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:72acf25261> > > > > > 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:8b45b11144> > > > 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:3ea3c77e93> > 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:f9ed235c9bR=ilnik@webrtc.org,stefan@webrtc.org BUG=webrtc:4172 Review-Url: https://codereview.webrtc.org/2789823002 Cr-Commit-Position: refs/heads/master@{#17498}
135 lines
3.6 KiB
Plaintext
135 lines
3.6 KiB
Plaintext
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
|
|
#
|
|
# Use of this source code is governed by a BSD-style license
|
|
# that can be found in the LICENSE file in the root of the source
|
|
# tree. An additional intellectual property rights grant can be found
|
|
# in the file PATENTS. All contributing project authors may
|
|
# be found in the AUTHORS file in the root of the source tree.
|
|
|
|
import("../webrtc.gni")
|
|
|
|
rtc_source_set("call_interfaces") {
|
|
sources = [
|
|
"audio_receive_stream.h",
|
|
"audio_send_stream.cc",
|
|
"audio_send_stream.h",
|
|
"audio_state.h",
|
|
"call.h",
|
|
"flexfec_receive_stream.h",
|
|
"rtp_transport_controller_send.h",
|
|
"syncable.cc",
|
|
"syncable.h",
|
|
]
|
|
deps = [
|
|
"..:webrtc_common",
|
|
"../api:audio_mixer_api",
|
|
"../api:transport_api",
|
|
"../api/audio_codecs:audio_codecs_api",
|
|
"../base:rtc_base",
|
|
"../base:rtc_base_approved",
|
|
"../modules/audio_coding:audio_encoder_interface",
|
|
]
|
|
}
|
|
|
|
rtc_static_library("call") {
|
|
sources = [
|
|
"bitrate_allocator.cc",
|
|
"call.cc",
|
|
"flexfec_receive_stream_impl.cc",
|
|
"flexfec_receive_stream_impl.h",
|
|
]
|
|
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
|
|
public_deps = [
|
|
":call_interfaces",
|
|
"../api:call_api",
|
|
]
|
|
|
|
deps = [
|
|
":call_interfaces",
|
|
"..:webrtc_common",
|
|
"../api:transport_api",
|
|
"../audio",
|
|
"../base:rtc_task_queue",
|
|
"../logging:rtc_event_log_api",
|
|
"../logging:rtc_event_log_impl",
|
|
"../modules/bitrate_controller",
|
|
"../modules/congestion_controller",
|
|
"../modules/pacing",
|
|
"../modules/rtp_rtcp",
|
|
"../modules/utility",
|
|
"../system_wrappers",
|
|
"../video",
|
|
]
|
|
}
|
|
|
|
if (rtc_include_tests) {
|
|
rtc_source_set("call_tests") {
|
|
testonly = true
|
|
sources = [
|
|
"bitrate_allocator_unittest.cc",
|
|
"bitrate_estimator_tests.cc",
|
|
"call_unittest.cc",
|
|
"flexfec_receive_stream_unittest.cc",
|
|
]
|
|
deps = [
|
|
":call",
|
|
"../base:rtc_base_approved",
|
|
"../logging:rtc_event_log_api",
|
|
"../modules/audio_device:mock_audio_device",
|
|
"../modules/audio_mixer",
|
|
"../modules/bitrate_controller",
|
|
"../modules/pacing",
|
|
"../modules/rtp_rtcp",
|
|
"../system_wrappers",
|
|
"../test:direct_transport",
|
|
"../test:test_common",
|
|
"../test:test_support",
|
|
"../test:video_test_common",
|
|
"//testing/gmock",
|
|
"//testing/gtest",
|
|
]
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
}
|
|
|
|
rtc_source_set("call_perf_tests") {
|
|
testonly = true
|
|
sources = [
|
|
"call_perf_tests.cc",
|
|
"rampup_tests.cc",
|
|
"rampup_tests.h",
|
|
]
|
|
deps = [
|
|
":call_interfaces",
|
|
"..:webrtc_common",
|
|
"../base:rtc_base_approved",
|
|
"../logging:rtc_event_log_api",
|
|
"../modules/audio_coding",
|
|
"../modules/audio_mixer:audio_mixer_impl",
|
|
"../modules/rtp_rtcp",
|
|
"../system_wrappers",
|
|
"../system_wrappers:metrics_default",
|
|
"../test:direct_transport",
|
|
"../test:fake_audio_device",
|
|
"../test:test_support",
|
|
"../test:video_test_common",
|
|
"../video",
|
|
"../voice_engine",
|
|
"//testing/gtest",
|
|
"//webrtc/test:field_trial",
|
|
"//webrtc/test:test_common",
|
|
]
|
|
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" ]
|
|
}
|
|
}
|
|
}
|