[svn-commits] file: branch group/media_formats r408268 - in /team/group/media_formats: incl...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Feb 17 08:16:51 CST 2014
Author: file
Date: Mon Feb 17 08:16:47 2014
New Revision: 408268
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=408268
Log:
More, more, more.
Modified:
team/group/media_formats/include/asterisk/audiohook.h
team/group/media_formats/include/asterisk/bridge_channel.h
team/group/media_formats/main/app.c
team/group/media_formats/main/audiohook.c
team/group/media_formats/main/bridge.c
team/group/media_formats/main/bridge_basic.c
team/group/media_formats/main/bridge_channel.c
team/group/media_formats/main/callerid.c
team/group/media_formats/main/ccss.c
Modified: team/group/media_formats/include/asterisk/audiohook.h
URL: http://svnview.digium.com/svn/asterisk/team/group/media_formats/include/asterisk/audiohook.h?view=diff&rev=408268&r1=408267&r2=408268
==============================================================================
--- team/group/media_formats/include/asterisk/audiohook.h (original)
+++ team/group/media_formats/include/asterisk/audiohook.h Mon Feb 17 08:16:47 2014
@@ -109,7 +109,7 @@
struct ast_slinfactory write_factory; /*!< Factory where frames written to the channel will go through */
struct timeval read_time; /*!< Last time read factory was fed */
struct timeval write_time; /*!< Last time write factory was fed */
- struct ast_format format; /*!< Format translation path is setup as */
+ struct ast_format *format; /*!< Format translation path is setup as */
struct ast_trans_pvt *trans_pvt; /*!< Translation path for reading frames */
ast_audiohook_manipulate_callback manipulate_callback; /*!< Manipulation callback */
struct ast_audiohook_options options; /*!< Applicable options */
Modified: team/group/media_formats/include/asterisk/bridge_channel.h
URL: http://svnview.digium.com/svn/asterisk/team/group/media_formats/include/asterisk/bridge_channel.h?view=diff&rev=408268&r1=408267&r2=408268
==============================================================================
--- team/group/media_formats/include/asterisk/bridge_channel.h (original)
+++ team/group/media_formats/include/asterisk/bridge_channel.h Mon Feb 17 08:16:47 2014
@@ -133,9 +133,9 @@
* optimizing based upon talk detection. */
struct ast_bridge_tech_optimizations tech_args;
/*! Copy of read format used by chan before join */
- struct ast_format read_format;
+ struct ast_format *read_format;
/*! Copy of write format used by chan before join */
- struct ast_format write_format;
+ struct ast_format *write_format;
/*! Call ID associated with bridge channel */
struct ast_callid *callid;
/*! A clone of the roles living on chan when the bridge channel joins the bridge. This may require some opacification */
Modified: team/group/media_formats/main/app.c
URL: http://svnview.digium.com/svn/asterisk/team/group/media_formats/main/app.c?view=diff&rev=408268&r1=408267&r2=408268
==============================================================================
--- team/group/media_formats/main/app.c (original)
+++ team/group/media_formats/main/app.c Mon Feb 17 08:16:47 2014
@@ -70,6 +70,7 @@
#include "asterisk/stasis.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/json.h"
+#include "asterisk/format_cache.h"
#define MWI_TOPIC_BUCKETS 57
@@ -765,16 +766,18 @@
int fd;
int autoclose;
int allowoverride;
- struct ast_format origwfmt;
+ struct ast_format *origwfmt;
};
static void linear_release(struct ast_channel *chan, void *params)
{
struct linear_state *ls = params;
- if (ls->origwfmt.id && ast_set_write_format(chan, &ls->origwfmt)) {
- ast_log(LOG_WARNING, "Unable to restore channel '%s' to format '%d'\n", ast_channel_name(chan), ls->origwfmt.id);
- }
+ if (ls->origwfmt && ast_set_write_format(chan, ls->origwfmt)) {
+ ast_log(LOG_WARNING, "Unable to restore channel '%s' to format '%s'\n", ast_channel_name(chan),
+ ls->origwfmt->codec->name);
+ }
+ ao2_cleanup(ls->origwfmt);
if (ls->autoclose) {
close(ls->fd);
@@ -789,12 +792,11 @@
struct linear_state *ls = data;
struct ast_frame f = {
.frametype = AST_FRAME_VOICE,
+ .subclass.format = ast_format_slin,
.data.ptr = buf + AST_FRIENDLY_OFFSET / 2,
.offset = AST_FRIENDLY_OFFSET,
};
int res;
-
- ast_format_set(&f.subclass.format, AST_FORMAT_SLINEAR, 0);
len = samples * 2;
if (len > sizeof(buf) - AST_FRIENDLY_OFFSET) {
@@ -828,10 +830,11 @@
ast_clear_flag(ast_channel_flags(chan), AST_FLAG_WRITE_INT);
}
- ast_format_copy(&ls->origwfmt, ast_channel_writeformat(chan));
-
- if (ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR)) {
+ ls->origwfmt = ast_format_copy(ast_channel_writeformat(chan));
+
+ if (ast_set_write_format(chan, ast_format_slin)) {
ast_log(LOG_WARNING, "Unable to set '%s' to linear format (write)\n", ast_channel_name(chan));
+ ao2_cleanup(ls->origwfmt);
ast_free(ls);
ls = params = NULL;
}
@@ -1107,7 +1110,7 @@
return NULL;
}
- if (orig->subclass.format.id != AST_FORMAT_SLINEAR) {
+ if (ast_format_cmp(orig->subclass.format, ast_format_slin) != AST_FORMAT_CMP_EQUAL) {
ast_log(LOG_WARNING, "Attempting to silence non-slin frame\n");
return NULL;
}
@@ -1129,12 +1132,11 @@
}
silence->mallocd = AST_MALLOCD_HDR;
+ silence->subclass.format = ast_format_copy(ast_format_slin);
silence->frametype = AST_FRAME_VOICE;
silence->data.ptr = (void *)(silence + 1);
silence->samples = samples;
silence->datalen = datalen;
-
- ast_format_set(&silence->subclass.format, AST_FORMAT_SLINEAR, 0);
return silence;
}
@@ -1149,13 +1151,13 @@
* \return 0 on success.
* \return -1 on error.
*/
-static int set_read_to_slin(struct ast_channel *chan, struct ast_format *orig_format)
-{
- if (!chan || !orig_format) {
+static int set_read_to_slin(struct ast_channel *chan, struct ast_format **orig_format)
+{
+ if (!chan) {
return -1;
}
- ast_format_copy(orig_format, ast_channel_readformat(chan));
- return ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR);
+ *orig_format = ast_format_copy(ast_channel_readformat(chan));
+ return ast_set_read_format(chan, ast_format_slin);
}
static int global_silence_threshold = 128;
@@ -1197,7 +1199,7 @@
int totalsilence = 0;
int dspsilence = 0;
int olddspsilence = 0;
- struct ast_format rfmt;
+ struct ast_format *rfmt = NULL;
struct ast_silence_generator *silgen = NULL;
char prependfile[PATH_MAX];
int ioflags; /* IO flags for writing output file */
@@ -1216,7 +1218,6 @@
break;
}
- ast_format_clear(&rfmt);
if (silencethreshold < 0) {
silencethreshold = global_silence_threshold;
}
@@ -1441,7 +1442,7 @@
* set the mode, if we haven't already
* for sildet
*/
- if (muted && !rfmt.id) {
+ if (muted && !rfmt) {
ast_verb(3, "Setting read format to linear mode\n");
res = set_read_to_slin(chan, &rfmt);
if (res < 0) {
@@ -1561,9 +1562,10 @@
ast_filedelete(prependfile, sfmt[x]);
}
}
- 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", rfmt->codec->name, ast_channel_name(chan));
+ }
+ ao2_cleanup(rfmt);
if ((outmsg == 2) && (!skip_confirmation_sound)) {
ast_stream_and_wait(chan, "auth-thankyou", "");
}
Modified: team/group/media_formats/main/audiohook.c
URL: http://svnview.digium.com/svn/asterisk/team/group/media_formats/main/audiohook.c?view=diff&rev=408268&r1=408267&r2=408268
==============================================================================
--- team/group/media_formats/main/audiohook.c (original)
+++ team/group/media_formats/main/audiohook.c Mon Feb 17 08:16:47 2014
@@ -41,13 +41,14 @@
#include "asterisk/slinfactory.h"
#include "asterisk/frame.h"
#include "asterisk/translate.h"
+#include "asterisk/format_cache.h"
#define AST_AUDIOHOOK_SYNC_TOLERANCE 100 /*!< Tolerance in milliseconds for audiohooks synchronization */
#define AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE 100 /*!< When small queue is enabled, this is the maximum amount of audio that can remain queued at a time. */
struct ast_audiohook_translate {
struct ast_trans_pvt *trans_pvt;
- struct ast_format format;
+ struct ast_format *format;
};
struct ast_audiohook_list {
@@ -67,7 +68,7 @@
static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate, int reset)
{
- struct ast_format slin;
+ struct ast_format *slin;
if (audiohook->hook_internal_samp_rate == rate) {
return 0;
@@ -75,7 +76,8 @@
audiohook->hook_internal_samp_rate = rate;
- ast_format_set(&slin, ast_format_slin_by_rate(rate), 0);
+ slin = ast_format_cache_get_slin_by_rate(rate);
+
/* Setup the factories that are needed for this audiohook type */
switch (audiohook->type) {
case AST_AUDIOHOOK_TYPE_SPY:
@@ -84,12 +86,15 @@
ast_slinfactory_destroy(&audiohook->read_factory);
ast_slinfactory_destroy(&audiohook->write_factory);
}
- ast_slinfactory_init_with_format(&audiohook->read_factory, &slin);
- ast_slinfactory_init_with_format(&audiohook->write_factory, &slin);
+ ast_slinfactory_init_with_format(&audiohook->read_factory, slin);
+ ast_slinfactory_init_with_format(&audiohook->write_factory, slin);
break;
default:
break;
}
+
+ ao2_ref(slin, -1);
+
return 0;
}
@@ -143,6 +148,8 @@
if (audiohook->trans_pvt)
ast_translator_free_path(audiohook->trans_pvt);
+ ao2_cleanup(audiohook->format);
+
/* Lock and trigger be gone! */
ast_cond_destroy(&audiohook->trigger);
ast_mutex_destroy(&audiohook->lock);
@@ -220,25 +227,33 @@
short buf[samples];
struct ast_frame frame = {
.frametype = AST_FRAME_VOICE,
+ .subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate),
.data.ptr = buf,
.datalen = sizeof(buf),
.samples = samples,
};
- ast_format_set(&frame.subclass.format, ast_format_slin_by_rate(audiohook->hook_internal_samp_rate), 0);
+ struct ast_frame *out;
/* Ensure the factory is able to give us the samples we want */
- if (samples > ast_slinfactory_available(factory))
+ if (samples > ast_slinfactory_available(factory)) {
+ ao2_ref(frame.subclass.format, -1);
return NULL;
+ }
/* Read data in from factory */
- if (!ast_slinfactory_read(factory, buf, samples))
+ if (!ast_slinfactory_read(factory, buf, samples)) {
+ ao2_ref(frame.subclass.format, -1);
return NULL;
+ }
/* If a volume adjustment needs to be applied apply it */
if (vol)
ast_frame_adjust_volume(&frame, vol);
- return ast_frdup(&frame);
+ out = ast_frdup(&frame);
+ ao2_ref(frame.subclass.format, -1);
+
+ return out;
}
static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples, struct ast_frame **read_reference, struct ast_frame **write_reference)
@@ -251,7 +266,7 @@
.datalen = sizeof(buf1),
.samples = samples,
};
- ast_format_set(&frame.subclass.format, ast_format_slin_by_rate(audiohook->hook_internal_samp_rate), 0);
+ struct ast_frame *out;
/* Make sure both factories have the required samples */
usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
@@ -341,24 +356,29 @@
/* Make the final buffer part of the frame, so it gets duplicated fine */
frame.data.ptr = final_buf;
+ frame.subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);
+
/* Yahoo, a combined copy of the audio! */
- return ast_frdup(&frame);
+ out = ast_frdup(&frame);
+ ao2_ref(frame.subclass.format, -1);
+
+ return out;
}
static struct ast_frame *audiohook_read_frame_helper(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format, struct ast_frame **read_reference, struct ast_frame **write_reference)
{
struct ast_frame *read_frame = NULL, *final_frame = NULL;
- struct ast_format tmp_fmt;
+ struct ast_format *slin;
int samples_converted;
/* the number of samples requested is based on the format they are requesting. Inorder
* to process this correctly samples must be converted to our internal sample rate */
- if (audiohook->hook_internal_samp_rate == ast_format_rate(format)) {
+ if (audiohook->hook_internal_samp_rate == format->codec->sample_rate) {
samples_converted = samples;
- } else if (audiohook->hook_internal_samp_rate > ast_format_rate(format)) {
- samples_converted = samples * (audiohook->hook_internal_samp_rate / (float) ast_format_rate(format));
+ } else if (audiohook->hook_internal_samp_rate > format->codec->sample_rate) {
+ samples_converted = samples * (audiohook->hook_internal_samp_rate / (float) format->codec->sample_rate);
} else {
- samples_converted = samples * (ast_format_rate(format) / (float) audiohook->hook_internal_samp_rate);
+ samples_converted = samples * (format->codec->sample_rate / (float) audiohook->hook_internal_samp_rate);
}
if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ?
@@ -367,27 +387,33 @@
return NULL;
}
+ slin = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);
+
/* If they don't want signed linear back out, we'll have to send it through the translation path */
- if (format->id != ast_format_slin_by_rate(audiohook->hook_internal_samp_rate)) {
+ if (ast_format_cmp(format, slin) != AST_FORMAT_CMP_EQUAL) {
/* Rebuild translation path if different format then previously */
- if (ast_format_cmp(format, &audiohook->format) == AST_FORMAT_CMP_NOT_EQUAL) {
+ if (ast_format_cmp(format, audiohook->format) == AST_FORMAT_CMP_NOT_EQUAL) {
if (audiohook->trans_pvt) {
ast_translator_free_path(audiohook->trans_pvt);
audiohook->trans_pvt = NULL;
}
/* Setup new translation path for this format... if we fail we can't very well return signed linear so free the frame and return nothing */
- if (!(audiohook->trans_pvt = ast_translator_build_path(format, ast_format_set(&tmp_fmt, ast_format_slin_by_rate(audiohook->hook_internal_samp_rate), 0)))) {
+ if (!(audiohook->trans_pvt = ast_translator_build_path(format, slin))) {
ast_frfree(read_frame);
+ ao2_ref(slin, -1);
return NULL;
}
- ast_format_copy(&audiohook->format, format);
+ ao2_ref(audiohook->format, -1);
+ audiohook->format = ast_format_copy(format);
}
/* Convert to requested format, and allow the read in frame to be freed */
final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1);
} else {
final_frame = read_frame;
}
+
+ ao2_ref(slin, -1);
return final_frame;
}
@@ -699,8 +725,7 @@
struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ?
&audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]);
struct ast_frame *new_frame = frame;
- struct ast_format tmp_fmt;
- enum ast_format_id slin_id;
+ struct ast_format *slin;
/* If we are capable of maintaining doing samplerates other that 8khz, update
* the internal audiohook_list's rate and higher samplerate audio arrives. By
@@ -708,24 +733,29 @@
* as the are written and read from. */
if (audiohook_list->native_slin_compatible) {
audiohook_list->list_internal_samp_rate =
- MAX(ast_format_rate(&frame->subclass.format), audiohook_list->list_internal_samp_rate);
- }
-
- slin_id = ast_format_slin_by_rate(audiohook_list->list_internal_samp_rate);
-
- if (frame->subclass.format.id == slin_id) {
+ MAX(frame->subclass.format->codec->sample_rate, audiohook_list->list_internal_samp_rate);
+ }
+
+ slin = ast_format_cache_get_slin_by_rate(audiohook_list->list_internal_samp_rate);
+ if (ast_format_cmp(frame->subclass.format, slin) == AST_FORMAT_CMP_EQUAL) {
+ ao2_ref(slin, -1);
return new_frame;
}
- if (ast_format_cmp(&frame->subclass.format, &in_translate->format) == AST_FORMAT_CMP_NOT_EQUAL) {
+ if (ast_format_cmp(frame->subclass.format, in_translate->format) == AST_FORMAT_CMP_NOT_EQUAL) {
if (in_translate->trans_pvt) {
ast_translator_free_path(in_translate->trans_pvt);
}
- if (!(in_translate->trans_pvt = ast_translator_build_path(ast_format_set(&tmp_fmt, slin_id, 0), &frame->subclass.format))) {
+ if (!(in_translate->trans_pvt = ast_translator_build_path(slin, frame->subclass.format))) {
+ ao2_ref(slin, -1);
return NULL;
}
- ast_format_copy(&in_translate->format, &frame->subclass.format);
- }
+ ao2_ref(in_translate->format, -1);
+ in_translate->format = ast_format_copy(frame->subclass.format);
+ }
+
+ ao2_ref(slin, -1);
+
if (!(new_frame = ast_translate(in_translate->trans_pvt, frame, 0))) {
return NULL;
}
@@ -738,16 +768,17 @@
{
struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
struct ast_frame *outframe = NULL;
- if (ast_format_cmp(&slin_frame->subclass.format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) {
+ if (ast_format_cmp(slin_frame->subclass.format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) {
/* rebuild translators if necessary */
- if (ast_format_cmp(&out_translate->format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) {
+ if (ast_format_cmp(out_translate->format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) {
if (out_translate->trans_pvt) {
ast_translator_free_path(out_translate->trans_pvt);
}
- if (!(out_translate->trans_pvt = ast_translator_build_path(outformat, &slin_frame->subclass.format))) {
+ if (!(out_translate->trans_pvt = ast_translator_build_path(outformat, slin_frame->subclass.format))) {
return NULL;
}
- ast_format_copy(&out_translate->format, outformat);
+ ao2_ref(out_translate->format, -1);
+ out_translate->format = ast_format_copy(outformat);
}
/* translate back to the format the frame came in as. */
if (!(outframe = ast_translate(out_translate->trans_pvt, slin_frame, 0))) {
@@ -872,7 +903,7 @@
/* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */
if (middle_frame_manipulated) {
- if (!(end_frame = audiohook_list_translate_to_native(audiohook_list, direction, middle_frame, &start_frame->subclass.format))) {
+ if (!(end_frame = audiohook_list_translate_to_native(audiohook_list, direction, middle_frame, start_frame->subclass.format))) {
/* translation failed, so just pass back the input frame */
end_frame = start_frame;
}
Modified: team/group/media_formats/main/bridge.c
URL: http://svnview.digium.com/svn/asterisk/team/group/media_formats/main/bridge.c?view=diff&rev=408268&r1=408267&r2=408268
==============================================================================
--- team/group/media_formats/main/bridge.c (original)
+++ team/group/media_formats/main/bridge.c Mon Feb 17 08:16:47 2014
@@ -925,61 +925,61 @@
static int bridge_make_compatible(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
{
- struct ast_format read_format;
- struct ast_format write_format;
- struct ast_format best_format;
+ struct ast_format *read_format;
+ struct ast_format *write_format;
+ struct ast_format *best_format;
char codec_buf[512];
- ast_format_copy(&read_format, ast_channel_readformat(bridge_channel->chan));
- ast_format_copy(&write_format, ast_channel_writeformat(bridge_channel->chan));
+ read_format = ast_channel_readformat(bridge_channel->chan);
+ write_format = ast_channel_writeformat(bridge_channel->chan);
/* Are the formats currently in use something this bridge can handle? */
- if (!ast_format_cap_iscompatible(bridge->technology->format_capabilities, ast_channel_readformat(bridge_channel->chan))) {
- ast_best_codec(bridge->technology->format_capabilities, &best_format);
+ if (!ast_format_cap_iscompatible_format(bridge->technology->format_capabilities, read_format)) {
+ best_format = ast_format_cap_get_format(bridge->technology->format_capabilities, 0);
/* Read format is a no go... */
ast_debug(1, "Bridge technology %s wants to read any of formats %s but channel has %s\n",
bridge->technology->name,
ast_getformatname_multiple(codec_buf, sizeof(codec_buf), bridge->technology->format_capabilities),
- ast_getformatname(&read_format));
+ read_format->codec->name);
/* Switch read format to the best one chosen */
- if (ast_set_read_format(bridge_channel->chan, &best_format)) {
+ if (ast_set_read_format(bridge_channel->chan, best_format)) {
ast_log(LOG_WARNING, "Failed to set channel %s to read format %s\n",
- ast_channel_name(bridge_channel->chan), ast_getformatname(&best_format));
+ ast_channel_name(bridge_channel->chan), best_format->codec->name);
return -1;
}
ast_debug(1, "Bridge %s put channel %s into read format %s\n",
bridge->uniqueid, ast_channel_name(bridge_channel->chan),
- ast_getformatname(&best_format));
+ best_format->codec->name);
} else {
ast_debug(1, "Bridge %s is happy that channel %s already has read format %s\n",
bridge->uniqueid, ast_channel_name(bridge_channel->chan),
- ast_getformatname(&read_format));
- }
-
- if (!ast_format_cap_iscompatible(bridge->technology->format_capabilities, &write_format)) {
- ast_best_codec(bridge->technology->format_capabilities, &best_format);
+ read_format->codec->name);
+ }
+
+ if (!ast_format_cap_iscompatible_format(bridge->technology->format_capabilities, write_format)) {
+ best_format = ast_format_cap_get_format(bridge->technology->format_capabilities, 0);
/* Write format is a no go... */
ast_debug(1, "Bridge technology %s wants to write any of formats %s but channel has %s\n",
bridge->technology->name,
ast_getformatname_multiple(codec_buf, sizeof(codec_buf), bridge->technology->format_capabilities),
- ast_getformatname(&write_format));
+ write_format->codec->name);
/* Switch write format to the best one chosen */
- if (ast_set_write_format(bridge_channel->chan, &best_format)) {
+ if (ast_set_write_format(bridge_channel->chan, best_format)) {
ast_log(LOG_WARNING, "Failed to set channel %s to write format %s\n",
- ast_channel_name(bridge_channel->chan), ast_getformatname(&best_format));
+ ast_channel_name(bridge_channel->chan), best_format->codec->name);
return -1;
}
ast_debug(1, "Bridge %s put channel %s into write format %s\n",
bridge->uniqueid, ast_channel_name(bridge_channel->chan),
- ast_getformatname(&best_format));
+ best_format->codec->name);
} else {
ast_debug(1, "Bridge %s is happy that channel %s already has write format %s\n",
bridge->uniqueid, ast_channel_name(bridge_channel->chan),
- ast_getformatname(&write_format));
+ write_format->codec->name);
}
return 0;
@@ -3458,7 +3458,7 @@
struct ast_bridge_video_talker_src_data *data;
/* If the channel doesn't support video, we don't care about it */
- if (!ast_format_cap_has_type(ast_channel_nativeformats(chan), AST_FORMAT_TYPE_VIDEO)) {
+ if (!ast_format_cap_has_type(ast_channel_nativeformats(chan), AST_MEDIA_TYPE_VIDEO)) {
return;
}
Modified: team/group/media_formats/main/bridge_basic.c
URL: http://svnview.digium.com/svn/asterisk/team/group/media_formats/main/bridge_basic.c?view=diff&rev=408268&r1=408267&r2=408268
==============================================================================
--- team/group/media_formats/main/bridge_basic.c (original)
+++ team/group/media_formats/main/bridge_basic.c Mon Feb 17 08:16:47 2014
@@ -46,6 +46,7 @@
#include "asterisk/dial.h"
#include "asterisk/stasis_bridges.h"
#include "asterisk/features.h"
+#include "asterisk/format_cache.h"
#define NORMAL_FLAGS (AST_BRIDGE_FLAG_DISSOLVE_HANGUP | AST_BRIDGE_FLAG_DISSOLVE_EMPTY \
| AST_BRIDGE_FLAG_SMART)
@@ -2218,14 +2219,13 @@
static int recalling_enter(struct attended_transfer_properties *props)
{
- RAII_VAR(struct ast_format_cap *, cap, ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK), ast_format_cap_destroy);
- struct ast_format fmt;
+ RAII_VAR(struct ast_format_cap *, cap, ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT), ao2_cleanup);
if (!cap) {
return -1;
}
- ast_format_cap_add(cap, ast_format_set(&fmt, AST_FORMAT_SLINEAR, 0));
+ ast_format_cap_add(cap, ast_format_slin, 0);
/* When we dial the transfer target, since we are communicating
* with a local channel, we can place the local channel in a bridge
@@ -2346,8 +2346,7 @@
static int retransfer_enter(struct attended_transfer_properties *props)
{
- RAII_VAR(struct ast_format_cap *, cap, ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK), ast_format_cap_destroy);
- struct ast_format fmt;
+ RAII_VAR(struct ast_format_cap *, cap, ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT), ao2_cleanup);
char destination[AST_MAX_EXTENSION + AST_MAX_CONTEXT + 2];
int cause;
@@ -2357,7 +2356,7 @@
snprintf(destination, sizeof(destination), "%s@%s", props->exten, props->context);
- ast_format_cap_add(cap, ast_format_set(&fmt, AST_FORMAT_SLINEAR, 0));
+ ast_format_cap_add(cap, ast_format_slin, 0);
/* Get a channel that is the destination we wish to call */
props->recall_target = ast_request("Local", cap, NULL, destination, &cause);
Modified: team/group/media_formats/main/bridge_channel.c
URL: http://svnview.digium.com/svn/asterisk/team/group/media_formats/main/bridge_channel.c?view=diff&rev=408268&r1=408267&r2=408268
==============================================================================
--- team/group/media_formats/main/bridge_channel.c (original)
+++ team/group/media_formats/main/bridge_channel.c Mon Feb 17 08:16:47 2014
@@ -184,24 +184,24 @@
void ast_bridge_channel_restore_formats(struct ast_bridge_channel *bridge_channel)
{
/* Restore original formats of the channel as they came in */
- if (ast_format_cmp(ast_channel_readformat(bridge_channel->chan), &bridge_channel->read_format) == AST_FORMAT_CMP_NOT_EQUAL) {
+ if (ast_format_cmp(ast_channel_readformat(bridge_channel->chan), bridge_channel->read_format) == AST_FORMAT_CMP_NOT_EQUAL) {
ast_debug(1, "Bridge is returning %p(%s) to read format %s\n",
bridge_channel, ast_channel_name(bridge_channel->chan),
- ast_getformatname(&bridge_channel->read_format));
- if (ast_set_read_format(bridge_channel->chan, &bridge_channel->read_format)) {
+ bridge_channel->read_format->codec->name);
+ if (ast_set_read_format(bridge_channel->chan, bridge_channel->read_format)) {
ast_debug(1, "Bridge failed to return %p(%s) to read format %s\n",
bridge_channel, ast_channel_name(bridge_channel->chan),
- ast_getformatname(&bridge_channel->read_format));
- }
- }
- if (ast_format_cmp(ast_channel_writeformat(bridge_channel->chan), &bridge_channel->write_format) == AST_FORMAT_CMP_NOT_EQUAL) {
+ bridge_channel->read_format->codec->name);
+ }
+ }
+ if (ast_format_cmp(ast_channel_writeformat(bridge_channel->chan), bridge_channel->write_format) == AST_FORMAT_CMP_NOT_EQUAL) {
ast_debug(1, "Bridge is returning %p(%s) to write format %s\n",
bridge_channel, ast_channel_name(bridge_channel->chan),
- ast_getformatname(&bridge_channel->write_format));
- if (ast_set_write_format(bridge_channel->chan, &bridge_channel->write_format)) {
+ bridge_channel->write_format->codec->name);
+ if (ast_set_write_format(bridge_channel->chan, bridge_channel->write_format)) {
ast_debug(1, "Bridge failed to return %p(%s) to write format %s\n",
bridge_channel, ast_channel_name(bridge_channel->chan),
- ast_getformatname(&bridge_channel->write_format));
+ bridge_channel->write_format->codec->name);
}
}
}
@@ -1969,8 +1969,8 @@
{
int res = 0;
- ast_format_copy(&bridge_channel->read_format, ast_channel_readformat(bridge_channel->chan));
- ast_format_copy(&bridge_channel->write_format, ast_channel_writeformat(bridge_channel->chan));
+ bridge_channel->read_format = ast_format_copy(ast_channel_readformat(bridge_channel->chan));
+ bridge_channel->write_format = ast_format_copy(ast_channel_writeformat(bridge_channel->chan));
ast_debug(1, "Bridge %s: %p(%s) is joining\n",
bridge_channel->bridge->uniqueid,
@@ -2211,6 +2211,9 @@
pipe_close(bridge_channel->alert_pipe);
ast_cond_destroy(&bridge_channel->cond);
+
+ ao2_cleanup(bridge_channel->write_format);
+ ao2_cleanup(bridge_channel->read_format);
}
struct ast_bridge_channel *bridge_channel_internal_alloc(struct ast_bridge *bridge)
Modified: team/group/media_formats/main/callerid.c
URL: http://svnview.digium.com/svn/asterisk/team/group/media_formats/main/callerid.c?view=diff&rev=408268&r1=408267&r2=408268
==============================================================================
--- team/group/media_formats/main/callerid.c (original)
+++ team/group/media_formats/main/callerid.c Mon Feb 17 08:16:47 2014
@@ -42,6 +42,7 @@
#include "asterisk/callerid.h"
#include "asterisk/fskmodem.h"
#include "asterisk/utils.h"
+#include "asterisk/format_cache.h"
struct callerid_state {
fsk_data fskd;
Modified: team/group/media_formats/main/ccss.c
URL: http://svnview.digium.com/svn/asterisk/team/group/media_formats/main/ccss.c?view=diff&rev=408268&r1=408267&r2=408268
==============================================================================
--- team/group/media_formats/main/ccss.c (original)
+++ team/group/media_formats/main/ccss.c Mon Feb 17 08:16:47 2014
@@ -52,6 +52,7 @@
#include "asterisk/manager.h"
#include "asterisk/causes.h"
#include "asterisk/stasis_system.h"
+#include "asterisk/format_cache.h"
/*** DOCUMENTATION
<application name="CallCompletionRequest" language="en_US">
@@ -2814,8 +2815,7 @@
const char *callback_macro = ast_get_cc_callback_macro(agent->cc_params);
const char *callback_sub = ast_get_cc_callback_sub(agent->cc_params);
unsigned int recall_timer = ast_get_cc_recall_timer(agent->cc_params) * 1000;
- struct ast_format tmp_fmt;
- struct ast_format_cap *tmp_cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
+ struct ast_format_cap *tmp_cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!tmp_cap) {
return NULL;
@@ -2826,17 +2826,17 @@
*target++ = '\0';
}
- ast_format_cap_add(tmp_cap, ast_format_set(&tmp_fmt, AST_FORMAT_SLINEAR, 0));
+ ast_format_cap_add(tmp_cap, ast_format_slin, 0);
if (!(chan = ast_request_and_dial(tech, tmp_cap, NULL, target, recall_timer, &reason, generic_pvt->cid_num, generic_pvt->cid_name))) {
/* Hmm, no channel. Sucks for you, bud.
*/
ast_log_dynamic_level(cc_logger_level, "Core %d: Failed to call back %s for reason %d\n",
agent->core_id, agent->device_name, reason);
ast_cc_failed(agent->core_id, "Failed to call back device %s/%s", tech, target);
- ast_format_cap_destroy(tmp_cap);
+ ao2_ref(tmp_cap, -1);
return NULL;
}
- ast_format_cap_destroy(tmp_cap);
+ ao2_ref(tmp_cap, -1);
/* We have a channel. It's time now to set up the datastore of recalled CC interfaces.
* This will be a common task for all recall functions. If it were possible, I'd have
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