webrtc_m130/logging/BUILD.gn

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# 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.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../webrtc.gni")
if (!build_with_mozilla) {
import("//third_party/protobuf/proto_library.gni")
}
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
group("logging") {
deps = [
":rtc_event_audio",
":rtc_event_bwe",
":rtc_event_log_impl_base",
":rtc_event_log_impl_encoder",
":rtc_event_log_impl_output",
":rtc_event_pacing",
":rtc_event_rtp_rtcp",
":rtc_event_video",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
]
}
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.cc",
"rtc_event_log/rtc_event_log.h",
"rtc_event_log/rtc_event_log_factory_interface.h",
]
deps = [
"../api:libjingle_logging_api",
"../rtc_base:ptr_util",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("rtc_stream_config") {
sources = [
"rtc_event_log/rtc_stream_config.cc",
"rtc_event_log/rtc_stream_config.h",
]
deps = [
":rtc_event_log_api",
"..:webrtc_common",
"../api:libjingle_peerconnection_api",
]
}
rtc_source_set("rtc_event_pacing") {
sources = [
"rtc_event_log/events/rtc_event_alr_state.cc",
"rtc_event_log/events/rtc_event_alr_state.h",
]
deps = [
":rtc_event_log_api",
"../rtc_base:ptr_util",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("rtc_event_audio") {
sources = [
"rtc_event_log/events/rtc_event_audio_network_adaptation.cc",
"rtc_event_log/events/rtc_event_audio_network_adaptation.h",
"rtc_event_log/events/rtc_event_audio_playout.cc",
"rtc_event_log/events/rtc_event_audio_playout.h",
"rtc_event_log/events/rtc_event_audio_receive_stream_config.cc",
"rtc_event_log/events/rtc_event_audio_receive_stream_config.h",
"rtc_event_log/events/rtc_event_audio_send_stream_config.cc",
"rtc_event_log/events/rtc_event_audio_send_stream_config.h",
]
deps = [
":rtc_event_log_api",
":rtc_stream_config",
"../modules/audio_coding:audio_network_adaptor_config",
"../rtc_base:ptr_util",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("rtc_event_bwe") {
sources = [
"rtc_event_log/events/rtc_event_bwe_update_delay_based.cc",
"rtc_event_log/events/rtc_event_bwe_update_delay_based.h",
"rtc_event_log/events/rtc_event_bwe_update_loss_based.cc",
"rtc_event_log/events/rtc_event_bwe_update_loss_based.h",
"rtc_event_log/events/rtc_event_probe_cluster_created.cc",
"rtc_event_log/events/rtc_event_probe_cluster_created.h",
"rtc_event_log/events/rtc_event_probe_result_failure.cc",
"rtc_event_log/events/rtc_event_probe_result_failure.h",
"rtc_event_log/events/rtc_event_probe_result_success.cc",
"rtc_event_log/events/rtc_event_probe_result_success.h",
]
deps = [
":rtc_event_log_api",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../rtc_base:ptr_util",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("rtc_event_rtp_rtcp") {
sources = [
"rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc",
"rtc_event_log/events/rtc_event_rtcp_packet_incoming.h",
"rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc",
"rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h",
"rtc_event_log/events/rtc_event_rtp_packet_incoming.cc",
"rtc_event_log/events/rtc_event_rtp_packet_incoming.h",
"rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc",
"rtc_event_log/events/rtc_event_rtp_packet_outgoing.h",
]
deps = [
":rtc_event_log_api",
"../api:array_view",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:ptr_util",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("rtc_event_video") {
sources = [
"rtc_event_log/events/rtc_event_video_receive_stream_config.cc",
"rtc_event_log/events/rtc_event_video_receive_stream_config.h",
"rtc_event_log/events/rtc_event_video_send_stream_config.cc",
"rtc_event_log/events/rtc_event_video_send_stream_config.h",
]
deps = [
":rtc_event_log_api",
":rtc_stream_config",
"../rtc_base:ptr_util",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_static_library("rtc_event_log_impl_encoder") {
visibility = [ "*" ]
sources = [
"rtc_event_log/encoder/blob_encoding.cc",
"rtc_event_log/encoder/blob_encoding.h",
"rtc_event_log/encoder/delta_encoding.cc",
"rtc_event_log/encoder/delta_encoding.h",
"rtc_event_log/encoder/varint.cc",
"rtc_event_log/encoder/varint.h",
]
defines = []
deps = [
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:strings",
"//third_party/abseil-cpp/absl/types:optional",
]
if (rtc_enable_protobuf) {
defines += [ "ENABLE_RTC_EVENT_LOG" ]
deps += [
":ice_log",
":rtc_event_audio",
":rtc_event_bwe",
":rtc_event_log2_proto",
":rtc_event_log_api",
":rtc_event_log_impl_output",
":rtc_event_log_proto",
":rtc_event_pacing",
":rtc_event_rtp_rtcp",
":rtc_event_video",
":rtc_stream_config",
"../api:array_view",
"../modules/audio_coding:audio_network_adaptor",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp:rtp_rtcp_format",
]
sources += [
"rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc",
"rtc_event_log/encoder/rtc_event_log_encoder_legacy.h",
"rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc",
"rtc_event_log/encoder/rtc_event_log_encoder_new_format.h",
]
}
}
rtc_source_set("rtc_event_log_impl_output") {
sources = [
"rtc_event_log/output/rtc_event_log_output_file.cc",
"rtc_event_log/output/rtc_event_log_output_file.h",
]
deps = [
":rtc_event_log_api",
"../api:libjingle_logging_api",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
]
}
rtc_static_library("rtc_event_log_impl_base") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
sources = [
"rtc_event_log/rtc_event_log_factory.cc",
"rtc_event_log/rtc_event_log_factory.h",
"rtc_event_log/rtc_event_log_impl.cc",
]
defines = []
deps = [
":ice_log",
":rtc_event_log_api",
"../api:libjingle_logging_api",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
Detangling target dependencies in rtc_base_approved. The eventual goal is to allow PlatformThread to use SequencedTaskChecker, but getting to that point will require some more detangling. Here are (roughly) the steps taken in this CL: * Make constructormagic a separate target. * Move atomicops and arraysize to separate targets * Move platform_thread_types to a separate target * Move criticalsection to a separate target * Move thread_checker to separate target * Make sequenced_task_checker not depend on base_approved * Move ptr_util to a separate target * Move scoped_ptr to ptr_util * Make rtc_task_queue_api not depend on base_approved * Make sequenced_task_checker depend on rtc_task_queue_api * Move rtc::Event to its own target * Move basictypes.h to constructormagic * Move format_macros and stringize_macros into constructormagic * Rename constructormagic target to... macromagic * Move stringencode to stringutils * New target for safe_conversions * Move timeutils to a new target. * Move logging to a new target. * Move platform_thread to a new target. * Make refcount a new target (refcount, refcountedobject, refcounter). * Remove rtc_base_approved from deps of TQ * Remove a circular dependency between event tracer and platform thread. Further steps will probably be to factor TaskQueue::Current() to not be a part of the TaskQueue class itself and have it declared+implemented in a target that's lower level than TQ itself. SequencedTaskChecker can then depend on that target and avoid the TQ dependency. Once we're there, PlatformThread will be able to depend on SequencedTaskChecker. Attempted but eventually removed from this CL: * Make TQ a part of rtc_base_approved * Remove direct dependencies on sequenced_task_checker. * Profit. A few include-what-you-use updates along the way. Fix a few targets that were depending on rtc_task_queue_api Change-Id: Iee79aa2e81d978444c51b3005db9df7dc12d92a9 Bug: webrtc:8957 Reviewed-on: https://webrtc-review.googlesource.com/58480 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22487}
2018-03-19 11:12:48 +01:00
"../rtc_base:rtc_task_queue",
"../rtc_base:safe_minmax",
"../rtc_base:sequenced_task_checker",
"//third_party/abseil-cpp/absl/memory",
]
if (rtc_enable_protobuf) {
defines += [ "ENABLE_RTC_EVENT_LOG" ]
deps += [ ":rtc_event_log_impl_encoder" ]
}
}
rtc_source_set("fake_rtc_event_log") {
testonly = true
sources = [
"rtc_event_log/fake_rtc_event_log.cc",
"rtc_event_log/fake_rtc_event_log.h",
"rtc_event_log/fake_rtc_event_log_factory.cc",
"rtc_event_log/fake_rtc_event_log_factory.h",
]
deps = [
":ice_log",
":rtc_event_log_api",
"../rtc_base:checks",
"../rtc_base:rtc_base",
]
}
if (rtc_enable_protobuf) {
proto_library("rtc_event_log_proto") {
sources = [
"rtc_event_log/rtc_event_log.proto",
]
proto_out_dir = "logging/rtc_event_log"
}
Reland "New protobuf format for event log."""" 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}
2017-12-15 17:12:42 +01:00
proto_library("rtc_event_log2_proto") {
sources = [
"rtc_event_log/rtc_event_log2.proto",
]
proto_out_dir = "logging/rtc_event_log"
}
rtc_static_library("rtc_event_log_parser") {
visibility = [ "*" ]
sources = [
Revert "Create new API for RtcEventLogParser." This reverts commit 9e336ec0b8a77c3461d13677cff3563c11c88daa. Reason for revert: Code can accidentally include the deprecated parser but link with the new one, or vice versa. Reverting to fix naming. Original change's description: > Create new API for RtcEventLogParser. > > The new API stores events gathered by event type. For example, it is > possible to ask fo a list of all incoming RTCP messages or all audio > playout events. > > The new API is experimental and may change over next few weeks. Once > it has stabilized and all unit tests and existing tools have been > ported to the new API, the old one will be removed. > > This CL also updates the event_log_visualizer tool to use the new > parser API. This is not a funcional change except for: > - Incoming and outgoing audio level are now drawn in two separate plots. > - Incoming and outgoing timstamps are now drawn in two separate plots. > - RTCP count is no longer split into Video and Audio. It also counts > all RTCP packets rather than only specific message types. > - Slight timing difference in sendside BWE simulation due to only > iterating over transport feedbacks and not over all RTCP packets. > This timing changes are not visible in the plots. > > > Media type for RTCP messages might not be identified correctly by > rtc_event_log2text anymore. On the other hand, assigning a specific > media type to an RTCP packet was a bit hacky to begin with. > > Bug: webrtc:8111 > Change-Id: I8e7168302beb69b2e163a097a2a142b86dd4a26b > Reviewed-on: https://webrtc-review.googlesource.com/60865 > Reviewed-by: Minyue Li <minyue@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23015} TBR=terelius@webrtc.org,srte@webrtc.org,minyue@webrtc.org Change-Id: Ib4bbcf0563423675a3cc1dce59ebf665e0c5dae9 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8111 Reviewed-on: https://webrtc-review.googlesource.com/72500 Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23026}
2018-04-25 14:23:01 +00:00
"rtc_event_log/rtc_event_log_parser.cc",
"rtc_event_log/rtc_event_log_parser.h",
2018-04-27 14:33:34 +02:00
"rtc_event_log/rtc_event_log_parser_new.cc",
"rtc_event_log/rtc_event_log_parser_new.h",
"rtc_event_log/rtc_event_processor.h",
]
deps = [
":ice_log",
":rtc_event_bwe",
Reland "New protobuf format for event log."""" 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}
2017-12-15 17:12:42 +01:00
":rtc_event_log2_proto",
":rtc_event_log_api",
":rtc_event_log_proto",
":rtc_stream_config",
"..:webrtc_common",
2018-04-27 14:33:34 +02:00
"../api:libjingle_peerconnection_api",
"../call:video_stream_api",
"../modules/audio_coding:audio_network_adaptor",
"../modules/congestion_controller/rtp:transport_feedback",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:protobuf_utils",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
}
if (rtc_include_tests) {
rtc_source_set("rtc_event_log_tests") {
testonly = true
assert(rtc_enable_protobuf)
defines = [ "ENABLE_RTC_EVENT_LOG" ]
sources = [
"rtc_event_log/encoder/blob_encoding_unittest.cc",
"rtc_event_log/encoder/delta_encoding_unittest.cc",
"rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc",
"rtc_event_log/output/rtc_event_log_output_file_unittest.cc",
"rtc_event_log/rtc_event_log_unittest.cc",
"rtc_event_log/rtc_event_log_unittest_helper.cc",
"rtc_event_log/rtc_event_log_unittest_helper.h",
"rtc_event_log/rtc_event_processor_unittest.cc",
]
deps = [
":ice_log",
":rtc_event_audio",
":rtc_event_bwe",
":rtc_event_log2_proto",
":rtc_event_log_api",
":rtc_event_log_impl_base",
":rtc_event_log_impl_encoder",
":rtc_event_log_impl_output",
":rtc_event_log_parser",
":rtc_event_log_proto",
":rtc_event_pacing",
":rtc_event_rtp_rtcp",
":rtc_event_video",
":rtc_stream_config",
"../api:libjingle_peerconnection_api",
"../call",
"../call:call_interfaces",
"../modules/audio_coding:audio_network_adaptor",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../test:fileutils",
"../test:test_support",
"//testing/gtest",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
rtc_test("rtc_event_log2rtp_dump") {
testonly = true
sources = [
"rtc_event_log/rtc_event_log2rtp_dump.cc",
]
deps = [
":rtc_event_log_api",
":rtc_event_log_parser",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:protobuf_utils",
"../rtc_base:rtc_base_approved",
"../test:rtp_test_utils",
"//third_party/abseil-cpp/absl/memory:memory",
"//third_party/abseil-cpp/absl/types:optional",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
}
if (rtc_include_tests) {
rtc_executable("rtc_event_log2text") {
testonly = true
sources = [
"rtc_event_log/rtc_event_log2text.cc",
]
deps = [
":rtc_event_log_api",
":rtc_event_log_parser",
"../:webrtc_common",
"../call:video_stream_api",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:protobuf_utils",
"../rtc_base:rtc_base_approved",
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"../rtc_base:stringutils",
# TODO(kwiberg): Remove this dependency.
"../api/audio_codecs:audio_codecs_api",
"../modules/audio_coding:audio_network_adaptor_config",
"../modules/rtp_rtcp",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
}
if (rtc_include_tests) {
rtc_executable("rtc_event_log2stats") {
testonly = true
sources = [
"rtc_event_log/rtc_event_log2stats.cc",
]
deps = [
":rtc_event_log_api",
":rtc_event_log_proto",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
]
}
}
}
rtc_source_set("ice_log") {
sources = [
"rtc_event_log/events/rtc_event_ice_candidate_pair.cc",
"rtc_event_log/events/rtc_event_ice_candidate_pair.h",
"rtc_event_log/events/rtc_event_ice_candidate_pair_config.cc",
"rtc_event_log/events/rtc_event_ice_candidate_pair_config.h",
"rtc_event_log/icelogger.cc",
"rtc_event_log/icelogger.h",
]
deps = [
":rtc_event_log_api",
"../api:libjingle_logging_api",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/memory",
]
}
if (rtc_include_tests) {
rtc_source_set("mocks") {
testonly = true
sources = [
Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets" This reverts commit 01aa210fad68f1006528d32d388b307c22990734. Reason for revert: downstream project adjusted Original change's description: > Revert "Enable and fix chromium clang warnings in rtp_rtcp test targets" > > This reverts commit 9486b117daac09c9f7ac8450ccda835938cf3150. > > Reason for revert: Breaks downstream project > > Original change's description: > > Enable and fix chromium clang warnings in rtp_rtcp test targets > > > > Bug: webrtc:163 > > Change-Id: I4ed3e63296d8bf06536a83196d597c7a906ba11c > > Reviewed-on: https://webrtc-review.googlesource.com/60802 > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22357} > > TBR=danilchap@webrtc.org,phoglund@webrtc.org,terelius@webrtc.org > > Change-Id: I2c3777ea9f26813bdb395e7fd68f6b49443586ea > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:163 > Reviewed-on: https://webrtc-review.googlesource.com/61060 > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22365} TBR=danilchap@webrtc.org,phoglund@webrtc.org,oprypin@webrtc.org,terelius@webrtc.org Change-Id: I0b4cb6d05b37caeb52cca9abf95417ad3ad6f76b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:163 Reviewed-on: https://webrtc-review.googlesource.com/61080 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22368}
2018-03-09 15:37:03 +00:00
"rtc_event_log/mock/mock_rtc_event_log.cc",
"rtc_event_log/mock/mock_rtc_event_log.h",
]
deps = [
":rtc_event_log_api",
"../test:test_support",
]
}
}