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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 * FEC and NACK added bitrate is handled outside class | 10 * FEC and NACK added bitrate is handled outside class |
11 */ | 11 */ |
12 | 12 |
13 #ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ | 13 #ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ |
14 #define WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ | 14 #define WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ |
15 | 15 |
16 #include <deque> | 16 #include <deque> |
17 | 17 |
18 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 18 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
19 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" | 19 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
20 | 20 |
21 namespace webrtc { | 21 namespace webrtc { |
| 22 |
| 23 class RtcEventLog; |
| 24 |
22 class SendSideBandwidthEstimation { | 25 class SendSideBandwidthEstimation { |
23 public: | 26 public: |
24 SendSideBandwidthEstimation(); | 27 SendSideBandwidthEstimation(); |
25 virtual ~SendSideBandwidthEstimation(); | 28 virtual ~SendSideBandwidthEstimation(); |
26 | 29 |
27 void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const; | 30 void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const; |
28 | 31 |
29 // Call periodically to update estimate. | 32 // Call periodically to update estimate. |
30 void UpdateEstimate(int64_t now_ms); | 33 void UpdateEstimate(int64_t now_ms); |
31 | 34 |
32 // Call when we receive a RTCP message with TMMBR or REMB. | 35 // Call when we receive a RTCP message with TMMBR or REMB. |
33 void UpdateReceiverEstimate(int64_t now_ms, uint32_t bandwidth); | 36 void UpdateReceiverEstimate(int64_t now_ms, uint32_t bandwidth); |
34 | 37 |
35 // Call when we receive a RTCP message with a ReceiveBlock. | 38 // Call when we receive a RTCP message with a ReceiveBlock. |
36 void UpdateReceiverBlock(uint8_t fraction_loss, | 39 void UpdateReceiverBlock(uint8_t fraction_loss, |
37 int64_t rtt, | 40 int64_t rtt, |
38 int number_of_packets, | 41 int number_of_packets, |
39 int64_t now_ms); | 42 int64_t now_ms); |
40 | 43 |
41 void SetSendBitrate(int bitrate); | 44 void SetSendBitrate(int bitrate); |
42 void SetMinMaxBitrate(int min_bitrate, int max_bitrate); | 45 void SetMinMaxBitrate(int min_bitrate, int max_bitrate); |
43 int GetMinBitrate() const; | 46 int GetMinBitrate() const; |
44 | 47 |
| 48 void SetEventLog(RtcEventLog* event_log); |
| 49 |
45 private: | 50 private: |
46 enum UmaState { kNoUpdate, kFirstDone, kDone }; | 51 enum UmaState { kNoUpdate, kFirstDone, kDone }; |
47 | 52 |
48 bool IsInStartPhase(int64_t now_ms) const; | 53 bool IsInStartPhase(int64_t now_ms) const; |
49 | 54 |
50 void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets); | 55 void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets); |
51 | 56 |
52 // Returns the input bitrate capped to the thresholds defined by the max, | 57 // Returns the input bitrate capped to the thresholds defined by the max, |
53 // min and incoming bandwidth. | 58 // min and incoming bandwidth. |
54 uint32_t CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate); | 59 uint32_t CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate); |
(...skipping 19 matching lines...) Expand all Loading... |
74 uint8_t last_fraction_loss_; | 79 uint8_t last_fraction_loss_; |
75 int64_t last_round_trip_time_ms_; | 80 int64_t last_round_trip_time_ms_; |
76 | 81 |
77 uint32_t bwe_incoming_; | 82 uint32_t bwe_incoming_; |
78 int64_t time_last_decrease_ms_; | 83 int64_t time_last_decrease_ms_; |
79 int64_t first_report_time_ms_; | 84 int64_t first_report_time_ms_; |
80 int initially_lost_packets_; | 85 int initially_lost_packets_; |
81 int bitrate_at_2_seconds_kbps_; | 86 int bitrate_at_2_seconds_kbps_; |
82 UmaState uma_update_state_; | 87 UmaState uma_update_state_; |
83 std::vector<bool> rampup_uma_stats_updated_; | 88 std::vector<bool> rampup_uma_stats_updated_; |
| 89 RtcEventLog* event_log_; |
84 }; | 90 }; |
85 } // namespace webrtc | 91 } // namespace webrtc |
86 #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ | 92 #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ |
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