/*
 *  Copyright (c) 2013 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_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_

#include <map>
#include <set>

#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/rtc_base/criticalsection.h"

namespace webrtc {

struct CodecInst;
class VideoCodec;

class RTPPayloadRegistry {
 public:
  RTPPayloadRegistry();
  ~RTPPayloadRegistry();

  // TODO(magjed): Split RTPPayloadRegistry into separate Audio and Video class
  // and simplify the code. http://crbug/webrtc/6743.

  // Replace all audio receive payload types with the given map.
  void SetAudioReceivePayloads(std::map<int, SdpAudioFormat> codecs);

  int32_t RegisterReceivePayload(const CodecInst& audio_codec,
                                 bool* created_new_payload_type);
  int32_t RegisterReceivePayload(const VideoCodec& video_codec);

  int32_t DeRegisterReceivePayload(int8_t payload_type);

  int32_t ReceivePayloadType(const CodecInst& audio_codec,
                             int8_t* payload_type) const;
  int32_t ReceivePayloadType(const VideoCodec& video_codec,
                             int8_t* payload_type) const;

  bool RtxEnabled() const;

  void SetRtxSsrc(uint32_t ssrc);

  bool GetRtxSsrc(uint32_t* ssrc) const;

  void SetRtxPayloadType(int payload_type, int associated_payload_type);

  bool IsRtx(const RTPHeader& header) const;

  bool RestoreOriginalPacket(uint8_t* restored_packet,
                             const uint8_t* packet,
                             size_t* packet_length,
                             uint32_t original_ssrc,
                             const RTPHeader& header);

  bool IsRed(const RTPHeader& header) const;

  // Returns true if the media of this RTP packet is encapsulated within an
  // extra header, such as RTX or RED.
  bool IsEncapsulated(const RTPHeader& header) const;

  bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const;

  int GetPayloadTypeFrequency(uint8_t payload_type) const;

  const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const;

  void ResetLastReceivedPayloadTypes() {
    rtc::CritScope cs(&crit_sect_);
    last_received_payload_type_ = -1;
    last_received_media_payload_type_ = -1;
  }

  // This sets the payload type of the packets being received from the network
  // on the media SSRC. For instance if packets are encapsulated with RED, this
  // payload type will be the RED payload type.
  void SetIncomingPayloadType(const RTPHeader& header);

  // Returns true if the new media payload type has not changed.
  bool ReportMediaPayloadType(uint8_t media_payload_type);

  int8_t red_payload_type() const { return GetPayloadTypeWithName("red"); }
  int8_t ulpfec_payload_type() const {
    return GetPayloadTypeWithName("ulpfec");
  }
  int8_t last_received_payload_type() const {
    rtc::CritScope cs(&crit_sect_);
    return last_received_payload_type_;
  }
  void set_last_received_payload_type(int8_t last_received_payload_type) {
    rtc::CritScope cs(&crit_sect_);
    last_received_payload_type_ = last_received_payload_type;
  }

  int8_t last_received_media_payload_type() const {
    rtc::CritScope cs(&crit_sect_);
    return last_received_media_payload_type_;
  }

 private:
  // Prunes the payload type map of the specific payload type, if it exists.
  void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
      const CodecInst& audio_codec);

  bool IsRtxInternal(const RTPHeader& header) const;
  // Returns the payload type for the payload with name |payload_name|, or -1 if
  // no such payload is registered.
  int8_t GetPayloadTypeWithName(const char* payload_name) const;

  rtc::CriticalSection crit_sect_;
  std::map<int, RtpUtility::Payload> payload_type_map_;
  int8_t incoming_payload_type_;
  int8_t last_received_payload_type_;
  int8_t last_received_media_payload_type_;
  bool rtx_;
  // Mapping rtx_payload_type_map_[rtx] = associated.
  std::map<int, int> rtx_payload_type_map_;
  uint32_t ssrc_rtx_;
  // Only warn once per payload type, if an RTX packet is received but
  // no associated payload type found in |rtx_payload_type_map_|.
  std::set<int> payload_types_with_suppressed_warnings_
      RTC_GUARDED_BY(crit_sect_);

  // As a first step in splitting this class up in separate cases for audio and
  // video, DCHECK that no instance is used for both audio and video.
#if RTC_DCHECK_IS_ON
  bool used_for_audio_ RTC_GUARDED_BY(crit_sect_) = false;
  bool used_for_video_ RTC_GUARDED_BY(crit_sect_) = false;
#endif
};

}  // namespace webrtc

#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_