mpv/audio/out/ao_oss.c

/*
 * OSS audio output driver
 *
 * This file is part of MPlayer.
 *
 * Support for >2 output channels added 2001-11-25
 * - Steve Davies <steve@daviesfam.org>
 *
 * 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 <sys/ioctl.h>
#include <unistd.h>
#include <sys/time.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <errno.h>
#include <string.h>

#include "config.h"
#include "core/mp_msg.h"
#include "audio/mixer.h"

#ifdef HAVE_SYS_SOUNDCARD_H
#include <sys/soundcard.h>
#else
#ifdef HAVE_SOUNDCARD_H
#include <soundcard.h>
#endif
#endif

#include "audio/format.h"

#include "ao.h"
#include "audio_out_internal.h"

static const ao_info_t info =
{
    "OSS/ioctl audio output",
    "oss",
    "A'rpi",
    ""
};

LIBAO_EXTERN(oss)

static int format2oss(int 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;
    case AF_FORMAT_U16_BE: return AFMT_U16_BE;
    case AF_FORMAT_S16_LE: return AFMT_S16_LE;
    case AF_FORMAT_S16_BE: return AFMT_S16_BE;
#ifdef AFMT_S24_PACKED
    case AF_FORMAT_S24_LE: return AFMT_S24_PACKED;
#endif
#ifdef AFMT_U32_LE
    case AF_FORMAT_U32_LE: return AFMT_U32_LE;
#endif
#ifdef AFMT_U32_BE
    case AF_FORMAT_U32_BE: return AFMT_U32_BE;
#endif
#ifdef AFMT_S32_LE
    case AF_FORMAT_S32_LE: return AFMT_S32_LE;
#endif
#ifdef AFMT_S32_BE
    case AF_FORMAT_S32_BE: return AFMT_S32_BE;
#endif
#ifdef AFMT_FLOAT
    case AF_FORMAT_FLOAT_NE: return AFMT_FLOAT;
#endif
        // SPECIALS
#ifdef AFMT_MPEG
    case AF_FORMAT_MPEG2: return AFMT_MPEG;
#endif
#ifdef AFMT_AC3
    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));
    return -1;
}

static int oss2format(int 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;
    case AFMT_U16_BE: return AF_FORMAT_U16_BE;
    case AFMT_S16_LE: return AF_FORMAT_S16_LE;
    case AFMT_S16_BE: return AF_FORMAT_S16_BE;
#ifdef AFMT_S24_PACKED
    case AFMT_S24_PACKED: return AF_FORMAT_S24_LE;
#endif
#ifdef AFMT_U32_LE
    case AFMT_U32_LE: return AF_FORMAT_U32_LE;
#endif
#ifdef AFMT_U32_BE
    case AFMT_U32_BE: return AF_FORMAT_U32_BE;
#endif
#ifdef AFMT_S32_LE
    case AFMT_S32_LE: return AF_FORMAT_S32_LE;
#endif
#ifdef AFMT_S32_BE
    case AFMT_S32_BE: return AF_FORMAT_S32_BE;
#endif
#ifdef AFMT_FLOAT
    case AFMT_FLOAT: return AF_FORMAT_FLOAT_NE;
#endif
        // SPECIALS
#ifdef AFMT_MPEG
    case AFMT_MPEG: return AF_FORMAT_MPEG2;
#endif
#ifdef AFMT_AC3
    case AFMT_AC3: return AF_FORMAT_AC3_NE;
#endif
    }
    mp_tmsg(MSGT_GLOBAL, MSGL_ERR, "[AO OSS] Unknown/Unsupported OSS format: %x.\n",
            format);
    return -1;
}

static char *dsp = PATH_DEV_DSP;
static audio_buf_info zz;
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)
{
    int v;

    if (audio_fd < 0)
        return CONTROL_ERROR;

    if (cmd == AOCONTROL_GET_VOLUME) {
        if (ioctl(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(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(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
        if (volume_oss4(vol, cmd) == CONTROL_OK)
            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;
    }
    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';
        }
        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; 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;
                    }
                    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 {
        oss_mixer_channel = SOUND_MIXER_PCM;
    }

    mp_msg(MSGT_AO, MSGL_V, "audio_setup: using '%s' dsp device\n", dsp);
    mp_msg(MSGT_AO, MSGL_V, "audio_setup: using '%s' mixer device\n",
           oss_mixer_device);
    mp_msg(MSGT_AO, MSGL_V, "audio_setup: using '%s' mixer device\n",
           mixer_channels[oss_mixer_channel]);

#ifdef __linux__
    audio_fd = open(dsp, O_WRONLY | O_NONBLOCK);
#else
    audio_fd = open(dsp, O_WRONLY);
#endif
    if (audio_fd < 0) {
        mp_tmsg(MSGT_AO, MSGL_ERR,
                "[AO OSS] audio_setup: Can't open audio device %s: %s\n",
                dsp, strerror(errno));
        return 0;
    }

#ifdef __linux__
    /* Remove the non-blocking flag */
    if (fcntl(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 0;
    }
#endif

#if defined(FD_CLOEXEC) && defined(F_SETFD)
    fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
#endif

    if (AF_FORMAT_IS_AC3(format)) {
        ao_data.samplerate = rate;
        ioctl(audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
    }

ac3_retry:
    if (AF_FORMAT_IS_AC3(format))
        format = AF_FORMAT_AC3_NE;
    ao_data.format = format;
    oss_format = format2oss(format);
    if (oss_format == -1) {
#if BYTE_ORDER == BIG_ENDIAN
        oss_format = AFMT_S16_BE;
#else
        oss_format = AFMT_S16_LE;
#endif
        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;
    }

    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;
    }

    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;
        }
        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
    }

    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;

    return 1;
}

// close audio device
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);
#endif
#ifdef SNDCTL_DSP_RESET
    if (immed)
        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;
    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;
    }

#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);
        }
        ioctl(audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
    }
}

// stop playing, keep buffers (for pause)
static void audio_pause(void)
{
    prepause_space = get_space();
    uninit(1);
}

// resume playing, after audio_pause()
static void audio_resume(void)
{
    int fillcnt;
    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);
    }
}


// return: how many bytes can be played without blocking
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;
    }
#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!
    }
#endif

    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)
        return len;
    if (len > ao_data.outburst || !(flags & AOPLAY_FINAL_CHUNK)) {
        len /= ao_data.outburst;
        len *= ao_data.outburst;
    }
    len = write(audio_fd, data, len);
    return len;
}

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) {
#ifdef SNDCTL_DSP_GETODELAY
        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;
}