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webrtc_m130/modules/rtp_rtcp/BUILD.gn

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GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../../webrtc.gni")
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
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_rtcp_format") {
Set public visibility for rtp_rtcp and video_coding targets Though discouraged, those folders are listed in native-api NOTRY=True Bug: webrtc:9808 Change-Id: I9407c8d69a0d75196cfa9435f5e459264c64e046 Reviewed-on: https://webrtc-review.googlesource.com/c/103364 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24953}
2018-10-03 14:54:43 +02:00
visibility = [ "*" ]
rtp_rtcp_format: Separate public and private source files There was one .h file that didn't have to be public. :-) BUG=webrtc:8159, webrtc:8255 Change-Id: I0998f0340384c57f52affdde30f6b4eb2eaa712b Reviewed-on: https://webrtc-review.googlesource.com/2400 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19915}
2017-09-20 21:46:54 +02:00
public = [
ReportBlockData and observer added, for stats collection in future CLs. The ReportBlockData contains information about a ReportBlock and additional data such as RTT. This will be used for the calculation of RTCRemoteInboundRtpStreamStats, see full picture here: https://webrtc-review.googlesource.com/c/src/+/134107 ReportBlockData is a class version of the previously internal struct RTCPReceiver::ReportBlockWithRtt. - The new name makes sense even if we add more info to it, which will be needed for future metrics. - The new location is modules/rtp_rtcp/include/report_block_data.h. The RTCPReceiver allows obtaining the ReportBlockData in two ways: 1. Using a ReportBlockDataObserver that is notified on receiving a report block. 2. Using the GetLatestReportBlockData(). Both codepaths will be needed; video stats uses observers and audio stats uses polling. Further plumbing will be done in follow-up CLs. Bug: webrtc:10455, webrtc:10456 Change-Id: Ic9e5b4f451b5f4b203efcd6fa3bbf9736487e1f4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136584 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27961}
2019-05-16 13:32:20 +02:00
"include/report_block_data.h",
Move RtcpStatistics from common_types.h to a new header file New location is modules/rtp_rtcp/include/rtcp_statistics.h. Bug: webrtc:5876 Change-Id: I85f55b58658588228ed77175226b3479352fd5de Reviewed-on: https://webrtc-review.googlesource.com/c/111961 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25799}
2018-11-27 14:05:08 +01:00
"include/rtcp_statistics.h",
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
"include/rtp_cvo.h",
"include/rtp_header_extension_map.h",
Move RtpPacketSender and merge it with RtpPacketPacer. This interface is intended to only handle packet-sending parts of the paced sender. See https://webrtc-review.googlesource.com/c/src/+/145212 for context Bug: webrtc:10809 Change-Id: I93f0b40e1865665c2d436db67021350a0ed0687b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145216 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28662}
2019-07-24 14:52:55 +02:00
"include/rtp_packet_sender.h",
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
"include/rtp_rtcp_defines.h",
"source/byte_io.h",
Move rtcp packet classes from rtp_rtcp to rtp_rtcp_format target Bug: None Change-Id: I353228fd5b75bd4fceeaee1bb6fd07b01dac56a1 Reviewed-on: https://webrtc-review.googlesource.com/1480 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19867}
2017-09-15 16:48:14 +02:00
"source/rtcp_packet.h",
"source/rtcp_packet/app.h",
"source/rtcp_packet/bye.h",
"source/rtcp_packet/common_header.h",
"source/rtcp_packet/compound_packet.h",
"source/rtcp_packet/dlrr.h",
"source/rtcp_packet/extended_reports.h",
"source/rtcp_packet/fir.h",
Create new RTCP feedback message - LossIndication Create a new RTCP feedback message for reporting the loss and/or non-decodability of video frames, to be used by the upcoming injectable VideoFrameBufferController. The new feedback message should report: 1. The sequence number of the last decoded non-discardable video frame. (TBD: If a multi-packet frame, should it be the sequence number of the first, last, or any of the packets?) 2. The sequence number of the last received RTP packet in the stream. 3. A decodability flag, whose specific meaning depends on the last-received RTP sequence number. The decodability flag is true if and only if all of the frame's dependencies are known to be decodable, and the frame itself is not yet known to be unassemblable. * Clarification #1: In a multi-packet frame, the first packet's dependencies are known, but it is not yet known whether all parts of the current frame will be received. * Clarification #2: In a multi-packet frame, the dependencies would be unknown if the first packet was not received. Then, the packet will be known-unassemblable. Bug: webrtc:10226 Change-Id: I1563c944477e3ed40235e82ab99a439414632aff Reviewed-on: https://webrtc-review.googlesource.com/c/118931 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26387}
2019-01-24 12:38:36 +01:00
"source/rtcp_packet/loss_notification.h",
Move rtcp packet classes from rtp_rtcp to rtp_rtcp_format target Bug: None Change-Id: I353228fd5b75bd4fceeaee1bb6fd07b01dac56a1 Reviewed-on: https://webrtc-review.googlesource.com/1480 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19867}
2017-09-15 16:48:14 +02:00
"source/rtcp_packet/nack.h",
"source/rtcp_packet/pli.h",
"source/rtcp_packet/psfb.h",
"source/rtcp_packet/rapid_resync_request.h",
"source/rtcp_packet/receiver_report.h",
"source/rtcp_packet/remb.h",
Adds remote estimate RTCP packet. This adds the RemoteEstimate rtcp packet and wires it up to GoogCC where it's used to improve congestion controller behavior. The functionality is negotiated using SDP. It's added with a field trial that allow disabling the functionality in case there's any issues. Bug: webrtc:10742 Change-Id: I1ea8e4216a27cd2b00505c99b42d1e38726256c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146602 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28654}
2019-07-24 11:38:03 +02:00
"source/rtcp_packet/remote_estimate.h",
Move rtcp packet classes from rtp_rtcp to rtp_rtcp_format target Bug: None Change-Id: I353228fd5b75bd4fceeaee1bb6fd07b01dac56a1 Reviewed-on: https://webrtc-review.googlesource.com/1480 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19867}
2017-09-15 16:48:14 +02:00
"source/rtcp_packet/report_block.h",
"source/rtcp_packet/rrtr.h",
"source/rtcp_packet/rtpfb.h",
"source/rtcp_packet/sdes.h",
"source/rtcp_packet/sender_report.h",
"source/rtcp_packet/target_bitrate.h",
"source/rtcp_packet/tmmb_item.h",
"source/rtcp_packet/tmmbn.h",
"source/rtcp_packet/tmmbr.h",
"source/rtcp_packet/transport_feedback.h",
Add trait to Build/Parse DependencyDescriptor rtp header extension TBR=aleloi@webrtc.org Bug: webrtc:10342 Change-Id: I9d321ec47ed748ccfac2be6793923c46d0a88d16 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144032 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28415}
2019-06-27 16:45:40 +02:00
"source/rtp_dependency_descriptor_extension.h",
Introduce list of fields to put into codec agnostic descriptor Bug: webrtc:9361 Change-Id: Iff44f289ffcecf7e4f997d5001958ab22124910f Reviewed-on: https://webrtc-review.googlesource.com/81241 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23600}
2018-06-13 15:51:43 +02:00
"source/rtp_generic_frame_descriptor.h",
Add Parsing/Building generic frame descriptor extension Bug: webrtc:9361 Change-Id: I7e85826117348e2d4f4726e8d515bb1d4a289966 Reviewed-on: https://webrtc-review.googlesource.com/83622 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23662}
2018-06-19 14:24:17 +02:00
"source/rtp_generic_frame_descriptor_extension.h",
rtp_rtcp_format: Separate public and private source files There was one .h file that didn't have to be public. :-) BUG=webrtc:8159, webrtc:8255 Change-Id: I0998f0340384c57f52affdde30f6b4eb2eaa712b Reviewed-on: https://webrtc-review.googlesource.com/2400 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19915}
2017-09-20 21:46:54 +02:00
"source/rtp_header_extensions.h",
Make rtp_packet.h public This would allow us to limit the visibility of RtpPacketReceived and RtpPacketToSend, when we only want to allocate memory to save the RTP header, and not the metadata. TBR=danilchap@webrtc.org Bug: webrtc:8111 Change-Id: Ic9339189ccc2081d82bdc8def0fb39677458356f Reviewed-on: https://webrtc-review.googlesource.com/5521 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20075}
2017-10-02 13:10:01 +02:00
"source/rtp_packet.h",
rtp_rtcp_format: Separate public and private source files There was one .h file that didn't have to be public. :-) BUG=webrtc:8159, webrtc:8255 Change-Id: I0998f0340384c57f52affdde30f6b4eb2eaa712b Reviewed-on: https://webrtc-review.googlesource.com/2400 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19915}
2017-09-20 21:46:54 +02:00
"source/rtp_packet_received.h",
"source/rtp_packet_to_send.h",
Unify helpers IsRtpPacket and IsRtcpPacket Bug: None Change-Id: Ibe942de433435d256cd6827440136936d4b274d6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225022 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34419}
2021-07-05 19:06:17 +02:00
"source/rtp_util.h",
Add experimental extension RtpVideoLayersAllocation The extension is suggested to be used for signaling per target bitrate, resolution and frame rate to a SFU to allow a SFU to know what video layers a client is currently targeting. It is hoped to replace the current Target bitrate RTCP XR message currently used only for screen share. Bug: webrtc:12000 Change-Id: Id7b55e7ddaf6304e31839fd0482b096e1dbe8925 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185980 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32313}
2020-10-05 13:51:47 +02:00
"source/rtp_video_layers_allocation_extension.h",
rtp_rtcp_format: Separate public and private source files There was one .h file that didn't have to be public. :-) BUG=webrtc:8159, webrtc:8255 Change-Id: I0998f0340384c57f52affdde30f6b4eb2eaa712b Reviewed-on: https://webrtc-review.googlesource.com/2400 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19915}
2017-09-20 21:46:54 +02:00
]
sources = [
ReportBlockData and observer added, for stats collection in future CLs. The ReportBlockData contains information about a ReportBlock and additional data such as RTT. This will be used for the calculation of RTCRemoteInboundRtpStreamStats, see full picture here: https://webrtc-review.googlesource.com/c/src/+/134107 ReportBlockData is a class version of the previously internal struct RTCPReceiver::ReportBlockWithRtt. - The new name makes sense even if we add more info to it, which will be needed for future metrics. - The new location is modules/rtp_rtcp/include/report_block_data.h. The RTCPReceiver allows obtaining the ReportBlockData in two ways: 1. Using a ReportBlockDataObserver that is notified on receiving a report block. 2. Using the GetLatestReportBlockData(). Both codepaths will be needed; video stats uses observers and audio stats uses polling. Further plumbing will be done in follow-up CLs. Bug: webrtc:10455, webrtc:10456 Change-Id: Ic9e5b4f451b5f4b203efcd6fa3bbf9736487e1f4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136584 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27961}
2019-05-16 13:32:20 +02:00
"include/report_block_data.cc",
Convert PayloadUnion from a union to a class, step 3 Remove PayloadUnion's public member variables, so that the outside world has to go through the accessors. This is good code hygiene in general. For example, it makes it possible to make the audio and video states Optional, so that exactly one of them can be live at any one time. BUG=webrtc:8159 Change-Id: Ie617b9038f961b329bd67b45478ff33d97148447 Reviewed-on: https://webrtc-review.googlesource.com/4428 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20064}
2017-10-02 09:54:48 +02:00
"include/rtp_rtcp_defines.cc",
rtp_rtcp_format: Separate public and private source files There was one .h file that didn't have to be public. :-) BUG=webrtc:8159, webrtc:8255 Change-Id: I0998f0340384c57f52affdde30f6b4eb2eaa712b Reviewed-on: https://webrtc-review.googlesource.com/2400 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19915}
2017-09-20 21:46:54 +02:00
"source/rtcp_packet.cc",
"source/rtcp_packet/app.cc",
"source/rtcp_packet/bye.cc",
"source/rtcp_packet/common_header.cc",
"source/rtcp_packet/compound_packet.cc",
"source/rtcp_packet/dlrr.cc",
"source/rtcp_packet/extended_reports.cc",
"source/rtcp_packet/fir.cc",
Create new RTCP feedback message - LossIndication Create a new RTCP feedback message for reporting the loss and/or non-decodability of video frames, to be used by the upcoming injectable VideoFrameBufferController. The new feedback message should report: 1. The sequence number of the last decoded non-discardable video frame. (TBD: If a multi-packet frame, should it be the sequence number of the first, last, or any of the packets?) 2. The sequence number of the last received RTP packet in the stream. 3. A decodability flag, whose specific meaning depends on the last-received RTP sequence number. The decodability flag is true if and only if all of the frame's dependencies are known to be decodable, and the frame itself is not yet known to be unassemblable. * Clarification #1: In a multi-packet frame, the first packet's dependencies are known, but it is not yet known whether all parts of the current frame will be received. * Clarification #2: In a multi-packet frame, the dependencies would be unknown if the first packet was not received. Then, the packet will be known-unassemblable. Bug: webrtc:10226 Change-Id: I1563c944477e3ed40235e82ab99a439414632aff Reviewed-on: https://webrtc-review.googlesource.com/c/118931 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26387}
2019-01-24 12:38:36 +01:00
"source/rtcp_packet/loss_notification.cc",
rtp_rtcp_format: Separate public and private source files There was one .h file that didn't have to be public. :-) BUG=webrtc:8159, webrtc:8255 Change-Id: I0998f0340384c57f52affdde30f6b4eb2eaa712b Reviewed-on: https://webrtc-review.googlesource.com/2400 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19915}
2017-09-20 21:46:54 +02:00
"source/rtcp_packet/nack.cc",
"source/rtcp_packet/pli.cc",
"source/rtcp_packet/psfb.cc",
"source/rtcp_packet/rapid_resync_request.cc",
"source/rtcp_packet/receiver_report.cc",
"source/rtcp_packet/remb.cc",
Adds remote estimate RTCP packet. This adds the RemoteEstimate rtcp packet and wires it up to GoogCC where it's used to improve congestion controller behavior. The functionality is negotiated using SDP. It's added with a field trial that allow disabling the functionality in case there's any issues. Bug: webrtc:10742 Change-Id: I1ea8e4216a27cd2b00505c99b42d1e38726256c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146602 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28654}
2019-07-24 11:38:03 +02:00
"source/rtcp_packet/remote_estimate.cc",
rtp_rtcp_format: Separate public and private source files There was one .h file that didn't have to be public. :-) BUG=webrtc:8159, webrtc:8255 Change-Id: I0998f0340384c57f52affdde30f6b4eb2eaa712b Reviewed-on: https://webrtc-review.googlesource.com/2400 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19915}
2017-09-20 21:46:54 +02:00
"source/rtcp_packet/report_block.cc",
"source/rtcp_packet/rrtr.cc",
"source/rtcp_packet/rtpfb.cc",
"source/rtcp_packet/sdes.cc",
"source/rtcp_packet/sender_report.cc",
"source/rtcp_packet/target_bitrate.cc",
"source/rtcp_packet/tmmb_item.cc",
"source/rtcp_packet/tmmbn.cc",
"source/rtcp_packet/tmmbr.cc",
"source/rtcp_packet/transport_feedback.cc",
Add trait to Build/Parse DependencyDescriptor rtp header extension TBR=aleloi@webrtc.org Bug: webrtc:10342 Change-Id: I9d321ec47ed748ccfac2be6793923c46d0a88d16 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144032 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28415}
2019-06-27 16:45:40 +02:00
"source/rtp_dependency_descriptor_extension.cc",
Implement Dependency Descriptor reader Bug: webrtc:10342 Change-Id: I671bf57368016b633546966cc994646095433519 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152823 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29197}
2019-09-16 10:18:59 +02:00
"source/rtp_dependency_descriptor_reader.cc",
Implement Dependency Descriptor writer Bug: webrtc:10342 Change-Id: I561825265c0990864e1d16aeed4afbdd98871940 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153350 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29232}
2019-09-18 18:11:23 +02:00
"source/rtp_dependency_descriptor_reader.h",
"source/rtp_dependency_descriptor_writer.cc",
"source/rtp_dependency_descriptor_writer.h",
Introduce list of fields to put into codec agnostic descriptor Bug: webrtc:9361 Change-Id: Iff44f289ffcecf7e4f997d5001958ab22124910f Reviewed-on: https://webrtc-review.googlesource.com/81241 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23600}
2018-06-13 15:51:43 +02:00
"source/rtp_generic_frame_descriptor.cc",
Add Parsing/Building generic frame descriptor extension Bug: webrtc:9361 Change-Id: I7e85826117348e2d4f4726e8d515bb1d4a289966 Reviewed-on: https://webrtc-review.googlesource.com/83622 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23662}
2018-06-19 14:24:17 +02:00
"source/rtp_generic_frame_descriptor_extension.cc",
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
"source/rtp_header_extension_map.cc",
"source/rtp_header_extensions.cc",
"source/rtp_packet.cc",
"source/rtp_packet_received.cc",
Add application_data field(s) to RtpPacketToSend and PacketOptions. Pass pointer to application_data from RtpPacketToSend arriving via RtpSender::SendToNetwork through to Transport::SendRtp, in PacketOptions. Bug: webrtc:8906 Change-Id: Ie75013ed472710f4efcfbcc160e46a6119a1f41d Reviewed-on: https://webrtc-review.googlesource.com/55600 Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Dino Radaković <dinor@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22174}
2018-02-22 14:18:06 +01:00
"source/rtp_packet_to_send.cc",
Unify helpers IsRtpPacket and IsRtcpPacket Bug: None Change-Id: Ibe942de433435d256cd6827440136936d4b274d6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225022 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34419}
2021-07-05 19:06:17 +02:00
"source/rtp_util.cc",
Add experimental extension RtpVideoLayersAllocation The extension is suggested to be used for signaling per target bitrate, resolution and frame rate to a SFU to allow a SFU to know what video layers a client is currently targeting. It is hoped to replace the current Target bitrate RTCP XR message currently used only for screen share. Bug: webrtc:12000 Change-Id: Id7b55e7ddaf6304e31839fd0482b096e1dbe8925 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185980 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32313}
2020-10-05 13:51:47 +02:00
"source/rtp_video_layers_allocation_extension.cc",
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
]
deps = [
[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
"..:module_api_public",
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
"../../api:array_view",
Move rtc::FunctionView to the public API Bug: webrtc:10138 Change-Id: Icc25a2a277a9608701aaddd546882366739991ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127898 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27227}
2019-03-21 14:37:36 +01:00
"../../api:function_view",
Add option to attach custom object to an rtp packet As an alternative to attaching custom array of bytes. Bug: b/178094662 Change-Id: I92dcbf04998d8206091125febc520ebfcc4bcebf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203264 Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33069}
2021-01-25 17:02:57 +01:00
"../../api:refcountedbase",
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",
Add option to attach custom object to an rtp packet As an alternative to attaching custom array of bytes. Bug: b/178094662 Change-Id: I92dcbf04998d8206091125febc520ebfcc4bcebf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203264 Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33069}
2021-01-25 17:02:57 +01:00
"../../api:scoped_refptr",
RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs This is needed in the general case, now that we aim to support codecs other than those built-in to WebRTC. BUG=webrtc:8159 Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706 Reviewed-on: https://webrtc-review.googlesource.com/5380 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20136}
2017-10-04 12:38:53 +02:00
"../../api/audio_codecs:audio_codecs_api",
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",
Publish DependencyDescriptor structures in the api The extension (and thus structures to carry it) are designed in particular for client<->SFU link. Putting the structure into api acknowledges it can be reused by SFU projects Bug: webrtc:10342 Change-Id: I8ca1f5046abadf6aa16200443c4892e9a2a928b4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166467 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30324}
2020-01-17 16:13:29 +01:00
"../../api/transport/rtp:dependency_descriptor",
Using unit types in TransportFeedbackAdapter. Bug: webrtc:9883 Change-Id: I6d7d653079bb969fa3bc6f62fd35f2aa870edab6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158792 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29705}
2019-11-05 13:25:47 +01:00
"../../api/units:time_delta",
Store RtpPacketReceived::arrival_time as Timestamp. Previously this value was rounded up to a millisecond value. This change is complementary to another change: https://webrtc-review.googlesource.com/c/src/+/216398 Bug: webrtc:12722 Change-Id: I0fd2baceb4608132615fb6ad241ec863e343edb1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217521 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33928}
2021-05-05 12:33:00 +02:00
"../../api/units:timestamp",
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",
VideoStreamEncoder report VideoLayersAllocation for simulcast Adds support for Vp8 simulcast. Bug: webrtc:12000 Change-Id: Ib24fd0542642b023ec35f7a7bdc4880d72365edf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187341 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32416}
2020-10-15 17:53:22 +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",
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
"../../common_video",
Move bit_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ia6e0f1c3017a968b22fdca512ef6adb37957d777 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258769 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36604}
2022-04-14 13:34:56 +02:00
"../../rtc_base:bit_buffer",
Migrate Dependency Descriptor parser from BitBuffer to BitstreamReader BitstreamReader itself uses idea of Read function that always succeed, and a separate function to check for errors. Thus extra layer in the DependencyDescriptorReader is not needed. Bug: None Change-Id: Ie58861f2cbecc02a5a1a9538232494b4442c9afd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231226 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34940}
2021-09-07 11:16:50 +02:00
"../../rtc_base:bitstream_reader",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
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",
Implement Dependency Descriptor writer Bug: webrtc:10342 Change-Id: I561825265c0990864e1d16aeed4afbdd98871940 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153350 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29232}
2019-09-18 18:11:23 +02:00
"../../rtc_base:divide_round",
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",
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 //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",
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
"../../system_wrappers",
[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
"../video_coding:codec_globals_headers",
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 Abseil container algorithms in modules/rtp_rtcp/ Bug: None Change-Id: Ica2e9795ec6195e044403f5ee25e476f6c47cf93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129600 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27361}
2019-03-28 10:56:11 -07:00
"//third_party/abseil-cpp/absl/algorithm:container",
Add functions IsLegalMidName and IsLegalRsidName This is a preparation for deleting the class StringRtpHeaderExtension. Bug: webrtc:10440 Change-Id: I3480e58d96e67d10c4d78597c8ab7f01b63e37ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128761 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27228}
2019-03-21 15:48:49 +01:00
"//third_party/abseil-cpp/absl/strings",
Replace rtc::Optional with absl::optional in modules/rtp_rtcp This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated using script: #!/bin/bash dir=modules/rtp_rtcp find $dir -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $dir -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc Reviewed-on: https://webrtc-review.googlesource.com/83584 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-14 12:59:38 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Implement PayloadUnion as variant instead of pair of optionals Bug: None Change-Id: I2e54f5a0561804bc59c4d4c8e35ccdaa9536b8e4 Reviewed-on: https://webrtc-review.googlesource.com/85366 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23745}
2018-06-26 10:44:13 +02:00
"//third_party/abseil-cpp/absl/types:variant",
Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies BUG=webrtc:8111 patch from issue 3011233002 at patchset 1 (http://crrev.com/3011233002#ps1) Review-Url: https://codereview.webrtc.org/3014463002 Cr-Commit-Position: refs/heads/master@{#19801}
2017-09-12 09:23:24 -07:00
]
}
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_rtcp") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
sources = [
Add a FlexfecReceiver class. This class is split in interface/implementation classes, since it will be referenced from the Call level. Its purpose is to interface the erasure code decoder with a new class FlexfecReceiveStream (for received packets), as well as with the main RTP pipeline (for recovered packets). BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2392663006 Cr-Commit-Position: refs/heads/master@{#14594}
2016-10-10 16:44:57 -07:00
"include/flexfec_receiver.h",
Add FlexfecSender. This class will interface RTPSenderVideo with the underlying erasure code. It is functionally similar to ProducerFec (to be renamed UlpfecGenerator). In fact, the FlexfecSender is a friend of ProducerFec, and reuses most of its implementation. Besides the fact that FlexfecSender outputs FlexFEC packets, the main difference with ProducerFec is that FlexfecSender allocates RTP sequence numbers, whereas ProducerFec does not do this for the RED-encapsulated ULPFEC packets. This class is split as interface/implementation, since it will be owned by VideoSendStream initially. Further along, it may be owned by PacedSender. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2441613002 Cr-Commit-Position: refs/heads/master@{#14922}
2016-11-03 09:22:33 -07:00
"include/flexfec_sender.h",
modules: more interface -> include renames This changes the following module directories: * webrtc/modules/audio_conference_mixer/interface * webrtc/modules/interface * webrtc/modules/media_file/interface * webrtc/modules/rtp_rtcp/interface * webrtc/modules/utility/interface To avoid breaking downstream, I followed this recipe: 1. Copy the interface dir to a new sibling directory: include 2. Update the header guards in the include directory to match the style guide. 3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code. 4. Add a pragma warning in the header files in the interface dir. Example: #pragma message("WARNING: webrtc/modules/interface is DEPRECATED; " "use webrtc/modules/include") 5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S) 6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*) BUG=5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417683006 . Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 08:31:52 +01:00
"include/receive_statistics.h",
"include/remote_ntp_time_estimator.h",
Make local to capturer clock offset a separate entry in PacketInfo. This also changes the meaning of |estimated_capture_clock_offset| in |absolute_capture_time_| to become a remote to capturer clock offset. Bug: chromium:1056230, webrtc:10739 Change-Id: Id658590e027bbe77ae0834ea224e1dc977a305f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219163 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/master@{#34067}
2021-05-19 14:38:25 +02:00
"source/absolute_capture_time_interpolator.cc",
"source/absolute_capture_time_interpolator.h",
Add helper classes to send and receive abs-capture-time extensions. This change adds helper classes to manipulate Absolute Capture Time header extensions. Both classes support the "timestamp interpolation" optimization. Bug: webrtc:10739 Change-Id: I08eff46eb8910842a6dbaa3288b976004fabe1c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149801 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28936}
2019-08-22 10:05:01 +02:00
"source/absolute_capture_time_sender.cc",
"source/absolute_capture_time_sender.h",
Add helper for DependencyDescriptor rtp header extension to decide when to set active_decode_target_bitmask field Bug: webrtc:10342 Change-Id: I348d7467a72b45651455f4574fe8fda3c77ebbae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177400 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31555}
2020-06-22 17:53:01 +02:00
"source/active_decode_targets_helper.cc",
"source/active_decode_targets_helper.h",
Make local to capturer clock offset a separate entry in PacketInfo. This also changes the meaning of |estimated_capture_clock_offset| in |absolute_capture_time_| to become a remote to capturer clock offset. Bug: chromium:1056230, webrtc:10739 Change-Id: Id658590e027bbe77ae0834ea224e1dc977a305f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219163 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/master@{#34067}
2021-05-19 14:38:25 +02:00
"source/capture_clock_offset_updater.cc",
"source/capture_clock_offset_updater.h",
Add implementations of the VideoRtpDepacketizer interface while suboptimal, these implementions are complete and allow to swap code from using RtpDepacketizer interface to VideoRtpDepacketizer Bug: webrtc:11152 Change-Id: Ie7823feeb5b0563b74754255aaedfada9d446ac5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161380 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30031}
2019-12-05 16:24:04 +01:00
"source/create_video_rtp_depacketizer.cc",
"source/create_video_rtp_depacketizer.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/dtmf_queue.cc",
"source/dtmf_queue.h",
Move the FEC private tables into .cc files. Change the arrays to be continuous uint8_t arrays instead being of arrays of arrays (of arrays). Code size difference is 17K for arm, ~42K for arm64. New lookup algorithm, tailored for these two tables + tests. Instead of returning a raw pointer into the table, the algorithm returns an ArrayView, which includes size information for how much memory can be read. Change-Id: I000b094520bac944e518eb8b51d8dbef4670f5d7 Bug: webrtc:9102 Reviewed-on: https://webrtc-review.googlesource.com/65920 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22736}
2018-04-04 13:48:17 +02:00
"source/fec_private_tables_bursty.cc",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/fec_private_tables_bursty.h",
Move the FEC private tables into .cc files. Change the arrays to be continuous uint8_t arrays instead being of arrays of arrays (of arrays). Code size difference is 17K for arm, ~42K for arm64. New lookup algorithm, tailored for these two tables + tests. Instead of returning a raw pointer into the table, the algorithm returns an ArrayView, which includes size information for how much memory can be read. Change-Id: I000b094520bac944e518eb8b51d8dbef4670f5d7 Bug: webrtc:9102 Reviewed-on: https://webrtc-review.googlesource.com/65920 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22736}
2018-04-04 13:48:17 +02:00
"source/fec_private_tables_random.cc",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/fec_private_tables_random.h",
Add FlexFEC header formatters. Add classes that can read and finalize FlexFEC headers. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2269903002 Cr-Commit-Position: refs/heads/master@{#14469}
2016-10-03 00:43:25 -07:00
"source/flexfec_header_reader_writer.cc",
"source/flexfec_header_reader_writer.h",
Make FlexfecReceiver a concrete class. There is no need for it to be an interface. In this CL, I also took the opportunity to make two small fixes: - remove the 'flexfec_' prefix from some member variables - remove unnecessary use of a stringstream object BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2471073003 Cr-Commit-Position: refs/heads/master@{#14919}
2016-11-03 08:18:27 -07:00
"source/flexfec_receiver.cc",
Add FlexfecSender. This class will interface RTPSenderVideo with the underlying erasure code. It is functionally similar to ProducerFec (to be renamed UlpfecGenerator). In fact, the FlexfecSender is a friend of ProducerFec, and reuses most of its implementation. Besides the fact that FlexfecSender outputs FlexFEC packets, the main difference with ProducerFec is that FlexfecSender allocates RTP sequence numbers, whereas ProducerFec does not do this for the RED-encapsulated ULPFEC packets. This class is split as interface/implementation, since it will be owned by VideoSendStream initially. Further along, it may be owned by PacedSender. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2441613002 Cr-Commit-Position: refs/heads/master@{#14922}
2016-11-03 09:22:33 -07:00
"source/flexfec_sender.cc",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/forward_error_correction.cc",
"source/forward_error_correction.h",
"source/forward_error_correction_internal.cc",
"source/forward_error_correction_internal.h",
Add statistics gathering for packet loss. Adds a class used to classify whether packet loss events are a single packet or multiple packets as well as how many packets have been lost. Also exposes a new function in the RtpRtcp interface to retrieve these statistics. BUG= Review URL: https://codereview.webrtc.org/1198853004 Cr-Commit-Position: refs/heads/master@{#9568}
2015-07-10 18:10:05 -07:00
"source/packet_loss_stats.cc",
"source/packet_loss_stats.h",
Extract sequencing from RtpSender This CL refactors RtpSender and extracts handling of sequence number assignment and timestamping of padding packets in a separate helper class. This is in preparation for allowing deferred sequencing to after the pacing stage. Bug: webrtc:11340 Change-Id: I5f8c67f3bb90780b3bdd24afa6ae28dbe9d839a0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208401 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33316}
2021-02-22 13:34:31 +01:00
"source/packet_sequencer.cc",
"source/packet_sequencer.h",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"source/receive_statistics_impl.cc",
"source/receive_statistics_impl.h",
"source/remote_ntp_time_estimator.cc",
Remove rtcp_utility as mostly unused. Since the only used class is RTCPUtilitiy::NackStats, rename it to RtcpNackStats and move it into dedicated file. BUG=webrtc:5565 Review-Url: https://codereview.webrtc.org/2680183004 Cr-Commit-Position: refs/heads/master@{#16515}
2017-02-09 05:21:42 -08:00
"source/rtcp_nack_stats.cc",
"source/rtcp_nack_stats.h",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"source/rtcp_receiver.cc",
"source/rtcp_receiver.h",
"source/rtcp_sender.cc",
"source/rtcp_sender.h",
Do not propagate generic descriptor on receiving frame It was used only for the frame decryptor. Decryptor needs only raw representation that it can recreate in a way compatible with the new version of the descriptor. This relands commit abf73de8eae90e9ac7e88ce1d52728e8102e824f. with adjustments. Change-Id: I935977179bef31d8e1023964b967658e9a7db92d Bug: webrtc:10342 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168489 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30532}
2020-02-12 19:25:57 +01:00
"source/rtp_descriptor_authentication.cc",
"source/rtp_descriptor_authentication.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/rtp_format.cc",
"source/rtp_format.h",
"source/rtp_format_h264.cc",
"source/rtp_format_h264.h",
"source/rtp_format_video_generic.cc",
"source/rtp_format_video_generic.h",
"source/rtp_format_vp8.cc",
"source/rtp_format_vp8.h",
Add support for VP9 packetization/depacketization. RTP payload format for VP9: https://www.ietf.org/id/draft-uberti-payload-vp9-01.txt BUG=webrtc:4148, webrtc:4168, chromium:500602 TBR=mflodman Review URL: https://codereview.webrtc.org/1232023006 Cr-Commit-Position: refs/heads/master@{#9649}
2015-07-28 04:02:54 -07:00
"source/rtp_format_vp9.cc",
"source/rtp_format_vp9.h",
Move RtpHeaderExtensionMap::GetTotalLengthInBytes into own file Rename to better match what it does, Adjust to support two-byte header extension Bug: webrtc:7990 Change-Id: I2786d70e7cf9cd3d722f54fb1d07c9cfaafab947 Reviewed-on: https://webrtc-review.googlesource.com/103201 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24958}
2018-10-03 10:15:36 +02:00
"source/rtp_header_extension_size.cc",
"source/rtp_header_extension_size.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/rtp_packet_history.cc",
"source/rtp_packet_history.h",
Implement AV1 RtpPacketizer class Bug: webrtc:11042 Change-Id: Id1fc0acfa87a4520344f2636f50cb4d4e7284829 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160416 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29947}
2019-11-28 13:42:50 +01:00
"source/rtp_packetizer_av1.cc",
"source/rtp_packetizer_av1.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/rtp_rtcp_config.h",
Add thread/sequence checks to ModuleRtpRtcpImpl. This ended up with needing to fork the current implementation in order to not break downstream projects that were inheriting from it. While those get updated, we'll move on with the forked class. Bug: webrtc:11581,b/8278269 Change-Id: I05b596cbda71aa5b72894c31a7119d17d4761883 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175500 Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31334}
2020-05-20 09:32:51 +02:00
"source/rtp_rtcp_impl2.cc",
"source/rtp_rtcp_impl2.h",
Remove dependency from RtpRtcp on the Module interface. The 'Module' part of the implementation must not be called via the RtpRtcp interface, but is rather a part of the contract with ProcessThread. That in turn is an implementation detail for how timers are currently implemented in the default implementation. Along the way I'm deprecating away the factory function which was inside the interface and tied it to one specific implementation. Instead, I'm moving that to the implementation itself and down the line, we don't have to go through it if we just want to create an instance of the class. The key change is in rtp_rtcp.h and the new rtp_rtcp_interface.h header file (things moved from rtp_rtcp.h), the rest falls from that. Change-Id: I294f13e947b9e3e4e649400ee94a11a81e8071ce Bug: webrtc:11581 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176419 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31440}
2020-06-03 22:55:33 +02:00
"source/rtp_rtcp_interface.h",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"source/rtp_sender.cc",
"source/rtp_sender.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/rtp_sender_audio.cc",
"source/rtp_sender_audio.h",
Split RTPSender into pre- and post-pacer parts. Post-pacer code now contained in RtpSenderEgress class. For now, this is a member of RTPSender. More refactoring is needed to make clean split. Bug: webrtc:11036 Change-Id: I95264d013de120601784f130ba81c7b234446980 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157172 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29519}
2019-10-17 16:56:22 +02:00
"source/rtp_sender_egress.cc",
"source/rtp_sender_egress.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"source/rtp_sender_video.cc",
"source/rtp_sender_video.h",
Transform encoded frame in RTPSenderVideo. This change is part of the implementation of the Insertable Streams Web API: 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: I491ecefc60d184b75128799274c7d7efcf907d2a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169128 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30666}
2020-03-03 05:39:28 +01:00
"source/rtp_sender_video_frame_transformer_delegate.cc",
"source/rtp_sender_video_frame_transformer_delegate.h",
Introduce RtpSequenceNumberMap The video encoder should be informed of incoming LossNotification RTCP messages. Since it is unaware of RTP sequence numbers, or anything else RTP-related, the sequence numbers mentioned in the RTCP message must first be mapped to timestamps, since those are meaningful to the encoder. This CL introduces RtpSequenceNumberMap, which maps RTP sequence numbers to timestamps, while providing: 1. Capping the number of entries. 2. Wrap-around handling. RtpSequenceNumberMap also remembers which packets were first and/or last in the frame. Later CLs will wire this up. Bug: webrtc:10501 Change-Id: Ie0662cdb5706a3bcf63aa2934816a9df88439357 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130497 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27448}
2019-04-04 11:43:54 +02:00
"source/rtp_sequence_number_map.cc",
"source/rtp_sequence_number_map.h",
Add a tracker for RTCRtpContributingSource and RTCRtpSynchronizationSource. This change adds a new SourceTracker class that can do spec-compliant tracking of RTCRtpContributingSource and RTCRtpSynchronizationSource when frames are delivered to the RTCRtpReceiver's MediaStreamTrack for playout. It will replace the existing spec-incompliant ContributingSources. Bug: webrtc:10545 webrtc:10668 Change-Id: I961adaba09d6337f2f36b301a4fabcd20de65271 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140948 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28249}
2019-06-12 12:13:22 +02:00
"source/source_tracker.cc",
"source/source_tracker.h",
rtt calculation handles time go backwards CompactNtpIntervalToMs renamed to CompactNtpRttToMs and handle special cases: large values consider negative/invalid and result in value of 1. 0 result consider too small and increases to 1. BUG=590996 R=asapersson@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1763823003 . Cr-Commit-Position: refs/heads/master@{#11928}
2016-03-09 15:14:35 +01:00
"source/time_util.cc",
"source/time_util.h",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"source/tmmbr_help.cc",
"source/tmmbr_help.h",
Rename ProducerFec to UlpfecGenerator. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2449783002 Cr-Commit-Position: refs/heads/master@{#14847}
2016-10-31 05:27:07 -07:00
"source/ulpfec_generator.cc",
"source/ulpfec_generator.h",
Generalize FEC header formatting. - Split out reading/writing of FEC headers to classes separate from ForwardErrorCorrection. This makes ForwardErrorCorrection oblivious to what FEC header scheme is used, and lets it focus on encoding/decoding the FEC payloads. - Add unit tests for FEC header readers/writers. - Split ForwardErrorCorrection::XorPackets into XorHeaders and XorPayloads and reuse these functions for both encoding and decoding. - Rename AttemptRecover -> AttemptRecovery in ForwardErrorCorrection. BUG=webrtc:5654 R=danilchap@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2260803002 . Cr-Commit-Position: refs/heads/master@{#14316}
2016-09-21 09:19:34 +02:00
"source/ulpfec_header_reader_writer.cc",
"source/ulpfec_header_reader_writer.h",
Remove UlpfecReceiver virtual interface. There's only one implementation. Bug: none Change-Id: I204c23e7f87102909fcf6ee8632ceeed84e901a1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271026 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37725}
2022-08-09 17:50:45 +02:00
"source/ulpfec_receiver.cc",
"source/ulpfec_receiver.h",
Reland "Reland "Refactors UlpFec and FlexFec to use a common interface."" This is a reland of 49734dc0faa69616a58a1a95c7fc61a4610793cf Patchset 2 contains a fix for the fuzzer set up. Since we now parse an RtpPacket out of the fuzzer data, the header needs to be correct, otherwise we fail before even reaching the FEC code that we actually want to test. Bug: webrtc:11340, chromium:1052323, chromium:1055974 TBR=stefan@webrtc.org Original change's description: > Reland "Refactors UlpFec and FlexFec to use a common interface." > > This is a reland of 11af1d7444fd7438766b7bc52cbd64752d72e32e > > Original change's description: > > Refactors UlpFec and FlexFec to use a common interface. > > > > The new VideoFecGenerator is now injected into RtpSenderVideo, > > and generalizes the usage. > > This also prepares for being able to genera FEC in the RTP egress > > module. > > > > Bug: webrtc:11340 > > Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347 > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Commit-Queue: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30515} > > Bug: webrtc:11340, chromium:1052323 > Change-Id: Id646047365f1c46cca9e6f3e8eefa5151207b4a0 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168608 > Commit-Queue: Erik Språng <sprang@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30593} Bug: webrtc:11340, chromium:1052323 Change-Id: Ib8925f44e2edfcfeadc95c845c3bfc23822604ed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169222 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30724}
2020-03-05 10:14:04 +01:00
"source/video_fec_generator.h",
Hide Av1 specfic logic from RtpVideoReceiver into depacketizer interface. Bug: None Change-Id: I0498d9e82cbc876d54bebc7f3265e3ae6da61614 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171062 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30872}
2020-03-24 13:24:42 +01:00
"source/video_rtp_depacketizer.cc",
Introduce VideoRtpDepacketizer interface to replace RtpDepacketizer Bug: webrtc:11152 Change-Id: I20fd81233080d45d8978e5e57d0be6b592f44f43 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161322 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30018}
2019-12-05 14:41:45 +01:00
"source/video_rtp_depacketizer.h",
Change Av1 depacketizer to implement VideoRtpDepacketizer interface Bug: webrtc:11152 Change-Id: I322115263f60439bee36277157a0acef9bd28e3e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165343 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30260}
2020-01-10 17:26:53 +01:00
"source/video_rtp_depacketizer_av1.cc",
"source/video_rtp_depacketizer_av1.h",
Add VideoRtpDepacketizerGeneric Bug: webrtc:11152 Change-Id: I27d6a62093d36a4e77dd35d4c115af8cdcc0178a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162202 Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30160}
2019-12-23 13:14:28 +01:00
"source/video_rtp_depacketizer_generic.cc",
"source/video_rtp_depacketizer_generic.h",
Move RtpDepacketizerH264 into own files Bug: webrtc:11152 Change-Id: Iab4975e9f378b177a2abf34559f9b74752e69843 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165582 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30212}
2020-01-09 18:55:02 +01:00
"source/video_rtp_depacketizer_h264.cc",
"source/video_rtp_depacketizer_h264.h",
Add implementations of the VideoRtpDepacketizer interface while suboptimal, these implementions are complete and allow to swap code from using RtpDepacketizer interface to VideoRtpDepacketizer Bug: webrtc:11152 Change-Id: Ie7823feeb5b0563b74754255aaedfada9d446ac5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161380 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30031}
2019-12-05 16:24:04 +01:00
"source/video_rtp_depacketizer_raw.cc",
"source/video_rtp_depacketizer_raw.h",
Move vp8 rtp depacketization to VideoRtpDepacketizerVp8 Bug: webrtc:11152 Change-Id: Ic2b7fd091cb4d095ce29fbe06196f6424c08fce1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161451 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30088}
2019-12-11 11:24:37 +01:00
"source/video_rtp_depacketizer_vp8.cc",
"source/video_rtp_depacketizer_vp8.h",
Move vp9 rtp depacketization to VideoRtpDepacketizerVp9 Bug: webrtc:11152 Change-Id: I560d4cd62fabae093e3df592f0e7cc4001c10657 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162420 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30102}
2019-12-16 13:19:28 +01:00
"source/video_rtp_depacketizer_vp9.cc",
"source/video_rtp_depacketizer_vp9.h",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
]
Enable BWE logging to command line when rtc_enable_bwe_test_logging is set to true This patch enables bwe related variable logging to the command line. This is useful to test congestion control algorithm over real networks. NOTRY=true Review-Url: https://codereview.webrtc.org/2296253002 Cr-Commit-Position: refs/heads/master@{#14209}
2016-09-14 05:04:36 -07:00
if (rtc_enable_bwe_test_logging) {
defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
} else {
defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
}
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
deps = [
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
":rtp_rtcp_format",
[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
":rtp_video_header",
"..:module_api_public",
"..:module_fec_api",
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",
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 RtpPacketHistory::SetSendTime() This method will be used instead of GetPacketAndSetSendTime() when the new pacer code path is used, where the packet isn't stored in the history during pacing. Bug: webrtc:10633 Change-Id: Ie168125d949cef617ade3868a1858ed1dffe909c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140892 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28240}
2019-06-11 18:59:39 +02:00
"../../api:function_view",
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) Reason for revert: Re-land, reverting did not fix bug. https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 Original issue's description: > Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) > > Reason for revert: > Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see > > https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 > > Original issue's description: > > Added the GetSources() to the RtpReceiverInterface and implemented > > it for the AudioRtpReceiver. > > > > This method returns a vector of RtpSource(both CSRC source and SSRC > > source) which contains the ID of a source, the timestamp, the source > > type (SSRC or CSRC) and the audio level. > > > > The RtpSource objects are buffered and maintained by the > > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called, > > the info of the contributing source will be pulled along the object > > chain: > > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel -> > > AudioReceiveStream -> voe::Channel -> RtpRtcp module > > > > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource > > > > BUG=chromium:703122 > > TBR=stefan@webrtc.org, danilchap@webrtc.org > > > > Review-Url: https://codereview.webrtc.org/2770233003 > > Cr-Commit-Position: refs/heads/master@{#17591} > > Committed: https://chromium.googlesource.com/external/webrtc/+/292084c3765d9f3ee406ca2ec86eae206b540053 > > TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=chromium:703122 > > Review-Url: https://codereview.webrtc.org/2809613002 > Cr-Commit-Position: refs/heads/master@{#17616} > Committed: https://chromium.googlesource.com/external/webrtc/+/fbcc5cb3869d1370008e40f24fc03ac8fb69c675 TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:703122 Review-Url: https://codereview.webrtc.org/2810623003 Cr-Commit-Position: refs/heads/master@{#17621}
2017-04-10 07:39:05 -07:00
"../../api:libjingle_peerconnection_api",
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",
Add a tracker for RTCRtpContributingSource and RTCRtpSynchronizationSource. This change adds a new SourceTracker class that can do spec-compliant tracking of RTCRtpContributingSource and RTCRtpSynchronizationSource when frames are delivered to the RTCRtpReceiver's MediaStreamTrack for playout. It will replace the existing spec-incompliant ContributingSources. Bug: webrtc:10545 webrtc:10668 Change-Id: I961adaba09d6337f2f36b301a4fabcd20de65271 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140948 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28249}
2019-06-12 12:13:22 +02:00
"../../api:rtp_packet_info",
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",
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",
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. transport.h defines an interface for sending rtp and rtcp packets, which is used by MediaChannel in webrtc/media/engine, {Audio|Video}{Send|Receive}Stream and in a few other places. It was part of the build target //webrtc:webrtc, which is a monolithic target with all webrtc production code. This CL moves the header to its own target in webrtc/api and deprecates the old location. Targets in webrtc/api should in general only depend on other targets in webrtc/api. The target webrtc/api:call_api depends on transport.h. This change also makes webrtc/voice_engine pass GN's header include checker and is needed in order for webrtc/api:call_api to pass it. transport.h will be completely removed in a follow-up CL in a few weeks after clients have updated their includes. NOTRY=True BUG=webrtc:5589, webrtc:5878, webrtc:6785 Review-Url: https://codereview.webrtc.org/2426563003 Cr-Commit-Position: refs/heads/master@{#15267}
2016-11-28 07:02:13 -08:00
"../../api:transport_api",
Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2774833003 Cr-Commit-Position: refs/heads/master@{#17391}
2017-03-27 07:15:49 -07:00
"../../api/audio_codecs:audio_codecs_api",
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",
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 to_queued_task.h and pending_task_safety_flag.h into public API Bug: b/235812579 Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-16 21:27:45 +02:00
"../../api/task_queue:pending_task_safety_flag",
Transform encoded frame in RTPSenderVideo. This change is part of the implementation of the Insertable Streams Web API: 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: I491ecefc60d184b75128799274c7d7efcf907d2a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169128 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30666}
2020-03-03 05:39:28 +01:00
"../../api/task_queue:task_queue",
Replace field trials with WebRtcKeyValueConfig in RtpRtcpModule Replaces use of field trials in RtpSender and RtpVideoSender with injectable WebRtcKeyValueConfig. Implementation still defaults to field trials. BUG: webrtc:10335 Change-Id: I5fc6d327ee5d011ccc41385734b38344df172627 Reviewed-on: https://webrtc-review.googlesource.com/c/123447 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26795}
2019-02-21 07:55:59 +01:00
"../../api/transport:field_trial_based_config",
in RtpSenderVideo add support for writing DependencyDescriptor header extension Bug: webrtc:10342 Change-Id: I12cca9c5e1606338bb914e58e13d268bbc6961f9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166532 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30427}
2020-01-30 16:28:53 +01:00
"../../api/transport/rtp:dependency_descriptor",
Split out RtpSource from libjingle_peerconnection_api And moved declaration into a new api directory, as api/transport/rtp/rtp_source.h. Bug: webrtc:8733 Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880 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@{#29039}
2019-09-02 15:16:49 +02:00
"../../api/transport/rtp:rtp_source",
Split RTPSender into pre- and post-pacer parts. Post-pacer code now contained in RtpSenderEgress class. For now, this is a member of RTPSender. More refactoring is needed to make clean split. Bug: webrtc:11036 Change-Id: I95264d013de120601784f130ba81c7b234446980 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157172 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29519}
2019-10-17 16:56:22 +02:00
"../../api/units:data_rate",
Add helper classes to send and receive abs-capture-time extensions. This change adds helper classes to manipulate Absolute Capture Time header extensions. Both classes support the "timestamp interpolation" optimization. Bug: webrtc:10739 Change-Id: I08eff46eb8910842a6dbaa3288b976004fabe1c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149801 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28936}
2019-08-22 10:05:01 +02:00
"../../api/units:time_delta",
"../../api/units:timestamp",
Transform encoded frame in RTPSenderVideo. This change is part of the implementation of the Insertable Streams Web API: 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: I491ecefc60d184b75128799274c7d7efcf907d2a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169128 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30666}
2020-03-03 05:39:28 +01:00
"../../api/video:encoded_frame",
Add RtpDepacketizerAv1::AssembleFrame function Bug: webrtc:11042 Change-Id: I677fc6a9affacf3b7c80adc2c3493c16806db1f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160003 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29862}
2019-11-21 14:08:28 +01:00
"../../api/video:encoded_image",
Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets Bug: webrtc:9271 Change-Id: Ic7415830588bef9d87bab92943460207890dada6 Reviewed-on: https://webrtc-review.googlesource.com/76960 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 12:35:56 +02:00
"../../api/video:video_bitrate_allocation",
Exposing video bitrate allocator into API In order to have public video bitrate allocator factory, the video bitrate allocator has be part of the api. Bug: webrtc:9513 Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2 Reviewed-on: https://webrtc-review.googlesource.com/88083 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#24073}
2018-07-20 15:49:43 -07:00
"../../api/video:video_bitrate_allocator",
Reject invalid spatial index We should reject invalid values explicitly in order to prevent DCHECK failures later, which affect fuzzing progress. Bug: chromium:1009172, chromium:1009073 Change-Id: I7f0dc417ecac7aab076a652143f5face2ff98da2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156340 Commit-Queue: Kuang-che Wu <kcwu@google.com> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29459}
2019-10-10 19:54:45 +08:00
"../../api/video:video_codec_constants",
[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
"../../api/video:video_frame",
Include module_common_types.h only where needed Bug: None Change-Id: I73d493f8f186b429c7be808f4dfac0398f150931 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153891 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29277}
2019-09-23 10:31:16 +02:00
"../../api/video:video_frame_type",
VideoStreamEncoder report VideoLayersAllocation for simulcast Adds support for Vp8 simulcast. Bug: webrtc:12000 Change-Id: Ib24fd0542642b023ec35f7a7bdc4880d72365edf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187341 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32416}
2020-10-15 17:53:22 +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",
Reland "Move class VideoCodec from common_types.h to its own api header file." This is a reland of efc71e565e9b36bcdfb4571f59e34bbd8fabd0cd Differs from the original cl by not widening the type of VideoCodec::width and VideoCodec::height. Original change's description: > Move class VideoCodec from common_types.h to its own api header file. > > Bug: webrtc:7660 > Change-Id: I91f19bfc2565461328f30081f8383e136419aefb > Reviewed-on: https://webrtc-review.googlesource.com/79881 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23544} Bug: webrtc:7660 Change-Id: I7cf74a85a61ea2b831e6f32b3b3e17514ebefec8 Reviewed-on: https://webrtc-review.googlesource.com/82140 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23569}
2018-06-08 14:03:44 +02:00
"../../api/video_codecs:video_codecs_api",
Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1 Add new method OnRtpPacket, but leave ReceiveStatisticsImpl::IncomingPacket and most of the implementation unchanged. Deleting the old method and converting implementation from RTPHeader to RtpPacketreceived is planned for a followup, after downstream code is updated. Bug: webrtc:7135, webrtc:8016 Change-Id: I697ec12804618859f8d69415622d1b957e1d0847 Reviewed-on: https://webrtc-review.googlesource.com/100104 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24889}
2018-09-14 08:26:32 +02:00
"../../call:rtp_interfaces",
Add H264 bitstream rewriting to limit frame reordering marker in header The VUI part an SPS may specify max_num_reorder_frames and max_dec_frame_buffering. These may cause a decoder to buffer a number of frame prior allowing decode, leading to delays, even if no frames using such references (ie B-frames) are sent. Because of this we update any SPS block emitted by the encoder. Also, a bunch of refactoring of H264-related code to reduce code duplication. BUG= Review-Url: https://codereview.webrtc.org/1979443004 Cr-Commit-Position: refs/heads/master@{#13010}
2016-06-02 02:43:32 -07:00
"../../common_video",
Reland "Break up rtc_event_log_api to solve circular dependencies." This is a reland of 001546da953275c7a39eb220592b440c9b47d756 Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@google.com> > Cr-Commit-Position: refs/heads/master@{#21811} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 11:04:46 -08:00
"../../logging:rtc_event_audio",
"../../logging:rtc_event_rtp_rtcp",
Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h Bug: webrtc:5876 Change-Id: I0c92f9410fcf0832bfa321229b3437134255dba6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128085 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27190}
2019-03-19 14:10:16 +01:00
"../../modules/audio_coding:audio_coding_module_typedefs",
Move bit_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ia6e0f1c3017a968b22fdca512ef6adb37957d777 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258769 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36604}
2022-04-14 13:34:56 +02:00
"../../rtc_base:bit_buffer",
Migrate VideoRtpDepacketizerVp9 implementation from BitBuffer to BitstreamReader The new version is subjectivly cleaner and objectively generates smaller binary size Bug: None Change-Id: I662596c41f30690f7e81a51804dd7feb263510c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231130 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34935}
2021-09-06 14:18:05 +02:00
"../../rtc_base:bitstream_reader",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
Move byte_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic7e912cba1218f1eed794cb8c393ac148106b16c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259310 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36634}
2022-04-19 17:35:04 +02:00
"../../rtc_base:byte_buffer",
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",
Delete modules/rtp_rtcp local DivideRoundToNearest in favor on one in rtc_base To resolve a TODO Bug: None Change-Id: I90e10af24718e1aafd7e72076731b34c1110bb4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153524 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29230}
2019-09-18 18:51:34 +02:00
"../../rtc_base:divide_round",
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 remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:gtest_prod",
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 mod_ops out of rtc_base_approved Bug: webrtc:9838 Change-Id: I9d974e027f114098d87da368fad0c040c072a893 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258771 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36608}
2022-04-14 13:41:58 +02:00
"../../rtc_base:mod_ops",
Move one_time_event out of rtc_base_approved Bug: webrtc:9838 Change-Id: If85ce079e6a437337f1f780fcee0147b4e95f9cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258765 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36575}
2022-04-14 13:18:04 +02:00
"../../rtc_base:one_time_event",
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",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:rate_limiter",
Move rate_statistics out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ia5a45291a403097c0527d50ced85c3881653fa37 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259306 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36627}
2022-04-19 15:58:39 +02:00
"../../rtc_base:rate_statistics",
Reland of "Use moving median filters in RemoteNtpTimeEstimator, RtpToNtpEstimator"" Use moving median filters in RemoteNtpTimeEstimator, RtpToNtpEstimator If Webrtc-ClockEstimation experiment is enabled, median filtering is applied to results of RtpToNtpEstimator and RemoteNtpEstimator to smooth out random errors introduced by incorrect RTCP SR reports and networking delays. Bug: webrtc:8468 Change-Id: I592c4083fefc0dbdebe7b3ff30b92e95ba337595 NOTRY=TRUE NOPRESUBMIT=TRUE Change-Id: I592c4083fefc0dbdebe7b3ff30b92e95ba337595 Reviewed-on: https://webrtc-review.googlesource.com/23263 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20708}
2017-11-15 16:48:04 +00:00
"../../rtc_base:rtc_numerics",
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",
Separate build target for rtc_base/numerics/safe_minmax.h So that we can avoid dependency cycles. Bug: none Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8 Reviewed-on: https://webrtc-review.googlesource.com/60500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:18:56 +01:00
"../../rtc_base:safe_minmax",
Remove //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",
Use flat_map in RTCPReceiver RTCPReceiver initially used a std::map, which made RTCPReceiver::IncomingPacket's use of std::map represent ~0.45% CPU in highly loaded media servers. Using std::unordered_map in change 216321 reduced it only slightly, to 0.39%. This is the second attempt to reduce it even further. By using a flat_map and taking advantage of the increased cache locality, the hope is that it will be reduced. These maps generally have low cardinality (indexed by SSRC), and are looked up often, but modified less often, which make them a potential candidate for flat_map. Bug: webrtc:12689 Change-Id: I6733ccf3484d1c54e661250fb6712971b80fa2a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225203 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34432}
2021-07-06 23:14:51 +02:00
"../../rtc_base/containers:flat_map",
Limits size of payload padding packets to 2x target size. This CL also slightly refactors unit test, to test fewer things each. Bug: webrtc:11508 Change-Id: I98553d2b185364132c626d373683f38891e36c6d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173620 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31087}
2020-04-16 15:07:56 +02:00
"../../rtc_base/experiments:field_trial_parser",
Migrate modules/rtp_rtcp to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I4c71f3a28ef875af2c232b1b553840d6e21649d3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178804 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31645}
2020-07-07 11:44:28 +02:00
"../../rtc_base/synchronization:mutex",
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 a timer from ModuleRtpRtcpImpl2 that runs 100 times a second. The timer fired a Notify call that goes to an object that already receives callbacks for every packet from RtpSenderEgress. Further optimizations will be realized by moving ownership of the stats to the worker thread and then be able to remove locking in a few classes that currently are tied to those variables and the callbacks that previously did not come from the same thread consistently. We could furthermore get rid of one of these callback interfaces and just use one. Bug: webrtc:11581 Change-Id: I56ca5893c0153a87a4cbbe87d7741c39f9e66e52 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177422 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31575}
2020-06-27 17:44:55 +02:00
"../../rtc_base/task_utils:repeating_task",
Move timestamp_extrapolator.h to rtc_base/time/ This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. BUG=webrtc:8445 Change-Id: I51dfe8879c28c91bd1c667fc47b4892373671e0f Reviewed-on: https://webrtc-review.googlesource.com/21540 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22569}
2018-03-22 15:29:03 +01:00
"../../rtc_base/time:timestamp_extrapolator",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"../../system_wrappers",
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:metrics",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"../remote_bitrate_estimator",
[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
"../video_coding:codec_globals_headers",
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 Abseil container algorithms in modules/rtp_rtcp/ Bug: None Change-Id: Ica2e9795ec6195e044403f5ee25e476f6c47cf93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129600 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27361}
2019-03-28 10:56:11 -07:00
"//third_party/abseil-cpp/absl/algorithm:container",
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
"//third_party/abseil-cpp/absl/base:core_headers",
in RtpPacketizerVp8 factor out payload splitter function so that it can be shared between different packetizers and thus easier to extend Bug: webrtc:9680 Change-Id: Ie5e904ad27afb8dd2ed35ef9e009f7f408017b2f Reviewed-on: https://webrtc-review.googlesource.com/97661 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24555}
2018-09-04 16:11:58 +02:00
"//third_party/abseil-cpp/absl/container:inlined_vector",
Add implementations of the VideoRtpDepacketizer interface while suboptimal, these implementions are complete and allow to swap code from using RtpDepacketizer interface to VideoRtpDepacketizer Bug: webrtc:11152 Change-Id: Ie7823feeb5b0563b74754255aaedfada9d446ac5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161380 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30031}
2019-12-05 16:24:04 +01:00
"//third_party/abseil-cpp/absl/memory",
Delete _strnicmp. Uses replaced with abseil functions. The replacements are absl::EqualsIgnoreCase and absl::StartsWithIgnoreCase. Also delete the alias RtpUtility::StringCompare. Bug: webrtc:6424 Change-Id: I4bed71540264450f85137ad0c2564125c5c6213f Reviewed-on: https://webrtc-review.googlesource.com/c/109006 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25481}
2018-11-02 10:54:56 +01:00
"//third_party/abseil-cpp/absl/strings",
Replace rtc::Optional with absl::optional in modules/rtp_rtcp This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated using script: #!/bin/bash dir=modules/rtp_rtcp find $dir -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $dir -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc Reviewed-on: https://webrtc-review.googlesource.com/83584 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-14 12:59:38 +02:00
"//third_party/abseil-cpp/absl/types:optional",
[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
"//third_party/abseil-cpp/absl/types:variant",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
]
}
GN target for test_packet_masks_metrics. NOTRY=True BUG=webrtc:6193 Review-Url: https://codereview.webrtc.org/2229473002 Cr-Commit-Position: refs/heads/master@{#13672}
2016-08-08 08:08:53 -07:00
Add new empty build targets rtp_rtcp_legacy and video_legacy. Initial step to be able to land https://webrtc-review.googlesource.com/c/src/+/196904 Bug: webrtc:11581 Change-Id: Iaab52e98f4562f701cf02e3f641b7b02a11b799e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197944 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32971}
2020-12-16 15:57:49 +01:00
rtc_source_set("rtp_rtcp_legacy") {
Move deprecated code to their own build targets. Moves the deprecated version of RtpRtcp module, and related classes in video/. Bug: webrtc:11581 Change-Id: Icc4cedb844fcd7c7372e8a907e5252f5b4fd955e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196904 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33025}
2021-01-14 12:00:57 +01:00
sources = [
"include/rtp_rtcp.h",
"source/deprecated/deprecated_rtp_sender_egress.cc",
"source/deprecated/deprecated_rtp_sender_egress.h",
"source/rtp_rtcp_impl.cc",
"source/rtp_rtcp_impl.h",
]
deps = [
":rtp_rtcp",
":rtp_rtcp_format",
"..:module_fec_api",
"../../api:rtp_headers",
"../../api:transport_api",
"../../api/rtc_event_log",
"../../api/transport:field_trial_based_config",
"../../api/units:data_rate",
Add Timestamp -> AbsoluteSendTime conversion function instead of ms -> AbsoluteSendTime helper Bug: webrtc:13757 Change-Id: I57389a66a43b4f4838023f9c224a985f2cd57107 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266024 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37350}
2022-06-17 19:07:30 +02:00
"../../api/units:timestamp",
Move deprecated code to their own build targets. Moves the deprecated version of RtpRtcp module, and related classes in video/. Bug: webrtc:11581 Change-Id: Icc4cedb844fcd7c7372e8a907e5252f5b4fd955e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196904 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33025}
2021-01-14 12:00:57 +01:00
"../../api/video:video_bitrate_allocation",
"../../logging:rtc_event_rtp_rtcp",
"../../rtc_base:checks",
"../../rtc_base:gtest_prod",
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 rate_statistics out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ia5a45291a403097c0527d50ced85c3881653fa37 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259306 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36627}
2022-04-19 15:58:39 +02:00
"../../rtc_base:rate_statistics",
Move deprecated code to their own build targets. Moves the deprecated version of RtpRtcp module, and related classes in video/. Bug: webrtc:11581 Change-Id: Icc4cedb844fcd7c7372e8a907e5252f5b4fd955e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196904 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33025}
2021-01-14 12:00:57 +01:00
"../../rtc_base/synchronization:mutex",
Add remote-outbound stats for audio streams Add missing members needed to surface `RTCRemoteOutboundRtpStreamStats` via `ChannelReceive::GetRTCPStatistics()` - i.e., audio streams. `GetSenderReportStats()` is added to both `ModuleRtpRtcpImpl` and `ModuleRtpRtcpImpl2` and used by `ChannelReceive::GetRTCPStatistics()`. Bug: webrtc:12529 Change-Id: Ia8f5dfe2e4cfc43e3ddd28f2f1149f5c00f9269d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211041 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33452}
2021-03-12 17:45:26 +01:00
"../../system_wrappers",
Move deprecated code to their own build targets. Moves the deprecated version of RtpRtcp module, and related classes in video/. Bug: webrtc:11581 Change-Id: Icc4cedb844fcd7c7372e8a907e5252f5b4fd955e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196904 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33025}
2021-01-14 12:00:57 +01:00
"../remote_bitrate_estimator",
]
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
Add new empty build targets rtp_rtcp_legacy and video_legacy. Initial step to be able to land https://webrtc-review.googlesource.com/c/src/+/196904 Bug: webrtc:11581 Change-Id: Iaab52e98f4562f701cf02e3f641b7b02a11b799e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197944 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32971}
2020-12-16 15:57:49 +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("rtcp_transceiver") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
public = [
Add Thread-safe wrapper for RtcpTransceiver Bug: webrtc:8239 Change-Id: I4cc2f7f2b27c764e1aae734f933902102b345614 Reviewed-on: https://webrtc-review.googlesource.com/21680 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20714}
2017-11-16 14:35:32 +01:00
"source/rtcp_transceiver.h",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
"source/rtcp_transceiver_config.h",
"source/rtcp_transceiver_impl.h",
]
sources = [
Add Thread-safe wrapper for RtcpTransceiver Bug: webrtc:8239 Change-Id: I4cc2f7f2b27c764e1aae734f933902102b345614 Reviewed-on: https://webrtc-review.googlesource.com/21680 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20714}
2017-11-16 14:35:32 +01:00
"source/rtcp_transceiver.cc",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
"source/rtcp_transceiver_config.cc",
"source/rtcp_transceiver_impl.cc",
]
deps = [
":rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
":rtp_rtcp_format",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
"../../api:array_view",
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",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
"../../api:transport_api",
Migrate RtcpTransceiver to use webrtc::TaskQueueBase instead of rtc::TaskQueue This changes removes an extra layer of indirection since RtcpTransceiver doesn't own TaskQueue it uses. Bug: None Change-Id: Ie1ef4cd8c3fb18a8e0b7ddaf0d6a319392b9e9f7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126040 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30704}
2020-03-05 15:43:24 +01:00
"../../api/task_queue",
In RtcpTransceiver add support for receiving network generic messages These message suppose to extract all information NetworkControllerInterface may need from rtcp. Bug: webrtc:8239 Change-Id: I21d9081ad147ca8abe1ae05ca7201568c6ff77d1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230421 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34876}
2021-08-30 14:42:58 +02:00
"../../api/units:data_rate",
"../../api/units:time_delta",
Use the injected clock in rtcp_transciever. Bug: webrtc:11327 Change-Id: Idb02842f2eb679f972c0449a01a81a26ceb85827 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219789 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Paul Hallak <phallak@google.com> Cr-Commit-Position: refs/heads/master@{#34080}
2021-05-21 18:08:04 +02:00
"../../api/units:timestamp",
Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets Bug: webrtc:9271 Change-Id: Ic7415830588bef9d87bab92943460207890dada6 Reviewed-on: https://webrtc-review.googlesource.com/76960 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 12:35:56 +02:00
"../../api/video:video_bitrate_allocation",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
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",
in RtcpTransceiver implement sending rtcp sender reports Bug: webrtc:8239 Change-Id: Id3298bf4e0eb18a3fc8072fb19416e67a126705f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249788 Reviewed-by: Emil Lundmark <lndmrk@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35995}
2022-02-14 15:24:30 +01:00
"../../rtc_base:divide_round",
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: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: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",
rtp_rtcp: use webrtc::flat_map for remote_senders This one is frequently accessed - Mainly by ::CreateReportBlocks and is visible in performance profiles (although not very much). By using webrtc::flat_map, better data cache locality is expected. Bug: webrtc:12689 Change-Id: Ic2ebcad806788074b2b4cb244a25395a48df1852 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232541 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35054}
2021-09-20 20:47:26 +02:00
"../../rtc_base/containers:flat_map",
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",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
Use absl_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 Abseil container algorithms in modules/rtp_rtcp/ Bug: None Change-Id: Ica2e9795ec6195e044403f5ee25e476f6c47cf93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129600 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27361}
2019-03-28 10:56:11 -07:00
"//third_party/abseil-cpp/absl/algorithm:container",
Reland "Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code" This is a reland of 8c2250eddc7263036397179a0794b9b17d7afb38 Original change's description: > Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code > > Bug: webrtc:12336 > Change-Id: If76f00d0883b5c8a90d3ef5554f5e22384b3fb58 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197620 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32978} Bug: webrtc:12336 Change-Id: I1cd017d45c1578528dec4532345950e9823f4a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201732 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33003}
2021-01-14 16:15:31 +01:00
"//third_party/abseil-cpp/absl/base:core_headers",
Migrate rtp_rtcp to absl::AnyInvocable based TaskQueueBase interface Bug: webrtc:14245 Change-Id: I037f964130648caf0bd1de86611f8681d475b078 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268146 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37481}
2022-07-07 13:34:25 +02:00
"//third_party/abseil-cpp/absl/cleanup",
"//third_party/abseil-cpp/absl/functional:any_invocable",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Replace rtc::Optional with absl::optional in modules/rtp_rtcp This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated using script: #!/bin/bash dir=modules/rtp_rtcp find $dir -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $dir -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc Reviewed-on: https://webrtc-review.googlesource.com/83584 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-14 12:59:38 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
]
}
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_video_header") {
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. Bug: none Change-Id: If28f57c5ae250afbb47c5d20c9854e9a11182642 Reviewed-on: https://webrtc-review.googlesource.com/87561 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23904}
2018-07-09 15:52:29 +02:00
visibility = [ "*" ]
sources = [
"source/rtp_video_header.cc",
"source/rtp_video_header.h",
]
deps = [
Allow providing the absolute capture time extension when packetizing a frame. Bug: b/150859541 Change-Id: Iffb6ee84f49ffa64fdb0633248864d2dfd6e9ff3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234868 Commit-Queue: Paul Hallak <phallak@google.com> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35194}
2021-10-13 12:44:02 +02:00
"../../api:rtp_headers",
in RtpSenderVideo add support for writing DependencyDescriptor header extension Bug: webrtc:10342 Change-Id: I12cca9c5e1606338bb914e58e13d268bbc6961f9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166532 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30427}
2020-01-30 16:28:53 +01:00
"../../api/transport/rtp:dependency_descriptor",
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. Bug: none Change-Id: If28f57c5ae250afbb47c5d20c9854e9a11182642 Reviewed-on: https://webrtc-review.googlesource.com/87561 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23904}
2018-07-09 15:52:29 +02:00
"../../api/video:video_frame",
Move frame_type member from RtpDepacketizer::ParsedPayload to RTPVideoHeader The latter is also a member of the former. This cleanup is also a preparation for dropping WebRtcRTPHeader::frameType (or deleting WebRtcRTPHeader right away), now that it's a video-specific member. Tbr: kwiberg@webrtc.org # Comment change in modules/include/ Bug: None Change-Id: I5c1f3f981f0d750713fc9b9b145278150fe32b5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133024 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27740}
2019-04-24 09:41:16 +02:00
"../../api/video:video_frame_type",
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",
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. Bug: none Change-Id: If28f57c5ae250afbb47c5d20c9854e9a11182642 Reviewed-on: https://webrtc-review.googlesource.com/87561 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23904}
2018-07-09 15:52:29 +02:00
"../../modules/video_coding:codec_globals_headers",
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 members for the codec agnostic descriptor to RTPVideoHeader. TBR=danilchap@webrtc.org Bug: webrtc:9361, webrtc:9582 Change-Id: I0303fc89bafab59e68ec81979e0e4372e79a4f51 Reviewed-on: https://webrtc-review.googlesource.com/91866 Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24170}
2018-08-01 17:13:08 +02:00
"//third_party/abseil-cpp/absl/container:inlined_vector",
[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
"//third_party/abseil-cpp/absl/types:optional",
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. Bug: none Change-Id: If28f57c5ae250afbb47c5d20c9854e9a11182642 Reviewed-on: https://webrtc-review.googlesource.com/87561 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23904}
2018-07-09 15:52:29 +02:00
"//third_party/abseil-cpp/absl/types:variant",
]
}
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("fec_test_helper") {
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
testonly = true
sources = [
"source/fec_test_helper.cc",
"source/fec_test_helper.h",
]
deps = [
":rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
":rtp_rtcp_format",
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 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",
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
]
}
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("mock_rtp_rtcp") {
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
testonly = true
Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets" This reverts commit 01aa210fad68f1006528d32d388b307c22990734. Reason for revert: downstream project adjusted Original change's description: > Revert "Enable and fix chromium clang warnings in rtp_rtcp test targets" > > This reverts commit 9486b117daac09c9f7ac8450ccda835938cf3150. > > Reason for revert: Breaks downstream project > > Original change's description: > > Enable and fix chromium clang warnings in rtp_rtcp test targets > > > > Bug: webrtc:163 > > Change-Id: I4ed3e63296d8bf06536a83196d597c7a906ba11c > > Reviewed-on: https://webrtc-review.googlesource.com/60802 > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22357} > > TBR=danilchap@webrtc.org,phoglund@webrtc.org,terelius@webrtc.org > > Change-Id: I2c3777ea9f26813bdb395e7fd68f6b49443586ea > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:163 > Reviewed-on: https://webrtc-review.googlesource.com/61060 > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22365} TBR=danilchap@webrtc.org,phoglund@webrtc.org,oprypin@webrtc.org,terelius@webrtc.org Change-Id: I0b4cb6d05b37caeb52cca9abf95417ad3ad6f76b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:163 Reviewed-on: https://webrtc-review.googlesource.com/61080 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22368}
2018-03-09 15:37:03 +00:00
public = [
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"mocks/mock_recovered_packet_receiver.h",
Using mock transport controller in audio unit tests. Using a mock of rtp transport controller send in audio send stream unit tests. This reduces the dependencies and makes the tests more focused on testing the functionality of audio send stream itself. Bug: webrtc:8415 Change-Id: Ia8d9cf47d93decc74b10ca75a6771f39df658dc2 Reviewed-on: https://webrtc-review.googlesource.com/56600 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22161}
2018-02-22 14:49:02 +01:00
"mocks/mock_rtcp_bandwidth_observer.h",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"mocks/mock_rtcp_rtt_stats.h",
"mocks/mock_rtp_rtcp.h",
]
deps = [
":rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
":rtp_rtcp_format",
Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets Bug: webrtc:9271 Change-Id: Ic7415830588bef9d87bab92943460207890dada6 Reviewed-on: https://webrtc-review.googlesource.com/76960 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 12:35:56 +02:00
"../../api/video:video_bitrate_allocation",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../../test:test_support",
]
Adopt absl::string_view in modules/rtp_rtcp Bug: webrtc:13579 Change-Id: Ic4e1431bedc69492358cb2e3749b50a941306f44 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262250 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36887}
2022-05-13 11:42:16 +02:00
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
}
AV1 OBU test helper. Bug: none Change-Id: I942319122f823e18e500c049274527b00e6feba6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223061 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34363}
2021-06-23 13:52:06 +02:00
rtc_library("rtp_packetizer_av1_test_helper") {
testonly = true
sources = [
"source/rtp_packetizer_av1_test_helper.cc",
"source/rtp_packetizer_av1_test_helper.h",
]
}
GN target for test_packet_masks_metrics. NOTRY=True BUG=webrtc:6193 Review-Url: https://codereview.webrtc.org/2229473002 Cr-Commit-Position: refs/heads/master@{#13672}
2016-08-08 08:08:53 -07:00
if (rtc_include_tests) {
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_executable("test_packet_masks_metrics") {
testonly = true
GN target for test_packet_masks_metrics. NOTRY=True BUG=webrtc:6193 Review-Url: https://codereview.webrtc.org/2229473002 Cr-Commit-Position: refs/heads/master@{#13672}
2016-08-08 08:08:53 -07:00
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 = [
"test/testFec/average_residual_loss_xor_codes.h",
"test/testFec/test_packet_masks_metrics.cc",
]
GN target for test_packet_masks_metrics. NOTRY=True BUG=webrtc:6193 Review-Url: https://codereview.webrtc.org/2229473002 Cr-Commit-Position: refs/heads/master@{#13672}
2016-08-08 08:08:53 -07:00
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
deps = [
":rtp_rtcp",
"../../test:fileutils",
"../../test:test_main",
"../../test:test_support",
"//testing/gtest",
]
} # test_packet_masks_metrics
}
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
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_rtcp_modules_tests") {
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
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
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
sources = [ "test/testFec/test_fec.cc" ]
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
deps = [
":rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
":rtp_rtcp_format",
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 the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"../../test:test_support",
]
}
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_rtcp_unittests") {
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
testonly = true
sources = [
Make local to capturer clock offset a separate entry in PacketInfo. This also changes the meaning of |estimated_capture_clock_offset| in |absolute_capture_time_| to become a remote to capturer clock offset. Bug: chromium:1056230, webrtc:10739 Change-Id: Id658590e027bbe77ae0834ea224e1dc977a305f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219163 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/master@{#34067}
2021-05-19 14:38:25 +02:00
"source/absolute_capture_time_interpolator_unittest.cc",
Add helper classes to send and receive abs-capture-time extensions. This change adds helper classes to manipulate Absolute Capture Time header extensions. Both classes support the "timestamp interpolation" optimization. Bug: webrtc:10739 Change-Id: I08eff46eb8910842a6dbaa3288b976004fabe1c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149801 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28936}
2019-08-22 10:05:01 +02:00
"source/absolute_capture_time_sender_unittest.cc",
Add helper for DependencyDescriptor rtp header extension to decide when to set active_decode_target_bitmask field Bug: webrtc:10342 Change-Id: I348d7467a72b45651455f4574fe8fda3c77ebbae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177400 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31555}
2020-06-22 17:53:01 +02:00
"source/active_decode_targets_helper_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/byte_io_unittest.cc",
Make local to capturer clock offset a separate entry in PacketInfo. This also changes the meaning of |estimated_capture_clock_offset| in |absolute_capture_time_| to become a remote to capturer clock offset. Bug: chromium:1056230, webrtc:10739 Change-Id: Id658590e027bbe77ae0834ea224e1dc977a305f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219163 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/master@{#34067}
2021-05-19 14:38:25 +02:00
"source/capture_clock_offset_updater_unittest.cc",
Move the FEC private tables into .cc files. Change the arrays to be continuous uint8_t arrays instead being of arrays of arrays (of arrays). Code size difference is 17K for arm, ~42K for arm64. New lookup algorithm, tailored for these two tables + tests. Instead of returning a raw pointer into the table, the algorithm returns an ArrayView, which includes size information for how much memory can be read. Change-Id: I000b094520bac944e518eb8b51d8dbef4670f5d7 Bug: webrtc:9102 Reviewed-on: https://webrtc-review.googlesource.com/65920 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22736}
2018-04-04 13:48:17 +02:00
"source/fec_private_tables_bursty_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/flexfec_header_reader_writer_unittest.cc",
"source/flexfec_receiver_unittest.cc",
"source/flexfec_sender_unittest.cc",
"source/nack_rtx_unittest.cc",
"source/packet_loss_stats_unittest.cc",
Refactor PacketSequencer in preparation for deferred sequencing. This CL is extracted from https://webrtc-review.googlesource.com/c/src/+/208584 PacketSequencer now has its own unit tests. They are maybe somewhat redundant with a few RtpSender unit tests, but will defer cleanup to a later CL. Bug: webrtc:11340, webrtc:12470 Change-Id: I1c31004b85ae075ddc696bdf1100d2a5044d4ef5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227343 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34638}
2021-08-03 20:24:13 +02:00
"source/packet_sequencer_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/receive_statistics_unittest.cc",
"source/remote_ntp_time_estimator_unittest.cc",
Remove rtcp_utility as mostly unused. Since the only used class is RTCPUtilitiy::NackStats, rename it to RtcpNackStats and move it into dedicated file. BUG=webrtc:5565 Review-Url: https://codereview.webrtc.org/2680183004 Cr-Commit-Position: refs/heads/master@{#16515}
2017-02-09 05:21:42 -08:00
"source/rtcp_nack_stats_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/rtcp_packet/app_unittest.cc",
"source/rtcp_packet/bye_unittest.cc",
"source/rtcp_packet/common_header_unittest.cc",
"source/rtcp_packet/compound_packet_unittest.cc",
"source/rtcp_packet/dlrr_unittest.cc",
"source/rtcp_packet/extended_reports_unittest.cc",
"source/rtcp_packet/fir_unittest.cc",
Create new RTCP feedback message - LossIndication Create a new RTCP feedback message for reporting the loss and/or non-decodability of video frames, to be used by the upcoming injectable VideoFrameBufferController. The new feedback message should report: 1. The sequence number of the last decoded non-discardable video frame. (TBD: If a multi-packet frame, should it be the sequence number of the first, last, or any of the packets?) 2. The sequence number of the last received RTP packet in the stream. 3. A decodability flag, whose specific meaning depends on the last-received RTP sequence number. The decodability flag is true if and only if all of the frame's dependencies are known to be decodable, and the frame itself is not yet known to be unassemblable. * Clarification #1: In a multi-packet frame, the first packet's dependencies are known, but it is not yet known whether all parts of the current frame will be received. * Clarification #2: In a multi-packet frame, the dependencies would be unknown if the first packet was not received. Then, the packet will be known-unassemblable. Bug: webrtc:10226 Change-Id: I1563c944477e3ed40235e82ab99a439414632aff Reviewed-on: https://webrtc-review.googlesource.com/c/118931 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26387}
2019-01-24 12:38:36 +01:00
"source/rtcp_packet/loss_notification_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/rtcp_packet/nack_unittest.cc",
"source/rtcp_packet/pli_unittest.cc",
"source/rtcp_packet/rapid_resync_request_unittest.cc",
"source/rtcp_packet/receiver_report_unittest.cc",
"source/rtcp_packet/remb_unittest.cc",
Adds remote estimate RTCP packet. This adds the RemoteEstimate rtcp packet and wires it up to GoogCC where it's used to improve congestion controller behavior. The functionality is negotiated using SDP. It's added with a field trial that allow disabling the functionality in case there's any issues. Bug: webrtc:10742 Change-Id: I1ea8e4216a27cd2b00505c99b42d1e38726256c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146602 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28654}
2019-07-24 11:38:03 +02:00
"source/rtcp_packet/remote_estimate_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/rtcp_packet/report_block_unittest.cc",
"source/rtcp_packet/rrtr_unittest.cc",
"source/rtcp_packet/sdes_unittest.cc",
"source/rtcp_packet/sender_report_unittest.cc",
"source/rtcp_packet/target_bitrate_unittest.cc",
"source/rtcp_packet/tmmbn_unittest.cc",
"source/rtcp_packet/tmmbr_unittest.cc",
"source/rtcp_packet/transport_feedback_unittest.cc",
"source/rtcp_packet_unittest.cc",
"source/rtcp_receiver_unittest.cc",
"source/rtcp_sender_unittest.cc",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
"source/rtcp_transceiver_impl_unittest.cc",
Add Thread-safe wrapper for RtcpTransceiver Bug: webrtc:8239 Change-Id: I4cc2f7f2b27c764e1aae734f933902102b345614 Reviewed-on: https://webrtc-review.googlesource.com/21680 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20714}
2017-11-16 14:35:32 +01:00
"source/rtcp_transceiver_unittest.cc",
in DependencyDescriptor writer zero remaing bytes adjust zeroing to support more than 64 bits. Bug: b/156802687 Change-Id: I42448b4dd6d5c04143eb9075cd61317e115ed936 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175564 Reviewed-by: Erik Varga <erikvarga@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31300}
2020-05-18 12:50:01 +02:00
"source/rtp_dependency_descriptor_extension_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/rtp_fec_unittest.cc",
"source/rtp_format_h264_unittest.cc",
in RtpPacketizerVp8 factor out payload splitter function so that it can be shared between different packetizers and thus easier to extend Bug: webrtc:9680 Change-Id: Ie5e904ad27afb8dd2ed35ef9e009f7f408017b2f Reviewed-on: https://webrtc-review.googlesource.com/97661 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24555}
2018-09-04 16:11:58 +02:00
"source/rtp_format_unittest.cc",
Fix packetization logic to leave space for extensions in the last packet Change packetizer interface to explicitly return number of packets instead of a last flag. Account for extra space needed in the last packet. BUG=webrtc:7588,webrtc:7594 Review-Url: https://codereview.webrtc.org/2871173008 Cr-Commit-Position: refs/heads/master@{#18244}
2017-05-23 09:34:21 -07:00
"source/rtp_format_video_generic_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/rtp_format_vp8_test_helper.cc",
"source/rtp_format_vp8_test_helper.h",
"source/rtp_format_vp8_unittest.cc",
"source/rtp_format_vp9_unittest.cc",
Add Parsing/Building generic frame descriptor extension Bug: webrtc:9361 Change-Id: I7e85826117348e2d4f4726e8d515bb1d4a289966 Reviewed-on: https://webrtc-review.googlesource.com/83622 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23662}
2018-06-19 14:24:17 +02:00
"source/rtp_generic_frame_descriptor_extension_unittest.cc",
Rename rtp_header_extension.h to rtp_header_extension_map.h Move it to include path of the rtp_rtcp module to indicate it is ok to include it outside of the module. Renamed to match the class it introduce and to reduce confusion with rtp_header_extensions.h Bug: webrtc:5565 Change-Id: Ic4b4e9f6b75cb9275e23539cd6e88632c1e7c8d2 Reviewed-on: https://chromium-review.googlesource.com/520947 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18402}
2017-06-01 17:10:51 +02:00
"source/rtp_header_extension_map_unittest.cc",
Move RtpHeaderExtensionMap::GetTotalLengthInBytes into own file Rename to better match what it does, Adjust to support two-byte header extension Bug: webrtc:7990 Change-Id: I2786d70e7cf9cd3d722f54fb1d07c9cfaafab947 Reviewed-on: https://webrtc-review.googlesource.com/103201 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24958}
2018-10-03 10:15:36 +02:00
"source/rtp_header_extension_size_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/rtp_packet_history_unittest.cc",
"source/rtp_packet_unittest.cc",
Implement AV1 RtpPacketizer class Bug: webrtc:11042 Change-Id: Id1fc0acfa87a4520344f2636f50cb4d4e7284829 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160416 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29947}
2019-11-28 13:42:50 +01:00
"source/rtp_packetizer_av1_unittest.cc",
Add thread/sequence checks to ModuleRtpRtcpImpl. This ended up with needing to fork the current implementation in order to not break downstream projects that were inheriting from it. While those get updated, we'll move on with the forked class. Bug: webrtc:11581,b/8278269 Change-Id: I05b596cbda71aa5b72894c31a7119d17d4761883 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175500 Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31334}
2020-05-20 09:32:51 +02:00
"source/rtp_rtcp_impl2_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/rtp_rtcp_impl_unittest.cc",
Move RtpSenderAudioTest to its own file Update RtpSenderAudioTest to call methods on RTPSenderAudio rather than RTPSender, when possible. In particular, avoid RTPSender::SendOutgoingData. Drop parameterization on the WebRTC-SendSideBwe-WithOverhead field trial, since that appears unrelated to these tests. Also delete some unused parts of the RtpSender test. Bug: webrtc:7135 Change-Id: I535bf48bb1720e2727f4a62fa3e49b2bb84394a0 Reviewed-on: https://webrtc-review.googlesource.com/c/120920 Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26516}
2019-02-01 14:13:29 +01:00
"source/rtp_sender_audio_unittest.cc",
Add pure RtpSenderEgress unit test fixture. The extracts and refactors some test code from rtp_sender_unittest.cc and puts it in a new target intended to only test RtpSenderEgress, and do it as pure unit test, rather than the unholy not-quite-unit-not-quite-integration-test thingy we have today. Only a first test case is actually ported with this CL, but it's a start... Bug: webrtc:11340 Change-Id: Ie2cdde63a00a6ff6eba7b8d443eeb76ce2a527c9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216180 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33841}
2021-04-26 17:14:27 +02:00
"source/rtp_sender_egress_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/rtp_sender_unittest.cc",
Move RtpSenderVideo tests to separate file. Also refactor most of the RtpSender tests to not use the frame-level method RTPSender::SendOutgoingData. Bug: webrtc:7135 Change-Id: I9b0af6aa45e9b908d8197e48b143fede7e2804c7 Reviewed-on: https://webrtc-review.googlesource.com/c/121461 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26577}
2019-02-06 18:23:44 +01:00
"source/rtp_sender_video_unittest.cc",
Introduce RtpSequenceNumberMap The video encoder should be informed of incoming LossNotification RTCP messages. Since it is unaware of RTP sequence numbers, or anything else RTP-related, the sequence numbers mentioned in the RTCP message must first be mapped to timestamps, since those are meaningful to the encoder. This CL introduces RtpSequenceNumberMap, which maps RTP sequence numbers to timestamps, while providing: 1. Capping the number of entries. 2. Wrap-around handling. RtpSequenceNumberMap also remembers which packets were first and/or last in the frame. Later CLs will wire this up. Bug: webrtc:10501 Change-Id: Ie0662cdb5706a3bcf63aa2934816a9df88439357 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130497 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27448}
2019-04-04 11:43:54 +02:00
"source/rtp_sequence_number_map_unittest.cc",
Unify helpers IsRtpPacket and IsRtcpPacket Bug: None Change-Id: Ibe942de433435d256cd6827440136936d4b274d6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225022 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34419}
2021-07-05 19:06:17 +02:00
"source/rtp_util_unittest.cc",
Add experimental extension RtpVideoLayersAllocation The extension is suggested to be used for signaling per target bitrate, resolution and frame rate to a SFU to allow a SFU to know what video layers a client is currently targeting. It is hoped to replace the current Target bitrate RTCP XR message currently used only for screen share. Bug: webrtc:12000 Change-Id: Id7b55e7ddaf6304e31839fd0482b096e1dbe8925 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185980 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32313}
2020-10-05 13:51:47 +02:00
"source/rtp_video_layers_allocation_extension_unittest.cc",
Add a tracker for RTCRtpContributingSource and RTCRtpSynchronizationSource. This change adds a new SourceTracker class that can do spec-compliant tracking of RTCRtpContributingSource and RTCRtpSynchronizationSource when frames are delivered to the RTCRtpReceiver's MediaStreamTrack for playout. It will replace the existing spec-incompliant ContributingSources. Bug: webrtc:10545 webrtc:10668 Change-Id: I961adaba09d6337f2f36b301a4fabcd20de65271 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140948 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28249}
2019-06-12 12:13:22 +02:00
"source/source_tracker_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"source/time_util_unittest.cc",
"source/ulpfec_generator_unittest.cc",
"source/ulpfec_header_reader_writer_unittest.cc",
"source/ulpfec_receiver_unittest.cc",
Change Av1 depacketizer to implement VideoRtpDepacketizer interface Bug: webrtc:11152 Change-Id: I322115263f60439bee36277157a0acef9bd28e3e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165343 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30260}
2020-01-10 17:26:53 +01:00
"source/video_rtp_depacketizer_av1_unittest.cc",
Add VideoRtpDepacketizerGeneric Bug: webrtc:11152 Change-Id: I27d6a62093d36a4e77dd35d4c115af8cdcc0178a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162202 Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30160}
2019-12-23 13:14:28 +01:00
"source/video_rtp_depacketizer_generic_unittest.cc",
Move RtpDepacketizerH264 into own files Bug: webrtc:11152 Change-Id: Iab4975e9f378b177a2abf34559f9b74752e69843 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165582 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30212}
2020-01-09 18:55:02 +01:00
"source/video_rtp_depacketizer_h264_unittest.cc",
Add implementations of the VideoRtpDepacketizer interface while suboptimal, these implementions are complete and allow to swap code from using RtpDepacketizer interface to VideoRtpDepacketizer Bug: webrtc:11152 Change-Id: Ie7823feeb5b0563b74754255aaedfada9d446ac5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161380 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30031}
2019-12-05 16:24:04 +01:00
"source/video_rtp_depacketizer_raw_unittest.cc",
Move vp8 rtp depacketization to VideoRtpDepacketizerVp8 Bug: webrtc:11152 Change-Id: Ic2b7fd091cb4d095ce29fbe06196f6424c08fce1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161451 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30088}
2019-12-11 11:24:37 +01:00
"source/video_rtp_depacketizer_vp8_unittest.cc",
Move vp9 rtp depacketization to VideoRtpDepacketizerVp9 Bug: webrtc:11152 Change-Id: I560d4cd62fabae093e3df592f0e7cc4001c10657 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162420 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30102}
2019-12-16 13:19:28 +01:00
"source/video_rtp_depacketizer_vp9_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 = [
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
":fec_test_helper",
":mock_rtp_rtcp",
Create skeleton of the rtcp transceiver. RtcpTransceiver name reserved for thread-safe version that planned to be wrapper of the RtcpTransceiverImpl BUG=webrtc:8239 Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f Reviewed-on: https://webrtc-review.googlesource.com/7920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 17:07:05 +02:00
":rtcp_transceiver",
AV1 OBU test helper. Bug: none Change-Id: I942319122f823e18e500c049274527b00e6feba6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223061 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34363}
2021-06-23 13:52:06 +02:00
":rtp_packetizer_av1_test_helper",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
":rtp_rtcp_format",
Add new empty build targets rtp_rtcp_legacy and video_legacy. Initial step to be able to land https://webrtc-review.googlesource.com/c/src/+/196904 Bug: webrtc:11581 Change-Id: Iaab52e98f4562f701cf02e3f641b7b02a11b799e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197944 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32971}
2020-12-16 15:57:49 +01:00
":rtp_rtcp_legacy",
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",
Restructure RtcpTransceiver tests to use fake time instead of real time to make tests faster and more determenistic. Bug: webrtc:8239 Change-Id: I18067251a1f1a349fda28bbfbb59bce333bfddca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201737 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36938}
2022-05-19 15:07:56 +02:00
"../../api:create_time_controller",
Move RtpExtension to api/ directory and config.h/.cc to call/. BUG=webrtc:5876 R=deadbeef@webrtc.org, solenberg@webrtc.org Review-Url: https://codereview.webrtc.org/3004723002 . Cr-Commit-Position: refs/heads/master@{#19639}
2017-09-01 15:29:28 +02:00
"../../api:libjingle_peerconnection_api",
Authenticate video header when dependency descriptor is sent same way as generic frame descriptor is authenticated. Bug: webrtc:10342 Change-Id: I50bb3ab343d66f1f628083183444da6e338f7db9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168681 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30578}
2020-02-20 15:01:15 +01:00
"../../api:mock_frame_encryptor",
Add a tracker for RTCRtpContributingSource and RTCRtpSynchronizationSource. This change adds a new SourceTracker class that can do spec-compliant tracking of RTCRtpContributingSource and RTCRtpSynchronizationSource when frames are delivered to the RTCRtpReceiver's MediaStreamTrack for playout. It will replace the existing spec-incompliant ContributingSources. Bug: webrtc:10545 webrtc:10668 Change-Id: I961adaba09d6337f2f36b301a4fabcd20de65271 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140948 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28249}
2019-06-12 12:13:22 +02:00
"../../api:rtp_headers",
"../../api:rtp_packet_info",
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",
Restructure RtcpTransceiver tests to use fake time instead of real time to make tests faster and more determenistic. Bug: webrtc:8239 Change-Id: I18067251a1f1a349fda28bbfbb59bce333bfddca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201737 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36938}
2022-05-19 15:07:56 +02:00
"../../api: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
"../../api:transport_api",
Stop using rtc_event.h forward header Bug: webrtc:10206 Change-Id: I16905ec745673178195d6715fda6175c31500163 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151601 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29149}
2019-09-05 11:29:59 +02:00
"../../api/rtc_event_log",
Replace field trials with WebRtcKeyValueConfig in RtpRtcpModule Replaces use of field trials in RtpSender and RtpVideoSender with injectable WebRtcKeyValueConfig. Implementation still defaults to field trials. BUG: webrtc:10335 Change-Id: I5fc6d327ee5d011ccc41385734b38344df172627 Reviewed-on: https://webrtc-review.googlesource.com/c/123447 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26795}
2019-02-21 07:55:59 +01:00
"../../api/transport:field_trial_based_config",
in RtpSenderVideo add support for writing DependencyDescriptor header extension Bug: webrtc:10342 Change-Id: I12cca9c5e1606338bb914e58e13d268bbc6961f9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166532 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30427}
2020-01-30 16:28:53 +01:00
"../../api/transport/rtp:dependency_descriptor",
In RtcpTransceiver add support for receiving network generic messages These message suppose to extract all information NetworkControllerInterface may need from rtcp. Bug: webrtc:8239 Change-Id: I21d9081ad147ca8abe1ae05ca7201568c6ff77d1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230421 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34876}
2021-08-30 14:42:58 +02:00
"../../api/units:data_rate",
Add pure RtpSenderEgress unit test fixture. The extracts and refactors some test code from rtp_sender_unittest.cc and puts it in a new target intended to only test RtpSenderEgress, and do it as pure unit test, rather than the unholy not-quite-unit-not-quite-integration-test thingy we have today. Only a first test case is actually ported with this CL, but it's a start... Bug: webrtc:11340 Change-Id: Ie2cdde63a00a6ff6eba7b8d443eeb76ce2a527c9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216180 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33841}
2021-04-26 17:14:27 +02:00
"../../api/units:data_size",
Use the injected clock in rtcp_transciever. Bug: webrtc:11327 Change-Id: Idb02842f2eb679f972c0449a01a81a26ceb85827 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219789 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Paul Hallak <phallak@google.com> Cr-Commit-Position: refs/heads/master@{#34080}
2021-05-21 18:08:04 +02:00
"../../api/units:time_delta",
ReportBlockData and observer added, for stats collection in future CLs. The ReportBlockData contains information about a ReportBlock and additional data such as RTT. This will be used for the calculation of RTCRemoteInboundRtpStreamStats, see full picture here: https://webrtc-review.googlesource.com/c/src/+/134107 ReportBlockData is a class version of the previously internal struct RTCPReceiver::ReportBlockWithRtt. - The new name makes sense even if we add more info to it, which will be needed for future metrics. - The new location is modules/rtp_rtcp/include/report_block_data.h. The RTCPReceiver allows obtaining the ReportBlockData in two ways: 1. Using a ReportBlockDataObserver that is notified on receiving a report block. 2. Using the GetLatestReportBlockData(). Both codepaths will be needed; video stats uses observers and audio stats uses polling. Further plumbing will be done in follow-up CLs. Bug: webrtc:10455, webrtc:10456 Change-Id: Ic9e5b4f451b5f4b203efcd6fa3bbf9736487e1f4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136584 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27961}
2019-05-16 13:32:20 +02:00
"../../api/units:timestamp",
Implement AV1 RtpPacketizer class Bug: webrtc:11042 Change-Id: Id1fc0acfa87a4520344f2636f50cb4d4e7284829 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160416 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29947}
2019-11-28 13:42:50 +01:00
"../../api/video:encoded_image",
Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets Bug: webrtc:9271 Change-Id: Ic7415830588bef9d87bab92943460207890dada6 Reviewed-on: https://webrtc-review.googlesource.com/76960 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 12:35:56 +02:00
"../../api/video:video_bitrate_allocation",
Exposing video bitrate allocator into API In order to have public video bitrate allocator factory, the video bitrate allocator has be part of the api. Bug: webrtc:9513 Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2 Reviewed-on: https://webrtc-review.googlesource.com/88083 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#24073}
2018-07-20 15:49:43 -07:00
"../../api/video:video_bitrate_allocator",
Move some video codec constants to separate file. kMaxSimulcastStreams, kMaxSpatialLayers and kMaxTemporalStreams don't really beling on VideoBitrateAllocation. common_types.h is going away and it feels dubious to requrie include of the full VideoEncoder api to use them. Therefore moving them into a seprate file/target. Also includes some remaining cleanup of includes. Bug: webrtc:9271 Change-Id: I7ded3d97a9a835ac756159700774445a2b93a697 Reviewed-on: https://webrtc-review.googlesource.com/c/117305 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26299}
2019-01-16 17:10:57 +01:00
"../../api/video:video_codec_constants",
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",
VideoStreamEncoder report VideoLayersAllocation for simulcast Adds support for Vp8 simulcast. Bug: webrtc:12000 Change-Id: Ib24fd0542642b023ec35f7a7bdc4880d72365edf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187341 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32416}
2020-10-15 17:53:22 +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",
Reland "Move class VideoCodec from common_types.h to its own api header file." This is a reland of efc71e565e9b36bcdfb4571f59e34bbd8fabd0cd Differs from the original cl by not widening the type of VideoCodec::width and VideoCodec::height. Original change's description: > Move class VideoCodec from common_types.h to its own api header file. > > Bug: webrtc:7660 > Change-Id: I91f19bfc2565461328f30081f8383e136419aefb > Reviewed-on: https://webrtc-review.googlesource.com/79881 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23544} Bug: webrtc:7660 Change-Id: I7cf74a85a61ea2b831e6f32b3b3e17514ebefec8 Reviewed-on: https://webrtc-review.googlesource.com/82140 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23569}
2018-06-08 14:03:44 +02:00
"../../api/video_codecs:video_codecs_api",
Reland of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #1 id:1 of https://codereview.webrtc.org/3011093002/ ) Reason for revert: The cl this change depended on has now been successfully relanded. Original issue's description: > Revert of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #3 id:40001 of https://codereview.webrtc.org/3006993002/ ) > > Reason for revert: > This has to be reverted to enable reverting cl https://codereview.webrtc.org/3006063002/, which seems to have broken ulpfec. > > Original issue's description: > > Delete Rtx-related methods from RTPPayloadRegistry. > > > > Delete methods IsRtx, IsEncapsulated and RestoreOriginalPacket. > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/3006993002 > > Cr-Commit-Position: refs/heads/master@{#19739} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5b4b52264132eefba10bc6880871715f9692da90 > > TBR=stefan@webrtc.org,danilchap@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/3011093002 > Cr-Commit-Position: refs/heads/master@{#19742} > Committed: https://chromium.googlesource.com/external/webrtc/+/a64685325c2f8f51873b67ae8a91f94ffb19d70b TBR=stefan@webrtc.org,danilchap@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/3012253002 Cr-Commit-Position: refs/heads/master@{#19821}
2017-09-13 07:13:57 -07:00
"../../call:rtp_receiver",
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",
in RtpSenderVideo add support for writing DependencyDescriptor header extension Bug: webrtc:10342 Change-Id: I12cca9c5e1606338bb914e58e13d268bbc6961f9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166532 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30427}
2020-01-30 16:28:53 +01:00
"../../common_video/generic_frame_descriptor",
Refactor creation of ColorSpace test data Bug: webrtc:8651 Change-Id: I2ebb5fcdc260af19d04513ab5f3d76f81a3b4ca9 Reviewed-on: https://webrtc-review.googlesource.com/c/114282 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26012}
2018-12-14 10:14:07 +01:00
"../../common_video/test:utilities",
Reland: Add mock_rtc_event_log.h. Bug: webrtc:7642 Change-Id: I3f97a8b603e34819e1563b335c1eeb51f89b11ac Reviewed-on: https://webrtc-review.googlesource.com/37081 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21480}
2018-01-03 09:08:20 +01:00
"../../logging:mocks",
Move bit_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ia6e0f1c3017a968b22fdca512ef6adb37957d777 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258769 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36604}
2022-04-14 13:34:56 +02:00
"../../rtc_base:bit_buffer",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
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",
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 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",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:rate_limiter",
Add Incoming packet handler to RtcpTransceiver Bug: webrtc:8239 Change-Id: Iebe1b6a2649f01e4f6e3780ac96cb05611d8671c Reviewed-on: https://webrtc-review.googlesource.com/17560 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20624}
2017-11-09 15:42:28 +01:00
"../../rtc_base:rtc_base_tests_utils",
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",
Introduce RtpSequenceNumberMap The video encoder should be informed of incoming LossNotification RTCP messages. Since it is unaware of RTP sequence numbers, or anything else RTP-related, the sequence numbers mentioned in the RTCP message must first be mapped to timestamps, since those are meaningful to the encoder. This CL introduces RtpSequenceNumberMap, which maps RTP sequence numbers to timestamps, while providing: 1. Capping the number of entries. 2. Wrap-around handling. RtpSequenceNumberMap also remembers which packets were first and/or last in the frame. Later CLs will wire this up. Bug: webrtc:10501 Change-Id: Ie0662cdb5706a3bcf63aa2934816a9df88439357 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130497 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27448}
2019-04-04 11:43:54 +02:00
"../../rtc_base:rtc_numerics",
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",
Avoid using global task queue factory in RtcpTransciever tests by not using convenient rtc::TaskQueue constructor and using TaskQueueForTest instead. Bug: webrtc:10284 Change-Id: I882ce4d1404a1cb9ab8bf97fca6beac3f89bb41d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128604 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27212}
2019-03-20 15:04:52 +01:00
"../../rtc_base:task_queue_for_test",
Delete TestListener and top-level thread wrapping. Instead use rtc::AutoThread in tests that need that. Bug: webrtc:9714 Change-Id: I1f33b1b2d321770d062504dd9ef86d66a345dd42 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254681 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36950}
2022-05-20 09:12:57 +02:00
"../../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",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../test:field_trial",
Define MockFrameTransformer in test/. Add MockFrameTransformer to test/, and remove definitions from unit test files. Bug: webrtc:11380 Change-Id: Ia709883e8d000852e3f71e7bfb87877072e22aeb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174001 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31151}
2020-04-29 22:39:13 +02:00
"../../test:mock_frame_transformer",
Refactoring mock_transport to be used separately Bug: webrtc:11251 Change-Id: I0a494c34c8d5c458b4d9b1b3616ae360d04df0d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168980 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Tim Na <natim@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30584}
2020-02-20 11:53:13 -08:00
"../../test:mock_transport",
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:rtp_test_utils",
Move RunLoop test class to its own build target To make it usable in tests without depending on all of CallTest. Bug: None Change-Id: Ie3102ab71bcfe3862dd6c35d3285098e961e54df Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262807 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36932}
2022-05-19 16:06:23 +02:00
"../../test:run_loop",
Remove unused WebRTC-LimitPaddingSize field trial Bug: webrtc:11508 Change-Id: Ib7d48e23bd44e2f948d51800090fc14b873d11eb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268122 Auto-Submit: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37485}
2022-07-07 14:49:18 +02:00
"../../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",
Add 1 sec timer to ModuleRtpRtcpImpl2 instead of frequent polling. This reduces the number of times we grab a few locks down from somewhere upwards of around a thousand time a second to a few times. * Update the RTT value on the worker thread and fire callbacks. * Trigger NotifyTmmbrUpdated() calls from the worker. * Update the tests to use a GlobalSimulatedTimeController. Change-Id: Ib81582494066b9460ae0aa84271f32311f30fbce Bug: webrtc:11581 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177664 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31602}
2020-07-01 08:53:21 +02:00
"../../test/time_controller:time_controller",
Cleanup RtpPacketizerVP8 tests Remove partition support for test helper and from tests. Merge Init function into constructor Replace extra macroses in favor of Bit helper function Replace extra members in favor of local variables Remove fixture Bug: None Change-Id: Ibf1600dda9f59abe5afd2bbe40c3e232a2d269ea Reviewed-on: https://webrtc-review.googlesource.com/96940 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24508}
2018-08-30 20:54:58 +02:00
"../video_coding:codec_globals_headers",
Use absl_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 Abseil container algorithms in modules/rtp_rtcp/ Bug: None Change-Id: Ica2e9795ec6195e044403f5ee25e476f6c47cf93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129600 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27361}
2019-03-28 10:56:11 -07:00
"//third_party/abseil-cpp/absl/algorithm:container",
Allowing buffering a LNTF (loss notification) feedback message in RTCPSender Loss notifications may either be sent immediately, or wait until another RTCP feedback message is sent. Bug: webrtc:10336 Change-Id: I40601d9fa1dec6c17b2ce905cb0c8cd2dcff7893 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139242 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28142}
2019-06-03 14:37:50 +02:00
"//third_party/abseil-cpp/absl/base:core_headers",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Adopt absl::string_view in modules/rtp_rtcp Bug: webrtc:13579 Change-Id: Ic4e1431bedc69492358cb2e3749b50a941306f44 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262250 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36887}
2022-05-13 11:42:16 +02:00
"//third_party/abseil-cpp/absl/strings",
Replace rtc::Optional with absl::optional in modules/rtp_rtcp This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated using script: #!/bin/bash dir=modules/rtp_rtcp find $dir -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $dir -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc Reviewed-on: https://webrtc-review.googlesource.com/83584 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-14 12:59:38 +02:00
"//third_party/abseil-cpp/absl/types:optional",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
]
}
GN target for test_packet_masks_metrics. NOTRY=True BUG=webrtc:6193 Review-Url: https://codereview.webrtc.org/2229473002 Cr-Commit-Position: refs/heads/master@{#13672}
2016-08-08 08:08:53 -07:00
}
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