modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc - src - Git at Google

 /*
  *  Copyright (c) 2019 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 "modules/rtp_rtcp/source/rtp_sender_audio.h"

 #include <memory>
 #include <vector>

 #include "api/transport/field_trial_based_config.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "test/field_trial.h"
 #include "test/gmock.h"
 #include "test/gtest.h"

 namespace webrtc {

 namespace {
 enum : int {  // The first valid value is 1.
   kAudioLevelExtensionId = 1,
   kAbsoluteCaptureTimeExtensionId = 2,
 };

 const uint16_t kSeqNum = 33;
 const uint32_t kSsrc = 725242;
 const uint8_t kAudioLevel = 0x5a;
 const uint64_t kStartTime = 123456789;

 using ::testing::ElementsAreArray;

 class LoopbackTransportTest : public webrtc::Transport {
  public:
   LoopbackTransportTest() {
     receivers_extensions_.Register<AudioLevel>(kAudioLevelExtensionId);
     receivers_extensions_.Register<AbsoluteCaptureTimeExtension>(
         kAbsoluteCaptureTimeExtensionId);
   }

   bool SendRtp(const uint8_t* data,
                size_t len,
                const PacketOptions& /*options*/) override {
     sent_packets_.push_back(RtpPacketReceived(&receivers_extensions_));
     EXPECT_TRUE(sent_packets_.back().Parse(data, len));
     return true;
   }
   bool SendRtcp(const uint8_t* data, size_t len) override { return false; }
   const RtpPacketReceived& last_sent_packet() { return sent_packets_.back(); }
   int packets_sent() { return sent_packets_.size(); }

  private:
   RtpHeaderExtensionMap receivers_extensions_;
   std::vector<RtpPacketReceived> sent_packets_;
 };

 }  // namespace

 class RtpSenderAudioTest : public ::testing::Test {
  public:
   RtpSenderAudioTest()
       : fake_clock_(kStartTime),
         rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
           RtpRtcpInterface::Configuration config;
           config.audio = true;
           config.clock = &fake_clock_;
           config.outgoing_transport = &transport_;
           config.local_media_ssrc = kSsrc;
           return config;
         }())),
         rtp_sender_audio_(
             std::make_unique<RTPSenderAudio>(&fake_clock_,
                                              rtp_module_->RtpSender())) {
     rtp_module_->SetSequenceNumber(kSeqNum);
   }

   SimulatedClock fake_clock_;
   LoopbackTransportTest transport_;
   std::unique_ptr<ModuleRtpRtcpImpl2> rtp_module_;
   std::unique_ptr<RTPSenderAudio> rtp_sender_audio_;
 };

 TEST_F(RtpSenderAudioTest, SendAudio) {
   const char payload_name[] = "PAYLOAD_NAME";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
                    payload_name, payload_type, 48000, 0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};

   ASSERT_TRUE(
       rtp_sender_audio_->SendAudio(AudioFrameType::kAudioFrameCN, payload_type,
                                    4321, payload, sizeof(payload),
                                    /*absolute_capture_timestamp_ms=*/0));

   auto sent_payload = transport_.last_sent_packet().payload();
   EXPECT_THAT(sent_payload, ElementsAreArray(payload));
 }

 TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
   EXPECT_EQ(0, rtp_sender_audio_->SetAudioLevel(kAudioLevel));
   rtp_module_->RegisterRtpHeaderExtension(AudioLevel::Uri(),
                                           kAudioLevelExtensionId);

   const char payload_name[] = "PAYLOAD_NAME";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
                    payload_name, payload_type, 48000, 0, 1500));

   uint8_t payload[] = {47, 11, 32, 93, 89};

   ASSERT_TRUE(
       rtp_sender_audio_->SendAudio(AudioFrameType::kAudioFrameCN, payload_type,
                                    4321, payload, sizeof(payload),
                                    /*absolute_capture_timestamp_ms=*/0));

   auto sent_payload = transport_.last_sent_packet().payload();
   EXPECT_THAT(sent_payload, ElementsAreArray(payload));
   // Verify AudioLevel extension.
   bool voice_activity;
   uint8_t audio_level;
   EXPECT_TRUE(transport_.last_sent_packet().GetExtension<AudioLevel>(
       &voice_activity, &audio_level));
   EXPECT_EQ(kAudioLevel, audio_level);
   EXPECT_FALSE(voice_activity);
 }

 TEST_F(RtpSenderAudioTest, SendAudioWithoutAbsoluteCaptureTime) {
   constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
   const char payload_name[] = "audio";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
                    payload_name, payload_type, 48000, 0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};

   ASSERT_TRUE(rtp_sender_audio_->SendAudio(
       AudioFrameType::kAudioFrameCN, payload_type, 4321, payload,
       sizeof(payload), kAbsoluteCaptureTimestampMs));

   EXPECT_FALSE(transport_.last_sent_packet()
                    .HasExtension<AbsoluteCaptureTimeExtension>());
 }

 // Essentially the same test as
 // SendAudioWithAbsoluteCaptureTimeWithCaptureClockOffset but with a field
 // trial. We will remove this test eventually.
 TEST_F(RtpSenderAudioTest, SendAudioWithAbsoluteCaptureTime) {
   // Recreate rtp_sender_audio_ with new field trial.
   test::ScopedFieldTrials field_trial(
       "WebRTC-IncludeCaptureClockOffset/Disabled/");
   rtp_sender_audio_ =
       std::make_unique<RTPSenderAudio>(&fake_clock_, rtp_module_->RtpSender());

   rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
                                           kAbsoluteCaptureTimeExtensionId);
   constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
   const char payload_name[] = "audio";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
                    payload_name, payload_type, 48000, 0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};

   ASSERT_TRUE(rtp_sender_audio_->SendAudio(
       AudioFrameType::kAudioFrameCN, payload_type, 4321, payload,
       sizeof(payload), kAbsoluteCaptureTimestampMs));

   auto absolute_capture_time =
       transport_.last_sent_packet()
           .GetExtension<AbsoluteCaptureTimeExtension>();
   EXPECT_TRUE(absolute_capture_time);
   EXPECT_EQ(
       absolute_capture_time->absolute_capture_timestamp,
       Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
           kAbsoluteCaptureTimestampMs)));
   EXPECT_FALSE(
       absolute_capture_time->estimated_capture_clock_offset.has_value());
 }

 TEST_F(RtpSenderAudioTest,
        SendAudioWithAbsoluteCaptureTimeWithCaptureClockOffset) {
   rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
                                           kAbsoluteCaptureTimeExtensionId);
   constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
   const char payload_name[] = "audio";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
                    payload_name, payload_type, 48000, 0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};

   ASSERT_TRUE(rtp_sender_audio_->SendAudio(
       AudioFrameType::kAudioFrameCN, payload_type, 4321, payload,
       sizeof(payload), kAbsoluteCaptureTimestampMs));

   auto absolute_capture_time =
       transport_.last_sent_packet()
           .GetExtension<AbsoluteCaptureTimeExtension>();
   EXPECT_TRUE(absolute_capture_time);
   EXPECT_EQ(
       absolute_capture_time->absolute_capture_timestamp,
       Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
           kAbsoluteCaptureTimestampMs)));
   EXPECT_TRUE(
       absolute_capture_time->estimated_capture_clock_offset.has_value());
   EXPECT_EQ(0, *absolute_capture_time->estimated_capture_clock_offset);
 }

 // As RFC4733, named telephone events are carried as part of the audio stream
 // and must use the same sequence number and timestamp base as the regular
 // audio channel.
 // This test checks the marker bit for the first packet and the consequent
 // packets of the same telephone event. Since it is specifically for DTMF
 // events, ignoring audio packets and sending kEmptyFrame instead of those.
 TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
   const char* kDtmfPayloadName = "telephone-event";
   const uint32_t kPayloadFrequency = 8000;
   const uint8_t kPayloadType = 126;
   ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
                    kDtmfPayloadName, kPayloadType, kPayloadFrequency, 0, 0));
   // For Telephone events, payload is not added to the registered payload list,
   // it will register only the payload used for audio stream.
   // Registering the payload again for audio stream with different payload name.
   const char* kPayloadName = "payload_name";
   ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
                    kPayloadName, kPayloadType, kPayloadFrequency, 1, 0));
   // Start time is arbitrary.
   uint32_t capture_timestamp = fake_clock_.TimeInMilliseconds();
   // DTMF event key=9, duration=500 and attenuationdB=10
   rtp_sender_audio_->SendTelephoneEvent(9, 500, 10);
   // During start, it takes the starting timestamp as last sent timestamp.
   // The duration is calculated as the difference of current and last sent
   // timestamp. So for first call it will skip since the duration is zero.
   ASSERT_TRUE(rtp_sender_audio_->SendAudio(
       AudioFrameType::kEmptyFrame, kPayloadType, capture_timestamp, nullptr, 0,
       /*absolute_capture_time_ms=0*/ 0));

   // DTMF Sample Length is (Frequency/1000) * Duration.
   // So in this case, it is (8000/1000) * 500 = 4000.
   // Sending it as two packets.
   ASSERT_TRUE(rtp_sender_audio_->SendAudio(AudioFrameType::kEmptyFrame,
                                            kPayloadType,
                                            capture_timestamp + 2000, nullptr, 0,
                                            /*absolute_capture_time_ms=0*/ 0));

   // Marker Bit should be set to 1 for first packet.
   EXPECT_TRUE(transport_.last_sent_packet().Marker());

   ASSERT_TRUE(rtp_sender_audio_->SendAudio(AudioFrameType::kEmptyFrame,
                                            kPayloadType,
                                            capture_timestamp + 4000, nullptr, 0,
                                            /*absolute_capture_time_ms=0*/ 0));
   // Marker Bit should be set to 0 for rest of the packets.
   EXPECT_FALSE(transport_.last_sent_packet().Marker());
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2019 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 "modules/rtp_rtcp/source/rtp_sender_audio.h"

	#include <memory>
	#include <vector>

	#include "api/transport/field_trial_based_config.h"
	#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
	#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
	#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
	#include "modules/rtp_rtcp/source/rtp_packet_received.h"
	#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
	#include "test/field_trial.h"
	#include "test/gmock.h"
	#include "test/gtest.h"

	namespace webrtc {

	namespace {
	enum : int { // The first valid value is 1.
	kAudioLevelExtensionId = 1,
	kAbsoluteCaptureTimeExtensionId = 2,
	};

	const uint16_t kSeqNum = 33;
	const uint32_t kSsrc = 725242;
	const uint8_t kAudioLevel = 0x5a;
	const uint64_t kStartTime = 123456789;

	using ::testing::ElementsAreArray;

	class LoopbackTransportTest : public webrtc::Transport {
	public:
	LoopbackTransportTest() {
	receivers_extensions_.Register<AudioLevel>(kAudioLevelExtensionId);
	receivers_extensions_.Register<AbsoluteCaptureTimeExtension>(
	kAbsoluteCaptureTimeExtensionId);
	}

	bool SendRtp(const uint8_t* data,
	size_t len,
	const PacketOptions& /options/) override {
	sent_packets_.push_back(RtpPacketReceived(&receivers_extensions_));
	EXPECT_TRUE(sent_packets_.back().Parse(data, len));
	return true;
	}
	bool SendRtcp(const uint8_t* data, size_t len) override { return false; }
	const RtpPacketReceived& last_sent_packet() { return sent_packets_.back(); }
	int packets_sent() { return sent_packets_.size(); }

	private:
	RtpHeaderExtensionMap receivers_extensions_;
	std::vector<RtpPacketReceived> sent_packets_;
	};

	} // namespace

	class RtpSenderAudioTest : public ::testing::Test {
	public:
	RtpSenderAudioTest()
	: fake_clock_(kStartTime),
	rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
	RtpRtcpInterface::Configuration config;
	config.audio = true;
	config.clock = &fake_clock_;
	config.outgoing_transport = &transport_;
	config.local_media_ssrc = kSsrc;
	return config;
	}())),
	rtp_sender_audio_(
	std::make_unique<RTPSenderAudio>(&fake_clock_,
	rtp_module_->RtpSender())) {
	rtp_module_->SetSequenceNumber(kSeqNum);
	}

	SimulatedClock fake_clock_;
	LoopbackTransportTest transport_;
	std::unique_ptr<ModuleRtpRtcpImpl2> rtp_module_;
	std::unique_ptr<RTPSenderAudio> rtp_sender_audio_;
	};

	TEST_F(RtpSenderAudioTest, SendAudio) {
	const char payload_name[] = "PAYLOAD_NAME";
	const uint8_t payload_type = 127;
	ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
	payload_name, payload_type, 48000, 0, 1500));
	uint8_t payload[] = {47, 11, 32, 93, 89};

	ASSERT_TRUE(
	rtp_sender_audio_->SendAudio(AudioFrameType::kAudioFrameCN, payload_type,
	4321, payload, sizeof(payload),
	/absolute_capture_timestamp_ms=/0));

	auto sent_payload = transport_.last_sent_packet().payload();
	EXPECT_THAT(sent_payload, ElementsAreArray(payload));
	}

	TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
	EXPECT_EQ(0, rtp_sender_audio_->SetAudioLevel(kAudioLevel));
	rtp_module_->RegisterRtpHeaderExtension(AudioLevel::Uri(),
	kAudioLevelExtensionId);

	const char payload_name[] = "PAYLOAD_NAME";
	const uint8_t payload_type = 127;
	ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
	payload_name, payload_type, 48000, 0, 1500));

	uint8_t payload[] = {47, 11, 32, 93, 89};

	ASSERT_TRUE(
	rtp_sender_audio_->SendAudio(AudioFrameType::kAudioFrameCN, payload_type,
	4321, payload, sizeof(payload),
	/absolute_capture_timestamp_ms=/0));

	auto sent_payload = transport_.last_sent_packet().payload();
	EXPECT_THAT(sent_payload, ElementsAreArray(payload));
	// Verify AudioLevel extension.
	bool voice_activity;
	uint8_t audio_level;
	EXPECT_TRUE(transport_.last_sent_packet().GetExtension<AudioLevel>(
	&voice_activity, &audio_level));
	EXPECT_EQ(kAudioLevel, audio_level);
	EXPECT_FALSE(voice_activity);
	}

	TEST_F(RtpSenderAudioTest, SendAudioWithoutAbsoluteCaptureTime) {
	constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
	const char payload_name[] = "audio";
	const uint8_t payload_type = 127;
	ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
	payload_name, payload_type, 48000, 0, 1500));
	uint8_t payload[] = {47, 11, 32, 93, 89};

	ASSERT_TRUE(rtp_sender_audio_->SendAudio(
	AudioFrameType::kAudioFrameCN, payload_type, 4321, payload,
	sizeof(payload), kAbsoluteCaptureTimestampMs));

	EXPECT_FALSE(transport_.last_sent_packet()
	.HasExtension<AbsoluteCaptureTimeExtension>());
	}

	// Essentially the same test as
	// SendAudioWithAbsoluteCaptureTimeWithCaptureClockOffset but with a field
	// trial. We will remove this test eventually.
	TEST_F(RtpSenderAudioTest, SendAudioWithAbsoluteCaptureTime) {
	// Recreate rtp_sender_audio_ with new field trial.
	test::ScopedFieldTrials field_trial(
	"WebRTC-IncludeCaptureClockOffset/Disabled/");
	rtp_sender_audio_ =
	std::make_unique<RTPSenderAudio>(&fake_clock_, rtp_module_->RtpSender());

	rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
	kAbsoluteCaptureTimeExtensionId);
	constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
	const char payload_name[] = "audio";
	const uint8_t payload_type = 127;
	ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
	payload_name, payload_type, 48000, 0, 1500));
	uint8_t payload[] = {47, 11, 32, 93, 89};

	ASSERT_TRUE(rtp_sender_audio_->SendAudio(
	AudioFrameType::kAudioFrameCN, payload_type, 4321, payload,
	sizeof(payload), kAbsoluteCaptureTimestampMs));

	auto absolute_capture_time =
	transport_.last_sent_packet()
	.GetExtension<AbsoluteCaptureTimeExtension>();
	EXPECT_TRUE(absolute_capture_time);
	EXPECT_EQ(
	absolute_capture_time->absolute_capture_timestamp,
	Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
	kAbsoluteCaptureTimestampMs)));
	EXPECT_FALSE(
	absolute_capture_time->estimated_capture_clock_offset.has_value());
	}

	TEST_F(RtpSenderAudioTest,
	SendAudioWithAbsoluteCaptureTimeWithCaptureClockOffset) {
	rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
	kAbsoluteCaptureTimeExtensionId);
	constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
	const char payload_name[] = "audio";
	const uint8_t payload_type = 127;
	ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
	payload_name, payload_type, 48000, 0, 1500));
	uint8_t payload[] = {47, 11, 32, 93, 89};

	ASSERT_TRUE(rtp_sender_audio_->SendAudio(
	AudioFrameType::kAudioFrameCN, payload_type, 4321, payload,
	sizeof(payload), kAbsoluteCaptureTimestampMs));

	auto absolute_capture_time =
	transport_.last_sent_packet()
	.GetExtension<AbsoluteCaptureTimeExtension>();
	EXPECT_TRUE(absolute_capture_time);
	EXPECT_EQ(
	absolute_capture_time->absolute_capture_timestamp,
	Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
	kAbsoluteCaptureTimestampMs)));
	EXPECT_TRUE(
	absolute_capture_time->estimated_capture_clock_offset.has_value());
	EXPECT_EQ(0, *absolute_capture_time->estimated_capture_clock_offset);
	}

	// As RFC4733, named telephone events are carried as part of the audio stream
	// and must use the same sequence number and timestamp base as the regular
	// audio channel.
	// This test checks the marker bit for the first packet and the consequent
	// packets of the same telephone event. Since it is specifically for DTMF
	// events, ignoring audio packets and sending kEmptyFrame instead of those.
	TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
	const char* kDtmfPayloadName = "telephone-event";
	const uint32_t kPayloadFrequency = 8000;
	const uint8_t kPayloadType = 126;
	ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
	kDtmfPayloadName, kPayloadType, kPayloadFrequency, 0, 0));
	// For Telephone events, payload is not added to the registered payload list,
	// it will register only the payload used for audio stream.
	// Registering the payload again for audio stream with different payload name.
	const char* kPayloadName = "payload_name";
	ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
	kPayloadName, kPayloadType, kPayloadFrequency, 1, 0));
	// Start time is arbitrary.
	uint32_t capture_timestamp = fake_clock_.TimeInMilliseconds();
	// DTMF event key=9, duration=500 and attenuationdB=10
	rtp_sender_audio_->SendTelephoneEvent(9, 500, 10);
	// During start, it takes the starting timestamp as last sent timestamp.
	// The duration is calculated as the difference of current and last sent
	// timestamp. So for first call it will skip since the duration is zero.
	ASSERT_TRUE(rtp_sender_audio_->SendAudio(
	AudioFrameType::kEmptyFrame, kPayloadType, capture_timestamp, nullptr, 0,
	/absolute_capture_time_ms=0/ 0));

	// DTMF Sample Length is (Frequency/1000) * Duration.
	// So in this case, it is (8000/1000) * 500 = 4000.
	// Sending it as two packets.
	ASSERT_TRUE(rtp_sender_audio_->SendAudio(AudioFrameType::kEmptyFrame,
	kPayloadType,
	capture_timestamp + 2000, nullptr, 0,
	/absolute_capture_time_ms=0/ 0));

	// Marker Bit should be set to 1 for first packet.
	EXPECT_TRUE(transport_.last_sent_packet().Marker());

	ASSERT_TRUE(rtp_sender_audio_->SendAudio(AudioFrameType::kEmptyFrame,
	kPayloadType,
	capture_timestamp + 4000, nullptr, 0,
	/absolute_capture_time_ms=0/ 0));
	// Marker Bit should be set to 0 for rest of the packets.
	EXPECT_FALSE(transport_.last_sent_packet().Marker());
	}

	} // namespace webrtc