blob: a601cce34066a20268656ac8bc7d4c4b8cc57cb0 [file] [log] [blame]
/*
* 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 {
// For SendSideBwe, Opus bitrate should be in the range between 6000 and 32000.
const int kOpusMinBitrateBps = 6000;
const int kOpusBitrateFbBps = 32000;
// OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
} // namespace
AudioAllocationSettings::AudioAllocationSettings()
: audio_send_side_bwe_("Enabled"),
allocate_audio_without_feedback_("Enabled"),
force_no_audio_feedback_("Enabled"),
send_side_bwe_with_overhead_("Enabled"),
default_min_bitrate_("min", DataRate::bps(kOpusMinBitrateBps)),
default_max_bitrate_("max", DataRate::bps(kOpusBitrateFbBps)),
priority_bitrate_("prio", DataRate::Zero()) {
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({&send_side_bwe_with_overhead_},
field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead"));
ParseFieldTrial(
{&default_min_bitrate_, &default_max_bitrate_, &priority_bitrate_},
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::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;
}
int AudioAllocationSettings::MinBitrateBps() const {
return default_min_bitrate_->bps() + min_overhead_bps_;
}
int AudioAllocationSettings::MaxBitrateBps(
absl::optional<int> rtp_parameter_max_bitrate_bps) const {
// We assume that the max is a hard limit on the payload bitrate, so we add
// min_overhead_bps to it to ensure that, when overhead is deducted, the
// payload rate never goes beyond the limit. Note: this also means that if a
// higher overhead is forced, we cannot reach the limit.
// TODO(minyue): Reconsider this when the signaling to BWE is done
// through a dedicated API.
// This means that when RtpParameters is reset, we may change the
// encoder's bit rate immediately (through ReconfigureAudioSendStream()),
// meanwhile change the cap to the output of BWE.
if (rtp_parameter_max_bitrate_bps)
return *rtp_parameter_max_bitrate_bps + min_overhead_bps_;
return default_max_bitrate_->bps() + min_overhead_bps_;
}
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;
}
} // namespace webrtc