/*! \internal \brief Disable talk detection on the channel */
static int remove_talk_detect(struct ast_channel *chan)
{
struct ast_datastore *datastore = NULL;
struct talk_detect_params *td_params;
SCOPED_CHANNELLOCK(chan_lock, chan);
datastore = ast_channel_datastore_find(chan, &talk_detect_datastore, NULL);
if (!datastore) {
ast_log(AST_LOG_WARNING, "Cannot remove TALK_DETECT from %s: TALK_DETECT not currently enabled\n",
ast_channel_name(chan));
return -1;
}
td_params = datastore->data;
if (ast_audiohook_remove(chan, &td_params->audiohook)) {
ast_log(AST_LOG_WARNING, "Failed to remove TALK_DETECT audiohook from channel %s\n",
ast_channel_name(chan));
return -1;
}
if (ast_channel_datastore_remove(chan, datastore)) {
ast_log(AST_LOG_WARNING, "Failed to remove TALK_DETECT datastore from channel %s\n",
ast_channel_name(chan));
return -1;
}
ast_datastore_free(datastore);
return 0;
}
static int speex_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
struct ast_datastore *datastore = NULL;
struct speex_info *si = NULL;
struct speex_direction_info **sdi = NULL;
int is_new = 0;
ast_channel_lock(chan);
if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
ast_channel_unlock(chan);
if (!(datastore = ast_datastore_alloc(&speex_datastore, NULL))) {
return 0;
}
if (!(si = ast_calloc(1, sizeof(*si)))) {
ast_datastore_free(datastore);
return 0;
}
ast_audiohook_init(&si->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "speex", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
si->audiohook.manipulate_callback = speex_callback;
si->lastrate = 8000;
is_new = 1;
} else {
ast_channel_unlock(chan);
si = datastore->data;
}
if (!strcasecmp(data, "rx")) {
sdi = &si->rx;
} else if (!strcasecmp(data, "tx")) {
sdi = &si->tx;
} else {
ast_log(LOG_ERROR, "Invalid argument provided to the %s function\n", cmd);
if (is_new) {
ast_datastore_free(datastore);
return -1;
}
}
if (!*sdi) {
if (!(*sdi = ast_calloc(1, sizeof(**sdi)))) {
return 0;
}
/* Right now, the audiohooks API will _only_ provide us 8 kHz slinear
* audio. When it supports 16 kHz (or any other sample rates, we will
* have to take that into account here. */
(*sdi)->samples = -1;
}
if (!strcasecmp(cmd, "agc")) {
if (!sscanf(value, "%30f", &(*sdi)->agclevel))
(*sdi)->agclevel = ast_true(value) ? DEFAULT_AGC_LEVEL : 0.0;
if ((*sdi)->agclevel > 32768.0) {
ast_log(LOG_WARNING, "AGC(%s)=%.01f is greater than 32768... setting to 32768 instead\n",
((*sdi == si->rx) ? "rx" : "tx"), (*sdi)->agclevel);
(*sdi)->agclevel = 32768.0;
}
(*sdi)->agc = !!((*sdi)->agclevel);
if ((*sdi)->state) {
speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC, &(*sdi)->agc);
if ((*sdi)->agc) {
speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &(*sdi)->agclevel);
}
}
} else if (!strcasecmp(cmd, "denoise")) {
(*sdi)->denoise = (ast_true(value) != 0);
if ((*sdi)->state) {
speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_DENOISE, &(*sdi)->denoise);
}
}
if (!(*sdi)->agc && !(*sdi)->denoise) {
if ((*sdi)->state)
speex_preprocess_state_destroy((*sdi)->state);
ast_free(*sdi);
*sdi = NULL;
}
if (!si->rx && !si->tx) {
if (is_new) {
is_new = 0;
} else {
ast_channel_lock(chan);
ast_channel_datastore_remove(chan, datastore);
ast_channel_unlock(chan);
ast_audiohook_remove(chan, &si->audiohook);
ast_audiohook_detach(&si->audiohook);
}
ast_datastore_free(datastore);
}
if (is_new) {
datastore->data = si;
ast_channel_lock(chan);
ast_channel_datastore_add(chan, datastore);
ast_channel_unlock(chan);
ast_audiohook_attach(chan, &si->audiohook);
}
return 0;
}
