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mpv/audio/out/ao_alsa.c
wm4 4873b32c59 Rename directories, move files (step 2 of 2) 
Finish renaming directories and moving files. Adjust all include
statements to make the previous commit compile.

The two commits are separate, because git is bad at tracking renames
and content changes at the same time.

Also take this as an opportunity to remove the separation between
"common" and "mplayer" sources in the Makefile. ("common" used to be
shared between mplayer and mencoder.)
2012-11-12 20:08:18 +01:00

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/*
* ALSA 0.9.x-1.x audio output driver
*
* Copyright (C) 2004 Alex Beregszaszi
*
* modified for real ALSA 0.9.0 support by Zsolt Barat <joy@streamminister.de>
* additional AC-3 passthrough support by Andy Lo A Foe <andy@alsaplayer.org>
* 08/22/2002 iec958-init rewritten and merged with common init, zsolt
* 04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka
* 04/25/2004 printfs converted to mp_msg, Zsolt.
*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <errno.h>
#include <sys/time.h>
#include <stdlib.h>
#include <stdarg.h>
#include <ctype.h>
#include <math.h>
#include <string.h>
#include <alloca.h>
#include "config.h"
#include "core/subopt-helper.h"
#include "audio/mixer.h"
#include "core/mp_msg.h"
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#include <alsa/asoundlib.h>
#include "ao.h"
#include "audio_out_internal.h"
#include "audio/format.h"
static const ao_info_t info =
{
"ALSA-0.9.x-1.x audio output",
"alsa",
"Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
"under development"
};
LIBAO_EXTERN(alsa)
static snd_pcm_t *alsa_handler;
static snd_pcm_format_t alsa_format;
#define BUFFER_TIME 500000 // 0.5 s
#define FRAGCOUNT 16
static size_t bytes_per_sample;
static int alsa_can_pause;
static snd_pcm_sframes_t prepause_frames;
#define ALSA_DEVICE_SIZE 256
static void alsa_error_handler(const char *file, int line, const char *function,
int err, const char *format, ...)
{
char tmp[0xc00];
va_list va;
va_start(va, format);
vsnprintf(tmp, sizeof tmp, format, va);
va_end(va);
if (err)
mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
file, line, function, tmp, snd_strerror(err));
else
mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
file, line, function, tmp);
}
/* to set/get/query special features/parameters */
static int control(int cmd, void *arg)
{
switch(cmd) {
case AOCONTROL_GET_MUTE:
case AOCONTROL_SET_MUTE:
case AOCONTROL_GET_VOLUME:
case AOCONTROL_SET_VOLUME:
{
int err;
snd_mixer_t *handle;
snd_mixer_elem_t *elem;
snd_mixer_selem_id_t *sid;
char *mix_name = "Master";
char *card = "default";
int mix_index = 0;
long pmin, pmax;
long get_vol, set_vol;
float f_multi;
if(AF_FORMAT_IS_AC3(ao_data.format) || AF_FORMAT_IS_IEC61937(ao_data.format))
return CONTROL_TRUE;
if(mixer_channel) {
char *test_mix_index;
mix_name = strdup(mixer_channel);
if ((test_mix_index = strchr(mix_name, ','))){
*test_mix_index = 0;
test_mix_index++;
mix_index = strtol(test_mix_index, &test_mix_index, 0);
if (*test_mix_index){
mp_tmsg(MSGT_AO,MSGL_ERR,
"[AO_ALSA] Invalid mixer index. Defaulting to 0.\n");
mix_index = 0 ;
}
}
}
if(mixer_device) card = mixer_device;
//allocate simple id
snd_mixer_selem_id_alloca(&sid);
//sets simple-mixer index and name
snd_mixer_selem_id_set_index(sid, mix_index);
snd_mixer_selem_id_set_name(sid, mix_name);
if (mixer_channel) {
free(mix_name);
mix_name = NULL;
}
if ((err = snd_mixer_open(&handle, 0)) < 0) {
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer open error: %s\n", snd_strerror(err));
return CONTROL_ERROR;
}
if ((err = snd_mixer_attach(handle, card)) < 0) {
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer attach %s error: %s\n",
card, snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer register error: %s\n", snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
err = snd_mixer_load(handle);
if (err < 0) {
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer load error: %s\n", snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
elem = snd_mixer_find_selem(handle, sid);
if (!elem) {
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to find simple control '%s',%i.\n",
snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
f_multi = (100 / (float)(pmax - pmin));
switch (cmd) {
case AOCONTROL_SET_VOLUME: {
ao_control_vol_t *vol = arg;
set_vol = vol->left / f_multi + pmin + 0.5;
//setting channels
if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting left channel, %s\n",
snd_strerror(err));
goto mixer_error;
}
mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
set_vol = vol->right / f_multi + pmin + 0.5;
if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting right channel, %s\n",
snd_strerror(err));
goto mixer_error;
}
mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
set_vol, pmin, pmax, f_multi);
break;
}
case AOCONTROL_GET_VOLUME: {
ao_control_vol_t *vol = arg;
snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
vol->left = (get_vol - pmin) * f_multi;
snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
vol->right = (get_vol - pmin) * f_multi;
mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
break;
}
case AOCONTROL_SET_MUTE: {
bool *mute = arg;
if (!snd_mixer_selem_has_playback_switch(elem))
goto mixer_error;
if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
snd_mixer_selem_set_playback_switch(
elem, SND_MIXER_SCHN_FRONT_RIGHT, !*mute);
}
snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT,
!*mute);
break;
}
case AOCONTROL_GET_MUTE: {
bool *mute = arg;
if (!snd_mixer_selem_has_playback_switch(elem))
goto mixer_error;
int tmp = 1;
snd_mixer_selem_get_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT,
&tmp);
*mute = !tmp;
if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
snd_mixer_selem_get_playback_switch(
elem, SND_MIXER_SCHN_FRONT_RIGHT, &tmp);
*mute &= !tmp;
}
break;
}
}
snd_mixer_close(handle);
return CONTROL_OK;
mixer_error:
snd_mixer_close(handle);
return CONTROL_ERROR;
}
} //end switch
return CONTROL_UNKNOWN;
}
static void parse_device (char *dest, const char *src, int len)
{
char *tmp;
memmove(dest, src, len);
dest[len] = 0;
while ((tmp = strrchr(dest, '.')))
tmp[0] = ',';
while ((tmp = strrchr(dest, '=')))
tmp[0] = ':';
}
static void print_help (void)
{
mp_tmsg (MSGT_AO, MSGL_FATAL,
"\n[AO_ALSA] -ao alsa commandline help:\n"\
"[AO_ALSA] Example: mpv -ao alsa:device=hw=0.3\n"\
"[AO_ALSA] Sets first card fourth hardware device.\n\n"\
"[AO_ALSA] Options:\n"\
"[AO_ALSA] noblock\n"\
"[AO_ALSA] Opens device in non-blocking mode.\n"\
"[AO_ALSA] device=<device-name>\n"\
"[AO_ALSA] Sets device (change , to . and : to =)\n");
}
static int str_maxlen(void *strp) {
strarg_t *str = strp;
return str->len <= ALSA_DEVICE_SIZE;
}
static int try_open_device(const char *device, int open_mode, int try_ac3)
{
int err, len;
char *ac3_device, *args;
if (try_ac3) {
/* to set the non-audio bit, use AES0=6 */
len = strlen(device);
ac3_device = malloc(len + 7 + 1);
if (!ac3_device)
return -ENOMEM;
strcpy(ac3_device, device);
args = strchr(ac3_device, ':');
if (!args) {
/* no existing parameters: add it behind device name */
strcat(ac3_device, ":AES0=6");
} else {
do
++args;
while (isspace(*args));
if (*args == '\0') {
/* ":" but no parameters */
strcat(ac3_device, "AES0=6");
} else if (*args != '{') {
/* a simple list of parameters: add it at the end of the list */
strcat(ac3_device, ",AES0=6");
} else {
/* parameters in config syntax: add it inside the { } block */
do
--len;
while (len > 0 && isspace(ac3_device[len]));
if (ac3_device[len] == '}')
strcpy(ac3_device + len, " AES0=6}");
}
}
err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
open_mode);
free(ac3_device);
if (!err)
return 0;
}
return snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
open_mode);
}
/*
open & setup audio device
return: 1=success 0=fail
*/
static int init(int rate_hz, 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:
case AF_FORMAT_IEC61937_LE:
alsa_format = SND_PCM_FORMAT_S16_LE;
break;
case AF_FORMAT_AC3_BE:
case AF_FORMAT_S16_BE:
case AF_FORMAT_IEC61937_BE:
alsa_format = SND_PCM_FORMAT_S16_BE;
break;
case AF_FORMAT_U32_LE:
alsa_format = SND_PCM_FORMAT_U32_LE;
break;
case AF_FORMAT_U32_BE:
alsa_format = SND_PCM_FORMAT_U32_BE;
break;
case AF_FORMAT_S32_LE:
alsa_format = SND_PCM_FORMAT_S32_LE;
break;
case AF_FORMAT_S32_BE:
alsa_format = SND_PCM_FORMAT_S32_BE;
break;
case AF_FORMAT_U24_LE:
alsa_format = SND_PCM_FORMAT_U24_3LE;
break;
case AF_FORMAT_U24_BE:
alsa_format = SND_PCM_FORMAT_U24_3BE;
break;
case AF_FORMAT_S24_LE:
alsa_format = SND_PCM_FORMAT_S24_3LE;
break;
case AF_FORMAT_S24_BE:
alsa_format = SND_PCM_FORMAT_S24_3BE;
break;
case AF_FORMAT_FLOAT_LE:
alsa_format = SND_PCM_FORMAT_FLOAT_LE;
break;
case AF_FORMAT_FLOAT_BE:
alsa_format = SND_PCM_FORMAT_FLOAT_BE;
break;
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) || AF_FORMAT_IS_IEC61937(format)) {
device.str = "iec958";
mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3/iec61937/iec958, %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);
alsa_can_pause = 1;
if (!alsa_handler) {
int open_mode = block ? 0 : SND_PCM_NONBLOCK;
int isac3 = AF_FORMAT_IS_AC3(format) || AF_FORMAT_IS_IEC61937(format);
//modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0)
{
if (err != -EBUSY && !block) {
mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
return 0;
}
} else {
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
return 0;
}
}
if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
mp_tmsg(MSGT_AO,MSGL_ERR,"[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(err));
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
}
snd_pcm_hw_params_t *alsa_hwparams;
snd_pcm_sw_params_t *alsa_swparams;
snd_pcm_hw_params_alloca(&alsa_hwparams);
snd_pcm_sw_params_alloca(&alsa_swparams);
// setting hw-parameters
if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
{
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get initial parameters: %s\n",
snd_strerror(err));
return 0;
}
err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set access type: %s\n",
snd_strerror(err));
return 0;
}
/* workaround for nonsupported formats
sets default format to S16_LE if the given formats aren't supported */
if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
alsa_format)) < 0)
{
mp_tmsg(MSGT_AO,MSGL_INFO,
"[AO_ALSA] Format %s is not supported by hardware, trying default.\n", af_fmt2str_short(format));
alsa_format = SND_PCM_FORMAT_S16_LE;
if (AF_FORMAT_IS_AC3(ao_data.format))
ao_data.format = AF_FORMAT_AC3_LE;
else if (AF_FORMAT_IS_IEC61937(ao_data.format))
ao_data.format = AF_FORMAT_IEC61937_LE;
else
ao_data.format = AF_FORMAT_S16_LE;
}
if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
alsa_format)) < 0)
{
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set format: %s\n",
snd_strerror(err));
return 0;
}
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 */
alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
snd_pcm_format_description(alsa_format));
} // end switch alsa_handler (spdif)
return 1;
} // end init
/* close audio device */
static void uninit(int immed)
{
if (alsa_handler) {
int err;
if (!immed)
snd_pcm_drain(alsa_handler);
if ((err = snd_pcm_close(alsa_handler)) < 0)
{
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm close error: %s\n", snd_strerror(err));
return;
}
else {
alsa_handler = NULL;
mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
}
}
else {
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] No handler defined!\n");
}
}
static void audio_pause(void)
{
int err;
if (alsa_can_pause) {
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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