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webrtc_m130/call/BUILD.gn

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Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
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")
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
Moved call.h and most of api/call/* into call/ BUG=webrtc:6716 Review-Url: https://codereview.webrtc.org/2550273003 Cr-Commit-Position: refs/heads/master@{#15460}
2016-12-07 04:52:58 -08:00
rtc_source_set("call_interfaces") {
sources = [
"audio_receive_stream.h",
"audio_send_stream.cc",
"audio_send_stream.h",
"audio_state.h",
"call.h",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"callfactoryinterface.h",
Move FlexfecReceiveStream from api/call/ to call/. Also rename internal::FlexfecReceiveStream to FlexfecReceiveStreamImpl. BUG=webrtc:6849 Review-Url: https://codereview.webrtc.org/2561123002 Cr-Commit-Position: refs/heads/master@{#15666}
2016-12-19 01:13:46 -08:00
"flexfec_receive_stream.h",
Stop using VoEVideoSync in Call/VideoReceiveStream. BUG=webrtc:4690 Review-Url: https://codereview.webrtc.org/2452163004 Cr-Commit-Position: refs/heads/master@{#16375}
2017-01-31 03:58:40 -08:00
"syncable.cc",
"syncable.h",
Moved call.h and most of api/call/* into call/ BUG=webrtc:6716 Review-Url: https://codereview.webrtc.org/2550273003 Cr-Commit-Position: refs/heads/master@{#15460}
2016-12-07 04:52:58 -08:00
]
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
deps = [
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
":rtp_interfaces",
Move video send/receive stream headers to webrtc/call. Moved the headers video_receive_stream.h and video_send_stream.h from webrtc/ into webrtc/call/ as part of the Slim and Modular work. The GN target webrtc:video_stream_api has moved to webrtc/call:video_stream_api. There are headers left in webrtc/ with the same name including the moved headers in webrtc/call/ for not breaking external projects depending on WebRTC. At the same time, some minor cleanup is done: Non-pure-virtual functions declared in the two affected headers now have definitions in the same target. After making this change, our 'chromium-style' plugin detected some style violations that have now been fixed: non-inlined constructors and destructors have been added to a number of classes, both inside the GN target of the two affected headers, and in other targets. BUG=webrtc:8107 Review-Url: https://codereview.webrtc.org/3000253002 Cr-Commit-Position: refs/heads/master@{#19448}
2017-08-22 05:43:23 -07:00
":video_stream_api",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"..:webrtc_common",
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
"../:typedefs",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../api:audio_mixer_api",
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",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../api:optional",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../api:transport_api",
"../api/audio_codecs:audio_codecs_api",
Use the new AudioProcessing statistics everywhere. The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM. Bug: webrtc:8563, b/67926135 Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24 Reviewed-on: https://webrtc-review.googlesource.com/25621 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20877}
2017-11-24 17:29:59 +01:00
"../modules/audio_processing:audio_processing_statistics",
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",
"../rtc_base:rtc_base_approved",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
]
Moved call.h and most of api/call/* into call/ BUG=webrtc:6716 Review-Url: https://codereview.webrtc.org/2550273003 Cr-Commit-Position: refs/heads/master@{#15460}
2016-12-07 04:52:58 -08:00
}
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
# TODO(nisse): These RTP targets should be moved elsewhere
Only one implementation of MockRtpPacketSink once MockRtpPacketSink has three identical implementations now, so time to move it to its own file. BUG=None Review-Url: https://codereview.webrtc.org/2988853002 Cr-Commit-Position: refs/heads/master@{#19183}
2017-07-28 10:05:45 -07:00
# when interfaces have stabilized. See also TODO for |mock_rtp_interfaces|.
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
rtc_source_set("rtp_interfaces") {
sources = [
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) Reason for revert: About to fix problem and reland. Original issue's description: > Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) > > Reason for revert: > Breaks Chromium FYI bots. > > The problem is in the BUILD.gn file. > > Sample failure: > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829 > > Sample logs: > use_goma = true > """ to /b/c/b/Linux_Builder/src/out/Release/args.gn. > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file. > "//webrtc/base:rtc_base_approved", > ^-------------------------------- > > Original issue's description: > > Create RtcpDemuxer. Capabilities: > > 1. Demux RTCP messages according to the sender-SSRC. > > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP). > > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks"). > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2943693003 > > Cr-Commit-Position: refs/heads/master@{#18763} > > Committed: https://chromium.googlesource.com/external/webrtc/+/cb83bdf01f2ec8b9ed254991edc2be053c9eed24 > > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2957763002 > Cr-Commit-Position: refs/heads/master@{#18764} > Committed: https://chromium.googlesource.com/external/webrtc/+/0e7e7869e74a29caf8197d02fb396d70748474ed BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2960623002 Cr-Commit-Position: refs/heads/master@{#18768}
2017-06-26 11:23:54 -07:00
"rtcp_packet_sink_interface.h",
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
"rtp_config.cc",
"rtp_config.h",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
"rtp_packet_sink_interface.h",
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. And implementation class RtpStreamReceiverController. It's responsible for demuxing, and acts as factory for RtpStreamReceiverInterface. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2886993005 Cr-Commit-Position: refs/heads/master@{#18696}
2017-06-21 01:05:22 -07:00
"rtp_stream_receiver_controller_interface.h",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
"rtp_transport_controller_send_interface.h",
]
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) Reason for revert: About to fix problem and reland. Original issue's description: > Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) > > Reason for revert: > Breaks Chromium FYI bots. > > The problem is in the BUILD.gn file. > > Sample failure: > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829 > > Sample logs: > use_goma = true > """ to /b/c/b/Linux_Builder/src/out/Release/args.gn. > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file. > "//webrtc/base:rtc_base_approved", > ^-------------------------------- > > Original issue's description: > > Create RtcpDemuxer. Capabilities: > > 1. Demux RTCP messages according to the sender-SSRC. > > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP). > > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks"). > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2943693003 > > Cr-Commit-Position: refs/heads/master@{#18763} > > Committed: https://chromium.googlesource.com/external/webrtc/+/cb83bdf01f2ec8b9ed254991edc2be053c9eed24 > > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2957763002 > Cr-Commit-Position: refs/heads/master@{#18764} > Committed: https://chromium.googlesource.com/external/webrtc/+/0e7e7869e74a29caf8197d02fb396d70748474ed BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2960623002 Cr-Commit-Position: refs/heads/master@{#18768}
2017-06-26 11:23:54 -07:00
deps = [
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",
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 Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) Reason for revert: About to fix problem and reland. Original issue's description: > Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) > > Reason for revert: > Breaks Chromium FYI bots. > > The problem is in the BUILD.gn file. > > Sample failure: > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829 > > Sample logs: > use_goma = true > """ to /b/c/b/Linux_Builder/src/out/Release/args.gn. > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file. > "//webrtc/base:rtc_base_approved", > ^-------------------------------- > > Original issue's description: > > Create RtcpDemuxer. Capabilities: > > 1. Demux RTCP messages according to the sender-SSRC. > > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP). > > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks"). > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2943693003 > > Cr-Commit-Position: refs/heads/master@{#18763} > > Committed: https://chromium.googlesource.com/external/webrtc/+/cb83bdf01f2ec8b9ed254991edc2be053c9eed24 > > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2957763002 > Cr-Commit-Position: refs/heads/master@{#18764} > Committed: https://chromium.googlesource.com/external/webrtc/+/0e7e7869e74a29caf8197d02fb396d70748474ed BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2960623002 Cr-Commit-Position: refs/heads/master@{#18768}
2017-06-26 11:23:54 -07:00
]
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
}
rtc_source_set("rtp_receiver") {
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 = [ "*" ]
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
sources = [
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) Reason for revert: About to fix problem and reland. Original issue's description: > Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) > > Reason for revert: > Breaks Chromium FYI bots. > > The problem is in the BUILD.gn file. > > Sample failure: > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829 > > Sample logs: > use_goma = true > """ to /b/c/b/Linux_Builder/src/out/Release/args.gn. > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file. > "//webrtc/base:rtc_base_approved", > ^-------------------------------- > > Original issue's description: > > Create RtcpDemuxer. Capabilities: > > 1. Demux RTCP messages according to the sender-SSRC. > > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP). > > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks"). > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2943693003 > > Cr-Commit-Position: refs/heads/master@{#18763} > > Committed: https://chromium.googlesource.com/external/webrtc/+/cb83bdf01f2ec8b9ed254991edc2be053c9eed24 > > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2957763002 > Cr-Commit-Position: refs/heads/master@{#18764} > Committed: https://chromium.googlesource.com/external/webrtc/+/0e7e7869e74a29caf8197d02fb396d70748474ed BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2960623002 Cr-Commit-Position: refs/heads/master@{#18768}
2017-06-26 11:23:54 -07:00
"rtcp_demuxer.cc",
"rtcp_demuxer.h",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
"rtp_demuxer.cc",
"rtp_demuxer.h",
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) Reason for revert: About to fix problem and reland. Original issue's description: > Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) > > Reason for revert: > Breaks Chromium FYI bots. > > The problem is in the BUILD.gn file. > > Sample failure: > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829 > > Sample logs: > use_goma = true > """ to /b/c/b/Linux_Builder/src/out/Release/args.gn. > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file. > "//webrtc/base:rtc_base_approved", > ^-------------------------------- > > Original issue's description: > > Create RtcpDemuxer. Capabilities: > > 1. Demux RTCP messages according to the sender-SSRC. > > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP). > > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks"). > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2943693003 > > Cr-Commit-Position: refs/heads/master@{#18763} > > Committed: https://chromium.googlesource.com/external/webrtc/+/cb83bdf01f2ec8b9ed254991edc2be053c9eed24 > > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2957763002 > Cr-Commit-Position: refs/heads/master@{#18764} > Committed: https://chromium.googlesource.com/external/webrtc/+/0e7e7869e74a29caf8197d02fb396d70748474ed BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2960623002 Cr-Commit-Position: refs/heads/master@{#18768}
2017-06-26 11:23:54 -07:00
"rtp_rtcp_demuxer_helper.cc",
"rtp_rtcp_demuxer_helper.h",
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. And implementation class RtpStreamReceiverController. It's responsible for demuxing, and acts as factory for RtpStreamReceiverInterface. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2886993005 Cr-Commit-Position: refs/heads/master@{#18696}
2017-06-21 01:05:22 -07:00
"rtp_stream_receiver_controller.cc",
"rtp_stream_receiver_controller.h",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
"rtx_receive_stream.cc",
"rtx_receive_stream.h",
Rename RsidResolutionObserver to SsrcBindingObserver. This rename prepares SsrcBindingObserver to be used for observing all the ways a sink can be bound to SSRCs (e.g., MID, payload types). Bug: webrtc:4050 Change-Id: I16b68481d01f921c326a33f5f2baf79d8b3f12e2 Reviewed-on: https://chromium-review.googlesource.com/590762 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19396}
2017-08-17 15:23:51 -07:00
"ssrc_binding_observer.h",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
]
deps = [
":rtp_interfaces",
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) Reason for revert: About to fix problem and reland. Original issue's description: > Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) > > Reason for revert: > Breaks Chromium FYI bots. > > The problem is in the BUILD.gn file. > > Sample failure: > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829 > > Sample logs: > use_goma = true > """ to /b/c/b/Linux_Builder/src/out/Release/args.gn. > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file. > "//webrtc/base:rtc_base_approved", > ^-------------------------------- > > Original issue's description: > > Create RtcpDemuxer. Capabilities: > > 1. Demux RTCP messages according to the sender-SSRC. > > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP). > > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks"). > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2943693003 > > Cr-Commit-Position: refs/heads/master@{#18763} > > Committed: https://chromium.googlesource.com/external/webrtc/+/cb83bdf01f2ec8b9ed254991edc2be053c9eed24 > > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2957763002 > Cr-Commit-Position: refs/heads/master@{#18764} > Committed: https://chromium.googlesource.com/external/webrtc/+/0e7e7869e74a29caf8197d02fb396d70748474ed BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2960623002 Cr-Commit-Position: refs/heads/master@{#18768}
2017-06-26 11:23:54 -07:00
"..: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",
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
"../api:libjingle_peerconnection_api",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../api:optional",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
"../modules/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
"../modules/rtp_rtcp:rtp_rtcp_format",
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",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
]
}
rtc_source_set("rtp_sender") {
sources = [
"rtp_transport_controller_send.cc",
"rtp_transport_controller_send.h",
]
deps = [
":rtp_interfaces",
Wire up RTP keep-alive in ortc api. [This CL is work in progress.] Wire up the rtp keep-alive in webrtc::Call::Config using new SetRtpTransportParameters() method on RtpTransportInterface. BUG=webrtc:7907 Review-Url: https://codereview.webrtc.org/2981513002 Cr-Commit-Position: refs/heads/master@{#19287}
2017-08-09 06:42:32 -07:00
"..:webrtc_common",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
"../modules/congestion_controller",
Delete deprecated constructor of SendSideCongestionController. Move packet_router #include to where it's needed, and delete unused MockPacketRouter. Bug: webrtc:6847 Change-Id: I03c86c6fb8b413f5a535a237fa1724cc10960ffa Reviewed-on: https://webrtc-review.googlesource.com/17320 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20564}
2017-10-31 10:19:10 +01:00
"../modules/pacing",
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",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
]
}
Fix Gn untracked headers in webrtc/call. This CL is the same CL we had at https://codereview.webrtc.org/3014543002/. Since we cannot land it with Rietveld anymore let's move the discussion to Gerrit. BUG=webrtc:7641 CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Change-Id: I5662bec318544b07f476c12ecada997d726e7361 Reviewed-on: https://webrtc-review.googlesource.com/7981 Reviewed-by: Henrik Kjellander <kjellander@google.com> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20224}
2017-10-10 14:38:11 +02:00
rtc_source_set("bitrate_allocator") {
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
sources = [
Move BitrateAllocator from BitrateController logic to Call. This is a step on the way to have variable bitrate for audio and is intended to be as much of a no-op as possible. BUG=webrtc:5079 Review URL: https://codereview.webrtc.org/1441673002 Cr-Commit-Position: refs/heads/master@{#10630}
2015-11-12 21:02:42 -08:00
"bitrate_allocator.cc",
Fix Gn untracked headers in webrtc/call. This CL is the same CL we had at https://codereview.webrtc.org/3014543002/. Since we cannot land it with Rietveld anymore let's move the discussion to Gerrit. BUG=webrtc:7641 CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Change-Id: I5662bec318544b07f476c12ecada997d726e7361 Reviewed-on: https://webrtc-review.googlesource.com/7981 Reviewed-by: Henrik Kjellander <kjellander@google.com> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20224}
2017-10-10 14:38:11 +02:00
"bitrate_allocator.h",
]
deps = [
"../modules/bitrate_controller",
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 Gn untracked headers in webrtc/call. This CL is the same CL we had at https://codereview.webrtc.org/3014543002/. Since we cannot land it with Rietveld anymore let's move the discussion to Gerrit. BUG=webrtc:7641 CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Change-Id: I5662bec318544b07f476c12ecada997d726e7361 Reviewed-on: https://webrtc-review.googlesource.com/7981 Reviewed-by: Henrik Kjellander <kjellander@google.com> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20224}
2017-10-10 14:38:11 +02:00
"../rtc_base:rtc_base_approved",
"../rtc_base:sequenced_task_checker",
"../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:metrics_api",
Fix Gn untracked headers in webrtc/call. This CL is the same CL we had at https://codereview.webrtc.org/3014543002/. Since we cannot land it with Rietveld anymore let's move the discussion to Gerrit. BUG=webrtc:7641 CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Change-Id: I5662bec318544b07f476c12ecada997d726e7361 Reviewed-on: https://webrtc-review.googlesource.com/7981 Reviewed-by: Henrik Kjellander <kjellander@google.com> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20224}
2017-10-10 14:38:11 +02:00
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
rtc_static_library("call") {
sources = [
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
"call.cc",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"callfactory.cc",
"callfactory.h",
Move FlexfecReceiveStream from api/call/ to call/. Also rename internal::FlexfecReceiveStream to FlexfecReceiveStreamImpl. BUG=webrtc:6849 Review-Url: https://codereview.webrtc.org/2561123002 Cr-Commit-Position: refs/heads/master@{#15666}
2016-12-19 01:13:46 -08:00
"flexfec_receive_stream_impl.cc",
"flexfec_receive_stream_impl.h",
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
}
deps = [
Fix Gn untracked headers in webrtc/call. This CL is the same CL we had at https://codereview.webrtc.org/3014543002/. Since we cannot land it with Rietveld anymore let's move the discussion to Gerrit. BUG=webrtc:7641 CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Change-Id: I5662bec318544b07f476c12ecada997d726e7361 Reviewed-on: https://webrtc-review.googlesource.com/7981 Reviewed-by: Henrik Kjellander <kjellander@google.com> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20224}
2017-10-10 14:38:11 +02:00
":bitrate_allocator",
Moved call.h and most of api/call/* into call/ BUG=webrtc:6716 Review-Url: https://codereview.webrtc.org/2550273003 Cr-Commit-Position: refs/heads/master@{#15460}
2016-12-07 04:52:58 -08:00
":call_interfaces",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
":rtp_interfaces",
":rtp_receiver",
":rtp_sender",
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
":video_stream_api",
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
"..:webrtc_common",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../api:optional",
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",
Add missing dependencies on audio, video and call to the new GN files. This caused linker failures when depending on the new `api` target. TBR=henrika@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2029323006 Cr-Commit-Position: refs/heads/master@{#13042}
2016-06-03 13:14:28 -07:00
"../audio",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../logging:rtc_event_log_api",
Moved RtcEventLog files from call/ to logging/ The RtcEventLog headers need to be accessible from any place which needs logging, and the implementation needs access to data structures that are logged. After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future). The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/). This change allows using RtcEventLog in the p2p/ directory, so that we can log STUN pings and ICE state transitions. BUG=webrtc:6393 R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org Review URL: https://codereview.webrtc.org/2380683005 . Cr-Commit-Position: refs/heads/master@{#14485}
2016-10-03 18:31:22 -07:00
"../logging:rtc_event_log_impl",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../modules/bitrate_controller",
Move congestion controller to a separate module. This allows other projects to more easily depend on this. The plan is to move remote_bitrate_estimator and bitrate_controller into this module and reduce the exposed interface to only a simplified version of congestion_controller.h. No functional changes in this CL. R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://codereview.webrtc.org/1718473002 . Cr-Commit-Position: refs/heads/master@{#11718}
2016-02-23 13:30:42 +01:00
"../modules/congestion_controller",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../modules/pacing",
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
"../modules/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
"../modules/rtp_rtcp:rtp_rtcp_format",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../modules/utility",
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",
Now that https://codereview.webrtc.org/3003643002 is landed we can remove rtc_base_approved from the public_deps list of rtc_task_queue. BUG=webrtc:8160 NOTRY=True Review-Url: https://codereview.webrtc.org/3008553002 Cr-Commit-Position: refs/heads/master@{#19551}
2017-08-27 23:40:10 -07:00
"../rtc_base:rtc_base_approved",
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_task_queue",
Change ThreadChecker to SequencedTaskChecker in internal::Call In preparation of running DirectTransport on a TaskQueue in unit-tests, change the thread-checkers to sequence-checkers. This is necessary for Mac and iOS, where the TaskQueue is guaranteed to run sequentially, but not guaranteed to do so on only one thread. TODO: Add the relevant BUGs. BUG=None Review-Url: https://codereview.webrtc.org/2999973002 Cr-Commit-Position: refs/heads/master@{#19404}
2017-08-18 02:47:08 -07:00
"../rtc_base:sequenced_task_checker",
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
"../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:metrics_api",
Add missing dependencies on audio, video and call to the new GN files. This caused linker failures when depending on the new `api` target. TBR=henrika@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2029323006 Cr-Commit-Position: refs/heads/master@{#13042}
2016-06-03 13:14:28 -07:00
"../video",
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
]
}
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
Move video send/receive stream headers to webrtc/call. Moved the headers video_receive_stream.h and video_send_stream.h from webrtc/ into webrtc/call/ as part of the Slim and Modular work. The GN target webrtc:video_stream_api has moved to webrtc/call:video_stream_api. There are headers left in webrtc/ with the same name including the moved headers in webrtc/call/ for not breaking external projects depending on WebRTC. At the same time, some minor cleanup is done: Non-pure-virtual functions declared in the two affected headers now have definitions in the same target. After making this change, our 'chromium-style' plugin detected some style violations that have now been fixed: non-inlined constructors and destructors have been added to a number of classes, both inside the GN target of the two affected headers, and in other targets. BUG=webrtc:8107 Review-Url: https://codereview.webrtc.org/3000253002 Cr-Commit-Position: refs/heads/master@{#19448}
2017-08-22 05:43:23 -07:00
rtc_source_set("video_stream_api") {
sources = [
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
"video_config.cc",
"video_config.h",
Move video send/receive stream headers to webrtc/call. Moved the headers video_receive_stream.h and video_send_stream.h from webrtc/ into webrtc/call/ as part of the Slim and Modular work. The GN target webrtc:video_stream_api has moved to webrtc/call:video_stream_api. There are headers left in webrtc/ with the same name including the moved headers in webrtc/call/ for not breaking external projects depending on WebRTC. At the same time, some minor cleanup is done: Non-pure-virtual functions declared in the two affected headers now have definitions in the same target. After making this change, our 'chromium-style' plugin detected some style violations that have now been fixed: non-inlined constructors and destructors have been added to a number of classes, both inside the GN target of the two affected headers, and in other targets. BUG=webrtc:8107 Review-Url: https://codereview.webrtc.org/3000253002 Cr-Commit-Position: refs/heads/master@{#19448}
2017-08-22 05:43:23 -07:00
"video_receive_stream.cc",
"video_receive_stream.h",
"video_send_stream.cc",
"video_send_stream.h",
]
deps = [
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
":rtp_interfaces",
Move videosourceinterface to api. VideoSourceInterface is clearly an integral part of mediastreaminterface.h already, so moving that interface to api makes sense. This also resolves a circular dependency in call/. Bug: webrc:6828 Change-Id: Ic1862f118363b0b55a235a9c0c35d9adc647184c Reviewed-on: https://webrtc-review.googlesource.com/37500 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21498}
2018-01-04 15:58:20 +01:00
"../:typedefs",
Move video send/receive stream headers to webrtc/call. Moved the headers video_receive_stream.h and video_send_stream.h from webrtc/ into webrtc/call/ as part of the Slim and Modular work. The GN target webrtc:video_stream_api has moved to webrtc/call:video_stream_api. There are headers left in webrtc/ with the same name including the moved headers in webrtc/call/ for not breaking external projects depending on WebRTC. At the same time, some minor cleanup is done: Non-pure-virtual functions declared in the two affected headers now have definitions in the same target. After making this change, our 'chromium-style' plugin detected some style violations that have now been fixed: non-inlined constructors and destructors have been added to a number of classes, both inside the GN target of the two affected headers, and in other targets. BUG=webrtc:8107 Review-Url: https://codereview.webrtc.org/3000253002 Cr-Commit-Position: refs/heads/master@{#19448}
2017-08-22 05:43:23 -07:00
"../:webrtc_common",
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",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../api:optional",
Move video send/receive stream headers to webrtc/call. Moved the headers video_receive_stream.h and video_send_stream.h from webrtc/ into webrtc/call/ as part of the Slim and Modular work. The GN target webrtc:video_stream_api has moved to webrtc/call:video_stream_api. There are headers left in webrtc/ with the same name including the moved headers in webrtc/call/ for not breaking external projects depending on WebRTC. At the same time, some minor cleanup is done: Non-pure-virtual functions declared in the two affected headers now have definitions in the same target. After making this change, our 'chromium-style' plugin detected some style violations that have now been fixed: non-inlined constructors and destructors have been added to a number of classes, both inside the GN target of the two affected headers, and in other targets. BUG=webrtc:8107 Review-Url: https://codereview.webrtc.org/3000253002 Cr-Commit-Position: refs/heads/master@{#19448}
2017-08-22 05:43:23 -07:00
"../api:transport_api",
Move videosourceinterface to api. VideoSourceInterface is clearly an integral part of mediastreaminterface.h already, so moving that interface to api makes sense. This also resolves a circular dependency in call/. Bug: webrc:6828 Change-Id: Ic1862f118363b0b55a235a9c0c35d9adc647184c Reviewed-on: https://webrtc-review.googlesource.com/37500 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21498}
2018-01-04 15:58:20 +01:00
"../api:video_frame_api",
Move video send/receive stream headers to webrtc/call. Moved the headers video_receive_stream.h and video_send_stream.h from webrtc/ into webrtc/call/ as part of the Slim and Modular work. The GN target webrtc:video_stream_api has moved to webrtc/call:video_stream_api. There are headers left in webrtc/ with the same name including the moved headers in webrtc/call/ for not breaking external projects depending on WebRTC. At the same time, some minor cleanup is done: Non-pure-virtual functions declared in the two affected headers now have definitions in the same target. After making this change, our 'chromium-style' plugin detected some style violations that have now been fixed: non-inlined constructors and destructors have been added to a number of classes, both inside the GN target of the two affected headers, and in other targets. BUG=webrtc:8107 Review-Url: https://codereview.webrtc.org/3000253002 Cr-Commit-Position: refs/heads/master@{#19448}
2017-08-22 05:43:23 -07:00
"../common_video:common_video",
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
"../modules/rtp_rtcp:rtp_rtcp_format",
Move videosourceinterface to api. VideoSourceInterface is clearly an integral part of mediastreaminterface.h already, so moving that interface to api makes sense. This also resolves a circular dependency in call/. Bug: webrc:6828 Change-Id: Ic1862f118363b0b55a235a9c0c35d9adc647184c Reviewed-on: https://webrtc-review.googlesource.com/37500 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21498}
2018-01-04 15:58:20 +01:00
"../rtc_base:checks",
Move video send/receive stream headers to webrtc/call. Moved the headers video_receive_stream.h and video_send_stream.h from webrtc/ into webrtc/call/ as part of the Slim and Modular work. The GN target webrtc:video_stream_api has moved to webrtc/call:video_stream_api. There are headers left in webrtc/ with the same name including the moved headers in webrtc/call/ for not breaking external projects depending on WebRTC. At the same time, some minor cleanup is done: Non-pure-virtual functions declared in the two affected headers now have definitions in the same target. After making this change, our 'chromium-style' plugin detected some style violations that have now been fixed: non-inlined constructors and destructors have been added to a number of classes, both inside the GN target of the two affected headers, and in other targets. BUG=webrtc:8107 Review-Url: https://codereview.webrtc.org/3000253002 Cr-Commit-Position: refs/heads/master@{#19448}
2017-08-22 05:43:23 -07:00
"../rtc_base:rtc_base_approved",
]
}
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
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_source_set("call_tests") {
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
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
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
sources = [
"bitrate_allocator_unittest.cc",
"bitrate_estimator_tests.cc",
"call_unittest.cc",
Add FlexfecReceiveStream. This class is logically parallel with the {Audio,Video}ReceiveStream classes. Its purpose is to describe a receive stream of FlexFEC packets, through the corresponding config. Functionally, this class simply forwards the received RTP packets to its FlexfecReceiver, which returns recovered packets to the Call level, for appropriate demultiplexing based on SSRC. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2397843005 Cr-Commit-Position: refs/heads/master@{#14704}
2016-10-20 04:54:48 -07:00
"flexfec_receive_stream_unittest.cc",
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) Reason for revert: About to fix problem and reland. Original issue's description: > Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) > > Reason for revert: > Breaks Chromium FYI bots. > > The problem is in the BUILD.gn file. > > Sample failure: > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829 > > Sample logs: > use_goma = true > """ to /b/c/b/Linux_Builder/src/out/Release/args.gn. > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file. > "//webrtc/base:rtc_base_approved", > ^-------------------------------- > > Original issue's description: > > Create RtcpDemuxer. Capabilities: > > 1. Demux RTCP messages according to the sender-SSRC. > > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP). > > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks"). > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2943693003 > > Cr-Commit-Position: refs/heads/master@{#18763} > > Committed: https://chromium.googlesource.com/external/webrtc/+/cb83bdf01f2ec8b9ed254991edc2be053c9eed24 > > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2957763002 > Cr-Commit-Position: refs/heads/master@{#18764} > Committed: https://chromium.googlesource.com/external/webrtc/+/0e7e7869e74a29caf8197d02fb396d70748474ed BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2960623002 Cr-Commit-Position: refs/heads/master@{#18768}
2017-06-26 11:23:54 -07:00
"rtcp_demuxer_unittest.cc",
Create unit tests for RtpDemuxer 1. Create unit tests for RtpDemuxer. 2. Add an RTC_DCHECK in RtpDemuxer that makes sure that the sink<->ssrc multimap does not allow multiple instances of the same association. BUG=None Review-Url: https://codereview.webrtc.org/2902823004 Cr-Commit-Position: refs/heads/master@{#18357}
2017-05-31 09:12:25 -07:00
"rtp_demuxer_unittest.cc",
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) Reason for revert: About to fix problem and reland. Original issue's description: > Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) > > Reason for revert: > Breaks Chromium FYI bots. > > The problem is in the BUILD.gn file. > > Sample failure: > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829 > > Sample logs: > use_goma = true > """ to /b/c/b/Linux_Builder/src/out/Release/args.gn. > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file. > "//webrtc/base:rtc_base_approved", > ^-------------------------------- > > Original issue's description: > > Create RtcpDemuxer. Capabilities: > > 1. Demux RTCP messages according to the sender-SSRC. > > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP). > > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks"). > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2943693003 > > Cr-Commit-Position: refs/heads/master@{#18763} > > Committed: https://chromium.googlesource.com/external/webrtc/+/cb83bdf01f2ec8b9ed254991edc2be053c9eed24 > > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2957763002 > Cr-Commit-Position: refs/heads/master@{#18764} > Committed: https://chromium.googlesource.com/external/webrtc/+/0e7e7869e74a29caf8197d02fb396d70748474ed BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2960623002 Cr-Commit-Position: refs/heads/master@{#18768}
2017-06-26 11:23:54 -07:00
"rtp_rtcp_demuxer_helper_unittest.cc",
New class RtxReceiveStream. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2888093002 Cr-Commit-Position: refs/heads/master@{#18212}
2017-05-19 06:15:19 -07:00
"rtx_receive_stream_unittest.cc",
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
]
deps = [
Fix Gn untracked headers in webrtc/call. This CL is the same CL we had at https://codereview.webrtc.org/3014543002/. Since we cannot land it with Rietveld anymore let's move the discussion to Gerrit. BUG=webrtc:7641 CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Change-Id: I5662bec318544b07f476c12ecada997d726e7361 Reviewed-on: https://webrtc-review.googlesource.com/7981 Reviewed-by: Henrik Kjellander <kjellander@google.com> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20224}
2017-10-10 14:38:11 +02:00
":bitrate_allocator",
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
":call",
Stop using public_deps in the call module. Bug: webrtc:8603 Change-Id: I048127bc86f213e638e6814ac8a86761cb8a64db Reviewed-on: https://webrtc-review.googlesource.com/28624 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21072}
2017-12-04 10:50:51 +01:00
":call_interfaces",
Only one implementation of MockRtpPacketSink once MockRtpPacketSink has three identical implementations now, so time to move it to its own file. BUG=None Review-Url: https://codereview.webrtc.org/2988853002 Cr-Commit-Position: refs/heads/master@{#19183}
2017-07-28 10:05:45 -07:00
":mock_rtp_interfaces",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
":rtp_interfaces",
":rtp_receiver",
":rtp_sender",
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) Reason for revert: About to fix problem and reland. Original issue's description: > Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) > > Reason for revert: > Breaks Chromium FYI bots. > > The problem is in the BUILD.gn file. > > Sample failure: > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829 > > Sample logs: > use_goma = true > """ to /b/c/b/Linux_Builder/src/out/Release/args.gn. > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file. > "//webrtc/base:rtc_base_approved", > ^-------------------------------- > > Original issue's description: > > Create RtcpDemuxer. Capabilities: > > 1. Demux RTCP messages according to the sender-SSRC. > > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP). > > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks"). > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2943693003 > > Cr-Commit-Position: refs/heads/master@{#18763} > > Committed: https://chromium.googlesource.com/external/webrtc/+/cb83bdf01f2ec8b9ed254991edc2be053c9eed24 > > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2957763002 > Cr-Commit-Position: refs/heads/master@{#18764} > Committed: https://chromium.googlesource.com/external/webrtc/+/0e7e7869e74a29caf8197d02fb396d70748474ed BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2960623002 Cr-Commit-Position: refs/heads/master@{#18768}
2017-06-26 11:23:54 -07:00
"..: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",
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
"../api:libjingle_peerconnection_api",
Store/restore RTP state for audio streams with same SSRC within a call This functionality already exists for video streams, so not having it for audio is unexpected and has lead to problems. BUG=webrtc:7631 Review-Url: https://codereview.webrtc.org/2887733002 Cr-Commit-Position: refs/heads/master@{#18231}
2017-05-23 06:07:11 -07:00
"../api:mock_audio_mixer",
Voice Engine: Require caller to supply an AudioDecoderFactory So that we don't have to be capable of creating one ourselves, which requires a dependency on the audio decoders. In the process, remove the default values for the VoEBase::Init() arguments, since there isn't a sensible default value for the audio decoder factory anymore. BUG=webrtc:8396 Change-Id: Idb433efa49e1a68e8206d369d27b3c255185777a Reviewed-on: https://webrtc-review.googlesource.com/18200 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20552}
2017-11-02 13:04:41 +01:00
"../api/audio_codecs:builtin_audio_decoder_factory",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../logging:rtc_event_log_api",
Added an empty AudioTransportProxy to AudioState. All audio in calls is now routed through AudioTransportProxy. The AudioTransport implemented by VoEBaseImpl is disconnected from AudioDevice and replaced by an empty proxy layer that forwards calls to the old Transport. This is a refactoring CL in preparation for landing https://codereview.webrtc.org/2436033002/, which will connect the new AudioMixer. In the planned configuration, the currently empty AudioTransportProxy will query the new mixer for audio instead of polling data from the old Transport. Mixed audio will be passed to an AudioProcessing interface. AudioTransportProxy is initialized with an AudioProcessing*, which is currently unused. No presubmit since we implement an interface with non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2454373002 Cr-Commit-Position: refs/heads/master@{#15133}
2016-11-17 06:28:59 -08:00
"../modules/audio_device:mock_audio_device",
Passed AudioMixer to AudioState::Config. This is a refactoring change in preparation for enabling AudioMixer with the goal to have a small CL as possible for passing audio through the new audio mixer in WebRTC. The dependent CL https://codereview.webrtc.org/2436033002/ enables the mixer. An object of class AudioState is shared across different webrtc audio connections. It is created in tests and in WebRTCVoiceEngine. AudioState is constructed by passing a Config struct, where one argument is scoped_refptr<AudioMixer>. Populating this field has now been mandatory. Tests and WebRTCVoiceEngine create and pass either a AudioMixerImpl. WebRTCVoiceEngine passes a real AudioMixer, which is currently unused. An alternative would have tests pass a mocked audio mixer. We chose not to do that, because we believe that tests should use the real thing unless there are reasons against it. Construction time is not an issue, because the real mixer is relatively lightweight. We couldn't find a way to test any mixer-related changes in AudioState before the mixes is connected. The next dependent CL https://codereview.webrtc.org/2436033002/ contains unit tests for mixer usage. BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2469743002 Cr-Commit-Position: refs/heads/master@{#15134}
2016-11-17 06:48:48 -08:00
"../modules/audio_mixer",
Stop using public_deps in modules/audio_mixer. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I74c01d5a0243c96dca504b2d696092ea35c36aa3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29860 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21105}
2017-12-05 15:33:16 +01:00
"../modules/audio_mixer:audio_mixer_impl",
Add missing files to audio_processing. Bug: webrtc:7621 Change-Id: I2cab764232fc4e084ed8a489f4cf3a3ac562c894 Reviewed-on: https://webrtc-review.googlesource.com/34658 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21461}
2017-12-19 16:44:45 +01:00
"../modules/audio_processing:mocks",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../modules/bitrate_controller",
Fix Gn untracked headers in webrtc/call. This CL is the same CL we had at https://codereview.webrtc.org/3014543002/. Since we cannot land it with Rietveld anymore let's move the discussion to Gerrit. BUG=webrtc:7641 CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Change-Id: I5662bec318544b07f476c12ecada997d726e7361 Reviewed-on: https://webrtc-review.googlesource.com/7981 Reviewed-by: Henrik Kjellander <kjellander@google.com> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20224}
2017-10-10 14:38:11 +02:00
"../modules/congestion_controller",
This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). FakeRtpTransportController moves to a common header and its constructor is changed to take a SendSideCongestionController to enable injecting the mock. BUG=webrtc:7395 Review-Url: https://codereview.webrtc.org/2834663003 Cr-Commit-Position: refs/heads/master@{#18055}
2017-05-08 11:52:38 -07:00
"../modules/congestion_controller:mock_congestion_controller",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../modules/pacing",
Tracking mock_paced_sender.h with a GN target Untracked headers fly under the 'gn check' radar and in the long term this can cause problems like unnoticed cyclic dependencies. This cl creates a synthetic target for this header since no other targets in webrtc/modules/pacing/BUILD.gn seem to be related to it. BUG=webrtc:7649 NOTRY=True Review-Url: https://codereview.webrtc.org/2887593003 Cr-Commit-Position: refs/heads/master@{#19656}
2017-09-04 04:57:11 -07:00
"../modules/pacing:mock_paced_sender",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../modules/rtp_rtcp",
Add untracked headers in modules/rtp_rtcp BUG=webrtc:7623 Review-Url: https://codereview.webrtc.org/2876533003 Cr-Commit-Position: refs/heads/master@{#18085}
2017-05-10 08:41:13 -07:00
"../modules/rtp_rtcp:mock_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
"../modules/rtp_rtcp:rtp_rtcp_format",
Tracking mock_process_thread with a GN target include/mock/mock_process_thread.h was not tracked by GN. This cl creates a target for it. The target is testonly because it depends on "webrtc/test:rtp_test_utils". This means that dependencies to this header cannot fly under the GN radar anymore. :) BUG=webrtc:7652 NOTRY=True Review-Url: https://codereview.webrtc.org/2881343003 Cr-Commit-Position: refs/heads/master@{#19234}
2017-08-03 05:57:11 -07:00
"../modules/utility:mock_process_thread",
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",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../system_wrappers",
Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ AudioDecoder and AudioDecoderFactory are in webrtc/api/ now, so move their mocks to someplace central where tests from all over WebRTC are allowed to #include them. BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2798063004 Cr-Commit-Position: refs/heads/master@{#17619}
2017-04-10 05:15:48 -07:00
"../test:audio_codec_mocks",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../test:direct_transport",
Passed AudioMixer to AudioState::Config. This is a refactoring change in preparation for enabling AudioMixer with the goal to have a small CL as possible for passing audio through the new audio mixer in WebRTC. The dependent CL https://codereview.webrtc.org/2436033002/ enables the mixer. An object of class AudioState is shared across different webrtc audio connections. It is created in tests and in WebRTCVoiceEngine. AudioState is constructed by passing a Config struct, where one argument is scoped_refptr<AudioMixer>. Populating this field has now been mandatory. Tests and WebRTCVoiceEngine create and pass either a AudioMixerImpl. WebRTCVoiceEngine passes a real AudioMixer, which is currently unused. An alternative would have tests pass a mocked audio mixer. We chose not to do that, because we believe that tests should use the real thing unless there are reasons against it. Construction time is not an issue, because the real mixer is relatively lightweight. We couldn't find a way to test any mixer-related changes in AudioState before the mixes is connected. The next dependent CL https://codereview.webrtc.org/2436033002/ contains unit tests for mixer usage. BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2469743002 Cr-Commit-Position: refs/heads/master@{#15134}
2016-11-17 06:48:48 -08:00
"../test:test_common",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../test:test_support",
"../test:video_test_common",
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
"//testing/gmock",
"//testing/gtest",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
}
}
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@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/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
rtc_source_set("call_perf_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 actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@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/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
sources = [
"call_perf_tests.cc",
"rampup_tests.cc",
"rampup_tests.h",
]
deps = [
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
":call_interfaces",
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
":video_stream_api",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"..:webrtc_common",
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
"../api/audio_codecs:builtin_audio_encoder_factory",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../logging:rtc_event_log_api",
"../modules/audio_coding",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/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",
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",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../system_wrappers",
"../system_wrappers:metrics_default",
"../test:direct_transport",
Reland Move fake_audio_device to its own target. Patchset 1 is patchset #5 id:80001 of https://codereview.webrtc.org/2717983003/ Patchset 2 fix call_perf_test dep on fake_audio_device. This reverts commit 985371bda999c6db51286586c5850d2ff58f3511. Original cl description: Move fake_audio_device to its own target. The purpose is to make it usefull for test targets that does not need or can use test_common. For some reason this also triggered override issues in rtp module tests that are fixed in the same cl. BUG=none TBR=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2718363002 Cr-Commit-Position: refs/heads/master@{#16922}
2017-02-28 14:41:05 -08:00
"../test:fake_audio_device",
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:field_trial",
Reland "iOS: Save perf results under Documents/perf_result.json" This will require a manual roll to downstream projects, since the //test:perf_test target was introduced. This is a reland of 10a8e7a9b5261a7e3ce19900ba3511be3b5911f8 Original change's description: > iOS: Save perf results under Documents/perf_result.json > > TBR=henrika@webrtc.org > > Bug: webrtc:7156 > Change-Id: Ib00992cce0007e0b5c9274340df1a892f810b0c5 > Reviewed-on: https://webrtc-review.googlesource.com/29202 > Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21244} R=henrika@webrtc.org, phoglund@webrtc.org Bug: webrtc:7156 Change-Id: I85fc7bc5fce0894af90017b71b9952b61b523424 Reviewed-on: https://webrtc-review.googlesource.com/37643 Reviewed-by: Patrik Höglund <phoglund@google.com> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21518}
2018-01-05 15:34:09 +01:00
"../test:perf_test",
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_common",
Enable GN check for webrtc/call BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2720503003 Cr-Commit-Position: refs/heads/master@{#16882}
2017-02-27 15:57:45 -08:00
"../test:test_support",
"../test:video_test_common",
"../video",
"../voice_engine",
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@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/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
"//testing/gtest",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
Only one implementation of MockRtpPacketSink once MockRtpPacketSink has three identical implementations now, so time to move it to its own file. BUG=None Review-Url: https://codereview.webrtc.org/2988853002 Cr-Commit-Position: refs/heads/master@{#19183}
2017-07-28 10:05:45 -07:00
# TODO(eladalon): This should be moved, as with the TODO for |rtp_interfaces|.
rtc_source_set("mock_rtp_interfaces") {
testonly = true
sources = [
Fix deps of audio:audio_tests. Bug: webrtc:6828 Change-Id: Iae9020fda37fe40221d9a9def38c3afcc387d359 Reviewed-on: https://webrtc-review.googlesource.com/22683 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20666}
2017-11-13 17:04:05 +01:00
"fake_rtp_transport_controller_send.h",
Only one implementation of MockRtpPacketSink once MockRtpPacketSink has three identical implementations now, so time to move it to its own file. BUG=None Review-Url: https://codereview.webrtc.org/2988853002 Cr-Commit-Position: refs/heads/master@{#19183}
2017-07-28 10:05:45 -07:00
"test/mock_rtp_packet_sink_interface.h",
]
deps = [
":rtp_interfaces",
Fix deps of audio:audio_tests. Bug: webrtc:6828 Change-Id: Iae9020fda37fe40221d9a9def38c3afcc387d359 Reviewed-on: https://webrtc-review.googlesource.com/22683 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20666}
2017-11-13 17:04:05 +01:00
"..:webrtc_common",
"../modules/congestion_controller:congestion_controller",
"../modules/pacing:pacing",
Only one implementation of MockRtpPacketSink once MockRtpPacketSink has three identical implementations now, so time to move it to its own file. BUG=None Review-Url: https://codereview.webrtc.org/2988853002 Cr-Commit-Position: refs/heads/master@{#19183}
2017-07-28 10:05:45 -07:00
"../test:test_support",
"//testing/gmock",
]
}
Remove voe::TransmitMixer TransmitMixer's functionality is moved into the AudioTransportProxy owned by AudioState. This removes the need for an AudioTransport implementation in VoEBaseImpl, which means that the proxy is no longer a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl. In the short term, AudioState needs to know which AudioDeviceModule is used, so it is added in AudioState::Config. AudioTransportImpl needs to know which AudioSendStream:s are currently enabled to send, so AudioState maintains a map of them, which is reduced into a simple vector for AudioTransportImpl. To encode and transmit audio, AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame) is introduced, which is used in both the Chromium and standalone use cases. This removes the need for two different instances of voe::Channel::ProcessAndEncodeAudio(), so there is now only one, taking an AudioFrame as argument. Callers need to allocate their own AudioFrame:s, which is wasteful but not a regression since this was already happening in the voe::Channel functions. Most of the logic changed resides in AudioTransportImpl::RecordedDataIsAvailable(), where two strange things were found: 1. The clock drift parameter was ineffective since apm->echo_cancellation()->enable_drift_compensation(false) is called during initialization. 2. The output parameter 'new_mic_volume' was never set - instead it was returned as a result, causing the ADM to never update the analog mic gain (https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100). Besides this, tests are updated, and some dead code is removed which was found in the process. Bug: webrtc:4690, webrtc:8591 Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2 Reviewed-on: https://webrtc-review.googlesource.com/26681 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
rtc_source_set("mock_call_interfaces") {
testonly = true
sources = [
"test/mock_audio_send_stream.h",
]
deps = [
":call_interfaces",
"//test:test_support",
]
}
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
}
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