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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
Add WebRTC code freshness version string. This CL adds a string to the resulting WebRTC library (trying to make sure the version string will be there no matter how WebRTC is packaged). This CL should be followed by some process to regularly and automatically update the version string. No-Try: True No-Presubmit: True Bug: webrtc:12159 Change-Id: I9143aeae2cd54d0d4048c138772888100d7873cb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191223 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32825}
2020-12-14 15:28:43 +01:00
rtc_library("version") {
sources = [
"version.cc",
"version.h",
]
visibility = [ ":*" ]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("call_interfaces") {
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
sources = [
Remove chromium clang style errors affecting sdk/android/media_jni Bug: webrtc:163 Change-Id: I1e98174817ca032ee13f9a6a386803382843389d Reviewed-on: https://webrtc-review.googlesource.com/67360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Paulina Hensman <phensman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22796}
2018-04-09 14:24:52 +02:00
"audio_receive_stream.cc",
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
"audio_receive_stream.h",
"audio_send_stream.h",
Remove chromium clang style errors affecting sdk/android/media_jni Bug: webrtc:163 Change-Id: I1e98174817ca032ee13f9a6a386803382843389d Reviewed-on: https://webrtc-review.googlesource.com/67360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Paulina Hensman <phensman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22796}
2018-04-09 14:24:52 +02:00
"audio_state.cc",
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
"audio_state.h",
"call.h",
Remove chromium clang style errors affecting sdk/android/media_jni Bug: webrtc:163 Change-Id: I1e98174817ca032ee13f9a6a386803382843389d Reviewed-on: https://webrtc-review.googlesource.com/67360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Paulina Hensman <phensman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22796}
2018-04-09 14:24:52 +02:00
"call_config.cc",
"call_config.h",
"flexfec_receive_stream.cc",
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",
Delete struct webrtc::PacketTime. Replaced by a int64_t representing time in us. Bug: webtrc:9584 Change-Id: I0505c020ef741ad940203ec300e8adb103856dda Reviewed-on: https://webrtc-review.googlesource.com/91840 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24204}
2018-08-07 11:03:12 +02:00
"packet_receiver.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
]
Update gn files to support Mozilla build Bug: webrtc:8670 No-Presubmit: true Change-Id: I085dc63daa8274b5068540cbf56b6330f40643fa Reviewed-on: https://webrtc-review.googlesource.com/38920 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21624}
2018-01-15 10:20:00 -05:00
if (!build_with_mozilla) {
sources += [ "audio_send_stream.cc" ]
}
Factor out common receive stream methods to a common interface. Also including common Rtp config members. Follow up changes will remove the ReceiveRtpConfig class in Call and copy of extension headers, instead use the config directly from the receive streams and not require stream recreation for changing the headers. Bug: webrtc:11993 Change-Id: I29ff3400d45d5bffddb3ad0a078403eb102afb65 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221983 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34283}
2021-06-14 10:54:20 +02:00
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 = [
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api Bug: webrtc:11251 Change-Id: Id3b6ff1814931d8250c4aaac59e494521fbe93ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164560 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tim Na <natim@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30238}
2020-01-10 10:33:05 -08:00
":audio_sender_interface",
Factor out common receive stream methods to a common interface. Also including common Rtp config members. Follow up changes will remove the ReceiveRtpConfig class in Call and copy of extension headers, instead use the config directly from the receive streams and not require stream recreation for changing the headers. Bug: webrtc:11993 Change-Id: I29ff3400d45d5bffddb3ad0a078403eb102afb65 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221983 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34283}
2021-06-14 10:54:20 +02:00
":receive_stream_interface",
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",
Make fec controller plug-able. Bug: webrtc:8656 Change-Id: I3d42ffc92a7c95266e5d53bab03f388bd0de2592 Reviewed-on: https://webrtc-review.googlesource.com/39760 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21710}
2018-01-19 17:58:57 +01:00
"../api:fec_controller_api",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
Insert audio frame transformer between encoder and packetizer. The frame transformer is passed from RTPSenderInterface through the library to be eventually set in ChannelSend, where the frame transformation will occur in the follow-up CL. Insertable Streams Web API explainer: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md Design doc for WebRTC library changes: http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk Bug: webrtc:11380 Change-Id: I01b2adc3c96b948d182d5401a9a4fe14cf5960a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171870 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30946}
2020-03-31 11:29:56 +02:00
"../api:frame_transformer_interface",
Add base class NetworkPredictor and NetworkPredictorFactory and wire up. Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe. Bug: webrtc:10492 Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201 Commit-Queue: Ying Wang <yinwa@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27543}
2019-04-10 13:48:24 +02:00
"../api:network_state_predictor_api",
Introduce api/crypto/BUILD.gn. No-Try: True Bug: webrtc:8733 Change-Id: I8679735be1e5069e371a9f1115a54e897e09964b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152622 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29186}
2019-09-13 14:42:15 +02:00
"../api:rtc_error",
Reland "Move api/rtp_headers.h to its own build target." This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e Original change's description: > Move api/rtp_headers.h to its own build target. > > Reduces dependencies on the libjingle_peerconnection_api target from > lower-level code. > > Bug: None > Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27213} Bug: None Tbr: kwiberg@webrtc.org Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-20 08:40:23 +01:00
"../api:rtp_headers",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
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",
[Adaptation] Adding adaptation resources from Call. This CL adds AddAdaptationResource to Call and AddAdaptationResource/GetAdaptationResources method to relevant VideoSendStream and VideoStreamEncoder interfaces and implementations. Unittests are added to ensure that resources can be added to the Call both before and after the creation of a VideoSendStream and that the resources always gets added to the streams. In a follow-up CL, we will continue to plumb the resources all the way to PeerConnectionInterface, and an integration test will then be added to ensure that injected resources are capable of triggering adaptation. Bug: webrtc:11525 Change-Id: I499e9c23c3e359df943414d420b2e0ce2e9b2d56 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177002 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31499}
2020-06-11 12:07:14 +02:00
"../api/adaptation:resource_adaptation_api",
Async audio processing API API to injecting a heavy audio processing operation into WebRTC audio capture pipeline Bug: webrtc:12003 Change-Id: I9f6f58f468bd84efd0a9d53d703db6229a03959e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165788 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32291}
2020-09-30 18:27:39 +02:00
"../api/audio:audio_frame_processor",
BUILD.gn file for api/audio. Targets containing files in api/audio are moved from api/BUILD.gn to api/audio/BUILD.gn. Bug: webrtc:8844 Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981 Reviewed-on: https://webrtc-review.googlesource.com/54360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22074}
2018-02-16 13:43:49 +01:00
"../api/audio:audio_mixer_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
"../api/audio_codecs:audio_codecs_api",
Introduce api/crypto/BUILD.gn. No-Try: True Bug: webrtc:8733 Change-Id: I8679735be1e5069e371a9f1115a54e897e09964b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152622 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29186}
2019-09-13 14:42:15 +02:00
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
Integrate sync decoding in video_receive_stream Wires up DecodeSynchronizer in Call if there is a Metronome injected into the PeerConnectionFactoryDependencies. Change-Id: I362cd12648bfa0c32e73111fcd0f3296fca2b275 Bug: webrtc:13658 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251341 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35996}
2022-02-14 14:09:57 +01:00
"../api/metronome",
Enable injection of a custom NetEqFactory into PeerConnectionFactory. Injecting both a custom NetEqFactory and an AudioDecoderFactory is not supported, in that case the AudioDecoderFactory should be wrapped inside the NetEqFactory. Bug: webrtc:11005 Change-Id: I4e311eb1bfa03c91bca587d70540e81829f881c9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158720 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29673}
2019-11-01 11:47:51 +01:00
"../api/neteq:neteq_api",
Inject TaskQueueFactory in RtpTransportControllerSend. Bug: webrtc:10365 Change-Id: I1656dcf774fb347afd8b28aa998acff8942cdd9f Reviewed-on: https://webrtc-review.googlesource.com/c/125180 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26928}
2019-03-01 14:45:21 +01:00
"../api/task_queue",
Remove api/bitrate_constraints.h. Bug: webrtc:8733 Change-Id: Iaeb26e07d399f25dc18b0c4af38ed400577a5d3a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153220 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29217}
2019-09-17 14:47:38 +02:00
"../api/transport:bitrate_settings",
Allows injection of network controller factory into peer connection factory. Bug: webrtc:9155 Change-Id: I0a17024042f154297aba20f5d2dc766feb27f3f7 Reviewed-on: https://webrtc-review.googlesource.com/73123 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23313}
2018-05-18 18:05:10 +02:00
"../api/transport:network_control",
Async audio processing API API to injecting a heavy audio processing operation into WebRTC audio capture pipeline Bug: webrtc:12003 Change-Id: I9f6f58f468bd84efd0a9d53d703db6229a03959e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165788 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32291}
2020-09-30 18:27:39 +02:00
"../modules/async_audio_processing",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../modules/audio_device",
"../modules/audio_processing",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../modules/audio_processing: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",
[audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo This is part of implementing RTCRemoteInboundRtpStreamStats. The CL was split up into smaller pieces for reviewability. Spec: https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats In [1], ReportBlockData was implemented and tested. This CL adds the plumbing to make it available in MediaSenderInfo for audio streams, but the code is not wired up to make use of these stats. In follow-up CL [2], ReportBlockData will be used to implement RTCRemoteInboundRtpStreamStats. The follow-up CL will add integration tests exercising the full code path. [1] https://webrtc-review.googlesource.com/c/src/+/136584 [2] https://webrtc-review.googlesource.com/c/src/+/138067 Bug: webrtc:10455 Change-Id: Id8940090cc9c121e8f06ccdb068a22ce24c07092 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138066 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28072}
2019-05-27 12:19:33 +02:00
"../modules/rtp_rtcp:rtp_rtcp_format",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../modules/utility",
"../rtc_base",
This CL removes all usages of our custom ostream << overloads. This prepares us for removing them altogether. Bug: webrtc:8982 Change-Id: I66002cc8d4bf0e07925766d568d2498422f0f38e Reviewed-on: https://webrtc-review.googlesource.com/64142 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Henrik Grunell <henrikg@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22707}
2018-04-03 13:40:05 +02:00
"../rtc_base:audio_format_to_string",
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 12:03:01 +02:00
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Remove //rtc_base:refcount from public deps Bug: webrtc:8603 Change-Id: Ib27a107ae809df739492846175f0e9c4af40d21a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257910 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36447}
2022-04-05 02:54:12 +02:00
"../rtc_base:refcount",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Removing unnecessary dependencies on socket.h. Since rtc:SentPacket was removed to a separate header. Some usages of socket.h can be replaced with sent_packet.h which defines a lot less things, making future maintenance simpler. Bug: webrtc:9586 Change-Id: If705edda293c389cf2a175117db52a6720a7be86 Reviewed-on: https://webrtc-review.googlesource.com/c/106144 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25201}
2018-10-15 21:06:26 +02:00
"../rtc_base/network:sent_packet",
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
]
Remove REMB throttling funcionality from PacketRouter This removes PacketRouter inheritance from RemoteBitrateObserver and TransportFeedbackSenderInterface. Call binds methods for sending REMB and transport feedback messages from RemoteCongestionController to PacketRouter. This is needed until the RTCPTranseiver is used instead of the RTP modules. Bug: webrtc:12693 Change-Id: I7088de497cd6d1e15c98788ff3e6b0a2c8897ea8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215965 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33993}
2021-05-12 09:02:43 +02:00
absl_deps = [
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/types:optional",
]
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
}
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api Bug: webrtc:11251 Change-Id: Id3b6ff1814931d8250c4aaac59e494521fbe93ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164560 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tim Na <natim@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30238}
2020-01-10 10:33:05 -08:00
rtc_source_set("audio_sender_interface") {
visibility = [ "*" ]
Reformat GN files. `gn format` recently [1] changed its formatting behavior for deps, source, and a few other elements when they are assigned (with =) single-element lists to be consistent with the formatting of updates (with +=) with single-element. Now that we've rolled in a GN binary with the change, reformat all files so that people don't get presubmit warnings due to this. CL generated with: $ git ls-files | grep BUILD.gn | xargs gn format $ gn format build_overrides/build.gni $ gn format build_overrides/gtest.gni $ gn format modules/audio_coding/audio_coding.gni $ gn format webrtc.gni $ gn format .gn Plus a few manual changes to add exceptions for "public_deps" (after changing these lines the presubmit started to complain). [1] - https://gn-review.googlesource.com/c/gn/+/6860 Bug: webrtc:11302 Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:10:10 +01:00
sources = [ "audio_sender.h" ]
deps = [ "../api/audio:audio_frame_api" ]
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api Bug: webrtc:11251 Change-Id: Id3b6ff1814931d8250c4aaac59e494521fbe93ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164560 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tim Na <natim@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30238}
2020-01-10 10:33:05 -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
Use backticks not vertical bars to denote variables in comments for /call Bug: webrtc:12338 Change-Id: I8f92127b61352bd4b98a0690e9a0435bb6c6f870 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226943 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34569}
2021-07-26 12:40:21 +02:00
# when interfaces have stabilized. See also TODO for `mock_rtp_interfaces`.
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("rtp_interfaces") {
Temporarily increase visibility of pacing and call/rtp_interfaces NOTRY=true Bug: webrtc:9808 Change-Id: I1ee8f6843167bb8904a367d1d2a249a6b15a18db Reviewed-on: https://webrtc-review.googlesource.com/c/103800 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24984}
2018-10-04 13:41:03 +02:00
# Client code SHOULD NOT USE THIS TARGET, but for now it needs to be public
# because there exists client code that uses it.
# TODO(bugs.webrtc.org/9808): Move to private visibility as soon as that
# client code gets updated.
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 = [
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",
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. This way we can have custom implementation of RtpTransportControllerSendInterface and pass it properly to Call. Call relies on RtpTransportControllerSendInterface already so this is natural way to customize RTP related classes. If there is custom factory present in dependencies it will be used, otherwise default factory will be used. Intention behind this change is to have ability to have custom QoS with custom parameters. Bug: webrtc:12778 Change-Id: I5b88957025621ef4bcd63eaa98c218ad213da9c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217769 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Philipp Hancke <phancke@nvidia.com> Cr-Commit-Position: refs/heads/master@{#34181}
2021-05-31 14:02:28 +02:00
"rtp_transport_config.h",
"rtp_transport_controller_send_factory_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",
Move FecController to RtpVideoSender. This also moves the packet feedback tracking to RtpVideoSender. Bug: webrtc:9517 Change-Id: Ifb1ff85051730108a0b0d1dd30f6f8595ad2af6e Reviewed-on: https://webrtc-review.googlesource.com/c/95920 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25019}
2018-10-04 15:21:55 +02:00
"../api:fec_controller_api",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
Insert frame transformer between Encoded and Packetizer. Add a new API in RTPSenderInterface, to be called from the browser side to insert a frame transformer between the Encoded and the Packetizer. The frame transformer is passed from RTPSenderInterface through the library to be eventually set in RTPSenderVideo, where the frame transformation will occur in the follow-up CL https://webrtc-review.googlesource.com/c/src/+/169128. Insertable Streams Web API explainer: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md Design doc for WebRTC library changes: http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk Bug: webrtc:11380 Change-Id: I46cd0d8a798c2736c837e90cbf90d8901c7d27fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169127 Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30642}
2020-02-27 16:16:55 +01:00
"../api:frame_transformer_interface",
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. This way we can have custom implementation of RtpTransportControllerSendInterface and pass it properly to Call. Call relies on RtpTransportControllerSendInterface already so this is natural way to customize RTP related classes. If there is custom factory present in dependencies it will be used, otherwise default factory will be used. Intention behind this change is to have ability to have custom QoS with custom parameters. Bug: webrtc:12778 Change-Id: I5b88957025621ef4bcd63eaa98c218ad213da9c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217769 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Philipp Hancke <phancke@nvidia.com> Cr-Commit-Position: refs/heads/master@{#34181}
2021-05-31 14:02:28 +02:00
"../api:network_state_predictor_api",
Reland "Move api/rtp_headers.h to its own build target." This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e Original change's description: > Move api/rtp_headers.h to its own build target. > > Reduces dependencies on the libjingle_peerconnection_api target from > lower-level code. > > Bug: None > Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27213} Bug: None Tbr: kwiberg@webrtc.org Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-20 08:40:23 +01:00
"../api:rtp_headers",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
Introduce api/crypto/BUILD.gn. No-Try: True Bug: webrtc:8733 Change-Id: I8679735be1e5069e371a9f1115a54e897e09964b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152622 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29186}
2019-09-13 14:42:15 +02:00
"../api/crypto:options",
Delete deprecated rtc_event_log header Bug: webrtc:10206 Change-Id: I9ed3148843c647372993729b87c0e74741ab540b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 12:24:53 +02:00
"../api/rtc_event_log",
New api struct BitrateSettings. Replaces both BitrateConstraintsMask and PeerConnectionInterface::BitrateParameters. The latter is kept temporarily for backwards compatibility. Bug: None Change-Id: Ibe1d043f2a76e56ff67809774e9c0f5e0ec9e00f Reviewed-on: https://webrtc-review.googlesource.com/74020 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23148}
2018-05-07 14:01:37 +02:00
"../api/transport:bitrate_settings",
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. This way we can have custom implementation of RtpTransportControllerSendInterface and pass it properly to Call. Call relies on RtpTransportControllerSendInterface already so this is natural way to customize RTP related classes. If there is custom factory present in dependencies it will be used, otherwise default factory will be used. Intention behind this change is to have ability to have custom QoS with custom parameters. Bug: webrtc:12778 Change-Id: I5b88957025621ef4bcd63eaa98c218ad213da9c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217769 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Philipp Hancke <phancke@nvidia.com> Cr-Commit-Position: refs/heads/master@{#34181}
2021-05-31 14:02:28 +02:00
"../api/transport:network_control",
Add RtpPacketPacer interface for pacer control The PacedSender is being reworked and will need an interface so we can inject different implementations of it. This CL introduces a new RtpPacketPacer interface inside the pacing module. This interface handles the details of _how_ packets should be paced, such as pacing rates/account for audio/max queue length etc. The RtpPacketSender interface exposed from the rtp_rtcp module handles only the actual sending of packets. Some minor cleanups are included here. Bug: webrtc:10809 Change-Id: I150b1a6262306d99e3f9d5f0b4afdb16a50e5ad8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145212 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28699}
2019-07-29 16:38:27 +02:00
"../api/units:timestamp",
Move FrameCounts and FrameCountObserver to common_video/frame_counts.h Bug: webrtc:7660 Change-Id: Ic42227508e0e26b5443de0ef0e0722126cbbb8fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182501 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32005}
2020-08-25 10:28:50 +02:00
"../common_video:frame_counts",
Refactoring PayloadRouter. - Move PayloadRouter to RtpTransportControllerInterface. - Move RetransmissionLimiter inside RtpTransportControllerSend from VideoSendStreamImpl. - Move video RTP specifics into PayloadRouter, in particular ownership of the RTP modules. - PayloadRouter now contains all video specific RTP code, and will be renamed in a follow-up to VideoRtpSender. - Introduce VideoRtpSenderInterface. Bug: webrtc:9517 Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38 Reviewed-on: https://webrtc-review.googlesource.com/88240 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24009}
2018-07-17 16:03:46 +02:00
"../modules/rtp_rtcp:rtp_rtcp_format",
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. This way we can have custom implementation of RtpTransportControllerSendInterface and pass it properly to Call. Call relies on RtpTransportControllerSendInterface already so this is natural way to customize RTP related classes. If there is custom factory present in dependencies it will be used, otherwise default factory will be used. Intention behind this change is to have ability to have custom QoS with custom parameters. Bug: webrtc:12778 Change-Id: I5b88957025621ef4bcd63eaa98c218ad213da9c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217769 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Philipp Hancke <phancke@nvidia.com> Cr-Commit-Position: refs/heads/master@{#34181}
2021-05-31 14:02:28 +02:00
"../modules/utility",
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats. --- Background --- The webrtc::VideoSendStream::StreamStats are converted into VideoSenderInfo objects which turn into "outbound-rtp" stats objects in getStats() (or "ssrc" objects in legacy getStats()). StreamStats are created for each type of substream: RTP media streams, RTX streams and FlexFEC streams - each with individual packet counters. The RTX stream is responsible for retransmissions of a referenced media stream and the FlexFEC stream is responsible for FEC of a referenced media stream. RTX/FEC streams do not show up as separate objects in getStats(). Only the media streams become "outbound-rtp" objects, but their packet and byte counters have to include the RTX and FEC counters. --- Overview of this CL --- This CL adds MergeInfoAboutOutboundRtpSubstreams(). It takes StreamStats of all kinds as input, and outputs media-only StreamStats - incorporating the RTX and FEC counters into the relevant media StreamStats. The merged StreamStats objects is a smaller set of objects than the non-merged counterparts, but when aggregating all packet counters together we end up with exact same packet and count as before. Because WebRtcVideoSendStream::GetVideoSenderInfo() currently aggregates the StreamStats into a single VideoSenderInfo (single "outbound-rtp"), this CL should not have any observable side-effects. Prior to this CL: aggregate StreamStats. After this CL: merge StreamStats and then aggregate them. However, when simulcast stats are implemented (WIP CL: https://webrtc-review.googlesource.com/c/src/+/168120) each RTP media stream should turn into an individual "outbound-rtp" object. We will then no longer aggregate all StreamStats into a single "info". This CL unblocks simulcast stats by providing StreamStats objects that could be turned into individual VideoSenderInfos. --- The Changes --- 1. Methods added to RtpConfig to be able to easily tell the relationship between RTP, RTX and FEC ssrcs. 2. StreamStats gets a StreamType (kMedia, kRtx or kFlexfec) that replaces the booleans (is_rtx, is_flexfec). 3. "referenced_media_ssrc" is added to StreamStats, making it possible to tell which kRtx/kFlexFec stream stats need to be merged with which kMedia StreamStats. 4. MergeInfoAboutOutboundRtpSubstreams() added and used. Bug: webrtc:11439 Change-Id: Iaf9002041169a054ddfd32c7ea06bd1dc36c6bca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170826 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30869}
2020-03-24 13:30:50 +01:00
"../rtc_base:checks",
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. This way we can have custom implementation of RtpTransportControllerSendInterface and pass it properly to Call. Call relies on RtpTransportControllerSendInterface already so this is natural way to customize RTP related classes. If there is custom factory present in dependencies it will be used, otherwise default factory will be used. Intention behind this change is to have ability to have custom QoS with custom parameters. Bug: webrtc:12778 Change-Id: I5b88957025621ef4bcd63eaa98c218ad213da9c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217769 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Philipp Hancke <phancke@nvidia.com> Cr-Commit-Position: refs/heads/master@{#34181}
2021-05-31 14:02:28 +02:00
"../rtc_base:rtc_task_queue",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats. --- Background --- The webrtc::VideoSendStream::StreamStats are converted into VideoSenderInfo objects which turn into "outbound-rtp" stats objects in getStats() (or "ssrc" objects in legacy getStats()). StreamStats are created for each type of substream: RTP media streams, RTX streams and FlexFEC streams - each with individual packet counters. The RTX stream is responsible for retransmissions of a referenced media stream and the FlexFEC stream is responsible for FEC of a referenced media stream. RTX/FEC streams do not show up as separate objects in getStats(). Only the media streams become "outbound-rtp" objects, but their packet and byte counters have to include the RTX and FEC counters. --- Overview of this CL --- This CL adds MergeInfoAboutOutboundRtpSubstreams(). It takes StreamStats of all kinds as input, and outputs media-only StreamStats - incorporating the RTX and FEC counters into the relevant media StreamStats. The merged StreamStats objects is a smaller set of objects than the non-merged counterparts, but when aggregating all packet counters together we end up with exact same packet and count as before. Because WebRtcVideoSendStream::GetVideoSenderInfo() currently aggregates the StreamStats into a single VideoSenderInfo (single "outbound-rtp"), this CL should not have any observable side-effects. Prior to this CL: aggregate StreamStats. After this CL: merge StreamStats and then aggregate them. However, when simulcast stats are implemented (WIP CL: https://webrtc-review.googlesource.com/c/src/+/168120) each RTP media stream should turn into an individual "outbound-rtp" object. We will then no longer aggregate all StreamStats into a single "info". This CL unblocks simulcast stats by providing StreamStats objects that could be turned into individual VideoSenderInfos. --- The Changes --- 1. Methods added to RtpConfig to be able to easily tell the relationship between RTP, RTX and FEC ssrcs. 2. StreamStats gets a StreamType (kMedia, kRtx or kFlexfec) that replaces the booleans (is_rtx, is_flexfec). 3. "referenced_media_ssrc" is added to StreamStats, making it possible to tell which kRtx/kFlexFec stream stats need to be merged with which kMedia StreamStats. 4. MergeInfoAboutOutboundRtpSubstreams() added and used. Bug: webrtc:11439 Change-Id: Iaf9002041169a054ddfd32c7ea06bd1dc36c6bca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170826 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30869}
2020-03-24 13:30:50 +01:00
"//third_party/abseil-cpp/absl/algorithm:container",
Replace rtc::Optional with absl::optional in audio, call and video This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameters 'audio call video': #!/bin/bash find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: I02c5db956846a88a268a300ba086703a02d62e36 Reviewed-on: https://webrtc-review.googlesource.com/83722 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23628}
2018-06-15 12:28:07 +02:00
"//third_party/abseil-cpp/absl/types:optional",
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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 = [
"rtp_demuxer.cc",
"rtp_demuxer.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",
]
deps = [
":rtp_interfaces",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../api:array_view",
Reland "Move api/rtp_headers.h to its own build target." This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e Original change's description: > Move api/rtp_headers.h to its own build target. > > Reduces dependencies on the libjingle_peerconnection_api target from > lower-level code. > > Bug: None > Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27213} Bug: None Tbr: kwiberg@webrtc.org Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-20 08:40:23 +01:00
"../api:rtp_headers",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
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 //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Use flat_map/flat_set in RtpDemuxer Except for a use of std::multimap (for which there currently is no drop-in replacement), use webrtc::flat_map and flat_set for increased performance. The number of ssrcs/mids/payload types/etc is likely to be small and is generally updated very rarely, but looked up a lot. RtpDemuxer::ResolveSink's use of std::map represents about 0.6% CPU in heavily loaded media servers. It does quite a few map lookups, and by taking advantage of the increased cache locality of the flat_map and flat_set containers, performance should be increased. A previous attempt to use std::unordered_map failed in change 216243, which was reverted. This is the second attempt. Bug: webrtc:12689 Change-Id: Ifdbec63b2fd8f2f52e45ebaf12834b11dd7a41c5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224662 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34429}
2021-07-01 21:42:02 +02:00
"../rtc_base/containers:flat_map",
"../rtc_base/containers:flat_set",
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
]
Add ctor to RtpDemuxerCriteria to allow supplying 'mid' and 'rsid'. Since another CL [1] grew a bit in scope, this CL needs to land first, coupled with downstream updates. [1]: https://webrtc-review.googlesource.com/c/src/+/244083 Bug: webrtc:12517, webrtc:11993 Change-Id: I80d826a0d3f3240a2cecf8cbe09842cba0902b16 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244085 Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35611}
2022-01-03 13:04:51 +00:00
absl_deps = [
"//third_party/abseil-cpp/absl/strings:strings",
"//third_party/abseil-cpp/absl/types: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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("rtp_sender") {
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 "Move allocation and rtp conversion logic out of payload router." This reverts commit c2406e4eaf7703c6c64d21318186adda791e09fd. Reason for revert: Reland by removing the conflict with the broken CL. Original change's description: > Revert "Move allocation and rtp conversion logic out of payload router." > > This reverts commit 1da4d79ba3275b3fa48cad3b2c0949e0d3b7afe7. > > Reason for revert: Need to revert https://webrtc-review.googlesource.com/c/src/+/88220 > > This causes a merge conflict. So need to revert this first. > > Original change's description: > > Move allocation and rtp conversion logic out of payload router. > > > > Makes it easier to write tests, and allows for moving rtp module > > ownership into the payload router in the future. > > > > The RtpPayloadParams class is split into declaration and definition and > > moved into separate files. > > > > Bug: webrtc:9517 > > Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad > > Reviewed-on: https://webrtc-review.googlesource.com/88564 > > Commit-Queue: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#23983} > > TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > Change-Id: I342c4bf483d975c87c706fe7f76f44e2dc60fe4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9517 > Reviewed-on: https://webrtc-review.googlesource.com/88821 > Reviewed-by: JT Teh <jtteh@webrtc.org> > Commit-Queue: JT Teh <jtteh@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23991} TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,lliuu@webrtc.org,jtteh@webrtc.org,tkchin@webrtc.org Change-Id: I154145cdbc668feee86dbe78860147a6954fee6c No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9517 Reviewed-on: https://webrtc-review.googlesource.com/89020 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23996}
2018-07-17 10:16:41 +02:00
"rtp_payload_params.cc",
"rtp_payload_params.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.cc",
"rtp_transport_controller_send.h",
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. This way we can have custom implementation of RtpTransportControllerSendInterface and pass it properly to Call. Call relies on RtpTransportControllerSendInterface already so this is natural way to customize RTP related classes. If there is custom factory present in dependencies it will be used, otherwise default factory will be used. Intention behind this change is to have ability to have custom QoS with custom parameters. Bug: webrtc:12778 Change-Id: I5b88957025621ef4bcd63eaa98c218ad213da9c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217769 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Philipp Hancke <phancke@nvidia.com> Cr-Commit-Position: refs/heads/master@{#34181}
2021-05-31 14:02:28 +02:00
"rtp_transport_controller_send_factory.h",
Rename PayloadRouter to RtpVideoSender. Bug: webrtc:9517 Change-Id: I18397a28067dbe5029fc80fe2eef360869abb339 Reviewed-on: https://webrtc-review.googlesource.com/89380 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24039}
2018-07-19 10:34:38 +02:00
"rtp_video_sender.cc",
"rtp_video_sender.h",
"rtp_video_sender_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
]
deps = [
Moved bitrate configuration to rtp controller Since rtp transport controller send owns the congestion controller it also should own the bitrate configuration logic, this way it can initialize the send side congestion controller with the bitrate configuration. Bug: webrtc:8415 Change-Id: Ifaa16139ca477cb1c80bf4aa24f17652af997553 Reviewed-on: https://webrtc-review.googlesource.com/54303 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22127}
2018-02-21 13:01:55 +01:00
":bitrate_configurator",
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",
RTPSenderVideo::GetSentRtpPacketInfo() over a set of sequence numbers Add a version of RTPSenderVideo::GetSentRtpPacketInfo() that operates over a set of numbers, so as to only grab the lock once. Bug: webrtc:10501 Change-Id: I9453b0cb44dcd6e2ce196390b2c5c9a7dd6d800a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132014 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27544}
2019-04-10 15:55:00 +02:00
"../api:array_view",
Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender Bug: webrtc:9883 Change-Id: I12d342ecd5eb0cc859123fe31fc759f6f60f7c8b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156940 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29492}
2019-10-15 15:58:37 +02:00
"../api:bitrate_allocation",
Move FecController to RtpVideoSender. This also moves the packet feedback tracking to RtpVideoSender. Bug: webrtc:9517 Change-Id: Ifb1ff85051730108a0b0d1dd30f6f8595ad2af6e Reviewed-on: https://webrtc-review.googlesource.com/c/95920 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25019}
2018-10-04 15:21:55 +02:00
"../api:fec_controller_api",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
Add base class NetworkPredictor and NetworkPredictorFactory and wire up. Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe. Bug: webrtc:10492 Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201 Commit-Queue: Ying Wang <yinwa@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27543}
2019-04-10 13:48:24 +02:00
"../api:network_state_predictor_api",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Refactoring PayloadRouter. - Move PayloadRouter to RtpTransportControllerInterface. - Move RetransmissionLimiter inside RtpTransportControllerSend from VideoSendStreamImpl. - Move video RTP specifics into PayloadRouter, in particular ownership of the RTP modules. - PayloadRouter now contains all video specific RTP code, and will be renamed in a follow-up to VideoRtpSender. - Introduce VideoRtpSenderInterface. Bug: webrtc:9517 Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38 Reviewed-on: https://webrtc-review.googlesource.com/88240 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24009}
2018-07-17 16:03:46 +02:00
"../api:transport_api",
Delete deprecated rtc_event_log header Bug: webrtc:10206 Change-Id: I9ed3148843c647372993729b87c0e74741ab540b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 12:24:53 +02:00
"../api/rtc_event_log",
Move ownership of RTPSenderVideo one more level up, to RtpVideoSender The idea is to let the RtpRtcp and RTPSender classes be responsible for media-agnostic RTP transport, and move out the media-specific processing, such as packetization and media-specific headers. Bug: webrtc:7135 Change-Id: Ib0ce45bf06713b3eb6c06acd91c5168856874e4e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123187 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26954}
2019-03-04 16:49:25 +01:00
"../api/transport:field_trial_based_config",
Merges RtpTransportControllerSend with SendSideCongestionController. Bug: webrtc:9586 Change-Id: I50332f2e128f107e40af7776be0ed530e20774d9 Reviewed-on: https://webrtc-review.googlesource.com/c/113183 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25922}
2018-12-05 17:35:35 +01:00
"../api/transport:goog_cc",
Moves network control interface to API. This prepares for allowing injection of a network controller. Bug: webrtc:9155 Change-Id: I5624f47738db9c5cd4750eac76cb6289e06a7aa3 Reviewed-on: https://webrtc-review.googlesource.com/73100 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23188}
2018-05-09 10:33:39 +02:00
"../api/transport:network_control",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api/units:data_rate",
"../api/units:time_delta",
"../api/units:timestamp",
"../api/video:video_frame",
Hookup VideoSendStreamImpl::OnVideoLayersAllocationUpdate to RtpVideoSender. Bug: webrtc:12000 Change-Id: Ieddbad8e6f4e7456441153d432f4dfb32e16d255 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188627 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32439}
2020-10-16 10:30:40 +02:00
"../api/video:video_layers_allocation",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../api/video:video_rtp_headers",
Moved PayloadRouter to call/. This is done in preparation for moving ownership of PayloadRouter to RtpTransportControllerSend. Bug: webrtc:9517 Change-Id: I4a5b449cbcfc23db594dc5bb68ca322dd8fa33b7 Reviewed-on: https://webrtc-review.googlesource.com/88241 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23936}
2018-07-11 17:11:31 +02:00
"../api/video_codecs:video_codecs_api",
Adds route changes in event logs. Bug: webrtc:10614 Change-Id: Ifd859c977fc66cb606914ddb38a3fb3618e3ad90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135952 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27924}
2019-05-13 11:57:42 +02:00
"../logging:rtc_event_bwe",
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",
Merges RtpTransportControllerSend with SendSideCongestionController. Bug: webrtc:9586 Change-Id: I50332f2e128f107e40af7776be0ed530e20774d9 Reviewed-on: https://webrtc-review.googlesource.com/c/113183 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25922}
2018-12-05 17:35:35 +01:00
"../modules/congestion_controller/rtp:control_handler",
"../modules/congestion_controller/rtp:transport_feedback",
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",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../modules/rtp_rtcp",
Moved PayloadRouter to call/. This is done in preparation for moving ownership of PayloadRouter to RtpTransportControllerSend. Bug: webrtc:9517 Change-Id: I4a5b449cbcfc23db594dc5bb68ca322dd8fa33b7 Reviewed-on: https://webrtc-review.googlesource.com/88241 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23936}
2018-07-11 17:11:31 +02:00
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/rtp_rtcp:rtp_video_header",
Moved pacer and congestion thread from call. Moving the module process thread responsible for running the pacer and the send side congestion controller to RtpTransportControllerSend since it already owns the pacer and the congestion controller. They are also moved to a common thread rather than using two separate threads. As part of the move, the remote bitrate estimator has been moved to the common process thread in the Call class. Previously it was run on the removed pacer thread. Bug: webrtc:8415 Change-Id: I4322eef30d8b97b9611f33af7e560703b710d232 Reviewed-on: https://webrtc-review.googlesource.com/55700 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22166}
2018-02-22 11:10:18 +01:00
"../modules/utility",
Calculate chain_diff for DependencyDescriptor from GenericFrameInfo Bug: webrtc:10342 Change-Id: Ic88d58dc011c641a3b251eef74064066975296fa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175652 Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31331}
2020-05-19 14:48:19 +02:00
"../modules/video_coding:chain_diff_calculator",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../modules/video_coding:codec_globals_headers",
Populate generic descriptor based on GenericFrameInfo when available. Bug: webrtc:10342 Change-Id: Iff769d2604fd79784bcb09874d2803793d20bde5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167000 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30505}
2020-02-10 16:22:57 +01:00
"../modules/video_coding:frame_dependencies_calculator",
Moved PayloadRouter to call/. This is done in preparation for moving ownership of PayloadRouter to RtpTransportControllerSend. Bug: webrtc:9517 Change-Id: I4a5b449cbcfc23db594dc5bb68ca322dd8fa33b7 Reviewed-on: https://webrtc-review.googlesource.com/88241 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23936}
2018-07-11 17:11:31 +02:00
"../modules/video_coding:video_codec_interface",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../rtc_base",
Moved PayloadRouter to call/. This is done in preparation for moving ownership of PayloadRouter to RtpTransportControllerSend. Bug: webrtc:9517 Change-Id: I4a5b449cbcfc23db594dc5bb68ca322dd8fa33b7 Reviewed-on: https://webrtc-review.googlesource.com/88241 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23936}
2018-07-11 17:11:31 +02:00
"../rtc_base:checks",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Move location out of rtc_base_approved Bug: webrtc:9838 Change-Id: I3510f3d01df532cb228a6cc14be830f6ab81a8e1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258764 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36576}
2022-04-14 13:13:56 +02:00
"../rtc_base:location",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Move race_checker out of rtc_base_approved Bug: webrtc:9838 Change-Id: If180abcca1ef598314de3aed70e4a6eb04f062d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258770 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36607}
2022-04-14 13:38:05 +02:00
"../rtc_base:race_checker",
Move random out of rtc_base_approved Bug: webrtc:9838 Change-Id: I64a5ef18c19d446139354d04aa6cb2a76d18aad0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258762 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36572}
2022-04-14 12:41:26 +02:00
"../rtc_base:random",
Refactoring PayloadRouter. - Move PayloadRouter to RtpTransportControllerInterface. - Move RetransmissionLimiter inside RtpTransportControllerSend from VideoSendStreamImpl. - Move video RTP specifics into PayloadRouter, in particular ownership of the RTP modules. - PayloadRouter now contains all video specific RTP code, and will be renamed in a follow-up to VideoRtpSender. - Introduce VideoRtpSenderInterface. Bug: webrtc:9517 Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38 Reviewed-on: https://webrtc-review.googlesource.com/88240 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24009}
2018-07-17 16:03:46 +02:00
"../rtc_base:rate_limiter",
Moving task queue from Call to transport controller. Moving ownership of worker task queue in Call to RtpTransportControllerSend. This CL also ensures that the task queue is not destroyed until the process thread running SendSideCongestionController is stopped. The worker queue should be owned by RtpTransportControllerSend since it is mainly used for rtp and transport related tasks such as bitrate allocation and signaling network state. Bug: webrtc:9232 Change-Id: I211edf1a3b9f9b2572875d5584cb788cb2449ef9 Reviewed-on: https://webrtc-review.googlesource.com/63023 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23119}
2018-05-04 14:08:15 +02:00
"../rtc_base:rtc_task_queue",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../rtc_base:timeutils",
Migrate call/ to webrtc::Mutex. Bug: webrtc:11567 Change-Id: Iab7142c77bc0c1a026cf5121b756094e05bccefe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176742 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31636}
2020-07-06 17:41:35 +02:00
"../rtc_base/synchronization:mutex",
Adds repeating task class. This CL adds a single class to manage the use case of having a task that repeats itself by a fixed or variable interval. It replaces the repeating task previously locally defined for rtp transport controller send as well as the cancelable periodic task. Furthermore, it is introduced where one off repeating tasks were created before. It provides the currently used functionality of the cancelable periodic task, but not some of the unused features, such as allowing cancellation of tasks before they are started and cancellation of a task after the owning task queue has been destroyed. Bug: webrtc:9883 Change-Id: Ifa7edee836c2a64fce16a7d0f682eb09c879eaca Reviewed-on: https://webrtc-review.googlesource.com/c/116182 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26313}
2019-01-18 10:30:54 +01:00
"../rtc_base/task_utils:repeating_task",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
Add support for early loss detection using transport feedback. Bug: webrtc:10676 Change-Id: Ifdef133e123a0c54204397fb323f4c671c40a464 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135881 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28106}
2019-05-29 13:02:24 +02:00
"//third_party/abseil-cpp/absl/algorithm:container",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"//third_party/abseil-cpp/absl/container:inlined_vector",
Reduces locking in RtpSenderVideo. This CL removes some unnecessary locking, since we are already serialized by the lock in VideoStreamEncoder. A simple RaceChecker is used to verify this. We also remove the usage of RegisterPayloadType() and replace it with a parameter in SendVideo instead. This way we are prepared for removing the payload type map and lock entirely. Some usage still exists downstream and needs to be removed before cleaning this up. Bug: webrtc:10809 Change-Id: Ie90163f15d11c8843f3beaf9a0df0dd2a1fd5ce6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154700 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29372}
2019-10-02 20:55:39 +02:00
"//third_party/abseil-cpp/absl/strings:strings",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
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
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("bitrate_configurator") {
Extracted bitrate configuration from call class. This separates the bitrate configuration logic from other call specific logic, creating a greater separation of concern and simplifying testing. The old call tests are kept but can be removed in the future reducing the dependencies on rtp transport control interface and congestion control in the system, which will simplify future refactoring. This also prepares for moving the bitrate configuration responsibility to the rtp transport controller in a later CL. Bug: webrtc:8415 Change-Id: I97126e89f30b63fc9b5d98a0bed1c29f18a6ed44 Reviewed-on: https://webrtc-review.googlesource.com/54401 Reviewed-by: Zach Stein <zstein@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22124}
2018-02-20 19:38:37 +01:00
sources = [
"rtp_bitrate_configurator.cc",
"rtp_bitrate_configurator.h",
]
deps = [
":rtp_interfaces",
Delete unneeded dependencies on libjingle_peerconnection_api Also annotate a few of the remaining uses, to guide further splits of that large build target. Bug: webrtc:8733 Change-Id: I16ac33ab48e6d39a1a8dbc2a3fc671d8db6dbfe9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150789 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29001}
2019-08-29 12:16:56 +02:00
# For api/bitrate_constraints.h
Convert video quality test from a TEST_F to a TEST fixture. The purpose is to make the fixture reusable in downstream projects. The CL adds the following things to API: - api/test/video_quality_test_fixture.h - api/test/create_video_quality_test_fixture.h The following things are moved to API: - call/bitrate_constraints.h (api/bitrate_constraints.h) - call/simulated_network.h (api/test/simulated_network.h) - call/media_type.h (api/mediatypes.h) These are required by the params struct passed to the fixture. I didn't attempt to split the params struct into an internal-only and public version in this CL, and as a result we need to pull in the above things. They are quite harmless though, so I think it's worth it in order to avoid splitting up the test config struct. This CL doesn't solve all the problems we need to implement downstream tests; we probably need to upstream tracing variants of FakeNetworkPipe for instance, but that will come later. This puts in place the basic structure for now. Bug: None Change-Id: I35e26ed126fad27bc7b2a465400291084f6ac911 Reviewed-on: https://webrtc-review.googlesource.com/69601 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23714}
2018-06-21 16:58:01 +02:00
"../api:libjingle_peerconnection_api",
New api struct BitrateSettings. Replaces both BitrateConstraintsMask and PeerConnectionInterface::BitrateParameters. The latter is kept temporarily for backwards compatibility. Bug: None Change-Id: Ibe1d043f2a76e56ff67809774e9c0f5e0ec9e00f Reviewed-on: https://webrtc-review.googlesource.com/74020 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23148}
2018-05-07 14:01:37 +02:00
"../api/transport:bitrate_settings",
Allow setting a bandwidth cap for relayed connections. For now the capping is experimental and applied via a field trial. Bug: webrtc:11434 Change-Id: Id8e6e9b948f099a0940974a9a431b5b0a43c32f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171226 Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30909}
2020-03-26 20:37:21 +01:00
"../api/units:data_rate",
Extracted bitrate configuration from call class. This separates the bitrate configuration logic from other call specific logic, creating a greater separation of concern and simplifying testing. The old call tests are kept but can be removed in the future reducing the dependencies on rtp transport control interface and congestion control in the system, which will simplify future refactoring. This also prepares for moving the bitrate configuration responsibility to the rtp transport controller in a later CL. Bug: webrtc:8415 Change-Id: I97126e89f30b63fc9b5d98a0bed1c29f18a6ed44 Reviewed-on: https://webrtc-review.googlesource.com/54401 Reviewed-by: Zach Stein <zstein@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22124}
2018-02-20 19:38:37 +01:00
"../rtc_base:checks",
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
Extracted bitrate configuration from call class. This separates the bitrate configuration logic from other call specific logic, creating a greater separation of concern and simplifying testing. The old call tests are kept but can be removed in the future reducing the dependencies on rtp transport control interface and congestion control in the system, which will simplify future refactoring. This also prepares for moving the bitrate configuration responsibility to the rtp transport controller in a later CL. Bug: webrtc:8415 Change-Id: I97126e89f30b63fc9b5d98a0bed1c29f18a6ed44 Reviewed-on: https://webrtc-review.googlesource.com/54401 Reviewed-by: Zach Stein <zstein@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22124}
2018-02-20 19:38:37 +01:00
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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 = [
Moves BitrateAllocationUpdate to api. This way it can be forwarded to lower layers. This makes it easier to add information without having to change signatures of intermediate classes. This will be used in a later CL to use the link capacity in the Opus decoder. Bug: webrtc:9718 Change-Id: I4a4c9d104fedb0e4a0bb7f14d169475940edbf7e Reviewed-on: https://webrtc-review.googlesource.com/c/111508 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25738}
2018-11-21 19:18:39 +01:00
"../api:bitrate_allocation",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Using struct for bitrate allocation limits. Bug: webrtc:9883 Change-Id: I855c29808ffa14626d78842491fdf81cd00589e6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153344 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29227}
2019-09-18 18:31:52 +02:00
"../api/transport:network_control",
Using unit classes in BitrateAllocationUpdate struct. This prepares for moving BitrateAllocationUpdate to API. Bug: webrtc:9718 Change-Id: Ib2bcedb6b68fde33b6a2466f40829e86438aa973 Reviewed-on: https://webrtc-review.googlesource.com/c/111507 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25737}
2018-11-21 19:13:07 +01:00
"../api/units:data_rate",
"../api/units:time_delta",
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 //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Propagating TargetRate struct to BitrateAllocator. Bug: webrtc:9883 Change-Id: I443ac7f1ef0f933e2165fdb2f912d314acc7f2f8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153345 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29241}
2019-09-19 14:54:43 +02:00
"../rtc_base:safe_minmax",
Reland "Introduce RTC_NO_UNIQUE_ADDRESS." This is a reland of f5e261aaf65cdf2eb903cdf40d651846be44f447 This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since there have been some issues. Original change's description: > Introduce RTC_NO_UNIQUE_ADDRESS. > > This macro introduces the possibility to suggest the compiler that a > data member doesn't need an address different from other non static > data members. > > The usage of a macro is to maintain portability since at the moment > the attribute [[no_unique_address]] is only supported by clang > with at least -std=c++11 but it should be supported by all the > compilers starting from C++20. > > Bug: webrtc:11495 > Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32246} Bug: webrtc:11495, webrtc:12218 Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32668}
2020-11-23 11:07:42 +01:00
"../rtc_base/system:no_unique_address",
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
"../system_wrappers",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../system_wrappers:field_trial",
"../system_wrappers:metrics",
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
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("call") {
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
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",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"call_factory.cc",
"call_factory.h",
Reland: Add ability to emulate degraded network in Call via field trial This is especially useful in Chrome, allowing use to emulate network conditions in incoming or outgoing media without the need for platform specific tools or hacks. It also doesn't interfere with the rest of the network traffic. Also includes some refactorings. TBR=stefan@webrtc.org, philipel@webrtc.org Originally reviewed on: https://webrtc-review.googlesource.com/33013 Bug: webrtc:8910 Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0 Reviewed-on: https://webrtc-review.googlesource.com/61782 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22430}
2018-03-14 15:16:50 +01:00
"degraded_call.cc",
"degraded_call.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",
Added receive time calculator under field trial. The receive time calculator combines the packet time stamps received from the socket interface with the system clock used in WebRTC. This means that the packet timestamps are set in the WebRTC clock timebase and that large jumps in the time stamps from the socket will not affect the reported receive time stamps. Bug: None Change-Id: I293925c41919829524a115bb9377027bf0a797fb Reviewed-on: https://webrtc-review.googlesource.com/61862 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22540}
2018-03-21 14:38:32 +01:00
"receive_time_calculator.cc",
"receive_time_calculator.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
]
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",
Reland: Add ability to emulate degraded network in Call via field trial This is especially useful in Chrome, allowing use to emulate network conditions in incoming or outgoing media without the need for platform specific tools or hacks. It also doesn't interfere with the rest of the network traffic. Also includes some refactorings. TBR=stefan@webrtc.org, philipel@webrtc.org Originally reviewed on: https://webrtc-review.googlesource.com/33013 Bug: webrtc:8910 Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0 Reviewed-on: https://webrtc-review.googlesource.com/61782 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22430}
2018-03-14 15:16:50 +01:00
":fake_network",
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",
Switch webrtc users from deprecated ctors. Stop using of deprecated ctors of FakeNetworkPipe in most part of webrtc codebase, except DirectTransport, where further refactoring will be continued in future CLs. Bug: webrtc:9630 Change-Id: I823404469e461601ddbc026eaeac668eeda8045f Reviewed-on: https://webrtc-review.googlesource.com/94763 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24327}
2018-08-17 13:00:54 +02:00
":simulated_network",
Add WebRTC code freshness version string. This CL adds a string to the resulting WebRTC library (trying to make sure the version string will be there no matter how WebRTC is packaged). This CL should be followed by some process to regularly and automatically update the version string. No-Try: True No-Presubmit: True Bug: webrtc:12159 Change-Id: I9143aeae2cd54d0d4048c138772888100d7873cb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191223 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32825}
2020-12-14 15:28:43 +01:00
":version",
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",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api:array_view",
Rename Call::Config to CallConfig, keep old name as alias. We want api/peerconnectioninterface.h (and corresponding build target) to not depend on call.h, and generally we treat Call as an internal, non-api, class. But we need CallFactoryInterface in the api in order to enable use of PeerConnection with or without support for media. Making CallConfig a top-level class makes it possible to forward declare it, together with Call, for use in callfactoryinterface.h and peerconnectioninterface.h. Delete the peerconnection_and_implicit_call_api target, replaced by new target callfactory_api, to link between Call and Peerconnection. Bug: webrtc:7504 Change-Id: I5e3978ef89bcd6705e94536f8676bcf89fc82fe1 Reviewed-on: https://webrtc-review.googlesource.com/46201 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22020}
2018-02-14 12:20:13 +01:00
"../api:callfactory_api",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api:fec_controller_api",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
Reland "Move api/rtp_headers.h to its own build target." This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e Original change's description: > Move api/rtp_headers.h to its own build target. > > Reduces dependencies on the libjingle_peerconnection_api target from > lower-level code. > > Bug: None > Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27213} Bug: None Tbr: kwiberg@webrtc.org Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-20 08:40:23 +01:00
"../api:rtp_headers",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Switch webrtc users from deprecated ctors. Stop using of deprecated ctors of FakeNetworkPipe in most part of webrtc codebase, except DirectTransport, where further refactoring will be continued in future CLs. Bug: webrtc:9630 Change-Id: I823404469e461601ddbc026eaeac668eeda8045f Reviewed-on: https://webrtc-review.googlesource.com/94763 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24327}
2018-08-17 13:00:54 +02:00
"../api:simulated_network_api",
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. transport.h defines an interface for sending rtp and rtcp packets, which is used by MediaChannel in webrtc/media/engine, {Audio|Video}{Send|Receive}Stream and in a few other places. It was part of the build target //webrtc:webrtc, which is a monolithic target with all webrtc production code. This CL moves the header to its own target in webrtc/api and deprecates the old location. Targets in webrtc/api should in general only depend on other targets in webrtc/api. The target webrtc/api:call_api depends on transport.h. This change also makes webrtc/voice_engine pass GN's header include checker and is needed in order for webrtc/api:call_api to pass it. transport.h will be completely removed in a follow-up CL in a few weeks after clients have updated their includes. NOTRY=True BUG=webrtc:5589, webrtc:5878, webrtc:6785 Review-Url: https://codereview.webrtc.org/2426563003 Cr-Commit-Position: refs/heads/master@{#15267}
2016-11-28 07:02:13 -08:00
"../api:transport_api",
Delete deprecated rtc_event_log header Bug: webrtc:10206 Change-Id: I9ed3148843c647372993729b87c0e74741ab540b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 12:24:53 +02:00
"../api/rtc_event_log",
Moves network control interface to API. This prepares for allowing injection of a network controller. Bug: webrtc:9155 Change-Id: I5624f47738db9c5cd4750eac76cb6289e06a7aa3 Reviewed-on: https://webrtc-review.googlesource.com/73100 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23188}
2018-05-09 10:33:39 +02:00
"../api/transport:network_control",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api/units:time_delta",
"../api/video_codecs:video_codecs_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",
Reland "Break up rtc_event_log_api to solve circular dependencies." This is a reland of 001546da953275c7a39eb220592b440c9b47d756 Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@google.com> > Cr-Commit-Position: refs/heads/master@{#21811} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 11:04:46 -08:00
"../logging:rtc_event_audio",
"../logging:rtc_event_rtp_rtcp",
"../logging:rtc_event_video",
"../logging:rtc_stream_config",
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",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../modules/video_coding",
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",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Move location out of rtc_base_approved Bug: webrtc:9838 Change-Id: I3510f3d01df532cb228a6cc14be830f6ab81a8e1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258764 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36576}
2022-04-14 13:13:56 +02:00
"../rtc_base:location",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Moved retransmission rate limiter to Call class. Ownership of the retransmission rate limiter for video is moved from send side congestion controller to Call. This is to reduce the interface on the rtp transport controller send. Bug: webrtc:8415 Change-Id: Ie9c7317400a9eb61a3c8325b9e527844ffc13769 Reviewed-on: https://webrtc-review.googlesource.com/58745 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22254}
2018-03-01 15:56:57 +01:00
"../rtc_base:rate_limiter",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_task_queue",
Removing AvailableBandwidth method on transport controller. Removing the Synchronous call AvailableBandwidth from the RtpTransportControllerSend interface. The bandwidth estimate is provided trough a new interface that communicates with a struct making it easier to add parameters in the future. This prepares for removing locking behavior in SendSideCongestionController that exists just to support this feature. To keep backwards compatibility with the old SendSideCongestionController, the struct TargetTransferRate is constructed in RtpTransportControllerSend. This step can be removed in the future when the old SendSideCongestionController is deprecated. Bug: webrtc:8415 Change-Id: I06f64a89848157de412901c989650d1ecf35246b Reviewed-on: https://webrtc-review.googlesource.com/60800 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22387}
2018-03-12 15:59:12 +01:00
"../rtc_base:safe_minmax",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../rtc_base:timeutils",
New method for precise packet reception time measurement. Bug: webrtc:9054 Change-Id: I43a32122e9af992b5e0ba8b187c9ad4f22aba80d Reviewed-on: https://webrtc-review.googlesource.com/c/104503 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25167}
2018-10-12 11:15:09 +02:00
"../rtc_base/experiments:field_trial_parser",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../rtc_base/network:sent_packet",
Reland "Introduce RTC_NO_UNIQUE_ADDRESS." This is a reland of f5e261aaf65cdf2eb903cdf40d651846be44f447 This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since there have been some issues. Original change's description: > Introduce RTC_NO_UNIQUE_ADDRESS. > > This macro introduces the possibility to suggest the compiler that a > data member doesn't need an address different from other non static > data members. > > The usage of a macro is to maintain portability since at the moment > the attribute [[no_unique_address]] is only supported by clang > with at least -std=c++11 but it should be supported by all the > compilers starting from C++20. > > Bug: webrtc:11495 > Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32246} Bug: webrtc:11495, webrtc:12218 Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32668}
2020-11-23 11:07:42 +01:00
"../rtc_base/system:no_unique_address",
Remove send_crit_, bitrate_crit_ and last_bandwidth_bps_crit_ locks. ...from the Call class. With minimal shifting of ownership of a few variables, we can maintain their state fully on the worker thread, where they're used. This change also removes the dependency on RWLockWrapper. Bug: webrtc:11612 Change-Id: Ia14be5bd6b50bd0b32d04f078b1e283080c00a19 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176122 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31360}
2020-05-26 19:35:16 +02:00
"../rtc_base/task_utils:pending_task_safety_flag",
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",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../system_wrappers:field_trial",
"../system_wrappers:metrics",
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",
Integrate sync decoding in video_receive_stream Wires up DecodeSynchronizer in Call if there is a Metronome injected into the PeerConnectionFactoryDependencies. Change-Id: I362cd12648bfa0c32e73111fcd0f3296fca2b275 Bug: webrtc:13658 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251341 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35996}
2022-02-14 14:09:57 +01:00
"../video:decode_synchronizer",
[Adaptation] Multi-processor support for injected Resources. Because a single Resource only has a single ResourceListener, injected Resources only gets wired up to the stream's ResourceAdaptationProcessor that was last to call SetResourceListener. This could potentially lead to the relevant stream not adapting based on the injected resource because it got wired up to the wrong stream's processor. This CL fixes this issue by introducing BroadcastResourceListener. By listening to 1 resource (the injected one), it can spawn N "adapter" resources that mirror's the injected resource's usage signal, allowing all ResourceAdaptationProcessor's to react to the signal. This is wired up in Call, and tests are updated to verify the signal gets through. Bug: chromium:1101263, webrtc:11720 Change-Id: I8a37284cb9a68f08ca1bdb1ee050b7144c451297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178386 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31612}
2020-07-01 15:48:46 +02:00
"adaptation:resource_adaptation",
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
]
Remove REMB throttling funcionality from PacketRouter This removes PacketRouter inheritance from RemoteBitrateObserver and TransportFeedbackSenderInterface. Call binds methods for sending REMB and transport feedback messages from RemoteCongestionController to PacketRouter. This is needed until the RTCPTranseiver is used instead of the RTP modules. Bug: webrtc:12693 Change-Id: I7088de497cd6d1e15c98788ff3e6b0a2c8897ea8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215965 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33993}
2021-05-12 09:02:43 +02:00
absl_deps = [
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/types:optional",
]
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
Factor out common receive stream methods to a common interface. Also including common Rtp config members. Follow up changes will remove the ReceiveRtpConfig class in Call and copy of extension headers, instead use the config directly from the receive streams and not require stream recreation for changing the headers. Bug: webrtc:11993 Change-Id: I29ff3400d45d5bffddb3ad0a078403eb102afb65 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221983 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34283}
2021-06-14 10:54:20 +02:00
rtc_source_set("receive_stream_interface") {
sources = [ "receive_stream.h" ]
deps = [
"../api:frame_transformer_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api/crypto:frame_decryptor_interface",
"../api/transport/rtp:rtp_source",
Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions. GetRtpExtensions() is still used in one corner case for audio receive streams, so GetRtpExtensions has migrated to AudioReceiveStream. Updated FlexfecReceiveStream config management (incl. pass by value) and now store an RtpHeaderExtensionMap in FlexfecReceiveStreamImpl. Call GetRtpExtensionMap() from call.cc instead of constructing one on the fly for each rtp packet (for video packets at least). Bug: webrtc:11993 Change-Id: Id90ec5d43ea368f58edd6f17cb39d8c54aec641f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261800 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36839}
2022-05-09 20:46:57 +00:00
"../modules/rtp_rtcp:rtp_rtcp_format",
Factor out common receive stream methods to a common interface. Also including common Rtp config members. Follow up changes will remove the ReceiveRtpConfig class in Call and copy of extension headers, instead use the config directly from the receive streams and not require stream recreation for changing the headers. Bug: webrtc:11993 Change-Id: I29ff3400d45d5bffddb3ad0a078403eb102afb65 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221983 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34283}
2021-06-14 10:54:20 +02:00
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("video_stream_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
sources = [
"video_receive_stream.cc",
"video_receive_stream.h",
"video_send_stream.cc",
"video_send_stream.h",
]
deps = [
Factor out common receive stream methods to a common interface. Also including common Rtp config members. Follow up changes will remove the ReceiveRtpConfig class in Call and copy of extension headers, instead use the config directly from the receive streams and not require stream recreation for changing the headers. Bug: webrtc:11993 Change-Id: I29ff3400d45d5bffddb3ad0a078403eb102afb65 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221983 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34283}
2021-06-14 10:54:20 +02:00
":receive_stream_interface",
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",
Insert frame transformer between Encoded and Packetizer. Add a new API in RTPSenderInterface, to be called from the browser side to insert a frame transformer between the Encoded and the Packetizer. The frame transformer is passed from RTPSenderInterface through the library to be eventually set in RTPSenderVideo, where the frame transformation will occur in the follow-up CL https://webrtc-review.googlesource.com/c/src/+/169128. Insertable Streams Web API explainer: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md Design doc for WebRTC library changes: http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk Bug: webrtc:11380 Change-Id: I46cd0d8a798c2736c837e90cbf90d8901c7d27fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169127 Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30642}
2020-02-27 16:16:55 +01:00
"../api:frame_transformer_interface",
Reland "Move api/rtp_headers.h to its own build target." This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e Original change's description: > Move api/rtp_headers.h to its own build target. > > Reduces dependencies on the libjingle_peerconnection_api target from > lower-level code. > > Bug: None > Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27213} Bug: None Tbr: kwiberg@webrtc.org Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-20 08:40:23 +01:00
"../api:rtp_headers",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
[Adaptation] Adding adaptation resources from Call. This CL adds AddAdaptationResource to Call and AddAdaptationResource/GetAdaptationResources method to relevant VideoSendStream and VideoStreamEncoder interfaces and implementations. Unittests are added to ensure that resources can be added to the Call both before and after the creation of a VideoSendStream and that the resources always gets added to the streams. In a follow-up CL, we will continue to plumb the resources all the way to PeerConnectionInterface, and an integration test will then be added to ensure that injected resources are capable of triggering adaptation. Bug: webrtc:11525 Change-Id: I499e9c23c3e359df943414d420b2e0ce2e9b2d56 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177002 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31499}
2020-06-11 12:07:14 +02:00
"../api:scoped_refptr",
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",
[Adaptation] Adding adaptation resources from Call. This CL adds AddAdaptationResource to Call and AddAdaptationResource/GetAdaptationResources method to relevant VideoSendStream and VideoStreamEncoder interfaces and implementations. Unittests are added to ensure that resources can be added to the Call both before and after the creation of a VideoSendStream and that the resources always gets added to the streams. In a follow-up CL, we will continue to plumb the resources all the way to PeerConnectionInterface, and an integration test will then be added to ensure that injected resources are capable of triggering adaptation. Bug: webrtc:11525 Change-Id: I499e9c23c3e359df943414d420b2e0ce2e9b2d56 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177002 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31499}
2020-06-11 12:07:14 +02:00
"../api/adaptation:resource_adaptation_api",
Introduce api/crypto/BUILD.gn. No-Try: True Bug: webrtc:8733 Change-Id: I8679735be1e5069e371a9f1115a54e897e09964b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152622 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29186}
2019-09-13 14:42:15 +02:00
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
VideoReceiveStream: Enable encoded frame sink. This change ultimately enables wiring up VideoRtpReceiver::OnGenerateKeyFrame and OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded frames can flow to sinks installed in VideoTrackSourceInterface. Bug: chromium:1013590 Change-Id: I0779932c251a2159880a39b2d42d5ce439cc88e6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161090 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29988}
2019-12-03 14:31:45 +01:00
"../api/video:recordable_encoded_frame",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../api/video:video_frame",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../api/video:video_rtp_headers",
New api function CreateVideoStreamEncoder. Bug: webrtc:8830 Change-Id: I01de86f601e48f76e6b41b4182ce006d397a190c Reviewed-on: https://webrtc-review.googlesource.com/78260 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24079}
2018-07-24 09:29:58 +02:00
"../api/video:video_stream_encoder",
Pass VideoEncoderFactory from WebrtcVideoEngine to VideoStreamEncoder. In preparation for also moving the responsibility for encoder creation. Bug: webrtc:8830 Change-Id: Ic3d2039a86cd31c1b4157f5df4e97b607c81f1d7 Reviewed-on: https://webrtc-review.googlesource.com/55264 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22630}
2018-03-27 16:39:01 +02:00
"../api/video_codecs:video_codecs_api",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../common_video",
Move FrameCounts and FrameCountObserver to common_video/frame_counts.h Bug: webrtc:7660 Change-Id: Ic42227508e0e26b5443de0ef0e0722126cbbb8fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182501 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32005}
2020-08-25 10:28:50 +02:00
"../common_video:frame_counts",
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",
Change the type of RTCVideoSourceStats.framesPerSecond Spec: https://w3c.github.io/webrtc-stats/#dom-rtcvideosourcestats-framespersecond Bug: webrtc:12905 Change-Id: If53e2e480e2d6f687c3f8bb95a9e1d1e386fe9c2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237420 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#35352}
2021-11-10 11:23:56 +09:00
"../rtc_base:stringutils",
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
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types: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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("simulated_network") {
Move SimulatedNetwork class to separate file. Bug: webrtc:9467 Change-Id: Iaf91f27ea7ad9e9e59991bbeb0ef3868578e6a8f Reviewed-on: https://webrtc-review.googlesource.com/92884 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24221}
2018-08-07 18:58:05 +02:00
sources = [
"simulated_network.cc",
"simulated_network.h",
]
deps = [
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Move SimulatedNetwork class to separate file. Bug: webrtc:9467 Change-Id: Iaf91f27ea7ad9e9e59991bbeb0ef3868578e6a8f Reviewed-on: https://webrtc-review.googlesource.com/92884 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24221}
2018-08-07 18:58:05 +02:00
"../api:simulated_network_api",
Fixing bug in SimulatedNetwork where packets stop. Bug: webrtc:9952 Change-Id: I68491f1d18fee317165999453776a35cea41e71f Reviewed-on: https://webrtc-review.googlesource.com/c/109009 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25531}
2018-11-06 19:18:28 +01:00
"../api/units:data_rate",
"../api/units:data_size",
"../api/units:time_delta",
Adds CoDel implementation to network simulation. Adds an implementation of the CoDel active queue management algorithm to the network simulation. It is loosely based on CoDel pseudocode from ACMQueue: https://queue.acm.org/appendices/codel.html Bug: webrtc:9510 Change-Id: Ice485be35a01dafa6169d697b51b5c1b33a49ba6 Reviewed-on: https://webrtc-review.googlesource.com/c/123581 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26834}
2019-02-25 10:24:46 +01:00
"../api/units:timestamp",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../rtc_base:checks",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Move race_checker out of rtc_base_approved Bug: webrtc:9838 Change-Id: If180abcca1ef598314de3aed70e4a6eb04f062d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258770 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36607}
2022-04-14 13:38:05 +02:00
"../rtc_base:race_checker",
Move random out of rtc_base_approved Bug: webrtc:9838 Change-Id: I64a5ef18c19d446139354d04aa6cb2a76d18aad0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258762 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36572}
2022-04-14 12:41:26 +02:00
"../rtc_base:random",
Migrate call/ to webrtc::Mutex. Bug: webrtc:11567 Change-Id: Iab7142c77bc0c1a026cf5121b756094e05bccefe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176742 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31636}
2020-07-06 17:41:35 +02:00
"../rtc_base/synchronization:mutex",
Move SimulatedNetwork class to separate file. Bug: webrtc:9467 Change-Id: Iaf91f27ea7ad9e9e59991bbeb0ef3868578e6a8f Reviewed-on: https://webrtc-review.googlesource.com/92884 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24221}
2018-08-07 18:58:05 +02:00
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
Move SimulatedNetwork class to separate file. Bug: webrtc:9467 Change-Id: Iaf91f27ea7ad9e9e59991bbeb0ef3868578e6a8f Reviewed-on: https://webrtc-review.googlesource.com/92884 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24221}
2018-08-07 18:58:05 +02:00
}
Introduce SimulatedNetworkReceiverInterface. Introduce SimulatedNetworkReceiverInterface and switch DirectTransport on this interface. Also switch part of related users on DefaultNetworkSimulationConfig. This two changes united into single CL to prevent work duplication. Most changes were done because of stop including fake_network_pipe.h into direct_transport.h, so splitting this into 2 CLs will require first fix all imports of fake_network_pipe.h and then replace them on new API imports again. Bug: webrtc:9630 Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6 Reviewed-on: https://webrtc-review.googlesource.com/94762 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24336}
2018-08-17 14:26:54 +02:00
rtc_source_set("simulated_packet_receiver") {
Reformat GN files. `gn format` recently [1] changed its formatting behavior for deps, source, and a few other elements when they are assigned (with =) single-element lists to be consistent with the formatting of updates (with +=) with single-element. Now that we've rolled in a GN binary with the change, reformat all files so that people don't get presubmit warnings due to this. CL generated with: $ git ls-files | grep BUILD.gn | xargs gn format $ gn format build_overrides/build.gni $ gn format build_overrides/gtest.gni $ gn format modules/audio_coding/audio_coding.gni $ gn format webrtc.gni $ gn format .gn Plus a few manual changes to add exceptions for "public_deps" (after changing these lines the presubmit started to complain). [1] - https://gn-review.googlesource.com/c/gn/+/6860 Bug: webrtc:11302 Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:10:10 +01:00
sources = [ "simulated_packet_receiver.h" ]
Introduce SimulatedNetworkReceiverInterface. Introduce SimulatedNetworkReceiverInterface and switch DirectTransport on this interface. Also switch part of related users on DefaultNetworkSimulationConfig. This two changes united into single CL to prevent work duplication. Most changes were done because of stop including fake_network_pipe.h into direct_transport.h, so splitting this into 2 CLs will require first fix all imports of fake_network_pipe.h and then replace them on new API imports again. Bug: webrtc:9630 Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6 Reviewed-on: https://webrtc-review.googlesource.com/94762 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24336}
2018-08-17 14:26:54 +02:00
deps = [
":call_interfaces",
"../api:simulated_network_api",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("fake_network") {
Reland: Add ability to emulate degraded network in Call via field trial This is especially useful in Chrome, allowing use to emulate network conditions in incoming or outgoing media without the need for platform specific tools or hacks. It also doesn't interfere with the rest of the network traffic. Also includes some refactorings. TBR=stefan@webrtc.org, philipel@webrtc.org Originally reviewed on: https://webrtc-review.googlesource.com/33013 Bug: webrtc:8910 Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0 Reviewed-on: https://webrtc-review.googlesource.com/61782 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22430}
2018-03-14 15:16:50 +01:00
sources = [
"fake_network_pipe.cc",
"fake_network_pipe.h",
]
deps = [
":call_interfaces",
Move SimulatedNetwork class to separate file. Bug: webrtc:9467 Change-Id: Iaf91f27ea7ad9e9e59991bbeb0ef3868578e6a8f Reviewed-on: https://webrtc-review.googlesource.com/92884 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24221}
2018-08-07 18:58:05 +02:00
":simulated_network",
Introduce SimulatedNetworkReceiverInterface. Introduce SimulatedNetworkReceiverInterface and switch DirectTransport on this interface. Also switch part of related users on DefaultNetworkSimulationConfig. This two changes united into single CL to prevent work duplication. Most changes were done because of stop including fake_network_pipe.h into direct_transport.h, so splitting this into 2 CLs will require first fix all imports of fake_network_pipe.h and then replace them on new API imports again. Bug: webrtc:9630 Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6 Reviewed-on: https://webrtc-review.googlesource.com/94762 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24336}
2018-08-17 14:26:54 +02:00
":simulated_packet_receiver",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Convert video quality test from a TEST_F to a TEST fixture. The purpose is to make the fixture reusable in downstream projects. The CL adds the following things to API: - api/test/video_quality_test_fixture.h - api/test/create_video_quality_test_fixture.h The following things are moved to API: - call/bitrate_constraints.h (api/bitrate_constraints.h) - call/simulated_network.h (api/test/simulated_network.h) - call/media_type.h (api/mediatypes.h) These are required by the params struct passed to the fixture. I didn't attempt to split the params struct into an internal-only and public version in this CL, and as a result we need to pull in the above things. They are quite harmless though, so I think it's worth it in order to avoid splitting up the test config struct. This CL doesn't solve all the problems we need to implement downstream tests; we probably need to upstream tracing variants of FakeNetworkPipe for instance, but that will come later. This puts in place the basic structure for now. Bug: None Change-Id: I35e26ed126fad27bc7b2a465400291084f6ac911 Reviewed-on: https://webrtc-review.googlesource.com/69601 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23714}
2018-06-21 16:58:01 +02:00
"../api:simulated_network_api",
Reland: Add ability to emulate degraded network in Call via field trial This is especially useful in Chrome, allowing use to emulate network conditions in incoming or outgoing media without the need for platform specific tools or hacks. It also doesn't interfere with the rest of the network traffic. Also includes some refactorings. TBR=stefan@webrtc.org, philipel@webrtc.org Originally reviewed on: https://webrtc-review.googlesource.com/33013 Bug: webrtc:8910 Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0 Reviewed-on: https://webrtc-review.googlesource.com/61782 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22430}
2018-03-14 15:16:50 +01:00
"../api:transport_api",
Allow FakeNetworkPipe to wake up its processing thread Bug: webrtc:9630 Change-Id: I2b09593f175e3f3e1fe0d990515aa70c2481161b Reviewed-on: https://webrtc-review.googlesource.com/c/95144 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25451}
2018-10-30 08:23:27 +01:00
"../modules/utility",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Migrate call/ to webrtc::Mutex. Bug: webrtc:11567 Change-Id: Iab7142c77bc0c1a026cf5121b756094e05bccefe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176742 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31636}
2020-07-06 17:41:35 +02:00
"../rtc_base/synchronization:mutex",
Reland: Add ability to emulate degraded network in Call via field trial This is especially useful in Chrome, allowing use to emulate network conditions in incoming or outgoing media without the need for platform specific tools or hacks. It also doesn't interfere with the rest of the network traffic. Also includes some refactorings. TBR=stefan@webrtc.org, philipel@webrtc.org Originally reviewed on: https://webrtc-review.googlesource.com/33013 Bug: webrtc:8910 Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0 Reviewed-on: https://webrtc-review.googlesource.com/61782 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22430}
2018-03-14 15:16:50 +01:00
"../system_wrappers",
]
}
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) {
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
if (!build_with_chromium) {
rtc_library("call_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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [
"bitrate_allocator_unittest.cc",
"bitrate_estimator_tests.cc",
"call_unittest.cc",
"flexfec_receive_stream_unittest.cc",
"receive_time_calculator_unittest.cc",
"rtp_bitrate_configurator_unittest.cc",
"rtp_demuxer_unittest.cc",
"rtp_payload_params_unittest.cc",
"rtp_video_sender_unittest.cc",
"rtx_receive_stream_unittest.cc",
]
deps = [
":bitrate_allocator",
":bitrate_configurator",
":call",
":call_interfaces",
":mock_rtp_interfaces",
":rtp_interfaces",
":rtp_receiver",
":rtp_sender",
":simulated_network",
"../api:array_view",
"../api:create_frame_generator",
"../api:mock_audio_mixer",
"../api:rtp_headers",
"../api:rtp_parameters",
"../api:transport_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/rtc_event_log",
"../api/task_queue:default_task_queue_factory",
"../api/test/video:function_video_factory",
"../api/transport:field_trial_based_config",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
"../audio",
"../modules:module_api",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/audio_processing:mocks",
"../modules/congestion_controller",
"../modules/pacing",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:mock_rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/utility:mock_process_thread",
"../modules/video_coding",
"../modules/video_coding:codec_globals_headers",
"../modules/video_coding:video_codec_interface",
"../rtc_base:checks",
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Move random out of rtc_base_approved Bug: webrtc:9838 Change-Id: I64a5ef18c19d446139354d04aa6cb2a76d18aad0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258762 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36572}
2022-04-14 12:41:26 +02:00
"../rtc_base:random",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../rtc_base:rate_limiter",
Remove //rtc_base:rtc_event from public deps Bug: webrtc:8603 Change-Id: Ib99f43043da17723c939b0fe2aa9f3e515462c93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257911 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36450}
2022-04-04 16:57:52 +02:00
"../rtc_base:rtc_event",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../rtc_base:safe_conversions",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../rtc_base:task_queue_for_test",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../rtc_base:timeutils",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../rtc_base/synchronization:mutex",
"../system_wrappers",
"../test:audio_codec_mocks",
"../test:direct_transport",
"../test:encoder_settings",
Calculate VP9 generic info from vp9 specific info Encoder wrapper can provide more accurate vp9 generic info, but each vp9 encoder wrapper would need to fill this structure. Inserting this code into the call allows to have some generic info for all vp9 encoder wrappers. Bug: webrtc:11999 Change-Id: I82490d24454815aa29bbb1c86f351e0b37292d59 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214491 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33836}
2021-04-22 15:20:28 +02:00
"../test:explicit_key_value_config",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../test:fake_video_codecs",
"../test:field_trial",
"../test:mock_frame_transformer",
"../test:mock_transport",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2 convert call/ (and the collaterals) Bug: webrtc:10335 Change-Id: I8f6bc13c032713aa2a947724b464f6f35454d39a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36165}
2022-03-09 15:12:07 +01:00
"../test:scoped_key_value_config",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../test:test_common",
"../test:test_support",
"../test:video_test_common",
"../test/scenario",
"../test/time_controller:time_controller",
"../video",
"adaptation:resource_adaptation_test_utilities",
"//testing/gmock",
"//testing/gtest",
]
absl_deps = [
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/memory",
Adopt absl::string_view in rtc_base/log_sinks This is part of a large scale effort to increase adoption of absl::string_view across the WebRTC code base. This CL adds absl::string_view versions of the OnLogMessage functions in rtc::LogSink. The const std::string& versions are kept for now since downstream clients use subclasses of LogSink and need to be migrated before these are removed. Bug: webrtc:13579 Change-Id: I57bb72ad503805ff0ca16f1d7aece2d44c65cb73 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253980 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Owners-Override: Xavier Lepaul‎ <xalep@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36189}
2022-03-14 13:32:04 +01:00
"//third_party/abseil-cpp/absl/strings",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
}
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_library("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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [
"call_perf_tests.cc",
"rampup_tests.cc",
"rampup_tests.h",
]
deps = [
":call_interfaces",
":simulated_network",
":video_stream_api",
"../api:rtc_event_log_output_file",
"../api:simulated_network_api",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/rtc_event_log",
"../api/rtc_event_log:rtc_event_log_factory",
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_bitrate_allocation",
"../api/video_codecs:video_codecs_api",
SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated. Results from test (CallPerfTest.TestEncodeFramerateVp8Simulcast): Simulcast streams: 0: max_fps:20 -> StreamStats.encode_frame_rate:15 (before), 20 (after) 1: max_fps:30 Bug: webrtc:13031 Change-Id: I30e6b2dcb2746859bd3e21b098bfa7b0fb3b2dda Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230120 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34867}
2021-08-26 12:04:27 +02:00
"../media:rtc_internal_video_codecs",
"../media:rtc_simulcast_encoder_adapter",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../modules/audio_coding",
"../modules/audio_device",
"../modules/audio_device:audio_device_impl",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base",
"../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Remove platform_thread from //rtc_base:rtc_base_approved public_deps While the target has a restricted visibility, since it was in rtc_base_approved public deps, a lot of targets were able to bypass the visibility check. So we remove the visibility restrictions and use the dependency explicitely everywhere instead. Bug: webrtc:8603 Change-Id: I94a03fdf7f94c54ab72081a58dd648e2cca73d17 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258944 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36566}
2022-04-14 12:18:28 +02:00
"../rtc_base:platform_thread",
Remove //rtc_base:rtc_event from public deps Bug: webrtc:8603 Change-Id: Ib99f43043da17723c939b0fe2aa9f3e515462c93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257911 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36450}
2022-04-04 16:57:52 +02:00
"../rtc_base:rtc_event",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../rtc_base:task_queue_for_test",
"../rtc_base:threading",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../rtc_base:timeutils",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../rtc_base/synchronization:mutex",
Fix race in CallPerfTest.Bitrate_Kbps_PadsToMinTransmitBitrate Task posted by OnSendRtp might be scheduled after `send_stream_` is destroyed. Fix by using a PendingTaskSafetyFlag, killed from the OnStreamsStopped callback. Bug: webrtc:12726 Change-Id: I935917a3d80e82c3536261d72059448fb7aac00d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228643 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34754}
2021-08-13 14:00:44 +02:00
"../rtc_base/task_utils:pending_task_safety_flag",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../rtc_base/task_utils:repeating_task",
"../system_wrappers",
"../system_wrappers:metrics",
"../test:direct_transport",
"../test:encoder_settings",
"../test:fake_video_codecs",
"../test:field_trial",
"../test:fileutils",
"../test:null_transport",
"../test:perf_test",
"../test:test_common",
"../test:test_support",
"../test:video_test_common",
"../video",
"//testing/gtest",
]
absl_deps = [ "//third_party/abseil-cpp/absl/flags:flag" ]
}
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
}
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
Use backticks not vertical bars to denote variables in comments for /call Bug: webrtc:12338 Change-Id: I8f92127b61352bd4b98a0690e9a0435bb6c6f870 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226943 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34569}
2021-07-26 12:40:21 +02:00
# TODO(eladalon): This should be moved, as with the TODO for `rtp_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
rtc_source_set("mock_rtp_interfaces") {
testonly = true
sources = [
"test/mock_rtp_packet_sink_interface.h",
Moved bitrate config interface from Call class. Moving usage of bitrate configuration related interface from Call interface to the corresponding methods in the RtpSendTransportController interface. SetBitrateConfig was replaced with SetSdpBitrateParameters SetBitrateConfigMask was replaced with SetClientBitratePreferences OnNetworkRouteChanged was replaced with OnNetworkRouteChanged This makes it more clear that RtpSendTransportController owns bitrate configuration and fits a longer term ambition to reduce the scope of the Call class. Bug: webrtc:8415 Change-Id: I6d04eaad22a54ecd5ed60096e01689b0c67e9c65 Reviewed-on: https://webrtc-review.googlesource.com/54365 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22131}
2018-02-21 13:07:13 +01:00
"test/mock_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
]
deps = [
":rtp_interfaces",
Insert frame transformer between Encoded and Packetizer. Add a new API in RTPSenderInterface, to be called from the browser side to insert a frame transformer between the Encoded and the Packetizer. The frame transformer is passed from RTPSenderInterface through the library to be eventually set in RTPSenderVideo, where the frame transformation will occur in the follow-up CL https://webrtc-review.googlesource.com/c/src/+/169128. Insertable Streams Web API explainer: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md Design doc for WebRTC library changes: http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk Bug: webrtc:11380 Change-Id: I46cd0d8a798c2736c837e90cbf90d8901c7d27fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169127 Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30642}
2020-02-27 16:16:55 +01:00
"../api:frame_transformer_interface",
Convert video quality test from a TEST_F to a TEST fixture. The purpose is to make the fixture reusable in downstream projects. The CL adds the following things to API: - api/test/video_quality_test_fixture.h - api/test/create_video_quality_test_fixture.h The following things are moved to API: - call/bitrate_constraints.h (api/bitrate_constraints.h) - call/simulated_network.h (api/test/simulated_network.h) - call/media_type.h (api/mediatypes.h) These are required by the params struct passed to the fixture. I didn't attempt to split the params struct into an internal-only and public version in this CL, and as a result we need to pull in the above things. They are quite harmless though, so I think it's worth it in order to avoid splitting up the test config struct. This CL doesn't solve all the problems we need to implement downstream tests; we probably need to upstream tracing variants of FakeNetworkPipe for instance, but that will come later. This puts in place the basic structure for now. Bug: None Change-Id: I35e26ed126fad27bc7b2a465400291084f6ac911 Reviewed-on: https://webrtc-review.googlesource.com/69601 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23714}
2018-06-21 16:58:01 +02:00
"../api:libjingle_peerconnection_api",
Introduce api/crypto/BUILD.gn. No-Try: True Bug: webrtc:8733 Change-Id: I8679735be1e5069e371a9f1115a54e897e09964b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152622 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29186}
2019-09-13 14:42:15 +02:00
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
Remove api/bitrate_constraints.h. Bug: webrtc:8733 Change-Id: Iaeb26e07d399f25dc18b0c4af38ed400577a5d3a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153220 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29217}
2019-09-17 14:47:38 +02:00
"../api/transport:bitrate_settings",
Added dependencies to mock transport controller send. Added dependencies used by MockRtpTransportControllerSend to its header file. The mocked interface can't be used properly without those. This prepares for later CLs utilizing the mock. Bug: None Change-Id: I5f8ca04032ad09810240f7c034cc628d700dbedb Reviewed-on: https://webrtc-review.googlesource.com/56181 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22154}
2018-02-21 17:30:36 +01:00
"../modules/pacing",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../rtc_base",
Added dependencies to mock transport controller send. Added dependencies used by MockRtpTransportControllerSend to its header file. The mocked interface can't be used properly without those. This prepares for later CLs utilizing the mock. Bug: None Change-Id: I5f8ca04032ad09810240f7c034cc628d700dbedb Reviewed-on: https://webrtc-review.googlesource.com/56181 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22154}
2018-02-21 17:30:36 +01:00
"../rtc_base:rate_limiter",
Removing unnecessary dependencies on socket.h. Since rtc:SentPacket was removed to a separate header. Some usages of socket.h can be replaced with sent_packet.h which defines a lot less things, making future maintenance simpler. Bug: webrtc:9586 Change-Id: If705edda293c389cf2a175117db52a6720a7be86 Reviewed-on: https://webrtc-review.googlesource.com/c/106144 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25201}
2018-10-15 21:06:26 +02:00
"../rtc_base/network:sent_packet",
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",
]
}
Adds mock bitrate allocator. This is used to allow mocking in tests in later CLs. Bug: None Change-Id: Id610471efb4a86c903530585dd4ee2fa1d1ea5bc Reviewed-on: https://webrtc-review.googlesource.com/70880 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22941}
2018-04-19 08:27:19 +02:00
rtc_source_set("mock_bitrate_allocator") {
testonly = true
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
Reformat GN files. `gn format` recently [1] changed its formatting behavior for deps, source, and a few other elements when they are assigned (with =) single-element lists to be consistent with the formatting of updates (with +=) with single-element. Now that we've rolled in a GN binary with the change, reformat all files so that people don't get presubmit warnings due to this. CL generated with: $ git ls-files | grep BUILD.gn | xargs gn format $ gn format build_overrides/build.gni $ gn format build_overrides/gtest.gni $ gn format modules/audio_coding/audio_coding.gni $ gn format webrtc.gni $ gn format .gn Plus a few manual changes to add exceptions for "public_deps" (after changing these lines the presubmit started to complain). [1] - https://gn-review.googlesource.com/c/gn/+/6860 Bug: webrtc:11302 Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:10:10 +01:00
sources = [ "test/mock_bitrate_allocator.h" ]
Adds mock bitrate allocator. This is used to allow mocking in tests in later CLs. Bug: None Change-Id: Id610471efb4a86c903530585dd4ee2fa1d1ea5bc Reviewed-on: https://webrtc-review.googlesource.com/70880 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22941}
2018-04-19 08:27:19 +02:00
deps = [
":bitrate_allocator",
Harmonize paths to dependent targets. This CL consistently use: * relative paths for WebRTC dependent targets (test_support) * absolute paths for shared dependent targets (abseil) This is a necessary (but insufficient) step to build WebRTC tests from Chromium tree (rtc_include_tests=true), since test/ doesn't sit anymore in the top level directory. We also make sure that target declarations and uses are consistent in regard to build_with_chromium flag. Bug: webrtc:9943 Bug: webrtc:9855 Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b Reviewed-on: https://webrtc-review.googlesource.com/c/108720 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Yves Gerey <yvesg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25445}
2018-10-30 21:12:42 +01:00
"../test:test_support",
Adds mock bitrate allocator. This is used to allow mocking in tests in later CLs. Bug: None Change-Id: Id610471efb4a86c903530585dd4ee2fa1d1ea5bc Reviewed-on: https://webrtc-review.googlesource.com/70880 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22941}
2018-04-19 08:27:19 +02:00
]
}
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
Reformat GN files. `gn format` recently [1] changed its formatting behavior for deps, source, and a few other elements when they are assigned (with =) single-element lists to be consistent with the formatting of updates (with +=) with single-element. Now that we've rolled in a GN binary with the change, reformat all files so that people don't get presubmit warnings due to this. CL generated with: $ git ls-files | grep BUILD.gn | xargs gn format $ gn format build_overrides/build.gni $ gn format build_overrides/gtest.gni $ gn format modules/audio_coding/audio_coding.gni $ gn format webrtc.gni $ gn format .gn Plus a few manual changes to add exceptions for "public_deps" (after changing these lines the presubmit started to complain). [1] - https://gn-review.googlesource.com/c/gn/+/6860 Bug: webrtc:11302 Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:10:10 +01:00
sources = [ "test/mock_audio_send_stream.h" ]
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
deps = [
":call_interfaces",
Harmonize paths to dependent targets. This CL consistently use: * relative paths for WebRTC dependent targets (test_support) * absolute paths for shared dependent targets (abseil) This is a necessary (but insufficient) step to build WebRTC tests from Chromium tree (rtc_include_tests=true), since test/ doesn't sit anymore in the top level directory. We also make sure that target declarations and uses are consistent in regard to build_with_chromium flag. Bug: webrtc:9943 Bug: webrtc:9855 Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b Reviewed-on: https://webrtc-review.googlesource.com/c/108720 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Yves Gerey <yvesg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25445}
2018-10-30 21:12:42 +01:00
"../test:test_support",
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
]
}
Reland: Add ability to emulate degraded network in Call via field trial This is especially useful in Chrome, allowing use to emulate network conditions in incoming or outgoing media without the need for platform specific tools or hacks. It also doesn't interfere with the rest of the network traffic. Also includes some refactorings. TBR=stefan@webrtc.org, philipel@webrtc.org Originally reviewed on: https://webrtc-review.googlesource.com/33013 Bug: webrtc:8910 Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0 Reviewed-on: https://webrtc-review.googlesource.com/61782 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22430}
2018-03-14 15:16:50 +01:00
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("fake_network_pipe_unittests") {
Adds FakeNetworkPipeTest to rtc_unittests. Also updating the tests so they pass. The previous target was not part of the targets that runs on the try bots, so they started breaking during a recent refactor. By adding it to the common target we'll avoid regression like this in the future. Bug: None Change-Id: I55391d454bd52964875818be39cdaa853360ae81 Reviewed-on: https://webrtc-review.googlesource.com/c/123199 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26820}
2019-02-21 16:21:51 +01:00
testonly = true
Extract functionality of test_main into separate library. Extract functionality of test_main into separate library to be able to reuse it if another main will be required. Bug: webrtc:5996 Change-Id: I2925b4240bd0e4fb884b43bb16667ca2d6216bbd Reviewed-on: https://webrtc-review.googlesource.com/c/105921 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25172}
2018-10-15 15:25:34 +02:00
sources = [
Adds FakeNetworkPipeTest to rtc_unittests. Also updating the tests so they pass. The previous target was not part of the targets that runs on the try bots, so they started breaking during a recent refactor. By adding it to the common target we'll avoid regression like this in the future. Bug: None Change-Id: I55391d454bd52964875818be39cdaa853360ae81 Reviewed-on: https://webrtc-review.googlesource.com/c/123199 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26820}
2019-02-21 16:21:51 +01:00
"fake_network_pipe_unittest.cc",
Adds CoDel implementation to network simulation. Adds an implementation of the CoDel active queue management algorithm to the network simulation. It is loosely based on CoDel pseudocode from ACMQueue: https://queue.acm.org/appendices/codel.html Bug: webrtc:9510 Change-Id: Ice485be35a01dafa6169d697b51b5c1b33a49ba6 Reviewed-on: https://webrtc-review.googlesource.com/c/123581 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26834}
2019-02-25 10:24:46 +01:00
"simulated_network_unittest.cc",
Extract functionality of test_main into separate library. Extract functionality of test_main into separate library to be able to reuse it if another main will be required. Bug: webrtc:5996 Change-Id: I2925b4240bd0e4fb884b43bb16667ca2d6216bbd Reviewed-on: https://webrtc-review.googlesource.com/c/105921 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25172}
2018-10-15 15:25:34 +02:00
]
Reland: Add ability to emulate degraded network in Call via field trial This is especially useful in Chrome, allowing use to emulate network conditions in incoming or outgoing media without the need for platform specific tools or hacks. It also doesn't interfere with the rest of the network traffic. Also includes some refactorings. TBR=stefan@webrtc.org, philipel@webrtc.org Originally reviewed on: https://webrtc-review.googlesource.com/33013 Bug: webrtc:8910 Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0 Reviewed-on: https://webrtc-review.googlesource.com/61782 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22430}
2018-03-14 15:16:50 +01:00
deps = [
":fake_network",
Do not use FakeNetworkPipe::SetConfig. Switch from using FakeNetworkPipe::SetConfig on holding direct reference on instances of NetworkSimulationInterface if you are required to reconfigure it in the runtime. Also add fake_network_unittests to built test targets. Bug: webrtc:9630 Change-Id: I5fd6b33904367aa6dc00ca2e2f5f780f433acf35 Reviewed-on: https://webrtc-review.googlesource.com/94510 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24314}
2018-08-16 15:51:07 +02:00
":simulated_network",
Adds CoDel implementation to network simulation. Adds an implementation of the CoDel active queue management algorithm to the network simulation. It is loosely based on CoDel pseudocode from ACMQueue: https://queue.acm.org/appendices/codel.html Bug: webrtc:9510 Change-Id: Ice485be35a01dafa6169d697b51b5c1b33a49ba6 Reviewed-on: https://webrtc-review.googlesource.com/c/123581 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26834}
2019-02-25 10:24:46 +01:00
"../api/units:data_rate",
Reland: Add ability to emulate degraded network in Call via field trial This is especially useful in Chrome, allowing use to emulate network conditions in incoming or outgoing media without the need for platform specific tools or hacks. It also doesn't interfere with the rest of the network traffic. Also includes some refactorings. TBR=stefan@webrtc.org, philipel@webrtc.org Originally reviewed on: https://webrtc-review.googlesource.com/33013 Bug: webrtc:8910 Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0 Reviewed-on: https://webrtc-review.googlesource.com/61782 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22430}
2018-03-14 15:16:50 +01:00
"../system_wrappers",
Harmonize paths to dependent targets. This CL consistently use: * relative paths for WebRTC dependent targets (test_support) * absolute paths for shared dependent targets (abseil) This is a necessary (but insufficient) step to build WebRTC tests from Chromium tree (rtc_include_tests=true), since test/ doesn't sit anymore in the top level directory. We also make sure that target declarations and uses are consistent in regard to build_with_chromium flag. Bug: webrtc:9943 Bug: webrtc:9855 Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b Reviewed-on: https://webrtc-review.googlesource.com/c/108720 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Yves Gerey <yvesg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25445}
2018-10-30 21:12:42 +01:00
"../test:test_support",
Reland: Add ability to emulate degraded network in Call via field trial This is especially useful in Chrome, allowing use to emulate network conditions in incoming or outgoing media without the need for platform specific tools or hacks. It also doesn't interfere with the rest of the network traffic. Also includes some refactorings. TBR=stefan@webrtc.org, philipel@webrtc.org Originally reviewed on: https://webrtc-review.googlesource.com/33013 Bug: webrtc:8910 Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0 Reviewed-on: https://webrtc-review.googlesource.com/61782 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22430}
2018-03-14 15:16:50 +01:00
"//testing/gtest",
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
Reland: Add ability to emulate degraded network in Call via field trial This is especially useful in Chrome, allowing use to emulate network conditions in incoming or outgoing media without the need for platform specific tools or hacks. It also doesn't interfere with the rest of the network traffic. Also includes some refactorings. TBR=stefan@webrtc.org, philipel@webrtc.org Originally reviewed on: https://webrtc-review.googlesource.com/33013 Bug: webrtc:8910 Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0 Reviewed-on: https://webrtc-review.googlesource.com/61782 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22430}
2018-03-14 15:16:50 +01:00
}
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
}
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