audio_buffer.h

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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"
#include "ardour/runtime_functions.h"

namespace ARDOUR {

class LIBARDOUR_API AudioBuffer : public Buffer
{
public:
        AudioBuffer(size_t capacity);
        ~AudioBuffer();

        void silence (framecnt_t len, framecnt_t offset = 0);

        void read_from (const Sample* src, framecnt_t len, framecnt_t dst_offset = 0, framecnt_t src_offset = 0) {
                assert(src != 0);
                assert(_capacity > 0);
                assert(len <= _capacity);
                memcpy(_data + dst_offset, src + src_offset, sizeof(Sample) * len);
                _silent = false;
                _written = true;
        }

        void read_from_with_gain (const Sample* src, framecnt_t len, gain_t gain, framecnt_t dst_offset = 0, framecnt_t src_offset = 0) {
                assert(src != 0);
                assert(_capacity > 0);
                assert(len <= _capacity);
                src += src_offset;
                for (framecnt_t n = 0; n < len; ++n) {
                        _data[dst_offset+n] = src[n] * gain;
                }
                _silent = false;
                _written = true;
        }

        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(dst_offset + len <= _capacity);
                assert( src_offset <= ((framecnt_t) src.capacity()-len));
                memcpy(_data + dst_offset, ((const 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;
        }

        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);
        }

        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;
        }

        void accumulate_from (const Sample* 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 + src_offset;

                mix_buffers_no_gain(dst_raw, src_raw, len);
                
                _silent = false;
                _written = true;
        }

        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;
        }

        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;
        }

        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);
        }

        void set_data (Sample* data, size_t size) {
                assert(!_owns_data); // prevent leaks
                _capacity = size;
                _data = data;
                _silent = false;
                _written = false;
        }

        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);
                _silent = false;
                return _data + offset;
        }

        bool check_silence (pframes_t, pframes_t&) const;

        void prepare () { _written = false; _silent = false; }
        bool written() const { return _written; }
        void set_written(bool w) { _written = w; }

  private:
        bool    _owns_data;
        bool    _written;
        Sample* _data; 
};


} // namespace ARDOUR

#endif // __ardour_audio_audio_buffer_h__