/*
 *  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 "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/typedefs.h"

namespace webrtc {

class CriticalSectionWrapper;

// Handles audio RTP packets. This class is thread-safe.
class RTPReceiverAudio : public RTPReceiverStrategy,
                         public TelephoneEventHandler {
 public:
  RTPReceiverAudio(RtpData* data_callback,
                   RtpAudioFeedback* incoming_messages_callback);
  virtual ~RTPReceiverAudio() {}

  // The following three methods implement the TelephoneEventHandler interface.
  // Forward DTMFs to decoder for playout.
  void SetTelephoneEventForwardToDecoder(bool forward_to_decoder);

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

  // Is TelephoneEvent configured with payload type payload_type
  bool TelephoneEventPayloadType(const int8_t payload_type) const;

  TelephoneEventHandler* GetTelephoneEventHandler() {
    return this;
  }

  // Returns true if CNG is configured with payload type payload_type. If so,
  // the frequency and cng_payload_type_has_changed are filled in.
  bool CNGPayloadType(const int8_t payload_type,
                      uint32_t* frequency,
                      bool* cng_payload_type_has_changed);

  int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
                         const PayloadUnion& specific_payload,
                         bool is_red,
                         const uint8_t* packet,
                         size_t payload_length,
                         int64_t timestamp_ms,
                         bool is_first_packet) override;

  int GetPayloadTypeFrequency() const override;

  RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;

  bool ShouldReportCsrcChanges(uint8_t payload_type) const override;

  int32_t OnNewPayloadTypeCreated(
      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
      int8_t payload_type,
      uint32_t frequency) override;

  int32_t InvokeOnInitializeDecoder(
      RtpFeedback* callback,
      int8_t payload_type,
      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
      const PayloadUnion& specific_payload) const override;

  // We do not allow codecs to have multiple payload types for audio, so we
  // need to override the default behavior (which is to do nothing).
  void PossiblyRemoveExistingPayloadType(
      RtpUtility::PayloadTypeMap* payload_type_map,
      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
      size_t payload_name_length,
      uint32_t frequency,
      uint8_t channels,
      uint32_t rate) const;

  // We need to look out for special payload types here and sometimes reset
  // statistics. In addition we sometimes need to tweak the frequency.
  void CheckPayloadChanged(int8_t payload_type,
                           PayloadUnion* specific_payload,
                           bool* should_discard_changes) override;

  int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;

 private:

  int32_t ParseAudioCodecSpecific(
      WebRtcRTPHeader* rtp_header,
      const uint8_t* payload_data,
      size_t payload_length,
      const AudioPayload& audio_specific,
      bool is_red);

  uint32_t last_received_frequency_;

  bool telephone_event_forward_to_decoder_;
  int8_t telephone_event_payload_type_;
  std::set<uint8_t> telephone_event_reported_;

  int8_t cng_nb_payload_type_;
  int8_t cng_wb_payload_type_;
  int8_t cng_swb_payload_type_;
  int8_t cng_fb_payload_type_;
  int8_t cng_payload_type_;

  // G722 is special since it use the wrong number of RTP samples in timestamp
  // VS. number of samples in the frame
  int8_t g722_payload_type_;
  bool last_received_g722_;

  uint8_t num_energy_;
  uint8_t current_remote_energy_[kRtpCsrcSize];

  RtpAudioFeedback* cb_audio_feedback_;
};
}  // namespace webrtc

#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_