1 /* 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 * 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 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ 12 #define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ 13 14 #include <stddef.h> 15 #include <stdint.h> 16 17 #include <memory> 18 19 #include "absl/strings/string_view.h" 20 #include "modules/audio_coding/include/audio_coding_module_typedefs.h" 21 #include "modules/rtp_rtcp/source/absolute_capture_time_sender.h" 22 #include "modules/rtp_rtcp/source/dtmf_queue.h" 23 #include "modules/rtp_rtcp/source/rtp_sender.h" 24 #include "rtc_base/constructor_magic.h" 25 #include "rtc_base/one_time_event.h" 26 #include "rtc_base/synchronization/mutex.h" 27 #include "rtc_base/thread_annotations.h" 28 #include "system_wrappers/include/clock.h" 29 30 namespace webrtc { 31 32 class RTPSenderAudio { 33 public: 34 RTPSenderAudio(Clock* clock, RTPSender* rtp_sender); 35 ~RTPSenderAudio(); 36 37 int32_t RegisterAudioPayload(absl::string_view payload_name, 38 int8_t payload_type, 39 uint32_t frequency, 40 size_t channels, 41 uint32_t rate); 42 43 bool SendAudio(AudioFrameType frame_type, 44 int8_t payload_type, 45 uint32_t rtp_timestamp, 46 const uint8_t* payload_data, 47 size_t payload_size); 48 49 bool SendAudio(AudioFrameType frame_type, 50 int8_t payload_type, 51 uint32_t rtp_timestamp, 52 const uint8_t* payload_data, 53 size_t payload_size, 54 int64_t absolute_capture_timestamp_ms); 55 56 // Store the audio level in dBov for 57 // header-extension-for-audio-level-indication. 58 // Valid range is [0,100]. Actual value is negative. 59 int32_t SetAudioLevel(uint8_t level_dbov); 60 61 // Send a DTMF tone using RFC 2833 (4733) 62 int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level); 63 64 protected: 65 bool SendTelephoneEventPacket( 66 bool ended, 67 uint32_t dtmf_timestamp, 68 uint16_t duration, 69 bool marker_bit); // set on first packet in talk burst 70 71 bool MarkerBit(AudioFrameType frame_type, int8_t payload_type); 72 73 private: 74 Clock* const clock_ = nullptr; 75 RTPSender* const rtp_sender_ = nullptr; 76 77 Mutex send_audio_mutex_; 78 79 // DTMF. 80 bool dtmf_event_is_on_ = false; 81 bool dtmf_event_first_packet_sent_ = false; 82 int8_t dtmf_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1; 83 uint32_t dtmf_payload_freq_ RTC_GUARDED_BY(send_audio_mutex_) = 8000; 84 uint32_t dtmf_timestamp_ = 0; 85 uint32_t dtmf_length_samples_ = 0; 86 int64_t dtmf_time_last_sent_ = 0; 87 uint32_t dtmf_timestamp_last_sent_ = 0; 88 DtmfQueue::Event dtmf_current_event_; 89 DtmfQueue dtmf_queue_; 90 91 // VAD detection, used for marker bit. 92 bool inband_vad_active_ RTC_GUARDED_BY(send_audio_mutex_) = false; 93 int8_t cngnb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1; 94 int8_t cngwb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1; 95 int8_t cngswb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1; 96 int8_t cngfb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1; 97 int8_t last_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1; 98 99 // Audio level indication. 100 // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/) 101 uint8_t audio_level_dbov_ RTC_GUARDED_BY(send_audio_mutex_) = 0; 102 OneTimeEvent first_packet_sent_; 103 104 absl::optional<uint32_t> encoder_rtp_timestamp_frequency_ 105 RTC_GUARDED_BY(send_audio_mutex_); 106 107 AbsoluteCaptureTimeSender absolute_capture_time_sender_; 108 109 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSenderAudio); 110 }; 111 112 } // namespace webrtc 113 114 #endif // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ 115