[Asterisk-code-review] format: Reintroduce smoother flags (asterisk[13])
Sean Bright
asteriskteam at digium.com
Tue May 30 08:47:01 CDT 2017
Sean Bright has uploaded a new change for review. ( https://gerrit.asterisk.org/5729 )
Change subject: format: Reintroduce smoother flags
......................................................................
format: Reintroduce smoother flags
In review 4843 (ASTERISK-24858), we added a hack that forced a smoother
creation when sending signed linear so that the byte order was adjusted
during transmission. This was needed because smoother flags were lost
during the new format work that was done in Asterisk 13.
Rather than rolling that same hack into res_rtp_multicast, re-introduce
smoother flags so that formats can dictate their own options.
Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16
---
M include/asterisk/codec.h
M include/asterisk/format.h
M include/asterisk/smoother.h
M main/codec_builtin.c
M main/format.c
M res/res_rtp_asterisk.c
M res/res_rtp_multicast.c
7 files changed, 122 insertions(+), 48 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/29/5729/1
diff --git a/include/asterisk/codec.h b/include/asterisk/codec.h
index 28befec..ea10c11 100644
--- a/include/asterisk/codec.h
+++ b/include/asterisk/codec.h
@@ -26,6 +26,9 @@
#ifndef _AST_CODEC_H_
#define _AST_CODEC_H_
+#define AST_SMOOTHER_FLAGS_PACK(x) ((x) << 1)
+#define AST_SMOOTHER_FLAGS_UNPACK(x) ((x) >> 1)
+
/*! \brief Types of media */
enum ast_media_type {
AST_MEDIA_TYPE_UNKNOWN = 0,
diff --git a/include/asterisk/format.h b/include/asterisk/format.h
index 2ce1b97..368e410 100644
--- a/include/asterisk/format.h
+++ b/include/asterisk/format.h
@@ -338,6 +338,17 @@
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
*
* \param format The media format
diff --git a/include/asterisk/smoother.h b/include/asterisk/smoother.h
index e63aa77..65ac889 100644
--- a/include/asterisk/smoother.h
+++ b/include/asterisk/smoother.h
@@ -33,6 +33,7 @@
#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 da03cce..5fdfa7e 100644
--- a/main/codec_builtin.c
+++ b/main/codec_builtin.c
@@ -37,6 +37,7 @@
#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);
@@ -264,7 +265,7 @@
.minimum_bytes = 160,
.samples_count = slin_samples,
.get_length = slin_length,
- .smooth = 1,
+ .smooth = AST_SMOOTHER_FLAGS_PACK(AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED),
};
static struct ast_codec slin12 = {
@@ -278,7 +279,7 @@
.minimum_bytes = 240,
.samples_count = slin_samples,
.get_length = slin_length,
- .smooth = 1,
+ .smooth = AST_SMOOTHER_FLAGS_PACK(AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED),
};
static struct ast_codec slin16 = {
@@ -292,7 +293,7 @@
.minimum_bytes = 320,
.samples_count = slin_samples,
.get_length = slin_length,
- .smooth = 1,
+ .smooth = AST_SMOOTHER_FLAGS_PACK(AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED),
};
static struct ast_codec slin24 = {
@@ -306,7 +307,7 @@
.minimum_bytes = 480,
.samples_count = slin_samples,
.get_length = slin_length,
- .smooth = 1,
+ .smooth = AST_SMOOTHER_FLAGS_PACK(AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED),
};
static struct ast_codec slin32 = {
@@ -320,7 +321,7 @@
.minimum_bytes = 640,
.samples_count = slin_samples,
.get_length = slin_length,
- .smooth = 1,
+ .smooth = AST_SMOOTHER_FLAGS_PACK(AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED),
};
static struct ast_codec slin44 = {
@@ -334,7 +335,7 @@
.minimum_bytes = 882,
.samples_count = slin_samples,
.get_length = slin_length,
- .smooth = 1,
+ .smooth = AST_SMOOTHER_FLAGS_PACK(AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED),
};
static struct ast_codec slin48 = {
@@ -348,7 +349,7 @@
.minimum_bytes = 960,
.samples_count = slin_samples,
.get_length = slin_length,
- .smooth = 1,
+ .smooth = AST_SMOOTHER_FLAGS_PACK(AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED),
};
static struct ast_codec slin96 = {
@@ -362,7 +363,7 @@
.minimum_bytes = 1920,
.samples_count = slin_samples,
.get_length = slin_length,
- .smooth = 1,
+ .smooth = AST_SMOOTHER_FLAGS_PACK(AST_SMOOTHER_FLAG_BE | AST_SMOOTHER_FLAG_FORCED),
};
static struct ast_codec slin192 = {
@@ -376,7 +377,7 @@
.minimum_bytes = 3840,
.samples_count = slin_samples,
.get_length = slin_length,
- .smooth = 1,
+ .smooth = AST_SMOOTHER_FLAGS_PACK(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 5bf38df..758a7fc 100644
--- a/main/format.c
+++ b/main/format.c
@@ -377,7 +377,13 @@
int ast_format_can_be_smoothed(const struct ast_format *format)
{
- return format->codec->smooth;
+ /* Coalesce to 1 if non-zero */
+ return format->codec->smooth ? 1 : 0;
+}
+
+int ast_format_get_smoother_flags(const struct ast_format *format)
+{
+ return AST_SMOOTHER_FLAGS_UNPACK(format->codec->smooth);
}
enum ast_media_type ast_format_get_type(const struct ast_format *format)
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 673f9c1..4743289 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -4014,10 +4014,10 @@
/* 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);
}
@@ -4028,9 +4028,7 @@
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);
}
}
diff --git a/res/res_rtp_multicast.c b/res/res_rtp_multicast.c
index ea31347..eee6fe1 100644
--- a/res/res_rtp_multicast.c
+++ b/res/res_rtp_multicast.c
@@ -56,6 +56,7 @@
#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
@@ -97,6 +98,7 @@
uint16_t seqno;
unsigned int lastts;
struct timeval txcore;
+ struct ast_smoother *smoother;
};
enum {
@@ -397,6 +399,10 @@
multicast_send_control_packet(instance, multicast, LINKSYS_MCAST_STOPCMD);
}
+ if (multicast->smoother) {
+ ast_smoother_free(multicast->smoother);
+ }
+
close(multicast->socket);
ast_free(multicast);
@@ -404,41 +410,24 @@
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 ast_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 */
- if ((codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 1, frame->subclass.format, 0)) < 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 +440,84 @@
/* 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 */
+ if ((codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 1, frame->subclass.format, 0)) < 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) {
+ ast_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) {
+ ast_rtp_raw_write(instance, f, codec);
+ }
+
+ if (f != frame) {
+ ast_frfree(f);
+ }
+ }
+
+ return 0;
}
/*! \brief Function called to read from a multicast instance */
--
To view, visit https://gerrit.asterisk.org/5729
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: newchange
Gerrit-Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Sean Bright <sean.bright at gmail.com>
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