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authorPaul Davis <paul@linuxaudiosystems.com>2007-02-27 03:59:37 +0000
committerPaul Davis <paul@linuxaudiosystems.com>2007-02-27 03:59:37 +0000
commit2e4fc4c3f13d192c774f38fd27561f728eee68de (patch)
tree7720446b670a164e877dbe37c105db7abbcb49e1 /libs
parent691b0c10ac8089efa015e88513d309768fe0f44b (diff)
totally redesign import code to avoid stupid intermediate file copy [beta]
git-svn-id: svn://localhost/ardour2/trunk@1520 d708f5d6-7413-0410-9779-e7cbd77b26cf
Diffstat (limited to 'libs')
-rw-r--r--libs/ardour/import.cc247
1 files changed, 117 insertions, 130 deletions
diff --git a/libs/ardour/import.cc b/libs/ardour/import.cc
index 5fbd4e0750..e6d3c5ba5e 100644
--- a/libs/ardour/import.cc
+++ b/libs/ardour/import.cc
@@ -50,6 +50,63 @@ using namespace PBD;
#define BLOCKSIZE 4096U
+class ImportableSource {
+ public:
+ ImportableSource (SNDFILE* sf, SF_INFO* info) : in (sf), sf_info (info) {}
+ virtual ~ImportableSource() {}
+
+ virtual nframes_t read (Sample* buffer, nframes_t nframes) {
+ nframes_t per_channel = nframes / sf_info->channels;
+ per_channel = sf_readf_float (in, buffer, per_channel);
+ return per_channel * sf_info->channels;
+ }
+
+ virtual float ratio() const { return 1.0f; }
+
+protected:
+ SNDFILE* in;
+ SF_INFO* sf_info;
+};
+
+class ResampledImportableSource : public ImportableSource {
+ public:
+ ResampledImportableSource (SNDFILE* sf, SF_INFO* info, nframes_t rate) : ImportableSource (sf, info) {
+
+ sf_seek (in, 0, SEEK_SET) ;
+
+ /* Initialize the sample rate converter. */
+
+ if ((src_state = src_new (SRC_SINC_BEST_QUALITY, sf_info->channels, &err)) == 0) {
+ error << string_compose(_("Import: src_new() failed : %1"), src_strerror (err)) << endmsg ;
+ throw failed_constructor ();
+ }
+
+ src_data.end_of_input = 0 ; /* Set this later. */
+
+ /* Start with zero to force load in while loop. */
+
+ src_data.input_frames = 0 ;
+ src_data.data_in = input ;
+
+ src_data.src_ratio = ((float) rate) / sf_info->samplerate ;
+
+ }
+
+ ~ResampledImportableSource () {
+ src_state = src_delete (src_state) ;
+ }
+
+ nframes_t read (Sample* buffer, nframes_t nframes);
+
+ float ratio() const { return src_data.src_ratio; }
+
+ private:
+ float input[BLOCKSIZE];
+ SRC_STATE* src_state;
+ SRC_DATA src_data;
+ int err;
+};
+
int
Session::import_audiofile (import_status& status)
{
@@ -68,32 +125,20 @@ Session::import_audiofile (import_status& status)
int ret = -1;
vector<string> new_paths;
struct tm* now;
- string tmp_convert_file;
-
+ ImportableSource* importable = 0;
+ const nframes_t nframes = BLOCKSIZE;
+
status.new_regions.clear ();
if ((in = sf_open (status.paths.front().c_str(), SFM_READ, &info)) == 0) {
error << string_compose(_("Import: cannot open input sound file \"%1\""), status.paths.front()) << endmsg;
return -1;
+ }
+
+ if ((nframes_t) info.samplerate != frame_rate()) {
+ importable = new ResampledImportableSource (in, &info, frame_rate());
} else {
- if ((uint32_t) info.samplerate != frame_rate()) {
- sf_close(in);
- status.doing_what = _("resampling audio");
- // resample to session frame_rate
- if (sample_rate_convert(status, status.paths.front(), tmp_convert_file)) {
- if ((in = sf_open (tmp_convert_file.c_str(), SFM_READ, &info)) == 0) {
- error << string_compose(_("Import: cannot open converted sound file \"%1\""), tmp_convert_file) << endmsg;
- return -1;
- }
- } else if (!status.cancel){
- // error
- error << string_compose(_("Import: error while resampling sound file \"%1\""), status.paths.front()) << endmsg;
- return -1;
- } else {
- // canceled
- goto out;
- }
- }
+ importable = new ImportableSource (in, &info);
}
for (n = 0; n < info.channels; ++n) {
@@ -149,12 +194,11 @@ Session::import_audiofile (import_status& status)
nfiles++;
}
-
- data = new float[BLOCKSIZE * info.channels];
+ data = new float[nframes * info.channels];
channel_data = new Sample * [ info.channels ];
for (n = 0; n < info.channels; ++n) {
- channel_data[n] = new Sample[BLOCKSIZE];
+ channel_data[n] = new Sample[nframes];
}
so_far = 0;
@@ -164,18 +208,19 @@ Session::import_audiofile (import_status& status)
while (!status.cancel) {
- long nread;
+ nframes_t nread;
long x;
long chn;
-
- if ((nread = sf_readf_float (in, data, BLOCKSIZE)) == 0) {
+
+ if ((nread = importable->read (data, nframes)) == 0) {
break;
}
/* de-interleave */
for (chn = 0; chn < info.channels; ++chn) {
- for (x = chn, n = 0; n < nread; x += info.channels, ++n) {
+
+ for (x = chn, n = 0; n < nframes; x += info.channels, ++n) {
channel_data[chn][n] = (Sample) data[x];
}
}
@@ -183,11 +228,15 @@ Session::import_audiofile (import_status& status)
/* flush to disk */
for (chn = 0; chn < info.channels; ++chn) {
- newfiles[chn]->write (channel_data[chn], nread);
+ newfiles[chn]->write (channel_data[chn], nread / info.channels);
}
so_far += nread;
- status.progress = so_far / (float) (info.frames * info.channels);
+ status.progress = so_far / (importable->ratio () * info.frames * info.channels);
+ }
+
+ if (status.cancel) {
+ goto out;
}
if (status.multichan) {
@@ -202,12 +251,9 @@ Session::import_audiofile (import_status& status)
time (&xnow);
now = localtime (&xnow);
- if (status.cancel) {
- goto out;
- }
-
if (status.multichan) {
/* all sources are used in a single multichannel region */
+
for (n = 0; n < nfiles && !status.cancel; ++n) {
/* flush the final length to the header */
newfiles[n]->update_header(0, *now, xnow);
@@ -275,116 +321,57 @@ Session::import_audiofile (import_status& status)
}
}
- if (tmp_convert_file.length()) {
- unlink(tmp_convert_file.c_str());
+ if (importable) {
+ delete importable;
}
-
- sf_close (in);
+
+ sf_close (in);
status.done = true;
+
return ret;
}
-string
-Session::build_tmp_convert_name(string infile)
+nframes_t
+ResampledImportableSource::read (Sample* output, nframes_t nframes)
{
- string tmp_name(_path + "/." + Glib::path_get_basename (infile.c_str()) + "XXXXXX");
- char* tmp = new char[tmp_name.length() + 1];
- tmp_name.copy(tmp, string::npos);
- tmp[tmp_name.length()] = 0;
- mkstemp(tmp);
- string outfile = tmp;
- delete [] tmp;
+ /* If the input buffer is empty, refill it. */
- return outfile;
-}
+ if (src_data.input_frames == 0) {
-bool
-Session::sample_rate_convert (import_status& status, string infile, string& outfile)
-{
- float input [BLOCKSIZE] ;
- float output [BLOCKSIZE] ;
-
- SF_INFO sf_info;
- SRC_STATE* src_state ;
- SRC_DATA src_data ;
- int err ;
- sf_count_t output_count = 0 ;
- sf_count_t input_count = 0;
-
- SNDFILE* in = sf_open(infile.c_str(), SFM_READ, &sf_info);
- if (!in) {
- error << string_compose(_("Import/SRC: could not open input file: %1"), outfile) << endmsg;
- return false;
- }
- sf_count_t total_input_frames = sf_info.frames;
-
- outfile = build_tmp_convert_name(infile);
- SNDFILE* out = sf_open(outfile.c_str(), SFM_RDWR, &sf_info);
- if (!out) {
- error << string_compose(_("Import/SRC: could not open output file: %1"), outfile) << endmsg;
- return false;
- }
-
- sf_seek (in, 0, SEEK_SET) ;
- sf_seek (out, 0, SEEK_SET) ;
-
- /* Initialize the sample rate converter. */
- if ((src_state = src_new (SRC_SINC_BEST_QUALITY, sf_info.channels, &err)) == 0) {
- error << string_compose(_("Import: src_new() failed : %1"), src_strerror (err)) << endmsg ;
- return false ;
- }
-
- src_data.end_of_input = 0 ; /* Set this later. */
+ src_data.input_frames = ImportableSource::read (input, BLOCKSIZE);
- /* Start with zero to force load in while loop. */
- src_data.input_frames = 0 ;
- src_data.data_in = input ;
+ /* The last read will not be a full buffer, so set end_of_input. */
- src_data.src_ratio = (1.0 * frame_rate()) / sf_info.samplerate ;
-
- src_data.data_out = output ;
- src_data.output_frames = BLOCKSIZE / sf_info.channels ;
-
- while (!status.cancel) {
- /* If the input buffer is empty, refill it. */
- if (src_data.input_frames == 0) {
- src_data.input_frames = sf_readf_float (in, input, BLOCKSIZE / sf_info.channels) ;
- src_data.data_in = input ;
-
- /* The last read will not be a full buffer, so snd_of_input. */
- if (src_data.input_frames < (int)BLOCKSIZE / sf_info.channels) {
- src_data.end_of_input = SF_TRUE ;
- }
- }
-
- if ((err = src_process (src_state, &src_data))) {
- error << string_compose(_("Import: %1"), src_strerror (err)) << endmsg ;
- return false ;
- }
-
- /* Terminate if at end */
- if (src_data.end_of_input && src_data.output_frames_gen == 0) {
- break ;
- }
+ if ((nframes_t) src_data.input_frames < BLOCKSIZE) {
+ src_data.end_of_input = SF_TRUE ;
+ }
- /* Write output. */
- sf_writef_float (out, output, src_data.output_frames_gen) ;
- output_count += src_data.output_frames_gen ;
- input_count += src_data.input_frames_used;
+ src_data.input_frames /= sf_info->channels;
+ src_data.data_in = input ;
+ }
+
+ src_data.data_out = output;
- src_data.data_in += src_data.input_frames_used * sf_info.channels ;
- src_data.input_frames -= src_data.input_frames_used ;
-
- status.progress = (float) input_count / total_input_frames;
+ if (!src_data.end_of_input) {
+ src_data.output_frames = nframes / sf_info->channels ;
+ } else {
+ src_data.output_frames = src_data.input_frames;
}
- src_state = src_delete (src_state) ;
- sf_close(in);
- sf_close(out);
-
- if (status.cancel) {
- return false;
- } else {
- return true ;
+ if ((err = src_process (src_state, &src_data))) {
+ error << string_compose(_("Import: %1"), src_strerror (err)) << endmsg ;
+ return false ;
+ }
+
+ /* Terminate if at end */
+
+ if (src_data.end_of_input && src_data.output_frames_gen == 0) {
+ return 0;
}
+
+ src_data.data_in += src_data.input_frames_used * sf_info->channels ;
+ src_data.input_frames -= src_data.input_frames_used ;
+
+ return nframes;
}
+