webrtc/modules/audio_processing/gain_control_impl.cc - platform/external/webrtc - Git at Google

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 #include "webrtc/modules/audio_processing/gain_control_impl.h"

 #include <assert.h>

 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"

 namespace webrtc {

 typedef void Handle;

 namespace {
 int16_t MapSetting(GainControl::Mode mode) {
   switch (mode) {
     case GainControl::kAdaptiveAnalog:
       return kAgcModeAdaptiveAnalog;
     case GainControl::kAdaptiveDigital:
       return kAgcModeAdaptiveDigital;
     case GainControl::kFixedDigital:
       return kAgcModeFixedDigital;
   }
   assert(false);
   return -1;
 }

 // Maximum length that a frame of samples can have.
 static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
 // Maximum number of frames to buffer in the render queue.
 // TODO(peah): Decrease this once we properly handle hugely unbalanced
 // reverse and forward call numbers.
 static const size_t kMaxNumFramesToBuffer = 100;

 }  // namespace

 GainControlImpl::GainControlImpl(const AudioProcessing* apm,
                                  rtc::CriticalSection* crit_render,
                                  rtc::CriticalSection* crit_capture)
     : ProcessingComponent(),
       apm_(apm),
       crit_render_(crit_render),
       crit_capture_(crit_capture),
       mode_(kAdaptiveAnalog),
       minimum_capture_level_(0),
       maximum_capture_level_(255),
       limiter_enabled_(true),
       target_level_dbfs_(3),
       compression_gain_db_(9),
       analog_capture_level_(0),
       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);
 }

 GainControlImpl::~GainControlImpl() {}

 int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
   rtc::CritScope cs(crit_render_);
   if (!is_component_enabled()) {
     return AudioProcessing::kNoError;
   }

   assert(audio->num_frames_per_band() <= 160);

   render_queue_buffer_.resize(0);
   for (int i = 0; i < num_handles(); i++) {
     Handle* my_handle = static_cast<Handle*>(handle(i));
     int err =
         WebRtcAgc_GetAddFarendError(my_handle, audio->num_frames_per_band());

     if (err != AudioProcessing::kNoError)
       return GetHandleError(my_handle);

     // Buffer the samples in the render queue.
     render_queue_buffer_.insert(
         render_queue_buffer_.end(), audio->mixed_low_pass_data(),
         (audio->mixed_low_pass_data() + audio->num_frames_per_band()));
   }

   // Insert the samples into the queue.
   if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
     // The data queue is full and needs to be emptied.
     ReadQueuedRenderData();

     // Retry the insert (should always work).
     RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
   }

   return AudioProcessing::kNoError;
 }

 // Read chunks of data that were received and queued on the render side from
 // a queue. All the data chunks are buffered into the farend signal of the AGC.
 void GainControlImpl::ReadQueuedRenderData() {
   rtc::CritScope cs(crit_capture_);

   if (!is_component_enabled()) {
     return;
   }

   while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
     size_t buffer_index = 0;
     const size_t num_frames_per_band =
         capture_queue_buffer_.size() / num_handles();
     for (int i = 0; i < num_handles(); i++) {
       Handle* my_handle = static_cast<Handle*>(handle(i));
       WebRtcAgc_AddFarend(my_handle, &capture_queue_buffer_[buffer_index],
                           num_frames_per_band);

       buffer_index += num_frames_per_band;
     }
   }
 }

 int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
   rtc::CritScope cs(crit_capture_);

   if (!is_component_enabled()) {
     return AudioProcessing::kNoError;
   }

   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == num_handles());

   int err = AudioProcessing::kNoError;

   if (mode_ == kAdaptiveAnalog) {
     capture_levels_.assign(num_handles(), analog_capture_level_);
     for (int i = 0; i < num_handles(); i++) {
       Handle* my_handle = static_cast<Handle*>(handle(i));
       err = WebRtcAgc_AddMic(
           my_handle,
           audio->split_bands(i),
           audio->num_bands(),
           audio->num_frames_per_band());

       if (err != AudioProcessing::kNoError) {
         return GetHandleError(my_handle);
       }
     }
   } else if (mode_ == kAdaptiveDigital) {

     for (int i = 0; i < num_handles(); i++) {
       Handle* my_handle = static_cast<Handle*>(handle(i));
       int32_t capture_level_out = 0;

       err = WebRtcAgc_VirtualMic(
           my_handle,
           audio->split_bands(i),
           audio->num_bands(),
           audio->num_frames_per_band(),
           analog_capture_level_,
           &capture_level_out);

       capture_levels_[i] = capture_level_out;

       if (err != AudioProcessing::kNoError) {
         return GetHandleError(my_handle);
       }

     }
   }

   return AudioProcessing::kNoError;
 }

 int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
   rtc::CritScope cs(crit_capture_);

   if (!is_component_enabled()) {
     return AudioProcessing::kNoError;
   }

   if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
     return AudioProcessing::kStreamParameterNotSetError;
   }

   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == num_handles());

   stream_is_saturated_ = false;
   for (int i = 0; i < num_handles(); i++) {
     Handle* my_handle = static_cast<Handle*>(handle(i));
     int32_t capture_level_out = 0;
     uint8_t saturation_warning = 0;

     // The call to stream_has_echo() is ok from a deadlock perspective
     // as the capture lock is allready held.
     int err = WebRtcAgc_Process(
         my_handle,
         audio->split_bands_const(i),
         audio->num_bands(),
         audio->num_frames_per_band(),
         audio->split_bands(i),
         capture_levels_[i],
         &capture_level_out,
         apm_->echo_cancellation()->stream_has_echo(),
         &saturation_warning);

     if (err != AudioProcessing::kNoError) {
       return GetHandleError(my_handle);
     }

     capture_levels_[i] = capture_level_out;
     if (saturation_warning == 1) {
       stream_is_saturated_ = true;
     }
   }

   if (mode_ == kAdaptiveAnalog) {
     // Take the analog level to be the average across the handles.
     analog_capture_level_ = 0;
     for (int i = 0; i < num_handles(); i++) {
       analog_capture_level_ += capture_levels_[i];
     }

     analog_capture_level_ /= num_handles();
   }

   was_analog_level_set_ = false;
   return AudioProcessing::kNoError;
 }

 // TODO(ajm): ensure this is called under kAdaptiveAnalog.
 int GainControlImpl::set_stream_analog_level(int level) {
   rtc::CritScope cs(crit_capture_);

   was_analog_level_set_ = true;
   if (level < minimum_capture_level_ || level > maximum_capture_level_) {
     return AudioProcessing::kBadParameterError;
   }
   analog_capture_level_ = level;

   return AudioProcessing::kNoError;
 }

 int GainControlImpl::stream_analog_level() {
   rtc::CritScope cs(crit_capture_);
   // TODO(ajm): enable this assertion?
   //assert(mode_ == kAdaptiveAnalog);

   return analog_capture_level_;
 }

 int GainControlImpl::Enable(bool enable) {
   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);
   return EnableComponent(enable);
 }

 bool GainControlImpl::is_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return is_component_enabled();
 }

 int GainControlImpl::set_mode(Mode mode) {
   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);
   if (MapSetting(mode) == -1) {
     return AudioProcessing::kBadParameterError;
   }

   mode_ = mode;
   return Initialize();
 }

 GainControl::Mode GainControlImpl::mode() const {
   rtc::CritScope cs(crit_capture_);
   return mode_;
 }

 int GainControlImpl::set_analog_level_limits(int minimum,
                                              int maximum) {
   rtc::CritScope cs(crit_capture_);
   if (minimum < 0) {
     return AudioProcessing::kBadParameterError;
   }

   if (maximum > 65535) {
     return AudioProcessing::kBadParameterError;
   }

   if (maximum < minimum) {
     return AudioProcessing::kBadParameterError;
   }

   minimum_capture_level_ = minimum;
   maximum_capture_level_ = maximum;

   return Initialize();
 }

 int GainControlImpl::analog_level_minimum() const {
   rtc::CritScope cs(crit_capture_);
   return minimum_capture_level_;
 }

 int GainControlImpl::analog_level_maximum() const {
   rtc::CritScope cs(crit_capture_);
   return maximum_capture_level_;
 }

 bool GainControlImpl::stream_is_saturated() const {
   rtc::CritScope cs(crit_capture_);
   return stream_is_saturated_;
 }

 int GainControlImpl::set_target_level_dbfs(int level) {
   rtc::CritScope cs(crit_capture_);
   if (level > 31 || level < 0) {
     return AudioProcessing::kBadParameterError;
   }

   target_level_dbfs_ = level;
   return Configure();
 }

 int GainControlImpl::target_level_dbfs() const {
   rtc::CritScope cs(crit_capture_);
   return target_level_dbfs_;
 }

 int GainControlImpl::set_compression_gain_db(int gain) {
   rtc::CritScope cs(crit_capture_);
   if (gain < 0 || gain > 90) {
     return AudioProcessing::kBadParameterError;
   }

   compression_gain_db_ = gain;
   return Configure();
 }

 int GainControlImpl::compression_gain_db() const {
   rtc::CritScope cs(crit_capture_);
   return compression_gain_db_;
 }

 int GainControlImpl::enable_limiter(bool enable) {
   rtc::CritScope cs(crit_capture_);
   limiter_enabled_ = enable;
   return Configure();
 }

 bool GainControlImpl::is_limiter_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return limiter_enabled_;
 }

 int GainControlImpl::Initialize() {
   int err = ProcessingComponent::Initialize();
   if (err != AudioProcessing::kNoError || !is_component_enabled()) {
     return err;
   }

   AllocateRenderQueue();

   rtc::CritScope cs_capture(crit_capture_);
   const int n = num_handles();
   RTC_CHECK_GE(n, 0) << "Bad number of handles: " << n;

   capture_levels_.assign(n, analog_capture_level_);
   return AudioProcessing::kNoError;
 }

 void GainControlImpl::AllocateRenderQueue() {
   const size_t new_render_queue_element_max_size =
       std::max<size_t>(static_cast<size_t>(1),
                        kMaxAllowedValuesOfSamplesPerFrame * num_handles());

   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);

   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_);

     render_signal_queue_.reset(
         new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
             kMaxNumFramesToBuffer, template_queue_element,
             RenderQueueItemVerifier<int16_t>(render_queue_element_max_size_)));

     render_queue_buffer_.resize(render_queue_element_max_size_);
     capture_queue_buffer_.resize(render_queue_element_max_size_);
   } else {
     render_signal_queue_->Clear();
   }
 }

 void* GainControlImpl::CreateHandle() const {
   return WebRtcAgc_Create();
 }

 void GainControlImpl::DestroyHandle(void* handle) const {
   WebRtcAgc_Free(static_cast<Handle*>(handle));
 }

 int GainControlImpl::InitializeHandle(void* handle) const {
   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);

   return WebRtcAgc_Init(static_cast<Handle*>(handle),
                           minimum_capture_level_,
                           maximum_capture_level_,
                           MapSetting(mode_),
                           apm_->proc_sample_rate_hz());
 }

 int GainControlImpl::ConfigureHandle(void* handle) const {
   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);
   WebRtcAgcConfig config;
   // TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
   //            change the interface.
   //assert(target_level_dbfs_ <= 0);
   //config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_);
   config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_);
   config.compressionGaindB =
       static_cast<int16_t>(compression_gain_db_);
   config.limiterEnable = limiter_enabled_;

   return WebRtcAgc_set_config(static_cast<Handle*>(handle), config);
 }

 int GainControlImpl::num_handles_required() const {
   // Not locked as it only relies on APM public API which is threadsafe.
   return apm_->num_output_channels();
 }

 int GainControlImpl::GetHandleError(void* handle) const {
   // The AGC has no get_error() function.
   // (Despite listing errors in its interface...)
   assert(handle != NULL);
   return AudioProcessing::kUnspecifiedError;
 }
 }  // namespace webrtc
	/*
	* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	#include "webrtc/modules/audio_processing/gain_control_impl.h"

	#include <assert.h>

	#include "webrtc/modules/audio_processing/audio_buffer.h"
	#include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"

	namespace webrtc {

	typedef void Handle;

	namespace {
	int16_t MapSetting(GainControl::Mode mode) {
	switch (mode) {
	case GainControl::kAdaptiveAnalog:
	return kAgcModeAdaptiveAnalog;
	case GainControl::kAdaptiveDigital:
	return kAgcModeAdaptiveDigital;
	case GainControl::kFixedDigital:
	return kAgcModeFixedDigital;
	}
	assert(false);
	return -1;
	}

	// Maximum length that a frame of samples can have.
	static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
	// Maximum number of frames to buffer in the render queue.
	// TODO(peah): Decrease this once we properly handle hugely unbalanced
	// reverse and forward call numbers.
	static const size_t kMaxNumFramesToBuffer = 100;

	} // namespace

	GainControlImpl::GainControlImpl(const AudioProcessing* apm,
	rtc::CriticalSection* crit_render,
	rtc::CriticalSection* crit_capture)
	: ProcessingComponent(),
	apm_(apm),
	crit_render_(crit_render),
	crit_capture_(crit_capture),
	mode_(kAdaptiveAnalog),
	minimum_capture_level_(0),
	maximum_capture_level_(255),
	limiter_enabled_(true),
	target_level_dbfs_(3),
	compression_gain_db_(9),
	analog_capture_level_(0),
	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);
	}

	GainControlImpl::~GainControlImpl() {}

	int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
	rtc::CritScope cs(crit_render_);
	if (!is_component_enabled()) {
	return AudioProcessing::kNoError;
	}

	assert(audio->num_frames_per_band() <= 160);

	render_queue_buffer_.resize(0);
	for (int i = 0; i < num_handles(); i++) {
	Handle* my_handle = static_cast<Handle*>(handle(i));
	int err =
	WebRtcAgc_GetAddFarendError(my_handle, audio->num_frames_per_band());

	if (err != AudioProcessing::kNoError)
	return GetHandleError(my_handle);

	// Buffer the samples in the render queue.
	render_queue_buffer_.insert(
	render_queue_buffer_.end(), audio->mixed_low_pass_data(),
	(audio->mixed_low_pass_data() + audio->num_frames_per_band()));
	}

	// Insert the samples into the queue.
	if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
	// The data queue is full and needs to be emptied.
	ReadQueuedRenderData();

	// Retry the insert (should always work).
	RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
	}

	return AudioProcessing::kNoError;
	}

	// Read chunks of data that were received and queued on the render side from
	// a queue. All the data chunks are buffered into the farend signal of the AGC.
	void GainControlImpl::ReadQueuedRenderData() {
	rtc::CritScope cs(crit_capture_);

	if (!is_component_enabled()) {
	return;
	}

	while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
	size_t buffer_index = 0;
	const size_t num_frames_per_band =
	capture_queue_buffer_.size() / num_handles();
	for (int i = 0; i < num_handles(); i++) {
	Handle* my_handle = static_cast<Handle*>(handle(i));
	WebRtcAgc_AddFarend(my_handle, &capture_queue_buffer_[buffer_index],
	num_frames_per_band);

	buffer_index += num_frames_per_band;
	}
	}
	}

	int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
	rtc::CritScope cs(crit_capture_);

	if (!is_component_enabled()) {
	return AudioProcessing::kNoError;
	}

	assert(audio->num_frames_per_band() <= 160);
	assert(audio->num_channels() == num_handles());

	int err = AudioProcessing::kNoError;

	if (mode_ == kAdaptiveAnalog) {
	capture_levels_.assign(num_handles(), analog_capture_level_);
	for (int i = 0; i < num_handles(); i++) {
	Handle* my_handle = static_cast<Handle*>(handle(i));
	err = WebRtcAgc_AddMic(
	my_handle,
	audio->split_bands(i),
	audio->num_bands(),
	audio->num_frames_per_band());

	if (err != AudioProcessing::kNoError) {
	return GetHandleError(my_handle);
	}
	}
	} else if (mode_ == kAdaptiveDigital) {

	for (int i = 0; i < num_handles(); i++) {
	Handle* my_handle = static_cast<Handle*>(handle(i));
	int32_t capture_level_out = 0;

	err = WebRtcAgc_VirtualMic(
	my_handle,
	audio->split_bands(i),
	audio->num_bands(),
	audio->num_frames_per_band(),
	analog_capture_level_,
	&capture_level_out);

	capture_levels_[i] = capture_level_out;

	if (err != AudioProcessing::kNoError) {
	return GetHandleError(my_handle);
	}

	}
	}

	return AudioProcessing::kNoError;
	}

	int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
	rtc::CritScope cs(crit_capture_);

	if (!is_component_enabled()) {
	return AudioProcessing::kNoError;
	}

	if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
	return AudioProcessing::kStreamParameterNotSetError;
	}

	assert(audio->num_frames_per_band() <= 160);
	assert(audio->num_channels() == num_handles());

	stream_is_saturated_ = false;
	for (int i = 0; i < num_handles(); i++) {
	Handle* my_handle = static_cast<Handle*>(handle(i));
	int32_t capture_level_out = 0;
	uint8_t saturation_warning = 0;

	// The call to stream_has_echo() is ok from a deadlock perspective
	// as the capture lock is allready held.
	int err = WebRtcAgc_Process(
	my_handle,
	audio->split_bands_const(i),
	audio->num_bands(),
	audio->num_frames_per_band(),
	audio->split_bands(i),
	capture_levels_[i],
	&capture_level_out,
	apm_->echo_cancellation()->stream_has_echo(),
	&saturation_warning);

	if (err != AudioProcessing::kNoError) {
	return GetHandleError(my_handle);
	}

	capture_levels_[i] = capture_level_out;
	if (saturation_warning == 1) {
	stream_is_saturated_ = true;
	}
	}

	if (mode_ == kAdaptiveAnalog) {
	// Take the analog level to be the average across the handles.
	analog_capture_level_ = 0;
	for (int i = 0; i < num_handles(); i++) {
	analog_capture_level_ += capture_levels_[i];
	}

	analog_capture_level_ /= num_handles();
	}

	was_analog_level_set_ = false;
	return AudioProcessing::kNoError;
	}

	// TODO(ajm): ensure this is called under kAdaptiveAnalog.
	int GainControlImpl::set_stream_analog_level(int level) {
	rtc::CritScope cs(crit_capture_);

	was_analog_level_set_ = true;
	if (level < minimum_capture_level_ \|\| level > maximum_capture_level_) {
	return AudioProcessing::kBadParameterError;
	}
	analog_capture_level_ = level;

	return AudioProcessing::kNoError;
	}

	int GainControlImpl::stream_analog_level() {
	rtc::CritScope cs(crit_capture_);
	// TODO(ajm): enable this assertion?
	//assert(mode_ == kAdaptiveAnalog);

	return analog_capture_level_;
	}

	int GainControlImpl::Enable(bool enable) {
	rtc::CritScope cs_render(crit_render_);
	rtc::CritScope cs_capture(crit_capture_);
	return EnableComponent(enable);
	}

	bool GainControlImpl::is_enabled() const {
	rtc::CritScope cs(crit_capture_);
	return is_component_enabled();
	}

	int GainControlImpl::set_mode(Mode mode) {
	rtc::CritScope cs_render(crit_render_);
	rtc::CritScope cs_capture(crit_capture_);
	if (MapSetting(mode) == -1) {
	return AudioProcessing::kBadParameterError;
	}

	mode_ = mode;
	return Initialize();
	}

	GainControl::Mode GainControlImpl::mode() const {
	rtc::CritScope cs(crit_capture_);
	return mode_;
	}

	int GainControlImpl::set_analog_level_limits(int minimum,
	int maximum) {
	rtc::CritScope cs(crit_capture_);
	if (minimum < 0) {
	return AudioProcessing::kBadParameterError;
	}

	if (maximum > 65535) {
	return AudioProcessing::kBadParameterError;
	}

	if (maximum < minimum) {
	return AudioProcessing::kBadParameterError;
	}

	minimum_capture_level_ = minimum;
	maximum_capture_level_ = maximum;

	return Initialize();
	}

	int GainControlImpl::analog_level_minimum() const {
	rtc::CritScope cs(crit_capture_);
	return minimum_capture_level_;
	}

	int GainControlImpl::analog_level_maximum() const {
	rtc::CritScope cs(crit_capture_);
	return maximum_capture_level_;
	}

	bool GainControlImpl::stream_is_saturated() const {
	rtc::CritScope cs(crit_capture_);
	return stream_is_saturated_;
	}

	int GainControlImpl::set_target_level_dbfs(int level) {
	rtc::CritScope cs(crit_capture_);
	if (level > 31 \|\| level < 0) {
	return AudioProcessing::kBadParameterError;
	}

	target_level_dbfs_ = level;
	return Configure();
	}

	int GainControlImpl::target_level_dbfs() const {
	rtc::CritScope cs(crit_capture_);
	return target_level_dbfs_;
	}

	int GainControlImpl::set_compression_gain_db(int gain) {
	rtc::CritScope cs(crit_capture_);
	if (gain < 0 \|\| gain > 90) {
	return AudioProcessing::kBadParameterError;
	}

	compression_gain_db_ = gain;
	return Configure();
	}

	int GainControlImpl::compression_gain_db() const {
	rtc::CritScope cs(crit_capture_);
	return compression_gain_db_;
	}

	int GainControlImpl::enable_limiter(bool enable) {
	rtc::CritScope cs(crit_capture_);
	limiter_enabled_ = enable;
	return Configure();
	}

	bool GainControlImpl::is_limiter_enabled() const {
	rtc::CritScope cs(crit_capture_);
	return limiter_enabled_;
	}

	int GainControlImpl::Initialize() {
	int err = ProcessingComponent::Initialize();
	if (err != AudioProcessing::kNoError \|\| !is_component_enabled()) {
	return err;
	}

	AllocateRenderQueue();

	rtc::CritScope cs_capture(crit_capture_);
	const int n = num_handles();
	RTC_CHECK_GE(n, 0) << "Bad number of handles: " << n;

	capture_levels_.assign(n, analog_capture_level_);
	return AudioProcessing::kNoError;
	}

	void GainControlImpl::AllocateRenderQueue() {
	const size_t new_render_queue_element_max_size =
	std::max<size_t>(static_cast<size_t>(1),
	kMaxAllowedValuesOfSamplesPerFrame * num_handles());

	rtc::CritScope cs_render(crit_render_);
	rtc::CritScope cs_capture(crit_capture_);

	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_);

	render_signal_queue_.reset(
	new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
	kMaxNumFramesToBuffer, template_queue_element,
	RenderQueueItemVerifier<int16_t>(render_queue_element_max_size_)));

	render_queue_buffer_.resize(render_queue_element_max_size_);
	capture_queue_buffer_.resize(render_queue_element_max_size_);
	} else {
	render_signal_queue_->Clear();
	}
	}

	void* GainControlImpl::CreateHandle() const {
	return WebRtcAgc_Create();
	}

	void GainControlImpl::DestroyHandle(void* handle) const {
	WebRtcAgc_Free(static_cast<Handle*>(handle));
	}

	int GainControlImpl::InitializeHandle(void* handle) const {
	rtc::CritScope cs_render(crit_render_);
	rtc::CritScope cs_capture(crit_capture_);

	return WebRtcAgc_Init(static_cast<Handle*>(handle),
	minimum_capture_level_,
	maximum_capture_level_,
	MapSetting(mode_),
	apm_->proc_sample_rate_hz());
	}

	int GainControlImpl::ConfigureHandle(void* handle) const {
	rtc::CritScope cs_render(crit_render_);
	rtc::CritScope cs_capture(crit_capture_);
	WebRtcAgcConfig config;
	// TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
	// change the interface.
	//assert(target_level_dbfs_ <= 0);
	//config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_);
	config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_);
	config.compressionGaindB =
	static_cast<int16_t>(compression_gain_db_);
	config.limiterEnable = limiter_enabled_;

	return WebRtcAgc_set_config(static_cast<Handle*>(handle), config);
	}

	int GainControlImpl::num_handles_required() const {
	// Not locked as it only relies on APM public API which is threadsafe.
	return apm_->num_output_channels();
	}

	int GainControlImpl::GetHandleError(void* handle) const {
	// The AGC has no get_error() function.
	// (Despite listing errors in its interface...)
	assert(handle != NULL);
	return AudioProcessing::kUnspecifiedError;
	}
	} // namespace webrtc