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914351de5c
(reverted in https://webrtc-review.googlesource.com/c/src/+/123182/1) Original cl description: Always offer transport sequence number header extension for audio If the extension is negotiated, it will only be used if the field trial WebRTC-Audio-SendSideBwe is enabled. This allows simpler experimentation if it should be used or not. Patchset 3 contain the only change: Add the field trial WebRTC-Audio-SendSideBwe to call/rampup_tests.cc TBR: srte@webrtc.org,ossu@webrtc.org Bug: webrtc:10309 webrtc:10286 Change-Id: I2c1224e8a9fab52c1030369c1364686322e88a0f Reviewed-on: https://webrtc-review.googlesource.com/c/123183 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26706}
97 lines
4.4 KiB
C++
97 lines
4.4 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 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 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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// 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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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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// Default bitrates to use as range if there's no user configured
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// bitrate range but audio bitrate allocation is enabled.
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FieldTrialParameter<DataRate> default_min_bitrate_;
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FieldTrialParameter<DataRate> default_max_bitrate_;
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FieldTrialParameter<DataRate> priority_bitrate_;
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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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