webrtc_m130/call/call_config.h
Ying Wang 0810a7c25a Add base class NetworkPredictor and NetworkPredictorFactory and wire up.
Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe.

Bug: webrtc:10492
Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27543}
2019-04-10 12:38:58 +00:00

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2.0 KiB
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/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef CALL_CALL_CONFIG_H_
#define CALL_CALL_CONFIG_H_
#include "api/bitrate_constraints.h"
#include "api/fec_controller.h"
#include "api/network_state_predictor.h"
#include "api/rtc_error.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/network_control.h"
#include "call/audio_state.h"
namespace webrtc {
class AudioProcessing;
class RtcEventLog;
struct CallConfig {
explicit CallConfig(RtcEventLog* event_log);
CallConfig(const CallConfig&);
~CallConfig();
RTC_DEPRECATED static constexpr int kDefaultStartBitrateBps = 300000;
// Bitrate config used until valid bitrate estimates are calculated. Also
// used to cap total bitrate used. This comes from the remote connection.
BitrateConstraints bitrate_config;
// AudioState which is possibly shared between multiple calls.
rtc::scoped_refptr<AudioState> audio_state;
// Audio Processing Module to be used in this call.
AudioProcessing* audio_processing = nullptr;
// RtcEventLog to use for this call. Required.
// Use webrtc::RtcEventLog::CreateNull() for a null implementation.
RtcEventLog* event_log = nullptr;
// FecController to use for this call.
FecControllerFactoryInterface* fec_controller_factory = nullptr;
// Task Queue Factory to be used in this call.
TaskQueueFactory* task_queue_factory = nullptr;
// NetworkStatePredictor to use for this call.
NetworkStatePredictorFactoryInterface* network_state_predictor_factory =
nullptr;
// Network controller factory to use for this call.
NetworkControllerFactoryInterface* network_controller_factory = nullptr;
};
} // namespace webrtc
#endif // CALL_CALL_CONFIG_H_