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/*
* 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/environment/environment.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::Environment;
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, Create)
.Times(AtLeast(2))
.WillRepeatedly(
[real_decoder_factory](
const webrtc::Environment& env,
const webrtc::SdpAudioFormat& format,
absl::optional<webrtc::AudioCodecPairId> codec_pair_id) {
auto real_decoder =
real_decoder_factory->Create(env, format, codec_pair_id);
return 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::CodecParameterMap 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::CodecParameterMap 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> Create(
const Environment& env,
const webrtc::SdpAudioFormat& format,
Options options) override {
EXPECT_TRUE(options.codec_pair_id.has_value());
codec_ids_->push_back(*options.codec_pair_id);
return fact_->Create(env, format, options);
}
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> Create(
const Environment& env,
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_->Create(env, 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.
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());
// 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));