Move some send stream configuration into webrtc::AudioSendStream.

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_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;
+
     EXPECT_CALL(voice_engine_,
-        RegisterVoiceEngineObserver(testing::_)).WillOnce(testing::Return(0));
+        RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
-        DeRegisterVoiceEngineObserver()).WillOnce(testing::Return(0));
+        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_,
+        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(
+        RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
+    stream_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
 
   void SetupMockForGetStats() {
+    using testing::DoAll;
+    using testing::SetArgPointee;
+    using testing::SetArgReferee;
+
     std::vector<ReportBlock> report_blocks;
     webrtc::ReportBlock block = kReportBlock;
     report_blocks.push_back(block);  // Has wrong SSRC.
     block.source_SSRC = kSsrc;
     report_blocks.push_back(block);  // Correct block.
     block.fraction_lost = 0;
     report_blocks.push_back(block);  // Duplicate SSRC, bad fraction_lost.
 
-    using testing::_;
-    using testing::DoAll;
-    using testing::Return;
-    using testing::SetArgPointee;
-    using testing::SetArgReferee;
     EXPECT_CALL(voice_engine_, GetLocalSSRC(kChannelId, _))
         .WillRepeatedly(DoAll(SetArgReferee<1>(0), Return(0)));
     EXPECT_CALL(voice_engine_, GetRTCPStatistics(kChannelId, _))
         .WillRepeatedly(DoAll(SetArgReferee<1>(kCallStats), Return(0)));
     EXPECT_CALL(voice_engine_, GetSendCodec(kChannelId, _))
         .WillRepeatedly(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
     EXPECT_CALL(voice_engine_, GetRemoteRTCPReportBlocks(kChannelId, _))
         .WillRepeatedly(DoAll(SetArgPointee<1>(report_blocks), Return(0)));
     EXPECT_CALL(voice_engine_, GetSpeechInputLevelFullRange(_))
         .WillRepeatedly(DoAll(SetArgReferee<0>(kSpeechInputLevel), Return(0)));
     EXPECT_CALL(voice_engine_, GetEcMetricsStatus(_))
         .WillRepeatedly(DoAll(SetArgReferee<0>(true), Return(0)));
     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:
-  MockVoiceEngine voice_engine_;
+  testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
 };
 }  // namespace
 
 TEST(AudioSendStreamTest, ConfigToString) {
-  const int kAbsSendTimeId = 3;
   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;
   config.cng_payload_type = 42;
   config.red_payload_type = 17;
   EXPECT_EQ(
       "{rtp: {ssrc: 1234, extensions: [{name: "
-      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}]}, "
-      "voe_channel_id: 1, cng_payload_type: 42, red_payload_type: 17}",
+      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
+      "c_name: foo_name}, voe_channel_id: 1, cng_payload_type: 42, "
+      "red_payload_type: 17}",
       config.ToString());
 }
 
 TEST(AudioSendStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
 }
 
 TEST(AudioSendStreamTest, GetStats) {
   ConfigHelper helper;
@@ skipping 32 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