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
85 lines
2.5 KiB
C++
85 lines
2.5 KiB
C++
/*
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* Copyright (c) 2012 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_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
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#include "webrtc/modules/audio_coding/main/test/Channel.h"
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#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class Config;
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class TestPack : public AudioPacketizationCallback {
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public:
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TestPack();
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~TestPack();
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void RegisterReceiverACM(AudioCodingModule* acm);
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int32_t SendData(FrameType frame_type,
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uint8_t payload_type,
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uint32_t timestamp,
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const uint8_t* payload_data,
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size_t payload_size,
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const RTPFragmentationHeader* fragmentation) override;
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size_t payload_size();
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uint32_t timestamp_diff();
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void reset_payload_size();
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private:
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AudioCodingModule* receiver_acm_;
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uint16_t sequence_number_;
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uint8_t payload_data_[60 * 32 * 2 * 2];
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uint32_t timestamp_diff_;
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uint32_t last_in_timestamp_;
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uint64_t total_bytes_;
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size_t payload_size_;
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};
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class TestAllCodecs : public ACMTest {
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public:
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explicit TestAllCodecs(int test_mode);
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~TestAllCodecs();
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void Perform() override;
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private:
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// The default value of '-1' indicates that the registration is based only on
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// codec name, and a sampling frequency matching is not required.
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// This is useful for codecs which support several sampling frequency.
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// Note! Only mono mode is tested in this test.
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void RegisterSendCodec(char side, char* codec_name, int32_t sampling_freq_hz,
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int rate, int packet_size, size_t extra_byte);
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void Run(TestPack* channel);
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void OpenOutFile(int test_number);
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void DisplaySendReceiveCodec();
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int test_mode_;
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rtc::scoped_ptr<AudioCodingModule> acm_a_;
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rtc::scoped_ptr<AudioCodingModule> acm_b_;
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TestPack* channel_a_to_b_;
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PCMFile infile_a_;
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PCMFile outfile_b_;
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int test_count_;
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int packet_size_samples_;
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size_t packet_size_bytes_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
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