xref: /aosp_15_r20/external/webrtc/modules/audio_coding/neteq/dtmf_buffer.h (revision d9f758449e529ab9291ac668be2861e7a55c2422)
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 MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
12 #define MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
13 
14 #include <stddef.h>
15 #include <stdint.h>
16 
17 #include <list>
18 
19 namespace webrtc {
20 
21 struct DtmfEvent {
22   uint32_t timestamp;
23   int event_no;
24   int volume;
25   int duration;
26   bool end_bit;
27 
28   // Constructors
DtmfEventDtmfEvent29   DtmfEvent()
30       : timestamp(0), event_no(0), volume(0), duration(0), end_bit(false) {}
DtmfEventDtmfEvent31   DtmfEvent(uint32_t ts, int ev, int vol, int dur, bool end)
32       : timestamp(ts), event_no(ev), volume(vol), duration(dur), end_bit(end) {}
33 };
34 
35 // This is the buffer holding DTMF events while waiting for them to be played.
36 class DtmfBuffer {
37  public:
38   enum BufferReturnCodes {
39     kOK = 0,
40     kInvalidPointer,
41     kPayloadTooShort,
42     kInvalidEventParameters,
43     kInvalidSampleRate
44   };
45 
46   // Set up the buffer for use at sample rate `fs_hz`.
47   explicit DtmfBuffer(int fs_hz);
48 
49   virtual ~DtmfBuffer();
50 
51   DtmfBuffer(const DtmfBuffer&) = delete;
52   DtmfBuffer& operator=(const DtmfBuffer&) = delete;
53 
54   // Flushes the buffer.
55   virtual void Flush();
56 
57   // Static method to parse 4 bytes from `payload` as a DTMF event (RFC 4733)
58   // and write the parsed information into the struct `event`. Input variable
59   // `rtp_timestamp` is simply copied into the struct.
60   static int ParseEvent(uint32_t rtp_timestamp,
61                         const uint8_t* payload,
62                         size_t payload_length_bytes,
63                         DtmfEvent* event);
64 
65   // Inserts `event` into the buffer. The method looks for a matching event and
66   // merges the two if a match is found.
67   virtual int InsertEvent(const DtmfEvent& event);
68 
69   // Checks if a DTMF event should be played at time `current_timestamp`. If so,
70   // the method returns true; otherwise false. The parameters of the event to
71   // play will be written to `event`.
72   virtual bool GetEvent(uint32_t current_timestamp, DtmfEvent* event);
73 
74   // Number of events in the buffer.
75   virtual size_t Length() const;
76 
77   virtual bool Empty() const;
78 
79   // Set a new sample rate.
80   virtual int SetSampleRate(int fs_hz);
81 
82  private:
83   typedef std::list<DtmfEvent> DtmfList;
84 
85   int max_extrapolation_samples_;
86   int frame_len_samples_;  // TODO(hlundin): Remove this later.
87 
88   // Compares two events and returns true if they are the same.
89   static bool SameEvent(const DtmfEvent& a, const DtmfEvent& b);
90 
91   // Merges `event` to the event pointed out by `it`. The method checks that
92   // the two events are the same (using the SameEvent method), and merges them
93   // if that was the case, returning true. If the events are not the same, false
94   // is returned.
95   bool MergeEvents(DtmfList::iterator it, const DtmfEvent& event);
96 
97   // Method used by the sort algorithm to rank events in the buffer.
98   static bool CompareEvents(const DtmfEvent& a, const DtmfEvent& b);
99 
100   DtmfList buffer_;
101 };
102 
103 }  // namespace webrtc
104 #endif  // MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
105