mirror of
https://github.com/mpv-player/mpv
synced 2024-12-25 00:02:13 +00:00
ac2fbcbf25
is available, convert to AFMT_S8 in software. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@1808 b3059339-0415-0410-9bf9-f77b7e298cf2
473 lines
12 KiB
C
473 lines
12 KiB
C
#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <unistd.h>
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#include <fcntl.h>
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#include <errno.h>
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#include <sys/ioctl.h>
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#include <sys/time.h>
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#include <sys/types.h>
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#include <sys/stat.h>
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#include <sys/audioio.h>
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#ifdef __svr4__
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#include <stropts.h>
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#endif
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#include "../config.h"
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#include "audio_out.h"
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#include "audio_out_internal.h"
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#include "afmt.h"
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static ao_info_t info =
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{
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"Sun audio output",
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"sun",
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"jk@tools.de",
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""
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};
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LIBAO_EXTERN(sun)
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/* These defines are missing on NetBSD */
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#ifndef AUDIO_PRECISION_8
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#define AUDIO_PRECISION_8 8
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#define AUDIO_PRECISION_16 16
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#endif
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#ifndef AUDIO_CHANNELS_MONO
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#define AUDIO_CHANNELS_MONO 1
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#define AUDIO_CHANNELS_STEREO 2
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#endif
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// there are some globals:
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// ao_samplerate
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// ao_channels
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// ao_format
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// ao_bps
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// ao_outburst
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// ao_buffersize
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static char *audio_dev = "/dev/audio";
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static int queued_bursts = 0;
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static int queued_samples = 0;
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static int bytes_per_sample = 0;
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static int convert_u8_s8;
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static int audio_fd = -1;
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static enum {
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RTSC_UNKNOWN = 0,
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RTSC_ENABLED,
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RTSC_DISABLED
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} enable_sample_timing;
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extern int verbose;
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// convert an OSS audio format specification into a sun audio encoding
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static int oss2sunfmt(int oss_format)
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{
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switch (oss_format){
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case AFMT_MU_LAW:
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return AUDIO_ENCODING_ULAW;
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case AFMT_A_LAW:
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return AUDIO_ENCODING_ALAW;
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case AFMT_S16_BE:
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case AFMT_S16_LE:
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return AUDIO_ENCODING_LINEAR;
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#ifdef AUDIO_ENCODING_LINEAR8 // Missing on SunOS 5.5.1...
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case AFMT_U8:
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return AUDIO_ENCODING_LINEAR8;
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#endif
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#ifdef AUDIO_ENCODING_DVI // Missing on NetBSD...
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case AFMT_IMA_ADPCM:
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return AUDIO_ENCODING_DVI;
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#endif
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default:
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return AUDIO_ENCODING_NONE;
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}
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}
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// try to figure out, if the soundcard driver provides usable (precise)
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// sample counter information
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static int realtime_samplecounter_available(char *dev)
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{
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int fd = -1;
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audio_info_t info;
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int rtsc_ok = RTSC_DISABLED;
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int len;
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void *silence = NULL;
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struct timeval start, end;
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struct timespec delay;
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int usec_delay;
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unsigned last_samplecnt;
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unsigned increment;
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unsigned min_increment;
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len = 44100 * 4 / 4; /* amount of data for 0.25sec of 44.1khz, stereo,
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* 16bit. 44kbyte can be sent to all supported
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* sun audio devices without blocking in the
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* "write" below.
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*/
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silence = calloc(1, len);
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if (silence == NULL)
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goto error;
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if ((fd = open(dev, O_WRONLY)) < 0)
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goto error;
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AUDIO_INITINFO(&info);
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info.play.sample_rate = 44100;
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info.play.channels = AUDIO_CHANNELS_STEREO;
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info.play.precision = AUDIO_PRECISION_16;
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info.play.encoding = AUDIO_ENCODING_LINEAR;
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info.play.samples = 0;
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if (ioctl(fd, AUDIO_SETINFO, &info)) {
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if (verbose)
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printf("rtsc: SETINFO failed\n");
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goto error;
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}
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if (write(fd, silence, len) != len) {
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if (verbose)
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printf("rtsc: write failed");
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goto error;
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}
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if (ioctl(fd, AUDIO_GETINFO, &info)) {
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if (verbose)
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perror("rtsc: GETINFO1");
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goto error;
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}
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last_samplecnt = info.play.samples;
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min_increment = ~0;
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gettimeofday(&start, NULL);
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for (;;) {
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delay.tv_sec = 0;
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delay.tv_nsec = 10000000;
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nanosleep(&delay, NULL);
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gettimeofday(&end, NULL);
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usec_delay = (end.tv_sec - start.tv_sec) * 1000000
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+ end.tv_usec - start.tv_usec;
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// stop monitoring sample counter after 0.2 seconds
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if (usec_delay > 200000)
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break;
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if (ioctl(fd, AUDIO_GETINFO, &info)) {
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if (verbose)
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perror("rtsc: GETINFO2 failed");
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goto error;
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}
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if (info.play.samples < last_samplecnt) {
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if (verbose)
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printf("rtsc: %d > %d?\n", last_samplecnt, info.play.samples);
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goto error;
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}
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if ((increment = info.play.samples - last_samplecnt) > 0) {
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if (verbose)
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printf("ao_sun: sample counter increment: %d\n", increment);
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if (increment < min_increment) {
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min_increment = increment;
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if (min_increment < 2000)
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break; // looks good
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}
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}
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last_samplecnt = info.play.samples;
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}
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/*
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* For 44.1kkz, stereo, 16-bit format we would send sound data in 16kbytes
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* chunks (== 4096 samples) to the audio device. If we see a minimum
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* sample counter increment from the soundcard driver of less than
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* 2000 samples, we assume that the driver provides a useable realtime
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* sample counter in the AUDIO_INFO play.samples field. Timing based
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* on sample counts should be much more accurate than counting whole
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* 16kbyte chunks.
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*/
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if (min_increment < 2000)
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rtsc_ok = RTSC_ENABLED;
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if (verbose)
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printf("ao_sun: minimum sample counter increment per 10msec interval: %d\n"
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"\t%susing sample counter based timing code\n",
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min_increment, rtsc_ok == RTSC_ENABLED ? "" : "not ");
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error:
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if (silence != NULL) free(silence);
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if (fd >= 0) {
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#ifdef __svr4__
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// remove the 0 bytes from the above measurement from the
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// audio driver's STREAMS queue
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ioctl(fd, I_FLUSH, FLUSHW);
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#endif
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//ioctl(fd, AUDIO_DRAIN, 0);
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close(fd);
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}
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return rtsc_ok;
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}
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// to set/get/query special features/parameters
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static int control(int cmd,int arg){
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switch(cmd){
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case AOCONTROL_SET_DEVICE:
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audio_dev=(char*)arg;
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return CONTROL_OK;
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case AOCONTROL_QUERY_FORMAT:
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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}
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// open & setup audio device
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// return: 1=success 0=fail
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static int init(int rate,int channels,int format,int flags){
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audio_info_t info;
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int byte_per_sec;
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int ok;
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if (ao_subdevice) audio_dev = ao_subdevice;
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if (enable_sample_timing == RTSC_UNKNOWN
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&& !getenv("AO_SUN_DISABLE_SAMPLE_TIMING")) {
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enable_sample_timing = realtime_samplecounter_available(audio_dev);
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}
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// printf("ao2: %d Hz %d chans %s [0x%X]\n",
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// rate,channels,audio_out_format_name(format),format);
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audio_fd=open(audio_dev, O_WRONLY);
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if(audio_fd<0){
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printf("Can't open audio device %s, %s -> nosound\n", audio_dev, strerror(errno));
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return 0;
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}
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ioctl(audio_fd, AUDIO_DRAIN, 0);
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AUDIO_INITINFO(&info);
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info.play.encoding = oss2sunfmt(ao_format = format);
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info.play.precision =
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(format==AFMT_S16_LE || format==AFMT_S16_BE
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? AUDIO_PRECISION_16
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: AUDIO_PRECISION_8);
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info.play.channels = ao_channels = channels;
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info.play.sample_rate = ao_samplerate = rate;
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convert_u8_s8 = 0;
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ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0;
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if (!ok && info.play.encoding == AUDIO_ENCODING_LINEAR8) {
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/* sun audiocs hardware does not support U8 format, try S8... */
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info.play.encoding = AUDIO_ENCODING_LINEAR;
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ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0;
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if (ok) {
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/* we must perform software U8 -> S8 conversion */
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convert_u8_s8 = 1;
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}
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}
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if (!ok) {
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printf("audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate\n",
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channels, audio_out_format_name(format), rate);
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return 0;
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}
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bytes_per_sample = channels * info.play.precision / 8;
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byte_per_sec = bytes_per_sample * rate;
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ao_outburst = byte_per_sec > 100000 ? 16384 : 8192;
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#ifdef __not_used__
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/*
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* hmm, ao_buffersize is currently not used in this driver, do there's
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* no need to measure it
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*/
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if(ao_buffersize==-1){
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// Measuring buffer size:
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void* data;
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ao_buffersize=0;
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#ifdef HAVE_AUDIO_SELECT
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data = malloc(ao_outburst);
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memset(data, format==AFMT_U8 ? 0x80 : 0, ao_outburst);
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while(ao_buffersize<0x40000){
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fd_set rfds;
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struct timeval tv;
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FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
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tv.tv_sec=0; tv.tv_usec = 0;
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if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
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write(audio_fd,data,ao_outburst);
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ao_buffersize+=ao_outburst;
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}
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free(data);
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if(ao_buffersize==0){
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printf("\n *** Your audio driver DOES NOT support select() ***\n");
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printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
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return 0;
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}
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#ifdef __svr4__
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// remove the 0 bytes from the above ao_buffersize measurement from the
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// audio driver's STREAMS queue
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ioctl(audio_fd, I_FLUSH, FLUSHW);
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#endif
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ioctl(audio_fd, AUDIO_DRAIN, 0);
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#endif
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}
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#endif /* __not_used__ */
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AUDIO_INITINFO(&info);
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info.play.samples = 0;
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info.play.eof = 0;
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info.play.error = 0;
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ioctl (audio_fd, AUDIO_SETINFO, &info);
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queued_bursts = 0;
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queued_samples = 0;
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return 1;
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}
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// close audio device
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static void uninit(){
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#ifdef __svr4__
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// throw away buffered data in the audio driver's STREAMS queue
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ioctl(audio_fd, I_FLUSH, FLUSHW);
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#endif
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close(audio_fd);
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}
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// stop playing and empty buffers (for seeking/pause)
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static void reset(){
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audio_info_t info;
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uninit();
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audio_fd=open(audio_dev, O_WRONLY);
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if(audio_fd<0){
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printf("\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE (%s) ***\n", strerror(errno));
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return;
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}
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ioctl(audio_fd, AUDIO_DRAIN, 0);
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AUDIO_INITINFO(&info);
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info.play.encoding = oss2sunfmt(ao_format);
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info.play.precision =
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(ao_format==AFMT_S16_LE || ao_format==AFMT_S16_BE
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? AUDIO_PRECISION_16
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: AUDIO_PRECISION_8);
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info.play.channels = ao_channels;
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info.play.sample_rate = ao_samplerate;
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info.play.samples = 0;
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info.play.eof = 0;
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info.play.error = 0;
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ioctl (audio_fd, AUDIO_SETINFO, &info);
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queued_bursts = 0;
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queued_samples = 0;
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}
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// stop playing, keep buffers (for pause)
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static void audio_pause()
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{
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struct audio_info info;
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AUDIO_INITINFO(&info);
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info.play.pause = 1;
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ioctl(audio_fd, AUDIO_SETINFO, &info);
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}
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// resume playing, after audio_pause()
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static void audio_resume()
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{
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struct audio_info info;
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AUDIO_INITINFO(&info);
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info.play.pause = 0;
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ioctl(audio_fd, AUDIO_SETINFO, &info);
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}
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// return: how many bytes can be played without blocking
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static int get_space(){
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int playsize = ao_outburst;
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audio_info_t info;
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// check buffer
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#ifdef HAVE_AUDIO_SELECT
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{
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fd_set rfds;
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struct timeval tv;
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FD_ZERO(&rfds);
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FD_SET(audio_fd, &rfds);
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tv.tv_sec = 0;
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tv.tv_usec = 0;
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if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block!
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}
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#endif
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ioctl(audio_fd, AUDIO_GETINFO, &info);
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if (queued_bursts - info.play.eof > 2)
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return 0;
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return ao_outburst;
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}
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// plays 'len' bytes of 'data'
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// it should round it down to outburst*n
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// return: number of bytes played
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static int play(void* data,int len,int flags){
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#if WORDS_BIGENDIAN
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int native_endian = AFMT_S16_BE;
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#else
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int native_endian = AFMT_S16_LE;
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#endif
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if (len < ao_outburst) return 0;
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len /= ao_outburst;
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len *= ao_outburst;
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/* 16-bit format using the 'wrong' byteorder? swap words */
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if ((ao_format == AFMT_S16_LE || ao_format == AFMT_S16_BE)
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&& ao_format != native_endian) {
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static void *swab_buf;
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static int swab_len;
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if (len > swab_len) {
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if (swab_buf)
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swab_buf = realloc(swab_buf, len);
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else
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swab_buf = malloc(len);
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swab_len = len;
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if (swab_buf == NULL) return 0;
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}
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swab(data, swab_buf, len);
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data = swab_buf;
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} else if (ao_format == AFMT_U8 && convert_u8_s8) {
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int i;
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unsigned char *p = data;
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for (i = 0, p = data; i < len; i++, p++)
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*p ^= 0x80;
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}
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len = write(audio_fd, data, len);
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if(len > 0) {
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queued_samples += len / bytes_per_sample;
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if (write(audio_fd,data,0) < 0)
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perror("ao_sun: send EOF audio record");
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else
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queued_bursts ++;
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}
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return len;
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}
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// return: how many unplayed bytes are in the buffer
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static int get_delay(){
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audio_info_t info;
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ioctl(audio_fd, AUDIO_GETINFO, &info);
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if (info.play.samples && enable_sample_timing == RTSC_ENABLED)
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return (queued_samples - info.play.samples) * bytes_per_sample;
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else
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return (queued_bursts - info.play.eof) * ao_outburst;
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}
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