mirror/mpv
0
0
Fork 0
mirror of https://github.com/mpv-player/mpv.git synced 2024-09-20 12:02:23 +02:00
Code
595de9b1fc
mpv/libao2/ao_alsa.c
uau 50e46d7681 ao_alsa: Fix get_space() return values larger than buffersize 
After a buffer underrun the ALSA get_space() function sometimes returned
values larger than the ao had set in ao_data.buffersize. Fix this by
replacing the old check against MAX_OUTBURST by one against
ao_data.buffersize. There should be no need for the MAX_OUTBURST check;
the current MPlayer side should no longer have any constant limit on the
amount of data an ao can buffer or request at once.

The get_space() values larger than ao_data.buffersize triggered errors
in audio decoding causing the current attempt to fill audio buffers to
be aborted. I'm not sure how often that caused behavior noticeably worse
then an underrun already is.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24610 b3059339-0415-0410-9bf9-f77b7e298cf2
2007-09-24 21:49:58 +00:00

890 lines
24 KiB
C
Raw Blame History

/*
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 <stdarg.h>
#include <ctype.h>
#include <math.h>
#include <string.h>
#include "config.h"
#include "subopt-helper.h"
#include "mixer.h"
#include "mp_msg.h"
#include "help_mp.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
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);
tmp[sizeof tmp - 1] = '\0';
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_QUERY_FORMAT:
return CONTROL_TRUE;
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,
MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero);
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,MSGTR_AO_ALSA_MixerOpenError, snd_strerror(err));
return CONTROL_ERROR;
}
if ((err = snd_mixer_attach(handle, card)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerAttachError,
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,MSGTR_AO_ALSA_MixerRegisterError, snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
err = snd_mixer_load(handle);
if (err < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerLoadError, 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,MSGTR_AO_ALSA_UnableToFindSimpleControl,
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,MSGTR_AO_ALSA_ErrorSettingLeftChannel,
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,MSGTR_AO_ALSA_ErrorSettingRightChannel,
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;
}
} //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,
MSGTR_AO_ALSA_CommandlineHelp);
}
static int str_maxlen(strarg_t *str) {
if (str->len > ALSA_DEVICE_SIZE)
return 0;
return 1;
}
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 (!try_ac3 || err < 0)
err = snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
open_mode);
return err;
}
/*
open & setup audio device
return: 1=success 0=fail
*/
static int init(int rate_hz, int channels, int format, int flags)
{
int err;
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;
#if SND_LIB_VERSION >= 0x010005
mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
#else
mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
#endif
snd_lib_error_set_handler(alsa_error_handler);
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) {
device.str = "iec958";
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,MSGTR_AO_ALSA_ChannelsNotSupported,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_V,"alsa-init: using device %s\n", 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 = try_open_device(alsa_device, open_mode, format == AF_FORMAT_AC3)) < 0)
{
if (err != -EBUSY && ao_noblock) {
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed);
if ((err = try_open_device(alsa_device, 0, format == AF_FORMAT_AC3)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
return(0);
}
} else {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
return(0);
}
}
if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err));
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened 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,MSGTR_AO_ALSA_UnableToGetInitialParameters,
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,MSGTR_AO_ALSA_UnableToSetAccessType,
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,
MSGTR_AO_ALSA_FormatNotSupportedByHardware, 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,MSGTR_AO_ALSA_UnableToSetFormat,
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,MSGTR_AO_ALSA_UnableToSetChannels,
snd_strerror(err));
return(0);
}
/* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
prefer our own resampler */
#if SND_LIB_VERSION >= 0x010009
if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
0)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToDisableResampling,
snd_strerror(err));
return(0);
}
#endif
if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
&ao_data.samplerate, NULL)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSamplerate2,
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,MSGTR_AO_ALSA_UnableToSetBufferTimeNear,
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,MSGTR_AO_ALSA_UnableToSetPeriodTime,
snd_strerror(err));
return 0;
}
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_BufferTimePeriodTime,
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,MSGTR_AO_ALSA_UnableToSetPeriodSize,
chunk_size, snd_strerror(err));
return 0;
}
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,MSGTR_AO_ALSA_UnableToSetPeriods,
snd_strerror(err));
return 0;
}
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,MSGTR_AO_ALSA_UnableToSetHwParameters,
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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBufferSize, 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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize, 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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
snd_strerror(err));
return 0;
}
#if SND_LIB_VERSION >= 0x000901
if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBoundary,
snd_strerror(err));
return 0;
}
#else
boundary = 0x7fffffff;
#endif
/* 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,MSGTR_AO_ALSA_UnableToSetStartThreshold,
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,MSGTR_AO_ALSA_UnableToSetStopThreshold,
snd_strerror(err));
return 0;
}
#if SND_LIB_VERSION >= 0x000901
/* 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,MSGTR_AO_ALSA_UnableToSetSilenceSize,
snd_strerror(err));
return 0;
}
#endif
if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
snd_strerror(err));
return 0;
}
/* end setting sw-params */
mp_msg(MSGT_AO,MSGL_V,"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,MSGTR_AO_ALSA_PcmCloseError, snd_strerror(err));
return;
}
else {
alsa_handler = NULL;
mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
}
}
else {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_NoHandlerDefined);
}
}
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,MSGTR_AO_ALSA_PcmPauseError, 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,MSGTR_AO_ALSA_PcmDropError, 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,MSGTR_AO_ALSA_PcmResumeError, 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,MSGTR_AO_ALSA_PcmPrepareError, 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,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
return;
}
if ((err = snd_pcm_prepare(alsa_handler)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, 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,MSGTR_AO_ALSA_DeviceConfigurationError);
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,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
sleep(1);
}
if (res < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_WriteError, snd_strerror(res));
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_TryingToResetSoundcard);
if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, 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,MSGTR_AO_ALSA_CannotGetPcmStatus, snd_strerror(ret));
return(0);
}
ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
if (ret > ao_data.buffersize) // Buffer underrun?
ret = ao_data.buffersize;
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);
}
}
View Git Blame Copy Permalink