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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

Issue 2365873002: Make RtpSenderAudio not inherit from DtmfQueue (Closed)
Patch Set: not inline Created 4 years, 2 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/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
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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