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/*
 * ALSA 0.9.x-1.x audio output driver
 *
 * Copyright (C) 2004 Alex Beregszaszi
 *
 * modified for real ALSA 0.9.0 support by Zsolt Barat <joy@streamminister.de>
 * additional AC-3 passthrough support by Andy Lo A Foe <andy@alsaplayer.org>
 * 08/22/2002 iec958-init rewritten and merged with common init, zsolt
 * 04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka
 * 04/25/2004 printfs converted to mp_msg, Zsolt.
 *
 * This file is part of MPlayer.
 *
 * 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 <errno.h>
#include <sys/time.h>
#include <stdlib.h>
#include <stdarg.h>
#include <ctype.h>
#include <math.h>
#include <string.h>
#include <alloca.h>

#include "config.h"
#include "core/subopt-helper.h"
#include "audio/mixer.h"
#include "core/mp_msg.h"

#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API

#include <alsa/asoundlib.h>

#include "ao.h"
#include "audio_out_internal.h"
#include "audio/format.h"
#include "audio/reorder_ch.h"

static const ao_info_t info =
{
    "ALSA-0.9.x-1.x audio output",
    "alsa",
    "Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
    "under development"
};

LIBAO_EXTERN(alsa)

static snd_pcm_t * alsa_handler;
static snd_pcm_format_t alsa_format;

#define BUFFER_TIME 500000  // 0.5 s
#define FRAGCOUNT 16

static size_t bytes_per_sample;

static int alsa_can_pause;
static snd_pcm_sframes_t prepause_frames;
static float delay_before_pause;

#define ALSA_DEVICE_SIZE 256

static void alsa_error_handler(const char *file, int line, const char *function,
                               int err, const char *format, ...)
{
    char tmp[0xc00];
    va_list va;

    va_start(va, format);
    vsnprintf(tmp, sizeof tmp, format, va);
    va_end(va);

    if (err)
        mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
               file, line, function, tmp, snd_strerror(err));
    else
        mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
               file, line, function, tmp);
}

/* to set/get/query special features/parameters */
static int control(int cmd, void *arg)
{
    switch (cmd) {
    case AOCONTROL_GET_MUTE:
    case AOCONTROL_SET_MUTE:
    case AOCONTROL_GET_VOLUME:
    case AOCONTROL_SET_VOLUME:
    {
        int err;
        snd_mixer_t *handle;
        snd_mixer_elem_t *elem;
        snd_mixer_selem_id_t *sid;

        char *mix_name = "Master";
        char *card = "default";
        int mix_index = 0;

        long pmin, pmax;
        long get_vol, set_vol;
        float f_multi;

        if (AF_FORMAT_IS_IEC61937(ao_data.format))
            return CONTROL_TRUE;

        if (mixer_channel) {
            char *test_mix_index;

            mix_name = strdup(mixer_channel);
            if ((test_mix_index = strchr(mix_name, ','))) {
                *test_mix_index = 0;
                test_mix_index++;
                mix_index = strtol(test_mix_index, &test_mix_index, 0);

                if (*test_mix_index) {
                    mp_tmsg(MSGT_AO, MSGL_ERR,
                            "[AO_ALSA] Invalid mixer index. Defaulting to 0.\n");
                    mix_index = 0;
                }
            }
        }
        if (mixer_device)
            card = mixer_device;

        //allocate simple id
        snd_mixer_selem_id_alloca(&sid);

        //sets simple-mixer index and name
        snd_mixer_selem_id_set_index(sid, mix_index);
        snd_mixer_selem_id_set_name(sid, mix_name);

        if (mixer_channel) {
            free(mix_name);
            mix_name = NULL;
        }

        if ((err = snd_mixer_open(&handle, 0)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Mixer open error: %s\n",
                    snd_strerror(
                        err));
            return CONTROL_ERROR;
        }

        if ((err = snd_mixer_attach(handle, card)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Mixer attach %s error: %s\n",
                    card, snd_strerror(err));
            snd_mixer_close(handle);
            return CONTROL_ERROR;
        }

        if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Mixer register error: %s\n",
                    snd_strerror(
                        err));
            snd_mixer_close(handle);
            return CONTROL_ERROR;
        }
        err = snd_mixer_load(handle);
        if (err < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Mixer load error: %s\n",
                    snd_strerror(
                        err));
            snd_mixer_close(handle);
            return CONTROL_ERROR;
        }

        elem = snd_mixer_find_selem(handle, sid);
        if (!elem) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] Unable to find simple control '%s',%i.\n",
                    snd_mixer_selem_id_get_name(
                        sid), snd_mixer_selem_id_get_index(sid));
            snd_mixer_close(handle);
            return CONTROL_ERROR;
        }

        snd_mixer_selem_get_playback_volume_range(elem, &pmin, &pmax);
        f_multi = (100 / (float)(pmax - pmin));

        switch (cmd) {
        case AOCONTROL_SET_VOLUME: {
            ao_control_vol_t *vol = arg;
            set_vol = vol->left / f_multi + pmin + 0.5;

            //setting channels
            if ((err =
                     snd_mixer_selem_set_playback_volume(elem,
                                                         SND_MIXER_SCHN_FRONT_LEFT,
                                                         set_vol)) < 0) {
                mp_tmsg(MSGT_AO, MSGL_ERR,
                        "[AO_ALSA] Error setting left channel, %s\n",
                        snd_strerror(
                            err));
                goto mixer_error;
            }
            mp_msg(MSGT_AO, MSGL_DBG2, "left=%li, ", set_vol);

            set_vol = vol->right / f_multi + pmin + 0.5;

            if ((err =
                     snd_mixer_selem_set_playback_volume(elem,
                                                         SND_MIXER_SCHN_FRONT_RIGHT,
                                                         set_vol)) < 0) {
                mp_tmsg(MSGT_AO, MSGL_ERR,
                        "[AO_ALSA] Error setting right channel, %s\n",
                        snd_strerror(
                            err));
                goto mixer_error;
            }
            mp_msg(MSGT_AO, MSGL_DBG2,
                   "right=%li, pmin=%li, pmax=%li, mult=%f\n",
                   set_vol, pmin, pmax,
                   f_multi);
            break;
        }
        case AOCONTROL_GET_VOLUME: {
            ao_control_vol_t *vol = arg;
            snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT,
                                                &get_vol);
            vol->left = (get_vol - pmin) * f_multi;
            snd_mixer_selem_get_playback_volume(elem,
                                                SND_MIXER_SCHN_FRONT_RIGHT,
                                                &get_vol);
            vol->right = (get_vol - pmin) * f_multi;
            mp_msg(MSGT_AO, MSGL_DBG2, "left=%f, right=%f\n", vol->left,
                   vol->right);
            break;
        }
        case AOCONTROL_SET_MUTE: {
            bool *mute = arg;
            if (!snd_mixer_selem_has_playback_switch(elem))
                goto mixer_error;
            if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
                snd_mixer_selem_set_playback_switch(
                    elem, SND_MIXER_SCHN_FRONT_RIGHT, !*mute);
            }
            snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT,
                                                !*mute);
            break;
        }
        case AOCONTROL_GET_MUTE: {
            bool *mute = arg;
            if (!snd_mixer_selem_has_playback_switch(elem))
                goto mixer_error;
            int tmp = 1;
            snd_mixer_selem_get_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT,
                                                &tmp);
            *mute = !tmp;
            if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
                snd_mixer_selem_get_playback_switch(
                    elem, SND_MIXER_SCHN_FRONT_RIGHT, &tmp);
                *mute &= !tmp;
            }
            break;
        }
        }
        snd_mixer_close(handle);
        return CONTROL_OK;
mixer_error:
        snd_mixer_close(handle);
        return CONTROL_ERROR;
    }

    } //end switch
    return CONTROL_UNKNOWN;
}

static void parse_device(char *dest, const char *src, int len)
{
    char *tmp;
    memmove(dest, src, len);
    dest[len] = 0;
    while ((tmp = strrchr(dest, '.')))
        tmp[0] = ',';
    while ((tmp = strrchr(dest, '=')))
        tmp[0] = ':';
}

static void print_help(void)
{
    mp_tmsg(MSGT_AO, MSGL_FATAL,
            "\n[AO_ALSA] -ao alsa commandline help:\n" \
            "[AO_ALSA] Example: mpv -ao alsa:device=hw=0.3\n" \
            "[AO_ALSA]   Sets first card fourth hardware device.\n\n" \
            "[AO_ALSA] Options:\n" \
            "[AO_ALSA]   noblock\n" \
            "[AO_ALSA]     Opens device in non-blocking mode.\n" \
            "[AO_ALSA]   device=<device-name>\n" \
            "[AO_ALSA]     Sets device (change , to . and : to =)\n");
}

static int str_maxlen(void *strp)
{
    strarg_t *str = strp;
    return str->len <= ALSA_DEVICE_SIZE;
}

static int try_open_device(const char *device, int open_mode, int try_ac3)
{
    int err, len;
    char *ac3_device, *args;

    if (try_ac3) {
        /* to set the non-audio bit, use AES0=6 */
        len = strlen(device);
        ac3_device = malloc(len + 7 + 1);
        if (!ac3_device)
            return -ENOMEM;
        strcpy(ac3_device, device);
        args = strchr(ac3_device, ':');
        if (!args) {
            /* no existing parameters: add it behind device name */
            strcat(ac3_device, ":AES0=6");
        } else {
            do
                ++args;
            while (isspace(*args));
            if (*args == '\0') {
                /* ":" but no parameters */
                strcat(ac3_device, "AES0=6");
            } else if (*args != '{') {
                /* a simple list of parameters: add it at the end of the list */
                strcat(ac3_device, ",AES0=6");
            } else {
                /* parameters in config syntax: add it inside the { } block */
                do
                    --len;
                while (len > 0 && isspace(ac3_device[len]));
                if (ac3_device[len] == '}')
                    strcpy(ac3_device + len, " AES0=6}");
            }
        }
        err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
                           open_mode);
        free(ac3_device);
        if (!err)
            return 0;
    }
    return snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
                        open_mode);
}

/*
    open & setup audio device
    return: 1=success 0=fail
 */
static int init(int rate_hz, const struct mp_chmap *channels, int format,
                int flags)
{
    int err;
    int block;
    strarg_t device;
    snd_pcm_uframes_t chunk_size;
    snd_pcm_uframes_t bufsize;
    snd_pcm_uframes_t boundary;
    const opt_t subopts[] = {
        {"block", OPT_ARG_BOOL, &block, NULL},
        {"device", OPT_ARG_STR, &device, str_maxlen},
        {NULL}
    };

    char alsa_device[ALSA_DEVICE_SIZE + 1];
    // make sure alsa_device is null-terminated even when using strncpy etc.
    memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);

    mp_msg(MSGT_AO, MSGL_V,
           "alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
           ao_data.channels.num,
           format);
    alsa_handler = NULL;
    mp_msg(MSGT_AO, MSGL_V, "alsa-init: using ALSA %s\n", snd_asoundlib_version());

    prepause_frames = 0;
    delay_before_pause = 0;

    snd_lib_error_set_handler(alsa_error_handler);

    switch (format) {
    case AF_FORMAT_S8:
        alsa_format = SND_PCM_FORMAT_S8;
        break;
    case AF_FORMAT_U8:
        alsa_format = SND_PCM_FORMAT_U8;
        break;
    case AF_FORMAT_U16_LE:
        alsa_format = SND_PCM_FORMAT_U16_LE;
        break;
    case AF_FORMAT_U16_BE:
        alsa_format = SND_PCM_FORMAT_U16_BE;
        break;
    case AF_FORMAT_AC3_LE:
    case AF_FORMAT_S16_LE:
    case AF_FORMAT_IEC61937_LE:
        alsa_format = SND_PCM_FORMAT_S16_LE;
        break;
    case AF_FORMAT_AC3_BE:
    case AF_FORMAT_S16_BE:
    case AF_FORMAT_IEC61937_BE:
        alsa_format = SND_PCM_FORMAT_S16_BE;
        break;
    case AF_FORMAT_U32_LE:
        alsa_format = SND_PCM_FORMAT_U32_LE;
        break;
    case AF_FORMAT_U32_BE:
        alsa_format = SND_PCM_FORMAT_U32_BE;
        break;
    case AF_FORMAT_S32_LE:
        alsa_format = SND_PCM_FORMAT_S32_LE;
        break;
    case AF_FORMAT_S32_BE:
        alsa_format = SND_PCM_FORMAT_S32_BE;
        break;
    case AF_FORMAT_U24_LE:
        alsa_format = SND_PCM_FORMAT_U24_3LE;
        break;
    case AF_FORMAT_U24_BE:
        alsa_format = SND_PCM_FORMAT_U24_3BE;
        break;
    case AF_FORMAT_S24_LE:
        alsa_format = SND_PCM_FORMAT_S24_3LE;
        break;
    case AF_FORMAT_S24_BE:
        alsa_format = SND_PCM_FORMAT_S24_3BE;
        break;
    case AF_FORMAT_FLOAT_LE:
        alsa_format = SND_PCM_FORMAT_FLOAT_LE;
        break;
    case AF_FORMAT_FLOAT_BE:
        alsa_format = SND_PCM_FORMAT_FLOAT_BE;
        break;

    default:
        alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
        break;
    }

    //subdevice parsing
    // set defaults
    block = 1;
    /* switch for spdif
     * sets opening sequence for SPDIF
     * sets also the playback and other switches 'on the fly'
     * while opening the abstract alias for the spdif subdevice
     * 'iec958'
     */
    if (AF_FORMAT_IS_IEC61937(format)) {
        device.str = "iec958";
        mp_msg(MSGT_AO, MSGL_V,
               "alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n",
               ao_data.channels.num);
    } else
        /* in any case for multichannel playback we should select
         * appropriate device
         */
        switch (ao_data.channels.num) {
        case 1:
        case 2:
            device.str = "default";
            mp_msg(MSGT_AO, MSGL_V, "alsa-init: setup for 1/2 channel(s)\n");
            break;
        case 4:
            if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
                // hack - use the converter plugin
                device.str = "plug:surround40";
            else
                device.str = "surround40";
            mp_msg(MSGT_AO, MSGL_V, "alsa-init: device set to surround40\n");
            break;
        case 6:
            if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
                device.str = "plug:surround51";
            else
                device.str = "surround51";
            mp_msg(MSGT_AO, MSGL_V, "alsa-init: device set to surround51\n");
            break;
        case 8:
            if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
                device.str = "plug:surround71";
            else
                device.str = "surround71";
            mp_msg(MSGT_AO, MSGL_V, "alsa-init: device set to surround71\n");
            break;
        default:
            device.str = "default";
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] %d channels are not supported.\n",
                    ao_data.channels.num);
        }
    device.len = strlen(device.str);
    if (subopt_parse(ao_subdevice, subopts) != 0) {
        print_help();
        return 0;
    }
    parse_device(alsa_device, device.str, device.len);

    mp_msg(MSGT_AO, MSGL_V, "alsa-init: using device %s\n", alsa_device);

    alsa_can_pause = 1;

    if (!alsa_handler) {
        int open_mode = block ? 0 : SND_PCM_NONBLOCK;
        int isac3 =  AF_FORMAT_IS_IEC61937(format);
        //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
        if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0) {
            if (err != -EBUSY && !block) {
                mp_tmsg(
                    MSGT_AO, MSGL_INFO,
                    "[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
                if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
                    mp_tmsg(MSGT_AO, MSGL_ERR,
                            "[AO_ALSA] Playback open error: %s\n", snd_strerror(
                                err));
                    return 0;
                }
            } else {
                mp_tmsg(MSGT_AO, MSGL_ERR,
                        "[AO_ALSA] Playback open error: %s\n", snd_strerror(
                            err));
                return 0;
            }
        }

        if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0)
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(
                        err));
        else
            mp_msg(MSGT_AO, MSGL_V, "alsa-init: pcm opened in blocking mode\n");

        snd_pcm_hw_params_t *alsa_hwparams;
        snd_pcm_sw_params_t *alsa_swparams;

        snd_pcm_hw_params_alloca(&alsa_hwparams);
        snd_pcm_sw_params_alloca(&alsa_swparams);

        // setting hw-parameters
        if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] Unable to get initial parameters: %s\n",
                    snd_strerror(
                        err));
            return 0;
        }

        err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
                                           SND_PCM_ACCESS_RW_INTERLEAVED);
        if (err < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] Unable to set access type: %s\n",
                    snd_strerror(
                        err));
            return 0;
        }

        /* workaround for nonsupported formats
           sets default format to S16_LE if the given formats aren't supported */
        if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
                                                 alsa_format)) < 0) {
            mp_tmsg(
                MSGT_AO, MSGL_INFO,
                "[AO_ALSA] Format %s is not supported by hardware, trying default.\n",
                af_fmt2str_short(format));
            alsa_format = SND_PCM_FORMAT_S16_LE;
            if (AF_FORMAT_IS_AC3(ao_data.format))
                ao_data.format = AF_FORMAT_AC3_LE;
            else if (AF_FORMAT_IS_IEC61937(ao_data.format))
                ao_data.format = AF_FORMAT_IEC61937_LE;
            else
                ao_data.format = AF_FORMAT_S16_LE;
        }

        if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
                                                alsa_format)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Unable to set format: %s\n",
                    snd_strerror(err));
            return 0;
        }

        int num_channels = ao_data.channels.num;
        if ((err =
                 snd_pcm_hw_params_set_channels_near(alsa_handler,
                                                     alsa_hwparams,
                                                     &num_channels)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Unable to set channels: %s\n",
                    snd_strerror(err));
            return 0;
        }
        mp_chmap_from_channels(&ao_data.channels, num_channels);
        if (!AF_FORMAT_IS_IEC61937(format))
            mp_chmap_reorder_to_alsa(&ao_data.channels);

        /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
           prefer our own resampler, since that allows users to choose the resampler,
           even per file if desired */
        if ((err =
                 snd_pcm_hw_params_set_rate_resample(alsa_handler,
                                                     alsa_hwparams,
                                                     0)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] Unable to disable resampling: %s\n",
                    snd_strerror(
                        err));
            return 0;
        }

        if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
                                                   &ao_data.samplerate,
                                                   NULL)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] Unable to set samplerate-2: %s\n",
                    snd_strerror(
                        err));
            return 0;
        }

        bytes_per_sample = af_fmt2bits(ao_data.format) / 8;
        bytes_per_sample *= ao_data.channels.num;
        ao_data.bps = ao_data.samplerate * bytes_per_sample;

        if ((err =
                 snd_pcm_hw_params_set_buffer_time_near(alsa_handler,
                                                        alsa_hwparams,
                                                        &(unsigned int){
                                                            BUFFER_TIME},
                                                        NULL)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] Unable to set buffer time near: %s\n",
                    snd_strerror(
                        err));
            return 0;
        }

        if ((err =
                 snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
                                                    &(unsigned int){FRAGCOUNT},
                                                    NULL)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Unable to set periods: %s\n",
                    snd_strerror(err));
            return 0;
        }

        /* finally install hardware parameters */
        if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] Unable to set hw-parameters: %s\n",
                    snd_strerror(
                        err));
            return 0;
        }
        // end setting hw-params


        // gets buffersize for control
        if ((err =
                 snd_pcm_hw_params_get_buffer_size(alsa_hwparams,
                                                   &bufsize)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] Unable to get buffersize: %s\n", snd_strerror(
                        err));
            return 0;
        } else {
            ao_data.buffersize = bufsize * bytes_per_sample;
            mp_msg(MSGT_AO, MSGL_V, "alsa-init: got buffersize=%i\n",
                   ao_data.buffersize);
        }

        if ((err =
                 snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size,
                                                   NULL)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO ALSA] Unable to get period size: %s\n", snd_strerror(
                        err));
            return 0;
        } else
            mp_msg(MSGT_AO, MSGL_V, "alsa-init: got period size %li\n",
                   chunk_size);
        ao_data.outburst = chunk_size * bytes_per_sample;

        /* setting software parameters */
        if ((err =
                 snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] Unable to get sw-parameters: %s\n",
                    snd_strerror(
                        err));
            return 0;
        }
        if ((err =
                 snd_pcm_sw_params_get_boundary(alsa_swparams,
                                                &boundary)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Unable to get boundary: %s\n",
                    snd_strerror(err));
            return 0;
        }
        /* start playing when one period has been written */
        if ((err =
                 snd_pcm_sw_params_set_start_threshold(alsa_handler,
                                                       alsa_swparams,
                                                       chunk_size)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] Unable to set start threshold: %s\n",
                    snd_strerror(
                        err));
            return 0;
        }
        /* disable underrun reporting */
        if ((err =
                 snd_pcm_sw_params_set_stop_threshold(alsa_handler,
                                                      alsa_swparams,
                                                      boundary)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] Unable to set stop threshold: %s\n",
                    snd_strerror(
                        err));
            return 0;
        }
        /* play silence when there is an underrun */
        if ((err =
                 snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams,
                                                    boundary)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] Unable to set silence size: %s\n",
                    snd_strerror(
                        err));
            return 0;
        }
        if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR,
                    "[AO_ALSA] Unable to get sw-parameters: %s\n",
                    snd_strerror(
                        err));
            return 0;
        }
        /* end setting sw-params */

        alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);

        mp_msg(MSGT_AO, MSGL_V,
               "alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
               ao_data.samplerate, ao_data.channels.num, (int)bytes_per_sample,
               ao_data.buffersize,
               snd_pcm_format_description(
                   alsa_format));

    } // end switch alsa_handler (spdif)
    return 1;
} // end init


/* close audio device */
static void uninit(int immed)
{

    if (alsa_handler) {
        int err;

        if (!immed)
            snd_pcm_drain(alsa_handler);

        if ((err = snd_pcm_close(alsa_handler)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm close error: %s\n",
                    snd_strerror(
                        err));
            return;
        } else {
            alsa_handler = NULL;
            mp_msg(MSGT_AO, MSGL_V, "alsa-uninit: pcm closed\n");
        }
    } else
        mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] No handler defined!\n");
}

static void audio_pause(void)
{
    int err;

    if (alsa_can_pause) {
        delay_before_pause = get_delay();
        if ((err = snd_pcm_pause(alsa_handler, 1)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm pause error: %s\n",
                    snd_strerror(
                        err));
            return;
        }
        mp_msg(MSGT_AO, MSGL_V, "alsa-pause: pause supported by hardware\n");
    } else {
        if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0
            || prepause_frames < 0)
            prepause_frames = 0;
        delay_before_pause = prepause_frames / (float)ao_data.samplerate;

        if ((err = snd_pcm_drop(alsa_handler)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm drop error: %s\n",
                    snd_strerror(
                        err));
            return;
        }
    }
}

static void audio_resume(void)
{
    int err;

    if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
        mp_tmsg(MSGT_AO, MSGL_INFO,
                "[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
        while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN)
            sleep(1);
    }
    if (alsa_can_pause) {
        if ((err = snd_pcm_pause(alsa_handler, 0)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm resume error: %s\n",
                    snd_strerror(
                        err));
            return;
        }
        mp_msg(MSGT_AO, MSGL_V, "alsa-resume: resume supported by hardware\n");
    } else {
        if ((err = snd_pcm_prepare(alsa_handler)) < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm prepare error: %s\n",
                    snd_strerror(
                        err));
            return;
        }
        if (prepause_frames) {
            void *silence = calloc(prepause_frames, bytes_per_sample);
            play(silence, prepause_frames * bytes_per_sample, 0);
            free(silence);
        }
    }
}

/* stop playing and empty buffers (for seeking/pause) */
static void reset(void)
{
    int err;

    prepause_frames = 0;
    delay_before_pause = 0;
    if ((err = snd_pcm_drop(alsa_handler)) < 0) {
        mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm prepare error: %s\n",
                snd_strerror(
                    err));
        return;
    }
    if ((err = snd_pcm_prepare(alsa_handler)) < 0) {
        mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm prepare error: %s\n",
                snd_strerror(
                    err));
        return;
    }
    return;
}

/*
    plays 'len' bytes of 'data'
    returns: number of bytes played
    modified last at 29.06.02 by jp
    thanxs for marius <marius@rospot.com> for giving us the light ;)
 */

static int play(void *data, int len, int flags)
{
    int num_frames;
    snd_pcm_sframes_t res = 0;
    if (!(flags & AOPLAY_FINAL_CHUNK))
        len = len / ao_data.outburst * ao_data.outburst;
    num_frames = len / bytes_per_sample;

    //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);

    if (!alsa_handler) {
        mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Device configuration error.");
        return 0;
    }

    if (num_frames == 0)
        return 0;

    do {
        res = snd_pcm_writei(alsa_handler, data, num_frames);

        if (res == -EINTR) {
            /* nothing to do */
            res = 0;
        } else if (res == -ESTRPIPE) {  /* suspend */
            mp_tmsg(MSGT_AO, MSGL_INFO,
                    "[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
            while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
                sleep(1);
        }
        if (res < 0) {
            mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Write error: %s\n",
                    snd_strerror(
                        res));
            mp_tmsg(MSGT_AO, MSGL_INFO,
                    "[AO_ALSA] Trying to reset soundcard.\n");
            if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
                mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] pcm prepare error: %s\n",
                        snd_strerror(
                            res));
                break;
            }
            res = 0;
        }
    } while (res == 0);

    return res < 0 ? 0 : res * bytes_per_sample;
}

/* how many byes are free in the buffer */
static int get_space(void)
{
    snd_pcm_status_t *status;
    int ret;

    snd_pcm_status_alloca(&status);

    if ((ret = snd_pcm_status(alsa_handler, status)) < 0) {
        mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Cannot get pcm status: %s\n",
                snd_strerror(
                    ret));
        return 0;
    }

    unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
    if (space > ao_data.buffersize) // Buffer underrun?
        space = ao_data.buffersize;
    return space;
}

/* delay in seconds between first and last sample in buffer */
static float get_delay(void)
{
    if (alsa_handler) {
        snd_pcm_sframes_t delay;

        if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_PAUSED)
            return delay_before_pause;

        if (snd_pcm_delay(alsa_handler, &delay) < 0)
            return 0;

        if (delay < 0) {
            /* underrun - move the application pointer forward to catch up */
            snd_pcm_forward(alsa_handler, -delay);
            delay = 0;
        }
        return (float)delay / (float)ao_data.samplerate;
    } else
        return 0;
}