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
120 lines
3.2 KiB
C++
120 lines
3.2 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_TESTSTEREO_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
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#include <math.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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#define PCMA_AND_PCMU
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namespace webrtc {
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enum StereoMonoMode {
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kNotSet,
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kMono,
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kStereo
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};
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class TestPackStereo : public AudioPacketizationCallback {
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public:
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TestPackStereo();
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~TestPackStereo();
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void RegisterReceiverACM(AudioCodingModule* acm);
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int32_t SendData(const FrameType frame_type,
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const uint8_t payload_type,
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const uint32_t timestamp,
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const uint8_t* payload_data,
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const size_t payload_size,
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const RTPFragmentationHeader* fragmentation) override;
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uint16_t payload_size();
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uint32_t timestamp_diff();
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void reset_payload_size();
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void set_codec_mode(StereoMonoMode mode);
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void set_lost_packet(bool lost);
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private:
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AudioCodingModule* receiver_acm_;
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int16_t seq_no_;
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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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int payload_size_;
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StereoMonoMode codec_mode_;
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// Simulate packet losses
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bool lost_packet_;
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};
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class TestStereo : public ACMTest {
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public:
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explicit TestStereo(int test_mode);
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~TestStereo();
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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 frequncy matching is not required. This is useful
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// for codecs which support several sampling frequency.
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void RegisterSendCodec(char side, char* codec_name, int32_t samp_freq_hz,
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int rate, int pack_size, int channels,
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int payload_type);
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void Run(TestPackStereo* channel, int in_channels, int out_channels,
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int percent_loss = 0);
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void OpenOutFile(int16_t 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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TestPackStereo* channel_a2b_;
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PCMFile* in_file_stereo_;
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PCMFile* in_file_mono_;
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PCMFile out_file_;
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int16_t test_cntr_;
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uint16_t pack_size_samp_;
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uint16_t pack_size_bytes_;
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int counter_;
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char* send_codec_name_;
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// Payload types for stereo codecs and CNG
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#ifdef WEBRTC_CODEC_G722
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int g722_pltype_;
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#endif
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#ifdef WEBRTC_CODEC_PCM16
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int l16_8khz_pltype_;
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int l16_16khz_pltype_;
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int l16_32khz_pltype_;
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#endif
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#ifdef PCMA_AND_PCMU
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int pcma_pltype_;
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int pcmu_pltype_;
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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int opus_pltype_;
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#endif
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
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