Mercurial > octave-nkf
changeset 19589:9bd2f0a07c95
improve performance of audio callback functions
* audiodevinfo.cc (octave_play_callback, portaudio_play_callback):
Avoid unnecessary copying of data.
| author | John W. Eaton <jwe@octave.org> |
|---|---|
| date | Fri, 09 Jan 2015 13:19:50 -0500 |
| parents | 398e42431dcc |
| children | d7057df2968e |
| files | libinterp/dldfcn/audiodevinfo.cc |
| diffstat | 1 files changed, 48 insertions(+), 70 deletions(-) [+] |
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--- a/libinterp/dldfcn/audiodevinfo.cc Fri Jan 09 13:17:51 2015 -0500 +++ b/libinterp/dldfcn/audiodevinfo.cc Fri Jan 09 13:19:50 2015 -0500 @@ -548,7 +548,7 @@ return paAbort; } - Matrix sound = retval(0).matrix_value (); + const Matrix sound = retval(0).matrix_value (); int return_status = retval(1).int_value (); if (error_state || frames != sound.rows () @@ -558,32 +558,13 @@ return paAbort; } - double scale_factor = 1.0; - - switch (player->get_nbits ()) - { - 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; - } - - 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 (); + // Don't multiply the audio data by scale_factor here. Although it + // does move the operation outside of the loops below, it also causes + // a second copy of the data array to be made. + + const ColumnVector sound_l = sound.column (0); + const ColumnVector sound_r = (sound.columns () == 1) + ? sound_l : sound.column (1); const double *p_l = sound_l.data (); const double *p_r = sound_r.data (); @@ -592,31 +573,37 @@ { case 8: { + static double scale_factor = pow (2.0, 7) - 1.0; + 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]; + buffer[2*i] = p_l[i] * scale_factor; + buffer[2*i+1] = p_r[i] * scale_factor; } } break; case 16: { + static double scale_factor = pow (2.0, 15) - 1.0; + 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]; + buffer[2*i] = p_l[i] * scale_factor; + buffer[2*i+1] = p_r[i] * scale_factor; } } break; case 24: { - int big_endian = is_big_endian (); + static double scale_factor = pow (2.0, 23) - 1.0; + + static int big_endian = is_big_endian (); uint8_t *buffer = static_cast<uint8_t *> (output); @@ -634,13 +621,13 @@ 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]; + buffer[offset+0] = _sample_l[0+big_endian] * scale_factor; + buffer[offset+1] = _sample_l[1+big_endian] * scale_factor; + buffer[offset+2] = _sample_l[2+big_endian] * scale_factor; + + buffer[offset+3] = _sample_r[0+big_endian] * scale_factor; + buffer[offset+4] = _sample_r[1+big_endian] * scale_factor; + buffer[offset+5] = _sample_r[2+big_endian] * scale_factor; } } break; @@ -666,29 +653,14 @@ return paAbort; } - double scale_factor = 1.0; - - switch (player->get_nbits ()) - { - 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; + // Don't multiply the audio data by scale_factor here. Although it + // would move the operation outside of the loops below, it also causes + // a second copy of the *entire* data array to be made when only a + // small portion (BUFFER_SIZE elements) is usually needed for this + // callback. + + const RowVector sound_l = player->get_left (); + const RowVector sound_r = player->get_right (); const double *pl = sound_l.data (); const double *pr = sound_l.data (); @@ -699,6 +671,8 @@ { case 8: { + static double scale_factor = pow (2.0, 7) - 1.0; + int8_t *buffer = static_cast<int8_t *> (output); for (unsigned long j = 0; j < frames; j++) @@ -710,8 +684,8 @@ unsigned long offset = j * 2; - buffer[offset+0] = pl[sample_number]; - buffer[offset+1] = pr[sample_number]; + buffer[offset+0] = pl[sample_number] * scale_factor; + buffer[offset+1] = pr[sample_number] * scale_factor; player->set_sample_number (sample_number + 1); } @@ -720,6 +694,8 @@ case 16: { + static double scale_factor = pow (2.0, 15) - 1.0; + int16_t *buffer = static_cast<int16_t *> (output); for (unsigned long j = 0; j < frames; j++) @@ -731,8 +707,8 @@ unsigned long offset = j * 2; - buffer[offset+0] = pl[sample_number]; - buffer[offset+1] = pr[sample_number]; + buffer[offset+0] = pl[sample_number] * scale_factor; + buffer[offset+1] = pr[sample_number] * scale_factor; player->set_sample_number (sample_number + 1); } @@ -741,6 +717,10 @@ case 24: { + static double scale_factor = pow (2.0, 23) - 1.0; + + static int big_endian = is_big_endian (); + uint8_t *buffer = static_cast<uint8_t *> (output); for (unsigned long j = 0; j < frames; j++) @@ -750,14 +730,12 @@ if (sample_number >= player->get_end_sample ()) return paComplete; - int32_t sample_l = pl[sample_number]; - int32_t sample_r = pr[sample_number]; + int32_t sample_l = pl[sample_number] * scale_factor; + int32_t sample_r = pr[sample_number] * scale_factor; 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);