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8f119ca0a7
This CL makes it possible to configure the priority of audio streams in bitrate allocations using field trials. It also adds the option to forcibly ignore any injected audio allocation strategy, so that experimentation with allocation won't be blocked on the work to remove the strategy injection. Bug: webrtc:10603 Change-Id: Ic36ceee6c15eb0fad275866f77e2a121066e516c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135467 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27881}
124 lines
4.4 KiB
C++
124 lines
4.4 KiB
C++
/*
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* Copyright 2019 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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#include "rtc_base/experiments/audio_allocation_settings.h"
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#include "system_wrappers/include/field_trial.h"
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namespace webrtc {
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namespace {
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// OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
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constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
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} // namespace
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AudioAllocationSettings::AudioAllocationSettings()
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: audio_send_side_bwe_("Enabled"),
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allocate_audio_without_feedback_("Enabled"),
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force_no_audio_feedback_("Enabled"),
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send_side_bwe_with_overhead_("Enabled"),
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min_bitrate_("min"),
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max_bitrate_("max"),
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priority_bitrate_("prio_rate", DataRate::Zero()),
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bitrate_priority_("rate_prio") {
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ParseFieldTrial({&audio_send_side_bwe_},
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field_trial::FindFullName("WebRTC-Audio-SendSideBwe"));
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ParseFieldTrial({&allocate_audio_without_feedback_},
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field_trial::FindFullName("WebRTC-Audio-ABWENoTWCC"));
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ParseFieldTrial({&force_no_audio_feedback_},
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field_trial::FindFullName("WebRTC-Audio-ForceNoTWCC"));
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ParseFieldTrial({&send_side_bwe_with_overhead_},
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field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead"));
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ParseFieldTrial(
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{&min_bitrate_, &max_bitrate_, &priority_bitrate_, &bitrate_priority_},
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field_trial::FindFullName("WebRTC-Audio-Allocation"));
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// TODO(mflodman): Keep testing this and set proper values.
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// Note: This is an early experiment currently only supported by Opus.
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if (send_side_bwe_with_overhead_) {
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constexpr int kMaxPacketSizeMs = WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60;
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min_overhead_bps_ = kOverheadPerPacket * 8 * 1000 / kMaxPacketSizeMs;
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}
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}
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AudioAllocationSettings::~AudioAllocationSettings() {}
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bool AudioAllocationSettings::ForceNoAudioFeedback() const {
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return force_no_audio_feedback_;
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}
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bool AudioAllocationSettings::IgnoreSeqNumIdChange() const {
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return !audio_send_side_bwe_;
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}
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bool AudioAllocationSettings::ConfigureRateAllocationRange() const {
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return audio_send_side_bwe_;
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}
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bool AudioAllocationSettings::ShouldSendTransportSequenceNumber(
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int transport_seq_num_extension_header_id) const {
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if (force_no_audio_feedback_)
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return false;
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return audio_send_side_bwe_ && !allocate_audio_without_feedback_ &&
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transport_seq_num_extension_header_id != 0;
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}
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bool AudioAllocationSettings::IncludeAudioInAllocationOnStart(
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int min_bitrate_bps,
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int max_bitrate_bps,
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bool has_dscp,
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int transport_seq_num_extension_header_id) const {
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if (has_dscp || min_bitrate_bps == -1 || max_bitrate_bps == -1)
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return false;
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if (transport_seq_num_extension_header_id != 0 && !force_no_audio_feedback_)
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return true;
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if (allocate_audio_without_feedback_)
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return true;
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if (audio_send_side_bwe_)
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return false;
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return true;
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}
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bool AudioAllocationSettings::IncludeAudioInAllocationOnReconfigure(
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int min_bitrate_bps,
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int max_bitrate_bps,
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bool has_dscp,
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int transport_seq_num_extension_header_id) const {
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// TODO(srte): Make this match include_audio_in_allocation_on_start.
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if (has_dscp || min_bitrate_bps == -1 || max_bitrate_bps == -1)
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return false;
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if (transport_seq_num_extension_header_id != 0)
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return true;
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if (audio_send_side_bwe_)
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return false;
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return true;
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}
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bool AudioAllocationSettings::IncludeOverheadInAudioAllocation() const {
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return send_side_bwe_with_overhead_;
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}
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absl::optional<DataRate> AudioAllocationSettings::MinBitrate() const {
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return min_bitrate_.GetOptional();
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}
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absl::optional<DataRate> AudioAllocationSettings::MaxBitrate() const {
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return max_bitrate_.GetOptional();
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}
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DataRate AudioAllocationSettings::DefaultPriorityBitrate() const {
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DataRate max_overhead = DataRate::Zero();
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if (send_side_bwe_with_overhead_) {
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const TimeDelta kMinPacketDuration = TimeDelta::ms(20);
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max_overhead = DataSize::bytes(kOverheadPerPacket) / kMinPacketDuration;
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}
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return priority_bitrate_.Get() + max_overhead;
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}
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absl::optional<double> AudioAllocationSettings::BitratePriority() const {
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return bitrate_priority_.GetOptional();
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}
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} // namespace webrtc
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