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Removed the ProcessingComponent class

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BUG=

Review URL: https://codereview.webrtc.org/1772553002

Cr-Commit-Position: refs/heads/master@{#11950}
This commit is contained in:
peah
2016-03-10 23:05:28 -08:00
committed by Commit bot
parent bfa971198d
commit 737f4b8d12
10 changed files with 42 additions and 215 deletions
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3
webrtc/modules/audio_processing/BUILD.gn
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@ -86,8 +86,7 @@ source_set("audio_processing") {
"logging/aec_logging_file_handling.h",
"noise_suppression_impl.cc",
"noise_suppression_impl.h",
"processing_component.cc",
"processing_component.h",
"render_queue_item_verifier.h",
"rms_level.cc",
"rms_level.h",
"splitting_filter.cc",

3
webrtc/modules/audio_processing/audio_processing.gypi
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@ -96,8 +96,7 @@
'logging/aec_logging_file_handling.h',
'noise_suppression_impl.cc',
'noise_suppression_impl.h',
'processing_component.cc',
'processing_component.h',
'render_queue_item_verifier.h',
'rms_level.cc',
'rms_level.h',
'splitting_filter.cc',

27
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -33,7 +33,6 @@
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
#include "webrtc/modules/audio_processing/level_estimator_impl.h"
#include "webrtc/modules/audio_processing/noise_suppression_impl.h"
#include "webrtc/modules/audio_processing/processing_component.h"
#include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
#include "webrtc/modules/audio_processing/voice_detection_impl.h"
#include "webrtc/modules/include/module_common_types.h"
@ -101,7 +100,6 @@ struct AudioProcessingImpl::ApmPrivateSubmodules {
explicit ApmPrivateSubmodules(Beamformer<float>* beamformer)
: beamformer(beamformer) {}
// Accessed internally from capture or during initialization
std::list<ProcessingComponent*> component_list;
std::unique_ptr<Beamformer<float>> beamformer;
std::unique_ptr<AgcManagerDirect> agc_manager;
};
@ -197,13 +195,6 @@ AudioProcessingImpl::~AudioProcessingImpl() {
private_submodules_->agc_manager.reset();
// Depends on gain_control_.
public_submodules_->gain_control_for_experimental_agc.reset();
while (!private_submodules_->component_list.empty()) {
ProcessingComponent* component =
private_submodules_->component_list.front();
component->Destroy();
delete component;
private_submodules_->component_list.pop_front();
}
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_dump_.debug_file->Open()) {
@ -308,14 +299,6 @@ int AudioProcessingImpl::InitializeLocked() {
fwd_audio_buffer_channels,
formats_.api_format.output_stream().num_frames()));
// Initialize all components.
for (auto item : private_submodules_->component_list) {
int err = item->Initialize();
if (err != kNoError) {
return err;
}
}
InitializeGainController();
InitializeEchoCanceller();
InitializeEchoControlMobile();
@ -416,9 +399,6 @@ void AudioProcessingImpl::SetExtraOptions(const Config& config) {
// Run in a single-threaded manner when setting the extra options.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
for (auto item : private_submodules_->component_list) {
item->SetExtraOptions(config);
}
public_submodules_->echo_cancellation->SetExtraOptions(config);
@ -1131,13 +1111,6 @@ bool AudioProcessingImpl::is_data_processed() const {
return true;
}
// All of the private submodules modify the data.
for (auto item : private_submodules_->component_list) {
if (item->is_component_enabled()) {
return true;
}
}
// The capture data is otherwise unchanged.
return false;
}

2
webrtc/modules/audio_processing/echo_cancellation_impl.h
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@ -17,7 +17,7 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/common_audio/swap_queue.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/processing_component.h"
#include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
namespace webrtc {

2
webrtc/modules/audio_processing/echo_control_mobile_impl.h
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@ -17,7 +17,7 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/common_audio/swap_queue.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/processing_component.h"
#include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
namespace webrtc {

2
webrtc/modules/audio_processing/gain_control_impl.h
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@ -19,7 +19,7 @@
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_audio/swap_queue.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/processing_component.h"
#include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
namespace webrtc {

2
webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
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@ -19,7 +19,7 @@
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/swap_queue.h"
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
#include "webrtc/modules/audio_processing/processing_component.h"
#include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
#include "webrtc/modules/audio_processing/vad/voice_activity_detector.h"
namespace webrtc {

111
webrtc/modules/audio_processing/processing_component.cc
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@ -1,111 +0,0 @@
/*
* Copyright (c) 2011 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/processing_component.h"
#include <assert.h>
#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace webrtc {
ProcessingComponent::ProcessingComponent()
: initialized_(false),
enabled_(false),
num_handles_(0) {}
ProcessingComponent::~ProcessingComponent() {
assert(initialized_ == false);
}
int ProcessingComponent::Destroy() {
while (!handles_.empty()) {
DestroyHandle(handles_.back());
handles_.pop_back();
}
initialized_ = false;
return AudioProcessing::kNoError;
}
int ProcessingComponent::EnableComponent(bool enable) {
if (enable && !enabled_) {
enabled_ = enable; // Must be set before Initialize() is called.
int err = Initialize();
if (err != AudioProcessing::kNoError) {
enabled_ = false;
return err;
}
} else {
enabled_ = enable;
}
return AudioProcessing::kNoError;
}
bool ProcessingComponent::is_component_enabled() const {
return enabled_;
}
void* ProcessingComponent::handle(size_t index) const {
assert(index < num_handles_);
return handles_[index];
}
size_t ProcessingComponent::num_handles() const {
return num_handles_;
}
int ProcessingComponent::Initialize() {
if (!enabled_) {
return AudioProcessing::kNoError;
}
num_handles_ = num_handles_required();
if (num_handles_ > handles_.size()) {
handles_.resize(num_handles_, NULL);
}
assert(handles_.size() >= num_handles_);
for (size_t i = 0; i < num_handles_; i++) {
if (handles_[i] == NULL) {
handles_[i] = CreateHandle();
if (handles_[i] == NULL) {
return AudioProcessing::kCreationFailedError;
}
}
int err = InitializeHandle(handles_[i]);
if (err != AudioProcessing::kNoError) {
return GetHandleError(handles_[i]);
}
}
initialized_ = true;
return Configure();
}
int ProcessingComponent::Configure() {
if (!initialized_) {
return AudioProcessing::kNoError;
}
assert(handles_.size() >= num_handles_);
for (size_t i = 0; i < num_handles_; i++) {
int err = ConfigureHandle(handles_[i]);
if (err != AudioProcessing::kNoError) {
return GetHandleError(handles_[i]);
}
}
return AudioProcessing::kNoError;
}
} // namespace webrtc

69
webrtc/modules/audio_processing/processing_component.h
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@ -1,69 +0,0 @@
/*
* 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_PROCESSING_COMPONENT_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_PROCESSING_COMPONENT_H_
#include <vector>
#include "webrtc/common.h"
namespace webrtc {
// Functor to use when supplying a verifier function for the queue item
// verifcation.
template <typename T>
class RenderQueueItemVerifier {
public:
explicit RenderQueueItemVerifier(size_t minimum_capacity)
: minimum_capacity_(minimum_capacity) {}
bool operator()(const std::vector<T>& v) const {
return v.capacity() >= minimum_capacity_;
}
private:
size_t minimum_capacity_;
};
class ProcessingComponent {
public:
ProcessingComponent();
virtual ~ProcessingComponent();
virtual int Initialize();
virtual void SetExtraOptions(const Config& config) {}
virtual int Destroy();
bool is_component_enabled() const;
protected:
virtual int Configure();
int EnableComponent(bool enable);
void* handle(size_t index) const;
size_t num_handles() const;
private:
virtual void* CreateHandle() const = 0;
virtual int InitializeHandle(void* handle) const = 0;
virtual int ConfigureHandle(void* handle) const = 0;
virtual void DestroyHandle(void* handle) const = 0;
virtual size_t num_handles_required() const = 0;
virtual int GetHandleError(void* handle) const = 0;
std::vector<void*> handles_;
bool initialized_;
bool enabled_;
size_t num_handles_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_PROCESSING_COMPONENT_H__

36
webrtc/modules/audio_processing/render_queue_item_verifier.h Normal file
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@ -0,0 +1,36 @@
/*
* Copyright (c) 2016 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RENDER_QUEUE_ITEM_VERIFIER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_RENDER_QUEUE_ITEM_VERIFIER_H_
#include <vector>
namespace webrtc {
// Functor to use when supplying a verifier function for the queue item
// verifcation.
template <typename T>
class RenderQueueItemVerifier {
public:
explicit RenderQueueItemVerifier(size_t minimum_capacity)
: minimum_capacity_(minimum_capacity) {}
bool operator()(const std::vector<T>& v) const {
return v.capacity() >= minimum_capacity_;
}
private:
size_t minimum_capacity_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_RENDER_QUEUE_ITEM_VERIFIER_H__