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
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# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("//build/config/linux/pkg_config.gni")
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2017-01-24 06:58:22 -08:00
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import("../webrtc.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
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group("media") {
|
2018-01-15 10:20:00 -05:00
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deps = []
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if (!build_with_mozilla) {
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deps += [
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":rtc_media",
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|
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":rtc_media_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
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}
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config("rtc_media_defines_config") {
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defines = [
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"HAVE_WEBRTC_VIDEO",
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"HAVE_WEBRTC_VOICE",
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]
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}
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2017-06-21 01:02:59 -07:00
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rtc_source_set("rtc_h264_profile_id") {
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2018-01-10 15:54:53 +00:00
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visibility = [ "*" ]
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2017-06-21 01:02:59 -07:00
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sources = [
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"base/h264_profile_level_id.cc",
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"base/h264_profile_level_id.h",
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]
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if (!build_with_chromium && is_clang) {
|
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
|
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|
}
|
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deps = [
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"..:webrtc_common",
|
Remove CodecInst pt.2
The following APIs on AudioCodingModule are deprecated with this CL:
static int NumberOfCodecs();
static int Codec(int, CodecInst*);
static int Codec(const char*, CodecInst*, int, size_t);
static int Codec(const char*, int, size_t);
absl::optional<CodecInst> SendCodec() const;
bool RegisterReceiveCodec(int, const SdpAudioFormat&);
int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&);
int UnregisterReceiveCodec(uint8_t);
int32_t ReceiveCodec(CodecInst*);
absl::optional<SdpAudioFormat> ReceiveFormat();
As well as this method on RtpRtcp module:
int32_t RegisterSendPayload(const CodecInst&);
Bug: webrtc:7626
Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/113816
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26025}
2018-12-11 12:22:10 +01:00
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"../rtc_base:checks",
|
2017-07-19 10:40:47 -07:00
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"../rtc_base:rtc_base",
|
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"../rtc_base:rtc_base_approved",
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2018-10-16 14:13:50 +02:00
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"../rtc_base/system:rtc_export",
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2018-06-15 15:58:38 +02:00
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"//third_party/abseil-cpp/absl/types:optional",
|
2017-06-21 01:02:59 -07:00
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]
|
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}
|
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2018-01-23 10:37:42 +01:00
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rtc_source_set("rtc_media_config") {
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visibility = [ "*" ]
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sources = [
|
2019-01-11 09:11:00 -08:00
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"base/media_config.h",
|
2019-01-11 09:13:07 -08:00
|
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|
"base/mediaconfig.h",
|
2018-01-23 10:37:42 +01:00
|
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|
]
|
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}
|
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2018-06-28 10:59:02 -07:00
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rtc_source_set("rtc_vp9_profile") {
|
2018-12-04 11:00:52 -05:00
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visibility = [ "*" ]
|
2018-06-28 10:59:02 -07:00
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sources = [
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"base/vp9_profile.cc",
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"base/vp9_profile.h",
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]
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deps = [
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"..:webrtc_common",
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"../api/video_codecs:video_codecs_api",
|
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|
"../rtc_base:rtc_base_approved",
|
|
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"//third_party/abseil-cpp/absl/types:optional",
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|
]
|
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}
|
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2017-06-22 01:28:59 -07:00
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rtc_static_library("rtc_media_base") {
|
2018-01-10 15:54:53 +00:00
|
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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 = []
|
|
|
|
|
libs = []
|
2017-12-13 16:05:42 +01:00
|
|
|
deps = [
|
2018-11-28 16:47:49 +01:00
|
|
|
"../api:array_view",
|
2018-01-19 11:28:54 +01:00
|
|
|
"../api:audio_options_api",
|
2017-12-13 16:05:42 +01:00
|
|
|
"../rtc_base:checks",
|
2018-03-19 11:12:48 +01:00
|
|
|
"../rtc_base:rtc_base_approved",
|
|
|
|
|
"../rtc_base:rtc_task_queue",
|
2017-12-13 16:05:42 +01:00
|
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"../rtc_base:sanitizer",
|
2018-02-06 09:29:17 +01:00
|
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"../rtc_base:sequenced_task_checker",
|
2017-12-15 14:40:10 +01:00
|
|
|
"../rtc_base:stringutils",
|
2017-12-13 16:05:42 +01:00
|
|
|
]
|
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
|
|
|
sources = [
|
2019-01-11 09:11:00 -08:00
|
|
|
"base/adapted_video_track_source.cc",
|
|
|
|
|
"base/adapted_video_track_source.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"base/adaptedvideotracksource.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"base/audio_source.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"base/audiosource.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
|
|
|
"base/codec.cc",
|
|
|
|
|
"base/codec.h",
|
|
|
|
|
"base/device.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"base/media_channel.cc",
|
|
|
|
|
"base/media_channel.h",
|
|
|
|
|
"base/media_constants.cc",
|
|
|
|
|
"base/media_constants.h",
|
|
|
|
|
"base/media_engine.cc",
|
|
|
|
|
"base/media_engine.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"base/mediachannel.h",
|
|
|
|
|
"base/mediaconstants.h",
|
|
|
|
|
"base/mediaengine.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"base/rid_description.cc",
|
|
|
|
|
"base/rid_description.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"base/riddescription.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"base/rtp_data_engine.cc",
|
|
|
|
|
"base/rtp_data_engine.h",
|
|
|
|
|
"base/rtp_utils.cc",
|
|
|
|
|
"base/rtp_utils.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"base/rtpdataengine.h",
|
|
|
|
|
"base/rtputils.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"base/stream_params.cc",
|
|
|
|
|
"base/stream_params.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"base/streamparams.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"base/turn_utils.cc",
|
|
|
|
|
"base/turn_utils.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"base/turnutils.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"base/video_adapter.cc",
|
|
|
|
|
"base/video_adapter.h",
|
|
|
|
|
"base/video_broadcaster.cc",
|
|
|
|
|
"base/video_broadcaster.h",
|
|
|
|
|
"base/video_capturer.cc",
|
|
|
|
|
"base/video_capturer.h",
|
|
|
|
|
"base/video_common.cc",
|
|
|
|
|
"base/video_common.h",
|
|
|
|
|
"base/video_source_base.cc",
|
|
|
|
|
"base/video_source_base.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"base/videoadapter.h",
|
|
|
|
|
"base/videobroadcaster.h",
|
|
|
|
|
"base/videocapturer.h",
|
|
|
|
|
"base/videocommon.h",
|
|
|
|
|
"base/videosourcebase.h",
|
2016-11-25 00:40:18 -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" ]
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
deps += [
|
2018-01-11 17:07:30 +01:00
|
|
|
":rtc_h264_profile_id",
|
2018-01-23 10:37:42 +01:00
|
|
|
":rtc_media_config",
|
2018-06-28 10:59:02 -07:00
|
|
|
":rtc_vp9_profile",
|
2016-11-25 00:40:18 -08:00
|
|
|
"..:webrtc_common",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../api:libjingle_peerconnection_api",
|
2017-11-28 15:46:08 +01:00
|
|
|
"../api/audio_codecs:audio_codecs_api",
|
2018-10-02 14:47:06 +02:00
|
|
|
"../api/video:video_bitrate_allocation",
|
2018-05-11 11:15:30 +02:00
|
|
|
"../api/video:video_frame",
|
|
|
|
|
"../api/video:video_frame_i420",
|
2017-11-28 15:46:08 +01:00
|
|
|
"../api/video_codecs:video_codecs_api",
|
|
|
|
|
"../call:call_interfaces",
|
2018-01-04 12:14:35 +01:00
|
|
|
"../common_video",
|
2017-11-28 15:46:08 +01:00
|
|
|
"../modules/audio_processing:audio_processing_statistics",
|
2017-07-19 10:40:47 -07:00
|
|
|
"../rtc_base:rtc_base",
|
|
|
|
|
"../rtc_base:rtc_base_approved",
|
Reland "Export symbols needed by the Chromium component build (part 1)."
This reverts commit 16fe3f290a524a136f71660a114d0b03ef501f10.
Reason for revert:
After discussing this problem with nisse@ and yvesg@, we decided to modify
how RTC_EXPORT works and avoid to depend on the macro COMPONENT_BUILD.
RTC_EXPORT will instead depend on a macro WEBRTC_COMPONENT_BUILD (which
can be set as a GN argument which defaults to false).
When all the symbols needed by Chromium will be marked with RTC_EXPORT we
will flip the GN arg in Chromium, setting to to `component_build` and from
that moment, Chromium will depend on a WebRTC shared library when
`component_build=true`.
Original change's description:
> Revert "Export symbols needed by the Chromium component build (part 1)."
>
> This reverts commit 99eea42fc1fe0be0ebed13c5eba7e1e42059bc5a.
>
> Reason for revert:
> lld-link: error: undefined symbol: "__declspec(dllimport) bool __cdecl cricket::UnwrapTurnPacket(unsigned char const *, unsigned int, unsigned int *, unsigned int *)" (__imp_?UnwrapTurnPacket@cricket@@YA_NPBEIPAI1@Z)
> >>> referenced by obj/services/network/network_service/socket_manager.obj:("virtual void __thiscall network::P2PSocketManager::DumpPacket(class base::span<unsigned char const, 4294967295>, bool)" (?DumpPacket@P2PSocketManager@network@@EAEXV?$span@$$CBE$0PPPPPPPP@@base@@_N@Z))
> lld-link: error: undefined symbol: "__declspec(dllimport) bool __cdecl cricket::ValidateRtpHeader(unsigned char const *, unsigned int, unsigned int *)" (__imp_?ValidateRtpHeader@cricket@@YA_NPBEIPAI@Z)
> >>> referenced by obj/services/network/network_service/socket_manager.obj:("virtual void __thiscall network::P2PSocketManager::DumpPacket(class base::span<unsigned char const, 4294967295>, bool)" (?DumpPacket@P2PSocketManager@network@@EAEXV?$span@$$CBE$0PPPPPPPP@@base@@_N@Z))
> lld-link: error: undefined symbol: "__declspec(dllimport) bool __cdecl cricket::ApplyPacketOptions(unsigned char *, unsigned int, struct rtc::PacketTimeUpdateParams const &, unsigned __int64)" (__imp_?ApplyPacketOptions@cricket@@YA_NPAEIABUPacketTimeUpdateParams@rtc@@_K@Z)
> >>> referenced by obj/services/network/network_service/socket_tcp.obj:("virtual void __thiscall network::P2PSocketTcp::DoSend(class net::IPEndPoint const &, class std::vector<signed char, class std::allocator<signed char>> const &, struct rtc::PacketOptions const &, struct net::NetworkTrafficAnnotationTag)" (?DoSend@P2PSocketTcp@network@@MAEXABVIPEndPoint@net@@ABV?$vector@CV?$allocator@C@std@@@std@@ABUPacketOptions@rtc@@UNetworkTrafficAnnotationTag@4@@Z))
> >>> referenced by obj/services/network/network_service/socket_tcp.obj:("virtual void __thiscall network::P2PSocketStunTcp::DoSend(class net::IPEndPoint const &, class std::vector<signed char, class std::allocator<signed char>> const &, struct rtc::PacketOptions const &, struct net::NetworkTrafficAnnotationTag)" (?DoSend@P2PSocketStunTcp@network@@MAEXABVIPEndPoint@net@@ABV?$vector@CV?$allocator@C@std@@@std@@ABUPacketOptions@rtc@@UNetworkTrafficAnnotationTag@4@@Z))
> lld-link: error: undefined symbol: "__declspec(dllimport) bool __cdecl cricket::ApplyPacketOptions(unsigned char *, unsigned int, struct rtc::PacketTimeUpdateParams const &, unsigned __int64)" (__imp_?ApplyPacketOptions@cricket@@YA_NPAEIABUPacketTimeUpdateParams@rtc@@_K@Z)
> >>> referenced by obj/services/network/network_service/socket_udp.obj:("bool __thiscall network::P2PSocketUdp::DoSend(struct network::P2PSocketUdp::PendingPacket const &)" (?DoSend@P2PSocketUdp@network@@AAE_NABUPendingPacket@12@@Z))
>
> Original change's description:
> > Reland "Reland "Export symbols needed by the Chromium component build (part 1).""
> >
> > This reverts commit b49520bfc08f5c5832dda1d642125f0bb898f974.
> >
> > Reason for revert: Problem fixed in https://chromium-review.googlesource.com/c/chromium/src/+/1261398.
> >
> > Original change's description:
> > > Revert "Reland "Export symbols needed by the Chromium component build (part 1).""
> > >
> > > This reverts commit 588f4642d1a29f7beaf28265dbd08728191b4c52.
> > >
> > > Reason for revert: Breaks WebRTC Chromium FYI Win Builder (dbg).
> > > lld-link: error: undefined symbol: "__declspec(dllimport) __thiscall webrtc::Config::Config(void)" (__imp_??0Config@webrtc@@QAE@XZ)
> > > [...]
> > >
> > > Original change's description:
> > > > Reland "Export symbols needed by the Chromium component build (part 1)."
> > > >
> > > > This reverts commit 2ea9af227517556136fd629dd2663c0d75d77c7b.
> > > >
> > > > Reason for revert: The problem will be fixed by
> > > > https://chromium-review.googlesource.com/c/chromium/src/+/1261122.
> > > >
> > > > Original change's description:
> > > > > Revert "Export symbols needed by the Chromium component build (part 1)."
> > > > >
> > > > > This reverts commit 9e24dcff167c4eb3555bf0ce6eaba090c10fbe53.
> > > > >
> > > > > Reason for revert: Breaks chromium.webrtc.fyi bots.
> > > > >
> > > > > Original change's description:
> > > > > > Export symbols needed by the Chromium component build (part 1).
> > > > > >
> > > > > > 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).
> > > > > >
> > > > > > [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
> > > > > >
> > > > > > Bug: webrtc:9419
> > > > > > Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/103505
> > > > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > > > > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > > > > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > > > > > Cr-Commit-Position: refs/heads/master@{#24969}
> > > > >
> > > > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
> > > > >
> > > > > Change-Id: I01f6e18f0d2c0f0309cdaa6c943c3927e1f1f49f
> > > > > No-Presubmit: true
> > > > > No-Tree-Checks: true
> > > > > No-Try: true
> > > > > Bug: webrtc:9419
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/103720
> > > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#24974}
> > > >
> > > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
> > > >
> > > > Change-Id: I83bbc7f550fc23e823c4d055e0a6f60c828960dd
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: webrtc:9419
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/103740
> > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#24980}
> > >
> > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
> > >
> > > Change-Id: I4b7cfe492f2c8eeda5c8ac52520e0cfc95ade9b0
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: webrtc:9419
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/103801
> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#24983}
> >
> > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
> >
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> >
> > Bug: webrtc:9419
> > Change-Id: Id986a0a03cdc2818690337784396882af067f7fa
> > Reviewed-on: https://webrtc-review.googlesource.com/c/104602
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25049}
>
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
>
> Change-Id: I6f58b9c90defccdb160307783fb55271ab424fa1
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9419
> Reviewed-on: https://webrtc-review.googlesource.com/c/104623
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25050}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
Change-Id: I4d01ed96ae40a8f9ca42c466be5c87653d75d7c1
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/104641
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25108}
2018-10-11 10:50:45 +02:00
|
|
|
"../rtc_base/system:rtc_export",
|
2018-07-25 15:04:28 +02:00
|
|
|
"../rtc_base/third_party/sigslot",
|
2018-10-19 15:29:54 +02:00
|
|
|
"//third_party/abseil-cpp/absl/strings",
|
2018-06-15 15:58:38 +02:00
|
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
2016-11-25 00:40:18 -08:00
|
|
|
]
|
2017-08-23 14:19:50 -07:00
|
|
|
|
2018-01-15 10:20:00 -05:00
|
|
|
if (!build_with_mozilla) {
|
|
|
|
|
deps += [ "../p2p" ]
|
|
|
|
|
}
|
|
|
|
|
|
2017-04-21 05:17:08 -07:00
|
|
|
if (is_nacl) {
|
|
|
|
|
deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
|
|
|
|
|
}
|
2016-11-25 00:40:18 -08:00
|
|
|
}
|
|
|
|
|
|
2018-01-04 15:10:22 +01:00
|
|
|
rtc_static_library("rtc_constants") {
|
2016-11-25 00:40:18 -08:00
|
|
|
defines = []
|
|
|
|
|
libs = []
|
2018-01-04 15:10:22 +01:00
|
|
|
deps = []
|
2016-11-25 00:40:18 -08:00
|
|
|
sources = [
|
2017-11-20 17:48:13 +05:30
|
|
|
"engine/constants.cc",
|
2017-09-14 13:53:02 -07:00
|
|
|
"engine/constants.h",
|
2018-01-04 15:10:22 +01:00
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
|
2018-10-19 00:51:18 -07:00
|
|
|
rtc_static_library("rtc_simulcast_encoder_adapter") {
|
|
|
|
|
visibility = [ "*" ]
|
|
|
|
|
defines = []
|
|
|
|
|
libs = []
|
|
|
|
|
sources = [
|
|
|
|
|
"engine/simulcast_encoder_adapter.cc",
|
|
|
|
|
"engine/simulcast_encoder_adapter.h",
|
|
|
|
|
]
|
|
|
|
|
deps = [
|
|
|
|
|
"../api/video:video_bitrate_allocation",
|
2018-11-28 16:47:49 +01:00
|
|
|
"../api/video:video_frame",
|
2018-10-19 00:51:18 -07:00
|
|
|
"../api/video:video_frame_i420",
|
|
|
|
|
"../api/video_codecs:video_codecs_api",
|
|
|
|
|
"../modules/video_coding:video_codec_interface",
|
|
|
|
|
"../modules/video_coding:video_coding_utility",
|
|
|
|
|
"../rtc_base:checks",
|
|
|
|
|
"../rtc_base:rtc_base_approved",
|
|
|
|
|
"../rtc_base:sequenced_task_checker",
|
|
|
|
|
"../system_wrappers",
|
|
|
|
|
"../system_wrappers:field_trial",
|
|
|
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
|
|
|
"//third_party/libyuv",
|
|
|
|
|
]
|
|
|
|
|
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" ]
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
|
|
2018-12-06 13:38:24 +01:00
|
|
|
rtc_static_library("rtc_encoder_simulcast_proxy") {
|
|
|
|
|
visibility = [ "*" ]
|
|
|
|
|
defines = []
|
|
|
|
|
libs = []
|
|
|
|
|
sources = [
|
|
|
|
|
"engine/encoder_simulcast_proxy.cc",
|
|
|
|
|
"engine/encoder_simulcast_proxy.h",
|
|
|
|
|
]
|
|
|
|
|
deps = [
|
|
|
|
|
":rtc_simulcast_encoder_adapter",
|
|
|
|
|
"../:webrtc_common",
|
|
|
|
|
"../api/video:video_bitrate_allocation",
|
|
|
|
|
"../api/video:video_frame",
|
|
|
|
|
"../api/video_codecs:video_codecs_api",
|
|
|
|
|
"../modules/video_coding:video_codec_interface",
|
|
|
|
|
]
|
|
|
|
|
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" ]
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
|
|
2018-01-04 15:10:22 +01:00
|
|
|
rtc_static_library("rtc_internal_video_codecs") {
|
2018-01-10 15:57:32 +00:00
|
|
|
visibility = [ "*" ]
|
2018-04-27 04:31:53 +02:00
|
|
|
allow_poison = [ "software_video_codecs" ]
|
2018-01-04 15:10:22 +01:00
|
|
|
defines = []
|
|
|
|
|
libs = []
|
2018-01-08 11:05:10 +01:00
|
|
|
deps = [
|
2018-12-06 13:38:24 +01:00
|
|
|
":rtc_encoder_simulcast_proxy",
|
2018-01-11 17:07:30 +01:00
|
|
|
":rtc_h264_profile_id",
|
2018-10-19 00:51:18 -07:00
|
|
|
":rtc_simulcast_encoder_adapter",
|
2018-11-28 16:47:49 +01:00
|
|
|
"../:webrtc_common",
|
|
|
|
|
"../api/video:encoded_image",
|
|
|
|
|
"../api/video:video_frame",
|
2018-01-08 11:05:10 +01:00
|
|
|
"../modules/video_coding:video_codec_interface",
|
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",
|
2018-01-08 11:05:10 +01:00
|
|
|
]
|
2018-01-04 15:10:22 +01:00
|
|
|
sources = [
|
2017-10-31 11:24:54 +01:00
|
|
|
"engine/convert_legacy_video_factory.cc",
|
|
|
|
|
"engine/convert_legacy_video_factory.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"engine/internal_decoder_factory.cc",
|
|
|
|
|
"engine/internal_decoder_factory.h",
|
|
|
|
|
"engine/internal_encoder_factory.cc",
|
|
|
|
|
"engine/internal_encoder_factory.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"engine/internaldecoderfactory.h",
|
|
|
|
|
"engine/internalencoderfactory.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"engine/multiplex_codec_factory.cc",
|
|
|
|
|
"engine/multiplex_codec_factory.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"engine/multiplexcodecfactory.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"engine/scoped_video_decoder.cc",
|
|
|
|
|
"engine/scoped_video_decoder.h",
|
|
|
|
|
"engine/scoped_video_encoder.cc",
|
|
|
|
|
"engine/scoped_video_encoder.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"engine/scopedvideodecoder.h",
|
|
|
|
|
"engine/scopedvideoencoder.h",
|
2018-10-24 17:21:36 -07:00
|
|
|
|
|
|
|
|
# TODO(bugs.webrtc.org/7925): stop exporting this header once downstream
|
2018-12-06 13:38:24 +01:00
|
|
|
# targets depend on :rtc_encoder_simulcast_proxy directly.
|
|
|
|
|
"engine/encoder_simulcast_proxy.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"engine/webrtc_video_decoder_factory.h",
|
|
|
|
|
"engine/webrtc_video_encoder_factory.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"engine/webrtcvideodecoderfactory.h",
|
|
|
|
|
"engine/webrtcvideoencoderfactory.h",
|
2018-01-04 15:10:22 +01:00
|
|
|
]
|
|
|
|
|
|
|
|
|
|
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" ]
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
include_dirs = []
|
|
|
|
|
|
|
|
|
|
public_configs = []
|
|
|
|
|
deps += [
|
|
|
|
|
":rtc_constants",
|
|
|
|
|
":rtc_media_base",
|
2018-05-22 15:37:23 +02:00
|
|
|
"../api/video:video_bitrate_allocation",
|
2018-11-20 11:06:58 +01:00
|
|
|
"../api/video:video_frame",
|
2018-06-12 11:15:56 +02:00
|
|
|
"../api/video_codecs:rtc_software_fallback_wrappers",
|
2018-01-04 15:10:22 +01:00
|
|
|
"../api/video_codecs:video_codecs_api",
|
|
|
|
|
"../call:call_interfaces",
|
|
|
|
|
"../call:video_stream_api",
|
|
|
|
|
"../modules/video_coding:webrtc_h264",
|
2018-01-25 13:01:09 -08:00
|
|
|
"../modules/video_coding:webrtc_multiplex",
|
2018-01-04 15:10:22 +01:00
|
|
|
"../modules/video_coding:webrtc_vp8",
|
|
|
|
|
"../modules/video_coding:webrtc_vp9",
|
|
|
|
|
"../rtc_base:checks",
|
2018-12-11 15:44:54 +01:00
|
|
|
"../rtc_base:deprecation",
|
2018-01-04 15:10:22 +01:00
|
|
|
"../rtc_base:rtc_base_approved",
|
2018-10-16 14:13:50 +02:00
|
|
|
"../rtc_base/system:rtc_export",
|
2018-10-23 10:07:25 +02:00
|
|
|
"//third_party/abseil-cpp/absl/strings",
|
2018-01-04 15:10:22 +01:00
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
rtc_static_library("rtc_audio_video") {
|
2018-01-10 15:54:53 +00:00
|
|
|
visibility = [ "*" ]
|
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.
|
|
|
|
|
]
|
2018-01-04 15:10:22 +01:00
|
|
|
defines = []
|
|
|
|
|
libs = []
|
|
|
|
|
deps = [
|
2018-11-28 16:47:49 +01:00
|
|
|
"../api/video:video_bitrate_allocation",
|
2018-11-08 10:02:56 -08:00
|
|
|
"../api/video:video_bitrate_allocator_factory",
|
2018-11-07 14:29:54 +00:00
|
|
|
"../modules/audio_processing:api",
|
2018-11-28 16:47:49 +01:00
|
|
|
"../modules/audio_processing:gain_control_interface",
|
2018-01-26 15:11:40 +01:00
|
|
|
"../modules/audio_processing/aec_dump:aec_dump",
|
2018-01-16 10:11:40 +01:00
|
|
|
"../modules/video_coding:video_codec_interface",
|
2018-01-08 11:05:10 +01:00
|
|
|
"../modules/video_coding:video_coding",
|
2018-06-21 16:16:38 +02:00
|
|
|
"../modules/video_coding:video_coding_utility",
|
2018-04-03 13:40:05 +02:00
|
|
|
"../rtc_base:audio_format_to_string",
|
2018-01-04 15:10:22 +01:00
|
|
|
"../rtc_base:checks",
|
2018-10-16 14:13:50 +02:00
|
|
|
"../rtc_base/system:rtc_export",
|
2018-07-23 17:38:12 +02:00
|
|
|
"../rtc_base/third_party/base64",
|
2018-09-28 08:51:10 +02:00
|
|
|
"../system_wrappers:field_trial",
|
|
|
|
|
"../system_wrappers:metrics",
|
2018-01-04 15:10:22 +01:00
|
|
|
]
|
|
|
|
|
|
|
|
|
|
sources = [
|
|
|
|
|
"engine/adm_helpers.cc",
|
|
|
|
|
"engine/adm_helpers.h",
|
|
|
|
|
"engine/apm_helpers.cc",
|
|
|
|
|
"engine/apm_helpers.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"engine/null_webrtc_video_engine.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"engine/nullwebrtcvideoengine.h",
|
2018-01-04 15:10:22 +01:00
|
|
|
"engine/payload_type_mapper.cc",
|
|
|
|
|
"engine/payload_type_mapper.h",
|
2018-01-30 10:32:13 +01:00
|
|
|
"engine/simulcast.cc",
|
|
|
|
|
"engine/simulcast.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"engine/webrtc_media_engine.cc",
|
|
|
|
|
"engine/webrtc_media_engine.h",
|
|
|
|
|
"engine/webrtc_video_decoder_factory.cc",
|
|
|
|
|
"engine/webrtc_video_decoder_factory.h",
|
|
|
|
|
"engine/webrtc_video_encoder_factory.cc",
|
|
|
|
|
"engine/webrtc_video_encoder_factory.h",
|
|
|
|
|
"engine/webrtc_video_engine.cc",
|
|
|
|
|
"engine/webrtc_video_engine.h",
|
|
|
|
|
"engine/webrtc_voice_engine.cc",
|
|
|
|
|
"engine/webrtc_voice_engine.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"engine/webrtcmediaengine.h",
|
|
|
|
|
"engine/webrtcvideodecoderfactory.h",
|
|
|
|
|
"engine/webrtcvideoencoderfactory.h",
|
|
|
|
|
"engine/webrtcvideoengine.h",
|
|
|
|
|
"engine/webrtcvoiceengine.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
|
|
|
]
|
|
|
|
|
|
2016-10-16 23:56:12 -07:00
|
|
|
if (!build_with_chromium && is_clang) {
|
2016-06-12 22:12:01 -07:00
|
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
2016-09-26 06:11:53 -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
|
|
|
}
|
|
|
|
|
|
2017-02-01 17:31:11 -08:00
|
|
|
if (rtc_opus_support_120ms_ptime) {
|
|
|
|
|
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
|
|
|
|
|
} else {
|
|
|
|
|
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
|
|
|
|
|
}
|
|
|
|
|
|
2016-08-19 01:07:30 -07:00
|
|
|
include_dirs = []
|
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
|
|
|
|
2016-09-05 06:10:18 -07:00
|
|
|
public_configs = []
|
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 (build_with_chromium) {
|
|
|
|
|
deps += [ "../modules/video_capture:video_capture" ]
|
|
|
|
|
} else {
|
|
|
|
|
public_configs += [ ":rtc_media_defines_config" ]
|
|
|
|
|
deps += [ "../modules/video_capture:video_capture_internal_impl" ]
|
|
|
|
|
}
|
Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2910633002/ )
Reason for revert:
Revert of revert of revert of revert of 'Activating..'. Or "reland of reland of 'Activate..'".
*Now* the internal projects are fixed and the fix is verified.
Original issue's description:
> Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2903153005/ )
>
> Reason for revert:
> Reverting again: internal project issues were apparently not completely fixed.
>
> Original issue's description:
> > Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2904893002/ )
> >
> > Reason for revert:
> > Revert the revert now that internal projects are updated.
> >
> > Original issue's description:
> > > Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ )
> > >
> > > Reason for revert:
> > > Breaks internal project.
> > >
> > > Original issue's description:
> > > > Activate 'offload debug dump recordings from audio thread to TaskQueue'.
> > > >
> > > > A low priority task queue is added to WebRTCVoiceEngine. The
> > > > start/stop debug calls make file logging happen on the task queue.
> > > >
> > > > In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> > > > so that it can be shared for low priority tasks between different
> > > > subcomponents. It will require some changes to MediaEngine,
> > > > CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> > > > projects.
> > > >
> > > > A task queue must be created and destroyed from the same thread. With
> > > > this CL that will be the worker thread, which creates and destroys
> > > > WebRTCVoiceEngine. With the dependent CL, it will probably change to
> > > > the signaling thread.
> > > >
> > > > NOTRY=True # tests just passed
> > > >
> > > > BUG=webrtc:7404
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2896813002
> > > > Cr-Commit-Position: refs/heads/master@{#18252}
> > > > Committed: https://chromium.googlesource.com/external/webrtc/+/c61bf947b4ac31f3500858ffcae6fee39d799930
> > >
> > > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:7404
> > >
> > > Review-Url: https://codereview.webrtc.org/2904893002
> > > Cr-Commit-Position: refs/heads/master@{#18255}
> > > Committed: https://chromium.googlesource.com/external/webrtc/+/be68b72cfad0686dcd892bba1368b199a7ee16ca
> >
> > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:7404
> >
> > Review-Url: https://codereview.webrtc.org/2903153005
> > Cr-Commit-Position: refs/heads/master@{#18270}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/d2303a2338106feab684860f1c133877b46bdd4f
>
> TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2910633002
> Cr-Commit-Position: refs/heads/master@{#18272}
> Committed: https://chromium.googlesource.com/external/webrtc/+/fe9ecb07ea8254d8a09605f25203a4d045b3ffee
TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2904423002
Cr-Commit-Position: refs/heads/master@{#18300}
2017-05-29 02:56:27 -07:00
|
|
|
if (rtc_enable_protobuf) {
|
|
|
|
|
deps += [ "../modules/audio_processing/aec_dump:aec_dump_impl" ]
|
|
|
|
|
} else {
|
|
|
|
|
deps += [ "../modules/audio_processing/aec_dump:null_aec_dump_factory" ]
|
|
|
|
|
}
|
2018-01-30 10:32:13 +01:00
|
|
|
if (rtc_use_builtin_sw_codecs) {
|
|
|
|
|
deps += [ ":rtc_internal_video_codecs" ]
|
|
|
|
|
}
|
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 += [
|
2018-01-04 15:10:22 +01:00
|
|
|
":rtc_constants",
|
2016-11-25 00:40:18 -08:00
|
|
|
":rtc_media_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
|
|
|
"..:webrtc_common",
|
2016-08-31 07:33:05 -07:00
|
|
|
"../api:call_api",
|
2017-06-15 12:52:32 -07:00
|
|
|
"../api:libjingle_peerconnection_api",
|
2016-11-28 07:02:13 -08:00
|
|
|
"../api:transport_api",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../api/audio_codecs:audio_codecs_api",
|
2018-11-08 10:02:56 -08:00
|
|
|
"../api/video:builtin_video_bitrate_allocator_factory",
|
2018-05-11 11:15:30 +02:00
|
|
|
"../api/video:video_frame",
|
|
|
|
|
"../api/video:video_frame_i420",
|
2018-06-12 11:15:56 +02:00
|
|
|
"../api/video_codecs:rtc_software_fallback_wrappers",
|
2017-04-05 03:02:20 -07:00
|
|
|
"../api/video_codecs:video_codecs_api",
|
2016-06-12 22:12:01 -07:00
|
|
|
"../call",
|
2018-01-04 15:10:22 +01:00
|
|
|
"../call:call_interfaces",
|
2017-08-22 05:43:23 -07:00
|
|
|
"../call:video_stream_api",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../common_video:common_video",
|
|
|
|
|
"../modules/audio_device:audio_device",
|
2018-03-15 12:44:12 +01:00
|
|
|
"../modules/audio_device:audio_device_impl",
|
2016-11-17 06:48:48 -08:00
|
|
|
"../modules/audio_mixer:audio_mixer_impl",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../modules/audio_processing:audio_processing",
|
|
|
|
|
"../modules/video_capture:video_capture_module",
|
2017-06-15 12:52:32 -07:00
|
|
|
"../pc:rtc_pc_base",
|
2017-07-19 10:40:47 -07:00
|
|
|
"../rtc_base:rtc_base",
|
|
|
|
|
"../rtc_base:rtc_task_queue",
|
2018-01-04 15:10:22 +01:00
|
|
|
"../rtc_base:stringutils",
|
2018-10-12 17:36:57 +02:00
|
|
|
"../rtc_base/experiments:normalize_simulcast_size_experiment",
|
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
|
|
|
"../system_wrappers",
|
2018-11-16 09:54:32 +01:00
|
|
|
"//third_party/abseil-cpp/absl/memory",
|
2018-10-19 15:29:54 +02:00
|
|
|
"//third_party/abseil-cpp/absl/strings",
|
2018-06-15 15:58:38 +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
|
|
|
]
|
|
|
|
|
}
|
2016-06-12 22:12:01 -07:00
|
|
|
|
2017-06-15 12:52:32 -07:00
|
|
|
rtc_static_library("rtc_data") {
|
|
|
|
|
defines = []
|
|
|
|
|
deps = []
|
|
|
|
|
|
|
|
|
|
if (rtc_enable_sctp) {
|
|
|
|
|
sources = [
|
2019-01-11 09:11:00 -08:00
|
|
|
"sctp/sctp_transport.cc",
|
|
|
|
|
"sctp/sctp_transport.h",
|
|
|
|
|
"sctp/sctp_transport_internal.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"sctp/sctptransport.h",
|
|
|
|
|
"sctp/sctptransportinternal.h",
|
2017-06-15 12:52:32 -07:00
|
|
|
]
|
2017-12-01 15:26:41 +01:00
|
|
|
} else {
|
|
|
|
|
# libtool on mac does not like empty targets.
|
|
|
|
|
sources = [
|
2017-12-13 07:40:54 +01:00
|
|
|
"sctp/noop.cc",
|
2017-12-01 15:26:41 +01:00
|
|
|
]
|
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" ]
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
if (rtc_enable_sctp && rtc_build_usrsctp) {
|
|
|
|
|
include_dirs = [
|
|
|
|
|
# TODO(jiayl): move this into the public_configs of
|
|
|
|
|
# //third_party/usrsctp/BUILD.gn.
|
|
|
|
|
"//third_party/usrsctp/usrsctplib",
|
|
|
|
|
]
|
|
|
|
|
deps += [ "//third_party/usrsctp" ]
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
deps += [
|
|
|
|
|
":rtc_media_base",
|
|
|
|
|
"..:webrtc_common",
|
|
|
|
|
"../api:call_api",
|
|
|
|
|
"../api:transport_api",
|
|
|
|
|
"../p2p:rtc_p2p",
|
2017-07-19 10:40:47 -07:00
|
|
|
"../rtc_base:rtc_base",
|
|
|
|
|
"../rtc_base:rtc_base_approved",
|
2018-07-25 15:04:28 +02:00
|
|
|
"../rtc_base/third_party/sigslot",
|
2017-06-15 12:52:32 -07:00
|
|
|
"../system_wrappers",
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
rtc_source_set("rtc_media") {
|
2018-01-10 15:54:53 +00:00
|
|
|
visibility = [ "*" ]
|
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.
|
|
|
|
|
]
|
2018-01-11 17:07:30 +01:00
|
|
|
deps = [
|
2017-06-15 12:52:32 -07:00
|
|
|
":rtc_audio_video",
|
|
|
|
|
":rtc_data",
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
|
2016-06-12 22:12:01 -07:00
|
|
|
if (rtc_include_tests) {
|
2017-04-25 00:35:35 -07:00
|
|
|
rtc_source_set("rtc_media_tests_utils") {
|
2016-06-12 22:12:01 -07:00
|
|
|
testonly = true
|
|
|
|
|
|
2018-05-14 09:48:06 +02:00
|
|
|
defines = []
|
2016-08-19 01:07:30 -07:00
|
|
|
include_dirs = []
|
2017-03-06 06:04:55 -08:00
|
|
|
deps = [
|
2018-01-11 17:07:30 +01:00
|
|
|
":rtc_audio_video",
|
2018-10-19 00:51:18 -07:00
|
|
|
":rtc_simulcast_encoder_adapter",
|
2018-01-04 12:14:35 +01:00
|
|
|
"../api:libjingle_peerconnection_api",
|
2018-11-28 16:47:49 +01:00
|
|
|
"../api/video:encoded_image",
|
2018-05-11 11:15:30 +02:00
|
|
|
"../api/video:video_frame_i420",
|
2017-08-22 05:43:23 -07:00
|
|
|
"../call:video_stream_api",
|
2018-01-04 12:14:35 +01:00
|
|
|
"../common_video:common_video",
|
2018-11-07 14:29:54 +00:00
|
|
|
"../modules/audio_processing:api",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../modules/audio_processing:audio_processing",
|
2017-12-06 07:51:33 +01:00
|
|
|
"../modules/rtp_rtcp:rtp_rtcp_format",
|
2018-01-08 11:05:10 +01:00
|
|
|
"../modules/video_coding:video_codec_interface",
|
2017-09-18 03:08:08 -07:00
|
|
|
"../modules/video_coding:video_coding_utility",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../p2p:rtc_p2p",
|
2017-12-13 16:05:42 +01:00
|
|
|
"../rtc_base:checks",
|
2018-11-23 13:15:08 +01:00
|
|
|
"../rtc_base:gunit_helpers",
|
2018-03-19 11:12:48 +01:00
|
|
|
"../rtc_base:rtc_task_queue",
|
2017-12-15 14:40:10 +01:00
|
|
|
"../rtc_base:stringutils",
|
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",
|
2018-10-30 22:23:02 +01:00
|
|
|
"//third_party/abseil-cpp/absl/strings",
|
2017-03-06 06:04:55 -08:00
|
|
|
]
|
2016-06-12 22:12:01 -07:00
|
|
|
sources = [
|
2019-01-11 09:11:00 -08:00
|
|
|
"base/fake_frame_source.cc",
|
|
|
|
|
"base/fake_frame_source.h",
|
|
|
|
|
"base/fake_media_engine.cc",
|
|
|
|
|
"base/fake_media_engine.h",
|
|
|
|
|
"base/fake_network_interface.h",
|
|
|
|
|
"base/fake_rtp.cc",
|
|
|
|
|
"base/fake_rtp.h",
|
|
|
|
|
"base/fake_video_capturer.cc",
|
|
|
|
|
"base/fake_video_capturer.h",
|
|
|
|
|
"base/fake_video_renderer.cc",
|
|
|
|
|
"base/fake_video_renderer.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"base/fakeframesource.h",
|
|
|
|
|
"base/fakemediaengine.h",
|
|
|
|
|
"base/fakenetworkinterface.h",
|
|
|
|
|
"base/fakertp.h",
|
|
|
|
|
"base/fakevideocapturer.h",
|
|
|
|
|
"base/fakevideorenderer.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"base/test_utils.cc",
|
|
|
|
|
"base/test_utils.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"base/testutils.h",
|
2019-01-11 09:11:00 -08:00
|
|
|
"engine/fake_webrtc_call.cc",
|
|
|
|
|
"engine/fake_webrtc_call.h",
|
|
|
|
|
"engine/fake_webrtc_video_engine.cc",
|
|
|
|
|
"engine/fake_webrtc_video_engine.h",
|
2019-01-11 09:13:07 -08:00
|
|
|
"engine/fakewebrtccall.h",
|
|
|
|
|
"engine/fakewebrtcvideoengine.h",
|
2016-06-12 22:12:01 -07:00
|
|
|
]
|
|
|
|
|
|
2016-10-16 23:56:12 -07:00
|
|
|
if (!build_with_chromium && is_clang) {
|
|
|
|
|
# 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" ]
|
2016-06-12 22:12:01 -07:00
|
|
|
}
|
|
|
|
|
|
2018-05-14 09:48:06 +02:00
|
|
|
if (rtc_use_h264) {
|
|
|
|
|
defines += [ "WEBRTC_USE_H264" ]
|
|
|
|
|
}
|
|
|
|
|
|
2016-06-12 22:12:01 -07:00
|
|
|
deps += [
|
2018-01-04 15:10:22 +01:00
|
|
|
":rtc_internal_video_codecs",
|
2017-03-06 06:04:55 -08:00
|
|
|
":rtc_media",
|
|
|
|
|
":rtc_media_base",
|
|
|
|
|
"..:webrtc_common",
|
|
|
|
|
"../api:call_api",
|
2018-05-22 15:37:23 +02:00
|
|
|
"../api/video:video_bitrate_allocation",
|
2018-05-11 11:15:30 +02:00
|
|
|
"../api/video:video_frame",
|
2017-04-05 03:02:20 -07:00
|
|
|
"../api/video_codecs:video_codecs_api",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../call:call_interfaces",
|
2018-02-21 13:07:13 +01:00
|
|
|
"../call:mock_rtp_interfaces",
|
2017-07-19 10:40:47 -07:00
|
|
|
"../rtc_base:rtc_base",
|
|
|
|
|
"../rtc_base:rtc_base_approved",
|
2018-02-13 19:47:50 +01:00
|
|
|
"../rtc_base:rtc_task_queue_for_test",
|
2018-07-25 15:04:28 +02:00
|
|
|
"../rtc_base/third_party/sigslot",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../test:test_support",
|
2016-06-12 22:12:01 -07:00
|
|
|
"//testing/gtest",
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
|
2016-09-14 05:10:01 -07:00
|
|
|
rtc_media_unittests_resources = [
|
2017-09-15 06:47:31 +02:00
|
|
|
"../resources/media/captured-320x240-2s-48.frames",
|
|
|
|
|
"../resources/media/faces.1280x720_P420.yuv",
|
|
|
|
|
"../resources/media/faces_I420.jpg",
|
|
|
|
|
"../resources/media/faces_I422.jpg",
|
|
|
|
|
"../resources/media/faces_I444.jpg",
|
|
|
|
|
"../resources/media/faces_I411.jpg",
|
|
|
|
|
"../resources/media/faces_I400.jpg",
|
2016-09-14 05:10:01 -07:00
|
|
|
]
|
2016-08-30 02:53:49 -07:00
|
|
|
|
|
|
|
|
if (is_ios) {
|
|
|
|
|
bundle_data("rtc_media_unittests_bundle_data") {
|
|
|
|
|
testonly = true
|
|
|
|
|
sources = rtc_media_unittests_resources
|
|
|
|
|
outputs = [
|
|
|
|
|
"{{bundle_resources_dir}}/{{source_file_part}}",
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
|
|
2016-09-02 04:10:34 -07:00
|
|
|
rtc_test("rtc_media_unittests") {
|
2016-06-12 22:12:01 -07:00
|
|
|
testonly = true
|
|
|
|
|
|
2016-08-26 02:59:47 -07:00
|
|
|
defines = []
|
2017-03-06 06:04:55 -08:00
|
|
|
deps = [
|
2018-01-11 17:07:30 +01:00
|
|
|
":rtc_audio_video",
|
2018-01-04 15:10:22 +01:00
|
|
|
":rtc_constants",
|
2018-01-11 17:07:30 +01:00
|
|
|
":rtc_data",
|
2018-11-28 16:47:49 +01:00
|
|
|
"../:webrtc_common",
|
2018-10-26 15:57:48 +02:00
|
|
|
"../api/test/video:function_video_factory",
|
2018-09-26 16:04:32 +02:00
|
|
|
"../api/units:time_delta",
|
2018-05-11 11:15:30 +02:00
|
|
|
"../api/video:video_frame_i420",
|
2018-11-07 14:29:54 +00:00
|
|
|
"../modules/audio_processing:api",
|
2017-12-19 16:44:45 +01:00
|
|
|
"../modules/audio_processing:mocks",
|
2018-07-03 10:40:40 +02:00
|
|
|
"../modules/rtp_rtcp",
|
2018-11-28 16:47:49 +01:00
|
|
|
|
|
|
|
|
# TODO(http://crbug.com/908819): Add this dependency when Chromium
|
|
|
|
|
# android templates stop to consider *_module to have a special meaning.
|
|
|
|
|
# "../modules/video_capture:video_capture_module",
|
2018-01-08 11:05:10 +01:00
|
|
|
"../modules/video_coding:video_codec_interface",
|
2018-06-21 16:16:38 +02:00
|
|
|
"../modules/video_coding:webrtc_vp8",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../pc:rtc_pc",
|
2017-12-06 11:23:19 +01:00
|
|
|
"../pc:rtc_pc_base",
|
2017-12-13 16:05:42 +01:00
|
|
|
"../rtc_base:checks",
|
2018-11-23 13:15:08 +01:00
|
|
|
"../rtc_base:gunit_helpers",
|
|
|
|
|
"../rtc_base:rtc_base_tests_utils",
|
2018-03-19 11:12:48 +01:00
|
|
|
"../rtc_base:rtc_task_queue",
|
2017-12-15 14:40:10 +01:00
|
|
|
"../rtc_base:stringutils",
|
2018-11-28 16:47:49 +01:00
|
|
|
"../rtc_base/third_party/sigslot:sigslot",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../test:field_trial",
|
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",
|
2018-10-30 22:23:02 +01:00
|
|
|
"//third_party/abseil-cpp/absl/strings",
|
2017-03-06 06:04:55 -08:00
|
|
|
]
|
2016-06-12 22:12:01 -07:00
|
|
|
sources = [
|
|
|
|
|
"base/codec_unittest.cc",
|
2019-01-11 09:11:00 -08:00
|
|
|
"base/rtp_data_engine_unittest.cc",
|
|
|
|
|
"base/rtp_utils_unittest.cc",
|
|
|
|
|
"base/stream_params_unittest.cc",
|
|
|
|
|
"base/turn_utils_unittest.cc",
|
|
|
|
|
"base/video_adapter_unittest.cc",
|
|
|
|
|
"base/video_broadcaster_unittest.cc",
|
|
|
|
|
"base/video_capturer_unittest.cc",
|
|
|
|
|
"base/video_common_unittest.cc",
|
2017-02-21 00:54:31 -08:00
|
|
|
"engine/apm_helpers_unittest.cc",
|
2018-12-06 13:38:24 +01:00
|
|
|
"engine/encoder_simulcast_proxy_unittest.cc",
|
2019-01-11 09:11:00 -08:00
|
|
|
"engine/internal_decoder_factory_unittest.cc",
|
|
|
|
|
"engine/multiplex_codec_factory_unittest.cc",
|
|
|
|
|
"engine/null_webrtc_video_engine_unittest.cc",
|
2016-08-17 02:45:41 -07:00
|
|
|
"engine/payload_type_mapper_unittest.cc",
|
2017-07-10 03:26:36 -07:00
|
|
|
"engine/simulcast_encoder_adapter_unittest.cc",
|
2018-06-27 10:44:56 +02:00
|
|
|
"engine/simulcast_unittest.cc",
|
2019-01-11 09:11:00 -08:00
|
|
|
"engine/webrtc_media_engine_unittest.cc",
|
|
|
|
|
"engine/webrtc_video_encoder_factory_unittest.cc",
|
|
|
|
|
"engine/webrtc_video_engine_unittest.cc",
|
2016-06-12 22:12:01 -07:00
|
|
|
]
|
|
|
|
|
|
2017-06-22 01:47:20 -07:00
|
|
|
# TODO(kthelgason): Reenable this test on iOS.
|
|
|
|
|
# See bugs.webrtc.org/5569
|
|
|
|
|
if (!is_ios) {
|
2019-01-11 09:11:00 -08:00
|
|
|
sources += [ "engine/webrtc_voice_engine_unittest.cc" ]
|
2017-06-22 01:47:20 -07:00
|
|
|
}
|
|
|
|
|
|
2016-12-22 10:53:38 -08:00
|
|
|
if (rtc_enable_sctp) {
|
2019-01-11 09:11:00 -08:00
|
|
|
sources += [ "sctp/sctp_transport_unittest.cc" ]
|
2016-12-22 10:53:38 -08:00
|
|
|
}
|
|
|
|
|
|
2016-08-26 02:59:47 -07:00
|
|
|
if (rtc_use_h264) {
|
|
|
|
|
defines += [ "WEBRTC_USE_H264" ]
|
|
|
|
|
}
|
2017-02-16 05:37:06 -08:00
|
|
|
|
|
|
|
|
if (rtc_opus_support_120ms_ptime) {
|
|
|
|
|
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
|
|
|
|
|
} else {
|
|
|
|
|
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
|
|
|
|
|
}
|
|
|
|
|
|
2016-10-16 23:56:12 -07:00
|
|
|
if (!build_with_chromium && is_clang) {
|
2016-09-02 04:10:34 -07:00
|
|
|
suppressed_configs += [
|
2016-10-16 23:56:12 -07:00
|
|
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
2016-06-12 22:12:01 -07:00
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"//build/config/clang:find_bad_constructs",
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]
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}
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|
2016-09-14 05:10:01 -07:00
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data = rtc_media_unittests_resources
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|
2016-06-12 22:12:01 -07:00
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if (is_android) {
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|
deps += [ "//testing/android/native_test:native_test_support" ]
|
2016-08-24 07:48:42 -07:00
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|
shard_timeout = 900
|
|
|
|
|
}
|
2016-08-30 02:53:49 -07:00
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|
|
|
|
if (is_ios) {
|
|
|
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|
deps += [ ":rtc_media_unittests_bundle_data" ]
|
2016-06-12 22:12:01 -07:00
|
|
|
}
|
|
|
|
|
|
|
|
|
|
deps += [
|
2018-12-06 13:38:24 +01:00
|
|
|
":rtc_encoder_simulcast_proxy",
|
2018-01-04 15:10:22 +01:00
|
|
|
":rtc_internal_video_codecs",
|
2016-06-12 22:12:01 -07:00
|
|
|
":rtc_media",
|
2017-03-06 06:04:55 -08:00
|
|
|
":rtc_media_base",
|
2017-04-25 00:35:35 -07:00
|
|
|
":rtc_media_tests_utils",
|
2018-10-19 00:51:18 -07:00
|
|
|
":rtc_simulcast_encoder_adapter",
|
2018-06-28 10:59:02 -07:00
|
|
|
":rtc_vp9_profile",
|
2018-05-31 12:53:00 +02:00
|
|
|
"../api:create_simulcast_test_fixture_api",
|
2017-12-22 09:36:42 -08:00
|
|
|
"../api:libjingle_peerconnection_api",
|
2018-11-08 10:02:56 -08:00
|
|
|
"../api:mock_video_bitrate_allocator",
|
|
|
|
|
"../api:mock_video_bitrate_allocator_factory",
|
2017-10-30 23:10:12 -07:00
|
|
|
"../api:mock_video_codec_factory",
|
2018-05-31 12:53:00 +02:00
|
|
|
"../api:simulcast_test_fixture_api",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../api/audio_codecs:builtin_audio_decoder_factory",
|
2017-05-02 06:46:30 -07:00
|
|
|
"../api/audio_codecs:builtin_audio_encoder_factory",
|
2018-11-08 10:02:56 -08:00
|
|
|
"../api/video:builtin_video_bitrate_allocator_factory",
|
2018-05-22 15:37:23 +02:00
|
|
|
"../api/video:video_bitrate_allocation",
|
2018-05-11 11:15:30 +02:00
|
|
|
"../api/video:video_frame",
|
2018-05-14 09:48:06 +02:00
|
|
|
"../api/video_codecs:builtin_video_decoder_factory",
|
|
|
|
|
"../api/video_codecs:builtin_video_encoder_factory",
|
2017-04-05 03:02:20 -07:00
|
|
|
"../api/video_codecs:video_codecs_api",
|
2016-06-12 22:12:01 -07:00
|
|
|
"../audio",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../call:call_interfaces",
|
|
|
|
|
"../common_video:common_video",
|
|
|
|
|
"../logging:rtc_event_log_api",
|
2018-02-01 11:04:46 -08:00
|
|
|
"../logging:rtc_event_log_impl_base",
|
2016-11-10 01:05:34 -08:00
|
|
|
"../modules/audio_device:mock_audio_device",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../modules/audio_processing:audio_processing",
|
2018-05-31 12:53:00 +02:00
|
|
|
"../modules/video_coding:simulcast_test_fixture_impl",
|
2017-04-18 15:49:09 -07:00
|
|
|
"../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_main",
|
2017-04-10 05:15:48 -07:00
|
|
|
"../test:audio_codec_mocks",
|
2017-03-06 06:04:55 -08:00
|
|
|
"../test:test_support",
|
2017-10-25 10:04:54 +02:00
|
|
|
"../test:video_test_common",
|
2016-06-12 22:12:01 -07:00
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
}
|