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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 <cstring>
#include "ardour/buffer.h"
namespace ARDOUR {
/** Buffer containing audio data. */
class AudioBuffer : public Buffer
{
public:
AudioBuffer(size_t capacity);
~AudioBuffer();
void silence (framecnt_t len, framecnt_t offset = 0) {
if (!_silent) {
assert(_capacity > 0);
assert(offset + len <= _capacity);
memset(_data + offset, 0, sizeof (Sample) * len);
if (len == _capacity) {
_silent = true;
}
}
_written = true;
}
/** Read @a len frames @a src starting at @a src_offset into self starting at @ dst_offset*/
void read_from (const Buffer& src, framecnt_t len, framecnt_t dst_offset = 0, framecnt_t src_offset = 0) {
assert(&src != this);
assert(_capacity > 0);
assert(src.type() == DataType::AUDIO);
assert(len <= _capacity);
memcpy(_data + dst_offset, ((AudioBuffer&)src).data() + src_offset, sizeof(Sample) * len);
if (dst_offset == 0 && src_offset == 0 && len == _capacity) {
_silent = src.silent();
} else {
_silent = _silent && src.silent();
}
_written = true;
}
/** Acumulate (add) @a len frames @a src starting at @a src_offset into self starting at @a dst_offset */
void merge_from (const Buffer& src, framecnt_t len, framecnt_t dst_offset = 0, framecnt_t src_offset = 0) {
const AudioBuffer* ab = dynamic_cast<const AudioBuffer*>(&src);
assert (ab);
accumulate_from (*ab, len, dst_offset, src_offset);
}
/** Acumulate (add) @a len frames @a src starting at @a src_offset into self starting at @a dst_offset */
void accumulate_from (const AudioBuffer& src, framecnt_t len, framecnt_t dst_offset = 0, framecnt_t src_offset = 0) {
assert(_capacity > 0);
assert(len <= _capacity);
Sample* const dst_raw = _data + dst_offset;
const Sample* const src_raw = src.data() + src_offset;
mix_buffers_no_gain(dst_raw, src_raw, len);
_silent = (src.silent() && _silent);
_written = true;
}
/** Acumulate (add) @a len frames @a src starting at @a src_offset into self starting at @dst_offset
* scaling by @a gain_coeff */
void accumulate_with_gain_from (const AudioBuffer& src, framecnt_t len, gain_t gain_coeff, framecnt_t dst_offset = 0, framecnt_t src_offset = 0) {
assert(_capacity > 0);
assert(len <= _capacity);
if (src.silent()) {
return;
}
Sample* const dst_raw = _data + dst_offset;
const Sample* const src_raw = src.data() + src_offset;
mix_buffers_with_gain (dst_raw, src_raw, len, gain_coeff);
_silent = ( (src.silent() && _silent) || (_silent && gain_coeff == 0) );
_written = true;
}
/** Accumulate (add) @a len frames FROM THE START OF @a src into self
* scaling by @a gain_coeff */
void accumulate_with_gain_from (const Sample* src_raw, framecnt_t len, gain_t gain_coeff, framecnt_t dst_offset = 0) {
assert(_capacity > 0);
assert(len <= _capacity);
Sample* const dst_raw = _data + dst_offset;
mix_buffers_with_gain (dst_raw, src_raw, len, gain_coeff);
_silent = (_silent && gain_coeff == 0);
_written = true;
}
/** Accumulate (add) @a len frames FROM THE START OF @a src into self
* scaling by @a gain_coeff */
void accumulate_with_ramped_gain_from (const Sample* src, framecnt_t len, gain_t initial, gain_t target, framecnt_t dst_offset = 0) {
assert(_capacity > 0);
assert(len <= _capacity);
Sample* dst = _data + dst_offset;
gain_t gain_delta = (target - initial)/len;
for (framecnt_t n = 0; n < len; ++n) {
*dst++ += (*src++ * initial);
initial += gain_delta;
}
_silent = (_silent && initial == 0 && target == 0);
_written = true;
}
void apply_gain (gain_t gain, framecnt_t len) {
apply_gain_to_buffer (_data, 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;
_written = 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 (framecnt_t offset = 0) const {
assert(offset <= _capacity);
return _data + offset;
}
Sample* data (framecnt_t offset = 0) {
assert(offset <= _capacity);
return _data + offset;
}
void prepare () { _written = false; _silent = false; }
bool written() const { return _written; }
private:
bool _owns_data;
bool _written;
Sample* _data; ///< Actual buffer contents
};
} // namespace ARDOUR
#endif // __ardour_audio_audio_buffer_h__
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