1/*
2 *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10#ifndef WEBRTC_CALL_H_
11#define WEBRTC_CALL_H_
12
13#include <string>
14#include <vector>
15
16#include "webrtc/common_types.h"
17#include "webrtc/video_receive_stream.h"
18#include "webrtc/video_send_stream.h"
19
20namespace webrtc {
21
22class VoiceEngine;
23
24const char* Version();
25
26class PacketReceiver {
27 public:
28  enum DeliveryStatus {
29    DELIVERY_OK,
30    DELIVERY_UNKNOWN_SSRC,
31    DELIVERY_PACKET_ERROR,
32  };
33
34  virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
35                                       size_t length) = 0;
36
37 protected:
38  virtual ~PacketReceiver() {}
39};
40
41// Callback interface for reporting when a system overuse is detected.
42// The detection is based on the jitter of incoming captured frames.
43class OveruseCallback {
44 public:
45  // Called as soon as an overuse is detected.
46  virtual void OnOveruse() = 0;
47  // Called periodically when the system is not overused any longer.
48  virtual void OnNormalUse() = 0;
49
50 protected:
51  virtual ~OveruseCallback() {}
52};
53
54// A Call instance can contain several send and/or receive streams. All streams
55// are assumed to have the same remote endpoint and will share bitrate estimates
56// etc.
57class Call {
58 public:
59  enum NetworkState {
60    kNetworkUp,
61    kNetworkDown,
62  };
63  struct Config {
64    explicit Config(newapi::Transport* send_transport)
65        : webrtc_config(NULL),
66          send_transport(send_transport),
67          voice_engine(NULL),
68          overuse_callback(NULL),
69          start_bitrate_bps(-1) {}
70
71    webrtc::Config* webrtc_config;
72
73    newapi::Transport* send_transport;
74
75    // VoiceEngine used for audio/video synchronization for this Call.
76    VoiceEngine* voice_engine;
77
78    // Callback for overuse and normal usage based on the jitter of incoming
79    // captured frames. 'NULL' disables the callback.
80    OveruseCallback* overuse_callback;
81
82    // Start bitrate used before a valid bitrate estimate is calculated. '-1'
83    // lets the call decide start bitrate.
84    // Note: This currently only affects video.
85    int start_bitrate_bps;
86  };
87
88  static Call* Create(const Call::Config& config);
89
90  static Call* Create(const Call::Config& config,
91                      const webrtc::Config& webrtc_config);
92
93  virtual VideoSendStream* CreateVideoSendStream(
94      const VideoSendStream::Config& config,
95      const VideoEncoderConfig& encoder_config) = 0;
96
97  virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
98
99  virtual VideoReceiveStream* CreateVideoReceiveStream(
100      const VideoReceiveStream::Config& config) = 0;
101  virtual void DestroyVideoReceiveStream(
102      VideoReceiveStream* receive_stream) = 0;
103
104  // All received RTP and RTCP packets for the call should be inserted to this
105  // PacketReceiver. The PacketReceiver pointer is valid as long as the
106  // Call instance exists.
107  virtual PacketReceiver* Receiver() = 0;
108
109  // Returns the estimated total send bandwidth. Note: this can differ from the
110  // actual encoded bitrate.
111  virtual uint32_t SendBitrateEstimate() = 0;
112
113  // Returns the total estimated receive bandwidth for the call. Note: this can
114  // differ from the actual receive bitrate.
115  virtual uint32_t ReceiveBitrateEstimate() = 0;
116
117  virtual void SignalNetworkState(NetworkState state) = 0;
118
119  virtual ~Call() {}
120};
121}  // namespace webrtc
122
123#endif  // WEBRTC_CALL_H_
124