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totally redesign import code to avoid stupid intermediate file copy [beta]

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git-svn-id: svn://localhost/ardour2/trunk@1520 d708f5d6-7413-0410-9779-e7cbd77b26cf
This commit is contained in:
Paul Davis
2007-02-27 03:59:37 +00:00
parent 691b0c10ac
commit 2e4fc4c3f1
1 changed files with 118 additions and 131 deletions
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249
libs/ardour/import.cc
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@@ -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] ;
src_data.input_frames = ImportableSource::read (input, 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;
/* The last read will not be a full buffer, so set end_of_input. */
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;
if ((nframes_t) src_data.input_frames < BLOCKSIZE) {
src_data.end_of_input = SF_TRUE ;
}
src_data.input_frames /= sf_info->channels;
src_data.data_in = input ;
}
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) ;
src_data.data_out = output;
/* 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. */
/* Start with zero to force load in while loop. */
src_data.input_frames = 0 ;
src_data.data_in = 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 ;
}
/* 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.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;
}
src_state = src_delete (src_state) ;
sf_close(in);
sf_close(out);
if (status.cancel) {
return false;
if (!src_data.end_of_input) {
src_data.output_frames = nframes / sf_info->channels ;
} else {
return true ;
src_data.output_frames = src_data.input_frames;
}
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;
}