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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 | 10 |
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286 virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0; | 286 virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0; |
287 | 287 |
288 virtual void OnReceivedRtcpReceiverReport( | 288 virtual void OnReceivedRtcpReceiverReport( |
289 const ReportBlockList& report_blocks, | 289 const ReportBlockList& report_blocks, |
290 int64_t rtt, | 290 int64_t rtt, |
291 int64_t now_ms) = 0; | 291 int64_t now_ms) = 0; |
292 | 292 |
293 virtual ~RtcpBandwidthObserver() {} | 293 virtual ~RtcpBandwidthObserver() {} |
294 }; | 294 }; |
295 | 295 |
| 296 class SendTimeObserver { |
| 297 public: |
| 298 SendTimeObserver() {} |
| 299 virtual ~SendTimeObserver() {} |
| 300 |
| 301 // Transport-wide sequence number and timestamp (system time in milliseconds), |
| 302 // of when the packet was put on the wire. |
| 303 virtual void OnPacketSent(uint16_t transport_sequence_number, |
| 304 int64_t send_time) = 0; |
| 305 }; |
| 306 |
296 class RtcpRttStats { | 307 class RtcpRttStats { |
297 public: | 308 public: |
298 virtual void OnRttUpdate(int64_t rtt) = 0; | 309 virtual void OnRttUpdate(int64_t rtt) = 0; |
299 | 310 |
300 virtual int64_t LastProcessedRtt() const = 0; | 311 virtual int64_t LastProcessedRtt() const = 0; |
301 | 312 |
302 virtual ~RtcpRttStats() {}; | 313 virtual ~RtcpRttStats() {}; |
303 }; | 314 }; |
304 | 315 |
305 // Null object version of RtpFeedback. | 316 // Null object version of RtpFeedback. |
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358 uint64_t single_packet_loss_count; | 369 uint64_t single_packet_loss_count; |
359 // The number of events in which more than one adjacent packet was lost. | 370 // The number of events in which more than one adjacent packet was lost. |
360 uint64_t multiple_packet_loss_event_count; | 371 uint64_t multiple_packet_loss_event_count; |
361 // The number of packets lost in events where more than one adjacent packet | 372 // The number of packets lost in events where more than one adjacent packet |
362 // was lost. | 373 // was lost. |
363 uint64_t multiple_packet_loss_packet_count; | 374 uint64_t multiple_packet_loss_packet_count; |
364 }; | 375 }; |
365 | 376 |
366 } // namespace webrtc | 377 } // namespace webrtc |
367 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_ | 378 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_ |
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