blob: 17f48bf326ed9f8e5a45f026c0bd45f9548e24fc [file] [log] [blame]
/*
* 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 <memory>
#include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/audio_decoder_factory_template.h"
#include "api/audio_codecs/audio_encoder_factory_template.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "rtc_base/gunit.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/stringencode.h"
#include "rtc_base/stringutils.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/androidtestinitializer.h"
#endif
#include "pc/test/peerconnectiontestwrapper.h"
// Notice that mockpeerconnectionobservers.h must be included after the above!
#include "pc/test/mockpeerconnectionobservers.h"
#include "test/mock_audio_decoder.h"
#include "test/mock_audio_decoder_factory.h"
using testing::AtLeast;
using testing::Invoke;
using testing::StrictMock;
using testing::_;
using webrtc::DataChannelInterface;
using webrtc::FakeConstraints;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStreamInterface;
using webrtc::PeerConnectionInterface;
namespace {
const int kMaxWait = 10000;
} // namespace
class PeerConnectionEndToEndTest
: public sigslot::has_slots<>,
public testing::Test {
public:
typedef std::vector<rtc::scoped_refptr<DataChannelInterface> >
DataChannelList;
PeerConnectionEndToEndTest() {
network_thread_ = rtc::Thread::CreateWithSocketServer();
worker_thread_ = rtc::Thread::Create();
RTC_CHECK(network_thread_->Start());
RTC_CHECK(worker_thread_->Start());
caller_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
"caller", network_thread_.get(), worker_thread_.get());
callee_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
"callee", 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);
#ifdef WEBRTC_ANDROID
webrtc::InitializeAndroidObjects();
#endif
}
void CreatePcs(const MediaConstraintsInterface* pc_constraints,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
audio_encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
audio_decoder_factory) {
EXPECT_TRUE(caller_->CreatePc(
pc_constraints, config_, audio_encoder_factory, audio_decoder_factory));
EXPECT_TRUE(callee_->CreatePc(
pc_constraints, config_, audio_encoder_factory, audio_decoder_factory));
PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
caller_->SignalOnDataChannel.connect(
this, &PeerConnectionEndToEndTest::OnCallerAddedDataChanel);
callee_->SignalOnDataChannel.connect(
this, &PeerConnectionEndToEndTest::OnCalleeAddedDataChannel);
}
void GetAndAddUserMedia() {
FakeConstraints audio_constraints;
FakeConstraints video_constraints;
GetAndAddUserMedia(true, audio_constraints, true, video_constraints);
}
void GetAndAddUserMedia(bool audio,
const FakeConstraints& audio_constraints,
bool video,
const FakeConstraints& video_constraints) {
caller_->GetAndAddUserMedia(audio, audio_constraints,
video, video_constraints);
callee_->GetAndAddUserMedia(audio, audio_constraints,
video, video_constraints);
}
void Negotiate() {
caller_->CreateOffer(NULL);
}
void WaitForCallEstablished() {
caller_->WaitForCallEstablished();
callee_->WaitForCallEstablished();
}
void WaitForConnection() {
caller_->WaitForConnection();
callee_->WaitForConnection();
}
void OnCallerAddedDataChanel(DataChannelInterface* dc) {
caller_signaled_data_channels_.push_back(dc);
}
void OnCalleeAddedDataChannel(DataChannelInterface* dc) {
callee_signaled_data_channels_.push_back(dc);
}
// Tests that |dc1| and |dc2| can send to and receive from each other.
void TestDataChannelSendAndReceive(
DataChannelInterface* dc1, DataChannelInterface* dc2) {
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 = "abcdefg";
webrtc::DataBuffer buffer(kDummyData);
EXPECT_TRUE(dc1->Send(buffer));
EXPECT_EQ_WAIT(kDummyData, dc2_observer->last_message(), kMaxWait);
EXPECT_TRUE(dc2->Send(buffer));
EXPECT_EQ_WAIT(kDummyData, dc1_observer->last_message(), kMaxWait);
EXPECT_EQ(1U, dc1_observer->received_message_count());
EXPECT_EQ(1U, dc2_observer->received_message_count());
}
void WaitForDataChannelsToOpen(DataChannelInterface* local_dc,
const DataChannelList& remote_dc_list,
size_t remote_dc_index) {
EXPECT_EQ_WAIT(DataChannelInterface::kOpen, local_dc->state(), kMaxWait);
EXPECT_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:
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_;
};
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 =
rtc::MakeUnique<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, IncomingPacket(_, _, _, _, _))
.Times(AtLeast(1))
.WillRepeatedly(Invoke([dec](const uint8_t* payload, size_t payload_len,
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) {
return dec->IncomingPacket(payload, payload_len, rtp_sequence_number,
rtp_timestamp, arrival_timestamp);
}));
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 =
new rtc::RefCountedObject<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,
std::unique_ptr<webrtc::AudioDecoder>* return_value) {
auto real_decoder = real_decoder_factory->MakeAudioDecoder(format);
*return_value =
real_decoder
? CreateForwardingMockDecoder(std::move(real_decoder))
: nullptr;
}));
return mock_decoder_factory;
}
struct AudioEncoderUnicornSparklesRainbow {
using Config = webrtc::AudioEncoderL16::Config;
static rtc::Optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
if (STR_CASE_CMP(format.name.c_str(), "UnicornSparklesRainbow") == 0) {
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 rtc::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) {
return webrtc::AudioEncoderL16::MakeAudioEncoder(config, payload_type);
}
};
struct AudioDecoderUnicornSparklesRainbow {
using Config = webrtc::AudioDecoderL16::Config;
static rtc::Optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
if (STR_CASE_CMP(format.name.c_str(), "UnicornSparklesRainbow") == 0) {
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 rtc::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) {
return webrtc::AudioDecoderL16::MakeAudioDecoder(config);
}
};
} // namespace
// Disabled for TSan v2, see
// https://bugs.chromium.org/p/webrtc/issues/detail?id=4719 for details.
// Disabled for Mac, see
// https://bugs.chromium.org/p/webrtc/issues/detail?id=5231 for details.
#if defined(THREAD_SANITIZER) || defined(WEBRTC_MAC)
#define MAYBE_Call DISABLED_Call
#else
#define MAYBE_Call Call
#endif
TEST_F(PeerConnectionEndToEndTest, MAYBE_Call) {
rtc::scoped_refptr<webrtc::AudioDecoderFactory> real_decoder_factory =
webrtc::CreateBuiltinAudioDecoderFactory();
CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
CreateForwardingMockDecoderFactory(real_decoder_factory.get()));
GetAndAddUserMedia();
Negotiate();
WaitForCallEstablished();
}
#if !defined(ADDRESS_SANITIZER)
TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
FakeConstraints pc_constraints;
pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
false);
CreatePcs(&pc_constraints, webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory());
GetAndAddUserMedia();
Negotiate();
WaitForCallEstablished();
}
#endif // !defined(ADDRESS_SANITIZER)
TEST_F(PeerConnectionEndToEndTest, CallWithCustomCodec) {
CreatePcs(
nullptr,
webrtc::CreateAudioEncoderFactory<AudioEncoderUnicornSparklesRainbow>(),
webrtc::CreateAudioDecoderFactory<AudioDecoderUnicornSparklesRainbow>());
GetAndAddUserMedia();
Negotiate();
WaitForCallEstablished();
}
#ifdef HAVE_SCTP
// Verifies that a DataChannel created before the negotiation can transition to
// "OPEN" and transfer data.
TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
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, callee_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[0]);
TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0);
CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
}
// Verifies that a DataChannel created after the negotiation can transition to
// "OPEN" and transfer data.
TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
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, 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, callee_signaled_data_channels_, 1);
WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
}
// Verifies that DataChannel IDs are even/odd based on the DTLS roles.
TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
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(1U, caller_dc_1->id() % 2);
EXPECT_EQ(0U, 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(1U, caller_dc_2->id() % 2);
EXPECT_EQ(0U, callee_dc_2->id() % 2);
}
// Verifies that the message is received by the right remote DataChannel when
// there are multiple DataChannels.
TEST_F(PeerConnectionEndToEndTest,
MessageTransferBetweenTwoPairsOfDataChannels) {
CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
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, callee_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(caller_dc_2, callee_signaled_data_channels_, 1);
std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer(
new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[0]));
std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer(
new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[1]));
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-uses the ID of the closed
// channel, and the closed channel was incorrectly still assigned to the id.
// TODO(deadbeef): This is disabled because there's currently a race condition
// caused by the fact that a data channel signals that it's closed before it
// really is. Re-enable this test once that's fixed.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4453
TEST_F(PeerConnectionEndToEndTest,
DISABLED_DataChannelFromOpenWorksAfterClose) {
CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
webrtc::DataChannelInit init;
rtc::scoped_refptr<DataChannelInterface> caller_dc(
caller_->CreateDataChannel("data", init));
Negotiate();
WaitForConnection();
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0);
// Create a new channel and ensure it works after closing the previous one.
caller_dc = caller_->CreateDataChannel("data2", init);
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
CloseDataChannels(caller_dc, 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_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
webrtc::DataChannelInit init;
rtc::scoped_refptr<DataChannelInterface> caller_dc(
caller_->CreateDataChannel("data", init));
Negotiate();
WaitForConnection();
WaitForDataChannelsToOpen(caller_dc, 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);
}
#endif // HAVE_SCTP