Mercurial > mplayer.hg
view libao2/ao_sun.c @ 22705:f8fe4c872e9e
Remove conditional addition of CONFIG_DARWIN to CFLAGS, it's added
unconditionally three lines above ...
author | diego |
---|---|
date | Sun, 18 Mar 2007 19:38:43 +0000 |
parents | 99e20a22d5d0 |
children | 0471ad67050e |
line wrap: on
line source
#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" #include "help_mp.h" static 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; // 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_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscSetinfoFailed); goto error; } if (write(fd, silence, len) != len) { if ( mp_msg_test(MSGT_AO,MSGL_V) ) mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscWriteFailed); 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: if (silence != NULL) free(silence); if (fd >= 0) { #ifdef __svr4__ // remove the 0 bytes from the above measurement from the // audio driver's STREAMS queue ioctl(fd, I_FLUSH, FLUSHW); #endif //ioctl(fd, AUDIO_DRAIN, 0); 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_SET_DEVICE: audio_dev=(char*)arg; return CONTROL_OK; case AOCONTROL_QUERY_FORMAT: return CONTROL_TRUE; 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_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_CantOpenAudioDev, audio_dev, strerror(errno)); return 0; } ioctl(audio_fd, AUDIO_DRAIN, 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_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_UnsupSampleRate, 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; #ifdef __not_used__ /* * hmm, ao_data.buffersize is currently not used in this driver, do there's * no need to measure it */ if(ao_data.buffersize==-1){ // Measuring buffer size: void* data; ao_data.buffersize=0; #ifdef HAVE_AUDIO_SELECT data = malloc(ao_data.outburst); memset(data, format==AF_FORMAT_U8 ? 0x80 : 0, ao_data.outburst); while(ao_data.buffersize<0x40000){ 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)) break; write(audio_fd,data,ao_data.outburst); ao_data.buffersize+=ao_data.outburst; } free(data); if(ao_data.buffersize==0){ mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_CantUseSelect); return 0; } #ifdef __svr4__ // remove the 0 bytes from the above ao_data.buffersize measurement from the // audio driver's STREAMS queue ioctl(audio_fd, I_FLUSH, FLUSHW); #endif ioctl(audio_fd, AUDIO_DRAIN, 0); #endif } #endif /* __not_used__ */ 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; return 1; } // close audio device static void uninit(int immed){ #ifdef __svr4__ // throw away buffered data in the audio driver's STREAMS queue if (immed) ioctl(audio_fd, I_FLUSH, FLUSHW); #endif close(audio_fd); } // stop playing and empty buffers (for seeking/pause) static void reset(void){ audio_info_t info; uninit(1); audio_fd=open(audio_dev, O_WRONLY); if(audio_fd<0){ mp_msg(MSGT_AO, MSGL_FATAL, MSGTR_AO_SUN_CantReopenReset, strerror(errno)); return; } ioctl(audio_fd, AUDIO_DRAIN, 0); AUDIO_INITINFO(&info); info.play.encoding = af2sunfmt(ao_data.format); info.play.precision = (ao_data.format==AF_FORMAT_S16_NE ? AUDIO_PRECISION_16 : AUDIO_PRECISION_8); info.play.channels = ao_data.channels; info.play.sample_rate = ao_data.samplerate; 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 #if !defined (__OpenBSD__) && !defined(__NetBSD__) ioctl(audio_fd, AUDIO_GETINFO, &info); if (queued_bursts - info.play.eof > 2) return 0; #endif #if defined(__NetBSD__) || defined(__OpenBSD__) ioctl(audio_fd, AUDIO_GETINFO, &info); return info.hiwat * info.blocksize - info.play.seek; #else return ao_data.outburst; #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 (len < ao_data.outburst) return 0; len /= ao_data.outburst; len *= ao_data.outburst; 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 }