f45ca3787f
--- Background --- The webrtc::VideoSendStream::StreamStats are converted into VideoSenderInfo objects which turn into "outbound-rtp" stats objects in getStats() (or "ssrc" objects in legacy getStats()). StreamStats are created for each type of substream: RTP media streams, RTX streams and FlexFEC streams - each with individual packet counters. The RTX stream is responsible for retransmissions of a referenced media stream and the FlexFEC stream is responsible for FEC of a referenced media stream. RTX/FEC streams do not show up as separate objects in getStats(). Only the media streams become "outbound-rtp" objects, but their packet and byte counters have to include the RTX and FEC counters. --- Overview of this CL --- This CL adds MergeInfoAboutOutboundRtpSubstreams(). It takes StreamStats of all kinds as input, and outputs media-only StreamStats - incorporating the RTX and FEC counters into the relevant media StreamStats. The merged StreamStats objects is a smaller set of objects than the non-merged counterparts, but when aggregating all packet counters together we end up with exact same packet and count as before. Because WebRtcVideoSendStream::GetVideoSenderInfo() currently aggregates the StreamStats into a single VideoSenderInfo (single "outbound-rtp"), this CL should not have any observable side-effects. Prior to this CL: aggregate StreamStats. After this CL: merge StreamStats and then aggregate them. However, when simulcast stats are implemented (WIP CL: https://webrtc-review.googlesource.com/c/src/+/168120) each RTP media stream should turn into an individual "outbound-rtp" object. We will then no longer aggregate all StreamStats into a single "info". This CL unblocks simulcast stats by providing StreamStats objects that could be turned into individual VideoSenderInfos. --- The Changes --- 1. Methods added to RtpConfig to be able to easily tell the relationship between RTP, RTX and FEC ssrcs. 2. StreamStats gets a StreamType (kMedia, kRtx or kFlexfec) that replaces the booleans (is_rtx, is_flexfec). 3. "referenced_media_ssrc" is added to StreamStats, making it possible to tell which kRtx/kFlexFec stream stats need to be merged with which kMedia StreamStats. 4. MergeInfoAboutOutboundRtpSubstreams() added and used. Bug: webrtc:11439 Change-Id: Iaf9002041169a054ddfd32c7ea06bd1dc36c6bca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170826 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30869}
557 lines
16 KiB
Plaintext
557 lines
16 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_library("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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":audio_sender_interface",
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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:network_state_predictor_api",
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"../api:rtc_error",
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"../api:rtp_headers",
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"../api:rtp_parameters",
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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/crypto:frame_decryptor_interface",
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"../api/crypto:frame_encryptor_interface",
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"../api/crypto:options",
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"../api/neteq:neteq_api",
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"../api/task_queue",
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"../api/transport:bitrate_settings",
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"../api/transport:network_control",
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"../api/transport:webrtc_key_value_config",
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"../api/transport/rtp:rtp_source",
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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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rtc_source_set("audio_sender_interface") {
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visibility = [ "*" ]
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sources = [ "audio_sender.h" ]
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deps = [ "../api/audio:audio_frame_api" ]
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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_library("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:frame_transformer_interface",
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"../api:rtp_headers",
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"../api:rtp_parameters",
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"../api/crypto:options",
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"../api/rtc_event_log",
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"../api/transport:bitrate_settings",
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"../api/units:timestamp",
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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/algorithm:container",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_library("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/types:optional",
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]
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}
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rtc_library("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:bitrate_allocation",
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"../api:fec_controller_api",
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"../api:network_state_predictor_api",
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"../api:rtp_parameters",
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"../api:transport_api",
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"../api/rtc_event_log",
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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:video_rtp_headers",
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"../api/video_codecs:video_codecs_api",
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"../logging:rtc_event_bwe",
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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:frame_dependencies_calculator",
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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/algorithm:container",
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"//third_party/abseil-cpp/absl/container:inlined_vector",
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"//third_party/abseil-cpp/absl/strings:strings",
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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_library("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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# For api/bitrate_constraints.h
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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_library("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/transport:network_control",
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"../api/units:data_rate",
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"../api/units:time_delta",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:safe_minmax",
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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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"//third_party/abseil-cpp/absl/algorithm:container",
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]
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}
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rtc_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:rtp_headers",
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"../api:rtp_parameters",
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"../api:simulated_network_api",
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"../api:transport_api",
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"../api/rtc_event_log",
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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_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/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/types:optional",
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]
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}
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rtc_library("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:frame_transformer_interface",
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"../api:rtp_headers",
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"../api:rtp_parameters",
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"../api:transport_api",
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"../api/crypto:frame_decryptor_interface",
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"../api/crypto:frame_encryptor_interface",
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"../api/crypto:options",
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"../api/transport/rtp:rtp_source",
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"../api/video:recordable_encoded_frame",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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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_library("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/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 = [ "simulated_packet_receiver.h" ]
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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_library("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:rtp_parameters",
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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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]
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}
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if (rtc_include_tests) {
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rtc_library("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:create_frame_generator",
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"../api:mock_audio_mixer",
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"../api:rtp_headers",
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"../api:rtp_parameters",
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"../api:transport_api",
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"../api/audio_codecs:builtin_audio_decoder_factory",
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"../api/rtc_event_log",
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"../api/task_queue:default_task_queue_factory",
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"../api/transport:field_trial_based_config",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../audio",
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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/congestion_controller",
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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: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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"../rtc_base:task_queue_for_test",
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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:mock_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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"../test/time_controller:time_controller",
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"../video",
|
|
"//test/scenario:scenario",
|
|
"//testing/gmock",
|
|
"//testing/gtest",
|
|
"//third_party/abseil-cpp/absl/container:inlined_vector",
|
|
"//third_party/abseil-cpp/absl/memory",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
}
|
|
|
|
rtc_library("call_perf_tests") {
|
|
testonly = true
|
|
|
|
sources = [
|
|
"call_perf_tests.cc",
|
|
"rampup_tests.cc",
|
|
"rampup_tests.h",
|
|
]
|
|
deps = [
|
|
":call_interfaces",
|
|
":simulated_network",
|
|
":video_stream_api",
|
|
"../api:rtc_event_log_output_file",
|
|
"../api:simulated_network_api",
|
|
"../api/audio_codecs:builtin_audio_encoder_factory",
|
|
"../api/rtc_event_log",
|
|
"../api/rtc_event_log:rtc_event_log_factory",
|
|
"../api/task_queue",
|
|
"../api/task_queue:default_task_queue_factory",
|
|
"../api/video:builtin_video_bitrate_allocator_factory",
|
|
"../api/video:video_bitrate_allocation",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../modules/audio_coding",
|
|
"../modules/audio_device",
|
|
"../modules/audio_device:audio_device_impl",
|
|
"../modules/audio_mixer:audio_mixer_impl",
|
|
"../modules/rtp_rtcp",
|
|
"../modules/rtp_rtcp:rtp_rtcp_format",
|
|
"../rtc_base",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:task_queue_for_test",
|
|
"../rtc_base:task_queue_for_test",
|
|
"../rtc_base/task_utils:repeating_task",
|
|
"../system_wrappers",
|
|
"../system_wrappers:metrics",
|
|
"../test:direct_transport",
|
|
"../test:encoder_settings",
|
|
"../test:fake_video_codecs",
|
|
"../test:field_trial",
|
|
"../test:fileutils",
|
|
"../test:null_transport",
|
|
"../test:perf_test",
|
|
"../test:rtp_test_utils",
|
|
"../test:test_common",
|
|
"../test:test_support",
|
|
"../test:video_test_common",
|
|
"../video",
|
|
"//testing/gtest",
|
|
"//third_party/abseil-cpp/absl/flags:flag",
|
|
]
|
|
}
|
|
|
|
# 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:frame_transformer_interface",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api/crypto:frame_encryptor_interface",
|
|
"../api/crypto:options",
|
|
"../api/transport:bitrate_settings",
|
|
"../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_library("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",
|
|
]
|
|
}
|
|
}
|