14665ff7d4
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh
Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h
The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"`
which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override
Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h
Remaining uses of OVERRIDE was fixed by search+replace.
Manual edits were done to fix virtual destructors that were
overriding inherited ones.
Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc
This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.
BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41069004
Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
127 lines
3.8 KiB
C++
127 lines
3.8 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_TEST_COMMON_CALL_TEST_H_
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#define WEBRTC_TEST_COMMON_CALL_TEST_H_
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#include <vector>
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#include "webrtc/call.h"
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#include "webrtc/system_wrappers/interface/scoped_vector.h"
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#include "webrtc/test/fake_decoder.h"
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#include "webrtc/test/fake_encoder.h"
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#include "webrtc/test/frame_generator_capturer.h"
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#include "webrtc/test/rtp_rtcp_observer.h"
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namespace webrtc {
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namespace test {
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class BaseTest;
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class CallTest : public ::testing::Test {
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public:
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CallTest();
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~CallTest();
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static const size_t kNumSsrcs = 3;
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static const unsigned int kDefaultTimeoutMs;
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static const unsigned int kLongTimeoutMs;
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static const uint8_t kSendPayloadType;
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static const uint8_t kSendRtxPayloadType;
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static const uint8_t kFakeSendPayloadType;
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static const uint8_t kRedPayloadType;
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static const uint8_t kUlpfecPayloadType;
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static const uint32_t kSendRtxSsrcs[kNumSsrcs];
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static const uint32_t kSendSsrcs[kNumSsrcs];
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static const uint32_t kReceiverLocalSsrc;
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static const int kNackRtpHistoryMs;
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protected:
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void RunBaseTest(BaseTest* test);
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void CreateCalls(const Call::Config& sender_config,
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const Call::Config& receiver_config);
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void CreateSenderCall(const Call::Config& config);
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void CreateReceiverCall(const Call::Config& config);
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void CreateSendConfig(size_t num_streams);
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void CreateMatchingReceiveConfigs();
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void CreateFrameGeneratorCapturer();
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void CreateStreams();
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void Start();
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void Stop();
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void DestroyStreams();
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Clock* const clock_;
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rtc::scoped_ptr<Call> sender_call_;
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VideoSendStream::Config send_config_;
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VideoEncoderConfig encoder_config_;
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VideoSendStream* send_stream_;
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rtc::scoped_ptr<Call> receiver_call_;
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std::vector<VideoReceiveStream::Config> receive_configs_;
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std::vector<VideoReceiveStream*> receive_streams_;
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rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
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test::FakeEncoder fake_encoder_;
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ScopedVector<VideoDecoder> allocated_decoders_;
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};
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class BaseTest : public RtpRtcpObserver {
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public:
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explicit BaseTest(unsigned int timeout_ms);
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BaseTest(unsigned int timeout_ms, const FakeNetworkPipe::Config& config);
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virtual ~BaseTest();
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virtual void PerformTest() = 0;
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virtual bool ShouldCreateReceivers() const = 0;
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virtual size_t GetNumStreams() const;
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virtual Call::Config GetSenderCallConfig();
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virtual Call::Config GetReceiverCallConfig();
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virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
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virtual void ModifyConfigs(
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VideoSendStream::Config* send_config,
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std::vector<VideoReceiveStream::Config>* receive_configs,
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VideoEncoderConfig* encoder_config);
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virtual void OnStreamsCreated(
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VideoSendStream* send_stream,
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const std::vector<VideoReceiveStream*>& receive_streams);
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virtual void OnFrameGeneratorCapturerCreated(
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FrameGeneratorCapturer* frame_generator_capturer);
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};
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class SendTest : public BaseTest {
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public:
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explicit SendTest(unsigned int timeout_ms);
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SendTest(unsigned int timeout_ms, const FakeNetworkPipe::Config& config);
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bool ShouldCreateReceivers() const override;
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};
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class EndToEndTest : public BaseTest {
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public:
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explicit EndToEndTest(unsigned int timeout_ms);
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EndToEndTest(unsigned int timeout_ms, const FakeNetworkPipe::Config& config);
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bool ShouldCreateReceivers() const override;
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_TEST_COMMON_CALL_TEST_H_
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