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git-svn-id: svn://localhost/ardour2/branches/3.0@10846 d708f5d6-7413-0410-9779-e7cbd77b26cf
184 lines
5.5 KiB
C++
184 lines
5.5 KiB
C++
/*
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Copyright (C) 2006 Paul Davis
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This program is free software; you can redistribute it and/or modify it
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under the terms of the GNU General Public License as published by the Free
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Software Foundation; either version 2 of the License, or (at your option)
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any later version.
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This program is distributed in the hope that it will be useful, but WITHOUT
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ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
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FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License
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for more details.
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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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675 Mass Ave, Cambridge, MA 02139, 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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namespace ARDOUR {
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/** Buffer containing audio data. */
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class 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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void silence (framecnt_t len, framecnt_t offset = 0) {
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if (!_silent) {
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assert(_capacity > 0);
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assert(offset + len <= _capacity);
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memset(_data + offset, 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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}
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_written = true;
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}
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/** Read @a len frames @a src starting at @a src_offset into self starting at @ dst_offset*/
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void read_from (const Buffer& src, framecnt_t len, framecnt_t dst_offset = 0, framecnt_t src_offset = 0) {
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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(len <= _capacity);
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assert( src_offset <= ((framecnt_t) src.capacity()-len));
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memcpy(_data + dst_offset, ((AudioBuffer&)src).data() + src_offset, sizeof(Sample) * len);
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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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/** Acumulate (add) @a len frames @a src starting at @a src_offset into self starting at @a dst_offset */
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void merge_from (const Buffer& src, framecnt_t len, framecnt_t dst_offset = 0, framecnt_t src_offset = 0) {
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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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/** Acumulate (add) @a len frames @a src starting at @a src_offset into self starting at @a dst_offset */
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void accumulate_from (const AudioBuffer& src, framecnt_t len, framecnt_t dst_offset = 0, framecnt_t src_offset = 0) {
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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.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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/** Acumulate (add) @a len frames @a src starting at @a src_offset into self starting at @dst_offset
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* scaling by @a gain_coeff */
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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) {
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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_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) @a len frames FROM THE START OF @a src into self
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* scaling by @a gain_coeff */
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void accumulate_with_gain_from (const Sample* src_raw, framecnt_t len, gain_t gain_coeff, framecnt_t dst_offset = 0) {
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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) @a len frames FROM THE START OF @a src into self
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* scaling by @a gain_coeff */
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void accumulate_with_ramped_gain_from (const Sample* src, framecnt_t len, gain_t initial, gain_t target, framecnt_t dst_offset = 0) {
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assert(_capacity > 0);
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assert(len <= _capacity);
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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 (framecnt_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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void apply_gain (gain_t gain, framecnt_t len) {
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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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assert(!_owns_data); // prevent leaks
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_capacity = size;
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_size = 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 @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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bool empty() const { return _size == 0; }
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const Sample* data (framecnt_t offset = 0) const {
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assert(offset <= _capacity);
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return _data + offset;
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}
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Sample* data (framecnt_t offset = 0) {
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assert(offset <= _capacity);
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return _data + offset;
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}
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void prepare () { _written = false; _silent = false; }
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bool written() const { return _written; }
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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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