b3c210fa56
This reverts commit 6cfbc35ad7e6c64874c1e2dbd58b8d7c4ab3a679. Reason for revert: Fixing downstream projects. Original change's description: > Revert "Revert "Revert "New protobuf format for event log.""" > > This reverts commit ef8f42040367b3809295a007d7eeeff4526e1b39. > > Reason for revert: New problems with downstream project. > > Original change's description: > > Revert "Revert "New protobuf format for event log."" > > > > This reverts commit 546373fc66e24d041e8eb8ffd2fc522847d841d1. > > > > Reason for revert: Downstream project fixed. > > > > Original change's description: > > > Revert "New protobuf format for event log." > > > > > > This reverts commit 99463c14dbbc88732f0991cb30e7bbfcdaeb3cdc. > > > > > > Reason for revert: Speculative revert for downstream project breakage. > > > > > > Original change's description: > > > > New protobuf format for event log. > > > > > > > > Bug: webrtc:6295 > > > > Change-Id: Ie20a2808a4f076b05fb6195f4fed73215f6fd3b2 > > > > Reviewed-on: https://webrtc-review.googlesource.com/8880 > > > > Commit-Queue: Björn Terelius <terelius@webrtc.org> > > > > Reviewed-by: Per Kjellander <perkj@webrtc.org> > > > > Reviewed-by: Dino Radaković <dinor@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#21291} > > > > > > TBR=terelius@webrtc.org,perkj@webrtc.org,dinor@webrtc.org > > > > > > Change-Id: Ic319170a7a777002ca29248d102cb4e26966d5ae > > > No-Presubmit: true > > > No-Tree-Checks: true > > > No-Try: true > > > Bug: webrtc:6295 > > > Reviewed-on: https://webrtc-review.googlesource.com/33400 > > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > > Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21292} > > > > TBR=brandtr@webrtc.org,terelius@webrtc.org,perkj@webrtc.org,dinor@webrtc.org > > > > Change-Id: I9e96e5007d0447e63178d47c7330488b2a8f2b6f > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: webrtc:6295 > > Reviewed-on: https://webrtc-review.googlesource.com/33440 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21296} > > TBR=brandtr@webrtc.org,terelius@webrtc.org,perkj@webrtc.org,dinor@webrtc.org > > Change-Id: I4eb15c809f67af13ffa7b7df6eb06088af21f63f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:6295 > Reviewed-on: https://webrtc-review.googlesource.com/33480 > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21297} TBR=brandtr@webrtc.org,terelius@webrtc.org,perkj@webrtc.org,dinor@webrtc.org Change-Id: I7895575f2b6e4ec2c36296fe81a7596147158601 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:6295 Reviewed-on: https://webrtc-review.googlesource.com/33520 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21299}
286 lines
9.9 KiB
Plaintext
286 lines
9.9 KiB
Plaintext
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../webrtc.gni")
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import("//third_party/protobuf/proto_library.gni")
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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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group("logging") {
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deps = [
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":rtc_event_log_impl",
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]
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if (rtc_enable_protobuf) {
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deps += [ ":rtc_event_log_parser" ]
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}
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}
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rtc_source_set("rtc_event_log_api") {
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sources = [
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"rtc_event_log/events/rtc_event.h",
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"rtc_event_log/events/rtc_event_alr_state.cc",
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"rtc_event_log/events/rtc_event_alr_state.h",
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"rtc_event_log/events/rtc_event_audio_network_adaptation.cc",
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"rtc_event_log/events/rtc_event_audio_network_adaptation.h",
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"rtc_event_log/events/rtc_event_audio_playout.cc",
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"rtc_event_log/events/rtc_event_audio_playout.h",
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"rtc_event_log/events/rtc_event_audio_receive_stream_config.cc",
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"rtc_event_log/events/rtc_event_audio_receive_stream_config.h",
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"rtc_event_log/events/rtc_event_audio_send_stream_config.cc",
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"rtc_event_log/events/rtc_event_audio_send_stream_config.h",
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"rtc_event_log/events/rtc_event_bwe_update_delay_based.cc",
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"rtc_event_log/events/rtc_event_bwe_update_delay_based.h",
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"rtc_event_log/events/rtc_event_bwe_update_loss_based.cc",
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"rtc_event_log/events/rtc_event_bwe_update_loss_based.h",
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"rtc_event_log/events/rtc_event_logging_started.cc",
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"rtc_event_log/events/rtc_event_logging_started.h",
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"rtc_event_log/events/rtc_event_logging_stopped.cc",
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"rtc_event_log/events/rtc_event_logging_stopped.h",
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"rtc_event_log/events/rtc_event_probe_cluster_created.cc",
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"rtc_event_log/events/rtc_event_probe_cluster_created.h",
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"rtc_event_log/events/rtc_event_probe_result_failure.cc",
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"rtc_event_log/events/rtc_event_probe_result_failure.h",
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"rtc_event_log/events/rtc_event_probe_result_success.cc",
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"rtc_event_log/events/rtc_event_probe_result_success.h",
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"rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc",
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"rtc_event_log/events/rtc_event_rtcp_packet_incoming.h",
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"rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc",
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"rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h",
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"rtc_event_log/events/rtc_event_rtp_packet_incoming.cc",
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"rtc_event_log/events/rtc_event_rtp_packet_incoming.h",
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"rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc",
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"rtc_event_log/events/rtc_event_rtp_packet_outgoing.h",
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"rtc_event_log/events/rtc_event_video_receive_stream_config.cc",
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"rtc_event_log/events/rtc_event_video_receive_stream_config.h",
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"rtc_event_log/events/rtc_event_video_send_stream_config.cc",
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"rtc_event_log/events/rtc_event_video_send_stream_config.h",
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"rtc_event_log/output/rtc_event_log_output_file.cc",
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"rtc_event_log/output/rtc_event_log_output_file.h",
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"rtc_event_log/rtc_event_log.h",
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"rtc_event_log/rtc_event_log_factory_interface.h",
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"rtc_event_log/rtc_stream_config.cc",
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"rtc_event_log/rtc_stream_config.h",
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]
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deps = [
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"..:webrtc_common",
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"../:typedefs",
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"../api:array_view",
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"../api:libjingle_logging_api",
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"../api:libjingle_peerconnection_api",
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"../call:video_stream_api",
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"../modules/audio_coding:audio_network_adaptor_config",
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"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers",
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]
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# TODO(eladalon): Remove this.
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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rtc_static_library("rtc_event_log_impl") {
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sources = [
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"rtc_event_log/encoder/rtc_event_log_encoder.h",
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"rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc",
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"rtc_event_log/encoder/rtc_event_log_encoder_legacy.h",
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"rtc_event_log/rtc_event_log.cc",
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"rtc_event_log/rtc_event_log_factory.cc",
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"rtc_event_log/rtc_event_log_factory.h",
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]
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defines = []
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deps = [
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":rtc_event_log_api",
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"..:webrtc_common",
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"../modules/audio_coding:audio_network_adaptor",
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"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:checks",
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"../rtc_base:protobuf_utils",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_task_queue",
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"../rtc_base:sequenced_task_checker",
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"../system_wrappers",
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]
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if (rtc_enable_protobuf) {
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defines += [ "ENABLE_RTC_EVENT_LOG" ]
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deps += [ ":rtc_event_log_proto" ]
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}
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# TODO(eladalon): Remove this.
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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if (rtc_enable_protobuf) {
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proto_library("rtc_event_log_proto") {
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sources = [
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"rtc_event_log/rtc_event_log.proto",
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]
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proto_out_dir = "logging/rtc_event_log"
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}
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proto_library("rtc_event_log2_proto") {
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sources = [
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"rtc_event_log/rtc_event_log2.proto",
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]
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proto_out_dir = "logging/rtc_event_log"
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}
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rtc_static_library("rtc_event_log_parser") {
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sources = [
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"rtc_event_log/rtc_event_log_parser.cc",
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"rtc_event_log/rtc_event_log_parser.h",
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]
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deps = [
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":rtc_event_log2_proto",
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":rtc_event_log_api",
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":rtc_event_log_proto",
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"..:webrtc_common",
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"../call:video_stream_api",
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"../modules/audio_coding:audio_network_adaptor",
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"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:checks",
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"../rtc_base:protobuf_utils",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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if (rtc_include_tests) {
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rtc_source_set("rtc_event_log_tests") {
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testonly = true
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assert(rtc_enable_protobuf)
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defines = [ "ENABLE_RTC_EVENT_LOG" ]
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if (rtc_use_memcheck) {
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defines += [ "WEBRTC_USE_MEMCHECK" ]
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}
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sources = [
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"rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc",
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"rtc_event_log/output/rtc_event_log_output_file_unittest.cc",
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"rtc_event_log/rtc_event_log_unittest.cc",
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"rtc_event_log/rtc_event_log_unittest_helper.cc",
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"rtc_event_log/rtc_event_log_unittest_helper.h",
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]
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deps = [
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":rtc_event_log_api",
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":rtc_event_log_impl",
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":rtc_event_log_parser",
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":rtc_event_log_proto",
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"../api:libjingle_peerconnection_api",
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"../call",
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"../call:call_interfaces",
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"../modules/audio_coding:audio_network_adaptor",
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"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_base_tests_utils",
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"../system_wrappers:metrics_default",
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"../test:test_support",
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"//testing/gmock",
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"//testing/gtest",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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rtc_test("rtc_event_log2rtp_dump") {
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testonly = true
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sources = [
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"rtc_event_log/rtc_event_log2rtp_dump.cc",
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]
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deps = [
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":rtc_event_log_api",
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":rtc_event_log_impl",
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":rtc_event_log_parser",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:checks",
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"../rtc_base:protobuf_utils",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers:field_trial_default",
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"../system_wrappers:metrics_default",
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"../test:rtp_test_utils",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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}
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if (rtc_include_tests) {
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rtc_executable("rtc_event_log2text") {
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testonly = true
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sources = [
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"rtc_event_log/rtc_event_log2text.cc",
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]
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deps = [
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":rtc_event_log_api",
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":rtc_event_log_impl",
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":rtc_event_log_parser",
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"../:webrtc_common",
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"../call:video_stream_api",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:checks",
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"../rtc_base:protobuf_utils",
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"../rtc_base:rtc_base_approved",
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# TODO(kwiberg): Remove this dependency.
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"../api/audio_codecs:audio_codecs_api",
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"../modules/audio_coding:audio_network_adaptor_config",
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"../modules/rtp_rtcp",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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}
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if (rtc_include_tests) {
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rtc_executable("rtc_event_log2stats") {
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testonly = true
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sources = [
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"rtc_event_log/rtc_event_log2stats.cc",
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]
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deps = [
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":rtc_event_log_api",
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":rtc_event_log_impl",
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":rtc_event_log_proto",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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}
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}
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