src/modules/rtp_rtcp/source/rtp_receiver_audio.h

/*
 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_

#include <set>

#include "rtp_rtcp_defines.h"
#include "rtp_utility.h"

#include "typedefs.h"

namespace webrtc {
class CriticalSectionWrapper;

class RTPReceiverAudio
{
public:
    RTPReceiverAudio(const WebRtc_Word32 id);
    virtual ~RTPReceiverAudio();

    virtual void ChangeUniqueId(const WebRtc_Word32 id);

    WebRtc_Word32 Init();

    WebRtc_Word32 RegisterIncomingAudioCallback(RtpAudioFeedback* incomingMessagesCallback);

    ModuleRTPUtility::Payload* RegisterReceiveAudioPayload(
        const char payloadName[RTP_PAYLOAD_NAME_SIZE],
        const WebRtc_Word8 payloadType,
        const WebRtc_UWord32 frequency,
        const WebRtc_UWord8 channels,
        const WebRtc_UWord32 rate);

    WebRtc_UWord32 AudioFrequency() const;

    // Outband TelephoneEvent (DTMF) detection
    WebRtc_Word32 SetTelephoneEventStatus(const bool enable,
                                        const bool forwardToDecoder,
                                        const bool detectEndOfTone);

    // Is outband DTMF(AVT) turned on/off?
    bool TelephoneEvent() const ;

    // Is forwarding of outband telephone events turned on/off?
    bool TelephoneEventForwardToDecoder() const ;

    // Is TelephoneEvent configured with payload type payloadType
    bool TelephoneEventPayloadType(const WebRtc_Word8 payloadType) const;

    // Is CNG configured with payload type payloadType
    bool CNGPayloadType(const WebRtc_Word8 payloadType, WebRtc_UWord32& frequency);

    WebRtc_Word32 ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
                                        const WebRtc_UWord8* payloadData,
                                        const WebRtc_UWord16 payloadLength,
                                        const ModuleRTPUtility::AudioPayload& audioSpecific,
                                        const bool isRED);

    virtual WebRtc_Word32 ResetStatistics() = 0;

protected:
    virtual WebRtc_Word32 CallbackOfReceivedPayloadData(const WebRtc_UWord8* payloadData,
                                                      const WebRtc_UWord16 payloadSize,
                                                      const WebRtcRTPHeader* rtpHeader) = 0;
private:
    WebRtc_Word32             _id;

    WebRtc_UWord32            _lastReceivedFrequency;

    bool                    _telephoneEvent;
    bool                    _telephoneEventForwardToDecoder;
    bool                    _telephoneEventDetectEndOfTone;
    WebRtc_Word8            _telephoneEventPayloadType;
    std::set<WebRtc_UWord8> _telephoneEventReported;

    WebRtc_Word8              _cngNBPayloadType;
    WebRtc_Word8              _cngWBPayloadType;
    WebRtc_Word8              _cngSWBPayloadType;
    WebRtc_Word8                _cngPayloadType;

    // G722 is special since it use the wrong number of RTP samples in timestamp VS. number of samples in the frame
    WebRtc_Word8              _G722PayloadType;
    bool                    _lastReceivedG722;

    CriticalSectionWrapper* _criticalSectionFeedback;
    RtpAudioFeedback*   _cbAudioFeedback;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_