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webrtc_m130/modules/rtp_rtcp/BUILD.gn

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GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
# Copyright (c) 2014 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")
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
rtc_source_set("rtp_rtcp_format") {
rtp_rtcp_format: Separate public and private source files There was one .h file that didn't have to be public. :-) BUG=webrtc:8159, webrtc:8255 Change-Id: I0998f0340384c57f52affdde30f6b4eb2eaa712b Reviewed-on: https://webrtc-review.googlesource.com/2400 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19915}
2017-09-20 21:46:54 +02:00
public = [
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
"include/rtp_cvo.h",
"include/rtp_header_extension_map.h",
"include/rtp_rtcp_defines.h",
"source/byte_io.h",
Move rtcp packet classes from rtp_rtcp to rtp_rtcp_format target Bug: None Change-Id: I353228fd5b75bd4fceeaee1bb6fd07b01dac56a1 Reviewed-on: https://webrtc-review.googlesource.com/1480 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19867}
2017-09-15 16:48:14 +02:00
"source/rtcp_packet.h",
"source/rtcp_packet/app.h",
"source/rtcp_packet/bye.h",
"source/rtcp_packet/common_header.h",
"source/rtcp_packet/compound_packet.h",
"source/rtcp_packet/dlrr.h",
"source/rtcp_packet/extended_jitter_report.h",
"source/rtcp_packet/extended_reports.h",
"source/rtcp_packet/fir.h",
"source/rtcp_packet/nack.h",
"source/rtcp_packet/pli.h",
"source/rtcp_packet/psfb.h",
"source/rtcp_packet/rapid_resync_request.h",
"source/rtcp_packet/receiver_report.h",
"source/rtcp_packet/remb.h",
"source/rtcp_packet/report_block.h",
"source/rtcp_packet/rrtr.h",
"source/rtcp_packet/rtpfb.h",
"source/rtcp_packet/sdes.h",
"source/rtcp_packet/sender_report.h",
"source/rtcp_packet/target_bitrate.h",
"source/rtcp_packet/tmmb_item.h",
"source/rtcp_packet/tmmbn.h",
"source/rtcp_packet/tmmbr.h",
"source/rtcp_packet/transport_feedback.h",
"source/rtcp_packet/voip_metric.h",
Introduce list of fields to put into codec agnostic descriptor Bug: webrtc:9361 Change-Id: Iff44f289ffcecf7e4f997d5001958ab22124910f Reviewed-on: https://webrtc-review.googlesource.com/81241 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23600}
2018-06-13 15:51:43 +02:00
"source/rtp_generic_frame_descriptor.h",
Add Parsing/Building generic frame descriptor extension Bug: webrtc:9361 Change-Id: I7e85826117348e2d4f4726e8d515bb1d4a289966 Reviewed-on: https://webrtc-review.googlesource.com/83622 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23662}
2018-06-19 14:24:17 +02:00
"source/rtp_generic_frame_descriptor_extension.h",
rtp_rtcp_format: Separate public and private source files There was one .h file that didn't have to be public. :-) BUG=webrtc:8159, webrtc:8255 Change-Id: I0998f0340384c57f52affdde30f6b4eb2eaa712b Reviewed-on: https://webrtc-review.googlesource.com/2400 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19915}
2017-09-20 21:46:54 +02:00
"source/rtp_header_extensions.h",
Make rtp_packet.h public This would allow us to limit the visibility of RtpPacketReceived and RtpPacketToSend, when we only want to allocate memory to save the RTP header, and not the metadata. TBR=danilchap@webrtc.org Bug: webrtc:8111 Change-Id: Ic9339189ccc2081d82bdc8def0fb39677458356f Reviewed-on: https://webrtc-review.googlesource.com/5521 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20075}
2017-10-02 13:10:01 +02:00
"source/rtp_packet.h",
rtp_rtcp_format: Separate public and private source files There was one .h file that didn't have to be public. :-) BUG=webrtc:8159, webrtc:8255 Change-Id: I0998f0340384c57f52affdde30f6b4eb2eaa712b Reviewed-on: https://webrtc-review.googlesource.com/2400 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19915}
2017-09-20 21:46:54 +02:00
"source/rtp_packet_received.h",
"source/rtp_packet_to_send.h",
]
sources = [
Convert PayloadUnion from a union to a class, step 3 Remove PayloadUnion's public member variables, so that the outside world has to go through the accessors. This is good code hygiene in general. For example, it makes it possible to make the audio and video states Optional, so that exactly one of them can be live at any one time. BUG=webrtc:8159 Change-Id: Ie617b9038f961b329bd67b45478ff33d97148447 Reviewed-on: https://webrtc-review.googlesource.com/4428 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20064}
2017-10-02 09:54:48 +02:00
"include/rtp_rtcp_defines.cc",
rtp_rtcp_format: Separate public and private source files There was one .h file that didn't have to be public. :-) BUG=webrtc:8159, webrtc:8255 Change-Id: I0998f0340384c57f52affdde30f6b4eb2eaa712b Reviewed-on: https://webrtc-review.googlesource.com/2400 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19915}
2017-09-20 21:46:54 +02:00
"source/rtcp_packet.cc",
"source/rtcp_packet/app.cc",
"source/rtcp_packet/bye.cc",
"source/rtcp_packet/common_header.cc",
"source/rtcp_packet/compound_packet.cc",
"source/rtcp_packet/dlrr.cc",
"source/rtcp_packet/extended_jitter_report.cc",
"source/rtcp_packet/extended_reports.cc",
"source/rtcp_packet/fir.cc",
"source/rtcp_packet/nack.cc",
"source/rtcp_packet/pli.cc",
"source/rtcp_packet/psfb.cc",
"source/rtcp_packet/rapid_resync_request.cc",
"source/rtcp_packet/receiver_report.cc",
"source/rtcp_packet/remb.cc",
"source/rtcp_packet/report_block.cc",
"source/rtcp_packet/rrtr.cc",
"source/rtcp_packet/rtpfb.cc",
"source/rtcp_packet/sdes.cc",
"source/rtcp_packet/sender_report.cc",
"source/rtcp_packet/target_bitrate.cc",
"source/rtcp_packet/tmmb_item.cc",
"source/rtcp_packet/tmmbn.cc",
"source/rtcp_packet/tmmbr.cc",
"source/rtcp_packet/transport_feedback.cc",
"source/rtcp_packet/voip_metric.cc",
Introduce list of fields to put into codec agnostic descriptor Bug: webrtc:9361 Change-Id: Iff44f289ffcecf7e4f997d5001958ab22124910f Reviewed-on: https://webrtc-review.googlesource.com/81241 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23600}
2018-06-13 15:51:43 +02:00
"source/rtp_generic_frame_descriptor.cc",
Add Parsing/Building generic frame descriptor extension Bug: webrtc:9361 Change-Id: I7e85826117348e2d4f4726e8d515bb1d4a289966 Reviewed-on: https://webrtc-review.googlesource.com/83622 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23662}
2018-06-19 14:24:17 +02:00
"source/rtp_generic_frame_descriptor_extension.cc",
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
"source/rtp_header_extension_map.cc",
"source/rtp_header_extensions.cc",
"source/rtp_packet.cc",
"source/rtp_packet_received.cc",
Add application_data field(s) to RtpPacketToSend and PacketOptions. Pass pointer to application_data from RtpPacketToSend arriving via RtpSender::SendToNetwork through to Transport::SendRtp, in PacketOptions. Bug: webrtc:8906 Change-Id: Ie75013ed472710f4efcfbcc160e46a6119a1f41d Reviewed-on: https://webrtc-review.googlesource.com/55600 Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Dino Radaković <dinor@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22174}
2018-02-22 14:18:06 +01:00
"source/rtp_packet_to_send.cc",
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
]
deps = [
"..:module_api",
"../..:webrtc_common",
"../../api:array_view",
"../../api:libjingle_peerconnection_api",
RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs This is needed in the general case, now that we aim to support codecs other than those built-in to WebRTC. BUG=webrtc:8159 Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706 Reviewed-on: https://webrtc-review.googlesource.com/5380 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20136}
2017-10-04 12:38:53 +02:00
"../../api/audio_codecs:audio_codecs_api",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../../api/video:video_frame",
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
"../../common_video",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../../rtc_base:deprecation",
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
Replace rtc::Optional with absl::optional in modules/rtp_rtcp This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated using script: #!/bin/bash dir=modules/rtp_rtcp find $dir -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $dir -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc Reviewed-on: https://webrtc-review.googlesource.com/83584 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-14 12:59:38 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Implement PayloadUnion as variant instead of pair of optionals Bug: None Change-Id: I2e54f5a0561804bc59c4d4c8e35ccdaa9536b8e4 Reviewed-on: https://webrtc-review.googlesource.com/85366 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23745}
2018-06-26 10:44:13 +02:00
"//third_party/abseil-cpp/absl/types:variant",
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
]
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("rtp_rtcp") {
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 = [ "*" ]
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
sources = [
Add a FlexfecReceiver class. This class is split in interface/implementation classes, since it will be referenced from the Call level. Its purpose is to interface the erasure code decoder with a new class FlexfecReceiveStream (for received packets), as well as with the main RTP pipeline (for recovered packets). BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2392663006 Cr-Commit-Position: refs/heads/master@{#14594}
2016-10-10 16:44:57 -07:00
"include/flexfec_receiver.h",
Add FlexfecSender. This class will interface RTPSenderVideo with the underlying erasure code. It is functionally similar to ProducerFec (to be renamed UlpfecGenerator). In fact, the FlexfecSender is a friend of ProducerFec, and reuses most of its implementation. Besides the fact that FlexfecSender outputs FlexFEC packets, the main difference with ProducerFec is that FlexfecSender allocates RTP sequence numbers, whereas ProducerFec does not do this for the RED-encapsulated ULPFEC packets. This class is split as interface/implementation, since it will be owned by VideoSendStream initially. Further along, it may be owned by PacedSender. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2441613002 Cr-Commit-Position: refs/heads/master@{#14922}
2016-11-03 09:22:33 -07:00
"include/flexfec_sender.h",
modules: more interface -> include renames This changes the following module directories: * webrtc/modules/audio_conference_mixer/interface * webrtc/modules/interface * webrtc/modules/media_file/interface * webrtc/modules/rtp_rtcp/interface * webrtc/modules/utility/interface To avoid breaking downstream, I followed this recipe: 1. Copy the interface dir to a new sibling directory: include 2. Update the header guards in the include directory to match the style guide. 3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code. 4. Add a pragma warning in the header files in the interface dir. Example: #pragma message("WARNING: webrtc/modules/interface is DEPRECATED; " "use webrtc/modules/include") 5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S) 6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*) BUG=5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417683006 . Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 08:31:52 +01:00
"include/receive_statistics.h",
"include/remote_ntp_time_estimator.h",
"include/rtp_header_parser.h",
"include/rtp_payload_registry.h",
"include/rtp_receiver.h",
"include/rtp_rtcp.h",
Rename FecReceiver to UlpfecReceiver. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2451643002 Cr-Commit-Position: refs/heads/master@{#14846}
2016-10-31 04:51:33 -07:00
"include/ulpfec_receiver.h",
Refactor RtpReceiverImpl, extracting CSRC book-keeping to its own class Bug: webrtc:7135 Change-Id: I7ce9afe575241542e4e3f7e2e8459ee3257eec76 Reviewed-on: https://webrtc-review.googlesource.com/93466 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24271}
2018-08-13 13:23:08 +02:00
"source/contributing_sources.cc",
"source/contributing_sources.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/dtmf_queue.cc",
"source/dtmf_queue.h",
Move the FEC private tables into .cc files. Change the arrays to be continuous uint8_t arrays instead being of arrays of arrays (of arrays). Code size difference is 17K for arm, ~42K for arm64. New lookup algorithm, tailored for these two tables + tests. Instead of returning a raw pointer into the table, the algorithm returns an ArrayView, which includes size information for how much memory can be read. Change-Id: I000b094520bac944e518eb8b51d8dbef4670f5d7 Bug: webrtc:9102 Reviewed-on: https://webrtc-review.googlesource.com/65920 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22736}
2018-04-04 13:48:17 +02:00
"source/fec_private_tables_bursty.cc",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/fec_private_tables_bursty.h",
Move the FEC private tables into .cc files. Change the arrays to be continuous uint8_t arrays instead being of arrays of arrays (of arrays). Code size difference is 17K for arm, ~42K for arm64. New lookup algorithm, tailored for these two tables + tests. Instead of returning a raw pointer into the table, the algorithm returns an ArrayView, which includes size information for how much memory can be read. Change-Id: I000b094520bac944e518eb8b51d8dbef4670f5d7 Bug: webrtc:9102 Reviewed-on: https://webrtc-review.googlesource.com/65920 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22736}
2018-04-04 13:48:17 +02:00
"source/fec_private_tables_random.cc",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/fec_private_tables_random.h",
Add FlexFEC header formatters. Add classes that can read and finalize FlexFEC headers. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2269903002 Cr-Commit-Position: refs/heads/master@{#14469}
2016-10-03 00:43:25 -07:00
"source/flexfec_header_reader_writer.cc",
"source/flexfec_header_reader_writer.h",
Make FlexfecReceiver a concrete class. There is no need for it to be an interface. In this CL, I also took the opportunity to make two small fixes: - remove the 'flexfec_' prefix from some member variables - remove unnecessary use of a stringstream object BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2471073003 Cr-Commit-Position: refs/heads/master@{#14919}
2016-11-03 08:18:27 -07:00
"source/flexfec_receiver.cc",
Add FlexfecSender. This class will interface RTPSenderVideo with the underlying erasure code. It is functionally similar to ProducerFec (to be renamed UlpfecGenerator). In fact, the FlexfecSender is a friend of ProducerFec, and reuses most of its implementation. Besides the fact that FlexfecSender outputs FlexFEC packets, the main difference with ProducerFec is that FlexfecSender allocates RTP sequence numbers, whereas ProducerFec does not do this for the RED-encapsulated ULPFEC packets. This class is split as interface/implementation, since it will be owned by VideoSendStream initially. Further along, it may be owned by PacedSender. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2441613002 Cr-Commit-Position: refs/heads/master@{#14922}
2016-11-03 09:22:33 -07:00
"source/flexfec_sender.cc",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/forward_error_correction.cc",
"source/forward_error_correction.h",
"source/forward_error_correction_internal.cc",
"source/forward_error_correction_internal.h",
Add statistics gathering for packet loss. Adds a class used to classify whether packet loss events are a single packet or multiple packets as well as how many packets have been lost. Also exposes a new function in the RtpRtcp interface to retrieve these statistics. BUG= Review URL: https://codereview.webrtc.org/1198853004 Cr-Commit-Position: refs/heads/master@{#9568}
2015-07-10 18:10:05 -07:00
"source/packet_loss_stats.cc",
"source/packet_loss_stats.h",
Add sender controlled playout delay limits This CL adds support for an extension on RTP frames to allow the sender to specify the minimum and maximum playout delay limits. The receiver makes a best-effort attempt to keep the capture-to-render delay within this range. This allows different types of application to specify different end-to-end delay goals. For example gaming can support rendering of frames as soon as received on receiver to minimize delay. A movie playback application can specify a minimum playout delay to allow fixed buffering in presence of network jitter. There are no tests at this time and most of testing is done with chromium webrtc prototype. On chromoting performance tests, this extension helps bring down end-to-end delay by about 150 ms on small frames. BUG=webrtc:5895 Review-Url: https://codereview.webrtc.org/2007743003 Cr-Commit-Position: refs/heads/master@{#13059}
2016-06-08 00:24:21 -07:00
"source/playout_delay_oracle.cc",
"source/playout_delay_oracle.h",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"source/receive_statistics_impl.cc",
"source/receive_statistics_impl.h",
"source/remote_ntp_time_estimator.cc",
Remove rtcp_utility as mostly unused. Since the only used class is RTCPUtilitiy::NackStats, rename it to RtcpNackStats and move it into dedicated file. BUG=webrtc:5565 Review-Url: https://codereview.webrtc.org/2680183004 Cr-Commit-Position: refs/heads/master@{#16515}
2017-02-09 05:21:42 -08:00
"source/rtcp_nack_stats.cc",
"source/rtcp_nack_stats.h",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"source/rtcp_receiver.cc",
"source/rtcp_receiver.h",
"source/rtcp_sender.cc",
"source/rtcp_sender.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/rtp_format.cc",
"source/rtp_format.h",
"source/rtp_format_h264.cc",
"source/rtp_format_h264.h",
"source/rtp_format_video_generic.cc",
"source/rtp_format_video_generic.h",
"source/rtp_format_vp8.cc",
"source/rtp_format_vp8.h",
Add support for VP9 packetization/depacketization. RTP payload format for VP9: https://www.ietf.org/id/draft-uberti-payload-vp9-01.txt BUG=webrtc:4148, webrtc:4168, chromium:500602 TBR=mflodman Review URL: https://codereview.webrtc.org/1232023006 Cr-Commit-Position: refs/heads/master@{#9649}
2015-07-28 04:02:54 -07:00
"source/rtp_format_vp9.cc",
"source/rtp_format_vp9.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/rtp_header_parser.cc",
"source/rtp_packet_history.cc",
"source/rtp_packet_history.h",
"source/rtp_payload_registry.cc",
"source/rtp_receiver_audio.cc",
"source/rtp_receiver_audio.h",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"source/rtp_receiver_impl.cc",
"source/rtp_receiver_impl.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/rtp_receiver_strategy.cc",
"source/rtp_receiver_strategy.h",
"source/rtp_receiver_video.cc",
"source/rtp_receiver_video.h",
"source/rtp_rtcp_config.h",
"source/rtp_rtcp_impl.cc",
"source/rtp_rtcp_impl.h",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"source/rtp_sender.cc",
"source/rtp_sender.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/rtp_sender_audio.cc",
"source/rtp_sender_audio.h",
"source/rtp_sender_video.cc",
"source/rtp_sender_video.h",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"source/rtp_utility.cc",
"source/rtp_utility.h",
rtt calculation handles time go backwards CompactNtpIntervalToMs renamed to CompactNtpRttToMs and handle special cases: large values consider negative/invalid and result in value of 1. 0 result consider too small and increases to 1. BUG=590996 R=asapersson@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1763823003 . Cr-Commit-Position: refs/heads/master@{#11928}
2016-03-09 15:14:35 +01:00
"source/time_util.cc",
"source/time_util.h",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"source/tmmbr_help.cc",
"source/tmmbr_help.h",
Rename ProducerFec to UlpfecGenerator. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2449783002 Cr-Commit-Position: refs/heads/master@{#14847}
2016-10-31 05:27:07 -07:00
"source/ulpfec_generator.cc",
"source/ulpfec_generator.h",
Generalize FEC header formatting. - Split out reading/writing of FEC headers to classes separate from ForwardErrorCorrection. This makes ForwardErrorCorrection oblivious to what FEC header scheme is used, and lets it focus on encoding/decoding the FEC payloads. - Add unit tests for FEC header readers/writers. - Split ForwardErrorCorrection::XorPackets into XorHeaders and XorPayloads and reuse these functions for both encoding and decoding. - Rename AttemptRecover -> AttemptRecovery in ForwardErrorCorrection. BUG=webrtc:5654 R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2260803002 . Cr-Commit-Position: refs/heads/master@{#14316}
2016-09-21 09:19:34 +02:00
"source/ulpfec_header_reader_writer.cc",
"source/ulpfec_header_reader_writer.h",
Rename FecReceiver to UlpfecReceiver. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2451643002 Cr-Commit-Position: refs/heads/master@{#14846}
2016-10-31 04:51:33 -07:00
"source/ulpfec_receiver_impl.cc",
"source/ulpfec_receiver_impl.h",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
]
Enable BWE logging to command line when rtc_enable_bwe_test_logging is set to true This patch enables bwe related variable logging to the command line. This is useful to test congestion control algorithm over real networks. NOTRY=true Review-Url: https://codereview.webrtc.org/2296253002 Cr-Commit-Position: refs/heads/master@{#14209}
2016-09-14 05:04:36 -07:00
if (rtc_enable_bwe_test_logging) {
defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
} else {
defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
}
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
deps = [
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
":rtp_rtcp_format",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Fix style violations in common_types.h and config.h Mostly, it's about moving constructors and descructors to the .cc files, so that they won't be inlined everywhere. The reason this CL is so big is that a lot of code was using common_types.h without declaring a dependency on webrtc_common, which broke the build once common_types.h started to depend on common_types.cc. BUG=163 R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26089004 Cr-Commit-Position: refs/heads/master@{#8516} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 13:59:22 +00:00
"../..:webrtc_common",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) Reason for revert: Re-land, reverting did not fix bug. https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 Original issue's description: > Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) > > Reason for revert: > Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see > > https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 > > Original issue's description: > > Added the GetSources() to the RtpReceiverInterface and implemented > > it for the AudioRtpReceiver. > > > > This method returns a vector of RtpSource(both CSRC source and SSRC > > source) which contains the ID of a source, the timestamp, the source > > type (SSRC or CSRC) and the audio level. > > > > The RtpSource objects are buffered and maintained by the > > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called, > > the info of the contributing source will be pulled along the object > > chain: > > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel -> > > AudioReceiveStream -> voe::Channel -> RtpRtcp module > > > > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource > > > > BUG=chromium:703122 > > TBR=stefan@webrtc.org, danilchap@webrtc.org > > > > Review-Url: https://codereview.webrtc.org/2770233003 > > Cr-Commit-Position: refs/heads/master@{#17591} > > Committed: https://chromium.googlesource.com/external/webrtc/+/292084c3765d9f3ee406ca2ec86eae206b540053 > > TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=chromium:703122 > > Review-Url: https://codereview.webrtc.org/2809613002 > Cr-Commit-Position: refs/heads/master@{#17616} > Committed: https://chromium.googlesource.com/external/webrtc/+/fbcc5cb3869d1370008e40f24fc03ac8fb69c675 TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:703122 Review-Url: https://codereview.webrtc.org/2810623003 Cr-Commit-Position: refs/heads/master@{#17621}
2017-04-10 07:39:05 -07:00
"../../api:libjingle_peerconnection_api",
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. transport.h defines an interface for sending rtp and rtcp packets, which is used by MediaChannel in webrtc/media/engine, {Audio|Video}{Send|Receive}Stream and in a few other places. It was part of the build target //webrtc:webrtc, which is a monolithic target with all webrtc production code. This CL moves the header to its own target in webrtc/api and deprecates the old location. Targets in webrtc/api should in general only depend on other targets in webrtc/api. The target webrtc/api:call_api depends on transport.h. This change also makes webrtc/voice_engine pass GN's header include checker and is needed in order for webrtc/api:call_api to pass it. transport.h will be completely removed in a follow-up CL in a few weeks after clients have updated their includes. NOTRY=True BUG=webrtc:5589, webrtc:5878, webrtc:6785 Review-Url: https://codereview.webrtc.org/2426563003 Cr-Commit-Position: refs/heads/master@{#15267}
2016-11-28 07:02:13 -08:00
"../../api:transport_api",
Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2774833003 Cr-Commit-Position: refs/heads/master@{#17391}
2017-03-27 07:15:49 -07:00
"../../api/audio_codecs:audio_codecs_api",
Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets Bug: webrtc:9271 Change-Id: Ic7415830588bef9d87bab92943460207890dada6 Reviewed-on: https://webrtc-review.googlesource.com/76960 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 12:35:56 +02:00
"../../api/video:video_bitrate_allocation",
Exposing video bitrate allocator into API In order to have public video bitrate allocator factory, the video bitrate allocator has be part of the api. Bug: webrtc:9513 Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2 Reviewed-on: https://webrtc-review.googlesource.com/88083 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#24073}
2018-07-20 15:49:43 -07:00
"../../api/video:video_bitrate_allocator",
Add unlimited retransmission experiment for screenshare Bug: webrtc:9659 Change-Id: Idcdc647c112ed2c7c027a7a0056b145ce8f45788 Reviewed-on: https://webrtc-review.googlesource.com/95724 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24422}
2018-08-23 17:07:29 +02:00
"../../api/video:video_frame",
Reland "Move class VideoCodec from common_types.h to its own api header file." This is a reland of efc71e565e9b36bcdfb4571f59e34bbd8fabd0cd Differs from the original cl by not widening the type of VideoCodec::width and VideoCodec::height. Original change's description: > Move class VideoCodec from common_types.h to its own api header file. > > Bug: webrtc:7660 > Change-Id: I91f19bfc2565461328f30081f8383e136419aefb > Reviewed-on: https://webrtc-review.googlesource.com/79881 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23544} Bug: webrtc:7660 Change-Id: I7cf74a85a61ea2b831e6f32b3b3e17514ebefec8 Reviewed-on: https://webrtc-review.googlesource.com/82140 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23569}
2018-06-08 14:03:44 +02:00
"../../api/video_codecs:video_codecs_api",
Add H264 bitstream rewriting to limit frame reordering marker in header The VUI part an SPS may specify max_num_reorder_frames and max_dec_frame_buffering. These may cause a decoder to buffer a number of frame prior allowing decode, leading to delays, even if no frames using such references (ie B-frames) are sent. Because of this we update any SPS block emitted by the encoder. Also, a bunch of refactoring of H264-related code to reduce code duplication. BUG= Review-Url: https://codereview.webrtc.org/1979443004 Cr-Commit-Position: refs/heads/master@{#13010}
2016-06-02 02:43:32 -07:00
"../../common_video",
Reland "Break up rtc_event_log_api to solve circular dependencies." This is a reland of 001546da953275c7a39eb220592b440c9b47d756 Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@google.com> > Cr-Commit-Position: refs/heads/master@{#21811} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 11:04:46 -08:00
"../../logging:rtc_event_audio",
Refactor webrtc/modules/rtp_rtcp for GN check This moves some GN check configurations out of .gn to individual targets. This commit also removes the source file 'mocks/mock_rtp_rtcp.h' from the static_library 'rtp_rtcp' because it depends on a 'testonly = true' target. After a check this seems only included in the unitest code: $ grep -Rn "mocks/mock_rtp_rtcp.h" webrtc/modules/rtp_rtcp/ webrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc:18:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc:17:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" This commit also removes the dependency on '//webrt/modules/video_coding' because it seems that the following include can be removed: #include "webrtc/modules/video_coding/include/video_coding_defines.h" The now checked target is: "//webrtc/modules/rtp_rtcp/*" BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2598963002 Cr-Commit-Position: refs/heads/master@{#15760}
2016-12-22 07:35:39 -08:00
"../../logging:rtc_event_log_api",
Reland "Break up rtc_event_log_api to solve circular dependencies." This is a reland of 001546da953275c7a39eb220592b440c9b47d756 Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@google.com> > Cr-Commit-Position: refs/heads/master@{#21811} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 11:04:46 -08:00
"../../logging:rtc_event_rtp_rtcp",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../../rtc_base:deprecation",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:gtest_prod",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:rate_limiter",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Reland of "Use moving median filters in RemoteNtpTimeEstimator, RtpToNtpEstimator"" Use moving median filters in RemoteNtpTimeEstimator, RtpToNtpEstimator If Webrtc-ClockEstimation experiment is enabled, median filtering is applied to results of RtpToNtpEstimator and RemoteNtpEstimator to smooth out random errors introduced by incorrect RTCP SR reports and networking delays. Bug: webrtc:8468 Change-Id: I592c4083fefc0dbdebe7b3ff30b92e95ba337595 NOTRY=TRUE NOPRESUBMIT=TRUE Change-Id: I592c4083fefc0dbdebe7b3ff30b92e95ba337595 Reviewed-on: https://webrtc-review.googlesource.com/23263 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20708}
2017-11-15 16:48:04 +00:00
"../../rtc_base:rtc_numerics",
Separate build target for rtc_base/numerics/safe_minmax.h So that we can avoid dependency cycles. Bug: none Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8 Reviewed-on: https://webrtc-review.googlesource.com/60500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:18:56 +01:00
"../../rtc_base:safe_minmax",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:sequenced_task_checker",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../../rtc_base:stringutils",
Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix Bug: chromium:805946 Change-Id: Ibb5dce9af27d0e48c9aee6b0a860b6f62b3c76a0 Reviewed-on: https://webrtc-review.googlesource.com/46145 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21889}
2018-02-05 10:33:35 +01:00
"../../rtc_base/system:fallthrough",
Move timestamp_extrapolator.h to rtc_base/time/ This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. BUG=webrtc:8445 Change-Id: I51dfe8879c28c91bd1c667fc47b4892373671e0f Reviewed-on: https://webrtc-review.googlesource.com/21540 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22569}
2018-03-22 15:29:03 +01:00
"../../rtc_base/time:timestamp_extrapolator",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"../../system_wrappers",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:field_trial_api",
"../../system_wrappers:metrics_api",
Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2774833003 Cr-Commit-Position: refs/heads/master@{#17391}
2017-03-27 07:15:49 -07:00
"../audio_coding:audio_format_conversion",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"../remote_bitrate_estimator",
in RtpPacketizerVp8 factor out payload splitter function so that it can be shared between different packetizers and thus easier to extend Bug: webrtc:9680 Change-Id: Ie5e904ad27afb8dd2ed35ef9e009f7f408017b2f Reviewed-on: https://webrtc-review.googlesource.com/97661 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24555}
2018-09-04 16:11:58 +02:00
"//third_party/abseil-cpp/absl/container:inlined_vector",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Replace rtc::Optional with absl::optional in modules/rtp_rtcp This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated using script: #!/bin/bash dir=modules/rtp_rtcp find $dir -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $dir -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc Reviewed-on: https://webrtc-review.googlesource.com/83584 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-14 12:59:38 +02:00
"//third_party/abseil-cpp/absl/types:optional",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
]
}
GN target for test_packet_masks_metrics. NOTRY=True BUG=webrtc:6193 Review-Url: https://codereview.webrtc.org/2229473002 Cr-Commit-Position: refs/heads/master@{#13672}
2016-08-08 08:08:53 -07:00
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
rtc_source_set("rtcp_transceiver") {
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 = [ "*" ]
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
public = [
Add Thread-safe wrapper for RtcpTransceiver Bug: webrtc:8239 Change-Id: I4cc2f7f2b27c764e1aae734f933902102b345614 Reviewed-on: https://webrtc-review.googlesource.com/21680 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20714}
2017-11-16 14:35:32 +01:00
"source/rtcp_transceiver.h",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
"source/rtcp_transceiver_config.h",
"source/rtcp_transceiver_impl.h",
]
sources = [
Add Thread-safe wrapper for RtcpTransceiver Bug: webrtc:8239 Change-Id: I4cc2f7f2b27c764e1aae734f933902102b345614 Reviewed-on: https://webrtc-review.googlesource.com/21680 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20714}
2017-11-16 14:35:32 +01:00
"source/rtcp_transceiver.cc",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
"source/rtcp_transceiver_config.cc",
"source/rtcp_transceiver_impl.cc",
]
deps = [
":rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
":rtp_rtcp_format",
Add rtcp observers for media receiver to RtcpTransceiverImpl Bug: webrtc:8239 Change-Id: I7b6735f2efb87e303d1b8076c965a751db4af250 Reviewed-on: https://webrtc-review.googlesource.com/31980 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21240}
2017-12-13 12:26:17 +01:00
"../../:webrtc_common",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
"../../api:array_view",
Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets Bug: webrtc:9271 Change-Id: Ic7415830588bef9d87bab92943460207890dada6 Reviewed-on: https://webrtc-review.googlesource.com/76960 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 12:35:56 +02:00
"../../api:libjingle_peerconnection_api",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
"../../api:transport_api",
Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets Bug: webrtc:9271 Change-Id: Ic7415830588bef9d87bab92943460207890dada6 Reviewed-on: https://webrtc-review.googlesource.com/76960 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 12:35:56 +02:00
"../../api/video:video_bitrate_allocation",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
"../../rtc_base:rtc_base_approved",
Replace WeakPtr with CancelablePeriodicTask in RtcpTransceiverImpl This allow to destroy the RtcpTransceiverImpl off the task queue with assumption it is destroyed after task queue. i.e. it allows to post destruction of RtcpTransceiverImpl to the TaskQueue without waiting. BUG: webrtc:8239 Change-Id: I4bea7a6d2edc97061ebd00f2f275c1ab827bc3c5 Reviewed-on: https://webrtc-review.googlesource.com/97160 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24574}
2018-09-05 12:24:07 +02:00
"../../rtc_base:rtc_cancelable_task",
Add periodic compound packet sending to RtcpTransceiver Bug: webrtc:8239 Change-Id: I1511db63a15e8c5101a933e55e66d3877ff963be Reviewed-on: https://webrtc-review.googlesource.com/15440 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20480}
2017-10-30 15:21:41 +01:00
"../../rtc_base:rtc_task_queue",
"../../rtc_base:weak_ptr",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Replace rtc::Optional with absl::optional in modules/rtp_rtcp This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated using script: #!/bin/bash dir=modules/rtp_rtcp find $dir -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $dir -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc Reviewed-on: https://webrtc-review.googlesource.com/83584 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-14 12:59:38 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
]
}
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. Bug: none Change-Id: If28f57c5ae250afbb47c5d20c9854e9a11182642 Reviewed-on: https://webrtc-review.googlesource.com/87561 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23904}
2018-07-09 15:52:29 +02:00
rtc_source_set("rtp_video_header") {
visibility = [ "*" ]
sources = [
"source/rtp_video_header.cc",
"source/rtp_video_header.h",
]
deps = [
"../../:webrtc_common",
"../../api/video:video_frame",
"../../modules/video_coding:codec_globals_headers",
Add members for the codec agnostic descriptor to RTPVideoHeader. TBR=danilchap@webrtc.org Bug: webrtc:9361, webrtc:9582 Change-Id: I0303fc89bafab59e68ec81979e0e4372e79a4f51 Reviewed-on: https://webrtc-review.googlesource.com/91866 Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24170}
2018-08-01 17:13:08 +02:00
"//third_party/abseil-cpp/absl/container:inlined_vector",
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. Bug: none Change-Id: If28f57c5ae250afbb47c5d20c9854e9a11182642 Reviewed-on: https://webrtc-review.googlesource.com/87561 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23904}
2018-07-09 15:52:29 +02:00
"//third_party/abseil-cpp/absl/types:variant",
]
}
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
rtc_source_set("fec_test_helper") {
testonly = true
sources = [
"source/fec_test_helper.cc",
"source/fec_test_helper.h",
]
deps = [
":rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
":rtp_rtcp_format",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
]
}
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
rtc_source_set("mock_rtp_rtcp") {
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
"mocks/mock_recovered_packet_receiver.cc",
"mocks/mock_rtcp_bandwidth_observer.cc",
"mocks/mock_rtcp_rtt_stats.cc",
"mocks/mock_rtp_rtcp.cc",
]
public = [
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"mocks/mock_recovered_packet_receiver.h",
Using mock transport controller in audio unit tests. Using a mock of rtp transport controller send in audio send stream unit tests. This reduces the dependencies and makes the tests more focused on testing the functionality of audio send stream itself. Bug: webrtc:8415 Change-Id: Ia8d9cf47d93decc74b10ca75a6771f39df658dc2 Reviewed-on: https://webrtc-review.googlesource.com/56600 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22161}
2018-02-22 14:49:02 +01:00
"mocks/mock_rtcp_bandwidth_observer.h",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"mocks/mock_rtcp_rtt_stats.h",
"mocks/mock_rtp_rtcp.h",
]
deps = [
":rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
":rtp_rtcp_format",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"..:module_api",
Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets Bug: webrtc:9271 Change-Id: Ic7415830588bef9d87bab92943460207890dada6 Reviewed-on: https://webrtc-review.googlesource.com/76960 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 12:35:56 +02:00
"../../api/video:video_bitrate_allocation",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../../test:test_support",
Replace rtc::Optional with absl::optional in modules/rtp_rtcp This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated using script: #!/bin/bash dir=modules/rtp_rtcp find $dir -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $dir -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc Reviewed-on: https://webrtc-review.googlesource.com/83584 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-14 12:59:38 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
]
}
GN target for test_packet_masks_metrics. NOTRY=True BUG=webrtc:6193 Review-Url: https://codereview.webrtc.org/2229473002 Cr-Commit-Position: refs/heads/master@{#13672}
2016-08-08 08:08:53 -07:00
if (rtc_include_tests) {
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("test_packet_masks_metrics") {
GN target for test_packet_masks_metrics. NOTRY=True BUG=webrtc:6193 Review-Url: https://codereview.webrtc.org/2229473002 Cr-Commit-Position: refs/heads/master@{#13672}
2016-08-08 08:08:53 -07:00
testonly = true
sources = [
"test/testFec/average_residual_loss_xor_codes.h",
"test/testFec/test_packet_masks_metrics.cc",
]
deps = [
":rtp_rtcp",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Use relative paths in GN files. BUG=webrtc:7952 TBR=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2974863003 Cr-Commit-Position: refs/heads/master@{#18970}
2017-07-11 06:20:45 -07:00
"../../test:test_main",
GN target for test_packet_masks_metrics. NOTRY=True BUG=webrtc:6193 Review-Url: https://codereview.webrtc.org/2229473002 Cr-Commit-Position: refs/heads/master@{#13672}
2016-08-08 08:08:53 -07:00
"//testing/gtest",
]
} # test_packet_masks_metrics
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
rtc_source_set("rtp_rtcp_modules_tests") {
testonly = true
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
sources = [
"test/testFec/test_fec.cc",
]
deps = [
":rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
":rtp_rtcp_format",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"../../test:test_support",
]
}
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
rtc_source_set("rtp_rtcp_unittests") {
testonly = true
sources = [
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/byte_io_unittest.cc",
Refactor RtpReceiverImpl, extracting CSRC book-keeping to its own class Bug: webrtc:7135 Change-Id: I7ce9afe575241542e4e3f7e2e8459ee3257eec76 Reviewed-on: https://webrtc-review.googlesource.com/93466 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24271}
2018-08-13 13:23:08 +02:00
"source/contributing_sources_unittest.cc",
Move the FEC private tables into .cc files. Change the arrays to be continuous uint8_t arrays instead being of arrays of arrays (of arrays). Code size difference is 17K for arm, ~42K for arm64. New lookup algorithm, tailored for these two tables + tests. Instead of returning a raw pointer into the table, the algorithm returns an ArrayView, which includes size information for how much memory can be read. Change-Id: I000b094520bac944e518eb8b51d8dbef4670f5d7 Bug: webrtc:9102 Reviewed-on: https://webrtc-review.googlesource.com/65920 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22736}
2018-04-04 13:48:17 +02:00
"source/fec_private_tables_bursty_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/flexfec_header_reader_writer_unittest.cc",
"source/flexfec_receiver_unittest.cc",
"source/flexfec_sender_unittest.cc",
"source/nack_rtx_unittest.cc",
"source/packet_loss_stats_unittest.cc",
"source/playout_delay_oracle_unittest.cc",
"source/receive_statistics_unittest.cc",
"source/remote_ntp_time_estimator_unittest.cc",
Remove rtcp_utility as mostly unused. Since the only used class is RTCPUtilitiy::NackStats, rename it to RtcpNackStats and move it into dedicated file. BUG=webrtc:5565 Review-Url: https://codereview.webrtc.org/2680183004 Cr-Commit-Position: refs/heads/master@{#16515}
2017-02-09 05:21:42 -08:00
"source/rtcp_nack_stats_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/rtcp_packet/app_unittest.cc",
"source/rtcp_packet/bye_unittest.cc",
"source/rtcp_packet/common_header_unittest.cc",
"source/rtcp_packet/compound_packet_unittest.cc",
"source/rtcp_packet/dlrr_unittest.cc",
"source/rtcp_packet/extended_jitter_report_unittest.cc",
"source/rtcp_packet/extended_reports_unittest.cc",
"source/rtcp_packet/fir_unittest.cc",
"source/rtcp_packet/nack_unittest.cc",
"source/rtcp_packet/pli_unittest.cc",
"source/rtcp_packet/rapid_resync_request_unittest.cc",
"source/rtcp_packet/receiver_report_unittest.cc",
"source/rtcp_packet/remb_unittest.cc",
"source/rtcp_packet/report_block_unittest.cc",
"source/rtcp_packet/rrtr_unittest.cc",
"source/rtcp_packet/sdes_unittest.cc",
"source/rtcp_packet/sender_report_unittest.cc",
"source/rtcp_packet/target_bitrate_unittest.cc",
"source/rtcp_packet/tmmbn_unittest.cc",
"source/rtcp_packet/tmmbr_unittest.cc",
"source/rtcp_packet/transport_feedback_unittest.cc",
"source/rtcp_packet/voip_metric_unittest.cc",
"source/rtcp_packet_unittest.cc",
"source/rtcp_receiver_unittest.cc",
"source/rtcp_sender_unittest.cc",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
"source/rtcp_transceiver_impl_unittest.cc",
Add Thread-safe wrapper for RtcpTransceiver Bug: webrtc:8239 Change-Id: I4cc2f7f2b27c764e1aae734f933902102b345614 Reviewed-on: https://webrtc-review.googlesource.com/21680 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20714}
2017-11-16 14:35:32 +01:00
"source/rtcp_transceiver_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/rtp_fec_unittest.cc",
"source/rtp_format_h264_unittest.cc",
in RtpPacketizerVp8 factor out payload splitter function so that it can be shared between different packetizers and thus easier to extend Bug: webrtc:9680 Change-Id: Ie5e904ad27afb8dd2ed35ef9e009f7f408017b2f Reviewed-on: https://webrtc-review.googlesource.com/97661 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24555}
2018-09-04 16:11:58 +02:00
"source/rtp_format_unittest.cc",
Fix packetization logic to leave space for extensions in the last packet Change packetizer interface to explicitly return number of packets instead of a last flag. Account for extra space needed in the last packet. BUG=webrtc:7588,webrtc:7594 Review-Url: https://codereview.webrtc.org/2871173008 Cr-Commit-Position: refs/heads/master@{#18244}
2017-05-23 09:34:21 -07:00
"source/rtp_format_video_generic_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/rtp_format_vp8_test_helper.cc",
"source/rtp_format_vp8_test_helper.h",
"source/rtp_format_vp8_unittest.cc",
"source/rtp_format_vp9_unittest.cc",
Add Parsing/Building generic frame descriptor extension Bug: webrtc:9361 Change-Id: I7e85826117348e2d4f4726e8d515bb1d4a289966 Reviewed-on: https://webrtc-review.googlesource.com/83622 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23662}
2018-06-19 14:24:17 +02:00
"source/rtp_generic_frame_descriptor_extension_unittest.cc",
Rename rtp_header_extension.h to rtp_header_extension_map.h Move it to include path of the rtp_rtcp module to indicate it is ok to include it outside of the module. Renamed to match the class it introduce and to reduce confusion with rtp_header_extensions.h Bug: webrtc:5565 Change-Id: Ic4b4e9f6b75cb9275e23539cd6e88632c1e7c8d2 Reviewed-on: https://chromium-review.googlesource.com/520947 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18402}
2017-06-01 17:10:51 +02:00
"source/rtp_header_extension_map_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/rtp_packet_history_unittest.cc",
"source/rtp_packet_unittest.cc",
"source/rtp_payload_registry_unittest.cc",
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) Reason for revert: Re-land, reverting did not fix bug. https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 Original issue's description: > Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) > > Reason for revert: > Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see > > https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 > > Original issue's description: > > Added the GetSources() to the RtpReceiverInterface and implemented > > it for the AudioRtpReceiver. > > > > This method returns a vector of RtpSource(both CSRC source and SSRC > > source) which contains the ID of a source, the timestamp, the source > > type (SSRC or CSRC) and the audio level. > > > > The RtpSource objects are buffered and maintained by the > > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called, > > the info of the contributing source will be pulled along the object > > chain: > > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel -> > > AudioReceiveStream -> voe::Channel -> RtpRtcp module > > > > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource > > > > BUG=chromium:703122 > > TBR=stefan@webrtc.org, danilchap@webrtc.org > > > > Review-Url: https://codereview.webrtc.org/2770233003 > > Cr-Commit-Position: refs/heads/master@{#17591} > > Committed: https://chromium.googlesource.com/external/webrtc/+/292084c3765d9f3ee406ca2ec86eae206b540053 > > TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=chromium:703122 > > Review-Url: https://codereview.webrtc.org/2809613002 > Cr-Commit-Position: refs/heads/master@{#17616} > Committed: https://chromium.googlesource.com/external/webrtc/+/fbcc5cb3869d1370008e40f24fc03ac8fb69c675 TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:703122 Review-Url: https://codereview.webrtc.org/2810623003 Cr-Commit-Position: refs/heads/master@{#17621}
2017-04-10 07:39:05 -07:00
"source/rtp_receiver_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/rtp_rtcp_impl_unittest.cc",
"source/rtp_sender_unittest.cc",
"source/rtp_utility_unittest.cc",
"source/time_util_unittest.cc",
"source/ulpfec_generator_unittest.cc",
"source/ulpfec_header_reader_writer_unittest.cc",
"source/ulpfec_receiver_unittest.cc",
"test/testAPI/test_api.cc",
"test/testAPI/test_api.h",
"test/testAPI/test_api_audio.cc",
"test/testAPI/test_api_rtcp.cc",
"test/testAPI/test_api_video.cc",
]
deps = [
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
":fec_test_helper",
":mock_rtp_rtcp",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
":rtcp_transceiver",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
":rtp_rtcp_format",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../..:webrtc_common",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Move RtpExtension to api/ directory and config.h/.cc to call/. BUG=webrtc:5876 R=deadbeef@webrtc.org, solenberg@webrtc.org Review-Url: https://codereview.webrtc.org/3004723002 . Cr-Commit-Position: refs/heads/master@{#19639}
2017-09-01 15:29:28 +02:00
"../../api:libjingle_peerconnection_api",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../api:transport_api",
Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets Bug: webrtc:9271 Change-Id: Ic7415830588bef9d87bab92943460207890dada6 Reviewed-on: https://webrtc-review.googlesource.com/76960 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 12:35:56 +02:00
"../../api/video:video_bitrate_allocation",
Exposing video bitrate allocator into API In order to have public video bitrate allocator factory, the video bitrate allocator has be part of the api. Bug: webrtc:9513 Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2 Reviewed-on: https://webrtc-review.googlesource.com/88083 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#24073}
2018-07-20 15:49:43 -07:00
"../../api/video:video_bitrate_allocator",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../../api/video:video_frame",
Reland "Move class VideoCodec from common_types.h to its own api header file." This is a reland of efc71e565e9b36bcdfb4571f59e34bbd8fabd0cd Differs from the original cl by not widening the type of VideoCodec::width and VideoCodec::height. Original change's description: > Move class VideoCodec from common_types.h to its own api header file. > > Bug: webrtc:7660 > Change-Id: I91f19bfc2565461328f30081f8383e136419aefb > Reviewed-on: https://webrtc-review.googlesource.com/79881 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23544} Bug: webrtc:7660 Change-Id: I7cf74a85a61ea2b831e6f32b3b3e17514ebefec8 Reviewed-on: https://webrtc-review.googlesource.com/82140 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23569}
2018-06-08 14:03:44 +02:00
"../../api/video_codecs:video_codecs_api",
Reland of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #1 id:1 of https://codereview.webrtc.org/3011093002/ ) Reason for revert: The cl this change depended on has now been successfully relanded. Original issue's description: > Revert of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #3 id:40001 of https://codereview.webrtc.org/3006993002/ ) > > Reason for revert: > This has to be reverted to enable reverting cl https://codereview.webrtc.org/3006063002/, which seems to have broken ulpfec. > > Original issue's description: > > Delete Rtx-related methods from RTPPayloadRegistry. > > > > Delete methods IsRtx, IsEncapsulated and RestoreOriginalPacket. > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/3006993002 > > Cr-Commit-Position: refs/heads/master@{#19739} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5b4b52264132eefba10bc6880871715f9692da90 > > TBR=stefan@webrtc.org,danilchap@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/3011093002 > Cr-Commit-Position: refs/heads/master@{#19742} > Committed: https://chromium.googlesource.com/external/webrtc/+/a64685325c2f8f51873b67ae8a91f94ffb19d70b TBR=stefan@webrtc.org,danilchap@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/3012253002 Cr-Commit-Position: refs/heads/master@{#19821}
2017-09-13 07:13:57 -07:00
"../../call:rtp_receiver",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../common_video:common_video",
Reland: Add mock_rtc_event_log.h. Bug: webrtc:7642 Change-Id: I3f97a8b603e34819e1563b335c1eeb51f89b11ac Reviewed-on: https://webrtc-review.googlesource.com/37081 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21480}
2018-01-03 09:08:20 +01:00
"../../logging:mocks",
Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead RtcEventLogImpl no longer hard-codes the way encoding is done. It now relies on RtcEventEncoder for it. This gives two benefits: 1. We can decide between the current encoding and the new encoding (which is still WIP) without code duplication (no need for RtcEventLogImplNew). 2. Encoding is done only when the event needs to be written to a file. This both avoids unnecessary encoding of events which don't end up getting written to a file, as well as is useful for the new, delta-based encoding, which is stateful. BUG=webrtc:8111 Change-Id: I9517132e5f96b8059002a66fde8d42d3a678c3bb Reviewed-on: https://webrtc-review.googlesource.com/1365 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20118}
2017-10-03 16:11:34 +02:00
"../../logging:rtc_event_log_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
"../../rtc_base:rate_limiter",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Add Incoming packet handler to RtcpTransceiver Bug: webrtc:8239 Change-Id: Iebe1b6a2649f01e4f6e3780ac96cb05611d8671c Reviewed-on: https://webrtc-review.googlesource.com/17560 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20624}
2017-11-09 15:42:28 +01:00
"../../rtc_base:rtc_base_tests_utils",
Add periodic compound packet sending to RtcpTransceiver Bug: webrtc:8239 Change-Id: I1511db63a15e8c5101a933e55e66d3877ff963be Reviewed-on: https://webrtc-review.googlesource.com/15440 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20480}
2017-10-30 15:21:41 +01:00
"../../rtc_base:rtc_task_queue",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../test:field_trial",
"../../test:rtp_test_utils",
"../../test:test_common",
"../../test:test_support",
RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs This is needed in the general case, now that we aim to support codecs other than those built-in to WebRTC. BUG=webrtc:8159 Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706 Reviewed-on: https://webrtc-review.googlesource.com/5380 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20136}
2017-10-04 12:38:53 +02:00
"../audio_coding:audio_format_conversion",
Cleanup RtpPacketizerVP8 tests Remove partition support for test helper and from tests. Merge Init function into constructor Replace extra macroses in favor of Bit helper function Replace extra members in favor of local variables Remove fixture Bug: None Change-Id: Ibf1600dda9f59abe5afd2bbe40c3e232a2d269ea Reviewed-on: https://webrtc-review.googlesource.com/96940 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24508}
2018-08-30 20:54:58 +02:00
"../video_coding:codec_globals_headers",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Replace rtc::Optional with absl::optional in modules/rtp_rtcp This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated using script: #!/bin/bash dir=modules/rtp_rtcp find $dir -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $dir -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc Reviewed-on: https://webrtc-review.googlesource.com/83584 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-14 12:59:38 +02:00
"//third_party/abseil-cpp/absl/types:optional",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
]
}
GN target for test_packet_masks_metrics. NOTRY=True BUG=webrtc:6193 Review-Url: https://codereview.webrtc.org/2229473002 Cr-Commit-Position: refs/heads/master@{#13672}
2016-08-08 08:08:53 -07:00
}
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