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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" |
15 #include "webrtc/base/criticalsection.h" | 16 #include "webrtc/base/criticalsection.h" |
16 #include "webrtc/base/onetimeevent.h" | 17 #include "webrtc/base/onetimeevent.h" |
17 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h" | 18 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h" |
18 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" | 19 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" |
19 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" | 20 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.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 RTPSenderAudio : public DTMFqueue { | 26 class RTPSenderAudio { |
26 public: | 27 public: |
27 RTPSenderAudio(Clock* clock, RTPSender* rtp_sender); | 28 RTPSenderAudio(Clock* clock, RTPSender* rtp_sender); |
28 virtual ~RTPSenderAudio(); | 29 ~RTPSenderAudio(); |
29 | 30 |
30 int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], | 31 int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
31 int8_t payload_type, | 32 int8_t payload_type, |
32 uint32_t frequency, | 33 uint32_t frequency, |
33 size_t channels, | 34 size_t channels, |
34 uint32_t rate, | 35 uint32_t rate, |
35 RtpUtility::Payload** payload); | 36 RtpUtility::Payload** payload); |
36 | 37 |
37 bool SendAudio(FrameType frame_type, | 38 bool SendAudio(FrameType frame_type, |
38 int8_t payload_type, | 39 int8_t payload_type, |
(...skipping 37 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
76 // DTMF. | 77 // DTMF. |
77 bool dtmf_event_is_on_; | 78 bool dtmf_event_is_on_; |
78 bool dtmf_event_first_packet_sent_; | 79 bool dtmf_event_first_packet_sent_; |
79 int8_t dtmf_payload_type_ GUARDED_BY(send_audio_critsect_); | 80 int8_t dtmf_payload_type_ GUARDED_BY(send_audio_critsect_); |
80 uint32_t dtmf_timestamp_; | 81 uint32_t dtmf_timestamp_; |
81 uint8_t dtmf_key_; | 82 uint8_t dtmf_key_; |
82 uint32_t dtmf_length_samples_; | 83 uint32_t dtmf_length_samples_; |
83 uint8_t dtmf_level_; | 84 uint8_t dtmf_level_; |
84 int64_t dtmf_time_last_sent_; | 85 int64_t dtmf_time_last_sent_; |
85 uint32_t dtmf_timestamp_last_sent_; | 86 uint32_t dtmf_timestamp_last_sent_; |
| 87 DTMFqueue dtmf_queue_; |
86 | 88 |
87 // VAD detection, used for marker bit. | 89 // VAD detection, used for marker bit. |
88 bool inband_vad_active_ GUARDED_BY(send_audio_critsect_); | 90 bool inband_vad_active_ GUARDED_BY(send_audio_critsect_); |
89 int8_t cngnb_payload_type_ GUARDED_BY(send_audio_critsect_); | 91 int8_t cngnb_payload_type_ GUARDED_BY(send_audio_critsect_); |
90 int8_t cngwb_payload_type_ GUARDED_BY(send_audio_critsect_); | 92 int8_t cngwb_payload_type_ GUARDED_BY(send_audio_critsect_); |
91 int8_t cngswb_payload_type_ GUARDED_BY(send_audio_critsect_); | 93 int8_t cngswb_payload_type_ GUARDED_BY(send_audio_critsect_); |
92 int8_t cngfb_payload_type_ GUARDED_BY(send_audio_critsect_); | 94 int8_t cngfb_payload_type_ GUARDED_BY(send_audio_critsect_); |
93 int8_t last_payload_type_ GUARDED_BY(send_audio_critsect_); | 95 int8_t last_payload_type_ GUARDED_BY(send_audio_critsect_); |
94 | 96 |
95 // Audio level indication. | 97 // Audio level indication. |
96 // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/) | 98 // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/) |
97 uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_); | 99 uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_); |
98 OneTimeEvent first_packet_sent_; | 100 OneTimeEvent first_packet_sent_; |
| 101 |
| 102 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSenderAudio); |
99 }; | 103 }; |
100 | 104 |
101 } // namespace webrtc | 105 } // namespace webrtc |
102 | 106 |
103 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ | 107 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ |
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