[svn-commits] tilghman: branch tilghman/codec_bits3 r227544 - in /team/tilghman/codec_bits3...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Nov 3 22:01:44 CST 2009
Author: tilghman
Date: Tue Nov 3 22:01:40 2009
New Revision: 227544
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=227544
Log:
Make SIP also work against extended codec bits
Modified:
team/tilghman/codec_bits3/channels/chan_sip.c
team/tilghman/codec_bits3/include/asterisk/rtp_engine.h
team/tilghman/codec_bits3/main/dsp.c
team/tilghman/codec_bits3/main/rtp_engine.c
Modified: team/tilghman/codec_bits3/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/channels/chan_sip.c?view=diff&rev=227544&r1=227543&r2=227544
==============================================================================
--- team/tilghman/codec_bits3/channels/chan_sip.c (original)
+++ team/tilghman/codec_bits3/channels/chan_sip.c Tue Nov 3 22:01:40 2009
@@ -1146,7 +1146,7 @@
#define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
#define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */
-#define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263);
+#define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_TESTLAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263);
#endif
/*@}*/
@@ -2501,7 +2501,7 @@
static const char *get_sdp(struct sip_request *req, const char *name);
static int find_sdp(struct sip_request *req);
static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action);
-static void add_codec_to_sdp(const struct sip_pvt *p, int codec,
+static void add_codec_to_sdp(const struct sip_pvt *p, format_t codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size);
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
@@ -6459,7 +6459,7 @@
/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */
static void try_suggested_sip_codec(struct sip_pvt *p)
{
- int fmt;
+ format_t fmt;
const char *codec;
struct ast_channel* chan;
@@ -6914,10 +6914,10 @@
{
struct ast_channel *tmp;
struct ast_variable *v = NULL;
- int fmt;
- int what;
- int video;
- int text;
+ format_t fmt;
+ format_t what;
+ format_t video;
+ format_t text;
format_t needvideo = 0;
int needtext = 0;
char buf[SIPBUFSIZE];
@@ -10185,7 +10185,7 @@
}
/*! \brief Add codec offer to SDP offer/answer body in INVITE or 200 OK */
-static void add_codec_to_sdp(const struct sip_pvt *p, int codec,
+static void add_codec_to_sdp(const struct sip_pvt *p, format_t codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size)
{
@@ -10194,7 +10194,7 @@
if (debug)
- ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
+ ast_verbose("Adding codec 0x%Lx (%s) to SDP\n", codec, ast_getformatname(codec));
if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, codec)) == -1)
return;
@@ -10392,7 +10392,7 @@
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38)
{
int len = 0;
- int alreadysent = 0;
+ format_t alreadysent = 0;
struct sockaddr_in sin = { 0, };
struct sockaddr_in vsin = { 0, };
@@ -10420,8 +10420,8 @@
struct ast_str *a_text = ast_str_alloca(1024); /* Attributes for text */
struct ast_str *a_modem = ast_str_alloca(1024); /* Attributes for modem */
- int x;
- int capability = 0;
+ format_t x;
+ format_t capability = 0;
int needaudio = FALSE;
int needvideo = FALSE;
int needtext = FALSE;
@@ -10543,15 +10543,15 @@
Note that p->prefcodec can include video codecs, so mask them out
*/
if (capability & p->prefcodec) {
- int codec = p->prefcodec & AST_FORMAT_AUDIO_MASK;
+ format_t codec = p->prefcodec & AST_FORMAT_AUDIO_MASK;
add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
alreadysent |= codec;
}
/* Start by sending our preferred audio/video codecs */
- for (x = 0; x < 32; x++) {
- int codec;
+ for (x = 0; x < 64; x++) {
+ format_t codec;
if (!(codec = ast_codec_pref_index(&p->prefs, x)))
break;
@@ -10567,7 +10567,7 @@
}
/* Now send any other common audio and video codecs, and non-codec formats: */
- for (x = 1; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) {
+ for (x = 1LL; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) {
if (!(capability & x)) /* Codec not requested */
continue;
@@ -10583,7 +10583,7 @@
}
/* Now add DTMF RFC2833 telephony-event as a codec */
- for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
+ for (x = 1LL; x <= AST_RTP_MAX; x <<= 1) {
if (!(p->jointnoncodeccapability & x))
continue;
@@ -15665,9 +15665,10 @@
/*! \brief Print codec list from preference to CLI/manager */
static void print_codec_to_cli(int fd, struct ast_codec_pref *pref)
{
- int x, codec;
-
- for(x = 0; x < 32 ; x++) {
+ int x;
+ format_t codec;
+
+ for(x = 0; x < 64 ; x++) {
codec = ast_codec_pref_index(pref, x);
if (!codec)
break;
@@ -15860,7 +15861,8 @@
struct ast_codec_pref *pref;
struct ast_variable *v;
struct sip_auth *auth;
- int x = 0, codec = 0, load_realtime;
+ int x = 0, load_realtime;
+ format_t codec = 0;
int realtimepeers;
realtimepeers = ast_check_realtime("sippeers");
@@ -16067,12 +16069,12 @@
astman_append(s, "%s\r\n", codec_buf);
astman_append(s, "CodecOrder: ");
pref = &peer->prefs;
- for(x = 0; x < 32 ; x++) {
+ for(x = 0; x < 64 ; x++) {
codec = ast_codec_pref_index(pref, x);
if (!codec)
break;
astman_append(s, "%s", ast_getformatname(codec));
- if (x < 31 && ast_codec_pref_index(pref, x+1))
+ if (x < 63 && ast_codec_pref_index(pref, x+1))
astman_append(s, ",");
}
@@ -17862,7 +17864,7 @@
}
} else if (!strncasecmp(colname, "codec[", 6)) {
char *codecnum;
- int codec = 0;
+ format_t codec = 0;
codecnum = colname + 6; /* move past the '[' */
codecnum = strsep(&codecnum, "]"); /* trim trailing ']' if any */
Modified: team/tilghman/codec_bits3/include/asterisk/rtp_engine.h
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/include/asterisk/rtp_engine.h?view=diff&rev=227544&r1=227543&r2=227544
==============================================================================
--- team/tilghman/codec_bits3/include/asterisk/rtp_engine.h (original)
+++ team/tilghman/codec_bits3/include/asterisk/rtp_engine.h Tue Nov 3 22:01:40 2009
@@ -994,7 +994,7 @@
*
* \since 1.6.3
*/
-unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, int code);
+unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, format_t code);
/*!
* \brief Retrieve all formats that were found
Modified: team/tilghman/codec_bits3/main/dsp.c
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/main/dsp.c?view=diff&rev=227544&r1=227543&r2=227544
==============================================================================
--- team/tilghman/codec_bits3/main/dsp.c (original)
+++ team/tilghman/codec_bits3/main/dsp.c Tue Nov 3 22:01:40 2009
@@ -1321,6 +1321,7 @@
len = af->datalen / 2;
break;
case AST_FORMAT_ULAW:
+ case AST_FORMAT_TESTLAW:
shortdata = alloca(af->datalen * 2);
for (x = 0;x < len; x++) {
shortdata[x] = AST_MULAW(odata[x]);
Modified: team/tilghman/codec_bits3/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/team/tilghman/codec_bits3/main/rtp_engine.c?view=diff&rev=227544&r1=227543&r2=227544
==============================================================================
--- team/tilghman/codec_bits3/main/rtp_engine.c (original)
+++ team/tilghman/codec_bits3/main/rtp_engine.c Tue Nov 3 22:01:40 2009
@@ -655,7 +655,7 @@
return "";
}
-unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, int code)
+unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, format_t code)
{
unsigned int i;
@@ -670,7 +670,8 @@
char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, const format_t capability, const int asterisk_format, enum ast_rtp_options options)
{
- int format, found = 0;
+ format_t format;
+ int found = 0;
if (!buf) {
return NULL;
@@ -1164,7 +1165,7 @@
enum ast_rtp_glue_result audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID, text_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID, text_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
enum ast_bridge_result res = AST_BRIDGE_FAILED;
- int codec0 = 0, codec1 = 0;
+ format_t codec0 = 0, codec1 = 0;
int unlock_chans = 1;
/* Lock both channels so we can look for the glue that binds them together */
@@ -1229,7 +1230,7 @@
codec0 = glue0->get_codec ? glue0->get_codec(c0) : 0;
codec1 = glue1->get_codec ? glue1->get_codec(c1) : 0;
if (codec0 && codec1 && !(codec0 & codec1)) {
- ast_debug(1, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
+ ast_debug(1, "Channel codec0 = %s is not codec1 = %s, cannot native bridge in RTP.\n", ast_getformatname(codec0), ast_getformatname(codec1));
res = AST_BRIDGE_FAILED_NOWARN;
goto done;
}
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