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-rw-r--r--apps/app_agent_pool.c4
-rw-r--r--apps/app_alarmreceiver.c13
-rw-r--r--apps/app_amd.c20
-rw-r--r--apps/app_chanspy.c38
-rw-r--r--apps/app_confbridge.c21
-rw-r--r--apps/app_dictate.c14
-rw-r--r--apps/app_dumpchan.c12
-rw-r--r--apps/app_echo.c5
-rw-r--r--apps/app_fax.c49
-rw-r--r--apps/app_festival.c26
-rw-r--r--apps/app_ices.c14
-rw-r--r--apps/app_jack.c29
-rw-r--r--apps/app_meetme.c43
-rw-r--r--apps/app_milliwatt.c15
-rw-r--r--apps/app_mixmonitor.c10
-rw-r--r--apps/app_mp3.c31
-rw-r--r--apps/app_nbscat.c32
-rw-r--r--apps/app_originate.c25
-rw-r--r--apps/app_record.c13
-rw-r--r--apps/app_sms.c17
-rw-r--r--apps/app_speech_utils.c9
-rw-r--r--apps/app_talkdetect.c16
-rw-r--r--apps/app_test.c17
-rw-r--r--apps/app_voicemail.c29
-rw-r--r--apps/app_waitforsilence.c11
-rw-r--r--apps/confbridge/conf_chan_record.c21
26 files changed, 282 insertions, 252 deletions
diff --git a/apps/app_agent_pool.c b/apps/app_agent_pool.c
index 5095001cf..f531c1613 100644
--- a/apps/app_agent_pool.c
+++ b/apps/app_agent_pool.c
@@ -2136,8 +2136,8 @@ static int agent_login_exec(struct ast_channel *chan, const char *data)
}
ast_verb(2, "Agent '%s' logged in (format %s/%s)\n", agent->username,
- ast_getformatname(ast_channel_readformat(chan)),
- ast_getformatname(ast_channel_writeformat(chan)));
+ ast_format_get_name(ast_channel_readformat(chan)),
+ ast_format_get_name(ast_channel_writeformat(chan)));
ast_channel_lock(chan);
send_agent_login(chan, agent->username);
ast_channel_unlock(chan);
diff --git a/apps/app_alarmreceiver.c b/apps/app_alarmreceiver.c
index 9063e1a60..4f9de364d 100644
--- a/apps/app_alarmreceiver.c
+++ b/apps/app_alarmreceiver.c
@@ -65,6 +65,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/astdb.h"
#include "asterisk/utils.h"
#include "asterisk/indications.h"
+#include "asterisk/format_cache.h"
#define ALMRCV_CONFIG "alarmreceiver.conf"
#define UNKNOWN_FORMAT "UNKNOWN_FORMAT"
@@ -798,19 +799,19 @@ static int alarmreceiver_exec(struct ast_channel *chan, const char *data)
char signalling_type[64] = "";
event_node_t *event_head = NULL;
- if (ast_channel_writeformat(chan)->id != AST_FORMAT_ALAW
- && ast_channel_writeformat(chan)->id != AST_FORMAT_ULAW) {
+ if ((ast_format_cmp(ast_channel_writeformat(chan), ast_format_ulaw) == AST_FORMAT_CMP_NOT_EQUAL) &&
+ (ast_format_cmp(ast_channel_writeformat(chan), ast_format_alaw) == AST_FORMAT_CMP_NOT_EQUAL)) {
ast_verb(4, "AlarmReceiver: Setting write format to Mu-law\n");
- if (ast_set_write_format_by_id(chan,AST_FORMAT_ULAW)) {
+ if (ast_set_write_format(chan, ast_format_ulaw)) {
ast_log(LOG_WARNING, "AlarmReceiver: Unable to set write format to Mu-law on %s\n",ast_channel_name(chan));
return -1;
}
}
- if (ast_channel_readformat(chan)->id != AST_FORMAT_ALAW
- && ast_channel_readformat(chan)->id != AST_FORMAT_ULAW) {
+ if ((ast_format_cmp(ast_channel_readformat(chan), ast_format_ulaw) == AST_FORMAT_CMP_NOT_EQUAL) &&
+ (ast_format_cmp(ast_channel_readformat(chan), ast_format_alaw) == AST_FORMAT_CMP_NOT_EQUAL)) {
ast_verb(4, "AlarmReceiver: Setting read format to Mu-law\n");
- if (ast_set_read_format_by_id(chan,AST_FORMAT_ULAW)) {
+ if (ast_set_read_format(chan, ast_format_ulaw)) {
ast_log(LOG_WARNING, "AlarmReceiver: Unable to set read format to Mu-law on %s\n",ast_channel_name(chan));
return -1;
}
diff --git a/apps/app_amd.c b/apps/app_amd.c
index d8f26b49b..3dd8cc617 100644
--- a/apps/app_amd.c
+++ b/apps/app_amd.c
@@ -52,6 +52,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/config.h"
#include "asterisk/app.h"
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<application name="AMD" language="en_US">
@@ -163,7 +164,7 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
struct ast_frame *f = NULL;
struct ast_dsp *silenceDetector = NULL;
int dspsilence = 0, framelength = 0;
- struct ast_format readFormat;
+ RAII_VAR(struct ast_format *, readFormat, NULL, ao2_cleanup);
int inInitialSilence = 1;
int inGreeting = 0;
int voiceDuration = 0;
@@ -202,11 +203,10 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
AST_APP_ARG(argMaximumWordLength);
);
- ast_format_clear(&readFormat);
ast_verb(3, "AMD: %s %s %s (Fmt: %s)\n", ast_channel_name(chan),
S_COR(ast_channel_caller(chan)->ani.number.valid, ast_channel_caller(chan)->ani.number.str, "(N/A)"),
S_COR(ast_channel_redirecting(chan)->from.number.valid, ast_channel_redirecting(chan)->from.number.str, "(N/A)"),
- ast_getformatname(ast_channel_readformat(chan)));
+ ast_format_get_name(ast_channel_readformat(chan)));
/* Lets parse the arguments. */
if (!ast_strlen_zero(parse)) {
@@ -255,8 +255,8 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
minimumWordLength, betweenWordsSilence, maximumNumberOfWords, silenceThreshold, maximumWordLength);
/* Set read format to signed linear so we get signed linear frames in */
- ast_format_copy(&readFormat, ast_channel_readformat(chan));
- if (ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR) < 0 ) {
+ readFormat = ao2_bump(ast_channel_readformat(chan));
+ if (ast_set_read_format(chan, ast_format_slin) < 0 ) {
ast_log(LOG_WARNING, "AMD: Channel [%s]. Unable to set to linear mode, giving up\n", ast_channel_name(chan));
pbx_builtin_setvar_helper(chan , "AMDSTATUS", "");
pbx_builtin_setvar_helper(chan , "AMDCAUSE", "");
@@ -289,7 +289,7 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_NULL || f->frametype == AST_FRAME_CNG) {
/* If the total time exceeds the analysis time then give up as we are not too sure */
if (f->frametype == AST_FRAME_VOICE) {
- framelength = (ast_codec_get_samples(f) / DEFAULT_SAMPLES_PER_MS);
+ framelength = (ast_codec_samples_count(f) / DEFAULT_SAMPLES_PER_MS);
} else {
framelength = 2 * maxWaitTimeForFrame;
}
@@ -412,7 +412,7 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
pbx_builtin_setvar_helper(chan , "AMDCAUSE" , amdCause);
/* Restore channel read format */
- if (readFormat.id && ast_set_read_format(chan, &readFormat))
+ if (readFormat && ast_set_read_format(chan, readFormat))
ast_log(LOG_WARNING, "AMD: Unable to restore read format on '%s'\n", ast_channel_name(chan));
/* Free the DSP used to detect silence */
@@ -510,10 +510,10 @@ static int unload_module(void)
*/
static int load_module(void)
{
- if (load_config(0))
+ if (load_config(0) || ast_register_application_xml(app, amd_exec)) {
return AST_MODULE_LOAD_DECLINE;
- if (ast_register_application_xml(app, amd_exec))
- return AST_MODULE_LOAD_FAILURE;
+ }
+
return AST_MODULE_LOAD_SUCCESS;
}
diff --git a/apps/app_chanspy.c b/apps/app_chanspy.c
index 47f755e5f..af69be303 100644
--- a/apps/app_chanspy.c
+++ b/apps/app_chanspy.c
@@ -57,6 +57,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/autochan.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/json.h"
+#include "asterisk/format_cache.h"
#define AST_NAME_STRLEN 256
#define NUM_SPYGROUPS 128
@@ -451,9 +452,6 @@ static int spy_generate(struct ast_channel *chan, void *data, int len, int sampl
{
struct chanspy_translation_helper *csth = data;
struct ast_frame *f, *cur;
- struct ast_format format_slin;
-
- ast_format_set(&format_slin, AST_FORMAT_SLINEAR, 0);
ast_audiohook_lock(&csth->spy_audiohook);
if (csth->spy_audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
@@ -464,9 +462,9 @@ static int spy_generate(struct ast_channel *chan, void *data, int len, int sampl
if (ast_test_flag(&csth->flags, OPTION_READONLY)) {
/* Option 'o' was set, so don't mix channel audio */
- f = ast_audiohook_read_frame(&csth->spy_audiohook, samples, AST_AUDIOHOOK_DIRECTION_READ, &format_slin);
+ f = ast_audiohook_read_frame(&csth->spy_audiohook, samples, AST_AUDIOHOOK_DIRECTION_READ, ast_format_slin);
} else {
- f = ast_audiohook_read_frame(&csth->spy_audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, &format_slin);
+ f = ast_audiohook_read_frame(&csth->spy_audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, ast_format_slin);
}
ast_audiohook_unlock(&csth->spy_audiohook);
@@ -1181,7 +1179,7 @@ static int chanspy_exec(struct ast_channel *chan, const char *data)
.volume = '#',
.exit = '\0',
};
- struct ast_format oldwf;
+ RAII_VAR(struct ast_format *, oldwf, NULL, ao2_cleanup);
int volfactor = 0;
int res;
char *mailbox = NULL;
@@ -1194,7 +1192,6 @@ static int chanspy_exec(struct ast_channel *chan, const char *data)
char *parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
- ast_format_clear(&oldwf);
if (args.spec && !strcmp(args.spec, "all"))
args.spec = NULL;
@@ -1258,8 +1255,8 @@ static int chanspy_exec(struct ast_channel *chan, const char *data)
ast_clear_flag(&flags, AST_FLAGS_ALL);
}
- ast_format_copy(&oldwf, ast_channel_writeformat(chan));
- if (ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR) < 0) {
+ oldwf = ao2_bump(ast_channel_writeformat(chan));
+ if (ast_set_write_format(chan, ast_format_slin) < 0) {
ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
return -1;
}
@@ -1279,7 +1276,7 @@ static int chanspy_exec(struct ast_channel *chan, const char *data)
if (fd)
close(fd);
- if (oldwf.id && ast_set_write_format(chan, &oldwf) < 0)
+ if (oldwf && ast_set_write_format(chan, oldwf) < 0)
ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
if (ast_test_flag(&flags, OPTION_EXITONHANGUP)) {
@@ -1301,7 +1298,7 @@ static int extenspy_exec(struct ast_channel *chan, const char *data)
.volume = '#',
.exit = '\0',
};
- struct ast_format oldwf;
+ RAII_VAR(struct ast_format *, oldwf, NULL, ao2_cleanup);
int volfactor = 0;
int res;
char *mailbox = NULL;
@@ -1313,7 +1310,6 @@ static int extenspy_exec(struct ast_channel *chan, const char *data)
char *parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
- ast_format_clear(&oldwf);
if (!ast_strlen_zero(args.context) && (ptr = strchr(args.context, '@'))) {
exten = args.context;
@@ -1383,8 +1379,8 @@ static int extenspy_exec(struct ast_channel *chan, const char *data)
ast_clear_flag(&flags, AST_FLAGS_ALL);
}
- ast_format_copy(&oldwf, ast_channel_writeformat(chan));
- if (ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR) < 0) {
+ oldwf = ao2_bump(ast_channel_writeformat(chan));
+ if (ast_set_write_format(chan, ast_format_slin) < 0) {
ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
return -1;
}
@@ -1405,7 +1401,7 @@ static int extenspy_exec(struct ast_channel *chan, const char *data)
if (fd)
close(fd);
- if (oldwf.id && ast_set_write_format(chan, &oldwf) < 0)
+ if (oldwf && ast_set_write_format(chan, oldwf) < 0)
ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
return res;
@@ -1420,13 +1416,13 @@ static int dahdiscan_exec(struct ast_channel *chan, const char *data)
.volume = '\0',
.exit = '*',
};
- struct ast_format oldwf;
+ struct ast_format *oldwf;
int res;
char *mygroup = NULL;
/* Coverity - This uninit_use should be ignored since this macro initializes the flags */
ast_clear_flag(&flags, AST_FLAGS_ALL);
- ast_format_clear(&oldwf);
+
if (!ast_strlen_zero(data)) {
mygroup = ast_strdupa(data);
}
@@ -1434,16 +1430,18 @@ static int dahdiscan_exec(struct ast_channel *chan, const char *data)
ast_set_flag(&flags, OPTION_DTMF_CYCLE);
ast_set_flag(&flags, OPTION_DAHDI_SCAN);
- ast_format_copy(&oldwf, ast_channel_writeformat(chan));
- if (ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR) < 0) {
+ oldwf = ao2_bump(ast_channel_writeformat(chan));
+ if (ast_set_write_format(chan, ast_format_slin) < 0) {
ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
+ ao2_cleanup(oldwf);
return -1;
}
res = common_exec(chan, &flags, 0, 0, &user_options, mygroup, NULL, spec, NULL, NULL, NULL, NULL);
- if (oldwf.id && ast_set_write_format(chan, &oldwf) < 0)
+ if (oldwf && ast_set_write_format(chan, oldwf) < 0)
ast_log(LOG_ERROR, "Could Not Set Write Format.\n");
+ ao2_cleanup(oldwf);
return res;
}
diff --git a/apps/app_confbridge.c b/apps/app_confbridge.c
index 9a704ac93..445b196d4 100644
--- a/apps/app_confbridge.c
+++ b/apps/app_confbridge.c
@@ -70,6 +70,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/stasis.h"
#include "asterisk/stasis_bridges.h"
#include "asterisk/json.h"
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<application name="ConfBridge" language="en_US">
@@ -676,7 +677,6 @@ static int conf_stop_record_thread(struct confbridge_conference *conference)
static int conf_start_record(struct confbridge_conference *conference)
{
struct ast_format_cap *cap;
- struct ast_format format;
if (conference->record_state != CONF_RECORD_STOP) {
return -1;
@@ -687,16 +687,16 @@ static int conf_start_record(struct confbridge_conference *conference)
return -1;
}
- cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
+ cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!cap) {
return -1;
}
- ast_format_cap_add(cap, ast_format_set(&format, AST_FORMAT_SLINEAR, 0));
+ ast_format_cap_append(cap, ast_format_slin, 0);
conference->record_chan = ast_request("CBRec", cap, NULL, NULL,
conference->name, NULL);
- cap = ast_format_cap_destroy(cap);
+ ao2_ref(cap, -1);
if (!conference->record_chan) {
return -1;
}
@@ -1374,16 +1374,15 @@ static void leave_conference(struct confbridge_user *user)
static int alloc_playback_chan(struct confbridge_conference *conference)
{
struct ast_format_cap *cap;
- struct ast_format format;
- cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
+ cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!cap) {
return -1;
}
- ast_format_cap_add(cap, ast_format_set(&format, AST_FORMAT_SLINEAR, 0));
+ ast_format_cap_append(cap, ast_format_slin, 0);
conference->playback_chan = ast_request("CBAnn", cap, NULL, NULL,
conference->name, NULL);
- cap = ast_format_cap_destroy(cap);
+ ao2_ref(cap, -1);
if (!conference->playback_chan) {
return -1;
}
@@ -3238,7 +3237,7 @@ void conf_remove_user_waiting(struct confbridge_conference *conference, struct c
static void unregister_channel_tech(struct ast_channel_tech *tech)
{
ast_channel_unregister(tech);
- tech->capabilities = ast_format_cap_destroy(tech->capabilities);
+ ao2_cleanup(tech->capabilities);
}
/*!
@@ -3253,11 +3252,11 @@ static void unregister_channel_tech(struct ast_channel_tech *tech)
*/
static int register_channel_tech(struct ast_channel_tech *tech)
{
- tech->capabilities = ast_format_cap_alloc(0);
+ tech->capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!tech->capabilities) {
return -1;
}
- ast_format_cap_add_all(tech->capabilities);
+ ast_format_cap_append_by_type(tech->capabilities, AST_MEDIA_TYPE_UNKNOWN);
if (ast_channel_register(tech)) {
ast_log(LOG_ERROR, "Unable to register channel technology %s(%s).\n",
tech->type, tech->description);
diff --git a/apps/app_dictate.c b/apps/app_dictate.c
index 1820a1f83..8f229f2a9 100644
--- a/apps/app_dictate.c
+++ b/apps/app_dictate.c
@@ -43,6 +43,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/module.h"
#include "asterisk/say.h"
#include "asterisk/app.h"
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<application name="Dictate" language="en_US">
@@ -108,8 +109,7 @@ static int dictate_exec(struct ast_channel *chan, const char *data)
len = 0,
maxlen = 0,
mode = 0;
- struct ast_format oldr;
- ast_format_clear(&oldr);
+ struct ast_format *oldr;
snprintf(dftbase, sizeof(dftbase), "%s/dictate", ast_config_AST_SPOOL_DIR);
if (!ast_strlen_zero(data)) {
@@ -126,9 +126,10 @@ static int dictate_exec(struct ast_channel *chan, const char *data)
if (args.argc > 1 && args.filename) {
filename = args.filename;
}
- ast_format_copy(&oldr, ast_channel_readformat(chan));
- if ((res = ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR)) < 0) {
+ oldr = ao2_bump(ast_channel_readformat(chan));
+ if ((res = ast_set_read_format(chan, ast_format_slin)) < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode.\n");
+ ao2_cleanup(oldr);
return -1;
}
@@ -335,8 +336,9 @@ static int dictate_exec(struct ast_channel *chan, const char *data)
ast_frfree(f);
}
}
- if (oldr.id) {
- ast_set_read_format(chan, &oldr);
+ if (oldr) {
+ ast_set_read_format(chan, oldr);
+ ao2_ref(oldr, -1);
}
return 0;
}
diff --git a/apps/app_dumpchan.c b/apps/app_dumpchan.c
index 5c5be18af..1794c2565 100644
--- a/apps/app_dumpchan.c
+++ b/apps/app_dumpchan.c
@@ -72,7 +72,7 @@ static int serialize_showchan(struct ast_channel *c, char *buf, size_t size)
{
long elapsed_seconds = 0;
int hour = 0, min = 0, sec = 0;
- char nf[256];
+ struct ast_str *format_buf = ast_str_alloca(64);
char cgrp[256];
char pgrp[256];
struct ast_str *write_transpath = ast_str_alloca(256);
@@ -143,11 +143,11 @@ static int serialize_showchan(struct ast_channel *c, char *buf, size_t size)
ast_state2str(ast_channel_state(c)),
ast_channel_state(c),
ast_channel_rings(c),
- ast_getformatname_multiple(nf, sizeof(nf), ast_channel_nativeformats(c)),
- ast_getformatname(ast_channel_writeformat(c)),
- ast_getformatname(ast_channel_readformat(c)),
- ast_getformatname(ast_channel_rawwriteformat(c)),
- ast_getformatname(ast_channel_rawreadformat(c)),
+ ast_format_cap_get_names(ast_channel_nativeformats(c), &format_buf),
+ ast_format_get_name(ast_channel_writeformat(c)),
+ ast_format_get_name(ast_channel_readformat(c)),
+ ast_format_get_name(ast_channel_rawwriteformat(c)),
+ ast_format_get_name(ast_channel_rawreadformat(c)),
ast_channel_writetrans(c) ? "Yes" : "No",
ast_translate_path_to_str(ast_channel_writetrans(c), &write_transpath),
ast_channel_readtrans(c) ? "Yes" : "No",
diff --git a/apps/app_echo.c b/apps/app_echo.c
index df5a914f1..d8b207967 100644
--- a/apps/app_echo.c
+++ b/apps/app_echo.c
@@ -58,11 +58,6 @@ static const char app[] = "Echo";
static int echo_exec(struct ast_channel *chan, const char *data)
{
int res = -1;
- struct ast_format format;
-
- ast_best_codec(ast_channel_nativeformats(chan), &format);
- ast_set_write_format(chan, &format);
- ast_set_read_format(chan, &format);
while (ast_waitfor(chan, -1) > -1) {
struct ast_frame *f = ast_read(chan);
diff --git a/apps/app_fax.c b/apps/app_fax.c
index e23ac431e..ceff38fef 100644
--- a/apps/app_fax.c
+++ b/apps/app_fax.c
@@ -45,6 +45,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/module.h"
#include "asterisk/stasis.h"
#include "asterisk/stasis_channels.h"
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<application name="SendFAX" language="en_US" module="app_fax">
@@ -332,9 +333,9 @@ static int fax_generator_generate(struct ast_channel *chan, void *data, int len,
struct ast_frame outf = {
.frametype = AST_FRAME_VOICE,
+ .subclass.format = ast_format_slin,
.src = __FUNCTION__,
};
- ast_format_set(&outf.subclass.format, AST_FORMAT_SLINEAR, 0);
if (samples > MAX_SAMPLES) {
ast_log(LOG_WARNING, "Only generating %d samples, where %d requested\n", MAX_SAMPLES, samples);
@@ -365,8 +366,8 @@ static struct ast_generator generator = {
static int transmit_audio(fax_session *s)
{
int res = -1;
- struct ast_format original_read_fmt;
- struct ast_format original_write_fmt;
+ struct ast_format *original_read_fmt;
+ struct ast_format *original_write_fmt = NULL;
fax_state_t fax;
t30_state_t *t30state;
struct ast_frame *inf = NULL;
@@ -386,9 +387,6 @@ static int transmit_audio(fax_session *s)
*/
};
- ast_format_clear(&original_read_fmt);
- ast_format_clear(&original_write_fmt);
-
/* if in called party mode, try to use T.38 */
if (s->caller_mode == FALSE) {
/* check if we are already in T.38 mode (unlikely), or if we can request
@@ -461,22 +459,18 @@ static int transmit_audio(fax_session *s)
t30state = &fax.t30_state;
#endif
- ast_format_copy(&original_read_fmt, ast_channel_readformat(s->chan));
- if (original_read_fmt.id != AST_FORMAT_SLINEAR) {
- res = ast_set_read_format_by_id(s->chan, AST_FORMAT_SLINEAR);
- if (res < 0) {
- ast_log(LOG_WARNING, "Unable to set to linear read mode, giving up\n");
- goto done;
- }
+ original_read_fmt = ao2_bump(ast_channel_readformat(s->chan));
+ res = ast_set_read_format(s->chan, ast_format_slin);
+ if (res < 0) {
+ ast_log(LOG_WARNING, "Unable to set to linear read mode, giving up\n");
+ goto done;
}
- ast_format_copy(&original_write_fmt, ast_channel_writeformat(s->chan));
- if (original_write_fmt.id != AST_FORMAT_SLINEAR) {
- res = ast_set_write_format_by_id(s->chan, AST_FORMAT_SLINEAR);
- if (res < 0) {
- ast_log(LOG_WARNING, "Unable to set to linear write mode, giving up\n");
- goto done;
- }
+ original_write_fmt = ao2_bump(ast_channel_writeformat(s->chan));
+ res = ast_set_write_format(s->chan, ast_format_slin);
+ if (res < 0) {
+ ast_log(LOG_WARNING, "Unable to set to linear write mode, giving up\n");
+ goto done;
}
/* Initialize T30 terminal */
@@ -529,12 +523,13 @@ static int transmit_audio(fax_session *s)
break;
}
- ast_debug(10, "frame %d/%u, len=%d\n", inf->frametype, (unsigned int) inf->subclass.format.id, inf->datalen);
+ ast_debug(10, "frame %d/%s, len=%d\n", inf->frametype, ast_format_get_name(inf->subclass.format), inf->datalen);
/* Check the frame type. Format also must be checked because there is a chance
that a frame in old format was already queued before we set channel format
to slinear so it will still be received by ast_read */
- if (inf->frametype == AST_FRAME_VOICE && inf->subclass.format.id == AST_FORMAT_SLINEAR) {
+ if (inf->frametype == AST_FRAME_VOICE &&
+ (ast_format_cmp(inf->subclass.format, ast_format_slin) == AST_FORMAT_CMP_EQUAL)) {
if (fax_rx(&fax, inf->data.ptr, inf->samples) < 0) {
/* I know fax_rx never returns errors. The check here is for good style only */
ast_log(LOG_WARNING, "fax_rx returned error\n");
@@ -588,14 +583,16 @@ static int transmit_audio(fax_session *s)
fax_release(&fax);
done:
- if (original_write_fmt.id != AST_FORMAT_SLINEAR) {
- if (ast_set_write_format(s->chan, &original_write_fmt) < 0)
+ if (original_write_fmt) {
+ if (ast_set_write_format(s->chan, original_write_fmt) < 0)
ast_log(LOG_WARNING, "Unable to restore write format on '%s'\n", ast_channel_name(s->chan));
+ ao2_ref(original_write_fmt, -1);
}
- if (original_read_fmt.id != AST_FORMAT_SLINEAR) {
- if (ast_set_read_format(s->chan, &original_read_fmt) < 0)
+ if (original_read_fmt) {
+ if (ast_set_read_format(s->chan, original_read_fmt) < 0)
ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", ast_channel_name(s->chan));
+ ao2_ref(original_read_fmt, -1);
}
return res;
diff --git a/apps/app_festival.c b/apps/app_festival.c
index 3ccacb195..3626563c6 100644
--- a/apps/app_festival.c
+++ b/apps/app_festival.c
@@ -63,6 +63,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/app.h"
#include "asterisk/endian.h"
+#include "asterisk/format_cache.h"
#define FESTIVAL_CONFIG "festival.conf"
#define MAXLEN 180
@@ -177,7 +178,7 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
int res = 0;
int fds[2];
int needed = 0;
- struct ast_format owriteformat;
+ struct ast_format *owriteformat;
struct ast_frame *f;
struct myframe {
struct ast_frame f;
@@ -187,7 +188,6 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
.f = { 0, },
};
- ast_format_clear(&owriteformat);
if (pipe(fds)) {
ast_log(LOG_WARNING, "Unable to create pipe\n");
return -1;
@@ -199,12 +199,19 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
ast_stopstream(chan);
ast_indicate(chan, -1);
- ast_format_copy(&owriteformat, ast_channel_writeformat(chan));
- res = ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR);
+ owriteformat = ao2_bump(ast_channel_writeformat(chan));
+ res = ast_set_write_format(chan, ast_format_slin);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
+ ao2_cleanup(owriteformat);
return -1;
}
+
+ myf.f.frametype = AST_FRAME_VOICE;
+ myf.f.subclass.format = ast_format_slin;
+ myf.f.offset = AST_FRIENDLY_OFFSET;
+ myf.f.src = __PRETTY_FUNCTION__;
+ myf.f.data.ptr = myf.frdata;
res = send_waveform_to_fd(waveform, length, fds[1]);
if (res >= 0) {
@@ -240,13 +247,8 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
}
res = read(fds[0], myf.frdata, needed);
if (res > 0) {
- myf.f.frametype = AST_FRAME_VOICE;
- ast_format_set(&myf.f.subclass.format, AST_FORMAT_SLINEAR, 0);
myf.f.datalen = res;
myf.f.samples = res / 2;
- myf.f.offset = AST_FRIENDLY_OFFSET;
- myf.f.src = __PRETTY_FUNCTION__;
- myf.f.data.ptr = myf.frdata;
if (ast_write(chan, &myf.f) < 0) {
res = -1;
ast_frfree(f);
@@ -269,8 +271,10 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
close(fds[0]);
close(fds[1]);
- if (!res && owriteformat.id)
- ast_set_write_format(chan, &owriteformat);
+ if (!res && owriteformat)
+ ast_set_write_format(chan, owriteformat);
+ ao2_cleanup(owriteformat);
+
return res;
}
diff --git a/apps/app_ices.c b/apps/app_ices.c
index bdccff43f..3122273c7 100644
--- a/apps/app_ices.c
+++ b/apps/app_ices.c
@@ -48,6 +48,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/app.h"
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<application name="ICES" language="en_US">
@@ -115,12 +116,11 @@ static int ices_exec(struct ast_channel *chan, const char *data)
int ms = -1;
int pid = -1;
int flags;
- struct ast_format oreadformat;
+ struct ast_format *oreadformat;
struct ast_frame *f;
char filename[256]="";
char *c;
- ast_format_clear(&oreadformat);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "ICES requires an argument (configfile.xml)\n");
return -1;
@@ -145,12 +145,13 @@ static int ices_exec(struct ast_channel *chan, const char *data)
return -1;
}
- ast_format_copy(&oreadformat, ast_channel_readformat(chan));
- res = ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR);
+ oreadformat = ao2_bump(ast_channel_readformat(chan));
+ res = ast_set_read_format(chan, ast_format_slin);
if (res < 0) {
close(fds[0]);
close(fds[1]);
ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
+ ao2_cleanup(oreadformat);
return -1;
}
if (((char *)data)[0] == '/')
@@ -197,8 +198,9 @@ static int ices_exec(struct ast_channel *chan, const char *data)
if (pid > -1)
kill(pid, SIGKILL);
- if (!res && oreadformat.id)
- ast_set_read_format(chan, &oreadformat);
+ if (!res && oreadformat)
+ ast_set_read_format(chan, oreadformat);
+ ao2_cleanup(oreadformat);
return res;
}
diff --git a/apps/app_jack.c b/apps/app_jack.c
index 9c59ceaf4..8adfbc4cb 100644
--- a/apps/app_jack.c
+++ b/apps/app_jack.c
@@ -58,6 +58,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/app.h"
#include "asterisk/pbx.h"
#include "asterisk/audiohook.h"
+#include "asterisk/format_cache.h"
#define RESAMPLE_QUALITY 1
@@ -129,7 +130,7 @@ struct jack_data {
jack_port_t *output_port;
jack_ringbuffer_t *input_rb;
jack_ringbuffer_t *output_rb;
- enum ast_format_id audiohook_format_id;
+ struct ast_format *audiohook_format;
unsigned int audiohook_rate;
unsigned int frame_datalen;
void *output_resampler;
@@ -394,7 +395,6 @@ static int init_jack_data(struct ast_channel *chan, struct jack_data *jack_data)
jack_status_t status = 0;
jack_options_t jack_options = JackNullOption;
- struct ast_format format_slin;
unsigned int channel_rate;
unsigned int ringbuffer_size;
@@ -402,19 +402,17 @@ static int init_jack_data(struct ast_channel *chan, struct jack_data *jack_data)
/* Deducing audiohook sample rate from channel format
ATTENTION: Might be problematic, if channel has different sampling than used by audiohook!
*/
- channel_rate = ast_format_rate(ast_channel_readformat(chan));
- jack_data->audiohook_format_id = ast_format_slin_by_rate(channel_rate);
-
- ast_format_set(&format_slin, jack_data->audiohook_format_id, 0);
- jack_data->audiohook_rate = ast_format_rate(&format_slin);
+ channel_rate = ast_format_get_sample_rate(ast_channel_readformat(chan));
+ jack_data->audiohook_format = ast_format_cache_get_slin_by_rate(channel_rate);
+ jack_data->audiohook_rate = ast_format_get_sample_rate(jack_data->audiohook_format);
/* Guessing frame->datalen assuming a ptime of 20ms */
jack_data->frame_datalen = jack_data->audiohook_rate / 50;
ringbuffer_size = jack_data->frame_datalen * RINGBUFFER_FRAME_CAPACITY;
- ast_debug(1, "Audiohook parameters: slin-format:%d, rate:%d, frame-len:%d, ringbuffer_size: %d\n",
- jack_data->audiohook_format_id, jack_data->audiohook_rate, jack_data->frame_datalen, ringbuffer_size);
+ ast_debug(1, "Audiohook parameters: slin-format:%s, rate:%d, frame-len:%d, ringbuffer_size: %d\n",
+ ast_format_get_name(jack_data->audiohook_format), jack_data->audiohook_rate, jack_data->frame_datalen, ringbuffer_size);
if (!ast_strlen_zero(jack_data->client_name)) {
client_name = jack_data->client_name;
@@ -628,12 +626,12 @@ static void handle_jack_audio(struct ast_channel *chan, struct jack_data *jack_d
short buf[jack_data->frame_datalen];
struct ast_frame f = {
.frametype = AST_FRAME_VOICE,
+ .subclass.format = jack_data->audiohook_format,
.src = "JACK",
.data.ptr = buf,
.datalen = sizeof(buf),
.samples = ARRAY_LEN(buf),
};
- ast_format_set(&f.subclass.format, jack_data->audiohook_format_id, 0);
for (;;) {
size_t res, read_len;
@@ -778,12 +776,12 @@ static int jack_exec(struct ast_channel *chan, const char *data)
return -1;
}
- if (ast_set_read_format_by_id(chan, jack_data->audiohook_format_id)) {
+ if (ast_set_read_format(chan, jack_data->audiohook_format)) {
destroy_jack_data(jack_data);
return -1;
}
- if (ast_set_write_format_by_id(chan, jack_data->audiohook_format_id)) {
+ if (ast_set_write_format(chan, jack_data->audiohook_format)) {
destroy_jack_data(jack_data);
return -1;
}
@@ -859,9 +857,10 @@ static int jack_hook_callback(struct ast_audiohook *audiohook, struct ast_channe
jack_data = datastore->data;
- if (frame->subclass.format.id != jack_data->audiohook_format_id) {
- ast_log(LOG_WARNING, "Expected frame in SLINEAR with id %d for the audiohook, but got format %s\n",
- jack_data->audiohook_format_id, ast_getformatname(&frame->subclass.format));
+ if (ast_format_cmp(frame->subclass.format, jack_data->audiohook_format) == AST_FORMAT_CMP_NOT_EQUAL) {
+ ast_log(LOG_WARNING, "Expected frame in %s for the audiohook, but got format %s\n",
+ ast_format_get_name(jack_data->audiohook_format),
+ ast_format_get_name(frame->subclass.format));
ast_channel_unlock(chan);
return 0;
}
diff --git a/apps/app_meetme.c b/apps/app_meetme.c
index 1a780e6ff..6d0b5ac1b 100644
--- a/apps/app_meetme.c
+++ b/apps/app_meetme.c
@@ -77,6 +77,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/stasis_channels.h"
#include "asterisk/stasis_message_router.h"
#include "asterisk/json.h"
+#include "asterisk/format_compatibility.h"
#include "enter.h"
#include "leave.h"
@@ -1606,8 +1607,7 @@ static struct ast_conference *build_conf(const char *confno, const char *pin,
struct ast_conference *cnf;
struct dahdi_confinfo dahdic = { 0, };
int confno_int = 0;
- struct ast_format_cap *cap_slin = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
- struct ast_format tmp_fmt;
+ struct ast_format_cap *cap_slin = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
AST_LIST_LOCK(&confs);
@@ -1619,7 +1619,7 @@ static struct ast_conference *build_conf(const char *confno, const char *pin,
if (cnf || (!make && !dynamic) || !cap_slin)
goto cnfout;
- ast_format_cap_add(cap_slin, ast_format_set(&tmp_fmt, AST_FORMAT_SLINEAR, 0));
+ ast_format_cap_append(cap_slin, ast_format_slin, 0);
/* Make a new one */
if (!(cnf = ast_calloc(1, sizeof(*cnf))) ||
!(cnf->usercontainer = ao2_container_alloc(1, NULL, user_no_cmp))) {
@@ -1667,8 +1667,8 @@ static struct ast_conference *build_conf(const char *confno, const char *pin,
/* Setup a new channel for playback of audio files */
cnf->chan = ast_request("DAHDI", cap_slin, NULL, chan, "pseudo", NULL);
if (cnf->chan) {
- ast_set_read_format_by_id(cnf->chan, AST_FORMAT_SLINEAR);
- ast_set_write_format_by_id(cnf->chan, AST_FORMAT_SLINEAR);
+ ast_set_read_format(cnf->chan, ast_format_slin);
+ ast_set_write_format(cnf->chan, ast_format_slin);
dahdic.chan = 0;
dahdic.confno = cnf->dahdiconf;
dahdic.confmode = DAHDI_CONF_CONFANN | DAHDI_CONF_CONFANNMON;
@@ -1704,7 +1704,7 @@ static struct ast_conference *build_conf(const char *confno, const char *pin,
conf_map[confno_int] = 1;
cnfout:
- cap_slin = ast_format_cap_destroy(cap_slin);
+ ao2_cleanup(cap_slin);
if (cnf)
ast_atomic_fetchadd_int(&cnf->refcount, refcount);
@@ -3089,8 +3089,8 @@ static void meetme_menu_admin_extended(enum menu_modes *menu_mode, int *dtmf,
if ((conf->recordthread == AST_PTHREADT_NULL) && ast_test_flag64(confflags, CONFFLAG_RECORDCONF) && ((conf->lchan = ast_request("DAHDI", cap_slin, NULL, chan, "pseudo", NULL)))) {
struct dahdi_confinfo dahdic;
- ast_set_read_format_by_id(conf->lchan, AST_FORMAT_SLINEAR);
- ast_set_write_format_by_id(conf->lchan, AST_FORMAT_SLINEAR);
+ ast_set_read_format(conf->lchan, ast_format_slin);
+ ast_set_write_format(conf->lchan, ast_format_slin);
dahdic.chan = 0;
dahdic.confno = conf->dahdiconf;
dahdic.confmode = DAHDI_CONF_CONFANN | DAHDI_CONF_CONFANNMON;
@@ -3217,13 +3217,12 @@ static int conf_run(struct ast_channel *chan, struct ast_conference *conf, struc
int setusercount = 0;
int confsilence = 0, totalsilence = 0;
char *mailbox, *context;
- struct ast_format_cap *cap_slin = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
- struct ast_format tmpfmt;
+ struct ast_format_cap *cap_slin = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!cap_slin) {
goto conf_run_cleanup;
}
- ast_format_cap_add(cap_slin, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0));
+ ast_format_cap_append(cap_slin, ast_format_slin, 0);
if (!(user = ao2_alloc(sizeof(*user), NULL))) {
goto conf_run_cleanup;
@@ -3345,8 +3344,8 @@ static int conf_run(struct ast_channel *chan, struct ast_conference *conf, struc
ast_mutex_lock(&conf->recordthreadlock);
if ((conf->recordthread == AST_PTHREADT_NULL) && ast_test_flag64(confflags, CONFFLAG_RECORDCONF) &&
((conf->lchan = ast_request("DAHDI", cap_slin, NULL, chan, "pseudo", NULL)))) {
- ast_set_read_format_by_id(conf->lchan, AST_FORMAT_SLINEAR);
- ast_set_write_format_by_id(conf->lchan, AST_FORMAT_SLINEAR);
+ ast_set_read_format(conf->lchan, ast_format_slin);
+ ast_set_write_format(conf->lchan, ast_format_slin);
dahdic.chan = 0;
dahdic.confno = conf->dahdiconf;
dahdic.confmode = DAHDI_CONF_CONFANN | DAHDI_CONF_CONFANNMON;
@@ -3572,12 +3571,12 @@ static int conf_run(struct ast_channel *chan, struct ast_conference *conf, struc
ast_indicate(chan, -1);
}
- if (ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR) < 0) {
+ if (ast_set_write_format(chan, ast_format_slin) < 0) {
ast_log(LOG_WARNING, "Unable to set '%s' to write linear mode\n", ast_channel_name(chan));
goto outrun;
}
- if (ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR) < 0) {
+ if (ast_set_read_format(chan, ast_format_slin) < 0) {
ast_log(LOG_WARNING, "Unable to set '%s' to read linear mode\n", ast_channel_name(chan));
goto outrun;
}
@@ -4131,7 +4130,7 @@ static int conf_run(struct ast_channel *chan, struct ast_conference *conf, struc
dtmfstr[1] = '\0';
}
- if ((f->frametype == AST_FRAME_VOICE) && (f->subclass.format.id == AST_FORMAT_SLINEAR)) {
+ if ((f->frametype == AST_FRAME_VOICE) && (ast_format_cmp(f->subclass.format, ast_format_slin) == AST_FORMAT_CMP_EQUAL)) {
if (user->talk.actual) {
ast_frame_adjust_volume(f, user->talk.actual);
}
@@ -4289,7 +4288,7 @@ static int conf_run(struct ast_channel *chan, struct ast_conference *conf, struc
if (res > 0) {
memset(&fr, 0, sizeof(fr));
fr.frametype = AST_FRAME_VOICE;
- ast_format_set(&fr.subclass.format, AST_FORMAT_SLINEAR, 0);
+ fr.subclass.format = ast_format_slin;
fr.datalen = res;
fr.samples = res / 2;
fr.data.ptr = buf;
@@ -4301,7 +4300,7 @@ static int conf_run(struct ast_channel *chan, struct ast_conference *conf, struc
)) {
int idx;
for (idx = 0; idx < AST_FRAME_BITS; idx++) {
- if (ast_format_to_old_bitfield(ast_channel_rawwriteformat(chan)) & (1 << idx)) {
+ if (ast_format_compatibility_format2bitfield(ast_channel_rawwriteformat(chan)) & (1 << idx)) {
break;
}
}
@@ -4319,11 +4318,7 @@ static int conf_run(struct ast_channel *chan, struct ast_conference *conf, struc
mohtempstopped = 1;
}
if (!conf->transpath[idx]) {
- struct ast_format src;
- struct ast_format dst;
- ast_format_set(&src, AST_FORMAT_SLINEAR, 0);
- ast_format_from_old_bitfield(&dst, (1 << idx));
- conf->transpath[idx] = ast_translator_build_path(&dst, &src);
+ conf->transpath[idx] = ast_translator_build_path(ast_channel_rawwriteformat(chan), ast_format_slin);
}
if (conf->transpath[idx]) {
conf->transframe[idx] = ast_translate(conf->transpath[idx], conf->origframe, 0);
@@ -4485,7 +4480,7 @@ bailoutandtrynormal:
conf_run_cleanup:
- cap_slin = ast_format_cap_destroy(cap_slin);
+ ao2_cleanup(cap_slin);
return ret;
}
diff --git a/apps/app_milliwatt.c b/apps/app_milliwatt.c
index a4adc08c5..a3509581d 100644
--- a/apps/app_milliwatt.c
+++ b/apps/app_milliwatt.c
@@ -37,6 +37,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/indications.h"
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<application name="Milliwatt" language="en_US">
@@ -79,13 +80,14 @@ static int milliwatt_generate(struct ast_channel *chan, void *data, int len, int
{
unsigned char buf[AST_FRIENDLY_OFFSET + 640];
const int maxsamples = ARRAY_LEN(buf) - (AST_FRIENDLY_OFFSET / sizeof(buf[0]));
- int i, *indexp = (int *) data;
+ int i, *indexp = (int *) data, res;
struct ast_frame wf = {
.frametype = AST_FRAME_VOICE,
.offset = AST_FRIENDLY_OFFSET,
.src = __FUNCTION__,
};
- ast_format_set(&wf.subclass.format, AST_FORMAT_ULAW, 0);
+
+ wf.subclass.format = ast_format_ulaw;
wf.data.ptr = buf + AST_FRIENDLY_OFFSET;
/* Instead of len, use samples, because channel.c generator_force
@@ -108,7 +110,10 @@ static int milliwatt_generate(struct ast_channel *chan, void *data, int len, int
*indexp &= 7;
}
- if (ast_write(chan,&wf) < 0) {
+ res = ast_write(chan, &wf);
+ ast_frfree(&wf);
+
+ if (res < 0) {
ast_log(LOG_WARNING,"Failed to write frame to '%s': %s\n",ast_channel_name(chan),strerror(errno));
return -1;
}
@@ -124,8 +129,8 @@ static struct ast_generator milliwattgen = {
static int old_milliwatt_exec(struct ast_channel *chan)
{
- ast_set_write_format_by_id(chan, AST_FORMAT_ULAW);
- ast_set_read_format_by_id(chan, AST_FORMAT_ULAW);
+ ast_set_write_format(chan, ast_format_ulaw);
+ ast_set_read_format(chan, ast_format_ulaw);
if (ast_channel_state(chan) != AST_STATE_UP) {
ast_answer(chan);
diff --git a/apps/app_mixmonitor.c b/apps/app_mixmonitor.c
index ab1d0bad1..1e7833cea 100644
--- a/apps/app_mixmonitor.c
+++ b/apps/app_mixmonitor.c
@@ -57,6 +57,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/linkedlists.h"
#include "asterisk/test.h"
#include "asterisk/mixmonitor.h"
+#include "asterisk/format_cache.h"
#include "asterisk/beep.h"
/*** DOCUMENTATION
@@ -616,7 +617,7 @@ static void mixmonitor_save_prep(struct mixmonitor *mixmonitor, char *filename,
*errflag = 1;
} else {
struct ast_filestream *tmp = *fs;
- mixmonitor->mixmonitor_ds->samp_rate = MAX(mixmonitor->mixmonitor_ds->samp_rate, ast_format_rate(&tmp->fmt->format));
+ mixmonitor->mixmonitor_ds->samp_rate = MAX(mixmonitor->mixmonitor_ds->samp_rate, ast_format_get_sample_rate(tmp->fmt->format));
}
}
}
@@ -635,7 +636,7 @@ static void *mixmonitor_thread(void *obj)
unsigned int oflags;
int errflag = 0;
- struct ast_format format_slin;
+ struct ast_format *format_slin;
/* Keep callid association before any log messages */
if (mixmonitor->callid) {
@@ -653,11 +654,10 @@ static void *mixmonitor_thread(void *obj)
mixmonitor_save_prep(mixmonitor, mixmonitor->filename_read, fs_read, &oflags, &errflag, &fs_read_ext);
mixmonitor_save_prep(mixmonitor, mixmonitor->filename_write, fs_write, &oflags, &errflag, &fs_write_ext);
- ast_format_set(&format_slin, ast_format_slin_by_rate(mixmonitor->mixmonitor_ds->samp_rate), 0);
+ format_slin = ast_format_cache_get_slin_by_rate(mixmonitor->mixmonitor_ds->samp_rate);
ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
-
/* The audiohook must enter and exit the loop locked */
ast_audiohook_lock(&mixmonitor->audiohook);
while (mixmonitor->audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING && !mixmonitor->mixmonitor_ds->fs_quit) {
@@ -665,7 +665,7 @@ static void *mixmonitor_thread(void *obj)
struct ast_frame *fr_read = NULL;
struct ast_frame *fr_write = NULL;
- if (!(fr = ast_audiohook_read_frame_all(&mixmonitor->audiohook, SAMPLES_PER_FRAME, &format_slin,
+ if (!(fr = ast_audiohook_read_frame_all(&mixmonitor->audiohook, SAMPLES_PER_FRAME, format_slin,
&fr_read, &fr_write))) {
ast_audiohook_trigger_wait(&mixmonitor->audiohook);
diff --git a/apps/app_mp3.c b/apps/app_mp3.c
index 463a91906..7e4e8e490 100644
--- a/apps/app_mp3.c
+++ b/apps/app_mp3.c
@@ -47,6 +47,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/app.h"
+#include "asterisk/format_cache.h"
#define LOCAL_MPG_123 "/usr/local/bin/mpg123"
#define MPG_123 "/usr/bin/mpg123"
@@ -143,7 +144,7 @@ static int mp3_exec(struct ast_channel *chan, const char *data)
int fds[2];
int ms = -1;
int pid = -1;
- struct ast_format owriteformat;
+ RAII_VAR(struct ast_format *, owriteformat, NULL, ao2_cleanup);
int timeout = 2000;
struct timeval next;
struct ast_frame *f;
@@ -155,7 +156,6 @@ static int mp3_exec(struct ast_channel *chan, const char *data)
.f = { 0, },
};
- ast_format_clear(&owriteformat);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "MP3 Playback requires an argument (filename)\n");
return -1;
@@ -168,12 +168,21 @@ static int mp3_exec(struct ast_channel *chan, const char *data)
ast_stopstream(chan);
- ast_format_copy(&owriteformat, ast_channel_writeformat(chan));
- res = ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR);
+ owriteformat = ao2_bump(ast_channel_writeformat(chan));
+ res = ast_set_write_format(chan, ast_format_slin);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
return -1;
}
+
+ myf.f.frametype = AST_FRAME_VOICE;
+ myf.f.subclass.format = ast_format_slin;
+ myf.f.mallocd = 0;
+ myf.f.offset = AST_FRIENDLY_OFFSET;
+ myf.f.src = __PRETTY_FUNCTION__;
+ myf.f.delivery.tv_sec = 0;
+ myf.f.delivery.tv_usec = 0;
+ myf.f.data.ptr = myf.frdata;
res = mp3play(data, fds[1]);
if (!strncasecmp(data, "http://", 7)) {
@@ -191,16 +200,8 @@ static int mp3_exec(struct ast_channel *chan, const char *data)
if (ms <= 0) {
res = timed_read(fds[0], myf.frdata, sizeof(myf.frdata), timeout);
if (res > 0) {
- myf.f.frametype = AST_FRAME_VOICE;
- ast_format_set(&myf.f.subclass.format, AST_FORMAT_SLINEAR, 0);
myf.f.datalen = res;
myf.f.samples = res / 2;
- myf.f.mallocd = 0;
- myf.f.offset = AST_FRIENDLY_OFFSET;
- myf.f.src = __PRETTY_FUNCTION__;
- myf.f.delivery.tv_sec = 0;
- myf.f.delivery.tv_usec = 0;
- myf.f.data.ptr = myf.frdata;
if (ast_write(chan, &myf.f) < 0) {
res = -1;
break;
@@ -241,8 +242,10 @@ static int mp3_exec(struct ast_channel *chan, const char *data)
if (pid > -1)
kill(pid, SIGKILL);
- if (!res && owriteformat.id)
- ast_set_write_format(chan, &owriteformat);
+ if (!res && owriteformat)
+ ast_set_write_format(chan, owriteformat);
+
+ ast_frfree(&myf.f);
return res;
}
diff --git a/apps/app_nbscat.c b/apps/app_nbscat.c
index 8cf02b369..435af8a29 100644
--- a/apps/app_nbscat.c
+++ b/apps/app_nbscat.c
@@ -46,6 +46,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/app.h"
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<application name="NBScat" language="en_US">
@@ -115,7 +116,7 @@ static int NBScat_exec(struct ast_channel *chan, const char *data)
int fds[2];
int ms = -1;
int pid = -1;
- struct ast_format owriteformat;
+ struct ast_format *owriteformat;
struct timeval next;
struct ast_frame *f;
struct myframe {
@@ -124,7 +125,6 @@ static int NBScat_exec(struct ast_channel *chan, const char *data)
short frdata[160];
} myf;
- ast_format_clear(&owriteformat);
if (socketpair(AF_LOCAL, SOCK_STREAM, 0, fds)) {
ast_log(LOG_WARNING, "Unable to create socketpair\n");
return -1;
@@ -132,12 +132,22 @@ static int NBScat_exec(struct ast_channel *chan, const char *data)
ast_stopstream(chan);
- ast_format_copy(&owriteformat, ast_channel_writeformat(chan));
- res = ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR);
+ owriteformat = ao2_bump(ast_channel_writeformat(chan));
+ res = ast_set_write_format(chan, ast_format_slin);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
+ ao2_cleanup(owriteformat);
return -1;
}
+
+ myf.f.frametype = AST_FRAME_VOICE;
+ myf.f.subclass.format = ast_format_slin;
+ myf.f.mallocd = 0;
+ myf.f.offset = AST_FRIENDLY_OFFSET;
+ myf.f.src = __PRETTY_FUNCTION__;
+ myf.f.delivery.tv_sec = 0;
+ myf.f.delivery.tv_usec = 0;
+ myf.f.data.ptr = myf.frdata;
res = NBScatplay(fds[1]);
/* Wait 1000 ms first */
@@ -152,16 +162,8 @@ static int NBScat_exec(struct ast_channel *chan, const char *data)
if (ms <= 0) {
res = timed_read(fds[0], myf.frdata, sizeof(myf.frdata));
if (res > 0) {
- myf.f.frametype = AST_FRAME_VOICE;
- ast_format_set(&myf.f.subclass.format, AST_FORMAT_SLINEAR, 0);
myf.f.datalen = res;
myf.f.samples = res / 2;
- myf.f.mallocd = 0;
- myf.f.offset = AST_FRIENDLY_OFFSET;
- myf.f.src = __PRETTY_FUNCTION__;
- myf.f.delivery.tv_sec = 0;
- myf.f.delivery.tv_usec = 0;
- myf.f.data.ptr = myf.frdata;
if (ast_write(chan, &myf.f) < 0) {
res = -1;
break;
@@ -199,11 +201,13 @@ static int NBScat_exec(struct ast_channel *chan, const char *data)
}
close(fds[0]);
close(fds[1]);
+ ast_frfree(&myf.f);
if (pid > -1)
kill(pid, SIGKILL);
- if (!res && owriteformat.id)
- ast_set_write_format(chan, &owriteformat);
+ if (!res && owriteformat)
+ ast_set_write_format(chan, owriteformat);
+ ao2_cleanup(owriteformat);
return res;
}
diff --git a/apps/app_originate.c b/apps/app_originate.c
index 9fceb0849..15898b240 100644
--- a/apps/app_originate.c
+++ b/apps/app_originate.c
@@ -45,6 +45,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/app.h"
+#include "asterisk/format_cache.h"
static const char app_originate[] = "Originate";
@@ -112,22 +113,22 @@ static int originate_exec(struct ast_channel *chan, const char *data)
int outgoing_status = 0;
unsigned int timeout = 30;
static const char default_exten[] = "s";
- struct ast_format tmpfmt;
- struct ast_format_cap *cap_slin = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
+ struct ast_format_cap *cap_slin = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
ast_autoservice_start(chan);
if (!cap_slin) {
goto return_cleanup;
}
- ast_format_cap_add(cap_slin, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0));
- ast_format_cap_add(cap_slin, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR12, 0));
- ast_format_cap_add(cap_slin, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0));
- ast_format_cap_add(cap_slin, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR24, 0));
- ast_format_cap_add(cap_slin, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR32, 0));
- ast_format_cap_add(cap_slin, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR44, 0));
- ast_format_cap_add(cap_slin, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR48, 0));
- ast_format_cap_add(cap_slin, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR96, 0));
- ast_format_cap_add(cap_slin, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR192, 0));
+
+ ast_format_cap_append(cap_slin, ast_format_slin, 0);
+ ast_format_cap_append(cap_slin, ast_format_slin12, 0);
+ ast_format_cap_append(cap_slin, ast_format_slin16, 0);
+ ast_format_cap_append(cap_slin, ast_format_slin24, 0);
+ ast_format_cap_append(cap_slin, ast_format_slin32, 0);
+ ast_format_cap_append(cap_slin, ast_format_slin44, 0);
+ ast_format_cap_append(cap_slin, ast_format_slin48, 0);
+ ast_format_cap_append(cap_slin, ast_format_slin96, 0);
+ ast_format_cap_append(cap_slin, ast_format_slin192, 0);
if (ast_strlen_zero(data)) {
ast_log(LOG_ERROR, "Originate() requires arguments\n");
@@ -222,7 +223,7 @@ return_cleanup:
break;
}
}
- cap_slin = ast_format_cap_destroy(cap_slin);
+ ao2_cleanup(cap_slin);
ast_autoservice_stop(chan);
return res;
diff --git a/apps/app_record.c b/apps/app_record.c
index 45f1d8602..4008fc012 100644
--- a/apps/app_record.c
+++ b/apps/app_record.c
@@ -39,6 +39,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/app.h"
#include "asterisk/channel.h"
#include "asterisk/dsp.h" /* use dsp routines for silence detection */
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<application name="Record" language="en_US">
@@ -196,7 +197,7 @@ static int record_exec(struct ast_channel *chan, const char *data)
int maxduration = 0; /* max duration of recording in milliseconds */
int gottimeout = 0; /* did we timeout for maxduration exceeded? */
int terminator = '#';
- struct ast_format rfmt;
+ RAII_VAR(struct ast_format *, rfmt, NULL, ao2_cleanup);
int ioflags;
struct ast_silence_generator *silgen = NULL;
struct ast_flags flags = { 0, };
@@ -209,8 +210,6 @@ static int record_exec(struct ast_channel *chan, const char *data)
int ms;
struct timeval start;
- ast_format_clear(&rfmt);
-
/* The next few lines of code parse out the filename and header from the input string */
if (ast_strlen_zero(data)) { /* no data implies no filename or anything is present */
ast_log(LOG_WARNING, "Record requires an argument (filename)\n");
@@ -331,8 +330,8 @@ static int record_exec(struct ast_channel *chan, const char *data)
/* The end of beep code. Now the recording starts */
if (silence > 0) {
- ast_format_copy(&rfmt, ast_channel_readformat(chan));
- res = ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR);
+ rfmt = ao2_bump(ast_channel_readformat(chan));
+ res = ast_set_read_format(chan, ast_format_slin);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
@@ -459,8 +458,8 @@ static int record_exec(struct ast_channel *chan, const char *data)
ast_channel_stop_silence_generator(chan, silgen);
out:
- if ((silence > 0) && rfmt.id) {
- res = ast_set_read_format(chan, &rfmt);
+ if ((silence > 0) && rfmt) {
+ res = ast_set_read_format(chan, rfmt);
if (res) {
ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", ast_channel_name(chan));
}
diff --git a/apps/app_sms.c b/apps/app_sms.c
index b7d247818..266f8abdc 100644
--- a/apps/app_sms.c
+++ b/apps/app_sms.c
@@ -56,6 +56,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/callerid.h"
#include "asterisk/utils.h"
#include "asterisk/app.h"
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<application name="SMS" language="en_US">
@@ -140,11 +141,11 @@ static const signed short wave[] = {
static unsigned char wavea[80];
typedef unsigned char output_t;
static const output_t *wave_out = wavea; /* outgoing samples */
-#define __OUT_FMT AST_FORMAT_ALAW;
+#define __OUT_FMT ast_format_alaw
#else
typedef signed short output_t;
static const output_t *wave_out = wave; /* outgoing samples */
-#define __OUT_FMT AST_FORMAT_SLINEAR
+#define __OUT_FMT ast_format_slin
#endif
#define OSYNC_BITS 80 /* initial sync bits */
@@ -1599,7 +1600,7 @@ static int sms_generate(struct ast_channel *chan, void *data, int len, int sampl
#define MAXSAMPLES (800)
output_t *buf;
sms_t *h = data;
- int i;
+ int i, res;
if (samples > MAXSAMPLES) {
ast_log(LOG_WARNING, "Only doing %d samples (%d requested)\n",
@@ -1610,7 +1611,7 @@ static int sms_generate(struct ast_channel *chan, void *data, int len, int sampl
buf = ast_alloca(len);
f.frametype = AST_FRAME_VOICE;
- ast_format_set(&f.subclass.format, __OUT_FMT, 0);
+ f.subclass.format = __OUT_FMT;
f.datalen = samples * sizeof(*buf);
f.offset = AST_FRIENDLY_OFFSET;
f.mallocd = 0;
@@ -1660,7 +1661,9 @@ static int sms_generate(struct ast_channel *chan, void *data, int len, int sampl
}
}
}
- if (ast_write(chan, &f) < 0) {
+ res = ast_write(chan, &f);
+ ast_frfree(&f);
+ if (res < 0) {
ast_log(LOG_WARNING, "Failed to write frame to '%s': %s\n", ast_channel_name(chan), strerror(errno));
return -1;
}
@@ -2012,9 +2015,9 @@ static int sms_exec(struct ast_channel *chan, const char *data)
sms_messagetx(&h);
}
- res = ast_set_write_format_by_id(chan, __OUT_FMT);
+ res = ast_set_write_format(chan, __OUT_FMT);
if (res >= 0) {
- res = ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR);
+ res = ast_set_read_format(chan, ast_format_slin);
}
if (res < 0) {
ast_log(LOG_ERROR, "Unable to set to linear mode, giving up\n");
diff --git a/apps/app_speech_utils.c b/apps/app_speech_utils.c
index e9ca63ea9..e52425d6a 100644
--- a/apps/app_speech_utils.c
+++ b/apps/app_speech_utils.c
@@ -672,7 +672,7 @@ static int speech_background(struct ast_channel *chan, const char *data)
int res = 0, done = 0, started = 0, quieted = 0, max_dtmf_len = 0;
struct ast_speech *speech = find_speech(chan);
struct ast_frame *f = NULL;
- struct ast_format oldreadformat;
+ RAII_VAR(struct ast_format *, oldreadformat, NULL, ao2_cleanup);
char dtmf[AST_MAX_EXTENSION] = "";
struct timeval start = { 0, 0 }, current;
struct ast_datastore *datastore = NULL;
@@ -688,7 +688,6 @@ static int speech_background(struct ast_channel *chan, const char *data)
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
- ast_format_clear(&oldreadformat);
if (speech == NULL)
return -1;
@@ -704,10 +703,10 @@ static int speech_background(struct ast_channel *chan, const char *data)
}
/* Record old read format */
- ast_format_copy(&oldreadformat, ast_channel_readformat(chan));
+ oldreadformat = ao2_bump(ast_channel_readformat(chan));
/* Change read format to be signed linear */
- if (ast_set_read_format(chan, &speech->format))
+ if (ast_set_read_format(chan, speech->format))
return -1;
if (!ast_strlen_zero(args.soundfile)) {
@@ -912,7 +911,7 @@ static int speech_background(struct ast_channel *chan, const char *data)
ast_channel_datastore_remove(chan, datastore);
} else {
/* Channel is okay so restore read format */
- ast_set_read_format(chan, &oldreadformat);
+ ast_set_read_format(chan, oldreadformat);
}
return 0;
diff --git a/apps/app_talkdetect.c b/apps/app_talkdetect.c
index 5ef80c33c..5ccf2607e 100644
--- a/apps/app_talkdetect.c
+++ b/apps/app_talkdetect.c
@@ -42,6 +42,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/utils.h"
#include "asterisk/dsp.h"
#include "asterisk/app.h"
+#include "asterisk/format.h"
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<application name="BackgroundDetect" language="en_US">
@@ -91,7 +93,7 @@ static int background_detect_exec(struct ast_channel *chan, const char *data)
int analysistime = -1;
int continue_analysis = 1;
int x;
- struct ast_format origrformat;
+ RAII_VAR(struct ast_format *, origrformat, NULL, ao2_cleanup);
struct ast_dsp *dsp = NULL;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(filename);
@@ -101,7 +103,6 @@ static int background_detect_exec(struct ast_channel *chan, const char *data)
AST_APP_ARG(analysistime);
);
- ast_format_clear(&origrformat);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "BackgroundDetect requires an argument (filename)\n");
return -1;
@@ -131,8 +132,8 @@ static int background_detect_exec(struct ast_channel *chan, const char *data)
}
}
- ast_format_copy(&origrformat, ast_channel_readformat(chan));
- if ((ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR))) {
+ origrformat = ao2_bump(ast_channel_readformat(chan));
+ if ((ast_set_read_format(chan, ast_format_slin))) {
ast_log(LOG_WARNING, "Unable to set read format to linear!\n");
res = -1;
break;
@@ -187,7 +188,8 @@ static int background_detect_exec(struct ast_channel *chan, const char *data)
ast_frfree(fr);
break;
}
- } else if ((fr->frametype == AST_FRAME_VOICE) && (fr->subclass.format.id == AST_FORMAT_SLINEAR) && continue_analysis) {
+ } else if ((fr->frametype == AST_FRAME_VOICE) &&
+ (ast_format_cmp(fr->subclass.format, ast_format_slin) == AST_FORMAT_CMP_EQUAL) && continue_analysis) {
int totalsilence;
int ms;
res = ast_dsp_silence(dsp, fr, &totalsilence);
@@ -233,9 +235,9 @@ static int background_detect_exec(struct ast_channel *chan, const char *data)
} while (0);
if (res > -1) {
- if (origrformat.id && ast_set_read_format(chan, &origrformat)) {
+ if (origrformat && ast_set_read_format(chan, origrformat)) {
ast_log(LOG_WARNING, "Failed to restore read format for %s to %s\n",
- ast_channel_name(chan), ast_getformatname(&origrformat));
+ ast_channel_name(chan), ast_format_get_name(origrformat));
}
}
if (dsp) {
diff --git a/apps/app_test.c b/apps/app_test.c
index 5b7b979e6..54139ba80 100644
--- a/apps/app_test.c
+++ b/apps/app_test.c
@@ -44,6 +44,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/app.h"
#include "asterisk/pbx.h"
#include "asterisk/utils.h"
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<application name="TestServer" language="en_US">
@@ -91,11 +92,12 @@ static int measurenoise(struct ast_channel *chan, int ms, char *who)
short *foo;
struct timeval start;
struct ast_frame *f;
- struct ast_format rformat;
+ struct ast_format *rformat;
- ast_format_copy(&rformat, ast_channel_readformat(chan));
- if (ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR)) {
+ rformat = ao2_bump(ast_channel_readformat(chan));
+ if (ast_set_read_format(chan, ast_format_slin)) {
ast_log(LOG_NOTICE, "Unable to set to linear mode!\n");
+ ao2_cleanup(rformat);
return -1;
}
start = ast_tvnow();
@@ -111,7 +113,8 @@ static int measurenoise(struct ast_channel *chan, int ms, char *who)
res = -1;
break;
}
- if ((f->frametype == AST_FRAME_VOICE) && (f->subclass.format.id == AST_FORMAT_SLINEAR)) {
+ if ((f->frametype == AST_FRAME_VOICE) &&
+ (ast_format_cmp(f->subclass.format, ast_format_slin) == AST_FORMAT_CMP_EQUAL)) {
foo = (short *)f->data.ptr;
for (x=0;x<f->samples;x++) {
noise += abs(foo[x]);
@@ -121,11 +124,13 @@ static int measurenoise(struct ast_channel *chan, int ms, char *who)
ast_frfree(f);
}
- if (rformat.id) {
- if (ast_set_read_format(chan, &rformat)) {
+ if (rformat) {
+ if (ast_set_read_format(chan, rformat)) {
ast_log(LOG_NOTICE, "Unable to restore original format!\n");
+ ao2_ref(rformat, -1);
return -1;
}
+ ao2_ref(rformat, -1);
}
if (res < 0)
return res;
diff --git a/apps/app_voicemail.c b/apps/app_voicemail.c
index 9c8a4b8ad..8a9c0f3ca 100644
--- a/apps/app_voicemail.c
+++ b/apps/app_voicemail.c
@@ -135,6 +135,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/astobj2.h"
#include "asterisk/taskprocessor.h"
#include "asterisk/test.h"
+#include "asterisk/format_cache.h"
#ifdef ODBC_STORAGE
#include "asterisk/res_odbc.h"
@@ -6112,13 +6113,16 @@ static int msg_create_from_file(struct ast_vm_recording_data *recdata)
if ((recording_fs = ast_readfile(recdata->recording_file, recdata->recording_ext, NULL, 0, 0, VOICEMAIL_DIR_MODE))) {
if (!ast_seekstream(recording_fs, 0, SEEK_END)) {
long framelength = ast_tellstream(recording_fs);
- struct ast_format result = {0,};
+ struct ast_format *result;
/* XXX This use of ast_getformatbyname seems incorrect here. The file extension does not necessarily correspond
* to the name of the format. For instance, if "raw" were passed in, I don't think ast_getformatbyname would
* find the slinear format
*/
- ast_getformatbyname(recdata->recording_ext, &result);
- duration = (int) (framelength / ast_format_rate(&result));
+ result = ast_format_cache_get(recdata->recording_ext);
+ if (result) {
+ duration = (int) (framelength / ast_format_get_sample_rate(result));
+ ao2_ref(result, -1);
+ }
}
}
@@ -14026,7 +14030,7 @@ AST_TEST_DEFINE(test_voicemail_vmsayname)
struct ast_channel *test_channel1 = NULL;
int res = -1;
- struct ast_format_cap *nativeformats;
+ struct ast_format_cap *capabilities;
static const struct ast_channel_tech fake_tech = {
.write = fake_write,
@@ -14051,12 +14055,17 @@ AST_TEST_DEFINE(test_voicemail_vmsayname)
}
/* normally this is done in the channel driver */
- ast_format_set(ast_channel_writeformat(test_channel1), AST_FORMAT_GSM, 0);
- nativeformats = ast_channel_nativeformats(test_channel1);
- ast_format_cap_add(nativeformats, ast_channel_writeformat(test_channel1));
- ast_format_set(ast_channel_rawwriteformat(test_channel1), AST_FORMAT_GSM, 0);
- ast_format_set(ast_channel_readformat(test_channel1), AST_FORMAT_GSM, 0);
- ast_format_set(ast_channel_rawreadformat(test_channel1), AST_FORMAT_GSM, 0);
+ capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+ if (!capabilities) {
+ goto exit_vmsayname_test;
+ }
+ ast_format_cap_append(capabilities, ast_format_gsm, 0);
+ ast_channel_nativeformats_set(test_channel1, capabilities);
+ ao2_ref(capabilities, -1);
+ ast_channel_set_writeformat(test_channel1, ast_format_gsm);
+ ast_channel_set_rawwriteformat(test_channel1, ast_format_gsm);
+ ast_channel_set_readformat(test_channel1, ast_format_gsm);
+ ast_channel_set_rawreadformat(test_channel1, ast_format_gsm);
ast_channel_tech_set(test_channel1, &fake_tech);
ast_channel_unlock(test_channel1);
diff --git a/apps/app_waitforsilence.c b/apps/app_waitforsilence.c
index 53870df30..f2111d720 100644
--- a/apps/app_waitforsilence.c
+++ b/apps/app_waitforsilence.c
@@ -51,6 +51,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/dsp.h"
#include "asterisk/module.h"
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<application name="WaitForSilence" language="en_US">
@@ -129,7 +130,7 @@ static char *app_noise = "WaitForNoise";
static int do_waiting(struct ast_channel *chan, int timereqd, time_t waitstart, int timeout, int wait_for_silence) {
struct ast_frame *f = NULL;
int dsptime = 0;
- struct ast_format rfmt;
+ RAII_VAR(struct ast_format *, rfmt, NULL, ao2_cleanup);
int res = 0;
struct ast_dsp *sildet; /* silence detector dsp */
time_t now;
@@ -138,8 +139,8 @@ static int do_waiting(struct ast_channel *chan, int timereqd, time_t waitstart,
int (*ast_dsp_func)(struct ast_dsp*, struct ast_frame*, int*) =
wait_for_silence ? ast_dsp_silence : ast_dsp_noise;
- ast_format_copy(&rfmt, ast_channel_readformat(chan)); /* Set to linear mode */
- if ((res = ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR)) < 0) {
+ rfmt = ao2_bump(ast_channel_readformat(chan));
+ if ((res = ast_set_read_format(chan, ast_format_slin)) < 0) {
ast_log(LOG_WARNING, "Unable to set channel to linear mode, giving up\n");
return -1;
}
@@ -199,8 +200,8 @@ static int do_waiting(struct ast_channel *chan, int timereqd, time_t waitstart,
}
- if (rfmt.id && ast_set_read_format(chan, &rfmt)) {
- ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_getformatname(&rfmt), ast_channel_name(chan));
+ if (rfmt && ast_set_read_format(chan, rfmt)) {
+ ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_format_get_name(rfmt), ast_channel_name(chan));
}
ast_dsp_free(sildet);
return res;
diff --git a/apps/confbridge/conf_chan_record.c b/apps/confbridge/conf_chan_record.c
index 34de5af3b..0e298e78d 100644
--- a/apps/confbridge/conf_chan_record.c
+++ b/apps/confbridge/conf_chan_record.c
@@ -33,6 +33,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/channel.h"
#include "asterisk/bridge.h"
+#include "asterisk/format_cache.h"
#include "include/confbridge.h"
/* ------------------------------------------------------------------- */
@@ -56,8 +57,14 @@ static int rec_write(struct ast_channel *ast, struct ast_frame *f)
static struct ast_channel *rec_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
{
struct ast_channel *chan;
- struct ast_format format;
const char *conf_name = data;
+ RAII_VAR(struct ast_format_cap *, capabilities, NULL, ao2_cleanup);
+
+ capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+ if (!capabilities) {
+ return NULL;
+ }
+ ast_format_cap_append_by_type(capabilities, AST_MEDIA_TYPE_UNKNOWN);
chan = ast_channel_alloc(1, AST_STATE_UP, NULL, NULL, NULL, NULL, NULL, NULL, NULL, 0,
"CBRec/conf-%s-uid-%d",
@@ -70,13 +77,13 @@ static struct ast_channel *rec_request(const char *type, struct ast_format_cap *
ast_channel_release(chan);
return NULL;
}
- ast_format_set(&format, AST_FORMAT_SLINEAR, 0);
+
ast_channel_tech_set(chan, conf_record_get_tech());
- ast_format_cap_add_all(ast_channel_nativeformats(chan));
- ast_format_copy(ast_channel_writeformat(chan), &format);
- ast_format_copy(ast_channel_rawwriteformat(chan), &format);
- ast_format_copy(ast_channel_readformat(chan), &format);
- ast_format_copy(ast_channel_rawreadformat(chan), &format);
+ ast_channel_nativeformats_set(chan, capabilities);
+ ast_channel_set_writeformat(chan, ast_format_slin);
+ ast_channel_set_rawwriteformat(chan, ast_format_slin);
+ ast_channel_set_readformat(chan, ast_format_slin);
+ ast_channel_set_rawreadformat(chan, ast_format_slin);
ast_channel_unlock(chan);
return chan;
}
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-09 08:07:42 +0000