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117 lines
3.4 KiB
C++
117 lines
3.4 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
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#include <list>
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#include <string> // size_t
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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struct DtmfEvent {
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uint32_t timestamp;
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int event_no;
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int volume;
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int duration;
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bool end_bit;
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// Constructors
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DtmfEvent()
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: timestamp(0),
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event_no(0),
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volume(0),
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duration(0),
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end_bit(false) {
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}
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DtmfEvent(uint32_t ts, int ev, int vol, int dur, bool end)
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: timestamp(ts),
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event_no(ev),
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volume(vol),
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duration(dur),
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end_bit(end) {
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}
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};
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// This is the buffer holding DTMF events while waiting for them to be played.
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class DtmfBuffer {
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public:
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enum BufferReturnCodes {
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kOK = 0,
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kInvalidPointer,
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kPayloadTooShort,
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kInvalidEventParameters,
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kInvalidSampleRate
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};
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// Set up the buffer for use at sample rate |fs_hz|.
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explicit DtmfBuffer(int fs_hz) {
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SetSampleRate(fs_hz);
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}
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virtual ~DtmfBuffer() {}
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// Flushes the buffer.
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virtual void Flush() { buffer_.clear(); }
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// Static method to parse 4 bytes from |payload| as a DTMF event (RFC 4733)
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// and write the parsed information into the struct |event|. Input variable
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// |rtp_timestamp| is simply copied into the struct.
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static int ParseEvent(uint32_t rtp_timestamp,
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const uint8_t* payload,
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int payload_length_bytes,
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DtmfEvent* event);
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// Inserts |event| into the buffer. The method looks for a matching event and
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// merges the two if a match is found.
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virtual int InsertEvent(const DtmfEvent& event);
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// Checks if a DTMF event should be played at time |current_timestamp|. If so,
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// the method returns true; otherwise false. The parameters of the event to
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// play will be written to |event|.
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virtual bool GetEvent(uint32_t current_timestamp, DtmfEvent* event);
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// Number of events in the buffer.
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virtual size_t Length() const { return buffer_.size(); }
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virtual bool Empty() const { return buffer_.empty(); }
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// Set a new sample rate.
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virtual int SetSampleRate(int fs_hz);
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private:
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typedef std::list<DtmfEvent> DtmfList;
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int max_extrapolation_samples_;
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int frame_len_samples_; // TODO(hlundin): Remove this later.
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// Compares two events and returns true if they are the same.
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static bool SameEvent(const DtmfEvent& a, const DtmfEvent& b);
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// Merges |event| to the event pointed out by |it|. The method checks that
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// the two events are the same (using the SameEvent method), and merges them
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// if that was the case, returning true. If the events are not the same, false
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// is returned.
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bool MergeEvents(DtmfList::iterator it, const DtmfEvent& event);
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// Method used by the sort algorithm to rank events in the buffer.
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static bool CompareEvents(const DtmfEvent& a, const DtmfEvent& b);
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DtmfList buffer_;
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DISALLOW_COPY_AND_ASSIGN(DtmfBuffer);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
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