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This class collects the field trial based configuration of audio allocation and bandwidth in one place. This makes it easier overview and prepares for future cleanup of the trials. Bug: webrtc:9718 Change-Id: I34a441c0165b423f1e2ee63894337484684146ac Reviewed-on: https://webrtc-review.googlesource.com/c/118282 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26370}
89 lines
4 KiB
C++
89 lines
4 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 the transport sequence number extension should be enabled.
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bool EnableTransportSequenceNumberExtension() const;
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// Returns true if audio traffic should be included in transport wide feedback
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// packets.
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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 IncludeAudioInFeedback(int transport_seq_num_extension_header_id) const;
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// Returns true if target bitrate for audio streams should be updated.
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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 UpdateAudioTargetBitrate(
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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 the min bitrate for audio rate allocation, potentially including
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// overhead.
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int MinBitrateBps() const;
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// Returns the max bitrate for audio rate allocation, potentially including
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// overhead. |rtp_parameter_max_bitrate_bps| max bitrate as configured in rtp
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// parameters, excluding overhead.
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int MaxBitrateBps(absl::optional<int> rtp_parameter_max_bitrate_bps) 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 audio_feedback_to_improve_video_bwe_;
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FieldTrialFlag send_side_bwe_with_overhead_;
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int min_overhead_bps_ = 0;
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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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