| /* |
| * 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_; |
| FieldTrialOptional<double> bitrate_priority_; |
| }; |
| } // namespace webrtc |
| |
| #endif // RTC_BASE_EXPERIMENTS_AUDIO_ALLOCATION_SETTINGS_H_ |