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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 <algorithm> | 13 #include <algorithm> |
14 #include <memory> | 14 #include <memory> |
15 #include <string> | 15 #include <string> |
16 #include <vector> | 16 #include <vector> |
17 | 17 |
18 #include "webrtc/api/rtcerror.h" | 18 #include "webrtc/api/rtcerror.h" |
19 #include "webrtc/call/audio_receive_stream.h" | 19 #include "webrtc/call/audio_receive_stream.h" |
20 #include "webrtc/call/audio_send_stream.h" | 20 #include "webrtc/call/audio_send_stream.h" |
21 #include "webrtc/call/audio_state.h" | 21 #include "webrtc/call/audio_state.h" |
22 #include "webrtc/call/flexfec_receive_stream.h" | 22 #include "webrtc/call/flexfec_receive_stream.h" |
23 #include "webrtc/call/rtp_transport_controller_send_interface.h" | 23 #include "webrtc/call/rtp_transport_controller_send_interface.h" |
24 #include "webrtc/common_types.h" | 24 #include "webrtc/common_types.h" |
| 25 #include "webrtc/rtc_base/bitrateallocationstrategy.h" |
25 #include "webrtc/rtc_base/networkroute.h" | 26 #include "webrtc/rtc_base/networkroute.h" |
26 #include "webrtc/rtc_base/platform_file.h" | 27 #include "webrtc/rtc_base/platform_file.h" |
27 #include "webrtc/rtc_base/socket.h" | 28 #include "webrtc/rtc_base/socket.h" |
28 #include "webrtc/video_receive_stream.h" | 29 #include "webrtc/video_receive_stream.h" |
29 #include "webrtc/video_send_stream.h" | 30 #include "webrtc/video_send_stream.h" |
30 | 31 |
31 namespace webrtc { | 32 namespace webrtc { |
32 | 33 |
33 class AudioProcessing; | 34 class AudioProcessing; |
34 class RtcEventLog; | 35 class RtcEventLog; |
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176 virtual void SetBitrateConfig( | 177 virtual void SetBitrateConfig( |
177 const Config::BitrateConfig& bitrate_config) = 0; | 178 const Config::BitrateConfig& bitrate_config) = 0; |
178 | 179 |
179 // The greater min and smaller max set by this and SetBitrateConfig will be | 180 // The greater min and smaller max set by this and SetBitrateConfig will be |
180 // used. The latest non-negative start value form either call will be used. | 181 // used. The latest non-negative start value form either call will be used. |
181 // Specifying a start bitrate will reset the current bitrate estimate. | 182 // Specifying a start bitrate will reset the current bitrate estimate. |
182 // Assumes 0 <= min <= start <= max holds for set parameters. | 183 // Assumes 0 <= min <= start <= max holds for set parameters. |
183 virtual void SetBitrateConfigMask( | 184 virtual void SetBitrateConfigMask( |
184 const Config::BitrateConfigMask& bitrate_mask) = 0; | 185 const Config::BitrateConfigMask& bitrate_mask) = 0; |
185 | 186 |
| 187 virtual void SetBitrateAllocationStrategy( |
| 188 rtc::BitrateAllocationStrategy* bitrate_allocation_strategy) = 0; |
| 189 |
186 // TODO(skvlad): When the unbundled case with multiple streams for the same | 190 // TODO(skvlad): When the unbundled case with multiple streams for the same |
187 // media type going over different networks is supported, track the state | 191 // media type going over different networks is supported, track the state |
188 // for each stream separately. Right now it's global per media type. | 192 // for each stream separately. Right now it's global per media type. |
189 virtual void SignalChannelNetworkState(MediaType media, | 193 virtual void SignalChannelNetworkState(MediaType media, |
190 NetworkState state) = 0; | 194 NetworkState state) = 0; |
191 | 195 |
192 virtual void OnTransportOverheadChanged( | 196 virtual void OnTransportOverheadChanged( |
193 MediaType media, | 197 MediaType media, |
194 int transport_overhead_per_packet) = 0; | 198 int transport_overhead_per_packet) = 0; |
195 | 199 |
196 virtual void OnNetworkRouteChanged( | 200 virtual void OnNetworkRouteChanged( |
197 const std::string& transport_name, | 201 const std::string& transport_name, |
198 const rtc::NetworkRoute& network_route) = 0; | 202 const rtc::NetworkRoute& network_route) = 0; |
199 | 203 |
200 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; | 204 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; |
201 | 205 |
202 virtual ~Call() {} | 206 virtual ~Call() {} |
203 }; | 207 }; |
204 | 208 |
205 } // namespace webrtc | 209 } // namespace webrtc |
206 | 210 |
207 #endif // WEBRTC_CALL_CALL_H_ | 211 #endif // WEBRTC_CALL_CALL_H_ |
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