video/end_to_end_tests/ssrc_tests.cc - src - Git at Google

 /*
  *  Copyright 2018 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 <memory>

 #include "api/test/simulated_network.h"
 #include "call/fake_network_pipe.h"
 #include "call/simulated_network.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "rtc_base/task_queue_for_test.h"
 #include "test/call_test.h"
 #include "test/gtest.h"
 #include "test/rtcp_packet_parser.h"

 namespace webrtc {
 class SsrcEndToEndTest : public test::CallTest {
  protected:
   void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
 };

 TEST_F(SsrcEndToEndTest, ReceiverUsesLocalSsrc) {
   class SyncRtcpObserver : public test::EndToEndTest {
    public:
     SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {}

     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet, length));
       EXPECT_EQ(kReceiverLocalVideoSsrc, parser.sender_ssrc());
       observation_complete_.Set();

       return SEND_PACKET;
     }

     void PerformTest() override {
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for a receiver RTCP packet to be sent.";
     }
   } test;

   RunBaseTest(&test);
 }

 TEST_F(SsrcEndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
   class PacketInputObserver : public PacketReceiver {
    public:
     explicit PacketInputObserver(PacketReceiver* receiver)
         : receiver_(receiver) {}

     bool Wait() { return delivered_packet_.Wait(kDefaultTimeoutMs); }

    private:
     DeliveryStatus DeliverPacket(MediaType media_type,
                                  rtc::CopyOnWriteBuffer packet,
                                  int64_t packet_time_us) override {
       if (RtpHeaderParser::IsRtcp(packet.cdata(), packet.size())) {
         return receiver_->DeliverPacket(media_type, std::move(packet),
                                         packet_time_us);
       }
       DeliveryStatus delivery_status = receiver_->DeliverPacket(
           media_type, std::move(packet), packet_time_us);
       EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status);
       delivered_packet_.Set();
       return delivery_status;
     }

     PacketReceiver* receiver_;
     rtc::Event delivered_packet_;
   };

   std::unique_ptr<test::DirectTransport> send_transport;
   std::unique_ptr<test::DirectTransport> receive_transport;
   std::unique_ptr<PacketInputObserver> input_observer;

   SendTask(
       RTC_FROM_HERE, task_queue(),
       [this, &send_transport, &receive_transport, &input_observer]() {
         CreateCalls();

         send_transport = std::make_unique<test::DirectTransport>(
             task_queue(),
             std::make_unique<FakeNetworkPipe>(
                 Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
                                                BuiltInNetworkBehaviorConfig())),
             sender_call_.get(), payload_type_map_);
         receive_transport = std::make_unique<test::DirectTransport>(
             task_queue(),
             std::make_unique<FakeNetworkPipe>(
                 Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
                                                BuiltInNetworkBehaviorConfig())),
             receiver_call_.get(), payload_type_map_);
         input_observer =
             std::make_unique<PacketInputObserver>(receiver_call_->Receiver());
         send_transport->SetReceiver(input_observer.get());
         receive_transport->SetReceiver(sender_call_->Receiver());

         CreateSendConfig(1, 0, 0, send_transport.get());
         CreateMatchingReceiveConfigs(receive_transport.get());

         CreateVideoStreams();
         CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
                                      kDefaultHeight);
         Start();

         receiver_call_->DestroyVideoReceiveStream(video_receive_streams_[0]);
         video_receive_streams_.clear();
       });

   // Wait() waits for a received packet.
   EXPECT_TRUE(input_observer->Wait());

   SendTask(RTC_FROM_HERE, task_queue(),
            [this, &send_transport, &receive_transport]() {
              Stop();
              DestroyStreams();
              send_transport.reset();
              receive_transport.reset();
              DestroyCalls();
            });
 }

 void SsrcEndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
                                          bool send_single_ssrc_first) {
   class SendsSetSsrcs : public test::EndToEndTest {
    public:
     SendsSetSsrcs(const uint32_t* ssrcs,
                   size_t num_ssrcs,
                   bool send_single_ssrc_first)
         : EndToEndTest(kDefaultTimeoutMs),
           num_ssrcs_(num_ssrcs),
           send_single_ssrc_first_(send_single_ssrc_first),
           ssrcs_to_observe_(num_ssrcs),
           expect_single_ssrc_(send_single_ssrc_first),
           send_stream_(nullptr) {
       for (size_t i = 0; i < num_ssrcs; ++i)
         valid_ssrcs_[ssrcs[i]] = true;
     }

    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet, length));

       EXPECT_TRUE(valid_ssrcs_[rtp_packet.Ssrc()])
           << "Received unknown SSRC: " << rtp_packet.Ssrc();

       if (!valid_ssrcs_[rtp_packet.Ssrc()])
         observation_complete_.Set();

       if (!is_observed_[rtp_packet.Ssrc()]) {
         is_observed_[rtp_packet.Ssrc()] = true;
         --ssrcs_to_observe_;
         if (expect_single_ssrc_) {
           expect_single_ssrc_ = false;
           observation_complete_.Set();
         }
       }

       if (ssrcs_to_observe_ == 0)
         observation_complete_.Set();

       return SEND_PACKET;
     }

     size_t GetNumVideoStreams() const override { return num_ssrcs_; }

     // This test use other VideoStream settings than the the default settings
     // implemented in DefaultVideoStreamFactory. Therefore  this test implement
     // its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
     // in ModifyVideoConfigs.
     class VideoStreamFactory
         : public VideoEncoderConfig::VideoStreamFactoryInterface {
      public:
       VideoStreamFactory() {}

      private:
       std::vector<VideoStream> CreateEncoderStreams(
           int width,
           int height,
           const VideoEncoderConfig& encoder_config) override {
         std::vector<VideoStream> streams =
             test::CreateVideoStreams(width, height, encoder_config);
         // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
         for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
           streams[i].min_bitrate_bps = 10000;
           streams[i].target_bitrate_bps = 15000;
           streams[i].max_bitrate_bps = 20000;
         }
         return streams;
       }
     };

     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       encoder_config->video_stream_factory =
           new rtc::RefCountedObject<VideoStreamFactory>();
       video_encoder_config_all_streams_ = encoder_config->Copy();
       if (send_single_ssrc_first_)
         encoder_config->number_of_streams = 1;
     }

     void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
     }

     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out while waiting for "
                           << (send_single_ssrc_first_ ? "first SSRC."
                                                       : "SSRCs.");

       if (send_single_ssrc_first_) {
         // Set full simulcast and continue with the rest of the SSRCs.
         send_stream_->ReconfigureVideoEncoder(
             std::move(video_encoder_config_all_streams_));
         EXPECT_TRUE(Wait()) << "Timed out while waiting on additional SSRCs.";
       }
     }

    private:
     std::map<uint32_t, bool> valid_ssrcs_;
     std::map<uint32_t, bool> is_observed_;

     const size_t num_ssrcs_;
     const bool send_single_ssrc_first_;

     size_t ssrcs_to_observe_;
     bool expect_single_ssrc_;

     VideoSendStream* send_stream_;
     VideoEncoderConfig video_encoder_config_all_streams_;
   } test(kVideoSendSsrcs, num_ssrcs, send_single_ssrc_first);

   RunBaseTest(&test);
 }

 TEST_F(SsrcEndToEndTest, SendsSetSsrc) {
   TestSendsSetSsrcs(1, false);
 }

 TEST_F(SsrcEndToEndTest, SendsSetSimulcastSsrcs) {
   TestSendsSetSsrcs(kNumSimulcastStreams, false);
 }

 TEST_F(SsrcEndToEndTest, CanSwitchToUseAllSsrcs) {
   TestSendsSetSsrcs(kNumSimulcastStreams, true);
 }

 TEST_F(SsrcEndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
   class ObserveRedundantPayloads : public test::EndToEndTest {
    public:
     ObserveRedundantPayloads()
         : EndToEndTest(kDefaultTimeoutMs),
           ssrcs_to_observe_(kNumSimulcastStreams) {
       for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
         registered_rtx_ssrc_[kSendRtxSsrcs[i]] = true;
       }
     }

    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RtpPacket rtp_packet;
       EXPECT_TRUE(rtp_packet.Parse(packet, length));

       if (!registered_rtx_ssrc_[rtp_packet.Ssrc()])
         return SEND_PACKET;

       const bool packet_is_redundant_payload = rtp_packet.payload_size() > 0;

       if (!packet_is_redundant_payload)
         return SEND_PACKET;

       if (!observed_redundant_retransmission_[rtp_packet.Ssrc()]) {
         observed_redundant_retransmission_[rtp_packet.Ssrc()] = true;
         if (--ssrcs_to_observe_ == 0)
           observation_complete_.Set();
       }

       return SEND_PACKET;
     }

     size_t GetNumVideoStreams() const override { return kNumSimulcastStreams; }

     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
       encoder_config->max_bitrate_bps = 50000;
       for (auto& layer : encoder_config->simulcast_layers) {
         layer.min_bitrate_bps = 10000;
         layer.target_bitrate_bps = 15000;
         layer.max_bitrate_bps = 20000;
       }
       send_config->rtp.rtx.payload_type = kSendRtxPayloadType;

       for (size_t i = 0; i < kNumSimulcastStreams; ++i)
         send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);

       // Significantly higher than max bitrates for all video streams -> forcing
       // padding to trigger redundant padding on all RTX SSRCs.
       encoder_config->min_transmit_bitrate_bps = 100000;
     }

     void PerformTest() override {
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for redundant payloads on all SSRCs.";
     }

    private:
     size_t ssrcs_to_observe_;
     std::map<uint32_t, bool> observed_redundant_retransmission_;
     std::map<uint32_t, bool> registered_rtx_ssrc_;
   } test;

   RunBaseTest(&test);
 }
 }  // namespace webrtc
	/*
	* Copyright 2018 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 <memory>

	#include "api/test/simulated_network.h"
	#include "call/fake_network_pipe.h"
	#include "call/simulated_network.h"
	#include "modules/rtp_rtcp/source/rtp_packet.h"
	#include "rtc_base/task_queue_for_test.h"
	#include "test/call_test.h"
	#include "test/gtest.h"
	#include "test/rtcp_packet_parser.h"

	namespace webrtc {
	class SsrcEndToEndTest : public test::CallTest {
	protected:
	void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
	};

	TEST_F(SsrcEndToEndTest, ReceiverUsesLocalSsrc) {
	class SyncRtcpObserver : public test::EndToEndTest {
	public:
	SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {}

	Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
	test::RtcpPacketParser parser;
	EXPECT_TRUE(parser.Parse(packet, length));
	EXPECT_EQ(kReceiverLocalVideoSsrc, parser.sender_ssrc());
	observation_complete_.Set();

	return SEND_PACKET;
	}

	void PerformTest() override {
	EXPECT_TRUE(Wait())
	<< "Timed out while waiting for a receiver RTCP packet to be sent.";
	}
	} test;

	RunBaseTest(&test);
	}

	TEST_F(SsrcEndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
	class PacketInputObserver : public PacketReceiver {
	public:
	explicit PacketInputObserver(PacketReceiver* receiver)
	: receiver_(receiver) {}

	bool Wait() { return delivered_packet_.Wait(kDefaultTimeoutMs); }

	private:
	DeliveryStatus DeliverPacket(MediaType media_type,
	rtc::CopyOnWriteBuffer packet,
	int64_t packet_time_us) override {
	if (RtpHeaderParser::IsRtcp(packet.cdata(), packet.size())) {
	return receiver_->DeliverPacket(media_type, std::move(packet),
	packet_time_us);
	}
	DeliveryStatus delivery_status = receiver_->DeliverPacket(
	media_type, std::move(packet), packet_time_us);
	EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status);
	delivered_packet_.Set();
	return delivery_status;
	}

	PacketReceiver* receiver_;
	rtc::Event delivered_packet_;
	};

	std::unique_ptr<test::DirectTransport> send_transport;
	std::unique_ptr<test::DirectTransport> receive_transport;
	std::unique_ptr<PacketInputObserver> input_observer;

	SendTask(
	RTC_FROM_HERE, task_queue(),
	[this, &send_transport, &receive_transport, &input_observer]() {
	CreateCalls();

	send_transport = std::make_unique<test::DirectTransport>(
	task_queue(),
	std::make_unique<FakeNetworkPipe>(
	Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
	BuiltInNetworkBehaviorConfig())),
	sender_call_.get(), payload_type_map_);
	receive_transport = std::make_unique<test::DirectTransport>(
	task_queue(),
	std::make_unique<FakeNetworkPipe>(
	Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
	BuiltInNetworkBehaviorConfig())),
	receiver_call_.get(), payload_type_map_);
	input_observer =
	std::make_unique<PacketInputObserver>(receiver_call_->Receiver());
	send_transport->SetReceiver(input_observer.get());
	receive_transport->SetReceiver(sender_call_->Receiver());

	CreateSendConfig(1, 0, 0, send_transport.get());
	CreateMatchingReceiveConfigs(receive_transport.get());

	CreateVideoStreams();
	CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
	kDefaultHeight);
	Start();

	receiver_call_->DestroyVideoReceiveStream(video_receive_streams_[0]);
	video_receive_streams_.clear();
	});

	// Wait() waits for a received packet.
	EXPECT_TRUE(input_observer->Wait());

	SendTask(RTC_FROM_HERE, task_queue(),
	[this, &send_transport, &receive_transport]() {
	Stop();
	DestroyStreams();
	send_transport.reset();
	receive_transport.reset();
	DestroyCalls();
	});
	}

	void SsrcEndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
	bool send_single_ssrc_first) {
	class SendsSetSsrcs : public test::EndToEndTest {
	public:
	SendsSetSsrcs(const uint32_t* ssrcs,
	size_t num_ssrcs,
	bool send_single_ssrc_first)
	: EndToEndTest(kDefaultTimeoutMs),
	num_ssrcs_(num_ssrcs),
	send_single_ssrc_first_(send_single_ssrc_first),
	ssrcs_to_observe_(num_ssrcs),
	expect_single_ssrc_(send_single_ssrc_first),
	send_stream_(nullptr) {
	for (size_t i = 0; i < num_ssrcs; ++i)
	valid_ssrcs_[ssrcs[i]] = true;
	}

	private:
	Action OnSendRtp(const uint8_t* packet, size_t length) override {
	RtpPacket rtp_packet;
	EXPECT_TRUE(rtp_packet.Parse(packet, length));

	EXPECT_TRUE(valid_ssrcs_[rtp_packet.Ssrc()])
	<< "Received unknown SSRC: " << rtp_packet.Ssrc();

	if (!valid_ssrcs_[rtp_packet.Ssrc()])
	observation_complete_.Set();

	if (!is_observed_[rtp_packet.Ssrc()]) {
	is_observed_[rtp_packet.Ssrc()] = true;
	--ssrcs_to_observe_;
	if (expect_single_ssrc_) {
	expect_single_ssrc_ = false;
	observation_complete_.Set();
	}
	}

	if (ssrcs_to_observe_ == 0)
	observation_complete_.Set();

	return SEND_PACKET;
	}

	size_t GetNumVideoStreams() const override { return num_ssrcs_; }

	// This test use other VideoStream settings than the the default settings
	// implemented in DefaultVideoStreamFactory. Therefore this test implement
	// its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
	// in ModifyVideoConfigs.
	class VideoStreamFactory
	: public VideoEncoderConfig::VideoStreamFactoryInterface {
	public:
	VideoStreamFactory() {}

	private:
	std::vector<VideoStream> CreateEncoderStreams(
	int width,
	int height,
	const VideoEncoderConfig& encoder_config) override {
	std::vector<VideoStream> streams =
	test::CreateVideoStreams(width, height, encoder_config);
	// Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
	for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
	streams[i].min_bitrate_bps = 10000;
	streams[i].target_bitrate_bps = 15000;
	streams[i].max_bitrate_bps = 20000;
	}
	return streams;
	}
	};

	void ModifyVideoConfigs(
	VideoSendStream::Config* send_config,
	std::vector<VideoReceiveStream::Config>* receive_configs,
	VideoEncoderConfig* encoder_config) override {
	encoder_config->video_stream_factory =
	new rtc::RefCountedObject<VideoStreamFactory>();
	video_encoder_config_all_streams_ = encoder_config->Copy();
	if (send_single_ssrc_first_)
	encoder_config->number_of_streams = 1;
	}

	void OnVideoStreamsCreated(
	VideoSendStream* send_stream,
	const std::vector<VideoReceiveStream*>& receive_streams) override {
	send_stream_ = send_stream;
	}

	void PerformTest() override {
	EXPECT_TRUE(Wait()) << "Timed out while waiting for "
	<< (send_single_ssrc_first_ ? "first SSRC."
	: "SSRCs.");

	if (send_single_ssrc_first_) {
	// Set full simulcast and continue with the rest of the SSRCs.
	send_stream_->ReconfigureVideoEncoder(
	std::move(video_encoder_config_all_streams_));
	EXPECT_TRUE(Wait()) << "Timed out while waiting on additional SSRCs.";
	}
	}

	private:
	std::map<uint32_t, bool> valid_ssrcs_;
	std::map<uint32_t, bool> is_observed_;

	const size_t num_ssrcs_;
	const bool send_single_ssrc_first_;

	size_t ssrcs_to_observe_;
	bool expect_single_ssrc_;

	VideoSendStream* send_stream_;
	VideoEncoderConfig video_encoder_config_all_streams_;
	} test(kVideoSendSsrcs, num_ssrcs, send_single_ssrc_first);

	RunBaseTest(&test);
	}

	TEST_F(SsrcEndToEndTest, SendsSetSsrc) {
	TestSendsSetSsrcs(1, false);
	}

	TEST_F(SsrcEndToEndTest, SendsSetSimulcastSsrcs) {
	TestSendsSetSsrcs(kNumSimulcastStreams, false);
	}

	TEST_F(SsrcEndToEndTest, CanSwitchToUseAllSsrcs) {
	TestSendsSetSsrcs(kNumSimulcastStreams, true);
	}

	TEST_F(SsrcEndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
	class ObserveRedundantPayloads : public test::EndToEndTest {
	public:
	ObserveRedundantPayloads()
	: EndToEndTest(kDefaultTimeoutMs),
	ssrcs_to_observe_(kNumSimulcastStreams) {
	for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
	registered_rtx_ssrc_[kSendRtxSsrcs[i]] = true;
	}
	}

	private:
	Action OnSendRtp(const uint8_t* packet, size_t length) override {
	RtpPacket rtp_packet;
	EXPECT_TRUE(rtp_packet.Parse(packet, length));

	if (!registered_rtx_ssrc_[rtp_packet.Ssrc()])
	return SEND_PACKET;

	const bool packet_is_redundant_payload = rtp_packet.payload_size() > 0;

	if (!packet_is_redundant_payload)
	return SEND_PACKET;

	if (!observed_redundant_retransmission_[rtp_packet.Ssrc()]) {
	observed_redundant_retransmission_[rtp_packet.Ssrc()] = true;
	if (--ssrcs_to_observe_ == 0)
	observation_complete_.Set();
	}

	return SEND_PACKET;
	}

	size_t GetNumVideoStreams() const override { return kNumSimulcastStreams; }

	void ModifyVideoConfigs(
	VideoSendStream::Config* send_config,
	std::vector<VideoReceiveStream::Config>* receive_configs,
	VideoEncoderConfig* encoder_config) override {
	// Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
	encoder_config->max_bitrate_bps = 50000;
	for (auto& layer : encoder_config->simulcast_layers) {
	layer.min_bitrate_bps = 10000;
	layer.target_bitrate_bps = 15000;
	layer.max_bitrate_bps = 20000;
	}
	send_config->rtp.rtx.payload_type = kSendRtxPayloadType;

	for (size_t i = 0; i < kNumSimulcastStreams; ++i)
	send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);

	// Significantly higher than max bitrates for all video streams -> forcing
	// padding to trigger redundant padding on all RTX SSRCs.
	encoder_config->min_transmit_bitrate_bps = 100000;
	}

	void PerformTest() override {
	EXPECT_TRUE(Wait())
	<< "Timed out while waiting for redundant payloads on all SSRCs.";
	}

	private:
	size_t ssrcs_to_observe_;
	std::map<uint32_t, bool> observed_redundant_retransmission_;
	std::map<uint32_t, bool> registered_rtx_ssrc_;
	} test;

	RunBaseTest(&test);
	}
	} // namespace webrtc