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https://github.com/mollyim/webrtc.git
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f089d7ea54
This is a reland of commit b39c2a8464
Original change's description:
> FrameCadenceAdapter: align video encoding to metronome
>
> This CL aligns the video encoding tasks to metronome tick which
> similar with the metronome decoding.
>
> Design doc: https://docs.google.com/document/d/18PvEgS-DehClK6twCSCATOlX-j9acmXd-3vjb0tR9-Y
>
> Bug: b/304158952
> Change-Id: I262bd4a5097fdaeed559b9d7391a059ae86e2d63
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327460
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
> Cr-Commit-Position: refs/heads/main@{#41469}
Bug: b/304158952
Change-Id: Icf4e1ad91f5c98f3c32a88ffe4d6277e907353e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333464
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41479}
83 lines
2.7 KiB
C++
83 lines
2.7 KiB
C++
/*
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* Copyright (c) 2018 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 CALL_CALL_CONFIG_H_
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#define CALL_CALL_CONFIG_H_
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#include "api/environment/environment.h"
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#include "api/fec_controller.h"
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#include "api/metronome/metronome.h"
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#include "api/neteq/neteq_factory.h"
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#include "api/network_state_predictor.h"
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#include "api/transport/bitrate_settings.h"
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#include "api/transport/network_control.h"
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#include "call/audio_state.h"
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#include "call/rtp_transport_config.h"
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#include "call/rtp_transport_controller_send_factory_interface.h"
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namespace webrtc {
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class AudioProcessing;
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struct CallConfig {
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// If `network_task_queue` is set to nullptr, Call will assume that network
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// related callbacks will be made on the same TQ as the Call instance was
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// constructed on.
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explicit CallConfig(const Environment& env,
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TaskQueueBase* network_task_queue = nullptr);
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CallConfig(const CallConfig&);
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~CallConfig();
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RtpTransportConfig ExtractTransportConfig() const;
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Environment env;
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// Bitrate config used until valid bitrate estimates are calculated. Also
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// used to cap total bitrate used. This comes from the remote connection.
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BitrateConstraints bitrate_config;
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// AudioState which is possibly shared between multiple calls.
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rtc::scoped_refptr<AudioState> audio_state;
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// Audio Processing Module to be used in this call.
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AudioProcessing* audio_processing = nullptr;
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// FecController to use for this call.
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FecControllerFactoryInterface* fec_controller_factory = nullptr;
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// NetworkStatePredictor to use for this call.
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NetworkStatePredictorFactoryInterface* network_state_predictor_factory =
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nullptr;
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// Network controller factory to use for this call.
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NetworkControllerFactoryInterface* network_controller_factory = nullptr;
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// NetEq factory to use for this call.
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NetEqFactory* neteq_factory = nullptr;
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TaskQueueBase* const network_task_queue_ = nullptr;
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// RtpTransportControllerSend to use for this call.
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RtpTransportControllerSendFactoryInterface*
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rtp_transport_controller_send_factory = nullptr;
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Metronome* decode_metronome = nullptr;
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Metronome* encode_metronome = nullptr;
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// The burst interval of the pacer, see TaskQueuePacedSender constructor.
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absl::optional<TimeDelta> pacer_burst_interval;
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// Enables send packet batching from the egress RTP sender.
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bool enable_send_packet_batching = false;
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};
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} // namespace webrtc
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#endif // CALL_CALL_CONFIG_H_
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