e4be6dad65
This CL removes direct access to SendSideCongestionController (SSCC) via the RtpTransportControllerSend interface and replaces all usages with calls on RtpTransportControllerSend which will in turn calls SSCC. This prepares for later refactor of RtpTransportControllerSend. Bug: webrtc:8415 Change-Id: I68363a3ab0203b95579f747402a1e7f58a5eeeb5 Reviewed-on: https://webrtc-review.googlesource.com/53860 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22044}
377 lines
12 KiB
Plaintext
377 lines
12 KiB
Plaintext
# Copyright (c) 2014 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("../webrtc.gni")
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rtc_static_library("video") {
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sources = [
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"call_stats.cc",
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"call_stats.h",
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"encoder_rtcp_feedback.cc",
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"encoder_rtcp_feedback.h",
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"overuse_frame_detector.cc",
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"overuse_frame_detector.h",
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"payload_router.cc",
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"payload_router.h",
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"quality_threshold.cc",
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"quality_threshold.h",
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"receive_statistics_proxy.cc",
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"receive_statistics_proxy.h",
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"report_block_stats.cc",
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"report_block_stats.h",
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"rtp_streams_synchronizer.cc",
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"rtp_streams_synchronizer.h",
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"rtp_video_stream_receiver.cc",
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"rtp_video_stream_receiver.h",
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"send_delay_stats.cc",
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"send_delay_stats.h",
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"send_statistics_proxy.cc",
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"send_statistics_proxy.h",
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"stats_counter.cc",
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"stats_counter.h",
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"stream_synchronization.cc",
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"stream_synchronization.h",
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"transport_adapter.cc",
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"transport_adapter.h",
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"video_receive_stream.cc",
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"video_receive_stream.h",
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"video_send_stream.cc",
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"video_send_stream.h",
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"video_stream_decoder.cc",
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"video_stream_decoder.h",
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"video_stream_encoder.cc",
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"video_stream_encoder.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",
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"../:typedefs",
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"../api:fec_controller_api",
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"../api:libjingle_peerconnection_api",
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"../api:optional",
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"../api:transport_api",
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"../api:video_frame_api",
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"../api:video_frame_api_i420",
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"../api/video_codecs:video_codecs_api",
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"../call:bitrate_allocator",
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"../call:call_interfaces",
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"../call:rtp_interfaces",
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"../call:video_stream_api",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/video_coding:video_codec_interface",
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"../rtc_base:checks",
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"../rtc_base/experiments:alr_experiment",
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"../rtc_base/system:fallthrough",
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"../system_wrappers:field_trial_api",
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"../system_wrappers:metrics_api",
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# For RtxReceiveStream.
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"../call:rtp_receiver",
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"../common_video",
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"../logging:rtc_event_log_api",
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"../modules:module_api",
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"../modules/bitrate_controller",
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"../modules/pacing",
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"../modules/remote_bitrate_estimator",
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"../modules/rtp_rtcp",
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"../modules/utility",
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"../modules/video_coding",
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"../modules/video_coding:video_coding_utility",
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"../modules/video_processing",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_numerics",
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"../rtc_base:rtc_task_queue",
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"../rtc_base:sequenced_task_checker",
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"../rtc_base:weak_ptr",
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"../system_wrappers",
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]
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if (!build_with_mozilla) {
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deps += [ "../media:rtc_media_base" ]
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}
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}
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if (rtc_include_tests) {
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rtc_source_set("video_quality_test") {
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testonly = true
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visibility = [ ":*" ] # Only targets in this file can depend on this.
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sources = [
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"video_quality_test.cc",
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"video_quality_test.h",
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]
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deps = [
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"../logging:rtc_event_log_api",
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"../logging:rtc_event_log_impl_output",
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"../media:rtc_audio_video",
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"../media:rtc_internal_video_codecs",
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/rtp_rtcp",
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"../modules/video_coding:webrtc_h264",
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"../modules/video_coding:webrtc_vp8",
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"../modules/video_coding:webrtc_vp9",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_base_tests_utils",
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"../system_wrappers",
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"../test:perf_test",
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"../test:rtp_test_utils",
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"../test:test_common",
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"../test:test_renderer",
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"../test:test_support",
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"../test:test_support_test_artifacts",
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"../test:video_test_common",
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"../test:video_test_support",
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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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rtc_source_set("video_full_stack_tests") {
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testonly = true
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sources = [
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"full_stack_tests.cc",
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]
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deps = [
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":video_quality_test",
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"../modules/pacing:pacing",
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"../rtc_base/experiments:alr_experiment",
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"../test:field_trial",
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"../test:test_common",
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"../test:test_support",
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"//testing/gtest",
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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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if (rtc_use_h264) {
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defines = [ "WEBRTC_USE_H264" ]
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}
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}
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rtc_executable("video_loopback") {
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testonly = true
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sources = [
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"video_loopback.cc",
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]
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deps = [
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":video_quality_test",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers:field_trial_default",
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"../system_wrappers:metrics_default",
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"../system_wrappers:runtime_enabled_features_default",
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"../test:field_trial",
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"../test:run_test",
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"../test:run_test_interface",
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"../test:test_common",
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"../test:test_renderer",
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"../test:test_support",
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"//testing/gtest",
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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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rtc_executable("screenshare_loopback") {
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testonly = true
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sources = [
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"screenshare_loopback.cc",
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]
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deps = [
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":video_quality_test",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers:field_trial_default",
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"../system_wrappers:metrics_default",
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"../system_wrappers:runtime_enabled_features_default",
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"../test:field_trial",
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"../test:run_test",
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"../test:run_test_interface",
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"../test:test_common",
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"../test:test_renderer",
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"../test:test_support",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from Chrome's Clang plugins.
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# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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rtc_executable("sv_loopback") {
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testonly = true
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sources = [
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"sv_loopback.cc",
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]
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deps = [
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":video_quality_test",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers:field_trial_default",
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"../system_wrappers:metrics_default",
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"../system_wrappers:runtime_enabled_features_default",
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"../test:field_trial",
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"../test:run_test",
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"../test:run_test_interface",
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"../test:test_common",
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"../test:test_renderer",
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"../test:test_support",
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"//testing/gtest",
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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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rtc_executable("video_replay") {
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testonly = true
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sources = [
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"replay.cc",
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]
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deps = [
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"..:webrtc_common",
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"../:typedefs",
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"../api/video_codecs:video_codecs_api",
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"../call:call_interfaces",
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"../common_video",
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"../logging:rtc_event_log_api",
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"../modules/rtp_rtcp",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers",
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"../system_wrappers:metrics_default",
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"../system_wrappers:runtime_enabled_features_default",
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"../test:field_trial",
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"../test:rtp_test_utils",
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"../test:run_test",
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"../test:run_test_interface",
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"../test:test_common",
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"../test:test_renderer",
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"../test:test_support",
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"../test:video_test_common",
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"../test:video_test_support",
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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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# TODO(pbos): Rename test suite.
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rtc_source_set("video_tests") {
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testonly = true
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defines = []
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sources = [
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"call_stats_unittest.cc",
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"encoder_rtcp_feedback_unittest.cc",
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"end_to_end_tests/bandwidth_tests.cc",
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"end_to_end_tests/call_operation_tests.cc",
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"end_to_end_tests/codec_tests.cc",
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"end_to_end_tests/config_tests.cc",
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"end_to_end_tests/extended_reports_tests.cc",
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"end_to_end_tests/fec_tests.cc",
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"end_to_end_tests/histogram_tests.cc",
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"end_to_end_tests/log_tests.cc",
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"end_to_end_tests/multi_stream_tester.cc",
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"end_to_end_tests/multi_stream_tester.h",
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"end_to_end_tests/multi_stream_tests.cc",
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"end_to_end_tests/network_state_tests.cc",
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"end_to_end_tests/probing_tests.cc",
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"end_to_end_tests/retransmission_tests.cc",
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"end_to_end_tests/rtp_rtcp_tests.cc",
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"end_to_end_tests/ssrc_tests.cc",
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"end_to_end_tests/stats_tests.cc",
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"end_to_end_tests/transport_feedback_tests.cc",
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"overuse_frame_detector_unittest.cc",
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"payload_router_unittest.cc",
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"picture_id_tests.cc",
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"quality_threshold_unittest.cc",
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"receive_statistics_proxy_unittest.cc",
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"report_block_stats_unittest.cc",
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"rtp_video_stream_receiver_unittest.cc",
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"send_delay_stats_unittest.cc",
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"send_statistics_proxy_unittest.cc",
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"stats_counter_unittest.cc",
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"stream_synchronization_unittest.cc",
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"video_receive_stream_unittest.cc",
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"video_send_stream_tests.cc",
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"video_stream_encoder_unittest.cc",
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]
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deps = [
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":video",
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"../api:optional",
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"../api:video_frame_api",
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"../api:video_frame_api_i420",
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"../api/video_codecs:video_codecs_api",
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"../call:call_interfaces",
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"../call:mock_rtp_interfaces",
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"../call:rtp_receiver",
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"../call:rtp_sender",
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"../call:video_stream_api",
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"../common_video",
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"../logging:rtc_event_log_api",
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"../media:rtc_audio_video",
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"../media:rtc_internal_video_codecs",
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"../media:rtc_media",
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"../media:rtc_media_base",
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"../media:rtc_media_tests_utils",
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"../modules:module_api",
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"../modules/pacing",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:mock_rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/utility",
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"../modules/video_coding",
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"../modules/video_coding:video_codec_interface",
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"../modules/video_coding:video_coding_utility",
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"../modules/video_coding:webrtc_h264",
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"../modules/video_coding:webrtc_multiplex",
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"../modules/video_coding:webrtc_vp8_helpers",
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"../modules/video_coding:webrtc_vp9",
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"../rtc_base:checks",
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"../rtc_base:rate_limiter",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_base_tests_utils",
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"../rtc_base:rtc_numerics",
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"../rtc_base/experiments:alr_experiment",
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"../system_wrappers",
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"../system_wrappers:field_trial_default",
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"../system_wrappers:metrics_api",
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"../system_wrappers:metrics_default",
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"../test:direct_transport",
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"../test:field_trial",
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"../test:perf_test",
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"../test:rtp_test_utils",
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"../test:test_common",
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"../test:test_support",
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"../test:video_test_common",
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"//testing/gtest",
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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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if (rtc_use_h264) {
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defines += [ "WEBRTC_USE_H264" ]
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}
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if (!build_with_mozilla) {
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deps += [ "../media:rtc_media_base" ]
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}
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}
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}
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