Mercurial > octave-nkf
changeset 19578:827394ba8eb2
Backed out changeset 5f313345912f
Undo accidental changes from previous changeset.
| author | John W. Eaton <jwe@octave.org> |
|---|---|
| date | Wed, 07 Jan 2015 18:11:05 -0500 |
| parents | 5f313345912f |
| children | e2b570e7224b |
| files | libinterp/corefcn/load-path.cc libinterp/dldfcn/audiodevinfo.cc libinterp/dldfcn/audioread.cc |
| diffstat | 3 files changed, 128 insertions(+), 338 deletions(-) [+] |
line wrap: on
line diff
--- a/libinterp/corefcn/load-path.cc Wed Jan 07 18:07:12 2015 -0500 +++ b/libinterp/corefcn/load-path.cc Wed Jan 07 18:11:05 2015 -0500 @@ -1430,7 +1430,7 @@ size_t dir_len = dir.length (); - if (dname_len > dir_len + if (dname_len >= dir_len && file_ops::is_dir_sep (dname[dname_len - dir_len - 1]) && dir.compare (dname.substr (dname_len - dir_len)) == 0) {
--- a/libinterp/dldfcn/audiodevinfo.cc Wed Jan 07 18:07:12 2015 -0500 +++ b/libinterp/dldfcn/audiodevinfo.cc Wed Jan 07 18:11:05 2015 -0500 @@ -458,8 +458,8 @@ void set_y (octave_function *fn); void set_y (std::string fn); Matrix& get_y (void); - RowVector get_left (void) const; - RowVector get_right (void) const; + RowVector *get_left (void); + RowVector *get_right (void); void set_fs (int fs); int get_fs (void); void set_nbits (int nbits); @@ -510,7 +510,7 @@ DECLARE_OV_TYPEID_FUNCTIONS_AND_DATA }; -#define BUFFER_SIZE 8192 +#define BUFFER_SIZE 512 DEFINE_OCTAVE_ALLOCATOR (audioplayer); DEFINE_OV_TYPEID_FUNCTIONS_AND_DATA (audioplayer, "audioplayer", "audioplayer"); @@ -532,125 +532,66 @@ PaStreamCallbackFlags, void *data) { audioplayer *player = static_cast<audioplayer *> (data); - - if (! player) - { - error ("audio player callback function called without player"); - return paAbort; - } - - octave_value_list retval = feval (player->octave_callback_function, - ovl (static_cast<double> (frames)), 1); - - if (error_state || retval.length () < 2) - { - error ("audio player callback function failed"); - return paAbort; - } - + int big_endian = is_big_endian (); + octave_value_list args, retval; + args(0) = frames; + retval = feval (player->octave_callback_function, args, 1); + RowVector sound_l, sound_r; Matrix sound = retval(0).matrix_value (); int return_status = retval(1).int_value (); - - if (error_state || frames != sound.rows () - || sound.columns () < 1 || sound.columns () > 2) - { - error ("audio player callback function failed"); - return paAbort; - } - - double scale_factor = 1.0; - - switch (player->get_nbits ()) + sound_l.resize (frames); + sound_r.resize (frames); + if (sound.cols () == 1) { - case 8: - scale_factor = pow (2.0, 7) - 1.0; - break; - - case 16: - scale_factor = pow (2.0, 15) - 1.0; - break; - - case 24: - scale_factor = pow (2.0, 23) - 1.0; - break; - - default: - error ("invalid player bit depth in callback function"); - break; + for (unsigned long i = 0; i < frames; i++) + { + sound_l(i) = sound(i, 0); + sound_r(i) = sound(i, 0); + } } - - sound = sound * scale_factor; - - const RowVector sound_l = (sound.column (0)).transpose (); - - const RowVector sound_r = (sound.columns () == 1) - ? sound_l : (sound.column (1)).transpose (); - - const double *p_l = sound_l.data (); - const double *p_r = sound_r.data (); - - switch (player->get_nbits ()) + else if (sound.cols () == 2) + { + for (unsigned long i = 0; i < frames; i++) + { + sound_l(i) = sound(i, 0); + sound_r(i) = sound(i, 1); + } + } + else + return paAbort; + + for (unsigned long i = 0; i < frames; i++) { - case 8: - { - int8_t *buffer = static_cast<int8_t *> (output); - - for (unsigned long i = 0; i < frames; i++) - { - buffer[2*i] = p_l[i]; - buffer[2*i+1] = p_r[i]; - } - } - break; - - case 16: - { - int16_t *buffer = static_cast<int16_t *> (output); - - for (unsigned long i = 0; i < frames; i++) - { - buffer[2*i] = p_l[i]; - buffer[2*i+1] = p_r[i]; - } - } - break; - - case 24: - { - int big_endian = is_big_endian (); - - uint8_t *buffer = static_cast<uint8_t *> (output); - - for (unsigned long i = 0; i < frames; i++) - { - int32_t sample_l = p_l[i]; - int32_t sample_r = p_r[i]; - - sample_l &= 0x00ffffff; - sample_r &= 0x00ffffff; - - // FIXME: Would a mask work better? - uint8_t *_sample_l = reinterpret_cast<uint8_t *> (&sample_l); - uint8_t *_sample_r = reinterpret_cast<uint8_t *> (&sample_r); - - unsigned long offset = i * 6; - - buffer[offset+0] = _sample_l[0+big_endian]; - buffer[offset+1] = _sample_l[1+big_endian]; - buffer[offset+2] = _sample_l[2+big_endian]; - - buffer[offset+3] = _sample_r[0+big_endian]; - buffer[offset+4] = _sample_r[1+big_endian]; - buffer[offset+5] = _sample_r[2+big_endian]; - } - } - break; - - default: - error ("invalid player bit depth in callback function"); - break; + if (player->get_nbits () == 8) + { + int8_t *buffer = static_cast<int8_t *> (output); + buffer[2 * i] = sound_l.elem (i) * (pow (2.0, 7) - 1); + buffer[2 * i + 1] = sound_r.elem (i) * (pow (2.0, 7) - 1); + } + else if (player->get_nbits () == 16) + { + int16_t *buffer = static_cast<int16_t *> (output); + buffer[2 * i] = sound_l.elem (i) * (pow (2.0, 15) - 1); + buffer[2 * i + 1] = sound_r.elem (i) * (pow (2.0, 15) - 1); + } + else if (player->get_nbits () == 24) + { + uint8_t *buffer = static_cast<uint8_t *> (output); + int32_t sample_l = sound_l.elem (i) * (pow (2.0, 23) - 1); + int32_t sample_r = sound_r.elem (i) * (pow (2.0, 23) - 1); + sample_l &= 0x00ffffff; + sample_r &= 0x00ffffff; + // FIXME: Would a mask work better? + uint8_t *_sample_l = reinterpret_cast<uint8_t *> (&sample_l); + uint8_t *_sample_r = reinterpret_cast<uint8_t *> (&sample_r); + buffer[i * 6 + 0] = _sample_l[0 + big_endian]; + buffer[i * 6 + 1] = _sample_l[1 + big_endian]; + buffer[i * 6 + 2] = _sample_l[2 + big_endian]; + buffer[i * 6 + 3] = _sample_r[0 + big_endian]; + buffer[i * 6 + 4] = _sample_r[1 + big_endian]; + buffer[i * 6 + 5] = _sample_r[2 + big_endian]; + } } - return return_status; } @@ -660,187 +601,74 @@ PaStreamCallbackFlags, void *data) { audioplayer *player = static_cast<audioplayer *> (data); - - if (! player) - { - error ("audio player callback function called without player"); - return paAbort; - } - - double scale_factor = 1.0; - - switch (player->get_nbits ()) + int big_endian = is_big_endian (); + int channels = player->get_channels (); + RowVector *sound_l = player->get_left (); + RowVector *sound_r; + + if (channels > 1) + sound_r = player->get_right (); + else + sound_r = sound_l; + + for (unsigned long j = 0, k = 0; j < frames; j++, k += 2) { - case 8: - scale_factor = pow (2.0, 7) - 1.0; - break; - - case 16: - scale_factor = pow (2.0, 15) - 1.0; - break; - - case 24: - scale_factor = pow (2.0, 23) - 1.0; - break; - - default: - error ("invalid player bit depth in callback function"); - break; - } - - const RowVector sound_l = player->get_left () * scale_factor; - const RowVector sound_r = player->get_right () * scale_factor; - - const double *pl = sound_l.data (); - const double *pr = sound_l.data (); - - if (player->get_type () == DOUBLE) - { - switch (player->get_nbits ()) + unsigned int sample_number = player->get_sample_number (); + if (sample_number > player->get_end_sample ()) + return paAbort; + + if (player->get_type () == DOUBLE) { - case 8: - { - int8_t *buffer = static_cast<int8_t *> (output); - - for (unsigned long j = 0; j < frames; j++) - { - unsigned int sample_number = player->get_sample_number (); - - if (sample_number >= player->get_end_sample ()) - return paComplete; - - unsigned long offset = j * 2; - - buffer[offset+0] = pl[sample_number]; - buffer[offset+1] = pr[sample_number]; - - player->set_sample_number (sample_number + 1); - } - } - break; - - case 16: - { - int16_t *buffer = static_cast<int16_t *> (output); - - for (unsigned long j = 0; j < frames; j++) - { - unsigned int sample_number = player->get_sample_number (); - - if (sample_number >= player->get_end_sample ()) - return paComplete; - - unsigned long offset = j * 2; - - buffer[offset+0] = pl[sample_number]; - buffer[offset+1] = pr[sample_number]; - - player->set_sample_number (sample_number + 1); - } - } - break; - - case 24: - { - uint8_t *buffer = static_cast<uint8_t *> (output); - - for (unsigned long j = 0; j < frames; j++) - { - unsigned int sample_number = player->get_sample_number (); - - if (sample_number >= player->get_end_sample ()) - return paComplete; - - int32_t sample_l = pl[sample_number]; - int32_t sample_r = pr[sample_number]; - - sample_l &= 0x00ffffff; - sample_r &= 0x00ffffff; - - int big_endian = is_big_endian (); - - // FIXME: Would a mask work better? - uint8_t *_sample_l = reinterpret_cast<uint8_t *> (&sample_l); - uint8_t *_sample_r = reinterpret_cast<uint8_t *> (&sample_r); - - unsigned long offset = j * 6; - - buffer[offset+0] = _sample_l[0+big_endian]; - buffer[offset+1] = _sample_l[1+big_endian]; - buffer[offset+2] = _sample_l[2+big_endian]; - - buffer[offset+3] = _sample_r[0+big_endian]; - buffer[offset+4] = _sample_r[1+big_endian]; - buffer[offset+5] = _sample_r[2+big_endian]; - - player->set_sample_number (sample_number + 1); - } - } - break; - - default: - error ("invalid player bit depth in callback function"); - break; + if (player->get_nbits () == 8) + { + int8_t *buffer = static_cast<int8_t *> (output); + buffer[k] = sound_l->elem (sample_number) * (pow (2.0, 7) - 1); + buffer[k + 1] = sound_r->elem (sample_number) * (pow (2.0, 7) - 1); + } + else if (player->get_nbits () == 16) + { + int16_t *buffer = static_cast<int16_t *> (output); + buffer[k] = sound_l->elem (sample_number) * (pow (2.0, 15) - 1); + buffer[k + 1] = sound_r->elem (sample_number) * (pow (2.0, 15) - 1); + } + else if (player->get_nbits () == 24) + { + uint8_t *buffer = static_cast<uint8_t *> (output); + int32_t sample_l = sound_l->elem (sample_number) * (pow (2.0, 23) - 1); + int32_t sample_r = sound_r->elem (sample_number) * (pow (2.0, 23) - 1); + sample_l &= 0x00ffffff; + sample_r &= 0x00ffffff; + // FIXME: Would a mask work better? + uint8_t *_sample_l = reinterpret_cast<uint8_t *> (&sample_l); + uint8_t *_sample_r = reinterpret_cast<uint8_t *> (&sample_r); + buffer[j * 6 + 0] = _sample_l[0 + big_endian]; + buffer[j * 6 + 1] = _sample_l[1 + big_endian]; + buffer[j * 6 + 2] = _sample_l[2 + big_endian]; + buffer[j * 6 + 3] = _sample_r[0 + big_endian]; + buffer[j * 6 + 4] = _sample_r[1 + big_endian]; + buffer[j * 6 + 5] = _sample_r[2 + big_endian]; + } } - } - else if (player->get_type () == INT8) - { - int8_t *buffer = static_cast<int8_t *> (output); - - for (unsigned long j = 0; j < frames; j++) + else if (player->get_type () == INT8) { - unsigned int sample_number = player->get_sample_number (); - - if (sample_number >= player->get_end_sample ()) - return paComplete; - - unsigned long offset = j * 2; - - buffer[offset+0] = pl[sample_number]; - buffer[offset+1] = pr[sample_number]; - - player->set_sample_number (sample_number + 1); + int8_t *buffer = static_cast<int8_t *> (output); + buffer[k] = sound_l->elem (sample_number); + buffer[k + 1] = sound_r->elem (sample_number); } - } - else if (player->get_type () == UINT8) - { - uint8_t *buffer = static_cast<uint8_t *> (output); - - for (unsigned long j = 0; j < frames; j++) + else if (player->get_type () == UINT8) { - unsigned int sample_number = player->get_sample_number (); - - if (sample_number >= player->get_end_sample ()) - return paComplete; - - unsigned long offset = j * 2; - - buffer[offset+0] = pl[sample_number]; - buffer[offset+1] = pr[sample_number]; - - player->set_sample_number (sample_number + 1); + uint8_t *buffer = static_cast<uint8_t *> (output); + buffer[k] = sound_l->elem (sample_number); + buffer[k + 1] = sound_r->elem (sample_number); } + else if (player->get_type () == INT16) + { + int16_t *buffer = static_cast<int16_t *> (output); + buffer[k] = sound_l->elem (sample_number); + buffer[k + 1] = sound_r->elem (sample_number); + } + player->set_sample_number (sample_number + 1); } - else if (player->get_type () == INT16) - { - int16_t *buffer = static_cast<int16_t *> (output); - - for (unsigned long j = 0; j < frames; j++) - { - unsigned int sample_number = player->get_sample_number (); - - if (sample_number >= player->get_end_sample ()) - return paComplete; - - unsigned long offset = j * 2; - - buffer[offset+0] = pl[sample_number]; - buffer[offset+1] = pr[sample_number]; - - player->set_sample_number (sample_number + 1); - } - } - return paContinue; } @@ -973,16 +801,16 @@ return y; } -RowVector -audioplayer::get_left (void) const +RowVector * +audioplayer::get_left (void) { - return left; + return &(left); } -RowVector -audioplayer::get_right (void) const +RowVector * +audioplayer::get_right (void) { - return channels == 1 ? left : right; + return &(right); } void @@ -2350,12 +2178,6 @@ int nargin = args.length (); - if (nargin < 2) - { - print_usage (); - return retval; - } - audioplayer* recorder = new audioplayer (); bool is_function = args(0).is_string () || args(0).is_function_handle () || args(0).is_inline_function (); @@ -2372,7 +2194,6 @@ case 3: recorder->set_nbits (args(2).int_value ()); break; - case 4: recorder->set_nbits (args(2).int_value ()); recorder->set_id (args(3).int_value ());
--- a/libinterp/dldfcn/audioread.cc Wed Jan 07 18:07:12 2015 -0500 +++ b/libinterp/dldfcn/audioread.cc Wed Jan 07 18:11:05 2015 -0500 @@ -24,8 +24,6 @@ #include <config.h> #endif -#define BOUNDS_CHECKING 1 - #include <string> #include <map> @@ -249,19 +247,14 @@ SNDFILE *file; SF_INFO info; - memset (&info, 0, sizeof (info)); - OCTAVE_LOCAL_BUFFER (float, data, audio.rows () * audio.cols ()); - size_t idx = 0; - for (octave_idx_type j = 0; j < audio.rows (); j++) + for (int i = 0; i < audio.cols (); i++) { - for (octave_idx_type i = 0; i < audio.cols (); i++) - data[idx++] = audio(j, i); + for (int j = 0; j < audio.rows (); j++) + data[j * audio.cols () + i] = audio(j, i); } - std::cerr << "idx: " << idx << std::endl; - if (extension == "ogg") info.format = SF_FORMAT_VORBIS; else @@ -315,14 +308,6 @@ info.channels = audio.cols (); info.format |= extension_to_format[extension]; - std::cerr << extension << std::endl; - std::cerr << extension_to_format[extension] << std::endl; - std::cerr << info.format << std::endl; - std::cerr << SF_FORMAT_OGG << std::endl; - std::cerr << SF_FORMAT_VORBIS << std::endl; - - // info.format = SF_FORMAT_OGG; - file = sf_open (filename.c_str (), SFM_WRITE, &info); if (title != "") @@ -334,24 +319,9 @@ if (comment != "") sf_set_string (file, SF_STR_COMMENT, comment.c_str ()); - sf_count_t items_to_write = audio.rows () * audio.cols (); - - memset (&info, 0, sizeof (info)) ; - info.format = SF_FORMAT_OGG | SF_FORMAT_VORBIS ; - info.channels = 2 ; - info.samplerate = 44100 ; - - sf_count_t items_written = sf_write_float (file, data, items_to_write); - + sf_write_float (file, data, audio.rows () * audio.cols ()); sf_close (file); - if (items_written != items_to_write) - { - error ("expected to write %ld items, wrote %ld items", - items_to_write, items_written); - return retval; - } - #else error ("sndfile not found on your system and thus audiowrite is not functional"); @@ -388,7 +358,6 @@ result.assign ("NumChannels", info.channels); result.assign ("SampleRate", info.samplerate); result.assign ("TotalSamples", info.frames); - result.assign ("Format", info.format); double dframes = info.frames; double drate = info.samplerate;