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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/constructormagic.h" |
| 16 #include "webrtc/base/criticalsection.h" |
| 17 #include "webrtc/base/onetimeevent.h" |
15 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h" | 18 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h" |
16 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" | 19 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" |
17 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" | 20 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" |
18 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 21 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
19 #include "webrtc/rtc_base/constructormagic.h" | |
20 #include "webrtc/rtc_base/criticalsection.h" | |
21 #include "webrtc/rtc_base/onetimeevent.h" | |
22 #include "webrtc/typedefs.h" | 22 #include "webrtc/typedefs.h" |
23 | 23 |
24 namespace webrtc { | 24 namespace webrtc { |
25 | 25 |
26 class RTPSenderAudio { | 26 class RTPSenderAudio { |
27 public: | 27 public: |
28 RTPSenderAudio(Clock* clock, RTPSender* rtp_sender); | 28 RTPSenderAudio(Clock* clock, RTPSender* rtp_sender); |
29 ~RTPSenderAudio(); | 29 ~RTPSenderAudio(); |
30 | 30 |
31 int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], | 31 int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
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89 // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/) | 89 // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/) |
90 uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_) = 0; | 90 uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_) = 0; |
91 OneTimeEvent first_packet_sent_; | 91 OneTimeEvent first_packet_sent_; |
92 | 92 |
93 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSenderAudio); | 93 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSenderAudio); |
94 }; | 94 }; |
95 | 95 |
96 } // namespace webrtc | 96 } // namespace webrtc |
97 | 97 |
98 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ | 98 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ |
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