X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=libs%2Fardour%2Fardour%2Faudio_buffer.h;h=051b75ab4bf4b1821557f20f31f19c2ebfe9ec78;hb=4537f5fb20b2f43394ef1b47aecfd320fce0c2bb;hp=054a1f7b453f7aed074c0ed38ee6a37494664033;hpb=15cee60021eada542b2dae0fafbb3150fcaa5010;p=ardour.git diff --git a/libs/ardour/ardour/audio_buffer.h b/libs/ardour/ardour/audio_buffer.h index 054a1f7b45..051b75ab4b 100644 --- a/libs/ardour/ardour/audio_buffer.h +++ b/libs/ardour/ardour/audio_buffer.h @@ -27,25 +27,24 @@ namespace ARDOUR { /** Buffer containing audio data. */ -class AudioBuffer : public Buffer +class LIBARDOUR_API AudioBuffer : public Buffer { public: AudioBuffer(size_t capacity); ~AudioBuffer(); - void silence (framecnt_t len, framecnt_t offset = 0) { - if (!_silent) { - assert(_capacity > 0); - assert(offset + len <= _capacity); - memset(_data + offset, 0, sizeof (Sample) * len); - if (len == _capacity) { - _silent = true; - } - } - _written = true; - } + /** silence buffer + * @param len number of samples to clear + * @laram offset start offset + */ + void silence (framecnt_t len, framecnt_t offset = 0); - /** Read @a len frames @a src starting at @a src_offset into self starting at @ dst_offset*/ + /** 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); @@ -55,7 +54,7 @@ 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, 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); @@ -67,12 +66,17 @@ public: _written = true; } - /** Read @a len frames @a src starting at @a src_offset into self starting at @ dst_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(len <= _capacity); + 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) { @@ -83,14 +87,14 @@ public: _written = true; } - /** Acumulate (add) @a len frames @a src starting at @a src_offset into self starting at @a dst_offset */ + /** 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(&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 */ + /** 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(len <= _capacity); @@ -104,7 +108,7 @@ public: _written = true; } - /** Acumulate (add) @a len frames @a src starting at @a src_offset into self starting at @a dst_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); @@ -113,12 +117,12 @@ 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 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, framecnt_t len, gain_t gain_coeff, framecnt_t dst_offset = 0, framecnt_t src_offset = 0) { @@ -172,6 +176,10 @@ public: _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); } @@ -183,7 +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; @@ -195,8 +202,6 @@ public: */ void resize (size_t nframes); - bool empty() const { return _size == 0; } - const Sample* data (framecnt_t offset = 0) const { assert(offset <= _capacity); return _data + offset; @@ -204,10 +209,16 @@ public: Sample* data (framecnt_t offset = 0) { assert(offset <= _capacity); + _silent = false; return _data + offset; } - bool check_silence (pframes_t, pframes_t&) const; + /** 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; void prepare () { _written = false; _silent = false; } bool written() const { return _written; }