modules/rtp_rtcp/test/testAPI/test_api_audio.cc - src/webrtc - Git at Google

 /*
  *  Copyright (c) 2012 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.
  */

 #include <algorithm>
 #include <vector>
 #include <gtest/gtest.h>

 #include "test_api.h"

 #include "common_types.h"
 #include "rtp_rtcp.h"
 #include "rtp_rtcp_defines.h"

 using namespace webrtc;

 #define test_rate 64000u

 class VerifyingAudioReceiver : public RtpData {
  public:
   virtual WebRtc_Word32 OnReceivedPayloadData(
       const WebRtc_UWord8* payloadData,
       const WebRtc_UWord16 payloadSize,
       const webrtc::WebRtcRTPHeader* rtpHeader) {
     if (rtpHeader->header.payloadType == 98 ||
         rtpHeader->header.payloadType == 99) {
       EXPECT_EQ(4, payloadSize);
       char str[5];
       memcpy(str, payloadData, payloadSize);
       str[4] = 0;
       // All our test vectors for payload type 96 and 97 even the stereo is on
       // a per channel base equal to the 4 chars "test".
       // Note there is no null termination so we add that to use the
       // test EXPECT_STRCASEEQ.
       EXPECT_STRCASEEQ("test", str);
       return 0;
     }
     if (rtpHeader->header.payloadType == 100 ||
         rtpHeader->header.payloadType == 101 ||
         rtpHeader->header.payloadType == 102) {
       if (rtpHeader->type.Audio.channel == 1) {
         if (payloadData[0] == 0xff) {
           // All our test vectors for payload type 100, 101 and 102 have the
           // first channel data being equal to 0xff.
           return 0;
         }
       }
       ADD_FAILURE() << "This code path should never happen.";
       return -1;
     }
     return 0;
   }
 };

 class RTPCallback : public RtpFeedback {
  public:
   virtual WebRtc_Word32 OnInitializeDecoder(
       const WebRtc_Word32 id,
       const WebRtc_Word8 payloadType,
       const char payloadName[RTP_PAYLOAD_NAME_SIZE],
       const int frequency,
       const WebRtc_UWord8 channels,
       const WebRtc_UWord32 rate) {
     if (payloadType == 96) {
       EXPECT_EQ(test_rate, rate) <<
           "The rate should be 64K for this payloadType";
     }
     return 0;
   }
   virtual void OnPacketTimeout(const WebRtc_Word32 id) {
   }
   virtual void OnReceivedPacket(const WebRtc_Word32 id,
                                 const RtpRtcpPacketType packetType) {
   }
   virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
                                      const RTPAliveType alive) {
   }
   virtual void OnIncomingSSRCChanged(const WebRtc_Word32 id,
                                      const WebRtc_UWord32 SSRC) {
   }
   virtual void OnIncomingCSRCChanged(const WebRtc_Word32 id,
                                      const WebRtc_UWord32 CSRC,
                                      const bool added) {
   }
 };

 class AudioFeedback : public RtpAudioFeedback {
   virtual void OnReceivedTelephoneEvent(const WebRtc_Word32 id,
                                         const WebRtc_UWord8 event,
                                         const bool end) {
     static WebRtc_UWord8 expectedEvent = 0;

     if (end) {
       WebRtc_UWord8 oldEvent = expectedEvent-1;
       if (expectedEvent == 32) {
         oldEvent = 15;
       }
       EXPECT_EQ(oldEvent, event);
     } else {
       EXPECT_EQ(expectedEvent, event);
       expectedEvent++;
     }
     if (expectedEvent == 16) {
       expectedEvent = 32;
     }
   }
   virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id,
                                     const WebRtc_UWord8 event,
                                     const WebRtc_UWord16 lengthMs,
                                     const WebRtc_UWord8 volume) {
   };
 };

 class RtpRtcpAudioTest : public ::testing::Test {
  protected:
   RtpRtcpAudioTest() : fake_clock(123456) {
     test_CSRC[0] = 1234;
     test_CSRC[2] = 2345;
     test_id = 123;
     test_ssrc = 3456;
     test_timestamp = 4567;
     test_sequence_number = 2345;
   }
   ~RtpRtcpAudioTest() {}

   virtual void SetUp() {
     audioFeedback = new AudioFeedback();
     data_receiver1 = new VerifyingAudioReceiver();
     data_receiver2 = new VerifyingAudioReceiver();
     rtp_callback = new RTPCallback();
     transport1 = new LoopBackTransport();
     transport2 = new LoopBackTransport();

     RtpRtcp::Configuration configuration;
     configuration.id = test_id;
     configuration.audio = true;
     configuration.clock = &fake_clock;
     configuration.incoming_data = data_receiver1;
     configuration.outgoing_transport = transport1;
     configuration.audio_messages = audioFeedback;

     module1 = RtpRtcp::CreateRtpRtcp(configuration);

     configuration.id = test_id + 1;
     configuration.incoming_data = data_receiver2;
     configuration.incoming_messages = rtp_callback;
     configuration.outgoing_transport = transport2;
     configuration.audio_messages = audioFeedback;

     module2 = RtpRtcp::CreateRtpRtcp(configuration);

     transport1->SetSendModule(module2);
     transport2->SetSendModule(module1);
   }

   virtual void TearDown() {
     delete module1;
     delete module2;
     delete transport1;
     delete transport2;
     delete audioFeedback;
     delete data_receiver1;
     delete data_receiver2;
     delete rtp_callback;
   }

   int test_id;
   RtpRtcp* module1;
   RtpRtcp* module2;
   VerifyingAudioReceiver* data_receiver1;
   VerifyingAudioReceiver* data_receiver2;
   LoopBackTransport* transport1;
   LoopBackTransport* transport2;
   AudioFeedback* audioFeedback;
   RTPCallback* rtp_callback;
   WebRtc_UWord32 test_ssrc;
   WebRtc_UWord32 test_timestamp;
   WebRtc_UWord16 test_sequence_number;
   WebRtc_UWord32 test_CSRC[webrtc::kRtpCsrcSize];
   SimulatedClock fake_clock;
 };

 TEST_F(RtpRtcpAudioTest, Basic) {
   EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
   EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));

   // Test detection at the end of a DTMF tone.
   EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));

   EXPECT_EQ(0, module1->SetSendingStatus(true));

   // Start basic RTP test.

   // Send an empty RTP packet.
   // Should fail since we have not registerd the payload type.
   EXPECT_EQ(-1, module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
                                           96, 0, -1, NULL, 0));

   CodecInst voiceCodec;
   voiceCodec.pltype = 96;
   voiceCodec.plfreq = 8000;
   memcpy(voiceCodec.plname, "PCMU", 5);

   EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
   EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
   EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
   voiceCodec.rate = test_rate;
   EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
   printf("4\n");

   const WebRtc_UWord8 test[5] = "test";
   EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                          0, -1, test, 4));

   EXPECT_EQ(test_ssrc, module2->RemoteSSRC());
   EXPECT_EQ(test_timestamp, module2->RemoteTimestamp());
 }

 TEST_F(RtpRtcpAudioTest, RED) {
   CodecInst voiceCodec;
   voiceCodec.pltype = 96;
   voiceCodec.plfreq = 8000;
   memcpy(voiceCodec.plname, "PCMU", 5);

   EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
   EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
   EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
   voiceCodec.rate = test_rate;
   EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));

   EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
   EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
   EXPECT_EQ(0, module1->SetSendingStatus(true));

   voiceCodec.pltype = 127;
   voiceCodec.plfreq = 8000;
   memcpy(voiceCodec.plname, "RED", 4);

   EXPECT_EQ(0, module1->SetSendREDPayloadType(voiceCodec.pltype));
   WebRtc_Word8 red = 0;
   EXPECT_EQ(0, module1->SendREDPayloadType(red));
   EXPECT_EQ(voiceCodec.pltype, red);
   EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
   EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));

   RTPFragmentationHeader fragmentation;
   fragmentation.fragmentationVectorSize = 2;
   fragmentation.fragmentationLength = new WebRtc_UWord32[2];
   fragmentation.fragmentationLength[0] = 4;
   fragmentation.fragmentationLength[1] = 4;
   fragmentation.fragmentationOffset = new WebRtc_UWord32[2];
   fragmentation.fragmentationOffset[0] = 0;
   fragmentation.fragmentationOffset[1] = 4;
   fragmentation.fragmentationTimeDiff = new WebRtc_UWord16[2];
   fragmentation.fragmentationTimeDiff[0] = 0;
   fragmentation.fragmentationTimeDiff[1] = 0;
   fragmentation.fragmentationPlType = new WebRtc_UWord8[2];
   fragmentation.fragmentationPlType[0] = 96;
   fragmentation.fragmentationPlType[1] = 96;

   const WebRtc_UWord8 test[5] = "test";
   // Send a RTP packet.
   EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
                                          96, 160, -1, test, 4,
                                          &fragmentation));

   EXPECT_EQ(0, module1->SetSendREDPayloadType(-1));
   EXPECT_EQ(-1, module1->SendREDPayloadType(red));
 }

 TEST_F(RtpRtcpAudioTest, DTMF) {
   CodecInst voiceCodec;
   voiceCodec.pltype = 96;
   voiceCodec.plfreq = 8000;
   memcpy(voiceCodec.plname, "PCMU", 5);

   EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
   EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
   EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
   voiceCodec.rate = test_rate;
   EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));

   EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
   EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
   EXPECT_EQ(0, module1->SetSendingStatus(true));

   // Prepare for DTMF.
   voiceCodec.pltype = 97;
   voiceCodec.plfreq = 8000;
   memcpy(voiceCodec.plname, "telephone-event", 16);

   EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
   EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));

   // Start DTMF test.
   WebRtc_UWord32 timeStamp = 160;

   // Send a DTMF tone using RFC 2833 (4733).
   for (int i = 0; i < 16; i++) {
     EXPECT_EQ(0, module1->SendTelephoneEventOutband(i, timeStamp, 10));
   }
   timeStamp += 160;  // Prepare for next packet.

   const WebRtc_UWord8 test[9] = "test";

   // Send RTP packets for 16 tones a 160 ms  100ms
   // pause between = 2560ms + 1600ms = 4160ms
   for (;timeStamp <= 250 * 160; timeStamp += 160) {
     EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                            timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
   EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));

   for (;timeStamp <= 740 * 160; timeStamp += 160) {
     EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                            timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
 }
	/*
	* Copyright (c) 2012 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.
	*/

	#include <algorithm>
	#include <vector>
	#include <gtest/gtest.h>

	#include "test_api.h"

	#include "common_types.h"
	#include "rtp_rtcp.h"
	#include "rtp_rtcp_defines.h"

	using namespace webrtc;

	#define test_rate 64000u

	class VerifyingAudioReceiver : public RtpData {
	public:
	virtual WebRtc_Word32 OnReceivedPayloadData(
	const WebRtc_UWord8* payloadData,
	const WebRtc_UWord16 payloadSize,
	const webrtc::WebRtcRTPHeader* rtpHeader) {
	if (rtpHeader->header.payloadType == 98 \|\|
	rtpHeader->header.payloadType == 99) {
	EXPECT_EQ(4, payloadSize);
	char str[5];
	memcpy(str, payloadData, payloadSize);
	str[4] = 0;
	// All our test vectors for payload type 96 and 97 even the stereo is on
	// a per channel base equal to the 4 chars "test".
	// Note there is no null termination so we add that to use the
	// test EXPECT_STRCASEEQ.
	EXPECT_STRCASEEQ("test", str);
	return 0;
	}
	if (rtpHeader->header.payloadType == 100 \|\|
	rtpHeader->header.payloadType == 101 \|\|
	rtpHeader->header.payloadType == 102) {
	if (rtpHeader->type.Audio.channel == 1) {
	if (payloadData[0] == 0xff) {
	// All our test vectors for payload type 100, 101 and 102 have the
	// first channel data being equal to 0xff.
	return 0;
	}
	}
	ADD_FAILURE() << "This code path should never happen.";
	return -1;
	}
	return 0;
	}
	};

	class RTPCallback : public RtpFeedback {
	public:
	virtual WebRtc_Word32 OnInitializeDecoder(
	const WebRtc_Word32 id,
	const WebRtc_Word8 payloadType,
	const char payloadName[RTP_PAYLOAD_NAME_SIZE],
	const int frequency,
	const WebRtc_UWord8 channels,
	const WebRtc_UWord32 rate) {
	if (payloadType == 96) {
	EXPECT_EQ(test_rate, rate) <<
	"The rate should be 64K for this payloadType";
	}
	return 0;
	}
	virtual void OnPacketTimeout(const WebRtc_Word32 id) {
	}
	virtual void OnReceivedPacket(const WebRtc_Word32 id,
	const RtpRtcpPacketType packetType) {
	}
	virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
	const RTPAliveType alive) {
	}
	virtual void OnIncomingSSRCChanged(const WebRtc_Word32 id,
	const WebRtc_UWord32 SSRC) {
	}
	virtual void OnIncomingCSRCChanged(const WebRtc_Word32 id,
	const WebRtc_UWord32 CSRC,
	const bool added) {
	}
	};

	class AudioFeedback : public RtpAudioFeedback {
	virtual void OnReceivedTelephoneEvent(const WebRtc_Word32 id,
	const WebRtc_UWord8 event,
	const bool end) {
	static WebRtc_UWord8 expectedEvent = 0;

	if (end) {
	WebRtc_UWord8 oldEvent = expectedEvent-1;
	if (expectedEvent == 32) {
	oldEvent = 15;
	}
	EXPECT_EQ(oldEvent, event);
	} else {
	EXPECT_EQ(expectedEvent, event);
	expectedEvent++;
	}
	if (expectedEvent == 16) {
	expectedEvent = 32;
	}
	}
	virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id,
	const WebRtc_UWord8 event,
	const WebRtc_UWord16 lengthMs,
	const WebRtc_UWord8 volume) {
	};
	};

	class RtpRtcpAudioTest : public ::testing::Test {
	protected:
	RtpRtcpAudioTest() : fake_clock(123456) {
	test_CSRC[0] = 1234;
	test_CSRC[2] = 2345;
	test_id = 123;
	test_ssrc = 3456;
	test_timestamp = 4567;
	test_sequence_number = 2345;
	}
	~RtpRtcpAudioTest() {}

	virtual void SetUp() {
	audioFeedback = new AudioFeedback();
	data_receiver1 = new VerifyingAudioReceiver();
	data_receiver2 = new VerifyingAudioReceiver();
	rtp_callback = new RTPCallback();
	transport1 = new LoopBackTransport();
	transport2 = new LoopBackTransport();

	RtpRtcp::Configuration configuration;
	configuration.id = test_id;
	configuration.audio = true;
	configuration.clock = &fake_clock;
	configuration.incoming_data = data_receiver1;
	configuration.outgoing_transport = transport1;
	configuration.audio_messages = audioFeedback;

	module1 = RtpRtcp::CreateRtpRtcp(configuration);

	configuration.id = test_id + 1;
	configuration.incoming_data = data_receiver2;
	configuration.incoming_messages = rtp_callback;
	configuration.outgoing_transport = transport2;
	configuration.audio_messages = audioFeedback;

	module2 = RtpRtcp::CreateRtpRtcp(configuration);

	transport1->SetSendModule(module2);
	transport2->SetSendModule(module1);
	}

	virtual void TearDown() {
	delete module1;
	delete module2;
	delete transport1;
	delete transport2;
	delete audioFeedback;
	delete data_receiver1;
	delete data_receiver2;
	delete rtp_callback;
	}

	int test_id;
	RtpRtcp* module1;
	RtpRtcp* module2;
	VerifyingAudioReceiver* data_receiver1;
	VerifyingAudioReceiver* data_receiver2;
	LoopBackTransport* transport1;
	LoopBackTransport* transport2;
	AudioFeedback* audioFeedback;
	RTPCallback* rtp_callback;
	WebRtc_UWord32 test_ssrc;
	WebRtc_UWord32 test_timestamp;
	WebRtc_UWord16 test_sequence_number;
	WebRtc_UWord32 test_CSRC[webrtc::kRtpCsrcSize];
	SimulatedClock fake_clock;
	};

	TEST_F(RtpRtcpAudioTest, Basic) {
	EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
	EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));

	// Test detection at the end of a DTMF tone.
	EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));

	EXPECT_EQ(0, module1->SetSendingStatus(true));

	// Start basic RTP test.

	// Send an empty RTP packet.
	// Should fail since we have not registerd the payload type.
	EXPECT_EQ(-1, module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
	96, 0, -1, NULL, 0));

	CodecInst voiceCodec;
	voiceCodec.pltype = 96;
	voiceCodec.plfreq = 8000;
	memcpy(voiceCodec.plname, "PCMU", 5);

	EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
	EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
	EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
	voiceCodec.rate = test_rate;
	EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
	printf("4\n");

	const WebRtc_UWord8 test[5] = "test";
	EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
	0, -1, test, 4));

	EXPECT_EQ(test_ssrc, module2->RemoteSSRC());
	EXPECT_EQ(test_timestamp, module2->RemoteTimestamp());
	}

	TEST_F(RtpRtcpAudioTest, RED) {
	CodecInst voiceCodec;
	voiceCodec.pltype = 96;
	voiceCodec.plfreq = 8000;
	memcpy(voiceCodec.plname, "PCMU", 5);

	EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
	EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
	EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
	voiceCodec.rate = test_rate;
	EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));

	EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
	EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
	EXPECT_EQ(0, module1->SetSendingStatus(true));

	voiceCodec.pltype = 127;
	voiceCodec.plfreq = 8000;
	memcpy(voiceCodec.plname, "RED", 4);

	EXPECT_EQ(0, module1->SetSendREDPayloadType(voiceCodec.pltype));
	WebRtc_Word8 red = 0;
	EXPECT_EQ(0, module1->SendREDPayloadType(red));
	EXPECT_EQ(voiceCodec.pltype, red);
	EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
	EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));

	RTPFragmentationHeader fragmentation;
	fragmentation.fragmentationVectorSize = 2;
	fragmentation.fragmentationLength = new WebRtc_UWord32[2];
	fragmentation.fragmentationLength[0] = 4;
	fragmentation.fragmentationLength[1] = 4;
	fragmentation.fragmentationOffset = new WebRtc_UWord32[2];
	fragmentation.fragmentationOffset[0] = 0;
	fragmentation.fragmentationOffset[1] = 4;
	fragmentation.fragmentationTimeDiff = new WebRtc_UWord16[2];
	fragmentation.fragmentationTimeDiff[0] = 0;
	fragmentation.fragmentationTimeDiff[1] = 0;
	fragmentation.fragmentationPlType = new WebRtc_UWord8[2];
	fragmentation.fragmentationPlType[0] = 96;
	fragmentation.fragmentationPlType[1] = 96;

	const WebRtc_UWord8 test[5] = "test";
	// Send a RTP packet.
	EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
	96, 160, -1, test, 4,
	&fragmentation));

	EXPECT_EQ(0, module1->SetSendREDPayloadType(-1));
	EXPECT_EQ(-1, module1->SendREDPayloadType(red));
	}

	TEST_F(RtpRtcpAudioTest, DTMF) {
	CodecInst voiceCodec;
	voiceCodec.pltype = 96;
	voiceCodec.plfreq = 8000;
	memcpy(voiceCodec.plname, "PCMU", 5);

	EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
	EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
	EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
	voiceCodec.rate = test_rate;
	EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));

	EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
	EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
	EXPECT_EQ(0, module1->SetSendingStatus(true));

	// Prepare for DTMF.
	voiceCodec.pltype = 97;
	voiceCodec.plfreq = 8000;
	memcpy(voiceCodec.plname, "telephone-event", 16);

	EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
	EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));

	// Start DTMF test.
	WebRtc_UWord32 timeStamp = 160;

	// Send a DTMF tone using RFC 2833 (4733).
	for (int i = 0; i < 16; i++) {
	EXPECT_EQ(0, module1->SendTelephoneEventOutband(i, timeStamp, 10));
	}
	timeStamp += 160; // Prepare for next packet.

	const WebRtc_UWord8 test[9] = "test";

	// Send RTP packets for 16 tones a 160 ms 100ms
	// pause between = 2560ms + 1600ms = 4160ms
	for (;timeStamp <= 250 * 160; timeStamp += 160) {
	EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
	timeStamp, -1, test, 4));
	fake_clock.AdvanceTimeMilliseconds(20);
	module1->Process();
	}
	EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));

	for (;timeStamp <= 740 * 160; timeStamp += 160) {
	EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
	timeStamp, -1, test, 4));
	fake_clock.AdvanceTimeMilliseconds(20);
	module1->Process();
	}
	}