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Add mute functionality. Add config option to not try to open capture device.

Adds "console {mute|unmute}" CLI command.
Adds mute and noaudiocapture config options (will update sample configs shortly).

(closes issue #14673)
Reported by: Nick_Lewis
Patches:
      chan_alsa.c-oneway3.patch uploaded by Nick Lewis (license 657)
Tested by: qwell


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229753 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
qwell 2009-11-12 23:37:36 +00:00
parent f1a42f813c
commit f4fff33b9b
1 changed files with 109 additions and 23 deletions

View File

@ -129,6 +129,8 @@ static int readdev = -1;
static int writedev = -1;
static int autoanswer = 1;
static int mute = 0;
static int noaudiocapture = 0;
static struct ast_channel *alsa_request(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause);
static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration);
@ -265,15 +267,22 @@ static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream)
static int soundcard_init(void)
{
alsa.icard = alsa_card_init(indevname, SND_PCM_STREAM_CAPTURE);
if (!noaudiocapture) {
alsa.icard = alsa_card_init(indevname, SND_PCM_STREAM_CAPTURE);
if (!alsa.icard) {
ast_log(LOG_ERROR, "Problem opening alsa capture device\n");
return -1;
}
}
alsa.ocard = alsa_card_init(outdevname, SND_PCM_STREAM_PLAYBACK);
if (!alsa.icard || !alsa.ocard) {
ast_log(LOG_ERROR, "Problem opening ALSA I/O devices\n");
if (!alsa.ocard) {
ast_log(LOG_ERROR, "Problem opening ALSA playback device\n");
return -1;
}
return readdev;
return writedev;
}
static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration)
@ -310,6 +319,9 @@ static int alsa_call(struct ast_channel *c, char *dest, int timeout)
ast_verbose(" << Call placed to '%s' on console >> \n", dest);
if (autoanswer) {
ast_verbose(" << Auto-answered >> \n");
if (mute) {
ast_verbose( " << Muted >> \n" );
}
grab_owner();
if (alsa.owner) {
f.subclass.integer = AST_CONTROL_ANSWER;
@ -326,8 +338,10 @@ static int alsa_call(struct ast_channel *c, char *dest, int timeout)
ast_indicate(alsa.owner, AST_CONTROL_RINGING);
}
}
snd_pcm_prepare(alsa.icard);
snd_pcm_start(alsa.icard);
if (!noaudiocapture) {
snd_pcm_prepare(alsa.icard);
snd_pcm_start(alsa.icard);
}
ast_mutex_unlock(&alsalock);
return 0;
@ -338,8 +352,10 @@ static int alsa_answer(struct ast_channel *c)
ast_mutex_lock(&alsalock);
ast_verbose(" << Console call has been answered >> \n");
ast_setstate(c, AST_STATE_UP);
snd_pcm_prepare(alsa.icard);
snd_pcm_start(alsa.icard);
if (!noaudiocapture) {
snd_pcm_prepare(alsa.icard);
snd_pcm_start(alsa.icard);
}
ast_mutex_unlock(&alsalock);
return 0;
@ -353,7 +369,9 @@ static int alsa_hangup(struct ast_channel *c)
ast_verbose(" << Hangup on console >> \n");
ast_module_unref(ast_module_info->self);
hookstate = 0;
snd_pcm_drop(alsa.icard);
if (!noaudiocapture) {
snd_pcm_drop(alsa.icard);
}
ast_mutex_unlock(&alsalock);
return 0;
@ -436,6 +454,12 @@ static struct ast_frame *alsa_read(struct ast_channel *chan)
f.delivery.tv_sec = 0;
f.delivery.tv_usec = 0;
if (noaudiocapture) {
/* Return null frame to asterisk*/
ast_mutex_unlock(&alsalock);
return &f;
}
state = snd_pcm_state(alsa.icard);
if ((state != SND_PCM_STATE_PREPARED) && (state != SND_PCM_STATE_RUNNING)) {
snd_pcm_prepare(alsa.icard);
@ -470,6 +494,12 @@ static struct ast_frame *alsa_read(struct ast_channel *chan)
ast_mutex_unlock(&alsalock);
return &f;
}
if (mute) {
/* Don't transmit if muted */
ast_mutex_unlock(&alsalock);
return &f;
}
f.frametype = AST_FRAME_VOICE;
f.subclass.codec = AST_FORMAT_SLINEAR;
f.samples = FRAME_SIZE;
@ -667,6 +697,9 @@ static char *console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_arg
ast_cli(a->fd, "No one is calling us\n");
res = CLI_FAILURE;
} else {
if (mute) {
ast_verbose( " << Muted >> \n" );
}
hookstate = 1;
grab_owner();
if (alsa.owner) {
@ -675,8 +708,10 @@ static char *console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_arg
}
}
snd_pcm_prepare(alsa.icard);
snd_pcm_start(alsa.icard);
if (!noaudiocapture) {
snd_pcm_prepare(alsa.icard);
snd_pcm_start(alsa.icard);
}
ast_mutex_unlock(&alsalock);
@ -835,12 +870,57 @@ static char *console_dial(struct ast_cli_entry *e, int cmd, struct ast_cli_args
return res;
}
static char *console_mute(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
int toggle = 0;
char *res = CLI_SUCCESS;
switch (cmd) {
case CLI_INIT:
e->command = "console {mute|unmute} [toggle]";
e->usage =
"Usage: console {mute|unmute} [toggle]\n"
" Mute/unmute the microphone.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc > 3) {
return CLI_SHOWUSAGE;
}
if (a->argc == 3) {
if (strcasecmp(a->argv[2], "toggle"))
return CLI_SHOWUSAGE;
toggle = 1;
}
if (a->argc < 2) {
return CLI_SHOWUSAGE;
}
if (!strcasecmp(a->argv[1], "mute")) {
mute = toggle ? !mute : 1;
} else if (!strcasecmp(a->argv[1], "unmute")) {
mute = toggle ? !mute : 0;
} else {
return CLI_SHOWUSAGE;
}
ast_cli(a->fd, "Console mic is %s\n", mute ? "off" : "on");
return res;
}
static struct ast_cli_entry cli_alsa[] = {
AST_CLI_DEFINE(console_answer, "Answer an incoming console call"),
AST_CLI_DEFINE(console_hangup, "Hangup a call on the console"),
AST_CLI_DEFINE(console_dial, "Dial an extension on the console"),
AST_CLI_DEFINE(console_sendtext, "Send text to the remote device"),
AST_CLI_DEFINE(console_autoanswer, "Sets/displays autoanswer"),
AST_CLI_DEFINE(console_mute, "Disable/Enable mic input"),
};
static int load_module(void)
@ -865,27 +945,33 @@ static int load_module(void)
v = ast_variable_browse(cfg, "general");
for (; v; v = v->next) {
/* handle jb conf */
if (!ast_jb_read_conf(&global_jbconf, v->name, v->value))
continue;
if (!strcasecmp(v->name, "autoanswer"))
if (!ast_jb_read_conf(&global_jbconf, v->name, v->value)) {
continue;
}
if (!strcasecmp(v->name, "autoanswer")) {
autoanswer = ast_true(v->value);
else if (!strcasecmp(v->name, "silencesuppression"))
} else if (!strcasecmp(v->name, "mute")) {
mute = ast_true(v->value);
} else if (!strcasecmp(v->name, "noaudiocapture")) {
noaudiocapture = ast_true(v->value);
} else if (!strcasecmp(v->name, "silencesuppression")) {
silencesuppression = ast_true(v->value);
else if (!strcasecmp(v->name, "silencethreshold"))
} else if (!strcasecmp(v->name, "silencethreshold")) {
silencethreshold = atoi(v->value);
else if (!strcasecmp(v->name, "context"))
} else if (!strcasecmp(v->name, "context")) {
ast_copy_string(context, v->value, sizeof(context));
else if (!strcasecmp(v->name, "language"))
} else if (!strcasecmp(v->name, "language")) {
ast_copy_string(language, v->value, sizeof(language));
else if (!strcasecmp(v->name, "extension"))
} else if (!strcasecmp(v->name, "extension")) {
ast_copy_string(exten, v->value, sizeof(exten));
else if (!strcasecmp(v->name, "input_device"))
} else if (!strcasecmp(v->name, "input_device")) {
ast_copy_string(indevname, v->value, sizeof(indevname));
else if (!strcasecmp(v->name, "output_device"))
} else if (!strcasecmp(v->name, "output_device")) {
ast_copy_string(outdevname, v->value, sizeof(outdevname));
else if (!strcasecmp(v->name, "mohinterpret"))
} else if (!strcasecmp(v->name, "mohinterpret")) {
ast_copy_string(mohinterpret, v->value, sizeof(mohinterpret));
}
}
ast_config_destroy(cfg);