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Diffstat (limited to 'libs/ardour/ardour/audio_buffer.h')
-rw-r--r--libs/ardour/ardour/audio_buffer.h48
1 files changed, 24 insertions, 24 deletions
diff --git a/libs/ardour/ardour/audio_buffer.h b/libs/ardour/ardour/audio_buffer.h
index 7d255c5c50..833d1dcb18 100644
--- a/libs/ardour/ardour/audio_buffer.h
+++ b/libs/ardour/ardour/audio_buffer.h
@@ -37,7 +37,7 @@ public:
* @param len number of samples to clear
* @laram offset start offset
*/
- void silence (framecnt_t len, framecnt_t offset = 0);
+ void silence (samplecnt_t len, samplecnt_t offset = 0);
/** Copy samples from src array starting at src_offset into self starting at dst_offset
* @param src array to read from
@@ -45,7 +45,7 @@ public:
* @param dst_offset offset in destination buffer
* @param src_offset start offset in src buffer
*/
- void read_from (const Sample* src, framecnt_t len, frameoffset_t dst_offset = 0, frameoffset_t src_offset = 0) {
+ void read_from (const Sample* src, samplecnt_t len, sampleoffset_t dst_offset = 0, sampleoffset_t src_offset = 0) {
assert(src != 0);
assert(_capacity > 0);
assert(len <= _capacity);
@@ -54,12 +54,12 @@ public:
_written = true;
}
- void read_from_with_gain (const Sample* src, framecnt_t len, gain_t gain, frameoffset_t dst_offset = 0, frameoffset_t src_offset = 0) {
+ void read_from_with_gain (const Sample* src, samplecnt_t len, gain_t gain, sampleoffset_t dst_offset = 0, sampleoffset_t src_offset = 0) {
assert(src != 0);
assert(_capacity > 0);
assert(len <= _capacity);
src += src_offset;
- for (framecnt_t n = 0; n < len; ++n) {
+ for (samplecnt_t n = 0; n < len; ++n) {
_data[dst_offset+n] = src[n] * gain;
}
_silent = false;
@@ -72,12 +72,12 @@ public:
* @param dst_offset offset in destination buffer
* @param src_offset start offset in src buffer
*/
- void read_from (const Buffer& src, framecnt_t len, frameoffset_t dst_offset = 0, frameoffset_t src_offset = 0) {
+ void read_from (const Buffer& src, samplecnt_t len, sampleoffset_t dst_offset = 0, sampleoffset_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));
+ assert( src_offset <= ((samplecnt_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();
@@ -87,15 +87,15 @@ public:
_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, frameoffset_t dst_offset = 0, frameoffset_t src_offset = 0) {
+ /** Accumulate (add) @a len samples @a src starting at @a src_offset into self starting at @a dst_offset */
+ void merge_from (const Buffer& src, samplecnt_t len, sampleoffset_t dst_offset = 0, sampleoffset_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 @a src starting at @a src_offset into self starting at @a dst_offset */
- void accumulate_from (const AudioBuffer& src, framecnt_t len, frameoffset_t dst_offset = 0, frameoffset_t src_offset = 0) {
+ /** Accumulate (add) @a len samples @a src starting at @a src_offset into self starting at @a dst_offset */
+ void accumulate_from (const AudioBuffer& src, samplecnt_t len, sampleoffset_t dst_offset = 0, sampleoffset_t src_offset = 0) {
assert(_capacity > 0);
assert(len <= _capacity);
@@ -108,8 +108,8 @@ public:
_written = true;
}
- /** 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, frameoffset_t dst_offset = 0, frameoffset_t src_offset = 0) {
+ /** Accumulate (add) @a len samples @a src starting at @a src_offset into self starting at @a dst_offset */
+ void accumulate_from (const Sample* src, samplecnt_t len, sampleoffset_t dst_offset = 0, sampleoffset_t src_offset = 0) {
assert(_capacity > 0);
assert(len <= _capacity);
@@ -122,9 +122,9 @@ public:
_written = true;
}
- /** Accumulate (add) @a len frames @a src starting at @a src_offset into self starting at @dst_offset
+ /** Accumulate (add) @a len samples @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, frameoffset_t dst_offset = 0, frameoffset_t src_offset = 0) {
+ void accumulate_with_gain_from (const AudioBuffer& src, samplecnt_t len, gain_t gain_coeff, sampleoffset_t dst_offset = 0, sampleoffset_t src_offset = 0) {
assert(_capacity > 0);
assert(len <= _capacity);
@@ -142,9 +142,9 @@ public:
_written = true;
}
- /** Accumulate (add) @a len frames FROM THE START OF @a src into self
+ /** Accumulate (add) @a len samples 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, frameoffset_t dst_offset = 0) {
+ void accumulate_with_gain_from (const Sample* src_raw, samplecnt_t len, gain_t gain_coeff, sampleoffset_t dst_offset = 0) {
assert(_capacity > 0);
assert(len <= _capacity);
@@ -157,9 +157,9 @@ public:
_written = true;
}
- /** Accumulate (add) @a len frames FROM THE START OF @a src into self
+ /** Accumulate (add) @a len samples 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, frameoffset_t dst_offset = 0) {
+ void accumulate_with_ramped_gain_from (const Sample* src, samplecnt_t len, gain_t initial, gain_t target, sampleoffset_t dst_offset = 0) {
assert(_capacity > 0);
assert(len <= _capacity);
@@ -167,7 +167,7 @@ public:
Sample* dst = _data + dst_offset;
gain_t gain_delta = (target - initial)/len;
- for (framecnt_t n = 0; n < len; ++n) {
+ for (samplecnt_t n = 0; n < len; ++n) {
*dst++ += (*src++ * initial);
initial += gain_delta;
}
@@ -178,9 +178,9 @@ public:
/** apply a fixed gain factor to the audio buffer
* @param gain gain factor
- * @param len number of frames to amplify
+ * @param len number of samples to amplify
*/
- void apply_gain (gain_t gain, framecnt_t len) {
+ void apply_gain (gain_t gain, samplecnt_t len) {
apply_gain_to_buffer (_data, len, gain);
}
@@ -202,19 +202,19 @@ public:
*/
void resize (size_t nframes);
- const Sample* data (framecnt_t offset = 0) const {
+ const Sample* data (samplecnt_t offset = 0) const {
assert(offset <= _capacity);
return _data + offset;
}
- Sample* data (framecnt_t offset = 0) {
+ Sample* data (samplecnt_t offset = 0) {
assert(offset <= _capacity);
_silent = false;
return _data + offset;
}
/** check buffer for silence
- * @param nframes number of frames to check
+ * @param nframes number of samples to check
* @param n first non zero sample (if any)
* @return true if all samples are zero
*/