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