9356252bfb
(Re-land reverted cr). Use (in order from lowest to highest precedence): -- fixed 32000bps -- fixed target bitrate from codec -- explicit values from the rtp encoding parameters -- Final precedence is given to field trial values from WebRTC-Audio-Allocation Bug: webrtc:10487 Change-Id: I573e996fa1f243e673785cdbe687e029fd5cbf4a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133483 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Daniel Lee <dklee@google.com> Cr-Commit-Position: refs/heads/master@{#27847}
119 lines
4.2 KiB
C++
119 lines
4.2 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", DataRate::Zero()) {
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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({&min_bitrate_, &max_bitrate_, &priority_bitrate_},
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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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} // namespace webrtc
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