chromium/third_party/webrtc/call/call.h

/*
 *  Copyright (c) 2013 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_H_
#define CALL_CALL_H_

#include <algorithm>
#include <memory>
#include <string>
#include <vector>

#include "absl/strings/string_view.h"
#include "api/adaptation/resource.h"
#include "api/media_types.h"
#include "api/task_queue/task_queue_base.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "call/call_config.h"
#include "call/flexfec_receive_stream.h"
#include "call/packet_receiver.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/ref_count.h"

namespace webrtc {

// A Call represents a two-way connection carrying zero or more outgoing
// and incoming media streams, transported over one or more RTP transports.

// A Call instance can contain several send and/or receive streams. All streams
// are assumed to have the same remote endpoint and will share bitrate estimates
// etc.

// When using the PeerConnection API, there is an one to one relationship
// between the PeerConnection and the Call.

class Call { … };

}  // namespace webrtc

#endif  // CALL_CALL_H_