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Side by Side Diff: webrtc/common_types.h

Issue 1720883002: Move RTCP histograms from vie_channel to video channel stats proxies. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 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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204 nack_requests += other.nack_requests; 204 nack_requests += other.nack_requests;
205 unique_nack_requests += other.unique_nack_requests; 205 unique_nack_requests += other.unique_nack_requests;
206 if (other.first_packet_time_ms != -1 && 206 if (other.first_packet_time_ms != -1 &&
207 (other.first_packet_time_ms < first_packet_time_ms || 207 (other.first_packet_time_ms < first_packet_time_ms ||
208 first_packet_time_ms == -1)) { 208 first_packet_time_ms == -1)) {
209 // Use oldest time. 209 // Use oldest time.
210 first_packet_time_ms = other.first_packet_time_ms; 210 first_packet_time_ms = other.first_packet_time_ms;
211 } 211 }
212 } 212 }
213 213
214 RtcpPacketTypeCounter Subtract(const RtcpPacketTypeCounter& other) const {
215 RtcpPacketTypeCounter diff = other;
216 diff.nack_packets -= nack_packets;
217 diff.fir_packets -= fir_packets;
218 diff.pli_packets -= pli_packets;
219 diff.nack_requests -= nack_requests;
220 diff.unique_nack_requests -= unique_nack_requests;
221 if (diff.first_packet_time_ms != -1 &&
222 diff.first_packet_time_ms < first_packet_time_ms) {
223 // Use youngest time.
224 diff.first_packet_time_ms = first_packet_time_ms;
225 }
226 return diff;
227 }
228
214 int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const { 229 int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
215 return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms); 230 return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
216 } 231 }
217 232
218 int UniqueNackRequestsInPercent() const { 233 int UniqueNackRequestsInPercent() const {
219 if (nack_requests == 0) { 234 if (nack_requests == 0) {
220 return 0; 235 return 0;
221 } 236 }
222 return static_cast<int>( 237 return static_cast<int>(
223 (unique_nack_requests * 100.0f / nack_requests) + 0.5f); 238 (unique_nack_requests * 100.0f / nack_requests) + 0.5f);
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838 uint32_t ssrc) = 0; 853 uint32_t ssrc) = 0;
839 }; 854 };
840 855
841 // RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size 856 // RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
842 // RTCP mode is described by RFC 5506. 857 // RTCP mode is described by RFC 5506.
843 enum class RtcpMode { kOff, kCompound, kReducedSize }; 858 enum class RtcpMode { kOff, kCompound, kReducedSize };
844 859
845 } // namespace webrtc 860 } // namespace webrtc
846 861
847 #endif // WEBRTC_COMMON_TYPES_H_ 862 #endif // WEBRTC_COMMON_TYPES_H_
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