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/*
* 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_SENDER_AUDIO_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class RTPSenderAudio: public DTMFqueue
{
public:
RTPSenderAudio(Clock* clock,
RTPSender* rtpSender,
RtpAudioFeedback* audio_feedback);
virtual ~RTPSenderAudio();
int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int8_t payloadType,
const uint32_t frequency,
const uint8_t channels,
const uint32_t rate,
RtpUtility::Payload*& payload);
int32_t SendAudio(const FrameType frameType,
const int8_t payloadType,
const uint32_t captureTimeStamp,
const uint8_t* payloadData,
const size_t payloadSize,
const RTPFragmentationHeader* fragmentation);
// set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
int32_t SetAudioPacketSize(const uint16_t packetSizeSamples);
// Store the audio level in dBov for header-extension-for-audio-level-indication.
// Valid range is [0,100]. Actual value is negative.
int32_t SetAudioLevel(const uint8_t level_dBov);
// Send a DTMF tone using RFC 2833 (4733)
int32_t SendTelephoneEvent(const uint8_t key,
const uint16_t time_ms,
const uint8_t level);
int AudioFrequency() const;
// Set payload type for Redundant Audio Data RFC 2198
int32_t SetRED(const int8_t payloadType);
// Get payload type for Redundant Audio Data RFC 2198
int32_t RED(int8_t& payloadType) const;
protected:
int32_t SendTelephoneEventPacket(bool ended,
int8_t dtmf_payload_type,
uint32_t dtmfTimeStamp,
uint16_t duration,
bool markerBit); // set on first packet in talk burst
bool MarkerBit(const FrameType frameType,
const int8_t payloadType);
private:
Clock* const _clock;
RTPSender* const _rtpSender;
RtpAudioFeedback* const _audioFeedback;
rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
// DTMF
bool _dtmfEventIsOn;
bool _dtmfEventFirstPacketSent;
int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect);
uint32_t _dtmfTimestamp;
uint8_t _dtmfKey;
uint32_t _dtmfLengthSamples;
uint8_t _dtmfLevel;
int64_t _dtmfTimeLastSent;
uint32_t _dtmfTimestampLastSent;
int8_t _REDPayloadType GUARDED_BY(_sendAudioCritsect);
// VAD detection, used for markerbit
bool _inbandVADactive GUARDED_BY(_sendAudioCritsect);
int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect);
int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect);
int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect);
int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
// Audio level indication
// (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
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