webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h - Issue 1922483002: De-flake VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly

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Issue 1922483002: De-flake VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Feedback fixes Created 4 years, 7 months ago

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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

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39	39

40 // Call when a new delay-based estimate is available.	40 // Call when a new delay-based estimate is available.

41 void UpdateDelayBasedEstimate(int64_t now_ms, uint32_t bitrate_bps);	41 void UpdateDelayBasedEstimate(int64_t now_ms, uint32_t bitrate_bps);

42	42

43 // Call when we receive a RTCP message with a ReceiveBlock.	43 // Call when we receive a RTCP message with a ReceiveBlock.

44 void UpdateReceiverBlock(uint8_t fraction_loss,	44 void UpdateReceiverBlock(uint8_t fraction_loss,

45 int64_t rtt,	45 int64_t rtt,

46 int number_of_packets,	46 int number_of_packets,

47 int64_t now_ms);	47 int64_t now_ms);

48	48

	49 void SetBitrates(int send_bitrate,

	50 int min_bitrate,

	51 int max_bitrate);

49 void SetSendBitrate(int bitrate);	52 void SetSendBitrate(int bitrate);
	stefan-webrtc 2016/04/28 11:21:28 Deprecated Deprecated
50 void SetMinMaxBitrate(int min_bitrate, int max_bitrate);	53 void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
	stefan-webrtc 2016/04/28 11:21:28 Deprecated Deprecated
51 int GetMinBitrate() const;	54 int GetMinBitrate() const;

52	55

53 void SetEventLog(RtcEventLog* event_log);	56 void SetEventLog(RtcEventLog* event_log);

54	57

55 private:	58 private:

56 enum UmaState { kNoUpdate, kFirstDone, kDone };	59 enum UmaState { kNoUpdate, kFirstDone, kDone };

57	60

58 bool IsInStartPhase(int64_t now_ms) const;	61 bool IsInStartPhase(int64_t now_ms) const;

59	62

60 void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets);	63 void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets);

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89 int64_t time_last_decrease_ms_;	92 int64_t time_last_decrease_ms_;

90 int64_t first_report_time_ms_;	93 int64_t first_report_time_ms_;

91 int initially_lost_packets_;	94 int initially_lost_packets_;

92 int bitrate_at_2_seconds_kbps_;	95 int bitrate_at_2_seconds_kbps_;

93 UmaState uma_update_state_;	96 UmaState uma_update_state_;

94 std::vector<bool> rampup_uma_stats_updated_;	97 std::vector<bool> rampup_uma_stats_updated_;

95 RtcEventLog* event_log_;	98 RtcEventLog* event_log_;

96 };	99 };

97 } // namespace webrtc	100 } // namespace webrtc

98 #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_	101 #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_

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