97 lines
3.8 KiB
C
97 lines
3.8 KiB
C
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/*
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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_RECEIVER_AUDIO_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
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#include "rtp_rtcp_defines.h"
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#include "rtp_utility.h"
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#include "typedefs.h"
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#include "map_wrapper.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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class RTPReceiverAudio
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{
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public:
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RTPReceiverAudio(const WebRtc_Word32 id);
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virtual ~RTPReceiverAudio();
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virtual void ChangeUniqueId(const WebRtc_Word32 id);
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WebRtc_Word32 Init();
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WebRtc_Word32 RegisterIncomingAudioCallback(RtpAudioFeedback* incomingMessagesCallback);
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ModuleRTPUtility::Payload* RegisterReceiveAudioPayload(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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WebRtc_UWord32 AudioFrequency() const;
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// Outband TelephoneEvent (DTMF) detection
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WebRtc_Word32 SetTelephoneEventStatus(const bool enable,
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const bool forwardToDecoder,
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const bool detectEndOfTone);
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// Is outband DTMF(AVT) turned on/off?
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bool TelephoneEvent() const ;
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// Is forwarding of outband telephone events turned on/off?
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bool TelephoneEventForwardToDecoder() const ;
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// Is TelephoneEvent configured with payload type payloadType
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bool TelephoneEventPayloadType(const WebRtc_Word8 payloadType) const;
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// Is CNG configured with payload type payloadType
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bool CNGPayloadType(const WebRtc_Word8 payloadType, WebRtc_UWord32& frequency);
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WebRtc_Word32 ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadLength,
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const ModuleRTPUtility::AudioPayload& audioSpecific,
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const bool isRED);
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virtual WebRtc_Word32 ResetStatistics() = 0;
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protected:
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virtual WebRtc_Word32 CallbackOfReceivedPayloadData(const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadSize,
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const WebRtcRTPHeader* rtpHeader) = 0;
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private:
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WebRtc_Word32 _id;
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WebRtc_UWord32 _lastReceivedFrequency;
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bool _telephoneEvent;
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bool _telephoneEventForwardToDecoder;
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bool _telephoneEventDetectEndOfTone;
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WebRtc_Word8 _telephoneEventPayloadType;
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MapWrapper _telephoneEventReported;
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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 _cngPayloadType;
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// G722 is special since it use the wrong number of RTP samples in timestamp VS. number of samples in the frame
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WebRtc_Word8 _G722PayloadType;
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bool _lastReceivedG722;
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CriticalSectionWrapper& _criticalSectionFeedback;
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RtpAudioFeedback* _cbAudioFeedback;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
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