/*
- Copyright (C) 2006 Paul Davis
-
+ 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.
#define __ardour_audio_buffer_h__
#include <cstring>
-#include <ardour/buffer.h>
+
+#include "ardour/buffer.h"
+#include "ardour/runtime_functions.h"
namespace ARDOUR {
-class AudioBuffer : public Buffer
+/** Buffer containing audio data. */
+class LIBARDOUR_API 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;
- }
+ /** silence buffer
+ * @param len number of samples to clear
+ * @laram offset start offset
+ */
+ void silence (framecnt_t len, framecnt_t offset = 0);
+
+ /** Copy samples from src array starting at src_offset into self starting at dst_offset
+ * @param src array to read from
+ * @param len number of samples to copy
+ * @param dst_offset offset in destination buffer
+ * @param src_offset start offset in src buffer
+ */
+ 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;
}
-
- /** 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) {
+
+ /** Copy samples from src buffer starting at src_offset into self starting at dst_offset
+ * @param src buffer to read from
+ * @param len number of samples to copy
+ * @param dst_offset offset in destination buffer
+ * @param src_offset start offset in src buffer
+ */
+ 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(offset + len <= _capacity);
- memcpy(_data + offset, ((AudioBuffer&)src).data(), sizeof(Sample) * len);
- _silent = src.silent();
+ 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;
+ }
+
+ /** Accumulate (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);
}
-
- /** 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) {
+
+ /** Accumulate (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(offset + len <= _capacity);
+ assert(len <= _capacity);
- Sample* const dst_raw = _data + offset;
- const Sample* const src_raw = src.data();
+ 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;
}
-
- /** Accumulate (add) @a len frames FROM THE START OF @a src into self at @a offset
+
+ /** Accumulate (add) @a len frames @a src starting at @a src_offset into self starting at @a dst_offset */
+ 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;
+ }
+
+ /** Accumulate (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, nframes_t len, nframes_t offset, gain_t 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(offset + len <= _capacity);
+ assert(len <= _capacity);
- Sample* const dst_raw = _data + offset;
- const Sample* const src_raw = src.data();
+ 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 apply_gain(gain_t gain, nframes_t len, nframes_t offset=0) {
- apply_gain_to_buffer (_data + offset, len, gain);
+
+ /** 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;
+ }
+
+ /** apply a fixed gain factor to the audio buffer
+ * @param gain gain factor
+ * @param len number of frames to amplify
+ */
+ 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 () const { return _data; }
- Sample* data () { return _data; }
+ 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;
+ }
- const Sample* data(nframes_t nframes, nframes_t offset) const
- { assert(offset + nframes <= _capacity); return _data + offset; }
+ /** check buffer for silence
+ * @param nframes number of frames to check
+ * @param n first non zero sample (if any)
+ * @return true if all samples are zero
+ */
+ bool check_silence (pframes_t nframes, pframes_t& n) const;
- Sample* data (nframes_t nframes, nframes_t offset)
- { assert(offset + nframes <= _capacity); return _data + offset; }
+ void prepare () {
+ if (!_owns_data) {
+ _data = 0;
+ }
+ _written = false;
+ _silent = false;
+ }
+ bool written() const { return _written; }
+ void set_written(bool w) { _written = w; }
-private:
+ private:
bool _owns_data;
+ bool _written;
Sample* _data; ///< Actual buffer contents
};
projects / ardour.git / blobdiff