6e436d1cc0
This is part of implementing RTCRemoteInboundRtpStreamStats. The CL was split up into smaller pieces for reviewability. Spec: https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats In [1], ReportBlockData was implemented and tested. This CL adds the plumbing to make it available in MediaSenderInfo for audio streams, but the code is not wired up to make use of these stats. In follow-up CL [2], ReportBlockData will be used to implement RTCRemoteInboundRtpStreamStats. The follow-up CL will add integration tests exercising the full code path. [1] https://webrtc-review.googlesource.com/c/src/+/136584 [2] https://webrtc-review.googlesource.com/c/src/+/138067 Bug: webrtc:10455 Change-Id: Id8940090cc9c121e8f06ccdb068a22ce24c07092 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138066 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28072}
523 lines
15 KiB
Plaintext
523 lines
15 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.cc",
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"audio_receive_stream.h",
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"audio_send_stream.h",
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"audio_state.cc",
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"audio_state.h",
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"call.h",
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"call_config.cc",
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"call_config.h",
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"flexfec_receive_stream.cc",
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"flexfec_receive_stream.h",
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"packet_receiver.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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"../api:fec_controller_api",
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"../api:libjingle_peerconnection_api",
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"../api:network_state_predictor_api",
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"../api:rtp_headers",
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"../api:scoped_refptr",
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"../api:transport_api",
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"../api/audio:audio_mixer_api",
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"../api/audio_codecs:audio_codecs_api",
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"../api/task_queue",
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"../api/transport:network_control",
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"../modules/audio_device",
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"../modules/audio_processing",
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"../modules/audio_processing:api",
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"../modules/audio_processing:audio_processing_statistics",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/utility",
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"../rtc_base",
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"../rtc_base:audio_format_to_string",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/network:sent_packet",
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"//third_party/abseil-cpp/absl/types:optional",
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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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# Client code SHOULD NOT USE THIS TARGET, but for now it needs to be public
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# because there exists client code that uses it.
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# TODO(bugs.webrtc.org/9808): Move to private visibility as soon as that
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# client code gets updated.
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visibility = [ "*" ]
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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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"../api:fec_controller_api",
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"../api:libjingle_peerconnection_api",
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"../api:rtp_headers",
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"../api/transport:bitrate_settings",
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"../logging:rtc_event_log_api",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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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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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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"../api:array_view",
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"../api:rtp_headers",
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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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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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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_payload_params.cc",
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"rtp_payload_params.h",
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"rtp_transport_controller_send.cc",
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"rtp_transport_controller_send.h",
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"rtp_video_sender.cc",
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"rtp_video_sender.h",
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"rtp_video_sender_interface.h",
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]
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deps = [
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":bitrate_configurator",
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":rtp_interfaces",
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"../api:array_view",
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"../api:fec_controller_api",
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"../api:network_state_predictor_api",
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"../api:transport_api",
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"../api/transport:field_trial_based_config",
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"../api/transport:goog_cc",
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"../api/transport:network_control",
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"../api/units:data_rate",
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"../api/units:time_delta",
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"../api/units:timestamp",
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"../api/video:video_frame",
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"../api/video_codecs:video_codecs_api",
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"../logging:rtc_event_bwe",
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"../logging:rtc_event_log_api",
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"../modules/congestion_controller",
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"../modules/congestion_controller/rtp:control_handler",
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"../modules/congestion_controller/rtp:transport_feedback",
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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/rtp_rtcp:rtp_video_header",
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"../modules/utility",
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"../modules/video_coding:codec_globals_headers",
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"../modules/video_coding:video_codec_interface",
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"../rtc_base",
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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_task_queue",
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"../rtc_base/task_utils:repeating_task",
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"../system_wrappers:field_trial",
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"//third_party/abseil-cpp/absl/container:inlined_vector",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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"//third_party/abseil-cpp/absl/types:variant",
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]
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}
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rtc_source_set("bitrate_configurator") {
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sources = [
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"rtp_bitrate_configurator.cc",
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"rtp_bitrate_configurator.h",
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]
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deps = [
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":rtp_interfaces",
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"../api:libjingle_peerconnection_api",
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"../api/transport:bitrate_settings",
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"../rtc_base:checks",
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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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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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"../api:bitrate_allocation",
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"../api/units:data_rate",
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"../api/units:time_delta",
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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/synchronization:sequence_checker",
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"../system_wrappers",
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"../system_wrappers:field_trial",
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"../system_wrappers:metrics",
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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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"call_factory.cc",
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"call_factory.h",
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"degraded_call.cc",
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"degraded_call.h",
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"flexfec_receive_stream_impl.cc",
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"flexfec_receive_stream_impl.h",
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"receive_time_calculator.cc",
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"receive_time_calculator.h",
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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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":fake_network",
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":rtp_interfaces",
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":rtp_receiver",
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":rtp_sender",
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":simulated_network",
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":video_stream_api",
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"../api:array_view",
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"../api:callfactory_api",
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"../api:fec_controller_api",
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"../api:libjingle_peerconnection_api",
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"../api:rtp_headers",
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"../api:simulated_network_api",
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"../api:transport_api",
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"../api/task_queue:global_task_queue_factory",
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"../api/transport:network_control",
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"../api/units:time_delta",
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"../api/video_codecs:video_codecs_api",
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"../audio",
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"../logging:rtc_event_audio",
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"../logging:rtc_event_log_api",
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"../logging:rtc_event_rtp_rtcp",
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"../logging:rtc_event_video",
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"../logging:rtc_stream_config",
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"../modules:module_api",
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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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"../modules/video_coding",
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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_task_queue",
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"../rtc_base:safe_minmax",
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"../rtc_base/experiments:field_trial_parser",
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"../rtc_base/network:sent_packet",
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"../rtc_base/synchronization:rw_lock_wrapper",
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"../rtc_base/synchronization:sequence_checker",
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"../system_wrappers",
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"../system_wrappers:field_trial",
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"../system_wrappers:metrics",
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"../video",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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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_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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"../api:libjingle_peerconnection_api",
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"../api:rtp_headers",
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"../api:transport_api",
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"../api/video:video_frame",
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"../api/video:video_stream_encoder",
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"../api/video_codecs:video_codecs_api",
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"../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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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_source_set("simulated_network") {
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sources = [
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"simulated_network.cc",
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"simulated_network.h",
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]
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deps = [
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"../api:simulated_network_api",
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"../api/units:data_rate",
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"../api/units:data_size",
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"../api/units:time_delta",
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"../api/units:timestamp",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/synchronization:sequence_checker",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_source_set("simulated_packet_receiver") {
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sources = [
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"simulated_packet_receiver.h",
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]
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deps = [
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":call_interfaces",
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"../api:simulated_network_api",
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]
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}
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rtc_source_set("fake_network") {
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sources = [
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"fake_network_pipe.cc",
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"fake_network_pipe.h",
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]
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deps = [
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":call_interfaces",
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":simulated_network",
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":simulated_packet_receiver",
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"../api:libjingle_peerconnection_api",
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"../api:simulated_network_api",
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"../api:transport_api",
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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/synchronization:sequence_checker",
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"../system_wrappers",
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"//third_party/abseil-cpp/absl/memory",
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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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"receive_time_calculator_unittest.cc",
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"rtcp_demuxer_unittest.cc",
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"rtp_bitrate_configurator_unittest.cc",
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"rtp_demuxer_unittest.cc",
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"rtp_payload_params_unittest.cc",
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"rtp_rtcp_demuxer_helper_unittest.cc",
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"rtp_video_sender_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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":bitrate_configurator",
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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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":simulated_network",
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"../api:array_view",
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"../api:fake_media_transport",
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"../api:fake_media_transport",
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"../api:libjingle_peerconnection_api",
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"../api:mock_audio_mixer",
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"../api:rtp_headers",
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"../api:transport_api",
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"../api/audio_codecs:builtin_audio_decoder_factory",
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"../api/task_queue:default_task_queue_factory",
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"../api/video:video_frame",
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"../audio",
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"../logging:rtc_event_log_api",
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"../logging:rtc_event_log_impl_base",
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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/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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"../modules/video_coding",
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"../modules/video_coding:codec_globals_headers",
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"../modules/video_coding:video_codec_interface",
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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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"../system_wrappers",
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"../test:audio_codec_mocks",
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"../test:direct_transport",
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"../test:encoder_settings",
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"../test:fake_video_codecs",
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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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"../test:video_test_common",
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"../video",
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"//testing/gmock",
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"//testing/gtest",
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"//third_party/abseil-cpp/absl/container:inlined_vector",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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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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":simulated_network",
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":video_stream_api",
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"../api:rtc_event_log_output_file",
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"../api:simulated_network_api",
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"../api/audio_codecs:builtin_audio_encoder_factory",
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"../api/task_queue",
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"../api/task_queue:default_task_queue_factory",
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"../api/video:builtin_video_bitrate_allocator_factory",
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"../api/video:video_bitrate_allocation",
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"../api/video_codecs:video_codecs_api",
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"../logging:rtc_event_log_api",
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"../logging:rtc_event_log_impl_base",
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"../modules/audio_coding",
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"../modules/audio_device",
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"../modules/audio_device:audio_device_impl",
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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",
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"../test:direct_transport",
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"../test:encoder_settings",
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"../test:fake_video_codecs",
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"../test:field_trial",
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"../test:fileutils",
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"../test:null_transport",
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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",
|
|
"//third_party/abseil-cpp/absl/memory",
|
|
]
|
|
}
|
|
|
|
# TODO(eladalon): This should be moved, as with the TODO for |rtp_interfaces|.
|
|
rtc_source_set("mock_rtp_interfaces") {
|
|
testonly = true
|
|
|
|
sources = [
|
|
"test/mock_rtp_packet_sink_interface.h",
|
|
"test/mock_rtp_transport_controller_send.h",
|
|
]
|
|
deps = [
|
|
":rtp_interfaces",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../modules/congestion_controller",
|
|
"../modules/pacing",
|
|
"../rtc_base",
|
|
"../rtc_base:rate_limiter",
|
|
"../rtc_base/network:sent_packet",
|
|
"../test:test_support",
|
|
]
|
|
}
|
|
rtc_source_set("mock_bitrate_allocator") {
|
|
testonly = true
|
|
|
|
sources = [
|
|
"test/mock_bitrate_allocator.h",
|
|
]
|
|
deps = [
|
|
":bitrate_allocator",
|
|
"../test:test_support",
|
|
]
|
|
}
|
|
rtc_source_set("mock_call_interfaces") {
|
|
testonly = true
|
|
|
|
sources = [
|
|
"test/mock_audio_send_stream.h",
|
|
]
|
|
deps = [
|
|
":call_interfaces",
|
|
"../test:test_support",
|
|
]
|
|
}
|
|
|
|
rtc_source_set("fake_network_pipe_unittests") {
|
|
testonly = true
|
|
|
|
sources = [
|
|
"fake_network_pipe_unittest.cc",
|
|
"simulated_network_unittest.cc",
|
|
]
|
|
deps = [
|
|
":fake_network",
|
|
":simulated_network",
|
|
"../api/units:data_rate",
|
|
"../system_wrappers",
|
|
"../test:test_support",
|
|
"//testing/gtest",
|
|
"//third_party/abseil-cpp/absl/algorithm:container",
|
|
"//third_party/abseil-cpp/absl/memory",
|
|
]
|
|
}
|
|
}
|