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mpv/audio/out/ao_alsa.c
wm4 4d3a2c7e0d audio/out: remove ao->outburst/buffersize fields 
The core didn't use these fields, and use of them was inconsistent
accross AOs. Some didn't use them at all. Some only set them; the values
were completely unused by the core. Some made full use of them.

Remove these fields. In places where they are still needed, make them
private AO state.

Remove the --abs option. It set the buffer size for ao_oss and ao_dsound
(being ignored by all other AOs), and was already marked as obsolete. If
it turns out that it's still needed for ao_oss or ao_dsound, their
default buffer sizes could be adjusted, and if even that doesn't help,
AO suboptions could be added in these cases.
2013-06-16 19:36:56 +02:00

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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/options.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/format.h"
#include "audio/reorder_ch.h"
struct priv {
snd_pcm_t *alsa;
snd_pcm_format_t alsa_fmt;
size_t bytes_per_sample;
int can_pause;
snd_pcm_sframes_t prepause_frames;
float delay_before_pause;
int buffersize;
int outburst;
};
#define BUFFER_TIME 500000 // 0.5 s
#define FRAGCOUNT 16
#define ALSA_DEVICE_SIZE 256
#define CHECK_ALSA_ERROR(message) \
do { \
if (err < 0) { \
mp_msg(MSGT_VO, MSGL_ERR, "[AO_ALSA] %s: %s\n", \
(message), snd_strerror(err)); \
goto alsa_error; \
} \
} while (0)
static float get_delay(struct ao *ao);
static int play(struct ao *ao, void *data, int len, int flags);
static void uninit(struct ao *ao, bool immed);
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(struct ao *ao, enum aocontrol cmd, void *arg)
{
snd_mixer_t *handle = NULL;
switch (cmd) {
case AOCONTROL_GET_MUTE:
case AOCONTROL_SET_MUTE:
case AOCONTROL_GET_VOLUME:
case AOCONTROL_SET_VOLUME:
{
int err;
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_IEC61937(ao->format))
return CONTROL_TRUE;
if (ao->opts->mixer_channel) {
char *test_mix_index;
mix_name = strdup(ao->opts->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 (ao->opts->mixer_device)
card = ao->opts->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 (ao->opts->mixer_channel) {
free(mix_name);
mix_name = NULL;
}
err = snd_mixer_open(&handle, 0);
CHECK_ALSA_ERROR("Mixer open error");
err = snd_mixer_attach(handle, card);
CHECK_ALSA_ERROR("Mixer attach error");
err = snd_mixer_selem_register(handle, NULL, NULL);
CHECK_ALSA_ERROR("Mixer register error");
err = snd_mixer_load(handle);
CHECK_ALSA_ERROR("Mixer load 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));
goto alsa_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
err = snd_mixer_selem_set_playback_volume
(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol);
CHECK_ALSA_ERROR("Error setting left channel");
mp_msg(MSGT_AO, MSGL_DBG2, "left=%li, ", set_vol);
set_vol = vol->right / f_multi + pmin + 0.5;
err = snd_mixer_selem_set_playback_volume
(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol);
CHECK_ALSA_ERROR("Error setting right channel");
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 alsa_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 alsa_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;
}
} //end switch
return CONTROL_UNKNOWN;
alsa_error:
if (handle)
snd_mixer_close(handle);
return CONTROL_ERROR;
}
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 const int mp_to_alsa_format[][2] = {
{AF_FORMAT_S8, SND_PCM_FORMAT_S8},
{AF_FORMAT_U8, SND_PCM_FORMAT_U8},
{AF_FORMAT_U16_LE, SND_PCM_FORMAT_U16_LE},
{AF_FORMAT_U16_BE, SND_PCM_FORMAT_U16_BE},
{AF_FORMAT_S16_LE, SND_PCM_FORMAT_S16_LE},
{AF_FORMAT_S16_BE, SND_PCM_FORMAT_S16_BE},
{AF_FORMAT_U32_LE, SND_PCM_FORMAT_U32_LE},
{AF_FORMAT_U32_BE, SND_PCM_FORMAT_U32_BE},
{AF_FORMAT_S32_LE, SND_PCM_FORMAT_S32_LE},
{AF_FORMAT_S32_BE, SND_PCM_FORMAT_S32_BE},
{AF_FORMAT_U24_LE, SND_PCM_FORMAT_U24_3LE},
{AF_FORMAT_U24_BE, SND_PCM_FORMAT_U24_3BE},
{AF_FORMAT_S24_LE, SND_PCM_FORMAT_S24_3LE},
{AF_FORMAT_S24_BE, SND_PCM_FORMAT_S24_3BE},
{AF_FORMAT_FLOAT_LE, SND_PCM_FORMAT_FLOAT_LE},
{AF_FORMAT_FLOAT_BE, SND_PCM_FORMAT_FLOAT_BE},
{AF_FORMAT_AC3_LE, SND_PCM_FORMAT_S16_LE},
{AF_FORMAT_AC3_BE, SND_PCM_FORMAT_S16_BE},
{AF_FORMAT_IEC61937_LE, SND_PCM_FORMAT_S16_LE},
{AF_FORMAT_IEC61937_BE, SND_PCM_FORMAT_S16_BE},
{AF_FORMAT_MPEG2, SND_PCM_FORMAT_MPEG},
{AF_FORMAT_UNKNOWN, SND_PCM_FORMAT_UNKNOWN},
};
static int find_alsa_format(int af_format)
{
for (int n = 0; mp_to_alsa_format[n][0] != AF_FORMAT_UNKNOWN; n++) {
if (mp_to_alsa_format[n][0] == af_format)
return mp_to_alsa_format[n][1];
}
return SND_PCM_FORMAT_UNKNOWN;
}
// Lists device names and their implied channel map.
// The second item must be resolvable with mp_chmap_from_str().
// Source: http://www.alsa-project.org/main/index.php/DeviceNames
// (Speaker names are slightly different from mpv's.)
static const char *device_channel_layouts[][2] = {
{"default", "fc"},
{"default", "fl-fr"},
{"rear", "bl-br"},
{"center_lfe", "fc-lfe"},
{"side", "sl-sr"},
{"surround40", "fl-fr-bl-br"},
{"surround50", "fl-fr-bl-br-fc"},
{"surround41", "fl-fr-bl-br-lfe"},
{"surround51", "fl-fr-bl-br-fc-lfe"},
{"surround71", "fl-fr-bl-br-fc-lfe-sl-sr"},
};
#define ARRAY_LEN(x) (sizeof(x) / sizeof((x)[0]))
#define NUM_ALSA_CHMAPS ARRAY_LEN(device_channel_layouts)
static const char *select_chmap(struct ao *ao)
{
struct mp_chmap_sel sel = {0};
struct mp_chmap maps[NUM_ALSA_CHMAPS];
for (int n = 0; n < NUM_ALSA_CHMAPS; n++) {
mp_chmap_from_str(&maps[n], bstr0(device_channel_layouts[n][1]));
mp_chmap_sel_add_map(&sel, &maps[n]);
};
if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
return NULL;
for (int n = 0; n < NUM_ALSA_CHMAPS; n++) {
if (mp_chmap_equals(&ao->channels, &maps[n]))
return device_channel_layouts[n][0];
}
char *name = mp_chmap_to_str(&ao->channels);
mp_tmsg(MSGT_AO, MSGL_ERR,
"[AO_ALSA] channel layout %s (%d ch) not supported.\n",
name, ao->channels.num);
talloc_free(name);
return "default";
}
static int try_open_device(struct ao *ao, const char *device, int open_mode,
int try_ac3)
{
struct priv *p = ao->priv;
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
(&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, open_mode);
free(ac3_device);
if (!err)
return 0;
}
return snd_pcm_open
(&p->alsa, device, SND_PCM_STREAM_PLAYBACK, open_mode);
}
/*
open & setup audio device
return: 0=success -1=fail
*/
static int init(struct ao *ao, char *params)
{
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}
};
struct priv *p = talloc_zero(ao, struct priv);
ao->priv = p;
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",
ao->samplerate, ao->channels.num, ao->format);
p->prepause_frames = 0;
p->delay_before_pause = 0;
//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'
*/
device.str = NULL;
if (AF_FORMAT_IS_IEC61937(ao->format)) {
device.str = "iec958";
mp_msg(MSGT_AO, MSGL_V,
"alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n",
ao->channels.num);
} else {
device.str = select_chmap(ao);
if (strcmp(device.str, "default") != 0 && ao->format == AF_FORMAT_FLOAT_NE)
{
// hack - use the converter plugin (why the heck?)
device.str = talloc_asprintf(ao, "plug:%s", device.str);
}
}
device.len = strlen(device.str);
if (subopt_parse(params, 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);
p->can_pause = 1;
mp_msg(MSGT_AO, MSGL_V, "alsa-init: using ALSA %s\n", snd_asoundlib_version());
snd_lib_error_set_handler(alsa_error_handler);
int open_mode = block ? 0 : SND_PCM_NONBLOCK;
int isac3 = AF_FORMAT_IS_IEC61937(ao->format);
//modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
err = try_open_device(ao, alsa_device, open_mode, isac3);
if (err < 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");
err = try_open_device(ao, alsa_device, 0, isac3);
}
CHECK_ALSA_ERROR("Playback open error");
}
err = snd_pcm_nonblock(p->alsa, 0);
if (err < 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
err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams);
CHECK_ALSA_ERROR("Unable to get initial parameters");
err = snd_pcm_hw_params_set_access
(p->alsa, alsa_hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
CHECK_ALSA_ERROR("Unable to set access type");
p->alsa_fmt = find_alsa_format(ao->format);
if (p->alsa_fmt == SND_PCM_FORMAT_UNKNOWN) {
p->alsa_fmt = SND_PCM_FORMAT_S16;
ao->format = AF_FORMAT_S16_NE;
}
err = snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt);
if (err < 0) {
mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Format %s is not supported "
"by hardware, trying default.\n", af_fmt2str_short(ao->format));
p->alsa_fmt = SND_PCM_FORMAT_S16_LE;
if (AF_FORMAT_IS_AC3(ao->format))
ao->format = AF_FORMAT_AC3_LE;
else if (AF_FORMAT_IS_IEC61937(ao->format))
ao->format = AF_FORMAT_IEC61937_LE;
else
ao->format = AF_FORMAT_S16_LE;
}
err = snd_pcm_hw_params_set_format(p->alsa, alsa_hwparams, p->alsa_fmt);
CHECK_ALSA_ERROR("Unable to set format");
int num_channels = ao->channels.num;
err = snd_pcm_hw_params_set_channels_near
(p->alsa, alsa_hwparams, &num_channels);
CHECK_ALSA_ERROR("Unable to set channels");
if (num_channels != ao->channels.num) {
mp_tmsg(MSGT_AO, MSGL_ERR,
"[AO_ALSA] Couldn't get requested number of channels.\n");
mp_chmap_from_channels_alsa(&ao->channels, num_channels);
}
/* 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 */
err = snd_pcm_hw_params_set_rate_resample(p->alsa, alsa_hwparams, 0);
CHECK_ALSA_ERROR("Unable to disable resampling");
err = snd_pcm_hw_params_set_rate_near
(p->alsa, alsa_hwparams, &ao->samplerate, NULL);
CHECK_ALSA_ERROR("Unable to set samplerate-2");
p->bytes_per_sample = af_fmt2bits(ao->format) / 8;
p->bytes_per_sample *= ao->channels.num;
err = snd_pcm_hw_params_set_buffer_time_near
(p->alsa, alsa_hwparams, &(unsigned int){BUFFER_TIME}, NULL);
CHECK_ALSA_ERROR("Unable to set buffer time near");
err = snd_pcm_hw_params_set_periods_near
(p->alsa, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL);
CHECK_ALSA_ERROR("Unable to set periods");
/* finally install hardware parameters */
err = snd_pcm_hw_params(p->alsa, alsa_hwparams);
CHECK_ALSA_ERROR("Unable to set hw-parameters");
// end setting hw-params
// gets buffersize for control
err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize);
CHECK_ALSA_ERROR("Unable to get buffersize");
p->buffersize = bufsize * p->bytes_per_sample;
mp_msg(MSGT_AO, MSGL_V, "alsa-init: got buffersize=%i\n",
p->buffersize);
err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL);
CHECK_ALSA_ERROR("Unable to get period size");
mp_msg(MSGT_AO, MSGL_V, "alsa-init: got period size %li\n", chunk_size);
p->outburst = chunk_size * p->bytes_per_sample;
/* setting software parameters */
err = snd_pcm_sw_params_current(p->alsa, alsa_swparams);
CHECK_ALSA_ERROR("Unable to get sw-parameters");
err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary);
CHECK_ALSA_ERROR("Unable to get boundary");
/* start playing when one period has been written */
err = snd_pcm_sw_params_set_start_threshold
(p->alsa, alsa_swparams, chunk_size);
CHECK_ALSA_ERROR("Unable to set start threshold");
/* disable underrun reporting */
err = snd_pcm_sw_params_set_stop_threshold
(p->alsa, alsa_swparams, boundary);
CHECK_ALSA_ERROR("Unable to set stop threshold");
/* play silence when there is an underrun */
err = snd_pcm_sw_params_set_silence_size
(p->alsa, alsa_swparams, boundary);
CHECK_ALSA_ERROR("Unable to set silence size");
err = snd_pcm_sw_params(p->alsa, alsa_swparams);
CHECK_ALSA_ERROR("Unable to get sw-parameters");
/* end setting sw-params */
p->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->samplerate, ao->channels.num, (int)p->bytes_per_sample,
p->buffersize, snd_pcm_format_description(p->alsa_fmt));
return 0;
alsa_error:
uninit(ao, true);
return -1;
} // end init
/* close audio device */
static void uninit(struct ao *ao, bool immed)
{
struct priv *p = ao->priv;
if (p->alsa) {
int err;
if (!immed)
snd_pcm_drain(p->alsa);
err = snd_pcm_close(p->alsa);
CHECK_ALSA_ERROR("pcm close error");
mp_msg(MSGT_AO, MSGL_V, "alsa-uninit: pcm closed\n");
}
alsa_error:
p->alsa = NULL;
snd_lib_error_set_handler(NULL);
}
static void audio_pause(struct ao *ao)
{
struct priv *p = ao->priv;
int err;
if (p->can_pause) {
p->delay_before_pause = get_delay(ao);
err = snd_pcm_pause(p->alsa, 1);
CHECK_ALSA_ERROR("pcm pause error");
mp_msg(MSGT_AO, MSGL_V, "alsa-pause: pause supported by hardware\n");
} else {
if (snd_pcm_delay(p->alsa, &p->prepause_frames) < 0
|| p->prepause_frames < 0)
p->prepause_frames = 0;
p->delay_before_pause = p->prepause_frames / (float)ao->samplerate;
err = snd_pcm_drop(p->alsa);
CHECK_ALSA_ERROR("pcm drop error");
}
alsa_error: ;
}
static void audio_resume(struct ao *ao)
{
struct priv *p = ao->priv;
int err;
if (snd_pcm_state(p->alsa) == 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(p->alsa)) == -EAGAIN)
sleep(1);
}
if (p->can_pause) {
err = snd_pcm_pause(p->alsa, 0);
CHECK_ALSA_ERROR("pcm resume error");
mp_msg(MSGT_AO, MSGL_V, "alsa-resume: resume supported by hardware\n");
} else {
err = snd_pcm_prepare(p->alsa);
CHECK_ALSA_ERROR("pcm prepare error");
if (p->prepause_frames) {
void *silence = calloc(p->prepause_frames, p->bytes_per_sample);
play(ao, silence, p->prepause_frames * p->bytes_per_sample, 0);
free(silence);
}
}
alsa_error: ;
}
/* stop playing and empty buffers (for seeking/pause) */
static void reset(struct ao *ao)
{
struct priv *p = ao->priv;
int err;
p->prepause_frames = 0;
p->delay_before_pause = 0;
err = snd_pcm_drop(p->alsa);
CHECK_ALSA_ERROR("pcm prepare error");
err = snd_pcm_prepare(p->alsa);
CHECK_ALSA_ERROR("pcm prepare error");
alsa_error: ;
}
/*
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(struct ao *ao, void *data, int len, int flags)
{
struct priv *p = ao->priv;
int num_frames;
snd_pcm_sframes_t res = 0;
if (!(flags & AOPLAY_FINAL_CHUNK))
len = len / p->outburst * p->outburst;
num_frames = len / p->bytes_per_sample;
//mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
if (!p->alsa) {
mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Device configuration error.");
return 0;
}
if (num_frames == 0)
return 0;
do {
res = snd_pcm_writei(p->alsa, 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(p->alsa)) == -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");
res = snd_pcm_prepare(p->alsa);
int err = res;
CHECK_ALSA_ERROR("pcm prepare error");
res = 0;
}
} while (res == 0);
return res < 0 ? 0 : res * p->bytes_per_sample;
alsa_error:
return 0;
}
/* how many byes are free in the buffer */
static int get_space(struct ao *ao)
{
struct priv *p = ao->priv;
snd_pcm_status_t *status;
int err;
snd_pcm_status_alloca(&status);
err = snd_pcm_status(p->alsa, status);
CHECK_ALSA_ERROR("cannot get pcm status");
unsigned space = snd_pcm_status_get_avail(status) * p->bytes_per_sample;
if (space > p->buffersize) // Buffer underrun?
space = p->buffersize;
return space;
alsa_error:
return 0;
}
/* delay in seconds between first and last sample in buffer */
static float get_delay(struct ao *ao)
{
struct priv *p = ao->priv;
if (p->alsa) {
snd_pcm_sframes_t delay;
if (snd_pcm_state(p->alsa) == SND_PCM_STATE_PAUSED)
return p->delay_before_pause;
if (snd_pcm_delay(p->alsa, &delay) < 0)
return 0;
if (delay < 0) {
/* underrun - move the application pointer forward to catch up */
snd_pcm_forward(p->alsa, -delay);
delay = 0;
}
return (float)delay / (float)ao->samplerate;
} else
return 0;
}
const struct ao_driver audio_out_alsa = {
.info = &(const struct ao_info) {
"ALSA-0.9.x-1.x audio output",
"alsa",
"Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
"under development"
},
.init = init,
.uninit = uninit,
.control = control,
.get_space = get_space,
.play = play,
.get_delay = get_delay,
.pause = audio_pause,
.resume = audio_resume,
.reset = reset,
};
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