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

diff --git a/channels/chan_alsa.c b/channels/chan_alsa.c
index ca6c07897624063577f2ba5a7a2de321d4098116..b52a03c08a0a35ecd61d77c0020043a5c6e1b571 100644 (file)
--- 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);