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Remove more CriticalSectionWrappers.

Browse Source 
BUG=webrtc:7035

Review-Url: https://codereview.webrtc.org/2785673002
Cr-Commit-Position: refs/heads/master@{#17465}
This commit is contained in:
kthelgason 2017-03-30 01:14:41 -07:00 committed by Commit bot
parent e4cd15d379
commit 6bfe49c00a
19 changed files with 173 additions and 229 deletions
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20
webrtc/modules/audio_conference_mixer/source/time_scheduler.cc
View File

@ -10,27 +10,17 @@
#include "webrtc/base/timeutils.h"
#include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
TimeScheduler::TimeScheduler(const int64_t periodicityInMs)
: _crit(CriticalSectionWrapper::CreateCriticalSection()),
_isStarted(false),
: _isStarted(false),
_lastPeriodMark(),
_periodicityInMs(periodicityInMs),
_periodicityInTicks(periodicityInMs * rtc::kNumNanosecsPerMillisec),
_missedPeriods(0)
{
}
_missedPeriods(0) {}
TimeScheduler::~TimeScheduler()
{
delete _crit;
}
int32_t TimeScheduler::UpdateScheduler()
{
CriticalSectionScoped cs(_crit);
int32_t TimeScheduler::UpdateScheduler() {
rtc::CritScope cs(&_crit);
if(!_isStarted)
{
_isStarted = true;
@ -79,7 +69,7 @@ int32_t TimeScheduler::UpdateScheduler()
int32_t TimeScheduler::TimeToNextUpdate(
int64_t& updateTimeInMS) const
{
CriticalSectionScoped cs(_crit);
rtc::CritScope cs(&_crit);
// Missed periods means that the next UpdateScheduler() should happen
// immediately.
if(_missedPeriods > 0)

9
webrtc/modules/audio_conference_mixer/source/time_scheduler.h
View File

@ -11,17 +11,18 @@
// The TimeScheduler class keeps track of periodic events. It is non-drifting
// and keeps track of any missed periods so that it is possible to catch up.
// (compare to a metronome)
#include "webrtc/base/criticalsection.h"
#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
namespace webrtc {
class CriticalSectionWrapper;
class TimeScheduler
{
class TimeScheduler {
public:
TimeScheduler(const int64_t periodicityInMs);
~TimeScheduler();
~TimeScheduler() = default;
// Signal that a periodic event has been triggered.
int32_t UpdateScheduler();
@ -31,7 +32,7 @@ public:
int32_t TimeToNextUpdate(int64_t& updateTimeInMS) const;
private:
CriticalSectionWrapper* _crit;
rtc::CriticalSection _crit;
bool _isStarted;
int64_t _lastPeriodMark; // In ns

23
webrtc/modules/audio_device/audio_device_impl.cc
View File

@ -53,7 +53,6 @@
#include "webrtc/modules/audio_device/dummy/audio_device_dummy.h"
#include "webrtc/modules/audio_device/dummy/file_audio_device.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#define CHECK_INITIALIZED() \
{ \
@ -118,10 +117,7 @@ rtc::scoped_refptr<AudioDeviceModule> AudioDeviceModule::Create(
AudioDeviceModuleImpl::AudioDeviceModuleImpl(const int32_t id,
const AudioLayer audioLayer)
: _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
_critSectEventCb(*CriticalSectionWrapper::CreateCriticalSection()),
_critSectAudioCb(*CriticalSectionWrapper::CreateCriticalSection()),
_ptrCbAudioDeviceObserver(NULL),
: _ptrCbAudioDeviceObserver(NULL),
_ptrAudioDevice(NULL),
_id(id),
_platformAudioLayer(audioLayer),
@ -358,15 +354,10 @@ int32_t AudioDeviceModuleImpl::AttachAudioBuffer() {
AudioDeviceModuleImpl::~AudioDeviceModuleImpl() {
LOG(INFO) << __FUNCTION__;
if (_ptrAudioDevice) {
delete _ptrAudioDevice;
_ptrAudioDevice = NULL;
}
delete &_critSect;
delete &_critSectEventCb;
delete &_critSectAudioCb;
}
// ============================================================================
@ -398,7 +389,7 @@ void AudioDeviceModuleImpl::Process() {
// kPlayoutWarning
if (_ptrAudioDevice->PlayoutWarning()) {
CriticalSectionScoped lock(&_critSectEventCb);
rtc::CritScope lock(&_critSectEventCb);
if (_ptrCbAudioDeviceObserver) {
LOG(WARNING) << "=> OnWarningIsReported(kPlayoutWarning)";
_ptrCbAudioDeviceObserver->OnWarningIsReported(
@ -409,7 +400,7 @@ void AudioDeviceModuleImpl::Process() {
// kPlayoutError
if (_ptrAudioDevice->PlayoutError()) {
CriticalSectionScoped lock(&_critSectEventCb);
rtc::CritScope lock(&_critSectEventCb);
if (_ptrCbAudioDeviceObserver) {
LOG(LERROR) << "=> OnErrorIsReported(kPlayoutError)";
_ptrCbAudioDeviceObserver->OnErrorIsReported(
@ -420,7 +411,7 @@ void AudioDeviceModuleImpl::Process() {
// kRecordingWarning
if (_ptrAudioDevice->RecordingWarning()) {
CriticalSectionScoped lock(&_critSectEventCb);
rtc::CritScope lock(&_critSectEventCb);
if (_ptrCbAudioDeviceObserver) {
LOG(WARNING) << "=> OnWarningIsReported(kRecordingWarning)";
_ptrCbAudioDeviceObserver->OnWarningIsReported(
@ -431,7 +422,7 @@ void AudioDeviceModuleImpl::Process() {
// kRecordingError
if (_ptrAudioDevice->RecordingError()) {
CriticalSectionScoped lock(&_critSectEventCb);
rtc::CritScope lock(&_critSectEventCb);
if (_ptrCbAudioDeviceObserver) {
LOG(LERROR) << "=> OnErrorIsReported(kRecordingError)";
_ptrCbAudioDeviceObserver->OnErrorIsReported(
@ -1459,7 +1450,7 @@ bool AudioDeviceModuleImpl::Recording() const {
int32_t AudioDeviceModuleImpl::RegisterEventObserver(
AudioDeviceObserver* eventCallback) {
LOG(INFO) << __FUNCTION__;
CriticalSectionScoped lock(&_critSectEventCb);
rtc::CritScope lock(&_critSectEventCb);
_ptrCbAudioDeviceObserver = eventCallback;
return 0;
@ -1472,7 +1463,7 @@ int32_t AudioDeviceModuleImpl::RegisterEventObserver(
int32_t AudioDeviceModuleImpl::RegisterAudioCallback(
AudioTransport* audioCallback) {
LOG(INFO) << __FUNCTION__;
CriticalSectionScoped lock(&_critSectAudioCb);
rtc::CritScope lock(&_critSectAudioCb);
return _audioDeviceBuffer.RegisterAudioCallback(audioCallback);
}

8
webrtc/modules/audio_device/audio_device_impl.h
View File

@ -16,6 +16,7 @@
#include <memory>
#include "webrtc/base/checks.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/modules/audio_device/audio_device_buffer.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
@ -23,7 +24,6 @@ namespace webrtc {
class AudioDeviceGeneric;
class AudioManager;
class CriticalSectionWrapper;
class AudioDeviceModuleImpl : public AudioDeviceModule {
public:
@ -203,9 +203,9 @@ class AudioDeviceModuleImpl : public AudioDeviceModule {
PlatformType Platform() const;
AudioLayer PlatformAudioLayer() const;
CriticalSectionWrapper& _critSect;
CriticalSectionWrapper& _critSectEventCb;
CriticalSectionWrapper& _critSectAudioCb;
rtc::CriticalSection _critSect;
rtc::CriticalSection _critSectEventCb;
rtc::CriticalSection _critSectAudioCb;
AudioDeviceObserver* _ptrCbAudioDeviceObserver;

13
webrtc/modules/audio_device/dummy/file_audio_device.cc
View File

@ -33,7 +33,6 @@ FileAudioDevice::FileAudioDevice(const int32_t id,
_playoutBuffer(NULL),
_recordingFramesLeft(0),
_playoutFramesLeft(0),
_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
_recordingBufferSizeIn10MS(0),
_recordingFramesIn10MS(0),
_playoutFramesIn10MS(0),
@ -161,7 +160,7 @@ int32_t FileAudioDevice::RecordingIsAvailable(bool& available) {
}
int32_t FileAudioDevice::InitRecording() {
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_recording) {
return -1;
@ -219,7 +218,7 @@ int32_t FileAudioDevice::StartPlayout() {
int32_t FileAudioDevice::StopPlayout() {
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_playing = false;
}
@ -229,7 +228,7 @@ int32_t FileAudioDevice::StopPlayout() {
_ptrThreadPlay.reset();
}
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_playoutFramesLeft = 0;
delete [] _playoutBuffer;
@ -280,7 +279,7 @@ int32_t FileAudioDevice::StartRecording() {
int32_t FileAudioDevice::StopRecording() {
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_recording = false;
}
@ -289,7 +288,7 @@ int32_t FileAudioDevice::StopRecording() {
_ptrThreadRec.reset();
}
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_recordingFramesLeft = 0;
if (_recordingBuffer) {
delete [] _recordingBuffer;
@ -456,7 +455,7 @@ void FileAudioDevice::ClearRecordingWarning() {}
void FileAudioDevice::ClearRecordingError() {}
void FileAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_ptrAudioBuffer = audioBuffer;

4
webrtc/modules/audio_device/dummy/file_audio_device.h
View File

@ -16,9 +16,9 @@
#include <memory>
#include <string>
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
namespace rtc {
@ -176,7 +176,7 @@ class FileAudioDevice : public AudioDeviceGeneric {
int8_t* _playoutBuffer; // In bytes.
uint32_t _recordingFramesLeft;
uint32_t _playoutFramesLeft;
CriticalSectionWrapper& _critSect;
rtc::CriticalSection _critSect;
size_t _recordingBufferSizeIn10MS;
size_t _recordingFramesIn10MS;

44
webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
View File

@ -62,7 +62,6 @@ static const unsigned int ALSA_CAPTURE_WAIT_TIMEOUT = 5; // in ms
AudioDeviceLinuxALSA::AudioDeviceLinuxALSA(const int32_t id) :
_ptrAudioBuffer(NULL),
_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
_id(id),
_mixerManager(id),
_inputDeviceIndex(0),
@ -130,13 +129,12 @@ AudioDeviceLinuxALSA::~AudioDeviceLinuxALSA()
delete [] _playoutBuffer;
_playoutBuffer = NULL;
}
delete &_critSect;
}
void AudioDeviceLinuxALSA::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_ptrAudioBuffer = audioBuffer;
@ -157,7 +155,7 @@ int32_t AudioDeviceLinuxALSA::ActiveAudioLayer(
}
AudioDeviceGeneric::InitStatus AudioDeviceLinuxALSA::Init() {
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
// Load libasound
if (!AlsaSymbolTable.Load()) {
@ -194,7 +192,7 @@ int32_t AudioDeviceLinuxALSA::Terminate()
return 0;
}
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_mixerManager.Close();
@ -243,7 +241,7 @@ bool AudioDeviceLinuxALSA::Initialized() const
int32_t AudioDeviceLinuxALSA::InitSpeaker()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_playing)
{
@ -258,7 +256,7 @@ int32_t AudioDeviceLinuxALSA::InitSpeaker()
int32_t AudioDeviceLinuxALSA::InitMicrophone()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_recording)
{
@ -560,7 +558,7 @@ int32_t AudioDeviceLinuxALSA::MicrophoneBoost(bool& enabled) const
int32_t AudioDeviceLinuxALSA::StereoRecordingIsAvailable(bool& available)
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
// If we already have initialized in stereo it's obviously available
if (_recIsInitialized && (2 == _recChannels))
@ -631,7 +629,7 @@ int32_t AudioDeviceLinuxALSA::StereoRecording(bool& enabled) const
int32_t AudioDeviceLinuxALSA::StereoPlayoutIsAvailable(bool& available)
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
// If we already have initialized in stereo it's obviously available
if (_playIsInitialized && (2 == _playChannels))
@ -1009,7 +1007,7 @@ int32_t AudioDeviceLinuxALSA::InitPlayout()
int errVal = 0;
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_playing)
{
return -1;
@ -1162,7 +1160,7 @@ int32_t AudioDeviceLinuxALSA::InitRecording()
int errVal = 0;
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_recording)
{
@ -1397,7 +1395,7 @@ int32_t AudioDeviceLinuxALSA::StopRecording()
{
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (!_recIsInitialized)
{
@ -1420,7 +1418,7 @@ int32_t AudioDeviceLinuxALSA::StopRecording()
_ptrThreadRec.reset();
}
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_recordingFramesLeft = 0;
if (_recordingBuffer)
{
@ -1523,7 +1521,7 @@ int32_t AudioDeviceLinuxALSA::StopPlayout()
{
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (!_playIsInitialized)
{
@ -1545,7 +1543,7 @@ int32_t AudioDeviceLinuxALSA::StopPlayout()
_ptrThreadPlay.reset();
}
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_playoutFramesLeft = 0;
delete [] _playoutBuffer;
@ -1635,49 +1633,49 @@ int32_t AudioDeviceLinuxALSA::CPULoad(uint16_t& load) const
bool AudioDeviceLinuxALSA::PlayoutWarning() const
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
return (_playWarning > 0);
}
bool AudioDeviceLinuxALSA::PlayoutError() const
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
return (_playError > 0);
}
bool AudioDeviceLinuxALSA::RecordingWarning() const
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
return (_recWarning > 0);
}
bool AudioDeviceLinuxALSA::RecordingError() const
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
return (_recError > 0);
}
void AudioDeviceLinuxALSA::ClearPlayoutWarning()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_playWarning = 0;
}
void AudioDeviceLinuxALSA::ClearPlayoutError()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_playError = 0;
}
void AudioDeviceLinuxALSA::ClearRecordingWarning()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_recWarning = 0;
}
void AudioDeviceLinuxALSA::ClearRecordingError()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_recError = 0;
}

13
webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
View File

@ -13,10 +13,10 @@
#include <memory>
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/platform_thread.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#if defined(USE_X11)
#include <X11/Xlib.h>
@ -145,7 +145,6 @@ public:
// CPU load
int32_t CPULoad(uint16_t& load) const override;
public:
bool PlayoutWarning() const override;
bool PlayoutError() const override;
bool RecordingWarning() const override;
@ -155,7 +154,6 @@ public:
void ClearRecordingWarning() override;
void ClearRecordingError() override;
public:
void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
private:
@ -166,26 +164,22 @@ private:
const int32_t ednLen = 0) const;
int32_t ErrorRecovery(int32_t error, snd_pcm_t* deviceHandle);
private:
bool KeyPressed() const;
private:
void Lock() EXCLUSIVE_LOCK_FUNCTION(_critSect) { _critSect.Enter(); };
void UnLock() UNLOCK_FUNCTION(_critSect) { _critSect.Leave(); };
private:
inline int32_t InputSanityCheckAfterUnlockedPeriod() const;
inline int32_t OutputSanityCheckAfterUnlockedPeriod() const;
private:
static bool RecThreadFunc(void*);
static bool PlayThreadFunc(void*);
bool RecThreadProcess();
bool PlayThreadProcess();
private:
AudioDeviceBuffer* _ptrAudioBuffer;
CriticalSectionWrapper& _critSect;
rtc::CriticalSection _critSect;
// TODO(pbos): Make plain members and start/stop instead of resetting these
// pointers. A thread can be reused.
@ -226,7 +220,6 @@ private:
AudioDeviceModule::BufferType _playBufType;
private:
bool _initialized;
bool _recording;
bool _playing;

42
webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
View File

@ -30,7 +30,6 @@ namespace webrtc
AudioDeviceLinuxPulse::AudioDeviceLinuxPulse(const int32_t id) :
_ptrAudioBuffer(NULL),
_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
_timeEventRec(*EventWrapper::Create()),
_timeEventPlay(*EventWrapper::Create()),
_recStartEvent(*EventWrapper::Create()),
@ -133,7 +132,6 @@ AudioDeviceLinuxPulse::~AudioDeviceLinuxPulse()
delete &_playStartEvent;
delete &_timeEventRec;
delete &_timeEventPlay;
delete &_critSect;
}
void AudioDeviceLinuxPulse::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
@ -751,7 +749,7 @@ int32_t AudioDeviceLinuxPulse::StereoPlayout(bool& enabled) const
int32_t AudioDeviceLinuxPulse::SetAGC(bool enable)
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_AGC = enable;
return 0;
@ -759,7 +757,7 @@ int32_t AudioDeviceLinuxPulse::SetAGC(bool enable)
bool AudioDeviceLinuxPulse::AGC() const
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
return _AGC;
}
@ -1346,7 +1344,7 @@ int32_t AudioDeviceLinuxPulse::StartRecording()
if (kEventTimeout == _recStartEvent.Wait(10000))
{
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_startRec = false;
}
StopRecording();
@ -1356,7 +1354,7 @@ int32_t AudioDeviceLinuxPulse::StartRecording()
}
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_recording)
{
// The recording state is set by the audio thread after recording
@ -1375,7 +1373,7 @@ int32_t AudioDeviceLinuxPulse::StartRecording()
int32_t AudioDeviceLinuxPulse::StopRecording()
{
RTC_DCHECK(thread_checker_.CalledOnValidThread());
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (!_recIsInitialized)
{
@ -1469,7 +1467,7 @@ int32_t AudioDeviceLinuxPulse::StartPlayout()
// Set state to ensure that playout starts from the audio thread.
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_startPlay = true;
}
@ -1481,7 +1479,7 @@ int32_t AudioDeviceLinuxPulse::StartPlayout()
if (kEventTimeout == _playStartEvent.Wait(10000))
{
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_startPlay = false;
}
StopPlayout();
@ -1491,7 +1489,7 @@ int32_t AudioDeviceLinuxPulse::StartPlayout()
}
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_playing)
{
// The playing state is set by the audio thread after playout
@ -1510,7 +1508,7 @@ int32_t AudioDeviceLinuxPulse::StartPlayout()
int32_t AudioDeviceLinuxPulse::StopPlayout()
{
RTC_DCHECK(thread_checker_.CalledOnValidThread());
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (!_playIsInitialized)
{
@ -1574,7 +1572,7 @@ int32_t AudioDeviceLinuxPulse::StopPlayout()
int32_t AudioDeviceLinuxPulse::PlayoutDelay(uint16_t& delayMS) const
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
delayMS = (uint16_t) _sndCardPlayDelay;
return 0;
}
@ -1631,49 +1629,49 @@ int32_t AudioDeviceLinuxPulse::CPULoad(uint16_t& /*load*/) const
bool AudioDeviceLinuxPulse::PlayoutWarning() const
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
return (_playWarning > 0);
}
bool AudioDeviceLinuxPulse::PlayoutError() const
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
return (_playError > 0);
}
bool AudioDeviceLinuxPulse::RecordingWarning() const
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
return (_recWarning > 0);
}
bool AudioDeviceLinuxPulse::RecordingError() const
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
return (_recError > 0);
}
void AudioDeviceLinuxPulse::ClearPlayoutWarning()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_playWarning = 0;
}
void AudioDeviceLinuxPulse::ClearPlayoutError()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_playError = 0;
}
void AudioDeviceLinuxPulse::ClearRecordingWarning()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_recWarning = 0;
}
void AudioDeviceLinuxPulse::ClearRecordingError()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_recError = 0;
}
@ -2602,7 +2600,7 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess()
return true;
}
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_startPlay)
{
@ -2843,7 +2841,7 @@ bool AudioDeviceLinuxPulse::RecThreadProcess()
return true;
}
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_startRec)
{

4
webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
View File

@ -13,11 +13,11 @@
#include <memory>
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/platform_thread.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include <X11/Xlib.h>
#include <pulse/pulseaudio.h>
@ -280,7 +280,7 @@ private:
AudioDeviceBuffer* _ptrAudioBuffer;
CriticalSectionWrapper& _critSect;
rtc::CriticalSection _critSect;
EventWrapper& _timeEventRec;
EventWrapper& _timeEventPlay;
EventWrapper& _recStartEvent;

33
webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
View File

@ -25,7 +25,6 @@ namespace webrtc
{
AudioMixerManagerLinuxALSA::AudioMixerManagerLinuxALSA(const int32_t id) :
_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
_id(id),
_outputMixerHandle(NULL),
_inputMixerHandle(NULL),
@ -43,10 +42,7 @@ AudioMixerManagerLinuxALSA::~AudioMixerManagerLinuxALSA()
{
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
"%s destructed", __FUNCTION__);
Close();
delete &_critSect;
}
// ============================================================================
@ -58,7 +54,7 @@ int32_t AudioMixerManagerLinuxALSA::Close()
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
__FUNCTION__);
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
CloseSpeaker();
CloseMicrophone();
@ -72,7 +68,7 @@ int32_t AudioMixerManagerLinuxALSA::CloseSpeaker()
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
__FUNCTION__);
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
int errVal = 0;
@ -113,7 +109,7 @@ int32_t AudioMixerManagerLinuxALSA::CloseMicrophone()
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
int errVal = 0;
@ -165,7 +161,7 @@ int32_t AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName)
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"AudioMixerManagerLinuxALSA::OpenSpeaker(name=%s)", deviceName);
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
int errVal = 0;
@ -259,7 +255,7 @@ int32_t AudioMixerManagerLinuxALSA::OpenMicrophone(char *deviceName)
"AudioMixerManagerLinuxALSA::OpenMicrophone(name=%s)",
deviceName);
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
int errVal = 0;
@ -378,7 +374,7 @@ int32_t AudioMixerManagerLinuxALSA::SetSpeakerVolume(
"AudioMixerManagerLinuxALSA::SetSpeakerVolume(volume=%u)",
volume);
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_outputMixerElement == NULL)
{
@ -642,7 +638,7 @@ int32_t AudioMixerManagerLinuxALSA::SetSpeakerMute(bool enable)
"AudioMixerManagerLinuxALSA::SetSpeakerMute(enable=%u)",
enable);
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_outputMixerElement == NULL)
{
@ -739,7 +735,7 @@ int32_t AudioMixerManagerLinuxALSA::SetMicrophoneMute(bool enable)
"AudioMixerManagerLinuxALSA::SetMicrophoneMute(enable=%u)",
enable);
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_inputMixerElement == NULL)
{
@ -838,7 +834,7 @@ int32_t AudioMixerManagerLinuxALSA::SetMicrophoneBoost(bool enable)
"AudioMixerManagerLinuxALSA::SetMicrophoneBoost(enable=%u)",
enable);
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_inputMixerHandle == NULL)
{
@ -900,7 +896,7 @@ int32_t AudioMixerManagerLinuxALSA::SetMicrophoneVolume(
"AudioMixerManagerLinuxALSA::SetMicrophoneVolume(volume=%u)",
volume);
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_inputMixerElement == NULL)
{
@ -1296,19 +1292,16 @@ void AudioMixerManagerLinuxALSA::GetControlName(char* controlName,
// controlName: "hw:CARD=Intel"
char* pos1 = strchr(deviceName, ':');
char* pos2 = strchr(deviceName, ',');
if (!pos2)
{
if (!pos2) {
// Can also be default:CARD=Intel
pos2 = &deviceName[strlen(deviceName)];
}
if (pos1 && pos2)
{
if (pos1 && pos2) {
strcpy(controlName, "hw");
int nChar = (int) (pos2 - pos1);
strncpy(&controlName[2], pos1, nChar);
controlName[2 + nChar] = '\0';
} else
{
} else {
strcpy(controlName, deviceName);
}

4
webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
View File

@ -11,9 +11,9 @@
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H
#define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H
#include "webrtc/base/criticalsection.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/modules/audio_device/linux/alsasymboltable_linux.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/typedefs.h"
#include <alsa/asoundlib.h>
@ -63,7 +63,7 @@ private:
void GetControlName(char *controlName, char* deviceName) const;
private:
CriticalSectionWrapper& _critSect;
rtc::CriticalSection _critSect;
int32_t _id;
mutable snd_mixer_t* _outputMixerHandle;
char _outputMixerStr[kAdmMaxDeviceNameSize];

77
webrtc/modules/audio_device/win/audio_device_core_win.cc
View File

@ -406,8 +406,6 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported()
AudioDeviceWindowsCore::AudioDeviceWindowsCore(const int32_t id) :
_comInit(ScopedCOMInitializer::kMTA),
_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
_volumeMutex(*CriticalSectionWrapper::CreateCriticalSection()),
_id(id),
_ptrAudioBuffer(NULL),
_ptrEnumerator(NULL),
@ -638,9 +636,6 @@ AudioDeviceWindowsCore::~AudioDeviceWindowsCore()
"AudioDeviceWindowsCore::~AudioDeviceWindowsCore() the Avrt DLL module is now unloaded");
}
}
delete &_critSect;
delete &_volumeMutex;
}
// ============================================================================
@ -680,7 +675,7 @@ int32_t AudioDeviceWindowsCore::ActiveAudioLayer(AudioDeviceModule::AudioLayer&
// ----------------------------------------------------------------------------
AudioDeviceGeneric::InitStatus AudioDeviceWindowsCore::Init() {
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_initialized) {
return InitStatus::OK;
@ -709,7 +704,7 @@ AudioDeviceGeneric::InitStatus AudioDeviceWindowsCore::Init() {
int32_t AudioDeviceWindowsCore::Terminate()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (!_initialized) {
return 0;
@ -751,7 +746,7 @@ bool AudioDeviceWindowsCore::Initialized() const
int32_t AudioDeviceWindowsCore::InitSpeaker()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_playing)
{
@ -833,7 +828,7 @@ int32_t AudioDeviceWindowsCore::InitSpeaker()
int32_t AudioDeviceWindowsCore::InitMicrophone()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_recording)
{
@ -923,7 +918,7 @@ bool AudioDeviceWindowsCore::MicrophoneIsInitialized() const
int32_t AudioDeviceWindowsCore::SpeakerVolumeIsAvailable(bool& available)
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_ptrDeviceOut == NULL)
{
@ -968,7 +963,7 @@ int32_t AudioDeviceWindowsCore::SetSpeakerVolume(uint32_t volume)
{
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (!_speakerIsInitialized)
{
@ -1011,7 +1006,7 @@ int32_t AudioDeviceWindowsCore::SpeakerVolume(uint32_t& volume) const
{
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (!_speakerIsInitialized)
{
@ -1123,7 +1118,7 @@ int32_t AudioDeviceWindowsCore::SpeakerVolumeStepSize(uint16_t& stepSize) const
int32_t AudioDeviceWindowsCore::SpeakerMuteIsAvailable(bool& available)
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_ptrDeviceOut == NULL)
{
@ -1162,7 +1157,7 @@ Exit:
int32_t AudioDeviceWindowsCore::SetSpeakerMute(bool enable)
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (!_speakerIsInitialized)
{
@ -1242,7 +1237,7 @@ Exit:
int32_t AudioDeviceWindowsCore::MicrophoneMuteIsAvailable(bool& available)
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_ptrDeviceIn == NULL)
{
@ -1402,7 +1397,7 @@ int32_t AudioDeviceWindowsCore::StereoRecordingIsAvailable(bool& available)
int32_t AudioDeviceWindowsCore::SetStereoRecording(bool enable)
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (enable)
{
@ -1453,7 +1448,7 @@ int32_t AudioDeviceWindowsCore::StereoPlayoutIsAvailable(bool& available)
int32_t AudioDeviceWindowsCore::SetStereoPlayout(bool enable)
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (enable)
{
@ -1492,7 +1487,7 @@ int32_t AudioDeviceWindowsCore::StereoPlayout(bool& enabled) const
int32_t AudioDeviceWindowsCore::SetAGC(bool enable)
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_AGC = enable;
return 0;
}
@ -1503,7 +1498,7 @@ int32_t AudioDeviceWindowsCore::SetAGC(bool enable)
bool AudioDeviceWindowsCore::AGC() const
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
return _AGC;
}
@ -1514,7 +1509,7 @@ bool AudioDeviceWindowsCore::AGC() const
int32_t AudioDeviceWindowsCore::MicrophoneVolumeIsAvailable(bool& available)
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_ptrDeviceIn == NULL)
{
@ -1553,7 +1548,7 @@ int32_t AudioDeviceWindowsCore::SetMicrophoneVolume(uint32_t volume)
WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::SetMicrophoneVolume(volume=%u)", volume);
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (!_microphoneIsInitialized)
{
@ -1594,7 +1589,7 @@ Exit:
int32_t AudioDeviceWindowsCore::MicrophoneVolume(uint32_t& volume) const
{
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (!_microphoneIsInitialized)
{
@ -1689,7 +1684,7 @@ int32_t AudioDeviceWindowsCore::MicrophoneVolumeStepSize(uint16_t& stepSize) con
int16_t AudioDeviceWindowsCore::PlayoutDevices()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_RefreshDeviceList(eRender) != -1)
{
@ -1720,7 +1715,7 @@ int32_t AudioDeviceWindowsCore::SetPlayoutDevice(uint16_t index)
return -1;
}
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
HRESULT hr(S_OK);
@ -1775,7 +1770,7 @@ int32_t AudioDeviceWindowsCore::SetPlayoutDevice(AudioDeviceModule::WindowsDevic
role = eCommunications;
}
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
// Refresh the list of rendering endpoint devices
_RefreshDeviceList(eRender);
@ -1845,7 +1840,7 @@ int32_t AudioDeviceWindowsCore::PlayoutDeviceName(
memset(guid, 0, kAdmMaxGuidSize);
}
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
int32_t ret(-1);
WCHAR szDeviceName[MAX_PATH];
@ -1925,7 +1920,7 @@ int32_t AudioDeviceWindowsCore::RecordingDeviceName(
memset(guid, 0, kAdmMaxGuidSize);
}
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
int32_t ret(-1);
WCHAR szDeviceName[MAX_PATH];
@ -1979,7 +1974,7 @@ int32_t AudioDeviceWindowsCore::RecordingDeviceName(
int16_t AudioDeviceWindowsCore::RecordingDevices()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_RefreshDeviceList(eCapture) != -1)
{
@ -2010,7 +2005,7 @@ int32_t AudioDeviceWindowsCore::SetRecordingDevice(uint16_t index)
return -1;
}
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
HRESULT hr(S_OK);
@ -2065,7 +2060,7 @@ int32_t AudioDeviceWindowsCore::SetRecordingDevice(AudioDeviceModule::WindowsDev
role = eCommunications;
}
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
// Refresh the list of capture endpoint devices
_RefreshDeviceList(eCapture);
@ -2155,7 +2150,7 @@ int32_t AudioDeviceWindowsCore::RecordingIsAvailable(bool& available)
int32_t AudioDeviceWindowsCore::InitPlayout()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_playing)
{
@ -2495,7 +2490,7 @@ int32_t AudioDeviceWindowsCore::InitRecordingDMO()
int32_t AudioDeviceWindowsCore::InitRecording()
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
if (_recording)
{
@ -2746,7 +2741,7 @@ int32_t AudioDeviceWindowsCore::StartRecording()
}
{
CriticalSectionScoped critScoped(&_critSect);
rtc::CritScope critScoped(&_critSect);
// Create thread which will drive the capturing
LPTHREAD_START_ROUTINE lpStartAddress = WSAPICaptureThread;
@ -2996,7 +2991,7 @@ int32_t AudioDeviceWindowsCore::StartPlayout()
}
{
CriticalSectionScoped critScoped(&_critSect);
rtc::CritScope critScoped(&_critSect);
// Create thread which will drive the rendering.
assert(_hPlayThread == NULL);
@ -3046,7 +3041,7 @@ int32_t AudioDeviceWindowsCore::StopPlayout()
}
{
CriticalSectionScoped critScoped(&_critSect) ;
rtc::CritScope critScoped(&_critSect) ;
if (_hPlayThread == NULL)
{
@ -3079,7 +3074,7 @@ int32_t AudioDeviceWindowsCore::StopPlayout()
}
{
CriticalSectionScoped critScoped(&_critSect);
rtc::CritScope critScoped(&_critSect);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"webrtc_core_audio_render_thread is now closed");
@ -3123,7 +3118,7 @@ int32_t AudioDeviceWindowsCore::StopPlayout()
int32_t AudioDeviceWindowsCore::PlayoutDelay(uint16_t& delayMS) const
{
CriticalSectionScoped critScoped(&_critSect);
rtc::CritScope critScoped(&_critSect);
delayMS = static_cast<uint16_t>(_sndCardPlayDelay);
return 0;
}
@ -3134,7 +3129,7 @@ int32_t AudioDeviceWindowsCore::PlayoutDelay(uint16_t& delayMS) const
int32_t AudioDeviceWindowsCore::RecordingDelay(uint16_t& delayMS) const
{
CriticalSectionScoped critScoped(&_critSect);
rtc::CritScope critScoped(&_critSect);
delayMS = static_cast<uint16_t>(_sndCardRecDelay);
return 0;
}
@ -3154,7 +3149,7 @@ bool AudioDeviceWindowsCore::Playing() const
int32_t AudioDeviceWindowsCore::SetPlayoutBuffer(const AudioDeviceModule::BufferType type, uint16_t sizeMS)
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
_playBufType = type;
@ -3172,7 +3167,7 @@ int32_t AudioDeviceWindowsCore::SetPlayoutBuffer(const AudioDeviceModule::Buffer
int32_t AudioDeviceWindowsCore::PlayoutBuffer(AudioDeviceModule::BufferType& type, uint16_t& sizeMS) const
{
CriticalSectionScoped lock(&_critSect);
rtc::CritScope lock(&_critSect);
type = _playBufType;
if (type == AudioDeviceModule::kFixedBufferSize)
@ -3773,7 +3768,7 @@ DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO()
while (keepRecording)
{
CriticalSectionScoped critScoped(&_critSect);
rtc::CritScope critScoped(&_critSect);
DWORD dwStatus = 0;
{

13
webrtc/modules/audio_device/win/audio_device_core_win.h
View File

@ -24,8 +24,8 @@
#include <endpointvolume.h>
#include <mediaobj.h> // IMediaObject
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
// Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority
#pragma comment( lib, "avrt.lib" )
@ -237,10 +237,8 @@ private: // thread functions
void _Lock() { _critSect.Enter(); };
void _UnLock() { _critSect.Leave(); };
private:
int32_t Id() {return _id;}
private:
int SetDMOProperties();
int SetBoolProperty(IPropertyStore* ptrPS,
@ -272,21 +270,18 @@ private:
int32_t InitRecordingDMO();
private:
ScopedCOMInitializer _comInit;
AudioDeviceBuffer* _ptrAudioBuffer;
CriticalSectionWrapper& _critSect;
CriticalSectionWrapper& _volumeMutex;
int32_t _id;
rtc::CriticalSection _critSect;
rtc::CriticalSection _volumeMutex;
int32_t _id;
private: // MMDevice
IMMDeviceEnumerator* _ptrEnumerator;
IMMDeviceCollection* _ptrRenderCollection;
IMMDeviceCollection* _ptrCaptureCollection;
IMMDevice* _ptrDeviceOut;
IMMDevice* _ptrDeviceIn;
private: // WASAPI
IAudioClient* _ptrClientOut;
IAudioClient* _ptrClientIn;
IAudioRenderClient* _ptrRenderClient;

1
webrtc/modules/bitrate_controller/include/bitrate_controller.h
View File

@ -24,7 +24,6 @@
namespace webrtc {
class CriticalSectionWrapper;
class RtcEventLog;
// Deprecated

46
webrtc/modules/media_file/media_file_impl.cc
View File

@ -12,7 +12,6 @@
#include "webrtc/base/format_macros.h"
#include "webrtc/modules/media_file/media_file_impl.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
@ -29,8 +28,6 @@ void MediaFile::DestroyMediaFile(MediaFile* module)
MediaFileImpl::MediaFileImpl(const int32_t id)
: _id(id),
_crit(CriticalSectionWrapper::CreateCriticalSection()),
_callbackCrit(CriticalSectionWrapper::CreateCriticalSection()),
_ptrFileUtilityObj(NULL),
codec_info_(),
_ptrInStream(NULL),
@ -57,7 +54,7 @@ MediaFileImpl::~MediaFileImpl()
{
WEBRTC_TRACE(kTraceMemory, kTraceFile, _id, "~MediaFileImpl()");
{
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
if(_playingActive)
{
@ -79,9 +76,6 @@ MediaFileImpl::~MediaFileImpl()
_ptrOutStream = NULL;
}
}
delete _crit;
delete _callbackCrit;
}
int64_t MediaFileImpl::TimeUntilNextProcess()
@ -119,7 +113,7 @@ int32_t MediaFileImpl::PlayoutAudioData(int8_t* buffer,
int32_t bytesRead = 0;
{
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
if(!_playingActive)
{
@ -213,7 +207,7 @@ void MediaFileImpl::HandlePlayCallbacks(int32_t bytesRead)
}
// Only _callbackCrit may and should be taken when making callbacks.
CriticalSectionScoped lock(_callbackCrit);
rtc::CritScope lock(&_callbackCrit);
if(_ptrCallback)
{
if(callbackNotifyMs)
@ -252,7 +246,7 @@ int32_t MediaFileImpl::PlayoutStereoData(
bool playEnded = false;
uint32_t callbackNotifyMs = 0;
{
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
if(!_playingActive || !_isStereo)
{
@ -313,7 +307,7 @@ int32_t MediaFileImpl::PlayoutStereoData(
}
}
CriticalSectionScoped lock(_callbackCrit);
rtc::CritScope lock(&_callbackCrit);
if(_ptrCallback)
{
if(callbackNotifyMs)
@ -386,7 +380,7 @@ int32_t MediaFileImpl::StartPlayingAudioFile(
return -1;
}
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
_openFile = true;
strncpy(_fileName, fileName, sizeof(_fileName));
_fileName[sizeof(_fileName) - 1] = '\0';
@ -424,7 +418,7 @@ int32_t MediaFileImpl::StartPlayingStream(
return -1;
}
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
if(_playingActive || _recordingActive)
{
WEBRTC_TRACE(
@ -555,7 +549,7 @@ int32_t MediaFileImpl::StartPlayingStream(
int32_t MediaFileImpl::StopPlaying()
{
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
_isStereo = false;
if(_ptrFileUtilityObj)
{
@ -590,7 +584,7 @@ int32_t MediaFileImpl::StopPlaying()
bool MediaFileImpl::IsPlaying()
{
WEBRTC_TRACE(kTraceStream, kTraceFile, _id, "MediaFileImpl::IsPlaying()");
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
return _playingActive;
}
@ -612,7 +606,7 @@ int32_t MediaFileImpl::IncomingAudioData(
bool recordingEnded = false;
uint32_t callbackNotifyMs = 0;
{
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
if(!_recordingActive)
{
@ -707,7 +701,7 @@ int32_t MediaFileImpl::IncomingAudioData(
}
// Only _callbackCrit may and should be taken when making callbacks.
CriticalSectionScoped lock(_callbackCrit);
rtc::CritScope lock(&_callbackCrit);
if(_ptrCallback)
{
if(callbackNotifyMs)
@ -767,7 +761,7 @@ int32_t MediaFileImpl::StartRecordingAudioFile(
return -1;
}
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
_openFile = true;
strncpy(_fileName, fileName, sizeof(_fileName));
_fileName[sizeof(_fileName) - 1] = '\0';
@ -786,7 +780,7 @@ int32_t MediaFileImpl::StartRecordingAudioStream(
return -1;
}
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
if(_recordingActive || _playingActive)
{
WEBRTC_TRACE(
@ -923,7 +917,7 @@ int32_t MediaFileImpl::StartRecordingAudioStream(
int32_t MediaFileImpl::StopRecording()
{
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
if(!_recordingActive)
{
WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
@ -967,14 +961,14 @@ int32_t MediaFileImpl::StopRecording()
bool MediaFileImpl::IsRecording()
{
WEBRTC_TRACE(kTraceStream, kTraceFile, _id, "MediaFileImpl::IsRecording()");
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
return _recordingActive;
}
int32_t MediaFileImpl::RecordDurationMs(uint32_t& durationMs)
{
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
if(!_recordingActive)
{
durationMs = 0;
@ -987,14 +981,14 @@ int32_t MediaFileImpl::RecordDurationMs(uint32_t& durationMs)
bool MediaFileImpl::IsStereo()
{
WEBRTC_TRACE(kTraceStream, kTraceFile, _id, "MediaFileImpl::IsStereo()");
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
return _isStereo;
}
int32_t MediaFileImpl::SetModuleFileCallback(FileCallback* callback)
{
CriticalSectionScoped lock(_callbackCrit);
rtc::CritScope lock(&_callbackCrit);
_ptrCallback = callback;
return 0;
@ -1038,7 +1032,7 @@ int32_t MediaFileImpl::FileDurationMs(const char* fileName,
int32_t MediaFileImpl::PlayoutPositionMs(uint32_t& positionMs) const
{
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
if(!_playingActive)
{
positionMs = 0;
@ -1050,7 +1044,7 @@ int32_t MediaFileImpl::PlayoutPositionMs(uint32_t& positionMs) const
int32_t MediaFileImpl::codec_info(CodecInst& codecInst) const
{
CriticalSectionScoped lock(_crit);
rtc::CritScope lock(&_crit);
if(!_playingActive && !_recordingActive)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,

6
webrtc/modules/media_file/media_file_impl.h
View File

@ -11,12 +11,12 @@
#ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
#include "webrtc/base/criticalsection.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/media_file/media_file.h"
#include "webrtc/modules/media_file/media_file_defines.h"
#include "webrtc/modules/media_file/media_file_utility.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
class MediaFileImpl : public MediaFile
@ -120,8 +120,8 @@ private:
const uint32_t stopPointMs);
int32_t _id;
CriticalSectionWrapper* _crit;
CriticalSectionWrapper* _callbackCrit;
rtc::CriticalSection _crit;
rtc::CriticalSection _callbackCrit;
ModuleFileUtility* _ptrFileUtilityObj;
CodecInst codec_info_;

38
webrtc/modules/pacing/paced_sender.cc
View File

@ -23,7 +23,6 @@
#include "webrtc/modules/pacing/bitrate_prober.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/field_trial.h"
namespace {
@ -253,7 +252,6 @@ PacedSender::PacedSender(const Clock* clock,
: clock_(clock),
packet_sender_(packet_sender),
alr_detector_(new AlrDetector()),
critsect_(CriticalSectionWrapper::CreateCriticalSection()),
paused_(false),
media_budget_(new paced_sender::IntervalBudget(0)),
padding_budget_(new paced_sender::IntervalBudget(0)),
@ -272,14 +270,14 @@ PacedSender::PacedSender(const Clock* clock,
PacedSender::~PacedSender() {}
void PacedSender::CreateProbeCluster(int bitrate_bps) {
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
prober_->CreateProbeCluster(bitrate_bps, clock_->TimeInMilliseconds());
}
void PacedSender::Pause() {
LOG(LS_INFO) << "PacedSender paused.";
{
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
paused_ = true;
}
// Tell the process thread to call our TimeUntilNextProcess() method to get
@ -291,7 +289,7 @@ void PacedSender::Pause() {
void PacedSender::Resume() {
LOG(LS_INFO) << "PacedSender resumed.";
{
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
paused_ = false;
}
// Tell the process thread to call our TimeUntilNextProcess() method to
@ -302,14 +300,14 @@ void PacedSender::Resume() {
void PacedSender::SetProbingEnabled(bool enabled) {
RTC_CHECK_EQ(0, packet_counter_);
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
prober_->SetEnabled(enabled);
}
void PacedSender::SetEstimatedBitrate(uint32_t bitrate_bps) {
if (bitrate_bps == 0)
LOG(LS_ERROR) << "PacedSender is not designed to handle 0 bitrate.";
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
estimated_bitrate_bps_ = bitrate_bps;
padding_budget_->set_target_rate_kbps(
std::min(estimated_bitrate_bps_ / 1000, max_padding_bitrate_kbps_));
@ -321,7 +319,7 @@ void PacedSender::SetEstimatedBitrate(uint32_t bitrate_bps) {
void PacedSender::SetSendBitrateLimits(int min_send_bitrate_bps,
int padding_bitrate) {
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
min_send_bitrate_kbps_ = min_send_bitrate_bps / 1000;
pacing_bitrate_kbps_ =
std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000) *
@ -337,7 +335,7 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) {
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
RTC_DCHECK(estimated_bitrate_bps_ > 0)
<< "SetEstimatedBitrate must be called before InsertPacket.";
@ -353,7 +351,7 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
}
int64_t PacedSender::ExpectedQueueTimeMs() const {
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
RTC_DCHECK_GT(pacing_bitrate_kbps_, 0);
return static_cast<int64_t>(packets_->SizeInBytes() * 8 /
pacing_bitrate_kbps_);
@ -361,17 +359,17 @@ int64_t PacedSender::ExpectedQueueTimeMs() const {
rtc::Optional<int64_t> PacedSender::GetApplicationLimitedRegionStartTime()
const {
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
return alr_detector_->GetApplicationLimitedRegionStartTime();
}
size_t PacedSender::QueueSizePackets() const {
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
return packets_->SizeInPackets();
}
int64_t PacedSender::QueueInMs() const {
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
int64_t oldest_packet = packets_->OldestEnqueueTimeMs();
if (oldest_packet == 0)
@ -381,13 +379,13 @@ int64_t PacedSender::QueueInMs() const {
}
int64_t PacedSender::AverageQueueTimeMs() {
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
return packets_->AverageQueueTimeMs();
}
int64_t PacedSender::TimeUntilNextProcess() {
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
if (paused_)
return 1000 * 60 * 60;
@ -403,7 +401,7 @@ int64_t PacedSender::TimeUntilNextProcess() {
void PacedSender::Process() {
int64_t now_us = clock_->TimeInMicroseconds();
CriticalSectionScoped cs(critsect_.get());
rtc::CritScope cs(&critsect_);
int64_t elapsed_time_ms = (now_us - time_last_update_us_ + 500) / 1000;
time_last_update_us_ = now_us;
int target_bitrate_kbps = pacing_bitrate_kbps_;
@ -489,11 +487,11 @@ bool PacedSender::SendPacket(const paced_sender::Packet& packet,
return false;
}
critsect_->Leave();
critsect_.Leave();
const bool success = packet_sender_->TimeToSendPacket(
packet.ssrc, packet.sequence_number, packet.capture_time_ms,
packet.retransmission, pacing_info);
critsect_->Enter();
critsect_.Enter();
if (success) {
// TODO(holmer): High priority packets should only be accounted for if we
@ -509,10 +507,10 @@ bool PacedSender::SendPacket(const paced_sender::Packet& packet,
size_t PacedSender::SendPadding(size_t padding_needed,
const PacedPacketInfo& pacing_info) {
critsect_->Leave();
critsect_.Leave();
size_t bytes_sent =
packet_sender_->TimeToSendPadding(padding_needed, pacing_info);
critsect_->Enter();
critsect_.Enter();
if (bytes_sent > 0) {
UpdateBudgetWithBytesSent(bytes_sent);

4
webrtc/modules/pacing/paced_sender.h
View File

@ -15,6 +15,7 @@
#include <memory>
#include <set>
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/optional.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/include/module.h"
@ -25,7 +26,6 @@ namespace webrtc {
class AlrDetector;
class BitrateProber;
class Clock;
class CriticalSectionWrapper;
class ProbeClusterCreatedObserver;
class RtcEventLog;
@ -162,7 +162,7 @@ class PacedSender : public Module, public RtpPacketSender {
PacketSender* const packet_sender_;
std::unique_ptr<AlrDetector> alr_detector_ GUARDED_BY(critsect_);
std::unique_ptr<CriticalSectionWrapper> critsect_;
rtc::CriticalSection critsect_;
bool paused_ GUARDED_BY(critsect_);
// This is the media budget, keeping track of how many bits of media
// we can pace out during the current interval.