1 /*
2  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
12 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
13 
14 #include <list>
15 #include <string>  // size_t
16 
17 #include "webrtc/base/constructormagic.h"
18 #include "webrtc/typedefs.h"
19 
20 namespace webrtc {
21 
22 struct DtmfEvent {
23   uint32_t timestamp;
24   int event_no;
25   int volume;
26   int duration;
27   bool end_bit;
28 
29   // Constructors
DtmfEventDtmfEvent30   DtmfEvent()
31       : timestamp(0),
32         event_no(0),
33         volume(0),
34         duration(0),
35         end_bit(false) {
36   }
DtmfEventDtmfEvent37   DtmfEvent(uint32_t ts, int ev, int vol, int dur, bool end)
38       : timestamp(ts),
39         event_no(ev),
40         volume(vol),
41         duration(dur),
42         end_bit(end) {
43   }
44 };
45 
46 // This is the buffer holding DTMF events while waiting for them to be played.
47 class DtmfBuffer {
48  public:
49   enum BufferReturnCodes {
50     kOK = 0,
51     kInvalidPointer,
52     kPayloadTooShort,
53     kInvalidEventParameters,
54     kInvalidSampleRate
55   };
56 
57   // Set up the buffer for use at sample rate |fs_hz|.
58   explicit DtmfBuffer(int fs_hz);
59 
60   virtual ~DtmfBuffer();
61 
62   // Flushes the buffer.
63   virtual void Flush();
64 
65   // Static method to parse 4 bytes from |payload| as a DTMF event (RFC 4733)
66   // and write the parsed information into the struct |event|. Input variable
67   // |rtp_timestamp| is simply copied into the struct.
68   static int ParseEvent(uint32_t rtp_timestamp,
69                         const uint8_t* payload,
70                         size_t payload_length_bytes,
71                         DtmfEvent* event);
72 
73   // Inserts |event| into the buffer. The method looks for a matching event and
74   // merges the two if a match is found.
75   virtual int InsertEvent(const DtmfEvent& event);
76 
77   // Checks if a DTMF event should be played at time |current_timestamp|. If so,
78   // the method returns true; otherwise false. The parameters of the event to
79   // play will be written to |event|.
80   virtual bool GetEvent(uint32_t current_timestamp, DtmfEvent* event);
81 
82   // Number of events in the buffer.
83   virtual size_t Length() const;
84 
85   virtual bool Empty() const;
86 
87   // Set a new sample rate.
88   virtual int SetSampleRate(int fs_hz);
89 
90  private:
91   typedef std::list<DtmfEvent> DtmfList;
92 
93   int max_extrapolation_samples_;
94   int frame_len_samples_;  // TODO(hlundin): Remove this later.
95 
96   // Compares two events and returns true if they are the same.
97   static bool SameEvent(const DtmfEvent& a, const DtmfEvent& b);
98 
99   // Merges |event| to the event pointed out by |it|. The method checks that
100   // the two events are the same (using the SameEvent method), and merges them
101   // if that was the case, returning true. If the events are not the same, false
102   // is returned.
103   bool MergeEvents(DtmfList::iterator it, const DtmfEvent& event);
104 
105   // Method used by the sort algorithm to rank events in the buffer.
106   static bool CompareEvents(const DtmfEvent& a, const DtmfEvent& b);
107 
108   DtmfList buffer_;
109 
110   RTC_DISALLOW_COPY_AND_ASSIGN(DtmfBuffer);
111 };
112 
113 }  // namespace webrtc
114 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
115