/* * 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 #include // 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 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_