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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_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_SENDER_AUDIO_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
13 13
14 #include "webrtc/common_types.h" 14 #include "webrtc/common_types.h"
15 #include "webrtc/base/onetimeevent.h"
15 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h" 16 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
16 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" 17 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
17 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" 18 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
18 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" 19 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
19 #include "webrtc/typedefs.h" 20 #include "webrtc/typedefs.h"
20 21
21 namespace webrtc { 22 namespace webrtc {
22 class RTPSenderAudio : public DTMFqueue { 23 class RTPSenderAudio : public DTMFqueue {
23 public: 24 public:
24 RTPSenderAudio(Clock* clock, RTPSender* rtpSender); 25 RTPSenderAudio(Clock* clock, RTPSender* rtpSender);
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93 bool _inbandVADactive GUARDED_BY(_sendAudioCritsect); 94 bool _inbandVADactive GUARDED_BY(_sendAudioCritsect);
94 int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect); 95 int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect);
95 int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect); 96 int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect);
96 int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect); 97 int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect);
97 int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect); 98 int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
98 int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect); 99 int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
99 100
100 // Audio level indication 101 // Audio level indication
101 // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/) 102 // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
102 uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect); 103 uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
104 OneTimeEvent first_packet_sent_;
103 }; 105 };
104 } // namespace webrtc 106 } // namespace webrtc
105 107
106 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ 108 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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