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webrtc_m130/modules/pacing/BUILD.gn

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GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../../webrtc.gni")
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
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("pacing") {
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 = [ "*" ]
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
sources = [
Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate. Also wires up a finch experiment to control this. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7505 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 11:57:05 +00:00
"bitrate_prober.cc",
"bitrate_prober.h",
Extract most of PacedSender into PacedSendingController. The Pacer now just handles interaction with Module/ProcessThread and forwarding packets to PacketRouter. All other logic is moved to PacedSendingController, including tests. PacedSender unittest are now just some basic sanity tests. Bug: webrtc:10809 Change-Id: I69223cd9d8300997375b03706d2e99c88e46241c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149041 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28886}
2019-08-14 10:43:47 +02:00
"pacing_controller.cc",
"pacing_controller.h",
Moving the pacer and the pacer thread to ChannelGroup. This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out. BUG=4323 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45549004 Cr-Commit-Position: refs/heads/master@{#8864}
2015-03-26 11:11:06 +01:00
"packet_router.cc",
Remove include dirs from modules/{media_file,pacing} Also move files out of media_file/source. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=asapersson@webrtc.org, perkj@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1435093002 . Cr-Commit-Position: refs/heads/master@{#10647}
2015-11-16 11:12:24 +01:00
"packet_router.h",
Add new prioritized packet queue. This queue is a more strict round robing queue, unlike the class named RoundRobinPacketQueue. That is, we don't have the same logic to prioritize lower-bitrate streams. The queue time mechanism is essentially directly copied from the previous implementation however. Bug: webrtc:11340 Change-Id: Ie38ba8ce27c985f5f1e907cec068d6a365089bcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260562 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36737}
2022-05-02 21:18:08 +02:00
"prioritized_packet_queue.cc",
"prioritized_packet_queue.h",
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
"rtp_packet_pacer.h",
Implements a task-queue based PacedSender, wires it up for field trials Bug: webrtc:10809 Change-Id: Ia181c16559f4598f32dd399c24802d0a289e250b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150942 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29946}
2019-11-26 17:48:49 +01:00
"task_queue_paced_sender.cc",
"task_queue_paced_sender.h",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
]
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
Removes redundant AlrDetector. This replaces the old AlrDetector used by the pacer with the one in GoogCC. This reduces the risk of accidentally changing only one version. Note that the pacer instance will be removed when moving over to the task queue based send side congestion controller. Bug: webrtc:8415 Change-Id: Id4b2000ee5a04b94565092c29a84572a7750d2f5 Reviewed-on: https://webrtc-review.googlesource.com/85363 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23791}
2018-06-25 17:56:08 +02:00
":interval_budget",
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",
"../../api:field_trials_view",
Wire up new PacedSender code path. This CL makes the new code path for paced sending functionally complete. By default, the field trial WebRTC-Pacer-ReferencePackets is Enabled, meaning that there is no behavior change unless the field trial is forced to Disabled. This is done in tests, and can be done on the command line for manual testing. Bug: webrtc:10633 Change-Id: I0d66c94ef83b5847dee437a785018f09ba3f828d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144050 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28497}
2019-07-05 16:53:43 +02:00
"../../api:function_view",
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",
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",
Implements a task-queue based PacedSender, wires it up for field trials Bug: webrtc:10809 Change-Id: Ia181c16559f4598f32dd399c24802d0a289e250b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150942 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29946}
2019-11-26 17:48:49 +01:00
"../../api/task_queue:task_queue",
Replace field trials with WebRtcKeyValueConfig in PacedSender Replaces use of field trials in PacedSender with injectable WebRtcKeyValueConfig. Implementation still defaults to field trials. BUG: webrtc:10335 Change-Id: Ie8870d93d51e996e762f2c2de7545bad261b6bb7 Reviewed-on: https://webrtc-review.googlesource.com/c/123521 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26818}
2019-02-22 13:09:32 +01:00
"../../api/transport:field_trial_based_config",
Cleanup modules_common_types Uninline RTPFragmentaion functions fix RTPFragmentation move constructor and assign operators (was recursive for win) replace assert with rtc::dchecked_cast Remove unused includes and dependencies. Fix other targets that used those includes transitively instead of directly Bug: None Change-Id: I647cb1eda107dc7d87d25234095545bc2842fa40 Reviewed-on: https://webrtc-review.googlesource.com/100500 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24759}
2018-09-17 13:11:50 +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:data_rate",
"../../api/units:data_size",
"../../api/units:time_delta",
"../../api/units:timestamp",
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_bwe",
"../../logging:rtc_event_pacing",
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 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",
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",
Add ability to control TaskQueuePacedSender holdback window. Holdback window can be specified as absolute time and in terms of packet send times. Example: WebRTC-TaskQueuePacer/Enabled,holdback_window:20ms,holdback_packet:3/ If current conditions have us running with 2000kbps pacing rate and 1250byte (10kbit) packets, each packet send time is 5ms. The holdback window would then be min(20ms, 3*5ms) = 15ms. The default is like before 1ms and packets no take into account when TQ pacer is used, parameters have no effect with legacy process thread pacer. Bug: webrtc:10809 Change-Id: I800de05107e2d4df461eabaaf1ca04fb4c5de51e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233421 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35266}
2021-10-26 16:19:03 +02:00
"../../rtc_base:rtc_numerics",
Implements a task-queue based PacedSender, wires it up for field trials Bug: webrtc:10809 Change-Id: Ia181c16559f4598f32dd399c24802d0a289e250b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150942 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29946}
2019-11-26 17:48:49 +01:00
"../../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",
Make pacing buffer send interval configurable. Bug: webrtc:10153 Change-Id: I2f71b5fc902e24b87f0b6cba474d988b8c27aefa Reviewed-on: https://webrtc-review.googlesource.com/c/115301 Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26076}
2018-12-20 15:46:03 +01:00
"../../rtc_base/experiments:field_trial_parser",
Migrate modules/pacing to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I4edded4bdad7c3d0be4c7cfa0d34219d942a467d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176856 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31494}
2020-06-10 15:12:15 +02:00
"../../rtc_base/synchronization:mutex",
Eliminate unnecessary `RTC_TRACE_EVENTS_ENABLED` Bug: webrtc:14073 Change-Id: I6365cc17393be52c11312dfa954783a3e135cb8c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262263 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36929}
2022-05-13 10:40:25 +08:00
"../../rtc_base/system:unused",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"../../system_wrappers",
Add probing histograms. This CL adds a handful of histograms that track how many probes we send, and some details about them. We'll track the total number of requested probe clusters and how many of them timed out, recorded when we destroy the prober. For the successful clusters we'll also track how many bytes it took, how many packets we sent and how long it took to send. Bug: webrtc:10413 Change-Id: I4a223caa6d3c2829a36788d4fa615a41286b183f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127884 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27270}
2019-03-25 10:10:31 +01:00
"../../system_wrappers:metrics",
Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it. BUG= Review URL: https://codereview.webrtc.org/1350163005 Cr-Commit-Position: refs/heads/master@{#10005}
2015-09-21 15:11:14 -07:00
"../rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
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 = [
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Replace std::string::find() == 0 with absl::StartsWith (part 2). This CL has been generated using clang-tidy [1] except for changes to BUILD.gn files. [1] - https://clang.llvm.org/extra/clang-tidy/checks/abseil-string-find-startswith.html Bug: None Change-Id: Ibf75601065a53bde28623b8eef57bec067235640 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172586 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30984}
2020-04-01 13:43:08 +02:00
"//third_party/abseil-cpp/absl/strings",
Replace rtc::Optional with absl::optional This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script from modules with parameters 'pacing video_coding congestion_controller remote_bitrate_estimator': 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: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118 Reviewed-on: https://webrtc-review.googlesource.com/83900 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23640}
2018-06-18 10:48:16 +02:00
"//third_party/abseil-cpp/absl/types:optional",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
}
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
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("interval_budget") {
Removes redundant AlrDetector. This replaces the old AlrDetector used by the pacer with the one in GoogCC. This reduces the risk of accidentally changing only one version. Note that the pacer instance will be removed when moving over to the task queue based send side congestion controller. Bug: webrtc:8415 Change-Id: Id4b2000ee5a04b94565092c29a84572a7750d2f5 Reviewed-on: https://webrtc-review.googlesource.com/85363 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23791}
2018-06-25 17:56:08 +02:00
sources = [
"interval_budget.cc",
"interval_budget.h",
]
deps = [
"../../rtc_base:checks",
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",
Removes redundant AlrDetector. This replaces the old AlrDetector used by the pacer with the one in GoogCC. This reduces the risk of accidentally changing only one version. Note that the pacer instance will be removed when moving over to the task queue based send side congestion controller. Bug: webrtc:8415 Change-Id: Id4b2000ee5a04b94565092c29a84572a7750d2f5 Reviewed-on: https://webrtc-review.googlesource.com/85363 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23791}
2018-06-25 17:56:08 +02:00
]
}
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
if (rtc_include_tests) {
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("pacing_unittests") {
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
testonly = true
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
sources = [
"bitrate_prober_unittest.cc",
Let alr detector use a budged to detect underuse. BUG=webrtc:7947 Review-Url: https://codereview.webrtc.org/2965233002 Cr-Commit-Position: refs/heads/master@{#18972}
2017-07-11 06:56:04 -07:00
"interval_budget_unittest.cc",
Extract most of PacedSender into PacedSendingController. The Pacer now just handles interaction with Module/ProcessThread and forwarding packets to PacketRouter. All other logic is moved to PacedSendingController, including tests. PacedSender unittest are now just some basic sanity tests. Bug: webrtc:10809 Change-Id: I69223cd9d8300997375b03706d2e99c88e46241c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149041 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28886}
2019-08-14 10:43:47 +02:00
"pacing_controller_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"packet_router_unittest.cc",
Add new prioritized packet queue. This queue is a more strict round robing queue, unlike the class named RoundRobinPacketQueue. That is, we don't have the same logic to prioritize lower-bitrate streams. The queue time mechanism is essentially directly copied from the previous implementation however. Bug: webrtc:11340 Change-Id: Ie38ba8ce27c985f5f1e907cec068d6a365089bcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260562 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36737}
2022-05-02 21:18:08 +02:00
"prioritized_packet_queue_unittest.cc",
Implements a task-queue based PacedSender, wires it up for field trials Bug: webrtc:10809 Change-Id: Ia181c16559f4598f32dd399c24802d0a289e250b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150942 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29946}
2019-11-26 17:48:49 +01:00
"task_queue_paced_sender_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
]
deps = [
Removes redundant AlrDetector. This replaces the old AlrDetector used by the pacer with the one in GoogCC. This reduces the risk of accidentally changing only one version. Note that the pacer instance will be removed when moving over to the task queue based send side congestion controller. Bug: webrtc:8415 Change-Id: Id4b2000ee5a04b94565092c29a84572a7750d2f5 Reviewed-on: https://webrtc-review.googlesource.com/85363 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23791}
2018-06-25 17:56:08 +02:00
":interval_budget",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":pacing",
Reland "Experimentally reduce TaskQueuePacedSender's delayed task precision." This is a reland of commit 43a69b3f46059b103c35079744dc09a0e2bb2948 Original change's description: > Experimentally reduce TaskQueuePacedSender's delayed task precision. > > This CL reduces the delayed task precision of non-probes in accordance > with DD go/slacked-task-queue-paced-sender. The precision is only > deduced if field trial "WebRTC-SlackedTaskQueuePacedSender" is enabled > though. > > Bug: webrtc:13824 > Change-Id: I37e53b24e343f4f08059be08a3cda74f5484cc05 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255341 > Reviewed-by: Erik Språng <sprang@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#36214} Bug: webrtc:13824 Change-Id: I86cace6f4f6bf23d51c75b3d18f8d24fff0f5b74 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255826 Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36221}
2022-03-16 10:16:29 +01:00
"../../api/task_queue:task_queue",
Minimizes risk of probes being late when using TaskQueuePacedSender. The time precision of delayed tasks is one millisecond, so the TaskQueuePacedSender makes sure that is the minimum sleep time, and then allows sending prior data as if it was on time. Furthermore, if there already exists a pending task within 1ms of a new desired process time - we don't schedule a new one with the same motivation as above. These two facts clashes somewhat with how BitrateProber works, and especially if they coincide it can result in scheduled ProcessPackets() that is 2ms late. The default timeout set in BitrateProber is 3ms, so there is a higher risk of probes timing out. This CL changes the TaskQueuePacedSender to allow scheduling a ProcesPackets() call as soon as possible if we are probing - even if that means executing up to 1ms earlier than expected (the BitrateProber will compensate for that). The PacingController is updated in order to allow early execution in this one case. Bug: webrtc:10809 Change-Id: Ia5097ddc39aa80c05ebfe56369310c94ef0e0baf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178901 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31778}
2020-07-19 12:48:16 +02:00
"../../api/transport:network_control",
Extract most of PacedSender into PacedSendingController. The Pacer now just handles interaction with Module/ProcessThread and forwarding packets to PacketRouter. All other logic is moved to PacedSendingController, including tests. PacedSender unittest are now just some basic sanity tests. Bug: webrtc:10809 Change-Id: I69223cd9d8300997375b03706d2e99c88e46241c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149041 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28886}
2019-08-14 10:43:47 +02:00
"../../api/units:data_rate",
[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",
[SlackedPacer] Don't slack while retransmissions or audio is in queue. This CL introduces PacketQueue::SizeInPacketsPerRtpPacketMediaType keeping track of the number of packets in the queue per RtpPacketMediaType. The TaskQueuePacedSender is updated not to apply slack if the queue contains any kRetransmission or kAudio packets. The hope is that not slacking retransmissions will make the NACK/retransmission regression of the SlackedPacer experiment go away. Wanting to not slack audio packets is unrelated to the regression but a sensible thing to due since audio is highest priority. This CL does not change anything when the SlackedPacer experiment is not running, since if its not running then none of the packets are slacked. Bug: webrtc:14161 Change-Id: I1e588599b6b64ebfd7d890706b6afd0b84fd746d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265160 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37139}
2022-06-06 16:19:11 +02:00
"../../api/units:timestamp",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_tests_utils",
Moved ALR experiment settings to new experiments folder. This replaces most of the existing dependencies on the application limited region(ALR) detector. This is to achieve a greater separation of concerns and will make further refactoring regarding the ALR Detector less invasive on other parts of the code base. Bug: webrtc:8415 Change-Id: I92912254c6d02285cce6a88f6789f0ac94794c88 Reviewed-on: https://webrtc-review.googlesource.com/37560 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21598}
2018-01-12 10:54:18 +01:00
"../../rtc_base/experiments:alr_experiment",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
Allows bitrate prober to discard delayed probes, unit type refactorings This CL adds a parameter to the BirateProber field trial config, which allows the prober to actually discard probe cluster is pacer scheduling is too delayed. Today it just keeps going at a too low rate. Some refactoring was needed anyway, so also switch to using unit types in more places. Initially keeps legacy behavior default, to verify no perf regressions. Bug: webrtc:11780 Change-Id: I9edd114773b10a8d86b54a1a0398a4052aab9dd5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179090 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31756}
2020-07-17 12:06:12 +02:00
"../../test:explicit_key_value_config",
Make KeyValueConfig mandatory in the pacer. This CL also removes dependency on the legacy field trial methods. Bug: webrtc:11926 Change-Id: I53feeee86b92878cf0f2b8ebdce3d101f9e04014 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255381 Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Auto-Submit: Erik Språng <sprang@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36205}
2022-03-15 15:13:25 +01:00
"../../test:scoped_key_value_config",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../test:test_support",
Implements a task-queue based PacedSender, wires it up for field trials Bug: webrtc:10809 Change-Id: Ia181c16559f4598f32dd399c24802d0a289e250b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150942 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29946}
2019-11-26 17:48:49 +01:00
"../../test/time_controller:time_controller",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../rtp_rtcp",
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
"../rtp_rtcp:mock_rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
]
Migrate pacing and video_coding to absl::AnyInvocable based TaskQueueBase interface Bug: webrtc:14245 Change-Id: Icfab3e6548055ea72a199a226eca5233b1ead20d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267983 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37467}
2022-07-06 13:17:54 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/functional:any_invocable" ]
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
}
}
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