call.h revision 6ae48c660934784b4df56ab1ac99402ce3745e9f 
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  struct Config {
60    explicit Config(newapi::Transport* send_transport)
61        : webrtc_config(NULL),
62          send_transport(send_transport),
63          voice_engine(NULL),
64          overuse_callback(NULL) {}
65
66    webrtc::Config* webrtc_config;
67
68    newapi::Transport* send_transport;
69
70    // VoiceEngine used for audio/video synchronization for this Call.
71    VoiceEngine* voice_engine;
72
73    // Callback for overuse and normal usage based on the jitter of incoming
74    // captured frames. 'NULL' disables the callback.
75    OveruseCallback* overuse_callback;
76  };
77
78  static Call* Create(const Call::Config& config);
79
80  static Call* Create(const Call::Config& config,
81                      const webrtc::Config& webrtc_config);
82
83  virtual VideoSendStream::Config GetDefaultSendConfig() = 0;
84
85  virtual VideoSendStream* CreateVideoSendStream(
86      const VideoSendStream::Config& config,
87      const std::vector<VideoStream>& video_streams,
88      const void* encoder_settings) = 0;
89
90  virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
91
92  virtual VideoReceiveStream::Config GetDefaultReceiveConfig() = 0;
93
94  virtual VideoReceiveStream* CreateVideoReceiveStream(
95      const VideoReceiveStream::Config& config) = 0;
96  virtual void DestroyVideoReceiveStream(
97      VideoReceiveStream* receive_stream) = 0;
98
99  // All received RTP and RTCP packets for the call should be inserted to this
100  // PacketReceiver. The PacketReceiver pointer is valid as long as the
101  // Call instance exists.
102  virtual PacketReceiver* Receiver() = 0;
103
104  // Returns the estimated total send bandwidth. Note: this can differ from the
105  // actual encoded bitrate.
106  virtual uint32_t SendBitrateEstimate() = 0;
107
108  // Returns the total estimated receive bandwidth for the call. Note: this can
109  // differ from the actual receive bitrate.
110  virtual uint32_t ReceiveBitrateEstimate() = 0;
111
112  virtual ~Call() {}
113};
114}  // namespace webrtc
115
116#endif  // WEBRTC_CALL_H_
117