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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/fec_controller.h"
#include "api/rtcerror.h"
#include "call/audio_state.h"
#include "call/bitrate_constraints.h"
#include "rtc_base/platform_file.h"
namespace webrtc {
class AudioProcessing;
class RtcEventLog;
struct CallConfig {
explicit CallConfig(RtcEventLog* event_log);
~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.
// TODO(solenberg): Change this to a shared_ptr once we can use C++11.
rtc::scoped_refptr<AudioState> audio_state;
// Audio Processing Module to be used in this call.
// TODO(solenberg): Change this to a shared_ptr once we can use C++11.
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;
};
} // namespace webrtc
#endif // CALL_CALL_CONFIG_H_