session-android/jni/webrtc/modules/audio_coding/neteq/dtmf_buffer.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_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_

#include <list>
#include <string>  // size_t

#include "webrtc/base/constructormagic.h"
#include "webrtc/typedefs.h"

namespace webrtc {

struct DtmfEvent {
  uint32_t timestamp;
  int event_no;
  int volume;
  int duration;
  bool end_bit;

  // Constructors
  DtmfEvent()
      : timestamp(0),
        event_no(0),
        volume(0),
        duration(0),
        end_bit(false) {
  }
  DtmfEvent(uint32_t ts, int ev, int vol, int dur, bool end)
      : timestamp(ts),
        event_no(ev),
        volume(vol),
        duration(dur),
        end_bit(end) {
  }
};

// This is the buffer holding DTMF events while waiting for them to be played.
class DtmfBuffer {
 public:
  enum BufferReturnCodes {
    kOK = 0,
    kInvalidPointer,
    kPayloadTooShort,
    kInvalidEventParameters,
    kInvalidSampleRate
  };

  // Set up the buffer for use at sample rate |fs_hz|.
  explicit DtmfBuffer(int fs_hz) {
    SetSampleRate(fs_hz);
  }

  virtual ~DtmfBuffer() {}

  // Flushes the buffer.
  virtual void Flush() { buffer_.clear(); }

  // Static method to parse 4 bytes from |payload| as a DTMF event (RFC 4733)
  // and write the parsed information into the struct |event|. Input variable
  // |rtp_timestamp| is simply copied into the struct.
  static int ParseEvent(uint32_t rtp_timestamp,
                        const uint8_t* payload,
                        int payload_length_bytes,
                        DtmfEvent* event);

  // Inserts |event| into the buffer. The method looks for a matching event and
  // merges the two if a match is found.
  virtual int InsertEvent(const DtmfEvent& event);

  // Checks if a DTMF event should be played at time |current_timestamp|. If so,
  // the method returns true; otherwise false. The parameters of the event to
  // play will be written to |event|.
  virtual bool GetEvent(uint32_t current_timestamp, DtmfEvent* event);

  // Number of events in the buffer.
  virtual size_t Length() const { return buffer_.size(); }

  virtual bool Empty() const { return buffer_.empty(); }

  // Set a new sample rate.
  virtual int SetSampleRate(int fs_hz);

 private:
  typedef std::list<DtmfEvent> DtmfList;

  int max_extrapolation_samples_;
  int frame_len_samples_;  // TODO(hlundin): Remove this later.

  // Compares two events and returns true if they are the same.
  static bool SameEvent(const DtmfEvent& a, const DtmfEvent& b);

  // Merges |event| to the event pointed out by |it|. The method checks that
  // the two events are the same (using the SameEvent method), and merges them
  // if that was the case, returning true. If the events are not the same, false
  // is returned.
  bool MergeEvents(DtmfList::iterator it, const DtmfEvent& event);

  // Method used by the sort algorithm to rank events in the buffer.
  static bool CompareEvents(const DtmfEvent& a, const DtmfEvent& b);

  DtmfList buffer_;

  DISALLOW_COPY_AND_ASSIGN(DtmfBuffer);
};

}  // namespace webrtc
#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_