Renamed old-style variables.
Review-Url: https://codereview.webrtc.org/2227633002
Cr-Commit-Position: refs/heads/master@{#13676}
diff --git a/webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.cc b/webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.cc
index 3afd9be..dbb46ff 100644
--- a/webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.cc
+++ b/webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.cc
@@ -98,12 +98,12 @@
return _mixHistory->IsMixed();
}
-NewMixHistory::NewMixHistory() : _isMixed(0) {}
+NewMixHistory::NewMixHistory() : is_mixed_(0) {}
NewMixHistory::~NewMixHistory() {}
bool NewMixHistory::IsMixed() const {
- return _isMixed;
+ return is_mixed_;
}
bool NewMixHistory::WasMixed() const {
@@ -113,12 +113,12 @@
}
int32_t NewMixHistory::SetIsMixed(const bool mixed) {
- _isMixed = mixed;
+ is_mixed_ = mixed;
return 0;
}
void NewMixHistory::ResetMixedStatus() {
- _isMixed = false;
+ is_mixed_ = false;
}
NewAudioConferenceMixer* NewAudioConferenceMixer::Create(int id) {
@@ -131,53 +131,53 @@
}
NewAudioConferenceMixerImpl::NewAudioConferenceMixerImpl(int id)
- : _id(id),
- _outputFrequency(kDefaultFrequency),
- _sampleSize(0),
+ : id_(id),
+ output_frequency_(kDefaultFrequency),
+ sample_size_(0),
audio_source_list_(),
additional_audio_source_list_(),
num_mixed_audio_sources_(0),
use_limiter_(true),
- _timeStamp(0) {
+ time_stamp_(0) {
thread_checker_.DetachFromThread();
}
bool NewAudioConferenceMixerImpl::Init() {
- _crit.reset(CriticalSectionWrapper::CreateCriticalSection());
- if (_crit.get() == NULL)
+ crit_.reset(CriticalSectionWrapper::CreateCriticalSection());
+ if (crit_.get() == NULL)
return false;
- _cbCrit.reset(CriticalSectionWrapper::CreateCriticalSection());
- if (_cbCrit.get() == NULL)
+ cb_crit_.reset(CriticalSectionWrapper::CreateCriticalSection());
+ if (cb_crit_.get() == NULL)
return false;
Config config;
config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
- _limiter.reset(AudioProcessing::Create(config));
- if (!_limiter.get())
+ limiter_.reset(AudioProcessing::Create(config));
+ if (!limiter_.get())
return false;
if (SetOutputFrequency(kDefaultFrequency) == -1)
return false;
- if (_limiter->gain_control()->set_mode(GainControl::kFixedDigital) !=
- _limiter->kNoError)
+ if (limiter_->gain_control()->set_mode(GainControl::kFixedDigital) !=
+ limiter_->kNoError)
return false;
// We smoothly limit the mixed frame to -7 dbFS. -6 would correspond to the
// divide-by-2 but -7 is used instead to give a bit of headroom since the
// AGC is not a hard limiter.
- if (_limiter->gain_control()->set_target_level_dbfs(7) != _limiter->kNoError)
+ if (limiter_->gain_control()->set_target_level_dbfs(7) != limiter_->kNoError)
return false;
- if (_limiter->gain_control()->set_compression_gain_db(0) !=
- _limiter->kNoError)
+ if (limiter_->gain_control()->set_compression_gain_db(0) !=
+ limiter_->kNoError)
return false;
- if (_limiter->gain_control()->enable_limiter(true) != _limiter->kNoError)
+ if (limiter_->gain_control()->enable_limiter(true) != limiter_->kNoError)
return false;
- if (_limiter->gain_control()->Enable(true) != _limiter->kNoError)
+ if (limiter_->gain_control()->Enable(true) != limiter_->kNoError)
return false;
return true;
@@ -193,7 +193,7 @@
AudioFrameList additionalFramesList;
std::map<int, MixerAudioSource*> mixedAudioSourcesMap;
{
- CriticalSectionScoped cs(_cbCrit.get());
+ CriticalSectionScoped cs(cb_crit_.get());
Frequency mixing_frequency;
switch (sample_rate) {
@@ -231,25 +231,25 @@
}
audio_frame_for_mixing->UpdateFrame(
- -1, _timeStamp, NULL, 0, _outputFrequency, AudioFrame::kNormalSpeech,
+ -1, time_stamp_, NULL, 0, output_frequency_, AudioFrame::kNormalSpeech,
AudioFrame::kVadPassive, number_of_channels);
- _timeStamp += static_cast<uint32_t>(_sampleSize);
+ time_stamp_ += static_cast<uint32_t>(sample_size_);
use_limiter_ = num_mixed_audio_sources_ > 1 &&
- _outputFrequency <= AudioProcessing::kMaxNativeSampleRateHz;
+ output_frequency_ <= AudioProcessing::kMaxNativeSampleRateHz;
// We only use the limiter if it supports the output sample rate and
// we're actually mixing multiple streams.
- MixFromList(audio_frame_for_mixing, mixList, _id, use_limiter_);
+ MixFromList(audio_frame_for_mixing, mixList, id_, use_limiter_);
{
- CriticalSectionScoped cs(_crit.get());
+ CriticalSectionScoped cs(crit_.get());
MixAnonomouslyFromList(audio_frame_for_mixing, additionalFramesList);
if (audio_frame_for_mixing->samples_per_channel_ == 0) {
// Nothing was mixed, set the audio samples to silence.
- audio_frame_for_mixing->samples_per_channel_ = _sampleSize;
+ audio_frame_for_mixing->samples_per_channel_ = sample_size_;
audio_frame_for_mixing->Mute();
} else {
// Only call the limiter if we have something to mix.
@@ -261,19 +261,19 @@
int32_t NewAudioConferenceMixerImpl::SetOutputFrequency(
const Frequency& frequency) {
- CriticalSectionScoped cs(_crit.get());
+ CriticalSectionScoped cs(crit_.get());
- _outputFrequency = frequency;
- _sampleSize =
- static_cast<size_t>((_outputFrequency * kProcessPeriodicityInMs) / 1000);
+ output_frequency_ = frequency;
+ sample_size_ =
+ static_cast<size_t>((output_frequency_ * kProcessPeriodicityInMs) / 1000);
return 0;
}
NewAudioConferenceMixer::Frequency
NewAudioConferenceMixerImpl::OutputFrequency() const {
- CriticalSectionScoped cs(_crit.get());
- return _outputFrequency;
+ CriticalSectionScoped cs(crit_.get());
+ return output_frequency_;
}
int32_t NewAudioConferenceMixerImpl::SetMixabilityStatus(
@@ -286,11 +286,11 @@
}
size_t numMixedAudioSources;
{
- CriticalSectionScoped cs(_cbCrit.get());
+ CriticalSectionScoped cs(cb_crit_.get());
const bool isMixed = IsAudioSourceInList(*audio_source, audio_source_list_);
// API must be called with a new state.
if (!(mixable ^ isMixed)) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, id_,
"Mixable is aready %s", isMixed ? "ON" : "off");
return -1;
}
@@ -301,7 +301,7 @@
success = RemoveAudioSourceFromList(audio_source, &audio_source_list_);
}
if (!success) {
- WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
+ WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
"failed to %s audio_source", mixable ? "add" : "remove");
RTC_NOTREACHED();
return -1;
@@ -317,28 +317,28 @@
// A MixerAudioSource was added or removed. Make sure the scratch
// buffer is updated if necessary.
// Note: The scratch buffer may only be updated in Process().
- CriticalSectionScoped cs(_crit.get());
+ CriticalSectionScoped cs(crit_.get());
num_mixed_audio_sources_ = numMixedAudioSources;
return 0;
}
bool NewAudioConferenceMixerImpl::MixabilityStatus(
const MixerAudioSource& audio_source) const {
- CriticalSectionScoped cs(_cbCrit.get());
+ CriticalSectionScoped cs(cb_crit_.get());
return IsAudioSourceInList(audio_source, audio_source_list_);
}
int32_t NewAudioConferenceMixerImpl::SetAnonymousMixabilityStatus(
MixerAudioSource* audio_source,
bool anonymous) {
- CriticalSectionScoped cs(_cbCrit.get());
+ CriticalSectionScoped cs(cb_crit_.get());
if (IsAudioSourceInList(*audio_source, additional_audio_source_list_)) {
if (anonymous) {
return 0;
}
if (!RemoveAudioSourceFromList(audio_source,
&additional_audio_source_list_)) {
- WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
+ WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
"unable to remove audio_source from anonymous list");
RTC_NOTREACHED();
return -1;
@@ -352,7 +352,7 @@
RemoveAudioSourceFromList(audio_source, &audio_source_list_);
if (!mixable) {
WEBRTC_TRACE(
- kTraceWarning, kTraceAudioMixerServer, _id,
+ kTraceWarning, kTraceAudioMixerServer, id_,
"audio_source must be registered before turning it into anonymous");
// Setting anonymous status is only possible if MixerAudioSource is
// already registered.
@@ -365,7 +365,7 @@
bool NewAudioConferenceMixerImpl::AnonymousMixabilityStatus(
const MixerAudioSource& audio_source) const {
- CriticalSectionScoped cs(_cbCrit.get());
+ CriticalSectionScoped cs(cb_crit_.get());
return IsAudioSourceInList(audio_source, additional_audio_source_list_);
}
@@ -377,13 +377,13 @@
// Get audio source audio and put it in the struct vector.
for (MixerAudioSource* audio_source : audio_source_list_) {
auto audio_frame_with_info = audio_source->GetAudioFrameWithMuted(
- _id, static_cast<int>(_outputFrequency));
+ id_, static_cast<int>(output_frequency_));
auto audio_frame_info = audio_frame_with_info.audio_frame_info;
AudioFrame* audio_source_audio_frame = audio_frame_with_info.audio_frame;
if (audio_frame_info == MixerAudioSource::AudioFrameInfo::kError) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, id_,
"failed to GetAudioFrameWithMuted() from participant");
continue;
}
@@ -429,7 +429,7 @@
void NewAudioConferenceMixerImpl::GetAdditionalAudio(
AudioFrameList* additionalFramesList) const {
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
"GetAdditionalAudio(additionalFramesList)");
// The GetAudioFrameWithMuted() callback may result in the audio source being
// removed from additionalAudioFramesList_. If that happens it will
@@ -444,11 +444,11 @@
additionalAudioSourceList.begin();
audio_source != additionalAudioSourceList.end(); ++audio_source) {
auto audio_frame_with_info =
- (*audio_source)->GetAudioFrameWithMuted(_id, _outputFrequency);
+ (*audio_source)->GetAudioFrameWithMuted(id_, output_frequency_);
auto ret = audio_frame_with_info.audio_frame_info;
AudioFrame* audio_frame = audio_frame_with_info.audio_frame;
if (ret == MixerAudioSource::AudioFrameInfo::kError) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
+ WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, id_,
"failed to GetAudioFrameWithMuted() from audio_source");
continue;
}
@@ -464,21 +464,16 @@
bool NewAudioConferenceMixerImpl::IsAudioSourceInList(
const MixerAudioSource& audio_source,
const MixerAudioSourceList& audioSourceList) const {
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
"IsAudioSourceInList(audio_source,audioSourceList)");
- for (MixerAudioSourceList::const_iterator iter = audioSourceList.begin();
- iter != audioSourceList.end(); ++iter) {
- if (&audio_source == *iter) {
- return true;
- }
- }
- return false;
+ return std::find(audioSourceList.begin(), audioSourceList.end(),
+ &audio_source) != audioSourceList.end();
}
bool NewAudioConferenceMixerImpl::AddAudioSourceToList(
MixerAudioSource* audio_source,
MixerAudioSourceList* audioSourceList) const {
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
"AddAudioSourceToList(audio_source, audioSourceList)");
audioSourceList->push_back(audio_source);
// Make sure that the mixed status is correct for new MixerAudioSource.
@@ -489,18 +484,18 @@
bool NewAudioConferenceMixerImpl::RemoveAudioSourceFromList(
MixerAudioSource* audio_source,
MixerAudioSourceList* audioSourceList) const {
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
"RemoveAudioSourceFromList(audio_source, audioSourceList)");
- for (MixerAudioSourceList::iterator iter = audioSourceList->begin();
- iter != audioSourceList->end(); ++iter) {
- if (*iter == audio_source) {
- audioSourceList->erase(iter);
- // AudioSource is no longer mixed, reset to default.
- audio_source->_mixHistory->ResetMixedStatus();
- return true;
- }
+ auto iter =
+ std::find(audioSourceList->begin(), audioSourceList->end(), audio_source);
+ if (iter != audioSourceList->end()) {
+ audioSourceList->erase(iter);
+ // AudioSource is no longer mixed, reset to default.
+ audio_source->_mixHistory->ResetMixedStatus();
+ return true;
+ } else {
+ return false;
}
- return false;
}
int32_t NewAudioConferenceMixerImpl::MixFromList(
@@ -551,7 +546,7 @@
int32_t NewAudioConferenceMixerImpl::MixAnonomouslyFromList(
AudioFrame* mixedAudio,
const AudioFrameList& audioFrameList) const {
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+ WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id_,
"MixAnonomouslyFromList(mixedAudio, audioFrameList)");
if (audioFrameList.empty())
@@ -573,7 +568,7 @@
}
// Smoothly limit the mixed frame.
- const int error = _limiter->ProcessStream(mixedAudio);
+ const int error = limiter_->ProcessStream(mixedAudio);
// And now we can safely restore the level. This procedure results in
// some loss of resolution, deemed acceptable.
@@ -587,8 +582,8 @@
// negative value is undefined).
*mixedAudio += *mixedAudio;
- if (error != _limiter->kNoError) {
- WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
+ if (error != limiter_->kNoError) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
"Error from AudioProcessing: %d", error);
RTC_NOTREACHED();
return false;
diff --git a/webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h b/webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h
index 1cca542..a6a3dea 100644
--- a/webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h
+++ b/webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h
@@ -53,7 +53,7 @@
void ResetMixedStatus();
private:
- bool _isMixed;
+ bool is_mixed_;
};
class NewAudioConferenceMixerImpl : public NewAudioConferenceMixer {
@@ -123,14 +123,14 @@
bool LimitMixedAudio(AudioFrame* mixedAudio) const;
- std::unique_ptr<CriticalSectionWrapper> _crit;
- std::unique_ptr<CriticalSectionWrapper> _cbCrit;
+ std::unique_ptr<CriticalSectionWrapper> crit_;
+ std::unique_ptr<CriticalSectionWrapper> cb_crit_;
- int32_t _id;
+ int32_t id_;
// The current sample frequency and sample size when mixing.
- Frequency _outputFrequency;
- size_t _sampleSize;
+ Frequency output_frequency_;
+ size_t sample_size_;
// List of all audio sources. Note all lists are disjunct
MixerAudioSourceList audio_source_list_; // May be mixed.
@@ -143,13 +143,13 @@
// mixing.
bool use_limiter_;
- uint32_t _timeStamp;
+ uint32_t time_stamp_;
// Ensures that Mix is called from the same thread.
rtc::ThreadChecker thread_checker_;
// Used for inhibiting saturation in mixing.
- std::unique_ptr<AudioProcessing> _limiter;
+ std::unique_ptr<AudioProcessing> limiter_;
};
} // namespace webrtc
