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: https://origsvn.digium.com/svn/asterisk/trunk@229753 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 109 insertions, 23 deletions

diff --git a/channels/chan_alsa.c b/channels/chan_alsa.c
index ca6c07897..b52a03c08 100644
--- a/channels/chan_alsa.c
+++ b/channels/chan_alsa.c
@@ -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);