mirror of
https://github.com/mollyim/webrtc.git
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63658d06ec
This reverts commit e47aee3b86.
Reason for revert: Breaks downstream project
Original change's description:
> Ensure that we always set values for min and max audio bitrate.
>
> Use (in order from lowest to highest precedence):
> -- fixed 32000bps
> -- fixed target bitrate from codec
> -- explicit values from the rtp encoding parameters
> -- Final precedence is given to field trial values from
> WebRTC-Audio-Allocation
>
> Bug: webrtc:10487
> Change-Id: I7e289f209a927785572058b6fbfdf60fa14edf05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126229
> Reviewed-by: Minyue Li <minyue@google.com>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Daniel Lee <dklee@google.com>
> Cr-Commit-Position: refs/heads/master@{#27667}
TBR=solenberg@webrtc.org,stefan@webrtc.org,srte@webrtc.org,crodbro@webrtc.org,minyue@webrtc.org,minyue@google.com,dklee@google.com
Change-Id: Ie975cf40e65105d1e4cfab417b220b6bfc34592b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10487
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133481
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Daniel Lee <dklee@google.com>
Cr-Commit-Position: refs/heads/master@{#27670}
90 lines
4.1 KiB
C++
90 lines
4.1 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 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 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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