8f119ca0a7
This CL makes it possible to configure the priority of audio streams in bitrate allocations using field trials. It also adds the option to forcibly ignore any injected audio allocation strategy, so that experimentation with allocation won't be blocked on the work to remove the strategy injection. Bug: webrtc:10603 Change-Id: Ic36ceee6c15eb0fad275866f77e2a121066e516c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135467 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27881}
95 lines
4.2 KiB
C++
95 lines
4.2 KiB
C++
/*
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* Copyright 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef RTC_BASE_EXPERIMENTS_AUDIO_ALLOCATION_SETTINGS_H_
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#define RTC_BASE_EXPERIMENTS_AUDIO_ALLOCATION_SETTINGS_H_
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#include "rtc_base/experiments/field_trial_parser.h"
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#include "rtc_base/experiments/field_trial_units.h"
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namespace webrtc {
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// This class encapsulates the logic that controls how allocation of audio
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// bitrate is done. This is primarily based on field trials, but also on the
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// values of audio parameters.
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class AudioAllocationSettings {
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public:
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AudioAllocationSettings();
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~AudioAllocationSettings();
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// Returns true if audio feedback should be force disabled.
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bool ForceNoAudioFeedback() const;
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// Returns true if changes in transport sequence number id should be ignored
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// as a trigger for reconfiguration.
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bool IgnoreSeqNumIdChange() const;
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// Returns true if the bitrate allocation range should be configured.
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bool ConfigureRateAllocationRange() const;
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// Returns true if sent audio packets should have transport wide sequence
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// numbers.
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// |transport_seq_num_extension_header_id| the extension header id for
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// transport sequence numbers. Set to 0 if not the extension is not
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// configured.
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bool ShouldSendTransportSequenceNumber(
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int transport_seq_num_extension_header_id) const;
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// Returns true if audio should be added to rate allocation when the audio
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// stream is started.
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// |min_bitrate_bps| the configured min bitrate, set to -1 if unset.
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// |max_bitrate_bps| the configured max bitrate, set to -1 if unset.
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// |has_dscp| true is dscp is enabled.
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// |transport_seq_num_extension_header_id| the extension header id for
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// transport sequence numbers. Set to 0 if not the extension is not
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// configured.
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bool IncludeAudioInAllocationOnStart(
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int min_bitrate_bps,
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int max_bitrate_bps,
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bool has_dscp,
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int transport_seq_num_extension_header_id) const;
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// Returns true if audio should be added to rate allocation when the audio
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// stream is reconfigured.
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// |min_bitrate_bps| the configured min bitrate, set to -1 if unset.
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// |max_bitrate_bps| the configured max bitrate, set to -1 if unset.
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// |has_dscp| true is dscp is enabled.
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// |transport_seq_num_extension_header_id| the extension header id for
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// transport sequence numbers. Set to 0 if not the extension is not
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// configured.
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bool IncludeAudioInAllocationOnReconfigure(
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int min_bitrate_bps,
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int max_bitrate_bps,
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bool has_dscp,
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int transport_seq_num_extension_header_id) const;
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// Returns true if we should include packet overhead in audio allocation.
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bool IncludeOverheadInAudioAllocation() const;
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// Returns the min bitrate for audio rate allocation.
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absl::optional<DataRate> MinBitrate() const;
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// Returns the max bitrate for audio rate allocation.
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absl::optional<DataRate> MaxBitrate() const;
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// Indicates the default priority bitrate for audio streams. The bitrate
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// allocator will prioritize audio until it reaches this bitrate and will
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// divide bitrate evently between audio and video above this bitrate.
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DataRate DefaultPriorityBitrate() const;
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// The bitrate priority is used to determine how much of the available bitrate
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// beyond the min or priority bitrate audio streams should receive.
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absl::optional<double> BitratePriority() const;
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private:
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FieldTrialFlag audio_send_side_bwe_;
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FieldTrialFlag allocate_audio_without_feedback_;
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FieldTrialFlag force_no_audio_feedback_;
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FieldTrialFlag send_side_bwe_with_overhead_;
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int min_overhead_bps_ = 0;
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// Field Trial configured bitrates to use as overrides over default/user
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// configured bitrate range when audio bitrate allocation is enabled.
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FieldTrialOptional<DataRate> min_bitrate_;
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FieldTrialOptional<DataRate> max_bitrate_;
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FieldTrialParameter<DataRate> priority_bitrate_;
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FieldTrialOptional<double> bitrate_priority_;
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};
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} // namespace webrtc
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#endif // RTC_BASE_EXPERIMENTS_AUDIO_ALLOCATION_SETTINGS_H_
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