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921d366aed
There are no plans to start using std::shared_ptr in WebRTC. Bug: webrtc:10198 No-Try: True Change-Id: I87a6c32b33b30d1b6b98eccda3400ce755a0ae95 Reviewed-on: https://webrtc-review.googlesource.com/c/117362 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26264}
54 lines
1.7 KiB
C++
54 lines
1.7 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/bitrate_constraints.h"
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#include "api/fec_controller.h"
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#include "api/rtc_error.h"
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#include "api/transport/network_control.h"
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#include "call/audio_state.h"
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namespace webrtc {
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class AudioProcessing;
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class RtcEventLog;
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struct CallConfig {
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explicit CallConfig(RtcEventLog* event_log);
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CallConfig(const CallConfig&);
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~CallConfig();
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RTC_DEPRECATED static constexpr int kDefaultStartBitrateBps = 300000;
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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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// RtcEventLog to use for this call. Required.
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// Use webrtc::RtcEventLog::CreateNull() for a null implementation.
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RtcEventLog* event_log = nullptr;
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// FecController to use for this call.
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FecControllerFactoryInterface* fec_controller_factory = 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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};
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} // namespace webrtc
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#endif // CALL_CALL_CONFIG_H_
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