diff options
Diffstat (limited to 'libao2/ao_sun.c')
| -rw-r--r-- | libao2/ao_sun.c | 692 |
1 files changed, 0 insertions, 692 deletions
diff --git a/libao2/ao_sun.c b/libao2/ao_sun.c deleted file mode 100644 index ecdb23d4af..0000000000 --- a/libao2/ao_sun.c +++ /dev/null @@ -1,692 +0,0 @@ -/* - * SUN audio output driver - * - * This file is part of MPlayer. - * - * MPlayer is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * MPlayer is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with MPlayer; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include <stdio.h> -#include <stdlib.h> -#include <string.h> - -#include <unistd.h> -#include <fcntl.h> -#include <errno.h> -#include <sys/ioctl.h> -#include <sys/time.h> -#include <sys/types.h> -#include <sys/stat.h> -#include <sys/audioio.h> -#ifdef AUDIO_SWFEATURE_MIXER /* solaris8 or newer? */ -# define HAVE_SYS_MIXER_H 1 -#endif -#if HAVE_SYS_MIXER_H -# include <sys/mixer.h> -#endif -#ifdef __svr4__ -#include <stropts.h> -#endif - -#include "config.h" -#include "mixer.h" - -#include "audio_out.h" -#include "audio_out_internal.h" -#include "libaf/af_format.h" -#include "mp_msg.h" - -static const ao_info_t info = -{ - "Sun audio output", - "sun", - "Juergen Keil", - "" -}; - -LIBAO_EXTERN(sun) - - -/* These defines are missing on NetBSD */ -#ifndef AUDIO_PRECISION_8 -#define AUDIO_PRECISION_8 8 -#define AUDIO_PRECISION_16 16 -#endif -#ifndef AUDIO_CHANNELS_MONO -#define AUDIO_CHANNELS_MONO 1 -#define AUDIO_CHANNELS_STEREO 2 -#endif - - -static char *sun_mixer_device = NULL; -static char *audio_dev = NULL; -static int queued_bursts = 0; -static int queued_samples = 0; -static int bytes_per_sample = 0; -static int byte_per_sec = 0; -static int audio_fd = -1; -static enum { - RTSC_UNKNOWN = 0, - RTSC_ENABLED, - RTSC_DISABLED -} enable_sample_timing; - - -static void flush_audio(int fd) { -#ifdef AUDIO_FLUSH - ioctl(fd, AUDIO_FLUSH, 0); -#elif defined(__svr4__) - ioctl(fd, I_FLUSH, FLUSHW); -#endif -} - -// convert an OSS audio format specification into a sun audio encoding -static int af2sunfmt(int format) -{ - switch (format){ - case AF_FORMAT_MU_LAW: - return AUDIO_ENCODING_ULAW; - case AF_FORMAT_A_LAW: - return AUDIO_ENCODING_ALAW; - case AF_FORMAT_S16_NE: - return AUDIO_ENCODING_LINEAR; -#ifdef AUDIO_ENCODING_LINEAR8 // Missing on SunOS 5.5.1... - case AF_FORMAT_U8: - return AUDIO_ENCODING_LINEAR8; -#endif - case AF_FORMAT_S8: - return AUDIO_ENCODING_LINEAR; -#ifdef AUDIO_ENCODING_DVI // Missing on NetBSD... - case AF_FORMAT_IMA_ADPCM: - return AUDIO_ENCODING_DVI; -#endif - default: - return AUDIO_ENCODING_NONE; - } -} - -// try to figure out, if the soundcard driver provides usable (precise) -// sample counter information -static int realtime_samplecounter_available(char *dev) -{ - int fd = -1; - audio_info_t info; - int rtsc_ok = RTSC_DISABLED; - int len; - void *silence = NULL; - struct timeval start, end; - struct timespec delay; - int usec_delay; - unsigned last_samplecnt; - unsigned increment; - unsigned min_increment; - - len = 44100 * 4 / 4; /* amount of data for 0.25sec of 44.1khz, stereo, - * 16bit. 44kbyte can be sent to all supported - * sun audio devices without blocking in the - * "write" below. - */ - silence = calloc(1, len); - if (silence == NULL) - goto error; - - if ((fd = open(dev, O_WRONLY)) < 0) - goto error; - - AUDIO_INITINFO(&info); - info.play.sample_rate = 44100; - info.play.channels = AUDIO_CHANNELS_STEREO; - info.play.precision = AUDIO_PRECISION_16; - info.play.encoding = AUDIO_ENCODING_LINEAR; - info.play.samples = 0; - if (ioctl(fd, AUDIO_SETINFO, &info)) { - if ( mp_msg_test(MSGT_AO,MSGL_V) ) - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SUN] rtsc: SETINFO failed.\n"); - goto error; - } - - if (write(fd, silence, len) != len) { - if ( mp_msg_test(MSGT_AO,MSGL_V) ) - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SUN] rtsc: write failed.\n"); - goto error; - } - - if (ioctl(fd, AUDIO_GETINFO, &info)) { - if ( mp_msg_test(MSGT_AO,MSGL_V) ) - perror("rtsc: GETINFO1"); - goto error; - } - - last_samplecnt = info.play.samples; - min_increment = ~0; - - gettimeofday(&start, NULL); - for (;;) { - delay.tv_sec = 0; - delay.tv_nsec = 10000000; - nanosleep(&delay, NULL); - gettimeofday(&end, NULL); - usec_delay = (end.tv_sec - start.tv_sec) * 1000000 - + end.tv_usec - start.tv_usec; - - // stop monitoring sample counter after 0.2 seconds - if (usec_delay > 200000) - break; - - if (ioctl(fd, AUDIO_GETINFO, &info)) { - if ( mp_msg_test(MSGT_AO,MSGL_V) ) - perror("rtsc: GETINFO2 failed"); - goto error; - } - if (info.play.samples < last_samplecnt) { - if ( mp_msg_test(MSGT_AO,MSGL_V) ) - mp_msg(MSGT_AO,MSGL_V,"rtsc: %d > %d?\n", last_samplecnt, info.play.samples); - goto error; - } - - if ((increment = info.play.samples - last_samplecnt) > 0) { - if ( mp_msg_test(MSGT_AO,MSGL_V) ) - mp_msg(MSGT_AO,MSGL_V,"ao_sun: sample counter increment: %d\n", increment); - if (increment < min_increment) { - min_increment = increment; - if (min_increment < 2000) - break; // looks good - } - } - last_samplecnt = info.play.samples; - } - - /* - * For 44.1kkz, stereo, 16-bit format we would send sound data in 16kbytes - * chunks (== 4096 samples) to the audio device. If we see a minimum - * sample counter increment from the soundcard driver of less than - * 2000 samples, we assume that the driver provides a useable realtime - * sample counter in the AUDIO_INFO play.samples field. Timing based - * on sample counts should be much more accurate than counting whole - * 16kbyte chunks. - */ - if (min_increment < 2000) - rtsc_ok = RTSC_ENABLED; - - if ( mp_msg_test(MSGT_AO,MSGL_V) ) - mp_msg(MSGT_AO,MSGL_V,"ao_sun: minimum sample counter increment per 10msec interval: %d\n" - "\t%susing sample counter based timing code\n", - min_increment, rtsc_ok == RTSC_ENABLED ? "" : "not "); - - -error: - free(silence); - if (fd >= 0) { - // remove the 0 bytes from the above measurement from the - // audio driver's STREAMS queue - flush_audio(fd); - close(fd); - } - - return rtsc_ok; -} - - -// match the requested sample rate |sample_rate| against the -// sample rates supported by the audio device |dev|. Return -// a supported sample rate, if that sample rate is close to -// (< 1% difference) the requested rate; return 0 otherwise. - -#define MAX_RATE_ERR 1 - -static unsigned -find_close_samplerate_match(int dev, unsigned sample_rate) -{ -#if HAVE_SYS_MIXER_H - am_sample_rates_t *sr; - unsigned i, num, err, best_err, best_rate; - - for (num = 16; num < 1024; num *= 2) { - sr = malloc(AUDIO_MIXER_SAMP_RATES_STRUCT_SIZE(num)); - if (!sr) - return 0; - sr->type = AUDIO_PLAY; - sr->flags = 0; - sr->num_samp_rates = num; - if (ioctl(dev, AUDIO_MIXER_GET_SAMPLE_RATES, sr)) { - free(sr); - return 0; - } - if (sr->num_samp_rates <= num) - break; - free(sr); - } - - if (sr->flags & MIXER_SR_LIMITS) { - /* - * HW can playback any rate between - * sr->samp_rates[0] .. sr->samp_rates[1] - */ - free(sr); - return 0; - } else { - /* HW supports fixed sample rates only */ - - best_err = 65535; - best_rate = 0; - - for (i = 0; i < sr->num_samp_rates; i++) { - err = abs(sr->samp_rates[i] - sample_rate); - if (err == 0) { - /* - * exact supported sample rate match, no need to - * retry something else - */ - best_rate = 0; - break; - } - if (err < best_err) { - best_err = err; - best_rate = sr->samp_rates[i]; - } - } - - free(sr); - - if (best_rate > 0 && (100/MAX_RATE_ERR)*best_err < sample_rate) { - /* found a supported sample rate with <1% error? */ - return best_rate; - } - return 0; - } -#else /* old audioio driver, cannot return list of supported rates */ - /* XXX: hardcoded sample rates */ - unsigned i, err; - unsigned audiocs_rates[] = { - 5510, 6620, 8000, 9600, 11025, 16000, 18900, 22050, - 27420, 32000, 33075, 37800, 44100, 48000, 0 - }; - - for (i = 0; audiocs_rates[i]; i++) { - err = abs(audiocs_rates[i] - sample_rate); - if (err == 0) { - /* - * exact supported sample rate match, no need to - * retry something elise - */ - return 0; - } - if ((100/MAX_RATE_ERR)*err < audiocs_rates[i]) { - /* <1% error? */ - return audiocs_rates[i]; - } - } - - return 0; -#endif -} - - -// return the highest sample rate supported by audio device |dev|. -static unsigned -find_highest_samplerate(int dev) -{ -#if HAVE_SYS_MIXER_H - am_sample_rates_t *sr; - unsigned i, num, max_rate; - - for (num = 16; num < 1024; num *= 2) { - sr = malloc(AUDIO_MIXER_SAMP_RATES_STRUCT_SIZE(num)); - if (!sr) - return 0; - sr->type = AUDIO_PLAY; - sr->flags = 0; - sr->num_samp_rates = num; - if (ioctl(dev, AUDIO_MIXER_GET_SAMPLE_RATES, sr)) { - free(sr); - return 0; - } - if (sr->num_samp_rates <= num) - break; - free(sr); - } - - if (sr->flags & MIXER_SR_LIMITS) { - /* - * HW can playback any rate between - * sr->samp_rates[0] .. sr->samp_rates[1] - */ - max_rate = sr->samp_rates[1]; - } else { - /* HW supports fixed sample rates only */ - max_rate = 0; - for (i = 0; i < sr->num_samp_rates; i++) { - if (sr->samp_rates[i] > max_rate) - max_rate = sr->samp_rates[i]; - } - } - free(sr); - return max_rate; - -#else /* old audioio driver, cannot return list of supported rates */ - return 44100; /* should be supported even on old ISA SB cards */ -#endif -} - - -static void setup_device_paths(void) -{ - if (audio_dev == NULL) { - if ((audio_dev = getenv("AUDIODEV")) == NULL) - audio_dev = "/dev/audio"; - } - - if (sun_mixer_device == NULL) { - if ((sun_mixer_device = mixer_device) == NULL || !sun_mixer_device[0]) { - sun_mixer_device = malloc(strlen(audio_dev) + 4); - strcpy(sun_mixer_device, audio_dev); - strcat(sun_mixer_device, "ctl"); - } - } - - if (ao_subdevice) audio_dev = ao_subdevice; -} - -// to set/get/query special features/parameters -static int control(int cmd,void *arg){ - switch(cmd){ - case AOCONTROL_GET_VOLUME: - { - int fd; - - if ( !sun_mixer_device ) /* control function is used before init? */ - setup_device_paths(); - - fd=open( sun_mixer_device,O_RDONLY ); - if ( fd != -1 ) - { - ao_control_vol_t *vol = (ao_control_vol_t *)arg; - float volume; - struct audio_info info; - ioctl( fd,AUDIO_GETINFO,&info); - volume = info.play.gain * 100. / AUDIO_MAX_GAIN; - if ( info.play.balance == AUDIO_MID_BALANCE ) { - vol->right = vol->left = volume; - } else if ( info.play.balance < AUDIO_MID_BALANCE ) { - vol->left = volume; - vol->right = volume * info.play.balance / AUDIO_MID_BALANCE; - } else { - vol->left = volume * (AUDIO_RIGHT_BALANCE-info.play.balance) - / AUDIO_MID_BALANCE; - vol->right = volume; - } - close( fd ); - return CONTROL_OK; - } - return CONTROL_ERROR; - } - case AOCONTROL_SET_VOLUME: - { - ao_control_vol_t *vol = (ao_control_vol_t *)arg; - int fd; - - if ( !sun_mixer_device ) /* control function is used before init? */ - setup_device_paths(); - - fd=open( sun_mixer_device,O_RDONLY ); - if ( fd != -1 ) - { - struct audio_info info; - float volume; - AUDIO_INITINFO(&info); - volume = vol->right > vol->left ? vol->right : vol->left; - if ( volume != 0 ) { - info.play.gain = volume * AUDIO_MAX_GAIN / 100; - if ( vol->right == vol->left ) - info.play.balance = AUDIO_MID_BALANCE; - else - info.play.balance = (vol->right - vol->left + volume) * AUDIO_RIGHT_BALANCE / (2*volume); - } -#if !defined (__OpenBSD__) && !defined (__NetBSD__) - info.output_muted = (volume == 0); -#endif - ioctl( fd,AUDIO_SETINFO,&info ); - close( fd ); - return CONTROL_OK; - } - return CONTROL_ERROR; - } - } - return CONTROL_UNKNOWN; -} - -// open & setup audio device -// return: 1=success 0=fail -static int init(int rate,int channels,int format,int flags){ - - audio_info_t info; - int pass; - int ok; - int convert_u8_s8; - - setup_device_paths(); - - if (enable_sample_timing == RTSC_UNKNOWN - && !getenv("AO_SUN_DISABLE_SAMPLE_TIMING")) { - enable_sample_timing = realtime_samplecounter_available(audio_dev); - } - - mp_msg(MSGT_AO,MSGL_STATUS,"ao2: %d Hz %d chans %s [0x%X]\n", - rate,channels,af_fmt2str_short(format),format); - - audio_fd=open(audio_dev, O_WRONLY); - if(audio_fd<0){ - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SUN] Can't open audio device %s, %s -> nosound.\n", audio_dev, strerror(errno)); - return 0; - } - - if (af2sunfmt(format) == AUDIO_ENCODING_NONE) - format = AF_FORMAT_S16_NE; - - for (ok = pass = 0; pass <= 5; pass++) { /* pass 6&7 not useful */ - - AUDIO_INITINFO(&info); - info.play.encoding = af2sunfmt(ao_data.format = format); - info.play.precision = - (format==AF_FORMAT_S16_NE - ? AUDIO_PRECISION_16 - : AUDIO_PRECISION_8); - info.play.channels = ao_data.channels = channels; - info.play.sample_rate = ao_data.samplerate = rate; - - convert_u8_s8 = 0; - - if (pass & 1) { - /* - * on some sun audio drivers, 8-bit unsigned LINEAR8 encoding is - * not supported, but 8-bit signed encoding is. - * - * Try S8, and if it works, use our own U8->S8 conversion before - * sending the samples to the sound driver. - */ -#ifdef AUDIO_ENCODING_LINEAR8 - if (info.play.encoding != AUDIO_ENCODING_LINEAR8) -#endif - continue; - info.play.encoding = AUDIO_ENCODING_LINEAR; - convert_u8_s8 = 1; - } - - if (pass & 2) { - /* - * on some sun audio drivers, only certain fixed sample rates are - * supported. - * - * In case the requested sample rate is very close to one of the - * supported rates, use the fixed supported rate instead. - */ - if (!(info.play.sample_rate = - find_close_samplerate_match(audio_fd, rate))) - continue; - - /* - * I'm not returning the correct sample rate in - * |ao_data.samplerate|, to avoid software resampling. - * - * ao_data.samplerate = info.play.sample_rate; - */ - } - - if (pass & 4) { - /* like "pass & 2", but use the highest supported sample rate */ - if (!(info.play.sample_rate - = ao_data.samplerate - = find_highest_samplerate(audio_fd))) - continue; - } - - ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0; - if (ok) { - /* audio format accepted by audio driver */ - break; - } - - /* - * format not supported? - * retry with different encoding and/or sample rate - */ - } - - if (!ok) { - char buf[128]; - mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SUN] audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate.\n", - channels, af_fmt2str(format, buf, 128), rate); - return 0; - } - - if (convert_u8_s8) - ao_data.format = AF_FORMAT_S8; - - bytes_per_sample = channels * info.play.precision / 8; - ao_data.bps = byte_per_sec = bytes_per_sample * ao_data.samplerate; - ao_data.outburst = byte_per_sec > 100000 ? 16384 : 8192; - - reset(); - - return 1; -} - -// close audio device -static void uninit(int immed){ - // throw away buffered data in the audio driver's STREAMS queue - if (immed) - flush_audio(audio_fd); - else - ioctl(audio_fd, AUDIO_DRAIN, 0); - close(audio_fd); -} - -// stop playing and empty buffers (for seeking/pause) -static void reset(void){ - audio_info_t info; - flush_audio(audio_fd); - - AUDIO_INITINFO(&info); - info.play.samples = 0; - info.play.eof = 0; - info.play.error = 0; - ioctl(audio_fd, AUDIO_SETINFO, &info); - - queued_bursts = 0; - queued_samples = 0; -} - -// stop playing, keep buffers (for pause) -static void audio_pause(void) -{ - struct audio_info info; - AUDIO_INITINFO(&info); - info.play.pause = 1; - ioctl(audio_fd, AUDIO_SETINFO, &info); -} - -// resume playing, after audio_pause() -static void audio_resume(void) -{ - struct audio_info info; - AUDIO_INITINFO(&info); - info.play.pause = 0; - ioctl(audio_fd, AUDIO_SETINFO, &info); -} - - -// return: how many bytes can be played without blocking -static int get_space(void){ - audio_info_t info; - - // check buffer -#ifdef HAVE_AUDIO_SELECT - { - fd_set rfds; - struct timeval tv; - FD_ZERO(&rfds); - FD_SET(audio_fd, &rfds); - tv.tv_sec = 0; - tv.tv_usec = 0; - if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block! - } -#endif - - ioctl(audio_fd, AUDIO_GETINFO, &info); -#if !defined (__OpenBSD__) && !defined(__NetBSD__) - if (queued_bursts - info.play.eof > 2) - return 0; - return ao_data.outburst; -#else - return info.hiwat * info.blocksize - info.play.seek; -#endif - -} - -// plays 'len' bytes of 'data' -// it should round it down to outburst*n -// return: number of bytes played -static int play(void* data,int len,int flags){ - if (!(flags & AOPLAY_FINAL_CHUNK)) { - len /= ao_data.outburst; - len *= ao_data.outburst; - } - if (len <= 0) return 0; - - len = write(audio_fd, data, len); - if(len > 0) { - queued_samples += len / bytes_per_sample; - if (write(audio_fd,data,0) < 0) - perror("ao_sun: send EOF audio record"); - else - queued_bursts ++; - } - return len; -} - - -// return: delay in seconds between first and last sample in buffer -static float get_delay(void){ - audio_info_t info; - ioctl(audio_fd, AUDIO_GETINFO, &info); -#if defined (__OpenBSD__) || defined(__NetBSD__) - return (float) info.play.seek/ (float)byte_per_sec ; -#else - if (info.play.samples && enable_sample_timing == RTSC_ENABLED) - return (float)(queued_samples - info.play.samples) / (float)ao_data.samplerate; - else - return (float)((queued_bursts - info.play.eof) * ao_data.outburst) / (float)byte_per_sec; -#endif -} |