not deemed necessary for tests yet.
+            }));
     EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
     EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
     EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
     EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1);
     EXPECT_CALL(*channel_proxy_,
                 SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
         .Times(1);
     if (audio_bwe_enabled) {
       EXPECT_CALL(*channel_proxy_,
                   EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
@@ skipping 84 matching lines @@
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
   testing::NiceMock<MockCongestionObserver> bitrate_observer_;
   MockAudioProcessing audio_processing_;
   MockTransmitMixer transmit_mixer_;
   AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
   FakeRtpTransportController fake_transport_;
   MockRtcEventLog event_log_;
+  MockRtpRtcp rtp_rtcp_;
   MockRtcpRttStats rtcp_rtt_stats_;
   testing::NiceMock<MockLimitObserver> limit_observer_;
   BitrateAllocator bitrate_allocator_;
   // |worker_queue| is defined last to ensure all pending tasks are cancelled
   // and deleted before any other members.
   rtc::TaskQueue worker_queue_;
 };
 }  // namespace
 
 TEST(AudioSendStreamTest, ConfigToString) {
@@ skipping 212 matching lines @@
   stream_config.send_codec_spec->cng_payload_type = rtc::Optional<int>(105);
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
       helper.rtcp_rtt_stats());
   send_stream.Reconfigure(stream_config);
 }
 
 }  // namespace test
 }  // namespace webrtc