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mpv/libao2/ao_alsa.c

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/*
ao_alsa9/1.x - ALSA-0.9.x-1.x output plugin for MPlayer
(C) Alex Beregszaszi
modified for real alsa-0.9.0-support by Zsolt Barat <joy@streamminister.de>
additional AC3 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.
Any bugreports regarding to this driver are welcome.
*/
#include <errno.h>
#include <sys/time.h>
#include <stdlib.h>
#include <math.h>
#include <string.h>
#include "config.h"
#include "subopt-helper.h"
#include "mixer.h"
#include "mp_msg.h"
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#if HAVE_SYS_ASOUNDLIB_H
#include <sys/asoundlib.h>
#elif HAVE_ALSA_ASOUNDLIB_H
#include <alsa/asoundlib.h>
#else
#error "asoundlib.h is not in sys/ or alsa/ - please bugreport"
#endif
#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"
static ao_info_t info =
{
"ALSA-0.9.x-1.x audio output",
"alsa",
"Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
"under developement"
};
LIBAO_EXTERN(alsa)
static snd_pcm_t *alsa_handler;
static snd_pcm_format_t alsa_format;
static snd_pcm_hw_params_t *alsa_hwparams;
static snd_pcm_sw_params_t *alsa_swparams;
/* 16 sets buffersize to 16 * chunksize is as default 1024
* which seems to be good avarge for most situations
* so buffersize is 16384 frames by default */
static int alsa_fragcount = 16;
static snd_pcm_uframes_t chunk_size = 1024;
static size_t bytes_per_sample;
static int ao_noblock = 0;
static int open_mode;
static int alsa_can_pause = 0;
#define ALSA_DEVICE_SIZE 256
#undef BUFFERTIME
#define SET_CHUNKSIZE
/* to set/get/query special features/parameters */
static int control(int cmd, void *arg)
{
switch(cmd) {
case AOCONTROL_QUERY_FORMAT:
return CONTROL_TRUE;
#ifndef WORDS_BIGENDIAN
case AOCONTROL_GET_VOLUME:
case AOCONTROL_SET_VOLUME:
{
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
int err;
snd_mixer_t *handle;
snd_mixer_elem_t *elem;
snd_mixer_selem_id_t *sid;
static char *mix_name = "PCM";
static char *card = "default";
static int mix_index = 0;
long pmin, pmax;
long get_vol, set_vol;
float f_multi;
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_msg(MSGT_AO,MSGL_ERR,
"alsa-control: invalid mixer index. Defaulting to 0\n");
mix_index = 0 ;
}
}
}
if(mixer_device) card = mixer_device;
if(ao_data.format == AF_FORMAT_AC3)
return CONTROL_TRUE;
//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_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer open error: %s\n", snd_strerror(err));
return CONTROL_ERROR;
}
if ((err = snd_mixer_attach(handle, card)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: 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_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer register error: %s\n", snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
err = snd_mixer_load(handle);
if (err < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: 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_msg(MSGT_AO,MSGL_ERR,"alsa-control: 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));
if (cmd == AOCONTROL_SET_VOLUME) {
set_vol = vol->left / f_multi + pmin + 0.5;
//setting channels
if ((err = snd_mixer_selem_set_playback_volume(elem, 0, set_vol)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: error setting left channel, %s\n",
snd_strerror(err));
return CONTROL_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, 1, set_vol)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: error setting right channel, %s\n",
snd_strerror(err));
return CONTROL_ERROR;
}
mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
set_vol, pmin, pmax, f_multi);
if (snd_mixer_selem_has_playback_switch(elem)) {
int lmute = (vol->left == 0.0);
int rmute = (vol->right == 0.0);
if (snd_mixer_selem_has_playback_switch_joined(elem)) {
lmute = rmute = lmute && rmute;
} else {
snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
}
snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
}
}
else {
snd_mixer_selem_get_playback_volume(elem, 0, &get_vol);
vol->left = (get_vol - pmin) * f_multi;
snd_mixer_selem_get_playback_volume(elem, 1, &get_vol);
vol->right = (get_vol - pmin) * f_multi;
mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
}
snd_mixer_close(handle);
return CONTROL_OK;
}
#endif
} //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_msg (MSGT_AO, MSGL_FATAL,
"\n-ao alsa commandline help:\n"
"Example: mplayer -ao alsa:device=hw=0.3\n"
" sets first card fourth hardware device\n"
"\nOptions:\n"
" noblock\n"
" Opens device in non-blocking mode\n"
" device=<device-name>\n"
" Sets device (change , to . and : to =)\n");
}
static int str_maxlen(strarg_t *str) {
if (str->len > ALSA_DEVICE_SIZE)
return 0;
return 1;
}
/*
open & setup audio device
return: 1=success 0=fail
*/
static int init(int rate_hz, int channels, int format, int flags)
{
int err;
int cards = -1;
int block;
strarg_t device;
snd_pcm_uframes_t bufsize;
snd_pcm_uframes_t boundary;
opt_t subopts[] = {
{"block", OPT_ARG_BOOL, &block, NULL},
{"device", OPT_ARG_STR, &device, (opt_test_f)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,
channels, format);
alsa_handler = NULL;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
if ((err = snd_card_next(&cards)) < 0 || cards < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: no soundcards found: %s\n", snd_strerror(err));
return(0);
}
ao_data.samplerate = rate_hz;
ao_data.format = format;
ao_data.channels = channels;
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;
#ifndef WORDS_BIGENDIAN
case AF_FORMAT_AC3:
#endif
case AF_FORMAT_S16_LE:
alsa_format = SND_PCM_FORMAT_S16_LE;
break;
#ifdef WORDS_BIGENDIAN
case AF_FORMAT_AC3:
#endif
case AF_FORMAT_S16_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_FLOAT_LE:
alsa_format = SND_PCM_FORMAT_FLOAT_LE;
break;
case AF_FORMAT_FLOAT_BE:
alsa_format = SND_PCM_FORMAT_FLOAT_BE;
break;
case AF_FORMAT_MU_LAW:
alsa_format = SND_PCM_FORMAT_MU_LAW;
break;
case AF_FORMAT_A_LAW:
alsa_format = SND_PCM_FORMAT_A_LAW;
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 (format == AF_FORMAT_AC3) {
unsigned char s[4];
s[0] = IEC958_AES0_NONAUDIO |
IEC958_AES0_CON_EMPHASIS_NONE;
s[1] = IEC958_AES1_CON_ORIGINAL |
IEC958_AES1_CON_PCM_CODER;
s[2] = 0;
s[3] = IEC958_AES3_CON_FS_48000;
snprintf(alsa_device, ALSA_DEVICE_SIZE,
"iec958:{CARD 0 AES0 0x%02x AES1 0x%02x AES2 0x%02x AES3 0x%02x}",
s[0], s[1], s[2], s[3]);
device.str = alsa_device;
mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
}
else
/* in any case for multichannel playback we should select
* appropriate device
*/
switch (channels) {
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;
default:
device.str = "default";
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: %d channels are not supported\n",channels);
}
device.len = strlen(device.str);
if (subopt_parse(ao_subdevice, subopts) != 0) {
print_help();
return 0;
}
ao_noblock = !block;
parse_device(alsa_device, device.str, device.len);
mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: %d soundcard%s found, using: %s\n", cards+1,(cards >= 0) ? "" : "s", alsa_device);
//setting modes for block or nonblock-mode
if (ao_noblock) {
open_mode = SND_PCM_NONBLOCK;
}
else {
open_mode = 0;
}
//sets buff/chunksize if its set manually
if (ao_data.buffersize) {
switch (ao_data.buffersize)
{
case 1:
alsa_fragcount = 16;
chunk_size = 512;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
break;
case 2:
alsa_fragcount = 8;
chunk_size = 1024;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
break;
case 3:
alsa_fragcount = 32;
chunk_size = 512;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
break;
case 4:
alsa_fragcount = 16;
chunk_size = 1024;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
break;
default:
alsa_fragcount = 16;
chunk_size = 1024;
break;
}
}
if (!alsa_handler) {
//modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
if ((err = snd_pcm_open(&alsa_handler, alsa_device, SND_PCM_STREAM_PLAYBACK, open_mode)) < 0)
{
if (err != -EBUSY && ao_noblock) {
mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: open in nonblock-mode failed, trying to open in block-mode\n");
if ((err = snd_pcm_open(&alsa_handler, alsa_device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: playback open error: %s\n", snd_strerror(err));
return(0);
}
} else {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: playback open error: %s\n", snd_strerror(err));
return(0);
}
}
if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: error set block-mode %s\n", snd_strerror(err));
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opend in blocking mode\n");
}
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_msg(MSGT_AO,MSGL_ERR,"alsa-init: 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_msg(MSGT_AO,MSGL_ERR,"alsa-init: 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_msg(MSGT_AO,MSGL_INFO,
"alsa-init: format %s are not supported by hardware, trying default\n", af_fmt2str_short(format));
alsa_format = SND_PCM_FORMAT_S16_LE;
ao_data.format = AF_FORMAT_S16_LE;
}
if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
alsa_format)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set format: %s\n",
snd_strerror(err));
return(0);
}
if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
&ao_data.channels)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set channels: %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_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set samplerate-2: %s\n",
snd_strerror(err));
return(0);
}
bytes_per_sample = snd_pcm_format_physical_width(alsa_format) / 8;
bytes_per_sample *= ao_data.channels;
ao_data.bps = ao_data.samplerate * bytes_per_sample;
#ifdef BUFFERTIME
{
int alsa_buffer_time = 500000; /* original 60 */
int alsa_period_time;
alsa_period_time = alsa_buffer_time/4;
if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
&alsa_buffer_time, NULL)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set buffer time near: %s\n",
snd_strerror(err));
return(0);
} else
alsa_buffer_time = err;
if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams,
&alsa_period_time, NULL)) < 0)
/* original: alsa_buffer_time/ao_data.bps */
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set period time: %s\n",
snd_strerror(err));
}
mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: buffer_time: %d, period_time :%d\n",
alsa_buffer_time, err);
}
#endif//end SET_BUFFERTIME
#ifdef SET_CHUNKSIZE
{
//set chunksize
if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams,
&chunk_size, NULL)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set periodsize(%ld): %s\n",
chunk_size, snd_strerror(err));
}
else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set to %li\n", chunk_size);
}
if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
&alsa_fragcount, NULL)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set periods: %s\n",
snd_strerror(err));
}
else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: fragcount=%i\n", alsa_fragcount);
}
}
#endif//end SET_CHUNKSIZE
/* finally install hardware parameters */
if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set hw-parameters: %s\n",
snd_strerror(err));
}
// end setting hw-params
// gets buffersize for control
if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get buffersize: %s\n", snd_strerror(err));
}
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_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get period size: %s\n", snd_strerror(err));
} 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_msg(MSGT_AO,MSGL_ERR,"alsa-init: 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_msg(MSGT_AO,MSGL_ERR,"alsa-init: 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_msg(MSGT_AO,MSGL_ERR,"alsa-init: 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_msg(MSGT_AO,MSGL_ERR,"alsa-init: 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_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set silence size: %s\n",
snd_strerror(err));
return 0;
}
if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set sw-parameters: %s\n",
snd_strerror(err));
return 0;
}
/* end setting sw-params */
mp_msg(MSGT_AO,MSGL_INFO,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
ao_data.samplerate, ao_data.channels, bytes_per_sample, ao_data.buffersize,
snd_pcm_format_description(alsa_format));
} // end switch alsa_handler (spdif)
alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
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_msg(MSGT_AO,MSGL_ERR,"alsa-uninit: pcm close error: %s\n", snd_strerror(err));
return;
}
else {
alsa_handler = NULL;
mp_msg(MSGT_AO,MSGL_INFO,"alsa-uninit: pcm closed\n");
}
}
else {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-uninit: no handler defined!\n");
}
}
static void audio_pause(void)
{
int err;
if (alsa_can_pause) {
if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-pause: pcm pause error: %s\n", snd_strerror(err));
return;
}
mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
} else {
if ((err = snd_pcm_drop(alsa_handler)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-pause: pcm drop error: %s\n", snd_strerror(err));
return;
}
}
}
static void audio_resume(void)
{
int err;
if (alsa_can_pause) {
if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-resume: 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_msg(MSGT_AO,MSGL_ERR,"alsa-resume: pcm prepare error: %s\n", snd_strerror(err));
return;
}
}
}
/* stop playing and empty buffers (for seeking/pause) */
static void reset(void)
{
int err;
if ((err = snd_pcm_drop(alsa_handler)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-reset: pcm drop error: %s\n", snd_strerror(err));
return;
}
if ((err = snd_pcm_prepare(alsa_handler)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-reset: 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 = len / bytes_per_sample;
snd_pcm_sframes_t res = 0;
//mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
if (!alsa_handler) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: 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_msg(MSGT_AO,MSGL_INFO,"alsa-play: pcm in suspend mode. trying to resume\n");
while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
sleep(1);
}
if (res < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: write error: %s\n", snd_strerror(res));
mp_msg(MSGT_AO,MSGL_INFO,"alsa-play: trying to reset soundcard\n");
if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: pcm prepare error: %s\n", snd_strerror(res));
return(0);
break;
}
}
} while (res == 0);
return res < 0 ? res : 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_msg(MSGT_AO,MSGL_ERR,"alsa-space: cannot get pcm status: %s\n", snd_strerror(ret));
return(0);
}
ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
if (ret > MAX_OUTBURST)
ret = MAX_OUTBURST;
return(ret);
}
/* 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_delay(alsa_handler, &delay) < 0)
return 0;
if (delay < 0) {
/* underrun - move the application pointer forward to catch up */
#if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
snd_pcm_forward(alsa_handler, -delay);
#endif
delay = 0;
}
return (float)delay / (float)ao_data.samplerate;
} else {
return(0);
}
}
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