webrtc/video_engine/test/send_stream_tests.cc

/*
 *  Copyright (c) 2013 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 "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/video_engine/test/common/fake_encoder.h"
#include "webrtc/video_engine/test/common/frame_generator.h"
#include "webrtc/video_engine/test/common/frame_generator_capturer.h"
#include "webrtc/video_engine/test/common/null_transport.h"
#include "webrtc/video_engine/new_include/call.h"
#include "webrtc/video_engine/new_include/video_send_stream.h"

namespace webrtc {

class SendTransportObserver : public test::NullTransport {
 public:
  explicit SendTransportObserver(unsigned long timeout_ms)
      : rtp_header_parser_(RtpHeaderParser::Create()),
        send_test_complete_(EventWrapper::Create()),
        timeout_ms_(timeout_ms) {}

  EventTypeWrapper Wait() { return send_test_complete_->Wait(timeout_ms_); }

 protected:
  scoped_ptr<RtpHeaderParser> rtp_header_parser_;
  scoped_ptr<EventWrapper> send_test_complete_;

 private:
  unsigned long timeout_ms_;
};

class VideoSendStreamTest : public ::testing::Test {
 public:
  VideoSendStreamTest() : fake_encoder_(Clock::GetRealTimeClock()) {}

 protected:
  static const uint32_t kSendSsrc;
  void RunSendTest(Call* call,
                   const VideoSendStream::Config& config,
                   SendTransportObserver* observer) {
    VideoSendStream* send_stream = call->CreateSendStream(config);
    scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
        test::FrameGeneratorCapturer::Create(
            send_stream->Input(),
            test::FrameGenerator::Create(320, 240, Clock::GetRealTimeClock()),
            30));
    send_stream->StartSend();
    frame_generator_capturer->Start();

    EXPECT_EQ(kEventSignaled, observer->Wait());

    frame_generator_capturer->Stop();
    send_stream->StopSend();
    call->DestroySendStream(send_stream);
  }

  VideoSendStream::Config GetSendTestConfig(Call* call) {
    VideoSendStream::Config config = call->GetDefaultSendConfig();
    config.encoder = &fake_encoder_;
    config.internal_source = false;
    config.rtp.ssrcs.push_back(kSendSsrc);
    test::FakeEncoder::SetCodecSettings(&config.codec, 1);
    return config;
  }

  test::FakeEncoder fake_encoder_;
};

const uint32_t VideoSendStreamTest::kSendSsrc = 0xC0FFEE;

TEST_F(VideoSendStreamTest, SendsSetSsrc) {
  class SendSsrcObserver : public SendTransportObserver {
   public:
    SendSsrcObserver() : SendTransportObserver(30 * 1000) {}

    virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
      RTPHeader header;
      EXPECT_TRUE(
          rtp_header_parser_->Parse(packet, static_cast<int>(length), &header));

      if (header.ssrc == kSendSsrc)
        send_test_complete_->Set();

      return true;
    }
  } observer;

  Call::Config call_config(&observer);
  scoped_ptr<Call> call(Call::Create(call_config));

  VideoSendStream::Config send_config = GetSendTestConfig(call.get());

  RunSendTest(call.get(), send_config, &observer);
}

TEST_F(VideoSendStreamTest, SupportsCName) {
  static std::string kCName = "PjQatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo=";
  class CNameObserver : public SendTransportObserver {
   public:
    CNameObserver() : SendTransportObserver(30 * 1000) {}

    virtual bool SendRTCP(const uint8_t* packet, size_t length) OVERRIDE {
      RTCPUtility::RTCPParserV2 parser(packet, length, true);
      EXPECT_TRUE(parser.IsValid());

      RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
      while (packet_type != RTCPUtility::kRtcpNotValidCode) {
        if (packet_type == RTCPUtility::kRtcpSdesChunkCode) {
          EXPECT_EQ(parser.Packet().CName.CName, kCName);
          send_test_complete_->Set();
        }

        packet_type = parser.Iterate();
      }

      return true;
    }
  } observer;

  Call::Config call_config(&observer);
  scoped_ptr<Call> call(Call::Create(call_config));

  VideoSendStream::Config send_config = GetSendTestConfig(call.get());
  send_config.rtp.c_name = kCName;

  RunSendTest(call.get(), send_config, &observer);
}

TEST_F(VideoSendStreamTest, RespondsToNack) {
  class NackObserver : public SendTransportObserver, webrtc::Transport {
   public:
    NackObserver()
        : SendTransportObserver(30 * 1000),
          thread_(ThreadWrapper::CreateThread(NackProcess, this)),
          send_call_receiver_(NULL),
          send_count_(0),
          ssrc_(0),
          nacked_sequence_number_(0) {}

    ~NackObserver() {
      EXPECT_TRUE(thread_->Stop());
    }

    void SetReceiver(PacketReceiver* send_call_receiver) {
      send_call_receiver_ = send_call_receiver;
    }

    // Sending NACKs must be done from a different "network" thread to prevent
    // violating locking orders. With this no locks are held prior to inserting
    // packets back into the sender.
    static bool NackProcess(void* observer) {
      return static_cast<NackObserver*>(observer)->SendNack();
    }

    bool SendNack() {
      NullReceiveStatistics null_stats;
      RTCPSender rtcp_sender(0, false, Clock::GetRealTimeClock(), &null_stats);
      EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(this));

      rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
      rtcp_sender.SetRemoteSSRC(ssrc_);

      RTCPSender::FeedbackState feedback_state;
      EXPECT_EQ(0, rtcp_sender.SendRTCP(
          feedback_state, kRtcpNack, 1, &nacked_sequence_number_));
      return false;
    }

    virtual int SendPacket(int channel, const void* data, int len) OVERRIDE {
      ADD_FAILURE()
          << "This should never be reached. Only a NACK should be sent.";
      return -1;
    }

    virtual int SendRTCPPacket(int channel,
                               const void* data,
                               int len) OVERRIDE {
      EXPECT_TRUE(send_call_receiver_->DeliverPacket(
          static_cast<const uint8_t*>(data), static_cast<size_t>(len)));
      return len;
    }

    virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
      EXPECT_TRUE(send_call_receiver_ != NULL);
      RTPHeader header;
      EXPECT_TRUE(
          rtp_header_parser_->Parse(packet, static_cast<int>(length), &header));

      // Nack second packet after receiving the third one.
      if (++send_count_ == 3) {
        ssrc_ = header.ssrc;
        nacked_sequence_number_ = header.sequenceNumber - 1;
        unsigned int id;
        EXPECT_TRUE(thread_->Start(id));
      }

      if (header.sequenceNumber == nacked_sequence_number_)
        send_test_complete_->Set();

      return true;
    }
   private:
    scoped_ptr<ThreadWrapper> thread_;
    PacketReceiver* send_call_receiver_;
    int send_count_;
    uint32_t ssrc_;
    uint16_t nacked_sequence_number_;
  } observer;

  Call::Config call_config(&observer);
  scoped_ptr<Call> call(Call::Create(call_config));
  observer.SetReceiver(call->Receiver());

  VideoSendStream::Config send_config = GetSendTestConfig(call.get());
  send_config.rtp.nack.rtp_history_ms = 1000;

  RunSendTest(call.get(), send_config, &observer);
}

}  // namespace webrtc