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authorwm4 <wm4@nowhere>2013-04-07 21:34:09 +0200
committerwm4 <wm4@nowhere>2013-05-12 21:24:55 +0200
commit7f0f33fc8f105144eaac9653564e91599692e1e7 (patch)
treea2fd19643f27af7cf854ebda3261d74f8c093759 /audio
parent30dd18eac165b393e89b03e53518aa77ccaa85a1 (diff)
downloadmpv-7f0f33fc8f105144eaac9653564e91599692e1e7.tar.bz2
mpv-7f0f33fc8f105144eaac9653564e91599692e1e7.tar.xz
ao_alsa: uncrustify
Diffstat (limited to 'audio')
-rw-r--r--audio/out/ao_alsa.c1340
1 files changed, 717 insertions, 623 deletions
diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
index a194ce0478..7c925687c9 100644
--- a/audio/out/ao_alsa.c
+++ b/audio/out/ao_alsa.c
@@ -60,7 +60,7 @@ static const ao_info_t info =
LIBAO_EXTERN(alsa)
-static snd_pcm_t *alsa_handler;
+static snd_pcm_t * alsa_handler;
static snd_pcm_format_t alsa_format;
#define BUFFER_TIME 500000 // 0.5 s
@@ -75,263 +75,289 @@ static float delay_before_pause;
#define ALSA_DEVICE_SIZE 256
static void alsa_error_handler(const char *file, int line, const char *function,
- int err, const char *format, ...)
+ 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);
+ 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:
+ 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_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;
+ 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_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;
+ }
+ }
}
- break;
- }
- }
- snd_mixer_close(handle);
- return CONTROL_OK;
- mixer_error:
- snd_mixer_close(handle);
- return CONTROL_ERROR;
+ 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;
+ } //end switch
+ return CONTROL_UNKNOWN;
}
-static void parse_device (char *dest, const char *src, int len)
+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] = ':';
+ 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)
+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");
+ 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 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}");
- }
+ 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;
}
- 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);
+ 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, const struct mp_chmap *channels, int format,
int flags)
{
@@ -342,84 +368,85 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
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}
+ {"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,
- ao_data.channels.num, format);
+ mp_msg(MSGT_AO, MSGL_V,
+ "alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
+ ao_data.channels.num,
+ format);
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());
prepause_frames = 0;
delay_before_pause = 0;
snd_lib_error_set_handler(alsa_error_handler);
- 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;
-
- default:
- alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
- break;
- }
+ 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;
+
+ default:
+ alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
+ break;
+ }
//subdevice parsing
// set defaults
@@ -431,45 +458,47 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
* 'iec958'
*/
if (AF_FORMAT_IS_IEC61937(format)) {
- device.str = "iec958";
- mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n", ao_data.channels.num);
- }
- else
+ device.str = "iec958";
+ mp_msg(MSGT_AO, MSGL_V,
+ "alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n",
+ ao_data.channels.num);
+ } else
/* in any case for multichannel playback we should select
* appropriate device
*/
switch (ao_data.channels.num) {
- 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",
- ao_data.channels.num);
+ 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",
+ ao_data.channels.num);
}
device.len = strlen(device.str);
if (subopt_parse(ao_subdevice, subopts) != 0) {
@@ -478,200 +507,254 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
}
parse_device(alsa_device, device.str, device.len);
- mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
+ 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_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;
- }
-
- int num_channels = ao_data.channels.num;
- if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
- &num_channels)) < 0)
- {
- mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set channels: %s\n",
- snd_strerror(err));
- return 0;
- }
- mp_chmap_from_channels(&ao_data.channels, num_channels);
- if (!AF_FORMAT_IS_IEC61937(format))
- mp_chmap_reorder_to_alsa(&ao_data.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 */
- 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.num;
- 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.num, (int)bytes_per_sample, ao_data.buffersize,
- snd_pcm_format_description(alsa_format));
+ int open_mode = block ? 0 : SND_PCM_NONBLOCK;
+ int isac3 = 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;
+ }
+
+ int num_channels = ao_data.channels.num;
+ if ((err =
+ snd_pcm_hw_params_set_channels_near(alsa_handler,
+ alsa_hwparams,
+ &num_channels)) < 0) {
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Unable to set channels: %s\n",
+ snd_strerror(err));
+ return 0;
+ }
+ mp_chmap_from_channels(&ao_data.channels, num_channels);
+ if (!AF_FORMAT_IS_IEC61937(format))
+ mp_chmap_reorder_to_alsa(&ao_data.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 */
+ 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.num;
+ 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.num, (int)bytes_per_sample,
+ ao_data.buffersize,
+ snd_pcm_format_description(
+ alsa_format));
} // end switch alsa_handler (spdif)
return 1;
@@ -682,25 +765,23 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
static void uninit(int immed)
{
- if (alsa_handler) {
- int err;
+ 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");
- }
+ 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)
@@ -709,21 +790,23 @@ static void audio_pause(void)
if (alsa_can_pause) {
delay_before_pause = get_delay();
- 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));
+ 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");
+ 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;
delay_before_pause = prepause_frames / (float)ao_data.samplerate;
- if ((err = snd_pcm_drop(alsa_handler)) < 0)
- {
- mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm drop error: %s\n", snd_strerror(err));
+ 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;
}
}
@@ -734,20 +817,24 @@ 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);
+ 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));
+ 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");
+ 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));
+ 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) {
@@ -765,15 +852,17 @@ static void reset(void)
prepause_frames = 0;
delay_before_pause = 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_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;
+ 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;
}
@@ -783,50 +872,55 @@ static void reset(void)
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)
+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));
- break;
- }
- res = 0;
- }
- } while (res == 0);
-
- return res < 0 ? 0 : res * bytes_per_sample;
+ 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));
+ break;
+ }
+ res = 0;
+ }
+ } while (res == 0);
+
+ return res < 0 ? 0 : res * bytes_per_sample;
}
/* how many byes are free in the buffer */
@@ -837,10 +931,11 @@ static int get_space(void)
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;
+ 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;
@@ -852,22 +947,21 @@ static int get_space(void)
/* delay in seconds between first and last sample in buffer */
static float get_delay(void)
{
- if (alsa_handler) {
- snd_pcm_sframes_t delay;
+ if (alsa_handler) {
+ snd_pcm_sframes_t delay;
- if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_PAUSED)
- return delay_before_pause;
+ if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_PAUSED)
+ return delay_before_pause;
- if (snd_pcm_delay(alsa_handler, &delay) < 0)
- return 0;
+ 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;
- }
+ 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;
}