globally change all use of "frame" to refer to audio into "sample".

[ardour.git] / libs / ardour / ardour / audio_buffer.h

diff --git a/libs/ardour/ardour/audio_buffer.h b/libs/ardour/ardour/audio_buffer.h
index a32e679b692ca59816a652eb8638270d86730ecf..833d1dcb189140f6f507bd93a5f1a074fabf6d57 100644 (file)
--- a/libs/ardour/ardour/audio_buffer.h
+++ b/libs/ardour/ardour/audio_buffer.h
@@ -33,10 +33,19 @@ public:
        AudioBuffer(size_t capacity);
        ~AudioBuffer();
 
-       void silence (framecnt_t len, framecnt_t offset = 0);
+       /** silence buffer
+        * @param len number of samples to clear
+        * @laram offset start offset
+        */
+       void silence (samplecnt_t len, samplecnt_t offset = 0);
 
-       /** Read @a len frames @a src starting at @a src_offset into self starting at @ dst_offset*/
-       void read_from (const Sample* src, framecnt_t len, framecnt_t dst_offset = 0, framecnt_t src_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, samplecnt_t len, sampleoffset_t dst_offset = 0, sampleoffset_t src_offset = 0) {
                assert(src != 0);
                assert(_capacity > 0);
                assert(len <= _capacity);
@@ -45,25 +54,30 @@ public:
                _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) {
+       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;
                _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) {
+       /** 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, 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(len <= _capacity);
-               assert( src_offset <= ((framecnt_t) src.capacity()-len));
+               assert(dst_offset + len <= _capacity);
+               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();
@@ -73,15 +87,15 @@ public:
                _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) {
+       /** 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);
        }
 
-       /** 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) {
+       /** 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);
 
@@ -94,8 +108,8 @@ public:
                _written = true;
        }
 
-       /** Acumulate (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) {
+       /** 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);
 
@@ -103,14 +117,14 @@ public:
                const Sample* const src_raw = src + src_offset;
 
                mix_buffers_no_gain(dst_raw, src_raw, len);
-               
+
                _silent = false;
                _written = true;
        }
 
-       /** Acumulate (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, framecnt_t dst_offset = 0, framecnt_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);
@@ -128,9 +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, framecnt_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);
@@ -143,9 +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, framecnt_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);
@@ -153,7 +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;
                }
@@ -162,7 +176,11 @@ public:
                _written = true;
        }
 
-       void apply_gain (gain_t gain, framecnt_t len) {
+       /** apply a fixed gain factor to the audio buffer
+        * @param gain gain factor
+        * @param len number of samples to amplify
+        */
+       void apply_gain (gain_t gain, samplecnt_t len) {
                apply_gain_to_buffer (_data, len, gain);
        }
 
@@ -173,7 +191,6 @@ public:
        void set_data (Sample* data, size_t size) {
                assert(!_owns_data); // prevent leaks
                _capacity = size;
-               _size = size;
                _data = data;
                _silent = false;
                _written = false;
@@ -185,22 +202,31 @@ public:
         */
        void resize (size_t nframes);
 
-       bool empty() const { return _size == 0; }
-
-       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;
        }
 
-       bool check_silence (pframes_t, bool, pframes_t&) const;
+       /** check buffer for silence
+        * @param nframes  number of samples 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;
 
-       void prepare () { _written = false; _silent = false; }
+       void prepare () {
+               if (!_owns_data) {
+                       _data = 0;
+               }
+               _written = false;
+               _silent = false;
+       }
        bool written() const { return _written; }
        void set_written(bool w) { _written = w; }