Multi frequency DTMF support - sender side

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 29 matching lines @@
 const int kEchoDelayMedian = 254;
 const int kEchoDelayStdDev = -3;
 const int kEchoReturnLoss = -65;
 const int kEchoReturnLossEnhancement = 101;
 const float kResidualEchoLikelihood = -1.0f;
 const unsigned int kSpeechInputLevel = 96;
 const CallStatistics kCallStats = {
     1345,  1678,  1901, 1234,  112, 13456, 17890, 1567, -1890, -1123};
 const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
 const int kTelephoneEventPayloadType = 123;
+const int kTelephoneEventPayloadFrequency = 65432;
 const int kTelephoneEventCode = 45;
 const int kTelephoneEventDuration = 6789;
 const CodecInst kIsacCodec = {103, "isac", 16000, 320, 1, 32000};
 
 class MockLimitObserver : public BitrateAllocator::LimitObserver {
  public:
   MOCK_METHOD2(OnAllocationLimitsChanged,
                void(uint32_t min_send_bitrate_bps,
                     uint32_t max_padding_bitrate_bps));
 };
@@ skipping 87 matching lines @@
     // Let |GetSendCodec| return -1 for the first time to indicate that no send
     // codec has been set.
     EXPECT_CALL(voice_engine_, GetSendCodec(kChannelId, _))
         .WillOnce(Return(-1));
     EXPECT_CALL(voice_engine_, SetSendCodec(kChannelId, _)).WillOnce(Return(0));
   }
 
   void SetupMockForSendTelephoneEvent() {
     EXPECT_TRUE(channel_proxy_);
     EXPECT_CALL(*channel_proxy_,
-        SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType))
+        SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType,
+                                         kTelephoneEventPayloadFrequency))
             .WillOnce(Return(true));
     EXPECT_CALL(*channel_proxy_,
         SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
             .WillOnce(Return(true));
   }
 
   void SetupMockForGetStats() {
     using testing::DoAll;
     using testing::SetArgReferee;
 
@@ skipping 93 matching lines @@
 }
 
 TEST(AudioSendStreamTest, SendTelephoneEvent) {
   ConfigHelper helper;
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
       helper.event_log());
   helper.SetupMockForSendTelephoneEvent();
   EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
-      kTelephoneEventCode, kTelephoneEventDuration));
+      kTelephoneEventPayloadFrequency, kTelephoneEventCode,
+      kTelephoneEventDuration));
 }
 
 TEST(AudioSendStreamTest, SetMuted) {
   ConfigHelper helper;
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
       helper.event_log());
   EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
   send_stream.SetMuted(true);
@@ skipping 104 matching lines @@
   EXPECT_CALL(*helper.voice_engine(), SetVADStatus(kChannelId, true, _, _))
       .WillOnce(Return(0));
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
       helper.event_log());
 }
 
 }  // namespace test
 }  // namespace webrtc