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Side by Side Diff: webrtc/call/video_receive_stream.h

Issue 3009793002: Implement googContentType GetStats metric reported on receive side. (Closed)
Patch Set: Fix ce Created 3 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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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80 rtc::Optional<uint64_t> qp_sum; 80 rtc::Optional<uint64_t> qp_sum;
81 81
82 int current_payload_type = -1; 82 int current_payload_type = -1;
83 83
84 int total_bitrate_bps = 0; 84 int total_bitrate_bps = 0;
85 int discarded_packets = 0; 85 int discarded_packets = 0;
86 86
87 int width = 0; 87 int width = 0;
88 int height = 0; 88 int height = 0;
89 89
90 // "screen" vs "realtime".
tommi 2017/08/30 15:03:27 if these are fixed constants, can we use const cha
ilnik 2017/08/30 15:30:24 Enum is absolutely not suitable here, as we want i
tommi 2017/08/30 17:54:42 Since stats gathering is a notoriously performance
ilnik 2017/08/30 18:12:18 If we introduce a second enum, it will be quite er
tommi 2017/08/31 07:28:48 Sorry for being slow but I'm still not fully under
ilnik 2017/08/31 07:51:11 Ok, I've made it enum everythere. Converting to st
91 std::string content_type;
92
90 int sync_offset_ms = std::numeric_limits<int>::max(); 93 int sync_offset_ms = std::numeric_limits<int>::max();
91 94
92 uint32_t ssrc = 0; 95 uint32_t ssrc = 0;
93 std::string c_name; 96 std::string c_name;
94 StreamDataCounters rtp_stats; 97 StreamDataCounters rtp_stats;
95 RtcpPacketTypeCounter rtcp_packet_type_counts; 98 RtcpPacketTypeCounter rtcp_packet_type_counts;
96 RtcpStatistics rtcp_stats; 99 RtcpStatistics rtcp_stats;
97 }; 100 };
98 101
99 struct Config { 102 struct Config {
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241 virtual void AddSecondarySink(RtpPacketSinkInterface* sink) = 0; 244 virtual void AddSecondarySink(RtpPacketSinkInterface* sink) = 0;
242 virtual void RemoveSecondarySink(const RtpPacketSinkInterface* sink) = 0; 245 virtual void RemoveSecondarySink(const RtpPacketSinkInterface* sink) = 0;
243 246
244 protected: 247 protected:
245 virtual ~VideoReceiveStream() {} 248 virtual ~VideoReceiveStream() {}
246 }; 249 };
247 250
248 } // namespace webrtc 251 } // namespace webrtc
249 252
250 #endif // WEBRTC_CALL_VIDEO_RECEIVE_STREAM_H_ 253 #endif // WEBRTC_CALL_VIDEO_RECEIVE_STREAM_H_
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