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mpv/libao2/ao_alsa.c

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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 "subopt-helper.h"
#include "mixer.h"
#include "mp_msg.h"
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#include <alsa/asoundlib.h>
#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.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;
static snd_pcm_hw_params_t *alsa_hwparams;
static snd_pcm_sw_params_t *alsa_swparams;
#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;
#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);
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_MUTE:
case AOCONTROL_SET_MUTE:
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;
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_AC3(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: {
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: {
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: {
if (!snd_mixer_selem_has_playback_switch(elem))
goto mixer_error;
bool m_l = vol->left == 0.0f, m_r = vol->right == 0.0f;
if (snd_mixer_selem_has_playback_switch_joined(elem)) {
m_l = m_l || m_r;
} else {
snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !m_r);
}
snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !m_l);
break;
}
case AOCONTROL_GET_MUTE: {
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);
vol->left = tmp ? 1.0f : 0.0f;
if (snd_mixer_selem_has_playback_switch_joined(elem)) {
vol->right = vol->left;
} else {
snd_mixer_selem_get_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, &tmp);
vol->right = tmp ? 1.0f : 0.0f;
}
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: mplayer -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, int 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,
channels, format);
alsa_handler = NULL;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
prepause_frames = 0;
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;
case AF_FORMAT_AC3_LE:
case AF_FORMAT_S16_LE:
alsa_format = SND_PCM_FORMAT_S16_LE;
break;
case AF_FORMAT_AC3_BE:
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_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;
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 (AF_FORMAT_IS_AC3(format)) {
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;
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",channels);
}
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);
if (!alsa_handler) {
int open_mode = block ? 0 : SND_PCM_NONBLOCK;
int isac3 = AF_FORMAT_IS_AC3(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_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
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;
}
if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
&ao_data.channels)) < 0)
{
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set channels: %s\n",
snd_strerror(err));
return 0;
}
/* 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;
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 */
mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
ao_data.samplerate, ao_data.channels, (int)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_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) {
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;
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;
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));
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_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_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;
}
}
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