From f8f42856710768fec2518b020c2bf367e2b165e8 Mon Sep 17 00:00:00 2001 From: wm4 Date: Fri, 7 Jun 2013 14:29:59 +0200 Subject: ao_oss: uncrustify --- audio/out/ao_oss.c | 632 ++++++++++++++++++++++++++++------------------------- 1 file changed, 334 insertions(+), 298 deletions(-) (limited to 'audio') diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c index 4ff97b30b6..566d321c19 100644 --- a/audio/out/ao_oss.c +++ b/audio/out/ao_oss.c @@ -3,6 +3,9 @@ * * This file is part of MPlayer. * + * Support for >2 output channels added 2001-11-25 + * - Steve Davies + * * 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 @@ -49,20 +52,17 @@ static const ao_info_t info = { - "OSS/ioctl audio output", - "oss", - "A'rpi", - "" + "OSS/ioctl audio output", + "oss", + "A'rpi", + "" }; -/* Support for >2 output channels added 2001-11-25 - Steve Davies */ - LIBAO_EXTERN(oss) static int format2oss(int format) { - switch(format) - { + switch (format) { case AF_FORMAT_U8: return AFMT_U8; case AF_FORMAT_S8: return AFMT_S8; case AF_FORMAT_U16_LE: return AFMT_U16_LE; @@ -87,7 +87,7 @@ static int format2oss(int format) #ifdef AFMT_FLOAT case AF_FORMAT_FLOAT_NE: return AFMT_FLOAT; #endif - // SPECIALS + // SPECIALS #ifdef AFMT_MPEG case AF_FORMAT_MPEG2: return AFMT_MPEG; #endif @@ -95,14 +95,14 @@ static int format2oss(int format) case AF_FORMAT_AC3_NE: return AFMT_AC3; #endif } - mp_msg(MSGT_AO, MSGL_V, "OSS: Unknown/not supported internal format: %s\n", af_fmt2str_short(format)); + mp_msg(MSGT_AO, MSGL_V, "OSS: Unknown/not supported internal format: %s\n", + af_fmt2str_short(format)); return -1; } static int oss2format(int format) { - switch(format) - { + switch (format) { case AFMT_U8: return AF_FORMAT_U8; case AFMT_S8: return AF_FORMAT_S8; case AFMT_U16_LE: return AF_FORMAT_U16_LE; @@ -127,7 +127,7 @@ static int oss2format(int format) #ifdef AFMT_FLOAT case AFMT_FLOAT: return AF_FORMAT_FLOAT_NE; #endif - // SPECIALS + // SPECIALS #ifdef AFMT_MPEG case AFMT_MPEG: return AF_FORMAT_MPEG2; #endif @@ -135,20 +135,22 @@ static int oss2format(int format) case AFMT_AC3: return AF_FORMAT_AC3_NE; #endif } - mp_tmsg(MSGT_GLOBAL,MSGL_ERR,"[AO OSS] Unknown/Unsupported OSS format: %x.\n", format); + mp_tmsg(MSGT_GLOBAL, MSGL_ERR, "[AO OSS] Unknown/Unsupported OSS format: %x.\n", + format); return -1; } -static char *dsp=PATH_DEV_DSP; +static char *dsp = PATH_DEV_DSP; static audio_buf_info zz; -static int audio_fd=-1; +static int audio_fd = -1; static int prepause_space; static const char *oss_mixer_device = PATH_DEV_MIXER; static int oss_mixer_channel = SOUND_MIXER_PCM; #ifdef SNDCTL_DSP_GETPLAYVOL -static int volume_oss4(ao_control_vol_t *vol, int cmd) { +static int volume_oss4(ao_control_vol_t *vol, int cmd) +{ int v; if (audio_fd < 0) @@ -171,13 +173,14 @@ static int volume_oss4(ao_control_vol_t *vol, int cmd) { #endif // to set/get/query special features/parameters -static int control(int cmd,void *arg){ - switch(cmd){ - case AOCONTROL_GET_VOLUME: - case AOCONTROL_SET_VOLUME: - { - ao_control_vol_t *vol = (ao_control_vol_t *)arg; - int fd, v, devs; +static int control(int cmd, void *arg) +{ + switch (cmd) { + case AOCONTROL_GET_VOLUME: + case AOCONTROL_SET_VOLUME: + { + ao_control_vol_t *vol = (ao_control_vol_t *)arg; + int fd, v, devs; #ifdef SNDCTL_DSP_GETPLAYVOL // Try OSS4 first @@ -185,294 +188,320 @@ static int control(int cmd,void *arg){ return CONTROL_OK; #endif - if(AF_FORMAT_IS_AC3(ao_data.format)) - return CONTROL_TRUE; - - if ((fd = open(oss_mixer_device, O_RDONLY)) != -1) - { - ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs); - if (devs & (1 << oss_mixer_channel)) - { - if (cmd == AOCONTROL_GET_VOLUME) - { - ioctl(fd, MIXER_READ(oss_mixer_channel), &v); - vol->right = (v & 0xFF00) >> 8; - vol->left = v & 0x00FF; - } - else - { - v = ((int)vol->right << 8) | (int)vol->left; - ioctl(fd, MIXER_WRITE(oss_mixer_channel), &v); - } - } - else - { - close(fd); - return CONTROL_ERROR; - } - close(fd); - return CONTROL_OK; - } - } - return CONTROL_ERROR; + if (AF_FORMAT_IS_AC3(ao_data.format)) + return CONTROL_TRUE; + + if ((fd = open(oss_mixer_device, O_RDONLY)) != -1) { + ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs); + if (devs & (1 << oss_mixer_channel)) { + if (cmd == AOCONTROL_GET_VOLUME) { + ioctl(fd, MIXER_READ(oss_mixer_channel), &v); + vol->right = (v & 0xFF00) >> 8; + vol->left = v & 0x00FF; + } else { + v = ((int)vol->right << 8) | (int)vol->left; + ioctl(fd, MIXER_WRITE(oss_mixer_channel), &v); + } + } else { + close(fd); + return CONTROL_ERROR; + } + 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,const struct mp_chmap *channels,int format,int flags){ - char *mixer_channels [SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES; - int oss_format; - char *mdev = mixer_device, *mchan = mixer_channel; - - mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,ao_data.channels.num, - af_fmt2str_short(format)); - - if (ao_subdevice) { - char *m,*c; - m = strchr(ao_subdevice,':'); - if(m) { - c = strchr(m+1,':'); - if(c) { - mchan = c+1; - c[0] = '\0'; - } - mdev = m+1; - m[0] = '\0'; +static int init(int rate, const struct mp_chmap *channels, int format, + int flags) +{ + char *mixer_channels[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES; + int oss_format; + char *mdev = mixer_device, *mchan = mixer_channel; + + mp_msg(MSGT_AO, MSGL_V, "ao2: %d Hz %d chans %s\n", rate, + ao_data.channels.num, af_fmt2str_short(format)); + + if (ao_subdevice) { + char *m, *c; + m = strchr(ao_subdevice, ':'); + if (m) { + c = strchr(m + 1, ':'); + if (c) { + mchan = c + 1; + c[0] = '\0'; + } + mdev = m + 1; + m[0] = '\0'; + } + dsp = ao_subdevice; } - dsp = ao_subdevice; - } - - if(mdev) - oss_mixer_device=mdev; - else - oss_mixer_device=PATH_DEV_MIXER; - - if(mchan){ - int fd, devs, i; - - if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){ - mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't open mixer device %s: %s\n", - oss_mixer_device, strerror(errno)); - }else{ - ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs); - close(fd); - - for (i=0; i2 channels, in case some drivers don't have it - if (reqchannels > 2) { - int nchannels = reqchannels; - if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &nchannels) == -1 || - nchannels != reqchannels ) { - mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", reqchannels); - return 0; - } + format = AF_FORMAT_S16_NE; + } + if (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format) < 0 || + oss_format != format2oss(format)) + { + mp_tmsg(MSGT_AO, MSGL_WARN, "[AO OSS] Can't set audio device %s to %s " + "output, trying %s...\n", dsp, af_fmt2str_short(format), + af_fmt2str_short(AF_FORMAT_S16_NE)); + format = AF_FORMAT_S16_NE; + goto ac3_retry; } - else { - int c = reqchannels-1; - if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) { - mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", reqchannels); - return 0; - } - if (!ao_chmap_sel_get_def(&ao_data, &sel, &ao_data.channels, c + 1)) + + ao_data.format = oss2format(oss_format); + if (ao_data.format == -1) return 0; + + mp_msg(MSGT_AO, MSGL_V, "audio_setup: sample format: %s (requested: %s)\n", + af_fmt2str_short(ao_data.format), af_fmt2str_short(format)); + + if (!AF_FORMAT_IS_AC3(format)) { + struct mp_chmap_sel sel = {0}; + mp_chmap_sel_add_alsa_def(&sel); + if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels)) + return 0; + int reqchannels = ao_data.channels.num; + // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it + if (reqchannels > 2) { + int nchannels = reqchannels; + if (ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &nchannels) == -1 || + nchannels != reqchannels) + { + mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS] audio_setup: Failed to " + "set audio device to %d channels.\n", reqchannels); + return 0; + } + } else { + int c = reqchannels - 1; + if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &c) == -1) { + mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS] audio_setup: Failed to " + "set audio device to %d channels.\n", reqchannels); + return 0; + } + if (!ao_chmap_sel_get_def(&ao_data, &sel, &ao_data.channels, c + 1)) + return 0; + } + mp_msg(MSGT_AO, MSGL_V, + "audio_setup: using %d channels (requested: %d)\n", + ao_data.channels.num, reqchannels); + // set rate + ao_data.samplerate = rate; + ioctl(audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); + mp_msg(MSGT_AO, MSGL_V, + "audio_setup: using %d Hz samplerate (requested: %d)\n", + ao_data.samplerate, rate); + } + + if (ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz) == -1) { + int r = 0; + mp_tmsg(MSGT_AO, MSGL_WARN, "[AO OSS] audio_setup: driver doesn't " + "support SNDCTL_DSP_GETOSPACE\n"); + if (ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r) == -1) + mp_msg(MSGT_AO, MSGL_V, "audio_setup: %d bytes/frag (config.h)\n", + ao_data.outburst); + else { + ao_data.outburst = r; + mp_msg(MSGT_AO, MSGL_V, "audio_setup: %d bytes/frag (GETBLKSIZE)\n", + ao_data.outburst); + } + } else { + mp_msg(MSGT_AO, MSGL_V, + "audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n", + zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes); + if (ao_data.buffersize == -1) + ao_data.buffersize = zz.bytes; + ao_data.outburst = zz.fragsize; } - mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels.num, reqchannels); - // set rate - ao_data.samplerate=rate; - ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); - mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate); - } - - if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){ - int r=0; - mp_tmsg(MSGT_AO,MSGL_WARN,"[AO OSS] audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n"); - if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){ - mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst); - } else { - ao_data.outburst=r; - mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst); - } - } else { - mp_msg(MSGT_AO,MSGL_V,"audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n", - zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes); - if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes; - ao_data.outburst=zz.fragsize; - } - - if(ao_data.buffersize==-1){ - // Measuring buffer size: - void* data; - ao_data.buffersize=0; + + 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,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; + data = malloc(ao_data.outburst); + memset(data, 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_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS]\n *** Your audio driver " + "DOES NOT support select() ***\n Recompile mpv with " + "#undef HAVE_AUDIO_SELECT in config.h !\n\n"); + return 0; + } +#endif } - free(data); - if(ao_data.buffersize==0){ - mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS]\n *** Your audio driver DOES NOT support select() ***\n Recompile mpv with #undef HAVE_AUDIO_SELECT in config.h !\n\n"); - return 0; + + ao_data.bps = ao_data.channels.num; + switch (ao_data.format & AF_FORMAT_BITS_MASK) { + case AF_FORMAT_8BIT: + break; + case AF_FORMAT_16BIT: + ao_data.bps *= 2; + break; + case AF_FORMAT_24BIT: + ao_data.bps *= 3; + break; + case AF_FORMAT_32BIT: + ao_data.bps *= 4; + break; } -#endif - } - - ao_data.bps=ao_data.channels.num; - switch (ao_data.format & AF_FORMAT_BITS_MASK) { - case AF_FORMAT_8BIT: - break; - case AF_FORMAT_16BIT: - ao_data.bps*=2; - break; - case AF_FORMAT_24BIT: - ao_data.bps*=3; - break; - case AF_FORMAT_32BIT: - ao_data.bps*=4; - break; - } - - ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down - ao_data.bps*=ao_data.samplerate; + + ao_data.outburst -= ao_data.outburst % ao_data.bps; // round down + ao_data.bps *= ao_data.samplerate; return 1; } // close audio device -static void uninit(int immed){ - if(audio_fd == -1) return; +static void uninit(int immed) +{ + if (audio_fd == -1) + return; #ifdef SNDCTL_DSP_SYNC // to get the buffer played if (!immed) - ioctl(audio_fd, SNDCTL_DSP_SYNC, NULL); + ioctl(audio_fd, SNDCTL_DSP_SYNC, NULL); #endif #ifdef SNDCTL_DSP_RESET if (immed) - ioctl(audio_fd, SNDCTL_DSP_RESET, NULL); + ioctl(audio_fd, SNDCTL_DSP_RESET, NULL); #endif close(audio_fd); audio_fd = -1; } // stop playing and empty buffers (for seeking/pause) -static void reset(void){ - int oss_format; +static void reset(void) +{ + int oss_format; uninit(1); - audio_fd=open(dsp, O_WRONLY); - if(audio_fd < 0){ - mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS]\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE *** %s\n", strerror(errno)); - return; + audio_fd = open(dsp, O_WRONLY); + if (audio_fd < 0) { + mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS]\nFatal error: *** CANNOT " + "RE-OPEN / RESET AUDIO DEVICE *** %s\n", strerror(errno)); + return; } #if defined(FD_CLOEXEC) && defined(F_SETFD) - fcntl(audio_fd, F_SETFD, FD_CLOEXEC); -#endif - - oss_format = format2oss(ao_data.format); - if(AF_FORMAT_IS_AC3(ao_data.format)) - ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); - ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format); - if(!AF_FORMAT_IS_AC3(ao_data.format)) { - if (ao_data.channels.num > 2) - ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels.num); - else { - int c = ao_data.channels.num-1; - ioctl (audio_fd, SNDCTL_DSP_STEREO, &c); + fcntl(audio_fd, F_SETFD, FD_CLOEXEC); +#endif + + oss_format = format2oss(ao_data.format); + if (AF_FORMAT_IS_AC3(ao_data.format)) + ioctl(audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); + ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format); + if (!AF_FORMAT_IS_AC3(ao_data.format)) { + if (ao_data.channels.num > 2) + ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels.num); + else { + int c = ao_data.channels.num - 1; + ioctl(audio_fd, SNDCTL_DSP_STEREO, &c); + } + ioctl(audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); } - ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); - } } // stop playing, keep buffers (for pause) @@ -489,72 +518,79 @@ static void audio_resume(void) reset(); fillcnt = get_space() - prepause_space; if (fillcnt > 0 && !(ao_data.format & AF_FORMAT_SPECIAL_MASK)) { - void *silence = calloc(fillcnt, 1); - play(silence, fillcnt, 0); - free(silence); + void *silence = calloc(fillcnt, 1); + play(silence, fillcnt, 0); + free(silence); } } // return: how many bytes can be played without blocking -static int get_space(void){ - int playsize=ao_data.outburst; +static int get_space(void) +{ + int playsize = ao_data.outburst; #ifdef SNDCTL_DSP_GETOSPACE - if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){ - // calculate exact buffer space: - playsize = zz.fragments*zz.fragsize; - return playsize; - } + if (ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz) != -1) { + // calculate exact buffer space: + playsize = zz.fragments * zz.fragsize; + return playsize; + } #endif // 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! + { + 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 - return ao_data.outburst; + return ao_data.outburst; } // 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==0) +static int play(void *data, int len, int flags) +{ + if (len == 0) return len; - if(len>ao_data.outburst || !(flags & AOPLAY_FINAL_CHUNK)) { - len/=ao_data.outburst; - len*=ao_data.outburst; + if (len > ao_data.outburst || !(flags & AOPLAY_FINAL_CHUNK)) { + len /= ao_data.outburst; + len *= ao_data.outburst; } - len=write(audio_fd,data,len); + len = write(audio_fd, data, len); return len; } -static int audio_delay_method=2; +static int audio_delay_method = 2; // return: delay in seconds between first and last sample in buffer -static float get_delay(void){ - /* Calculate how many bytes/second is sent out */ - if(audio_delay_method==2){ +static float get_delay(void) +{ + /* Calculate how many bytes/second is sent out */ + if (audio_delay_method == 2) { #ifdef SNDCTL_DSP_GETODELAY - int r=0; - if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1) - return ((float)r)/(float)ao_data.bps; + int r = 0; + if (ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r) != -1) + return ((float)r) / (float)ao_data.bps; #endif - audio_delay_method=1; // fallback if not supported - } - if(audio_delay_method==1){ - // SNDCTL_DSP_GETOSPACE - if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1) - return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps; - audio_delay_method=0; // fallback if not supported - } - return ((float)ao_data.buffersize)/(float)ao_data.bps; + audio_delay_method = 1; // fallback if not supported + } + if (audio_delay_method == 1) { + // SNDCTL_DSP_GETOSPACE + if (ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz) != -1) { + return ((float)(ao_data.buffersize - + zz.bytes)) / (float)ao_data.bps; + } + audio_delay_method = 0; // fallback if not supported + } + return ((float)ao_data.buffersize) / (float)ao_data.bps; } -- cgit v1.2.3