[Asterisk-code-review] codecs: Add Codec 2 mode 2400. (asterisk[master])
Joshua Colp
asteriskteam at digium.com
Sun Sep 4 14:11:34 CDT 2016
Joshua Colp has submitted this change and it was merged.
Change subject: codecs: Add Codec 2 mode 2400.
......................................................................
codecs: Add Codec 2 mode 2400.
ASTERISK-26217 #close
Change-Id: I1e45d8084683fab5f2b272bf35f4a149cea8b8d6
---
M build_tools/menuselect-deps.in
A codecs/codec_codec2.c
A codecs/ex_codec2.h
M configure
M configure.ac
M include/asterisk/autoconfig.h.in
M include/asterisk/format_cache.h
M main/codec_builtin.c
M main/format_cache.c
M makeopts.in
10 files changed, 446 insertions(+), 1 deletion(-)
Approvals:
Mark Michelson: Looks good to me, but someone else must approve
Matt Jordan: Looks good to me, approved
Joshua Colp: Looks good to me, but someone else must approve; Verified
diff --git a/build_tools/menuselect-deps.in b/build_tools/menuselect-deps.in
index f194482..a044409 100644
--- a/build_tools/menuselect-deps.in
+++ b/build_tools/menuselect-deps.in
@@ -4,6 +4,7 @@
CRYPTO=@PBX_CRYPTO@
BFD=@PBX_BFD@
BISON=@PBX_BISON@
+CODEC2=@PBX_CODEC2@
CURL=@PBX_CURL@
DAHDI=@PBX_DAHDI@
DLADDR=@PBX_DLADDR@
diff --git a/codecs/codec_codec2.c b/codecs/codec_codec2.c
new file mode 100644
index 0000000..e446854
--- /dev/null
+++ b/codecs/codec_codec2.c
@@ -0,0 +1,222 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2016, Alexander Traud
+ *
+ * Alexander Traud <pabstraud at compuserve.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Translate between signed linear and Codec 2
+ *
+ * \author Alexander Traud <pabstraud at compuserve.com>
+ *
+ * \note http://www.rowetel.com/codec2.html
+ *
+ * \ingroup codecs
+ */
+
+/*** MODULEINFO
+ <depend>codec2</depend>
+ <support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/codec.h" /* for AST_MEDIA_TYPE_AUDIO */
+#include "asterisk/frame.h" /* for ast_frame */
+#include "asterisk/linkedlists.h" /* for AST_LIST_NEXT, etc */
+#include "asterisk/logger.h" /* for ast_log, etc */
+#include "asterisk/module.h"
+#include "asterisk/rtp_engine.h" /* ast_rtp_engine_(un)load_format */
+#include "asterisk/translate.h" /* for ast_trans_pvt, etc */
+
+#include <codec2/codec2.h>
+
+#define BUFFER_SAMPLES 8000
+#define CODEC2_SAMPLES 160 /* consider codec2_samples_per_frame(.) */
+#define CODEC2_FRAME_LEN 6 /* consider codec2_bits_per_frame(.) */
+
+/* Sample frame data */
+#include "asterisk/slin.h"
+#include "ex_codec2.h"
+
+struct codec2_translator_pvt {
+ struct CODEC2 *state; /* May be encoder or decoder */
+ int16_t buf[BUFFER_SAMPLES];
+};
+
+static int codec2_new(struct ast_trans_pvt *pvt)
+{
+ struct codec2_translator_pvt *tmp = pvt->pvt;
+
+ tmp->state = codec2_create(CODEC2_MODE_2400);
+
+ if (!tmp->state) {
+ ast_log(LOG_ERROR, "Error creating Codec 2 conversion\n");
+ return -1;
+ }
+
+ return 0;
+}
+
+/*! \brief decode and store in outbuf. */
+static int codec2tolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
+{
+ struct codec2_translator_pvt *tmp = pvt->pvt;
+ int x;
+
+ for (x = 0; x < f->datalen; x += CODEC2_FRAME_LEN) {
+ unsigned char *src = f->data.ptr + x;
+ int16_t *dst = pvt->outbuf.i16 + pvt->samples;
+
+ codec2_decode(tmp->state, dst, src);
+
+ pvt->samples += CODEC2_SAMPLES;
+ pvt->datalen += CODEC2_SAMPLES * 2;
+ }
+
+ return 0;
+}
+
+/*! \brief store samples into working buffer for later decode */
+static int lintocodec2_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
+{
+ struct codec2_translator_pvt *tmp = pvt->pvt;
+
+ memcpy(tmp->buf + pvt->samples, f->data.ptr, f->datalen);
+ pvt->samples += f->samples;
+
+ return 0;
+}
+
+/*! \brief encode and produce a frame */
+static struct ast_frame *lintocodec2_frameout(struct ast_trans_pvt *pvt)
+{
+ struct codec2_translator_pvt *tmp = pvt->pvt;
+ struct ast_frame *result = NULL;
+ struct ast_frame *last = NULL;
+ int samples = 0; /* output samples */
+
+ while (pvt->samples >= CODEC2_SAMPLES) {
+ struct ast_frame *current;
+
+ /* Encode a frame of data */
+ codec2_encode(tmp->state, pvt->outbuf.uc, tmp->buf + samples);
+
+ samples += CODEC2_SAMPLES;
+ pvt->samples -= CODEC2_SAMPLES;
+
+ current = ast_trans_frameout(pvt, CODEC2_FRAME_LEN, CODEC2_SAMPLES);
+
+ if (!current) {
+ continue;
+ } else if (last) {
+ AST_LIST_NEXT(last, frame_list) = current;
+ } else {
+ result = current;
+ }
+ last = current;
+ }
+
+ /* Move the data at the end of the buffer to the front */
+ if (samples) {
+ memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2);
+ }
+
+ return result;
+}
+
+static void codec2_destroy_stuff(struct ast_trans_pvt *pvt)
+{
+ struct codec2_translator_pvt *tmp = pvt->pvt;
+
+ if (tmp->state) {
+ codec2_destroy(tmp->state);
+ }
+}
+
+static struct ast_translator codec2tolin = {
+ .name = "codec2tolin",
+ .src_codec = {
+ .name = "codec2",
+ .type = AST_MEDIA_TYPE_AUDIO,
+ .sample_rate = 8000,
+ },
+ .dst_codec = {
+ .name = "slin",
+ .type = AST_MEDIA_TYPE_AUDIO,
+ .sample_rate = 8000,
+ },
+ .format = "slin",
+ .newpvt = codec2_new,
+ .framein = codec2tolin_framein,
+ .destroy = codec2_destroy_stuff,
+ .sample = codec2_sample,
+ .desc_size = sizeof(struct codec2_translator_pvt),
+ .buffer_samples = BUFFER_SAMPLES,
+ .buf_size = BUFFER_SAMPLES * 2,
+};
+
+static struct ast_translator lintocodec2 = {
+ .name = "lintocodec2",
+ .src_codec = {
+ .name = "slin",
+ .type = AST_MEDIA_TYPE_AUDIO,
+ .sample_rate = 8000,
+ },
+ .dst_codec = {
+ .name = "codec2",
+ .type = AST_MEDIA_TYPE_AUDIO,
+ .sample_rate = 8000,
+ },
+ .format = "codec2",
+ .newpvt = codec2_new,
+ .framein = lintocodec2_framein,
+ .frameout = lintocodec2_frameout,
+ .destroy = codec2_destroy_stuff,
+ .sample = slin8_sample,
+ .desc_size = sizeof(struct codec2_translator_pvt),
+ .buffer_samples = BUFFER_SAMPLES,
+ .buf_size = (BUFFER_SAMPLES * CODEC2_FRAME_LEN + CODEC2_SAMPLES - 1) / CODEC2_SAMPLES,
+};
+
+static int unload_module(void)
+{
+ int res = 0;
+
+ res |= ast_rtp_engine_unload_format(ast_format_codec2);
+ res |= ast_unregister_translator(&lintocodec2);
+ res |= ast_unregister_translator(&codec2tolin);
+
+ return res;
+}
+
+static int load_module(void)
+{
+ int res = 0;
+
+ res |= ast_register_translator(&codec2tolin);
+ res |= ast_register_translator(&lintocodec2);
+ res |= ast_rtp_engine_load_format(ast_format_codec2);
+
+ if (res) {
+ unload_module();
+ return AST_MODULE_LOAD_FAILURE;
+ }
+
+ return AST_MODULE_LOAD_SUCCESS;
+}
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Codec 2 Coder/Decoder");
diff --git a/codecs/ex_codec2.h b/codecs/ex_codec2.h
new file mode 100644
index 0000000..f2f4c97
--- /dev/null
+++ b/codecs/ex_codec2.h
@@ -0,0 +1,32 @@
+/*! \file
+ * \brief 8-bit raw data
+ *
+ * Copyright (C) 2016, Alexander Traud
+ *
+ * Distributed under the terms of the GNU General Public License
+ *
+ */
+
+#include "asterisk/format_cache.h" /* for ast_format_codec2 */
+#include "asterisk/frame.h" /* for ast_frame, etc */
+
+static uint8_t ex_codec2[] = {
+ 0xea, 0xca, 0x14, 0x85, 0x91, 0x78,
+};
+
+static struct ast_frame *codec2_sample(void)
+{
+ static struct ast_frame f = {
+ .frametype = AST_FRAME_VOICE,
+ .datalen = sizeof(ex_codec2),
+ .samples = CODEC2_SAMPLES,
+ .mallocd = 0,
+ .offset = 0,
+ .src = __PRETTY_FUNCTION__,
+ .data.ptr = ex_codec2,
+ };
+
+ f.subclass.format = ast_format_codec2;
+
+ return &f;
+}
diff --git a/configure b/configure
index 6d3faf2..64c7683 100755
--- a/configure
+++ b/configure
@@ -1152,6 +1152,10 @@
COROSYNC_DIR
COROSYNC_INCLUDE
COROSYNC_LIB
+PBX_CODEC2
+CODEC2_DIR
+CODEC2_INCLUDE
+CODEC2_LIB
PBX_CAP
CAP_DIR
CAP_INCLUDE
@@ -1326,6 +1330,7 @@
with_execinfo
with_bluetooth
with_cap
+with_codec2
with_cpg
with_curses
with_crypt
@@ -2065,6 +2070,7 @@
--with-execinfo=PATH use Stack Backtrace files in PATH
--with-bluetooth=PATH use Bluetooth files in PATH
--with-cap=PATH use POSIX 1.e capabilities files in PATH
+ --with-codec2=PATH use Codec 2 Audio Decoder/Encoder files in PATH
--with-cpg=PATH use Corosync files in PATH
--with-curses=PATH use curses files in PATH
--with-crypt=PATH use password and data encryption files in PATH
@@ -8976,6 +8982,38 @@
*)
CAP_DIR="${withval}"
ac_mandatory_list="${ac_mandatory_list} CAP"
+ ;;
+ esac
+
+fi
+
+
+
+
+
+
+
+
+ CODEC2_DESCRIP="Codec 2 Audio Decoder/Encoder"
+ CODEC2_OPTION="codec2"
+ PBX_CODEC2=0
+
+# Check whether --with-codec2 was given.
+if test "${with_codec2+set}" = set; then :
+ withval=$with_codec2;
+ case ${withval} in
+ n|no)
+ USE_CODEC2=no
+ # -1 is a magic value used by menuselect to know that the package
+ # was disabled, other than 'not found'
+ PBX_CODEC2=-1
+ ;;
+ y|ye|yes)
+ ac_mandatory_list="${ac_mandatory_list} CODEC2"
+ ;;
+ *)
+ CODEC2_DIR="${withval}"
+ ac_mandatory_list="${ac_mandatory_list} CODEC2"
;;
esac
@@ -30372,6 +30410,111 @@
fi
+if test "x${PBX_CODEC2}" != "x1" -a "${USE_CODEC2}" != "no"; then
+ pbxlibdir=""
+ # if --with-CODEC2=DIR has been specified, use it.
+ if test "x${CODEC2_DIR}" != "x"; then
+ if test -d ${CODEC2_DIR}/lib; then
+ pbxlibdir="-L${CODEC2_DIR}/lib"
+ else
+ pbxlibdir="-L${CODEC2_DIR}"
+ fi
+ fi
+ pbxfuncname="codec2_create"
+ if test "x${pbxfuncname}" = "x" ; then # empty lib, assume only headers
+ AST_CODEC2_FOUND=yes
+ else
+ ast_ext_lib_check_save_CFLAGS="${CFLAGS}"
+ CFLAGS="${CFLAGS} "
+ as_ac_Lib=`$as_echo "ac_cv_lib_codec2_${pbxfuncname}" | $as_tr_sh`
+{ $as_echo "$as_me:${as_lineno-$LINENO}: checking for ${pbxfuncname} in -lcodec2" >&5
+$as_echo_n "checking for ${pbxfuncname} in -lcodec2... " >&6; }
+if eval \${$as_ac_Lib+:} false; then :
+ $as_echo_n "(cached) " >&6
+else
+ ac_check_lib_save_LIBS=$LIBS
+LIBS="-lcodec2 ${pbxlibdir} $LIBS"
+cat confdefs.h - <<_ACEOF >conftest.$ac_ext
+/* end confdefs.h. */
+
+/* Override any GCC internal prototype to avoid an error.
+ Use char because int might match the return type of a GCC
+ builtin and then its argument prototype would still apply. */
+#ifdef __cplusplus
+extern "C"
+#endif
+char ${pbxfuncname} ();
+int
+main ()
+{
+return ${pbxfuncname} ();
+ ;
+ return 0;
+}
+_ACEOF
+if ac_fn_c_try_link "$LINENO"; then :
+ eval "$as_ac_Lib=yes"
+else
+ eval "$as_ac_Lib=no"
+fi
+rm -f core conftest.err conftest.$ac_objext \
+ conftest$ac_exeext conftest.$ac_ext
+LIBS=$ac_check_lib_save_LIBS
+fi
+eval ac_res=\$$as_ac_Lib
+ { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_res" >&5
+$as_echo "$ac_res" >&6; }
+if eval test \"x\$"$as_ac_Lib"\" = x"yes"; then :
+ AST_CODEC2_FOUND=yes
+else
+ AST_CODEC2_FOUND=no
+fi
+
+ CFLAGS="${ast_ext_lib_check_save_CFLAGS}"
+ fi
+
+ # now check for the header.
+ if test "${AST_CODEC2_FOUND}" = "yes"; then
+ CODEC2_LIB="${pbxlibdir} -lcodec2 "
+ # if --with-CODEC2=DIR has been specified, use it.
+ if test "x${CODEC2_DIR}" != "x"; then
+ CODEC2_INCLUDE="-I${CODEC2_DIR}/include"
+ fi
+ CODEC2_INCLUDE="${CODEC2_INCLUDE} "
+ if test "xcodec2/codec2.h" = "x" ; then # no header, assume found
+ CODEC2_HEADER_FOUND="1"
+ else # check for the header
+ ast_ext_lib_check_saved_CPPFLAGS="${CPPFLAGS}"
+ CPPFLAGS="${CPPFLAGS} ${CODEC2_INCLUDE}"
+ ac_fn_c_check_header_mongrel "$LINENO" "codec2/codec2.h" "ac_cv_header_codec2_codec2_h" "$ac_includes_default"
+if test "x$ac_cv_header_codec2_codec2_h" = xyes; then :
+ CODEC2_HEADER_FOUND=1
+else
+ CODEC2_HEADER_FOUND=0
+fi
+
+
+ CPPFLAGS="${ast_ext_lib_check_saved_CPPFLAGS}"
+ fi
+ if test "x${CODEC2_HEADER_FOUND}" = "x0" ; then
+ CODEC2_LIB=""
+ CODEC2_INCLUDE=""
+ else
+ if test "x${pbxfuncname}" = "x" ; then # only checking headers -> no library
+ CODEC2_LIB=""
+ fi
+ PBX_CODEC2=1
+ cat >>confdefs.h <<_ACEOF
+#define HAVE_CODEC2 1
+_ACEOF
+
+ fi
+ fi
+fi
+
+
+
+
if test "x${PBX_COROSYNC}" != "x1" -a "${USE_COROSYNC}" != "no"; then
pbxlibdir=""
# if --with-COROSYNC=DIR has been specified, use it.
diff --git a/configure.ac b/configure.ac
index dedfd8a..96a20d0 100644
--- a/configure.ac
+++ b/configure.ac
@@ -407,6 +407,7 @@
AST_EXT_LIB_SETUP([BKTR], [Stack Backtrace], [execinfo])
AST_EXT_LIB_SETUP([BLUETOOTH], [Bluetooth], [bluetooth])
AST_EXT_LIB_SETUP([CAP], [POSIX 1.e capabilities], [cap])
+AST_EXT_LIB_SETUP([CODEC2], [Codec 2 Audio Decoder/Encoder], [codec2])
AST_EXT_LIB_SETUP([COROSYNC], [Corosync], [cpg])
AST_EXT_LIB_SETUP_OPTIONAL([COROSYNC_CFG_STATE_TRACK], [A callback only in corosync 1.x], [COROSYNC], [cfg])
AST_EXT_LIB_SETUP([CURSES], [curses], [curses])
@@ -2336,6 +2337,8 @@
AST_EXT_LIB_CHECK([RADIUS], [radiusclient-ng], [rc_read_config], [radiusclient-ng.h])
fi
+AST_EXT_LIB_CHECK([CODEC2], [codec2], [codec2_create], [codec2/codec2.h])
+
AST_EXT_LIB_CHECK([COROSYNC], [cpg], [cpg_join], [corosync/cpg.h], [-lcfg])
AST_EXT_LIB_CHECK([COROSYNC_CFG_STATE_TRACK], [cfg], [corosync_cfg_state_track], [corosync/cfg.h], [-lcfg])
diff --git a/include/asterisk/autoconfig.h.in b/include/asterisk/autoconfig.h.in
index 51f0f14..7d2a08c 100644
--- a/include/asterisk/autoconfig.h.in
+++ b/include/asterisk/autoconfig.h.in
@@ -142,6 +142,9 @@
/* Define to 1 if you have the `closefrom' function. */
#undef HAVE_CLOSEFROM
+/* Define to 1 if you have the Codec 2 Audio Decoder/Encoder library. */
+#undef HAVE_CODEC2
+
/* Define to 1 if you have the Corosync library. */
#undef HAVE_COROSYNC
diff --git a/include/asterisk/format_cache.h b/include/asterisk/format_cache.h
index 3894ad2..6099c59 100644
--- a/include/asterisk/format_cache.h
+++ b/include/asterisk/format_cache.h
@@ -209,6 +209,11 @@
extern struct ast_format *ast_format_opus;
/*!
+ * \brief Built-in cached Codec 2 format.
+ */
+extern struct ast_format *ast_format_codec2;
+
+/*!
* \brief Built-in cached t140 format.
*/
extern struct ast_format *ast_format_t140;
diff --git a/main/codec_builtin.c b/main/codec_builtin.c
index 50fbf55..6fc0fd8 100644
--- a/main/codec_builtin.c
+++ b/main/codec_builtin.c
@@ -107,6 +107,30 @@
.get_length = g723_length,
};
+static int codec2_samples(struct ast_frame *frame)
+{
+ return 160 * (frame->datalen / 6);
+}
+
+static int codec2_length(unsigned int samples)
+{
+ return (samples / 160) * 6;
+}
+
+static struct ast_codec codec2 = {
+ .name = "codec2",
+ .description = "Codec 2",
+ .type = AST_MEDIA_TYPE_AUDIO,
+ .sample_rate = 8000,
+ .minimum_ms = 20,
+ .maximum_ms = 300,
+ .default_ms = 20,
+ .minimum_bytes = 6,
+ .samples_count = codec2_samples,
+ .get_length = codec2_length,
+ .smooth = 1,
+};
+
static int none_samples(struct ast_frame *frame)
{
return frame->datalen;
@@ -863,6 +887,7 @@
{
int res = 0;
+ res |= CODEC_REGISTER_AND_CACHE(codec2);
res |= CODEC_REGISTER_AND_CACHE(g723);
res |= CODEC_REGISTER_AND_CACHE(ulaw);
res |= CODEC_REGISTER_AND_CACHE(alaw);
diff --git a/main/format_cache.c b/main/format_cache.c
index b4d4260..c704f1c 100644
--- a/main/format_cache.c
+++ b/main/format_cache.c
@@ -218,6 +218,11 @@
struct ast_format *ast_format_opus;
/*!
+ * \brief Built-in cached codec2 format.
+ */
+struct ast_format *ast_format_codec2;
+
+/*!
* \brief Built-in cached t140 format.
*/
struct ast_format *ast_format_t140;
@@ -328,6 +333,7 @@
ao2_replace(ast_format_testlaw, NULL);
ao2_replace(ast_format_g719, NULL);
ao2_replace(ast_format_opus, NULL);
+ ao2_replace(ast_format_codec2, NULL);
ao2_replace(ast_format_jpeg, NULL);
ao2_replace(ast_format_png, NULL);
ao2_replace(ast_format_h261, NULL);
@@ -360,7 +366,9 @@
static void set_cached_format(const char *name, struct ast_format *format)
{
- if (!strcmp(name, "g723")) {
+ if (!strcmp(name, "codec2")) {
+ ao2_replace(ast_format_codec2, format);
+ } else if (!strcmp(name, "g723")) {
ao2_replace(ast_format_g723, format);
} else if (!strcmp(name, "ulaw")) {
ao2_replace(ast_format_ulaw, format);
diff --git a/makeopts.in b/makeopts.in
index d4347da..f0b0d0e 100644
--- a/makeopts.in
+++ b/makeopts.in
@@ -126,6 +126,9 @@
BLUETOOTH_INCLUDE=@BLUETOOTH_INCLUDE@
BLUETOOTH_LIB=@BLUETOOTH_LIB@
+CODEC2_INCLUDE=@CODEC2_INCLUDE@
+CODEC2_LIB=@CODEC2_LIB@
+
CURL_INCLUDE=@CURL_INCLUDE@
CURL_LIB=@CURL_LIB@
--
To view, visit https://gerrit.asterisk.org/3244
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: I1e45d8084683fab5f2b272bf35f4a149cea8b8d6
Gerrit-PatchSet: 6
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
Gerrit-Reviewer: Matthew Fredrickson <creslin at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
More information about the asterisk-code-review
mailing list