mirror of
https://github.com/mollyim/webrtc.git
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Removes legacy bitrate controller.
Bug: webrtc:9883 Change-Id: I66af2597059fc1f38c78682f6884361a4d16c4a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141408 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28232}
This commit is contained in:
Sebastian Jansson
Commit Bot
parent
171bd2644d
commit
7742b21839
7 changed files with 1 additions and 1000 deletions
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@ -75,7 +75,6 @@ LEGACY_API_DIRS = (
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'common_audio/include',
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'common_audio/include',
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'modules/audio_coding/include',
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'modules/audio_coding/include',
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'modules/audio_processing/include',
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'modules/audio_processing/include',
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'modules/bitrate_controller/include',
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'modules/congestion_controller/include',
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'modules/congestion_controller/include',
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'modules/include',
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'modules/include',
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'modules/remote_bitrate_estimator/include',
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'modules/remote_bitrate_estimator/include',
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@ -11,10 +11,6 @@ import("../../webrtc.gni")
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rtc_static_library("bitrate_controller") {
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rtc_static_library("bitrate_controller") {
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visibility = [ "*" ]
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visibility = [ "*" ]
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sources = [
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sources = [
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"bitrate_controller.cc",
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"bitrate_controller_impl.cc",
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"bitrate_controller_impl.h",
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"include/bitrate_controller.h",
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"loss_based_bandwidth_estimation.cc",
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"loss_based_bandwidth_estimation.cc",
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"loss_based_bandwidth_estimation.h",
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"loss_based_bandwidth_estimation.h",
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"send_side_bandwidth_estimation.cc",
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"send_side_bandwidth_estimation.cc",
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@ -28,7 +24,6 @@ rtc_static_library("bitrate_controller") {
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}
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}
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deps = [
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deps = [
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"..:module_api",
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"../../api/transport:network_control",
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"../../api/transport:network_control",
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"../../api/units:data_rate",
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"../../api/units:data_rate",
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"../../api/units:time_delta",
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"../../api/units:time_delta",
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@ -36,14 +31,10 @@ rtc_static_library("bitrate_controller") {
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"../../logging:rtc_event_bwe",
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"../../logging:rtc_event_bwe",
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"../../logging:rtc_event_log_api",
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"../../logging:rtc_event_log_api",
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"../../rtc_base:checks",
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"../../rtc_base:checks",
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"../../rtc_base:deprecation",
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"../../rtc_base:logging",
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"../../rtc_base:rtc_base_approved",
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"../../rtc_base/experiments:field_trial_parser",
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"../../rtc_base/experiments:field_trial_parser",
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"../../system_wrappers",
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"../../system_wrappers:field_trial",
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"../../system_wrappers:field_trial",
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"../../system_wrappers:metrics",
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"../../system_wrappers:metrics",
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"../congestion_controller/goog_cc:delay_based_bwe",
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"../pacing",
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"../remote_bitrate_estimator",
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"../remote_bitrate_estimator",
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"../rtp_rtcp",
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"../rtp_rtcp",
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"../rtp_rtcp:rtp_rtcp_format",
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"../rtp_rtcp:rtp_rtcp_format",
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@ -57,7 +48,6 @@ if (rtc_include_tests) {
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testonly = true
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testonly = true
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sources = [
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sources = [
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"bitrate_controller_unittest.cc",
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"send_side_bandwidth_estimation_unittest.cc",
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"send_side_bandwidth_estimation_unittest.cc",
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]
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]
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deps = [
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deps = [
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@ -65,14 +55,7 @@ if (rtc_include_tests) {
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"../../logging:mocks",
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"../../logging:mocks",
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"../../logging:rtc_event_bwe",
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"../../logging:rtc_event_bwe",
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"../../logging:rtc_event_log_api",
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"../../logging:rtc_event_log_api",
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"../../system_wrappers",
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"../../test:field_trial",
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"../../test:test_support",
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"../../test:test_support",
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"../congestion_controller/goog_cc:delay_based_bwe",
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"../pacing",
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"../pacing:mock_paced_sender",
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"../remote_bitrate_estimator",
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"../rtp_rtcp:rtp_rtcp_format",
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]
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]
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}
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}
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}
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}
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@ -1,23 +0,0 @@
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/*
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* Copyright (c) 2018 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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*
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* Usage: this class will register multiple RtcpBitrateObserver's one at each
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* RTCP module. It will aggregate the results and run one bandwidth estimation
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* and push the result to the encoders via BitrateObserver(s).
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*/
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#include "modules/bitrate_controller/include/bitrate_controller.h"
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namespace webrtc {
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size_t BitrateObserver::pacer_queue_size_in_bytes() {
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return 0;
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}
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} // namespace webrtc
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@ -1,292 +0,0 @@
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/*
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* Copyright (c) 2012 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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*
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*/
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#include "modules/bitrate_controller/bitrate_controller_impl.h"
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#include <algorithm>
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#include <utility>
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#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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namespace {
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absl::optional<DataRate> ToOptionalDataRate(int send_bitrate_bps) {
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if (send_bitrate_bps > 0)
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return DataRate::bps(send_bitrate_bps);
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return absl::nullopt;
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}
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DataRate MaxRate(int max_bitrate_bps) {
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if (max_bitrate_bps == -1)
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return DataRate::Infinity();
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return DataRate::bps(max_bitrate_bps);
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}
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} // namespace
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class BitrateControllerImpl::RtcpBandwidthObserverImpl
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: public RtcpBandwidthObserver {
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public:
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explicit RtcpBandwidthObserverImpl(BitrateControllerImpl* owner)
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: owner_(owner) {}
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~RtcpBandwidthObserverImpl() override = default;
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// Received RTCP REMB or TMMBR.
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void OnReceivedEstimatedBitrate(uint32_t bitrate) override {
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owner_->OnReceivedEstimatedBitrate(bitrate);
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}
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// Received RTCP receiver block.
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void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
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int64_t rtt,
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int64_t now_ms) override {
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owner_->OnReceivedRtcpReceiverReport(report_blocks, rtt, now_ms);
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}
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private:
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BitrateControllerImpl* const owner_;
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};
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BitrateController* BitrateController::CreateBitrateController(
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Clock* clock,
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BitrateObserver* observer,
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RtcEventLog* event_log) {
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return new BitrateControllerImpl(clock, observer, event_log);
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}
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BitrateController* BitrateController::CreateBitrateController(
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Clock* clock,
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RtcEventLog* event_log) {
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return CreateBitrateController(clock, nullptr, event_log);
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}
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BitrateControllerImpl::BitrateControllerImpl(Clock* clock,
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BitrateObserver* observer,
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RtcEventLog* event_log)
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: clock_(clock),
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observer_(observer),
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last_bitrate_update_ms_(clock_->TimeInMilliseconds()),
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event_log_(event_log),
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bandwidth_estimation_(event_log),
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last_bitrate_bps_(0),
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last_fraction_loss_(0),
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last_rtt_ms_(0) {
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// This calls the observer_ if set, which means that the observer provided by
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// the user must be ready to accept a bitrate update when it constructs the
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// controller. We do this to avoid having to keep synchronized initial values
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// in both the controller and the allocator.
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MaybeTriggerOnNetworkChanged();
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}
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RtcpBandwidthObserver* BitrateControllerImpl::CreateRtcpBandwidthObserver() {
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return new RtcpBandwidthObserverImpl(this);
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}
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void BitrateControllerImpl::SetStartBitrate(int start_bitrate_bps) {
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{
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rtc::CritScope cs(&critsect_);
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bandwidth_estimation_.SetSendBitrate(
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DataRate::bps(start_bitrate_bps),
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Timestamp::ms(clock_->TimeInMilliseconds()));
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}
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MaybeTriggerOnNetworkChanged();
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}
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void BitrateControllerImpl::SetMinMaxBitrate(int min_bitrate_bps,
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int max_bitrate_bps) {
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{
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rtc::CritScope cs(&critsect_);
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bandwidth_estimation_.SetMinMaxBitrate(DataRate::bps(min_bitrate_bps),
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DataRate::bps(max_bitrate_bps));
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}
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MaybeTriggerOnNetworkChanged();
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}
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void BitrateControllerImpl::SetBitrates(int start_bitrate_bps,
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int min_bitrate_bps,
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int max_bitrate_bps) {
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{
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rtc::CritScope cs(&critsect_);
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bandwidth_estimation_.SetBitrates(
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ToOptionalDataRate(start_bitrate_bps), DataRate::bps(min_bitrate_bps),
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MaxRate(max_bitrate_bps), Timestamp::ms(clock_->TimeInMilliseconds()));
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}
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MaybeTriggerOnNetworkChanged();
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}
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void BitrateControllerImpl::ResetBitrates(int bitrate_bps,
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int min_bitrate_bps,
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int max_bitrate_bps) {
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{
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rtc::CritScope cs(&critsect_);
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bandwidth_estimation_ = SendSideBandwidthEstimation(event_log_);
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bandwidth_estimation_.SetBitrates(
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ToOptionalDataRate(bitrate_bps), DataRate::bps(min_bitrate_bps),
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MaxRate(max_bitrate_bps), Timestamp::ms(clock_->TimeInMilliseconds()));
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}
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MaybeTriggerOnNetworkChanged();
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}
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// This is called upon reception of REMB or TMMBR.
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void BitrateControllerImpl::OnReceivedEstimatedBitrate(uint32_t bitrate) {
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{
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rtc::CritScope cs(&critsect_);
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bandwidth_estimation_.UpdateReceiverEstimate(
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Timestamp::ms(clock_->TimeInMilliseconds()), DataRate::bps(bitrate));
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BWE_TEST_LOGGING_PLOT(1, "REMB_kbps", clock_->TimeInMilliseconds(),
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bitrate / 1000);
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}
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MaybeTriggerOnNetworkChanged();
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}
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void BitrateControllerImpl::OnDelayBasedBweResult(
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const DelayBasedBwe::Result& result) {
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if (!result.updated)
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return;
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{
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rtc::CritScope cs(&critsect_);
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if (result.probe) {
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bandwidth_estimation_.SetSendBitrate(
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result.target_bitrate, Timestamp::ms(clock_->TimeInMilliseconds()));
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}
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// Since SetSendBitrate now resets the delay-based estimate, we have to call
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// UpdateDelayBasedEstimate after SetSendBitrate.
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bandwidth_estimation_.UpdateDelayBasedEstimate(
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Timestamp::ms(clock_->TimeInMilliseconds()), result.target_bitrate);
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}
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MaybeTriggerOnNetworkChanged();
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}
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int64_t BitrateControllerImpl::TimeUntilNextProcess() {
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const int64_t kBitrateControllerUpdateIntervalMs = 25;
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rtc::CritScope cs(&critsect_);
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int64_t time_since_update_ms =
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clock_->TimeInMilliseconds() - last_bitrate_update_ms_;
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return std::max<int64_t>(
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kBitrateControllerUpdateIntervalMs - time_since_update_ms, 0);
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}
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void BitrateControllerImpl::Process() {
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{
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rtc::CritScope cs(&critsect_);
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bandwidth_estimation_.UpdateEstimate(
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Timestamp::ms(clock_->TimeInMilliseconds()));
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}
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MaybeTriggerOnNetworkChanged();
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last_bitrate_update_ms_ = clock_->TimeInMilliseconds();
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}
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void BitrateControllerImpl::OnReceivedRtcpReceiverReport(
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const ReportBlockList& report_blocks,
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int64_t rtt,
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int64_t now_ms) {
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if (report_blocks.empty())
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return;
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{
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rtc::CritScope cs(&critsect_);
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int fraction_lost_aggregate = 0;
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int total_number_of_packets = 0;
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// Compute the a weighted average of the fraction loss from all report
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// blocks.
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for (const RTCPReportBlock& report_block : report_blocks) {
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std::map<uint32_t, uint32_t>::iterator seq_num_it =
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ssrc_to_last_received_extended_high_seq_num_.find(
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report_block.source_ssrc);
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int number_of_packets = 0;
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if (seq_num_it != ssrc_to_last_received_extended_high_seq_num_.end()) {
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number_of_packets =
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report_block.extended_highest_sequence_number - seq_num_it->second;
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}
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fraction_lost_aggregate += number_of_packets * report_block.fraction_lost;
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total_number_of_packets += number_of_packets;
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// Update last received for this SSRC.
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ssrc_to_last_received_extended_high_seq_num_[report_block.source_ssrc] =
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report_block.extended_highest_sequence_number;
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}
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if (total_number_of_packets < 0) {
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RTC_LOG(LS_WARNING)
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<< "Received report block where extended high sequence "
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"number goes backwards, ignoring.";
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return;
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}
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if (total_number_of_packets == 0)
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fraction_lost_aggregate = 0;
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else
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fraction_lost_aggregate =
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(fraction_lost_aggregate + total_number_of_packets / 2) /
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total_number_of_packets;
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if (fraction_lost_aggregate > 255)
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return;
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RTC_DCHECK_GE(total_number_of_packets, 0);
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bandwidth_estimation_.UpdateReceiverBlock(
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fraction_lost_aggregate, TimeDelta::ms(rtt), total_number_of_packets,
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Timestamp::ms(now_ms));
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}
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MaybeTriggerOnNetworkChanged();
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}
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void BitrateControllerImpl::MaybeTriggerOnNetworkChanged() {
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if (!observer_)
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return;
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uint32_t bitrate_bps;
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uint8_t fraction_loss;
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int64_t rtt;
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if (GetNetworkParameters(&bitrate_bps, &fraction_loss, &rtt))
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observer_->OnNetworkChanged(bitrate_bps, fraction_loss, rtt);
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}
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bool BitrateControllerImpl::GetNetworkParameters(uint32_t* bitrate,
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uint8_t* fraction_loss,
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int64_t* rtt) {
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rtc::CritScope cs(&critsect_);
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int current_bitrate;
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|
||||||
bandwidth_estimation_.CurrentEstimate(¤t_bitrate, fraction_loss, rtt);
|
|
||||||
*bitrate = current_bitrate;
|
|
||||||
|
|
||||||
bool new_bitrate = false;
|
|
||||||
if (*bitrate != last_bitrate_bps_ || *fraction_loss != last_fraction_loss_ ||
|
|
||||||
*rtt != last_rtt_ms_) {
|
|
||||||
last_bitrate_bps_ = *bitrate;
|
|
||||||
last_fraction_loss_ = *fraction_loss;
|
|
||||||
last_rtt_ms_ = *rtt;
|
|
||||||
new_bitrate = true;
|
|
||||||
}
|
|
||||||
|
|
||||||
BWE_TEST_LOGGING_PLOT(1, "fraction_loss_%", clock_->TimeInMilliseconds(),
|
|
||||||
(last_fraction_loss_ * 100) / 256);
|
|
||||||
BWE_TEST_LOGGING_PLOT(1, "rtt_ms", clock_->TimeInMilliseconds(),
|
|
||||||
last_rtt_ms_);
|
|
||||||
BWE_TEST_LOGGING_PLOT(1, "Target_bitrate_kbps", clock_->TimeInMilliseconds(),
|
|
||||||
last_bitrate_bps_ / 1000);
|
|
||||||
|
|
||||||
return new_bitrate;
|
|
||||||
}
|
|
||||||
|
|
||||||
bool BitrateControllerImpl::AvailableBandwidth(uint32_t* bandwidth) const {
|
|
||||||
rtc::CritScope cs(&critsect_);
|
|
||||||
int bitrate;
|
|
||||||
uint8_t fraction_loss;
|
|
||||||
int64_t rtt;
|
|
||||||
bandwidth_estimation_.CurrentEstimate(&bitrate, &fraction_loss, &rtt);
|
|
||||||
if (bitrate > 0) {
|
|
||||||
*bandwidth = bitrate;
|
|
||||||
return true;
|
|
||||||
}
|
|
||||||
return false;
|
|
||||||
}
|
|
||||||
} // namespace webrtc
|
|
||||||
|
|
@ -1,103 +0,0 @@
|
||||||
/*
|
|
||||||
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
||||||
*
|
|
||||||
* Use of this source code is governed by a BSD-style license
|
|
||||||
* that can be found in the LICENSE file in the root of the source
|
|
||||||
* tree. An additional intellectual property rights grant can be found
|
|
||||||
* in the file PATENTS. All contributing project authors may
|
|
||||||
* be found in the AUTHORS file in the root of the source tree.
|
|
||||||
*
|
|
||||||
* Usage: this class will register multiple RtcpBitrateObserver's one at each
|
|
||||||
* RTCP module. It will aggregate the results and run one bandwidth estimation
|
|
||||||
* and push the result to the encoder via VideoEncoderCallback.
|
|
||||||
*/
|
|
||||||
|
|
||||||
#ifndef MODULES_BITRATE_CONTROLLER_BITRATE_CONTROLLER_IMPL_H_
|
|
||||||
#define MODULES_BITRATE_CONTROLLER_BITRATE_CONTROLLER_IMPL_H_
|
|
||||||
|
|
||||||
#include "modules/bitrate_controller/include/bitrate_controller.h"
|
|
||||||
|
|
||||||
#include <list>
|
|
||||||
#include <map>
|
|
||||||
#include <utility>
|
|
||||||
#include <vector>
|
|
||||||
|
|
||||||
#include "modules/bitrate_controller/send_side_bandwidth_estimation.h"
|
|
||||||
#include "rtc_base/constructor_magic.h"
|
|
||||||
#include "rtc_base/critical_section.h"
|
|
||||||
|
|
||||||
namespace webrtc {
|
|
||||||
|
|
||||||
class BitrateControllerImpl : public BitrateController {
|
|
||||||
public:
|
|
||||||
// TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
|
|
||||||
// |observer| is left for project that is not yet updated.
|
|
||||||
BitrateControllerImpl(Clock* clock,
|
|
||||||
BitrateObserver* observer,
|
|
||||||
RtcEventLog* event_log);
|
|
||||||
virtual ~BitrateControllerImpl() {}
|
|
||||||
|
|
||||||
bool AvailableBandwidth(uint32_t* bandwidth) const override;
|
|
||||||
|
|
||||||
RTC_DEPRECATED RtcpBandwidthObserver* CreateRtcpBandwidthObserver() override;
|
|
||||||
|
|
||||||
// Deprecated
|
|
||||||
void SetStartBitrate(int start_bitrate_bps) override;
|
|
||||||
// Deprecated
|
|
||||||
void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) override;
|
|
||||||
|
|
||||||
void SetBitrates(int start_bitrate_bps,
|
|
||||||
int min_bitrate_bps,
|
|
||||||
int max_bitrate_bps) override;
|
|
||||||
|
|
||||||
void ResetBitrates(int bitrate_bps,
|
|
||||||
int min_bitrate_bps,
|
|
||||||
int max_bitrate_bps) override;
|
|
||||||
|
|
||||||
// Returns true if the parameters have changed since the last call.
|
|
||||||
bool GetNetworkParameters(uint32_t* bitrate,
|
|
||||||
uint8_t* fraction_loss,
|
|
||||||
int64_t* rtt) override;
|
|
||||||
|
|
||||||
void OnDelayBasedBweResult(const DelayBasedBwe::Result& result) override;
|
|
||||||
|
|
||||||
int64_t TimeUntilNextProcess() override;
|
|
||||||
void Process() override;
|
|
||||||
|
|
||||||
private:
|
|
||||||
class RtcpBandwidthObserverImpl;
|
|
||||||
|
|
||||||
// Called by BitrateObserver's direct from the RTCP module.
|
|
||||||
// Implements RtcpBandwidthObserver.
|
|
||||||
void OnReceivedEstimatedBitrate(uint32_t bitrate) override;
|
|
||||||
|
|
||||||
void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
|
|
||||||
int64_t rtt,
|
|
||||||
int64_t now_ms) override;
|
|
||||||
|
|
||||||
// Deprecated
|
|
||||||
void MaybeTriggerOnNetworkChanged();
|
|
||||||
|
|
||||||
void OnNetworkChanged(uint32_t bitrate,
|
|
||||||
uint8_t fraction_loss, // 0 - 255.
|
|
||||||
int64_t rtt) RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
|
|
||||||
|
|
||||||
// Used by process thread.
|
|
||||||
Clock* const clock_;
|
|
||||||
BitrateObserver* const observer_;
|
|
||||||
int64_t last_bitrate_update_ms_;
|
|
||||||
RtcEventLog* const event_log_;
|
|
||||||
|
|
||||||
rtc::CriticalSection critsect_;
|
|
||||||
std::map<uint32_t, uint32_t> ssrc_to_last_received_extended_high_seq_num_
|
|
||||||
RTC_GUARDED_BY(critsect_);
|
|
||||||
SendSideBandwidthEstimation bandwidth_estimation_ RTC_GUARDED_BY(critsect_);
|
|
||||||
|
|
||||||
uint32_t last_bitrate_bps_ RTC_GUARDED_BY(critsect_);
|
|
||||||
uint8_t last_fraction_loss_ RTC_GUARDED_BY(critsect_);
|
|
||||||
int64_t last_rtt_ms_ RTC_GUARDED_BY(critsect_);
|
|
||||||
|
|
||||||
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(BitrateControllerImpl);
|
|
||||||
};
|
|
||||||
} // namespace webrtc
|
|
||||||
#endif // MODULES_BITRATE_CONTROLLER_BITRATE_CONTROLLER_IMPL_H_
|
|
||||||
|
|
@ -1,461 +0,0 @@
|
||||||
/*
|
|
||||||
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
||||||
*
|
|
||||||
* Use of this source code is governed by a BSD-style license
|
|
||||||
* that can be found in the LICENSE file in the root of the source
|
|
||||||
* tree. An additional intellectual property rights grant can be found
|
|
||||||
* in the file PATENTS. All contributing project authors may
|
|
||||||
* be found in the AUTHORS file in the root of the source tree.
|
|
||||||
*/
|
|
||||||
|
|
||||||
#include <stdint.h>
|
|
||||||
#include <memory>
|
|
||||||
|
|
||||||
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
|
|
||||||
#include "modules/bitrate_controller/include/bitrate_controller.h"
|
|
||||||
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
|
|
||||||
#include "modules/pacing/paced_sender.h"
|
|
||||||
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
|
|
||||||
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
|
||||||
#include "system_wrappers/include/clock.h"
|
|
||||||
#include "test/field_trial.h"
|
|
||||||
#include "test/gmock.h"
|
|
||||||
#include "test/gtest.h"
|
|
||||||
|
|
||||||
using ::testing::Exactly;
|
|
||||||
using ::testing::Return;
|
|
||||||
|
|
||||||
using webrtc::BitrateController;
|
|
||||||
using webrtc::BitrateObserver;
|
|
||||||
using webrtc::PacedSender;
|
|
||||||
using webrtc::RtcpBandwidthObserver;
|
|
||||||
|
|
||||||
uint8_t WeightedLoss(int num_packets1,
|
|
||||||
uint8_t fraction_loss1,
|
|
||||||
int num_packets2,
|
|
||||||
uint8_t fraction_loss2) {
|
|
||||||
int weighted_sum =
|
|
||||||
num_packets1 * fraction_loss1 + num_packets2 * fraction_loss2;
|
|
||||||
int total_num_packets = num_packets1 + num_packets2;
|
|
||||||
return (weighted_sum + total_num_packets / 2) / total_num_packets;
|
|
||||||
}
|
|
||||||
|
|
||||||
webrtc::RTCPReportBlock CreateReportBlock(
|
|
||||||
uint32_t remote_ssrc,
|
|
||||||
uint32_t source_ssrc,
|
|
||||||
uint8_t fraction_lost,
|
|
||||||
uint32_t extended_high_sequence_number) {
|
|
||||||
return webrtc::RTCPReportBlock(remote_ssrc, source_ssrc, fraction_lost, 0,
|
|
||||||
extended_high_sequence_number, 0, 0, 0);
|
|
||||||
}
|
|
||||||
|
|
||||||
class TestBitrateObserver : public BitrateObserver {
|
|
||||||
public:
|
|
||||||
TestBitrateObserver()
|
|
||||||
: last_bitrate_(0), last_fraction_loss_(0), last_rtt_(0) {}
|
|
||||||
|
|
||||||
virtual void OnNetworkChanged(uint32_t bitrate,
|
|
||||||
uint8_t fraction_loss,
|
|
||||||
int64_t rtt) {
|
|
||||||
last_bitrate_ = static_cast<int>(bitrate);
|
|
||||||
last_fraction_loss_ = fraction_loss;
|
|
||||||
last_rtt_ = rtt;
|
|
||||||
}
|
|
||||||
int last_bitrate_;
|
|
||||||
uint8_t last_fraction_loss_;
|
|
||||||
int64_t last_rtt_;
|
|
||||||
};
|
|
||||||
|
|
||||||
class BitrateControllerTest : public ::testing::Test {
|
|
||||||
protected:
|
|
||||||
BitrateControllerTest() : clock_(0) {}
|
|
||||||
~BitrateControllerTest() {}
|
|
||||||
|
|
||||||
virtual void SetUp() {
|
|
||||||
controller_.reset(BitrateController::CreateBitrateController(
|
|
||||||
&clock_, &bitrate_observer_, &event_log_));
|
|
||||||
controller_->SetStartBitrate(kStartBitrateBps);
|
|
||||||
EXPECT_EQ(kStartBitrateBps, bitrate_observer_.last_bitrate_);
|
|
||||||
controller_->SetMinMaxBitrate(kMinBitrateBps, kMaxBitrateBps);
|
|
||||||
EXPECT_EQ(kStartBitrateBps, bitrate_observer_.last_bitrate_);
|
|
||||||
bandwidth_observer_ = controller_.get();
|
|
||||||
}
|
|
||||||
|
|
||||||
virtual void TearDown() {}
|
|
||||||
|
|
||||||
const int kMinBitrateBps = 100000;
|
|
||||||
const int kStartBitrateBps = 200000;
|
|
||||||
const int kMaxBitrateBps = 300000;
|
|
||||||
|
|
||||||
const int kDefaultMinBitrateBps = 10000;
|
|
||||||
const int kDefaultMaxBitrateBps = 1000000000;
|
|
||||||
|
|
||||||
webrtc::SimulatedClock clock_;
|
|
||||||
TestBitrateObserver bitrate_observer_;
|
|
||||||
std::unique_ptr<BitrateController> controller_;
|
|
||||||
RtcpBandwidthObserver* bandwidth_observer_;
|
|
||||||
::testing::NiceMock<webrtc::MockRtcEventLog> event_log_;
|
|
||||||
};
|
|
||||||
|
|
||||||
TEST_F(BitrateControllerTest, DefaultMinMaxBitrate) {
|
|
||||||
// Receive successively lower REMBs, verify the reserved bitrate is deducted.
|
|
||||||
controller_->SetMinMaxBitrate(0, 0);
|
|
||||||
EXPECT_EQ(kStartBitrateBps, bitrate_observer_.last_bitrate_);
|
|
||||||
bandwidth_observer_->OnReceivedEstimatedBitrate(kDefaultMinBitrateBps / 2);
|
|
||||||
EXPECT_EQ(webrtc::congestion_controller::GetMinBitrateBps(),
|
|
||||||
bitrate_observer_.last_bitrate_);
|
|
||||||
bandwidth_observer_->OnReceivedEstimatedBitrate(2 * kDefaultMaxBitrateBps);
|
|
||||||
clock_.AdvanceTimeMilliseconds(1000);
|
|
||||||
controller_->Process();
|
|
||||||
EXPECT_EQ(kDefaultMaxBitrateBps, bitrate_observer_.last_bitrate_);
|
|
||||||
}
|
|
||||||
|
|
||||||
TEST_F(BitrateControllerTest, OneBitrateObserverOneRtcpObserver) {
|
|
||||||
// First REMB applies immediately.
|
|
||||||
int64_t time_ms = 1001;
|
|
||||||
webrtc::ReportBlockList report_blocks;
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 1));
|
|
||||||
bandwidth_observer_->OnReceivedEstimatedBitrate(200000);
|
|
||||||
EXPECT_EQ(200000, bitrate_observer_.last_bitrate_);
|
|
||||||
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
|
|
||||||
EXPECT_EQ(0, bitrate_observer_.last_rtt_);
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
report_blocks.clear();
|
|
||||||
time_ms += 2000;
|
|
||||||
|
|
||||||
// Receive a high remb, test bitrate inc.
|
|
||||||
bandwidth_observer_->OnReceivedEstimatedBitrate(400000);
|
|
||||||
|
|
||||||
// Test bitrate increase 8% per second.
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 21));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
EXPECT_EQ(217000, bitrate_observer_.last_bitrate_);
|
|
||||||
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
|
|
||||||
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
|
|
||||||
time_ms += 1000;
|
|
||||||
|
|
||||||
report_blocks.clear();
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 41));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
EXPECT_EQ(235360, bitrate_observer_.last_bitrate_);
|
|
||||||
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
|
|
||||||
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
|
|
||||||
time_ms += 1000;
|
|
||||||
|
|
||||||
report_blocks.clear();
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 61));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
EXPECT_EQ(255189, bitrate_observer_.last_bitrate_);
|
|
||||||
time_ms += 1000;
|
|
||||||
|
|
||||||
report_blocks.clear();
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 81));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
EXPECT_EQ(276604, bitrate_observer_.last_bitrate_);
|
|
||||||
time_ms += 1000;
|
|
||||||
|
|
||||||
report_blocks.clear();
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 101));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
EXPECT_EQ(299732, bitrate_observer_.last_bitrate_);
|
|
||||||
time_ms += 1000;
|
|
||||||
|
|
||||||
// Reach max cap.
|
|
||||||
report_blocks.clear();
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 121));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
EXPECT_EQ(300000, bitrate_observer_.last_bitrate_);
|
|
||||||
time_ms += 1000;
|
|
||||||
|
|
||||||
report_blocks.clear();
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 141));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
EXPECT_EQ(300000, bitrate_observer_.last_bitrate_);
|
|
||||||
|
|
||||||
// Test that a low delay-based estimate limits the combined estimate.
|
|
||||||
webrtc::DelayBasedBwe::Result result(false, webrtc::DataRate::kbps(280));
|
|
||||||
controller_->OnDelayBasedBweResult(result);
|
|
||||||
EXPECT_EQ(280000, bitrate_observer_.last_bitrate_);
|
|
||||||
|
|
||||||
// Test that a low REMB limits the combined estimate.
|
|
||||||
bandwidth_observer_->OnReceivedEstimatedBitrate(250000);
|
|
||||||
EXPECT_EQ(250000, bitrate_observer_.last_bitrate_);
|
|
||||||
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
|
|
||||||
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
|
|
||||||
|
|
||||||
bandwidth_observer_->OnReceivedEstimatedBitrate(1000);
|
|
||||||
EXPECT_EQ(100000, bitrate_observer_.last_bitrate_);
|
|
||||||
}
|
|
||||||
|
|
||||||
TEST_F(BitrateControllerTest, OneBitrateObserverTwoRtcpObservers) {
|
|
||||||
const uint32_t kSenderSsrc1 = 1;
|
|
||||||
const uint32_t kSenderSsrc2 = 2;
|
|
||||||
const uint32_t kMediaSsrc1 = 3;
|
|
||||||
const uint32_t kMediaSsrc2 = 4;
|
|
||||||
|
|
||||||
int64_t time_ms = 1;
|
|
||||||
webrtc::ReportBlockList report_blocks;
|
|
||||||
report_blocks = {CreateReportBlock(kSenderSsrc1, kMediaSsrc1, 0, 1)};
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
time_ms += 500;
|
|
||||||
|
|
||||||
RtcpBandwidthObserver* second_bandwidth_observer = controller_.get();
|
|
||||||
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 21)};
|
|
||||||
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
|
|
||||||
time_ms);
|
|
||||||
|
|
||||||
// Test start bitrate.
|
|
||||||
EXPECT_EQ(200000, bitrate_observer_.last_bitrate_);
|
|
||||||
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
|
|
||||||
EXPECT_EQ(100, bitrate_observer_.last_rtt_);
|
|
||||||
time_ms += 500;
|
|
||||||
|
|
||||||
// Test bitrate increase 8% per second.
|
|
||||||
report_blocks = {CreateReportBlock(kSenderSsrc1, kMediaSsrc1, 0, 21)};
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
time_ms += 500;
|
|
||||||
|
|
||||||
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 21)};
|
|
||||||
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
|
|
||||||
time_ms);
|
|
||||||
EXPECT_EQ(217000, bitrate_observer_.last_bitrate_);
|
|
||||||
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
|
|
||||||
EXPECT_EQ(100, bitrate_observer_.last_rtt_);
|
|
||||||
time_ms += 500;
|
|
||||||
|
|
||||||
// Extra report should not change estimate.
|
|
||||||
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 31)};
|
|
||||||
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
|
|
||||||
time_ms);
|
|
||||||
EXPECT_EQ(217000, bitrate_observer_.last_bitrate_);
|
|
||||||
time_ms += 500;
|
|
||||||
|
|
||||||
report_blocks = {CreateReportBlock(kSenderSsrc1, kMediaSsrc1, 0, 41)};
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
EXPECT_EQ(235360, bitrate_observer_.last_bitrate_);
|
|
||||||
|
|
||||||
// Second report should not change estimate.
|
|
||||||
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 41)};
|
|
||||||
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
|
|
||||||
time_ms);
|
|
||||||
EXPECT_EQ(235360, bitrate_observer_.last_bitrate_);
|
|
||||||
time_ms += 1000;
|
|
||||||
|
|
||||||
// Reports from only one bandwidth observer is ok.
|
|
||||||
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 61)};
|
|
||||||
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
|
|
||||||
time_ms);
|
|
||||||
EXPECT_EQ(255189, bitrate_observer_.last_bitrate_);
|
|
||||||
time_ms += 1000;
|
|
||||||
|
|
||||||
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 81)};
|
|
||||||
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
|
|
||||||
time_ms);
|
|
||||||
EXPECT_EQ(276604, bitrate_observer_.last_bitrate_);
|
|
||||||
time_ms += 1000;
|
|
||||||
|
|
||||||
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 121)};
|
|
||||||
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
|
|
||||||
time_ms);
|
|
||||||
EXPECT_EQ(299732, bitrate_observer_.last_bitrate_);
|
|
||||||
time_ms += 1000;
|
|
||||||
|
|
||||||
// Reach max cap.
|
|
||||||
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 141)};
|
|
||||||
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
|
|
||||||
time_ms);
|
|
||||||
EXPECT_EQ(300000, bitrate_observer_.last_bitrate_);
|
|
||||||
|
|
||||||
// Test that a low REMB trigger immediately.
|
|
||||||
// We don't care which bandwidth observer that delivers the REMB.
|
|
||||||
second_bandwidth_observer->OnReceivedEstimatedBitrate(250000);
|
|
||||||
EXPECT_EQ(250000, bitrate_observer_.last_bitrate_);
|
|
||||||
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
|
|
||||||
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
|
|
||||||
|
|
||||||
// Min cap.
|
|
||||||
bandwidth_observer_->OnReceivedEstimatedBitrate(1000);
|
|
||||||
EXPECT_EQ(100000, bitrate_observer_.last_bitrate_);
|
|
||||||
}
|
|
||||||
|
|
||||||
TEST_F(BitrateControllerTest, OneBitrateObserverMultipleReportBlocks) {
|
|
||||||
uint32_t sequence_number[2] = {0, 0xFF00};
|
|
||||||
const int kStartBitrate = 200000;
|
|
||||||
const int kMinBitrate = 100000;
|
|
||||||
const int kMaxBitrate = 300000;
|
|
||||||
controller_->SetStartBitrate(kStartBitrate);
|
|
||||||
controller_->SetMinMaxBitrate(kMinBitrate, kMaxBitrate);
|
|
||||||
|
|
||||||
// REMBs during the first 2 seconds apply immediately.
|
|
||||||
int64_t time_ms = 1001;
|
|
||||||
webrtc::ReportBlockList report_blocks;
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, sequence_number[0]));
|
|
||||||
bandwidth_observer_->OnReceivedEstimatedBitrate(kStartBitrate);
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
report_blocks.clear();
|
|
||||||
time_ms += 2000;
|
|
||||||
|
|
||||||
// Receive a high REMB, test bitrate increase.
|
|
||||||
bandwidth_observer_->OnReceivedEstimatedBitrate(400000);
|
|
||||||
|
|
||||||
int last_bitrate = 0;
|
|
||||||
// Ramp up to max bitrate.
|
|
||||||
for (int i = 0; i < 7; ++i) {
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, sequence_number[0]));
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 3, 0, sequence_number[1]));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50,
|
|
||||||
time_ms);
|
|
||||||
EXPECT_GT(bitrate_observer_.last_bitrate_, last_bitrate);
|
|
||||||
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
|
|
||||||
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
|
|
||||||
last_bitrate = bitrate_observer_.last_bitrate_;
|
|
||||||
time_ms += 1000;
|
|
||||||
sequence_number[0] += 20;
|
|
||||||
sequence_number[1] += 1;
|
|
||||||
report_blocks.clear();
|
|
||||||
}
|
|
||||||
|
|
||||||
EXPECT_EQ(kMaxBitrate, bitrate_observer_.last_bitrate_);
|
|
||||||
|
|
||||||
// Packet loss on the first stream. Verify that bitrate decreases.
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 50, sequence_number[0]));
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 3, 0, sequence_number[1]));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
EXPECT_LT(bitrate_observer_.last_bitrate_, last_bitrate);
|
|
||||||
EXPECT_EQ(WeightedLoss(20, 50, 1, 0), bitrate_observer_.last_fraction_loss_);
|
|
||||||
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
|
|
||||||
last_bitrate = bitrate_observer_.last_bitrate_;
|
|
||||||
sequence_number[0] += 20;
|
|
||||||
sequence_number[1] += 20;
|
|
||||||
time_ms += 1000;
|
|
||||||
report_blocks.clear();
|
|
||||||
|
|
||||||
// Packet loss on the second stream. Verify that bitrate decreases.
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, sequence_number[0]));
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 3, 75, sequence_number[1]));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
EXPECT_LT(bitrate_observer_.last_bitrate_, last_bitrate);
|
|
||||||
EXPECT_EQ(WeightedLoss(20, 0, 20, 75), bitrate_observer_.last_fraction_loss_);
|
|
||||||
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
|
|
||||||
last_bitrate = bitrate_observer_.last_bitrate_;
|
|
||||||
sequence_number[0] += 20;
|
|
||||||
sequence_number[1] += 1;
|
|
||||||
time_ms += 1000;
|
|
||||||
report_blocks.clear();
|
|
||||||
|
|
||||||
// All packets lost on stream with few packets, no back-off.
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, sequence_number[0]));
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 3, 255, sequence_number[1]));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
|
|
||||||
EXPECT_EQ(bitrate_observer_.last_bitrate_, last_bitrate);
|
|
||||||
EXPECT_EQ(WeightedLoss(20, 0, 1, 255), bitrate_observer_.last_fraction_loss_);
|
|
||||||
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
|
|
||||||
last_bitrate = bitrate_observer_.last_bitrate_;
|
|
||||||
sequence_number[0] += 20;
|
|
||||||
sequence_number[1] += 1;
|
|
||||||
report_blocks.clear();
|
|
||||||
}
|
|
||||||
|
|
||||||
TEST_F(BitrateControllerTest, TimeoutsWithoutFeedback) {
|
|
||||||
{
|
|
||||||
webrtc::test::ScopedFieldTrials override_field_trials(
|
|
||||||
"WebRTC-FeedbackTimeout/Enabled/");
|
|
||||||
SetUp();
|
|
||||||
int expected_bitrate_bps = 300000;
|
|
||||||
controller_->SetBitrates(300000, kDefaultMinBitrateBps,
|
|
||||||
kDefaultMaxBitrateBps);
|
|
||||||
|
|
||||||
webrtc::ReportBlockList report_blocks;
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 1));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(
|
|
||||||
report_blocks, 50, clock_.TimeInMilliseconds());
|
|
||||||
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
|
|
||||||
clock_.AdvanceTimeMilliseconds(500);
|
|
||||||
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 21));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(
|
|
||||||
report_blocks, 50, clock_.TimeInMilliseconds());
|
|
||||||
report_blocks.clear();
|
|
||||||
expected_bitrate_bps = expected_bitrate_bps * 1.08 + 1000;
|
|
||||||
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
|
|
||||||
clock_.AdvanceTimeMilliseconds(1500);
|
|
||||||
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 41));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(
|
|
||||||
report_blocks, 50, clock_.TimeInMilliseconds());
|
|
||||||
expected_bitrate_bps = expected_bitrate_bps * 1.08 + 1000;
|
|
||||||
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
|
|
||||||
clock_.AdvanceTimeMilliseconds(4000);
|
|
||||||
|
|
||||||
// 4 seconds since feedback, expect increase.
|
|
||||||
controller_->Process();
|
|
||||||
expected_bitrate_bps = expected_bitrate_bps * 1.08 + 1000;
|
|
||||||
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
|
|
||||||
clock_.AdvanceTimeMilliseconds(2000);
|
|
||||||
|
|
||||||
// 6 seconds since feedback, expect no increase.
|
|
||||||
controller_->Process();
|
|
||||||
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
|
|
||||||
clock_.AdvanceTimeMilliseconds(9001);
|
|
||||||
|
|
||||||
// More than 15 seconds since feedback, expect decrease.
|
|
||||||
controller_->Process();
|
|
||||||
expected_bitrate_bps *= 0.8;
|
|
||||||
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
|
|
||||||
clock_.AdvanceTimeMilliseconds(500);
|
|
||||||
|
|
||||||
// Only one timeout every second.
|
|
||||||
controller_->Process();
|
|
||||||
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
|
|
||||||
clock_.AdvanceTimeMilliseconds(501);
|
|
||||||
|
|
||||||
// New timeout allowed.
|
|
||||||
controller_->Process();
|
|
||||||
expected_bitrate_bps *= 0.8;
|
|
||||||
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
TEST_F(BitrateControllerTest, StopIncreaseWithoutPacketReports) {
|
|
||||||
int expected_bitrate_bps = 300000;
|
|
||||||
controller_->SetBitrates(300000, kDefaultMinBitrateBps,
|
|
||||||
kDefaultMaxBitrateBps);
|
|
||||||
|
|
||||||
webrtc::ReportBlockList report_blocks;
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 1));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(
|
|
||||||
report_blocks, 50, clock_.TimeInMilliseconds());
|
|
||||||
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
|
|
||||||
clock_.AdvanceTimeMilliseconds(500);
|
|
||||||
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 21));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(
|
|
||||||
report_blocks, 50, clock_.TimeInMilliseconds());
|
|
||||||
report_blocks.clear();
|
|
||||||
expected_bitrate_bps = expected_bitrate_bps * 1.08 + 1000;
|
|
||||||
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
|
|
||||||
clock_.AdvanceTimeMilliseconds(1500);
|
|
||||||
|
|
||||||
// 1.2 seconds without packets reported as received, no increase.
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 21));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(
|
|
||||||
report_blocks, 50, clock_.TimeInMilliseconds());
|
|
||||||
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
|
|
||||||
clock_.AdvanceTimeMilliseconds(1000);
|
|
||||||
|
|
||||||
// 5 packets reported as received since last, too few, no increase.
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 26));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(
|
|
||||||
report_blocks, 50, clock_.TimeInMilliseconds());
|
|
||||||
report_blocks.clear();
|
|
||||||
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
|
|
||||||
clock_.AdvanceTimeMilliseconds(100);
|
|
||||||
|
|
||||||
// 15 packets reported as received since last, enough to increase.
|
|
||||||
report_blocks.push_back(CreateReportBlock(1, 2, 0, 41));
|
|
||||||
bandwidth_observer_->OnReceivedRtcpReceiverReport(
|
|
||||||
report_blocks, 50, clock_.TimeInMilliseconds());
|
|
||||||
expected_bitrate_bps = expected_bitrate_bps * 1.08 + 1000;
|
|
||||||
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
|
|
||||||
clock_.AdvanceTimeMilliseconds(1000);
|
|
||||||
}
|
|
||||||
|
|
@ -1,102 +0,0 @@
|
||||||
/*
|
|
||||||
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
||||||
*
|
|
||||||
* Use of this source code is governed by a BSD-style license
|
|
||||||
* that can be found in the LICENSE file in the root of the source
|
|
||||||
* tree. An additional intellectual property rights grant can be found
|
|
||||||
* in the file PATENTS. All contributing project authors may
|
|
||||||
* be found in the AUTHORS file in the root of the source tree.
|
|
||||||
*
|
|
||||||
* Usage: this class will register multiple RtcpBitrateObserver's one at each
|
|
||||||
* RTCP module. It will aggregate the results and run one bandwidth estimation
|
|
||||||
* and push the result to the encoders via BitrateObserver(s).
|
|
||||||
*/
|
|
||||||
|
|
||||||
#ifndef MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
|
|
||||||
#define MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
|
|
||||||
|
|
||||||
#include <stddef.h>
|
|
||||||
#include <stdint.h>
|
|
||||||
|
|
||||||
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
|
|
||||||
#include "modules/include/module.h"
|
|
||||||
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
|
||||||
#include "rtc_base/deprecation.h"
|
|
||||||
|
|
||||||
namespace webrtc {
|
|
||||||
|
|
||||||
class Clock;
|
|
||||||
class RtcEventLog;
|
|
||||||
|
|
||||||
// Deprecated
|
|
||||||
// TODO(perkj): Remove BitrateObserver when no implementations use it.
|
|
||||||
class BitrateObserver {
|
|
||||||
// Observer class for bitrate changes announced due to change in bandwidth
|
|
||||||
// estimate or due to bitrate allocation changes. Fraction loss and rtt is
|
|
||||||
// also part of this callback to allow the obsevrer to optimize its settings
|
|
||||||
// for different types of network environments. The bitrate does not include
|
|
||||||
// packet headers and is measured in bits per second.
|
|
||||||
public:
|
|
||||||
virtual void OnNetworkChanged(uint32_t bitrate_bps,
|
|
||||||
uint8_t fraction_loss, // 0 - 255.
|
|
||||||
int64_t rtt_ms) = 0;
|
|
||||||
// TODO(gnish): Merge these two into one function.
|
|
||||||
virtual void OnNetworkChanged(uint32_t bitrate_for_encoder_bps,
|
|
||||||
uint32_t bitrate_for_pacer_bps,
|
|
||||||
bool in_probe_rtt,
|
|
||||||
int64_t target_set_time,
|
|
||||||
uint64_t congestion_window) {}
|
|
||||||
virtual void OnBytesAcked(size_t bytes) {}
|
|
||||||
virtual size_t pacer_queue_size_in_bytes();
|
|
||||||
virtual ~BitrateObserver() {}
|
|
||||||
};
|
|
||||||
|
|
||||||
class BitrateController : public Module, public RtcpBandwidthObserver {
|
|
||||||
// This class collects feedback from all streams sent to a peer (via
|
|
||||||
// RTCPBandwidthObservers). It does one aggregated send side bandwidth
|
|
||||||
// estimation and divide the available bitrate between all its registered
|
|
||||||
// BitrateObservers.
|
|
||||||
public:
|
|
||||||
static const int kDefaultStartBitratebps = 300000;
|
|
||||||
|
|
||||||
// Deprecated:
|
|
||||||
// TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
|
|
||||||
// Remove this method once other other projects does not use it.
|
|
||||||
static BitrateController* CreateBitrateController(Clock* clock,
|
|
||||||
BitrateObserver* observer,
|
|
||||||
RtcEventLog* event_log);
|
|
||||||
|
|
||||||
static BitrateController* CreateBitrateController(Clock* clock,
|
|
||||||
RtcEventLog* event_log);
|
|
||||||
|
|
||||||
~BitrateController() override {}
|
|
||||||
|
|
||||||
// Deprecated, use raw pointer to BitrateController instance instead.
|
|
||||||
// Creates RtcpBandwidthObserver caller responsible to delete.
|
|
||||||
RTC_DEPRECATED virtual RtcpBandwidthObserver*
|
|
||||||
CreateRtcpBandwidthObserver() = 0;
|
|
||||||
|
|
||||||
// Deprecated
|
|
||||||
virtual void SetStartBitrate(int start_bitrate_bps) = 0;
|
|
||||||
// Deprecated
|
|
||||||
virtual void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) = 0;
|
|
||||||
virtual void SetBitrates(int start_bitrate_bps,
|
|
||||||
int min_bitrate_bps,
|
|
||||||
int max_bitrate_bps) = 0;
|
|
||||||
|
|
||||||
virtual void ResetBitrates(int bitrate_bps,
|
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int min_bitrate_bps,
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int max_bitrate_bps) = 0;
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virtual void OnDelayBasedBweResult(const DelayBasedBwe::Result& result) = 0;
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// Gets the available payload bandwidth in bits per second. Note that
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// this bandwidth excludes packet headers.
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virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
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virtual bool GetNetworkParameters(uint32_t* bitrate,
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uint8_t* fraction_loss,
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int64_t* rtt) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
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Reference in a new issue