Delete deprecated rtc_event_log header

Bug: webrtc:10206
Change-Id: I9ed3148843c647372993729b87c0e74741ab540b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28791}
This commit is contained in:
Danil Chapovalov 2019-08-07 12:24:53 +02:00 committed by Commit Bot
parent e08648dc70
commit 83bbe91398
71 changed files with 103 additions and 128 deletions

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@ -49,6 +49,7 @@ rtc_static_library("audio") {
"../api/audio:audio_frame_api",
"../api/audio:audio_mixer_api",
"../api/audio_codecs:audio_codecs_api",
"../api/rtc_event_log",
"../api/task_queue",
"../call:bitrate_allocator",
"../call:call_interfaces",
@ -56,7 +57,6 @@ rtc_static_library("audio") {
"../common_audio",
"../common_audio:common_audio_c",
"../logging:rtc_event_audio",
"../logging:rtc_event_log_api",
"../logging:rtc_stream_config",
"../modules/audio_coding",
"../modules/audio_coding:audio_coding_module_typedefs",
@ -133,6 +133,7 @@ if (rtc_include_tests) {
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs/opus:audio_decoder_opus",
"../api/audio_codecs/opus:audio_encoder_opus",
"../api/rtc_event_log",
"../api/task_queue:default_task_queue_factory",
"../api/units:time_delta",
"../call:mock_bitrate_allocator",
@ -143,7 +144,6 @@ if (rtc_include_tests) {
"../call:rtp_sender",
"../common_audio",
"../logging:mocks",
"../logging:rtc_event_log_api",
"../modules/audio_device:audio_device_impl", # For TestAudioDeviceModule
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer:audio_mixer_impl",

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@ -22,6 +22,7 @@
#include "api/crypto/frame_encryptor_interface.h"
#include "api/function_view.h"
#include "api/media_transport_config.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "audio/audio_state.h"
#include "audio/channel_send.h"
#include "audio/conversion.h"
@ -29,7 +30,6 @@
#include "call/rtp_transport_controller_send_interface.h"
#include "common_audio/vad/include/vad.h"
#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_processing/include/audio_processing.h"

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@ -18,11 +18,11 @@
#include <vector>
#include "absl/memory/memory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "audio/audio_level.h"
#include "audio/channel_send.h"
#include "audio/utility/audio_frame_operations.h"
#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
#include "modules/audio_device/include/audio_device.h"

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@ -21,10 +21,10 @@
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "audio/utility/audio_frame_operations.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_processing/rms_level.h"

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@ -14,6 +14,7 @@
#include "api/audio_codecs/opus/audio_decoder_opus.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "api/media_transport_config.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/loopback_media_transport.h"
#include "api/test/mock_audio_mixer.h"
@ -21,7 +22,6 @@
#include "audio/audio_send_stream.h"
#include "call/rtp_transport_controller_send.h"
#include "call/test/mock_bitrate_allocator.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_processing/include/mock_audio_processing.h"

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@ -76,9 +76,9 @@ rtc_source_set("rtp_interfaces") {
"../api:fec_controller_api",
"../api:libjingle_peerconnection_api",
"../api:rtp_headers",
"../api/rtc_event_log",
"../api/transport:bitrate_settings",
"../api/units:timestamp",
"../logging:rtc_event_log_api",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/types:optional",
@ -130,6 +130,7 @@ rtc_source_set("rtp_sender") {
"../api:fec_controller_api",
"../api:network_state_predictor_api",
"../api:transport_api",
"../api/rtc_event_log",
"../api/transport:field_trial_based_config",
"../api/transport:goog_cc",
"../api/transport:network_control",
@ -140,7 +141,6 @@ rtc_source_set("rtp_sender") {
"../api/video:video_rtp_headers",
"../api/video_codecs:video_codecs_api",
"../logging:rtc_event_bwe",
"../logging:rtc_event_log_api",
"../modules/congestion_controller",
"../modules/congestion_controller/rtp:control_handler",
"../modules/congestion_controller/rtp:transport_feedback",
@ -228,12 +228,12 @@ rtc_static_library("call") {
"../api:rtp_headers",
"../api:simulated_network_api",
"../api:transport_api",
"../api/rtc_event_log",
"../api/transport:network_control",
"../api/units:time_delta",
"../api/video_codecs:video_codecs_api",
"../audio",
"../logging:rtc_event_audio",
"../logging:rtc_event_log_api",
"../logging:rtc_event_rtp_rtcp",
"../logging:rtc_event_video",
"../logging:rtc_stream_config",
@ -372,11 +372,11 @@ if (rtc_include_tests) {
"../api:rtp_headers",
"../api:transport_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/rtc_event_log",
"../api/task_queue:default_task_queue_factory",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
"../audio",
"../logging:rtc_event_log_api",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer",
"../modules/audio_mixer:audio_mixer_impl",
@ -427,13 +427,13 @@ if (rtc_include_tests) {
"../api:rtc_event_log_output_file",
"../api:simulated_network_api",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/rtc_event_log",
"../api/rtc_event_log:rtc_event_log_factory",
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_bitrate_allocation",
"../api/video_codecs:video_codecs_api",
"../logging:rtc_event_log_api",
"../modules/audio_coding",
"../modules/audio_device",
"../modules/audio_device:audio_device_impl",

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@ -21,6 +21,7 @@
#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/network_control.h"
#include "audio/audio_receive_stream.h"
#include "audio/audio_send_stream.h"
@ -35,7 +36,6 @@
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
#include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
#include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
#include "modules/rtp_rtcp/include/flexfec_receiver.h"

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@ -15,6 +15,7 @@
#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/test/simulated_network.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/video_bitrate_allocation.h"
@ -23,7 +24,6 @@
#include "call/call.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"

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@ -17,13 +17,13 @@
#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/fake_media_transport.h"
#include "api/test/mock_audio_mixer.h"
#include "audio/audio_receive_stream.h"
#include "audio/audio_send_stream.h"
#include "call/audio_state.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "modules/pacing/mock/mock_paced_sender.h"
@ -54,7 +54,7 @@ struct CallHelper {
webrtc::Call* operator->() { return call_.get(); }
private:
webrtc::RtcEventLogNullImpl event_log_;
webrtc::RtcEventLogNull event_log_;
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
std::unique_ptr<webrtc::Call> call_;
};

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@ -16,10 +16,10 @@
#include <utility>
#include <vector>
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/test/simulated_network.h"
#include "call/call.h"
#include "call/simulated_network.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "rtc_base/event.h"
#include "test/call_test.h"

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@ -22,10 +22,10 @@
#include "api/bitrate_constraints.h"
#include "api/crypto/crypto_options.h"
#include "api/fec_controller.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/bitrate_settings.h"
#include "api/units/timestamp.h"
#include "call/rtp_config.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
#include "modules/rtp_rtcp/include/rtp_packet_sender.h"

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@ -21,12 +21,12 @@
#include "api/call/transport.h"
#include "api/fec_controller.h"
#include "api/fec_controller_override.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/video_codecs/video_encoder.h"
#include "call/rtp_config.h"
#include "call/rtp_payload_params.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/rtp_video_sender_interface.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/rtp_rtcp/include/flexfec_sender.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/rtp_rtcp/source/rtp_sender_video.h"

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@ -167,7 +167,7 @@ class RtpVideoSenderTestFixture {
NiceMock<MockTransport> transport_;
NiceMock<MockRtcpIntraFrameObserver> encoder_feedback_;
SimulatedClock clock_;
RtcEventLogNullImpl event_log_;
RtcEventLogNull event_log_;
VideoSendStream::Config config_;
SendDelayStats send_delay_stats_;
BitrateConstraints bitrate_config_;

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@ -30,7 +30,6 @@ rtc_source_set("rtc_event_log_api") {
sources = [
"rtc_event_log/encoder/rtc_event_log_encoder.h",
"rtc_event_log/events/rtc_event.h",
"rtc_event_log/rtc_event_log.h",
"rtc_event_log/rtc_event_log_factory_interface.h",
]
@ -277,6 +276,7 @@ rtc_source_set("fake_rtc_event_log") {
deps = [
":ice_log",
":rtc_event_log_api",
"../api/rtc_event_log",
"../rtc_base",
"../rtc_base:checks",
]
@ -314,13 +314,13 @@ if (rtc_enable_protobuf) {
":ice_log",
":rtc_event_bwe",
":rtc_event_log2_proto",
":rtc_event_log_api",
":rtc_event_log_impl_encoder",
":rtc_event_log_proto",
":rtc_stream_config",
"../api:function_view",
"../api:libjingle_peerconnection_api",
"../api:rtp_headers",
"../api/rtc_event_log",
"../api/units:data_rate",
"../api/units:time_delta",
"../api/units:timestamp",
@ -361,7 +361,6 @@ if (rtc_enable_protobuf) {
":rtc_event_bwe",
":rtc_event_generic_packet_events",
":rtc_event_log2_proto",
":rtc_event_log_api",
":rtc_event_log_impl_encoder",
":rtc_event_log_parser",
":rtc_event_log_proto",
@ -373,6 +372,7 @@ if (rtc_enable_protobuf) {
"../api:libjingle_peerconnection_api",
"../api:rtc_event_log_output_file",
"../api:rtp_headers",
"../api/rtc_event_log",
"../api/rtc_event_log:rtc_event_log_factory",
"../api/task_queue:default_task_queue_factory",
"../call",
@ -398,11 +398,11 @@ if (rtc_enable_protobuf) {
"rtc_event_log/rtc_event_log2rtp_dump.cc",
]
deps = [
":rtc_event_log_api",
":rtc_event_log_parser",
"../api:array_view",
"../api:libjingle_peerconnection_api",
"../api:rtp_headers",
"../api/rtc_event_log",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
@ -437,6 +437,7 @@ rtc_source_set("ice_log") {
":rtc_event_log_api",
"../api:libjingle_logging_api",
"../api:libjingle_peerconnection_api",
"../api/rtc_event_log",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/memory",
]
@ -450,7 +451,7 @@ if (rtc_include_tests) {
"rtc_event_log/mock/mock_rtc_event_log.h",
]
deps = [
":rtc_event_log_api",
"../api/rtc_event_log",
"../test:test_support",
]
}

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@ -14,8 +14,8 @@
#include <map>
#include <memory>
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "rtc_base/async_invoker.h"
#include "rtc_base/thread.h"

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@ -10,8 +10,8 @@
#include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/fake_rtc_event_log.h"
#include "logging/rtc_event_log/rtc_event_log.h"
namespace webrtc {

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@ -11,7 +11,7 @@
#include "logging/rtc_event_log/ice_logger.h"
#include "absl/memory/memory.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log/rtc_event_log.h"
namespace webrtc {

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@ -13,7 +13,7 @@
#include <memory>
#include "logging/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "test/gmock.h"
namespace webrtc {

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@ -1,23 +0,0 @@
/*
* Copyright (c) 2015 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 LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
#define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
// TODO(bugs.webrtc.org/10206): For backwards compatibility; Delete as soon as
// dependencies are updated.
#include "api/rtc_event_log/rtc_event_log.h"
namespace webrtc {
using RtcEventLogNullImpl = ::webrtc::RtcEventLogNull;
} // namespace webrtc
#endif // LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_

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@ -23,8 +23,8 @@
#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_headers.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "logging/rtc_event_log/rtc_event_processor.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"

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@ -22,12 +22,12 @@
#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "logging/rtc_event_log/encoder/blob_encoding.h"
#include "logging/rtc_event_log/encoder/delta_encoding.h"
#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_common.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_event_processor.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "modules/include/module_common_types.h"

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@ -18,10 +18,10 @@
#include <utility> // pair
#include <vector>
#include "api/rtc_event_log/rtc_event_log.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "logging/rtc_event_log/logged_events.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "rtc_base/ignore_wundef.h"

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@ -8,8 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "logging/rtc_event_log/rtc_event_log.h"
#include <algorithm>
#include <limits>
#include <map>
@ -20,6 +18,7 @@
#include <vector>
#include "absl/memory/memory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/rtc_event_log_output_file.h"
#include "api/task_queue/default_task_queue_factory.h"

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@ -521,6 +521,7 @@ if (rtc_include_tests) {
"../api:simulcast_test_fixture_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/rtc_event_log",
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
"../api/test/video:function_video_factory",
@ -536,7 +537,6 @@ if (rtc_include_tests) {
"../audio",
"../call:call_interfaces",
"../common_video",
"../logging:rtc_event_log_api",
"../modules/audio_device:mock_audio_device",
"../modules/audio_processing",
"../modules/audio_processing:api",

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@ -20,6 +20,7 @@
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "api/media_transport_config.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_parameters.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/fake_media_transport.h"
@ -39,7 +40,6 @@
#include "api/video_codecs/video_encoder_factory.h"
#include "call/flexfec_receive_stream.h"
#include "common_video/h264/profile_level_id.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/base/fake_frame_source.h"
#include "media/base/fake_network_interface.h"
#include "media/base/fake_video_renderer.h"
@ -272,7 +272,7 @@ class WebRtcVideoEngineTest : public ::testing::Test {
// race condition in the clock access.
rtc::ScopedFakeClock fake_clock_;
std::unique_ptr<webrtc::test::ScopedFieldTrials> override_field_trials_;
webrtc::RtcEventLogNullImpl event_log_;
webrtc::RtcEventLogNull event_log_;
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
// Used in WebRtcVideoEngineVoiceTest, but defined here so it's properly
// initialized when the constructor is called.
@ -1146,7 +1146,7 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
.WillOnce(::testing::Return(decoder));
// Create a call.
webrtc::RtcEventLogNullImpl event_log;
webrtc::RtcEventLogNull event_log;
auto task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
webrtc::Call::Config call_config(&event_log);
call_config.task_queue_factory = task_queue_factory.get();
@ -1216,7 +1216,7 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullDecoder) {
.WillOnce(::testing::Return(nullptr));
// Create a call.
webrtc::RtcEventLogNullImpl event_log;
webrtc::RtcEventLogNull event_log;
auto task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
webrtc::Call::Config call_config(&event_log);
call_config.task_queue_factory = task_queue_factory.get();
@ -1491,7 +1491,7 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
return cricket::StreamParams::CreateLegacy(kSsrc);
}
webrtc::RtcEventLogNullImpl event_log_;
webrtc::RtcEventLogNull event_log_;
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
std::unique_ptr<webrtc::Call> call_;
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
@ -7539,7 +7539,7 @@ class WebRtcVideoChannelSimulcastTest : public ::testing::Test {
return streams[streams.size() - 1];
}
webrtc::RtcEventLogNullImpl event_log_;
webrtc::RtcEventLogNull event_log_;
FakeCall fake_call_;
cricket::FakeWebRtcVideoEncoderFactory* encoder_factory_;
cricket::FakeWebRtcVideoDecoderFactory* decoder_factory_;

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@ -18,11 +18,11 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/media_transport_config.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "call/call.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/base/fake_media_engine.h"
#include "media/base/fake_network_interface.h"
#include "media/base/fake_rtp.h"
@ -3449,7 +3449,7 @@ TEST(WebRtcVoiceEngineTest, StartupShutdown) {
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNullImpl event_log;
webrtc::RtcEventLogNull event_log;
webrtc::Call::Config call_config(&event_log);
call_config.task_queue_factory = task_queue_factory.get();
auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
@ -3477,7 +3477,7 @@ TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNullImpl event_log;
webrtc::RtcEventLogNull event_log;
webrtc::Call::Config call_config(&event_log);
call_config.task_queue_factory = task_queue_factory.get();
auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
@ -3551,7 +3551,7 @@ TEST(WebRtcVoiceEngineTest, Has32Channels) {
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNullImpl event_log;
webrtc::RtcEventLogNull event_log;
webrtc::Call::Config call_config(&event_log);
call_config.task_queue_factory = task_queue_factory.get();
auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
@ -3594,7 +3594,7 @@ TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNullImpl event_log;
webrtc::RtcEventLogNull event_log;
webrtc::Call::Config call_config(&event_log);
call_config.task_queue_factory = task_queue_factory.get();
auto call = absl::WrapUnique(webrtc::Call::Create(call_config));

View File

@ -50,7 +50,6 @@ rtc_static_library("audio_coding") {
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../common_audio:common_audio_c",
"../../logging:rtc_event_log_api",
"../../rtc_base:audio_format_to_string",
"../../rtc_base:checks",
"../../rtc_base:deprecation",
@ -917,6 +916,7 @@ rtc_static_library("audio_network_adaptor") {
deps = [
"../../api/audio_codecs:audio_codecs_api",
"../../api/rtc_event_log",
"../../common_audio",
"../../logging:rtc_event_audio",
"../../logging:rtc_event_log_api",

View File

@ -18,9 +18,9 @@
#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event.h"
#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "rtc_base/checks.h"
namespace webrtc {

View File

@ -24,6 +24,7 @@ rtc_static_library("bitrate_controller") {
}
deps = [
"../../api/rtc_event_log",
"../../api/transport:network_control",
"../../api/units:data_rate",
"../../api/units:time_delta",

View File

@ -16,9 +16,9 @@
#include <string>
#include "absl/memory/memory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event.h"
#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"

View File

@ -32,6 +32,7 @@ rtc_static_library("goog_cc") {
"../..:module_api",
"../../..:webrtc_common",
"../../../api:network_state_predictor_api",
"../../../api/rtc_event_log",
"../../../api/transport:field_trial_based_config",
"../../../api/transport:network_control",
"../../../api/transport:webrtc_key_value_config",
@ -39,7 +40,6 @@ rtc_static_library("goog_cc") {
"../../../api/units:data_size",
"../../../api/units:time_delta",
"../../../api/units:timestamp",
"../../../logging:rtc_event_log_api",
"../../../logging:rtc_event_pacing",
"../../../rtc_base:checks",
"../../../rtc_base:logging",
@ -88,6 +88,7 @@ rtc_source_set("alr_detector") {
"alr_detector.h",
]
deps = [
"../../../api/rtc_event_log",
"../../../api/transport:field_trial_based_config",
"../../../api/transport:webrtc_key_value_config",
"../../../logging:rtc_event_log_api",
@ -120,12 +121,12 @@ rtc_source_set("estimators") {
deps = [
"../../../api:network_state_predictor_api",
"../../../api/rtc_event_log",
"../../../api/transport:network_control",
"../../../api/transport:webrtc_key_value_config",
"../../../api/units:data_rate",
"../../../api/units:timestamp",
"../../../logging:rtc_event_bwe",
"../../../logging:rtc_event_log_api",
"../../../rtc_base:checks",
"../../../rtc_base:logging",
"../../../rtc_base:macromagic",
@ -149,6 +150,7 @@ rtc_source_set("delay_based_bwe") {
deps = [
":estimators",
"../../../api:network_state_predictor_api",
"../../../api/rtc_event_log",
"../../../api/transport:network_control",
"../../../api/transport:network_control",
"../../../api/transport:webrtc_key_value_config",
@ -172,13 +174,13 @@ rtc_source_set("probe_controller") {
]
deps = [
"../../../api/rtc_event_log",
"../../../api/transport:network_control",
"../../../api/transport:webrtc_key_value_config",
"../../../api/units:data_rate",
"../../../api/units:time_delta",
"../../../api/units:timestamp",
"../../../logging:rtc_event_bwe",
"../../../logging:rtc_event_log_api",
"../../../logging:rtc_event_pacing",
"../../../rtc_base:checks",
"../../../rtc_base:logging",
@ -204,10 +206,10 @@ if (rtc_include_tests) {
":delay_based_bwe",
":estimators",
":goog_cc",
"../../../api/rtc_event_log",
"../../../api/transport:goog_cc",
"../../../api/transport:network_control",
"../../../api/units:timestamp",
"../../../logging:rtc_event_log_api",
"../../../rtc_base:checks",
"../../../test/logging:log_writer",
"../../remote_bitrate_estimator",

View File

@ -14,9 +14,9 @@
#include <cstdio>
#include "absl/memory/memory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event.h"
#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/time_utils.h"

View File

@ -17,9 +17,9 @@
#include <utility>
#include "absl/memory/memory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event.h"
#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "rtc_base/checks.h"

View File

@ -19,6 +19,7 @@
#include "absl/types/optional.h"
#include "api/network_state_predictor.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/field_trial_based_config.h"
#include "api/transport/network_control.h"
#include "api/transport/network_types.h"
@ -26,7 +27,6 @@
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
#include "api/units/timestamp.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/bitrate_controller/send_side_bandwidth_estimation.h"
#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
#include "modules/congestion_controller/goog_cc/alr_detector.h"

View File

@ -13,9 +13,9 @@
#include <algorithm>
#include "absl/memory/memory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
#include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"

View File

@ -17,9 +17,9 @@
#include <vector>
#include "absl/types/optional.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/network_control.h"
#include "api/transport/webrtc_key_value_config.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/system/unused.h"

View File

@ -14,11 +14,11 @@
#include <memory>
#include <string>
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/goog_cc_factory.h"
#include "api/transport/network_control.h"
#include "api/transport/network_types.h"
#include "api/units/timestamp.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
#include "test/logging/log_writer.h"

View File

@ -30,6 +30,7 @@ rtc_static_library("pacing") {
":interval_budget",
"..:module_api",
"../../api:function_view",
"../../api/rtc_event_log",
"../../api/transport:field_trial_based_config",
"../../api/transport:network_control",
"../../api/transport:webrtc_key_value_config",

View File

@ -13,9 +13,9 @@
#include <algorithm>
#include "absl/memory/memory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event.h"
#include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/metrics.h"

View File

@ -15,7 +15,7 @@
#include <vector>
#include "absl/memory/memory.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "modules/pacing/bitrate_prober.h"
#include "modules/pacing/interval_budget.h"
#include "modules/utility/include/process_thread.h"

View File

@ -221,6 +221,7 @@ rtc_static_library("rtp_rtcp") {
"../../api:scoped_refptr",
"../../api:transport_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/rtc_event_log",
"../../api/transport:field_trial_based_config",
"../../api/transport:webrtc_key_value_config",
"../../api/video:video_bitrate_allocation",
@ -231,7 +232,6 @@ rtc_static_library("rtp_rtcp") {
"../../call:rtp_interfaces",
"../../common_video",
"../../logging:rtc_event_audio",
"../../logging:rtc_event_log_api",
"../../logging:rtc_event_rtp_rtcp",
"../../modules/audio_coding:audio_coding_module_typedefs",
"../../rtc_base:checks",

View File

@ -16,8 +16,8 @@
#include <utility>
#include "absl/memory/memory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"

View File

@ -18,9 +18,9 @@
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "api/array_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/field_trial_based_config.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"

View File

@ -90,9 +90,9 @@ rtc_static_library("rtc_p2p") {
deps = [
"../api:libjingle_peerconnection_api",
"../api:scoped_refptr",
"../api/rtc_event_log",
"../api/transport:enums",
"../logging:ice_log",
"../logging:rtc_event_log_api",
"../rtc_base",
"../rtc_base:checks",

View File

@ -15,9 +15,9 @@
#include <utility>
#include "absl/memory/memory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event_dtls_transport_state.h"
#include "logging/rtc_event_log/events/rtc_event_dtls_writable_state.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "p2p/base/packet_transport_internal.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"

View File

@ -15,9 +15,9 @@
#include <utility>
#include "absl/memory/memory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event_dtls_transport_state.h"
#include "logging/rtc_event_log/events/rtc_event_dtls_writable_state.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "p2p/base/packet_transport_internal.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"

View File

@ -84,6 +84,7 @@ rtc_static_library("rtc_pc_base") {
"../api:libjingle_peerconnection_api",
"../api:rtp_headers",
"../api:scoped_refptr",
"../api/rtc_event_log",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
@ -92,7 +93,6 @@ rtc_static_library("rtc_pc_base") {
"../call:rtp_receiver",
"../common_video",
"../logging:ice_log",
"../logging:rtc_event_log_api",
"../media:rtc_data",
"../media:rtc_h264_profile_id",
"../media:rtc_media_base",
@ -219,6 +219,7 @@ rtc_static_library("peerconnection") {
"../api:rtc_event_log_output_file",
"../api:rtc_stats_api",
"../api:scoped_refptr",
"../api/rtc_event_log",
"../api/task_queue",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_frame",
@ -227,7 +228,6 @@ rtc_static_library("peerconnection") {
"../call:call_interfaces",
"../common_video",
"../logging:ice_log",
"../logging:rtc_event_log_api",
"../media:rtc_data",
"../media:rtc_media_base",
"../modules/rtp_rtcp:rtp_rtcp_format",
@ -307,7 +307,6 @@ if (rtc_include_tests) {
"../api/video:builtin_video_bitrate_allocator_factory",
"../call:rtp_interfaces",
"../call:rtp_receiver",
"../logging:rtc_event_log_api",
"../media:rtc_data",
"../media:rtc_media_base",
"../media:rtc_media_tests_utils",
@ -443,7 +442,6 @@ if (rtc_include_tests) {
"../api/video_codecs:builtin_video_encoder_factory",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../logging:rtc_event_log_api",
"../media:rtc_data",
"../media:rtc_media",
"../media:rtc_media_base",
@ -576,7 +574,6 @@ if (rtc_include_tests) {
"../api/video_codecs:builtin_video_encoder_factory",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../logging:rtc_event_log_api",
"../media:rtc_audio_video",
"../media:rtc_data", # TODO(phoglund): AFAIK only used for one sctp constant.
"../media:rtc_media_base",

View File

@ -18,9 +18,9 @@
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event_dtls_transport_state.h"
#include "logging/rtc_event_log/events/rtc_event_dtls_writable_state.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"

View File

@ -22,7 +22,7 @@
#include "api/media_transport_config.h"
#include "api/media_transport_interface.h"
#include "api/peer_connection_interface.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "media/sctp/sctp_transport_internal.h"
#include "p2p/base/dtls_transport.h"
#include "p2p/base/p2p_transport_channel.h"

View File

@ -25,13 +25,13 @@
#include "api/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log_output_file.h"
#include "api/rtp_parameters.h"
#include "api/uma_metrics.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "call/call.h"
#include "logging/rtc_event_log/ice_logger.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/base/rid_description.h"
#include "media/sctp/sctp_transport.h"
#include "pc/audio_rtp_receiver.h"

View File

@ -22,9 +22,9 @@
#include "api/network_state_predictor.h"
#include "api/peer_connection_factory_proxy.h"
#include "api/peer_connection_proxy.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/turn_customizer.h"
#include "api/video_track_source_proxy.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/base/rtp_data_engine.h"
#include "media/sctp/sctp_transport.h"
#include "p2p/base/basic_packet_socket_factory.h"
@ -327,7 +327,7 @@ std::unique_ptr<RtcEventLog> PeerConnectionFactory::CreateRtcEventLog_w() {
encoding_type = RtcEventLog::EncodingType::NewFormat;
return event_log_factory_
? event_log_factory_->CreateRtcEventLog(encoding_type)
: absl::make_unique<RtcEventLogNullImpl>();
: absl::make_unique<RtcEventLogNull>();
}
std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w(

View File

@ -26,12 +26,12 @@
#include "api/dtmf_sender_interface.h"
#include "api/media_stream_interface.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/test/fake_frame_decryptor.h"
#include "api/test/fake_frame_encryptor.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/base/codec.h"
#include "media/base/fake_media_engine.h"
#include "media/base/media_channel.h"
@ -489,7 +489,7 @@ class RtpSenderReceiverTest
protected:
rtc::Thread* const network_thread_;
rtc::Thread* const worker_thread_;
webrtc::RtcEventLogNullImpl event_log_;
webrtc::RtcEventLogNull event_log_;
// The |rtp_dtls_transport_| and |rtp_transport_| should be destroyed after
// the |channel_manager|.
std::unique_ptr<cricket::DtlsTransportInternal> rtp_dtls_transport_;

View File

@ -155,6 +155,7 @@ if (!build_with_chromium) {
deps = [
"../api:libjingle_peerconnection_api",
"../api:transport_api",
"../api/rtc_event_log",
"../api/task_queue:default_task_queue_factory",
"../api/task_queue:task_queue",
"../api/video:builtin_video_bitrate_allocator_factory",
@ -169,7 +170,6 @@ if (!build_with_chromium) {
"../call:simulated_network",
"../call:simulated_packet_receiver",
"../call:video_stream_api",
"../logging:rtc_event_log_api",
"../media:rtc_audio_video",
"../media:rtc_media_base",
"../rtc_base",
@ -289,7 +289,6 @@ if (!build_with_chromium) {
"../api/transport:network_control",
"../call:call_interfaces",
"../call:video_stream_api",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_parser",
"../logging:rtc_stream_config",
"../modules/audio_coding:ana_debug_dump_proto",
@ -325,7 +324,7 @@ if (rtc_include_tests) {
defines = [ "ENABLE_RTC_EVENT_LOG" ]
deps = [
":event_log_visualizer_utils",
"../logging:rtc_event_log_api",
"../api/rtc_event_log",
"../logging:rtc_event_log_parser",
"../modules/audio_coding:neteq",
"../modules/rtp_rtcp:rtp_rtcp_format",

View File

@ -1218,7 +1218,7 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
SimulatedClock clock(0);
BitrateObserver observer;
RtcEventLogNullImpl null_event_log;
RtcEventLogNull null_event_log;
PacketRouter packet_router;
PacedSender pacer(&clock, &packet_router, &null_event_log);
TransportFeedbackAdapter transport_feedback;

View File

@ -45,7 +45,7 @@ class LogBasedNetworkControllerSimulation {
void OnFeedback(const LoggedRtcpPacketTransportFeedback& feedback);
void OnReceiverReport(const LoggedRtcpPacketReceiverReport& report);
void OnIceConfig(const LoggedIceCandidatePairConfig& candidate);
RtcEventLogNullImpl null_event_log_;
RtcEventLogNull null_event_log_;
const std::function<void(const NetworkControlUpdate&, Timestamp)>
update_handler_;

View File

@ -24,7 +24,7 @@
#include "absl/flags/usage.h"
#include "absl/flags/usage_config.h"
#include "absl/strings/match.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"

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@ -17,6 +17,7 @@
#include "api/call/transport.h"
#include "api/media_types.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video_codecs/video_decoder_factory.h"
@ -25,7 +26,6 @@
#include "call/call.h"
#include "call/rtp_config.h"
#include "call/video_send_stream.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/engine/webrtc_video_engine.h"
#include "rtc_base/constructor_magic.h"
#include "test/frame_generator.h"

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@ -740,6 +740,7 @@ rtc_source_set("test_common") {
"../api:transport_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/rtc_event_log",
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
"../api/test/video:function_video_factory",
@ -756,7 +757,6 @@ rtc_source_set("test_common") {
"../call:simulated_network",
"../call:simulated_packet_receiver",
"../call:video_stream_api",
"../logging:rtc_event_log_api",
"../media:rtc_internal_video_codecs",
"../media:rtc_media_base",
"../modules/audio_device",

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@ -16,12 +16,12 @@
#include <vector>
#include "absl/types/optional.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/test/video/function_video_decoder_factory.h"
#include "api/test/video/function_video_encoder_factory.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "call/call.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "test/encoder_settings.h"
#include "test/fake_decoder.h"

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@ -15,13 +15,13 @@ rtc_source_set("rtp_replayer") {
"rtp_replayer.h",
]
deps = [
"../../../api/rtc_event_log",
"../../../api/task_queue:default_task_queue_factory",
"../../../api/test/video:function_video_factory",
"../../../api/video_codecs:video_codecs_api",
"../../../call",
"../../../call:call_interfaces",
"../../../common_video",
"../../../logging:rtc_event_log_api",
"../../../media:rtc_internal_video_codecs",
"../../../modules/rtp_rtcp",
"../../../rtc_base:checks",

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@ -51,7 +51,7 @@ void RtpReplayer::Replay(
}
// Setup the video streams based on the configuration.
webrtc::RtcEventLogNullImpl event_log;
webrtc::RtcEventLogNull event_log;
std::unique_ptr<TaskQueueFactory> task_queue_factory =
CreateDefaultTaskQueueFactory();
Call::Config call_config(&event_log);

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@ -18,10 +18,10 @@
#include <string>
#include <vector>
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/test/video/function_video_decoder_factory.h"
#include "api/video_codecs/video_decoder.h"
#include "call/call.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/engine/internal_decoder_factory.h"
#include "rtc_base/time_utils.h"
#include "test/null_transport.h"

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@ -265,11 +265,11 @@ if (rtc_include_tests) {
"../../../api:rtc_event_log_output_file",
"../../../api:scoped_refptr",
"../../../api:video_quality_analyzer_api",
"../../../api/rtc_event_log",
"../../../api/task_queue",
"../../../api/task_queue:default_task_queue_factory",
"../../../api/units:time_delta",
"../../../api/units:timestamp",
"../../../logging:rtc_event_log_api",
"../../../pc:pc_test_utils",
"../../../pc:peerconnection",
"../../../rtc_base",

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@ -17,12 +17,12 @@
#include "api/jsep.h"
#include "api/media_stream_interface.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log_output_file.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/video_quality_analyzer_interface.h"
#include "api/units/time_delta.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "pc/sdp_utils.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "rtc_base/bind.h"

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@ -105,7 +105,6 @@ if (rtc_include_tests) {
"../../call:simulated_network",
"../../call:video_stream_api",
"../../common_video",
"../../logging:rtc_event_log_api",
"../../media:rtc_audio_video",
"../../media:rtc_internal_video_codecs",
"../../media:rtc_media_base",

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@ -16,8 +16,8 @@
#include <utility>
#include <vector>
#include "api/rtc_event_log/rtc_event_log.h"
#include "call/call.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "modules/congestion_controller/goog_cc/test/goog_cc_printer.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"

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@ -57,6 +57,7 @@ rtc_static_library("video") {
"../api:libjingle_peerconnection_api",
"../api:scoped_refptr",
"../api:transport_api",
"../api/rtc_event_log",
"../api/task_queue",
"../api/video:encoded_image",
"../api/video:video_bitrate_allocation",
@ -74,7 +75,6 @@ rtc_static_library("video") {
"../call:rtp_sender",
"../call:video_stream_api",
"../common_video",
"../logging:rtc_event_log_api",
"../media:rtc_h264_profile_id",
"../modules:module_api",
"../modules:module_api_public",
@ -269,7 +269,6 @@ if (rtc_include_tests) {
"../call:fake_network",
"../call:simulated_network",
"../common_video",
"../logging:rtc_event_log_api",
"../media:rtc_audio_video",
"../media:rtc_encoder_simulcast_proxy",
"../media:rtc_internal_video_codecs",
@ -476,11 +475,11 @@ if (rtc_include_tests) {
"video_replay.cc",
]
deps = [
"../api/rtc_event_log",
"../api/test/video:function_video_factory",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../common_video",
"../logging:rtc_event_log_api",
"../media:rtc_internal_video_codecs",
"../modules/rtp_rtcp",
"../rtc_base:checks",
@ -567,6 +566,7 @@ if (rtc_include_tests) {
"../api:rtp_headers",
"../api:scoped_refptr",
"../api:simulated_network_api",
"../api/rtc_event_log",
"../api/task_queue:default_task_queue_factory",
"../api/test/video:function_video_factory",
"../api/units:data_rate",
@ -592,7 +592,6 @@ if (rtc_include_tests) {
"../call:video_stream_api",
"../common_video",
"../common_video/test:utilities",
"../logging:rtc_event_log_api",
"../media:rtc_audio_video",
"../media:rtc_internal_video_codecs",
"../media:rtc_media",

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@ -15,13 +15,13 @@
#include <vector>
#include "absl/memory/memory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/simulated_network.h"
#include "api/test/video/function_video_encoder_factory.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/engine/internal_decoder_factory.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "test/call_test.h"
@ -45,7 +45,7 @@ MultiStreamTester::MultiStreamTester(
MultiStreamTester::~MultiStreamTester() {}
void MultiStreamTester::RunTest() {
webrtc::RtcEventLogNullImpl event_log;
webrtc::RtcEventLogNull event_log;
auto task_queue_factory = CreateDefaultTaskQueueFactory();
Call::Config config(&event_log);
config.task_queue_factory = task_queue_factory.get();

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@ -17,11 +17,11 @@
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
#include "absl/memory/memory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/test/video/function_video_decoder_factory.h"
#include "api/video_codecs/video_decoder.h"
#include "call/call.h"
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/engine/internal_decoder_factory.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
#include "rtc_base/checks.h"
@ -262,7 +262,7 @@ class RtpReplayer final {
// Replay a rtp dump with an optional json configuration.
static void Replay(const std::string& replay_config_path,
const std::string& rtp_dump_path) {
webrtc::RtcEventLogNullImpl event_log;
webrtc::RtcEventLogNull event_log;
Call::Config call_config(&event_log);
std::unique_ptr<Call> call(Call::Create(call_config));
std::unique_ptr<StreamState> stream_state;

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@ -20,6 +20,7 @@
#include "absl/types/optional.h"
#include "api/fec_controller.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/video/encoded_image.h"
#include "api/video/video_bitrate_allocation.h"
#include "api/video/video_bitrate_allocator.h"
@ -30,7 +31,6 @@
#include "call/rtp_config.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/rtp_video_sender_interface.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/include/module_common_types.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/utility/include/process_thread.h"

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@ -14,10 +14,10 @@
#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "call/rtp_video_sender.h"
#include "call/test/mock_bitrate_allocator.h"
#include "call/test/mock_rtp_transport_controller_send.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
#include "modules/utility/include/process_thread.h"
#include "modules/video_coding/fec_controller_default.h"
@ -142,7 +142,7 @@ class VideoSendStreamImplTest : public ::testing::Test {
bool rtp_video_sender_active_ = false;
SimulatedClock clock_;
RtcEventLogNullImpl event_log_;
RtcEventLogNull event_log_;
VideoSendStream::Config config_;
SendDelayStats send_delay_stats_;
TaskQueueForTest test_queue_;