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Code Issues Releases Wiki Activity

ao_alsa: switch to new AO API

This commit is contained in:
wm4 2013-04-19 00:16:23 +02:00
parent 74487b8430
commit e1207f2ceb
1 changed files with 54 additions and 39 deletions
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93
audio/out/ao_alsa.c
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@ -36,6 +36,7 @@
#include <alloca.h> #include <alloca.h>
#include "config.h" #include "config.h"
#include "core/options.h"
#include "core/subopt-helper.h" #include "core/subopt-helper.h"
#include "audio/mixer.h" #include "audio/mixer.h"
#include "core/mp_msg.h" #include "core/mp_msg.h"
@ -46,19 +47,11 @@
#include <alsa/asoundlib.h> #include <alsa/asoundlib.h>
#include "ao.h" #include "ao.h"
#include "audio_out_internal.h"
#include "audio/format.h" #include "audio/format.h"
#include "audio/reorder_ch.h" #include "audio/reorder_ch.h"
static const ao_info_t info = extern struct ao *global_ao;
{ #define ao_data (*global_ao)
"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_t *alsa_handler;
static snd_pcm_format_t alsa_format; static snd_pcm_format_t alsa_format;
@ -83,6 +76,9 @@ static float delay_before_pause;
} \ } \
} while (0) } while (0)
static float get_delay(struct ao *ao);
static int play(struct ao *ao, void *data, int len, int flags);
static void alsa_error_handler(const char *file, int line, const char *function, static void alsa_error_handler(const char *file, int line, const char *function,
int err, const char *format, ...) int err, const char *format, ...)
{ {
@ -103,7 +99,7 @@ static void alsa_error_handler(const char *file, int line, const char *function,
} }
/* to set/get/query special features/parameters */ /* to set/get/query special features/parameters */
static int control(int cmd, void *arg) static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{ {
snd_mixer_t *handle = NULL; snd_mixer_t *handle = NULL;
switch (cmd) { switch (cmd) {
@ -127,10 +123,10 @@ static int control(int cmd, void *arg)
if (AF_FORMAT_IS_IEC61937(ao_data.format)) if (AF_FORMAT_IS_IEC61937(ao_data.format))
return CONTROL_TRUE; return CONTROL_TRUE;
if (mixer_channel) { if (global_ao->opts->mixer_channel) {
char *test_mix_index; char *test_mix_index;
mix_name = strdup(mixer_channel); mix_name = strdup(global_ao->opts->mixer_channel);
if ((test_mix_index = strchr(mix_name, ','))) { if ((test_mix_index = strchr(mix_name, ','))) {
*test_mix_index = 0; *test_mix_index = 0;
test_mix_index++; test_mix_index++;
@ -143,8 +139,8 @@ static int control(int cmd, void *arg)
} }
} }
} }
if (mixer_device) if (global_ao->opts->mixer_device)
card = mixer_device; card = global_ao->opts->mixer_device;
//allocate simple id //allocate simple id
snd_mixer_selem_id_alloca(&sid); snd_mixer_selem_id_alloca(&sid);
@ -153,7 +149,7 @@ static int control(int cmd, void *arg)
snd_mixer_selem_id_set_index(sid, mix_index); snd_mixer_selem_id_set_index(sid, mix_index);
snd_mixer_selem_id_set_name(sid, mix_name); snd_mixer_selem_id_set_name(sid, mix_name);
if (mixer_channel) { if (global_ao->opts->mixer_channel) {
free(mix_name); free(mix_name);
mix_name = NULL; mix_name = NULL;
} }
@ -402,10 +398,9 @@ static int try_open_device(const char *device, int open_mode, int try_ac3)
/* /*
open & setup audio device open & setup audio device
return: 1=success 0=fail return: 0=success -1=fail
*/ */
static int init(int rate_hz, const struct mp_chmap *channels, int format, static int init(struct ao *ao, char *params)
int flags)
{ {
int err; int err;
int block; int block;
@ -419,14 +414,15 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
{NULL} {NULL}
}; };
global_ao = ao;
char alsa_device[ALSA_DEVICE_SIZE + 1]; char alsa_device[ALSA_DEVICE_SIZE + 1];
// make sure alsa_device is null-terminated even when using strncpy etc. // make sure alsa_device is null-terminated even when using strncpy etc.
memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1); memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
mp_msg(MSGT_AO, MSGL_V, mp_msg(MSGT_AO, MSGL_V,
"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz, "alsa-init: requested format: %d Hz, %d channels, %x\n",
ao_data.channels.num, ao->samplerate, ao->channels.num, ao->format);
format);
alsa_handler = NULL; alsa_handler = NULL;
mp_msg(MSGT_AO, MSGL_V, "alsa-init: using ALSA %s\n", snd_asoundlib_version()); mp_msg(MSGT_AO, MSGL_V, "alsa-init: using ALSA %s\n", snd_asoundlib_version());
@ -435,7 +431,7 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
snd_lib_error_set_handler(alsa_error_handler); snd_lib_error_set_handler(alsa_error_handler);
alsa_format = find_alsa_format(format); alsa_format = find_alsa_format(ao->format);
//subdevice parsing //subdevice parsing
// set defaults // set defaults
@ -447,7 +443,7 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
* 'iec958' * 'iec958'
*/ */
device.str = NULL; device.str = NULL;
if (AF_FORMAT_IS_IEC61937(format)) { if (AF_FORMAT_IS_IEC61937(ao->format)) {
device.str = "iec958"; device.str = "iec958";
mp_msg(MSGT_AO, MSGL_V, mp_msg(MSGT_AO, MSGL_V,
"alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n", "alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n",
@ -463,14 +459,14 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
name, ao_data.channels.num); name, ao_data.channels.num);
talloc_free(name); talloc_free(name);
} }
if (strcmp(device.str, "default") != 0 && format == AF_FORMAT_FLOAT_NE) if (strcmp(device.str, "default") != 0 && ao->format == AF_FORMAT_FLOAT_NE)
{ {
// hack - use the converter plugin (why the heck?) // hack - use the converter plugin (why the heck?)
device.str = talloc_asprintf(global_ao, "plug:%s", device.str); device.str = talloc_asprintf(global_ao, "plug:%s", device.str);
} }
} }
device.len = strlen(device.str); device.len = strlen(device.str);
if (subopt_parse(ao_subdevice, subopts) != 0) { if (subopt_parse(params, subopts) != 0) {
print_help(); print_help();
return 0; return 0;
} }
@ -481,7 +477,7 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
alsa_can_pause = 1; alsa_can_pause = 1;
int open_mode = block ? 0 : SND_PCM_NONBLOCK; int open_mode = block ? 0 : SND_PCM_NONBLOCK;
int isac3 = AF_FORMAT_IS_IEC61937(format); int isac3 = AF_FORMAT_IS_IEC61937(ao->format);
//modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
err = try_open_device(alsa_device, open_mode, isac3); err = try_open_device(alsa_device, open_mode, isac3);
if (err < 0) { if (err < 0) {
@ -522,7 +518,7 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
(alsa_handler, alsa_hwparams, alsa_format); (alsa_handler, alsa_hwparams, alsa_format);
if (err < 0) { if (err < 0) {
mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Format %s is not supported " mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Format %s is not supported "
"by hardware, trying default.\n", af_fmt2str_short(format)); "by hardware, trying default.\n", af_fmt2str_short(ao->format));
alsa_format = SND_PCM_FORMAT_S16_LE; alsa_format = SND_PCM_FORMAT_S16_LE;
if (AF_FORMAT_IS_AC3(ao_data.format)) if (AF_FORMAT_IS_AC3(ao_data.format))
ao_data.format = AF_FORMAT_AC3_LE; ao_data.format = AF_FORMAT_AC3_LE;
@ -619,15 +615,15 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
ao_data.samplerate, ao_data.channels.num, (int)bytes_per_sample, ao_data.samplerate, ao_data.channels.num, (int)bytes_per_sample,
ao_data.buffersize, snd_pcm_format_description(alsa_format)); ao_data.buffersize, snd_pcm_format_description(alsa_format));
return 1; return 0;
alsa_error: alsa_error:
return 0; return -1;
} // end init } // end init
/* close audio device */ /* close audio device */
static void uninit(int immed) static void uninit(struct ao *ao, bool immed)
{ {
if (alsa_handler) { if (alsa_handler) {
@ -648,12 +644,12 @@ static void uninit(int immed)
alsa_error: ; alsa_error: ;
} }
static void audio_pause(void) static void audio_pause(struct ao *ao)
{ {
int err; int err;
if (alsa_can_pause) { if (alsa_can_pause) {
delay_before_pause = get_delay(); delay_before_pause = get_delay(ao);
err = snd_pcm_pause(alsa_handler, 1); err = snd_pcm_pause(alsa_handler, 1);
CHECK_ALSA_ERROR("pcm pause error"); CHECK_ALSA_ERROR("pcm pause error");
mp_msg(MSGT_AO, MSGL_V, "alsa-pause: pause supported by hardware\n"); mp_msg(MSGT_AO, MSGL_V, "alsa-pause: pause supported by hardware\n");
@ -670,7 +666,7 @@ static void audio_pause(void)
alsa_error: ; alsa_error: ;
} }
static void audio_resume(void) static void audio_resume(struct ao *ao)
{ {
int err; int err;
@ -689,7 +685,7 @@ static void audio_resume(void)
CHECK_ALSA_ERROR("pcm prepare error"); CHECK_ALSA_ERROR("pcm prepare error");
if (prepause_frames) { if (prepause_frames) {
void *silence = calloc(prepause_frames, bytes_per_sample); void *silence = calloc(prepause_frames, bytes_per_sample);
play(silence, prepause_frames * bytes_per_sample, 0); play(ao, silence, prepause_frames * bytes_per_sample, 0);
free(silence); free(silence);
} }
} }
@ -698,7 +694,7 @@ alsa_error: ;
} }
/* stop playing and empty buffers (for seeking/pause) */ /* stop playing and empty buffers (for seeking/pause) */
static void reset(void) static void reset(struct ao *ao)
{ {
int err; int err;
@ -719,7 +715,7 @@ alsa_error: ;
thanxs for marius <marius@rospot.com> for giving us the light ;) thanxs for marius <marius@rospot.com> for giving us the light ;)
*/ */
static int play(void *data, int len, int flags) static int play(struct ao *ao, void *data, int len, int flags)
{ {
int num_frames; int num_frames;
snd_pcm_sframes_t res = 0; snd_pcm_sframes_t res = 0;
@ -768,7 +764,7 @@ alsa_error:
} }
/* how many byes are free in the buffer */ /* how many byes are free in the buffer */
static int get_space(void) static int get_space(struct ao *ao)
{ {
snd_pcm_status_t *status; snd_pcm_status_t *status;
int err; int err;
@ -788,7 +784,7 @@ alsa_error:
} }
/* delay in seconds between first and last sample in buffer */ /* delay in seconds between first and last sample in buffer */
static float get_delay(void) static float get_delay(struct ao *ao)
{ {
if (alsa_handler) { if (alsa_handler) {
snd_pcm_sframes_t delay; snd_pcm_sframes_t delay;
@ -808,3 +804,22 @@ static float get_delay(void)
} else } else
return 0; return 0;
} }
const struct ao_driver audio_out_alsa = {
.is_new = true,
.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,
};