/*
 *  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_