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

webrtc/audio/audio_send_stream_unittest.cc

Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 227ec8379971f3e6c2ad73b52acf75872e27a08a..592509aebf2964ffb22edd09ddd6118c61add660 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -11,11 +11,35 @@
 #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/base/scoped_ptr.h"
 #include "webrtc/test/fake_voice_engine.h"
 
 namespace webrtc {
 namespace test {
+namespace {
+
+struct ConfigHelper {
+  ConfigHelper() : stream_config_(nullptr) {
+    AudioState::Config config;
+    config.voice_engine = &voice_engine_;
+    audio_state_.reset(new internal::AudioState(config));
+  }
+
+  AudioSendStream::Config& config() {
+    return stream_config_;
+  }
+  internal::AudioState* audio_state() {
+    return audio_state_.get();
+  }
+
+ private:
+  FakeVoiceEngine voice_engine_;
+  rtc::scoped_ptr<internal::AudioState> audio_state_;
+  AudioSendStream::Config stream_config_;
+};
+}  // namespace
 
 TEST(AudioSendStreamTest, ConfigToString) {
   const int kAbsSendTimeId = 3;
@@ -33,18 +57,16 @@ TEST(AudioSendStreamTest, ConfigToString) {
 }
 
 TEST(AudioSendStreamTest, ConstructDestruct) {
-  FakeVoiceEngine voice_engine;
-  AudioSendStream::Config config(nullptr);
-  config.voe_channel_id = 1;
-  internal::AudioSendStream send_stream(config, &voice_engine);
+  ConfigHelper helper;
+  helper.config().voe_channel_id = 1;
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
 }
 
 TEST(AudioSendStreamTest, GetStats) {
-  FakeVoiceEngine voice_engine;
-  AudioSendStream::Config config(nullptr);
-  config.rtp.ssrc = FakeVoiceEngine::kSendSsrc;
-  config.voe_channel_id = FakeVoiceEngine::kSendChannelId;
-  internal::AudioSendStream send_stream(config, &voice_engine);
+  ConfigHelper helper;
+  helper.config().rtp.ssrc = FakeVoiceEngine::kSendSsrc;
+  helper.config().voe_channel_id = FakeVoiceEngine::kSendChannelId;
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
 
   AudioSendStream::Stats stats = send_stream.GetStats();
   const CallStatistics& call_stats = FakeVoiceEngine::kSendCallStats;
@@ -72,5 +94,18 @@ TEST(AudioSendStreamTest, GetStats) {
             stats.echo_return_loss_enhancement);
   EXPECT_FALSE(stats.typing_noise_detected);
 }
+
+TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
+  ConfigHelper helper;
+  helper.config().voe_channel_id = 1;
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+  EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
+
+  internal::AudioState* audio_state = helper.audio_state();
+  audio_state->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
+  EXPECT_TRUE(send_stream.GetStats().typing_noise_detected);
+  audio_state->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
+  EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
+}
 }  // namespace test
 }  // namespace webrtc