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102 lines
4.2 KiB
C++
102 lines
4.2 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_SYNC_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class SyncBuffer : public AudioMultiVector {
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public:
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SyncBuffer(size_t channels, size_t length)
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: AudioMultiVector(channels, length),
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next_index_(length),
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end_timestamp_(0),
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dtmf_index_(0) {}
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virtual ~SyncBuffer() {}
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// Returns the number of samples yet to play out form the buffer.
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size_t FutureLength() const;
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// Adds the contents of |append_this| to the back of the SyncBuffer. Removes
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// the same number of samples from the beginning of the SyncBuffer, to
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// maintain a constant buffer size. The |next_index_| is updated to reflect
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// the move of the beginning of "future" data.
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void PushBack(const AudioMultiVector& append_this);
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// Adds |length| zeros to the beginning of each channel. Removes
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// the same number of samples from the end of the SyncBuffer, to
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// maintain a constant buffer size. The |next_index_| is updated to reflect
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// the move of the beginning of "future" data.
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// Note that this operation may delete future samples that are waiting to
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// be played.
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void PushFrontZeros(size_t length);
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// Inserts |length| zeros into each channel at index |position|. The size of
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// the SyncBuffer is kept constant, which means that the last |length|
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// elements in each channel will be purged.
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virtual void InsertZerosAtIndex(size_t length, size_t position);
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// Overwrites each channel in this SyncBuffer with values taken from
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// |insert_this|. The values are taken from the beginning of |insert_this| and
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// are inserted starting at |position|. |length| values are written into each
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// channel. The size of the SyncBuffer is kept constant. That is, if |length|
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// and |position| are selected such that the new data would extend beyond the
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// end of the current SyncBuffer, the buffer is not extended.
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// The |next_index_| is not updated.
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virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
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size_t length,
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size_t position);
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// Same as the above method, but where all of |insert_this| is written (with
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// the same constraints as above, that the SyncBuffer is not extended).
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virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
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size_t position);
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// Reads |requested_len| samples from each channel and writes them interleaved
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// into |output|. The |next_index_| is updated to point to the sample to read
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// next time.
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size_t GetNextAudioInterleaved(size_t requested_len, int16_t* output);
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// Adds |increment| to |end_timestamp_|.
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void IncreaseEndTimestamp(uint32_t increment);
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// Flushes the buffer. The buffer will contain only zeros after the flush, and
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// |next_index_| will point to the end, like when the buffer was first
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// created.
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void Flush();
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const AudioVector& Channel(size_t n) const { return *channels_[n]; }
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AudioVector& Channel(size_t n) { return *channels_[n]; }
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// Accessors and mutators.
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size_t next_index() const { return next_index_; }
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void set_next_index(size_t value);
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uint32_t end_timestamp() const { return end_timestamp_; }
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void set_end_timestamp(uint32_t value) { end_timestamp_ = value; }
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size_t dtmf_index() const { return dtmf_index_; }
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void set_dtmf_index(size_t value);
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private:
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size_t next_index_;
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uint32_t end_timestamp_; // The timestamp of the last sample in the buffer.
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size_t dtmf_index_; // Index to the first non-DTMF sample in the buffer.
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DISALLOW_COPY_AND_ASSIGN(SyncBuffer);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
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