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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ 13 14 #include "webrtc/common_types.h" 15 #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_sender.h" 18 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" 19 #include "webrtc/typedefs.h" 20 21 namespace webrtc { 22 class RTPSenderAudio: public DTMFqueue 23 { 24 public: 25 RTPSenderAudio(const int32_t id, Clock* clock, 26 RTPSender* rtpSender); 27 virtual ~RTPSenderAudio(); 28 29 int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], 30 const int8_t payloadType, 31 const uint32_t frequency, 32 const uint8_t channels, 33 const uint32_t rate, 34 RtpUtility::Payload*& payload); 35 36 int32_t SendAudio(const FrameType frameType, 37 const int8_t payloadType, 38 const uint32_t captureTimeStamp, 39 const uint8_t* payloadData, 40 const uint32_t payloadSize, 41 const RTPFragmentationHeader* fragmentation); 42 43 // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG) 44 int32_t SetAudioPacketSize(const uint16_t packetSizeSamples); 45 46 // Store the audio level in dBov for header-extension-for-audio-level-indication. 47 // Valid range is [0,100]. Actual value is negative. 48 int32_t SetAudioLevel(const uint8_t level_dBov); 49 50 // Send a DTMF tone using RFC 2833 (4733) 51 int32_t SendTelephoneEvent(const uint8_t key, 52 const uint16_t time_ms, 53 const uint8_t level); 54 55 bool SendTelephoneEventActive(int8_t& telephoneEvent) const; 56 57 void SetAudioFrequency(const uint32_t f); 58 59 int AudioFrequency() const; 60 61 // Set payload type for Redundant Audio Data RFC 2198 62 int32_t SetRED(const int8_t payloadType); 63 64 // Get payload type for Redundant Audio Data RFC 2198 65 int32_t RED(int8_t& payloadType) const; 66 67 int32_t RegisterAudioCallback(RtpAudioFeedback* messagesCallback); 68 69 protected: 70 int32_t SendTelephoneEventPacket(const bool ended, 71 const uint32_t dtmfTimeStamp, 72 const uint16_t duration, 73 const bool markerBit); // set on first packet in talk burst 74 75 bool MarkerBit(const FrameType frameType, 76 const int8_t payloadType); 77 78 private: 79 int32_t _id; 80 Clock* _clock; 81 RTPSender* _rtpSender; 82 CriticalSectionWrapper* _audioFeedbackCritsect; 83 RtpAudioFeedback* _audioFeedback; 84 85 CriticalSectionWrapper* _sendAudioCritsect; 86 87 uint32_t _frequency; 88 uint16_t _packetSizeSamples; 89 90 // DTMF 91 bool _dtmfEventIsOn; 92 bool _dtmfEventFirstPacketSent; 93 int8_t _dtmfPayloadType; 94 uint32_t _dtmfTimestamp; 95 uint8_t _dtmfKey; 96 uint32_t _dtmfLengthSamples; 97 uint8_t _dtmfLevel; 98 int64_t _dtmfTimeLastSent; 99 uint32_t _dtmfTimestampLastSent; 100 101 int8_t _REDPayloadType; 102 103 // VAD detection, used for markerbit 104 bool _inbandVADactive; 105 int8_t _cngNBPayloadType; 106 int8_t _cngWBPayloadType; 107 int8_t _cngSWBPayloadType; 108 int8_t _cngFBPayloadType; 109 int8_t _lastPayloadType; 110 111 // Audio level indication (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/) 112 uint8_t _audioLevel_dBov; 113 }; 114 } // namespace webrtc 115 116 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ 117
