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.
|
|
|
|
|
|
2017-01-24 06:58:22 -08:00
|
|
|
import("../webrtc.gni")
|
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") {
|
2016-09-15 00:53:26 -07:00
|
|
|
public_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") {
|
2017-03-06 11:32:22 -08:00
|
|
|
defines = []
|
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
|
|
|
}
|
|
|
|
|
|
2017-06-15 12:52:32 -07:00
|
|
|
rtc_static_library("rtc_pc_base") {
|
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 = [
|
|
|
|
|
"audiomonitor.cc",
|
|
|
|
|
"audiomonitor.h",
|
|
|
|
|
"bundlefilter.cc",
|
|
|
|
|
"bundlefilter.h",
|
|
|
|
|
"channel.cc",
|
|
|
|
|
"channel.h",
|
|
|
|
|
"channelmanager.cc",
|
|
|
|
|
"channelmanager.h",
|
|
|
|
|
"currentspeakermonitor.cc",
|
|
|
|
|
"currentspeakermonitor.h",
|
2017-03-06 00:14:10 -08:00
|
|
|
"externalhmac.cc",
|
|
|
|
|
"externalhmac.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
|
|
|
"mediamonitor.cc",
|
|
|
|
|
"mediamonitor.h",
|
|
|
|
|
"mediasession.cc",
|
|
|
|
|
"mediasession.h",
|
|
|
|
|
"rtcpmuxfilter.cc",
|
|
|
|
|
"rtcpmuxfilter.h",
|
2017-04-04 19:45:57 -07:00
|
|
|
"rtptransport.cc",
|
|
|
|
|
"rtptransport.h",
|
2017-07-19 13:38:02 -07:00
|
|
|
"rtptransportinternal.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",
|
2017-07-07 14:26:25 -07:00
|
|
|
"srtpsession.cc",
|
|
|
|
|
"srtpsession.h",
|
2017-07-19 13:38:02 -07:00
|
|
|
"srtptransport.cc",
|
|
|
|
|
"srtptransport.h",
|
2017-09-29 10:51:43 -07:00
|
|
|
"transportcontroller.cc",
|
|
|
|
|
"transportcontroller.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
|
|
|
"voicechannel.h",
|
|
|
|
|
]
|
|
|
|
|
|
|
|
|
|
deps = [
|
2017-06-01 13:01:48 -07:00
|
|
|
"..:webrtc_common",
|
2016-08-31 07:33:05 -07:00
|
|
|
"../api:call_api",
|
2017-06-01 13:01:48 -07:00
|
|
|
"../api:libjingle_peerconnection_api",
|
2017-09-05 08:43:13 -07:00
|
|
|
"../api:optional",
|
2017-06-01 13:01:48 -07:00
|
|
|
"../api:ortc_api",
|
2017-06-15 12:52:32 -07:00
|
|
|
"../media:rtc_data",
|
2017-06-21 01:02:59 -07:00
|
|
|
"../media:rtc_h264_profile_id",
|
2017-06-15 12:52:32 -07:00
|
|
|
"../media:rtc_media_base",
|
2017-06-01 13:01:48 -07:00
|
|
|
"../p2p:rtc_p2p",
|
2017-07-19 10:40:47 -07:00
|
|
|
"../rtc_base:rtc_base",
|
|
|
|
|
"../rtc_base:rtc_task_queue",
|
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" ]
|
|
|
|
|
}
|
|
|
|
|
|
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
|
|
|
|
2016-10-16 23:56:12 -07:00
|
|
|
if (!build_with_chromium && is_clang) {
|
2016-06-13 00:08:24 -07:00
|
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
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
|
|
|
}
|
|
|
|
|
}
|
2016-06-13 00:08:24 -07:00
|
|
|
|
2017-06-15 12:52:32 -07:00
|
|
|
rtc_source_set("rtc_pc") {
|
|
|
|
|
public_deps = [
|
|
|
|
|
":rtc_pc_base",
|
|
|
|
|
]
|
|
|
|
|
|
|
|
|
|
deps = [
|
|
|
|
|
"../media:rtc_audio_video",
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
|
2017-01-23 04:56:25 -08:00
|
|
|
config("libjingle_peerconnection_warnings_config") {
|
|
|
|
|
# GN orders flags on a target before flags from configs. The default config
|
|
|
|
|
# adds these flags so to cancel them out they need to come from a config and
|
|
|
|
|
# cannot be on the target directly.
|
|
|
|
|
if (!is_win && !is_clang) {
|
|
|
|
|
cflags = [ "-Wno-maybe-uninitialized" ] # Only exists for GCC.
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
|
|
2017-06-15 12:52:32 -07:00
|
|
|
rtc_static_library("peerconnection") {
|
2017-01-23 04:56:25 -08:00
|
|
|
cflags = []
|
|
|
|
|
sources = [
|
|
|
|
|
"audiotrack.cc",
|
|
|
|
|
"audiotrack.h",
|
|
|
|
|
"datachannel.cc",
|
|
|
|
|
"datachannel.h",
|
|
|
|
|
"dtmfsender.cc",
|
|
|
|
|
"dtmfsender.h",
|
2017-03-29 21:08:16 -07:00
|
|
|
"iceserverparsing.cc",
|
|
|
|
|
"iceserverparsing.h",
|
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",
|
|
|
|
|
"remoteaudiosource.cc",
|
|
|
|
|
"remoteaudiosource.h",
|
|
|
|
|
"rtcstatscollector.cc",
|
|
|
|
|
"rtcstatscollector.h",
|
|
|
|
|
"rtpreceiver.cc",
|
|
|
|
|
"rtpreceiver.h",
|
|
|
|
|
"rtpsender.cc",
|
|
|
|
|
"rtpsender.h",
|
|
|
|
|
"sctputils.cc",
|
|
|
|
|
"sctputils.h",
|
2017-10-06 10:14:03 -07:00
|
|
|
"sdputils.cc",
|
|
|
|
|
"sdputils.h",
|
2017-01-23 04:56:25 -08:00
|
|
|
"statscollector.cc",
|
|
|
|
|
"statscollector.h",
|
|
|
|
|
"streamcollection.h",
|
2017-02-11 19:07:31 -08:00
|
|
|
"trackmediainfomap.cc",
|
|
|
|
|
"trackmediainfomap.h",
|
2017-01-23 04:56:25 -08:00
|
|
|
"videocapturertracksource.cc",
|
|
|
|
|
"videocapturertracksource.h",
|
|
|
|
|
"videotrack.cc",
|
|
|
|
|
"videotrack.h",
|
|
|
|
|
"videotracksource.cc",
|
|
|
|
|
"videotracksource.h",
|
|
|
|
|
"webrtcsdp.cc",
|
|
|
|
|
"webrtcsdp.h",
|
|
|
|
|
"webrtcsession.cc",
|
|
|
|
|
"webrtcsession.h",
|
|
|
|
|
"webrtcsessiondescriptionfactory.cc",
|
|
|
|
|
"webrtcsessiondescriptionfactory.h",
|
|
|
|
|
]
|
|
|
|
|
|
|
|
|
|
configs += [ ":libjingle_peerconnection_warnings_config" ]
|
|
|
|
|
|
|
|
|
|
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 = [
|
2017-06-15 12:52:32 -07:00
|
|
|
":rtc_pc_base",
|
2017-06-01 13:01:48 -07:00
|
|
|
"..:webrtc_common",
|
2017-01-23 04:56:25 -08:00
|
|
|
"../api:call_api",
|
2017-09-05 08:43:13 -07:00
|
|
|
"../api:optional",
|
2017-01-23 04:56:25 -08:00
|
|
|
"../api:rtc_stats_api",
|
2017-04-05 03:02:20 -07:00
|
|
|
"../api/video_codecs:video_codecs_api",
|
2017-06-15 12:52:32 -07:00
|
|
|
"../call:call_interfaces",
|
2017-06-01 13:01:48 -07:00
|
|
|
"../logging:rtc_event_log_api",
|
2017-06-15 12:52:32 -07:00
|
|
|
"../media:rtc_data",
|
|
|
|
|
"../media:rtc_media_base",
|
2017-06-01 13:01:48 -07:00
|
|
|
"../p2p:rtc_p2p",
|
2017-07-19 10:40:47 -07:00
|
|
|
"../rtc_base:rtc_base",
|
|
|
|
|
"../rtc_base:rtc_base_approved",
|
2017-01-23 04:56:25 -08:00
|
|
|
"../stats",
|
2017-06-01 13:01:48 -07:00
|
|
|
"../system_wrappers:system_wrappers",
|
2017-01-23 04:56:25 -08:00
|
|
|
]
|
|
|
|
|
|
2017-01-23 07:37:43 -08:00
|
|
|
public_deps = [
|
|
|
|
|
"../api:libjingle_peerconnection_api",
|
|
|
|
|
]
|
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.
|
2017-06-22 01:28:59 -07:00
|
|
|
rtc_static_library("create_pc_factory") {
|
2017-06-15 12:52:32 -07:00
|
|
|
sources = [
|
|
|
|
|
"createpeerconnectionfactory.cc",
|
|
|
|
|
]
|
|
|
|
|
|
|
|
|
|
deps = [
|
|
|
|
|
"../api:audio_mixer_api",
|
|
|
|
|
"../api:libjingle_peerconnection_api",
|
|
|
|
|
"../api/audio_codecs:audio_codecs_api",
|
|
|
|
|
"../api/audio_codecs:builtin_audio_decoder_factory",
|
|
|
|
|
"../api/audio_codecs:builtin_audio_encoder_factory",
|
2017-09-15 19:02:47 +02:00
|
|
|
"../api/video_codecs:video_codecs_api",
|
2017-06-15 12:52:32 -07:00
|
|
|
"../call",
|
|
|
|
|
"../call:call_interfaces",
|
|
|
|
|
"../logging:rtc_event_log_api",
|
|
|
|
|
"../media:rtc_audio_video",
|
|
|
|
|
"../modules/audio_device:audio_device",
|
2017-06-29 08:32:09 -07:00
|
|
|
"../modules/audio_processing:audio_processing",
|
2017-07-19 10:40:47 -07:00
|
|
|
"../rtc_base:rtc_base",
|
|
|
|
|
"../rtc_base:rtc_base_approved",
|
2017-06-15 12:52:32 -07:00
|
|
|
]
|
|
|
|
|
|
|
|
|
|
configs += [ ":libjingle_peerconnection_warnings_config" ]
|
|
|
|
|
|
|
|
|
|
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") {
|
|
|
|
|
public_deps = [
|
|
|
|
|
":create_pc_factory",
|
|
|
|
|
":peerconnection",
|
|
|
|
|
"../api:libjingle_peerconnection_api",
|
|
|
|
|
]
|
2017-01-23 07:37:43 -08:00
|
|
|
|
2017-01-23 04:56:25 -08:00
|
|
|
if (rtc_use_quic) {
|
|
|
|
|
sources += [
|
|
|
|
|
"quicdatachannel.cc",
|
|
|
|
|
"quicdatachannel.h",
|
|
|
|
|
"quicdatatransport.cc",
|
|
|
|
|
"quicdatatransport.h",
|
|
|
|
|
]
|
|
|
|
|
deps += [ "//third_party/libquic" ]
|
|
|
|
|
public_deps = [
|
|
|
|
|
"//third_party/libquic",
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
|
|
2016-06-13 00:08:24 -07:00
|
|
|
if (rtc_include_tests) {
|
|
|
|
|
config("rtc_pc_unittests_config") {
|
|
|
|
|
# GN orders flags on a target before flags from configs. The default config
|
|
|
|
|
# adds -Wall, and this flag have to be after -Wall -- so they need to
|
|
|
|
|
# come from a config and can't be on the target directly.
|
|
|
|
|
if (!is_win && !is_clang) {
|
|
|
|
|
cflags = [ "-Wno-maybe-uninitialized" ] # Only exists for GCC.
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
|
|
2016-09-02 04:10:34 -07:00
|
|
|
rtc_test("rtc_pc_unittests") {
|
2016-06-13 00:08:24 -07:00
|
|
|
testonly = true
|
|
|
|
|
|
|
|
|
|
sources = [
|
|
|
|
|
"bundlefilter_unittest.cc",
|
|
|
|
|
"channel_unittest.cc",
|
|
|
|
|
"channelmanager_unittest.cc",
|
|
|
|
|
"currentspeakermonitor_unittest.cc",
|
|
|
|
|
"mediasession_unittest.cc",
|
|
|
|
|
"rtcpmuxfilter_unittest.cc",
|
2017-04-24 16:54:35 -07:00
|
|
|
"rtptransport_unittest.cc",
|
2017-07-19 13:38:02 -07:00
|
|
|
"rtptransporttestutil.h",
|
2016-06-13 00:08:24 -07:00
|
|
|
"srtpfilter_unittest.cc",
|
2017-07-07 14:26:25 -07:00
|
|
|
"srtpsession_unittest.cc",
|
|
|
|
|
"srtptestutil.h",
|
2017-07-19 13:38:02 -07:00
|
|
|
"srtptransport_unittest.cc",
|
2017-09-29 10:51:43 -07:00
|
|
|
"transportcontroller_unittest.cc",
|
2016-06-13 00:08:24 -07:00
|
|
|
]
|
|
|
|
|
|
|
|
|
|
include_dirs = [ "//third_party/libsrtp/srtp" ]
|
|
|
|
|
|
2016-09-05 01:35:44 -07:00
|
|
|
configs += [ ":rtc_pc_unittests_config" ]
|
2016-06-13 00:08:24 -07:00
|
|
|
|
2016-10-16 23:56:12 -07:00
|
|
|
if (!build_with_chromium && is_clang) {
|
2016-06-13 00:08:24 -07:00
|
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
2016-09-02 04:10:34 -07:00
|
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
2016-06-13 00:08:24 -07:00
|
|
|
}
|
|
|
|
|
|
|
|
|
|
if (is_win) {
|
|
|
|
|
libs = [ "strmiids.lib" ]
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
deps = [
|
2017-01-23 04:56:25 -08:00
|
|
|
":libjingle_peerconnection",
|
2017-09-29 10:51:43 -07:00
|
|
|
":pc_test_utils",
|
2016-06-13 00:08:24 -07:00
|
|
|
":rtc_pc",
|
2017-09-04 05:43:17 -07:00
|
|
|
"../api:array_view",
|
2017-06-01 13:01:48 -07:00
|
|
|
"../logging:rtc_event_log_api",
|
|
|
|
|
"../media:rtc_media_base",
|
2017-04-25 00:35:35 -07:00
|
|
|
"../media:rtc_media_tests_utils",
|
2017-06-01 13:01:48 -07:00
|
|
|
"../p2p:p2p_test_utils",
|
|
|
|
|
"../p2p:rtc_p2p",
|
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",
|
2016-06-13 00:08:24 -07:00
|
|
|
"../system_wrappers:metrics_default",
|
2017-07-19 13:38:02 -07:00
|
|
|
"../test:test_support",
|
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" ]
|
|
|
|
|
}
|
|
|
|
|
}
|
2017-01-23 04:56:25 -08:00
|
|
|
|
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",
|
|
|
|
|
"test/fakeperiodicvideocapturer.h",
|
|
|
|
|
"test/fakertccertificategenerator.h",
|
2017-09-29 10:51:43 -07:00
|
|
|
"test/faketransportcontroller.h",
|
2017-02-25 18:15:09 -08:00
|
|
|
"test/fakevideotrackrenderer.h",
|
|
|
|
|
"test/fakevideotracksource.h",
|
|
|
|
|
"test/mock_datachannel.h",
|
|
|
|
|
"test/mock_peerconnection.h",
|
|
|
|
|
"test/mock_webrtcsession.h",
|
|
|
|
|
"test/mockpeerconnectionobservers.h",
|
|
|
|
|
"test/peerconnectiontestwrapper.cc",
|
|
|
|
|
"test/peerconnectiontestwrapper.h",
|
|
|
|
|
"test/rtcstatsobtainer.h",
|
|
|
|
|
"test/testsdpstrings.h",
|
|
|
|
|
]
|
|
|
|
|
|
|
|
|
|
deps = [
|
|
|
|
|
":libjingle_peerconnection",
|
2017-09-29 10:51:43 -07:00
|
|
|
":rtc_pc_base",
|
2017-06-01 13:01:48 -07:00
|
|
|
"..:webrtc_common",
|
|
|
|
|
"../api:libjingle_peerconnection_test_api",
|
|
|
|
|
"../api:rtc_stats_api",
|
2017-06-15 12:52:32 -07:00
|
|
|
"../call:call_interfaces",
|
|
|
|
|
"../logging:rtc_event_log_api",
|
2017-06-01 13:01:48 -07:00
|
|
|
"../media:rtc_media",
|
|
|
|
|
"../media:rtc_media_tests_utils",
|
|
|
|
|
"../modules/audio_device:audio_device",
|
|
|
|
|
"../p2p:p2p_test_utils",
|
2017-07-19 10:40:47 -07:00
|
|
|
"../rtc_base:rtc_base",
|
|
|
|
|
"../rtc_base:rtc_base_approved",
|
|
|
|
|
"../rtc_base:rtc_base_tests_utils",
|
2017-06-01 13:01:48 -07:00
|
|
|
"../test:test_support",
|
2017-02-25 18:15:09 -08:00
|
|
|
"//testing/gmock",
|
|
|
|
|
]
|
|
|
|
|
|
|
|
|
|
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" ]
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
|
|
2017-01-23 04:56:25 -08:00
|
|
|
config("peerconnection_unittests_config") {
|
|
|
|
|
# The warnings below are enabled by default. Since GN orders compiler flags
|
|
|
|
|
# for a target before flags from configs, the only way to disable such
|
|
|
|
|
# warnings is by having them in a separate config, loaded from the target.
|
|
|
|
|
# TODO(kjellander): Make the code compile without disabling these flags.
|
|
|
|
|
# See https://bugs.webrtc.org/3307.
|
|
|
|
|
if (is_clang && is_win) {
|
|
|
|
|
cflags = [
|
|
|
|
|
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=6267
|
|
|
|
|
# for -Wno-sign-compare
|
|
|
|
|
"-Wno-sign-compare",
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
if (!is_win) {
|
|
|
|
|
cflags = [ "-Wno-sign-compare" ]
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
rtc_test("peerconnection_unittests") {
|
|
|
|
|
testonly = true
|
|
|
|
|
sources = [
|
|
|
|
|
"datachannel_unittest.cc",
|
|
|
|
|
"dtmfsender_unittest.cc",
|
2017-03-29 21:08:16 -07:00
|
|
|
"iceserverparsing_unittest.cc",
|
2017-01-23 04:56:25 -08:00
|
|
|
"jsepsessiondescription_unittest.cc",
|
|
|
|
|
"localaudiosource_unittest.cc",
|
|
|
|
|
"mediaconstraintsinterface_unittest.cc",
|
|
|
|
|
"mediastream_unittest.cc",
|
2017-10-06 11:20:31 -07:00
|
|
|
"peerconnection_crypto_unittest.cc",
|
2017-03-29 21:08:16 -07:00
|
|
|
"peerconnection_integrationtest.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",
|
2017-01-23 04:56:25 -08:00
|
|
|
"peerconnectionendtoend_unittest.cc",
|
|
|
|
|
"peerconnectionfactory_unittest.cc",
|
|
|
|
|
"peerconnectioninterface_unittest.cc",
|
2017-09-26 16:20:19 -07:00
|
|
|
"peerconnectionwrapper.cc",
|
|
|
|
|
"peerconnectionwrapper.h",
|
2017-01-23 04:56:25 -08:00
|
|
|
"proxy_unittest.cc",
|
|
|
|
|
"rtcstats_integrationtest.cc",
|
|
|
|
|
"rtcstatscollector_unittest.cc",
|
|
|
|
|
"rtpsenderreceiver_unittest.cc",
|
|
|
|
|
"sctputils_unittest.cc",
|
|
|
|
|
"statscollector_unittest.cc",
|
|
|
|
|
"test/fakeaudiocapturemodule_unittest.cc",
|
|
|
|
|
"test/testsdpstrings.h",
|
2017-02-11 19:07:31 -08:00
|
|
|
"trackmediainfomap_unittest.cc",
|
2017-01-23 04:56:25 -08:00
|
|
|
"videocapturertracksource_unittest.cc",
|
|
|
|
|
"videotrack_unittest.cc",
|
|
|
|
|
"webrtcsdp_unittest.cc",
|
|
|
|
|
"webrtcsession_unittest.cc",
|
|
|
|
|
]
|
|
|
|
|
|
|
|
|
|
if (rtc_enable_sctp) {
|
|
|
|
|
defines = [ "HAVE_SCTP" ]
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
configs += [ ":peerconnection_unittests_config" ]
|
|
|
|
|
|
|
|
|
|
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" ]
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
# TODO(jschuh): Bug 1348: fix this warning.
|
|
|
|
|
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
|
|
|
|
|
|
|
|
|
|
if (is_win) {
|
|
|
|
|
cflags = [
|
|
|
|
|
"/wd4245", # conversion from int to size_t, signed/unsigned mismatch.
|
|
|
|
|
"/wd4389", # signed/unsigned mismatch.
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
if (rtc_use_quic) {
|
|
|
|
|
public_deps = [
|
|
|
|
|
"//third_party/libquic",
|
|
|
|
|
]
|
|
|
|
|
sources += [
|
|
|
|
|
"quicdatachannel_unittest.cc",
|
|
|
|
|
"quicdatatransport_unittest.cc",
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
deps = []
|
|
|
|
|
if (is_android) {
|
2017-09-15 09:04:28 +02:00
|
|
|
deps += [ ":android_black_magic" ]
|
2017-01-23 04:56:25 -08:00
|
|
|
}
|
|
|
|
|
|
|
|
|
|
deps += [
|
|
|
|
|
":libjingle_peerconnection",
|
2017-02-25 18:15:09 -08:00
|
|
|
":pc_test_utils",
|
2017-01-23 04:56:25 -08:00
|
|
|
"..:webrtc_common",
|
|
|
|
|
"../api:fakemetricsobserver",
|
2017-09-15 09:04:28 +02:00
|
|
|
"../api:libjingle_peerconnection_test_api",
|
|
|
|
|
"../api:rtc_stats_api",
|
|
|
|
|
"../api/audio_codecs:builtin_audio_decoder_factory",
|
|
|
|
|
"../api/audio_codecs:builtin_audio_encoder_factory",
|
|
|
|
|
"../media:rtc_audio_video",
|
|
|
|
|
"../media:rtc_data", # TODO(phoglund): AFAIK only used for one sctp constant.
|
|
|
|
|
"../media:rtc_media_base",
|
2017-04-25 00:35:35 -07:00
|
|
|
"../media:rtc_media_tests_utils",
|
2017-09-15 09:04:28 +02:00
|
|
|
"../modules/audio_processing:audio_processing",
|
|
|
|
|
"../modules/utility:utility",
|
|
|
|
|
"../p2p:p2p_test_utils",
|
|
|
|
|
"../p2p:rtc_p2p",
|
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",
|
2017-09-15 09:04:28 +02:00
|
|
|
"../rtc_base:rtc_base_approved",
|
2017-07-19 10:40:47 -07:00
|
|
|
"../rtc_base:rtc_base_tests_main",
|
|
|
|
|
"../rtc_base:rtc_base_tests_utils",
|
2017-01-23 04:56:25 -08:00
|
|
|
"../system_wrappers:metrics_default",
|
2017-06-14 03:31:17 -07:00
|
|
|
"../test:audio_codec_mocks",
|
2017-09-15 09:04:28 +02:00
|
|
|
"../test:test_support",
|
2017-01-23 04:56:25 -08:00
|
|
|
]
|
|
|
|
|
|
|
|
|
|
if (is_android) {
|
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",
|
|
|
|
|
]
|
2017-01-23 04:56:25 -08:00
|
|
|
|
|
|
|
|
shard_timeout = 900
|
|
|
|
|
}
|
|
|
|
|
}
|
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",
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
}
|
2016-06-13 00:08:24 -07:00
|
|
|
}
|