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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 #ifndef WEBRTC_CALL_H_ | |
11 #define WEBRTC_CALL_H_ | |
12 | 10 |
13 #include <string> | 11 // This file is deprecated. It has been moved to the location below. Please |
14 #include <vector> | 12 // update your includes! See: http://bugs.webrtc.org/6716 |
15 | 13 #include "webrtc/call/call.h" |
16 #include "webrtc/api/call/audio_receive_stream.h" | |
17 #include "webrtc/api/call/audio_send_stream.h" | |
18 #include "webrtc/api/call/audio_state.h" | |
19 #include "webrtc/api/call/flexfec_receive_stream.h" | |
20 #include "webrtc/base/networkroute.h" | |
21 #include "webrtc/base/platform_file.h" | |
22 #include "webrtc/base/socket.h" | |
23 #include "webrtc/common_types.h" | |
24 #include "webrtc/video_receive_stream.h" | |
25 #include "webrtc/video_send_stream.h" | |
26 | |
27 namespace webrtc { | |
28 | |
29 class AudioProcessing; | |
30 class RtcEventLog; | |
31 | |
32 const char* Version(); | |
33 | |
34 enum class MediaType { | |
35 ANY, | |
36 AUDIO, | |
37 VIDEO, | |
38 DATA | |
39 }; | |
40 | |
41 class PacketReceiver { | |
42 public: | |
43 enum DeliveryStatus { | |
44 DELIVERY_OK, | |
45 DELIVERY_UNKNOWN_SSRC, | |
46 DELIVERY_PACKET_ERROR, | |
47 }; | |
48 | |
49 virtual DeliveryStatus DeliverPacket(MediaType media_type, | |
50 const uint8_t* packet, | |
51 size_t length, | |
52 const PacketTime& packet_time) = 0; | |
53 | |
54 protected: | |
55 virtual ~PacketReceiver() {} | |
56 }; | |
57 | |
58 // A Call instance can contain several send and/or receive streams. All streams | |
59 // are assumed to have the same remote endpoint and will share bitrate estimates | |
60 // etc. | |
61 class Call { | |
62 public: | |
63 struct Config { | |
64 explicit Config(RtcEventLog* event_log) : event_log(event_log) { | |
65 RTC_DCHECK(event_log); | |
66 } | |
67 | |
68 static const int kDefaultStartBitrateBps; | |
69 | |
70 // Bitrate config used until valid bitrate estimates are calculated. Also | |
71 // used to cap total bitrate used. | |
72 struct BitrateConfig { | |
73 int min_bitrate_bps = 0; | |
74 int start_bitrate_bps = kDefaultStartBitrateBps; | |
75 int max_bitrate_bps = -1; | |
76 } bitrate_config; | |
77 | |
78 // AudioState which is possibly shared between multiple calls. | |
79 // TODO(solenberg): Change this to a shared_ptr once we can use C++11. | |
80 rtc::scoped_refptr<AudioState> audio_state; | |
81 | |
82 // Audio Processing Module to be used in this call. | |
83 // TODO(solenberg): Change this to a shared_ptr once we can use C++11. | |
84 AudioProcessing* audio_processing = nullptr; | |
85 | |
86 // RtcEventLog to use for this call. Required. | |
87 // Use webrtc::RtcEventLog::CreateNull() for a null implementation. | |
88 RtcEventLog* event_log = nullptr; | |
89 }; | |
90 | |
91 struct Stats { | |
92 std::string ToString(int64_t time_ms) const; | |
93 | |
94 int send_bandwidth_bps = 0; // Estimated available send bandwidth. | |
95 int max_padding_bitrate_bps = 0; // Cumulative configured max padding. | |
96 int recv_bandwidth_bps = 0; // Estimated available receive bandwidth. | |
97 int64_t pacer_delay_ms = 0; | |
98 int64_t rtt_ms = -1; | |
99 }; | |
100 | |
101 static Call* Create(const Call::Config& config); | |
102 | |
103 virtual AudioSendStream* CreateAudioSendStream( | |
104 const AudioSendStream::Config& config) = 0; | |
105 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; | |
106 | |
107 virtual AudioReceiveStream* CreateAudioReceiveStream( | |
108 const AudioReceiveStream::Config& config) = 0; | |
109 virtual void DestroyAudioReceiveStream( | |
110 AudioReceiveStream* receive_stream) = 0; | |
111 | |
112 virtual VideoSendStream* CreateVideoSendStream( | |
113 VideoSendStream::Config config, | |
114 VideoEncoderConfig encoder_config) = 0; | |
115 virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0; | |
116 | |
117 virtual VideoReceiveStream* CreateVideoReceiveStream( | |
118 VideoReceiveStream::Config configuration) = 0; | |
119 virtual void DestroyVideoReceiveStream( | |
120 VideoReceiveStream* receive_stream) = 0; | |
121 | |
122 virtual FlexfecReceiveStream* CreateFlexfecReceiveStream( | |
123 FlexfecReceiveStream::Config configuration) = 0; | |
124 virtual void DestroyFlexfecReceiveStream( | |
125 FlexfecReceiveStream* receive_stream) = 0; | |
126 | |
127 // All received RTP and RTCP packets for the call should be inserted to this | |
128 // PacketReceiver. The PacketReceiver pointer is valid as long as the | |
129 // Call instance exists. | |
130 virtual PacketReceiver* Receiver() = 0; | |
131 | |
132 // Returns the call statistics, such as estimated send and receive bandwidth, | |
133 // pacing delay, etc. | |
134 virtual Stats GetStats() const = 0; | |
135 | |
136 // TODO(pbos): Like BitrateConfig above this is currently per-stream instead | |
137 // of maximum for entire Call. This should be fixed along with the above. | |
138 // Specifying a start bitrate (>0) will currently reset the current bitrate | |
139 // estimate. This is due to how the 'x-google-start-bitrate' flag is currently | |
140 // implemented. | |
141 virtual void SetBitrateConfig( | |
142 const Config::BitrateConfig& bitrate_config) = 0; | |
143 | |
144 // TODO(skvlad): When the unbundled case with multiple streams for the same | |
145 // media type going over different networks is supported, track the state | |
146 // for each stream separately. Right now it's global per media type. | |
147 virtual void SignalChannelNetworkState(MediaType media, | |
148 NetworkState state) = 0; | |
149 | |
150 virtual void OnTransportOverheadChanged( | |
151 MediaType media, | |
152 int transport_overhead_per_packet) = 0; | |
153 | |
154 virtual void OnNetworkRouteChanged( | |
155 const std::string& transport_name, | |
156 const rtc::NetworkRoute& network_route) = 0; | |
157 | |
158 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; | |
159 | |
160 virtual ~Call() {} | |
161 }; | |
162 | |
163 } // namespace webrtc | |
164 | |
165 #endif // WEBRTC_CALL_H_ | |
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