2015-09-25 13:58:30 +02:00
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# Copyright (c) 2015 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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2017-01-24 06:58:22 -08:00
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import("../webrtc.gni")
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2015-09-25 13:58:30 +02:00
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2016-12-07 04:52:58 -08:00
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rtc_source_set("call_interfaces") {
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sources = [
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"audio_receive_stream.h",
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"audio_send_stream.cc",
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"audio_send_stream.h",
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"audio_state.h",
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"call.h",
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2016-12-19 01:13:46 -08:00
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"flexfec_receive_stream.h",
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2017-04-24 05:53:20 -07:00
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"rtp_transport_controller_send_interface.h",
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2017-01-31 03:58:40 -08:00
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"syncable.cc",
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"syncable.h",
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2016-12-07 04:52:58 -08:00
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]
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2017-02-27 15:57:45 -08:00
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deps = [
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2017-04-25 23:42:15 -07:00
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"..:video_stream_api",
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2017-02-27 15:57:45 -08:00
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"..:webrtc_common",
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"../api:audio_mixer_api",
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Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
Reason for revert:
Re-land, reverting did not fix bug.
https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
Original issue's description:
> Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
>
> Reason for revert:
> Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
>
> https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
>
> Original issue's description:
> > Added the GetSources() to the RtpReceiverInterface and implemented
> > it for the AudioRtpReceiver.
> >
> > This method returns a vector of RtpSource(both CSRC source and SSRC
> > source) which contains the ID of a source, the timestamp, the source
> > type (SSRC or CSRC) and the audio level.
> >
> > The RtpSource objects are buffered and maintained by the
> > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> > the info of the contributing source will be pulled along the object
> > chain:
> > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> > AudioReceiveStream -> voe::Channel -> RtpRtcp module
> >
> > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
> >
> > BUG=chromium:703122
> > TBR=stefan@webrtc.org, danilchap@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2770233003
> > Cr-Commit-Position: refs/heads/master@{#17591}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/292084c3765d9f3ee406ca2ec86eae206b540053
>
> TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:703122
>
> Review-Url: https://codereview.webrtc.org/2809613002
> Cr-Commit-Position: refs/heads/master@{#17616}
> Committed: https://chromium.googlesource.com/external/webrtc/+/fbcc5cb3869d1370008e40f24fc03ac8fb69c675
TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122
Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
2017-04-10 07:39:05 -07:00
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"../api:libjingle_peerconnection_api",
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2017-02-27 15:57:45 -08:00
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"../api:transport_api",
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"../api/audio_codecs:audio_codecs_api",
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"../base:rtc_base",
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"../base:rtc_base_approved",
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2017-04-27 02:08:52 -07:00
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"../modules/audio_coding:audio_encoder_factory_interface",
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2017-02-27 15:57:45 -08:00
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"../modules/audio_coding:audio_encoder_interface",
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]
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2016-12-07 04:52:58 -08:00
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}
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2016-09-23 00:38:52 -07:00
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rtc_static_library("call") {
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2015-09-25 13:58:30 +02:00
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sources = [
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2015-11-12 21:02:42 -08:00
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"bitrate_allocator.cc",
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2015-09-25 13:58:30 +02:00
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"call.cc",
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2016-12-19 01:13:46 -08:00
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"flexfec_receive_stream_impl.cc",
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"flexfec_receive_stream_impl.h",
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2017-04-24 05:53:20 -07:00
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"rtp_transport_controller_send.cc",
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"rtp_transport_controller_send.h",
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2015-09-25 13:58:30 +02:00
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]
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2016-10-16 23:56:12 -07:00
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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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2016-09-02 04:10:34 -07:00
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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2015-09-25 13:58:30 +02:00
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}
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2016-11-17 06:28:59 -08:00
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public_deps = [
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2016-12-07 04:52:58 -08:00
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":call_interfaces",
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2016-11-17 06:28:59 -08:00
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"../api:call_api",
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]
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2015-09-25 13:58:30 +02:00
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deps = [
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2016-12-07 04:52:58 -08:00
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":call_interfaces",
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2015-09-25 13:58:30 +02:00
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"..:webrtc_common",
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2016-11-28 07:02:13 -08:00
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"../api:transport_api",
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2016-06-03 13:14:28 -07:00
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"../audio",
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2016-09-01 01:17:40 -07:00
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"../base:rtc_task_queue",
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2017-02-27 15:57:45 -08:00
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"../logging:rtc_event_log_api",
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Moved RtcEventLog files from call/ to logging/
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.
After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).
The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).
This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.
BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2380683005 .
Cr-Commit-Position: refs/heads/master@{#14485}
2016-10-03 18:31:22 -07:00
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"../logging:rtc_event_log_impl",
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2017-02-27 15:57:45 -08:00
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"../modules/bitrate_controller",
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2016-02-23 13:30:42 +01:00
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"../modules/congestion_controller",
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2017-02-27 15:57:45 -08:00
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"../modules/pacing",
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2015-09-25 13:58:30 +02:00
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"../modules/rtp_rtcp",
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2017-02-27 15:57:45 -08:00
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"../modules/utility",
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2015-09-25 13:58:30 +02:00
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"../system_wrappers",
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2016-06-03 13:14:28 -07:00
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"../video",
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2015-09-25 13:58:30 +02:00
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]
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}
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2016-06-14 12:52:54 +02:00
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if (rtc_include_tests) {
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2016-09-02 04:10:34 -07:00
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rtc_source_set("call_tests") {
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2016-06-14 12:52:54 +02:00
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testonly = true
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2017-04-25 04:04:50 -07:00
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# Skip restricting visibility on mobile platforms since the tests on those
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# gets additional generated targets which would require many lines here to
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# cover (which would be confusing to read and hard to maintain).
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if (!is_android && !is_ios) {
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visibility = [ "//webrtc:video_engine_tests" ]
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}
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2016-06-14 12:52:54 +02:00
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sources = [
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"bitrate_allocator_unittest.cc",
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"bitrate_estimator_tests.cc",
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"call_unittest.cc",
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2016-10-20 04:54:48 -07:00
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"flexfec_receive_stream_unittest.cc",
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2016-06-14 12:52:54 +02:00
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]
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deps = [
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":call",
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2016-11-17 06:48:48 -08:00
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"../base:rtc_base_approved",
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2017-02-27 15:57:45 -08:00
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"../logging:rtc_event_log_api",
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2016-11-17 06:28:59 -08:00
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"../modules/audio_device:mock_audio_device",
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2016-11-17 06:48:48 -08:00
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"../modules/audio_mixer",
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2017-02-27 15:57:45 -08:00
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"../modules/bitrate_controller",
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"../modules/pacing",
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"../modules/rtp_rtcp",
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"../system_wrappers",
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2017-04-10 05:15:48 -07:00
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"../test:audio_codec_mocks",
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2017-02-27 15:57:45 -08:00
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"../test:direct_transport",
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2016-11-17 06:48:48 -08:00
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"../test:test_common",
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2017-02-27 15:57:45 -08:00
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"../test:test_support",
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"../test:video_test_common",
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2016-06-14 12:52:54 +02:00
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"//testing/gmock",
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"//testing/gtest",
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]
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2016-10-16 23:56:12 -07:00
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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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2016-09-02 04:10:34 -07:00
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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2016-06-14 12:52:54 +02:00
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}
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}
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2017-01-05 07:09:50 -08:00
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rtc_source_set("call_perf_tests") {
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testonly = true
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2017-04-25 04:04:50 -07:00
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# Skip restricting visibility on mobile platforms since the tests on those
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# gets additional generated targets which would require many lines here to
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# cover (which would be confusing to read and hard to maintain).
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if (!is_android && !is_ios) {
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visibility = [ "//webrtc:webrtc_perf_tests" ]
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}
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2017-01-05 07:09:50 -08:00
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sources = [
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"call_perf_tests.cc",
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"rampup_tests.cc",
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"rampup_tests.h",
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]
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deps = [
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2017-02-27 15:57:45 -08:00
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":call_interfaces",
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"..:webrtc_common",
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"../base:rtc_base_approved",
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"../logging:rtc_event_log_api",
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"../modules/audio_coding",
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2017-04-27 02:08:52 -07:00
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"../modules/audio_coding:builtin_audio_encoder_factory",
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2017-02-27 15:57:45 -08:00
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/rtp_rtcp",
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"../system_wrappers",
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"../system_wrappers:metrics_default",
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"../test:direct_transport",
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2017-02-28 14:41:05 -08:00
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"../test:fake_audio_device",
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2017-02-27 15:57:45 -08:00
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"../test:test_support",
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"../test:video_test_common",
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"../video",
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"../voice_engine",
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2017-01-05 07:09:50 -08:00
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"//testing/gtest",
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Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
Reason for revert:
Seem to be a flaky test rather than an issue with this cl. Creating reland, will add code to reduce flakiness to that test.
Original issue's description:
> Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
>
> Reason for revert:
> This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780
>
> Original issue's description:
> > Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ )
> >
> > Reason for revert:
> > Found issue with test case, will add fix to reland cl.
> >
> > Original issue's description:
> > > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
> > >
> > > Reason for revert:
> > > Breaks perf tests:
> > > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679
> > > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325
> > >
> > > Original issue's description:
> > > > Add framerate to VideoSinkWants and ability to signal on overuse
> > > >
> > > > In ViEEncoder, try to reduce framerate instead of resolution if the
> > > > current degradation preference is maintain-resolution rather than
> > > > balanced.
> > > >
> > > > BUG=webrtc:4172
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2716643002
> > > > Cr-Commit-Position: refs/heads/master@{#17327}
> > > > Committed: https://chromium.googlesource.com/external/webrtc/+/72acf2526177bb4dbb5103cd6e165eb4361a5ae6
> > >
> > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:4172
> > >
> > > Review-Url: https://codereview.webrtc.org/2764133002
> > > Cr-Commit-Position: refs/heads/master@{#17331}
> > > Committed: https://chromium.googlesource.com/external/webrtc/+/8b45b11144c968b4173215c76f78c710c9a2ed0b
> >
> > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:4172
> >
> > Review-Url: https://codereview.webrtc.org/2781433002
> > Cr-Commit-Position: refs/heads/master@{#17474}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/3ea3c77e93121b1ab9d5e46641e6764f2cca0d51
>
> TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4172
>
> Review-Url: https://codereview.webrtc.org/2783183003
> Cr-Commit-Position: refs/heads/master@{#17477}
> Committed: https://chromium.googlesource.com/external/webrtc/+/f9ed235c9b7248694edcb46feb1f29ce7456ab59
R=ilnik@webrtc.org,stefan@webrtc.org
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2789823002
Cr-Commit-Position: refs/heads/master@{#17498}
2017-04-02 23:53:04 -07:00
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"//webrtc/test:field_trial",
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2017-01-05 07:09:50 -08:00
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"//webrtc/test:test_common",
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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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}
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2016-06-14 12:52:54 +02:00
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}
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