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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h

Issue 1765443002: Added log messages for important call setup events: first packet sent/received (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Moved first packet logging from rtp_receiver_impl.cc to rtp_receiver_audio/video.cc Created 4 years, 9 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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47 int8_t payload_type, 47 int8_t payload_type,
48 uint32_t frequency) override; 48 uint32_t frequency) override;
49 49
50 int32_t InvokeOnInitializeDecoder( 50 int32_t InvokeOnInitializeDecoder(
51 RtpFeedback* callback, 51 RtpFeedback* callback,
52 int8_t payload_type, 52 int8_t payload_type,
53 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 53 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
54 const PayloadUnion& specific_payload) const override; 54 const PayloadUnion& specific_payload) const override;
55 55
56 void SetPacketOverHead(uint16_t packet_over_head); 56 void SetPacketOverHead(uint16_t packet_over_head);
57
58 private:
59 bool first_packet_received_ = false;
pthatcher1 2016/03/16 04:55:27 Strange indentation
skvlad 2016/03/16 19:07:04 Are you talking about the single space before "pri
pthatcher1 2016/03/16 19:17:26 Oh, you're right. I guess I'm not used to seeing
57 }; 60 };
58 } // namespace webrtc 61 } // namespace webrtc
59 62
60 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_ 63 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
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