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Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
# Copyright (c) 2016 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")
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
group("pc") {
Stop using public_deps in pc/. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29864 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-06 11:23:19 +01:00
deps = [
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
":rtc_pc",
]
}
config("rtc_pc_config") {
Remove HAVE_SRTP define and unmaintained code. It was defined unconditionally and the code for non-HAVE_SRTP was unmaintained and failed to compile. BUG=webrtc:7294 Review-Url: https://codereview.webrtc.org/2729373002 Cr-Commit-Position: refs/heads/master@{#17074}
2017-03-06 11:32:22 -08:00
defines = []
Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define. This allows building without SCTP support (and even building/running tests). The "HAVE_SCTP" define has been functional for a while, but there wasn't any easy way to turn it on/off. NOTRY=True BUG=webrtc:6933 Review-Url: https://codereview.webrtc.org/2593313002 Cr-Commit-Position: refs/heads/master@{#15763}
2016-12-22 10:53:38 -08:00
if (rtc_enable_sctp) {
defines += [ "HAVE_SCTP" ]
}
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
}
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
rtc_static_library("rtc_pc_base") {
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 = [ "*" ]
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
defines = []
sources = [
"channel.cc",
"channel.h",
Prevent channels being set on stopped transceiver. Fixing bug that allows a channel to be set on a stopped transceiver. This CL contains the following refactoring: 1. Extracted ChannelInterface from BaseChannel 2. Unified SetXxxMediaChannel (Voice, Video) into SetMediaChannel Bug: webrtc:9932 Change-Id: I2fbf00c823b7848ad4f2acb6e80b1b58ac45ee38 Reviewed-on: https://webrtc-review.googlesource.com/c/110564 Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Amit Hilbuch <amithi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25641}
2018-11-13 16:26:05 -08:00
"channelinterface.h",
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
"channelmanager.cc",
"channelmanager.h",
Created the DtlsSrtpTransport. The DtlsSrtpTransport is designed to take DTLS responsibilities from BaseChannel. DtlsSrtpTransport is responsible for exporting keys from DtlsTransport and setting up the wrapped SrtpTransport. The DtlsSrtpTransport is not hooked up to BaseChannel yet in this CL. Bug: webrtc:7013 Change-Id: I318c00dadf9b1e033ec842de6e1536e9227ab713 Reviewed-on: https://webrtc-review.googlesource.com/6700 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20804}
2017-11-20 14:35:11 -08:00
"dtlssrtptransport.cc",
"dtlssrtptransport.h",
Reland: Improve testing of SRTP external auth code paths. This CL is a reland of https://codereview.webrtc.org/2722423003 which got reverted due to compile errors when rolling into Chromium. Original CL description: Improve testing of SRTP external auth code paths. Previously code behind ENABLE_EXTERNAL_AUTH was only compiled with Chromium but developed in WebRTC, which made testing rather complicated. This caused some trouble in the past (e.g. https://crbug.com/628400#c1) This CL helps in that the external auth code is now compiled with WebRTC and the srtpfilter integration gets tested. BUG=chromium:628400 Review-Url: https://codereview.webrtc.org/2735613002 Cr-Commit-Position: refs/heads/master@{#17052}
2017-03-06 00:14:10 -08:00
"externalhmac.cc",
"externalhmac.h",
Replace BundleFilter with RtpDemuxer in RtpTransport. BundleFilter is replaced by RtpDemuxer in RtpTransport for payload type-based demuxing. RtpTransport will support MID-based demuxing later. Each BaseChannel has its own RTP demuxing criteria and when connecting to the RtpTransport, BaseChannel will register itself as a demuxer sink. The inheritance model is changed. New inheritance chain: DtlsSrtpTransport->SrtpTransport->RtpTranpsort The JsepTransport2 is renamed to JsepTransport. NOTE: When RTCP packets are received, Call::DeliverRtcp will be called for multiple times (webrtc:9035) which is an existing issue. With this CL, it will become more of a problem and should be fixed. Bug: webrtc:8587 Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62 Reviewed-on: https://webrtc-review.googlesource.com/65786 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22867}
2018-04-13 16:44:34 -07:00
"jseptransport.cc",
"jseptransport.h",
Create the JsepTransportController and JsepTransport2. JsepTransportController process the entire SDP and handle the RTCP-mux, SRTP setup, BUNDLE related logic internally. This will replace the current TransportController. JsepTransport2 is used by the JsepTransportController which processes the transport part of SDP and owns the DtlsTransport created internally. JsepTransport2 will replace JsepTransport and be renamed eventually. Bug: webrtc:8587 Change-Id: Ib02dfa52fe9b7a5b8b132afcc8e4363eb8bd9cf4 Reviewed-on: https://webrtc-review.googlesource.com/48841 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22164}
2018-02-22 15:26:27 -08:00
"jseptransportcontroller.cc",
"jseptransportcontroller.h",
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
"mediasession.cc",
"mediasession.h",
"rtcpmuxfilter.cc",
"rtcpmuxfilter.h",
Remove cricket::RtpTransceiverDirection Replaces cricket::RtpTransceiverDirection with webrtc::RtpTransceiverDirection, which is part of the public API. Bug: webrtc:8558 Change-Id: Ibfc9373e25187e98fb969e7ac937a1371c8fa4c7 Reviewed-on: https://webrtc-review.googlesource.com/24129 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20899}
2017-11-27 14:30:09 -08:00
"rtpmediautils.cc",
"rtpmediautils.h",
Add a minimal RtpTransport class for use by BaseChannel. This will eventually implement webrtc::RtpTransportInterface from api/ortc. It needs to live in the pc build target until the pc <- ortc dependency is inverted. BUG=webrtc:7013 Review-Url: https://codereview.webrtc.org/2792223002 Cr-Commit-Position: refs/heads/master@{#17534}
2017-04-04 19:45:57 -07:00
"rtptransport.cc",
"rtptransport.h",
Introduce RtpTransportInternal and SrtpTransport. SrtpTransport currently just delegates everything to RtpTransport. Also makes BaseChannel::rtp_transport_ an RtpTransportInternal and constructs an SrtpTransport if srtp is required. BUG=webrtc:7013 Review-Url: https://codereview.webrtc.org/2981013002 Cr-Commit-Position: refs/heads/master@{#19095}
2017-07-19 13:38:02 -07:00
"rtptransportinternal.h",
Created the DtlsSrtpTransport. The DtlsSrtpTransport is designed to take DTLS responsibilities from BaseChannel. DtlsSrtpTransport is responsible for exporting keys from DtlsTransport and setting up the wrapped SrtpTransport. The DtlsSrtpTransport is not hooked up to BaseChannel yet in this CL. Bug: webrtc:7013 Change-Id: I318c00dadf9b1e033ec842de6e1536e9227ab713 Reviewed-on: https://webrtc-review.googlesource.com/6700 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20804}
2017-11-20 14:35:11 -08:00
"rtptransportinternaladapter.h",
Move sessiondescription.h/cc from p2p/base to pc/ SDP is a detail of PeerConnection and is not used by anything in p2p, so it belongs in the pc/ directory. This also allows MediaContentDescription to be inlined in the future. Bug: webrtc:8620 Change-Id: I38b65ede9942e29eb15035ab29f2be988da1e5ce Reviewed-on: https://webrtc-review.googlesource.com/33781 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21376}
2017-12-19 14:26:11 -08:00
"sessiondescription.cc",
"sessiondescription.h",
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
"srtpfilter.cc",
"srtpfilter.h",
Move SrtpSession and tests to their own files. BUG=None Review-Url: https://codereview.webrtc.org/2976443002 Cr-Commit-Position: refs/heads/master@{#18935}
2017-07-07 14:26:25 -07:00
"srtpsession.cc",
"srtpsession.h",
Introduce RtpTransportInternal and SrtpTransport. SrtpTransport currently just delegates everything to RtpTransport. Also makes BaseChannel::rtp_transport_ an RtpTransportInternal and constructs an SrtpTransport if srtp is required. BUG=webrtc:7013 Review-Url: https://codereview.webrtc.org/2981013002 Cr-Commit-Position: refs/heads/master@{#19095}
2017-07-19 13:38:02 -07:00
"srtptransport.cc",
"srtptransport.h",
Create the JsepTransportController and JsepTransport2. JsepTransportController process the entire SDP and handle the RTCP-mux, SRTP setup, BUNDLE related logic internally. This will replace the current TransportController. JsepTransport2 is used by the JsepTransportController which processes the transport part of SDP and owns the DtlsTransport created internally. JsepTransport2 will replace JsepTransport and be renamed eventually. Bug: webrtc:8587 Change-Id: Ib02dfa52fe9b7a5b8b132afcc8e4363eb8bd9cf4 Reviewed-on: https://webrtc-review.googlesource.com/48841 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22164}
2018-02-22 15:26:27 -08:00
"transportstats.cc",
"transportstats.h",
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
]
deps = [
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"..:webrtc_common",
Securely clear memory containing key information / passwords before freeing. The previously used "memset(ptr, 0, size)" can get optimized away by compilers if "ptr" is not used afterwards. A new class "ZeroOnFreeBuffer" is introduced that can hold sensitive data and that automatically clears underlying memory when it's no longer used. Bug: webrtc:8806, webrtc:8897, webrtc:8905 Change-Id: Iedddddf80790f9af0addaab3346ec5bff102917d Reviewed-on: https://webrtc-review.googlesource.com/41941 Commit-Queue: Joachim Bauch <jbauch@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22334}
2018-03-07 20:02:26 +01:00
"../api:array_view",
Move webrtc/audio_*.h to webrtc/api/call BUG=webrtc:5878 NOTRY=True Review-Url: https://codereview.webrtc.org/2059703002 Cr-Commit-Position: refs/heads/master@{#13996}
2016-08-31 07:33:05 -07:00
"../api:call_api",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"../api:libjingle_peerconnection_api",
"../api:ortc_api",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../api/video:video_frame",
Replace BundleFilter with RtpDemuxer in RtpTransport. BundleFilter is replaced by RtpDemuxer in RtpTransport for payload type-based demuxing. RtpTransport will support MID-based demuxing later. Each BaseChannel has its own RTP demuxing criteria and when connecting to the RtpTransport, BaseChannel will register itself as a demuxer sink. The inheritance model is changed. New inheritance chain: DtlsSrtpTransport->SrtpTransport->RtpTranpsort The JsepTransport2 is renamed to JsepTransport. NOTE: When RTCP packets are received, Call::DeliverRtcp will be called for multiple times (webrtc:9035) which is an existing issue. With this CL, it will become more of a problem and should be fixed. Bug: webrtc:8587 Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62 Reviewed-on: https://webrtc-review.googlesource.com/65786 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22867}
2018-04-13 16:44:34 -07:00
"../call:rtp_interfaces",
"../call:rtp_receiver",
Move videosinkinterface.h to common_video to solve a circular dep. I updated some dependency enforcement rules to allow examples and pc to depend on common_video. I reckoned depending on common_video is not controversial when they already dependend on media/base, which is a lower-level abstraction. Bug: webrtc:6828 Change-Id: I77dbeb10187b4e70dda1d873a29994fa76070758 Reviewed-on: https://webrtc-review.googlesource.com/34187 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21495}
2018-01-04 12:14:35 +01:00
"../common_video:common_video",
Pass the RtcEventLog instance to ICE via JsepTransportController. This CL fixes a bug that the RtcEventLog owned by PeerConnection was not passed to P2PTransportChannel after JsepTransportController was introduced to deprecate the legacy TransportController. Bug: webrtc:9337 Change-Id: I406cd9c0761dfe67f969aa99c6141e1ab38249d5 Reviewed-on: https://webrtc-review.googlesource.com/79964 Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23572}
2018-06-11 20:15:46 -07:00
"../logging:rtc_event_log_api",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"../media:rtc_data",
Reland of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2949953003/ ) Reason for revert: Relanding the orginal CL. The breakage would be a flakey build. Original issue's description: > Revert of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2945233002/ ) > > Reason for revert: > The Android 32 (more config) bot is broken. > > Original issue's description: > > Try to fix the binary size increase issue on Chromium. > > > > The target common_video used to depend on rtc_media_base which introduces > > the dependency on p2p. This probably causes the binary size increase on Win > > Chromium. Some chromium targets like src/media/gpu:gpu depends on common_video directly. > > > > BUG=chromium:734631 > > > > Review-Url: https://codereview.webrtc.org/2945233002 > > Cr-Commit-Position: refs/heads/master@{#18693} > > Committed: https://chromium.googlesource.com/external/webrtc/+/9ed16097375fb8d9b45623c58d9086d33e503760 > > TBR=kjellander@webrtc.org,deadbeef@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=chromium:734631 > > Review-Url: https://codereview.webrtc.org/2949953003 > Cr-Commit-Position: refs/heads/master@{#18694} > Committed: https://chromium.googlesource.com/external/webrtc/+/c2e208a9249452590fa282ef5aba43e480bc5794 TBR=kjellander@webrtc.org,deadbeef@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:734631 Review-Url: https://codereview.webrtc.org/2949883003 Cr-Commit-Position: refs/heads/master@{#18695}
2017-06-21 01:02:59 -07:00
"../media:rtc_h264_profile_id",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"../media:rtc_media_base",
Replace BundleFilter with RtpDemuxer in RtpTransport. BundleFilter is replaced by RtpDemuxer in RtpTransport for payload type-based demuxing. RtpTransport will support MID-based demuxing later. Each BaseChannel has its own RTP demuxing criteria and when connecting to the RtpTransport, BaseChannel will register itself as a demuxer sink. The inheritance model is changed. New inheritance chain: DtlsSrtpTransport->SrtpTransport->RtpTranpsort The JsepTransport2 is renamed to JsepTransport. NOTE: When RTCP packets are received, Call::DeliverRtcp will be called for multiple times (webrtc:9035) which is an existing issue. With this CL, it will become more of a problem and should be fixed. Bug: webrtc:8587 Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62 Reviewed-on: https://webrtc-review.googlesource.com/65786 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22867}
2018-04-13 16:44:34 -07:00
"../modules/rtp_rtcp:rtp_rtcp_format",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"../p2p:rtc_p2p",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base",
"../rtc_base:rtc_task_queue",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../rtc_base:stringutils",
Move base64.h to the proper location. Move base64.h to the proper location and put redirect header into the old place to be able to switch downstream users on new location. Bug: webrtc:8366 Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291 Reviewed-on: https://webrtc-review.googlesource.com/88223 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24069}
2018-07-23 17:38:12 +02:00
"../rtc_base/third_party/base64",
Move sigslot to proper third_party directory Extract sigslot into separate target and move it to proper third_party directory. Bug: webrtc:8366 Change-Id: Id2e0712bd020bfad811947803c94553dce06d976 Reviewed-on: https://webrtc-review.googlesource.com/84141 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24099}
2018-07-25 15:04:28 +02:00
"../rtc_base/third_party/sigslot",
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",
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 _stricmp with absl::EqualsIgnoreCase All uses check only for equality. Bug: webrtc:6424 Change-Id: I8755dde02370c89dbc2226bb703664c9e4f88bdb Reviewed-on: https://webrtc-review.googlesource.com/c/106383 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25274}
2018-10-19 15:29:54 +02:00
"//third_party/abseil-cpp/absl/strings",
Replace rtc::Optional with absl::optional in pc This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'pc' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12 Reviewed-on: https://webrtc-review.googlesource.com/84181 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23669}
2018-06-19 16:47:43 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
]
if (rtc_build_libsrtp) {
deps += [ "//third_party/libsrtp" ]
}
GN Templates: Move common_inherited_config to the template. Remove common_inherited_config from the targets and add it to the template instead. BUG=webrtc:6187 NOTRY=True Review-Url: https://codereview.webrtc.org/2311843002 Cr-Commit-Position: refs/heads/master@{#14069}
2016-09-05 06:10:18 -07:00
public_configs = [ ":rtc_pc_config" ]
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
}
}
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
rtc_source_set("rtc_pc") {
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 = [ "*" ]
Mark built-in software video codecs as poisonous. The goal is to make these injectable, and only VP8 and VP9 specific targets should depend on them. Bug: webrtc:7925 Change-Id: Ie9239a54d197fe70c93de0582797211fef6997a2 Reviewed-on: https://webrtc-review.googlesource.com/72082 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23021}
2018-04-24 16:39:05 +02:00
allow_poison = [
"audio_codecs", # TODO(bugs.webrtc.org/8396): Remove.
"software_video_codecs", # TODO(bugs.webrtc.org/7925): Remove.
]
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
deps = [
Stop using public_deps in pc/. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29864 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-06 11:23:19 +01:00
":rtc_pc_base",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"../media:rtc_audio_video",
]
}
rtc_static_library("peerconnection") {
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 //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
cflags = []
sources = [
"audiotrack.cc",
"audiotrack.h",
"datachannel.cc",
"datachannel.h",
"dtmfsender.cc",
"dtmfsender.h",
Rewrite PeerConnection integration tests using better testing practices. Also renames "peerconnection_unittests" to "peerconnection_integrationtests", and moves the ICE URL parsing code to separate files. The main problem previously was that the test assertions occurred in various places in the main test class, and this shared test code was overly complex and stateful. As a result, it was difficult to tell what a test even does, let alone what assertions it's meant to be making. And writing a new test that does what you want can be a frustrating ordeal. The new code still uses helper methods, but they have intuitive names and a smaller role; all of the important parts of the test's logic are in the test case itself. We're planning on merging PeerConnection and WebRtcSession at some point soon, so it seemed valuable to do this, so that the WebRtcSession tests can be rewritten as PeerConnection tests using better patterns. BUG=None Review-Url: https://codereview.webrtc.org/2738353003 Cr-Commit-Position: refs/heads/master@{#17458}
2017-03-29 21:08:16 -07:00
"iceserverparsing.cc",
"iceserverparsing.h",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"jsepicecandidate.cc",
"jsepsessiondescription.cc",
"localaudiosource.cc",
"localaudiosource.h",
"mediastream.cc",
"mediastream.h",
"mediastreamobserver.cc",
"mediastreamobserver.h",
"mediastreamtrack.h",
"peerconnection.cc",
"peerconnection.h",
"peerconnectionfactory.cc",
"peerconnectionfactory.h",
Move internal PeerConnection methods to PeerConnectionInternal PeerConnectionInternal is being introduced so that it can be mocked in tests and so that a fake can be written for it to be used by stats tests. Bug: webrtc:8764 Change-Id: I375d12ce352523e8ac584402685a7870bc399fac Reviewed-on: https://webrtc-review.googlesource.com/43202 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21747}
2018-01-23 16:38:46 -08:00
"peerconnectioninternal.h",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"remoteaudiosource.cc",
"remoteaudiosource.h",
"rtcstatscollector.cc",
"rtcstatscollector.h",
Stats traversal algorithm added. This is part of the work to add a selector argument to getStats(). Changes: - TakeReferencedStats() added, which traverses the stats graph and takes any stats from the report that are directly or indirectly accessible from the starting stats objects in the stats graph. The result is returned as a stats report. - GetStatsReferencedIds(), an efficient helper function for getting neighbor stats object IDs. - RTCStatsReport::Take(), removed the stats object with the given ID and returns ownership of it (so that it can be added to another report). TakeReferencedStats() is tested with a bunch of sample stats graphs. GetStatsReferencedIds() is tested in the rtcstats_integrationttest.cc, making sure the expected IDs are returned. The expected IDs are the values of the stats object members with the "Id" or "Ids" suffix. Design doc: https://docs.google.com/document/d/18BywbtXgHCjsbR5nWBedpzqDjAfXrFSTJNiADnzoK0w/edit?usp=sharing Bug: chromium:680172 Change-Id: I5da9da8250da0cb05adb864015901393a4290776 Reviewed-on: https://webrtc-review.googlesource.com/60869 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22381}
2018-03-12 10:27:55 +01:00
"rtcstatstraversal.cc",
"rtcstatstraversal.h",
Add PeerConnection GetRtpSender/ReceiverCapabilities Those are static functions in the spec, so implemented as member functions of the PeerConnectionFactory instead. Bug: webrtc:7577, webrtc:9441 Change-Id: Iccb24180e096e713d24e7e25ecfd5d7bbd7638f9 Reviewed-on: https://webrtc-review.googlesource.com/85341 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23768}
2018-06-28 14:09:33 +02:00
"rtpparametersconversion.cc",
"rtpparametersconversion.h",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"rtpreceiver.cc",
"rtpreceiver.h",
"rtpsender.cc",
"rtpsender.h",
Add RtpTransceiverInterface and implementing class Introduces the public API interface corresponding to the standardized RtpTransceiver object in the WebRTC spec. https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver The RtpTransceiver will be the internal representation for both Plan B and Unified Plan SDP, but the public API interface will only support Unified Plan (existing users should continue to use GetSenders/GetReceivers, which will still be supported). Bug: webrtc:7600 Change-Id: I417ffda683209ba9a9b4cbd274f91ca8295779a7 Reviewed-on: https://webrtc-review.googlesource.com/21460 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20659}
2017-11-13 10:44:53 -08:00
"rtptransceiver.cc",
"rtptransceiver.h",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"sctputils.cc",
"sctputils.h",
Add sdputils.h with useful functions for working with session descriptions Bug: webrtc:8222 Change-Id: Iedeb65dad493721c5a4eddab254097171dfdb522 Reviewed-on: https://webrtc-review.googlesource.com/7120 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20192}
2017-10-06 10:14:03 -07:00
"sdputils.cc",
"sdputils.h",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"statscollector.cc",
"statscollector.h",
"streamcollection.h",
Move trackmediainfomap files from api/ to pc/. It looks like this was left out of the original api/pc move CL since it had been added recently. BUG=webrtc:5883 TBR=ossu@webrtc.org Review-Url: https://codereview.webrtc.org/2690793003 Cr-Commit-Position: refs/heads/master@{#16560}
2017-02-11 19:07:31 -08:00
"trackmediainfomap.cc",
"trackmediainfomap.h",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"videocapturertracksource.cc",
"videocapturertracksource.h",
"videotrack.cc",
"videotrack.h",
"videotracksource.cc",
"videotracksource.h",
"webrtcsdp.cc",
"webrtcsdp.h",
"webrtcsessiondescriptionfactory.cc",
"webrtcsessiondescriptionfactory.h",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
":rtc_pc_base",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"..:webrtc_common",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"../api:call_api",
Revert "Revert "Enables PeerConnectionFactory using external fec controller"" This reverts commit 00733015fafbbc61ddc12dfdc88b21a9fcd9d122. Reason for revert: The reason for a downstream test failure on the original commit and a workaround has been found. Solution is to keep a PeerConnectionFactory constructor implementation as the same as before. Original change's description: > Revert "Enables PeerConnectionFactory using external fec controller" > > This reverts commit 4f07bdb25567d8ef528311e0b50a62c61d543fc3. > > Reason for revert: Speculatively reverting, because downstream test is now hitting "PeerConnectionFactory.initialize was not called before creating a PeerConnectionFactory" error, even though it did call initialize. I don't see how any change in this CL could cause that, but it's the only CL on the blamelist, and it does modify PeerConnectionFactory.java > > Original change's description: > > Enables PeerConnectionFactory using external fec controller > > > > Bug: webrtc:8799 > > Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8 > > Reviewed-on: https://webrtc-review.googlesource.com/43961 > > Commit-Queue: Ying Wang <yinwa@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22038} > > TBR=sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org > > Change-Id: I95868c35d6f9973e0ebf563814cd71d0fcbd433d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8799 > Reviewed-on: https://webrtc-review.googlesource.com/54080 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22040} TBR=deadbeef@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org Bug: webrtc:8799 Change-Id: If9f3292bfcc739782967530c49f006d0abbc38a8 Reviewed-on: https://webrtc-review.googlesource.com/55400 Commit-Queue: Ying Wang <yinwa@webrtc.org> Reviewed-by: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22100}
2018-02-20 12:50:27 +01:00
"../api:fec_controller_api",
Stop using public_deps in pc/. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29864 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-06 11:23:19 +01:00
"../api:libjingle_peerconnection_api",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"../api:rtc_stats_api",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../api/video:video_frame",
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) Reason for revert: Reland with temporary deprecated API to not break chromium and google3. Original issue's description: > Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) > > Reason for revert: > Suspect of breaking Chrome FYI bots. > > See > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065 > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder > > Example logs: > ../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory > #include "third_party/webrtc/video_encoder.h" > ^ > > Original issue's description: > > Move video_encoder.h and video_decoder.h to /api and create GN targets for them > > > > BUG=webrtc:5881 > > # Because PRESUBMIT ignores LINT blacklist for moved files and these > > # headers have some not easy to resolve issues. > > NOPRESUBMIT=True > > > > Review-Url: https://codereview.webrtc.org/2780943003 > > Cr-Commit-Position: refs/heads/master@{#17511} > > Committed: https://chromium.googlesource.com/external/webrtc/+/c42f54057050c933008a49d57582577bfb9aed25 > > TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5881 > > Review-Url: https://codereview.webrtc.org/2794033002 > Cr-Commit-Position: refs/heads/master@{#17514} > Committed: https://chromium.googlesource.com/external/webrtc/+/716d7ac5c1ed6e392e264b34065800bbf03772b3 TBR=solenberg@webrtc.org,sprang@webrtc.org,guidou@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5881 Review-Url: https://codereview.webrtc.org/2795163002 Cr-Commit-Position: refs/heads/master@{#17537}
2017-04-05 03:02:20 -07:00
"../api/video_codecs:video_codecs_api",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"../call:call_interfaces",
Move videosinkinterface.h to common_video to solve a circular dep. I updated some dependency enforcement rules to allow examples and pc to depend on common_video. I reckoned depending on common_video is not controversial when they already dependend on media/base, which is a lower-level abstraction. Bug: webrtc:6828 Change-Id: I77dbeb10187b4e70dda1d873a29994fa76070758 Reviewed-on: https://webrtc-review.googlesource.com/34187 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21495}
2018-01-04 12:14:35 +01:00
"../common_video:common_video",
Bind the structured ICE logging with P2PTransportChannel. This change list passes the instance of RtcEventLog from Peerconnection down to P2PTransportChannel, and binds the structured ICE logging with ICE layer objects. Logs of ICE connectivity checks are injected for candidate pairs. TBR=terelius@webrtc.org Bug: None Change-Id: Ia979dbbac6d31dcf0f8988da1065bdfc3e461821 Reviewed-on: https://webrtc-review.googlesource.com/34660 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21884}
2018-01-30 17:13:09 -08:00
"../logging:ice_log",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"../logging:rtc_event_log_api",
Reland "Break up rtc_event_log_api to solve circular dependencies." This is a reland of 001546da953275c7a39eb220592b440c9b47d756 Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@google.com> > Cr-Commit-Position: refs/heads/master@{#21811} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 11:04:46 -08:00
"../logging:rtc_event_log_impl_output",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"../media:rtc_data",
"../media:rtc_media_base",
Allows injection of network controller factory into peer connection factory. Bug: webrtc:9155 Change-Id: I0a17024042f154297aba20f5d2dc766feb27f3f7 Reviewed-on: https://webrtc-review.googlesource.com/73123 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23313}
2018-05-18 18:05:10 +02:00
"../modules/congestion_controller/bbr",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"../p2p:rtc_p2p",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../rtc_base:stringutils",
Allows injection of network controller factory into peer connection factory. Bug: webrtc:9155 Change-Id: I0a17024042f154297aba20f5d2dc766feb27f3f7 Reviewed-on: https://webrtc-review.googlesource.com/73123 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23313}
2018-05-18 18:05:10 +02:00
"../rtc_base/experiments:congestion_controller_experiment",
Export symbols needed by the Chromium component build (part 2). This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) to mark WebRTC symbols as visible from a shared library, this doesn't mean these symbols are part of the public API (please continue to refer to [1] for info about what is considered public WebRTC API). Bug: webrtc:9419 Change-Id: I6f27003001548ea9d54412fdf62d5dd7a39cfd46 Reviewed-on: https://webrtc-review.googlesource.com/c/106022 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25187}
2018-10-15 17:15:12 +02:00
"../rtc_base/system:rtc_export",
Move base64.h to the proper location. Move base64.h to the proper location and put redirect header into the old place to be able to switch downstream users on new location. Bug: webrtc:8366 Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291 Reviewed-on: https://webrtc-review.googlesource.com/88223 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24069}
2018-07-23 17:38:12 +02:00
"../rtc_base/third_party/base64",
Move sigslot to proper third_party directory Extract sigslot into separate target and move it to proper third_party directory. Bug: webrtc:8366 Change-Id: Id2e0712bd020bfad811947803c94553dce06d976 Reviewed-on: https://webrtc-review.googlesource.com/84141 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24099}
2018-07-25 15:04:28 +02:00
"../rtc_base/third_party/sigslot",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"../stats",
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",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../system_wrappers:field_trial",
"../system_wrappers:metrics",
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",
Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase" This is a reland of 80cd25bcfb2264fa0f1192de942a6f063879dd42 Original change's description: > Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase > > Bug: None > Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad > Reviewed-on: https://webrtc-review.googlesource.com/c/107303 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25312} Tbr: kwiberg@webrtc.org Bug: None Change-Id: Id43a93bada9d6d66a4d0f0286f583066156aa2fc Reviewed-on: https://webrtc-review.googlesource.com/c/107716 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25368}
2018-10-23 10:07:25 +02:00
"//third_party/abseil-cpp/absl/strings",
Replace rtc::Optional with absl::optional in pc This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'pc' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12 Reviewed-on: https://webrtc-review.googlesource.com/84181 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23669}
2018-06-19 16:47:43 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
]
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
}
# This target implements CreatePeerConnectionFactory methods that will create a
# PeerConnection will full functionality (audio, video and data). Applications
# that wish to reduce their binary size by ommitting functionality they don't
# need should use CreateModularCreatePeerConnectionFactory instead, using the
# "peerconnection" build target and other targets specific to their
# requrements. See comment in peerconnectionfactoryinterface.h.
Fix the binary size regression on Chromium Windows. There is a dependency chain from Chromium windows main_dll to Opus which should never exist. We used to rely on rtc_static_library to break this chain. So this CL replaced some rtc_source_set with rtc_static_library. libvpx fix (https://chromium-review.googlesource.com/c/544107/) for ios-simulator linking issue is landed and this CL can be sumbitted once the new Chromium is rolled into WebRTC. BUG=chromium:734631 Review-Url: https://codereview.webrtc.org/2947273002 Cr-Commit-Position: refs/heads/master@{#18709}
2017-06-22 01:28:59 -07:00
rtc_static_library("create_pc_factory") {
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
sources = [
"createpeerconnectionfactory.cc",
]
deps = [
Rename Call::Config to CallConfig, keep old name as alias. We want api/peerconnectioninterface.h (and corresponding build target) to not depend on call.h, and generally we treat Call as an internal, non-api, class. But we need CallFactoryInterface in the api in order to enable use of PeerConnection with or without support for media. Making CallConfig a top-level class makes it possible to forward declare it, together with Call, for use in callfactoryinterface.h and peerconnectioninterface.h. Delete the peerconnection_and_implicit_call_api target, replaced by new target callfactory_api, to link between Call and Peerconnection. Bug: webrtc:7504 Change-Id: I5e3978ef89bcd6705e94536f8676bcf89fc82fe1 Reviewed-on: https://webrtc-review.googlesource.com/46201 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22020}
2018-02-14 12:20:13 +01:00
"../api:callfactory_api",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"../api:libjingle_peerconnection_api",
BUILD.gn file for api/audio. Targets containing files in api/audio are moved from api/BUILD.gn to api/audio/BUILD.gn. Bug: webrtc:8844 Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981 Reviewed-on: https://webrtc-review.googlesource.com/54360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22074}
2018-02-16 13:43:49 +01:00
"../api/audio:audio_mixer_api",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"../api/audio_codecs:audio_codecs_api",
Expose new video codec factories in the PeerConnectionFactory API This CL exposes the new type of video codec factories that represent all video codecs in the PeerConnectionFactory API, i.e. no extra internal SW video codecs will be added. Clients of the new functions will be responsible for adding all SW video codecs themselves, and also handling SW fallback and simulcast. BUG=webrtc:7925 R=deadbeef@webrtc.org Review-Url: https://codereview.webrtc.org/3004353002 . Cr-Commit-Position: refs/heads/master@{#19866}
2017-09-15 19:02:47 +02:00
"../api/video_codecs:video_codecs_api",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"../call",
"../call:call_interfaces",
"../logging:rtc_event_log_api",
Reland "Break up rtc_event_log_api to solve circular dependencies." This is a reland of 001546da953275c7a39eb220592b440c9b47d756 Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@google.com> > Cr-Commit-Position: refs/heads/master@{#21811} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 11:04:46 -08:00
"../logging:rtc_event_log_impl_base",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"../media:rtc_audio_video",
Fix some dependencies for peerconnection_and_implicit_call_api. Some targets depend on "api/peerconnectioninterface.h" which is part of //api:peerconnection_and_implicit_call_api. Furthermore, peerconnection_and_implicit_call_api depends among other things of headers in //media:rtc_media_base, so we should add it as a dependency as well. Bug: webrtc:7504 Change-Id: Ifab69253d52532876509b3507917b1c93d99a2ac Reviewed-on: https://webrtc-review.googlesource.com/24660 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20815}
2017-11-20 18:41:50 +01:00
"../media:rtc_media_base",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"../modules/audio_device:audio_device",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../modules/audio_processing:api",
Allow an external audio processing module to be used in WebRTC [This CL is a rebase of an original CL by solenberg@: https://codereview.webrtc.org/2948763002/ which in turn was a rebase of an original CL by peah@: https://chromium-review.googlesource.com/c/527032/] Allow an external audio processing module to be used in WebRTC This CL adds support for optionally using an externally created audio processing module in a peerconnection. The ownership is shared between the peerconnection and the external creator of the module. As part of this the internal ownership of the audio processing module is moved from VoiceEngine to WebRtcVoiceEngine. BUG=webrtc:7775 Review-Url: https://codereview.webrtc.org/2961723004 Cr-Commit-Position: refs/heads/master@{#18837}
2017-06-29 08:32:09 -07:00
"../modules/audio_processing:audio_processing",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
rtc_source_set("libjingle_peerconnection") {
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 = [ "*" ]
Mark built-in software video codecs as poisonous. The goal is to make these injectable, and only VP8 and VP9 specific targets should depend on them. Bug: webrtc:7925 Change-Id: Ie9239a54d197fe70c93de0582797211fef6997a2 Reviewed-on: https://webrtc-review.googlesource.com/72082 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23021}
2018-04-24 16:39:05 +02:00
allow_poison = [
"audio_codecs", # TODO(bugs.webrtc.org/8396): Remove.
"software_video_codecs", # TODO(bugs.webrtc.org/7925): Remove.
]
Stop using public_deps in pc/. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29864 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-06 11:23:19 +01:00
deps = [
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
":create_pc_factory",
":peerconnection",
"../api:libjingle_peerconnection_api",
]
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
}
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
if (rtc_include_tests) {
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_test("rtc_pc_unittests") {
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
testonly = true
sources = [
"channel_unittest.cc",
"channelmanager_unittest.cc",
Created the DtlsSrtpTransport. The DtlsSrtpTransport is designed to take DTLS responsibilities from BaseChannel. DtlsSrtpTransport is responsible for exporting keys from DtlsTransport and setting up the wrapped SrtpTransport. The DtlsSrtpTransport is not hooked up to BaseChannel yet in this CL. Bug: webrtc:7013 Change-Id: I318c00dadf9b1e033ec842de6e1536e9227ab713 Reviewed-on: https://webrtc-review.googlesource.com/6700 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20804}
2017-11-20 14:35:11 -08:00
"dtlssrtptransport_unittest.cc",
Replace BundleFilter with RtpDemuxer in RtpTransport. BundleFilter is replaced by RtpDemuxer in RtpTransport for payload type-based demuxing. RtpTransport will support MID-based demuxing later. Each BaseChannel has its own RTP demuxing criteria and when connecting to the RtpTransport, BaseChannel will register itself as a demuxer sink. The inheritance model is changed. New inheritance chain: DtlsSrtpTransport->SrtpTransport->RtpTranpsort The JsepTransport2 is renamed to JsepTransport. NOTE: When RTCP packets are received, Call::DeliverRtcp will be called for multiple times (webrtc:9035) which is an existing issue. With this CL, it will become more of a problem and should be fixed. Bug: webrtc:8587 Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62 Reviewed-on: https://webrtc-review.googlesource.com/65786 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22867}
2018-04-13 16:44:34 -07:00
"jseptransport_unittest.cc",
Create the JsepTransportController and JsepTransport2. JsepTransportController process the entire SDP and handle the RTCP-mux, SRTP setup, BUNDLE related logic internally. This will replace the current TransportController. JsepTransport2 is used by the JsepTransportController which processes the transport part of SDP and owns the DtlsTransport created internally. JsepTransport2 will replace JsepTransport and be renamed eventually. Bug: webrtc:8587 Change-Id: Ib02dfa52fe9b7a5b8b132afcc8e4363eb8bd9cf4 Reviewed-on: https://webrtc-review.googlesource.com/48841 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22164}
2018-02-22 15:26:27 -08:00
"jseptransportcontroller_unittest.cc",
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
"mediasession_unittest.cc",
"rtcpmuxfilter_unittest.cc",
Move ready to send logic from BaseChannel to RtpTransport. BUG=webrtc:7013 Review-Url: https://codereview.webrtc.org/2812243005 Cr-Commit-Position: refs/heads/master@{#17853}
2017-04-24 16:54:35 -07:00
"rtptransport_unittest.cc",
Introduce RtpTransportInternal and SrtpTransport. SrtpTransport currently just delegates everything to RtpTransport. Also makes BaseChannel::rtp_transport_ an RtpTransportInternal and constructs an SrtpTransport if srtp is required. BUG=webrtc:7013 Review-Url: https://codereview.webrtc.org/2981013002 Cr-Commit-Position: refs/heads/master@{#19095}
2017-07-19 13:38:02 -07:00
"rtptransporttestutil.h",
Refactor of extmap-allow-mixed in SessionDescription - Rename member variable and access functions to show connection to the extmap-allow-mixed attribute. - Transfer extmap-allow-mixed setting when new content is added to a SessionDescription. - Add boolean function that shows support of extmap-allow-mixed. - Add unit tests of extmap-allow-mixed logic. Bug: webrtc:7990 Change-Id: Ic330fbab7be3e1df81c2d36ce522acc254617178 Reviewed-on: https://webrtc-review.googlesource.com/c/105308 Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25163}
2018-10-12 10:54:26 +02:00
"sessiondescription_unittest.cc",
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
"srtpfilter_unittest.cc",
Move SrtpSession and tests to their own files. BUG=None Review-Url: https://codereview.webrtc.org/2976443002 Cr-Commit-Position: refs/heads/master@{#18935}
2017-07-07 14:26:25 -07:00
"srtpsession_unittest.cc",
"srtptestutil.h",
Introduce RtpTransportInternal and SrtpTransport. SrtpTransport currently just delegates everything to RtpTransport. Also makes BaseChannel::rtp_transport_ an RtpTransportInternal and constructs an SrtpTransport if srtp is required. BUG=webrtc:7013 Review-Url: https://codereview.webrtc.org/2981013002 Cr-Commit-Position: refs/heads/master@{#19095}
2017-07-19 13:38:02 -07:00
"srtptransport_unittest.cc",
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
]
include_dirs = [ "//third_party/libsrtp/srtp" ]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
}
if (is_win) {
libs = [ "strmiids.lib" ]
}
deps = [
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
":libjingle_peerconnection",
Move the TransportController from p2p/base to pc/. The TransportController was in p2p/base before and it cannot depend on pc/ or media/ level targets because of the circular dependency. To make the TransportController be responsible for creating and managing the RtpTransport related objects which are pc/ level targets, the TransportController is moved from p2p/base to pc/. The TransportController makes more sense in pc/ anyway, since its main responsibility is processing the "transport" parts of SDP which is PeerConnection-specific. This is also easier than moving RtpTransport related objects to p2p/base because those objects also depend on other media/ and pc/ level targets such as srtpfilter, cryptoparams etc. Bug: webrtc:7013 Change-Id: Ic48dd5c454046ff3c81331f4b459f96a3255f328 Reviewed-on: https://webrtc-review.googlesource.com/4560 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20049}
2017-09-29 10:51:43 -07:00
":pc_test_utils",
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
":rtc_pc",
Stop using public_deps in pc/. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29864 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-06 11:23:19 +01:00
":rtc_pc_base",
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",
Integration of media_transport in JSepTransportController Basic integration of media_transport in JSepTransportController. - Creates media_transport if media transport factory provided in jsep transport controller configuration. - Unittest that makes sure media_transport is created with correct caller or callee setting. - Added fake_media_transport, for now simple implementation which only stores caller/callee, but in the future fake media transport will be expanded to pass frames between two fake media_transports, which will enable audio / video integration tests. NEXT STEPS: Once integration of media_transport with PeerConnection (https://webrtc-review.googlesource.com/c/src/+/103860) lands, we can start passing media transport factory from peer connection to jsep transport controller. NOTE: Includes missing include change from https://webrtc-review.googlesource.com/c/src/+/103540 (otherwise this change will not compile) Bug: webrtc:9719 Change-Id: I1e8a521beab445aa9f7ea93c8d7a537dc137d11c Reviewed-on: https://webrtc-review.googlesource.com/c/104400 Commit-Queue: Anton Sukhanov <sukhanov@google.com> Reviewed-by: Peter Slatala <psla@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25096}
2018-10-10 10:34:49 -07:00
"../api:fake_media_transport",
Stop using public_deps in pc/. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29864 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-06 11:23:19 +01:00
"../api:libjingle_peerconnection_api",
Replace BundleFilter with RtpDemuxer in RtpTransport. BundleFilter is replaced by RtpDemuxer in RtpTransport for payload type-based demuxing. RtpTransport will support MID-based demuxing later. Each BaseChannel has its own RTP demuxing criteria and when connecting to the RtpTransport, BaseChannel will register itself as a demuxer sink. The inheritance model is changed. New inheritance chain: DtlsSrtpTransport->SrtpTransport->RtpTranpsort The JsepTransport2 is renamed to JsepTransport. NOTE: When RTCP packets are received, Call::DeliverRtcp will be called for multiple times (webrtc:9035) which is an existing issue. With this CL, it will become more of a problem and should be fixed. Bug: webrtc:8587 Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62 Reviewed-on: https://webrtc-review.googlesource.com/65786 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22867}
2018-04-13 16:44:34 -07:00
"../call:rtp_interfaces",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"../logging:rtc_event_log_api",
"../media:rtc_media_base",
Enabling 'gn check' on webrtc/video. I disabled the check on "video_tests" because it pulls "//webrtc/media/rtc_unittest_main" as a dependency and it defines the _main (that is already defined by "//webrtc/test:test_main"). I will file a bug to solve this in another CL. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2832063003 Cr-Commit-Position: refs/heads/master@{#17859}
2017-04-25 00:35:35 -07:00
"../media:rtc_media_tests_utils",
Replace BundleFilter with RtpDemuxer in RtpTransport. BundleFilter is replaced by RtpDemuxer in RtpTransport for payload type-based demuxing. RtpTransport will support MID-based demuxing later. Each BaseChannel has its own RTP demuxing criteria and when connecting to the RtpTransport, BaseChannel will register itself as a demuxer sink. The inheritance model is changed. New inheritance chain: DtlsSrtpTransport->SrtpTransport->RtpTranpsort The JsepTransport2 is renamed to JsepTransport. NOTE: When RTCP packets are received, Call::DeliverRtcp will be called for multiple times (webrtc:9035) which is an existing issue. With this CL, it will become more of a problem and should be fixed. Bug: webrtc:8587 Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62 Reviewed-on: https://webrtc-review.googlesource.com/65786 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22867}
2018-04-13 16:44:34 -07:00
"../modules/rtp_rtcp:rtp_rtcp_format",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"../p2p:p2p_test_utils",
"../p2p:rtc_p2p",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_main",
"../rtc_base:rtc_base_tests_utils",
Move sigslot to proper third_party directory Extract sigslot into separate target and move it to proper third_party directory. Bug: webrtc:8366 Change-Id: Id2e0712bd020bfad811947803c94553dce06d976 Reviewed-on: https://webrtc-review.googlesource.com/84141 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24099}
2018-07-25 15:04:28 +02:00
"../rtc_base/third_party/sigslot",
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",
Introduce RtpTransportInternal and SrtpTransport. SrtpTransport currently just delegates everything to RtpTransport. Also makes BaseChannel::rtp_transport_ an RtpTransportInternal and constructs an SrtpTransport if srtp is required. BUG=webrtc:7013 Review-Url: https://codereview.webrtc.org/2981013002 Cr-Commit-Position: refs/heads/master@{#19095}
2017-07-19 13:38:02 -07:00
"../test:test_support",
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",
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
]
if (rtc_build_libsrtp) {
deps += [ "//third_party/libsrtp" ]
}
if (is_android) {
deps += [ "//testing/android/native_test:native_test_support" ]
}
}
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
A new PeerConnection level perf test. This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-22 15:40:16 -07:00
rtc_source_set("peerconnection_perf_tests") {
testonly = true
sources = [
"peerconnection_rampup_tests.cc",
"peerconnectionwrapper.cc",
"peerconnectionwrapper.h",
]
deps = [
":pc_test_utils",
"../api:libjingle_peerconnection_api",
"../api:rtc_stats_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/video_codecs:builtin_video_decoder_factory",
"../api/video_codecs:builtin_video_encoder_factory",
"../media:rtc_media_tests_utils",
"../p2p:p2p_test_utils",
"../p2p:rtc_p2p",
"../pc:peerconnection",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../test:perf_test",
"../test:test_support",
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",
A new PeerConnection level perf test. This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-22 15:40:16 -07:00
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
2017-02-25 18:15:09 -08:00
rtc_source_set("pc_test_utils") {
testonly = true
sources = [
"test/fakeaudiocapturemodule.cc",
"test/fakeaudiocapturemodule.h",
"test/fakedatachannelprovider.h",
Add FakePeerConnectionBase This provides an intermediate class for defining default, null implementations of all the PeerConnectionInterface/ PeerConnectionInternal methods. Specific fake PeerConnections then can inherit from this and only override the methods pertaining to the scenarios it will be used in. Bug: webrtc:8764 Change-Id: I7614303b5673747244053b54b839e58aada43d10 Reviewed-on: https://webrtc-review.googlesource.com/43245 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21752}
2018-01-24 15:29:17 -08:00
"test/fakepeerconnectionbase.h",
Rewrite StatsCollector tests to use a fake PeerConnection This removes use of the MockPeerConnection and replaces it with a fake implementation of PeerConnection tailored to the needs of StatsCollector and (soon) RTCStatsCollector. The stats collector tests really care about the PeerConnection only so much as to set up scenarios to test the StatsCollector with. Since each scenario (e.g., adding a track) affects the results of multiple methods (e.g., voice_channel and SessionStats), the tests were needing to manually configure these dependent operations which was tedious, error prone and difficult to change. The new fake lets the tests express the scenario more concisely (e.g., AddVoiceChannel) while filling in all the affected methods on the PeerConnection automatically. Furthermore, this can be expanded to use with the RTCStatsCollector and to cover more scenarios in the future. Bug: webrtc:8764 Change-Id: I195074174684256543f7cdc27c3834e5ff0b4702 Reviewed-on: https://webrtc-review.googlesource.com/43521 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21778}
2018-01-26 10:25:53 -08:00
"test/fakepeerconnectionforstats.h",
Delete ortc methods using cricket::VideoCapturer. For testing, introduce new class FakePeriodicVideoSource. Bug: webrtc:6353 Change-Id: I234dbd4176ea23d21b8f876ee02d3662477f225e Reviewed-on: https://webrtc-review.googlesource.com/49220 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22554}
2018-03-21 09:20:21 +01:00
"test/fakeperiodicvideosource.h",
New class FakePeriodicVideoTrackSource, simplifying shutdown logic. Previous code had a FakePeriodicVideoSource and a VideoTrackSource, where the latter is reference counted and outlives the former. That results in potential races when RemoveSink is called on the VideoTrackSource after the FakePeriodicVideoSource is destroyed, with a complicated sequence to do correct shutdown. The new class, FakePeriodicVideoTrackSource, owns a FakePeriodicVideoSource, and they get the same lifetime. Bug: webrtc:6353 Change-Id: Ic33b393e00a31fa28893dce2018948d3f90e0a9e Reviewed-on: https://webrtc-review.googlesource.com/76961 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23320}
2018-05-17 09:16:41 +02:00
"test/fakeperiodicvideotracksource.h",
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
2017-02-25 18:15:09 -08:00
"test/fakertccertificategenerator.h",
Rewrite WebRtcSession data channel tests as PeerConnection tests Bug: webrtc:8222 Change-Id: I1382a0727b04dfd33e79992841d885f640b3a032 Reviewed-on: https://webrtc-review.googlesource.com/8281 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20398}
2017-10-23 11:41:54 -07:00
"test/fakesctptransport.h",
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
2017-02-25 18:15:09 -08:00
"test/fakevideotrackrenderer.h",
"test/fakevideotracksource.h",
A new PeerConnection level perf test. This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-22 15:40:16 -07:00
"test/framegeneratorcapturervideotracksource.h",
Prevent channels being set on stopped transceiver. Fixing bug that allows a channel to be set on a stopped transceiver. This CL contains the following refactoring: 1. Extracted ChannelInterface from BaseChannel 2. Unified SetXxxMediaChannel (Voice, Video) into SetMediaChannel Bug: webrtc:9932 Change-Id: I2fbf00c823b7848ad4f2acb6e80b1b58ac45ee38 Reviewed-on: https://webrtc-review.googlesource.com/c/110564 Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Amit Hilbuch <amithi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25641}
2018-11-13 16:26:05 -08:00
"test/mock_channelinterface.h",
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
2017-02-25 18:15:09 -08:00
"test/mock_datachannel.h",
Reland "Update RTCStatsCollector to work with RtpTransceivers" Original change's description: > Update RTCStatsCollector to work with RtpTransceivers > > Bug: webrtc:8764 > Change-Id: I8b442345869eb6d8b65fd12241ed7cb6e7d7ce3d > Reviewed-on: https://webrtc-review.googlesource.com/49580 > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22026} Bug: webrtc:8764 Change-Id: I6a682824febf3f4f41397fc1a8dd7396c4ffa8e3 Reviewed-on: https://webrtc-review.googlesource.com/54160 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22064}
2018-02-15 15:19:50 -08:00
"test/mock_rtpreceiverinternal.h",
"test/mock_rtpsenderinternal.h",
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
2017-02-25 18:15:09 -08:00
"test/mockpeerconnectionobservers.h",
"test/peerconnectiontestwrapper.cc",
"test/peerconnectiontestwrapper.h",
"test/rtcstatsobtainer.h",
"test/testsdpstrings.h",
]
deps = [
":libjingle_peerconnection",
Stop using public_deps in pc/. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29864 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-06 11:23:19 +01:00
":peerconnection",
Move the TransportController from p2p/base to pc/. The TransportController was in p2p/base before and it cannot depend on pc/ or media/ level targets because of the circular dependency. To make the TransportController be responsible for creating and managing the RtpTransport related objects which are pc/ level targets, the TransportController is moved from p2p/base to pc/. The TransportController makes more sense in pc/ anyway, since its main responsibility is processing the "transport" parts of SDP which is PeerConnection-specific. This is also easier than moving RtpTransport related objects to p2p/base because those objects also depend on other media/ and pc/ level targets such as srtpfilter, cryptoparams etc. Bug: webrtc:7013 Change-Id: Ic48dd5c454046ff3c81331f4b459f96a3255f328 Reviewed-on: https://webrtc-review.googlesource.com/4560 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20049}
2017-09-29 10:51:43 -07:00
":rtc_pc_base",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"..:webrtc_common",
Stop using public_deps in api/. Bug: webrtc:8603 Change-Id: I12f91365fa9a3ed12be1a9658999b6f6aad7e5e3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/33220 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21584}
2018-01-11 10:13:56 +01:00
"../api:libjingle_peerconnection_api",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"../api:libjingle_peerconnection_test_api",
"../api:rtc_stats_api",
Move VideoStreamEncoderInterface to api/. Bug: webrtc:8830 Change-Id: I17908b4ef6a043acf22e2110b9672012d5fa7fc0 Reviewed-on: https://webrtc-review.googlesource.com/74481 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23334}
2018-05-21 14:09:31 +02:00
"../api/video:video_frame",
Built in video codec factories. To prepare for making the software codecs optional and injectable, these codec factories provide a way to pass in identical factories as were the default old behaviour. Bug: webrtc:7925 Change-Id: I0c70fa3c56c999e9d1af6e172eff2fbba849e921 Reviewed-on: https://webrtc-review.googlesource.com/71162 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23096}
2018-05-03 11:28:29 +02:00
"../api/video_codecs:builtin_video_decoder_factory",
"../api/video_codecs:builtin_video_encoder_factory",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"../call:call_interfaces",
"../logging:rtc_event_log_api",
Stop using public_deps in media/. Bug: webrtc:8603 Change-Id: I7a6dad323ac298dc784feb5aa1fdc2ae5876cb5c No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/33180 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21591}
2018-01-11 17:07:30 +01:00
"../media:rtc_data",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"../media:rtc_media",
Fix some dependencies for peerconnection_and_implicit_call_api. Some targets depend on "api/peerconnectioninterface.h" which is part of //api:peerconnection_and_implicit_call_api. Furthermore, peerconnection_and_implicit_call_api depends among other things of headers in //media:rtc_media_base, so we should add it as a dependency as well. Bug: webrtc:7504 Change-Id: Ifab69253d52532876509b3507917b1c93d99a2ac Reviewed-on: https://webrtc-review.googlesource.com/24660 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20815}
2017-11-20 18:41:50 +01:00
"../media:rtc_media_base",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"../media:rtc_media_tests_utils",
"../modules/audio_device:audio_device",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../modules/audio_processing:api",
Built in video codec factories. To prepare for making the software codecs optional and injectable, these codec factories provide a way to pass in identical factories as were the default old behaviour. Bug: webrtc:7925 Change-Id: I0c70fa3c56c999e9d1af6e172eff2fbba849e921 Reviewed-on: https://webrtc-review.googlesource.com/71162 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23096}
2018-05-03 11:28:29 +02:00
"../modules/audio_processing:audio_processing",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"../p2p:p2p_test_utils",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
Detangling target dependencies in rtc_base_approved. The eventual goal is to allow PlatformThread to use SequencedTaskChecker, but getting to that point will require some more detangling. Here are (roughly) the steps taken in this CL: * Make constructormagic a separate target. * Move atomicops and arraysize to separate targets * Move platform_thread_types to a separate target * Move criticalsection to a separate target * Move thread_checker to separate target * Make sequenced_task_checker not depend on base_approved * Move ptr_util to a separate target * Move scoped_ptr to ptr_util * Make rtc_task_queue_api not depend on base_approved * Make sequenced_task_checker depend on rtc_task_queue_api * Move rtc::Event to its own target * Move basictypes.h to constructormagic * Move format_macros and stringize_macros into constructormagic * Rename constructormagic target to... macromagic * Move stringencode to stringutils * New target for safe_conversions * Move timeutils to a new target. * Move logging to a new target. * Move platform_thread to a new target. * Make refcount a new target (refcount, refcountedobject, refcounter). * Remove rtc_base_approved from deps of TQ * Remove a circular dependency between event tracer and platform thread. Further steps will probably be to factor TaskQueue::Current() to not be a part of the TaskQueue class itself and have it declared+implemented in a target that's lower level than TQ itself. SequencedTaskChecker can then depend on that target and avoid the TQ dependency. Once we're there, PlatformThread will be able to depend on SequencedTaskChecker. Attempted but eventually removed from this CL: * Make TQ a part of rtc_base_approved * Remove direct dependencies on sequenced_task_checker. * Profit. A few include-what-you-use updates along the way. Fix a few targets that were depending on rtc_task_queue_api Change-Id: Iee79aa2e81d978444c51b3005db9df7dc12d92a9 Bug: webrtc:8957 Reviewed-on: https://webrtc-review.googlesource.com/58480 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22487}
2018-03-19 11:12:48 +01:00
"../rtc_base:rtc_task_queue",
Move sigslot to proper third_party directory Extract sigslot into separate target and move it to proper third_party directory. Bug: webrtc:8366 Change-Id: Id2e0712bd020bfad811947803c94553dce06d976 Reviewed-on: https://webrtc-review.googlesource.com/84141 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24099}
2018-07-25 15:04:28 +02:00
"../rtc_base/third_party/sigslot",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"../test:test_support",
A new PeerConnection level perf test. This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-22 15:40:16 -07:00
"../test:video_test_common",
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",
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
2017-02-25 18:15:09 -08:00
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
rtc_test("peerconnection_unittests") {
testonly = true
sources = [
"datachannel_unittest.cc",
"dtmfsender_unittest.cc",
Rewrite PeerConnection integration tests using better testing practices. Also renames "peerconnection_unittests" to "peerconnection_integrationtests", and moves the ICE URL parsing code to separate files. The main problem previously was that the test assertions occurred in various places in the main test class, and this shared test code was overly complex and stateful. As a result, it was difficult to tell what a test even does, let alone what assertions it's meant to be making. And writing a new test that does what you want can be a frustrating ordeal. The new code still uses helper methods, but they have intuitive names and a smaller role; all of the important parts of the test's logic are in the test case itself. We're planning on merging PeerConnection and WebRtcSession at some point soon, so it seemed valuable to do this, so that the WebRtcSession tests can be rewritten as PeerConnection tests using better patterns. BUG=None Review-Url: https://codereview.webrtc.org/2738353003 Cr-Commit-Position: refs/heads/master@{#17458}
2017-03-29 21:08:16 -07:00
"iceserverparsing_unittest.cc",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"jsepsessiondescription_unittest.cc",
"localaudiosource_unittest.cc",
"mediaconstraintsinterface_unittest.cc",
"mediastream_unittest.cc",
Reland "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests" This is a reland of b49b66109ea8a0a33a3192ebccf91366af2e49ae. Original change's description: > Rewrite WebRtcSession BUNDLE tests as PeerConnection tests > > Bug: webrtc:8222 > Change-Id: Id47e4544dc073564ad7e63d02865ca80dd5a85ff > Reviewed-on: https://webrtc-review.googlesource.com/8280 > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20365} Bug: webrtc:8222 Change-Id: If3dcd8090875c641881e2b9e92fc1db387ba1de5 Reviewed-on: https://webrtc-review.googlesource.com/14400 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20397}
2017-10-23 09:39:20 -07:00
"peerconnection_bundle_unittest.cc",
Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests Bug: webrtc:8222 Change-Id: I6be2c5a5735b77a5c577472b88ff830204dd69eb Reviewed-on: https://webrtc-review.googlesource.com/1160 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20193}
2017-10-06 11:20:31 -07:00
"peerconnection_crypto_unittest.cc",
Rewrite WebRtcSession data channel tests as PeerConnection tests Bug: webrtc:8222 Change-Id: I1382a0727b04dfd33e79992841d885f640b3a032 Reviewed-on: https://webrtc-review.googlesource.com/8281 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20398}
2017-10-23 11:41:54 -07:00
"peerconnection_datachannel_unittest.cc",
Ensure that PC usage is recorded if a PC is alive for 60 seconds. Bug: chromium:718508 Change-Id: Id2cbcb370b56cb8a6a6c821e0f89c51089cc8e6b Reviewed-on: https://webrtc-review.googlesource.com/83140 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23723}
2018-06-25 12:03:50 +02:00
"peerconnection_histogram_unittest.cc",
Rewrite WebRtcSession ICE tests as PeerConnection tests Bug: webrtc:8222 Change-Id: Ie138712beaae964f08150b29ba5a8ed17ea0e956 Reviewed-on: https://webrtc-review.googlesource.com/4640 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20296}
2017-10-13 11:13:35 -07:00
"peerconnection_ice_unittest.cc",
Rewrite PeerConnection integration tests using better testing practices. Also renames "peerconnection_unittests" to "peerconnection_integrationtests", and moves the ICE URL parsing code to separate files. The main problem previously was that the test assertions occurred in various places in the main test class, and this shared test code was overly complex and stateful. As a result, it was difficult to tell what a test even does, let alone what assertions it's meant to be making. And writing a new test that does what you want can be a frustrating ordeal. The new code still uses helper methods, but they have intuitive names and a smaller role; all of the important parts of the test's logic are in the test case itself. We're planning on merging PeerConnection and WebRtcSession at some point soon, so it seemed valuable to do this, so that the WebRtcSession tests can be rewritten as PeerConnection tests using better patterns. BUG=None Review-Url: https://codereview.webrtc.org/2738353003 Cr-Commit-Position: refs/heads/master@{#17458}
2017-03-29 21:08:16 -07:00
"peerconnection_integrationtest.cc",
Add support for JSEP offer/answer with transceivers This change adds support to PeerConnection's CreateOffer/ CreateAnswer/SetLocalDescription/SetRemoteDescription for Unified Plan SDP mapping to/from RtpTransceivers. This behavior is enabled using the kUnifiedPlan SDP semantics in the PeerConnection configuration. Bug: webrtc:7600 Change-Id: I4b44f5d3690887d387bf9c47eac00db8ec974571 Reviewed-on: https://webrtc-review.googlesource.com/28341 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21442}
2017-12-22 16:02:54 -08:00
"peerconnection_jsep_unittest.cc",
Reland "Rewrite WebRtcSession media tests as PeerConnection tests" This is a reland of 3df5dcac9b339ba4d3f4969602f094c2c8035b51 Original change's description: > Rewrite WebRtcSession media tests as PeerConnection tests > > Bug: webrtc:8222 > Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad > Reviewed-on: https://webrtc-review.googlesource.com/6640 > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20364} Bug: webrtc:8222 Change-Id: I0a5398170d469eb9223bc781bfb417a85a72a2d2 Reviewed-on: https://webrtc-review.googlesource.com/14380 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20377}
2017-10-20 15:30:51 -07:00
"peerconnection_media_unittest.cc",
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
"peerconnection_rtp_unittest.cc",
Reland "Rewrite WebRtcSession media tests as PeerConnection tests" This is a reland of 3df5dcac9b339ba4d3f4969602f094c2c8035b51 Original change's description: > Rewrite WebRtcSession media tests as PeerConnection tests > > Bug: webrtc:8222 > Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad > Reviewed-on: https://webrtc-review.googlesource.com/6640 > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20364} Bug: webrtc:8222 Change-Id: I0a5398170d469eb9223bc781bfb417a85a72a2d2 Reviewed-on: https://webrtc-review.googlesource.com/14380 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20377}
2017-10-20 15:30:51 -07:00
"peerconnection_signaling_unittest.cc",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"peerconnectionendtoend_unittest.cc",
"peerconnectionfactory_unittest.cc",
"peerconnectioninterface_unittest.cc",
Add base fixture and PeerConnection wrapper for unit tests This lays the groundwork for splitting up the PeerConnectionInterface unit tests into multiple files so that the tests can be organized better. The intent is for each unit test file to declare a test fixture which subclasses PeerConnectionUnitTestFixture and creates PeerConnectionWrappers to write assertions against. Bug: webrtc:8222 Change-Id: I21175b1e1828a6cd5012305a8a27faaf4eecf81c Reviewed-on: https://webrtc-review.googlesource.com/1120 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20004}
2017-09-26 16:20:19 -07:00
"peerconnectionwrapper.cc",
"peerconnectionwrapper.h",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"proxy_unittest.cc",
"rtcstats_integrationtest.cc",
"rtcstatscollector_unittest.cc",
Stats traversal algorithm added. This is part of the work to add a selector argument to getStats(). Changes: - TakeReferencedStats() added, which traverses the stats graph and takes any stats from the report that are directly or indirectly accessible from the starting stats objects in the stats graph. The result is returned as a stats report. - GetStatsReferencedIds(), an efficient helper function for getting neighbor stats object IDs. - RTCStatsReport::Take(), removed the stats object with the given ID and returns ownership of it (so that it can be added to another report). TakeReferencedStats() is tested with a bunch of sample stats graphs. GetStatsReferencedIds() is tested in the rtcstats_integrationttest.cc, making sure the expected IDs are returned. The expected IDs are the values of the stats object members with the "Id" or "Ids" suffix. Design doc: https://docs.google.com/document/d/18BywbtXgHCjsbR5nWBedpzqDjAfXrFSTJNiADnzoK0w/edit?usp=sharing Bug: chromium:680172 Change-Id: I5da9da8250da0cb05adb864015901393a4290776 Reviewed-on: https://webrtc-review.googlesource.com/60869 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22381}
2018-03-12 10:27:55 +01:00
"rtcstatstraversal_unittest.cc",
Remove cricket::RtpTransceiverDirection Replaces cricket::RtpTransceiverDirection with webrtc::RtpTransceiverDirection, which is part of the public API. Bug: webrtc:8558 Change-Id: Ibfc9373e25187e98fb969e7ac937a1371c8fa4c7 Reviewed-on: https://webrtc-review.googlesource.com/24129 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20899}
2017-11-27 14:30:09 -08:00
"rtpmediautils_unittest.cc",
Add PeerConnection GetRtpSender/ReceiverCapabilities Those are static functions in the spec, so implemented as member functions of the PeerConnectionFactory instead. Bug: webrtc:7577, webrtc:9441 Change-Id: Iccb24180e096e713d24e7e25ecfd5d7bbd7638f9 Reviewed-on: https://webrtc-review.googlesource.com/85341 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23768}
2018-06-28 14:09:33 +02:00
"rtpparametersconversion_unittest.cc",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"rtpsenderreceiver_unittest.cc",
Prevent channels being set on stopped transceiver. Fixing bug that allows a channel to be set on a stopped transceiver. This CL contains the following refactoring: 1. Extracted ChannelInterface from BaseChannel 2. Unified SetXxxMediaChannel (Voice, Video) into SetMediaChannel Bug: webrtc:9932 Change-Id: I2fbf00c823b7848ad4f2acb6e80b1b58ac45ee38 Reviewed-on: https://webrtc-review.googlesource.com/c/110564 Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Amit Hilbuch <amithi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25641}
2018-11-13 16:26:05 -08:00
"rtptransceiver_unittest.cc",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"sctputils_unittest.cc",
"statscollector_unittest.cc",
"test/fakeaudiocapturemodule_unittest.cc",
"test/testsdpstrings.h",
Move trackmediainfomap files from api/ to pc/. It looks like this was left out of the original api/pc move CL since it had been added recently. BUG=webrtc:5883 TBR=ossu@webrtc.org Review-Url: https://codereview.webrtc.org/2690793003 Cr-Commit-Position: refs/heads/master@{#16560}
2017-02-11 19:07:31 -08:00
"trackmediainfomap_unittest.cc",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"videocapturertracksource_unittest.cc",
"videotrack_unittest.cc",
"webrtcsdp_unittest.cc",
]
if (rtc_enable_sctp) {
defines = [ "HAVE_SCTP" ]
}
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
Stop using public_deps in pc/. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29864 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-06 11:23:19 +01:00
deps = [
":peerconnection",
":rtc_pc_base",
Fixes use after free error when setting a new FrameEncryptor on ChannelSend. This change corrects a potential race condition when updating a FrameEncryptor for the audio send channel. If a FrameEncryptor is set on an active audio stream it is possible for the current FrameEncryptor attached to the audio channel to be deallocated due to the FrameEncryptors reference count reaching zero before the new FrameEncryptor is set on the channel. To address this issue the ChannelSend is now holds a scoped_reftptr<FrameEncryptor> to only allow deallocation when it is actually set on the encoder queue. ChannelSend is unique in this respect as the Audio Receiver a long with the Video Sender and Video Receiver streams all recreate themselves when they have a configuration change. ChannelSend instead reconfigures itself using the existing channel object. Added Seth as TBR as this only introduces mocks. TBR=shampson@webrtc.org Bug: webrtc:9907 Change-Id: Ibf391dc9cecdbed1874e0252ff5c2cb92a5c64f4 Reviewed-on: https://webrtc-review.googlesource.com/c/107664 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25374}
2018-10-25 09:52:57 -07:00
"../api:fake_frame_decryptor",
"../api:fake_frame_encryptor",
Stop using public_deps in api/. Bug: webrtc:8603 Change-Id: I12f91365fa9a3ed12be1a9658999b6f6aad7e5e3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/33220 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21584}
2018-01-11 10:13:56 +01:00
"../api:libjingle_peerconnection_api",
Implement data channels over media transport. This changes PeerConnection to allow sending and receiving data channel messages over the media transport. If |use_media_transport_for_data_channels| is set, PeerConnection will use a DCT_MEDIA_TRANSPORT mode for data channels. DCT_MEDIA_TRANSPORT acts exactly like DCT_SCTP within the data channel and peer connection layers. On the transport layer, it uses the media transport instead of SCTP. It appears as an RTP data channel in SDP (just as media over media-transport appears as RTP in SDP). Bug: webrtc:9719 Change-Id: I6a90142bd3f43668479c825ed02689dcd0d58b78 Reviewed-on: https://webrtc-review.googlesource.com/c/109740 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25575}
2018-11-08 11:23:22 -08:00
"../api:loopback_media_transport",
Add missing mock headers to api. R=mbonadei@webrtc.org Bug: webrtc:7618 Change-Id: Ia622a7623b2fa05ec14b52d5d31d158d1bd0ef6d Reviewed-on: https://webrtc-review.googlesource.com/34644 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21364}
2017-12-19 10:32:11 +01:00
"../api:mock_rtp",
Replacing rtc::TimeDelta with webrtc::TimeDelta. This removes the redundant type and replaces all usages. A slight change in behavior is that we no longer get nanosecond resolution. This should not matter since no current code requires nanosecond resolution. Bug: webrtc:9155 Change-Id: I04334e08c686d95731621a6c8a7e40400d0ae3b2 Reviewed-on: https://webrtc-review.googlesource.com/71163 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23174}
2018-05-08 14:52:22 +02:00
"../api/units:time_delta",
Pass the RtcEventLog instance to ICE via JsepTransportController. This CL fixes a bug that the RtcEventLog owned by PeerConnection was not passed to P2PTransportChannel after JsepTransportController was introduced to deprecate the legacy TransportController. Bug: webrtc:9337 Change-Id: I406cd9c0761dfe67f969aa99c6141e1ab38249d5 Reviewed-on: https://webrtc-review.googlesource.com/79964 Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23572}
2018-06-11 20:15:46 -07:00
"../logging:fake_rtc_event_log",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../rtc_base:checks",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../rtc_base:stringutils",
Move base64.h to the proper location. Move base64.h to the proper location and put redirect header into the old place to be able to switch downstream users on new location. Bug: webrtc:8366 Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291 Reviewed-on: https://webrtc-review.googlesource.com/88223 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24069}
2018-07-23 17:38:12 +02:00
"../rtc_base/third_party/base64",
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",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../test:fileutils",
Use 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",
Depend directly on destination targets. Makes 'gn check' happy. Followup to https://webrtc-review.googlesource.com/c/src/+/106820 Bug: webrtc:5876, webrtc:9855 Change-Id: I33fa2c31ba26dc10c9a9c17da0ffed255c1f4d5b Reviewed-on: https://webrtc-review.googlesource.com/c/108760 Commit-Queue: Yves Gerey <yvesg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25447}
2018-10-30 22:23:02 +01:00
"//third_party/abseil-cpp/absl/strings",
Stop using public_deps in pc/. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29864 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-06 11:23:19 +01:00
]
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
if (is_android) {
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
deps += [ ":android_black_magic" ]
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
}
deps += [
":libjingle_peerconnection",
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
2017-02-25 18:15:09 -08:00
":pc_test_utils",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"..:webrtc_common",
Rename Call::Config to CallConfig, keep old name as alias. We want api/peerconnectioninterface.h (and corresponding build target) to not depend on call.h, and generally we treat Call as an internal, non-api, class. But we need CallFactoryInterface in the api in order to enable use of PeerConnection with or without support for media. Making CallConfig a top-level class makes it possible to forward declare it, together with Call, for use in callfactoryinterface.h and peerconnectioninterface.h. Delete the peerconnection_and_implicit_call_api target, replaced by new target callfactory_api, to link between Call and Peerconnection. Bug: webrtc:7504 Change-Id: I5e3978ef89bcd6705e94536f8676bcf89fc82fe1 Reviewed-on: https://webrtc-review.googlesource.com/46201 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22020}
2018-02-14 12:20:13 +01:00
"../api:callfactory_api",
Reland "Reland "Propagate media transport to media channel."" This is a reland of da65ed2adcfa57ff3288ce01c1602c973fcab00d Original change's description: > Reland "Propagate media transport to media channel." > > This reverts commit 37cf2455a420124b341ad06ac27fa3c4dbd29d3c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Propagate media transport to media channel." > > > > This reverts commit 8c16f745ab92cb6d305283e87fa8a661ae500ce4. > > > > Reason for revert: Breaks downstream project > > > > Original change's description: > > > Propagate media transport to media channel. > > > > > > 1. Pass media transport factory to JSEP transport controller. > > > 2. Pass media transport to voice media channel. > > > 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel. > > > > > > Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71 > > > Bug: webrtc:9719 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/105542 > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > > Reviewed-by: Peter Slatala <psla@webrtc.org> > > > Commit-Queue: Anton Sukhanov <sukhanov@google.com> > > > Cr-Commit-Position: refs/heads/master@{#25152} > > > > TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:9719 > > Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1 > > Reviewed-on: https://webrtc-review.googlesource.com/c/105840 > > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25154} > > TBR=steveanton@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com > > Change-Id: I505ff3451eae81573531faef155ff35d7f894022 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9719 > Reviewed-on: https://webrtc-review.googlesource.com/c/106500 > Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> > Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25220} Bug: webrtc:9719 Tbr: Steve Anton <steveanton@webrtc.org> Tbr: Niels Moller <nisse@webrtc.org> Change-Id: Ib45691ba8be9abb89ff8c6dac1861bdf59be4c8d Reviewed-on: https://webrtc-review.googlesource.com/c/106561 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25240}
2018-10-17 13:15:42 -07:00
"../api:fake_media_transport",
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
"../api:libjingle_peerconnection_test_api",
"../api:rtc_stats_api",
PeerConnection end-to-end test with a non-builtin codec BUG=webrtc:8159 Change-Id: I27f3fc8a6204115c480d70238225b0565c41bd81 Reviewed-on: https://webrtc-review.googlesource.com/6283 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20245}
2017-10-10 23:17:17 +02:00
"../api/audio_codecs:audio_codecs_api",
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
PeerConnection end-to-end test with a non-builtin codec BUG=webrtc:8159 Change-Id: I27f3fc8a6204115c480d70238225b0565c41bd81 Reviewed-on: https://webrtc-review.googlesource.com/6283 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20245}
2017-10-10 23:17:17 +02:00
"../api/audio_codecs/L16:audio_decoder_L16",
"../api/audio_codecs/L16:audio_encoder_L16",
Built in video codec factories. To prepare for making the software codecs optional and injectable, these codec factories provide a way to pass in identical factories as were the default old behaviour. Bug: webrtc:7925 Change-Id: I0c70fa3c56c999e9d1af6e172eff2fbba849e921 Reviewed-on: https://webrtc-review.googlesource.com/71162 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23096}
2018-05-03 11:28:29 +02:00
"../api/video_codecs:builtin_video_decoder_factory",
"../api/video_codecs:builtin_video_encoder_factory",
"../api/video_codecs:video_codecs_api",
Reland "Rewrite WebRtcSession media tests as PeerConnection tests" This is a reland of 3df5dcac9b339ba4d3f4969602f094c2c8035b51 Original change's description: > Rewrite WebRtcSession media tests as PeerConnection tests > > Bug: webrtc:8222 > Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad > Reviewed-on: https://webrtc-review.googlesource.com/6640 > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20364} Bug: webrtc:8222 Change-Id: I0a5398170d469eb9223bc781bfb417a85a72a2d2 Reviewed-on: https://webrtc-review.googlesource.com/14380 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20377}
2017-10-20 15:30:51 -07:00
"../call:call_interfaces",
Add PeerConnection::StartRtcEventLog version that takes RtcEventLogOutput as parameter This will allow Chrome to provide a RtcEventLogOutput object that reports the log back to Chrome, allowing Chrome to manage the log by itself - write it to a file, upload it to a server, etc. Bug: webrtc:8111 Change-Id: I6a2a6945fc8586ef10e0fb9c56eaa8fda00dfc98 Reviewed-on: https://webrtc-review.googlesource.com/8081 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20295}
2017-10-13 16:29:40 +02:00
"../logging:rtc_event_log_api",
Reland "Break up rtc_event_log_api to solve circular dependencies." This is a reland of 001546da953275c7a39eb220592b440c9b47d756 Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@google.com> > Cr-Commit-Position: refs/heads/master@{#21811} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 11:04:46 -08:00
"../logging:rtc_event_log_impl_base",
"../logging:rtc_event_log_impl_output",
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
"../media:rtc_audio_video",
"../media:rtc_data", # TODO(phoglund): AFAIK only used for one sctp constant.
"../media:rtc_media_base",
Enabling 'gn check' on webrtc/video. I disabled the check on "video_tests" because it pulls "//webrtc/media/rtc_unittest_main" as a dependency and it defines the _main (that is already defined by "//webrtc/test:test_main"). I will file a bug to solve this in another CL. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2832063003 Cr-Commit-Position: refs/heads/master@{#17859}
2017-04-25 00:35:35 -07:00
"../media:rtc_media_tests_utils",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../modules/audio_processing:api",
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
"../modules/audio_processing:audio_processing",
"../modules/utility:utility",
"../p2p:p2p_test_utils",
"../p2p:rtc_p2p",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"../pc:rtc_pc",
Reland "Added PeerConnectionObserver::OnRemoveTrack." This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40. Reason for revert: Fixed the flake. Original change's description: > Revert "Added PeerConnectionObserver::OnRemoveTrack." > > This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. > > Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. > > Original change's description: > > Added PeerConnectionObserver::OnRemoveTrack. > > > > This corresponds to processing the removal of a remote track step of > > the spec, with processing the addition of a remote track already > > covered by OnAddTrack. > > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > > > Bug: webrtc:8260, webrtc:8315 > > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > > Reviewed-on: https://webrtc-review.googlesource.com/4722 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20214} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org > > Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8260, webrtc:8315 > Reviewed-on: https://webrtc-review.googlesource.com/7940 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20218} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org Change-Id: Iab7500bebf98535754b102874259de43831fff6b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/8180 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20227}
2017-10-10 10:05:16 -07:00
"../rtc_base:rtc_base",
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
"../rtc_base:rtc_base_approved",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base_tests_main",
"../rtc_base:rtc_base_tests_utils",
Detangling target dependencies in rtc_base_approved. The eventual goal is to allow PlatformThread to use SequencedTaskChecker, but getting to that point will require some more detangling. Here are (roughly) the steps taken in this CL: * Make constructormagic a separate target. * Move atomicops and arraysize to separate targets * Move platform_thread_types to a separate target * Move criticalsection to a separate target * Move thread_checker to separate target * Make sequenced_task_checker not depend on base_approved * Move ptr_util to a separate target * Move scoped_ptr to ptr_util * Make rtc_task_queue_api not depend on base_approved * Make sequenced_task_checker depend on rtc_task_queue_api * Move rtc::Event to its own target * Move basictypes.h to constructormagic * Move format_macros and stringize_macros into constructormagic * Rename constructormagic target to... macromagic * Move stringencode to stringutils * New target for safe_conversions * Move timeutils to a new target. * Move logging to a new target. * Move platform_thread to a new target. * Make refcount a new target (refcount, refcountedobject, refcounter). * Remove rtc_base_approved from deps of TQ * Remove a circular dependency between event tracer and platform thread. Further steps will probably be to factor TaskQueue::Current() to not be a part of the TaskQueue class itself and have it declared+implemented in a target that's lower level than TQ itself. SequencedTaskChecker can then depend on that target and avoid the TQ dependency. Once we're there, PlatformThread will be able to depend on SequencedTaskChecker. Attempted but eventually removed from this CL: * Make TQ a part of rtc_base_approved * Remove direct dependencies on sequenced_task_checker. * Profit. A few include-what-you-use updates along the way. Fix a few targets that were depending on rtc_task_queue_api Change-Id: Iee79aa2e81d978444c51b3005db9df7dc12d92a9 Bug: webrtc:8957 Reviewed-on: https://webrtc-review.googlesource.com/58480 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22487}
2018-03-19 11:12:48 +01:00
"../rtc_base:rtc_task_queue",
Removing warning suppression flags from pc/. Bug: webrtc:9251 Change-Id: Ic12126fc03309448fe71a17e6b65343949496f4f Reviewed-on: https://webrtc-review.googlesource.com/86820 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23838}
2018-07-03 12:53:23 +02:00
"../rtc_base:safe_conversions",
Fix Chromium style checker warnings for MockAudioDecoder BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2934143003 Cr-Commit-Position: refs/heads/master@{#18587}
2017-06-14 03:31:17 -07:00
"../test:audio_codec_mocks",
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
"../test:test_support",
Replace rtc::Optional with absl::optional in pc This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'pc' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12 Reviewed-on: https://webrtc-review.googlesource.com/84181 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23669}
2018-06-19 16:47:43 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
]
if (is_android) {
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
deps += [
"//testing/android/native_test:native_test_support",
# We need to depend on this one directly, or classloads will fail for
# the voice engine BuildInfo, for instance.
"../sdk/android:libjingle_peerconnection_java",
]
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
shard_timeout = 900
}
}
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
if (is_android) {
rtc_source_set("android_black_magic") {
# The android code uses hacky includes to chromium-base and the ssl code;
# having this in a separate target enables us to keep the peerconnection
# unit tests clean.
check_includes = false
testonly = true
sources = [
"test/androidtestinitializer.cc",
"test/androidtestinitializer.h",
]
deps = [
"../sdk/android:libjingle_peerconnection_jni",
"//testing/android/native_test:native_test_support",
]
}
}
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
}
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