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/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;
   AudioSendStream::Config config(nullptr);
   config.rtp.ssrc = 1234;
   config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
   config.voe_channel_id = 1;
   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) {
-  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;
   const CodecInst& codec_inst = FakeVoiceEngine::kSendCodecInst;
   const ReportBlock& report_block = FakeVoiceEngine::kSendReportBlock;
   EXPECT_EQ(FakeVoiceEngine::kSendSsrc, stats.local_ssrc);
   EXPECT_EQ(static_cast<int64_t>(call_stats.bytesSent), stats.bytes_sent);
   EXPECT_EQ(call_stats.packetsSent, stats.packets_sent);
   EXPECT_EQ(static_cast<int32_t>(report_block.cumulative_num_packets_lost),
             stats.packets_lost);
   EXPECT_EQ(Q8ToFloat(report_block.fraction_lost), stats.fraction_lost);
   EXPECT_EQ(std::string(codec_inst.plname), stats.codec_name);
   EXPECT_EQ(static_cast<int32_t>(report_block.extended_highest_sequence_number),
             stats.ext_seqnum);
   EXPECT_EQ(static_cast<int32_t>(report_block.interarrival_jitter /
                 (codec_inst.plfreq / 1000)), stats.jitter_ms);
   EXPECT_EQ(call_stats.rttMs, stats.rtt_ms);
   EXPECT_EQ(static_cast<int32_t>(FakeVoiceEngine::kSendSpeechInputLevel),
             stats.audio_level);
   EXPECT_EQ(-1, stats.aec_quality_min);
   EXPECT_EQ(FakeVoiceEngine::kSendEchoDelayMedian, stats.echo_delay_median_ms);
   EXPECT_EQ(FakeVoiceEngine::kSendEchoDelayStdDev, stats.echo_delay_std_ms);
   EXPECT_EQ(FakeVoiceEngine::kSendEchoReturnLoss, stats.echo_return_loss);
   EXPECT_EQ(FakeVoiceEngine::kSendEchoReturnLossEnhancement,
             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