/* * OSS audio output driver * * 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 * (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 #include #include #include #include #include #include #include #include #include #include "config.h" #include "core/options.h" #include "core/mp_msg.h" #include "audio/mixer.h" #ifdef HAVE_SYS_SOUNDCARD_H #include #else #ifdef HAVE_SOUNDCARD_H #include #endif #endif #include "audio/format.h" #include "ao.h" struct priv { int audio_fd; int prepause_space; int oss_mixer_channel; audio_buf_info zz; int audio_delay_method; int buffersize; int outburst; char *dsp; char *oss_mixer_device; char *cfg_oss_mixer_channel; }; static const char *mixer_channels[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES; static int format_table[][2] = { {AFMT_U8, AF_FORMAT_U8}, {AFMT_S8, AF_FORMAT_S8}, {AFMT_U16_LE, AF_FORMAT_U16_LE}, {AFMT_U16_BE, AF_FORMAT_U16_BE}, {AFMT_S16_LE, AF_FORMAT_S16_LE}, {AFMT_S16_BE, AF_FORMAT_S16_BE}, #ifdef AFMT_S24_PACKED {AFMT_S24_PACKED, AF_FORMAT_S24_LE}, #endif #ifdef AFMT_U32_LE {AFMT_U32_LE, AF_FORMAT_U32_LE}, #endif #ifdef AFMT_U32_BE {AFMT_U32_BE, AF_FORMAT_U32_BE}, #endif #ifdef AFMT_S32_LE {AFMT_S32_LE, AF_FORMAT_S32_LE}, #endif #ifdef AFMT_S32_BE {AFMT_S32_BE, AF_FORMAT_S32_BE}, #endif #ifdef AFMT_FLOAT {AFMT_FLOAT, AF_FORMAT_FLOAT_NE}, #endif // SPECIALS #ifdef AFMT_MPEG {AFMT_MPEG, AF_FORMAT_MPEG2}, #endif #ifdef AFMT_AC3 {AFMT_AC3, AF_FORMAT_AC3_NE}, #endif {-1, -1} }; static int format2oss(int format) { for (int n = 0; format_table[n][0] != -1; n++) { if (format_table[n][1] == format) return format_table[n][0]; } 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) { for (int n = 0; format_table[n][0] != -1; n++) { if (format_table[n][0] == format) return format_table[n][1]; } mp_tmsg(MSGT_GLOBAL, MSGL_ERR, "[AO OSS] Unknown/Unsupported OSS format: %x.\n", format); return -1; } #ifdef SNDCTL_DSP_GETPLAYVOL static int volume_oss4(struct ao *ao, ao_control_vol_t *vol, int cmd) { struct priv *p = ao->priv; int v; if (p->audio_fd < 0) return CONTROL_ERROR; if (cmd == AOCONTROL_GET_VOLUME) { if (ioctl(p->audio_fd, SNDCTL_DSP_GETPLAYVOL, &v) == -1) return CONTROL_ERROR; vol->right = (v & 0xff00) >> 8; vol->left = v & 0x00ff; return CONTROL_OK; } else if (cmd == AOCONTROL_SET_VOLUME) { v = ((int) vol->right << 8) | (int) vol->left; if (ioctl(p->audio_fd, SNDCTL_DSP_SETPLAYVOL, &v) == -1) return CONTROL_ERROR; return CONTROL_OK; } else return CONTROL_UNKNOWN; } #endif // to set/get/query special features/parameters static int control(struct ao *ao, enum aocontrol cmd, void *arg) { struct priv *p = ao->priv; 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 if (volume_oss4(ao, vol, cmd) == CONTROL_OK) return CONTROL_OK; #endif if (AF_FORMAT_IS_AC3(ao->format)) return CONTROL_TRUE; if ((fd = open(p->oss_mixer_device, O_RDONLY)) != -1) { ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs); if (devs & (1 << p->oss_mixer_channel)) { if (cmd == AOCONTROL_GET_VOLUME) { ioctl(fd, MIXER_READ(p->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(p->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: 0=success -1=fail static int init(struct ao *ao, char *params) { struct priv *p = ao->priv; int oss_format; const char *mchan = NULL; if (p->cfg_oss_mixer_channel && p->cfg_oss_mixer_channel[0]) mchan = p->cfg_oss_mixer_channel; mp_msg(MSGT_AO, MSGL_V, "ao2: %d Hz %d chans %s\n", ao->samplerate, ao->channels.num, af_fmt2str_short(ao->format)); if (mchan) { int fd, devs, i; if ((fd = open(p->oss_mixer_device, O_RDONLY)) == -1) { mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS] audio_setup: Can't open mixer device %s: %s\n", p->oss_mixer_device, strerror(errno)); } else { ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs); close(fd); for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) { if (!strcasecmp(mixer_channels[i], mchan)) { if (!(devs & (1 << i))) { mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS] audio_setup: " "Audio card mixer does not have channel '%s', " "using default.\n", mchan); i = SOUND_MIXER_NRDEVICES + 1; break; } p->oss_mixer_channel = i; break; } } if (i == SOUND_MIXER_NRDEVICES) { mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS] audio_setup: Audio card " "mixer does not have channel '%s', using default.\n", mchan); } } } else { p->oss_mixer_channel = SOUND_MIXER_PCM; } mp_msg(MSGT_AO, MSGL_V, "audio_setup: using '%s' dsp device\n", p->dsp); mp_msg(MSGT_AO, MSGL_V, "audio_setup: using '%s' mixer device\n", p->oss_mixer_device); mp_msg(MSGT_AO, MSGL_V, "audio_setup: using '%s' mixer device\n", mixer_channels[p->oss_mixer_channel]); #ifdef __linux__ p->audio_fd = open(p->dsp, O_WRONLY | O_NONBLOCK); #else p->audio_fd = open(p->dsp, O_WRONLY); #endif if (p->audio_fd < 0) { mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS] audio_setup: Can't open audio device %s: %s\n", p->dsp, strerror(errno)); return -1; } #ifdef __linux__ /* Remove the non-blocking flag */ if (fcntl(p->audio_fd, F_SETFL, 0) < 0) { mp_tmsg(MSGT_AO, MSGL_ERR, "[AO OSS] audio_setup: Can't make file " "descriptor blocking: %s\n", strerror(errno)); return -1; } #endif #if defined(FD_CLOEXEC) && defined(F_SETFD) fcntl(p->audio_fd, F_SETFD, FD_CLOEXEC); #endif if (AF_FORMAT_IS_AC3(ao->format)) { ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &ao->samplerate); } ac3_retry: if (AF_FORMAT_IS_AC3(ao->format)) ao->format = AF_FORMAT_AC3_NE; oss_format = format2oss(ao->format); if (oss_format == -1) { #if BYTE_ORDER == BIG_ENDIAN oss_format = AFMT_S16_BE; #else oss_format = AFMT_S16_LE; #endif ao->format = AF_FORMAT_S16_NE; } if (ioctl(p->audio_fd, SNDCTL_DSP_SETFMT, &oss_format) < 0 || oss_format != format2oss(ao->format)) { mp_tmsg(MSGT_AO, MSGL_WARN, "[AO OSS] Can't set audio device %s to %s " "output, trying %s...\n", p->dsp, af_fmt2str_short(ao->format), af_fmt2str_short(AF_FORMAT_S16_NE)); ao->format = AF_FORMAT_S16_NE; goto ac3_retry; } ao->format = oss2format(oss_format); if (ao->format == -1) return -1; mp_msg(MSGT_AO, MSGL_V, "audio_setup: sample format: %s\n", af_fmt2str_short(ao->format)); if (!AF_FORMAT_IS_AC3(ao->format)) { struct mp_chmap_sel sel = {0}; mp_chmap_sel_add_alsa_def(&sel); if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels)) return -1; int reqchannels = ao->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(p->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 -1; } } else { int c = reqchannels - 1; if (ioctl(p->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 -1; } if (!ao_chmap_sel_get_def(ao, &sel, &ao->channels, c + 1)) return -1; } mp_msg(MSGT_AO, MSGL_V, "audio_setup: using %d channels (requested: %d)\n", ao->channels.num, reqchannels); // set rate ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &ao->samplerate); mp_msg(MSGT_AO, MSGL_V, "audio_setup: using %d Hz samplerate\n", ao->samplerate); } if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &p->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(p->audio_fd, SNDCTL_DSP_GETBLKSIZE, &r) == -1) mp_msg(MSGT_AO, MSGL_V, "audio_setup: %d bytes/frag (config.h)\n", p->outburst); else { p->outburst = r; mp_msg(MSGT_AO, MSGL_V, "audio_setup: %d bytes/frag (GETBLKSIZE)\n", p->outburst); } } else { mp_msg(MSGT_AO, MSGL_V, "audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n", p->zz.fragments, p->zz.fragstotal, p->zz.fragsize, p->zz.bytes); p->buffersize = p->zz.bytes; p->outburst = p->zz.fragsize; } if (p->buffersize == -1) { // Measuring buffer size: void *data; p->buffersize = 0; #ifdef HAVE_AUDIO_SELECT data = malloc(p->outburst); memset(data, 0, p->outburst); while (p->buffersize < 0x40000) { fd_set rfds; struct timeval tv; FD_ZERO(&rfds); FD_SET(p->audio_fd, &rfds); tv.tv_sec = 0; tv.tv_usec = 0; if (!select(p->audio_fd + 1, NULL, &rfds, NULL, &tv)) break; write(p->audio_fd, data, p->outburst); p->buffersize += p->outburst; } free(data); if (p->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 -1; } #endif } ao->bps = ao->channels.num * (af_fmt2bits(ao->format) / 8); p->outburst -= p->outburst % ao->bps; // round down ao->bps *= ao->samplerate; return 0; } // close audio device static void uninit(struct ao *ao, bool immed) { struct priv *p = ao->priv; if (p->audio_fd == -1) return; #ifdef SNDCTL_DSP_SYNC // to get the buffer played if (!immed) ioctl(p->audio_fd, SNDCTL_DSP_SYNC, NULL); #endif #ifdef SNDCTL_DSP_RESET if (immed) ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL); #endif close(p->audio_fd); p->audio_fd = -1; } static void close_device(struct ao *ao) { struct priv *p = ao->priv; #ifdef SNDCTL_DSP_RESET ioctl(p->audio_fd, SNDCTL_DSP_RESET, NULL); #endif close(p->audio_fd); p->audio_fd = -1; } // stop playing and empty buffers (for seeking/pause) static void reset(struct ao *ao) { struct priv *p = ao->priv; int oss_format; close_device(ao); p->audio_fd = open(p->dsp, O_WRONLY); if (p->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(p->audio_fd, F_SETFD, FD_CLOEXEC); #endif oss_format = format2oss(ao->format); if (AF_FORMAT_IS_AC3(ao->format)) ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &ao->samplerate); ioctl(p->audio_fd, SNDCTL_DSP_SETFMT, &oss_format); if (!AF_FORMAT_IS_AC3(ao->format)) { if (ao->channels.num > 2) ioctl(p->audio_fd, SNDCTL_DSP_CHANNELS, &ao->channels.num); else { int c = ao->channels.num - 1; ioctl(p->audio_fd, SNDCTL_DSP_STEREO, &c); } ioctl(p->audio_fd, SNDCTL_DSP_SPEED, &ao->samplerate); } } // return: how many bytes can be played without blocking static int get_space(struct ao *ao) { struct priv *p = ao->priv; int playsize = p->outburst; #ifdef SNDCTL_DSP_GETOSPACE if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &p->zz) != -1) { // calculate exact buffer space: playsize = p->zz.fragments * p->zz.fragsize; return playsize; } #endif // check buffer #ifdef HAVE_AUDIO_SELECT { fd_set rfds; struct timeval tv; FD_ZERO(&rfds); FD_SET(p->audio_fd, &rfds); tv.tv_sec = 0; tv.tv_usec = 0; if (!select(p->audio_fd + 1, NULL, &rfds, NULL, &tv)) return 0; // not block! } #endif return p->outburst; } // stop playing, keep buffers (for pause) static void audio_pause(struct ao *ao) { struct priv *p = ao->priv; p->prepause_space = get_space(ao); close_device(ao); } // plays 'len' bytes of 'data' // it should round it down to outburst*n // return: number of bytes played static int play(struct ao *ao, void *data, int len, int flags) { struct priv *p = ao->priv; if (len == 0) return len; if (len > p->outburst || !(flags & AOPLAY_FINAL_CHUNK)) { len /= p->outburst; len *= p->outburst; } len = write(p->audio_fd, data, len); return len; } // resume playing, after audio_pause() static void audio_resume(struct ao *ao) { struct priv *p = ao->priv; int fillcnt; reset(ao); fillcnt = get_space(ao) - p->prepause_space; if (fillcnt > 0 && !(ao->format & AF_FORMAT_SPECIAL_MASK)) { void *silence = calloc(fillcnt, 1); play(ao, silence, fillcnt, 0); free(silence); } } // return: delay in seconds between first and last sample in buffer static float get_delay(struct ao *ao) { struct priv *p = ao->priv; /* Calculate how many bytes/second is sent out */ if (p->audio_delay_method == 2) { #ifdef SNDCTL_DSP_GETODELAY int r = 0; if (ioctl(p->audio_fd, SNDCTL_DSP_GETODELAY, &r) != -1) return ((float)r) / (float)ao->bps; #endif p->audio_delay_method = 1; // fallback if not supported } if (p->audio_delay_method == 1) { // SNDCTL_DSP_GETOSPACE if (ioctl(p->audio_fd, SNDCTL_DSP_GETOSPACE, &p->zz) != -1) { return ((float)(p->buffersize - p->zz.bytes)) / (float)ao->bps; } p->audio_delay_method = 0; // fallback if not supported } return ((float)p->buffersize) / (float)ao->bps; } #define OPT_BASE_STRUCT struct priv const struct ao_driver audio_out_oss = { .info = &(const struct ao_info) { "OSS/ioctl audio output", "oss", "A'rpi", "" }, .init = init, .uninit = uninit, .control = control, .get_space = get_space, .play = play, .get_delay = get_delay, .pause = audio_pause, .resume = audio_resume, .reset = reset, .priv_size = sizeof(struct priv), .priv_defaults = &(const struct priv) { .audio_fd = -1, .audio_delay_method = 2, .buffersize = -1, .outburst = 512, .oss_mixer_channel = SOUND_MIXER_PCM, .dsp = PATH_DEV_DSP, .oss_mixer_device = PATH_DEV_MIXER, }, .options = (const struct m_option[]) { OPT_STRING("device", dsp, 0), OPT_STRING("mixer-device", oss_mixer_device, 0), OPT_STRING("mixer-channel", cfg_oss_mixer_channel, 0), {0} }, };