Removed the dependency on AudioProcessingImpl in GainControlImpl
BUG=webrtc:5353
Review URL: https://codereview.webrtc.org/1801003002
Cr-Commit-Position: refs/heads/master@{#11994}
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 67dcd90..32ebb0a 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -172,7 +172,7 @@
public_submodules_->echo_control_mobile.reset(
new EchoControlMobileImpl(this, &crit_render_, &crit_capture_));
public_submodules_->gain_control.reset(
- new GainControlImpl(this, &crit_capture_, &crit_capture_));
+ new GainControlImpl(&crit_capture_, &crit_capture_));
public_submodules_->high_pass_filter.reset(
new HighPassFilterImpl(&crit_capture_));
public_submodules_->level_estimator.reset(
@@ -718,7 +718,8 @@
ca->split_bands_const(0)[kBand0To8kHz], ca->num_frames_per_band(),
capture_nonlocked_.split_rate);
}
- RETURN_ON_ERR(public_submodules_->gain_control->ProcessCaptureAudio(ca));
+ RETURN_ON_ERR(public_submodules_->gain_control->ProcessCaptureAudio(
+ ca, echo_cancellation()->stream_has_echo()));
if (synthesis_needed(data_processed)) {
ca->MergeFrequencyBands();
@@ -1221,7 +1222,8 @@
}
void AudioProcessingImpl::InitializeGainController() {
- public_submodules_->gain_control->Initialize();
+ public_submodules_->gain_control->Initialize(num_proc_channels(),
+ proc_sample_rate_hz());
}
void AudioProcessingImpl::InitializeEchoControlMobile() {
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc
index 936a286..db1c585 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -90,11 +90,9 @@
RTC_DISALLOW_COPY_AND_ASSIGN(GainController);
};
-GainControlImpl::GainControlImpl(const AudioProcessing* apm,
- rtc::CriticalSection* crit_render,
+GainControlImpl::GainControlImpl(rtc::CriticalSection* crit_render,
rtc::CriticalSection* crit_capture)
- : apm_(apm),
- crit_render_(crit_render),
+ : crit_render_(crit_render),
crit_capture_(crit_capture),
mode_(kAdaptiveAnalog),
minimum_capture_level_(0),
@@ -106,7 +104,6 @@
was_analog_level_set_(false),
stream_is_saturated_(false),
render_queue_element_max_size_(0) {
- RTC_DCHECK(apm);
RTC_DCHECK(crit_render);
RTC_DCHECK(crit_capture);
}
@@ -159,8 +156,10 @@
while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
size_t buffer_index = 0;
+ RTC_DCHECK(num_proc_channels_);
+ RTC_DCHECK_LT(0ul, *num_proc_channels_);
const size_t num_frames_per_band =
- capture_queue_buffer_.size() / num_handles_required();
+ capture_queue_buffer_.size() / (*num_proc_channels_);
for (auto& gain_controller : gain_controllers_) {
WebRtcAgc_AddFarend(gain_controller->state(),
&capture_queue_buffer_[buffer_index],
@@ -178,9 +177,10 @@
return AudioProcessing::kNoError;
}
+ RTC_DCHECK(num_proc_channels_);
RTC_DCHECK_GE(160u, audio->num_frames_per_band());
- RTC_DCHECK_EQ(audio->num_channels(), num_handles_required());
- RTC_DCHECK_LE(num_handles_required(), gain_controllers_.size());
+ RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
+ RTC_DCHECK_LE(*num_proc_channels_, gain_controllers_.size());
if (mode_ == kAdaptiveAnalog) {
int capture_channel = 0;
@@ -216,7 +216,8 @@
return AudioProcessing::kNoError;
}
-int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio,
+ bool stream_has_echo) {
rtc::CritScope cs(crit_capture_);
if (!enabled_) {
@@ -227,8 +228,9 @@
return AudioProcessing::kStreamParameterNotSetError;
}
+ RTC_DCHECK(num_proc_channels_);
RTC_DCHECK_GE(160u, audio->num_frames_per_band());
- RTC_DCHECK_EQ(audio->num_channels(), num_handles_required());
+ RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
stream_is_saturated_ = false;
int capture_channel = 0;
@@ -243,7 +245,7 @@
audio->num_bands(), audio->num_frames_per_band(),
audio->split_bands(capture_channel),
gain_controller->get_capture_level(), &capture_level_out,
- apm_->echo_cancellation()->stream_has_echo(), &saturation_warning);
+ stream_has_echo, &saturation_warning);
if (err != AudioProcessing::kNoError) {
return AudioProcessing::kUnspecifiedError;
@@ -257,6 +259,7 @@
++capture_channel;
}
+ RTC_DCHECK_LT(0ul, *num_proc_channels_);
if (mode_ == kAdaptiveAnalog) {
// Take the analog level to be the average across the handles.
analog_capture_level_ = 0;
@@ -264,7 +267,7 @@
analog_capture_level_ += gain_controller->get_capture_level();
}
- analog_capture_level_ /= num_handles_required();
+ analog_capture_level_ /= (*num_proc_channels_);
}
was_analog_level_set_ = false;
@@ -297,7 +300,10 @@
rtc::CritScope cs_capture(crit_capture_);
if (enable && !enabled_) {
enabled_ = enable; // Must be set before Initialize() is called.
- Initialize();
+
+ RTC_DCHECK(num_proc_channels_);
+ RTC_DCHECK(sample_rate_hz_);
+ Initialize(*num_proc_channels_, *sample_rate_hz_);
} else {
enabled_ = enable;
}
@@ -317,7 +323,9 @@
}
mode_ = mode;
- Initialize();
+ RTC_DCHECK(num_proc_channels_);
+ RTC_DCHECK(sample_rate_hz_);
+ Initialize(*num_proc_channels_, *sample_rate_hz_);
return AudioProcessing::kNoError;
}
@@ -344,7 +352,9 @@
minimum_capture_level_ = minimum;
maximum_capture_level_ = maximum;
- Initialize();
+ RTC_DCHECK(num_proc_channels_);
+ RTC_DCHECK(sample_rate_hz_);
+ Initialize(*num_proc_channels_, *sample_rate_hz_);
return AudioProcessing::kNoError;
}
@@ -408,21 +418,24 @@
return limiter_enabled_;
}
-void GainControlImpl::Initialize() {
+void GainControlImpl::Initialize(size_t num_proc_channels, int sample_rate_hz) {
rtc::CritScope cs_render(crit_render_);
rtc::CritScope cs_capture(crit_capture_);
+
+ num_proc_channels_ = rtc::Optional<size_t>(num_proc_channels);
+ sample_rate_hz_ = rtc::Optional<int>(sample_rate_hz);
+
if (!enabled_) {
return;
}
- int sample_rate_hz = apm_->proc_sample_rate_hz();
- gain_controllers_.resize(num_handles_required());
+ gain_controllers_.resize(*num_proc_channels_);
for (auto& gain_controller : gain_controllers_) {
if (!gain_controller) {
gain_controller.reset(new GainController());
}
gain_controller->Initialize(minimum_capture_level_, maximum_capture_level_,
- mode_, sample_rate_hz, analog_capture_level_);
+ mode_, *sample_rate_hz_, analog_capture_level_);
}
Configure();
@@ -431,13 +444,14 @@
}
void GainControlImpl::AllocateRenderQueue() {
- const size_t new_render_queue_element_max_size = std::max<size_t>(
- static_cast<size_t>(1),
- kMaxAllowedValuesOfSamplesPerFrame * num_handles_required());
-
rtc::CritScope cs_render(crit_render_);
rtc::CritScope cs_capture(crit_capture_);
+ RTC_DCHECK(num_proc_channels_);
+ const size_t new_render_queue_element_max_size = std::max<size_t>(
+ static_cast<size_t>(1),
+ kMaxAllowedValuesOfSamplesPerFrame * (*num_proc_channels_));
+
if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
render_queue_element_max_size_ = new_render_queue_element_max_size;
std::vector<int16_t> template_queue_element(render_queue_element_max_size_);
@@ -477,9 +491,4 @@
}
return error;
}
-
-size_t GainControlImpl::num_handles_required() const {
- // Not locked as it only relies on APM public API which is threadsafe.
- return apm_->num_proc_channels();
-}
} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/gain_control_impl.h b/webrtc/modules/audio_processing/gain_control_impl.h
index b5fe1e3..6841d5d 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.h
+++ b/webrtc/modules/audio_processing/gain_control_impl.h
@@ -27,16 +27,15 @@
class GainControlImpl : public GainControl {
public:
- GainControlImpl(const AudioProcessing* apm,
- rtc::CriticalSection* crit_render,
+ GainControlImpl(rtc::CriticalSection* crit_render,
rtc::CriticalSection* crit_capture);
~GainControlImpl() override;
int ProcessRenderAudio(AudioBuffer* audio);
int AnalyzeCaptureAudio(AudioBuffer* audio);
- int ProcessCaptureAudio(AudioBuffer* audio);
+ int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo);
- void Initialize();
+ void Initialize(size_t num_proc_channels, int sample_rate_hz);
// GainControl implementation.
bool is_enabled() const override;
@@ -64,14 +63,9 @@
int analog_level_maximum() const override;
bool stream_is_saturated() const override;
- size_t num_handles_required() const;
-
void AllocateRenderQueue();
int Configure();
- // Not guarded as its public API is thread safe.
- const AudioProcessing* apm_;
-
rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
rtc::CriticalSection* const crit_capture_;
@@ -99,6 +93,9 @@
std::vector<std::unique_ptr<GainController>> gain_controllers_;
+ rtc::Optional<size_t> num_proc_channels_ GUARDED_BY(crit_capture_);
+ rtc::Optional<int> sample_rate_hz_ GUARDED_BY(crit_capture_);
+
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl);
};
} // namespace webrtc