Multi frequency DTMF support - sender side

webrtc/media/engine/webrtcvoiceengine_unittest.cc

 /*
  *  Copyright (c) 2008 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 203 matching lines @@
                     const cricket::AudioOptions* options = nullptr) {
     EXPECT_CALL(apm_, set_output_will_be_muted(!enable));
     ASSERT_TRUE(channel_);
     if (enable && options) {
       EXPECT_CALL(apm_, ApplyConfig(testing::_));
       EXPECT_CALL(apm_, SetExtraOptions(testing::_));
     }
     EXPECT_TRUE(channel_->SetAudioSend(ssrc, enable, options, source));
   }
 
-  void TestInsertDtmf(uint32_t ssrc, bool caller) {
+  void TestInsertDtmf(uint32_t ssrc, bool caller,
+                      const cricket::AudioCodec& codec) {
     EXPECT_TRUE(SetupChannel());
     if (caller) {
       // If this is a caller, local description will be applied and add the
       // send stream.
       EXPECT_TRUE(channel_->AddSendStream(
           cricket::StreamParams::CreateLegacy(kSsrc1)));
     }
 
     // Test we can only InsertDtmf when the other side supports telephone-event.
     SetSendParameters(send_parameters_);
     SetSend(true);
     EXPECT_FALSE(channel_->CanInsertDtmf());
     EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111));
-    send_parameters_.codecs.push_back(kTelephoneEventCodec1);
+    send_parameters_.codecs.push_back(codec);
     SetSendParameters(send_parameters_);
     EXPECT_TRUE(channel_->CanInsertDtmf());
 
     if (!caller) {
       // If this is callee, there's no active send channel yet.
       EXPECT_FALSE(channel_->InsertDtmf(ssrc, 2, 123));
       EXPECT_TRUE(channel_->AddSendStream(
           cricket::StreamParams::CreateLegacy(kSsrc1)));
     }
 
     // Check we fail if the ssrc is invalid.
     EXPECT_FALSE(channel_->InsertDtmf(-1, 1, 111));
 
     // Test send.
     cricket::FakeAudioSendStream::TelephoneEvent telephone_event =
         GetSendStream(kSsrc1).GetLatestTelephoneEvent();
     EXPECT_EQ(-1, telephone_event.payload_type);
     EXPECT_TRUE(channel_->InsertDtmf(ssrc, 2, 123));
     telephone_event = GetSendStream(kSsrc1).GetLatestTelephoneEvent();
-    EXPECT_EQ(kTelephoneEventCodec1.id, telephone_event.payload_type);
+    EXPECT_EQ(codec.id, telephone_event.payload_type);
+    EXPECT_EQ(codec.clockrate, telephone_event.payload_frequency);
     EXPECT_EQ(2, telephone_event.event_code);
     EXPECT_EQ(123, telephone_event.duration_ms);
   }
 
   // Test that send bandwidth is set correctly.
   // |codec| is the codec under test.
   // |max_bitrate| is a parameter to set to SetMaxSendBandwidth().
   // |expected_result| is the expected result from SetMaxSendBandwidth().
   // |expected_bitrate| is the expected audio bitrate afterward.
   void TestMaxSendBandwidth(const cricket::AudioCodec& codec,
@@ skipping 1608 matching lines @@
   const auto& gcodec = GetSendStreamConfig(kSsrc1).send_codec_spec.codec_inst;
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
   EXPECT_TRUE(channel_->CanInsertDtmf());
 }
 
 // Test that payload type range is limited for telephone-event codec.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) {
   EXPECT_TRUE(SetupSendStream());
   cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kTelephoneEventCodec1);
+  parameters.codecs.push_back(kTelephoneEventCodec2);
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs[0].id = 0;  // DTMF
   parameters.codecs[1].id = 96;
   SetSendParameters(parameters);
   EXPECT_TRUE(channel_->CanInsertDtmf());
   parameters.codecs[0].id = 128;  // DTMF
   EXPECT_FALSE(channel_->SetSendParameters(parameters));
   EXPECT_FALSE(channel_->CanInsertDtmf());
   parameters.codecs[0].id = 127;
   SetSendParameters(parameters);
@@ skipping 47 matching lines @@
 
 // Test that we set VAD and DTMF types correctly as callee.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
   EXPECT_TRUE(SetupChannel());
   cricket::AudioSendParameters parameters;
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs.push_back(kPcmuCodec);
   // TODO(juberti): cn 32000
   parameters.codecs.push_back(kCn16000Codec);
   parameters.codecs.push_back(kCn8000Codec);
-  parameters.codecs.push_back(kTelephoneEventCodec1);
+  parameters.codecs.push_back(kTelephoneEventCodec2);
   parameters.codecs[0].id = 96;
   parameters.codecs[2].id = 97;  // wideband CN
   parameters.codecs[4].id = 98;  // DTMF
   SetSendParameters(parameters);
   EXPECT_TRUE(channel_->AddSendStream(
       cricket::StreamParams::CreateLegacy(kSsrc1)));
 
   const auto& send_codec_spec = GetSendStreamConfig(kSsrc1).send_codec_spec;
   EXPECT_EQ(96, send_codec_spec.codec_inst.pltype);
   EXPECT_STREQ("ISAC", send_codec_spec.codec_inst.plname);
@@ skipping 759 matching lines @@
   // Add recv stream 2 should work.
   EXPECT_TRUE(AddRecvStream(2));
   int new_channel = voe_.GetLastChannel();
   EXPECT_NE(channel, new_channel);
   // The last created channel is deleted too.
   EXPECT_EQ(0, voe_.DeleteChannel(new_channel));
 }
 
 // Test the InsertDtmf on default send stream as caller.
 TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCaller) {
-  TestInsertDtmf(0, true);
+  TestInsertDtmf(0, true, kTelephoneEventCodec1);
 }
 
 // Test the InsertDtmf on default send stream as callee
 TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCallee) {
-  TestInsertDtmf(0, false);
+  TestInsertDtmf(0, false, kTelephoneEventCodec2);
 }
 
 // Test the InsertDtmf on specified send stream as caller.
 TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCaller) {
-  TestInsertDtmf(kSsrc1, true);
+  TestInsertDtmf(kSsrc1, true, kTelephoneEventCodec2);
 }
 
 // Test the InsertDtmf on specified send stream as callee.
 TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) {
-  TestInsertDtmf(kSsrc1, false);
+  TestInsertDtmf(kSsrc1, false, kTelephoneEventCodec1);
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
   EXPECT_TRUE(SetupSendStream());
   EXPECT_CALL(adm_,
               BuiltInAECIsAvailable()).Times(9).WillRepeatedly(Return(false));
   EXPECT_CALL(adm_,
               BuiltInAGCIsAvailable()).Times(4).WillRepeatedly(Return(false));
   EXPECT_CALL(adm_,
               BuiltInNSIsAvailable()).Times(2).WillRepeatedly(Return(false));
@@ skipping 768 matching lines @@
       nullptr, webrtc::CreateBuiltinAudioDecoderFactory());
   webrtc::RtcEventLogNullImpl event_log;
   std::unique_ptr<webrtc::Call> call(
       webrtc::Call::Create(webrtc::Call::Config(&event_log)));
   cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
                                            cricket::AudioOptions(), call.get());
   cricket::AudioRecvParameters parameters;
   parameters.codecs = engine.recv_codecs();
   EXPECT_TRUE(channel.SetRecvParameters(parameters));
 }