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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_CALL_H_ | 10 #ifndef WEBRTC_CALL_CALL_H_ |
11 #define WEBRTC_CALL_CALL_H_ | 11 #define WEBRTC_CALL_CALL_H_ |
12 | 12 |
13 #include <string> | 13 #include <string> |
14 #include <vector> | 14 #include <vector> |
15 | 15 |
| 16 #include "webrtc/api/rtcerror.h" |
16 #include "webrtc/base/networkroute.h" | 17 #include "webrtc/base/networkroute.h" |
17 #include "webrtc/base/platform_file.h" | 18 #include "webrtc/base/platform_file.h" |
18 #include "webrtc/base/socket.h" | 19 #include "webrtc/base/socket.h" |
19 #include "webrtc/call/audio_receive_stream.h" | 20 #include "webrtc/call/audio_receive_stream.h" |
20 #include "webrtc/call/audio_send_stream.h" | 21 #include "webrtc/call/audio_send_stream.h" |
21 #include "webrtc/call/audio_state.h" | 22 #include "webrtc/call/audio_state.h" |
22 #include "webrtc/call/flexfec_receive_stream.h" | 23 #include "webrtc/call/flexfec_receive_stream.h" |
23 #include "webrtc/common_types.h" | 24 #include "webrtc/common_types.h" |
24 #include "webrtc/video_receive_stream.h" | 25 #include "webrtc/video_receive_stream.h" |
25 #include "webrtc/video_send_stream.h" | 26 #include "webrtc/video_send_stream.h" |
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68 static const int kDefaultStartBitrateBps; | 69 static const int kDefaultStartBitrateBps; |
69 | 70 |
70 // Bitrate config used until valid bitrate estimates are calculated. Also | 71 // Bitrate config used until valid bitrate estimates are calculated. Also |
71 // used to cap total bitrate used. | 72 // used to cap total bitrate used. |
72 struct BitrateConfig { | 73 struct BitrateConfig { |
73 int min_bitrate_bps = 0; | 74 int min_bitrate_bps = 0; |
74 int start_bitrate_bps = kDefaultStartBitrateBps; | 75 int start_bitrate_bps = kDefaultStartBitrateBps; |
75 int max_bitrate_bps = -1; | 76 int max_bitrate_bps = -1; |
76 } bitrate_config; | 77 } bitrate_config; |
77 | 78 |
| 79 // TODO(zstein): Consider using PeerConnectionInterface::BitrateParameters |
| 80 // instead (and move BitrateParameters to its own file in api/). |
| 81 struct BitrateConfigMask { |
| 82 rtc::Optional<int> min_bitrate_bps; |
| 83 rtc::Optional<int> start_bitrate_bps; |
| 84 rtc::Optional<int> max_bitrate_bps; |
| 85 }; |
| 86 |
78 // AudioState which is possibly shared between multiple calls. | 87 // AudioState which is possibly shared between multiple calls. |
79 // TODO(solenberg): Change this to a shared_ptr once we can use C++11. | 88 // TODO(solenberg): Change this to a shared_ptr once we can use C++11. |
80 rtc::scoped_refptr<AudioState> audio_state; | 89 rtc::scoped_refptr<AudioState> audio_state; |
81 | 90 |
82 // Audio Processing Module to be used in this call. | 91 // Audio Processing Module to be used in this call. |
83 // TODO(solenberg): Change this to a shared_ptr once we can use C++11. | 92 // TODO(solenberg): Change this to a shared_ptr once we can use C++11. |
84 AudioProcessing* audio_processing = nullptr; | 93 AudioProcessing* audio_processing = nullptr; |
85 | 94 |
86 // RtcEventLog to use for this call. Required. | 95 // RtcEventLog to use for this call. Required. |
87 // Use webrtc::RtcEventLog::CreateNull() for a null implementation. | 96 // Use webrtc::RtcEventLog::CreateNull() for a null implementation. |
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129 | 138 |
130 // All received RTP and RTCP packets for the call should be inserted to this | 139 // All received RTP and RTCP packets for the call should be inserted to this |
131 // PacketReceiver. The PacketReceiver pointer is valid as long as the | 140 // PacketReceiver. The PacketReceiver pointer is valid as long as the |
132 // Call instance exists. | 141 // Call instance exists. |
133 virtual PacketReceiver* Receiver() = 0; | 142 virtual PacketReceiver* Receiver() = 0; |
134 | 143 |
135 // Returns the call statistics, such as estimated send and receive bandwidth, | 144 // Returns the call statistics, such as estimated send and receive bandwidth, |
136 // pacing delay, etc. | 145 // pacing delay, etc. |
137 virtual Stats GetStats() const = 0; | 146 virtual Stats GetStats() const = 0; |
138 | 147 |
139 // TODO(pbos): Like BitrateConfig above this is currently per-stream instead | 148 // The min and start values will only be used if they are not set by |
140 // of maximum for entire Call. This should be fixed along with the above. | 149 // SetBitrateConfigMask. The minimum max set by the two calls will be used. |
141 // Specifying a start bitrate (>0) will currently reset the current bitrate | 150 // Specifying a start bitrate (>0) will reset the current bitrate estimate. |
142 // estimate. This is due to how the 'x-google-start-bitrate' flag is currently | 151 // This is due to how the 'x-google-start-bitrate' flag is currently |
143 // implemented. | 152 // implemented. |
144 virtual void SetBitrateConfig( | 153 virtual void SetBitrateConfig( |
145 const Config::BitrateConfig& bitrate_config) = 0; | 154 const Config::BitrateConfig& bitrate_config) = 0; |
146 | 155 |
| 156 // The min and start values set here are preferred to values set by |
| 157 // SetBitrateConfig. The minimum of the max set by the two calls will be used. |
| 158 // Assumes 0 <= min <= start <= max holds for set parameters. |
| 159 virtual RTCError SetBitrateConfigMask( |
| 160 const Config::BitrateConfigMask& bitrate_mask) = 0; |
| 161 |
147 // TODO(skvlad): When the unbundled case with multiple streams for the same | 162 // TODO(skvlad): When the unbundled case with multiple streams for the same |
148 // media type going over different networks is supported, track the state | 163 // media type going over different networks is supported, track the state |
149 // for each stream separately. Right now it's global per media type. | 164 // for each stream separately. Right now it's global per media type. |
150 virtual void SignalChannelNetworkState(MediaType media, | 165 virtual void SignalChannelNetworkState(MediaType media, |
151 NetworkState state) = 0; | 166 NetworkState state) = 0; |
152 | 167 |
153 virtual void OnTransportOverheadChanged( | 168 virtual void OnTransportOverheadChanged( |
154 MediaType media, | 169 MediaType media, |
155 int transport_overhead_per_packet) = 0; | 170 int transport_overhead_per_packet) = 0; |
156 | 171 |
157 virtual void OnNetworkRouteChanged( | 172 virtual void OnNetworkRouteChanged( |
158 const std::string& transport_name, | 173 const std::string& transport_name, |
159 const rtc::NetworkRoute& network_route) = 0; | 174 const rtc::NetworkRoute& network_route) = 0; |
160 | 175 |
161 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; | 176 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; |
162 | 177 |
163 virtual ~Call() {} | 178 virtual ~Call() {} |
164 }; | 179 }; |
165 | 180 |
166 } // namespace webrtc | 181 } // namespace webrtc |
167 | 182 |
168 #endif // WEBRTC_CALL_CALL_H_ | 183 #endif // WEBRTC_CALL_CALL_H_ |
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