Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID.

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 "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
+#include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 using testing::_;
 using testing::Return;
 
 const int kChannelId = 1;
 const uint32_t kSsrc = 1234;
 const char* kCName = "foo_name";
 const int kAudioLevelId = 2;
 const int kAbsSendTimeId = 3;
 const int kEchoDelayMedian = 254;
 const int kEchoDelayStdDev = -3;
 const int kEchoReturnLoss = -65;
 const int kEchoReturnLossEnhancement = 101;
 const unsigned int kSpeechInputLevel = 96;
 const CallStatistics kCallStats = {
     1345,  1678,  1901, 1234,  112, 13456, 17890, 1567, -1890, -1123};
 const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, -451, -671};
 const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
 
 struct ConfigHelper {
   ConfigHelper() : stream_config_(nullptr) {
-    using testing::StrEq;
+    using testing::Invoke;
 
     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
     AudioState::Config config;
     config.voice_engine = &voice_engine_;
     audio_state_ = AudioState::Create(config);
 
-    EXPECT_CALL(voice_engine_, SetRTCPStatus(kChannelId, true))
-        .WillOnce(Return(0));
-    EXPECT_CALL(voice_engine_, SetLocalSSRC(kChannelId, kSsrc))
-        .WillOnce(Return(0));
-    EXPECT_CALL(voice_engine_, SetRTCP_CNAME(kChannelId, StrEq(kCName)))
-        .WillOnce(Return(0));
+    EXPECT_CALL(voice_engine_, GetChannelProxy(kChannelId))
+        .WillOnce(Invoke(this, &ConfigHelper::CreateChannelProxy));

[kwiberg-webrtc, 2015/11/25 10:44:59]

See earlier comment about lambda and whether channel_proxy_ is needed.

[the sun, 2015/11/25 12:28:01]

channel_proxy_ *will* be needed. Once the methods for obtaining stats are added
to ChannelProxy, the below method SetupMockForGetStats() will start adding
expectations on the channel_proxy_ mock.

     EXPECT_CALL(voice_engine_,
         SetSendAbsoluteSenderTimeStatus(kChannelId, true, kAbsSendTimeId))
             .WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         SetSendAudioLevelIndicationStatus(kChannelId, true, kAudioLevelId))
             .WillOnce(Return(0));
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.ssrc = kSsrc;
     stream_config_.rtp.c_name = kCName;
     stream_config_.rtp.extensions.push_back(
@@ skipping 31 matching lines @@
     EXPECT_CALL(voice_engine_, GetEchoMetrics(_, _, _, _))
         .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoReturnLoss),
                         SetArgReferee<1>(kEchoReturnLossEnhancement),
                         Return(0)));
     EXPECT_CALL(voice_engine_, GetEcDelayMetrics(_, _, _))
         .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoDelayMedian),
                         SetArgReferee<1>(kEchoDelayStdDev), Return(0)));
   }
 
  private:
+  voe::ChannelProxy* CreateChannelProxy(int channel_id) {
+    using testing::StrEq;
+
+    EXPECT_FALSE(channel_proxy_);
+    channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
+    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);
+    return channel_proxy_;
+  }
+
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
+  testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
 };
 }  // namespace
 
 TEST(AudioSendStreamTest, ConfigToString) {
   AudioSendStream::Config config(nullptr);
   config.rtp.ssrc = kSsrc;
   config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
   config.rtp.c_name = kCName;
   config.voe_channel_id = kChannelId;
@@ skipping 49 matching lines @@
       static_cast<internal::AudioState*>(helper.audio_state().get());
   VoiceEngineObserver* voe_observer =
       static_cast<VoiceEngineObserver*>(internal_audio_state);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
   EXPECT_TRUE(send_stream.GetStats().typing_noise_detected);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
   EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
 }
 }  // namespace test
 }  // namespace webrtc