webrtc/modules/rtp_rtcp/source/rtp_sender_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_SENDER_AUDIO_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_

#include "webrtc/common_types.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/onetimeevent.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* rtp_sender);
  virtual ~RTPSenderAudio();

  int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                               int8_t payload_type,
                               uint32_t frequency,
                               size_t channels,
                               uint32_t rate,
                               RtpUtility::Payload** payload);

  bool SendAudio(FrameType frame_type,
                 int8_t payload_type,
                 uint32_t capture_timestamp,
                 const uint8_t* payload_data,
                 size_t payload_size,
                 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(uint16_t packet_size_samples);

  // 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(uint8_t level_dbov);

  // Send a DTMF tone using RFC 2833 (4733)
  int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);

  int AudioFrequency() const;

 protected:
  bool SendTelephoneEventPacket(
      bool ended,
      int8_t dtmf_payload_type,
      uint32_t dtmf_timestamp,
      uint16_t duration,
      bool marker_bit);  // set on first packet in talk burst

  bool MarkerBit(FrameType frame_type, int8_t payload_type);

 private:
  Clock* const clock_;
  RTPSender* const rtp_sender_;

  rtc::CriticalSection send_audio_critsect_;

  uint16_t packet_size_samples_ GUARDED_BY(send_audio_critsect_);

  // DTMF.
  bool dtmf_event_is_on_;
  bool dtmf_event_first_packet_sent_;
  int8_t dtmf_payload_type_ GUARDED_BY(send_audio_critsect_);
  uint32_t dtmf_timestamp_;
  uint8_t dtmf_key_;
  uint32_t dtmf_length_samples_;
  uint8_t dtmf_level_;
  int64_t dtmf_time_last_sent_;
  uint32_t dtmf_timestamp_last_sent_;

  // VAD detection, used for marker bit.
  bool inband_vad_active_ GUARDED_BY(send_audio_critsect_);
  int8_t cngnb_payload_type_ GUARDED_BY(send_audio_critsect_);
  int8_t cngwb_payload_type_ GUARDED_BY(send_audio_critsect_);
  int8_t cngswb_payload_type_ GUARDED_BY(send_audio_critsect_);
  int8_t cngfb_payload_type_ GUARDED_BY(send_audio_critsect_);
  int8_t last_payload_type_ GUARDED_BY(send_audio_critsect_);

  // Audio level indication.
  // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
  uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_);
  OneTimeEvent first_packet_sent_;
};

}  // namespace webrtc

#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_