📄 oss_a.c
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/* TiMidity++ -- MIDI to WAVE converter and player Copyright (C) 1999-2002 Masanao Izumo <mo@goice.co.jp> Copyright (C) 1995 Tuukka Toivonen <tt@cgs.fi> This program 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. This program 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 this program; if not, write to the Free Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA oss_a.c (linux_a.c) Functions to play sound on the OSS audio driver (Linux or FreeBSD)*/#ifdef HAVE_CONFIG_H#include "config.h"#endif /* HAVE_CONFIG_H */#define _GNU_SOURCE#include <stdio.h>#include <unistd.h>#include <fcntl.h>#ifndef NO_STRING_H#include <string.h>#else#include <strings.h>#endif#if defined(HAVE_SYS_SOUNDCARD_H)#include <sys/soundcard.h>#elif defined(linux)#include <sys/ioctl.h> /* new with 1.2.0? Didn't need this under 1.1.64 */#include <linux/soundcard.h>#elif defined(__FreeBSD__)#include <machine/soundcard.h>#include <sys/filio.h>#elif defined(HAVE_SOUNDCARD_H)#include <soundcard.h>#else#include <sys/soundcard.h>#endif /* HAVE_SYS_SOUNDCARD_H */#include "timidity.h"#include "common.h"#include "output.h"#include "controls.h"#include "timer.h"#include "instrum.h"#include "playmidi.h"#include "miditrace.h"#ifndef AFMT_S16_NE#ifdef LITTLE_ENDIAN#define AFMT_S16_NE AFMT_S16_LE#else#define AFMT_S16_NE AFMT_S16_BE#endif#endif /* AFMT_S16_NE */static int open_output(void); /* 0=success, 1=warning, -1=fatal error */static void close_output(void);static int output_data(char *buf, int32 nbytes);static int acntl(int request, void *arg);static int detect(void);static int output_counter;static int total_bytes; /* Maximum buffer size in bytes *//* export the playback mode */#define dpm oss_play_modePlayMode dpm = { DEFAULT_RATE, PE_16BIT|PE_SIGNED, PF_PCM_STREAM|PF_CAN_TRACE|PF_BUFF_FRAGM_OPT, -1, {0}, /* default: get all the buffer fragments you can */ "dsp device", 'd',#ifdef OSS_DEVICE OSS_DEVICE,#else "/dev/dsp",#endif open_output, close_output, output_data, acntl, detect};static int detect(void){ int fd; fd = open(dpm.name, O_WRONLY | O_NONBLOCK); if (fd < 0) return 0; close(fd); return 1; /* found */}/*************************************************************************//* We currently only honor the PE_MONO bit, the sample rate, and the number of buffer fragments. We try 16-bit signed data first, and then 8-bit unsigned if it fails. If you have a sound device that can't handle either, let me know. */static int open_output(void){ int fd, tmp, i, warnings = 0; int include_enc, exclude_enc;#ifdef SNDCTL_DSP_GETOSPACE audio_buf_info info;#endif /* SNDCTL_DSP_GETOSPACE */ /* Open the audio device */ fd = open(dpm.name, O_WRONLY); if(fd < 0) { ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "%s: %s", dpm.name, strerror(errno)); return -1; } /* * Modified: Fri Nov 20 1998 * * Reported from http://www.ife.ee.ethz.ch/~sailer/linux/pciaudio.html * by Thomas Sailer <sailer@ife.ee.ethz.ch> * OSS/Free sets nonblocking mode with an ioctl, unlike the rest of the * kernel, which uses the O_NONBLOCK flag. I want to regularize that API, * and this trips on Timidity. */ fcntl(fd, F_SETFL, fcntl(fd, F_GETFL, 0) & ~O_NDELAY); include_enc = 0; exclude_enc = PE_ULAW|PE_ALAW|PE_BYTESWAP; /* They can't mean these */ if(dpm.encoding & PE_16BIT) include_enc |= PE_SIGNED; else exclude_enc |= PE_SIGNED; dpm.encoding = validate_encoding(dpm.encoding, include_enc, exclude_enc); /* Set sample width to whichever the user wants. If it fails, try the other one. */ i = tmp = (dpm.encoding & PE_16BIT) ? AFMT_S16_NE : AFMT_U8; if(ioctl(fd, SNDCTL_DSP_SAMPLESIZE, &tmp) < 0 || tmp != i) { /* Try the other one */ i = tmp = (dpm.encoding & PE_16BIT) ? AFMT_U8 : AFMT_S16_NE; if(ioctl(fd, SNDCTL_DSP_SAMPLESIZE, &tmp) < 0 || tmp != i) { ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "%s doesn't support 16- or 8-bit sample width", dpm.name); close(fd); return -1; } ctl->cmsg(CMSG_WARNING, VERB_VERBOSE, "Sample width adjusted to %d bits", tmp); dpm.encoding ^= PE_16BIT; warnings = 1; } /* Try stereo or mono, whichever the user wants. If it fails, try the other. */ i = tmp = (dpm.encoding & PE_MONO) ? 0 : 1; if((ioctl(fd, SNDCTL_DSP_STEREO, &tmp) < 0) || tmp != i) { i = tmp = (dpm.encoding & PE_MONO) ? 1 : 0; if((ioctl(fd, SNDCTL_DSP_STEREO, &tmp) < 0) || tmp != i) { ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "%s doesn't support mono or stereo samples", dpm.name); close(fd); return -1; } if(tmp == 0) dpm.encoding |= PE_MONO; else dpm.encoding &= ~PE_MONO; ctl->cmsg(CMSG_WARNING, VERB_VERBOSE, "Sound adjusted to %sphonic", (tmp==0) ? "mono" : "stereo"); warnings = 1; } /* Set the sample rate */ tmp = dpm.rate; if(ioctl(fd, SNDCTL_DSP_SPEED, &tmp) < 0) { ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "%s doesn't support a %d Hz sample rate", dpm.name, dpm.rate); close(fd); return -1; } if(tmp != dpm.rate) { ctl->cmsg(CMSG_WARNING, VERB_VERBOSE, "Output rate adjusted to %d Hz (requested %d Hz)", tmp, dpm.rate); dpm.rate = tmp; warnings = 1; } /* Older VoxWare drivers don't have buffer fragment capabilities */#ifdef SNDCTL_DSP_SETFRAGMENT /* Set buffer fragments (in extra_param[0]) */ tmp = audio_buffer_bits; if(!(dpm.encoding & PE_MONO)) tmp++; if(dpm.encoding & PE_16BIT) tmp++; i = tmp; tmp |= (dpm.extra_param[0] << 16); if(ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &tmp) < 0) { ctl->cmsg(CMSG_WARNING, VERB_NORMAL, "%s doesn't support %d-byte buffer fragments (%d)", dpm.name, 1<<i, i); /* It should still work in some fashion. We should use a secondary buffer anyway -- 64k isn't enough. */ warnings = 1; }#else if(dpm.extra_param[0]) { ctl->cmsg(CMSG_WARNING, VERB_NORMAL, "%s doesn't support buffer fragments", dpm.name); warnings = 1; }#endif#ifdef SNDCTL_DSP_GETOSPACE if(ioctl(fd, SNDCTL_DSP_GETOSPACE, &info) != -1) { total_bytes = info.fragstotal * info.fragsize; ctl->cmsg(CMSG_INFO, VERB_NOISY, "Audio device buffer: %d x %d bytes", info.fragstotal, info.fragsize); } else#endif /* SNDCTL_DSP_GETOSPACE */ total_bytes = -1; /* Unknown */ dpm.fd = fd; output_counter = 0; return warnings;}static int output_data(char *buf, int32 nbytes){ int n; while(nbytes > 0) { if((n = write(dpm.fd, buf, nbytes)) == -1) {#if 0 if(errno != XXX) /* I don't know what error should be ignored. * XXX := EWOULDBLOCK?? */ { usleep(100000); continue; } ctl->cmsg(CMSG_WARNING, VERB_VERBOSE, "%s: %s", dpm.name, strerror(errno)); return -1;#endif /* Now, Ignore error */ usleep(100000); continue; } buf += n; nbytes -= n;#ifdef SNDCTL_DSP_GETODELAY output_counter += n;#endif } return 0;}static void close_output(void){ if(dpm.fd == -1) return; close(dpm.fd); dpm.fd = -1;}static int acntl(int request, void *arg){#ifndef SNDCTL_DSP_GETODELAY# ifdef SNDCTL_DSP_GETOSPACE audio_buf_info info;# endif /* SNDCTL_DSP_GETOSPACE */ count_info cinfo;#endif /* SNDCTL_DSP_GETODELAY */ int i; switch(request) { case PM_REQ_RATE: i = *(int *)arg; /* sample rate in and out */ if(ioctl(dpm.fd, SNDCTL_DSP_SPEED, &i) < 0) return -1; play_mode->rate = i; return 0; case PM_REQ_GETQSIZ: if(total_bytes <= 0) return -1; *((int *)arg) = total_bytes; return 0;#ifdef SNDCTL_DSP_GETODELAY case PM_REQ_DISCARD: output_counter = 0; return ioctl(dpm.fd, SNDCTL_DSP_RESET, NULL); case PM_REQ_FLUSH: output_counter = 0; return ioctl(dpm.fd, SNDCTL_DSP_SYNC, NULL); case PM_REQ_GETFILLED: if(total_bytes <= 0 || ioctl(dpm.fd, SNDCTL_DSP_GETODELAY, &i) == -1) return -1; if (i > total_bytes) i = total_bytes; if(!(dpm.encoding & PE_MONO)) i >>= 1; if(dpm.encoding & PE_16BIT) i >>= 1; *((int *)arg) = i; return 0; case PM_REQ_GETFILLABLE: if(total_bytes <= 0 || ioctl(dpm.fd, SNDCTL_DSP_GETODELAY, &i) == -1) return -1; if (i > total_bytes) i = 0; else i = total_bytes - i; if(!(dpm.encoding & PE_MONO)) i >>= 1; if(dpm.encoding & PE_16BIT) i >>= 1; *((int *)arg) = i; return 0; case PM_REQ_GETSAMPLES: if(ioctl(dpm.fd, SNDCTL_DSP_GETODELAY, &i) == -1) return -1; i = output_counter - i; if(!(dpm.encoding & PE_MONO)) i >>= 1; if(dpm.encoding & PE_16BIT) i >>= 1; *((int *)arg) = i; return 0;#else /* SNDCTL_DSP_GETODELAY */ case PM_REQ_DISCARD: if(ioctl(dpm.fd, SNDCTL_DSP_RESET, NULL) == -1) return -1; if(ioctl(dpm.fd, SNDCTL_DSP_GETOPTR, &cinfo) == -1) return -1; output_counter = cinfo.bytes; return 0; case PM_REQ_FLUSH: if(ioctl(dpm.fd, SNDCTL_DSP_SYNC, NULL) == -1) return -1; if(ioctl(dpm.fd, SNDCTL_DSP_GETOPTR, &cinfo) == -1) return -1; output_counter = cinfo.bytes; return 0;# ifdef SNDCTL_DSP_GETOSPACE case PM_REQ_GETFILLABLE: if(total_bytes <= 0 || ioctl(dpm.fd, SNDCTL_DSP_GETOSPACE, &info) == -1) return -1; if (info.bytes > total_bytes) i = total_bytes; else i = info.bytes; if(!(dpm.encoding & PE_MONO)) i >>= 1; if(dpm.encoding & PE_16BIT) i >>= 1; *((int *)arg) = i; return 0; case PM_REQ_GETFILLED: if(total_bytes <= 0 || ioctl(dpm.fd, SNDCTL_DSP_GETOSPACE, &info) == -1) return -1; if (info.bytes > total_bytes) i = 0; else i = total_bytes - info.bytes; if(!(dpm.encoding & PE_MONO)) i >>= 1; if(dpm.encoding & PE_16BIT) i >>= 1; *((int *)arg) = i; return 0;# endif /* SNDCTL_DSP_GETOSPACE */ case PM_REQ_GETSAMPLES: if(ioctl(dpm.fd, SNDCTL_DSP_GETOPTR, &cinfo) == -1) return -1; i = cinfo.bytes - output_counter; if(!(dpm.encoding & PE_MONO)) i >>= 1; if(dpm.encoding & PE_16BIT) i >>= 1; *((int *)arg) = i; return 0;#endif /* SNDCTL_DSP_GETODELAY */ } return -1;}
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