webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc - src - 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 <memory>
 #include <vector>
 #include "testing/gtest/include/gtest/gtest.h"

 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"

 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"

 namespace webrtc {
 namespace {
 #define test_rate 64000u

 class VerifyingAudioReceiver : public NullRtpData {
  public:
   int32_t OnReceivedPayloadData(
       const uint8_t* payloadData,
       size_t payloadSize,
       const webrtc::WebRtcRTPHeader* rtpHeader) override {
     if (rtpHeader->header.payloadType == 98 ||
         rtpHeader->header.payloadType == 99) {
       EXPECT_EQ(4u, 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 NullRtpFeedback {
  public:
   int32_t OnInitializeDecoder(const int8_t payloadType,
                               const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                               const int frequency,
                               const size_t channels,
                               const uint32_t rate) override {
     if (payloadType == 96) {
       EXPECT_EQ(test_rate, rate) <<
           "The rate should be 64K for this payloadType";
     }
     return 0;
   }
 };

 class RtpRtcpAudioTest : public ::testing::Test {
  protected:
   RtpRtcpAudioTest()
       : fake_clock(123456), retransmission_rate_limiter_(&fake_clock, 1000) {
     test_CSRC[0] = 1234;
     test_CSRC[2] = 2345;
     test_ssrc = 3456;
     test_timestamp = 4567;
     test_sequence_number = 2345;
   }
   ~RtpRtcpAudioTest() {}

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

     receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock));
     receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock));

     rtp_payload_registry1_.reset(new RTPPayloadRegistry(
         RTPPayloadStrategy::CreateStrategy(true)));
     rtp_payload_registry2_.reset(new RTPPayloadRegistry(
         RTPPayloadStrategy::CreateStrategy(true)));

     RtpRtcp::Configuration configuration;
     configuration.audio = true;
     configuration.clock = &fake_clock;
     configuration.receive_statistics = receive_statistics1_.get();
     configuration.outgoing_transport = transport1;
     configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;

     module1 = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
         &fake_clock, data_receiver1, NULL, rtp_payload_registry1_.get()));

     configuration.receive_statistics = receive_statistics2_.get();
     configuration.outgoing_transport = transport2;

     module2 = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
         &fake_clock, data_receiver2, NULL, rtp_payload_registry2_.get()));

     transport1->SetSendModule(module2, rtp_payload_registry2_.get(),
                               rtp_receiver2_.get(), receive_statistics2_.get());
     transport2->SetSendModule(module1, rtp_payload_registry1_.get(),
                               rtp_receiver1_.get(), receive_statistics1_.get());
   }

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

   RtpRtcp* module1;
   RtpRtcp* module2;
   std::unique_ptr<ReceiveStatistics> receive_statistics1_;
   std::unique_ptr<ReceiveStatistics> receive_statistics2_;
   std::unique_ptr<RtpReceiver> rtp_receiver1_;
   std::unique_ptr<RtpReceiver> rtp_receiver2_;
   std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
   std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
   VerifyingAudioReceiver* data_receiver1;
   VerifyingAudioReceiver* data_receiver2;
   LoopBackTransport* transport1;
   LoopBackTransport* transport2;
   RTPCallback* rtp_callback;
   uint32_t test_ssrc;
   uint32_t test_timestamp;
   uint16_t test_sequence_number;
   uint32_t test_CSRC[webrtc::kRtpCsrcSize];
   SimulatedClock fake_clock;
   RateLimiter retransmission_rate_limiter_;
 };

 TEST_F(RtpRtcpAudioTest, Basic) {
   module1->SetSSRC(test_ssrc);
   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 registered the payload type.
   EXPECT_FALSE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
                                          nullptr, 0, nullptr, nullptr,
                                          nullptr));

   CodecInst voice_codec;
   memset(&voice_codec, 0, sizeof(voice_codec));
   voice_codec.pltype = 96;
   voice_codec.plfreq = 8000;
   memcpy(voice_codec.plname, "PCMU", 5);

   EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
   EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
       voice_codec.plname,
       voice_codec.pltype,
       voice_codec.plfreq,
       voice_codec.channels,
       (voice_codec.rate < 0) ? 0 : voice_codec.rate));
   EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec));
   voice_codec.rate = test_rate;
   EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
       voice_codec.plname,
       voice_codec.pltype,
       voice_codec.plfreq,
       voice_codec.channels,
       (voice_codec.rate < 0) ? 0 : voice_codec.rate));

   const uint8_t test[5] = "test";
   EXPECT_EQ(true,
             module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
                                       test, 4, nullptr, nullptr, nullptr));

   EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
   uint32_t timestamp;
   EXPECT_TRUE(rtp_receiver2_->Timestamp(&timestamp));
   EXPECT_EQ(test_timestamp, timestamp);
 }

 TEST_F(RtpRtcpAudioTest, DTMF) {
   CodecInst voice_codec;
   memset(&voice_codec, 0, sizeof(voice_codec));
   voice_codec.pltype = 96;
   voice_codec.plfreq = 8000;
   memcpy(voice_codec.plname, "PCMU", 5);

   EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
   EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
       voice_codec.plname,
       voice_codec.pltype,
       voice_codec.plfreq,
       voice_codec.channels,
       (voice_codec.rate < 0) ? 0 : voice_codec.rate));
   EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec));
   voice_codec.rate = test_rate;
   EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
       voice_codec.plname,
       voice_codec.pltype,
       voice_codec.plfreq,
       voice_codec.channels,
       (voice_codec.rate < 0) ? 0 : voice_codec.rate));

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

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

   EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
   EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
         voice_codec.plname,
         voice_codec.pltype,
         voice_codec.plfreq,
         voice_codec.channels,
         (voice_codec.rate < 0) ? 0 : voice_codec.rate));

   // Start DTMF test.
   int 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 uint8_t 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_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                           timeStamp, -1, test, 4, nullptr,
                                           nullptr, nullptr));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
   EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));

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

 }  // namespace
 }  // namespace webrtc
	/*
	* 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 <memory>
	#include <vector>
	#include "testing/gtest/include/gtest/gtest.h"

	#include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"

	#include "webrtc/base/rate_limiter.h"
	#include "webrtc/common_types.h"
	#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
	#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
	#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"

	namespace webrtc {
	namespace {
	#define test_rate 64000u

	class VerifyingAudioReceiver : public NullRtpData {
	public:
	int32_t OnReceivedPayloadData(
	const uint8_t* payloadData,
	size_t payloadSize,
	const webrtc::WebRtcRTPHeader* rtpHeader) override {
	if (rtpHeader->header.payloadType == 98 \|\|
	rtpHeader->header.payloadType == 99) {
	EXPECT_EQ(4u, 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 NullRtpFeedback {
	public:
	int32_t OnInitializeDecoder(const int8_t payloadType,
	const char payloadName[RTP_PAYLOAD_NAME_SIZE],
	const int frequency,
	const size_t channels,
	const uint32_t rate) override {
	if (payloadType == 96) {
	EXPECT_EQ(test_rate, rate) <<
	"The rate should be 64K for this payloadType";
	}
	return 0;
	}
	};

	class RtpRtcpAudioTest : public ::testing::Test {
	protected:
	RtpRtcpAudioTest()
	: fake_clock(123456), retransmission_rate_limiter_(&fake_clock, 1000) {
	test_CSRC[0] = 1234;
	test_CSRC[2] = 2345;
	test_ssrc = 3456;
	test_timestamp = 4567;
	test_sequence_number = 2345;
	}
	~RtpRtcpAudioTest() {}

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

	receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock));
	receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock));

	rtp_payload_registry1_.reset(new RTPPayloadRegistry(
	RTPPayloadStrategy::CreateStrategy(true)));
	rtp_payload_registry2_.reset(new RTPPayloadRegistry(
	RTPPayloadStrategy::CreateStrategy(true)));

	RtpRtcp::Configuration configuration;
	configuration.audio = true;
	configuration.clock = &fake_clock;
	configuration.receive_statistics = receive_statistics1_.get();
	configuration.outgoing_transport = transport1;
	configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;

	module1 = RtpRtcp::CreateRtpRtcp(configuration);
	rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
	&fake_clock, data_receiver1, NULL, rtp_payload_registry1_.get()));

	configuration.receive_statistics = receive_statistics2_.get();
	configuration.outgoing_transport = transport2;

	module2 = RtpRtcp::CreateRtpRtcp(configuration);
	rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
	&fake_clock, data_receiver2, NULL, rtp_payload_registry2_.get()));

	transport1->SetSendModule(module2, rtp_payload_registry2_.get(),
	rtp_receiver2_.get(), receive_statistics2_.get());
	transport2->SetSendModule(module1, rtp_payload_registry1_.get(),
	rtp_receiver1_.get(), receive_statistics1_.get());
	}

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

	RtpRtcp* module1;
	RtpRtcp* module2;
	std::unique_ptr<ReceiveStatistics> receive_statistics1_;
	std::unique_ptr<ReceiveStatistics> receive_statistics2_;
	std::unique_ptr<RtpReceiver> rtp_receiver1_;
	std::unique_ptr<RtpReceiver> rtp_receiver2_;
	std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
	std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
	VerifyingAudioReceiver* data_receiver1;
	VerifyingAudioReceiver* data_receiver2;
	LoopBackTransport* transport1;
	LoopBackTransport* transport2;
	RTPCallback* rtp_callback;
	uint32_t test_ssrc;
	uint32_t test_timestamp;
	uint16_t test_sequence_number;
	uint32_t test_CSRC[webrtc::kRtpCsrcSize];
	SimulatedClock fake_clock;
	RateLimiter retransmission_rate_limiter_;
	};

	TEST_F(RtpRtcpAudioTest, Basic) {
	module1->SetSSRC(test_ssrc);
	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 registered the payload type.
	EXPECT_FALSE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
	nullptr, 0, nullptr, nullptr,
	nullptr));

	CodecInst voice_codec;
	memset(&voice_codec, 0, sizeof(voice_codec));
	voice_codec.pltype = 96;
	voice_codec.plfreq = 8000;
	memcpy(voice_codec.plname, "PCMU", 5);

	EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
	EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
	voice_codec.plname,
	voice_codec.pltype,
	voice_codec.plfreq,
	voice_codec.channels,
	(voice_codec.rate < 0) ? 0 : voice_codec.rate));
	EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec));
	voice_codec.rate = test_rate;
	EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
	voice_codec.plname,
	voice_codec.pltype,
	voice_codec.plfreq,
	voice_codec.channels,
	(voice_codec.rate < 0) ? 0 : voice_codec.rate));

	const uint8_t test[5] = "test";
	EXPECT_EQ(true,
	module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
	test, 4, nullptr, nullptr, nullptr));

	EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
	uint32_t timestamp;
	EXPECT_TRUE(rtp_receiver2_->Timestamp(&timestamp));
	EXPECT_EQ(test_timestamp, timestamp);
	}

	TEST_F(RtpRtcpAudioTest, DTMF) {
	CodecInst voice_codec;
	memset(&voice_codec, 0, sizeof(voice_codec));
	voice_codec.pltype = 96;
	voice_codec.plfreq = 8000;
	memcpy(voice_codec.plname, "PCMU", 5);

	EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
	EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
	voice_codec.plname,
	voice_codec.pltype,
	voice_codec.plfreq,
	voice_codec.channels,
	(voice_codec.rate < 0) ? 0 : voice_codec.rate));
	EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec));
	voice_codec.rate = test_rate;
	EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
	voice_codec.plname,
	voice_codec.pltype,
	voice_codec.plfreq,
	voice_codec.channels,
	(voice_codec.rate < 0) ? 0 : voice_codec.rate));

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

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

	EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
	EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
	voice_codec.plname,
	voice_codec.pltype,
	voice_codec.plfreq,
	voice_codec.channels,
	(voice_codec.rate < 0) ? 0 : voice_codec.rate));

	// Start DTMF test.
	int 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 uint8_t 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_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
	timeStamp, -1, test, 4, nullptr,
	nullptr, nullptr));
	fake_clock.AdvanceTimeMilliseconds(20);
	module1->Process();
	}
	EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));

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

	} // namespace
	} // namespace webrtc