Refactor WVoE DTMF handling #2

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.
  */
 
@@ skipping 22 matching lines @@
 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};
+const int kTelephoneEventPayloadType = 123;
+const uint8_t kTelephoneEventCode = 45;
+const uint32_t kTelephoneEventDuration = 6789;
 
 struct ConfigHelper {
   ConfigHelper() : stream_config_(nullptr) {
     using testing::Invoke;
     using testing::StrEq;
 
     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
@@ skipping 19 matching lines @@
     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 SetupMockForSendTelephoneEvent() {
+    EXPECT_TRUE(channel_proxy_);
+    EXPECT_CALL(*channel_proxy_,
+        SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType))
+            .WillOnce(Return(true));
+    EXPECT_CALL(*channel_proxy_,
+        SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
+            .WillOnce(Return(true));
+  }
+
   void SetupMockForGetStats() {
     using testing::DoAll;
     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;
@@ skipping 43 matching lines @@
       "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, SendTelephoneEvent) {
+  ConfigHelper helper;
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+  helper.SetupMockForSendTelephoneEvent();
+  EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
+      kTelephoneEventCode, kTelephoneEventDuration));
+}
+
 TEST(AudioSendStreamTest, GetStats) {
   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);
@@ skipping 24 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