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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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