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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.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: Corrected naming Created 4 years, 8 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
11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
13 13
14 #include <set> 14 #include <set>
15 15
16 #include "webrtc/base/onetimeevent.h"
16 #include "webrtc/base/scoped_ptr.h" 17 #include "webrtc/base/scoped_ptr.h"
17 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" 18 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
18 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 19 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
19 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" 20 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
20 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" 21 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
21 #include "webrtc/typedefs.h" 22 #include "webrtc/typedefs.h"
22 23
23 namespace webrtc { 24 namespace webrtc {
24 25
25 class CriticalSectionWrapper; 26 class CriticalSectionWrapper;
(...skipping 85 matching lines...) Expand 10 before | Expand all | Expand 10 after
111 int8_t cng_fb_payload_type_; 112 int8_t cng_fb_payload_type_;
112 int8_t cng_payload_type_; 113 int8_t cng_payload_type_;
113 114
114 // G722 is special since it use the wrong number of RTP samples in timestamp 115 // G722 is special since it use the wrong number of RTP samples in timestamp
115 // VS. number of samples in the frame 116 // VS. number of samples in the frame
116 int8_t g722_payload_type_; 117 int8_t g722_payload_type_;
117 bool last_received_g722_; 118 bool last_received_g722_;
118 119
119 uint8_t num_energy_; 120 uint8_t num_energy_;
120 uint8_t current_remote_energy_[kRtpCsrcSize]; 121 uint8_t current_remote_energy_[kRtpCsrcSize];
122
123 ThreadUnsafeOneTimeEvent first_packet_received_;
121 }; 124 };
122 } // namespace webrtc 125 } // namespace webrtc
123 126
124 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ 127 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
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