From 30b087ab3d28f1585987fa3f6ae006562ae192e3 Mon Sep 17 00:00:00 2001 From: Paul Davis Date: Mon, 18 Sep 2017 12:39:17 -0400 Subject: globally change all use of "frame" to refer to audio into "sample". Generated by tools/f2s. Some hand-editing will be required in a few places to fix up comments related to timecode and video in order to keep the legible --- libs/ardour/audiosource.cc | 160 ++++++++++++++++++++++----------------------- 1 file changed, 80 insertions(+), 80 deletions(-) (limited to 'libs/ardour/audiosource.cc') diff --git a/libs/ardour/audiosource.cc b/libs/ardour/audiosource.cc index 2db06230f6..aea0203394 100644 --- a/libs/ardour/audiosource.cc +++ b/libs/ardour/audiosource.cc @@ -152,14 +152,14 @@ AudioSource::empty () const return _length == 0; } -framecnt_t -AudioSource::length (framepos_t /*pos*/) const +samplecnt_t +AudioSource::length (samplepos_t /*pos*/) const { return _length; } void -AudioSource::update_length (framecnt_t len) +AudioSource::update_length (samplecnt_t len) { if (len > _length) { _length = len; @@ -308,8 +308,8 @@ AudioSource::initialize_peakfile (const string& audio_path, const bool in_sessio return 0; } -framecnt_t -AudioSource::read (Sample *dst, framepos_t start, framecnt_t cnt, int /*channel*/) const +samplecnt_t +AudioSource::read (Sample *dst, samplepos_t start, samplecnt_t cnt, int /*channel*/) const { assert (cnt >= 0); @@ -317,8 +317,8 @@ AudioSource::read (Sample *dst, framepos_t start, framecnt_t cnt, int /*channel* return read_unlocked (dst, start, cnt); } -framecnt_t -AudioSource::write (Sample *dst, framecnt_t cnt) +samplecnt_t +AudioSource::write (Sample *dst, samplecnt_t cnt) { Glib::Threads::Mutex::Lock lm (_lock); /* any write makes the file not removable */ @@ -327,7 +327,7 @@ AudioSource::write (Sample *dst, framecnt_t cnt) } int -AudioSource::read_peaks (PeakData *peaks, framecnt_t npeaks, framepos_t start, framecnt_t cnt, double samples_per_visual_peak) const +AudioSource::read_peaks (PeakData *peaks, samplecnt_t npeaks, samplepos_t start, samplecnt_t cnt, double samples_per_visual_peak) const { return read_peaks_with_fpp (peaks, npeaks, start, cnt, samples_per_visual_peak, _FPP); } @@ -337,8 +337,8 @@ AudioSource::read_peaks (PeakData *peaks, framecnt_t npeaks, framepos_t start, f */ int -AudioSource::read_peaks_with_fpp (PeakData *peaks, framecnt_t npeaks, framepos_t start, framecnt_t cnt, - double samples_per_visual_peak, framecnt_t samples_per_file_peak) const +AudioSource::read_peaks_with_fpp (PeakData *peaks, samplecnt_t npeaks, samplepos_t start, samplecnt_t cnt, + double samples_per_visual_peak, samplecnt_t samples_per_file_peak) const { Glib::Threads::Mutex::Lock lm (_lock); double scale; @@ -353,8 +353,8 @@ AudioSource::read_peaks_with_fpp (PeakData *peaks, framecnt_t npeaks, framepos_t #else const int bufsize = sysconf(_SC_PAGESIZE); #endif - framecnt_t read_npeaks = npeaks; - framecnt_t zero_fill = 0; + samplecnt_t read_npeaks = npeaks; + samplecnt_t zero_fill = 0; GStatBuf statbuf; @@ -413,7 +413,7 @@ AudioSource::read_peaks_with_fpp (PeakData *peaks, framecnt_t npeaks, framepos_t if (cnt > _length - start) { // cerr << "too close to end @ " << _length << " given " << start << " + " << cnt << " (" << _length - start << ")" << endl; cnt = _length - start; - read_npeaks = min ((framecnt_t) floor (cnt / samples_per_visual_peak), npeaks); + read_npeaks = min ((samplecnt_t) floor (cnt / samples_per_visual_peak), npeaks); zero_fill = npeaks - read_npeaks; expected_peaks = (cnt / (double) samples_per_file_peak); scale = npeaks/expected_peaks; @@ -436,7 +436,7 @@ AudioSource::read_peaks_with_fpp (PeakData *peaks, framecnt_t npeaks, framepos_t return -1; } - for (framecnt_t i = 0; i < npeaks; ++i) { + for (samplecnt_t i = 0; i < npeaks; ++i) { peaks[i].max = raw_staging[i]; peaks[i].min = raw_staging[i]; } @@ -518,7 +518,7 @@ AudioSource::read_peaks_with_fpp (PeakData *peaks, framecnt_t npeaks, framepos_t /* the caller wants: - - more frames-per-peak (lower resolution) than the peakfile, or to put it another way, + - more samples-per-peak (lower resolution) than the peakfile, or to put it another way, - less peaks than the peakfile holds for the same range So, read a block into a staging area, and then downsample from there. @@ -526,15 +526,15 @@ AudioSource::read_peaks_with_fpp (PeakData *peaks, framecnt_t npeaks, framepos_t to avoid confusion, I'll refer to the requested peaks as visual_peaks and the peakfile peaks as stored_peaks */ - const framecnt_t chunksize = (framecnt_t) expected_peaks; // we read all the peaks we need in one hit. + const samplecnt_t chunksize = (samplecnt_t) expected_peaks; // we read all the peaks we need in one hit. - /* compute the rounded up frame position */ + /* compute the rounded up sample position */ - framepos_t current_stored_peak = (framepos_t) ceil (start / (double) samples_per_file_peak); - framepos_t next_visual_peak = (framepos_t) ceil (start / samples_per_visual_peak); - double next_visual_peak_frame = next_visual_peak * samples_per_visual_peak; - framepos_t stored_peak_before_next_visual_peak = (framepos_t) next_visual_peak_frame / samples_per_file_peak; - framecnt_t nvisual_peaks = 0; + samplepos_t current_stored_peak = (samplepos_t) ceil (start / (double) samples_per_file_peak); + samplepos_t next_visual_peak = (samplepos_t) ceil (start / samples_per_visual_peak); + double next_visual_peak_sample = next_visual_peak * samples_per_visual_peak; + samplepos_t stored_peak_before_next_visual_peak = (samplepos_t) next_visual_peak_sample / samples_per_file_peak; + samplecnt_t nvisual_peaks = 0; uint32_t i = 0; /* handle the case where the initial visual peak is on a pixel boundary */ @@ -608,8 +608,8 @@ AudioSource::read_peaks_with_fpp (PeakData *peaks, framecnt_t npeaks, framepos_t peak_cache[nvisual_peaks].max = xmax; peak_cache[nvisual_peaks].min = xmin; ++nvisual_peaks; - next_visual_peak_frame = min ((double) start + cnt, (next_visual_peak_frame + samples_per_visual_peak)); - stored_peak_before_next_visual_peak = (uint32_t) next_visual_peak_frame / samples_per_file_peak; + next_visual_peak_sample = min ((double) start + cnt, (next_visual_peak_sample + samples_per_visual_peak)); + stored_peak_before_next_visual_peak = (uint32_t) next_visual_peak_sample / samples_per_file_peak; } if (zero_fill) { @@ -630,35 +630,35 @@ AudioSource::read_peaks_with_fpp (PeakData *peaks, framecnt_t npeaks, framepos_t /* the caller wants - - less frames-per-peak (more resolution) + - less samples-per-peak (more resolution) - more peaks than stored in the Peakfile So, fetch data from the raw source, and generate peak data on the fly. */ - framecnt_t frames_read = 0; - framepos_t current_frame = start; - framecnt_t i = 0; - framecnt_t nvisual_peaks = 0; - framecnt_t chunksize = (framecnt_t) min (cnt, (framecnt_t) 4096); + samplecnt_t samples_read = 0; + samplepos_t current_sample = start; + samplecnt_t i = 0; + samplecnt_t nvisual_peaks = 0; + samplecnt_t chunksize = (samplecnt_t) min (cnt, (samplecnt_t) 4096); boost::scoped_array raw_staging(new Sample[chunksize]); - framepos_t frame_pos = start; - double pixel_pos = floor (frame_pos / samples_per_visual_peak); - double next_pixel_pos = ceil (frame_pos / samples_per_visual_peak); - double pixels_per_frame = 1.0 / samples_per_visual_peak; + samplepos_t sample_pos = start; + double pixel_pos = floor (sample_pos / samples_per_visual_peak); + double next_pixel_pos = ceil (sample_pos / samples_per_visual_peak); + double pixels_per_sample = 1.0 / samples_per_visual_peak; xmin = 1.0; xmax = -1.0; while (nvisual_peaks < read_npeaks) { - if (i == frames_read) { + if (i == samples_read) { - to_read = min (chunksize, (framecnt_t)(_length - current_frame)); + to_read = min (chunksize, (samplecnt_t)(_length - current_sample)); - if (current_frame >= _length) { + if (current_sample >= _length) { /* hmm, error condition - we've reached the end of the file without generating all the peak data. cook up a zero-filled @@ -671,12 +671,12 @@ AudioSource::read_peaks_with_fpp (PeakData *peaks, framecnt_t npeaks, framepos_t } else { - to_read = min (chunksize, (_length - current_frame)); + to_read = min (chunksize, (_length - current_sample)); - if ((frames_read = read_unlocked (raw_staging.get(), current_frame, to_read)) == 0) { + if ((samples_read = read_unlocked (raw_staging.get(), current_sample, to_read)) == 0) { error << string_compose(_("AudioSource[%1]: peak read - cannot read %2 samples at offset %3 of %4 (%5)"), - _name, to_read, current_frame, _length, strerror (errno)) + _name, to_read, current_sample, _length, strerror (errno)) << endmsg; return -1; } @@ -688,8 +688,8 @@ AudioSource::read_peaks_with_fpp (PeakData *peaks, framecnt_t npeaks, framepos_t xmax = max (xmax, raw_staging[i]); xmin = min (xmin, raw_staging[i]); ++i; - ++current_frame; - pixel_pos += pixels_per_frame; + ++current_sample; + pixel_pos += pixels_per_sample; if (pixel_pos >= next_pixel_pos) { @@ -715,7 +715,7 @@ AudioSource::read_peaks_with_fpp (PeakData *peaks, framecnt_t npeaks, framepos_t int AudioSource::build_peaks_from_scratch () { - const framecnt_t bufsize = 65536; // 256kB per disk read for mono data is about ideal + const samplecnt_t bufsize = 65536; // 256kB per disk read for mono data is about ideal DEBUG_TRACE (DEBUG::Peaks, "Building peaks from scratch\n"); @@ -730,18 +730,18 @@ AudioSource::build_peaks_from_scratch () goto out; } - framecnt_t current_frame = 0; - framecnt_t cnt = _length; + samplecnt_t current_sample = 0; + samplecnt_t cnt = _length; _peaks_built = false; boost::scoped_array buf(new Sample[bufsize]); while (cnt) { - framecnt_t frames_to_read = min (bufsize, cnt); - framecnt_t frames_read; + samplecnt_t samples_to_read = min (bufsize, cnt); + samplecnt_t samples_read; - if ((frames_read = read_unlocked (buf.get(), current_frame, frames_to_read)) != frames_to_read) { + if ((samples_read = read_unlocked (buf.get(), current_sample, samples_to_read)) != samples_to_read) { error << string_compose(_("%1: could not write read raw data for peak computation (%2)"), _name, strerror (errno)) << endmsg; done_with_peakfile_writes (false); goto out; @@ -756,12 +756,12 @@ AudioSource::build_peaks_from_scratch () goto out; } - if (compute_and_write_peaks (buf.get(), current_frame, frames_read, true, false, _FPP)) { + if (compute_and_write_peaks (buf.get(), current_sample, samples_read, true, false, _FPP)) { break; } - current_frame += frames_read; - cnt -= frames_read; + current_sample += samples_read; + cnt -= samples_read; lp.acquire(); } @@ -840,24 +840,24 @@ AudioSource::done_with_peakfile_writes (bool done) _peakfile_fd = -1; } -/** @param first_frame Offset from the source start of the first frame to +/** @param first_sample Offset from the source start of the first sample to * process. _lock MUST be held by caller. */ int -AudioSource::compute_and_write_peaks (Sample* buf, framecnt_t first_frame, framecnt_t cnt, +AudioSource::compute_and_write_peaks (Sample* buf, samplecnt_t first_sample, samplecnt_t cnt, bool force, bool intermediate_peaks_ready) { - return compute_and_write_peaks (buf, first_frame, cnt, force, intermediate_peaks_ready, _FPP); + return compute_and_write_peaks (buf, first_sample, cnt, force, intermediate_peaks_ready, _FPP); } int -AudioSource::compute_and_write_peaks (Sample* buf, framecnt_t first_frame, framecnt_t cnt, - bool force, bool intermediate_peaks_ready, framecnt_t fpp) +AudioSource::compute_and_write_peaks (Sample* buf, samplecnt_t first_sample, samplecnt_t cnt, + bool force, bool intermediate_peaks_ready, samplecnt_t fpp) { - framecnt_t to_do; + samplecnt_t to_do; uint32_t peaks_computed; - framepos_t current_frame; - framecnt_t frames_done; + samplepos_t current_sample; + samplecnt_t samples_done; const size_t blocksize = (128 * 1024); off_t first_peak_byte; boost::scoped_array buf2; @@ -871,7 +871,7 @@ AudioSource::compute_and_write_peaks (Sample* buf, framecnt_t first_frame, frame restart: if (peak_leftover_cnt) { - if (first_frame != peak_leftover_frame + peak_leftover_cnt) { + if (first_sample != peak_leftover_sample + peak_leftover_cnt) { /* uh-oh, ::seek() since the last ::compute_and_write_peaks(), and we have leftovers. flush a single peak (since the leftovers @@ -883,7 +883,7 @@ AudioSource::compute_and_write_peaks (Sample* buf, framecnt_t first_frame, frame x.min = peak_leftovers[0]; x.max = peak_leftovers[0]; - off_t byte = (peak_leftover_frame / fpp) * sizeof (PeakData); + off_t byte = (peak_leftover_sample / fpp) * sizeof (PeakData); off_t offset = lseek (_peakfile_fd, byte, SEEK_SET); @@ -901,7 +901,7 @@ AudioSource::compute_and_write_peaks (Sample* buf, framecnt_t first_frame, frame { Glib::Threads::Mutex::Lock lm (_peaks_ready_lock); - PeakRangeReady (peak_leftover_frame, peak_leftover_cnt); /* EMIT SIGNAL */ + PeakRangeReady (peak_leftover_sample, peak_leftover_cnt); /* EMIT SIGNAL */ if (intermediate_peaks_ready) { PeaksReady (); /* EMIT SIGNAL */ } @@ -936,7 +936,7 @@ AudioSource::compute_and_write_peaks (Sample* buf, framecnt_t first_frame, frame /* make sure that when we write into the peakfile, we startup where we left off */ - first_frame = peak_leftover_frame; + first_sample = peak_leftover_sample; } else { to_do = cnt; @@ -944,12 +944,12 @@ AudioSource::compute_and_write_peaks (Sample* buf, framecnt_t first_frame, frame boost::scoped_array peakbuf(new PeakData[(to_do/fpp)+1]); peaks_computed = 0; - current_frame = first_frame; - frames_done = 0; + current_sample = first_sample; + samples_done = 0; while (to_do) { - /* if some frames were passed in (i.e. we're not flushing leftovers) + /* if some samples were passed in (i.e. we're not flushing leftovers) and there are less than fpp to do, save them till next time */ @@ -964,14 +964,14 @@ AudioSource::compute_and_write_peaks (Sample* buf, framecnt_t first_frame, frame } memcpy (peak_leftovers, buf, to_do * sizeof (Sample)); peak_leftover_cnt = to_do; - peak_leftover_frame = current_frame; + peak_leftover_sample = current_sample; /* done for now */ break; } - framecnt_t this_time = min (fpp, to_do); + samplecnt_t this_time = min (fpp, to_do); peakbuf[peaks_computed].max = buf[0]; peakbuf[peaks_computed].min = buf[0]; @@ -981,11 +981,11 @@ AudioSource::compute_and_write_peaks (Sample* buf, framecnt_t first_frame, frame peaks_computed++; buf += this_time; to_do -= this_time; - frames_done += this_time; - current_frame += this_time; + samples_done += this_time; + current_sample += this_time; } - first_peak_byte = (first_frame / fpp) * sizeof (PeakData); + first_peak_byte = (first_sample / fpp) * sizeof (PeakData); if (can_truncate_peaks()) { @@ -1025,9 +1025,9 @@ AudioSource::compute_and_write_peaks (Sample* buf, framecnt_t first_frame, frame _peak_byte_max = max (_peak_byte_max, (off_t) (first_peak_byte + bytes_to_write)); - if (frames_done) { + if (samples_done) { Glib::Threads::Mutex::Lock lm (_peaks_ready_lock); - PeakRangeReady (first_frame, frames_done); /* EMIT SIGNAL */ + PeakRangeReady (first_sample, samples_done); /* EMIT SIGNAL */ if (intermediate_peaks_ready) { PeaksReady (); /* EMIT SIGNAL */ } @@ -1058,7 +1058,7 @@ AudioSource::truncate_peakfile () } } -framecnt_t +samplecnt_t AudioSource::available_peaks (double zoom_factor) const { if (zoom_factor < _FPP) { @@ -1087,7 +1087,7 @@ AudioSource::mark_streaming_write_completed (const Lock& lock) } void -AudioSource::allocate_working_buffers (framecnt_t framerate) +AudioSource::allocate_working_buffers (samplecnt_t framerate) { Glib::Threads::Mutex::Lock lm (_level_buffer_lock); @@ -1104,18 +1104,18 @@ AudioSource::allocate_working_buffers (framecnt_t framerate) } void -AudioSource::ensure_buffers_for_level (uint32_t level, framecnt_t frame_rate) +AudioSource::ensure_buffers_for_level (uint32_t level, samplecnt_t sample_rate) { Glib::Threads::Mutex::Lock lm (_level_buffer_lock); - ensure_buffers_for_level_locked (level, frame_rate); + ensure_buffers_for_level_locked (level, sample_rate); } void -AudioSource::ensure_buffers_for_level_locked (uint32_t level, framecnt_t frame_rate) +AudioSource::ensure_buffers_for_level_locked (uint32_t level, samplecnt_t sample_rate) { - framecnt_t nframes = (framecnt_t) floor (Config->get_audio_playback_buffer_seconds() * frame_rate); + samplecnt_t nframes = (samplecnt_t) floor (Config->get_audio_playback_buffer_seconds() * sample_rate); - /* this may be called because either "level" or "frame_rate" have + /* this may be called because either "level" or "sample_rate" have * changed. and it may be called with "level" smaller than the current * number of buffers, because a new compound region has been created at * a more shallow level than the deepest one we currently have. -- cgit v1.2.3