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/*
* 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.
*/
#ifndef RTC_BASE_EXPERIMENTS_AUDIO_ALLOCATION_SETTINGS_H_
#define RTC_BASE_EXPERIMENTS_AUDIO_ALLOCATION_SETTINGS_H_
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/experiments/field_trial_units.h"
namespace webrtc {
// This class encapsulates the logic that controls how allocation of audio
// bitrate is done. This is primarily based on field trials, but also on the
// values of audio parameters.
class AudioAllocationSettings {
public:
AudioAllocationSettings();
~AudioAllocationSettings();
// Returns true if audio feedback should be force disabled.
bool ForceNoAudioFeedback() const;
// Returns true if changes in transport sequence number id should be ignored
// as a trigger for reconfiguration.
bool IgnoreSeqNumIdChange() const;
// Returns true if the bitrate allocation range should be configured.
bool ConfigureRateAllocationRange() const;
// Returns true if sent audio packets should have transport wide sequence
// numbers.
// |transport_seq_num_extension_header_id| the extension header id for
// transport sequence numbers. Set to 0 if not the extension is not
// configured.
bool ShouldSendTransportSequenceNumber(
int transport_seq_num_extension_header_id) const;
// Returns true if audio should be added to rate allocation when the audio
// stream is started.
// |min_bitrate_bps| the configured min bitrate, set to -1 if unset.
// |max_bitrate_bps| the configured max bitrate, set to -1 if unset.
// |has_dscp| true is dscp is enabled.
// |transport_seq_num_extension_header_id| the extension header id for
// transport sequence numbers. Set to 0 if not the extension is not
// configured.
bool IncludeAudioInAllocationOnStart(
int min_bitrate_bps,
int max_bitrate_bps,
bool has_dscp,
int transport_seq_num_extension_header_id) const;
// Returns true if audio should be added to rate allocation when the audio
// stream is reconfigured.
// |min_bitrate_bps| the configured min bitrate, set to -1 if unset.
// |max_bitrate_bps| the configured max bitrate, set to -1 if unset.
// |has_dscp| true is dscp is enabled.
// |transport_seq_num_extension_header_id| the extension header id for
// transport sequence numbers. Set to 0 if not the extension is not
// configured.
bool IncludeAudioInAllocationOnReconfigure(
int min_bitrate_bps,
int max_bitrate_bps,
bool has_dscp,
int transport_seq_num_extension_header_id) const;
// Returns the min bitrate for audio rate allocation, potentially including
// overhead.
int MinBitrateBps() const;
// Returns the max bitrate for audio rate allocation, potentially including
// overhead. |rtp_parameter_max_bitrate_bps| max bitrate as configured in rtp
// parameters, excluding overhead.
int MaxBitrateBps(absl::optional<int> rtp_parameter_max_bitrate_bps) const;
// Indicates the default priority bitrate for audio streams. The bitrate
// allocator will prioritize audio until it reaches this bitrate and will
// divide bitrate evently between audio and video above this bitrate.
DataRate DefaultPriorityBitrate() const;
private:
FieldTrialFlag audio_send_side_bwe_;
FieldTrialFlag allocate_audio_without_feedback_;
FieldTrialFlag force_no_audio_feedback_;
FieldTrialFlag send_side_bwe_with_overhead_;
int min_overhead_bps_ = 0;
// Default bitrates to use as range if there's no user configured
// bitrate range but audio bitrate allocation is enabled.
FieldTrialParameter<DataRate> default_min_bitrate_;
FieldTrialParameter<DataRate> default_max_bitrate_;
FieldTrialParameter<DataRate> priority_bitrate_;
};
} // namespace webrtc
#endif // RTC_BASE_EXPERIMENTS_AUDIO_ALLOCATION_SETTINGS_H_