mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
7f8dbe18ea
Bug: webrtc:10815 Change-Id: I6a498d6c6bcd4fe4ba6ccc4d6f407d686528d946 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145333 Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28590}
100 lines
4.5 KiB
C++
100 lines
4.5 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.
|
|
*/
|
|
#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 request ALR probing from network controller.
|
|
bool RequestAlrProbing() 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 true if we should include packet overhead in audio allocation.
|
|
bool IncludeOverheadInAudioAllocation() const;
|
|
|
|
// Returns the min bitrate for audio rate allocation.
|
|
absl::optional<DataRate> MinBitrate() const;
|
|
// Returns the max bitrate for audio rate allocation.
|
|
absl::optional<DataRate> MaxBitrate() 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;
|
|
|
|
// The bitrate priority is used to determine how much of the available bitrate
|
|
// beyond the min or priority bitrate audio streams should receive.
|
|
absl::optional<double> BitratePriority() const;
|
|
|
|
private:
|
|
const bool audio_send_side_bwe_;
|
|
const bool allocate_audio_without_feedback_;
|
|
const bool force_no_audio_feedback_;
|
|
const bool enable_audio_alr_probing_;
|
|
const bool send_side_bwe_with_overhead_;
|
|
int min_overhead_bps_ = 0;
|
|
// Field Trial configured bitrates to use as overrides over default/user
|
|
// configured bitrate range when audio bitrate allocation is enabled.
|
|
FieldTrialOptional<DataRate> min_bitrate_;
|
|
FieldTrialOptional<DataRate> max_bitrate_;
|
|
FieldTrialParameter<DataRate> priority_bitrate_;
|
|
// By default the priority_bitrate is compensated for packet overhead.
|
|
// Use this flag to configure a raw value instead.
|
|
FieldTrialOptional<DataRate> priority_bitrate_raw_;
|
|
FieldTrialOptional<double> bitrate_priority_;
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // RTC_BASE_EXPERIMENTS_AUDIO_ALLOCATION_SETTINGS_H_
|