Mercurial > mplayer.hg
view libao2/ao_sun.c @ 35805:5d163370a554
Restructure code for video window in uiChangeSkin().
Compact the conditions. It's much clearer now what's happening.
Additionally, change comment to indicate that the video window
will not be reloaded (like all other windows).
author | ib |
---|---|
date | Sat, 26 Jan 2013 15:32:37 +0000 |
parents | 2468dc935bed |
children |
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/* * 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" #include "help_mp.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)) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscSetinfoFailed); goto error; } if (write(fd, silence, len) != len) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscWriteFailed); goto error; } if (ioctl(fd, AUDIO_GETINFO, &info)) { 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)) { perror("rtsc: GETINFO2 failed"); goto error; } if (info.play.samples < last_samplecnt) { mp_msg(MSGT_AO, MSGL_ERR, "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_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; } 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; 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 }