Refactor WVoE DTMF handling #2

talk/media/webrtc/webrtcvoiceengine_unittest.cc

 /*
  * libjingle
  * Copyright 2008 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 46 matching lines @@
 const uint32_t kSsrc1 = 0x99;
 const uint32_t kSsrc2 = 0x98;
 const uint32_t kSsrcs4[] = { 1, 2, 3, 4 };
 
 class FakeVoEWrapper : public cricket::VoEWrapper {
  public:
   explicit FakeVoEWrapper(cricket::FakeWebRtcVoiceEngine* engine)
       : cricket::VoEWrapper(engine,  // processing
                             engine,  // base
                             engine,  // codec
-                            engine,  // dtmf
                             engine,  // hw
                             engine,  // network
                             engine,  // rtp
                             engine) {  // volume
   }
 };
 }  // namespace
 
 class WebRtcVoiceEngineTestFake : public testing::Test {
  public:
@@ skipping 36 matching lines @@
   }
   void DeliverPacket(const void* data, int len) {
     rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len);
     channel_->OnPacketReceived(&packet, rtc::PacketTime());
   }
   void TearDown() override {
     delete channel_;
     engine_.Terminate();
   }
 
+  const cricket::FakeAudioSendStream& GetSendStream(uint32_t ssrc) {
+    const auto* send_stream = call_.GetAudioSendStream(ssrc);
+    EXPECT_TRUE(send_stream);
+    return *send_stream;
+  }
+
   const webrtc::AudioSendStream::Config& GetSendStreamConfig(uint32_t ssrc) {
     const auto* send_stream = call_.GetAudioSendStream(ssrc);
     EXPECT_TRUE(send_stream);
     return send_stream->GetConfig();
   }
 
   const webrtc::AudioReceiveStream::Config& GetRecvStreamConfig(uint32_t ssrc) {
     const auto* recv_stream = call_.GetAudioReceiveStream(ssrc);
     EXPECT_TRUE(recv_stream);
     return recv_stream->GetConfig();
@@ skipping 22 matching lines @@
     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
-    int channel_id = voe_.GetLastChannel();
-    EXPECT_FALSE(voe_.WasSendTelephoneEventCalled(channel_id, 2, 123));
+    // Test send.
+    cricket::FakeAudioSendStream::TelephoneEvent telephone_event =
+        GetSendStream(kSsrc1).GetLatestTelephoneEvent();
+    EXPECT_EQ(-1, telephone_event.payload_type);
     EXPECT_TRUE(channel_->InsertDtmf(ssrc, 2, 123));
-    EXPECT_TRUE(voe_.WasSendTelephoneEventCalled(channel_id, 2, 123));
+    telephone_event = GetSendStream(kSsrc1).GetLatestTelephoneEvent();
+    EXPECT_EQ(kTelephoneEventCodec.id, telephone_event.payload_type);
+    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 TestSendBandwidth(const cricket::AudioCodec& codec,
                          int max_bitrate,
                          bool expected_result,
@@ skipping 578 matching lines @@
   int channel_num = voe_.GetLastChannel();
   webrtc::CodecInst gcodec;
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_EQ(48000, gcodec.rate);
   EXPECT_STREQ("ISAC", gcodec.plname);
   EXPECT_FALSE(voe_.GetVAD(channel_num));
   EXPECT_FALSE(voe_.GetRED(channel_num));
   EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
   EXPECT_EQ(105, voe_.GetSendCNPayloadType(channel_num, true));
-  EXPECT_EQ(106, voe_.GetSendTelephoneEventPayloadType(channel_num));
+  EXPECT_FALSE(channel_->CanInsertDtmf());
 }
 
 // Test that VoE Channel doesn't call SetSendCodec again if same codec is tried
 // to apply.
 TEST_F(WebRtcVoiceEngineTestFake, DontResetSetSendCodec) {
   EXPECT_TRUE(SetupEngineWithSendStream());
   cricket::AudioSendParameters parameters;
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs.push_back(kPcmuCodec);
   parameters.codecs.push_back(kRedCodec);
@@ skipping 820 matching lines @@
   parameters.codecs.push_back(kTelephoneEventCodec);
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs.push_back(kPcmuCodec);
   parameters.codecs[0].id = 98;  // DTMF
   parameters.codecs[1].id = 96;
   EXPECT_TRUE(channel_->SetSendParameters(parameters));
   webrtc::CodecInst gcodec;
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
-  EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
+  EXPECT_TRUE(channel_->CanInsertDtmf());
 }
 
 // Test that we can set send codecs even with CN codec as the first
 // one on the list.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNOnTop) {
   EXPECT_TRUE(SetupEngineWithSendStream());
   int channel_num = voe_.GetLastChannel();
   cricket::AudioSendParameters parameters;
   parameters.codecs.push_back(kCn16000Codec);
   parameters.codecs.push_back(kIsacCodec);
@@ skipping 25 matching lines @@
   parameters.codecs[4].id = 98;  // DTMF
   EXPECT_TRUE(channel_->SetSendParameters(parameters));
   webrtc::CodecInst gcodec;
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
   EXPECT_TRUE(voe_.GetVAD(channel_num));
   EXPECT_FALSE(voe_.GetRED(channel_num));
   EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
   EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
-  EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
+  EXPECT_TRUE(channel_->CanInsertDtmf());
 }
 
 // Test that we set VAD and DTMF types correctly as callee.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
   EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
   channel_ = engine_.CreateChannel(&call_, cricket::AudioOptions());
   EXPECT_TRUE(channel_ != nullptr);
 
   cricket::AudioSendParameters parameters;
   parameters.codecs.push_back(kIsacCodec);
@@ skipping 12 matching lines @@
   int channel_num = voe_.GetLastChannel();
 
   webrtc::CodecInst gcodec;
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
   EXPECT_TRUE(voe_.GetVAD(channel_num));
   EXPECT_FALSE(voe_.GetRED(channel_num));
   EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
   EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
-  EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
+  EXPECT_TRUE(channel_->CanInsertDtmf());
 }
 
 // Test that we only apply VAD if we have a CN codec that matches the
 // send codec clockrate.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) {
   EXPECT_TRUE(SetupEngineWithSendStream());
   int channel_num = voe_.GetLastChannel();
   cricket::AudioSendParameters parameters;
   // Set ISAC(16K) and CN(16K). VAD should be activated.
   parameters.codecs.push_back(kIsacCodec);
@@ skipping 43 matching lines @@
   parameters.codecs[4].id = 98;  // DTMF
   EXPECT_TRUE(channel_->SetSendParameters(parameters));
   webrtc::CodecInst gcodec;
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
   EXPECT_TRUE(voe_.GetVAD(channel_num));
   EXPECT_FALSE(voe_.GetRED(channel_num));
   EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
   EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
-  EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
+  EXPECT_TRUE(channel_->CanInsertDtmf());
 }
 
 // Test that we set up RED correctly as caller.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDAsCaller) {
   EXPECT_TRUE(SetupEngineWithSendStream());
   int channel_num = voe_.GetLastChannel();
   cricket::AudioSendParameters parameters;
   parameters.codecs.push_back(kRedCodec);
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs.push_back(kPcmuCodec);
@@ skipping 1579 matching lines @@
   cricket::WebRtcVoiceEngine engine;
   EXPECT_TRUE(engine.Init(rtc::Thread::Current()));
   rtc::scoped_ptr<webrtc::Call> call(
       webrtc::Call::Create(webrtc::Call::Config()));
   cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::AudioOptions(),
                                            call.get());
   cricket::AudioRecvParameters parameters;
   parameters.codecs = engine.codecs();
   EXPECT_TRUE(channel.SetRecvParameters(parameters));
 }