diff options
Diffstat (limited to 'src/xine-engine/audio_out.c')
| -rw-r--r-- | src/xine-engine/audio_out.c | 110 |
1 files changed, 100 insertions, 10 deletions
diff --git a/src/xine-engine/audio_out.c b/src/xine-engine/audio_out.c index 19273fdfd..24fff6929 100644 --- a/src/xine-engine/audio_out.c +++ b/src/xine-engine/audio_out.c @@ -17,10 +17,12 @@ * along with self program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA * - * $Id: audio_out.c,v 1.3 2001/08/21 19:39:50 jcdutton Exp $ + * $Id: audio_out.c,v 1.4 2001/08/22 10:51:05 jcdutton Exp $ * + * 22-8-2001 James imported some useful AC3 sections from the previous alsa driver. + * (c) 2001 Andy Lo A Foe <andy@alsaplayer.org> * 20-8-2001 First implementation of Audio sync and Audio driver separation. - * Copyright (C) 2001 James Courtier-Dutton James@superbug.demon.co.uk + * (c) 2001 James Courtier-Dutton James@superbug.demon.co.uk * * General Programming Guidelines: - * New concept of an "audio_frame". @@ -32,6 +34,11 @@ * when dealing with audio_bytes instead of audio_frames. * * The number of samples passed to/from the audio driver is also sent in units of audio_frames. + * + * Currently, James has tested with OSS: Standard stereo out, SPDIF PCM, SPDIF AC3 + * ALSA: Standard stereo out + * No testing has been done of ALSA SPDIF AC3 or any 4,5,5.1 channel output. + * Currently, I don't think resampling functions, as I cannot test it. */ /* required for swab() */ @@ -100,6 +107,55 @@ #define DSP_TEMPLATE "/dev/dsp%d" #endif +struct frmsize_s +{ + uint16_t bit_rate; + uint16_t frm_size[3]; +}; + + +static const struct frmsize_s frmsizecod_tbl[64] = +{ + { 32 ,{64 ,69 ,96 } }, + { 32 ,{64 ,70 ,96 } }, + { 40 ,{80 ,87 ,120 } }, + { 40 ,{80 ,88 ,120 } }, + { 48 ,{96 ,104 ,144 } }, + { 48 ,{96 ,105 ,144 } }, + { 56 ,{112 ,121 ,168 } }, + { 56 ,{112 ,122 ,168 } }, + { 64 ,{128 ,139 ,192 } }, + { 64 ,{128 ,140 ,192 } }, + { 80 ,{160 ,174 ,240 } }, + { 80 ,{160 ,175 ,240 } }, + { 96 ,{192 ,208 ,288 } }, + { 96 ,{192 ,209 ,288 } }, + { 112 ,{224 ,243 ,336 } }, + { 112 ,{224 ,244 ,336 } }, + { 128 ,{256 ,278 ,384 } }, + { 128 ,{256 ,279 ,384 } }, + { 160 ,{320 ,348 ,480 } }, + { 160 ,{320 ,349 ,480 } }, + { 192 ,{384 ,417 ,576 } }, + { 192 ,{384 ,418 ,576 } }, + { 224 ,{448 ,487 ,672 } }, + { 224 ,{448 ,488 ,672 } }, + { 256 ,{512 ,557 ,768 } }, + { 256 ,{512 ,558 ,768 } }, + { 320 ,{640 ,696 ,960 } }, + { 320 ,{640 ,697 ,960 } }, + { 384 ,{768 ,835 ,1152 } }, + { 384 ,{768 ,836 ,1152 } }, + { 448 ,{896 ,975 ,1344 } }, + { 448 ,{896 ,976 ,1344 } }, + { 512 ,{1024 ,1114 ,1536 } }, + { 512 ,{1024 ,1115 ,1536 } }, + { 576 ,{1152 ,1253 ,1728 } }, + { 576 ,{1152 ,1254 ,1728 } }, + { 640 ,{1280 ,1393 ,1920 } }, + { 640 ,{1280 ,1394 ,1920 } } +}; + /* * open the audio device for writing to */ @@ -111,17 +167,17 @@ static int ao_open(ao_instance_t *self, printf("open failed!\n"); return -1; }; -// self->frame_rate_factor = (double) self->output_frame_rate / (double) self->input_frame_rate; - self->mode = mode; + self->mode = mode; self->input_frame_rate = rate; - self->frames_in_buffer = 0; - self->audio_started = 0; - self->last_audio_vpts = 0; + self->frames_in_buffer = 0; + self->audio_started = 0; + self->last_audio_vpts = 0; + self->do_resample = 0; /* Resampling currently not working. */ self->output_frame_rate=rate; self->num_channels = self->driver->num_channels(self->driver); - self->frame_rate_factor = (double) 1 / (double) 1; + self->frame_rate_factor = (double) self->output_frame_rate / (double) self->input_frame_rate; /* Alway produces 1 at the moment */ self->audio_step = (uint32_t) 90000 * (uint32_t) 32768 / self->input_frame_rate; self->frames_per_kpts = self->output_frame_rate * self->num_channels * 2 * 1024 / 90000; @@ -160,7 +216,37 @@ static void ao_fill_gap (ao_instance_t *self, uint32_t pts_len) { } } +/* + * This routine is currently not used, but I do not want to loose it. + * I think "(c) 2001 Andy Lo A Foe <andy@alsaplayer.org>" originally added it + * to ./xine-lib/src/audio_out/audio_alsa_out.c before the architecture changes + * So it has moved to here. + */ + +void write_pause_burst(alsa_instance_t *this,int error) +{ +#define BURST_SIZE 6144 + + unsigned char buf[8192]; + unsigned short *sbuf = (unsigned short *)&buf[0]; + + sbuf[0] = 0xf872; + sbuf[1] = 0x4e1f; + if (error == 0) + // Audio ES Channel empty, wait for DD Decoder or pause + sbuf[2] = 0x0003; + else + // user stop, skip or error + sbuf[2] = 0x0103; + + sbuf[3] = 0x0020; + sbuf[4] = 0x0000; + sbuf[5] = 0x0000; + + memset(&sbuf[6], 0, BURST_SIZE - 96); + self->driver->write(self->driver, u_char * sbuf, BURST_SIZE / 4); +} static int ao_write(ao_instance_t *self, int16_t* output_frames, uint32_t num_frames, @@ -276,8 +362,12 @@ static int ao_write(ao_instance_t *self, self->frame_buffer[0] = 0xf872; //spdif syncword self->frame_buffer[1] = 0x4e1f; // ............. self->frame_buffer[2] = 0x0001; // AC3 data - self->frame_buffer[3] = num_frames * 8; - self->frame_buffer[4] = 0x0b77; // AC3 syncwork already in output_frames + + data = (uint8_t *)&output_samples[1]; // skip AC3 sync + fscod = (data[2] >> 6) & 0x3; + frmsizecod = data[2] & 0x3f; + frame_size = frmsizecod_tbl[frmsizecod].frm_size[fscod] << 4; + sample_buffer[3] = frame_size; // ac3 seems to be swabbed data swab(output_frames,self->frame_buffer+4, num_frames ); |