Repo created
This commit is contained in:
parent
81b91f4139
commit
f8c34fa5ee
22732 changed files with 4815320 additions and 2 deletions
83
TMessagesProj/jni/voip/webrtc/call/call_config.h
Normal file
83
TMessagesProj/jni/voip/webrtc/call/call_config.h
Normal file
|
|
@ -0,0 +1,83 @@
|
|||
/*
|
||||
* 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/environment/environment.h"
|
||||
#include "api/fec_controller.h"
|
||||
#include "api/metronome/metronome.h"
|
||||
#include "api/neteq/neteq_factory.h"
|
||||
#include "api/network_state_predictor.h"
|
||||
#include "api/transport/bitrate_settings.h"
|
||||
#include "api/transport/network_control.h"
|
||||
#include "call/audio_state.h"
|
||||
#include "call/rtp_transport_config.h"
|
||||
#include "call/rtp_transport_controller_send_factory_interface.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
class AudioProcessing;
|
||||
|
||||
struct CallConfig {
|
||||
// If `network_task_queue` is set to nullptr, Call will assume that network
|
||||
// related callbacks will be made on the same TQ as the Call instance was
|
||||
// constructed on.
|
||||
explicit CallConfig(const Environment& env,
|
||||
TaskQueueBase* network_task_queue = nullptr);
|
||||
|
||||
CallConfig(const CallConfig&);
|
||||
|
||||
~CallConfig();
|
||||
|
||||
RtpTransportConfig ExtractTransportConfig() const;
|
||||
|
||||
Environment env;
|
||||
|
||||
// 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;
|
||||
|
||||
// FecController to use for this call.
|
||||
FecControllerFactoryInterface* fec_controller_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;
|
||||
|
||||
// NetEq factory to use for this call.
|
||||
NetEqFactory* neteq_factory = nullptr;
|
||||
|
||||
TaskQueueBase* const network_task_queue_ = nullptr;
|
||||
// RtpTransportControllerSend to use for this call.
|
||||
RtpTransportControllerSendFactoryInterface*
|
||||
rtp_transport_controller_send_factory = nullptr;
|
||||
|
||||
Metronome* decode_metronome = nullptr;
|
||||
Metronome* encode_metronome = nullptr;
|
||||
|
||||
// The burst interval of the pacer, see TaskQueuePacedSender constructor.
|
||||
absl::optional<TimeDelta> pacer_burst_interval;
|
||||
|
||||
// Enables send packet batching from the egress RTP sender.
|
||||
bool enable_send_packet_batching = false;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // CALL_CALL_CONFIG_H_
|
||||
Loading…
Add table
Add a link
Reference in a new issue