Replace AudioConferenceMixer with AudioMixer.

webrtc/audio/audio_receive_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 <vector>
 
+#include "webrtc/api/test/mock_audio_mixer.h"
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
-#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
 
@@ skipping 36 matching lines @@
     123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0};
 const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
 
 struct ConfigHelper {
   ConfigHelper()
       : simulated_clock_(123456),
         decoder_factory_(new rtc::RefCountedObject<MockAudioDecoderFactory>),
         congestion_controller_(&simulated_clock_,
                                &bitrate_observer_,
                                &remote_bitrate_observer_,
-                               &event_log_) {
+                               &event_log_),
+        audio_mixer_(new rtc::RefCountedObject<MockAudioMixer>()) {
     using testing::Invoke;
 
     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_, audio_processing());
     EXPECT_CALL(voice_engine_, audio_device_module());
     EXPECT_CALL(voice_engine_, audio_transport());
 
     AudioState::Config config;
     config.voice_engine = &voice_engine_;
-    config.audio_mixer = AudioMixerImpl::Create();
+    config.audio_mixer = audio_mixer_;
     audio_state_ = AudioState::Create(config);
 
     EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
         .WillOnce(Invoke([this](int channel_id) {
           EXPECT_FALSE(channel_proxy_);
           channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
           EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
           EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1);
           EXPECT_CALL(*channel_proxy_,
               SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
@@ skipping 16 matching lines @@
               .WillOnce(ReturnRef(decoder_factory_));
           testing::Expectation expect_set =
               EXPECT_CALL(*channel_proxy_, SetRtcEventLog(&event_log_))
                   .Times(1);
           EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
               .Times(1)
               .After(expect_set);
           EXPECT_CALL(*channel_proxy_, DisassociateSendChannel()).Times(1);
           return channel_proxy_;
         }));
-    EXPECT_CALL(voice_engine_, StopPlayout(kChannelId)).WillOnce(Return(0));
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 300;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
     stream_config_.decoder_factory = decoder_factory_;
   }
 
   MockCongestionController* congestion_controller() {
     return &congestion_controller_;
   }
   MockRemoteBitrateEstimator* remote_bitrate_estimator() {
     return &remote_bitrate_estimator_;
   }
   MockRtcEventLog* event_log() { return &event_log_; }
   AudioReceiveStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
+  rtc::scoped_refptr<MockAudioMixer> audio_mixer() { return audio_mixer_; }
   MockVoiceEngine& voice_engine() { return voice_engine_; }
   MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
 
   void SetupMockForBweFeedback(bool send_side_bwe) {
     EXPECT_CALL(congestion_controller_,
                 GetRemoteBitrateEstimator(send_side_bwe))
         .WillOnce(Return(&remote_bitrate_estimator_));
     EXPECT_CALL(remote_bitrate_estimator_,
                 RemoveStream(stream_config_.rtp.remote_ssrc));
   }
@@ skipping 22 matching lines @@
   SimulatedClock simulated_clock_;
   PacketRouter packet_router_;
   testing::NiceMock<MockCongestionObserver> bitrate_observer_;
   testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
   MockCongestionController congestion_controller_;
   MockRemoteBitrateEstimator remote_bitrate_estimator_;
   MockRtcEventLog event_log_;
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
+  rtc::scoped_refptr<MockAudioMixer> audio_mixer_;
   AudioReceiveStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
 };
 
 void BuildOneByteExtension(std::vector<uint8_t>::iterator it,
                            int id,
                            uint32_t extension_value,
                            size_t value_length) {
   const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
   ByteWriter<uint16_t>::WriteBigEndian(&(*it), kRtpOneByteHeaderExtensionId);
@@ skipping 163 matching lines @@
 
 TEST(AudioReceiveStreamTest, SetGain) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.congestion_controller(), helper.config(), helper.audio_state(),
       helper.event_log());
   EXPECT_CALL(*helper.channel_proxy(),
       SetChannelOutputVolumeScaling(FloatEq(0.765f)));
   recv_stream.SetGain(0.765f);
 }
+
+TEST(AudioReceiveStreamTest, StreamShouldNotBeAddedToMixerWhenVoEReturnsError) {
+  ConfigHelper helper;
+  internal::AudioReceiveStream recv_stream(
+      helper.congestion_controller(), helper.config(), helper.audio_state(),
+      helper.event_log());
+
+  EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(-1));
+  EXPECT_CALL(*helper.audio_mixer(), AddSource(_)).Times(0);
+
+  recv_stream.Start();
+}
+
+TEST(AudioReceiveStreamTest, StreamShouldBeAddedToMixerOnStart) {
+  ConfigHelper helper;
+  internal::AudioReceiveStream recv_stream(
+      helper.congestion_controller(), helper.config(), helper.audio_state(),
+      helper.event_log());
+
+  EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(0));
+  EXPECT_CALL(helper.voice_engine(), StopPlayout(_));
+  EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream))
+      .WillOnce(Return(true));
+
+  recv_stream.Start();
+}
 }  // namespace test
 }  // namespace webrtc