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-rw-r--r--channels/chan_sip.c54
-rw-r--r--configs/codecs.conf.sample19
-rw-r--r--include/asterisk/format.h2
-rw-r--r--main/channel.c12
-rw-r--r--main/format.c54
-rw-r--r--main/frame.c4
-rw-r--r--main/rtp_engine.c4
-rw-r--r--res/res_rtp_asterisk.c1
8 files changed, 120 insertions, 30 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 9e2ddf7ed..9ab409265 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -9468,6 +9468,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
if ((format = ast_rtp_codecs_get_payload_format(newaudiortp, codec))) {
unsigned int bit_rate;
+ int val = 0;
switch ((int) format->id) {
case AST_FORMAT_SIREN7:
@@ -9500,20 +9501,21 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
}
}
break;
+ case AST_FORMAT_CELT:
+ if (sscanf(fmtp_string, "framesize=%30u", &val) == 1) {
+ ast_format_append(format, CELT_ATTR_KEY_FRAME_SIZE, val, AST_FORMAT_ATTR_END);
+ }
case AST_FORMAT_SILK:
- {
- int val = 0;
- if (sscanf(fmtp_string, "maxaveragebitrate=%30u", &val) == 1) {
- ast_format_append(format, SILK_ATTR_KEY_MAX_BITRATE, val, AST_FORMAT_ATTR_END);
- }
- if (sscanf(fmtp_string, "usedtx=%30u", &val) == 1) {
- ast_format_append(format, SILK_ATTR_KEY_DTX, val ? 1 : 0, AST_FORMAT_ATTR_END);
- }
- if (sscanf(fmtp_string, "useinbandfec=%30u", &val) == 1) {
- ast_format_append(format, SILK_ATTR_KEY_FEC, val ? 1 : 0, AST_FORMAT_ATTR_END);
- }
- break;
+ if (sscanf(fmtp_string, "maxaveragebitrate=%30u", &val) == 1) {
+ ast_format_append(format, SILK_ATTR_KEY_MAX_BITRATE, val, AST_FORMAT_ATTR_END);
+ }
+ if (sscanf(fmtp_string, "usedtx=%30u", &val) == 1) {
+ ast_format_append(format, SILK_ATTR_KEY_DTX, val ? 1 : 0, AST_FORMAT_ATTR_END);
+ }
+ if (sscanf(fmtp_string, "useinbandfec=%30u", &val) == 1) {
+ ast_format_append(format, SILK_ATTR_KEY_FEC, val ? 1 : 0, AST_FORMAT_ATTR_END);
}
+ break;
}
}
}
@@ -10829,7 +10831,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
{
int rtp_code;
struct ast_format_list fmt;
-
+ int val = 0;
if (debug)
ast_verbose("Adding codec %d (%s) to SDP\n", format->id, ast_getformatname(format));
@@ -10872,20 +10874,22 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
/* Indicate that we only expect 64Kbps */
ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=64000\r\n", rtp_code);
break;
+ case AST_FORMAT_CELT:
+ if (!ast_format_get_value(format, CELT_ATTR_KEY_FRAME_SIZE, &val) && val > 0) {
+ ast_str_append(a_buf, 0, "a=fmtp:%d framesize=%u\r\n", rtp_code, val);
+ }
+ break;
case AST_FORMAT_SILK:
- {
- int val = 0;
- if (!ast_format_get_value(format, SILK_ATTR_KEY_MAX_BITRATE, &val) && val > 5000 && val < 40000) {
- ast_str_append(a_buf, 0, "a=fmtp:%d maxaveragebitrate=%u\r\n", rtp_code, val);
- }
- if (!ast_format_get_value(format, SILK_ATTR_KEY_DTX, &val)) {
- ast_str_append(a_buf, 0, "a=fmtp:%d usedtx=%u\r\n", rtp_code, val ? 1 : 0);
- }
- if (!ast_format_get_value(format, SILK_ATTR_KEY_FEC, &val)) {
- ast_str_append(a_buf, 0, "a=fmtp:%d useinbandfec=%u\r\n", rtp_code, val ? 1 : 0);
- }
- break;
+ if (!ast_format_get_value(format, SILK_ATTR_KEY_MAX_BITRATE, &val) && val > 5000 && val < 40000) {
+ ast_str_append(a_buf, 0, "a=fmtp:%d maxaveragebitrate=%u\r\n", rtp_code, val);
+ }
+ if (!ast_format_get_value(format, SILK_ATTR_KEY_DTX, &val)) {
+ ast_str_append(a_buf, 0, "a=fmtp:%d usedtx=%u\r\n", rtp_code, val ? 1 : 0);
}
+ if (!ast_format_get_value(format, SILK_ATTR_KEY_FEC, &val)) {
+ ast_str_append(a_buf, 0, "a=fmtp:%d useinbandfec=%u\r\n", rtp_code, val ? 1 : 0);
+ }
+ break;
}
if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
diff --git a/configs/codecs.conf.sample b/configs/codecs.conf.sample
index 4404d4a0c..9b5738577 100644
--- a/configs/codecs.conf.sample
+++ b/configs/codecs.conf.sample
@@ -126,7 +126,6 @@ maxbitrate=20000
fec=true
packetloss_percentage=10;
-
[silk24]
type=silk
samprate=24000
@@ -134,3 +133,21 @@ maxbitrate=30000
fec=true
packetloss_percentage=10;
+
+; Default custom CELT codec definitions. Only one custom CELT definition is allowed
+; per a sample rate.
+;[celt44]
+;type=celt
+;samprate=44100 ; The samplerate in hz. This option is required.
+;framesize=480 ; The framesize option represents the duration of each frame in samples.
+ ; This must be a factor of 2. This option is only advertised in an SDP
+ ; when it is set. Otherwise a default of framesize of 480 is assumed
+ ; internally
+
+;[celt48]
+;type=celt
+;samprate=48000
+
+;[celt32]
+;type=celt
+;samprate=32000
diff --git a/include/asterisk/format.h b/include/asterisk/format.h
index 67e4178a2..d0f1021d5 100644
--- a/include/asterisk/format.h
+++ b/include/asterisk/format.h
@@ -28,6 +28,7 @@
#include "asterisk/astobj2.h"
#include "asterisk/silk.h"
+#include "asterisk/celt.h"
#define AST_FORMAT_ATTR_SIZE 128
#define AST_FORMAT_INC 100000
@@ -99,6 +100,7 @@ enum ast_format_id {
/*! Raw 16-bit Signed Linear (192000 Hz) PCM. maybe we're taking this too far. */
AST_FORMAT_SLINEAR192 = 27 + AST_FORMAT_TYPE_AUDIO,
AST_FORMAT_SPEEX32 = 28 + AST_FORMAT_TYPE_AUDIO,
+ AST_FORMAT_CELT = 29 + AST_FORMAT_TYPE_AUDIO,
/*! H.261 Video */
AST_FORMAT_H261 = 1 + AST_FORMAT_TYPE_VIDEO,
diff --git a/main/channel.c b/main/channel.c
index ed84da1e6..902c79d85 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -1081,6 +1081,8 @@ struct ast_format *ast_best_codec(struct ast_format_cap *cap, struct ast_format
AST_FORMAT_SPEEX,
/*! SILK is pretty awesome. */
AST_FORMAT_SILK,
+ /*! CELT supports crazy high sample rates */
+ AST_FORMAT_CELT,
/*! Ick, LPC10 sounds terrible, but at least we have code for it, if you're tacky enough
to use it */
AST_FORMAT_LPC10,
@@ -5019,12 +5021,13 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr)
from the single frame we passed in; if so, feed each one of them to the
channel, freeing each one after it has been written */
if ((f != fr) && AST_LIST_NEXT(f, frame_list)) {
- struct ast_frame *cur, *next;
+ struct ast_frame *cur, *next = NULL;
unsigned int skip = 0;
- for (cur = f, next = AST_LIST_NEXT(cur, frame_list);
- cur;
- cur = next, next = cur ? AST_LIST_NEXT(cur, frame_list) : NULL) {
+ cur = f;
+ while (cur) {
+ next = AST_LIST_NEXT(cur, frame_list);
+ AST_LIST_NEXT(cur, frame_list) = NULL;
if (!skip) {
if ((res = chan->tech->write(chan, cur)) < 0) {
chan->_softhangup |= AST_SOFTHANGUP_DEV;
@@ -5037,6 +5040,7 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr)
}
}
ast_frfree(cur);
+ cur = next;
}
/* reset f so the code below doesn't attempt to free it */
diff --git a/main/format.c b/main/format.c
index 28b15ae21..1c60ab770 100644
--- a/main/format.c
+++ b/main/format.c
@@ -748,6 +748,15 @@ int ast_format_rate(const struct ast_format *format)
} else {
return 8000;
}
+ case AST_FORMAT_CELT:
+ {
+ int samplerate;
+ if (!(ast_format_get_value(format,
+ CELT_ATTR_KEY_SAMP_RATE,
+ &samplerate))) {
+ return samplerate;
+ }
+ }
default:
return 8000;
}
@@ -1085,6 +1094,32 @@ init_cleanup:
return -1;
}
+static int custom_celt_format(struct ast_format_list *entry, unsigned int maxbitrate, unsigned int framesize)
+{
+ if (!entry->samplespersecond) {
+ ast_log(LOG_WARNING, "Custom CELT format definition '%s' requires sample rate to be defined.\n", entry->name);
+ }
+ ast_format_set(&entry->format, AST_FORMAT_CELT, 0);
+ if (!has_interface(&entry->format)) {
+ return -1;
+ }
+
+ snprintf(entry->desc, sizeof(entry->desc), "CELT Custom Format %dkhz", entry->samplespersecond/1000);
+
+ ast_format_append(&entry->format,
+ CELT_ATTR_KEY_SAMP_RATE, entry->samplespersecond,
+ CELT_ATTR_KEY_MAX_BITRATE, maxbitrate,
+ CELT_ATTR_KEY_FRAME_SIZE, framesize,
+ AST_FORMAT_ATTR_END);
+
+ entry->fr_len = 80;
+ entry->min_ms = 20;
+ entry->max_ms = 20;
+ entry->inc_ms = 20;
+ entry->def_ms = 20;
+ return 0;
+}
+
static int custom_silk_format(struct ast_format_list *entry, unsigned int maxbitrate, int usedtx, int usefec, int packetloss_percentage)
{
if (!entry->samplespersecond) {
@@ -1144,6 +1179,8 @@ static int conf_process_format_name(const char *name, enum ast_format_id *id)
{
if (!strcasecmp(name, "silk")) {
*id = AST_FORMAT_SILK;
+ } else if (!strcasecmp(name, "celt")) {
+ *id = AST_FORMAT_CELT;
} else {
*id = 0;
return -1;
@@ -1163,8 +1200,14 @@ static int conf_process_sample_rate(const char *rate, unsigned int *result)
*result = 24000;
} else if (!strcasecmp(rate, "32000")) {
*result = 32000;
+ } else if (!strcasecmp(rate, "44100")) {
+ *result = 44100;
} else if (!strcasecmp(rate, "48000")) {
*result = 48000;
+ } else if (!strcasecmp(rate, "96000")) {
+ *result = 96000;
+ } else if (!strcasecmp(rate, "192000")) {
+ *result = 192000;
} else {
*result = 0;
return -1;
@@ -1184,6 +1227,7 @@ static int load_format_config(void)
struct {
enum ast_format_id id;
unsigned int maxbitrate;
+ unsigned int framesize;
unsigned int packetloss_percentage;
int usefec;
int usedtx;
@@ -1221,6 +1265,11 @@ static int load_format_config(void)
ast_log(LOG_WARNING, "maxbitrate '%s' at line %d of %s is not supported.\n",
var->value, var->lineno, FORMAT_CONFIG);
}
+ } else if (!strcasecmp(var->name, "framesize")) {
+ if (sscanf(var->value, "%30u", &settings.framesize) != 1) {
+ ast_log(LOG_WARNING, "framesize '%s' at line %d of %s is not supported.\n",
+ var->value, var->lineno, FORMAT_CONFIG);
+ }
} else if (!strcasecmp(var->name, "dtx")) {
settings.usedtx = ast_true(var->value) ? 1 : 0;
} else if (!strcasecmp(var->name, "fec")) {
@@ -1239,6 +1288,11 @@ static int load_format_config(void)
add_it = 1;
}
break;
+ case AST_FORMAT_CELT:
+ if (!(custom_celt_format(&entry, settings.maxbitrate, settings.framesize))) {
+ add_it = 1;
+ }
+ break;
default:
ast_log(LOG_WARNING, "Can not create custom format %s\n", entry.name);
}
diff --git a/main/frame.c b/main/frame.c
index bb32386ca..b664e2e84 100644
--- a/main/frame.c
+++ b/main/frame.c
@@ -1011,6 +1011,10 @@ int ast_codec_get_samples(struct ast_frame *f)
} else {
return 160;
}
+ case AST_FORMAT_CELT:
+ /* TODO The assumes 20ms delivery right now, which is incorrect */
+ samples = ast_format_rate(&f->subclass.format) / 50;
+ break;
default:
ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(&f->subclass.format));
}
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index a3d9c4d8e..4ad3c9024 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -1931,6 +1931,10 @@ int ast_rtp_engine_load_format(const struct ast_format *format)
set_next_mime_type(format, 0, "audio", "SILK", ast_format_rate(format));
add_static_payload(-1, format, 0);
break;
+ case AST_FORMAT_CELT:
+ set_next_mime_type(format, 0, "audio", "CELT", ast_format_rate(format));
+ add_static_payload(-1, format, 0);
+ break;
default:
break;
}
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 60f7edacf..5948e664c 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -1254,6 +1254,7 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
case AST_FORMAT_SPEEX16:
case AST_FORMAT_SPEEX32:
case AST_FORMAT_SILK:
+ case AST_FORMAT_CELT:
case AST_FORMAT_G723_1:
case AST_FORMAT_SIREN7:
case AST_FORMAT_SIREN14: