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Revert of Move smoothing filter to common audio. (patchset #3 id:60001 of https://codereview.webrtc.org/2484153002/ )

Browse Source 
Reason for revert:
Breaks downstream projects:
error: undefined reference to 'rtc::ExpFilter::kValueUndefined'
error: undefined reference to 'rtc::ExpFilter::Apply(float, float)'
error: undefined reference to 'rtc::ExpFilter::Reset(float)'
rror: undefined reference to 'rtc::ExpFilter::UpdateBase(float)'

Original issue's description:
> Move smoothing filter to common audio.
>
> This will make the smoothing filter a basic tool that is going to be used by both voice engine and ANA.
>
> BUG=webrtc:6443
>
> Committed: https://crrev.com/a82395bf7cd15b7396456df06fe952ede8db0c39
> Cr-Commit-Position: refs/heads/master@{#15146}

TBR=minyue@webrtc.org,solenberg@webrtc.org,perkj@webrtc.org,tommi@webrtc.org,michaelt@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6443

Review-Url: https://codereview.webrtc.org/2510373002
Cr-Commit-Position: refs/heads/master@{#15147}
This commit is contained in:
magjed 2016-11-18 01:31:16 -08:00 committed by Commit bot
parent a82395bf7c
commit d7ac0a9bcc
12 changed files with 23 additions and 26 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
webrtc/BUILD.gn
View File

@ -536,7 +536,6 @@ 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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@ -22,7 +22,6 @@ group("base") {
public_deps = [
":rtc_base",
":rtc_base_approved",
":rtc_exp_filter",
":rtc_task_queue",
]
if (is_android) {
@ -125,6 +124,8 @@ 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",
@ -337,13 +338,6 @@ 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
View File

@ -86,8 +86,6 @@ 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",
@ -111,7 +109,6 @@ rtc_static_library("common_audio") {
]
deps = [
"../base:rtc_exp_filter",
"../system_wrappers",
]
@ -260,7 +257,6 @@ 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",

1
webrtc/modules/BUILD.gn
View File

@ -264,6 +264,7 @@ 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

@ -757,6 +757,8 @@ 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
View File

@ -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 {
@ -55,6 +55,7 @@ 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(
@ -261,6 +262,7 @@ 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(
@ -291,6 +293,7 @@ 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(

10
webrtc/common_audio/mocks/mock_smoothing_filter.h → webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_smoothing_filter.h
View File

@ -8,20 +8,22 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_COMMON_AUDIO_MOCKS_MOCK_SMOOTHING_FILTER_H_
#define WEBRTC_COMMON_AUDIO_MOCKS_MOCK_SMOOTHING_FILTER_H_
#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_
#include "webrtc/common_audio/smoothing_filter.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/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_COMMON_AUDIO_MOCKS_MOCK_SMOOTHING_FILTER_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_SMOOTHING_FILTER_H_

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

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

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

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_
#define WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_
#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_SMOOTHING_FILTER_H_
#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_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_IMPLICIT_CONSTRUCTORS(SmoothingFilterImpl);
RTC_DISALLOW_COPY_AND_ASSIGN(SmoothingFilterImpl);
};
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_SMOOTHING_FILTER_H_

2
webrtc/common_audio/smoothing_filter_unittest.cc → webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc
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@ -10,7 +10,7 @@
#include <memory>
#include "webrtc/common_audio/smoothing_filter.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/smoothing_filter.h"
#include "webrtc/test/gtest.h"
namespace webrtc {

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 {