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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 <sys/poll.h>
#include "../config.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 "afmt.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;
/* possible 4096, original 8192
* was only needed for calculating chunksize? */
static int alsa_fragsize = 4096;
/* 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;//is alsa_fragsize / 4
#define MIN_CHUNK_SIZE 1024
static size_t bits_per_sample, bytes_per_sample, bits_per_frame;
static size_t chunk_bytes;
int ao_mmap = 0;
int ao_noblock = 0;
int first = 1;
static int open_mode;
static int set_block_mode;
static int alsa_can_pause = 0;
#define ALSA_DEVICE_SIZE 256
#undef BUFFERTIME
#define SET_CHUNKSIZE
#undef USE_POLL
/* 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";
long pmin, pmax;
long get_vol, set_vol;
float f_multi;
if(mixer_channel) mix_name = mixer_channel;
if(mixer_device) card = mixer_device;
if(ao_data.format == AFMT_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, 0);
snd_mixer_selem_id_set_name(sid, mix_name);
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);
}
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, char *src, int len)
{
char *tmp;
strncpy (dest, src, len);
while ((tmp = strrchr(dest, '.')))
tmp[0] = ',';
while ((tmp = strrchr(dest, '=')))
tmp[0] = ':';
}
static void print_help ()
{
mp_msg (MSGT_AO, MSGL_FATAL,
"\n-ao alsa commandline help:\n"
"Example: mplayer -ao alsa:mmap:device=hw=0.3\n"
" sets mmap-mode and first card fourth device\n"
"\nOptions:\n"
" mmap\n"
" Set memory-mapped mode, experimental\n"
" noblock\n"
" Sets non-blocking mode\n"
" device=<device-name>\n"
" Sets device (change , to . and : to =)\n");
}
/*
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;
snd_pcm_info_t *alsa_info;
char *str_block_mode;
int device_set = 0;
int dir = 0;
snd_pcm_uframes_t bufsize;
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, %s\n", rate_hz,
channels, audio_out_format_name(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.bps = channels * rate_hz;
ao_data.format = format;
ao_data.channels = channels;
ao_data.outburst = OUTBURST;
switch (format)
{
case AFMT_S8:
alsa_format = SND_PCM_FORMAT_S8;
break;
case AFMT_U8:
alsa_format = SND_PCM_FORMAT_U8;
break;
case AFMT_U16_LE:
alsa_format = SND_PCM_FORMAT_U16_LE;
break;
case AFMT_U16_BE:
alsa_format = SND_PCM_FORMAT_U16_BE;
break;
#ifndef WORDS_BIGENDIAN
case AFMT_AC3:
#endif
case AFMT_S16_LE:
alsa_format = SND_PCM_FORMAT_S16_LE;
break;
#ifdef WORDS_BIGENDIAN
case AFMT_AC3:
#endif
case AFMT_S16_BE:
alsa_format = SND_PCM_FORMAT_S16_BE;
break;
case AFMT_S32_LE:
alsa_format = SND_PCM_FORMAT_S32_LE;
break;
case AFMT_S32_BE:
alsa_format = SND_PCM_FORMAT_S32_BE;
break;
case AFMT_FLOAT:
alsa_format = SND_PCM_FORMAT_FLOAT_LE;
break;
default:
alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
break;
}
//setting bw according to the input-format. resolution seems to be always s16_le or
//u16_le so 32bit is probably obsolet.
switch(alsa_format)
{
case SND_PCM_FORMAT_S8:
case SND_PCM_FORMAT_U8:
ao_data.bps *= 1;
break;
case SND_PCM_FORMAT_S16_LE:
case SND_PCM_FORMAT_U16_LE:
case SND_PCM_FORMAT_S16_BE:
case SND_PCM_FORMAT_U16_BE:
ao_data.bps *= 2;
break;
case SND_PCM_FORMAT_S32_LE:
case SND_PCM_FORMAT_S32_BE:
case SND_PCM_FORMAT_FLOAT_LE:
ao_data.bps *= 4;
break;
case -1:
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: invalid format (%s) requested - output disabled\n",
audio_out_format_name(format));
return(0);
break;
default:
ao_data.bps *= 2;
mp_msg(MSGT_AO,MSGL_WARN,"alsa-init: couldn't convert to right format. setting bps to: %d", ao_data.bps);
}
//subdevice parsing
if (ao_subdevice) {
int parse_err = 0;
char *parse_pos = &ao_subdevice[0];
while (parse_pos[0] && !parse_err) {
if (strncmp (parse_pos, "mmap", 4) == 0) {
parse_pos = &parse_pos[4];
ao_mmap = 1;
} else if (strncmp (parse_pos, "noblock", 7) == 0) {
parse_pos = &parse_pos[7];
ao_noblock = 1;
} else if (strncmp (parse_pos, "device=", 7) == 0) {
int name_len;
parse_pos = &parse_pos[7];
name_len = strcspn (parse_pos, ":");
if (name_len < 0 || name_len > ALSA_DEVICE_SIZE) {
parse_err = 1;
break;
}
parse_device (alsa_device, parse_pos, name_len);
parse_pos = &parse_pos[name_len];
device_set = 1;
}
if (parse_pos[0] == ':') parse_pos = &parse_pos[1];
else if (parse_pos[0]) parse_err = 1;
}
if (parse_err) {
print_help();
return 0;
}
} else { //end parsing ao_subdevice
/* in any case for multichannel playback we should select
* appropriate device
*/
switch (channels) {
case 1:
case 2:
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
strncpy(alsa_device, "plug:surround40", ALSA_DEVICE_SIZE);
else
strncpy(alsa_device, "surround40", ALSA_DEVICE_SIZE);
device_set = 1;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
break;
case 6:
if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
strncpy(alsa_device, "plug:surround51", ALSA_DEVICE_SIZE);
else
strncpy(alsa_device, "surround51", ALSA_DEVICE_SIZE);
device_set = 1;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
break;
default:
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: %d channels are not supported\n",channels);
}
}
/* 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 == AFMT_AC3) {
unsigned char s[4];
switch (channels) {
case 1:
case 2:
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:AES0=0x%x,AES1=0x%x,AES2=0x%x,AES3=0x%x",
s[0], s[1], s[2], s[3]);
mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
device_set = 1;
break;
case 4:
strncpy(alsa_device, "surround40", ALSA_DEVICE_SIZE);
device_set = 1;
break;
case 6:
strncpy(alsa_device, "surround51", ALSA_DEVICE_SIZE);
device_set = 1;
break;
default:
mp_msg(MSGT_AO,MSGL_ERR,"alsa-spdif-init: %d channels are not supported\n", channels);
}
}
if (!device_set)
{
int tmp_device, tmp_subdevice, err;
snd_pcm_info_alloca(&alsa_info);
if ((tmp_device = snd_pcm_info_get_device(alsa_info)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: can't get device\n");
}
if ((tmp_subdevice = snd_pcm_info_get_subdevice(alsa_info)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: can't get subdevice\n");
}
mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: got device=%i, subdevice=%i\n",
tmp_device, tmp_subdevice);
//we are setting here device to default cause it could be configured by the user
//if its not set by the user, it defaults to hw:0,0
if ((err = snprintf(alsa_device, ALSA_DEVICE_SIZE, "default")) <= 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: can't write device-id\n");
}
mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: %d soundcard%s found, using: %s\n", cards+1,(cards >= 0) ? "" : "s", alsa_device);
} else {
mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: soundcard set to %s\n", alsa_device);
}
//setting modes for block or nonblock-mode
if (ao_noblock) {
open_mode = SND_PCM_NONBLOCK;
set_block_mode = 1;
str_block_mode = "nonblock-mode";
}
else {
open_mode = 0;
set_block_mode = 0;
str_block_mode = "block-mode";
}
//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;
if (ao_mmap)
chunk_size = 512;
else
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 {
set_block_mode = 0;
str_block_mode = "block-mode";
}
} 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, set_block_mode)) < 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 %s\n", str_block_mode);
}
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);
}
if (ao_mmap) {
snd_pcm_access_mask_t *mask = alloca(snd_pcm_access_mask_sizeof());
snd_pcm_access_mask_none(mask);
snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_INTERLEAVED);
snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_NONINTERLEAVED);
snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_COMPLEX);
err = snd_pcm_hw_params_set_access_mask(alsa_handler, alsa_hwparams, mask);
mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: mmap set\n");
} else {
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",
audio_out_format_name(format));
alsa_format = SND_PCM_FORMAT_S16_LE;
ao_data.format = AFMT_S16_LE;
ao_data.bps = channels * rate_hz * 2;
}
bytes_per_sample = ao_data.bps / ao_data.samplerate; //it should be here
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));
}
if ((err = snd_pcm_hw_params_set_channels(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));
}
if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
&ao_data.samplerate, &dir)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set samplerate-2: %s\n",
snd_strerror(err));
return(0);
}
#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, &dir)) < 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, &dir)) < 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
dir=0;
if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams,
&chunk_size, &dir)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set periodsize(%d): %s\n",
chunk_size, snd_strerror(err));
}
else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set to %i\n", chunk_size);
}
if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
&alsa_fragcount, &dir)) < 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);
}
// setting sw-params (only avail-min) if noblocking mode was choosed
if (ao_noblock)
{
if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get parameters: %s\n",
snd_strerror(err));
}
//set min available frames to consider pcm ready (4)
//increased for nonblock-mode should be set dynamically later
if ((err = snd_pcm_sw_params_set_avail_min(alsa_handler, alsa_swparams, 4)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set avail_min %s\n",
snd_strerror(err));
}
if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to install sw-params\n");
}
bits_per_sample = snd_pcm_format_physical_width(alsa_format);
bits_per_frame = bits_per_sample * channels;
chunk_bytes = chunk_size * bits_per_frame / 8;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: bits per sample (bps)=%i, bits per frame (bpf)=%i, chunk_bytes=%i\n",bits_per_sample,bits_per_frame,chunk_bytes);}
//end swparams
if ((err = snd_pcm_prepare(alsa_handler)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: pcm prepare error: %s\n", snd_strerror(err));
}
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 (!ao_noblock) {
if ((err = snd_pcm_drop(alsa_handler)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-uninit: pcm drop error: %s\n", snd_strerror(err));
return;
}
}
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()
{
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()
{
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()
{
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;
}
#ifdef USE_POLL
static int wait_for_poll(snd_pcm_t *handle, struct pollfd *ufds, unsigned int count)
{
unsigned short revents;
while (1) {
poll(ufds, count, -1);
snd_pcm_poll_descriptors_revents(handle, ufds, count, &revents);
if (revents & POLLERR)
return -EIO;
if (revents & POLLOUT)
return 0;
}
}
#endif
#ifndef timersub
#define timersub(a, b, result) \
do { \
(result)->tv_sec = (a)->tv_sec - (b)->tv_sec; \
(result)->tv_usec = (a)->tv_usec - (b)->tv_usec; \
if ((result)->tv_usec < 0) { \
--(result)->tv_sec; \
(result)->tv_usec += 1000000; \
} \
} while (0)
#endif
/* I/O error handler */
static int xrun(u_char *str_mode)
{
int err;
snd_pcm_status_t *status;
snd_pcm_status_alloca(&status);
if ((err = snd_pcm_status(alsa_handler, status))<0) {
mp_msg(MSGT_AO,MSGL_ERR,"status error: %s", snd_strerror(err));
return(0);
}
if (snd_pcm_status_get_state(status) == SND_PCM_STATE_XRUN) {
struct timeval now, diff, tstamp;
gettimeofday(&now, 0);
snd_pcm_status_get_trigger_tstamp(status, &tstamp);
timersub(&now, &tstamp, &diff);
mp_msg(MSGT_AO,MSGL_INFO,"alsa-%s: xrun of at least %.3f msecs. resetting stream\n",
str_mode,
diff.tv_sec * 1000 + diff.tv_usec / 1000.0);
}
if ((err = snd_pcm_prepare(alsa_handler))<0) {
mp_msg(MSGT_AO,MSGL_ERR,"xrun: prepare error: %s", snd_strerror(err));
return(0);
}
return(1); /* ok, data should be accepted again */
}
static int play_normal(void* data, int len);
static int play_mmap(void* data, int len);
static int play(void* data, int len, int flags)
{
int result;
if (ao_mmap)
result = play_mmap(data, len);
else
result = play_normal(data, len);
return result;
}
/*
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_normal(void* data, int len)
{
//bytes_per_sample is always 4 for 2 chn S16_LE
int num_frames = len / bytes_per_sample;
char *output_samples = (char *)data;
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;
}
while (num_frames > 0) {
res = snd_pcm_writei(alsa_handler, (void *)output_samples, num_frames);
if (res == -EAGAIN) {
snd_pcm_wait(alsa_handler, 1000);
}
else if (res == -EPIPE) { /* underrun */
if (xrun("play") <= 0) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: xrun reset error");
return(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);
}
else if (res < 0) {
mp_msg(MSGT_AO,MSGL_INFO,"alsa-play: unknown status, trying to reset soundcard\n");
if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: snd prepare error");
return(0);
break;
}
}
if (res > 0) {
/* output_samples += ao_data.channels * res; */
output_samples += res * bytes_per_sample;
num_frames -= res;
}
} //end while
if (res < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: write error %s", snd_strerror(res));
return 0;
}
return len - len % bytes_per_sample;
}
/* mmap-mode mainly based on descriptions by Joshua Haberman <joshua@haberman.com>
* 'An overview of the ALSA API' http://people.debian.org/~joshua/x66.html
* and some help by Paul Davis <pbd@op.net> */
static int play_mmap(void* data, int len)
{
snd_pcm_sframes_t commitres, frames_available;
snd_pcm_uframes_t frames_transmit, size, offset;
const snd_pcm_channel_area_t *area;
void *outbuffer;
int result;
#ifdef USE_POLL //seems not really be needed
struct pollfd *ufds;
int count;
count = snd_pcm_poll_descriptors_count (alsa_handler);
ufds = malloc(sizeof(struct pollfd) * count);
snd_pcm_poll_descriptors(alsa_handler, ufds, count);
//first wait_for_poll
if (err = (wait_for_poll(alsa_handler, ufds, count) < 0)) {
if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_XRUN ||
snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
xrun("play");
}
}
#endif
outbuffer = alloca(ao_data.buffersize);
//don't trust get_space() ;)
frames_available = snd_pcm_avail_update(alsa_handler) * bytes_per_sample;
if (frames_available < 0)
xrun("play");
if (frames_available < 4) {
if (first) {
first = 0;
snd_pcm_start(alsa_handler);
}
else { //FIXME should break and return 0?
snd_pcm_wait(alsa_handler, -1);
first = 1;
}
}
/* len is simply the available bufferspace got by get_space()
* but real avail_buffer in frames is ab/bytes_per_sample */
size = len / bytes_per_sample;
//mp_msg(MSGT_AO,MSGL_V,"len: %i size %i, f_avail %i, bps %i ...\n", len, size, frames_available, bytes_per_sample);
frames_transmit = size;
/* prepare areas and set sw-pointers
* frames_transmit returns the real available buffer-size
* sometimes != frames_available cause of ringbuffer 'emulation' */
snd_pcm_mmap_begin(alsa_handler, &area, &offset, &frames_transmit);
/* this is specific to interleaved streams (or non-interleaved
* streams with only one channel) */
outbuffer = ((char *) area->addr + (area->first + area->step * offset) / 8); //8
//write data
memcpy(outbuffer, data, (frames_transmit * bytes_per_sample));
commitres = snd_pcm_mmap_commit(alsa_handler, offset, frames_transmit);
if (commitres < 0 || commitres != frames_transmit) {
if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_XRUN ||
snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
xrun("play");
}
}
//mp_msg(MSGT_AO,MSGL_V,"mmap ft: %i, cres: %i\n", frames_transmit, commitres);
/* err = snd_pcm_area_copy(&area, offset, &data, offset, len, alsa_format); */
/* if (err < 0) { */
/* mp_msg(MSGT_AO,MSGL_ERR,"area-copy-error\n"); */
/* return 0; */
/* } */
//calculate written frames!
result = commitres * bytes_per_sample;
/* if (verbose) { */
/* if (len == result) */
/* mp_msg(MSGT_AO,MSGL_V,"result: %i, frames written: %i ...\n", result, frames_transmit); */
/* else */
/* mp_msg(MSGT_AO,MSGL_V,"result: %i, frames written: %i, result != len ...\n", result, frames_transmit); */
/* } */
//mplayer doesn't like -result
if (result < 0)
result = 0;
#ifdef USE_POLL
free(ufds);
#endif
return result;
}
/* how many byes are free in the buffer */
static int get_space()
{
snd_pcm_status_t *status;
int ret;
char *str_status;
//snd_pcm_sframes_t avail_frames = 0;
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);
}
switch(snd_pcm_status_get_state(status))
{
case SND_PCM_STATE_OPEN:
str_status = "open";
ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
break;
case SND_PCM_STATE_PREPARED:
str_status = "prepared";
first = 1;
ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
if (ret == 0) //ugly workaround for hang in mmap-mode
ret = 10;
break;
case SND_PCM_STATE_RUNNING:
ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
//avail_frames = snd_pcm_avail_update(alsa_handler) * bytes_per_sample;
if (str_status != "open" && str_status != "prepared")
str_status = "running";
break;
case SND_PCM_STATE_PAUSED:
mp_msg(MSGT_AO,MSGL_V,"alsa-space: paused");
str_status = "paused";
ret = 0;
break;
case SND_PCM_STATE_XRUN:
xrun("space");
str_status = "xrun";
first = 1;
ret = 0;
break;
default:
str_status = "undefined";
ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
if (ret <= 0) {
xrun("space");
}
}
if (snd_pcm_status_get_state(status) != SND_PCM_STATE_RUNNING)
mp_msg(MSGT_AO,MSGL_V,"alsa-space: free space = %i, %s --\n", ret, str_status);
if (ret < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"negative value!!\n");
ret = 0;
}
// workaround for too small value returned
if (ret < MIN_CHUNK_SIZE)
ret = 0;
return(ret);
}
/* delay in seconds between first and last sample in buffer */
static float get_delay()
{
if (alsa_handler) {
snd_pcm_status_t *status;
float ret;
snd_pcm_status_alloca(&status);
if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,"alsa-delay: cannot get pcm status: %s\n", snd_strerror(ret));
}
switch(snd_pcm_status_get_state(status))
{
case SND_PCM_STATE_OPEN:
case SND_PCM_STATE_PREPARED:
case SND_PCM_STATE_RUNNING:
ret = (float)snd_pcm_status_get_delay(status)/(float)ao_data.samplerate;
break;
default:
ret = 0;
}
if (ret < 0)
ret = 0;
return(ret);
} else {
return(0);
}
}
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