diff options
Diffstat (limited to 'webrtc/test')
73 files changed, 1649 insertions, 1184 deletions
diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn index ed1dc75078..3ecd903522 100644 --- a/webrtc/test/BUILD.gn +++ b/webrtc/test/BUILD.gn @@ -27,6 +27,7 @@ source_set("field_trial") { deps = [ "..:webrtc_common", "../system_wrappers", + "../system_wrappers:field_trial_default", ] configs += [ "..:common_config" ] @@ -58,7 +59,6 @@ source_set("test_support") { "testsupport/frame_reader.h", "testsupport/frame_writer.cc", "testsupport/frame_writer.h", - "testsupport/gtest_disable.h", "testsupport/mock/mock_frame_reader.h", "testsupport/mock/mock_frame_writer.h", "testsupport/packet_reader.cc", @@ -70,10 +70,10 @@ source_set("test_support") { ] deps = [ - "//testing/gmock", - "//testing/gtest", "..:gtest_prod", "../system_wrappers", + "//testing/gmock", + "//testing/gtest", ] if (is_android) { diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc index 0a8b686974..e9651e33f5 100644 --- a/webrtc/test/call_test.cc +++ b/webrtc/test/call_test.cc @@ -7,8 +7,15 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ +#include "webrtc/base/checks.h" +#include "webrtc/common.h" +#include "webrtc/config.h" #include "webrtc/test/call_test.h" #include "webrtc/test/encoder_settings.h" +#include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/voice_engine/include/voe_base.h" +#include "webrtc/voice_engine/include/voe_codec.h" +#include "webrtc/voice_engine/include/voe_network.h" namespace webrtc { namespace test { @@ -19,25 +26,43 @@ const int kVideoRotationRtpExtensionId = 4; CallTest::CallTest() : clock_(Clock::GetRealTimeClock()), - send_config_(nullptr), - send_stream_(NULL), - fake_encoder_(clock_) { -} + video_send_config_(nullptr), + video_send_stream_(nullptr), + audio_send_config_(nullptr), + audio_send_stream_(nullptr), + fake_encoder_(clock_), + num_video_streams_(1), + num_audio_streams_(0), + fake_send_audio_device_(nullptr), + fake_recv_audio_device_(nullptr) {} CallTest::~CallTest() { } -void CallTest::RunBaseTest(BaseTest* test, - const FakeNetworkPipe::Config& config) { - CreateSenderCall(test->GetSenderCallConfig()); - if (test->ShouldCreateReceivers()) - CreateReceiverCall(test->GetReceiverCallConfig()); - send_transport_.reset(new PacketTransport( - sender_call_.get(), test, test::PacketTransport::kSender, config)); - receive_transport_.reset(new PacketTransport( - nullptr, test, test::PacketTransport::kReceiver, config)); - test->OnTransportsCreated(send_transport_.get(), receive_transport_.get()); +void CallTest::RunBaseTest(BaseTest* test) { + num_video_streams_ = test->GetNumVideoStreams(); + num_audio_streams_ = test->GetNumAudioStreams(); + RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0); + Call::Config send_config(test->GetSenderCallConfig()); + if (num_audio_streams_ > 0) { + CreateVoiceEngines(); + AudioState::Config audio_state_config; + audio_state_config.voice_engine = voe_send_.voice_engine; + send_config.audio_state = AudioState::Create(audio_state_config); + } + CreateSenderCall(send_config); + if (test->ShouldCreateReceivers()) { + Call::Config recv_config(test->GetReceiverCallConfig()); + if (num_audio_streams_ > 0) { + AudioState::Config audio_state_config; + audio_state_config.voice_engine = voe_recv_.voice_engine; + recv_config.audio_state = AudioState::Create(audio_state_config); + } + CreateReceiverCall(recv_config); + } test->OnCallsCreated(sender_call_.get(), receiver_call_.get()); + send_transport_.reset(test->CreateSendTransport(sender_call_.get())); + receive_transport_.reset(test->CreateReceiveTransport()); if (test->ShouldCreateReceivers()) { send_transport_->SetReceiver(receiver_call_->Receiver()); @@ -48,16 +73,34 @@ void CallTest::RunBaseTest(BaseTest* test, receive_transport_->SetReceiver(nullptr); } - CreateSendConfig(test->GetNumStreams(), send_transport_.get()); + CreateSendConfig(num_video_streams_, num_audio_streams_, + send_transport_.get()); if (test->ShouldCreateReceivers()) { CreateMatchingReceiveConfigs(receive_transport_.get()); } - test->ModifyConfigs(&send_config_, &receive_configs_, &encoder_config_); - CreateStreams(); - test->OnStreamsCreated(send_stream_, receive_streams_); + if (num_audio_streams_ > 0) + SetupVoiceEngineTransports(send_transport_.get(), receive_transport_.get()); + + if (num_video_streams_ > 0) { + test->ModifyVideoConfigs(&video_send_config_, &video_receive_configs_, + &video_encoder_config_); + } + if (num_audio_streams_ > 0) + test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_); - CreateFrameGeneratorCapturer(); - test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get()); + if (num_video_streams_ > 0) { + CreateVideoStreams(); + test->OnVideoStreamsCreated(video_send_stream_, video_receive_streams_); + } + if (num_audio_streams_ > 0) { + CreateAudioStreams(); + test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_); + } + + if (num_video_streams_ > 0) { + CreateFrameGeneratorCapturer(); + test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get()); + } Start(); test->PerformTest(); @@ -66,12 +109,28 @@ void CallTest::RunBaseTest(BaseTest* test, Stop(); DestroyStreams(); + DestroyCalls(); + if (num_audio_streams_ > 0) + DestroyVoiceEngines(); } void CallTest::Start() { - send_stream_->Start(); - for (size_t i = 0; i < receive_streams_.size(); ++i) - receive_streams_[i]->Start(); + if (video_send_stream_) + video_send_stream_->Start(); + for (VideoReceiveStream* video_recv_stream : video_receive_streams_) + video_recv_stream->Start(); + if (audio_send_stream_) { + fake_send_audio_device_->Start(); + audio_send_stream_->Start(); + EXPECT_EQ(0, voe_send_.base->StartSend(voe_send_.channel_id)); + } + for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_) + audio_recv_stream->Start(); + if (!audio_receive_streams_.empty()) { + fake_recv_audio_device_->Start(); + EXPECT_EQ(0, voe_recv_.base->StartPlayout(voe_recv_.channel_id)); + EXPECT_EQ(0, voe_recv_.base->StartReceive(voe_recv_.channel_id)); + } if (frame_generator_capturer_.get() != NULL) frame_generator_capturer_->Start(); } @@ -79,9 +138,22 @@ void CallTest::Start() { void CallTest::Stop() { if (frame_generator_capturer_.get() != NULL) frame_generator_capturer_->Stop(); - for (size_t i = 0; i < receive_streams_.size(); ++i) - receive_streams_[i]->Stop(); - send_stream_->Stop(); + if (!audio_receive_streams_.empty()) { + fake_recv_audio_device_->Stop(); + EXPECT_EQ(0, voe_recv_.base->StopReceive(voe_recv_.channel_id)); + EXPECT_EQ(0, voe_recv_.base->StopPlayout(voe_recv_.channel_id)); + } + for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_) + audio_recv_stream->Stop(); + if (audio_send_stream_) { + fake_send_audio_device_->Stop(); + EXPECT_EQ(0, voe_send_.base->StopSend(voe_send_.channel_id)); + audio_send_stream_->Stop(); + } + for (VideoReceiveStream* video_recv_stream : video_receive_streams_) + video_recv_stream->Stop(); + if (video_send_stream_) + video_send_stream_->Stop(); } void CallTest::CreateCalls(const Call::Config& sender_config, @@ -99,92 +171,203 @@ void CallTest::CreateReceiverCall(const Call::Config& config) { } void CallTest::DestroyCalls() { - sender_call_.reset(nullptr); - receiver_call_.reset(nullptr); + sender_call_.reset(); + receiver_call_.reset(); } -void CallTest::CreateSendConfig(size_t num_streams, +void CallTest::CreateSendConfig(size_t num_video_streams, + size_t num_audio_streams, Transport* send_transport) { - assert(num_streams <= kNumSsrcs); - send_config_ = VideoSendStream::Config(send_transport); - send_config_.encoder_settings.encoder = &fake_encoder_; - send_config_.encoder_settings.payload_name = "FAKE"; - send_config_.encoder_settings.payload_type = kFakeSendPayloadType; - send_config_.rtp.extensions.push_back( - RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId)); - encoder_config_.streams = test::CreateVideoStreams(num_streams); - for (size_t i = 0; i < num_streams; ++i) - send_config_.rtp.ssrcs.push_back(kSendSsrcs[i]); - send_config_.rtp.extensions.push_back( - RtpExtension(RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId)); -} - -void CallTest::CreateMatchingReceiveConfigs( - Transport* rtcp_send_transport) { - assert(!send_config_.rtp.ssrcs.empty()); - assert(receive_configs_.empty()); - assert(allocated_decoders_.empty()); - VideoReceiveStream::Config config(rtcp_send_transport); - config.rtp.remb = true; - config.rtp.local_ssrc = kReceiverLocalSsrc; - for (const RtpExtension& extension : send_config_.rtp.extensions) - config.rtp.extensions.push_back(extension); - for (size_t i = 0; i < send_config_.rtp.ssrcs.size(); ++i) { - VideoReceiveStream::Decoder decoder = - test::CreateMatchingDecoder(send_config_.encoder_settings); - allocated_decoders_.push_back(decoder.decoder); - config.decoders.clear(); - config.decoders.push_back(decoder); - config.rtp.remote_ssrc = send_config_.rtp.ssrcs[i]; - receive_configs_.push_back(config); + RTC_DCHECK(num_video_streams <= kNumSsrcs); + RTC_DCHECK_LE(num_audio_streams, 1u); + RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0); + if (num_video_streams > 0) { + video_send_config_ = VideoSendStream::Config(send_transport); + video_send_config_.encoder_settings.encoder = &fake_encoder_; + video_send_config_.encoder_settings.payload_name = "FAKE"; + video_send_config_.encoder_settings.payload_type = + kFakeVideoSendPayloadType; + video_send_config_.rtp.extensions.push_back( + RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId)); + video_encoder_config_.streams = test::CreateVideoStreams(num_video_streams); + for (size_t i = 0; i < num_video_streams; ++i) + video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); + video_send_config_.rtp.extensions.push_back(RtpExtension( + RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId)); + } + + if (num_audio_streams > 0) { + audio_send_config_ = AudioSendStream::Config(send_transport); + audio_send_config_.voe_channel_id = voe_send_.channel_id; + audio_send_config_.rtp.ssrc = kAudioSendSsrc; + } +} + +void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) { + RTC_DCHECK(video_receive_configs_.empty()); + RTC_DCHECK(allocated_decoders_.empty()); + if (num_video_streams_ > 0) { + RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty()); + VideoReceiveStream::Config video_config(rtcp_send_transport); + video_config.rtp.remb = true; + video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc; + for (const RtpExtension& extension : video_send_config_.rtp.extensions) + video_config.rtp.extensions.push_back(extension); + for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) { + VideoReceiveStream::Decoder decoder = + test::CreateMatchingDecoder(video_send_config_.encoder_settings); + allocated_decoders_.push_back(decoder.decoder); + video_config.decoders.clear(); + video_config.decoders.push_back(decoder); + video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i]; + video_receive_configs_.push_back(video_config); + } + } + + RTC_DCHECK(num_audio_streams_ <= 1); + if (num_audio_streams_ == 1) { + RTC_DCHECK(voe_send_.channel_id >= 0); + AudioReceiveStream::Config audio_config; + audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; + audio_config.rtcp_send_transport = rtcp_send_transport; + audio_config.voe_channel_id = voe_recv_.channel_id; + audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc; + audio_receive_configs_.push_back(audio_config); } } void CallTest::CreateFrameGeneratorCapturer() { - VideoStream stream = encoder_config_.streams.back(); - frame_generator_capturer_.reset( - test::FrameGeneratorCapturer::Create(send_stream_->Input(), - stream.width, - stream.height, - stream.max_framerate, - clock_)); + VideoStream stream = video_encoder_config_.streams.back(); + frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create( + video_send_stream_->Input(), stream.width, stream.height, + stream.max_framerate, clock_)); } -void CallTest::CreateStreams() { - assert(send_stream_ == NULL); - assert(receive_streams_.empty()); +void CallTest::CreateFakeAudioDevices() { + fake_send_audio_device_.reset(new FakeAudioDevice( + clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"))); + fake_recv_audio_device_.reset(new FakeAudioDevice( + clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"))); +} - send_stream_ = - sender_call_->CreateVideoSendStream(send_config_, encoder_config_); +void CallTest::CreateVideoStreams() { + RTC_DCHECK(video_send_stream_ == nullptr); + RTC_DCHECK(video_receive_streams_.empty()); + RTC_DCHECK(audio_send_stream_ == nullptr); + RTC_DCHECK(audio_receive_streams_.empty()); + + video_send_stream_ = sender_call_->CreateVideoSendStream( + video_send_config_, video_encoder_config_); + for (size_t i = 0; i < video_receive_configs_.size(); ++i) { + video_receive_streams_.push_back( + receiver_call_->CreateVideoReceiveStream(video_receive_configs_[i])); + } +} - for (size_t i = 0; i < receive_configs_.size(); ++i) { - receive_streams_.push_back( - receiver_call_->CreateVideoReceiveStream(receive_configs_[i])); +void CallTest::CreateAudioStreams() { + audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_); + for (size_t i = 0; i < audio_receive_configs_.size(); ++i) { + audio_receive_streams_.push_back( + receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i])); } + CodecInst isac = {kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000}; + EXPECT_EQ(0, voe_send_.codec->SetSendCodec(voe_send_.channel_id, isac)); } void CallTest::DestroyStreams() { - if (send_stream_ != NULL) - sender_call_->DestroyVideoSendStream(send_stream_); - send_stream_ = NULL; - for (size_t i = 0; i < receive_streams_.size(); ++i) - receiver_call_->DestroyVideoReceiveStream(receive_streams_[i]); - receive_streams_.clear(); + if (video_send_stream_) + sender_call_->DestroyVideoSendStream(video_send_stream_); + video_send_stream_ = nullptr; + for (VideoReceiveStream* video_recv_stream : video_receive_streams_) + receiver_call_->DestroyVideoReceiveStream(video_recv_stream); + + if (audio_send_stream_) + sender_call_->DestroyAudioSendStream(audio_send_stream_); + audio_send_stream_ = nullptr; + for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_) + receiver_call_->DestroyAudioReceiveStream(audio_recv_stream); + video_receive_streams_.clear(); + allocated_decoders_.clear(); } -const unsigned int CallTest::kDefaultTimeoutMs = 30 * 1000; -const unsigned int CallTest::kLongTimeoutMs = 120 * 1000; -const uint8_t CallTest::kSendPayloadType = 100; -const uint8_t CallTest::kFakeSendPayloadType = 125; +void CallTest::CreateVoiceEngines() { + CreateFakeAudioDevices(); + voe_send_.voice_engine = VoiceEngine::Create(); + voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine); + voe_send_.network = VoENetwork::GetInterface(voe_send_.voice_engine); + voe_send_.codec = VoECodec::GetInterface(voe_send_.voice_engine); + EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr)); + Config voe_config; + voe_config.Set<VoicePacing>(new VoicePacing(true)); + voe_send_.channel_id = voe_send_.base->CreateChannel(voe_config); + EXPECT_GE(voe_send_.channel_id, 0); + + voe_recv_.voice_engine = VoiceEngine::Create(); + voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine); + voe_recv_.network = VoENetwork::GetInterface(voe_recv_.voice_engine); + voe_recv_.codec = VoECodec::GetInterface(voe_recv_.voice_engine); + EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr)); + voe_recv_.channel_id = voe_recv_.base->CreateChannel(); + EXPECT_GE(voe_recv_.channel_id, 0); +} + +void CallTest::SetupVoiceEngineTransports(PacketTransport* send_transport, + PacketTransport* recv_transport) { + voe_send_.transport_adapter.reset( + new internal::TransportAdapter(send_transport)); + voe_send_.transport_adapter->Enable(); + EXPECT_EQ(0, voe_send_.network->RegisterExternalTransport( + voe_send_.channel_id, *voe_send_.transport_adapter.get())); + + voe_recv_.transport_adapter.reset( + new internal::TransportAdapter(recv_transport)); + voe_recv_.transport_adapter->Enable(); + EXPECT_EQ(0, voe_recv_.network->RegisterExternalTransport( + voe_recv_.channel_id, *voe_recv_.transport_adapter.get())); +} + +void CallTest::DestroyVoiceEngines() { + voe_recv_.base->DeleteChannel(voe_recv_.channel_id); + voe_recv_.channel_id = -1; + voe_recv_.base->Release(); + voe_recv_.base = nullptr; + voe_recv_.network->Release(); + voe_recv_.network = nullptr; + voe_recv_.codec->Release(); + voe_recv_.codec = nullptr; + + voe_send_.base->DeleteChannel(voe_send_.channel_id); + voe_send_.channel_id = -1; + voe_send_.base->Release(); + voe_send_.base = nullptr; + voe_send_.network->Release(); + voe_send_.network = nullptr; + voe_send_.codec->Release(); + voe_send_.codec = nullptr; + + VoiceEngine::Delete(voe_send_.voice_engine); + voe_send_.voice_engine = nullptr; + VoiceEngine::Delete(voe_recv_.voice_engine); + voe_recv_.voice_engine = nullptr; +} + +const int CallTest::kDefaultTimeoutMs = 30 * 1000; +const int CallTest::kLongTimeoutMs = 120 * 1000; +const uint8_t CallTest::kVideoSendPayloadType = 100; +const uint8_t CallTest::kFakeVideoSendPayloadType = 125; const uint8_t CallTest::kSendRtxPayloadType = 98; const uint8_t CallTest::kRedPayloadType = 118; const uint8_t CallTest::kRtxRedPayloadType = 99; const uint8_t CallTest::kUlpfecPayloadType = 119; +const uint8_t CallTest::kAudioSendPayloadType = 103; const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE, 0xBADCAFF}; -const uint32_t CallTest::kSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE, 0xC0FFEF}; -const uint32_t CallTest::kReceiverLocalSsrc = 0x123456; +const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE, + 0xC0FFEF}; +const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF; +const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456; +const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567; const int CallTest::kNackRtpHistoryMs = 1000; BaseTest::BaseTest(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) { @@ -204,23 +387,40 @@ Call::Config BaseTest::GetReceiverCallConfig() { void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) { } -void BaseTest::OnTransportsCreated(PacketTransport* send_transport, - PacketTransport* receive_transport) {} +test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) { + return new PacketTransport(sender_call, this, test::PacketTransport::kSender, + FakeNetworkPipe::Config()); +} -size_t BaseTest::GetNumStreams() const { +test::PacketTransport* BaseTest::CreateReceiveTransport() { + return new PacketTransport(nullptr, this, test::PacketTransport::kReceiver, + FakeNetworkPipe::Config()); +} + +size_t BaseTest::GetNumVideoStreams() const { return 1; } -void BaseTest::ModifyConfigs( +size_t BaseTest::GetNumAudioStreams() const { + return 0; +} + +void BaseTest::ModifyVideoConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - VideoEncoderConfig* encoder_config) { -} + VideoEncoderConfig* encoder_config) {} -void BaseTest::OnStreamsCreated( +void BaseTest::OnVideoStreamsCreated( VideoSendStream* send_stream, - const std::vector<VideoReceiveStream*>& receive_streams) { -} + const std::vector<VideoReceiveStream*>& receive_streams) {} + +void BaseTest::ModifyAudioConfigs( + AudioSendStream::Config* send_config, + std::vector<AudioReceiveStream::Config>* receive_configs) {} + +void BaseTest::OnAudioStreamsCreated( + AudioSendStream* send_stream, + const std::vector<AudioReceiveStream*>& receive_streams) {} void BaseTest::OnFrameGeneratorCapturerCreated( FrameGeneratorCapturer* frame_generator_capturer) { diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h index cf024d9c65..251d7f6044 100644 --- a/webrtc/test/call_test.h +++ b/webrtc/test/call_test.h @@ -7,19 +7,26 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_TEST_COMMON_CALL_TEST_H_ -#define WEBRTC_TEST_COMMON_CALL_TEST_H_ +#ifndef WEBRTC_TEST_CALL_TEST_H_ +#define WEBRTC_TEST_CALL_TEST_H_ #include <vector> #include "webrtc/call.h" +#include "webrtc/call/transport_adapter.h" #include "webrtc/system_wrappers/include/scoped_vector.h" +#include "webrtc/test/fake_audio_device.h" #include "webrtc/test/fake_decoder.h" #include "webrtc/test/fake_encoder.h" #include "webrtc/test/frame_generator_capturer.h" #include "webrtc/test/rtp_rtcp_observer.h" namespace webrtc { + +class VoEBase; +class VoECodec; +class VoENetwork; + namespace test { class BaseTest; @@ -27,25 +34,31 @@ class BaseTest; class CallTest : public ::testing::Test { public: CallTest(); - ~CallTest(); + virtual ~CallTest(); static const size_t kNumSsrcs = 3; - static const unsigned int kDefaultTimeoutMs; - static const unsigned int kLongTimeoutMs; - static const uint8_t kSendPayloadType; + static const int kDefaultTimeoutMs; + static const int kLongTimeoutMs; + static const uint8_t kVideoSendPayloadType; static const uint8_t kSendRtxPayloadType; - static const uint8_t kFakeSendPayloadType; + static const uint8_t kFakeVideoSendPayloadType; static const uint8_t kRedPayloadType; static const uint8_t kRtxRedPayloadType; static const uint8_t kUlpfecPayloadType; + static const uint8_t kAudioSendPayloadType; static const uint32_t kSendRtxSsrcs[kNumSsrcs]; - static const uint32_t kSendSsrcs[kNumSsrcs]; - static const uint32_t kReceiverLocalSsrc; + static const uint32_t kVideoSendSsrcs[kNumSsrcs]; + static const uint32_t kAudioSendSsrc; + static const uint32_t kReceiverLocalVideoSsrc; + static const uint32_t kReceiverLocalAudioSsrc; static const int kNackRtpHistoryMs; protected: - void RunBaseTest(BaseTest* test, const FakeNetworkPipe::Config& config); + // RunBaseTest overwrites the audio_state and the voice_engine of the send and + // receive Call configs to simplify test code and avoid having old VoiceEngine + // APIs in the tests. + void RunBaseTest(BaseTest* test); void CreateCalls(const Call::Config& sender_config, const Call::Config& receiver_config); @@ -53,12 +66,16 @@ class CallTest : public ::testing::Test { void CreateReceiverCall(const Call::Config& config); void DestroyCalls(); - void CreateSendConfig(size_t num_streams, Transport* send_transport); + void CreateSendConfig(size_t num_video_streams, + size_t num_audio_streams, + Transport* send_transport); void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport); void CreateFrameGeneratorCapturer(); + void CreateFakeAudioDevices(); - void CreateStreams(); + void CreateVideoStreams(); + void CreateAudioStreams(); void Start(); void Stop(); void DestroyStreams(); @@ -67,18 +84,57 @@ class CallTest : public ::testing::Test { rtc::scoped_ptr<Call> sender_call_; rtc::scoped_ptr<PacketTransport> send_transport_; - VideoSendStream::Config send_config_; - VideoEncoderConfig encoder_config_; - VideoSendStream* send_stream_; + VideoSendStream::Config video_send_config_; + VideoEncoderConfig video_encoder_config_; + VideoSendStream* video_send_stream_; + AudioSendStream::Config audio_send_config_; + AudioSendStream* audio_send_stream_; rtc::scoped_ptr<Call> receiver_call_; rtc::scoped_ptr<PacketTransport> receive_transport_; - std::vector<VideoReceiveStream::Config> receive_configs_; - std::vector<VideoReceiveStream*> receive_streams_; + std::vector<VideoReceiveStream::Config> video_receive_configs_; + std::vector<VideoReceiveStream*> video_receive_streams_; + std::vector<AudioReceiveStream::Config> audio_receive_configs_; + std::vector<AudioReceiveStream*> audio_receive_streams_; rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_; test::FakeEncoder fake_encoder_; ScopedVector<VideoDecoder> allocated_decoders_; + size_t num_video_streams_; + size_t num_audio_streams_; + + private: + // TODO(holmer): Remove once VoiceEngine is fully refactored to the new API. + // These methods are used to set up legacy voice engines and channels which is + // necessary while voice engine is being refactored to the new stream API. + struct VoiceEngineState { + VoiceEngineState() + : voice_engine(nullptr), + base(nullptr), + network(nullptr), + codec(nullptr), + channel_id(-1), + transport_adapter(nullptr) {} + + VoiceEngine* voice_engine; + VoEBase* base; + VoENetwork* network; + VoECodec* codec; + int channel_id; + rtc::scoped_ptr<internal::TransportAdapter> transport_adapter; + }; + + void CreateVoiceEngines(); + void SetupVoiceEngineTransports(PacketTransport* send_transport, + PacketTransport* recv_transport); + void DestroyVoiceEngines(); + + VoiceEngineState voe_send_; + VoiceEngineState voe_recv_; + + // The audio devices must outlive the voice engines. + rtc::scoped_ptr<test::FakeAudioDevice> fake_send_audio_device_; + rtc::scoped_ptr<test::FakeAudioDevice> fake_recv_audio_device_; }; class BaseTest : public RtpRtcpObserver { @@ -89,22 +145,31 @@ class BaseTest : public RtpRtcpObserver { virtual void PerformTest() = 0; virtual bool ShouldCreateReceivers() const = 0; - virtual size_t GetNumStreams() const; + virtual size_t GetNumVideoStreams() const; + virtual size_t GetNumAudioStreams() const; virtual Call::Config GetSenderCallConfig(); virtual Call::Config GetReceiverCallConfig(); virtual void OnCallsCreated(Call* sender_call, Call* receiver_call); - virtual void OnTransportsCreated(PacketTransport* send_transport, - PacketTransport* receive_transport); - virtual void ModifyConfigs( + virtual test::PacketTransport* CreateSendTransport(Call* sender_call); + virtual test::PacketTransport* CreateReceiveTransport(); + + virtual void ModifyVideoConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, VideoEncoderConfig* encoder_config); - virtual void OnStreamsCreated( + virtual void OnVideoStreamsCreated( VideoSendStream* send_stream, const std::vector<VideoReceiveStream*>& receive_streams); + virtual void ModifyAudioConfigs( + AudioSendStream::Config* send_config, + std::vector<AudioReceiveStream::Config>* receive_configs); + virtual void OnAudioStreamsCreated( + AudioSendStream* send_stream, + const std::vector<AudioReceiveStream*>& receive_streams); + virtual void OnFrameGeneratorCapturerCreated( FrameGeneratorCapturer* frame_generator_capturer); }; @@ -126,4 +191,4 @@ class EndToEndTest : public BaseTest { } // namespace test } // namespace webrtc -#endif // WEBRTC_TEST_COMMON_CALL_TEST_H_ +#endif // WEBRTC_TEST_CALL_TEST_H_ diff --git a/webrtc/test/channel_transport/channel_transport.cc b/webrtc/test/channel_transport/channel_transport.cc index 25eb59d887..38eefe54a2 100644 --- a/webrtc/test/channel_transport/channel_transport.cc +++ b/webrtc/test/channel_transport/channel_transport.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/test/channel_transport/include/channel_transport.h" +#include "webrtc/test/channel_transport/channel_transport.h" #include <stdio.h> @@ -16,7 +16,6 @@ #include "testing/gtest/include/gtest/gtest.h" #endif #include "webrtc/test/channel_transport/udp_transport.h" -#include "webrtc/video_engine/vie_defines.h" #include "webrtc/voice_engine/include/voe_network.h" #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) @@ -66,10 +65,11 @@ void VoiceChannelTransport::IncomingRTCPPacket( } int VoiceChannelTransport::SetLocalReceiver(uint16_t rtp_port) { + static const int kNumReceiveSocketBuffers = 500; int return_value = socket_transport_->InitializeReceiveSockets(this, rtp_port); if (return_value == 0) { - return socket_transport_->StartReceiving(kViENumReceiveSocketBuffers); + return socket_transport_->StartReceiving(kNumReceiveSocketBuffers); } return return_value; } diff --git a/webrtc/test/channel_transport/include/channel_transport.h b/webrtc/test/channel_transport/channel_transport.h index 8b84517de0..bab7c59181 100644 --- a/webrtc/test/channel_transport/include/channel_transport.h +++ b/webrtc/test/channel_transport/channel_transport.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_TEST_CHANNEL_TRANSPORT_INCLUDE_CHANNEL_TRANSPORT_H_ -#define WEBRTC_TEST_CHANNEL_TRANSPORT_INCLUDE_CHANNEL_TRANSPORT_H_ +#ifndef WEBRTC_TEST_CHANNEL_TRANSPORT_CHANNEL_TRANSPORT_H_ +#define WEBRTC_TEST_CHANNEL_TRANSPORT_CHANNEL_TRANSPORT_H_ #include "webrtc/test/channel_transport/udp_transport.h" @@ -53,4 +53,4 @@ class VoiceChannelTransport : public UdpTransportData { } // namespace test } // namespace webrtc -#endif // WEBRTC_TEST_CHANNEL_TRANSPORT_INCLUDE_CHANNEL_TRANSPORT_H_ +#endif // WEBRTC_TEST_CHANNEL_TRANSPORT_CHANNEL_TRANSPORT_H_ diff --git a/webrtc/test/channel_transport/udp_socket2_manager_win.cc b/webrtc/test/channel_transport/udp_socket2_manager_win.cc index 5a11abbd6e..9f40350287 100644 --- a/webrtc/test/channel_transport/udp_socket2_manager_win.cc +++ b/webrtc/test/channel_transport/udp_socket2_manager_win.cc @@ -520,8 +520,8 @@ int32_t UdpSocket2WorkerWindows::_numOfWorkers = 0; UdpSocket2WorkerWindows::UdpSocket2WorkerWindows(HANDLE ioCompletionHandle) : _ioCompletionHandle(ioCompletionHandle), - _init(false) -{ + _pThread(Run, this, "UdpSocket2ManagerWindows_thread"), + _init(false) { _workerNumber = _numOfWorkers++; WEBRTC_TRACE(kTraceMemory, kTraceTransport, -1, "UdpSocket2WorkerWindows created"); @@ -537,10 +537,9 @@ bool UdpSocket2WorkerWindows::Start() { WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1, "Start UdpSocket2WorkerWindows"); - if (!_pThread->Start()) - return false; + _pThread.Start(); - _pThread->SetPriority(kRealtimePriority); + _pThread.SetPriority(rtc::kRealtimePriority); return true; } @@ -548,18 +547,14 @@ bool UdpSocket2WorkerWindows::Stop() { WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1, "Stop UdpSocket2WorkerWindows"); - return _pThread->Stop(); + _pThread.Stop(); + return true; } int32_t UdpSocket2WorkerWindows::Init() { - if(!_init) - { - const char* threadName = "UdpSocket2ManagerWindows_thread"; - _pThread = ThreadWrapper::CreateThread(Run, this, threadName); - _init = true; - } - return 0; + _init = true; + return 0; } bool UdpSocket2WorkerWindows::Run(void* obj) diff --git a/webrtc/test/channel_transport/udp_socket2_manager_win.h b/webrtc/test/channel_transport/udp_socket2_manager_win.h index c6af03a702..e762dccd0d 100644 --- a/webrtc/test/channel_transport/udp_socket2_manager_win.h +++ b/webrtc/test/channel_transport/udp_socket2_manager_win.h @@ -17,7 +17,7 @@ #include "webrtc/system_wrappers/include/atomic32.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/event_wrapper.h" -#include "webrtc/system_wrappers/include/thread_wrapper.h" +#include "webrtc/base/platform_thread.h" #include "webrtc/test/channel_transport/udp_socket2_win.h" #include "webrtc/test/channel_transport/udp_socket_manager_wrapper.h" #include "webrtc/test/channel_transport/udp_transport.h" @@ -47,7 +47,7 @@ struct PerIoContext { int fromLen; // Should be set to true if the I/O context was passed to the system by // a thread not controlled by the socket implementation. - bool ioInitiatedByThreadWrapper; + bool ioInitiatedByPlatformThread; // TODO (hellner): Not used. Delete it. PerIoContext* pNextFree; }; @@ -105,7 +105,7 @@ protected: bool Process(); private: HANDLE _ioCompletionHandle; - rtc::scoped_ptr<ThreadWrapper> _pThread; + rtc::PlatformThread _pThread; static int32_t _numOfWorkers; int32_t _workerNumber; volatile bool _stop; diff --git a/webrtc/test/channel_transport/udp_socket2_win.cc b/webrtc/test/channel_transport/udp_socket2_win.cc index 4c63dc938d..adeb46a9d2 100644 --- a/webrtc/test/channel_transport/udp_socket2_win.cc +++ b/webrtc/test/channel_transport/udp_socket2_win.cc @@ -432,13 +432,13 @@ void UdpSocket2Windows::IOCompleted(PerIoContext* pIOContext, if(pIOContext == NULL || error == ERROR_OPERATION_ABORTED) { if ((pIOContext != NULL) && - !pIOContext->ioInitiatedByThreadWrapper && + !pIOContext->ioInitiatedByPlatformThread && (error == ERROR_OPERATION_ABORTED) && (pIOContext->ioOperation == OP_READ) && _outstandingCallsDisabled) { - // !pIOContext->initiatedIOByThreadWrapper indicate that the I/O - // was not initiated by a ThreadWrapper thread. + // !pIOContext->initiatedIOByPlatformThread indicate that the I/O + // was not initiated by a PlatformThread thread. // This may happen if the thread that initiated receiving (e.g. // by calling StartListen())) is deleted before any packets have // been received. @@ -519,7 +519,7 @@ void UdpSocket2Windows::IOCompleted(PerIoContext* pIOContext, { // The PerIoContext was posted by a thread controlled by the socket // implementation. - pIOContext->ioInitiatedByThreadWrapper = true; + pIOContext->ioInitiatedByPlatformThread = true; } OutstandingCallCompleted(); return; @@ -546,7 +546,7 @@ int32_t UdpSocket2Windows::PostRecv() } // This function may have been called by thread not controlled by the socket // implementation. - pIoContext->ioInitiatedByThreadWrapper = false; + pIoContext->ioInitiatedByPlatformThread = false; return PostRecv(pIoContext); } diff --git a/webrtc/test/channel_transport/udp_socket_manager_posix.cc b/webrtc/test/channel_transport/udp_socket_manager_posix.cc index 145efcbc58..6b1a466bf2 100644 --- a/webrtc/test/channel_transport/udp_socket_manager_posix.cc +++ b/webrtc/test/channel_transport/udp_socket_manager_posix.cc @@ -184,12 +184,11 @@ bool UdpSocketManagerPosix::RemoveSocket(UdpSocketWrapper* s) return retVal; } - UdpSocketManagerPosixImpl::UdpSocketManagerPosixImpl() -{ - _critSectList = CriticalSectionWrapper::CreateCriticalSection(); - _thread = ThreadWrapper::CreateThread(UdpSocketManagerPosixImpl::Run, this, - "UdpSocketManagerPosixImplThread"); + : _thread(UdpSocketManagerPosixImpl::Run, + this, + "UdpSocketManagerPosixImplThread"), + _critSectList(CriticalSectionWrapper::CreateCriticalSection()) { FD_ZERO(&_readFds); WEBRTC_TRACE(kTraceMemory, kTraceTransport, -1, "UdpSocketManagerPosix created"); @@ -220,29 +219,19 @@ UdpSocketManagerPosixImpl::~UdpSocketManagerPosixImpl() bool UdpSocketManagerPosixImpl::Start() { - if (!_thread) - { - return false; - } - WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1, "Start UdpSocketManagerPosix"); - if (!_thread->Start()) - return false; - _thread->SetPriority(kRealtimePriority); + _thread.Start(); + _thread.SetPriority(rtc::kRealtimePriority); return true; } bool UdpSocketManagerPosixImpl::Stop() { - if (!_thread) - { - return true; - } - WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1, "Stop UdpSocketManagerPosix"); - return _thread->Stop(); + _thread.Stop(); + return true; } bool UdpSocketManagerPosixImpl::Process() diff --git a/webrtc/test/channel_transport/udp_socket_manager_posix.h b/webrtc/test/channel_transport/udp_socket_manager_posix.h index 64156fd20f..45e55af99a 100644 --- a/webrtc/test/channel_transport/udp_socket_manager_posix.h +++ b/webrtc/test/channel_transport/udp_socket_manager_posix.h @@ -17,8 +17,8 @@ #include <list> #include <map> +#include "webrtc/base/platform_thread.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" -#include "webrtc/system_wrappers/include/thread_wrapper.h" #include "webrtc/test/channel_transport/udp_socket_manager_wrapper.h" #include "webrtc/test/channel_transport/udp_socket_wrapper.h" @@ -75,7 +75,7 @@ protected: private: typedef std::list<UdpSocketWrapper*> SocketList; typedef std::list<SOCKET> FdList; - rtc::scoped_ptr<ThreadWrapper> _thread; + rtc::PlatformThread _thread; CriticalSectionWrapper* _critSectList; fd_set _readFds; diff --git a/webrtc/test/common_unittest.cc b/webrtc/test/common_unittest.cc index 082c18c2c7..a239dade73 100644 --- a/webrtc/test/common_unittest.cc +++ b/webrtc/test/common_unittest.cc @@ -15,6 +15,7 @@ namespace webrtc { namespace { struct MyExperiment { + static const ConfigOptionID identifier = ConfigOptionID::kMyExperimentForTest; static const int kDefaultFactor; static const int kDefaultOffset; @@ -56,6 +57,8 @@ TEST(Config, SetNullSetsTheOptionBackToDefault) { } struct Algo1_CostFunction { + static const ConfigOptionID identifier = + ConfigOptionID::kAlgo1CostFunctionForTest; Algo1_CostFunction() {} virtual int cost(int x) const { diff --git a/webrtc/test/configurable_frame_size_encoder.cc b/webrtc/test/configurable_frame_size_encoder.cc index 2cd47504a5..831e481bd8 100644 --- a/webrtc/test/configurable_frame_size_encoder.cc +++ b/webrtc/test/configurable_frame_size_encoder.cc @@ -14,8 +14,8 @@ #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/common_video/interface/video_image.h" -#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h" +#include "webrtc/common_video/include/video_image.h" +#include "webrtc/modules/video_coding/include/video_codec_interface.h" namespace webrtc { namespace test { diff --git a/webrtc/test/direct_transport.cc b/webrtc/test/direct_transport.cc index 6dcba81c88..591e154b14 100644 --- a/webrtc/test/direct_transport.cc +++ b/webrtc/test/direct_transport.cc @@ -18,26 +18,17 @@ namespace webrtc { namespace test { DirectTransport::DirectTransport(Call* send_call) - : send_call_(send_call), - packet_event_(EventWrapper::Create()), - thread_( - ThreadWrapper::CreateThread(NetworkProcess, this, "NetworkProcess")), - clock_(Clock::GetRealTimeClock()), - shutting_down_(false), - fake_network_(FakeNetworkPipe::Config()) { - EXPECT_TRUE(thread_->Start()); -} + : DirectTransport(FakeNetworkPipe::Config(), send_call) {} DirectTransport::DirectTransport(const FakeNetworkPipe::Config& config, Call* send_call) : send_call_(send_call), - packet_event_(EventWrapper::Create()), - thread_( - ThreadWrapper::CreateThread(NetworkProcess, this, "NetworkProcess")), + packet_event_(false, false), + thread_(NetworkProcess, this, "NetworkProcess"), clock_(Clock::GetRealTimeClock()), shutting_down_(false), - fake_network_(config) { - EXPECT_TRUE(thread_->Start()); + fake_network_(clock_, config) { + thread_.Start(); } DirectTransport::~DirectTransport() { StopSending(); } @@ -52,8 +43,8 @@ void DirectTransport::StopSending() { shutting_down_ = true; } - packet_event_->Set(); - EXPECT_TRUE(thread_->Stop()); + packet_event_.Set(); + thread_.Stop(); } void DirectTransport::SetReceiver(PacketReceiver* receiver) { @@ -69,16 +60,20 @@ bool DirectTransport::SendRtp(const uint8_t* data, send_call_->OnSentPacket(sent_packet); } fake_network_.SendPacket(data, length); - packet_event_->Set(); + packet_event_.Set(); return true; } bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) { fake_network_.SendPacket(data, length); - packet_event_->Set(); + packet_event_.Set(); return true; } +int DirectTransport::GetAverageDelayMs() { + return fake_network_.AverageDelay(); +} + bool DirectTransport::NetworkProcess(void* transport) { return static_cast<DirectTransport*>(transport)->SendPackets(); } @@ -87,15 +82,7 @@ bool DirectTransport::SendPackets() { fake_network_.Process(); int64_t wait_time_ms = fake_network_.TimeUntilNextProcess(); if (wait_time_ms > 0) { - switch (packet_event_->Wait(static_cast<unsigned long>(wait_time_ms))) { - case kEventSignaled: - break; - case kEventTimeout: - break; - case kEventError: - // TODO(pbos): Log a warning here? - return true; - } + packet_event_.Wait(static_cast<int>(wait_time_ms)); } rtc::CritScope crit(&lock_); return shutting_down_ ? false : true; diff --git a/webrtc/test/direct_transport.h b/webrtc/test/direct_transport.h index 241a5bc110..d68bc7184e 100644 --- a/webrtc/test/direct_transport.h +++ b/webrtc/test/direct_transport.h @@ -7,17 +7,17 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_DIRECT_TRANSPORT_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_DIRECT_TRANSPORT_H_ +#ifndef WEBRTC_TEST_DIRECT_TRANSPORT_H_ +#define WEBRTC_TEST_DIRECT_TRANSPORT_H_ #include <assert.h> #include <deque> #include "webrtc/base/criticalsection.h" +#include "webrtc/base/event.h" +#include "webrtc/base/platform_thread.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/system_wrappers/include/event_wrapper.h" -#include "webrtc/system_wrappers/include/thread_wrapper.h" #include "webrtc/test/fake_network_pipe.h" #include "webrtc/transport.h" @@ -46,14 +46,16 @@ class DirectTransport : public Transport { const PacketOptions& options) override; bool SendRtcp(const uint8_t* data, size_t length) override; + int GetAverageDelayMs(); + private: static bool NetworkProcess(void* transport); bool SendPackets(); rtc::CriticalSection lock_; Call* const send_call_; - rtc::scoped_ptr<EventWrapper> packet_event_; - rtc::scoped_ptr<ThreadWrapper> thread_; + rtc::Event packet_event_; + rtc::PlatformThread thread_; Clock* const clock_; bool shutting_down_; @@ -63,4 +65,4 @@ class DirectTransport : public Transport { } // namespace test } // namespace webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_DIRECT_TRANSPORT_H_ +#endif // WEBRTC_TEST_DIRECT_TRANSPORT_H_ diff --git a/webrtc/test/fake_audio_device.cc b/webrtc/test/fake_audio_device.cc index e307dd7664..31cebda652 100644 --- a/webrtc/test/fake_audio_device.cc +++ b/webrtc/test/fake_audio_device.cc @@ -13,11 +13,11 @@ #include <algorithm> #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/media_file/source/media_file_utility.h" +#include "webrtc/base/platform_thread.h" +#include "webrtc/modules/media_file/media_file_utility.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/file_wrapper.h" -#include "webrtc/system_wrappers/include/thread_wrapper.h" namespace webrtc { namespace test { @@ -30,6 +30,7 @@ FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename) last_playout_ms_(-1), clock_(clock), tick_(EventTimerWrapper::Create()), + thread_(FakeAudioDevice::Run, this, "FakeAudioDevice"), file_utility_(new ModuleFileUtility(0)), input_stream_(FileWrapper::Create()) { memset(captured_audio_, 0, sizeof(captured_audio_)); @@ -42,8 +43,7 @@ FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename) FakeAudioDevice::~FakeAudioDevice() { Stop(); - if (thread_.get() != NULL) - thread_->Stop(); + thread_.Stop(); } int32_t FakeAudioDevice::Init() { @@ -53,15 +53,8 @@ int32_t FakeAudioDevice::Init() { if (!tick_->StartTimer(true, 10)) return -1; - thread_ = ThreadWrapper::CreateThread(FakeAudioDevice::Run, this, - "FakeAudioDevice"); - if (thread_.get() == NULL) - return -1; - if (!thread_->Start()) { - thread_.reset(); - return -1; - } - thread_->SetPriority(webrtc::kHighPriority); + thread_.Start(); + thread_.SetPriority(rtc::kHighPriority); return 0; } diff --git a/webrtc/test/fake_audio_device.h b/webrtc/test/fake_audio_device.h index bdc672892c..7ca657bbb6 100644 --- a/webrtc/test/fake_audio_device.h +++ b/webrtc/test/fake_audio_device.h @@ -13,6 +13,7 @@ #include <string> #include "webrtc/base/criticalsection.h" +#include "webrtc/base/platform_thread.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/audio_device/include/fake_audio_device.h" #include "webrtc/typedefs.h" @@ -23,7 +24,6 @@ class Clock; class EventTimerWrapper; class FileWrapper; class ModuleFileUtility; -class ThreadWrapper; namespace test { @@ -59,7 +59,7 @@ class FakeAudioDevice : public FakeAudioDeviceModule { Clock* clock_; rtc::scoped_ptr<EventTimerWrapper> tick_; mutable rtc::CriticalSection lock_; - rtc::scoped_ptr<ThreadWrapper> thread_; + rtc::PlatformThread thread_; rtc::scoped_ptr<ModuleFileUtility> file_utility_; rtc::scoped_ptr<FileWrapper> input_stream_; }; diff --git a/webrtc/test/fake_decoder.cc b/webrtc/test/fake_decoder.cc index 63316e0dab..dbdd580e88 100644 --- a/webrtc/test/fake_decoder.cc +++ b/webrtc/test/fake_decoder.cc @@ -53,10 +53,16 @@ int32_t FakeDecoder::RegisterDecodeCompleteCallback( int32_t FakeDecoder::Release() { return WEBRTC_VIDEO_CODEC_OK; } + int32_t FakeDecoder::Reset() { return WEBRTC_VIDEO_CODEC_OK; } +const char* FakeDecoder::kImplementationName = "fake_decoder"; +const char* FakeDecoder::ImplementationName() const { + return kImplementationName; +} + int32_t FakeH264Decoder::Decode(const EncodedImage& input, bool missing_frames, const RTPFragmentationHeader* fragmentation, diff --git a/webrtc/test/fake_decoder.h b/webrtc/test/fake_decoder.h index 593af512f8..0da961d9a0 100644 --- a/webrtc/test/fake_decoder.h +++ b/webrtc/test/fake_decoder.h @@ -8,12 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_DECODER_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_DECODER_H_ +#ifndef WEBRTC_TEST_FAKE_DECODER_H_ +#define WEBRTC_TEST_FAKE_DECODER_H_ #include <vector> -#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h" +#include "webrtc/modules/video_coding/include/video_codec_interface.h" #include "webrtc/system_wrappers/include/clock.h" namespace webrtc { @@ -39,6 +39,10 @@ class FakeDecoder : public VideoDecoder { int32_t Release() override; int32_t Reset() override; + const char* ImplementationName() const override; + + static const char* kImplementationName; + private: VideoCodec config_; VideoFrame frame_; @@ -71,4 +75,4 @@ class FakeNullDecoder : public FakeDecoder { } // namespace test } // namespace webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_DECODER_H_ +#endif // WEBRTC_TEST_FAKE_DECODER_H_ diff --git a/webrtc/test/fake_encoder.cc b/webrtc/test/fake_encoder.cc index a3ade6e97a..72df40f9a5 100644 --- a/webrtc/test/fake_encoder.cc +++ b/webrtc/test/fake_encoder.cc @@ -12,7 +12,7 @@ #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h" +#include "webrtc/modules/video_coding/include/video_codec_interface.h" #include "webrtc/system_wrappers/include/sleep.h" namespace webrtc { @@ -57,6 +57,11 @@ int32_t FakeEncoder::Encode(const VideoFrame& input_image, // at the display time of the previous frame. time_since_last_encode_ms = time_now_ms - last_encode_time_ms_; } + if (time_since_last_encode_ms > 3 * 1000 / config_.maxFramerate) { + // Rudimentary check to make sure we don't widely overshoot bitrate target + // when resuming encoding after a suspension. + time_since_last_encode_ms = 3 * 1000 / config_.maxFramerate; + } size_t bits_available = static_cast<size_t>(target_bitrate_kbps_ * time_since_last_encode_ms); @@ -127,6 +132,11 @@ int32_t FakeEncoder::SetRates(uint32_t new_target_bitrate, uint32_t framerate) { return 0; } +const char* FakeEncoder::kImplementationName = "fake_encoder"; +const char* FakeEncoder::ImplementationName() const { + return kImplementationName; +} + FakeH264Encoder::FakeH264Encoder(Clock* clock) : FakeEncoder(clock), callback_(NULL), idr_counter_(0) { FakeEncoder::RegisterEncodeCompleteCallback(this); diff --git a/webrtc/test/fake_encoder.h b/webrtc/test/fake_encoder.h index d677b92e4d..6bff00e2a3 100644 --- a/webrtc/test/fake_encoder.h +++ b/webrtc/test/fake_encoder.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_ENCODER_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_ENCODER_H_ +#ifndef WEBRTC_TEST_FAKE_ENCODER_H_ +#define WEBRTC_TEST_FAKE_ENCODER_H_ #include <vector> @@ -39,6 +39,9 @@ class FakeEncoder : public VideoEncoder { int32_t Release() override; int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override; int32_t SetRates(uint32_t new_target_bitrate, uint32_t framerate) override; + const char* ImplementationName() const override; + + static const char* kImplementationName; protected: Clock* const clock_; @@ -82,4 +85,4 @@ class DelayedEncoder : public test::FakeEncoder { } // namespace test } // namespace webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_FAKE_ENCODER_H_ +#endif // WEBRTC_TEST_FAKE_ENCODER_H_ diff --git a/webrtc/test/fake_network_pipe.cc b/webrtc/test/fake_network_pipe.cc index c36059356a..491a0526b9 100644 --- a/webrtc/test/fake_network_pipe.cc +++ b/webrtc/test/fake_network_pipe.cc @@ -16,7 +16,7 @@ #include <algorithm> #include "webrtc/call.h" -#include "webrtc/system_wrappers/include/tick_util.h" +#include "webrtc/system_wrappers/include/clock.h" namespace webrtc { @@ -70,14 +70,15 @@ class NetworkPacket { int64_t arrival_time_; }; -FakeNetworkPipe::FakeNetworkPipe(const FakeNetworkPipe::Config& config) - : packet_receiver_(NULL), +FakeNetworkPipe::FakeNetworkPipe(Clock* clock, + const FakeNetworkPipe::Config& config) + : clock_(clock), + packet_receiver_(NULL), config_(config), dropped_packets_(0), sent_packets_(0), total_packet_delay_(0), - next_process_time_(TickTime::MillisecondTimestamp()) { -} + next_process_time_(clock_->TimeInMilliseconds()) {} FakeNetworkPipe::~FakeNetworkPipe() { while (!capacity_link_.empty()) { @@ -112,7 +113,7 @@ void FakeNetworkPipe::SendPacket(const uint8_t* data, size_t data_length) { return; } - int64_t time_now = TickTime::MillisecondTimestamp(); + int64_t time_now = clock_->TimeInMilliseconds(); // Delay introduced by the link capacity. int64_t capacity_delay_ms = 0; @@ -145,11 +146,12 @@ int FakeNetworkPipe::AverageDelay() { if (sent_packets_ == 0) return 0; - return total_packet_delay_ / static_cast<int>(sent_packets_); + return static_cast<int>(total_packet_delay_ / + static_cast<int64_t>(sent_packets_)); } void FakeNetworkPipe::Process() { - int64_t time_now = TickTime::MillisecondTimestamp(); + int64_t time_now = clock_->TimeInMilliseconds(); std::queue<NetworkPacket*> packets_to_deliver; { rtc::CritScope crit(&lock_); @@ -210,8 +212,8 @@ int64_t FakeNetworkPipe::TimeUntilNextProcess() const { const int64_t kDefaultProcessIntervalMs = 30; if (capacity_link_.size() == 0 || delay_link_.size() == 0) return kDefaultProcessIntervalMs; - return std::max<int64_t>( - next_process_time_ - TickTime::MillisecondTimestamp(), 0); + return std::max<int64_t>(next_process_time_ - clock_->TimeInMilliseconds(), + 0); } } // namespace webrtc diff --git a/webrtc/test/fake_network_pipe.h b/webrtc/test/fake_network_pipe.h index 74189a594c..5d589d86f0 100644 --- a/webrtc/test/fake_network_pipe.h +++ b/webrtc/test/fake_network_pipe.h @@ -16,11 +16,11 @@ #include "webrtc/base/constructormagic.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/typedefs.h" namespace webrtc { +class Clock; class CriticalSectionWrapper; class NetworkPacket; class PacketReceiver; @@ -33,26 +33,20 @@ class PacketReceiver; class FakeNetworkPipe { public: struct Config { - Config() - : queue_length_packets(0), - queue_delay_ms(0), - delay_standard_deviation_ms(0), - link_capacity_kbps(0), - loss_percent(0) { - } + Config() {} // Queue length in number of packets. - size_t queue_length_packets; + size_t queue_length_packets = 0; // Delay in addition to capacity induced delay. - int queue_delay_ms; + int queue_delay_ms = 0; // Standard deviation of the extra delay. - int delay_standard_deviation_ms; + int delay_standard_deviation_ms = 0; // Link capacity in kbps. - int link_capacity_kbps; + int link_capacity_kbps = 0; // Random packet loss. - int loss_percent; + int loss_percent = 0; }; - explicit FakeNetworkPipe(const FakeNetworkPipe::Config& config); + FakeNetworkPipe(Clock* clock, const FakeNetworkPipe::Config& config); ~FakeNetworkPipe(); // Must not be called in parallel with SendPacket or Process. @@ -76,6 +70,7 @@ class FakeNetworkPipe { size_t sent_packets() { return sent_packets_; } private: + Clock* const clock_; mutable rtc::CriticalSection lock_; PacketReceiver* packet_receiver_; std::queue<NetworkPacket*> capacity_link_; @@ -87,7 +82,7 @@ class FakeNetworkPipe { // Statistics. size_t dropped_packets_; size_t sent_packets_; - int total_packet_delay_; + int64_t total_packet_delay_; int64_t next_process_time_; diff --git a/webrtc/test/fake_network_pipe_unittest.cc b/webrtc/test/fake_network_pipe_unittest.cc index 02438c59f3..ff18993829 100644 --- a/webrtc/test/fake_network_pipe_unittest.cc +++ b/webrtc/test/fake_network_pipe_unittest.cc @@ -13,7 +13,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/call.h" -#include "webrtc/system_wrappers/include/tick_util.h" +#include "webrtc/system_wrappers/include/clock.h" #include "webrtc/test/fake_network_pipe.h" using ::testing::_; @@ -39,9 +39,11 @@ class MockReceiver : public PacketReceiver { }; class FakeNetworkPipeTest : public ::testing::Test { + public: + FakeNetworkPipeTest() : fake_clock_(12345) {} + protected: virtual void SetUp() { - TickTime::UseFakeClock(12345); receiver_.reset(new MockReceiver()); ON_CALL(*receiver_, DeliverPacket(_, _, _, _)) .WillByDefault(Return(PacketReceiver::DELIVERY_OK)); @@ -61,6 +63,7 @@ class FakeNetworkPipeTest : public ::testing::Test { return 8 * kPacketSize / capacity_kbps; } + SimulatedClock fake_clock_; rtc::scoped_ptr<MockReceiver> receiver_; }; @@ -71,7 +74,8 @@ TEST_F(FakeNetworkPipeTest, CapacityTest) { FakeNetworkPipe::Config config; config.queue_length_packets = 20; config.link_capacity_kbps = 80; - rtc::scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(config)); + rtc::scoped_ptr<FakeNetworkPipe> pipe( + new FakeNetworkPipe(&fake_clock_, config)); pipe->SetReceiver(receiver_.get()); // Add 10 packets of 1000 bytes, = 80 kb, and verify it takes one second to @@ -89,17 +93,17 @@ TEST_F(FakeNetworkPipeTest, CapacityTest) { pipe->Process(); // Advance enough time to release one packet. - TickTime::AdvanceFakeClock(kPacketTimeMs); + fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1); pipe->Process(); // Release all but one packet - TickTime::AdvanceFakeClock(9 * kPacketTimeMs - 1); + fake_clock_.AdvanceTimeMilliseconds(9 * kPacketTimeMs - 1); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(8); pipe->Process(); // And the last one. - TickTime::AdvanceFakeClock(1); + fake_clock_.AdvanceTimeMilliseconds(1); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1); pipe->Process(); } @@ -110,7 +114,8 @@ TEST_F(FakeNetworkPipeTest, ExtraDelayTest) { config.queue_length_packets = 20; config.queue_delay_ms = 100; config.link_capacity_kbps = 80; - rtc::scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(config)); + rtc::scoped_ptr<FakeNetworkPipe> pipe( + new FakeNetworkPipe(&fake_clock_, config)); pipe->SetReceiver(receiver_.get()); const int kNumPackets = 2; @@ -122,17 +127,17 @@ TEST_F(FakeNetworkPipeTest, ExtraDelayTest) { kPacketSize); // Increase more than kPacketTimeMs, but not more than the extra delay. - TickTime::AdvanceFakeClock(kPacketTimeMs); + fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0); pipe->Process(); // Advance the network delay to get the first packet. - TickTime::AdvanceFakeClock(config.queue_delay_ms); + fake_clock_.AdvanceTimeMilliseconds(config.queue_delay_ms); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1); pipe->Process(); // Advance one more kPacketTimeMs to get the last packet. - TickTime::AdvanceFakeClock(kPacketTimeMs); + fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1); pipe->Process(); } @@ -143,7 +148,8 @@ TEST_F(FakeNetworkPipeTest, QueueLengthTest) { FakeNetworkPipe::Config config; config.queue_length_packets = 2; config.link_capacity_kbps = 80; - rtc::scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(config)); + rtc::scoped_ptr<FakeNetworkPipe> pipe( + new FakeNetworkPipe(&fake_clock_, config)); pipe->SetReceiver(receiver_.get()); const int kPacketSize = 1000; @@ -155,7 +161,7 @@ TEST_F(FakeNetworkPipeTest, QueueLengthTest) { // Increase time enough to deliver all three packets, verify only two are // delivered. - TickTime::AdvanceFakeClock(3 * kPacketTimeMs); + fake_clock_.AdvanceTimeMilliseconds(3 * kPacketTimeMs); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(2); pipe->Process(); } @@ -166,7 +172,8 @@ TEST_F(FakeNetworkPipeTest, StatisticsTest) { config.queue_length_packets = 2; config.queue_delay_ms = 20; config.link_capacity_kbps = 80; - rtc::scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(config)); + rtc::scoped_ptr<FakeNetworkPipe> pipe( + new FakeNetworkPipe(&fake_clock_, config)); pipe->SetReceiver(receiver_.get()); const int kPacketSize = 1000; @@ -175,7 +182,8 @@ TEST_F(FakeNetworkPipeTest, StatisticsTest) { // Send three packets and verify only 2 are delivered. SendPackets(pipe.get(), 3, kPacketSize); - TickTime::AdvanceFakeClock(3 * kPacketTimeMs + config.queue_delay_ms); + fake_clock_.AdvanceTimeMilliseconds(3 * kPacketTimeMs + + config.queue_delay_ms); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(2); pipe->Process(); @@ -194,7 +202,8 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithEmptyPipeTest) { FakeNetworkPipe::Config config; config.queue_length_packets = 20; config.link_capacity_kbps = 80; - rtc::scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(config)); + rtc::scoped_ptr<FakeNetworkPipe> pipe( + new FakeNetworkPipe(&fake_clock_, config)); pipe->SetReceiver(receiver_.get()); // Add 10 packets of 1000 bytes, = 80 kb, and verify it takes one second to @@ -212,7 +221,7 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithEmptyPipeTest) { // Advance time in steps to release one packet at a time. for (int i = 0; i < kNumPackets; ++i) { - TickTime::AdvanceFakeClock(packet_time_ms); + fake_clock_.AdvanceTimeMilliseconds(packet_time_ms); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1); pipe->Process(); } @@ -234,14 +243,14 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithEmptyPipeTest) { // Advance time in steps to release one packet at a time. for (int i = 0; i < kNumPackets; ++i) { - TickTime::AdvanceFakeClock(packet_time_ms); + fake_clock_.AdvanceTimeMilliseconds(packet_time_ms); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1); pipe->Process(); } // Check that all the packets were sent. EXPECT_EQ(static_cast<size_t>(2 * kNumPackets), pipe->sent_packets()); - TickTime::AdvanceFakeClock(pipe->TimeUntilNextProcess()); + fake_clock_.AdvanceTimeMilliseconds(pipe->TimeUntilNextProcess()); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0); pipe->Process(); } @@ -252,7 +261,8 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithPacketsInPipeTest) { FakeNetworkPipe::Config config; config.queue_length_packets = 20; config.link_capacity_kbps = 80; - rtc::scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(config)); + rtc::scoped_ptr<FakeNetworkPipe> pipe( + new FakeNetworkPipe(&fake_clock_, config)); pipe->SetReceiver(receiver_.get()); // Add 10 packets of 1000 bytes, = 80 kb. @@ -280,21 +290,21 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithPacketsInPipeTest) { // Advance time in steps to release one packet at a time. for (int i = 0; i < kNumPackets; ++i) { - TickTime::AdvanceFakeClock(packet_time_1_ms); + fake_clock_.AdvanceTimeMilliseconds(packet_time_1_ms); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1); pipe->Process(); } // Advance time in steps to release one packet at a time. for (int i = 0; i < kNumPackets; ++i) { - TickTime::AdvanceFakeClock(packet_time_2_ms); + fake_clock_.AdvanceTimeMilliseconds(packet_time_2_ms); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(1); pipe->Process(); } // Check that all the packets were sent. EXPECT_EQ(static_cast<size_t>(2 * kNumPackets), pipe->sent_packets()); - TickTime::AdvanceFakeClock(pipe->TimeUntilNextProcess()); + fake_clock_.AdvanceTimeMilliseconds(pipe->TimeUntilNextProcess()); EXPECT_CALL(*receiver_, DeliverPacket(_, _, _, _)).Times(0); pipe->Process(); } diff --git a/webrtc/test/fake_texture_frame.cc b/webrtc/test/fake_texture_frame.cc new file mode 100644 index 0000000000..5d46eec4b6 --- /dev/null +++ b/webrtc/test/fake_texture_frame.cc @@ -0,0 +1,27 @@ +/* + * Copyright (c) 2016 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 "webrtc/test/fake_texture_frame.h" + +namespace webrtc { +namespace test { + +VideoFrame FakeNativeHandle::CreateFrame(FakeNativeHandle* native_handle, + int width, + int height, + uint32_t timestamp, + int64_t render_time_ms, + VideoRotation rotation) { + return VideoFrame(new rtc::RefCountedObject<FakeNativeHandleBuffer>( + native_handle, width, height), + timestamp, render_time_ms, rotation); +} +} // namespace test +} // namespace webrtc diff --git a/webrtc/test/fake_texture_frame.h b/webrtc/test/fake_texture_frame.h index dc6abaf745..9575fae469 100644 --- a/webrtc/test/fake_texture_frame.h +++ b/webrtc/test/fake_texture_frame.h @@ -11,13 +11,21 @@ #define WEBRTC_TEST_FAKE_TEXTURE_FRAME_H_ #include "webrtc/base/checks.h" -#include "webrtc/common_video/interface/video_frame_buffer.h" +#include "webrtc/common_video/include/video_frame_buffer.h" #include "webrtc/video_frame.h" namespace webrtc { namespace test { -class FakeNativeHandle {}; +class FakeNativeHandle { + public: + static VideoFrame CreateFrame(FakeNativeHandle* native_handle, + int width, + int height, + uint32_t timestamp, + int64_t render_time_ms, + VideoRotation rotation); +}; class FakeNativeHandleBuffer : public NativeHandleBuffer { public: @@ -41,16 +49,6 @@ class FakeNativeHandleBuffer : public NativeHandleBuffer { } }; -static VideoFrame CreateFakeNativeHandleFrame(FakeNativeHandle* native_handle, - int width, - int height, - uint32_t timestamp, - int64_t render_time_ms, - VideoRotation rotation) { - return VideoFrame(new rtc::RefCountedObject<FakeNativeHandleBuffer>( - native_handle, width, height), - timestamp, render_time_ms, rotation); -} } // namespace test } // namespace webrtc #endif // WEBRTC_TEST_FAKE_TEXTURE_FRAME_H_ diff --git a/webrtc/test/fake_voice_engine.cc b/webrtc/test/fake_voice_engine.cc deleted file mode 100644 index 1a32e082b7..0000000000 --- a/webrtc/test/fake_voice_engine.cc +++ /dev/null @@ -1,70 +0,0 @@ -/* - * Copyright (c) 2015 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 "webrtc/test/fake_voice_engine.h" - -namespace { - -webrtc::AudioDecodingCallStats MakeAudioDecodingCallStats() { - webrtc::AudioDecodingCallStats stats; - stats.calls_to_silence_generator = 234; - stats.calls_to_neteq = 567; - stats.decoded_normal = 890; - stats.decoded_plc = 123; - stats.decoded_cng = 456; - stats.decoded_plc_cng = 789; - return stats; -} -} // namespace - -namespace webrtc { -namespace test { - -const int FakeVoiceEngine::kSendChannelId = 1; -const int FakeVoiceEngine::kRecvChannelId = 2; -const uint32_t FakeVoiceEngine::kSendSsrc = 665; -const uint32_t FakeVoiceEngine::kRecvSsrc = 667; -const int FakeVoiceEngine::kSendEchoDelayMedian = 254; -const int FakeVoiceEngine::kSendEchoDelayStdDev = -3; -const int FakeVoiceEngine::kSendEchoReturnLoss = -65; -const int FakeVoiceEngine::kSendEchoReturnLossEnhancement = 101; -const int FakeVoiceEngine::kRecvJitterBufferDelay = -7; -const int FakeVoiceEngine::kRecvPlayoutBufferDelay = 302; -const unsigned int FakeVoiceEngine::kSendSpeechInputLevel = 96; -const unsigned int FakeVoiceEngine::kRecvSpeechOutputLevel = 99; - -const CallStatistics FakeVoiceEngine::kSendCallStats = { - 1345, 1678, 1901, 1234, 112, 13456, 17890, 1567, -1890, -1123 -}; - -const CodecInst FakeVoiceEngine::kSendCodecInst = { - -121, "codec_name_send", 48000, -231, -451, -671 -}; - -const ReportBlock FakeVoiceEngine::kSendReportBlock = { - 456, 780, 123, 567, 890, 132, 143, 13354 -}; - -const CallStatistics FakeVoiceEngine::kRecvCallStats = { - 345, 678, 901, 234, -12, 3456, 7890, 567, 890, 123 -}; - -const CodecInst FakeVoiceEngine::kRecvCodecInst = { - 123, "codec_name_recv", 96000, -187, -198, -103 -}; - -const NetworkStatistics FakeVoiceEngine::kRecvNetworkStats = { - 123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0 -}; - -const AudioDecodingCallStats FakeVoiceEngine::kRecvAudioDecodingCallStats = - MakeAudioDecodingCallStats(); -} // namespace test -} // namespace webrtc diff --git a/webrtc/test/fake_voice_engine.h b/webrtc/test/fake_voice_engine.h deleted file mode 100644 index 8f08929720..0000000000 --- a/webrtc/test/fake_voice_engine.h +++ /dev/null @@ -1,504 +0,0 @@ -/* - * Copyright (c) 2015 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. - */ - -#ifndef WEBRTC_AUDIO_FAKE_VOICE_ENGINE_H_ -#define WEBRTC_AUDIO_FAKE_VOICE_ENGINE_H_ - -#include <vector> - -#include "testing/gtest/include/gtest/gtest.h" - -#include "webrtc/voice_engine/voice_engine_impl.h" - -namespace webrtc { -namespace test { - -// NOTE: This class inherits from VoiceEngineImpl so that its clients will be -// able to get the various interfaces as usual, via T::GetInterface(). -class FakeVoiceEngine final : public VoiceEngineImpl { - public: - static const int kSendChannelId; - static const int kRecvChannelId; - static const uint32_t kSendSsrc; - static const uint32_t kRecvSsrc; - static const int kSendEchoDelayMedian; - static const int kSendEchoDelayStdDev; - static const int kSendEchoReturnLoss; - static const int kSendEchoReturnLossEnhancement; - static const int kRecvJitterBufferDelay; - static const int kRecvPlayoutBufferDelay; - static const unsigned int kSendSpeechInputLevel; - static const unsigned int kRecvSpeechOutputLevel; - static const CallStatistics kSendCallStats; - static const CodecInst kSendCodecInst; - static const ReportBlock kSendReportBlock; - static const CallStatistics kRecvCallStats; - static const CodecInst kRecvCodecInst; - static const NetworkStatistics kRecvNetworkStats; - static const AudioDecodingCallStats kRecvAudioDecodingCallStats; - - FakeVoiceEngine() : VoiceEngineImpl(new Config(), true) { - // Increase ref count so this object isn't automatically deleted whenever - // interfaces are Release():d. - ++_ref_count; - } - ~FakeVoiceEngine() override { - // Decrease ref count before base class d-tor is called; otherwise it will - // trigger an assertion. - --_ref_count; - } - - // VoEAudioProcessing - int SetNsStatus(bool enable, NsModes mode = kNsUnchanged) override { - return -1; - } - int GetNsStatus(bool& enabled, NsModes& mode) override { return -1; } - int SetAgcStatus(bool enable, AgcModes mode = kAgcUnchanged) override { - return -1; - } - int GetAgcStatus(bool& enabled, AgcModes& mode) override { return -1; } - int SetAgcConfig(AgcConfig config) override { return -1; } - int GetAgcConfig(AgcConfig& config) override { return -1; } - int SetEcStatus(bool enable, EcModes mode = kEcUnchanged) override { - return -1; - } - int GetEcStatus(bool& enabled, EcModes& mode) override { return -1; } - int EnableDriftCompensation(bool enable) override { return -1; } - bool DriftCompensationEnabled() override { return false; } - void SetDelayOffsetMs(int offset) override {} - int DelayOffsetMs() override { return -1; } - int SetAecmMode(AecmModes mode = kAecmSpeakerphone, - bool enableCNG = true) override { return -1; } - int GetAecmMode(AecmModes& mode, bool& enabledCNG) override { return -1; } - int EnableHighPassFilter(bool enable) override { return -1; } - bool IsHighPassFilterEnabled() override { return false; } - int SetRxNsStatus(int channel, - bool enable, - NsModes mode = kNsUnchanged) override { return -1; } - int GetRxNsStatus(int channel, bool& enabled, NsModes& mode) override { - return -1; - } - int SetRxAgcStatus(int channel, - bool enable, - AgcModes mode = kAgcUnchanged) override { return -1; } - int GetRxAgcStatus(int channel, bool& enabled, AgcModes& mode) override { - return -1; - } - int SetRxAgcConfig(int channel, AgcConfig config) override { return -1; } - int GetRxAgcConfig(int channel, AgcConfig& config) override { return -1; } - int RegisterRxVadObserver(int channel, - VoERxVadCallback& observer) override { return -1; } - int DeRegisterRxVadObserver(int channel) override { return -1; } - int VoiceActivityIndicator(int channel) override { return -1; } - int SetEcMetricsStatus(bool enable) override { return -1; } - int GetEcMetricsStatus(bool& enabled) override { - enabled = true; - return 0; - } - int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP) override { - ERL = kSendEchoReturnLoss; - ERLE = kSendEchoReturnLossEnhancement; - RERL = -123456789; - A_NLP = 123456789; - return 0; - } - int GetEcDelayMetrics(int& delay_median, - int& delay_std, - float& fraction_poor_delays) override { - delay_median = kSendEchoDelayMedian; - delay_std = kSendEchoDelayStdDev; - fraction_poor_delays = -12345.7890f; - return 0; - } - int StartDebugRecording(const char* fileNameUTF8) override { return -1; } - int StartDebugRecording(FILE* file_handle) override { return -1; } - int StopDebugRecording() override { return -1; } - int SetTypingDetectionStatus(bool enable) override { return -1; } - int GetTypingDetectionStatus(bool& enabled) override { return -1; } - int TimeSinceLastTyping(int& seconds) override { return -1; } - int SetTypingDetectionParameters(int timeWindow, - int costPerTyping, - int reportingThreshold, - int penaltyDecay, - int typeEventDelay = 0) override { - return -1; - } - void EnableStereoChannelSwapping(bool enable) override {} - bool IsStereoChannelSwappingEnabled() override { return false; } - - // VoEBase - int RegisterVoiceEngineObserver(VoiceEngineObserver& observer) override { - return -1; - } - int DeRegisterVoiceEngineObserver() override { return -1; } - int Init(AudioDeviceModule* external_adm = NULL, - AudioProcessing* audioproc = NULL) override { return -1; } - AudioProcessing* audio_processing() override { return nullptr; } - int Terminate() override { return -1; } - int CreateChannel() override { return -1; } - int CreateChannel(const Config& config) override { return -1; } - int DeleteChannel(int channel) override { return -1; } - int StartReceive(int channel) override { return -1; } - int StopReceive(int channel) override { return -1; } - int StartPlayout(int channel) override { return -1; } - int StopPlayout(int channel) override { return -1; } - int StartSend(int channel) override { return -1; } - int StopSend(int channel) override { return -1; } - int GetVersion(char version[1024]) override { return -1; } - int LastError() override { return -1; } - AudioTransport* audio_transport() { return nullptr; } - int AssociateSendChannel(int channel, int accociate_send_channel) override { - return -1; - } - - // VoECodec - int NumOfCodecs() override { return -1; } - int GetCodec(int index, CodecInst& codec) override { return -1; } - int SetSendCodec(int channel, const CodecInst& codec) override { return -1; } - int GetSendCodec(int channel, CodecInst& codec) override { - EXPECT_EQ(channel, kSendChannelId); - codec = kSendCodecInst; - return 0; - } - int SetBitRate(int channel, int bitrate_bps) override { return -1; } - int GetRecCodec(int channel, CodecInst& codec) override { - EXPECT_EQ(channel, kRecvChannelId); - codec = kRecvCodecInst; - return 0; - } - int SetRecPayloadType(int channel, const CodecInst& codec) override { - return -1; - } - int GetRecPayloadType(int channel, CodecInst& codec) override { return -1; } - int SetSendCNPayloadType(int channel, int type, - PayloadFrequencies frequency = kFreq16000Hz) override { return -1; } - int SetVADStatus(int channel, - bool enable, - VadModes mode = kVadConventional, - bool disableDTX = false) override { return -1; } - int GetVADStatus(int channel, - bool& enabled, - VadModes& mode, - bool& disabledDTX) override { return -1; } - int SetOpusMaxPlaybackRate(int channel, int frequency_hz) override { - return -1; - } - int SetOpusDtx(int channel, bool enable_dtx) override { return -1; } - RtcEventLog* GetEventLog() override { return nullptr; } - - // VoEDtmf - int SendTelephoneEvent(int channel, - int eventCode, - bool outOfBand = true, - int lengthMs = 160, - int attenuationDb = 10) override { return -1; } - int SetSendTelephoneEventPayloadType(int channel, - unsigned char type) override { - return -1; - } - int GetSendTelephoneEventPayloadType(int channel, - unsigned char& type) override { - return -1; - } - int SetDtmfFeedbackStatus(bool enable, - bool directFeedback = false) override { return -1; } - int GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback) override { - return -1; - } - int PlayDtmfTone(int eventCode, - int lengthMs = 200, - int attenuationDb = 10) override { return -1; } - - // VoEExternalMedia - int RegisterExternalMediaProcessing( - int channel, - ProcessingTypes type, - VoEMediaProcess& processObject) override { return -1; } - int DeRegisterExternalMediaProcessing(int channel, - ProcessingTypes type) override { - return -1; - } - int GetAudioFrame(int channel, - int desired_sample_rate_hz, - AudioFrame* frame) override { return -1; } - int SetExternalMixing(int channel, bool enable) override { return -1; } - - // VoEFile - int StartPlayingFileLocally( - int channel, - const char fileNameUTF8[1024], - bool loop = false, - FileFormats format = kFileFormatPcm16kHzFile, - float volumeScaling = 1.0, - int startPointMs = 0, - int stopPointMs = 0) override { return -1; } - int StartPlayingFileLocally( - int channel, - InStream* stream, - FileFormats format = kFileFormatPcm16kHzFile, - float volumeScaling = 1.0, - int startPointMs = 0, - int stopPointMs = 0) override { return -1; } - int StopPlayingFileLocally(int channel) override { return -1; } - int IsPlayingFileLocally(int channel) override { return -1; } - int StartPlayingFileAsMicrophone( - int channel, - const char fileNameUTF8[1024], - bool loop = false, - bool mixWithMicrophone = false, - FileFormats format = kFileFormatPcm16kHzFile, - float volumeScaling = 1.0) override { return -1; } - int StartPlayingFileAsMicrophone( - int channel, - InStream* stream, - bool mixWithMicrophone = false, - FileFormats format = kFileFormatPcm16kHzFile, - float volumeScaling = 1.0) override { return -1; } - int StopPlayingFileAsMicrophone(int channel) override { return -1; } - int IsPlayingFileAsMicrophone(int channel) override { return -1; } - int StartRecordingPlayout(int channel, - const char* fileNameUTF8, - CodecInst* compression = NULL, - int maxSizeBytes = -1) override { return -1; } - int StopRecordingPlayout(int channel) override { return -1; } - int StartRecordingPlayout(int channel, - OutStream* stream, - CodecInst* compression = NULL) override { - return -1; - } - int StartRecordingMicrophone(const char* fileNameUTF8, - CodecInst* compression = NULL, - int maxSizeBytes = -1) override { return -1; } - int StartRecordingMicrophone(OutStream* stream, - CodecInst* compression = NULL) override { - return -1; - } - int StopRecordingMicrophone() override { return -1; } - - // VoEHardware - int GetNumOfRecordingDevices(int& devices) override { return -1; } - - // Gets the number of audio devices available for playout. - int GetNumOfPlayoutDevices(int& devices) override { return -1; } - - // Gets the name of a specific recording device given by an |index|. - // On Windows Vista/7, it also retrieves an additional unique ID - // (GUID) for the recording device. - int GetRecordingDeviceName(int index, - char strNameUTF8[128], - char strGuidUTF8[128]) override { return -1; } - - // Gets the name of a specific playout device given by an |index|. - // On Windows Vista/7, it also retrieves an additional unique ID - // (GUID) for the playout device. - int GetPlayoutDeviceName(int index, - char strNameUTF8[128], - char strGuidUTF8[128]) override { return -1; } - - // Sets the audio device used for recording. - int SetRecordingDevice( - int index, - StereoChannel recordingChannel = kStereoBoth) override { return -1; } - - // Sets the audio device used for playout. - int SetPlayoutDevice(int index) override { return -1; } - - // Sets the type of audio device layer to use. - int SetAudioDeviceLayer(AudioLayers audioLayer) override { return -1; } - - // Gets the currently used (active) audio device layer. - int GetAudioDeviceLayer(AudioLayers& audioLayer) override { return -1; } - - // Native sample rate controls (samples/sec) - int SetRecordingSampleRate(unsigned int samples_per_sec) override { - return -1; - } - int RecordingSampleRate(unsigned int* samples_per_sec) const override { - return -1; - } - int SetPlayoutSampleRate(unsigned int samples_per_sec) override { - return -1; - } - int PlayoutSampleRate(unsigned int* samples_per_sec) const override { - return -1; - } - - // Queries and controls platform audio effects on Android devices. - bool BuiltInAECIsAvailable() const override { return false; } - int EnableBuiltInAEC(bool enable) override { return -1; } - bool BuiltInAGCIsAvailable() const override { return false; } - int EnableBuiltInAGC(bool enable) override { return -1; } - bool BuiltInNSIsAvailable() const override { return false; } - int EnableBuiltInNS(bool enable) override { return -1; } - - // VoENetwork - int RegisterExternalTransport(int channel, Transport& transport) override { - return -1; - } - int DeRegisterExternalTransport(int channel) override { return -1; } - int ReceivedRTPPacket(int channel, - const void* data, - size_t length) override { return -1; } - int ReceivedRTPPacket(int channel, - const void* data, - size_t length, - const PacketTime& packet_time) override { return -1; } - int ReceivedRTCPPacket(int channel, - const void* data, - size_t length) { return -1; } - - // VoENetEqStats - int GetNetworkStatistics(int channel, NetworkStatistics& stats) override { - EXPECT_EQ(channel, kRecvChannelId); - stats = kRecvNetworkStats; - return 0; - } - int GetDecodingCallStatistics(int channel, - AudioDecodingCallStats* stats) const override { - EXPECT_EQ(channel, kRecvChannelId); - EXPECT_NE(nullptr, stats); - *stats = kRecvAudioDecodingCallStats; - return 0; - } - - // VoERTP_RTCP - int SetLocalSSRC(int channel, unsigned int ssrc) override { return -1; } - int GetLocalSSRC(int channel, unsigned int& ssrc) override { - EXPECT_EQ(channel, kSendChannelId); - ssrc = 0; - return 0; - } - int GetRemoteSSRC(int channel, unsigned int& ssrc) override { - EXPECT_EQ(channel, kRecvChannelId); - ssrc = 0; - return 0; - } - int SetSendAudioLevelIndicationStatus(int channel, - bool enable, - unsigned char id = 1) override { - return -1; - } - int SetSendAbsoluteSenderTimeStatus(int channel, - bool enable, - unsigned char id) override { return -1; } - int SetReceiveAbsoluteSenderTimeStatus(int channel, - bool enable, - unsigned char id) override { - return -1; - } - int SetRTCPStatus(int channel, bool enable) override { return -1; } - int GetRTCPStatus(int channel, bool& enabled) override { return -1; } - int SetRTCP_CNAME(int channel, const char cName[256]) override { return -1; } - int GetRTCP_CNAME(int channel, char cName[256]) { return -1; } - int GetRemoteRTCP_CNAME(int channel, char cName[256]) override { return -1; } - int GetRemoteRTCPData(int channel, - unsigned int& NTPHigh, - unsigned int& NTPLow, - unsigned int& timestamp, - unsigned int& playoutTimestamp, - unsigned int* jitter = NULL, - unsigned short* fractionLost = NULL) override { - return -1; - } - int GetRTPStatistics(int channel, - unsigned int& averageJitterMs, - unsigned int& maxJitterMs, - unsigned int& discardedPackets) override { return -1; } - int GetRTCPStatistics(int channel, CallStatistics& stats) override { - if (channel == kSendChannelId) { - stats = kSendCallStats; - } else { - EXPECT_EQ(channel, kRecvChannelId); - stats = kRecvCallStats; - } - return 0; - } - int GetRemoteRTCPReportBlocks( - int channel, - std::vector<ReportBlock>* receive_blocks) override { - EXPECT_EQ(channel, kSendChannelId); - EXPECT_NE(receive_blocks, nullptr); - EXPECT_EQ(receive_blocks->size(), 0u); - webrtc::ReportBlock block = kSendReportBlock; - receive_blocks->push_back(block); // Has wrong SSRC. - block.source_SSRC = kSendSsrc; - receive_blocks->push_back(block); // Correct block. - block.fraction_lost = 0; - receive_blocks->push_back(block); // Duplicate SSRC, bad fraction_lost. - return 0; - } - int SetNACKStatus(int channel, bool enable, int maxNoPackets) override { - return -1; - } - - // VoEVideoSync - int GetPlayoutBufferSize(int& buffer_ms) override { return -1; } - int SetMinimumPlayoutDelay(int channel, int delay_ms) override { return -1; } - int SetInitialPlayoutDelay(int channel, int delay_ms) override { return -1; } - int GetDelayEstimate(int channel, - int* jitter_buffer_delay_ms, - int* playout_buffer_delay_ms) override { - EXPECT_EQ(channel, kRecvChannelId); - *jitter_buffer_delay_ms = kRecvJitterBufferDelay; - *playout_buffer_delay_ms = kRecvPlayoutBufferDelay; - return 0; - } - int GetLeastRequiredDelayMs(int channel) const override { return -1; } - int SetInitTimestamp(int channel, unsigned int timestamp) override { - return -1; - } - int SetInitSequenceNumber(int channel, short sequenceNumber) override { - return -1; - } - int GetPlayoutTimestamp(int channel, unsigned int& timestamp) override { - return -1; - } - int GetRtpRtcp(int channel, - RtpRtcp** rtpRtcpModule, - RtpReceiver** rtp_receiver) override { return -1; } - - // VoEVolumeControl - int SetSpeakerVolume(unsigned int volume) override { return -1; } - int GetSpeakerVolume(unsigned int& volume) override { return -1; } - int SetMicVolume(unsigned int volume) override { return -1; } - int GetMicVolume(unsigned int& volume) override { return -1; } - int SetInputMute(int channel, bool enable) override { return -1; } - int GetInputMute(int channel, bool& enabled) override { return -1; } - int GetSpeechInputLevel(unsigned int& level) override { return -1; } - int GetSpeechOutputLevel(int channel, unsigned int& level) override { - return -1; - } - int GetSpeechInputLevelFullRange(unsigned int& level) override { - level = kSendSpeechInputLevel; - return 0; - } - int GetSpeechOutputLevelFullRange(int channel, - unsigned int& level) override { - EXPECT_EQ(channel, kRecvChannelId); - level = kRecvSpeechOutputLevel; - return 0; - } - int SetChannelOutputVolumeScaling(int channel, float scaling) override { - return -1; - } - int GetChannelOutputVolumeScaling(int channel, float& scaling) override { - return -1; - } - int SetOutputVolumePan(int channel, float left, float right) override { - return -1; - } - int GetOutputVolumePan(int channel, float& left, float& right) override { - return -1; - } -}; -} // namespace test -} // namespace webrtc - -#endif // WEBRTC_AUDIO_FAKE_VOICE_ENGINE_H_ diff --git a/webrtc/test/field_trial.cc b/webrtc/test/field_trial.cc index 613fb67679..c40d0783d8 100644 --- a/webrtc/test/field_trial.cc +++ b/webrtc/test/field_trial.cc @@ -18,27 +18,13 @@ #include <string> #include "webrtc/system_wrappers/include/field_trial.h" +#include "webrtc/system_wrappers/include/field_trial_default.h" namespace webrtc { namespace { -// Clients of this library have show a clear intent to setup field trials by -// linking with it. As so try to crash if they forget to call -// InitFieldTrialsFromString before webrtc tries to access a field trial. bool field_trials_initiated_ = false; -std::map<std::string, std::string> field_trials_; } // namespace -namespace field_trial { -std::string FindFullName(const std::string& trial_name) { - assert(field_trials_initiated_); - std::map<std::string, std::string>::const_iterator it = - field_trials_.find(trial_name); - if (it == field_trials_.end()) - return std::string(); - return it->second; -} -} // namespace field_trial - namespace test { // Note: this code is copied from src/base/metrics/field_trial.cc since the aim // is to mimic chromium --force-fieldtrials. @@ -53,6 +39,7 @@ void InitFieldTrialsFromString(const std::string& trials_string) { return; size_t next_item = 0; + std::map<std::string, std::string> field_trials; while (next_item < trials_string.length()) { size_t name_end = trials_string.find(kPersistentStringSeparator, next_item); if (name_end == trials_string.npos || next_item == name_end) @@ -67,15 +54,18 @@ void InitFieldTrialsFromString(const std::string& trials_string) { next_item = group_name_end + 1; // Fail if duplicate with different group name. - if (field_trials_.find(name) != field_trials_.end() && - field_trials_.find(name)->second != group_name) + if (field_trials.find(name) != field_trials.end() && + field_trials.find(name)->second != group_name) { break; + } - field_trials_[name] = group_name; + field_trials[name] = group_name; // Successfully parsed all field trials from the string. - if (next_item == trials_string.length()) + if (next_item == trials_string.length()) { + webrtc::field_trial::InitFieldTrialsFromString(trials_string.c_str()); return; + } } // Using fprintf as LOG does not print when this is called early in main. fprintf(stderr, "Invalid field trials string.\n"); @@ -85,18 +75,18 @@ void InitFieldTrialsFromString(const std::string& trials_string) { } ScopedFieldTrials::ScopedFieldTrials(const std::string& config) - : previous_field_trials_(field_trials_) { + : previous_field_trials_(webrtc::field_trial::GetFieldTrialString()) { assert(field_trials_initiated_); field_trials_initiated_ = false; - field_trials_.clear(); - InitFieldTrialsFromString(config); + current_field_trials_ = config; + InitFieldTrialsFromString(current_field_trials_); } ScopedFieldTrials::~ScopedFieldTrials() { // Should still be initialized, since InitFieldTrials is called from ctor. // That's why we don't restore the flag. assert(field_trials_initiated_); - field_trials_ = previous_field_trials_; + webrtc::field_trial::InitFieldTrialsFromString(previous_field_trials_); } } // namespace test diff --git a/webrtc/test/field_trial.h b/webrtc/test/field_trial.h index d448f3411d..735aa1f833 100644 --- a/webrtc/test/field_trial.h +++ b/webrtc/test/field_trial.h @@ -39,7 +39,8 @@ class ScopedFieldTrials { explicit ScopedFieldTrials(const std::string& config); ~ScopedFieldTrials(); private: - const std::map<std::string, std::string> previous_field_trials_; + std::string current_field_trials_; + const char* previous_field_trials_; }; } // namespace test diff --git a/webrtc/test/frame_generator_capturer.cc b/webrtc/test/frame_generator_capturer.cc index 70e2c85698..35ce6168a2 100644 --- a/webrtc/test/frame_generator_capturer.cc +++ b/webrtc/test/frame_generator_capturer.cc @@ -11,11 +11,11 @@ #include "webrtc/test/frame_generator_capturer.h" #include "webrtc/base/criticalsection.h" -#include "webrtc/test/frame_generator.h" +#include "webrtc/base/platform_thread.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/sleep.h" -#include "webrtc/system_wrappers/include/thread_wrapper.h" +#include "webrtc/test/frame_generator.h" #include "webrtc/video_send_stream.h" namespace webrtc { @@ -65,6 +65,7 @@ FrameGeneratorCapturer::FrameGeneratorCapturer(Clock* clock, clock_(clock), sending_(false), tick_(EventTimerWrapper::Create()), + thread_(FrameGeneratorCapturer::Run, this, "FrameGeneratorCapturer"), frame_generator_(frame_generator), target_fps_(target_fps), first_frame_capture_time_(-1) { @@ -76,8 +77,7 @@ FrameGeneratorCapturer::FrameGeneratorCapturer(Clock* clock, FrameGeneratorCapturer::~FrameGeneratorCapturer() { Stop(); - if (thread_.get() != NULL) - thread_->Stop(); + thread_.Stop(); } bool FrameGeneratorCapturer::Init() { @@ -88,15 +88,8 @@ bool FrameGeneratorCapturer::Init() { if (!tick_->StartTimer(true, 1000 / target_fps_)) return false; - thread_ = ThreadWrapper::CreateThread(FrameGeneratorCapturer::Run, this, - "FrameGeneratorCapturer"); - if (thread_.get() == NULL) - return false; - if (!thread_->Start()) { - thread_.reset(); - return false; - } - thread_->SetPriority(webrtc::kHighPriority); + thread_.Start(); + thread_.SetPriority(rtc::kHighPriority); return true; } diff --git a/webrtc/test/frame_generator_capturer.h b/webrtc/test/frame_generator_capturer.h index aff906dfa2..6bd0e0b327 100644 --- a/webrtc/test/frame_generator_capturer.h +++ b/webrtc/test/frame_generator_capturer.h @@ -7,12 +7,13 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_FRAME_GENERATOR_CAPTURER_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_FRAME_GENERATOR_CAPTURER_H_ +#ifndef WEBRTC_TEST_FRAME_GENERATOR_CAPTURER_H_ +#define WEBRTC_TEST_FRAME_GENERATOR_CAPTURER_H_ #include <string> #include "webrtc/base/criticalsection.h" +#include "webrtc/base/platform_thread.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/test/video_capturer.h" #include "webrtc/typedefs.h" @@ -21,7 +22,6 @@ namespace webrtc { class CriticalSectionWrapper; class EventTimerWrapper; -class ThreadWrapper; namespace test { @@ -64,7 +64,7 @@ class FrameGeneratorCapturer : public VideoCapturer { rtc::scoped_ptr<EventTimerWrapper> tick_; rtc::CriticalSection lock_; - rtc::scoped_ptr<ThreadWrapper> thread_; + rtc::PlatformThread thread_; rtc::scoped_ptr<FrameGenerator> frame_generator_; int target_fps_; @@ -74,4 +74,4 @@ class FrameGeneratorCapturer : public VideoCapturer { } // test } // webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_FRAME_GENERATOR_CAPTURER_H_ +#endif // WEBRTC_TEST_FRAME_GENERATOR_CAPTURER_H_ diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn new file mode 100644 index 0000000000..6a43548ec9 --- /dev/null +++ b/webrtc/test/fuzzers/BUILD.gn @@ -0,0 +1,115 @@ +# Copyright (c) 2015 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. + +import("//build/config/features.gni") +import("webrtc_fuzzer.gni") + +static_library("webrtc_fuzzer_main") { + public_configs = [ "../..:common_inherited_config" ] + sources = [ + "webrtc_fuzzer_main.cc", + ] + deps = [ + "../../system_wrappers:field_trial_default", + "../../system_wrappers:metrics_default", + "//testing/libfuzzer:libfuzzer_main", + ] +} + +webrtc_fuzzer_test("h264_depacketizer_fuzzer") { + sources = [ + "h264_depacketizer_fuzzer.cc", + ] + deps = [ + "../../modules/rtp_rtcp", + ] +} + +webrtc_fuzzer_test("vp8_depacketizer_fuzzer") { + sources = [ + "vp8_depacketizer_fuzzer.cc", + ] + deps = [ + "../../modules/rtp_rtcp", + ] +} + +webrtc_fuzzer_test("vp9_depacketizer_fuzzer") { + sources = [ + "vp9_depacketizer_fuzzer.cc", + ] + deps = [ + "../../modules/rtp_rtcp", + ] +} + +webrtc_fuzzer_test("vp8_qp_parser_fuzzer") { + sources = [ + "vp8_qp_parser_fuzzer.cc", + ] + deps = [ + "../../modules/video_coding/", + ] +} + +webrtc_fuzzer_test("producer_fec_fuzzer") { + sources = [ + "producer_fec_fuzzer.cc", + ] + deps = [ + "../../modules/rtp_rtcp/", + ] +} + +source_set("audio_decoder_fuzzer") { + public_configs = [ "../..:common_inherited_config" ] + sources = [ + "audio_decoder_fuzzer.cc", + "audio_decoder_fuzzer.h", + ] +} + +webrtc_fuzzer_test("audio_decoder_ilbc_fuzzer") { + sources = [ + "audio_decoder_ilbc_fuzzer.cc", + ] + deps = [ + ":audio_decoder_fuzzer", + "../../modules/audio_coding:ilbc", + ] +} + +webrtc_fuzzer_test("audio_decoder_isac_fuzzer") { + sources = [ + "audio_decoder_isac_fuzzer.cc", + ] + deps = [ + ":audio_decoder_fuzzer", + "../../modules/audio_coding:isac", + ] +} + +webrtc_fuzzer_test("audio_decoder_isacfix_fuzzer") { + sources = [ + "audio_decoder_isacfix_fuzzer.cc", + ] + deps = [ + ":audio_decoder_fuzzer", + "../../modules/audio_coding:isac_fix", + ] +} + +webrtc_fuzzer_test("audio_decoder_opus_fuzzer") { + sources = [ + "audio_decoder_opus_fuzzer.cc", + ] + deps = [ + ":audio_decoder_fuzzer", + "../../modules/audio_coding:webrtc_opus", + ] +} diff --git a/webrtc/test/fuzzers/OWNERS b/webrtc/test/fuzzers/OWNERS new file mode 100644 index 0000000000..6782b61fca --- /dev/null +++ b/webrtc/test/fuzzers/OWNERS @@ -0,0 +1 @@ +pbos@webrtc.org diff --git a/webrtc/test/fuzzers/audio_decoder_fuzzer.cc b/webrtc/test/fuzzers/audio_decoder_fuzzer.cc new file mode 100644 index 0000000000..fb5adb6cd8 --- /dev/null +++ b/webrtc/test/fuzzers/audio_decoder_fuzzer.cc @@ -0,0 +1,49 @@ +/* + * Copyright (c) 2015 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 "webrtc/test/fuzzers/audio_decoder_fuzzer.h" + +#include "webrtc/base/checks.h" +#include "webrtc/modules/audio_coding/codecs/audio_decoder.h" + +namespace webrtc { +namespace { +size_t PacketSizeFromTwoBytes(const uint8_t* data, size_t size) { + if (size < 2) + return 0; + return static_cast<size_t>((data[0] << 8) + data[1]); +} +} // namespace + +// This function reads two bytes from the beginning of |data|, interprets them +// as the first packet length, and reads this many bytes if available. The +// payload is inserted into the decoder, and the process continues until no more +// data is available. +void FuzzAudioDecoder(const uint8_t* data, + size_t size, + AudioDecoder* decoder, + int sample_rate_hz, + size_t max_decoded_bytes, + int16_t* decoded) { + const uint8_t* data_ptr = data; + size_t remaining_size = size; + size_t packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size); + while (packet_len != 0 && packet_len <= remaining_size - 2) { + data_ptr += 2; + remaining_size -= 2; + AudioDecoder::SpeechType speech_type; + decoder->Decode(data_ptr, packet_len, sample_rate_hz, max_decoded_bytes, + decoded, &speech_type); + data_ptr += packet_len; + remaining_size -= packet_len; + packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size); + } +} +} // namespace webrtc diff --git a/webrtc/test/fuzzers/audio_decoder_fuzzer.h b/webrtc/test/fuzzers/audio_decoder_fuzzer.h new file mode 100644 index 0000000000..cdd8574300 --- /dev/null +++ b/webrtc/test/fuzzers/audio_decoder_fuzzer.h @@ -0,0 +1,31 @@ +/* + * Copyright (c) 2015 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. + */ + +#ifndef WEBRTC_TEST_FUZZERS_AUDIO_DECODER_FUZZER_H_ +#define WEBRTC_TEST_FUZZERS_AUDIO_DECODER_FUZZER_H_ + +#include <stddef.h> + +#include "webrtc/typedefs.h" + +namespace webrtc { + +class AudioDecoder; + +void FuzzAudioDecoder(const uint8_t* data, + size_t size, + AudioDecoder* decoder, + int sample_rate_hz, + size_t max_decoded_bytes, + int16_t* decoded); + +} // namespace webrtc + +#endif // WEBRTC_TEST_FUZZERS_AUDIO_DECODER_FUZZER_H_ diff --git a/webrtc/test/fuzzers/audio_decoder_ilbc_fuzzer.cc b/webrtc/test/fuzzers/audio_decoder_ilbc_fuzzer.cc new file mode 100644 index 0000000000..d2a87f0cb6 --- /dev/null +++ b/webrtc/test/fuzzers/audio_decoder_ilbc_fuzzer.cc @@ -0,0 +1,22 @@ +/* + * Copyright (c) 2015 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 "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h" +#include "webrtc/test/fuzzers/audio_decoder_fuzzer.h" + +namespace webrtc { +void FuzzOneInput(const uint8_t* data, size_t size) { + AudioDecoderIlbc dec; + static const int kSampleRateHz = 8000; + static const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10; + int16_t output[kAllocatedOuputSizeSamples]; + FuzzAudioDecoder(data, size, &dec, kSampleRateHz, sizeof(output), output); +} +} // namespace webrtc diff --git a/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc b/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc new file mode 100644 index 0000000000..984cfda398 --- /dev/null +++ b/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc @@ -0,0 +1,22 @@ +/* + * Copyright (c) 2015 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 "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h" +#include "webrtc/test/fuzzers/audio_decoder_fuzzer.h" + +namespace webrtc { +void FuzzOneInput(const uint8_t* data, size_t size) { + AudioDecoderIsac dec(nullptr); + const int sample_rate_hz = size % 2 == 0 ? 16000 : 32000; // 16 or 32 kHz. + static const size_t kAllocatedOuputSizeSamples = 32000 / 10; // 100 ms. + int16_t output[kAllocatedOuputSizeSamples]; + FuzzAudioDecoder(data, size, &dec, sample_rate_hz, sizeof(output), output); +} +} // namespace webrtc diff --git a/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc b/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc new file mode 100644 index 0000000000..83fb8c2d62 --- /dev/null +++ b/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc @@ -0,0 +1,22 @@ +/* + * Copyright (c) 2015 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 "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h" +#include "webrtc/test/fuzzers/audio_decoder_fuzzer.h" + +namespace webrtc { +void FuzzOneInput(const uint8_t* data, size_t size) { + AudioDecoderIsacFix dec(nullptr); + static const int kSampleRateHz = 16000; + static const size_t kAllocatedOuputSizeSamples = 16000 / 10; // 100 ms. + int16_t output[kAllocatedOuputSizeSamples]; + FuzzAudioDecoder(data, size, &dec, kSampleRateHz, sizeof(output), output); +} +} // namespace webrtc diff --git a/webrtc/test/fuzzers/audio_decoder_opus_fuzzer.cc b/webrtc/test/fuzzers/audio_decoder_opus_fuzzer.cc new file mode 100644 index 0000000000..3d70ec507d --- /dev/null +++ b/webrtc/test/fuzzers/audio_decoder_opus_fuzzer.cc @@ -0,0 +1,23 @@ +/* + * Copyright (c) 2015 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 "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h" +#include "webrtc/test/fuzzers/audio_decoder_fuzzer.h" + +namespace webrtc { +void FuzzOneInput(const uint8_t* data, size_t size) { + const size_t channels = (size % 2) + 1; // 1 or 2 channels. + AudioDecoderOpus dec(channels); + const int kSampleRateHz = 48000; + const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10; // 100 ms. + int16_t output[kAllocatedOuputSizeSamples]; + FuzzAudioDecoder(data, size, &dec, kSampleRateHz, sizeof(output), output); +} +} // namespace webrtc diff --git a/webrtc/test/fuzzers/h264_depacketizer_fuzzer.cc b/webrtc/test/fuzzers/h264_depacketizer_fuzzer.cc new file mode 100644 index 0000000000..ca73d9495f --- /dev/null +++ b/webrtc/test/fuzzers/h264_depacketizer_fuzzer.cc @@ -0,0 +1,18 @@ +/* + * Copyright (c) 2015 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 "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h" + +namespace webrtc { +void FuzzOneInput(const uint8_t* data, size_t size) { + RtpDepacketizerH264 depacketizer; + RtpDepacketizer::ParsedPayload parsed_payload; + depacketizer.Parse(&parsed_payload, data, size); +} +} // namespace webrtc diff --git a/webrtc/test/fuzzers/producer_fec_fuzzer.cc b/webrtc/test/fuzzers/producer_fec_fuzzer.cc new file mode 100644 index 0000000000..7322fed4bf --- /dev/null +++ b/webrtc/test/fuzzers/producer_fec_fuzzer.cc @@ -0,0 +1,60 @@ +/* + * Copyright (c) 2015 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 "webrtc/base/checks.h" +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/modules/rtp_rtcp/source/byte_io.h" +#include "webrtc/modules/rtp_rtcp/source/producer_fec.h" + +namespace webrtc { + +void FuzzOneInput(const uint8_t* data, size_t size) { + ForwardErrorCorrection fec; + ProducerFec producer(&fec); + size_t i = 0; + if (size < 4) + return; + FecProtectionParams params = {data[i++] % 128, data[i++] % 1, + static_cast<int>(data[i++] % 10), + kFecMaskBursty}; + producer.SetFecParameters(¶ms, 0); + uint16_t seq_num = data[i++]; + + while (i + 3 < size) { + size_t rtp_header_length = data[i++] % 10 + 12; + size_t payload_size = data[i++] % 10; + if (i + payload_size + rtp_header_length + 2 > size) + break; + rtc::scoped_ptr<uint8_t[]> packet( + new uint8_t[payload_size + rtp_header_length]); + memcpy(packet.get(), &data[i], payload_size + rtp_header_length); + ByteWriter<uint16_t>::WriteBigEndian(&packet[2], seq_num++); + i += payload_size + rtp_header_length; + // Make sure sequence numbers are increasing. + const int kRedPayloadType = 98; + rtc::scoped_ptr<RedPacket> red_packet(producer.BuildRedPacket( + packet.get(), payload_size, rtp_header_length, kRedPayloadType)); + bool protect = static_cast<bool>(data[i++] % 2); + if (protect) { + producer.AddRtpPacketAndGenerateFec(packet.get(), payload_size, + rtp_header_length); + } + uint16_t num_fec_packets = producer.NumAvailableFecPackets(); + std::vector<RedPacket*> fec_packets; + if (num_fec_packets > 0) { + fec_packets = + producer.GetFecPackets(kRedPayloadType, 99, 100, rtp_header_length); + RTC_CHECK_EQ(num_fec_packets, fec_packets.size()); + } + for (RedPacket* fec_packet : fec_packets) { + delete fec_packet; + } + } +} +} // namespace webrtc diff --git a/webrtc/test/fuzzers/vp8_depacketizer_fuzzer.cc b/webrtc/test/fuzzers/vp8_depacketizer_fuzzer.cc new file mode 100644 index 0000000000..d048372456 --- /dev/null +++ b/webrtc/test/fuzzers/vp8_depacketizer_fuzzer.cc @@ -0,0 +1,18 @@ +/* + * Copyright (c) 2015 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 "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h" + +namespace webrtc { +void FuzzOneInput(const uint8_t* data, size_t size) { + RtpDepacketizerVp8 depacketizer; + RtpDepacketizer::ParsedPayload parsed_payload; + depacketizer.Parse(&parsed_payload, data, size); +} +} // namespace webrtc diff --git a/webrtc/test/fuzzers/vp8_qp_parser_fuzzer.cc b/webrtc/test/fuzzers/vp8_qp_parser_fuzzer.cc new file mode 100644 index 0000000000..5135f1a471 --- /dev/null +++ b/webrtc/test/fuzzers/vp8_qp_parser_fuzzer.cc @@ -0,0 +1,17 @@ +/* + * Copyright (c) 2015 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 "webrtc/modules/video_coding/utility/vp8_header_parser.h" + +namespace webrtc { +void FuzzOneInput(const uint8_t* data, size_t size) { + int qp; + vp8::GetQp(data, size, &qp); +} +} // namespace webrtc diff --git a/webrtc/test/fuzzers/vp9_depacketizer_fuzzer.cc b/webrtc/test/fuzzers/vp9_depacketizer_fuzzer.cc new file mode 100644 index 0000000000..02a7cc0f81 --- /dev/null +++ b/webrtc/test/fuzzers/vp9_depacketizer_fuzzer.cc @@ -0,0 +1,18 @@ +/* + * Copyright (c) 2015 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 "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h" + +namespace webrtc { +void FuzzOneInput(const uint8_t* data, size_t size) { + RtpDepacketizerVp9 depacketizer; + RtpDepacketizer::ParsedPayload parsed_payload; + depacketizer.Parse(&parsed_payload, data, size); +} +} // namespace webrtc diff --git a/webrtc/test/fuzzers/webrtc_fuzzer.gni b/webrtc/test/fuzzers/webrtc_fuzzer.gni new file mode 100644 index 0000000000..d264392c07 --- /dev/null +++ b/webrtc/test/fuzzers/webrtc_fuzzer.gni @@ -0,0 +1,28 @@ +# Copyright (c) 2015 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. + +import("//testing/test.gni") + +template("webrtc_fuzzer_test") { + assert(defined(invoker.sources), "Need sources in $target_name.") + + test(target_name) { + forward_variables_from(invoker, [ "sources" ]) + deps = [ + ":webrtc_fuzzer_main", + ] + if (defined(invoker.deps)) { + deps += invoker.deps + } + if (is_clang) { + # Suppress warnings from Chrome's Clang plugins. + # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. + configs -= [ "//build/config/clang:find_bad_constructs" ] + } + } +} diff --git a/webrtc/test/fuzzers/webrtc_fuzzer_main.cc b/webrtc/test/fuzzers/webrtc_fuzzer_main.cc new file mode 100644 index 0000000000..50a513c094 --- /dev/null +++ b/webrtc/test/fuzzers/webrtc_fuzzer_main.cc @@ -0,0 +1,41 @@ +/* + * Copyright (c) 2015 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. + */ + +// This file is intended to provide a common interface for fuzzing functions. +// It's intended to set sane defaults, such as removing logging for further +// fuzzing efficiency. + +#include "webrtc/base/logging.h" + +namespace { +bool g_initialized = false; +void InitializeWebRtcFuzzDefaults() { + if (g_initialized) + return; + + // Remove default logging to prevent huge slowdowns. + // TODO(pbos): Disable in Chromium: http://crbug.com/561667 +#if !defined(WEBRTC_CHROMIUM_BUILD) + rtc::LogMessage::LogToDebug(rtc::LS_NONE); +#endif // !defined(WEBRTC_CHROMIUM_BUILD) + + g_initialized = true; +} +} + +namespace webrtc { +extern void FuzzOneInput(const uint8_t* data, size_t size); +} // namespace webrtc + +extern "C" int LLVMFuzzerTestOneInput(const unsigned char *data, size_t size) { + InitializeWebRtcFuzzDefaults(); + webrtc::FuzzOneInput(data, size); + return 0; +} diff --git a/webrtc/test/gl/gl_renderer.h b/webrtc/test/gl/gl_renderer.h index 27749468a2..7682d3c918 100644 --- a/webrtc/test/gl/gl_renderer.h +++ b/webrtc/test/gl/gl_renderer.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_GL_GL_RENDERER_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_GL_GL_RENDERER_H_ +#ifndef WEBRTC_TEST_GL_GL_RENDERER_H_ +#define WEBRTC_TEST_GL_GL_RENDERER_H_ #ifdef WEBRTC_MAC #include <OpenGL/gl.h> @@ -48,4 +48,4 @@ class GlRenderer : public VideoRenderer { } // test } // webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_GL_GL_RENDERER_H_ +#endif // WEBRTC_TEST_GL_GL_RENDERER_H_ diff --git a/webrtc/test/histogram.cc b/webrtc/test/histogram.cc index 6fcdb6864f..2893e4389a 100644 --- a/webrtc/test/histogram.cc +++ b/webrtc/test/histogram.cc @@ -12,6 +12,7 @@ #include <map> +#include "webrtc/base/checks.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/thread_annotations.h" #include "webrtc/system_wrappers/include/metrics.h" @@ -22,10 +23,10 @@ namespace webrtc { namespace { struct SampleInfo { - SampleInfo(int sample) - : last(sample), total(1) {} - int last; // Last added sample. - int total; // Total number of added samples. + SampleInfo(const std::string& name) : name_(name), last_(-1), total_(0) {} + const std::string name_; + int last_; // Last added sample. + int total_; // Total number of added samples. }; rtc::CriticalSection histogram_crit_; @@ -36,21 +37,33 @@ std::map<std::string, SampleInfo> histograms_ GUARDED_BY(histogram_crit_); namespace metrics { Histogram* HistogramFactoryGetCounts(const std::string& name, int min, int max, - int bucket_count) { return NULL; } + int bucket_count) { + rtc::CritScope cs(&histogram_crit_); + if (histograms_.find(name) == histograms_.end()) { + histograms_.insert(std::make_pair(name, SampleInfo(name))); + } + auto it = histograms_.find(name); + return reinterpret_cast<Histogram*>(&it->second); +} Histogram* HistogramFactoryGetEnumeration(const std::string& name, - int boundary) { return NULL; } + int boundary) { + rtc::CritScope cs(&histogram_crit_); + if (histograms_.find(name) == histograms_.end()) { + histograms_.insert(std::make_pair(name, SampleInfo(name))); + } + auto it = histograms_.find(name); + return reinterpret_cast<Histogram*>(&it->second); +} void HistogramAdd( Histogram* histogram_pointer, const std::string& name, int sample) { rtc::CritScope cs(&histogram_crit_); - auto it = histograms_.find(name); - if (it == histograms_.end()) { - histograms_.insert(std::make_pair(name, SampleInfo(sample))); - return; - } - it->second.last = sample; - ++it->second.total; + SampleInfo* ptr = reinterpret_cast<SampleInfo*>(histogram_pointer); + // The name should not vary. + RTC_CHECK(ptr->name_ == name); + ptr->last_ = sample; + ++ptr->total_; } } // namespace metrics @@ -61,7 +74,7 @@ int LastHistogramSample(const std::string& name) { if (it == histograms_.end()) { return -1; } - return it->second.last; + return it->second.last_; } int NumHistogramSamples(const std::string& name) { @@ -70,13 +83,15 @@ int NumHistogramSamples(const std::string& name) { if (it == histograms_.end()) { return 0; } - return it->second.total; + return it->second.total_; } void ClearHistograms() { rtc::CritScope cs(&histogram_crit_); - histograms_.clear(); + for (auto& it : histograms_) { + it.second.last_ = -1; + it.second.total_ = 0; + } } } // namespace test } // namespace webrtc - diff --git a/webrtc/test/histogram.h b/webrtc/test/histogram.h index 44ce32b4f4..3c8e743aa1 100644 --- a/webrtc/test/histogram.h +++ b/webrtc/test/histogram.h @@ -23,7 +23,7 @@ int LastHistogramSample(const std::string& name); // Returns the number of added samples to a histogram. int NumHistogramSamples(const std::string& name); -// Removes all histograms. +// Removes all histogram samples. void ClearHistograms(); } // namespace test diff --git a/webrtc/test/layer_filtering_transport.cc b/webrtc/test/layer_filtering_transport.cc index a4ebf47f93..41d63ad6e7 100644 --- a/webrtc/test/layer_filtering_transport.cc +++ b/webrtc/test/layer_filtering_transport.cc @@ -9,8 +9,7 @@ */ #include "webrtc/base/checks.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" @@ -24,33 +23,35 @@ LayerFilteringTransport::LayerFilteringTransport( Call* send_call, uint8_t vp8_video_payload_type, uint8_t vp9_video_payload_type, - uint8_t tl_discard_threshold, - uint8_t sl_discard_threshold) + int selected_tl, + int selected_sl) : test::DirectTransport(config, send_call), vp8_video_payload_type_(vp8_video_payload_type), vp9_video_payload_type_(vp9_video_payload_type), - tl_discard_threshold_(tl_discard_threshold), - sl_discard_threshold_(sl_discard_threshold) {} + selected_tl_(selected_tl), + selected_sl_(selected_sl), + discarded_last_packet_(false) {} -uint16_t LayerFilteringTransport::NextSequenceNumber(uint32_t ssrc) { - auto it = current_seq_nums_.find(ssrc); - if (it == current_seq_nums_.end()) - return current_seq_nums_[ssrc] = 10000; - return ++it->second; +bool LayerFilteringTransport::DiscardedLastPacket() const { + return discarded_last_packet_; } bool LayerFilteringTransport::SendRtp(const uint8_t* packet, size_t length, const PacketOptions& options) { - if (tl_discard_threshold_ == 0 && sl_discard_threshold_ == 0) { + if (selected_tl_ == -1 && selected_sl_ == -1) { // Nothing to change, forward the packet immediately. return test::DirectTransport::SendRtp(packet, length, options); } bool set_marker_bit = false; - rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); + RtpUtility::RtpHeaderParser parser(packet, length); RTPHeader header; - parser->Parse(packet, length, &header); + parser.Parse(&header); + + RTC_DCHECK_LE(length, static_cast<size_t>(IP_PACKET_SIZE)); + uint8_t temp_buffer[IP_PACKET_SIZE]; + memcpy(temp_buffer, packet, length); if (header.payloadType == vp8_video_payload_type_ || header.payloadType == vp9_video_payload_type_) { @@ -65,40 +66,38 @@ bool LayerFilteringTransport::SendRtp(const uint8_t* packet, RtpDepacketizer::Create(is_vp8 ? kRtpVideoVp8 : kRtpVideoVp9)); RtpDepacketizer::ParsedPayload parsed_payload; if (depacketizer->Parse(&parsed_payload, payload, payload_data_length)) { - const uint8_t temporalIdx = + const int temporal_idx = static_cast<int>( is_vp8 ? parsed_payload.type.Video.codecHeader.VP8.temporalIdx - : parsed_payload.type.Video.codecHeader.VP9.temporal_idx; - const uint8_t spatialIdx = + : parsed_payload.type.Video.codecHeader.VP9.temporal_idx); + const int spatial_idx = static_cast<int>( is_vp8 ? kNoSpatialIdx - : parsed_payload.type.Video.codecHeader.VP9.spatial_idx; - if (sl_discard_threshold_ > 0 && - spatialIdx == sl_discard_threshold_ - 1 && + : parsed_payload.type.Video.codecHeader.VP9.spatial_idx); + if (selected_sl_ >= 0 && spatial_idx == selected_sl_ && parsed_payload.type.Video.codecHeader.VP9.end_of_frame) { // This layer is now the last in the superframe. set_marker_bit = true; - } - if ((tl_discard_threshold_ > 0 && temporalIdx != kNoTemporalIdx && - temporalIdx >= tl_discard_threshold_) || - (sl_discard_threshold_ > 0 && spatialIdx != kNoSpatialIdx && - spatialIdx >= sl_discard_threshold_)) { - return true; // Discard the packet. + } else if ((selected_tl_ >= 0 && temporal_idx != kNoTemporalIdx && + temporal_idx > selected_tl_) || + (selected_sl_ >= 0 && spatial_idx != kNoSpatialIdx && + spatial_idx > selected_sl_)) { + // Truncate packet to a padding packet. + length = header.headerLength + 1; + temp_buffer[0] |= (1 << 5); // P = 1. + temp_buffer[1] &= 0x7F; // M = 0. + discarded_last_packet_ = true; + temp_buffer[header.headerLength] = 1; // One byte of padding. } } else { RTC_NOTREACHED() << "Parse error"; } } - uint8_t temp_buffer[IP_PACKET_SIZE]; - memcpy(temp_buffer, packet, length); - // We are discarding some of the packets (specifically, whole layers), so // make sure the marker bit is set properly, and that sequence numbers are // continuous. if (set_marker_bit) temp_buffer[1] |= kRtpMarkerBitMask; - uint16_t seq_num = NextSequenceNumber(header.ssrc); - ByteWriter<uint16_t>::WriteBigEndian(&temp_buffer[2], seq_num); return test::DirectTransport::SendRtp(temp_buffer, length, options); } diff --git a/webrtc/test/layer_filtering_transport.h b/webrtc/test/layer_filtering_transport.h index 3f2389a51b..d453556235 100644 --- a/webrtc/test/layer_filtering_transport.h +++ b/webrtc/test/layer_filtering_transport.h @@ -26,23 +26,22 @@ class LayerFilteringTransport : public test::DirectTransport { Call* send_call, uint8_t vp8_video_payload_type, uint8_t vp9_video_payload_type, - uint8_t tl_discard_threshold, - uint8_t sl_discard_threshold); + int selected_tl, + int selected_sl); + bool DiscardedLastPacket() const; bool SendRtp(const uint8_t* data, size_t length, const PacketOptions& options) override; private: - uint16_t NextSequenceNumber(uint32_t ssrc); // Used to distinguish between VP8 and VP9. const uint8_t vp8_video_payload_type_; const uint8_t vp9_video_payload_type_; - // Discard all temporal/spatial layers with id greater or equal the - // threshold. 0 to disable. - const uint8_t tl_discard_threshold_; - const uint8_t sl_discard_threshold_; - // Current sequence number for each SSRC separately. - std::map<uint32_t, uint16_t> current_seq_nums_; + // Discard or invalidate all temporal/spatial layers with id greater than the + // selected one. -1 to disable filtering. + const int selected_tl_; + const int selected_sl_; + bool discarded_last_packet_; }; } // namespace test diff --git a/webrtc/test/linux/glx_renderer.h b/webrtc/test/linux/glx_renderer.h index 517f22a038..c117281cf1 100644 --- a/webrtc/test/linux/glx_renderer.h +++ b/webrtc/test/linux/glx_renderer.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_LINUX_GLX_RENDERER_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_LINUX_GLX_RENDERER_H_ +#ifndef WEBRTC_TEST_LINUX_GLX_RENDERER_H_ +#define WEBRTC_TEST_LINUX_GLX_RENDERER_H_ #include <GL/glx.h> #include <X11/Xlib.h> @@ -45,4 +45,4 @@ class GlxRenderer : public GlRenderer { } // test } // webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_LINUX_GLX_RENDERER_H_ +#endif // WEBRTC_TEST_LINUX_GLX_RENDERER_H_ diff --git a/webrtc/test/mac/video_renderer_mac.h b/webrtc/test/mac/video_renderer_mac.h index 2e55538954..7baf794744 100644 --- a/webrtc/test/mac/video_renderer_mac.h +++ b/webrtc/test/mac/video_renderer_mac.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_MAC_VIDEO_RENDERER_MAC_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_MAC_VIDEO_RENDERER_MAC_H_ +#ifndef WEBRTC_TEST_MAC_VIDEO_RENDERER_MAC_H_ +#define WEBRTC_TEST_MAC_VIDEO_RENDERER_MAC_H_ #include "webrtc/base/constructormagic.h" #include "webrtc/test/gl/gl_renderer.h" @@ -38,4 +38,4 @@ class MacRenderer : public GlRenderer { } // test } // webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_MAC_VIDEO_RENDERER_MAC_H_ +#endif // WEBRTC_TEST_MAC_VIDEO_RENDERER_MAC_H_ diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h new file mode 100644 index 0000000000..b5d79c18ea --- /dev/null +++ b/webrtc/test/mock_voe_channel_proxy.h @@ -0,0 +1,48 @@ +/* + * Copyright (c) 2015 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. + */ + +#ifndef WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_ +#define WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_ + +#include <string> +#include "testing/gmock/include/gmock/gmock.h" +#include "webrtc/voice_engine/channel_proxy.h" + +namespace webrtc { +namespace test { + +class MockVoEChannelProxy : public voe::ChannelProxy { + public: + MOCK_METHOD1(SetRTCPStatus, void(bool enable)); + MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc)); + MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name)); + MOCK_METHOD2(SetSendAbsoluteSenderTimeStatus, void(bool enable, int id)); + MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id)); + MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id)); + MOCK_METHOD2(SetReceiveAbsoluteSenderTimeStatus, void(bool enable, int id)); + MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id)); + MOCK_METHOD3(SetCongestionControlObjects, + void(RtpPacketSender* rtp_packet_sender, + TransportFeedbackObserver* transport_feedback_observer, + PacketRouter* seq_num_allocator)); + MOCK_CONST_METHOD0(GetRTCPStatistics, CallStatistics()); + MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>()); + MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics()); + MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats()); + MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int32_t()); + MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t()); + MOCK_METHOD1(SetSendTelephoneEventPayloadType, bool(int payload_type)); + MOCK_METHOD2(SendTelephoneEventOutband, bool(uint8_t event, + uint32_t duration_ms)); +}; +} // namespace test +} // namespace webrtc + +#endif // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_ diff --git a/webrtc/test/mock_voice_engine.h b/webrtc/test/mock_voice_engine.h new file mode 100644 index 0000000000..28a75f8063 --- /dev/null +++ b/webrtc/test/mock_voice_engine.h @@ -0,0 +1,337 @@ +/* + * Copyright (c) 2015 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. + */ + +#ifndef WEBRTC_AUDIO_MOCK_VOICE_ENGINE_H_ +#define WEBRTC_AUDIO_MOCK_VOICE_ENGINE_H_ + +#include "testing/gmock/include/gmock/gmock.h" +#include "webrtc/test/mock_voe_channel_proxy.h" +#include "webrtc/voice_engine/voice_engine_impl.h" + +namespace webrtc { +namespace test { + +// NOTE: This class inherits from VoiceEngineImpl so that its clients will be +// able to get the various interfaces as usual, via T::GetInterface(). +class MockVoiceEngine : public VoiceEngineImpl { + public: + MockVoiceEngine() : VoiceEngineImpl(new Config(), true) { + // Increase ref count so this object isn't automatically deleted whenever + // interfaces are Release():d. + ++_ref_count; + // We add this default behavior to make the mock easier to use in tests. It + // will create a NiceMock of a voe::ChannelProxy. + ON_CALL(*this, ChannelProxyFactory(testing::_)) + .WillByDefault( + testing::Invoke([](int channel_id) { + return new testing::NiceMock<MockVoEChannelProxy>(); + })); + } + ~MockVoiceEngine() override { + // Decrease ref count before base class d-tor is called; otherwise it will + // trigger an assertion. + --_ref_count; + } + // Allows injecting a ChannelProxy factory. + MOCK_METHOD1(ChannelProxyFactory, voe::ChannelProxy*(int channel_id)); + + // VoiceEngineImpl + rtc::scoped_ptr<voe::ChannelProxy> GetChannelProxy(int channel_id) override { + return rtc::scoped_ptr<voe::ChannelProxy>(ChannelProxyFactory(channel_id)); + } + + // VoEAudioProcessing + MOCK_METHOD2(SetNsStatus, int(bool enable, NsModes mode)); + MOCK_METHOD2(GetNsStatus, int(bool& enabled, NsModes& mode)); + MOCK_METHOD2(SetAgcStatus, int(bool enable, AgcModes mode)); + MOCK_METHOD2(GetAgcStatus, int(bool& enabled, AgcModes& mode)); + MOCK_METHOD1(SetAgcConfig, int(AgcConfig config)); + MOCK_METHOD1(GetAgcConfig, int(AgcConfig& config)); + MOCK_METHOD2(SetEcStatus, int(bool enable, EcModes mode)); + MOCK_METHOD2(GetEcStatus, int(bool& enabled, EcModes& mode)); + MOCK_METHOD1(EnableDriftCompensation, int(bool enable)); + MOCK_METHOD0(DriftCompensationEnabled, bool()); + MOCK_METHOD1(SetDelayOffsetMs, void(int offset)); + MOCK_METHOD0(DelayOffsetMs, int()); + MOCK_METHOD2(SetAecmMode, int(AecmModes mode, bool enableCNG)); + MOCK_METHOD2(GetAecmMode, int(AecmModes& mode, bool& enabledCNG)); + MOCK_METHOD1(EnableHighPassFilter, int(bool enable)); + MOCK_METHOD0(IsHighPassFilterEnabled, bool()); + MOCK_METHOD3(SetRxNsStatus, int(int channel, bool enable, NsModes mode)); + MOCK_METHOD3(GetRxNsStatus, int(int channel, bool& enabled, NsModes& mode)); + MOCK_METHOD3(SetRxAgcStatus, int(int channel, bool enable, AgcModes mode)); + MOCK_METHOD3(GetRxAgcStatus, int(int channel, bool& enabled, AgcModes& mode)); + MOCK_METHOD2(SetRxAgcConfig, int(int channel, AgcConfig config)); + MOCK_METHOD2(GetRxAgcConfig, int(int channel, AgcConfig& config)); + MOCK_METHOD2(RegisterRxVadObserver, + int(int channel, VoERxVadCallback& observer)); + MOCK_METHOD1(DeRegisterRxVadObserver, int(int channel)); + MOCK_METHOD1(VoiceActivityIndicator, int(int channel)); + MOCK_METHOD1(SetEcMetricsStatus, int(bool enable)); + MOCK_METHOD1(GetEcMetricsStatus, int(bool& enabled)); + MOCK_METHOD4(GetEchoMetrics, int(int& ERL, int& ERLE, int& RERL, int& A_NLP)); + MOCK_METHOD3(GetEcDelayMetrics, + int(int& delay_median, + int& delay_std, + float& fraction_poor_delays)); + MOCK_METHOD1(StartDebugRecording, int(const char* fileNameUTF8)); + MOCK_METHOD1(StartDebugRecording, int(FILE* file_handle)); + MOCK_METHOD0(StopDebugRecording, int()); + MOCK_METHOD1(SetTypingDetectionStatus, int(bool enable)); + MOCK_METHOD1(GetTypingDetectionStatus, int(bool& enabled)); + MOCK_METHOD1(TimeSinceLastTyping, int(int& seconds)); + MOCK_METHOD5(SetTypingDetectionParameters, + int(int timeWindow, + int costPerTyping, + int reportingThreshold, + int penaltyDecay, + int typeEventDelay)); + MOCK_METHOD1(EnableStereoChannelSwapping, void(bool enable)); + MOCK_METHOD0(IsStereoChannelSwappingEnabled, bool()); + + // VoEBase + MOCK_METHOD1(RegisterVoiceEngineObserver, int(VoiceEngineObserver& observer)); + MOCK_METHOD0(DeRegisterVoiceEngineObserver, int()); + MOCK_METHOD2(Init, + int(AudioDeviceModule* external_adm, + AudioProcessing* audioproc)); + MOCK_METHOD0(audio_processing, AudioProcessing*()); + MOCK_METHOD0(Terminate, int()); + MOCK_METHOD0(CreateChannel, int()); + MOCK_METHOD1(CreateChannel, int(const Config& config)); + MOCK_METHOD1(DeleteChannel, int(int channel)); + MOCK_METHOD1(StartReceive, int(int channel)); + MOCK_METHOD1(StopReceive, int(int channel)); + MOCK_METHOD1(StartPlayout, int(int channel)); + MOCK_METHOD1(StopPlayout, int(int channel)); + MOCK_METHOD1(StartSend, int(int channel)); + MOCK_METHOD1(StopSend, int(int channel)); + MOCK_METHOD1(GetVersion, int(char version[1024])); + MOCK_METHOD0(LastError, int()); + MOCK_METHOD0(audio_transport, AudioTransport*()); + MOCK_METHOD2(AssociateSendChannel, + int(int channel, int accociate_send_channel)); + + // VoECodec + MOCK_METHOD0(NumOfCodecs, int()); + MOCK_METHOD2(GetCodec, int(int index, CodecInst& codec)); + MOCK_METHOD2(SetSendCodec, int(int channel, const CodecInst& codec)); + MOCK_METHOD2(GetSendCodec, int(int channel, CodecInst& codec)); + MOCK_METHOD2(SetBitRate, int(int channel, int bitrate_bps)); + MOCK_METHOD2(GetRecCodec, int(int channel, CodecInst& codec)); + MOCK_METHOD2(SetRecPayloadType, int(int channel, const CodecInst& codec)); + MOCK_METHOD2(GetRecPayloadType, int(int channel, CodecInst& codec)); + MOCK_METHOD3(SetSendCNPayloadType, + int(int channel, int type, PayloadFrequencies frequency)); + MOCK_METHOD2(SetFECStatus, int(int channel, bool enable)); + MOCK_METHOD2(GetFECStatus, int(int channel, bool& enabled)); + MOCK_METHOD4(SetVADStatus, + int(int channel, bool enable, VadModes mode, bool disableDTX)); + MOCK_METHOD4( + GetVADStatus, + int(int channel, bool& enabled, VadModes& mode, bool& disabledDTX)); + MOCK_METHOD2(SetOpusMaxPlaybackRate, int(int channel, int frequency_hz)); + MOCK_METHOD2(SetOpusDtx, int(int channel, bool enable_dtx)); + MOCK_METHOD0(GetEventLog, RtcEventLog*()); + + // VoEDtmf + MOCK_METHOD5(SendTelephoneEvent, + int(int channel, + int eventCode, + bool outOfBand, + int lengthMs, + int attenuationDb)); + MOCK_METHOD2(SetSendTelephoneEventPayloadType, + int(int channel, unsigned char type)); + MOCK_METHOD2(GetSendTelephoneEventPayloadType, + int(int channel, unsigned char& type)); + MOCK_METHOD2(SetDtmfFeedbackStatus, int(bool enable, bool directFeedback)); + MOCK_METHOD2(GetDtmfFeedbackStatus, int(bool& enabled, bool& directFeedback)); + MOCK_METHOD3(PlayDtmfTone, + int(int eventCode, int lengthMs, int attenuationDb)); + + // VoEExternalMedia + MOCK_METHOD3(RegisterExternalMediaProcessing, + int(int channel, + ProcessingTypes type, + VoEMediaProcess& processObject)); + MOCK_METHOD2(DeRegisterExternalMediaProcessing, + int(int channel, ProcessingTypes type)); + MOCK_METHOD3(GetAudioFrame, + int(int channel, int desired_sample_rate_hz, AudioFrame* frame)); + MOCK_METHOD2(SetExternalMixing, int(int channel, bool enable)); + + // VoEFile + MOCK_METHOD7(StartPlayingFileLocally, + int(int channel, + const char fileNameUTF8[1024], + bool loop, + FileFormats format, + float volumeScaling, + int startPointMs, + int stopPointMs)); + MOCK_METHOD6(StartPlayingFileLocally, + int(int channel, + InStream* stream, + FileFormats format, + float volumeScaling, + int startPointMs, + int stopPointMs)); + MOCK_METHOD1(StopPlayingFileLocally, int(int channel)); + MOCK_METHOD1(IsPlayingFileLocally, int(int channel)); + MOCK_METHOD6(StartPlayingFileAsMicrophone, + int(int channel, + const char fileNameUTF8[1024], + bool loop, + bool mixWithMicrophone, + FileFormats format, + float volumeScaling)); + MOCK_METHOD5(StartPlayingFileAsMicrophone, + int(int channel, + InStream* stream, + bool mixWithMicrophone, + FileFormats format, + float volumeScaling)); + MOCK_METHOD1(StopPlayingFileAsMicrophone, int(int channel)); + MOCK_METHOD1(IsPlayingFileAsMicrophone, int(int channel)); + MOCK_METHOD4(StartRecordingPlayout, + int(int channel, + const char* fileNameUTF8, + CodecInst* compression, + int maxSizeBytes)); + MOCK_METHOD1(StopRecordingPlayout, int(int channel)); + MOCK_METHOD3(StartRecordingPlayout, + int(int channel, OutStream* stream, CodecInst* compression)); + MOCK_METHOD3(StartRecordingMicrophone, + int(const char* fileNameUTF8, + CodecInst* compression, + int maxSizeBytes)); + MOCK_METHOD2(StartRecordingMicrophone, + int(OutStream* stream, CodecInst* compression)); + MOCK_METHOD0(StopRecordingMicrophone, int()); + + // VoEHardware + MOCK_METHOD1(GetNumOfRecordingDevices, int(int& devices)); + MOCK_METHOD1(GetNumOfPlayoutDevices, int(int& devices)); + MOCK_METHOD3(GetRecordingDeviceName, + int(int index, char strNameUTF8[128], char strGuidUTF8[128])); + MOCK_METHOD3(GetPlayoutDeviceName, + int(int index, char strNameUTF8[128], char strGuidUTF8[128])); + MOCK_METHOD2(SetRecordingDevice, + int(int index, StereoChannel recordingChannel)); + MOCK_METHOD1(SetPlayoutDevice, int(int index)); + MOCK_METHOD1(SetAudioDeviceLayer, int(AudioLayers audioLayer)); + MOCK_METHOD1(GetAudioDeviceLayer, int(AudioLayers& audioLayer)); + MOCK_METHOD1(SetRecordingSampleRate, int(unsigned int samples_per_sec)); + MOCK_CONST_METHOD1(RecordingSampleRate, int(unsigned int* samples_per_sec)); + MOCK_METHOD1(SetPlayoutSampleRate, int(unsigned int samples_per_sec)); + MOCK_CONST_METHOD1(PlayoutSampleRate, int(unsigned int* samples_per_sec)); + MOCK_CONST_METHOD0(BuiltInAECIsAvailable, bool()); + MOCK_METHOD1(EnableBuiltInAEC, int(bool enable)); + MOCK_CONST_METHOD0(BuiltInAGCIsAvailable, bool()); + MOCK_METHOD1(EnableBuiltInAGC, int(bool enable)); + MOCK_CONST_METHOD0(BuiltInNSIsAvailable, bool()); + MOCK_METHOD1(EnableBuiltInNS, int(bool enable)); + + // VoENetEqStats + MOCK_METHOD2(GetNetworkStatistics, + int(int channel, NetworkStatistics& stats)); + MOCK_CONST_METHOD2(GetDecodingCallStatistics, + int(int channel, AudioDecodingCallStats* stats)); + + // VoENetwork + MOCK_METHOD2(RegisterExternalTransport, + int(int channel, Transport& transport)); + MOCK_METHOD1(DeRegisterExternalTransport, int(int channel)); + MOCK_METHOD3(ReceivedRTPPacket, + int(int channel, const void* data, size_t length)); + MOCK_METHOD4(ReceivedRTPPacket, + int(int channel, + const void* data, + size_t length, + const PacketTime& packet_time)); + MOCK_METHOD3(ReceivedRTCPPacket, + int(int channel, const void* data, size_t length)); + + // VoERTP_RTCP + MOCK_METHOD2(SetLocalSSRC, int(int channel, unsigned int ssrc)); + MOCK_METHOD2(GetLocalSSRC, int(int channel, unsigned int& ssrc)); + MOCK_METHOD2(GetRemoteSSRC, int(int channel, unsigned int& ssrc)); + MOCK_METHOD3(SetSendAudioLevelIndicationStatus, + int(int channel, bool enable, unsigned char id)); + MOCK_METHOD3(SetReceiveAudioLevelIndicationStatus, + int(int channel, bool enable, unsigned char id)); + MOCK_METHOD3(SetSendAbsoluteSenderTimeStatus, + int(int channel, bool enable, unsigned char id)); + MOCK_METHOD3(SetReceiveAbsoluteSenderTimeStatus, + int(int channel, bool enable, unsigned char id)); + MOCK_METHOD2(SetRTCPStatus, int(int channel, bool enable)); + MOCK_METHOD2(GetRTCPStatus, int(int channel, bool& enabled)); + MOCK_METHOD2(SetRTCP_CNAME, int(int channel, const char cName[256])); + MOCK_METHOD2(GetRTCP_CNAME, int(int channel, char cName[256])); + MOCK_METHOD2(GetRemoteRTCP_CNAME, int(int channel, char cName[256])); + MOCK_METHOD7(GetRemoteRTCPData, + int(int channel, + unsigned int& NTPHigh, + unsigned int& NTPLow, + unsigned int& timestamp, + unsigned int& playoutTimestamp, + unsigned int* jitter, + unsigned short* fractionLost)); + MOCK_METHOD4(GetRTPStatistics, + int(int channel, + unsigned int& averageJitterMs, + unsigned int& maxJitterMs, + unsigned int& discardedPackets)); + MOCK_METHOD2(GetRTCPStatistics, int(int channel, CallStatistics& stats)); + MOCK_METHOD2(GetRemoteRTCPReportBlocks, + int(int channel, std::vector<ReportBlock>* receive_blocks)); + MOCK_METHOD3(SetREDStatus, int(int channel, bool enable, int redPayloadtype)); + MOCK_METHOD3(GetREDStatus, + int(int channel, bool& enable, int& redPayloadtype)); + MOCK_METHOD3(SetNACKStatus, int(int channel, bool enable, int maxNoPackets)); + + // VoEVideoSync + MOCK_METHOD1(GetPlayoutBufferSize, int(int& buffer_ms)); + MOCK_METHOD2(SetMinimumPlayoutDelay, int(int channel, int delay_ms)); + MOCK_METHOD3(GetDelayEstimate, + int(int channel, + int* jitter_buffer_delay_ms, + int* playout_buffer_delay_ms)); + MOCK_CONST_METHOD1(GetLeastRequiredDelayMs, int(int channel)); + MOCK_METHOD2(SetInitTimestamp, int(int channel, unsigned int timestamp)); + MOCK_METHOD2(SetInitSequenceNumber, int(int channel, short sequenceNumber)); + MOCK_METHOD2(GetPlayoutTimestamp, int(int channel, unsigned int& timestamp)); + MOCK_METHOD3(GetRtpRtcp, + int(int channel, + RtpRtcp** rtpRtcpModule, + RtpReceiver** rtp_receiver)); + + // VoEVolumeControl + MOCK_METHOD1(SetSpeakerVolume, int(unsigned int volume)); + MOCK_METHOD1(GetSpeakerVolume, int(unsigned int& volume)); + MOCK_METHOD1(SetMicVolume, int(unsigned int volume)); + MOCK_METHOD1(GetMicVolume, int(unsigned int& volume)); + MOCK_METHOD2(SetInputMute, int(int channel, bool enable)); + MOCK_METHOD2(GetInputMute, int(int channel, bool& enabled)); + MOCK_METHOD1(GetSpeechInputLevel, int(unsigned int& level)); + MOCK_METHOD2(GetSpeechOutputLevel, int(int channel, unsigned int& level)); + MOCK_METHOD1(GetSpeechInputLevelFullRange, int(unsigned int& level)); + MOCK_METHOD2(GetSpeechOutputLevelFullRange, + int(int channel, unsigned& level)); + MOCK_METHOD2(SetChannelOutputVolumeScaling, int(int channel, float scaling)); + MOCK_METHOD2(GetChannelOutputVolumeScaling, int(int channel, float& scaling)); + MOCK_METHOD3(SetOutputVolumePan, int(int channel, float left, float right)); + MOCK_METHOD3(GetOutputVolumePan, int(int channel, float& left, float& right)); +}; +} // namespace test +} // namespace webrtc + +#endif // WEBRTC_AUDIO_MOCK_VOICE_ENGINE_H_ diff --git a/webrtc/test/null_transport.h b/webrtc/test/null_transport.h index f4b704634d..c49883e1dc 100644 --- a/webrtc/test/null_transport.h +++ b/webrtc/test/null_transport.h @@ -7,8 +7,8 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_NULL_TRANSPORT_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_NULL_TRANSPORT_H_ +#ifndef WEBRTC_TEST_NULL_TRANSPORT_H_ +#define WEBRTC_TEST_NULL_TRANSPORT_H_ #include "webrtc/transport.h" @@ -27,4 +27,4 @@ class NullTransport : public Transport { } // namespace test } // namespace webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_NULL_TRANSPORT_H_ +#endif // WEBRTC_TEST_NULL_TRANSPORT_H_ diff --git a/webrtc/test/random.cc b/webrtc/test/random.cc deleted file mode 100644 index c4c405f6b8..0000000000 --- a/webrtc/test/random.cc +++ /dev/null @@ -1,57 +0,0 @@ -/* - * Copyright (c) 2015 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 "webrtc/test/random.h" - -#include <math.h> - -#include "webrtc/base/checks.h" - -namespace webrtc { - -namespace test { - -Random::Random(uint32_t seed) : a_(0x531FDB97 ^ seed), b_(0x6420ECA8 + seed) { -} - -float Random::Rand() { - const double kScale = 1.0f / (static_cast<uint64_t>(1) << 32); - double result = kScale * b_; - a_ ^= b_; - b_ += a_; - return static_cast<float>(result); -} - -int Random::Rand(int low, int high) { - RTC_DCHECK(low <= high); - float uniform = Rand() * (high - low + 1) + low; - return static_cast<int>(uniform); -} - -int Random::Gaussian(int mean, int standard_deviation) { - // Creating a Normal distribution variable from two independent uniform - // variables based on the Box-Muller transform, which is defined on the - // interval (0, 1], hence the mask+add below. - const double kPi = 3.14159265358979323846; - const double kScale = 1.0 / 0x80000000ul; - double u1 = kScale * ((a_ & 0x7ffffffful) + 1); - double u2 = kScale * ((b_ & 0x7ffffffful) + 1); - a_ ^= b_; - b_ += a_; - return static_cast<int>( - mean + standard_deviation * sqrt(-2 * log(u1)) * cos(2 * kPi * u2)); -} - -int Random::Exponential(float lambda) { - float uniform = Rand(); - return static_cast<int>(-log(uniform) / lambda); -} -} // namespace test -} // namespace webrtc diff --git a/webrtc/test/random.h b/webrtc/test/random.h deleted file mode 100644 index 5cc54f2129..0000000000 --- a/webrtc/test/random.h +++ /dev/null @@ -1,49 +0,0 @@ -/* - * Copyright (c) 2015 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. - */ - -#ifndef WEBRTC_TEST_RANDOM_H_ -#define WEBRTC_TEST_RANDOM_H_ - -#include "webrtc/typedefs.h" -#include "webrtc/base/constructormagic.h" - -namespace webrtc { - -namespace test { - -class Random { - public: - explicit Random(uint32_t seed); - - // Return pseudo-random number in the interval [0.0, 1.0). - float Rand(); - - // Return pseudo-random number mapped to the interval [low, high]. - int Rand(int low, int high); - - // Normal Distribution. - int Gaussian(int mean, int standard_deviation); - - // Exponential Distribution. - int Exponential(float lambda); - - // TODO(solenberg): Random from histogram. - // template<typename T> int Distribution(const std::vector<T> histogram) { - - private: - uint32_t a_; - uint32_t b_; - - RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Random); -}; -} // namespace test -} // namespace webrtc - -#endif // WEBRTC_TEST_RANDOM_H_ diff --git a/webrtc/test/rtp_file_reader.cc b/webrtc/test/rtp_file_reader.cc index cb0e40705f..1413f00797 100644 --- a/webrtc/test/rtp_file_reader.cc +++ b/webrtc/test/rtp_file_reader.cc @@ -458,7 +458,7 @@ class PcapReader : public RtpFileReaderImpl { rtp_parser.ParseRtcp(&marker.rtp_header); packets_.push_back(marker); } else { - if (!rtp_parser.Parse(marker.rtp_header, NULL)) { + if (!rtp_parser.Parse(&marker.rtp_header, nullptr)) { DEBUG_LOG("Not recognized as RTP/RTCP"); return kResultSkip; } diff --git a/webrtc/test/rtp_file_reader_unittest.cc b/webrtc/test/rtp_file_reader_unittest.cc index 929813f999..15a456ccf6 100644 --- a/webrtc/test/rtp_file_reader_unittest.cc +++ b/webrtc/test/rtp_file_reader_unittest.cc @@ -85,7 +85,8 @@ class TestPcapFileReader : public ::testing::Test { while (rtp_packet_source_->NextPacket(&packet)) { RtpUtility::RtpHeaderParser rtp_header_parser(packet.data, packet.length); webrtc::RTPHeader header; - if (!rtp_header_parser.RTCP() && rtp_header_parser.Parse(header, NULL)) { + if (!rtp_header_parser.RTCP() && + rtp_header_parser.Parse(&header, nullptr)) { pps[header.ssrc]++; } } diff --git a/webrtc/test/rtp_rtcp_observer.h b/webrtc/test/rtp_rtcp_observer.h index 89b6dd06bd..5eb88d3f0d 100644 --- a/webrtc/test/rtp_rtcp_observer.h +++ b/webrtc/test/rtp_rtcp_observer.h @@ -7,8 +7,8 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_ +#ifndef WEBRTC_TEST_RTP_RTCP_OBSERVER_H_ +#define WEBRTC_TEST_RTP_RTCP_OBSERVER_H_ #include <map> #include <vector> @@ -16,7 +16,8 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/criticalsection.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/base/event.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/test/constants.h" #include "webrtc/test/direct_transport.h" #include "webrtc/typedefs.h" @@ -36,10 +37,7 @@ class RtpRtcpObserver { virtual ~RtpRtcpObserver() {} - virtual EventTypeWrapper Wait() { - EventTypeWrapper result = observation_complete_->Wait(timeout_ms_); - return result; - } + virtual bool Wait() { return observation_complete_.Wait(timeout_ms_); } virtual Action OnSendRtp(const uint8_t* packet, size_t length) { return SEND_PACKET; @@ -58,8 +56,8 @@ class RtpRtcpObserver { } protected: - explicit RtpRtcpObserver(unsigned int event_timeout_ms) - : observation_complete_(EventWrapper::Create()), + explicit RtpRtcpObserver(int event_timeout_ms) + : observation_complete_(false, false), parser_(RtpHeaderParser::Create()), timeout_ms_(event_timeout_ms) { parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset, @@ -70,11 +68,11 @@ class RtpRtcpObserver { kTransportSequenceNumberExtensionId); } - const rtc::scoped_ptr<EventWrapper> observation_complete_; + rtc::Event observation_complete_; const rtc::scoped_ptr<RtpHeaderParser> parser_; private: - unsigned int timeout_ms_; + const int timeout_ms_; }; class PacketTransport : public test::DirectTransport { @@ -138,4 +136,4 @@ class PacketTransport : public test::DirectTransport { } // namespace test } // namespace webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_ +#endif // WEBRTC_TEST_RTP_RTCP_OBSERVER_H_ diff --git a/webrtc/test/run_loop.h b/webrtc/test/run_loop.h index 31012525e2..238e2dc282 100644 --- a/webrtc/test/run_loop.h +++ b/webrtc/test/run_loop.h @@ -7,8 +7,8 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_RUN_LOOP_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_RUN_LOOP_H_ +#ifndef WEBRTC_TEST_RUN_LOOP_H_ +#define WEBRTC_TEST_RUN_LOOP_H_ namespace webrtc { namespace test { @@ -19,4 +19,4 @@ void PressEnterToContinue(); } // namespace test } // namespace webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RUN_LOOP_H_ +#endif // WEBRTC_TEST_RUN_LOOP_H_ diff --git a/webrtc/test/statistics.h b/webrtc/test/statistics.h index 0fc3a04ea9..d4a111e061 100644 --- a/webrtc/test/statistics.h +++ b/webrtc/test/statistics.h @@ -7,8 +7,8 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_STATISTICS_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_STATISTICS_H_ +#ifndef WEBRTC_TEST_STATISTICS_H_ +#define WEBRTC_TEST_STATISTICS_H_ #include "webrtc/typedefs.h" @@ -33,4 +33,4 @@ class Statistics { } // namespace test } // namespace webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_STATISTICS_H_ +#endif // WEBRTC_TEST_STATISTICS_H_ diff --git a/webrtc/test/test.gyp b/webrtc/test/test.gyp index 0d251dffdd..5bb7793842 100644 --- a/webrtc/test/test.gyp +++ b/webrtc/test/test.gyp @@ -22,7 +22,7 @@ ], 'sources': [ 'channel_transport/channel_transport.cc', - 'channel_transport/include/channel_transport.h', + 'channel_transport/channel_transport.h', 'channel_transport/traffic_control_win.cc', 'channel_transport/traffic_control_win.h', 'channel_transport/udp_socket_manager_posix.cc', @@ -41,11 +41,29 @@ 'channel_transport/udp_transport_impl.cc', 'channel_transport/udp_transport_impl.h', ], + 'conditions': [ + ['OS=="win" and clang==1', { + 'msvs_settings': { + 'VCCLCompilerTool': { + 'AdditionalOptions': [ + # Disable warnings failing when compiling with Clang on Windows. + # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366 + '-Wno-parentheses-equality', + '-Wno-reorder', + '-Wno-tautological-constant-out-of-range-compare', + '-Wno-unused-private-field', + ], + }, + }, + }], + ], # conditions. }, { - 'target_name': 'frame_generator', + 'target_name': 'fake_video_frames', 'type': 'static_library', 'sources': [ + 'fake_texture_frame.cc', + 'fake_texture_frame.h', 'frame_generator.cc', 'frame_generator.h', ], @@ -79,6 +97,7 @@ ], 'dependencies': [ '<(webrtc_root)/common.gyp:webrtc_common', + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default', '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', ], }, @@ -124,7 +143,6 @@ 'testsupport/frame_reader.h', 'testsupport/frame_writer.cc', 'testsupport/frame_writer.h', - 'testsupport/gtest_disable.h', 'testsupport/iosfileutils.mm', 'testsupport/mock/mock_frame_reader.h', 'testsupport/mock/mock_frame_writer.h', diff --git a/webrtc/test/test_main.cc b/webrtc/test/test_main.cc index 733831f5be..a435575f88 100644 --- a/webrtc/test/test_main.cc +++ b/webrtc/test/test_main.cc @@ -10,6 +10,7 @@ #include "gflags/gflags.h" #include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/logging.h" #include "webrtc/test/field_trial.h" #include "webrtc/test/testsupport/fileutils.h" @@ -21,6 +22,11 @@ DEFINE_string(force_fieldtrials, "", int main(int argc, char* argv[]) { ::testing::InitGoogleTest(&argc, argv); + // Default to LS_INFO, even for release builds to provide better test logging. + // TODO(pbos): Consider adding a command-line override. + if (rtc::LogMessage::GetLogToDebug() > rtc::LS_INFO) + rtc::LogMessage::LogToDebug(rtc::LS_INFO); + // AllowCommandLineParsing allows us to ignore flags passed on to us by // Chromium build bots without having to explicitly disable them. google::AllowCommandLineReparsing(); diff --git a/webrtc/test/testsupport/fileutils_unittest.cc b/webrtc/test/testsupport/fileutils_unittest.cc index dff7f2249b..e205db3ecf 100644 --- a/webrtc/test/testsupport/fileutils_unittest.cc +++ b/webrtc/test/testsupport/fileutils_unittest.cc @@ -16,7 +16,6 @@ #include <string> #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/test/testsupport/gtest_disable.h" #ifdef WIN32 #define chdir _chdir @@ -66,7 +65,14 @@ TEST_F(FileUtilsTest, ProjectRootPath) { } // Similar to the above test, but for the output dir -TEST_F(FileUtilsTest, DISABLED_ON_ANDROID(OutputPathFromUnchangedWorkingDir)) { +#if defined(WEBRTC_ANDROID) +#define MAYBE_OutputPathFromUnchangedWorkingDir \ + DISABLED_OutputPathFromUnchangedWorkingDir +#else +#define MAYBE_OutputPathFromUnchangedWorkingDir \ + OutputPathFromUnchangedWorkingDir +#endif +TEST_F(FileUtilsTest, MAYBE_OutputPathFromUnchangedWorkingDir) { std::string path = webrtc::test::OutputPath(); std::string expected_end = "out"; expected_end = kPathDelimiter + expected_end + kPathDelimiter; @@ -75,7 +81,12 @@ TEST_F(FileUtilsTest, DISABLED_ON_ANDROID(OutputPathFromUnchangedWorkingDir)) { // Tests with current working directory set to a directory higher up in the // directory tree than the project root dir. -TEST_F(FileUtilsTest, DISABLED_ON_ANDROID(OutputPathFromRootWorkingDir)) { +#if defined(WEBRTC_ANDROID) +#define MAYBE_OutputPathFromRootWorkingDir DISABLED_OutputPathFromRootWorkingDir +#else +#define MAYBE_OutputPathFromRootWorkingDir OutputPathFromRootWorkingDir +#endif +TEST_F(FileUtilsTest, MAYBE_OutputPathFromRootWorkingDir) { ASSERT_EQ(0, chdir(kPathDelimiter)); ASSERT_EQ("./", webrtc::test::OutputPath()); } diff --git a/webrtc/test/testsupport/gtest_disable.h b/webrtc/test/testsupport/gtest_disable.h deleted file mode 100644 index fdc56acc05..0000000000 --- a/webrtc/test/testsupport/gtest_disable.h +++ /dev/null @@ -1,57 +0,0 @@ -/* - * Copyright (c) 2012 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. - */ -#ifndef TEST_TESTSUPPORT_INCLUDE_GTEST_DISABLE_H_ -#define TEST_TESTSUPPORT_INCLUDE_GTEST_DISABLE_H_ - -// Helper macros for platform disables. These can be chained. Example use: -// TEST_F(ViEStandardIntegrationTest, -// DISABLED_ON_LINUX(RunsBaseTestWithoutErrors)) { // ... -// -// Or, you can disable a whole test class by wrapping all mentions of the test -// class name inside one of these macros. -// -// The platform #defines we are looking at here are set by the build system. -#ifdef WEBRTC_LINUX -#define DISABLED_ON_LINUX(test) DISABLED_##test -#else -#define DISABLED_ON_LINUX(test) test -#endif - -#ifdef WEBRTC_MAC -#define DISABLED_ON_MAC(test) DISABLED_##test -#else -#define DISABLED_ON_MAC(test) test -#endif - -#ifdef _WIN32 -#define DISABLED_ON_WIN(test) DISABLED_##test -#else -#define DISABLED_ON_WIN(test) test -#endif - -// Using some extra magic here to be able to chain Android and iOS macros. -// http://stackoverflow.com/questions/8231966/why-do-i-need-double-layer-of-indirection-for-macros -#ifdef WEBRTC_ANDROID -#define DISABLED_ON_ANDROID_HIDDEN(test) DISABLED_##test -#define DISABLED_ON_ANDROID(test) DISABLED_ON_ANDROID_HIDDEN(test) -#else -#define DISABLED_ON_ANDROID_HIDDEN(test) test -#define DISABLED_ON_ANDROID(test) DISABLED_ON_ANDROID_HIDDEN(test) -#endif - -#ifdef WEBRTC_IOS -#define DISABLED_ON_IOS_HIDDEN(test) DISABLED_##test -#define DISABLED_ON_IOS(test) DISABLED_ON_IOS_HIDDEN(test) -#else -#define DISABLED_ON_IOS_HIDDEN(test) test -#define DISABLED_ON_IOS(test) DISABLED_ON_IOS_HIDDEN(test) -#endif - -#endif // TEST_TESTSUPPORT_INCLUDE_GTEST_DISABLE_H_ diff --git a/webrtc/test/vcm_capturer.cc b/webrtc/test/vcm_capturer.cc index 1c6b91915e..0a82236c98 100644 --- a/webrtc/test/vcm_capturer.cc +++ b/webrtc/test/vcm_capturer.cc @@ -10,7 +10,7 @@ #include "webrtc/test/vcm_capturer.h" -#include "webrtc/modules/video_capture/include/video_capture_factory.h" +#include "webrtc/modules/video_capture/video_capture_factory.h" #include "webrtc/video_send_stream.h" namespace webrtc { diff --git a/webrtc/test/vcm_capturer.h b/webrtc/test/vcm_capturer.h index 53d61fc53a..6c30dd50e0 100644 --- a/webrtc/test/vcm_capturer.h +++ b/webrtc/test/vcm_capturer.h @@ -7,13 +7,13 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_VCM_CAPTURER_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_VCM_CAPTURER_H_ +#ifndef WEBRTC_TEST_VCM_CAPTURER_H_ +#define WEBRTC_TEST_VCM_CAPTURER_H_ #include "webrtc/base/criticalsection.h" #include "webrtc/common_types.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" -#include "webrtc/modules/video_capture/include/video_capture.h" +#include "webrtc/modules/video_capture/video_capture.h" #include "webrtc/test/video_capturer.h" namespace webrtc { @@ -47,4 +47,4 @@ class VcmCapturer : public VideoCapturer, public VideoCaptureDataCallback { } // test } // webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_VCM_CAPTURER_H_ +#endif // WEBRTC_TEST_VCM_CAPTURER_H_ diff --git a/webrtc/test/video_capturer.h b/webrtc/test/video_capturer.h index 3fe86f1998..169fd7151d 100644 --- a/webrtc/test/video_capturer.h +++ b/webrtc/test/video_capturer.h @@ -7,8 +7,8 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_CAPTURER_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_CAPTURER_H_ +#ifndef WEBRTC_TEST_VIDEO_CAPTURER_H_ +#define WEBRTC_TEST_VIDEO_CAPTURER_H_ #include <stddef.h> @@ -39,4 +39,4 @@ class VideoCapturer { } // test } // webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_CAPTURER_H_ +#endif // WEBRTC_TEST_VIDEO_CAPTURER_H_ diff --git a/webrtc/test/video_renderer.h b/webrtc/test/video_renderer.h index c8623270a7..3739522d7a 100644 --- a/webrtc/test/video_renderer.h +++ b/webrtc/test/video_renderer.h @@ -7,8 +7,8 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_RENDERER_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_RENDERER_H_ +#ifndef WEBRTC_TEST_VIDEO_RENDERER_H_ +#define WEBRTC_TEST_VIDEO_RENDERER_H_ #include <stddef.h> @@ -36,4 +36,4 @@ class VideoRenderer : public webrtc::VideoRenderer { } // namespace test } // namespace webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_VIDEO_RENDERER_H_ +#endif // WEBRTC_TEST_VIDEO_RENDERER_H_ diff --git a/webrtc/test/webrtc_test_common.gyp b/webrtc/test/webrtc_test_common.gyp index d075cb470b..07ea2c9b74 100644 --- a/webrtc/test/webrtc_test_common.gyp +++ b/webrtc/test/webrtc_test_common.gyp @@ -32,17 +32,15 @@ 'fake_encoder.h', 'fake_network_pipe.cc', 'fake_network_pipe.h', - 'fake_voice_engine.cc', - 'fake_voice_engine.h', 'frame_generator_capturer.cc', 'frame_generator_capturer.h', 'layer_filtering_transport.cc', 'layer_filtering_transport.h', 'mock_transport.h', + 'mock_voe_channel_proxy.h', + 'mock_voice_engine.h', 'null_transport.cc', 'null_transport.h', - 'random.cc', - 'random.h', 'rtp_rtcp_observer.h', 'run_loop.cc', 'run_loop.h', @@ -62,13 +60,14 @@ }], ], 'dependencies': [ + '<(DEPTH)/testing/gmock.gyp:gmock', '<(DEPTH)/testing/gtest.gyp:gtest', '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', '<(webrtc_root)/base/base.gyp:rtc_base', '<(webrtc_root)/common.gyp:webrtc_common', '<(webrtc_root)/modules/modules.gyp:media_file', '<(webrtc_root)/modules/modules.gyp:video_render', - '<(webrtc_root)/test/test.gyp:frame_generator', + '<(webrtc_root)/test/test.gyp:fake_video_frames', '<(webrtc_root)/test/test.gyp:test_support', '<(webrtc_root)/test/test.gyp:rtp_test_utils', '<(webrtc_root)/webrtc.gyp:webrtc', @@ -116,11 +115,24 @@ '<(directx_sdk_path)/Include', ], }], + ['OS=="win" and clang==1', { + 'msvs_settings': { + 'VCCLCompilerTool': { + 'AdditionalOptions': [ + # Disable warnings failing when compiling with Clang on Windows. + # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366 + '-Wno-bool-conversion', + '-Wno-comment', + '-Wno-delete-non-virtual-dtor', + ], + }, + }, + }], ], 'dependencies': [ '<(DEPTH)/testing/gtest.gyp:gtest', '<(webrtc_root)/modules/modules.gyp:media_file', - '<(webrtc_root)/test/test.gyp:frame_generator', + '<(webrtc_root)/test/test.gyp:fake_video_frames', '<(webrtc_root)/test/test.gyp:test_support', ], 'direct_dependent_settings': { diff --git a/webrtc/test/win/d3d_renderer.h b/webrtc/test/win/d3d_renderer.h index 46ce266460..cf2319edc4 100644 --- a/webrtc/test/win/d3d_renderer.h +++ b/webrtc/test/win/d3d_renderer.h @@ -7,8 +7,8 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_WIN_D3D_RENDERER_H_ -#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_WIN_D3D_RENDERER_H_ +#ifndef WEBRTC_TEST_WIN_D3D_RENDERER_H_ +#define WEBRTC_TEST_WIN_D3D_RENDERER_H_ #include <Windows.h> #include <d3d9.h> @@ -51,4 +51,4 @@ class D3dRenderer : public VideoRenderer { } // namespace test } // namespace webrtc -#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_WIN_D3D_RENDERER_H_ +#endif // WEBRTC_TEST_WIN_D3D_RENDERER_H_ |