blob: 89cd062e91df318c5e4a4aa740a9d8637447e241 [file] [log] [blame]
* 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.
#include <set>
#include "rtp_rtcp_defines.h"
#include "rtp_utility.h"
#include "typedefs.h"
namespace webrtc {
class CriticalSectionWrapper;
class RTPReceiverAudio
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;
virtual WebRtc_Word32 CallbackOfReceivedPayloadData(const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize,
const WebRtcRTPHeader* rtpHeader) = 0;
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