2ffe3e80db
This is an unchanged patch after dependency fixes in downstream projects are implemented. Original patch was reviewed here: https://webrtc-review.googlesource.com/c/src/+/39008 TBR=phoglund@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org,philipel@webrtc.org,lliuu@webrtc.org Change-Id: I648bbf63d34282a48cabc854615005ec65b28cb3 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8287 Reviewed-on: https://webrtc-review.googlesource.com/40420 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21674}
313 lines
8.4 KiB
Plaintext
313 lines
8.4 KiB
Plaintext
# 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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import("../webrtc.gni")
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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.h",
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"audio_state.h",
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"call.h",
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"callfactoryinterface.h",
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"flexfec_receive_stream.h",
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"syncable.cc",
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"syncable.h",
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]
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if (!build_with_mozilla) {
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sources += [ "audio_send_stream.cc" ]
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}
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deps = [
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":rtp_interfaces",
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":video_stream_api",
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"..:webrtc_common",
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"../:typedefs",
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"../api:audio_mixer_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/audio_codecs:audio_codecs_api",
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"../modules/audio_processing:audio_processing_statistics",
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_base_approved",
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]
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}
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# TODO(nisse): These RTP targets should be moved elsewhere
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# when interfaces have stabilized. See also TODO for |mock_rtp_interfaces|.
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rtc_source_set("rtp_interfaces") {
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sources = [
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"rtcp_packet_sink_interface.h",
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"rtp_config.cc",
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"rtp_config.h",
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"rtp_packet_sink_interface.h",
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"rtp_stream_receiver_controller_interface.h",
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"rtp_transport_controller_send_interface.h",
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]
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deps = [
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"../api:array_view",
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"../rtc_base:rtc_base_approved",
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]
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}
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rtc_source_set("rtp_receiver") {
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visibility = [ "*" ]
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sources = [
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"rtcp_demuxer.cc",
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"rtcp_demuxer.h",
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"rtp_demuxer.cc",
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"rtp_demuxer.h",
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"rtp_rtcp_demuxer_helper.cc",
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"rtp_rtcp_demuxer_helper.h",
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"rtp_stream_receiver_controller.cc",
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"rtp_stream_receiver_controller.h",
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"rtx_receive_stream.cc",
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"rtx_receive_stream.h",
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"ssrc_binding_observer.h",
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]
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deps = [
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":rtp_interfaces",
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"..:webrtc_common",
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"../api:array_view",
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"../api:libjingle_peerconnection_api",
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"../api:optional",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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]
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}
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rtc_source_set("rtp_sender") {
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sources = [
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"rtp_transport_controller_send.cc",
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"rtp_transport_controller_send.h",
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]
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deps = [
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":rtp_interfaces",
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"..:webrtc_common",
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"../modules/congestion_controller",
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"../modules/pacing",
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"../rtc_base:rtc_base_approved",
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]
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}
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rtc_source_set("bitrate_allocator") {
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sources = [
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"bitrate_allocator.cc",
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"bitrate_allocator.h",
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]
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deps = [
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"../modules/bitrate_controller",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:sequenced_task_checker",
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"../system_wrappers",
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"../system_wrappers:metrics_api",
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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_static_library("call") {
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sources = [
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"call.cc",
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"callfactory.cc",
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"callfactory.h",
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"flexfec_receive_stream_impl.cc",
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"flexfec_receive_stream_impl.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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":bitrate_allocator",
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":call_interfaces",
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":rtp_interfaces",
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":rtp_receiver",
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":rtp_sender",
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":video_stream_api",
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"..:webrtc_common",
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"../api:optional",
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"../api:transport_api",
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"../audio",
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"../logging:rtc_event_log_api",
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"../logging:rtc_event_log_impl",
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"../modules/bitrate_controller",
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"../modules/congestion_controller",
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"../modules/pacing",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/utility",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_task_queue",
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"../rtc_base:sequenced_task_checker",
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"../system_wrappers",
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"../system_wrappers:metrics_api",
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"../video",
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]
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}
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rtc_source_set("video_stream_api") {
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sources = [
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"video_config.cc",
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"video_config.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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]
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deps = [
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":rtp_interfaces",
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"../:typedefs",
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"../:webrtc_common",
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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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"../common_video:common_video",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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]
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}
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if (rtc_include_tests) {
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rtc_source_set("call_tests") {
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testonly = true
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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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"flexfec_receive_stream_unittest.cc",
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"rtcp_demuxer_unittest.cc",
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"rtp_demuxer_unittest.cc",
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"rtp_rtcp_demuxer_helper_unittest.cc",
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"rtx_receive_stream_unittest.cc",
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]
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deps = [
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":bitrate_allocator",
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":call",
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":call_interfaces",
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":mock_rtp_interfaces",
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":rtp_interfaces",
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":rtp_receiver",
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":rtp_sender",
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"..:webrtc_common",
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"../api:array_view",
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"../api:libjingle_peerconnection_api",
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"../api:mock_audio_mixer",
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"../api/audio_codecs:builtin_audio_decoder_factory",
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"../audio:audio",
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"../logging:rtc_event_log_api",
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"../modules/audio_device:mock_audio_device",
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"../modules/audio_mixer",
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/audio_processing:mocks",
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"../modules/bitrate_controller",
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"../modules/congestion_controller",
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"../modules/congestion_controller:mock_congestion_controller",
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"../modules/pacing",
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"../modules/pacing:mock_paced_sender",
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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:mock_process_thread",
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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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"../test:audio_codec_mocks",
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"../test:direct_transport",
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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/gmock",
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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_source_set("call_perf_tests") {
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testonly = true
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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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":call_interfaces",
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":video_stream_api",
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"..:webrtc_common",
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"../api/audio_codecs:builtin_audio_encoder_factory",
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"../logging:rtc_event_log_api",
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"../modules/audio_coding",
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"../modules/audio_mixer:audio_mixer_impl",
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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:direct_transport",
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"../test:fake_audio_device",
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"../test:field_trial",
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"../test:perf_test",
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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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"../video",
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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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# TODO(eladalon): This should be moved, as with the TODO for |rtp_interfaces|.
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rtc_source_set("mock_rtp_interfaces") {
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testonly = true
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sources = [
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"fake_rtp_transport_controller_send.h",
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"test/mock_rtp_packet_sink_interface.h",
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]
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deps = [
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":rtp_interfaces",
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"..:webrtc_common",
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"../modules/congestion_controller:congestion_controller",
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"../modules/pacing:pacing",
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"../test:test_support",
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"//testing/gmock",
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]
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}
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rtc_source_set("mock_call_interfaces") {
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testonly = true
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sources = [
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"test/mock_audio_send_stream.h",
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]
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deps = [
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":call_interfaces",
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"//test:test_support",
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]
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}
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}
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