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/*
* 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_
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