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/*
Copyright (C) 2006 Paul Davis
This program is free software; you can redistribute it and/or modify it
under the terms of the GNU General Public License as published by the Free
Software Foundation; either version 2 of the License, or (at your option)
any later version.
This program is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License
for more details.
You should have received a copy of the GNU General Public License along
with this program; if not, write to the Free Software Foundation, Inc.,
675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __ardour_audio_buffer_h__
#define __ardour_audio_buffer_h__
#include <ardour/buffer.h>
namespace ARDOUR {
/** Buffer containing 32-bit floating point (audio) data. */
class AudioBuffer : public Buffer
{
public:
AudioBuffer(size_t capacity);
~AudioBuffer();
void silence(nframes_t len, nframes_t offset=0)
{
if (!_silent) {
assert(_capacity > 0);
assert(offset + len <= _capacity);
memset(_data + offset, 0, sizeof (Sample) * len);
if (offset == 0 && len == _capacity) {
_silent = true;
}
}
}
/** Read @a len frames FROM THE START OF @a src into self at @a offset */
void read_from(const Buffer& src, nframes_t len, nframes_t offset)
{
assert(_capacity > 0);
assert(src.type() == DataType::AUDIO);
assert(offset + len <= _capacity);
memcpy(_data + offset, ((AudioBuffer&)src).data(), sizeof(Sample) * len);
_silent = src.silent();
}
/** Accumulate (add)@a len frames FROM THE START OF @a src into self at @a offset */
void accumulate_from(const AudioBuffer& src, nframes_t len, nframes_t offset)
{
assert(_capacity > 0);
assert(offset + len <= _capacity);
Sample* const dst_raw = _data + offset;
const Sample* const src_raw = src.data();
mix_buffers_no_gain(dst_raw, src_raw, len);
_silent = (src.silent() && _silent);
}
/** Accumulate (add) @a len frames FROM THE START OF @a src into self at @a offset
* scaling by @a gain_coeff */
void accumulate_with_gain_from(const AudioBuffer& src, nframes_t len, nframes_t offset, gain_t gain_coeff)
{
assert(_capacity > 0);
assert(offset + len <= _capacity);
Sample* const dst_raw = _data + offset;
const Sample* const src_raw = src.data();
mix_buffers_with_gain (dst_raw, src_raw, len, gain_coeff);
_silent = ( (src.silent() && _silent) || (_silent && gain_coeff == 0) );
}
void apply_gain(gain_t gain, nframes_t len, nframes_t offset=0) {
apply_gain_to_buffer (_data + offset, len, gain);
}
/** Set the data contained by this buffer manually (for setting directly to jack buffer).
*
* Constructor MUST have been passed capacity=0 or this will die (to prevent mem leaks).
*/
void set_data(Sample* data, size_t size)
{
assert(!_owns_data); // prevent leaks
_capacity = size;
_size = size;
_data = data;
_silent = false;
}
/** Reallocate the buffer used internally to handle at least @nframes of data
*
* Constructor MUST have been passed capacity!=0 or this will die (to prevent mem leaks).
*/
void resize (size_t nframes);
const Sample* data () const { return _data; }
Sample* data () { return _data; }
const Sample* data(nframes_t nframes, nframes_t offset) const
{ assert(offset + nframes <= _capacity); return _data + offset; }
Sample* data (nframes_t nframes, nframes_t offset)
{ assert(offset + nframes <= _capacity); return _data + offset; }
private:
bool _owns_data;
Sample* _data; ///< Actual buffer contents
};
} // namespace ARDOUR
#endif // __ardour_audio_audio_buffer_h__
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