Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 767 insertions, 0 deletions

diff --git a/third_party/libwebrtc/pc/peer_connection_end_to_end_unittest.cc b/third_party/libwebrtc/pc/peer_connection_end_to_end_unittest.cc
new file mode 100644
index 0000000000..a21d455ec5
--- /dev/null
+++ b/third_party/libwebrtc/pc/peer_connection_end_to_end_unittest.cc
@@ -0,0 +1,767 @@
+/*
+ *  Copyright 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 <stdint.h>
+
+#include <cstddef>
+#include <limits>
+#include <memory>
+#include <string>
+#include <type_traits>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/L16/audio_decoder_L16.h"
+#include "api/audio_codecs/L16/audio_encoder_L16.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_decoder_factory_template.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory_template.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus_audio_decoder_factory.h"
+#include "api/audio_codecs/opus_audio_encoder_factory.h"
+#include "api/audio_options.h"
+#include "api/data_channel_interface.h"
+#include "api/media_stream_interface.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_error.h"
+#include "api/scoped_refptr.h"
+#include "media/sctp/sctp_transport_internal.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/physical_socket_server.h"
+#include "rtc_base/third_party/sigslot/sigslot.h"
+#include "rtc_base/thread.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+#ifdef WEBRTC_ANDROID
+#include "pc/test/android_test_initializer.h"
+#endif
+#include "pc/test/peer_connection_test_wrapper.h"
+// Notice that mockpeerconnectionobservers.h must be included after the above!
+#include "pc/test/mock_peer_connection_observers.h"
+#include "test/mock_audio_decoder.h"
+#include "test/mock_audio_decoder_factory.h"
+#include "test/mock_audio_encoder_factory.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Invoke;
+using ::testing::StrictMock;
+using ::testing::Values;
+
+using webrtc::DataChannelInterface;
+using webrtc::MediaStreamInterface;
+using webrtc::PeerConnectionInterface;
+using webrtc::SdpSemantics;
+
+namespace {
+
+const int kMaxWait = 25000;
+
+}  // namespace
+
+class PeerConnectionEndToEndBaseTest : public sigslot::has_slots<>,
+                                       public ::testing::Test {
+ public:
+  typedef std::vector<rtc::scoped_refptr<DataChannelInterface>> DataChannelList;
+
+  explicit PeerConnectionEndToEndBaseTest(SdpSemantics sdp_semantics)
+      : network_thread_(std::make_unique<rtc::Thread>(&pss_)),
+        worker_thread_(rtc::Thread::Create()) {
+    RTC_CHECK(network_thread_->Start());
+    RTC_CHECK(worker_thread_->Start());
+    caller_ = rtc::make_ref_counted<PeerConnectionTestWrapper>(
+        "caller", &pss_, network_thread_.get(), worker_thread_.get());
+    callee_ = rtc::make_ref_counted<PeerConnectionTestWrapper>(
+        "callee", &pss_, network_thread_.get(), worker_thread_.get());
+    webrtc::PeerConnectionInterface::IceServer ice_server;
+    ice_server.uri = "stun:stun.l.google.com:19302";
+    config_.servers.push_back(ice_server);
+    config_.sdp_semantics = sdp_semantics;
+
+#ifdef WEBRTC_ANDROID
+    webrtc::InitializeAndroidObjects();
+#endif
+  }
+
+  void CreatePcs(
+      rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory1,
+      rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory1,
+      rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory2,
+      rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory2) {
+    EXPECT_TRUE(caller_->CreatePc(config_, audio_encoder_factory1,
+                                  audio_decoder_factory1));
+    EXPECT_TRUE(callee_->CreatePc(config_, audio_encoder_factory2,
+                                  audio_decoder_factory2));
+    PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
+
+    caller_->SignalOnDataChannel.connect(
+        this, &PeerConnectionEndToEndBaseTest::OnCallerAddedDataChanel);
+    callee_->SignalOnDataChannel.connect(
+        this, &PeerConnectionEndToEndBaseTest::OnCalleeAddedDataChannel);
+  }
+
+  void CreatePcs(
+      rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
+      rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory) {
+    CreatePcs(audio_encoder_factory, audio_decoder_factory,
+              audio_encoder_factory, audio_decoder_factory);
+  }
+
+  void GetAndAddUserMedia() {
+    cricket::AudioOptions audio_options;
+    GetAndAddUserMedia(true, audio_options, true);
+  }
+
+  void GetAndAddUserMedia(bool audio,
+                          const cricket::AudioOptions& audio_options,
+                          bool video) {
+    caller_->GetAndAddUserMedia(audio, audio_options, video);
+    callee_->GetAndAddUserMedia(audio, audio_options, video);
+  }
+
+  void Negotiate() {
+    caller_->CreateOffer(
+        webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
+  }
+
+  void WaitForCallEstablished() {
+    caller_->WaitForCallEstablished();
+    callee_->WaitForCallEstablished();
+  }
+
+  void WaitForConnection() {
+    caller_->WaitForConnection();
+    callee_->WaitForConnection();
+  }
+
+  void OnCallerAddedDataChanel(DataChannelInterface* dc) {
+    caller_signaled_data_channels_.push_back(
+        rtc::scoped_refptr<DataChannelInterface>(dc));
+  }
+
+  void OnCalleeAddedDataChannel(DataChannelInterface* dc) {
+    callee_signaled_data_channels_.push_back(
+        rtc::scoped_refptr<DataChannelInterface>(dc));
+  }
+
+  // Tests that `dc1` and `dc2` can send to and receive from each other.
+  void TestDataChannelSendAndReceive(DataChannelInterface* dc1,
+                                     DataChannelInterface* dc2,
+                                     size_t size = 6) {
+    std::unique_ptr<webrtc::MockDataChannelObserver> dc1_observer(
+        new webrtc::MockDataChannelObserver(dc1));
+
+    std::unique_ptr<webrtc::MockDataChannelObserver> dc2_observer(
+        new webrtc::MockDataChannelObserver(dc2));
+
+    static const std::string kDummyData =
+        "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/";
+    webrtc::DataBuffer buffer("");
+
+    size_t sizeLeft = size;
+    while (sizeLeft > 0) {
+      size_t chunkSize =
+          sizeLeft > kDummyData.length() ? kDummyData.length() : sizeLeft;
+      buffer.data.AppendData(kDummyData.data(), chunkSize);
+      sizeLeft -= chunkSize;
+    }
+
+    EXPECT_TRUE(dc1->Send(buffer));
+    EXPECT_EQ_WAIT(buffer.data,
+                   rtc::CopyOnWriteBuffer(dc2_observer->last_message()),
+                   kMaxWait);
+
+    EXPECT_TRUE(dc2->Send(buffer));
+    EXPECT_EQ_WAIT(buffer.data,
+                   rtc::CopyOnWriteBuffer(dc1_observer->last_message()),
+                   kMaxWait);
+
+    EXPECT_EQ(1U, dc1_observer->received_message_count());
+    EXPECT_EQ(size, dc1_observer->last_message().length());
+    EXPECT_EQ(1U, dc2_observer->received_message_count());
+    EXPECT_EQ(size, dc2_observer->last_message().length());
+  }
+
+  void WaitForDataChannelsToOpen(DataChannelInterface* local_dc,
+                                 const DataChannelList& remote_dc_list,
+                                 size_t remote_dc_index) {
+    EXPECT_EQ_WAIT(DataChannelInterface::kOpen, local_dc->state(), kMaxWait);
+
+    ASSERT_TRUE_WAIT(remote_dc_list.size() > remote_dc_index, kMaxWait);
+    EXPECT_EQ_WAIT(DataChannelInterface::kOpen,
+                   remote_dc_list[remote_dc_index]->state(), kMaxWait);
+    EXPECT_EQ(local_dc->id(), remote_dc_list[remote_dc_index]->id());
+  }
+
+  void CloseDataChannels(DataChannelInterface* local_dc,
+                         const DataChannelList& remote_dc_list,
+                         size_t remote_dc_index) {
+    local_dc->Close();
+    EXPECT_EQ_WAIT(DataChannelInterface::kClosed, local_dc->state(), kMaxWait);
+    EXPECT_EQ_WAIT(DataChannelInterface::kClosed,
+                   remote_dc_list[remote_dc_index]->state(), kMaxWait);
+  }
+
+ protected:
+  rtc::AutoThread main_thread_;
+  rtc::PhysicalSocketServer pss_;
+  std::unique_ptr<rtc::Thread> network_thread_;
+  std::unique_ptr<rtc::Thread> worker_thread_;
+  rtc::scoped_refptr<PeerConnectionTestWrapper> caller_;
+  rtc::scoped_refptr<PeerConnectionTestWrapper> callee_;
+  DataChannelList caller_signaled_data_channels_;
+  DataChannelList callee_signaled_data_channels_;
+  webrtc::PeerConnectionInterface::RTCConfiguration config_;
+};
+
+class PeerConnectionEndToEndTest
+    : public PeerConnectionEndToEndBaseTest,
+      public ::testing::WithParamInterface<SdpSemantics> {
+ protected:
+  PeerConnectionEndToEndTest() : PeerConnectionEndToEndBaseTest(GetParam()) {}
+};
+
+namespace {
+
+std::unique_ptr<webrtc::AudioDecoder> CreateForwardingMockDecoder(
+    std::unique_ptr<webrtc::AudioDecoder> real_decoder) {
+  class ForwardingMockDecoder : public StrictMock<webrtc::MockAudioDecoder> {
+   public:
+    explicit ForwardingMockDecoder(std::unique_ptr<AudioDecoder> decoder)
+        : decoder_(std::move(decoder)) {}
+
+   private:
+    std::unique_ptr<AudioDecoder> decoder_;
+  };
+
+  const auto dec = real_decoder.get();  // For lambda capturing.
+  auto mock_decoder =
+      std::make_unique<ForwardingMockDecoder>(std::move(real_decoder));
+  EXPECT_CALL(*mock_decoder, Channels())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke([dec] { return dec->Channels(); }));
+  EXPECT_CALL(*mock_decoder, DecodeInternal(_, _, _, _, _))
+      .Times(AtLeast(1))
+      .WillRepeatedly(
+          Invoke([dec](const uint8_t* encoded, size_t encoded_len,
+                       int sample_rate_hz, int16_t* decoded,
+                       webrtc::AudioDecoder::SpeechType* speech_type) {
+            return dec->Decode(encoded, encoded_len, sample_rate_hz,
+                               std::numeric_limits<size_t>::max(), decoded,
+                               speech_type);
+          }));
+  EXPECT_CALL(*mock_decoder, Die());
+  EXPECT_CALL(*mock_decoder, HasDecodePlc()).WillRepeatedly(Invoke([dec] {
+    return dec->HasDecodePlc();
+  }));
+  EXPECT_CALL(*mock_decoder, PacketDuration(_, _))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke([dec](const uint8_t* encoded, size_t encoded_len) {
+        return dec->PacketDuration(encoded, encoded_len);
+      }));
+  EXPECT_CALL(*mock_decoder, SampleRateHz())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke([dec] { return dec->SampleRateHz(); }));
+
+  return std::move(mock_decoder);
+}
+
+rtc::scoped_refptr<webrtc::AudioDecoderFactory>
+CreateForwardingMockDecoderFactory(
+    webrtc::AudioDecoderFactory* real_decoder_factory) {
+  rtc::scoped_refptr<webrtc::MockAudioDecoderFactory> mock_decoder_factory =
+      rtc::make_ref_counted<StrictMock<webrtc::MockAudioDecoderFactory>>();
+  EXPECT_CALL(*mock_decoder_factory, GetSupportedDecoders())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke([real_decoder_factory] {
+        return real_decoder_factory->GetSupportedDecoders();
+      }));
+  EXPECT_CALL(*mock_decoder_factory, IsSupportedDecoder(_))
+      .Times(AtLeast(1))
+      .WillRepeatedly(
+          Invoke([real_decoder_factory](const webrtc::SdpAudioFormat& format) {
+            return real_decoder_factory->IsSupportedDecoder(format);
+          }));
+  EXPECT_CALL(*mock_decoder_factory, MakeAudioDecoderMock(_, _, _))
+      .Times(AtLeast(2))
+      .WillRepeatedly(
+          Invoke([real_decoder_factory](
+                     const webrtc::SdpAudioFormat& format,
+                     absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
+                     std::unique_ptr<webrtc::AudioDecoder>* return_value) {
+            auto real_decoder =
+                real_decoder_factory->MakeAudioDecoder(format, codec_pair_id);
+            *return_value =
+                real_decoder
+                    ? CreateForwardingMockDecoder(std::move(real_decoder))
+                    : nullptr;
+          }));
+  return mock_decoder_factory;
+}
+
+struct AudioEncoderUnicornSparklesRainbow {
+  using Config = webrtc::AudioEncoderL16::Config;
+  static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
+    if (absl::EqualsIgnoreCase(format.name, "UnicornSparklesRainbow")) {
+      const webrtc::SdpAudioFormat::Parameters expected_params = {
+          {"num_horns", "1"}};
+      EXPECT_EQ(expected_params, format.parameters);
+      format.parameters.clear();
+      format.name = "L16";
+      return webrtc::AudioEncoderL16::SdpToConfig(format);
+    } else {
+      return absl::nullopt;
+    }
+  }
+  static void AppendSupportedEncoders(
+      std::vector<webrtc::AudioCodecSpec>* specs) {
+    std::vector<webrtc::AudioCodecSpec> new_specs;
+    webrtc::AudioEncoderL16::AppendSupportedEncoders(&new_specs);
+    for (auto& spec : new_specs) {
+      spec.format.name = "UnicornSparklesRainbow";
+      EXPECT_TRUE(spec.format.parameters.empty());
+      spec.format.parameters.emplace("num_horns", "1");
+      specs->push_back(spec);
+    }
+  }
+  static webrtc::AudioCodecInfo QueryAudioEncoder(const Config& config) {
+    return webrtc::AudioEncoderL16::QueryAudioEncoder(config);
+  }
+  static std::unique_ptr<webrtc::AudioEncoder> MakeAudioEncoder(
+      const Config& config,
+      int payload_type,
+      absl::optional<webrtc::AudioCodecPairId> codec_pair_id = absl::nullopt) {
+    return webrtc::AudioEncoderL16::MakeAudioEncoder(config, payload_type,
+                                                     codec_pair_id);
+  }
+};
+
+struct AudioDecoderUnicornSparklesRainbow {
+  using Config = webrtc::AudioDecoderL16::Config;
+  static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
+    if (absl::EqualsIgnoreCase(format.name, "UnicornSparklesRainbow")) {
+      const webrtc::SdpAudioFormat::Parameters expected_params = {
+          {"num_horns", "1"}};
+      EXPECT_EQ(expected_params, format.parameters);
+      format.parameters.clear();
+      format.name = "L16";
+      return webrtc::AudioDecoderL16::SdpToConfig(format);
+    } else {
+      return absl::nullopt;
+    }
+  }
+  static void AppendSupportedDecoders(
+      std::vector<webrtc::AudioCodecSpec>* specs) {
+    std::vector<webrtc::AudioCodecSpec> new_specs;
+    webrtc::AudioDecoderL16::AppendSupportedDecoders(&new_specs);
+    for (auto& spec : new_specs) {
+      spec.format.name = "UnicornSparklesRainbow";
+      EXPECT_TRUE(spec.format.parameters.empty());
+      spec.format.parameters.emplace("num_horns", "1");
+      specs->push_back(spec);
+    }
+  }
+  static std::unique_ptr<webrtc::AudioDecoder> MakeAudioDecoder(
+      const Config& config,
+      absl::optional<webrtc::AudioCodecPairId> codec_pair_id = absl::nullopt) {
+    return webrtc::AudioDecoderL16::MakeAudioDecoder(config, codec_pair_id);
+  }
+};
+
+}  // namespace
+
+TEST_P(PeerConnectionEndToEndTest, Call) {
+  rtc::scoped_refptr<webrtc::AudioDecoderFactory> real_decoder_factory =
+      webrtc::CreateOpusAudioDecoderFactory();
+  CreatePcs(webrtc::CreateOpusAudioEncoderFactory(),
+            CreateForwardingMockDecoderFactory(real_decoder_factory.get()));
+  GetAndAddUserMedia();
+  Negotiate();
+  WaitForCallEstablished();
+}
+
+#if defined(IS_FUCHSIA)
+TEST_P(PeerConnectionEndToEndTest, CallWithSdesKeyNegotiation) {
+  config_.enable_dtls_srtp = false;
+  CreatePcs(webrtc::CreateOpusAudioEncoderFactory(),
+            webrtc::CreateOpusAudioDecoderFactory());
+  GetAndAddUserMedia();
+  Negotiate();
+  WaitForCallEstablished();
+}
+#endif
+
+TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) {
+  class IdLoggingAudioEncoderFactory : public webrtc::AudioEncoderFactory {
+   public:
+    IdLoggingAudioEncoderFactory(
+        rtc::scoped_refptr<AudioEncoderFactory> real_factory,
+        std::vector<webrtc::AudioCodecPairId>* const codec_ids)
+        : fact_(real_factory), codec_ids_(codec_ids) {}
+    std::vector<webrtc::AudioCodecSpec> GetSupportedEncoders() override {
+      return fact_->GetSupportedEncoders();
+    }
+    absl::optional<webrtc::AudioCodecInfo> QueryAudioEncoder(
+        const webrtc::SdpAudioFormat& format) override {
+      return fact_->QueryAudioEncoder(format);
+    }
+    std::unique_ptr<webrtc::AudioEncoder> MakeAudioEncoder(
+        int payload_type,
+        const webrtc::SdpAudioFormat& format,
+        absl::optional<webrtc::AudioCodecPairId> codec_pair_id) override {
+      EXPECT_TRUE(codec_pair_id.has_value());
+      codec_ids_->push_back(*codec_pair_id);
+      return fact_->MakeAudioEncoder(payload_type, format, codec_pair_id);
+    }
+
+   private:
+    const rtc::scoped_refptr<webrtc::AudioEncoderFactory> fact_;
+    std::vector<webrtc::AudioCodecPairId>* const codec_ids_;
+  };
+
+  class IdLoggingAudioDecoderFactory : public webrtc::AudioDecoderFactory {
+   public:
+    IdLoggingAudioDecoderFactory(
+        rtc::scoped_refptr<AudioDecoderFactory> real_factory,
+        std::vector<webrtc::AudioCodecPairId>* const codec_ids)
+        : fact_(real_factory), codec_ids_(codec_ids) {}
+    std::vector<webrtc::AudioCodecSpec> GetSupportedDecoders() override {
+      return fact_->GetSupportedDecoders();
+    }
+    bool IsSupportedDecoder(const webrtc::SdpAudioFormat& format) override {
+      return fact_->IsSupportedDecoder(format);
+    }
+    std::unique_ptr<webrtc::AudioDecoder> MakeAudioDecoder(
+        const webrtc::SdpAudioFormat& format,
+        absl::optional<webrtc::AudioCodecPairId> codec_pair_id) override {
+      EXPECT_TRUE(codec_pair_id.has_value());
+      codec_ids_->push_back(*codec_pair_id);
+      return fact_->MakeAudioDecoder(format, codec_pair_id);
+    }
+
+   private:
+    const rtc::scoped_refptr<webrtc::AudioDecoderFactory> fact_;
+    std::vector<webrtc::AudioCodecPairId>* const codec_ids_;
+  };
+
+  std::vector<webrtc::AudioCodecPairId> encoder_id1, encoder_id2, decoder_id1,
+      decoder_id2;
+  CreatePcs(rtc::make_ref_counted<IdLoggingAudioEncoderFactory>(
+                webrtc::CreateAudioEncoderFactory<
+                    AudioEncoderUnicornSparklesRainbow>(),
+                &encoder_id1),
+            rtc::make_ref_counted<IdLoggingAudioDecoderFactory>(
+                webrtc::CreateAudioDecoderFactory<
+                    AudioDecoderUnicornSparklesRainbow>(),
+                &decoder_id1),
+            rtc::make_ref_counted<IdLoggingAudioEncoderFactory>(
+                webrtc::CreateAudioEncoderFactory<
+                    AudioEncoderUnicornSparklesRainbow>(),
+                &encoder_id2),
+            rtc::make_ref_counted<IdLoggingAudioDecoderFactory>(
+                webrtc::CreateAudioDecoderFactory<
+                    AudioDecoderUnicornSparklesRainbow>(),
+                &decoder_id2));
+  GetAndAddUserMedia();
+  Negotiate();
+  WaitForCallEstablished();
+
+  // Each codec factory has been used to create one codec. The first pair got
+  // the same ID because they were passed to the same PeerConnectionFactory,
+  // and the second pair got the same ID---but these two IDs are not equal,
+  // because each PeerConnectionFactory has its own ID.
+  EXPECT_EQ(1U, encoder_id1.size());
+  EXPECT_EQ(1U, encoder_id2.size());
+  EXPECT_EQ(encoder_id1, decoder_id1);
+  EXPECT_EQ(encoder_id2, decoder_id2);
+  EXPECT_NE(encoder_id1, encoder_id2);
+}
+
+#ifdef WEBRTC_HAVE_SCTP
+// Verifies that a DataChannel created before the negotiation can transition to
+// "OPEN" and transfer data.
+TEST_P(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+  rtc::scoped_refptr<DataChannelInterface> caller_dc(
+      caller_->CreateDataChannel("data", init));
+  rtc::scoped_refptr<DataChannelInterface> callee_dc(
+      callee_->CreateDataChannel("data", init));
+
+  Negotiate();
+  WaitForConnection();
+
+  WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 0);
+  WaitForDataChannelsToOpen(callee_dc.get(), caller_signaled_data_channels_, 0);
+
+  TestDataChannelSendAndReceive(caller_dc.get(),
+                                callee_signaled_data_channels_[0].get());
+  TestDataChannelSendAndReceive(callee_dc.get(),
+                                caller_signaled_data_channels_[0].get());
+
+  CloseDataChannels(caller_dc.get(), callee_signaled_data_channels_, 0);
+  CloseDataChannels(callee_dc.get(), caller_signaled_data_channels_, 0);
+}
+
+// Verifies that a DataChannel created after the negotiation can transition to
+// "OPEN" and transfer data.
+TEST_P(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+
+  // This DataChannel is for creating the data content in the negotiation.
+  rtc::scoped_refptr<DataChannelInterface> dummy(
+      caller_->CreateDataChannel("data", init));
+  Negotiate();
+  WaitForConnection();
+
+  // Wait for the data channel created pre-negotiation to be opened.
+  WaitForDataChannelsToOpen(dummy.get(), callee_signaled_data_channels_, 0);
+
+  // Create new DataChannels after the negotiation and verify their states.
+  rtc::scoped_refptr<DataChannelInterface> caller_dc(
+      caller_->CreateDataChannel("hello", init));
+  rtc::scoped_refptr<DataChannelInterface> callee_dc(
+      callee_->CreateDataChannel("hello", init));
+
+  WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 1);
+  WaitForDataChannelsToOpen(callee_dc.get(), caller_signaled_data_channels_, 0);
+
+  TestDataChannelSendAndReceive(caller_dc.get(),
+                                callee_signaled_data_channels_[1].get());
+  TestDataChannelSendAndReceive(callee_dc.get(),
+                                caller_signaled_data_channels_[0].get());
+
+  CloseDataChannels(caller_dc.get(), callee_signaled_data_channels_, 1);
+  CloseDataChannels(callee_dc.get(), caller_signaled_data_channels_, 0);
+}
+
+// Verifies that a DataChannel created can transfer large messages.
+TEST_P(PeerConnectionEndToEndTest, CreateDataChannelLargeTransfer) {
+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+
+  // This DataChannel is for creating the data content in the negotiation.
+  rtc::scoped_refptr<DataChannelInterface> dummy(
+      caller_->CreateDataChannel("data", init));
+  Negotiate();
+  WaitForConnection();
+
+  // Wait for the data channel created pre-negotiation to be opened.
+  WaitForDataChannelsToOpen(dummy.get(), callee_signaled_data_channels_, 0);
+
+  // Create new DataChannels after the negotiation and verify their states.
+  rtc::scoped_refptr<DataChannelInterface> caller_dc(
+      caller_->CreateDataChannel("hello", init));
+  rtc::scoped_refptr<DataChannelInterface> callee_dc(
+      callee_->CreateDataChannel("hello", init));
+
+  WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 1);
+  WaitForDataChannelsToOpen(callee_dc.get(), caller_signaled_data_channels_, 0);
+
+  TestDataChannelSendAndReceive(
+      caller_dc.get(), callee_signaled_data_channels_[1].get(), 256 * 1024);
+  TestDataChannelSendAndReceive(
+      callee_dc.get(), caller_signaled_data_channels_[0].get(), 256 * 1024);
+
+  CloseDataChannels(caller_dc.get(), callee_signaled_data_channels_, 1);
+  CloseDataChannels(callee_dc.get(), caller_signaled_data_channels_, 0);
+}
+
+// Verifies that DataChannel IDs are even/odd based on the DTLS roles.
+TEST_P(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+  rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
+      caller_->CreateDataChannel("data", init));
+  rtc::scoped_refptr<DataChannelInterface> callee_dc_1(
+      callee_->CreateDataChannel("data", init));
+
+  Negotiate();
+  WaitForConnection();
+
+  EXPECT_EQ(1, caller_dc_1->id() % 2);
+  EXPECT_EQ(0, callee_dc_1->id() % 2);
+
+  rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
+      caller_->CreateDataChannel("data", init));
+  rtc::scoped_refptr<DataChannelInterface> callee_dc_2(
+      callee_->CreateDataChannel("data", init));
+
+  EXPECT_EQ(1, caller_dc_2->id() % 2);
+  EXPECT_EQ(0, callee_dc_2->id() % 2);
+}
+
+// Verifies that the message is received by the right remote DataChannel when
+// there are multiple DataChannels.
+TEST_P(PeerConnectionEndToEndTest,
+       MessageTransferBetweenTwoPairsOfDataChannels) {
+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+
+  rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
+      caller_->CreateDataChannel("data", init));
+  rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
+      caller_->CreateDataChannel("data", init));
+
+  Negotiate();
+  WaitForConnection();
+  WaitForDataChannelsToOpen(caller_dc_1.get(), callee_signaled_data_channels_,
+                            0);
+  WaitForDataChannelsToOpen(caller_dc_2.get(), callee_signaled_data_channels_,
+                            1);
+
+  std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer(
+      new webrtc::MockDataChannelObserver(
+          callee_signaled_data_channels_[0].get()));
+
+  std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer(
+      new webrtc::MockDataChannelObserver(
+          callee_signaled_data_channels_[1].get()));
+
+  const std::string message_1 = "hello 1";
+  const std::string message_2 = "hello 2";
+
+  caller_dc_1->Send(webrtc::DataBuffer(message_1));
+  EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
+
+  caller_dc_2->Send(webrtc::DataBuffer(message_2));
+  EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
+
+  EXPECT_EQ(1U, dc_1_observer->received_message_count());
+  EXPECT_EQ(1U, dc_2_observer->received_message_count());
+}
+
+// Verifies that a DataChannel added from an OPEN message functions after
+// a channel has been previously closed (webrtc issue 3778).
+// This previously failed because the new channel re-used the ID of the closed
+// channel, and the closed channel was incorrectly still assigned to the ID.
+TEST_P(PeerConnectionEndToEndTest,
+       DataChannelFromOpenWorksAfterPreviousChannelClosed) {
+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+  rtc::scoped_refptr<DataChannelInterface> caller_dc(
+      caller_->CreateDataChannel("data", init));
+
+  Negotiate();
+  WaitForConnection();
+
+  WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 0);
+  int first_channel_id = caller_dc->id();
+  // Wait for the local side to say it's closed, but not the remote side.
+  // Previously, the channel on which Close is called reported being closed
+  // prematurely, and this caused issues; see bugs.webrtc.org/4453.
+  caller_dc->Close();
+  EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait);
+
+  // Create a new channel and ensure it works after closing the previous one.
+  caller_dc = caller_->CreateDataChannel("data2", init);
+  WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 1);
+  // Since the second channel was created after the first finished closing, it
+  // should be able to re-use the first one's ID.
+  EXPECT_EQ(first_channel_id, caller_dc->id());
+  TestDataChannelSendAndReceive(caller_dc.get(),
+                                callee_signaled_data_channels_[1].get());
+
+  CloseDataChannels(caller_dc.get(), callee_signaled_data_channels_, 1);
+}
+
+// This tests that if a data channel is closed remotely while not referenced
+// by the application (meaning only the PeerConnection contributes to its
+// reference count), no memory access violation will occur.
+// See: https://code.google.com/p/chromium/issues/detail?id=565048
+TEST_P(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+  rtc::scoped_refptr<DataChannelInterface> caller_dc(
+      caller_->CreateDataChannel("data", init));
+
+  Negotiate();
+  WaitForConnection();
+
+  WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 0);
+  // This removes the reference to the remote data channel that we hold.
+  callee_signaled_data_channels_.clear();
+  caller_dc->Close();
+  EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait);
+
+  // Wait for a bit longer so the remote data channel will receive the
+  // close message and be destroyed.
+  rtc::Thread::Current()->ProcessMessages(100);
+}
+
+// Test behavior of creating too many datachannels.
+TEST_P(PeerConnectionEndToEndTest, TooManyDataChannelsOpenedBeforeConnecting) {
+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+  std::vector<rtc::scoped_refptr<DataChannelInterface>> channels;
+  for (int i = 0; i <= cricket::kMaxSctpStreams / 2; i++) {
+    rtc::scoped_refptr<DataChannelInterface> caller_dc(
+        caller_->CreateDataChannel("data", init));
+    channels.push_back(std::move(caller_dc));
+  }
+  Negotiate();
+  WaitForConnection();
+  EXPECT_EQ_WAIT(callee_signaled_data_channels_.size(),
+                 static_cast<size_t>(cricket::kMaxSctpStreams / 2), kMaxWait);
+  EXPECT_EQ(DataChannelInterface::kOpen,
+            channels[(cricket::kMaxSctpStreams / 2) - 1]->state());
+  EXPECT_EQ(DataChannelInterface::kClosed,
+            channels[cricket::kMaxSctpStreams / 2]->state());
+}
+
+#endif  // WEBRTC_HAVE_SCTP
+
+TEST_P(PeerConnectionEndToEndTest, CanRestartIce) {
+  rtc::scoped_refptr<webrtc::AudioDecoderFactory> real_decoder_factory =
+      webrtc::CreateOpusAudioDecoderFactory();
+  CreatePcs(webrtc::CreateOpusAudioEncoderFactory(),
+            CreateForwardingMockDecoderFactory(real_decoder_factory.get()));
+  GetAndAddUserMedia();
+  Negotiate();
+  WaitForCallEstablished();
+  // Cause ICE restart to be requested.
+  auto config = caller_->pc()->GetConfiguration();
+  ASSERT_NE(PeerConnectionInterface::kRelay, config.type);
+  config.type = PeerConnectionInterface::kRelay;
+  ASSERT_TRUE(caller_->pc()->SetConfiguration(config).ok());
+  // When solving https://crbug.com/webrtc/10504, all we need to check
+  // is that we do not crash. We should also be testing that restart happens.
+}
+
+INSTANTIATE_TEST_SUITE_P(PeerConnectionEndToEndTest,
+                         PeerConnectionEndToEndTest,
+                         Values(SdpSemantics::kPlanB_DEPRECATED,
+                                SdpSemantics::kUnifiedPlan));