Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i…

webrtc/audio/audio_receive_stream_unittest.cc

Index: webrtc/audio/audio_receive_stream_unittest.cc
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 4e267f17389fa9fbc45e6e122f915fe035c6043c..1c6b0eeb129f09c1ad6b073a4ccde56a1d2d7ecc 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -11,7 +11,9 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/audio/audio_receive_stream.h"
+#include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
+#include "webrtc/base/scoped_ptr.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/test/fake_voice_engine.h"
@@ -56,6 +58,30 @@ size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header,
   rtp_header_length += kAbsoluteSendTimeLength;
   return rtp_header_length;
 }
+
+struct ConfigHelper {
+  ConfigHelper() {
+    webrtc::AudioState::Config config;
+    config.voice_engine = &voice_engine_;
+    audio_state_.reset(new webrtc::internal::AudioState(config));
+  }
+
+  webrtc::MockRemoteBitrateEstimator* remote_bitrate_estimator() {
+    return &remote_bitrate_estimator_;
+  }
+  webrtc::AudioReceiveStream::Config& config() {
+    return stream_config_;
+  }
+  webrtc::internal::AudioState* audio_state() {
+    return audio_state_.get();
+  }
+
+ private:
+  webrtc::MockRemoteBitrateEstimator remote_bitrate_estimator_;
+  webrtc::test::FakeVoiceEngine voice_engine_;
+  rtc::scoped_ptr<webrtc::internal::AudioState> audio_state_;
+  webrtc::AudioReceiveStream::Config stream_config_;
+};
 }  // namespace
 
 namespace webrtc {
@@ -77,31 +103,27 @@ TEST(AudioReceiveStreamTest, ConfigToString) {
 }
 
 TEST(AudioReceiveStreamTest, ConstructDestruct) {
-  MockRemoteBitrateEstimator remote_bitrate_estimator;
-  FakeVoiceEngine voice_engine;
-  AudioReceiveStream::Config config;
-  config.voe_channel_id = 1;
-  internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
-                                           &voice_engine);
+  ConfigHelper helper;
+  helper.config().voe_channel_id = 1;
+  internal::AudioReceiveStream recv_stream(
+      helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
 }
 
 TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
-  MockRemoteBitrateEstimator remote_bitrate_estimator;
-  FakeVoiceEngine voice_engine;
-  AudioReceiveStream::Config config;
-  config.combined_audio_video_bwe = true;
-  config.voe_channel_id = FakeVoiceEngine::kRecvChannelId;
+  ConfigHelper helper;
+  helper.config().combined_audio_video_bwe = true;
+  helper.config().voe_channel_id = FakeVoiceEngine::kRecvChannelId;
   const int kAbsSendTimeId = 3;
-  config.rtp.extensions.push_back(
+  helper.config().rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
-  internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
-                                           &voice_engine);
+  internal::AudioReceiveStream recv_stream(
+      helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
   uint8_t rtp_packet[30];
   const int kAbsSendTimeValue = 1234;
   CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
   PacketTime packet_time(5678000, 0);
   const size_t kExpectedHeaderLength = 20;
-  EXPECT_CALL(remote_bitrate_estimator,
+  EXPECT_CALL(*helper.remote_bitrate_estimator(),
       IncomingPacket(packet_time.timestamp / 1000,
           sizeof(rtp_packet) - kExpectedHeaderLength, testing::_, false))
       .Times(1);
@@ -110,13 +132,11 @@ TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
 }
 
 TEST(AudioReceiveStreamTest, GetStats) {
-  MockRemoteBitrateEstimator remote_bitrate_estimator;
-  FakeVoiceEngine voice_engine;
-  AudioReceiveStream::Config config;
-  config.rtp.remote_ssrc = FakeVoiceEngine::kRecvSsrc;
-  config.voe_channel_id = FakeVoiceEngine::kRecvChannelId;
-  internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
-                                           &voice_engine);
+  ConfigHelper helper;
+  helper.config().rtp.remote_ssrc = FakeVoiceEngine::kRecvSsrc;
+  helper.config().voe_channel_id = FakeVoiceEngine::kRecvChannelId;
+  internal::AudioReceiveStream recv_stream(
+      helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
 
   AudioReceiveStream::Stats stats = recv_stream.GetStats();
   const CallStatistics& call_stats = FakeVoiceEngine::kRecvCallStats;