Revert of Reland "Move smoothing filter to common audio". (patchset #5 id:100001 of https://codereview.webrtc.org/2520003005/ )

Reason for revert:
Internal bots failed.

Original issue's description:
> Reland "Move smoothing filter to common audio".
>
> The original CL was this https://codereview.webrtc.org/2484153002/
>
> Due to failure with internal trial servers, it was reverted. This CL tries to reland it.
>
> BUG=webrtc:6443
>
> Committed: https://crrev.com/223641f1b903e41e77d88f03199b4cdb65255ec8
> Cr-Commit-Position: refs/heads/master@{#15227}

TBR=tommi@webrtc.org,solenberg@webrtc.org,terelius@webrtc.org,kjellander@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/2529943002
Cr-Commit-Position: refs/heads/master@{#15228}
This commit is contained in:
minyue 2016-11-24 07:13:18 -08:00 committed by Commit bot
parent 223641f1b9
commit 3c3aef44de
24 changed files with 37 additions and 52 deletions

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@ -363,7 +363,6 @@ if (rtc_include_tests) {
testonly = true
sources = [
"api/fakemetricsobserver.cc",
"base/analytics/exp_filter_unittest.cc",
"base/array_view_unittest.cc",
"base/atomicops_unittest.cc",
"base/autodetectproxy_unittest.cc",
@ -381,6 +380,7 @@ if (rtc_include_tests) {
"base/criticalsection_unittest.cc",
"base/event_tracer_unittest.cc",
"base/event_unittest.cc",
"base/exp_filter_unittest.cc",
"base/file_unittest.cc",
"base/filerotatingstream_unittest.cc",
"base/fileutils_unittest.cc",
@ -511,7 +511,6 @@ if (rtc_include_tests) {
"base:rtc_base",
"base:rtc_base_tests_utils",
"base:rtc_task_queue",
"base/analytics:rtc_analytics",
"p2p:libstunprober",
"p2p:rtc_p2p",
"//testing/gmock",

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@ -124,6 +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",

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@ -1,16 +0,0 @@
# 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.
import("../../build/webrtc.gni")
rtc_static_library("rtc_analytics") {
sources = [
"exp_filter.cc",
"exp_filter.h",
]
}

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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/base/analytics/exp_filter.h"
#include "webrtc/base/exp_filter.h"
#include <math.h>

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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_BASE_ANALYTICS_EXP_FILTER_H_
#define WEBRTC_BASE_ANALYTICS_EXP_FILTER_H_
#ifndef WEBRTC_BASE_EXP_FILTER_H_
#define WEBRTC_BASE_EXP_FILTER_H_
namespace rtc {
@ -46,4 +46,4 @@ class ExpFilter {
};
} // namespace rtc
#endif // WEBRTC_BASE_ANALYTICS_EXP_FILTER_H_
#endif // WEBRTC_BASE_EXP_FILTER_H_

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@ -10,8 +10,8 @@
#include <math.h>
#include "webrtc/base/analytics/exp_filter.h"
#include "webrtc/test/gtest.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/exp_filter.h"
namespace rtc {

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@ -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/analytics:rtc_analytics",
"../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",

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@ -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",

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@ -709,7 +709,6 @@ rtc_static_library("webrtc_opus") {
":audio_encoder_interface",
":audio_network_adaptor",
"../../base:rtc_base_approved",
"../../base/analytics:rtc_analytics",
]
defines = []
@ -765,6 +764,8 @@ 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 = [

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 {

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(

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@ -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_

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@ -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 {

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@ -8,11 +8,11 @@
* 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/analytics/exp_filter.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/exp_filter.h"
#include "webrtc/base/optional.h"
#include "webrtc/system_wrappers/include/clock.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_

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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 {

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@ -13,8 +13,8 @@
#include <algorithm>
#include <iterator>
#include "webrtc/base/analytics/exp_filter.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/exp_filter.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"

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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 {

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@ -93,7 +93,6 @@ rtc_static_library("video_coding") {
":webrtc_vp8",
":webrtc_vp9",
"../..:webrtc_common",
"../../base/analytics:rtc_analytics",
"../../common_video",
"../../system_wrappers",
]
@ -125,7 +124,6 @@ rtc_static_library("video_coding_utility") {
}
deps = [
"../../base/analytics:rtc_analytics",
"../../common_video",
"../../system_wrappers",
]

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@ -16,7 +16,7 @@
#include <memory>
#include "webrtc/base/analytics/exp_filter.h"
#include "webrtc/base/exp_filter.h"
#include "webrtc/modules/video_coding/internal_defines.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/typedefs.h"

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@ -13,7 +13,7 @@
#include <cstddef>
#include "webrtc/base/analytics/exp_filter.h"
#include "webrtc/base/exp_filter.h"
#include "webrtc/typedefs.h"
namespace webrtc {

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@ -58,7 +58,6 @@ rtc_static_library("video") {
"..:webrtc_common",
"../base:rtc_base_approved",
"../base:rtc_task_queue",
"../base/analytics:rtc_analytics",
"../common_video",
"../logging:rtc_event_log_api",
"../modules/bitrate_controller",

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@ -17,8 +17,8 @@
#include <list>
#include <map>
#include "webrtc/base/analytics/exp_filter.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/exp_filter.h"
#include "webrtc/base/logging.h"
#include "webrtc/common_video/include/frame_callback.h"
#include "webrtc/system_wrappers/include/clock.h"

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@ -14,9 +14,9 @@
#include <list>
#include <memory>
#include "webrtc/base/analytics/exp_filter.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/optional.h"
#include "webrtc/base/exp_filter.h"
#include "webrtc/base/sequenced_task_checker.h"
#include "webrtc/base/task_queue.h"
#include "webrtc/base/thread_annotations.h"

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@ -16,8 +16,8 @@
#include <string>
#include <vector>
#include "webrtc/base/analytics/exp_filter.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/exp_filter.h"
#include "webrtc/base/ratetracker.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"