webrtc/video/video_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 <algorithm>  // max

#include "testing/gtest/include/gtest/gtest.h"

#include "webrtc/call.h"
#include "webrtc/common_video/interface/i420_video_frame.h"
#include "webrtc/frame_callback.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/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/sleep.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/configurable_frame_size_encoder.h"
#include "webrtc/test/frame_generator_capturer.h"
#include "webrtc/test/null_transport.h"
#include "webrtc/test/rtp_rtcp_observer.h"
#include "webrtc/video/transport_adapter.h"
#include "webrtc/video_send_stream.h"

namespace webrtc {

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

 protected:
  void RunSendTest(Call* call,
                   const VideoSendStream::Config& config,
                   test::RtpRtcpObserver* observer) {
    send_stream_ = call->CreateVideoSendStream(config);
    scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
        test::FrameGeneratorCapturer::Create(
            send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock()));
    send_stream_->StartSending();
    frame_generator_capturer->Start();

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

    observer->StopSending();
    frame_generator_capturer->Stop();
    send_stream_->StopSending();
    call->DestroyVideoSendStream(send_stream_);
  }

  VideoSendStream::Config GetSendTestConfig(Call* call,
                                            size_t number_of_streams) {
    assert(number_of_streams <= kNumSendSsrcs);
    VideoSendStream::Config config = call->GetDefaultSendConfig();
    config.encoder = &fake_encoder_;
    config.internal_source = false;
    for (size_t i = 0; i < number_of_streams; ++i)
      config.rtp.ssrcs.push_back(kSendSsrcs[i]);
    config.pacing = true;
    test::FakeEncoder::SetCodecSettings(&config.codec, number_of_streams);
    config.codec.plType = kFakeSendPayloadType;
    return config;
  }

  void TestNackRetransmission(uint32_t retransmit_ssrc,
                              uint8_t retransmit_payload_type,
                              bool enable_pacing);

  void SendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);

  enum { kNumSendSsrcs = 3 };
  static const uint8_t kSendPayloadType;
  static const uint8_t kSendRtxPayloadType;
  static const uint8_t kFakeSendPayloadType;
  static const uint32_t kSendSsrc;
  static const uint32_t kSendRtxSsrc;
  static const uint32_t kSendSsrcs[kNumSendSsrcs];

  VideoSendStream* send_stream_;
  test::FakeEncoder fake_encoder_;
};

const uint8_t VideoSendStreamTest::kSendPayloadType = 100;
const uint8_t VideoSendStreamTest::kFakeSendPayloadType = 125;
const uint8_t VideoSendStreamTest::kSendRtxPayloadType = 98;
const uint32_t VideoSendStreamTest::kSendRtxSsrc = 0xBADCAFE;
const uint32_t VideoSendStreamTest::kSendSsrcs[kNumSendSsrcs] = { 0xC0FFED,
    0xC0FFEE, 0xC0FFEF };
const uint32_t VideoSendStreamTest::kSendSsrc =
    VideoSendStreamTest::kSendSsrcs[0];

void VideoSendStreamTest::SendsSetSsrcs(size_t num_ssrcs,
                                        bool send_single_ssrc_first) {
  class SendSsrcObserver : public test::RtpRtcpObserver {
   public:
    SendSsrcObserver(const uint32_t* ssrcs,
                     size_t num_ssrcs,
                     bool send_single_ssrc_first)
        : RtpRtcpObserver(30 * 1000),
          ssrcs_to_observe_(num_ssrcs),
          expect_single_ssrc_(send_single_ssrc_first) {
      for (size_t i = 0; i < num_ssrcs; ++i)
        valid_ssrcs_[ssrcs[i]] = true;
    }

    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
      RTPHeader header;
      EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header));

      // TODO(pbos): Reenable this part of the test when #1695 is resolved and
      //             all SSRCs are allocated on startup. This test was observed
      //             to fail on TSan as the codec gets set before the SSRCs are
      //             set up and some frames are sent on a random-generated SSRC
      //             before the correct SSRC gets set.
      //EXPECT_TRUE(valid_ssrcs_[header.ssrc])
      //    << "Received unknown SSRC: " << header.ssrc;
      //
      //if (!valid_ssrcs_[header.ssrc])
      //  observation_complete_->Set();

      if (!is_observed_[header.ssrc]) {
        is_observed_[header.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;
    }

   private:
    std::map<uint32_t, bool> valid_ssrcs_;
    std::map<uint32_t, bool> is_observed_;
    size_t ssrcs_to_observe_;
    bool expect_single_ssrc_;
  } observer(kSendSsrcs, num_ssrcs, send_single_ssrc_first);

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

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

  if (num_ssrcs > 1) {
    // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
    for (size_t i = 0; i < num_ssrcs; ++i) {
      send_config.codec.simulcastStream[i].minBitrate = 10;
      send_config.codec.simulcastStream[i].targetBitrate = 10;
      send_config.codec.simulcastStream[i].maxBitrate = 10;
    }
  }

  if (send_single_ssrc_first)
    send_config.codec.numberOfSimulcastStreams = 1;

  send_stream_ = call->CreateVideoSendStream(send_config);
  scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
      test::FrameGeneratorCapturer::Create(
          send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock()));
  send_stream_->StartSending();
  frame_generator_capturer->Start();

  EXPECT_EQ(kEventSignaled, observer.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_config.codec.numberOfSimulcastStreams =
        static_cast<unsigned char>(num_ssrcs);
    send_stream_->SetCodec(send_config.codec);
    EXPECT_EQ(kEventSignaled, observer.Wait())
        << "Timed out while waiting on additional SSRCs.";
  }

  observer.StopSending();
  frame_generator_capturer->Stop();
  send_stream_->StopSending();
  call->DestroyVideoSendStream(send_stream_);
}

TEST_F(VideoSendStreamTest, SendsSetSsrc) { SendsSetSsrcs(1, false); }

TEST_F(VideoSendStreamTest, SendsSetSimulcastSsrcs) {
  SendsSetSsrcs(kNumSendSsrcs, false);
}

TEST_F(VideoSendStreamTest, CanSwitchToUseAllSsrcs) {
  SendsSetSsrcs(kNumSendSsrcs, true);
}

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

    virtual Action OnSendRtcp(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);
          observation_complete_->Set();
        }

        packet_type = parser.Iterate();
      }

      return SEND_PACKET;
    }
  } observer;

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

  VideoSendStream::Config send_config = GetSendTestConfig(call.get(), 1);
  send_config.rtp.c_name = kCName;

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

TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
  static const uint8_t kAbsSendTimeExtensionId = 13;
  class AbsoluteSendTimeObserver : public test::RtpRtcpObserver {
   public:
    AbsoluteSendTimeObserver() : RtpRtcpObserver(30 * 1000) {
      EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
          kRtpExtensionAbsoluteSendTime, kAbsSendTimeExtensionId));
    }

    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
      RTPHeader header;
      EXPECT_TRUE(
          parser_->Parse(packet, static_cast<int>(length), &header));

      if (header.extension.absoluteSendTime > 0)
        observation_complete_->Set();

      return SEND_PACKET;
    }
  } observer;

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

  VideoSendStream::Config send_config = GetSendTestConfig(call.get(), 1);
  send_config.rtp.extensions.push_back(
      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));

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

TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
  static const uint8_t kTOffsetExtensionId = 13;
  class DelayedEncoder : public test::FakeEncoder {
   public:
    explicit DelayedEncoder(Clock* clock) : test::FakeEncoder(clock) {}
    virtual int32_t Encode(
        const I420VideoFrame& input_image,
        const CodecSpecificInfo* codec_specific_info,
        const std::vector<VideoFrameType>* frame_types) OVERRIDE {
      // A delay needs to be introduced to assure that we get a timestamp
      // offset.
      SleepMs(5);
      return FakeEncoder::Encode(input_image, codec_specific_info, frame_types);
    }
  } encoder(Clock::GetRealTimeClock());

  class TransmissionTimeOffsetObserver : public test::RtpRtcpObserver {
   public:
    TransmissionTimeOffsetObserver() : RtpRtcpObserver(30 * 1000) {
      EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
          kRtpExtensionTransmissionTimeOffset, kTOffsetExtensionId));
    }

    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
      RTPHeader header;
      EXPECT_TRUE(
          parser_->Parse(packet, static_cast<int>(length), &header));

      EXPECT_GT(header.extension.transmissionTimeOffset, 0);
      observation_complete_->Set();

      return SEND_PACKET;
    }
  } observer;

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

  VideoSendStream::Config send_config = GetSendTestConfig(call.get(), 1);
  send_config.encoder = &encoder;
  send_config.rtp.extensions.push_back(
      RtpExtension(RtpExtension::kTOffset, kTOffsetExtensionId));

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

class FakeReceiveStatistics : public NullReceiveStatistics {
 public:
  FakeReceiveStatistics(uint32_t send_ssrc,
                        uint32_t last_sequence_number,
                        uint32_t cumulative_lost,
                        uint8_t fraction_lost)
      : lossy_stats_(new LossyStatistician(last_sequence_number,
                                           cumulative_lost,
                                           fraction_lost)) {
    stats_map_[send_ssrc] = lossy_stats_.get();
  }

  virtual StatisticianMap GetActiveStatisticians() const OVERRIDE {
    return stats_map_;
  }

  virtual StreamStatistician* GetStatistician(uint32_t ssrc) const OVERRIDE {
    return lossy_stats_.get();
  }

 private:
  class LossyStatistician : public StreamStatistician {
   public:
    LossyStatistician(uint32_t extended_max_sequence_number,
                      uint32_t cumulative_lost,
                      uint8_t fraction_lost) {
      stats_.fraction_lost = fraction_lost;
      stats_.cumulative_lost = cumulative_lost;
      stats_.extended_max_sequence_number = extended_max_sequence_number;
    }
    virtual bool GetStatistics(Statistics* statistics, bool reset) OVERRIDE {
      *statistics = stats_;
      return true;
    }
    virtual void GetDataCounters(uint32_t* bytes_received,
                                 uint32_t* packets_received) const OVERRIDE {
      *bytes_received = 0;
      *packets_received = 0;
    }
    virtual uint32_t BitrateReceived() const OVERRIDE { return 0; }
    virtual void ResetStatistics() OVERRIDE {}
    virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
                                         int min_rtt) const OVERRIDE {
      return false;
    }

    virtual bool IsPacketInOrder(uint16_t sequence_number) const OVERRIDE {
      return true;
    }
    Statistics stats_;
  };

  scoped_ptr<LossyStatistician> lossy_stats_;
  StatisticianMap stats_map_;
};

TEST_F(VideoSendStreamTest, SupportsFec) {
  static const int kRedPayloadType = 118;
  static const int kUlpfecPayloadType = 119;
  class FecObserver : public test::RtpRtcpObserver {
   public:
    FecObserver()
        : RtpRtcpObserver(30 * 1000),
          transport_adapter_(SendTransport()),
          send_count_(0),
          received_media_(false),
          received_fec_(false) {}

    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
      RTPHeader header;
      EXPECT_TRUE(
          parser_->Parse(packet, static_cast<int>(length), &header));

      // Send lossy receive reports to trigger FEC enabling.
      if (send_count_++ % 2 != 0) {
        // Receive statistics reporting having lost 50% of the packets.
        FakeReceiveStatistics lossy_receive_stats(
            kSendSsrc, header.sequenceNumber, send_count_ / 2, 127);
        RTCPSender rtcp_sender(
            0, false, Clock::GetRealTimeClock(), &lossy_receive_stats);
        EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));

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

        RTCPSender::FeedbackState feedback_state;

        EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
      }

      EXPECT_EQ(kRedPayloadType, header.payloadType);

      uint8_t encapsulated_payload_type = packet[header.headerLength];

      if (encapsulated_payload_type == kUlpfecPayloadType) {
        received_fec_ = true;
      } else {
        received_media_ = true;
      }

      if (received_media_ && received_fec_)
        observation_complete_->Set();

      return SEND_PACKET;
    }

   private:
    internal::TransportAdapter transport_adapter_;
    int send_count_;
    bool received_media_;
    bool received_fec_;
  } observer;

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

  observer.SetReceivers(call->Receiver(), NULL);

  VideoSendStream::Config send_config = GetSendTestConfig(call.get(), 1);
  send_config.rtp.fec.red_payload_type = kRedPayloadType;
  send_config.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;

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

void VideoSendStreamTest::TestNackRetransmission(
    uint32_t retransmit_ssrc,
    uint8_t retransmit_payload_type,
    bool enable_pacing) {
  class NackObserver : public test::RtpRtcpObserver {
   public:
    explicit NackObserver(uint32_t retransmit_ssrc,
                          uint8_t retransmit_payload_type)
        : RtpRtcpObserver(30 * 1000),
          transport_adapter_(SendTransport()),
          send_count_(0),
          retransmit_ssrc_(retransmit_ssrc),
          retransmit_payload_type_(retransmit_payload_type),
          nacked_sequence_number_(0) {}

    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
      RTPHeader header;
      EXPECT_TRUE(
          parser_->Parse(packet, static_cast<int>(length), &header));

      // Nack second packet after receiving the third one.
      if (++send_count_ == 3) {
        nacked_sequence_number_ = header.sequenceNumber - 1;
        NullReceiveStatistics null_stats;
        RTCPSender rtcp_sender(
            0, false, Clock::GetRealTimeClock(), &null_stats);
        EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));

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

        RTCPSender::FeedbackState feedback_state;

        EXPECT_EQ(0,
                  rtcp_sender.SendRTCP(
                      feedback_state, kRtcpNack, 1, &nacked_sequence_number_));
      }

      uint16_t sequence_number = header.sequenceNumber;

      if (header.ssrc == retransmit_ssrc_ && retransmit_ssrc_ != kSendSsrc) {
        // Not kSendSsrc, assume correct RTX packet. Extract sequence number.
        const uint8_t* rtx_header = packet + header.headerLength;
        sequence_number = (rtx_header[0] << 8) + rtx_header[1];
      }

      if (sequence_number == nacked_sequence_number_) {
        EXPECT_EQ(retransmit_ssrc_, header.ssrc);
        EXPECT_EQ(retransmit_payload_type_, header.payloadType);
        observation_complete_->Set();
      }

      return SEND_PACKET;
    }

   private:
    internal::TransportAdapter transport_adapter_;
    int send_count_;
    uint32_t retransmit_ssrc_;
    uint8_t retransmit_payload_type_;
    uint16_t nacked_sequence_number_;
  } observer(retransmit_ssrc, retransmit_payload_type);

  Call::Config call_config(observer.SendTransport());
  scoped_ptr<Call> call(Call::Create(call_config));
  observer.SetReceivers(call->Receiver(), NULL);

  VideoSendStream::Config send_config = GetSendTestConfig(call.get(), 1);
  send_config.rtp.nack.rtp_history_ms = 1000;
  send_config.rtp.rtx.rtx_payload_type = retransmit_payload_type;
  send_config.pacing = enable_pacing;
  if (retransmit_ssrc != kSendSsrc)
    send_config.rtp.rtx.ssrcs.push_back(retransmit_ssrc);

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

TEST_F(VideoSendStreamTest, RetransmitsNack) {
  // Normal NACKs should use the send SSRC.
  TestNackRetransmission(kSendSsrc, kFakeSendPayloadType, false);
}

TEST_F(VideoSendStreamTest, RetransmitsNackOverRtx) {
  // NACKs over RTX should use a separate SSRC.
  TestNackRetransmission(kSendRtxSsrc, kSendRtxPayloadType, false);
}

TEST_F(VideoSendStreamTest, RetransmitsNackOverRtxWithPacing) {
  // NACKs over RTX should use a separate SSRC.
  TestNackRetransmission(kSendRtxSsrc, kSendRtxPayloadType, true);
}

TEST_F(VideoSendStreamTest, FragmentsAccordingToMaxPacketSize) {
  // Observer that verifies that the expected number of packets and bytes
  // arrive for each frame size, from start_size to stop_size.
  class FrameFragmentationObserver : public test::RtpRtcpObserver,
                                     public EncodedFrameObserver {
   public:
    FrameFragmentationObserver(size_t max_packet_size,
                               uint32_t start_size,
                               uint32_t stop_size,
                               test::ConfigurableFrameSizeEncoder* encoder)
        : RtpRtcpObserver(30 * 1000),
          max_packet_size_(max_packet_size),
          accumulated_size_(0),
          accumulated_payload_(0),
          stop_size_(stop_size),
          current_size_rtp_(start_size),
          current_size_frame_(start_size),
          encoder_(encoder) {
      // Fragmentation required, this test doesn't make sense without it.
      assert(stop_size > max_packet_size);
    }

    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
      RTPHeader header;
      EXPECT_TRUE(
          parser_->Parse(packet, static_cast<int>(length), &header));

      EXPECT_LE(length, max_packet_size_);

      accumulated_size_ += length;
      // Payload size = packet size - minus RTP header, padding and one byte
      // generic header.
      accumulated_payload_ +=
          length - (header.headerLength + header.paddingLength + 1);

      // Marker bit set indicates last packet of a frame.
      if (header.markerBit) {
        EXPECT_GE(accumulated_size_, current_size_rtp_);
        EXPECT_EQ(accumulated_payload_, current_size_rtp_);

        // Last packet of frame; reset counters.
        accumulated_size_ = 0;
        accumulated_payload_ = 0;
        if (current_size_rtp_ == stop_size_) {
          // Done! (Don't increase size again, might arrive more @ stop_size).
          observation_complete_->Set();
        } else {
          // Increase next expected frame size.
          ++current_size_rtp_;
        }
      }

      return SEND_PACKET;
    }

    virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
      // Increase frame size for next encoded frame, in the context of the
      // encoder thread.
      if (current_size_frame_ < stop_size_) {
        ++current_size_frame_;
      }
      encoder_->SetFrameSize(current_size_frame_);
    }

   private:
    size_t max_packet_size_;
    size_t accumulated_size_;
    size_t accumulated_payload_;

    uint32_t stop_size_;
    uint32_t current_size_rtp_;
    uint32_t current_size_frame_;
    test::ConfigurableFrameSizeEncoder* encoder_;
  };

  // Use a fake encoder to output a frame of every size in the range [90, 290],
  // for each size making sure that the exact number of payload bytes received
  // is correct and that packets are fragmented to respect max packet size.
  static const uint32_t kMaxPacketSize = 128;
  static const uint32_t start = 90;
  static const uint32_t stop = 290;

  test::ConfigurableFrameSizeEncoder encoder(stop);
  encoder.SetFrameSize(start);

  FrameFragmentationObserver observer(kMaxPacketSize, start, stop, &encoder);
  Call::Config call_config(observer.SendTransport());
  scoped_ptr<Call> call(Call::Create(call_config));

  VideoSendStream::Config send_config = GetSendTestConfig(call.get(), 1);
  send_config.encoder = &encoder;
  send_config.rtp.max_packet_size = kMaxPacketSize;
  send_config.post_encode_callback = &observer;

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

TEST_F(VideoSendStreamTest, CanChangeSendCodec) {
  static const uint8_t kFirstPayloadType = 121;
  static const uint8_t kSecondPayloadType = 122;

  class CodecChangeObserver : public test::RtpRtcpObserver {
   public:
    CodecChangeObserver(VideoSendStream** send_stream_ptr)
        : RtpRtcpObserver(30 * 1000),
          received_first_payload_(EventWrapper::Create()),
          send_stream_ptr_(send_stream_ptr) {}

    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
      RTPHeader header;
      EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header));

      if (header.payloadType == kFirstPayloadType) {
        received_first_payload_->Set();
      } else if (header.payloadType == kSecondPayloadType) {
        observation_complete_->Set();
      }

      return SEND_PACKET;
    }

    virtual EventTypeWrapper Wait() OVERRIDE {
      EXPECT_EQ(kEventSignaled, received_first_payload_->Wait(30 * 1000))
          << "Timed out while waiting for first payload.";

      EXPECT_TRUE((*send_stream_ptr_)->SetCodec(second_codec_));

      EXPECT_EQ(kEventSignaled, RtpRtcpObserver::Wait())
          << "Timed out while waiting for second payload type.";

      // Return OK regardless, prevents double error reporting.
      return kEventSignaled;
    }

    void SetSecondCodec(const VideoCodec& codec) {
      second_codec_ = codec;
    }

   private:
    scoped_ptr<EventWrapper> received_first_payload_;
    VideoSendStream** send_stream_ptr_;
    VideoCodec second_codec_;
  } observer(&send_stream_);

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

  std::vector<VideoCodec> codecs = call->GetVideoCodecs();
  ASSERT_GE(codecs.size(), 2u)
      << "Test needs at least 2 separate codecs to work.";
  codecs[0].plType = kFirstPayloadType;
  codecs[1].plType = kSecondPayloadType;
  observer.SetSecondCodec(codecs[1]);

  VideoSendStream::Config send_config = GetSendTestConfig(call.get(), 1);
  send_config.codec = codecs[0];
  send_config.encoder = NULL;

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

// The test will go through a number of phases.
// 1. Start sending packets.
// 2. As soon as the RTP stream has been detected, signal a low REMB value to
//    suspend the stream.
// 3. Wait until |kSuspendTimeFrames| have been captured without seeing any RTP
//    packets.
// 4. Signal a high REMB and then wait for the RTP stream to start again.
//    When the stream is detected again, the test ends.
TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
  static const int kSuspendTimeFrames = 60;  // Suspend for 2 seconds @ 30 fps.

  class RembObserver : public test::RtpRtcpObserver, public I420FrameCallback {
   public:
    RembObserver()
        : RtpRtcpObserver(30 * 1000),  // Timeout after 30 seconds.
          transport_adapter_(&transport_),
          clock_(Clock::GetRealTimeClock()),
          test_state_(kBeforeSuspend),
          rtp_count_(0),
          last_sequence_number_(0),
          suspended_frame_count_(0),
          low_remb_bps_(0),
          high_remb_bps_(0),
          crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}

    void SetReceiver(PacketReceiver* receiver) {
      transport_.SetReceiver(receiver);
    }

    virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
      // Receive statistics reporting having lost 0% of the packets.
      // This is needed for the send-side bitrate controller to work properly.
      CriticalSectionScoped lock(crit_sect_.get());
      SendRtcpFeedback(0);  // REMB is only sent if value is > 0.
      return SEND_PACKET;
    }

    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
      CriticalSectionScoped lock(crit_sect_.get());
      ++rtp_count_;
      RTPHeader header;
      EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header));
      last_sequence_number_ = header.sequenceNumber;

      if (test_state_ == kBeforeSuspend) {
        // The stream has started. Try to suspend it.
        SendRtcpFeedback(low_remb_bps_);
        test_state_ = kDuringSuspend;
      } else if (test_state_ == kDuringSuspend) {
        if (header.paddingLength == 0) {
          // Received non-padding packet during suspension period. Reset the
          // counter.
          // TODO(hlundin): We should probably make this test more advanced in
          // the future, so that it verifies that the bitrate can go below the
          // min_bitrate. This requires that the fake encoder sees the
          // min_bitrate, and never goes below it. See WebRTC Issue 2655.
          suspended_frame_count_ = 0;
        }
      } else if (test_state_ == kWaitingForPacket) {
        if (header.paddingLength == 0) {
          // Non-padding packet observed. Test is complete.
          observation_complete_->Set();
        }
      }

      return SEND_PACKET;
    }

    // This method implements the I420FrameCallback.
    void FrameCallback(I420VideoFrame* video_frame) OVERRIDE {
      CriticalSectionScoped lock(crit_sect_.get());
      if (test_state_ == kDuringSuspend &&
          ++suspended_frame_count_ > kSuspendTimeFrames) {
        SendRtcpFeedback(high_remb_bps_);
        test_state_ = kWaitingForPacket;
      }
    }

    void set_low_remb_bps(int value) { low_remb_bps_ = value; }

    void set_high_remb_bps(int value) { high_remb_bps_ = value; }

    virtual void Stop() { transport_.StopSending(); }

   private:
    enum TestState {
      kBeforeSuspend,
      kDuringSuspend,
      kWaitingForPacket,
      kAfterSuspend
    };

    virtual void SendRtcpFeedback(int remb_value) {
      FakeReceiveStatistics receive_stats(
          kSendSsrc, last_sequence_number_, rtp_count_, 0);
      RTCPSender rtcp_sender(0, false, clock_, &receive_stats);
      EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));

      rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
      rtcp_sender.SetRemoteSSRC(kSendSsrc);
      if (remb_value > 0) {
        rtcp_sender.SetREMBStatus(true);
        rtcp_sender.SetREMBData(remb_value, 0, NULL);
      }
      RTCPSender::FeedbackState feedback_state;
      EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
    }

    internal::TransportAdapter transport_adapter_;
    test::DirectTransport transport_;
    Clock* clock_;
    TestState test_state_;
    int rtp_count_;
    int last_sequence_number_;
    int suspended_frame_count_;
    int low_remb_bps_;
    int high_remb_bps_;
    scoped_ptr<CriticalSectionWrapper> crit_sect_;
  } observer;

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

  VideoSendStream::Config send_config = GetSendTestConfig(call.get(), 1);
  send_config.rtp.nack.rtp_history_ms = 1000;
  send_config.pre_encode_callback = &observer;
  send_config.suspend_below_min_bitrate = true;
  unsigned int min_bitrate_bps =
      send_config.codec.simulcastStream[0].minBitrate * 1000;
  observer.set_low_remb_bps(min_bitrate_bps - 10000);
  unsigned int threshold_window = std::max(min_bitrate_bps / 10, 10000u);
  ASSERT_GT(send_config.codec.simulcastStream[0].maxBitrate * 1000,
            min_bitrate_bps + threshold_window + 5000);
  observer.set_high_remb_bps(min_bitrate_bps + threshold_window + 5000);

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

TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
  class PacketObserver : public test::RtpRtcpObserver {
   public:
    PacketObserver()
        : RtpRtcpObserver(30 * 1000),  // Timeout after 30 seconds.
          clock_(Clock::GetRealTimeClock()),
          last_packet_time_ms_(-1),
          transport_adapter_(ReceiveTransport()),
          capturer_(NULL),
          crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}

    void SetCapturer(test::FrameGeneratorCapturer* capturer) {
      capturer_ = capturer;
    }

    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
      CriticalSectionScoped lock(crit_sect_.get());
      last_packet_time_ms_ = clock_->TimeInMilliseconds();
      capturer_->Stop();
      return SEND_PACKET;
    }

    virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
      CriticalSectionScoped lock(crit_sect_.get());
      const int kVideoMutedThresholdMs = 10000;
      if (last_packet_time_ms_ > 0 && clock_->TimeInMilliseconds() -
          last_packet_time_ms_ > kVideoMutedThresholdMs)
        observation_complete_->Set();
      // Receive statistics reporting having lost 50% of the packets.
      FakeReceiveStatistics receive_stats(kSendSsrcs[0], 1, 1, 0);
      RTCPSender rtcp_sender(
          0, false, Clock::GetRealTimeClock(), &receive_stats);
      EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));

      rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
      rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);

      RTCPSender::FeedbackState feedback_state;

      EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
      return SEND_PACKET;
    }

   private:
    Clock* clock_;
    int64_t last_packet_time_ms_;
    internal::TransportAdapter transport_adapter_;
    test::FrameGeneratorCapturer* capturer_;
    scoped_ptr<CriticalSectionWrapper> crit_sect_;
  } observer;

  Call::Config call_config(observer.SendTransport());
  scoped_ptr<Call> call(Call::Create(call_config));
  observer.SetReceivers(call->Receiver(), call->Receiver());

  VideoSendStream::Config send_config = GetSendTestConfig(call.get(), 3);

  send_stream_ = call->CreateVideoSendStream(send_config);
  scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
      test::FrameGeneratorCapturer::Create(
          send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock()));
  observer.SetCapturer(frame_generator_capturer.get());
  send_stream_->StartSending();
  frame_generator_capturer->Start();

  EXPECT_EQ(kEventSignaled, observer.Wait())
      << "Timed out while waiting for RTP packets to stop being sent.";

  observer.StopSending();
  frame_generator_capturer->Stop();
  send_stream_->StopSending();
  call->DestroyVideoSendStream(send_stream_);
}

}  // namespace webrtc