132 lines
5.0 KiB
C++
132 lines
5.0 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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#include "rtp_rtcp_config.h" // misc. defines (e.g. MAX_PACKET_LENGTH)
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#include "common_types.h" // Transport
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#include "map_wrapper.h"
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#include "typedefs.h"
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#include "dtmf_queue.h"
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#include "rtp_utility.h"
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#include "rtp_sender.h"
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namespace webrtc {
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class RTPSenderAudio: public DTMFqueue
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{
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public:
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RTPSenderAudio(const WebRtc_Word32 id, RTPSenderInterface* rtpSender);
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virtual ~RTPSenderAudio();
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void ChangeUniqueId(const WebRtc_Word32 id);
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WebRtc_Word32 Init();
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WebRtc_Word32 RegisterAudioPayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 frequency,
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const WebRtc_UWord8 channels,
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const WebRtc_UWord32 rate,
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ModuleRTPUtility::Payload*& payload);
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WebRtc_Word32 SendAudio(const FrameType frameType,
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 captureTimeStamp,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord32 payloadSize,
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const RTPFragmentationHeader* fragmentation);
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// set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
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WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
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// Set status and ID for header-extension-for-audio-level-indication.
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// Valid ID range is [1,14].
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WebRtc_Word32 SetAudioLevelIndicationStatus(const bool enable,
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const WebRtc_UWord8 ID);
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// Get status and ID for header-extension-for-audio-level-indication.
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WebRtc_Word32 AudioLevelIndicationStatus(bool& enable,
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WebRtc_UWord8& ID) const;
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// Store the audio level in dBov for header-extension-for-audio-level-indication.
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// Valid range is [0,100]. Actual value is negative.
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WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
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// Send a DTMF tone using RFC 2833 (4733)
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WebRtc_Word32 SendTelephoneEvent(const WebRtc_UWord8 key,
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const WebRtc_UWord16 time_ms,
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const WebRtc_UWord8 level);
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bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
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void SetAudioFrequency(const WebRtc_UWord32 f);
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WebRtc_UWord32 AudioFrequency() const;
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// Set payload type for Redundant Audio Data RFC 2198
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WebRtc_Word32 SetRED(const WebRtc_Word8 payloadType);
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// Get payload type for Redundant Audio Data RFC 2198
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WebRtc_Word32 RED(WebRtc_Word8& payloadType) const;
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WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback);
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protected:
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WebRtc_Word32 SendTelephoneEventPacket(const bool ended,
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const WebRtc_UWord32 dtmfTimeStamp,
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const WebRtc_UWord16 duration,
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const bool markerBit); // set on first packet in talk burst
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bool MarkerBit(const FrameType frameType,
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const WebRtc_Word8 payloadType);
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private:
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WebRtc_Word32 _id;
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RTPSenderInterface* _rtpSender;
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CriticalSectionWrapper& _audioFeedbackCritsect;
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RtpAudioFeedback* _audioFeedback;
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CriticalSectionWrapper& _sendAudioCritsect;
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WebRtc_UWord32 _frequency;
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WebRtc_UWord16 _packetSizeSamples;
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// DTMF
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bool _dtmfEventIsOn;
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bool _dtmfEventFirstPacketSent;
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WebRtc_Word8 _dtmfPayloadType;
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WebRtc_UWord32 _dtmfTimestamp;
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WebRtc_UWord8 _dtmfKey;
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WebRtc_UWord32 _dtmfLengthSamples;
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WebRtc_UWord8 _dtmfLevel;
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WebRtc_UWord32 _dtmfTimeLastSent;
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WebRtc_UWord32 _dtmfTimestampLastSent;
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WebRtc_Word8 _REDPayloadType;
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// VAD detection, used for markerbit
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bool _inbandVADactive;
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WebRtc_Word8 _cngNBPayloadType;
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WebRtc_Word8 _cngWBPayloadType;
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WebRtc_Word8 _cngSWBPayloadType;
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WebRtc_Word8 _lastPayloadType;
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// Audio level indication (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
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bool _includeAudioLevelIndication;
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WebRtc_UWord8 _audioLevelIndicationID;
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WebRtc_UWord8 _audioLevel_dBov;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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