diff --git a/include/asterisk/codec.h b/include/asterisk/codec.h
index 2f5756cd14f2043936a5d167c4a6bec832f66364..79798acd085b89a1966dad4659b2c21f92464c44 100644
--- a/include/asterisk/codec.h
+++ b/include/asterisk/codec.h
@@ -76,6 +76,8 @@ struct ast_codec {
int (*get_length)(unsigned int samples);
/*! \brief Whether the media can be smoothed or not */
unsigned int smooth;
+ /*! \brief Flags to be passed to the smoother */
+ unsigned int smoother_flags;
/*! \brief The module that registered this codec */
struct ast_module *mod;
};
diff --git a/include/asterisk/format.h b/include/asterisk/format.h
index b01592d16e9a3197a7cf62441f06f37247c21d84..0bad96dccc1b9b5fe70ac13b9ef32bed3b35ed2d 100644
--- a/include/asterisk/format.h
+++ b/include/asterisk/format.h
@@ -355,6 +355,17 @@ const char *ast_format_get_codec_name(const struct ast_format *format);
*/
int ast_format_can_be_smoothed(const struct ast_format *format);
+/*!
+ * \since 13.17.0
+ *
+ * \brief Get smoother flags for this format
+ *
+ * \param format The media format
+ *
+ * \return smoother flags for the provided format
+ */
+int ast_format_get_smoother_flags(const struct ast_format *format);
+
/*!
* \brief Get the media type of a format
*
diff --git a/include/asterisk/smoother.h b/include/asterisk/smoother.h
index e63aa77bd684ca8c6324373a337d25f7a672ec83..65ac88921fa86998db662d0443c580f318b53c18 100644
--- a/include/asterisk/smoother.h
+++ b/include/asterisk/smoother.h
@@ -33,6 +33,7 @@ extern "C" {
#define AST_SMOOTHER_FLAG_G729 (1 << 0)
#define AST_SMOOTHER_FLAG_BE (1 << 1)
+#define AST_SMOOTHER_FLAG_FORCED (1 << 2)
/*! \name AST_Smoother
*/
diff --git a/main/codec_builtin.c b/main/codec_builtin.c
index 3320900c2e34f9a1e897284cb09458c20ddf0735..32ec12d3d428fcd69ddff73ea7533c58b166f8f5 100644
--- a/main/codec_builtin.c
+++ b/main/codec_builtin.c
@@ -36,6 +36,7 @@
#include "asterisk/format.h"
#include "asterisk/format_cache.h"
#include "asterisk/frame.h"
+#include "asterisk/smoother.h"
int __ast_codec_register_with_format(struct ast_codec *codec, const char *format_name,
struct ast_module *mod);
@@ -288,6 +289,7 @@ static struct ast_codec slin8 = {
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
+ .smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin12 = {
@@ -302,6 +304,7 @@ static struct ast_codec slin12 = {
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
+ .smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin16 = {
@@ -316,6 +319,7 @@ static struct ast_codec slin16 = {
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
+ .smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin24 = {
@@ -330,6 +334,7 @@ static struct ast_codec slin24 = {
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
+ .smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin32 = {
@@ -344,6 +349,7 @@ static struct ast_codec slin32 = {
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
+ .smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin44 = {
@@ -358,6 +364,7 @@ static struct ast_codec slin44 = {
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
+ .smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin48 = {
@@ -372,6 +379,7 @@ static struct ast_codec slin48 = {
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
+ .smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin96 = {
@@ -386,6 +394,7 @@ static struct ast_codec slin96 = {
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
+ .smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static struct ast_codec slin192 = {
@@ -400,6 +409,7 @@ static struct ast_codec slin192 = {
.samples_count = slin_samples,
.get_length = slin_length,
.smooth = 1,
+ .smoother_flags = AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED,
};
static int lpc10_samples(struct ast_frame *frame)
diff --git a/main/format.c b/main/format.c
index 5ae5ad98623bd60bb4e56d4599850d5961d01509..09e736cf5f49017583b8a757517efe0a76328941 100644
--- a/main/format.c
+++ b/main/format.c
@@ -391,6 +391,11 @@ int ast_format_can_be_smoothed(const struct ast_format *format)
return format->codec->smooth;
}
+int ast_format_get_smoother_flags(const struct ast_format *format)
+{
+ return format->codec->smoother_flags;
+}
+
enum ast_media_type ast_format_get_type(const struct ast_format *format)
{
return format->codec->type;
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 18987cee9eb95daffab57eb4fbea0ac197fb7076..c120fc1452eb98e1be3cbba8ba3d76ecc791bfe1 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -3747,7 +3747,7 @@ static int ast_rtcp_write(const void *data)
}
/*! \pre instance is locked */
-static int ast_rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *frame, int codec)
+static int rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *frame, int codec)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
int pred, mark = 0;
@@ -4016,10 +4016,10 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
/* If no smoother is present see if we have to set one up */
if (!rtp->smoother && ast_format_can_be_smoothed(format)) {
+ unsigned int smoother_flags = ast_format_get_smoother_flags(format);
unsigned int framing_ms = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(instance));
- int is_slinear = ast_format_cache_is_slinear(format);
- if (!framing_ms && is_slinear) {
+ if (!framing_ms && (smoother_flags & AST_SMOOTHER_FLAG_FORCED)) {
framing_ms = ast_format_get_default_ms(format);
}
@@ -4030,9 +4030,7 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
ast_format_get_name(format), framing_ms, ast_format_get_minimum_bytes(format));
return -1;
}
- if (is_slinear) {
- ast_smoother_set_flags(rtp->smoother, AST_SMOOTHER_FLAG_BE);
- }
+ ast_smoother_set_flags(rtp->smoother, smoother_flags);
}
}
@@ -4047,7 +4045,7 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
}
while ((f = ast_smoother_read(rtp->smoother)) && (f->data.ptr)) {
- ast_rtp_raw_write(instance, f, codec);
+ rtp_raw_write(instance, f, codec);
}
} else {
int hdrlen = 12;
@@ -4059,7 +4057,7 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
f = frame;
}
if (f->data.ptr) {
- ast_rtp_raw_write(instance, f, codec);
+ rtp_raw_write(instance, f, codec);
}
if (f != frame) {
ast_frfree(f);
diff --git a/res/res_rtp_multicast.c b/res/res_rtp_multicast.c
index 42de11f65e8239fbcd3e0e065c4ff6755a981552..14176da413ad697fcec73873e1d312196e555303 100644
--- a/res/res_rtp_multicast.c
+++ b/res/res_rtp_multicast.c
@@ -54,6 +54,7 @@
#include "asterisk/format_cache.h"
#include "asterisk/multicast_rtp.h"
#include "asterisk/app.h"
+#include "asterisk/smoother.h"
/*! Command value used for Linksys paging to indicate we are starting */
#define LINKSYS_MCAST_STARTCMD 6
@@ -95,6 +96,7 @@ struct multicast_rtp {
uint16_t seqno;
unsigned int lastts;
struct timeval txcore;
+ struct ast_smoother *smoother;
};
enum {
@@ -395,6 +397,10 @@ static int multicast_rtp_destroy(struct ast_rtp_instance *instance)
multicast_send_control_packet(instance, multicast, LINKSYS_MCAST_STOPCMD);
}
+ if (multicast->smoother) {
+ ast_smoother_free(multicast->smoother);
+ }
+
close(multicast->socket);
ast_free(multicast);
@@ -402,43 +408,24 @@ static int multicast_rtp_destroy(struct ast_rtp_instance *instance)
return 0;
}
-/*! \brief Function called to broadcast some audio on a multicast instance */
-static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
+static int rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *frame, int codec)
{
struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);
- struct ast_frame *f = frame;
- struct ast_sockaddr remote_address;
- int hdrlen = 12, res = 0, codec;
- unsigned char *rtpheader;
unsigned int ms = calc_txstamp(multicast, &frame->delivery);
+ unsigned char *rtpheader;
+ struct ast_sockaddr remote_address = { {0,} };
int rate = rtp_get_rate(frame->subclass.format) / 1000;
+ int hdrlen = 12;
- /* We only accept audio, nothing else */
- if (frame->frametype != AST_FRAME_VOICE) {
- return 0;
- }
-
- /* Grab the actual payload number for when we create the RTP packet */
- codec = ast_rtp_codecs_payload_code_tx(ast_rtp_instance_get_codecs(instance),
- 1, frame->subclass.format, 0);
- if (codec < 0) {
- return -1;
- }
-
- /* If we do not have space to construct an RTP header duplicate the frame so we get some */
- if (frame->offset < hdrlen) {
- f = ast_frdup(frame);
- }
-
- /* Calucate last TS */
+ /* Calculate last TS */
multicast->lastts = multicast->lastts + ms * rate;
-
+
/* Construct an RTP header for our packet */
- rtpheader = (unsigned char *)(f->data.ptr - hdrlen);
+ rtpheader = (unsigned char *)(frame->data.ptr - hdrlen);
put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (multicast->seqno)));
-
- if (ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO)) {
- put_unaligned_uint32(rtpheader + 4, htonl(f->ts * 8));
+
+ if (ast_test_flag(frame, AST_FRFLAG_HAS_TIMING_INFO)) {
+ put_unaligned_uint32(rtpheader + 4, htonl(frame->ts * 8));
} else {
put_unaligned_uint32(rtpheader + 4, htonl(multicast->lastts));
}
@@ -451,19 +438,86 @@ static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_fra
/* Finally send it out to the eager phones listening for us */
ast_rtp_instance_get_remote_address(instance, &remote_address);
- if (ast_sendto(multicast->socket, (void *) rtpheader, f->datalen + hdrlen, 0, &remote_address) < 0) {
+ if (ast_sendto(multicast->socket, (void *) rtpheader, frame->datalen + hdrlen, 0, &remote_address) < 0) {
ast_log(LOG_ERROR, "Multicast RTP Transmission error to %s: %s\n",
ast_sockaddr_stringify(&remote_address),
strerror(errno));
- res = -1;
+ return -1;
}
- /* If we were forced to duplicate the frame free the new one */
- if (frame != f) {
- ast_frfree(f);
+ return 0;
+}
+
+/*! \brief Function called to broadcast some audio on a multicast instance */
+static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
+{
+ struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);
+ struct ast_format *format;
+ struct ast_frame *f;
+ int codec;
+
+ /* We only accept audio, nothing else */
+ if (frame->frametype != AST_FRAME_VOICE) {
+ return 0;
}
- return res;
+ /* Grab the actual payload number for when we create the RTP packet */
+ codec = ast_rtp_codecs_payload_code_tx(ast_rtp_instance_get_codecs(instance),
+ 1, frame->subclass.format, 0);
+ if (codec < 0) {
+ return -1;
+ }
+
+ format = frame->subclass.format;
+ if (!multicast->smoother && ast_format_can_be_smoothed(format)) {
+ unsigned int smoother_flags = ast_format_get_smoother_flags(format);
+ unsigned int framing_ms = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(instance));
+
+ if (!framing_ms && (smoother_flags & AST_SMOOTHER_FLAG_FORCED)) {
+ framing_ms = ast_format_get_default_ms(format);
+ }
+
+ if (framing_ms) {
+ multicast->smoother = ast_smoother_new((framing_ms * ast_format_get_minimum_bytes(format)) / ast_format_get_minimum_ms(format));
+ if (!multicast->smoother) {
+ ast_log(LOG_WARNING, "Unable to create smoother: format %s ms: %u len %u\n",
+ ast_format_get_name(format), framing_ms, ast_format_get_minimum_bytes(format));
+ return -1;
+ }
+ ast_smoother_set_flags(multicast->smoother, smoother_flags);
+ }
+ }
+
+ if (multicast->smoother) {
+ if (ast_smoother_test_flag(multicast->smoother, AST_SMOOTHER_FLAG_BE)) {
+ ast_smoother_feed_be(multicast->smoother, frame);
+ } else {
+ ast_smoother_feed(multicast->smoother, frame);
+ }
+
+ while ((f = ast_smoother_read(multicast->smoother)) && f->data.ptr) {
+ rtp_raw_write(instance, f, codec);
+ }
+ } else {
+ int hdrlen = 12;
+
+ /* If we do not have space to construct an RTP header duplicate the frame so we get some */
+ if (frame->offset < hdrlen) {
+ f = ast_frdup(frame);
+ } else {
+ f = frame;
+ }
+
+ if (f->data.ptr) {
+ rtp_raw_write(instance, f, codec);
+ }
+
+ if (f != frame) {
+ ast_frfree(f);
+ }
+ }
+
+ return 0;
}
/*! \brief Function called to read from a multicast instance */