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Move smoothing filter to common audio.

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This will make the smoothing filter a basic tool that is going to be used by both voice engine and ANA.

BUG=webrtc:6443

Review-Url: https://codereview.webrtc.org/2484153002
Cr-Commit-Position: refs/heads/master@{#15146}
This commit is contained in:
michaelt 2016-11-18 00:23:19 -08:00 committed by Commit bot
parent 610c454cf9
commit a82395bf7c
12 changed files with 26 additions and 23 deletions
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1
webrtc/BUILD.gn
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@ -536,6 +536,7 @@ if (rtc_include_tests) {
deps = [
"base:rtc_base",
"base:rtc_base_tests_utils",
"base:rtc_exp_filter",
"base:rtc_task_queue",
"p2p:libstunprober",
"p2p:rtc_p2p",

10
webrtc/base/BUILD.gn
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@ -22,6 +22,7 @@ group("base") {
public_deps = [
":rtc_base",
":rtc_base_approved",
":rtc_exp_filter",
":rtc_task_queue",
]
if (is_android) {
@ -124,8 +125,6 @@ rtc_static_library("rtc_base_approved") {
"event.h",
"event_tracer.cc",
"event_tracer.h",
"exp_filter.cc",
"exp_filter.h",
"file.cc",
"file.h",
"format_macros.h",
@ -338,6 +337,13 @@ rtc_static_library("rtc_task_queue") {
}
}
rtc_static_library("rtc_exp_filter") {
sources = [
"exp_filter.cc",
"exp_filter.h",
]
}
config("rtc_base_warnings_config") {
if (is_win && is_clang) {
cflags = [

4
webrtc/common_audio/BUILD.gn
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@ -86,6 +86,8 @@ rtc_static_library("common_audio") {
"signal_processing/splitting_filter.c",
"signal_processing/sqrt_of_one_minus_x_squared.c",
"signal_processing/vector_scaling_operations.c",
"smoothing_filter.cc",
"smoothing_filter.h",
"sparse_fir_filter.cc",
"sparse_fir_filter.h",
"vad/include/vad.h",
@ -109,6 +111,7 @@ rtc_static_library("common_audio") {
]
deps = [
"../base:rtc_exp_filter",
"../system_wrappers",
]
@ -257,6 +260,7 @@ if (rtc_include_tests) {
"ring_buffer_unittest.cc",
"signal_processing/real_fft_unittest.cc",
"signal_processing/signal_processing_unittest.cc",
"smoothing_filter_unittest.cc",
"sparse_fir_filter_unittest.cc",
"vad/vad_core_unittest.cc",
"vad/vad_filterbank_unittest.cc",

10
webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_smoothing_filter.h → webrtc/common_audio/mocks/mock_smoothing_filter.h
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@ -8,22 +8,20 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_SMOOTHING_FILTER_H_
#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_SMOOTHING_FILTER_H_
#ifndef WEBRTC_COMMON_AUDIO_MOCKS_MOCK_SMOOTHING_FILTER_H_
#define WEBRTC_COMMON_AUDIO_MOCKS_MOCK_SMOOTHING_FILTER_H_
#include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
#include "webrtc/common_audio/smoothing_filter.h"
#include "webrtc/test/gmock.h"
namespace webrtc {
class MockSmoothingFilter : public SmoothingFilter {
public:
virtual ~MockSmoothingFilter() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_METHOD1(AddSample, void(float));
MOCK_CONST_METHOD0(GetAverage, rtc::Optional<float>());
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_SMOOTHING_FILTER_H_
#endif // WEBRTC_COMMON_AUDIO_MOCKS_MOCK_SMOOTHING_FILTER_H_

2
webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.cc → webrtc/common_audio/smoothing_filter.cc
View File

@ -10,7 +10,7 @@
#include <cmath>
#include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
#include "webrtc/common_audio/smoothing_filter.h"
namespace webrtc {

8
webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h → webrtc/common_audio/smoothing_filter.h
View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_SMOOTHING_FILTER_H_
#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_SMOOTHING_FILTER_H_
#ifndef WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_
#define WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/exp_filter.h"
@ -46,9 +46,9 @@ class SmoothingFilterImpl final : public SmoothingFilter {
int64_t last_sample_time_ms_;
rtc::ExpFilter filter_;
RTC_DISALLOW_COPY_AND_ASSIGN(SmoothingFilterImpl);
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SmoothingFilterImpl);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_SMOOTHING_FILTER_H_
#endif // WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_

2
webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc → webrtc/common_audio/smoothing_filter_unittest.cc
View File

@ -10,7 +10,7 @@
#include <memory>
#include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
#include "webrtc/common_audio/smoothing_filter.h"
#include "webrtc/test/gtest.h"
namespace webrtc {

1
webrtc/modules/BUILD.gn
View File

@ -264,7 +264,6 @@ if (rtc_include_tests) {
"audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc",
"audio_coding/audio_network_adaptor/mock/mock_controller.h",
"audio_coding/audio_network_adaptor/mock/mock_controller_manager.h",
"audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc",
]
deps = [
"audio_coding:audio_network_adaptor",

2
webrtc/modules/audio_coding/BUILD.gn
View File

@ -757,8 +757,6 @@ rtc_static_library("audio_network_adaptor") {
"audio_network_adaptor/frame_length_controller.cc",
"audio_network_adaptor/frame_length_controller.h",
"audio_network_adaptor/include/audio_network_adaptor.h",
"audio_network_adaptor/smoothing_filter.cc",
"audio_network_adaptor/smoothing_filter.h",
]
deps = [

2
webrtc/modules/audio_coding/audio_network_adaptor/fec_controller.h
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@ -14,8 +14,8 @@
#include <memory>
#include "webrtc/base/constructormagic.h"
#include "webrtc/common_audio/smoothing_filter.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
namespace webrtc {

5
webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_unittest.cc
View File

@ -10,8 +10,8 @@
#include <utility>
#include "webrtc/common_audio/mocks/mock_smoothing_filter.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_smoothing_filter.h"
#include "webrtc/test/gtest.h"
namespace webrtc {
@ -55,7 +55,6 @@ FecControllerStates CreateFecController(bool initial_fec_enabled) {
std::unique_ptr<MockSmoothingFilter> mock_smoothing_filter(
new NiceMock<MockSmoothingFilter>());
states.packet_loss_smoothed = mock_smoothing_filter.get();
EXPECT_CALL(*states.packet_loss_smoothed, Die());
using Threshold = FecController::Config::Threshold;
states.controller.reset(new FecController(
FecController::Config(
@ -262,7 +261,6 @@ TEST(FecControllerTest, CheckBehaviorOnSpecialCurves) {
std::unique_ptr<MockSmoothingFilter> mock_smoothing_filter(
new NiceMock<MockSmoothingFilter>());
states.packet_loss_smoothed = mock_smoothing_filter.get();
EXPECT_CALL(*states.packet_loss_smoothed, Die());
using Threshold = FecController::Config::Threshold;
states.controller.reset(new FecController(
FecController::Config(
@ -293,7 +291,6 @@ TEST(FecControllerDeathTest, InvalidConfig) {
std::unique_ptr<MockSmoothingFilter> mock_smoothing_filter(
new NiceMock<MockSmoothingFilter>());
states.packet_loss_smoothed = mock_smoothing_filter.get();
EXPECT_CALL(*states.packet_loss_smoothed, Die());
using Threshold = FecController::Config::Threshold;
EXPECT_DEATH(
states.controller.reset(new FecController(

2
webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
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@ -9,8 +9,8 @@
*/
#include <memory>
#include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
#include "webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/gtest.h"
namespace webrtc {