272 lines
7.8 KiB
C++
272 lines
7.8 KiB
C++
/*
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* Copyright (C) 2007-2012 David Robillard <d@drobilla.net>
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* Copyright (C) 2007-2017 Paul Davis <paul@linuxaudiosystems.com>
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* Copyright (C) 2010-2012 Carl Hetherington <carl@carlh.net>
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* Copyright (C) 2013-2016 Robin Gareus <robin@gareus.org>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#ifndef __ardour_audio_buffer_h__
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#define __ardour_audio_buffer_h__
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#include <cstring>
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#include "ardour/buffer.h"
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#include "ardour/runtime_functions.h"
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namespace ARDOUR
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{
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/** Buffer containing audio data. */
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class LIBARDOUR_API AudioBuffer : public Buffer
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{
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public:
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AudioBuffer (size_t capacity);
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~AudioBuffer ();
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/** silence buffer
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* @param len number of samples to clear
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* @param offset start offset
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*/
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void silence (samplecnt_t len, samplecnt_t offset = 0);
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bool silent_data() const;
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/** Copy samples from src array starting at src_offset into self starting at dst_offset
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* @param src array to read from
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* @param len number of samples to copy
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* @param dst_offset offset in destination buffer
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* @param src_offset start offset in src buffer
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*/
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void read_from (const Sample* src, samplecnt_t len, sampleoffset_t dst_offset = 0, sampleoffset_t src_offset = 0)
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{
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assert (src != 0);
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assert (_capacity > 0);
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assert (len <= _capacity);
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copy_vector (_data + dst_offset, src + src_offset, len);
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_silent = false;
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_written = true;
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}
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/** Copy samples from src buffer starting at src_offset into self starting at dst_offset
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* @param src buffer to read from
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* @param len number of samples to copy
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* @param dst_offset offset in destination buffer
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* @param src_offset start offset in src buffer
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*/
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void read_from (const Buffer& src, samplecnt_t len, sampleoffset_t dst_offset = 0, sampleoffset_t src_offset = 0)
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{
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assert (&src != this);
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assert (_capacity > 0);
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assert (src.type () == DataType::AUDIO);
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assert (dst_offset + len <= _capacity);
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assert (src_offset <= ((samplecnt_t)src.capacity () - len));
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if (src.silent ()) {
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memset (_data + dst_offset, 0, sizeof (Sample) * len);
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} else {
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copy_vector (_data + dst_offset, ((const AudioBuffer&)src).data () + src_offset, len);
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}
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if (dst_offset == 0 && src_offset == 0 && len == _capacity) {
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_silent = src.silent ();
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} else {
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_silent = _silent && src.silent ();
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}
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_written = true;
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}
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/** Accumulate (add) \p len samples from \p src starting at \p src_offset into self starting at \p dst_offset */
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void merge_from (const Buffer& src, samplecnt_t len, sampleoffset_t dst_offset = 0, sampleoffset_t src_offset = 0)
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{
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const AudioBuffer* ab = dynamic_cast<const AudioBuffer*> (&src);
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assert (ab);
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accumulate_from (*ab, len, dst_offset, src_offset);
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}
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/** Accumulate (add) \p len samples from \p src starting at \p src_offset into self starting at \p dst_offset */
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void accumulate_from (const AudioBuffer& src, samplecnt_t len, sampleoffset_t dst_offset = 0, sampleoffset_t src_offset = 0)
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{
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assert (_capacity > 0);
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assert (len <= _capacity);
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if (src.silent ()) {
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return;
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}
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Sample* const dst_raw = _data + dst_offset;
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const Sample* const src_raw = src.data () + src_offset;
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mix_buffers_no_gain (dst_raw, src_raw, len);
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_silent = (src.silent () && _silent);
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_written = true;
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}
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/** Accumulate (add) \p len samples of \p src starting at \p src_offset into self
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* starting at \p dst_offset
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*/
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void accumulate_from (const Sample* src, samplecnt_t len, sampleoffset_t dst_offset = 0, sampleoffset_t src_offset = 0)
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{
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assert (_capacity > 0);
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assert (len <= _capacity);
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Sample* const dst_raw = _data + dst_offset;
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const Sample* const src_raw = src + src_offset;
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mix_buffers_no_gain (dst_raw, src_raw, len);
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_silent = false;
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_written = true;
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}
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/** Accumulate (add) \p len samples if \p src starting at \p src_offset into self
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* starting at \p dst_offset scaling by \p gain_coeff
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*/
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void accumulate_with_gain_from (const AudioBuffer& src, samplecnt_t len, gain_t gain_coeff, sampleoffset_t dst_offset = 0, sampleoffset_t src_offset = 0)
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{
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assert (_capacity > 0);
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assert (len <= _capacity);
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if (src.silent () || gain_coeff == 0) {
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return;
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}
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Sample* const dst_raw = _data + dst_offset;
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const Sample* const src_raw = src.data () + src_offset;
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mix_buffers_with_gain (dst_raw, src_raw, len, gain_coeff);
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_silent = ((src.silent () && _silent) || (_silent && gain_coeff == 0));
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_written = true;
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}
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/** Accumulate (add) \p len samples from the start of \p src_raw into self at \p dst_offset
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* scaling by \p gain_coeff
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*/
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void accumulate_with_gain_from (const Sample* src_raw, samplecnt_t len, gain_t gain_coeff, sampleoffset_t dst_offset = 0)
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{
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assert (_capacity > 0);
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assert (len <= _capacity);
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Sample* const dst_raw = _data + dst_offset;
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mix_buffers_with_gain (dst_raw, src_raw, len, gain_coeff);
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_silent = (_silent && gain_coeff == 0);
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_written = true;
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}
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/** Accumulate (add) \p len samples from the start of \p src into self at \p dst_offset
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* using a linear gain ramp from \p initial to \p target .
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*/
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void accumulate_with_ramped_gain_from (const Sample* src, samplecnt_t len, gain_t initial, gain_t target, sampleoffset_t dst_offset = 0)
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{
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assert (_capacity > 0);
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assert (len <= _capacity);
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if (initial == 0 && target == 0) {
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return;
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}
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Sample* dst = _data + dst_offset;
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gain_t gain_delta = (target - initial) / len;
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for (samplecnt_t n = 0; n < len; ++n) {
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*dst++ += (*src++ * initial);
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initial += gain_delta;
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}
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_silent = (_silent && initial == 0 && target == 0);
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_written = true;
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}
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/** Apply a fixed gain factor to the audio buffer
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*
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* @param gain gain factor
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* @param len number of samples to amplify
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*/
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void apply_gain (gain_t gain, samplecnt_t len)
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{
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if (gain == 0) {
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memset (_data, 0, sizeof (Sample) * len);
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if (len == _capacity) {
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_silent = true;
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}
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return;
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}
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apply_gain_to_buffer (_data, len, gain);
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}
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/** Set the data contained by this buffer manually (for setting directly to jack buffer).
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*
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* Constructor MUST have been passed capacity=0 or this will die (to prevent mem leaks).
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*/
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void set_data (Sample* data, size_t size)
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{
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assert (!_owns_data); // prevent leaks
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_capacity = size;
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_data = data;
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_silent = false;
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_written = false;
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}
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/** Reallocate the buffer used internally to handle at least \p nframes of data
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*
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* Constructor MUST have been passed capacity!=0 or this will die (to prevent mem leaks).
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*/
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void resize (size_t nframes);
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const Sample* data (samplecnt_t offset = 0) const
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{
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assert (offset <= _capacity);
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return _data + offset;
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}
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Sample* data (samplecnt_t offset = 0)
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{
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assert (offset <= _capacity);
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_silent = false;
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return _data + offset;
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}
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/** Check buffer for silence
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*
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* @param nframes number of samples to check
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* @param n first non zero sample (if any)
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* @return true if all samples are zero
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*/
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bool check_silence (pframes_t nframes, pframes_t& n) const;
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void prepare ()
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{
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if (!_owns_data) {
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_data = 0;
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}
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_written = false;
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_silent = false;
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}
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bool written () const { return _written; }
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void set_written (bool w) { _written = w; }
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private:
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bool _owns_data;
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bool _written;
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Sample* _data; ///< Actual buffer contents
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};
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} // namespace ARDOUR
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#endif // __ardour_audio_audio_buffer_h__
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