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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 #include <utility> | 17 #include <utility> |
18 #include <vector> | 18 #include <vector> |
19 | 19 |
20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
21 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" | 21 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
22 | 22 |
23 namespace webrtc { | 23 namespace webrtc { |
24 | 24 |
25 class RtcEventLog; | 25 class RtcEventLog; |
26 | 26 |
27 class SendSideBandwidthEstimation { | 27 class SendSideBandwidthEstimation { |
28 public: | 28 public: |
29 SendSideBandwidthEstimation(); | 29 SendSideBandwidthEstimation() = delete; |
| 30 explicit SendSideBandwidthEstimation(RtcEventLog* event_log); |
30 virtual ~SendSideBandwidthEstimation(); | 31 virtual ~SendSideBandwidthEstimation(); |
31 | 32 |
32 void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const; | 33 void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const; |
33 | 34 |
34 // Call periodically to update estimate. | 35 // Call periodically to update estimate. |
35 void UpdateEstimate(int64_t now_ms); | 36 void UpdateEstimate(int64_t now_ms); |
36 | 37 |
37 // Call when we receive a RTCP message with TMMBR or REMB. | 38 // Call when we receive a RTCP message with TMMBR or REMB. |
38 void UpdateReceiverEstimate(int64_t now_ms, uint32_t bandwidth); | 39 void UpdateReceiverEstimate(int64_t now_ms, uint32_t bandwidth); |
39 | 40 |
40 // Call when a new delay-based estimate is available. | 41 // Call when a new delay-based estimate is available. |
41 void UpdateDelayBasedEstimate(int64_t now_ms, uint32_t bitrate_bps); | 42 void UpdateDelayBasedEstimate(int64_t now_ms, uint32_t bitrate_bps); |
42 | 43 |
43 // Call when we receive a RTCP message with a ReceiveBlock. | 44 // Call when we receive a RTCP message with a ReceiveBlock. |
44 void UpdateReceiverBlock(uint8_t fraction_loss, | 45 void UpdateReceiverBlock(uint8_t fraction_loss, |
45 int64_t rtt, | 46 int64_t rtt, |
46 int number_of_packets, | 47 int number_of_packets, |
47 int64_t now_ms); | 48 int64_t now_ms); |
48 | 49 |
49 void SetBitrates(int send_bitrate, | 50 void SetBitrates(int send_bitrate, |
50 int min_bitrate, | 51 int min_bitrate, |
51 int max_bitrate); | 52 int max_bitrate); |
52 void SetSendBitrate(int bitrate); | 53 void SetSendBitrate(int bitrate); |
53 void SetMinMaxBitrate(int min_bitrate, int max_bitrate); | 54 void SetMinMaxBitrate(int min_bitrate, int max_bitrate); |
54 int GetMinBitrate() const; | 55 int GetMinBitrate() const; |
55 | 56 |
56 void SetEventLog(RtcEventLog* event_log); | |
57 | |
58 private: | 57 private: |
59 enum UmaState { kNoUpdate, kFirstDone, kDone }; | 58 enum UmaState { kNoUpdate, kFirstDone, kDone }; |
60 | 59 |
61 bool IsInStartPhase(int64_t now_ms) const; | 60 bool IsInStartPhase(int64_t now_ms) const; |
62 | 61 |
63 void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets); | 62 void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets); |
64 | 63 |
65 // Returns the input bitrate capped to the thresholds defined by the max, | 64 // Returns the input bitrate capped to the thresholds defined by the max, |
66 // min and incoming bandwidth. | 65 // min and incoming bandwidth. |
67 uint32_t CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate); | 66 uint32_t CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate); |
(...skipping 24 matching lines...) Expand all Loading... |
92 int64_t time_last_decrease_ms_; | 91 int64_t time_last_decrease_ms_; |
93 int64_t first_report_time_ms_; | 92 int64_t first_report_time_ms_; |
94 int initially_lost_packets_; | 93 int initially_lost_packets_; |
95 int bitrate_at_2_seconds_kbps_; | 94 int bitrate_at_2_seconds_kbps_; |
96 UmaState uma_update_state_; | 95 UmaState uma_update_state_; |
97 std::vector<bool> rampup_uma_stats_updated_; | 96 std::vector<bool> rampup_uma_stats_updated_; |
98 RtcEventLog* event_log_; | 97 RtcEventLog* event_log_; |
99 }; | 98 }; |
100 } // namespace webrtc | 99 } // namespace webrtc |
101 #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ | 100 #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ |
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