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Diffstat (limited to 'modules/audio_coding/neteq/dtmf_buffer.h')
-rw-r--r-- | modules/audio_coding/neteq/dtmf_buffer.h | 116 |
1 files changed, 116 insertions, 0 deletions
diff --git a/modules/audio_coding/neteq/dtmf_buffer.h b/modules/audio_coding/neteq/dtmf_buffer.h new file mode 100644 index 00000000..5dd31c2d --- /dev/null +++ b/modules/audio_coding/neteq/dtmf_buffer.h @@ -0,0 +1,116 @@ +/* + * 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_ |