mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 06:40:43 +01:00
a352248c43
Bug: webrtc:10200 Change-Id: Ifc5ea100cd66a7ccd6b777259d6531c93118eeb1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138064 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28035}
131 lines
4.7 KiB
C++
131 lines
4.7 KiB
C++
/*
|
|
* Copyright 2019 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 "rtc_base/experiments/audio_allocation_settings.h"
|
|
|
|
#include "system_wrappers/include/field_trial.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
// OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
|
|
constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
|
|
} // namespace
|
|
AudioAllocationSettings::AudioAllocationSettings()
|
|
: audio_send_side_bwe_(/*Flag key*/ "Enabled"),
|
|
allocate_audio_without_feedback_(/*Flag key*/ "Enabled"),
|
|
force_no_audio_feedback_(/*Flag key*/ "Enabled"),
|
|
disable_audio_alr_request_(/*Flag key*/ "Disabled"),
|
|
send_side_bwe_with_overhead_(/*Flag key*/ "Enabled"),
|
|
min_bitrate_("min"),
|
|
max_bitrate_("max"),
|
|
priority_bitrate_("prio_rate", DataRate::Zero()),
|
|
bitrate_priority_("rate_prio") {
|
|
ParseFieldTrial({&audio_send_side_bwe_},
|
|
field_trial::FindFullName("WebRTC-Audio-SendSideBwe"));
|
|
ParseFieldTrial({&allocate_audio_without_feedback_},
|
|
field_trial::FindFullName("WebRTC-Audio-ABWENoTWCC"));
|
|
ParseFieldTrial({&force_no_audio_feedback_},
|
|
field_trial::FindFullName("WebRTC-Audio-ForceNoTWCC"));
|
|
ParseFieldTrial({&disable_audio_alr_request_},
|
|
field_trial::FindFullName("WebRTC-Audio-AlrProbing"));
|
|
|
|
ParseFieldTrial({&send_side_bwe_with_overhead_},
|
|
field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead"));
|
|
ParseFieldTrial(
|
|
{&min_bitrate_, &max_bitrate_, &priority_bitrate_, &bitrate_priority_},
|
|
field_trial::FindFullName("WebRTC-Audio-Allocation"));
|
|
|
|
// TODO(mflodman): Keep testing this and set proper values.
|
|
// Note: This is an early experiment currently only supported by Opus.
|
|
if (send_side_bwe_with_overhead_) {
|
|
constexpr int kMaxPacketSizeMs = WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60;
|
|
min_overhead_bps_ = kOverheadPerPacket * 8 * 1000 / kMaxPacketSizeMs;
|
|
}
|
|
}
|
|
|
|
AudioAllocationSettings::~AudioAllocationSettings() {}
|
|
|
|
bool AudioAllocationSettings::ForceNoAudioFeedback() const {
|
|
return force_no_audio_feedback_;
|
|
}
|
|
|
|
bool AudioAllocationSettings::IgnoreSeqNumIdChange() const {
|
|
return !audio_send_side_bwe_;
|
|
}
|
|
|
|
bool AudioAllocationSettings::ConfigureRateAllocationRange() const {
|
|
return audio_send_side_bwe_;
|
|
}
|
|
|
|
bool AudioAllocationSettings::ShouldSendTransportSequenceNumber(
|
|
int transport_seq_num_extension_header_id) const {
|
|
if (force_no_audio_feedback_)
|
|
return false;
|
|
return audio_send_side_bwe_ && !allocate_audio_without_feedback_ &&
|
|
transport_seq_num_extension_header_id != 0;
|
|
}
|
|
|
|
bool AudioAllocationSettings::RequestAlrProbing() const {
|
|
return !disable_audio_alr_request_;
|
|
}
|
|
|
|
bool AudioAllocationSettings::IncludeAudioInAllocationOnStart(
|
|
int min_bitrate_bps,
|
|
int max_bitrate_bps,
|
|
bool has_dscp,
|
|
int transport_seq_num_extension_header_id) const {
|
|
if (has_dscp || min_bitrate_bps == -1 || max_bitrate_bps == -1)
|
|
return false;
|
|
if (transport_seq_num_extension_header_id != 0 && !force_no_audio_feedback_)
|
|
return true;
|
|
if (allocate_audio_without_feedback_)
|
|
return true;
|
|
if (audio_send_side_bwe_)
|
|
return false;
|
|
return true;
|
|
}
|
|
|
|
bool AudioAllocationSettings::IncludeAudioInAllocationOnReconfigure(
|
|
int min_bitrate_bps,
|
|
int max_bitrate_bps,
|
|
bool has_dscp,
|
|
int transport_seq_num_extension_header_id) const {
|
|
// TODO(srte): Make this match include_audio_in_allocation_on_start.
|
|
if (has_dscp || min_bitrate_bps == -1 || max_bitrate_bps == -1)
|
|
return false;
|
|
if (transport_seq_num_extension_header_id != 0)
|
|
return true;
|
|
if (audio_send_side_bwe_)
|
|
return false;
|
|
return true;
|
|
}
|
|
|
|
bool AudioAllocationSettings::IncludeOverheadInAudioAllocation() const {
|
|
return send_side_bwe_with_overhead_;
|
|
}
|
|
|
|
absl::optional<DataRate> AudioAllocationSettings::MinBitrate() const {
|
|
return min_bitrate_.GetOptional();
|
|
}
|
|
absl::optional<DataRate> AudioAllocationSettings::MaxBitrate() const {
|
|
return max_bitrate_.GetOptional();
|
|
}
|
|
DataRate AudioAllocationSettings::DefaultPriorityBitrate() const {
|
|
DataRate max_overhead = DataRate::Zero();
|
|
if (send_side_bwe_with_overhead_) {
|
|
const TimeDelta kMinPacketDuration = TimeDelta::ms(20);
|
|
max_overhead = DataSize::bytes(kOverheadPerPacket) / kMinPacketDuration;
|
|
}
|
|
return priority_bitrate_.Get() + max_overhead;
|
|
}
|
|
absl::optional<double> AudioAllocationSettings::BitratePriority() const {
|
|
return bitrate_priority_.GetOptional();
|
|
}
|
|
|
|
} // namespace webrtc
|