Multi frequency DTMF support - receiver 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 23 matching lines @@
 
 namespace {
 
 const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1);
 const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1);
 const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 64000, 2);
 const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1);
 const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1);
 const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1);
 const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1);
-const cricket::AudioCodec kTelephoneEventCodec(106,
-                                               "telephone-event",
-                                               8000,
-                                               0,
-                                               1);
+const cricket::AudioCodec
+    kTelephoneEventCodec1(106, "telephone-event", 8000, 0, 1);
+const cricket::AudioCodec
+    kTelephoneEventCodec2(107, "telephone-event", 32000, 0, 1);
+
 const uint32_t kSsrc1 = 0x99;
 const uint32_t kSsrc2 = 2;
 const uint32_t kSsrc3 = 3;
 const uint32_t kSsrcs4[] = { 1, 2, 3, 4 };
 
 constexpr int kRtpHistoryMs = 5000;
 
 class FakeVoEWrapper : public cricket::VoEWrapper {
  public:
   explicit FakeVoEWrapper(cricket::FakeWebRtcVoiceEngine* engine)
@@ skipping 137 matching lines @@
       // send stream.
       EXPECT_TRUE(channel_->AddSendStream(
           cricket::StreamParams::CreateLegacy(kSsrc1)));
     }
 
     // Test we can only InsertDtmf when the other side supports telephone-event.
     EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
     SetSend(channel_, true);
     EXPECT_FALSE(channel_->CanInsertDtmf());
     EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111));
-    send_parameters_.codecs.push_back(kTelephoneEventCodec);
+    send_parameters_.codecs.push_back(kTelephoneEventCodec1);
     EXPECT_TRUE(channel_->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(kTelephoneEventCodec.id, telephone_event.payload_type);
+    EXPECT_EQ(kTelephoneEventCodec1.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 TestMaxSendBandwidth(const cricket::AudioCodec& codec,
@@ skipping 331 matching lines @@
 // Tests that we can find codecs by name or id, and that we interpret the
 // clockrate and bitrate fields properly.
 TEST_F(WebRtcVoiceEngineTestFake, FindCodec) {
   cricket::AudioCodec codec;
   webrtc::CodecInst codec_inst;
   // Find PCMU with explicit clockrate and bitrate.
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kPcmuCodec, &codec_inst));
   // Find ISAC with explicit clockrate and 0 bitrate.
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kIsacCodec, &codec_inst));
   // Find telephone-event with explicit clockrate and 0 bitrate.
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kTelephoneEventCodec,
+  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kTelephoneEventCodec1,
+                                                      &codec_inst));
+  // Find telephone-event with explicit clockrate and 0 bitrate.
+  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kTelephoneEventCodec2,
                                                       &codec_inst));
   // Find ISAC with a different payload id.
   codec = kIsacCodec;
   codec.id = 127;
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst));
   EXPECT_EQ(codec.id, codec_inst.pltype);
   // Find PCMU with a 0 clockrate.
   codec = kPcmuCodec;
   codec.clockrate = 0;
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst));
@@ skipping 12 matching lines @@
   EXPECT_EQ(codec.id, codec_inst.pltype);
   EXPECT_EQ(32000, codec_inst.rate);
 }
 
 // Test that we set our inbound codecs properly, including changing PT.
 TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecs) {
   EXPECT_TRUE(SetupChannel());
   cricket::AudioRecvParameters parameters;
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs.push_back(kPcmuCodec);
-  parameters.codecs.push_back(kTelephoneEventCodec);
-  parameters.codecs[0].id = 106;  // collide with existing telephone-event
+  parameters.codecs.push_back(kTelephoneEventCodec1);
+  parameters.codecs.push_back(kTelephoneEventCodec2);
+  parameters.codecs[0].id = 106;  // collide with existing CN 32k
   parameters.codecs[2].id = 126;
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(AddRecvStream(kSsrc1));
   int channel_num = voe_.GetLastChannel();
+
   webrtc::CodecInst gcodec;
   rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
   gcodec.plfreq = 16000;
   gcodec.channels = 1;
   EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
   EXPECT_EQ(106, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
+
   rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event");
   gcodec.plfreq = 8000;
   EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
   EXPECT_EQ(126, gcodec.pltype);
   EXPECT_STREQ("telephone-event", gcodec.plname);
+
+  gcodec.plfreq = 32000;
+  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
+  EXPECT_EQ(107, gcodec.pltype);
+  EXPECT_STREQ("telephone-event", gcodec.plname);
 }
 
 // Test that we fail to set an unknown inbound codec.
 TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsUnsupportedCodec) {
   EXPECT_TRUE(SetupChannel());
   cricket::AudioRecvParameters parameters;
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs.push_back(cricket::AudioCodec(127, "XYZ", 32000, 0, 1));
   EXPECT_FALSE(channel_->SetRecvParameters(parameters));
 }
@@ skipping 71 matching lines @@
   EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, opus));
   EXPECT_EQ(111, opus.pltype);
 }
 
 // Test that changes to recv codecs are applied to all streams.
 TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) {
   EXPECT_TRUE(SetupChannel());
   cricket::AudioRecvParameters parameters;
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs.push_back(kPcmuCodec);
-  parameters.codecs.push_back(kTelephoneEventCodec);
-  parameters.codecs[0].id = 106;  // collide with existing telephone-event
+  parameters.codecs.push_back(kTelephoneEventCodec1);
+  parameters.codecs.push_back(kTelephoneEventCodec2);
+  parameters.codecs[0].id = 106;  // collide with existing CN 32k
   parameters.codecs[2].id = 126;
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(AddRecvStream(kSsrc1));
   int channel_num2 = voe_.GetLastChannel();
+
   webrtc::CodecInst gcodec;
   rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
   gcodec.plfreq = 16000;
   gcodec.channels = 1;
   EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
   EXPECT_EQ(106, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
+
   rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event");
   gcodec.plfreq = 8000;
   gcodec.channels = 1;
   EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
   EXPECT_EQ(126, gcodec.pltype);
   EXPECT_STREQ("telephone-event", gcodec.plname);
+
+  gcodec.plfreq = 32000;
+  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
+  EXPECT_EQ(107, gcodec.pltype);
+  EXPECT_STREQ("telephone-event", gcodec.plname);
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) {
   EXPECT_TRUE(SetupRecvStream());
   cricket::AudioRecvParameters parameters;
   parameters.codecs.push_back(kIsacCodec);
-  parameters.codecs[0].id = 106;  // collide with existing telephone-event
+  parameters.codecs[0].id = 106;  // collide with existing CN 32k
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
 
   int channel_num2 = voe_.GetLastChannel();
   webrtc::CodecInst gcodec;
   rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
   gcodec.plfreq = 16000;
   gcodec.channels = 1;
   EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
   EXPECT_EQ(106, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
@@ skipping 1150 matching lines @@
   cricket::AudioSendParameters parameters;
   EXPECT_FALSE(channel_->SetSendParameters(parameters));
 }
 
 // Test that we can set send codecs even with telephone-event codec as the first
 // one on the list.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) {
   EXPECT_TRUE(SetupSendStream());
   int channel_num = voe_.GetLastChannel();
   cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kTelephoneEventCodec);
+  parameters.codecs.push_back(kTelephoneEventCodec1);
   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_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(kTelephoneEventCodec);
+  parameters.codecs.push_back(kTelephoneEventCodec1);
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs[0].id = 0;  // DTMF
   parameters.codecs[1].id = 96;
   EXPECT_TRUE(channel_->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;
   EXPECT_TRUE(channel_->SetSendParameters(parameters));
@@ skipping 25 matching lines @@
 // Test that we set VAD and DTMF types correctly as caller.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) {
   EXPECT_TRUE(SetupSendStream());
   int channel_num = voe_.GetLastChannel();
   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(kTelephoneEventCodec);
+  parameters.codecs.push_back(kTelephoneEventCodec1);
   parameters.codecs[0].id = 96;
   parameters.codecs[2].id = 97;  // wideband CN
   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_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
   EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
   EXPECT_TRUE(channel_->CanInsertDtmf());
 }
 
 // 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(kTelephoneEventCodec);
+  parameters.codecs.push_back(kTelephoneEventCodec1);
   parameters.codecs[0].id = 96;
   parameters.codecs[2].id = 97;  // wideband CN
   parameters.codecs[4].id = 98;  // DTMF
   EXPECT_TRUE(channel_->SetSendParameters(parameters));
   EXPECT_TRUE(channel_->AddSendStream(
       cricket::StreamParams::CreateLegacy(kSsrc1)));
   int channel_num = voe_.GetLastChannel();
 
   webrtc::CodecInst gcodec;
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
@@ skipping 44 matching lines @@
 
 // Test that we perform case-insensitive matching of codec names.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) {
   EXPECT_TRUE(SetupSendStream());
   int channel_num = voe_.GetLastChannel();
   cricket::AudioSendParameters parameters;
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs.push_back(kPcmuCodec);
   parameters.codecs.push_back(kCn16000Codec);
   parameters.codecs.push_back(kCn8000Codec);
-  parameters.codecs.push_back(kTelephoneEventCodec);
+  parameters.codecs.push_back(kTelephoneEventCodec1);
   parameters.codecs[0].name = "iSaC";
   parameters.codecs[0].id = 96;
   parameters.codecs[2].id = 97;  // wideband CN
   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));
@@ skipping 1337 matching lines @@
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(96, "G722", 8000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(96, "CN", 32000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(96, "CN", 16000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(96, "CN", 8000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(96, "telephone-event", 8000, 0, 1), nullptr));
+  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+      cricket::AudioCodec(96, "telephone-event", 16000, 0, 1), nullptr));
+  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+      cricket::AudioCodec(96, "telephone-event", 32000, 0, 1), nullptr));
+  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+      cricket::AudioCodec(96, "telephone-event", 48000, 0, 1), nullptr));
   // Check codecs with an id by id.
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(0, "", 8000, 0, 1), nullptr));  // PCMU
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(8, "", 8000, 0, 1), nullptr));  // PCMA
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(9, "", 8000, 0, 1), nullptr));  // G722
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(13, "", 8000, 0, 1), nullptr));  // CN
   // Check sample/bitrate matching.
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(0, "PCMU", 8000, 64000, 1), nullptr));
   // Check that bad codecs fail.
   EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(99, "ABCD", 0, 0, 1), nullptr));
   EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(88, "", 0, 0, 1), nullptr));
   EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(0, "", 0, 0, 2), nullptr));
   EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(0, "", 5000, 0, 1), nullptr));
   EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
       cricket::AudioCodec(0, "", 0, 5000, 1), nullptr));
 
   // Verify the payload id of common audio codecs, including CN, ISAC, and G722.
   // TODO(ossu): Why are the payload types of codecs with non-static payload
   // type assignments checked here? It shouldn't really matter.
   cricket::WebRtcVoiceEngine engine(
       nullptr, webrtc::MockAudioDecoderFactory::CreateUnusedFactory());
-  for (std::vector<cricket::AudioCodec>::const_iterator it =
-           engine.send_codecs().begin();
-       it != engine.send_codecs().end(); ++it) {
-    if (it->name == "CN" && it->clockrate == 16000) {
-      EXPECT_EQ(105, it->id);
-    } else if (it->name == "CN" && it->clockrate == 32000) {
-      EXPECT_EQ(106, it->id);
-    } else if (it->name == "ISAC" && it->clockrate == 16000) {
-      EXPECT_EQ(103, it->id);
-    } else if (it->name == "ISAC" && it->clockrate == 32000) {
-      EXPECT_EQ(104, it->id);
-    } else if (it->name == "G722" && it->clockrate == 8000) {
-      EXPECT_EQ(9, it->id);
-    } else if (it->name == "telephone-event") {
-      EXPECT_EQ(126, it->id);
-    } else if (it->name == "opus") {
-      EXPECT_EQ(111, it->id);
-      ASSERT_TRUE(it->params.find("minptime") != it->params.end());
-      EXPECT_EQ("10", it->params.find("minptime")->second);
-      ASSERT_TRUE(it->params.find("useinbandfec") != it->params.end());
-      EXPECT_EQ("1", it->params.find("useinbandfec")->second);
+  for (const cricket::AudioCodec& codec : engine.send_codecs()) {
+    if (codec.name == "CN" && codec.clockrate == 16000) {
+      EXPECT_EQ(105, codec.id);
+    } else if (codec.name == "CN" && codec.clockrate == 32000) {
+      EXPECT_EQ(106, codec.id);
+    } else if (codec.name == "ISAC" && codec.clockrate == 16000) {
+      EXPECT_EQ(103, codec.id);
+    } else if (codec.name == "ISAC" && codec.clockrate == 32000) {
+      EXPECT_EQ(104, codec.id);
+    } else if (codec.name == "G722" && codec.clockrate == 8000) {
+      EXPECT_EQ(9, codec.id);
+    } else if (codec.name == "telephone-event" && codec.clockrate == 8000) {
+      EXPECT_EQ(126, codec.id);
+    // TODO(solenberg): 16k, 32k, 48k DTMF should be dynamically assigned.
+    // Remove these checks once both send and receive side assigns payload types
+    // dynamically.
+    } else if (codec.name == "telephone-event" && codec.clockrate == 16000) {
+      EXPECT_EQ(113, codec.id);
+    } else if (codec.name == "telephone-event" && codec.clockrate == 32000) {
+      EXPECT_EQ(112, codec.id);
+    } else if (codec.name == "telephone-event" && codec.clockrate == 48000) {
+      EXPECT_EQ(110, codec.id);
+    } else if (codec.name == "opus") {
+      EXPECT_EQ(111, codec.id);
+      ASSERT_TRUE(codec.params.find("minptime") != codec.params.end());
+      EXPECT_EQ("10", codec.params.find("minptime")->second);
+      ASSERT_TRUE(codec.params.find("useinbandfec") != codec.params.end());
+      EXPECT_EQ("1", codec.params.find("useinbandfec")->second);
     }
   }
 }
 
 // Tests that VoE supports at least 32 channels
 TEST(WebRtcVoiceEngineTest, Has32Channels) {
   cricket::WebRtcVoiceEngine engine(
       nullptr, webrtc::MockAudioDecoderFactory::CreateUnusedFactory());
   webrtc::RtcEventLogNullImpl event_log;
   std::unique_ptr<webrtc::Call> call(
@@ skipping 30 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));
 }