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

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 {
 
 const int kChannelId = 1;
 const uint32_t kSsrc = 1234;
+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) {
+    EXPECT_CALL(voice_engine_,
+        RegisterVoiceEngineObserver(testing::_)).WillOnce(testing::Return(0));
+    EXPECT_CALL(voice_engine_,
+        DeRegisterVoiceEngineObserver()).WillOnce(testing::Return(0));
+    AudioState::Config config;
+    config.voice_engine = &voice_engine_;
+    audio_state_ = AudioState::Create(config);
+    stream_config_.voe_channel_id = kChannelId;
+    stream_config_.rtp.ssrc = kSsrc;
+  }
+
+  AudioSendStream::Config& config() { return stream_config_; }
+  rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
+
+  void SetupMockForGetStats() {
+    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_;
+  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.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}",
       config.ToString());
 }
 
 TEST(AudioSendStreamTest, ConstructDestruct) {
-  MockVoiceEngine voice_engine;
-  AudioSendStream::Config config(nullptr);
-  config.voe_channel_id = kChannelId;
-  internal::AudioSendStream send_stream(config, &voice_engine);
+  ConfigHelper helper;
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
 }
 
 TEST(AudioSendStreamTest, GetStats) {
-  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};
-
-  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.
-  }
-
-  MockVoiceEngine voice_engine;
-  AudioSendStream::Config config(nullptr);
-  config.rtp.ssrc = kSsrc;
-  config.voe_channel_id = kChannelId;
-  internal::AudioSendStream send_stream(config, &voice_engine);
-
-  using testing::_;
-  using testing::DoAll;
-  using testing::Return;
-  using testing::SetArgPointee;
-  using testing::SetArgReferee;
-  EXPECT_CALL(voice_engine, GetLocalSSRC(kChannelId, _))
-      .WillOnce(DoAll(SetArgReferee<1>(0), Return(0)));
-  EXPECT_CALL(voice_engine, GetRTCPStatistics(kChannelId, _))
-      .WillOnce(DoAll(SetArgReferee<1>(kCallStats), Return(0)));
-  EXPECT_CALL(voice_engine, GetSendCodec(kChannelId, _))
-      .WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
-  EXPECT_CALL(voice_engine, GetRemoteRTCPReportBlocks(kChannelId, _))
-      .WillOnce(DoAll(SetArgPointee<1>(report_blocks), Return(0)));
-  EXPECT_CALL(voice_engine, GetSpeechInputLevelFullRange(_))
-      .WillOnce(DoAll(SetArgReferee<0>(kSpeechInputLevel), Return(0)));
-  EXPECT_CALL(voice_engine, GetEcMetricsStatus(_))
-      .WillOnce(DoAll(SetArgReferee<0>(true), Return(0)));
-  EXPECT_CALL(voice_engine, GetEchoMetrics(_, _, _, _))
-      .WillOnce(DoAll(SetArgReferee<0>(kEchoReturnLoss),
-                      SetArgReferee<1>(kEchoReturnLossEnhancement), Return(0)));
-  EXPECT_CALL(voice_engine, GetEcDelayMetrics(_, _, _))
-      .WillOnce(DoAll(SetArgReferee<0>(kEchoDelayMedian),
-                      SetArgReferee<1>(kEchoDelayStdDev), Return(0)));
-
+  ConfigHelper helper;
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+  helper.SetupMockForGetStats();
   AudioSendStream::Stats stats = send_stream.GetStats();
   EXPECT_EQ(kSsrc, stats.local_ssrc);
   EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent);
   EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
   EXPECT_EQ(static_cast<int32_t>(kReportBlock.cumulative_num_packets_lost),
             stats.packets_lost);
   EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost);
   EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name);
   EXPECT_EQ(static_cast<int32_t>(kReportBlock.extended_highest_sequence_number),
             stats.ext_seqnum);
   EXPECT_EQ(static_cast<int32_t>(kReportBlock.interarrival_jitter /
                                  (kCodecInst.plfreq / 1000)),
             stats.jitter_ms);
   EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms);
   EXPECT_EQ(static_cast<int32_t>(kSpeechInputLevel), stats.audio_level);
   EXPECT_EQ(-1, stats.aec_quality_min);
   EXPECT_EQ(kEchoDelayMedian, stats.echo_delay_median_ms);
   EXPECT_EQ(kEchoDelayStdDev, stats.echo_delay_std_ms);
   EXPECT_EQ(kEchoReturnLoss, stats.echo_return_loss);
   EXPECT_EQ(kEchoReturnLossEnhancement, stats.echo_return_loss_enhancement);
   EXPECT_FALSE(stats.typing_noise_detected);
 }
+
+TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
+  ConfigHelper helper;
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+  helper.SetupMockForGetStats();
+  EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
+
+  internal::AudioState* internal_audio_state =
+      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