mirror of
https://github.com/asterisk/asterisk.git
synced 2025-10-17 10:11:53 +00:00
Remove as much trailing whitespace as possible.
Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0
This commit is contained in:
@@ -21,7 +21,7 @@
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* \brief Generic Linux Telephony Interface driver
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*
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* \author Mark Spencer <markster@digium.com>
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*
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*
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* \ingroup channel_drivers
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*/
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@@ -114,7 +114,7 @@ AST_MUTEX_DEFINE_STATIC(monlock);
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/* Boolean value whether the monitoring thread shall continue. */
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static unsigned int monitor;
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/* This is the thread for the monitor which checks for input on the channels
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which are not currently in use. */
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static pthread_t monitor_thread = AST_PTHREADT_NULL;
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@@ -123,7 +123,7 @@ static int restart_monitor(void);
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/* The private structures of the Phone Jack channels are linked for
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selecting outgoing channels */
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#define MODE_DIALTONE 1
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#define MODE_IMMEDIATE 2
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#define MODE_FXO 3
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@@ -336,7 +336,7 @@ static int phone_call(struct ast_channel *ast, const char *dest, int timeout)
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start = IXJ_PHONE_RING_START(cid);
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if (start == -1)
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return -1;
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if (p->mode == MODE_FXS) {
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const char *digit = strchr(dest, '/');
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if (digit)
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@@ -346,7 +346,7 @@ static int phone_call(struct ast_channel *ast, const char *dest, int timeout)
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phone_digit_end(ast, *digit++, 0);
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}
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}
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ast_setstate(ast, AST_STATE_RINGING);
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ast_queue_control(ast, AST_CONTROL_RINGING);
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return 0;
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@@ -449,7 +449,7 @@ static int phone_setup(struct ast_channel *ast)
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if (!p->lastinput || (ast_format_cmp(p->lastinput, ast_channel_rawreadformat(ast)) == AST_FORMAT_CMP_NOT_EQUAL)) {
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ao2_replace(p->lastinput, ast_channel_rawreadformat(ast));
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if (ioctl(p->fd, PHONE_REC_CODEC, ast_channel_rawreadformat(ast))) {
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ast_log(LOG_WARNING, "Failed to set codec to %s\n",
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ast_log(LOG_WARNING, "Failed to set codec to %s\n",
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ast_format_get_name(ast_channel_rawreadformat(ast)));
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return -1;
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}
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@@ -515,11 +515,11 @@ static struct ast_frame *phone_exception(struct ast_channel *ast)
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p->fr.offset = 0;
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p->fr.mallocd=0;
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p->fr.delivery = ast_tv(0,0);
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phonee.bytes = ioctl(p->fd, PHONE_EXCEPTION);
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if (phonee.bits.dtmf_ready) {
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ast_debug(1, "phone_exception(): DTMF\n");
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/* We've got a digit -- Just handle this nicely and easily */
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digit = ioctl(p->fd, PHONE_GET_DTMF_ASCII);
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p->fr.subclass.integer = digit;
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@@ -541,7 +541,7 @@ static struct ast_frame *phone_exception(struct ast_channel *ast)
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phone_setup(ast);
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ast_setstate(ast, AST_STATE_UP);
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return &p->fr;
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} else
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} else
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ast_log(LOG_WARNING, "Got off hook in weird state %u\n", ast_channel_state(ast));
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}
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}
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@@ -564,7 +564,7 @@ static struct ast_frame *phone_read(struct ast_channel *ast)
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{
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int res;
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struct phone_pvt *p = ast_channel_tech_pvt(ast);
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/* Some nice norms */
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p->fr.datalen = 0;
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@@ -645,7 +645,7 @@ static int phone_write_buf(struct phone_pvt *p, const char *buf, int len, int fr
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}
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p->obuflen -= frlen;
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/* Move memory if necessary */
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if (p->obuflen)
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if (p->obuflen)
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memmove(p->obuf, p->obuf + frlen, p->obuflen);
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}
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return len;
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@@ -654,7 +654,7 @@ static int phone_write_buf(struct phone_pvt *p, const char *buf, int len, int fr
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static int phone_send_text(struct ast_channel *ast, const char *text)
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{
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int length = strlen(text);
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return phone_write_buf(ast_channel_tech_pvt(ast), text, length, length, 0) ==
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return phone_write_buf(ast_channel_tech_pvt(ast), text, length, length, 0) ==
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length ? 0 : -1;
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}
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@@ -685,7 +685,7 @@ static int phone_write(struct ast_channel *ast, struct ast_frame *frame)
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/* Don't try tos end audio on-hook */
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return 0;
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}
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#endif
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#endif
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if (ast_format_cmp(frame->subclass.format, ast_format_g729) == AST_FORMAT_CMP_EQUAL) {
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if (!p->lastformat || (ast_format_cmp(p->lastformat, ast_format_g729) != AST_FORMAT_CMP_EQUAL)) {
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ioctl(p->fd, PHONE_PLAY_STOP);
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@@ -812,7 +812,7 @@ static int phone_write(struct ast_channel *ast, struct ast_frame *frame)
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expected = frame->datalen - sofar;
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if (maxfr < expected)
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expected = maxfr;
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/* XXX Internet Phone Jack does not handle the 4-byte VAD frame properly! XXX
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/* XXX Internet Phone Jack does not handle the 4-byte VAD frame properly! XXX
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we have to pad it to 24 bytes still. */
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if (frame->datalen == 4) {
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if (p->silencesupression) {
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@@ -832,7 +832,7 @@ static int phone_write(struct ast_channel *ast, struct ast_frame *frame)
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}
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if (res != expected) {
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if ((errno != EAGAIN) && (errno != EINTR)) {
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if (res < 0)
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if (res < 0)
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ast_log(LOG_WARNING, "Write returned error (%s)\n", strerror(errno));
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/*
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* Card is in non-blocking mode now and it works well now, but there are
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@@ -1024,8 +1024,8 @@ static void phone_check_exception(struct phone_pvt *i)
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}
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if (phonee.bits.caller_id)
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ast_verbose("We have caller ID\n");
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}
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static void *do_monitor(void *data)
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@@ -1191,7 +1191,7 @@ static struct phone_pvt *mkif(const char *iface, int mode, int txgain, int rxgai
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{
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/* Make a phone_pvt structure for this interface */
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struct phone_pvt *tmp;
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tmp = ast_calloc(1, sizeof(*tmp));
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if (tmp) {
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tmp->fd = open(iface, O_RDWR);
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@@ -1205,7 +1205,7 @@ static struct phone_pvt *mkif(const char *iface, int mode, int txgain, int rxgai
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ast_debug(1, "Unable to set port to PSTN\n");
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}
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} else {
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if (ioctl(tmp->fd, IXJCTL_PORT, PORT_POTS))
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if (ioctl(tmp->fd, IXJCTL_PORT, PORT_POTS))
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if (mode != MODE_FXS)
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ast_debug(1, "Unable to set port to POTS\n");
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}
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@@ -1217,7 +1217,7 @@ static struct phone_pvt *mkif(const char *iface, int mode, int txgain, int rxgai
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ast_debug(1, "ioctl(PHONE_PSTN_SET_STATE) failed on %s (%s)\n",iface, strerror(errno));
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if (echocancel != AEC_OFF)
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ioctl(tmp->fd, IXJCTL_AEC_START, echocancel);
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if (silencesupression)
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if (silencesupression)
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tmp->silencesupression = 1;
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#ifdef PHONE_VAD
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ioctl(tmp->fd, PHONE_VAD, tmp->silencesupression);
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@@ -1300,7 +1300,7 @@ static int parse_gain_value(const char *gain_type, const char *value)
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return DEFAULT_GAIN;
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}
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/* multiplicate gain by 1.0 gain value */
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/* multiplicate gain by 1.0 gain value */
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gain = gain * (float)DEFAULT_GAIN;
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/* percentage? */
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@@ -1424,7 +1424,7 @@ static int load_module(void)
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if (tmp) {
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tmp->next = iflist;
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iflist = tmp;
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} else {
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ast_log(LOG_ERROR, "Unable to register channel '%s'\n", v->value);
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ast_config_destroy(cfg);
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@@ -1439,7 +1439,7 @@ static int load_module(void)
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} else if (!strcasecmp(v->name, "callerid")) {
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ast_callerid_split(v->value, cid_name, sizeof(cid_name), cid_num, sizeof(cid_num));
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} else if (!strcasecmp(v->name, "mode")) {
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if (!strncasecmp(v->value, "di", 2))
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if (!strncasecmp(v->value, "di", 2))
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mode = MODE_DIALTONE;
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else if (!strncasecmp(v->value, "sig", 3))
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mode = MODE_SIGMA;
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@@ -1483,13 +1483,13 @@ static int load_module(void)
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echocancel = AEC_MED;
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} else if (!strcasecmp(v->value, "high")) {
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echocancel = AEC_HIGH;
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} else
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} else
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ast_log(LOG_WARNING, "Unknown echo cancellation '%s'\n", v->value);
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} else if (!strcasecmp(v->name, "txgain")) {
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txgain = parse_gain_value(v->name, v->value);
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} else if (!strcasecmp(v->name, "rxgain")) {
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rxgain = parse_gain_value(v->name, v->value);
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}
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}
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v = v->next;
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}
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ast_mutex_unlock(&iflock);
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@@ -1515,4 +1515,3 @@ static int load_module(void)
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}
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AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Linux Telephony API Support");
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