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Side by Side Diff: webrtc/video/payload_router.h

Issue 1652983002: Remove mutable from rtc::CriticalSections. (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) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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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62 // Returns the maximum allowed data payload length, given the configured MTU 62 // Returns the maximum allowed data payload length, given the configured MTU
63 // and RTP headers. 63 // and RTP headers.
64 size_t MaxPayloadLength() const; 64 size_t MaxPayloadLength() const;
65 65
66 void AddRef() { ++ref_count_; } 66 void AddRef() { ++ref_count_; }
67 void Release() { if (--ref_count_ == 0) { delete this; } } 67 void Release() { if (--ref_count_ == 0) { delete this; } }
68 68
69 private: 69 private:
70 // TODO(mflodman): When the new video API has launched, remove crit_ and 70 // TODO(mflodman): When the new video API has launched, remove crit_ and
71 // assume rtp_modules_ will never change during a call. 71 // assume rtp_modules_ will never change during a call.
72 mutable rtc::CriticalSection crit_; 72 rtc::CriticalSection crit_;
73 73
74 // Active sending RTP modules, in layer order. 74 // Active sending RTP modules, in layer order.
75 std::vector<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_); 75 std::vector<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_);
76 bool active_ GUARDED_BY(crit_); 76 bool active_ GUARDED_BY(crit_);
77 77
78 Atomic32 ref_count_; 78 Atomic32 ref_count_;
79 79
80 RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter); 80 RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
81 }; 81 };
82 82
83 } // namespace webrtc 83 } // namespace webrtc
84 84
85 #endif // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_ 85 #endif // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
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