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Side by Side Diff: webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h

Issue 2061193002: Remove audio/video distinction for probe packets. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase + feedback Created 4 years, 6 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 10
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51 assert(false); 51 assert(false);
52 } 52 }
53 53
54 // Called for each incoming packet. Updates the incoming payload bitrate 54 // Called for each incoming packet. Updates the incoming payload bitrate
55 // estimate and the over-use detector. If an over-use is detected the 55 // estimate and the over-use detector. If an over-use is detected the
56 // remote bitrate estimate will be updated. Note that |payload_size| is the 56 // remote bitrate estimate will be updated. Note that |payload_size| is the
57 // packet size excluding headers. 57 // packet size excluding headers.
58 // Note that |arrival_time_ms| can be of an arbitrary time base. 58 // Note that |arrival_time_ms| can be of an arbitrary time base.
59 virtual void IncomingPacket(int64_t arrival_time_ms, 59 virtual void IncomingPacket(int64_t arrival_time_ms,
60 size_t payload_size, 60 size_t payload_size,
61 const RTPHeader& header, 61 const RTPHeader& header) = 0;
62 bool was_paced) = 0;
63 62
64 // Removes all data for |ssrc|. 63 // Removes all data for |ssrc|.
65 virtual void RemoveStream(uint32_t ssrc) = 0; 64 virtual void RemoveStream(uint32_t ssrc) = 0;
66 65
67 // Returns true if a valid estimate exists and sets |bitrate_bps| to the 66 // Returns true if a valid estimate exists and sets |bitrate_bps| to the
68 // estimated payload bitrate in bits per second. |ssrcs| is the list of ssrcs 67 // estimated payload bitrate in bits per second. |ssrcs| is the list of ssrcs
69 // currently being received and of which the bitrate estimate is based upon. 68 // currently being received and of which the bitrate estimate is based upon.
70 virtual bool LatestEstimate(std::vector<uint32_t>* ssrcs, 69 virtual bool LatestEstimate(std::vector<uint32_t>* ssrcs,
71 uint32_t* bitrate_bps) const = 0; 70 uint32_t* bitrate_bps) const = 0;
72 71
73 // TODO(holmer): Remove when all implementations have been updated. 72 // TODO(holmer): Remove when all implementations have been updated.
74 virtual bool GetStats(ReceiveBandwidthEstimatorStats* output) const { 73 virtual bool GetStats(ReceiveBandwidthEstimatorStats* output) const {
75 return false; 74 return false;
76 } 75 }
77 76
78 virtual void SetMinBitrate(int min_bitrate_bps) = 0; 77 virtual void SetMinBitrate(int min_bitrate_bps) = 0;
79 78
80 protected: 79 protected:
81 static const int64_t kProcessIntervalMs = 500; 80 static const int64_t kProcessIntervalMs = 500;
82 static const int64_t kStreamTimeOutMs = 2000; 81 static const int64_t kStreamTimeOutMs = 2000;
83 }; 82 };
84 83
85 } // namespace webrtc 84 } // namespace webrtc
86 85
87 #endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMA TOR_H_ 86 #endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMA TOR_H_
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