--- /dev/null
+diff -urN asterisk-11.8.1-orig/build_tools/menuselect-deps.in asterisk-11.8.1-codec2/build_tools/menuselect-deps.in
+--- asterisk-11.8.1-orig/build_tools/menuselect-deps.in 2013-04-11 21:59:35.000000000 +0200
++++ asterisk-11.8.1-codec2/build_tools/menuselect-deps.in 2014-04-01 10:24:29.368859813 +0200
+@@ -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 -urN asterisk-11.8.1-orig/codecs/codec_codec2.c asterisk-11.8.1-codec2/codecs/codec_codec2.c
+--- asterisk-11.8.1-orig/codecs/codec_codec2.c 1970-01-01 01:00:00.000000000 +0100
++++ asterisk-11.8.1-codec2/codecs/codec_codec2.c 2014-04-01 01:28:47.000000000 +0200
+@@ -0,0 +1,190 @@
++/*
++ * Codec 2 module for Asterisk.
++ *
++ * Credit: codec_gsm.c used as a starting point.
++ *
++ * Copyright (C) 2012 Ed W and David Rowe
++ *
++ * 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
++ *
++ * \ingroup codecs
++ */
++
++/*** MODULEINFO
++ <depend>codec2</depend>
++ <support_level>core</support_level>
++ ***/
++
++#include "asterisk.h"
++
++#include "asterisk/translate.h"
++#include "asterisk/config.h"
++#include "asterisk/module.h"
++#include "asterisk/utils.h"
++
++#include <codec2/codec2.h>
++
++#define BUFFER_SAMPLES 8000
++#define CODEC2_SAMPLES 160
++#define CODEC2_FRAME_LEN 6
++
++/* Sample frame data */
++
++#include "asterisk/slin.h"
++#include "ex_codec2.h"
++
++struct codec2_translator_pvt { /* both codec2tolin and codec2togsm */
++ struct CODEC2 *codec2;
++ int16_t buf[BUFFER_SAMPLES]; /* lintocodec2, temporary storage */
++};
++
++static int codec2_new(struct ast_trans_pvt *pvt)
++{
++ struct codec2_translator_pvt *tmp = pvt->pvt;
++
++ tmp->codec2 = codec2_create(CODEC2_MODE_2400);
++ if (!tmp)
++ 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;
++ int16_t *dst = pvt->outbuf.i16;
++ int flen = CODEC2_FRAME_LEN;
++
++ for (x=0; x < f->datalen; x += flen) {
++ unsigned char *src;
++ int len;
++ len = CODEC2_SAMPLES;
++ src = f->data.ptr + x;
++
++ codec2_decode(tmp->codec2, dst + pvt->samples, src);
++
++ pvt->samples += CODEC2_SAMPLES;
++ pvt->datalen += 2 * CODEC2_SAMPLES;
++ }
++ 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;
++
++ if (pvt->samples + f->samples > BUFFER_SAMPLES) {
++ ast_log(LOG_WARNING, "Out of buffer space\n");
++ return -1;
++ }
++ 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;
++ int datalen = 0;
++ int samples = 0;
++
++ /* We can't work on anything less than a frame in size */
++ if (pvt->samples < CODEC2_SAMPLES)
++ return NULL;
++ while (pvt->samples >= CODEC2_SAMPLES) {
++ /* Encode a frame of data */
++ codec2_encode(tmp->codec2, (unsigned char*)(pvt->outbuf.c + datalen), tmp->buf + samples);
++ datalen += CODEC2_FRAME_LEN;
++ samples += CODEC2_SAMPLES;
++ pvt->samples -= CODEC2_SAMPLES;
++ }
++
++ /* Move the data at the end of the buffer to the front */
++ if (pvt->samples)
++ memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2);
++
++ return ast_trans_frameout(pvt, datalen, samples);
++}
++
++static void codec2_destroy_stuff(struct ast_trans_pvt *pvt)
++{
++ struct codec2_translator_pvt *tmp = pvt->pvt;
++ if (tmp->codec2)
++ codec2_destroy(tmp->codec2);
++}
++
++static struct ast_translator codec2tolin = {
++ .name = "codec2tolin",
++ .newpvt = codec2_new,
++ .framein = codec2tolin_framein,
++ .destroy = codec2_destroy_stuff,
++ .sample = codec2_sample,
++ .buffer_samples = BUFFER_SAMPLES,
++ .buf_size = BUFFER_SAMPLES * 2,
++ .desc_size = sizeof (struct codec2_translator_pvt ),
++};
++
++static struct ast_translator lintocodec2 = {
++ .name = "lintocodec2",
++ .newpvt = codec2_new,
++ .framein = lintocodec2_framein,
++ .frameout = lintocodec2_frameout,
++ .destroy = codec2_destroy_stuff,
++ .sample = slin8_sample,
++ .desc_size = sizeof (struct codec2_translator_pvt ),
++ .buf_size = (BUFFER_SAMPLES * CODEC2_FRAME_LEN + CODEC2_SAMPLES - 1)/CODEC2_SAMPLES,
++};
++
++/*! \brief standard module glue */
++static int reload(void)
++{
++ return AST_MODULE_LOAD_SUCCESS;
++}
++
++static int unload_module(void)
++{
++ int res;
++
++ res = ast_unregister_translator(&lintocodec2);
++ if (!res)
++ res = ast_unregister_translator(&codec2tolin);
++
++ return res;
++}
++
++static int load_module(void)
++{
++ int res;
++
++ ast_format_set(&codec2tolin.src_format, AST_FORMAT_CODEC2, 0);
++ ast_format_set(&codec2tolin.dst_format, AST_FORMAT_SLINEAR, 0);
++
++ ast_format_set(&lintocodec2.src_format, AST_FORMAT_SLINEAR, 0);
++ ast_format_set(&lintocodec2.dst_format, AST_FORMAT_CODEC2, 0);
++
++ res = ast_register_translator(&codec2tolin);
++ if (!res)
++ res=ast_register_translator(&lintocodec2);
++ else
++ ast_unregister_translator(&codec2tolin);
++ if (res)
++ return AST_MODULE_LOAD_FAILURE;
++ return AST_MODULE_LOAD_SUCCESS;
++}
++
++AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Codec 2 Coder/Decoder",
++ .load = load_module,
++ .unload = unload_module,
++ .reload = reload,
++ );
+diff -urN asterisk-11.8.1-orig/codecs/ex_codec2.h asterisk-11.8.1-codec2/codecs/ex_codec2.h
+--- asterisk-11.8.1-orig/codecs/ex_codec2.h 1970-01-01 01:00:00.000000000 +0100
++++ asterisk-11.8.1-codec2/codecs/ex_codec2.h 2014-04-01 01:30:29.000000000 +0200
+@@ -0,0 +1,29 @@
++/*! \file
++ * \brief 8-bit raw data
++ *
++ * Copyright (C) 2012, 2012 Ed W and David Rowe
++ *
++ * Distributed under the terms of the GNU General Public License
++ *
++ */
++
++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,
++ };
++
++ ast_format_set(&f.subclass.format, AST_FORMAT_CODEC2, 0);
++
++ return &f;
++}
+diff -urN asterisk-11.8.1-orig/configure.ac asterisk-11.8.1-codec2/configure.ac
+--- asterisk-11.8.1-orig/configure.ac 2014-01-08 17:17:32.000000000 +0100
++++ asterisk-11.8.1-codec2/configure.ac 2014-04-01 10:27:08.503390997 +0200
+@@ -384,6 +384,7 @@
+ 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], [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])
+@@ -2116,6 +2117,8 @@
+
+ AST_EXT_LIB_CHECK([RADIUS], [radiusclient-ng], [rc_read_config], [radiusclient-ng.h])
+
++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 -urN asterisk-11.8.1-orig/include/asterisk/format.h asterisk-11.8.1-codec2/include/asterisk/format.h
+--- asterisk-11.8.1-orig/include/asterisk/format.h 2012-07-13 20:41:07.000000000 +0200
++++ asterisk-11.8.1-codec2/include/asterisk/format.h 2014-04-01 10:03:16.120355835 +0200
+@@ -101,6 +101,7 @@
+ AST_FORMAT_SLINEAR192 = 27 + AST_FORMAT_TYPE_AUDIO,
+ AST_FORMAT_SPEEX32 = 28 + AST_FORMAT_TYPE_AUDIO,
+ AST_FORMAT_CELT = 29 + AST_FORMAT_TYPE_AUDIO,
++ AST_FORMAT_CODEC2 = 31 + AST_FORMAT_TYPE_AUDIO,
+
+ /*! H.261 Video */
+ AST_FORMAT_H261 = 1 + AST_FORMAT_TYPE_VIDEO,
+diff -urN asterisk-11.8.1-orig/main/channel.c asterisk-11.8.1-codec2/main/channel.c
+--- asterisk-11.8.1-orig/main/channel.c 2013-12-31 00:16:04.000000000 +0100
++++ asterisk-11.8.1-codec2/main/channel.c 2014-04-01 10:06:55.659929991 +0200
+@@ -918,6 +918,8 @@
+ AST_FORMAT_SILK,
+ /*! CELT supports crazy high sample rates */
+ AST_FORMAT_CELT,
++ /* Codec 2 */
++ AST_FORMAT_CODEC2,
+ /*! Ick, LPC10 sounds terrible, but at least we have code for it, if you're tacky enough
+ to use it */
+ AST_FORMAT_LPC10,
+diff -urN asterisk-11.8.1-orig/main/format.c asterisk-11.8.1-codec2/main/format.c
+--- asterisk-11.8.1-orig/main/format.c 2013-06-12 04:25:23.000000000 +0200
++++ asterisk-11.8.1-codec2/main/format.c 2014-04-01 10:11:05.319972320 +0200
+@@ -430,6 +430,9 @@
+ /*! SpeeX Wideband (16kHz) Free Compression */
+ case AST_FORMAT_SPEEX16:
+ return (1ULL << 33);
++ /*! Codec 2 (8KHz) */
++ case AST_FORMAT_CODEC2:
++ return (1ULL << 35);
+ /*! Raw mu-law data (G.711) */
+ case AST_FORMAT_TESTLAW:
+ return (1ULL << 47);
+@@ -532,6 +535,9 @@
+ /*! SpeeX Wideband (16kHz) Free Compression */
+ case (1ULL << 33):
+ return ast_format_set(dst, AST_FORMAT_SPEEX16, 0);
++ /*! Codec 2 (8KHz) */
++ case (1ULL << 35):
++ return ast_format_set(dst, AST_FORMAT_CODEC2, 0);
+ /*! Raw mu-law data (G.711) */
+ case (1ULL << 47):
+ return ast_format_set(dst, AST_FORMAT_TESTLAW, 0);
+@@ -1071,6 +1077,8 @@
+ format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR48, 0), "slin48", 48000, "16 bit Signed Linear PCM (48kHz)", 960, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE, 0);/*!< Signed linear (48kHz) */
+ format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR96, 0), "slin96", 96000, "16 bit Signed Linear PCM (96kHz)", 1920, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE, 0);/*!< Signed linear (96kHz) */
+ format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR192, 0), "slin192", 192000, "16 bit Signed Linear PCM (192kHz)", 3840, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE, 0);/*!< Signed linear (192kHz) */
++ /* Codec 2 */
++ format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_CODEC2, 0), "codec2", 8000, "Codec 2", 6, 20, 20, 20, 20, 0, 0); /*!< codec_codec2.c */
+
+ return 0;
+ }
+diff -urN asterisk-11.8.1-orig/main/frame.c asterisk-11.8.1-codec2/main/frame.c
+--- asterisk-11.8.1-orig/main/frame.c 2012-07-24 18:54:26.000000000 +0200
++++ asterisk-11.8.1-codec2/main/frame.c 2014-04-01 10:13:35.626395684 +0200
+@@ -1083,6 +1083,10 @@
+ /* TODO The assumes 20ms delivery right now, which is incorrect */
+ samples = ast_format_rate(&f->subclass.format) / 50;
+ break;
++ /* Codec 2 */
++ case AST_FORMAT_CODEC2:
++ samples = 160 * (f->datalen / 6);
++ break;
+ default:
+ ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(&f->subclass.format));
+ }
+@@ -1134,6 +1138,10 @@
+ /* 48,000 samples per second at 64kbps is 8,000 bytes per second */
+ len = (int) samples / ((float) 48000 / 8000);
+ break;
++ /* Codec 2 */
++ case AST_FORMAT_CODEC2:
++ len = (samples / 160) * 6;
++ break;
+ default:
+ ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format));
+ }
+diff -urN asterisk-11.8.1-orig/main/rtp_engine.c asterisk-11.8.1-codec2/main/rtp_engine.c
+--- asterisk-11.8.1-orig/main/rtp_engine.c 2013-12-18 00:35:07.000000000 +0100
++++ asterisk-11.8.1-codec2/main/rtp_engine.c 2014-04-01 10:17:00.121681465 +0200
+@@ -2289,6 +2289,8 @@
+ set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN7, 0), 0, "audio", "G7221", 16000);
+ set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SIREN14, 0), 0, "audio", "G7221", 32000);
+ set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_G719, 0), 0, "audio", "G719", 48000);
++ /* Codec 2 */
++ set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_CODEC2, 0), 0, "audio", "CODEC2", 8000);
+
+ /* Define the static rtp payload mappings */
+ add_static_payload(0, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0);
+@@ -2330,6 +2332,8 @@
+ add_static_payload(118, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR16, 0), 0); /* 16 Khz signed linear */
+ add_static_payload(119, ast_format_set(&tmpfmt, AST_FORMAT_SPEEX32, 0), 0);
+ add_static_payload(121, NULL, AST_RTP_CISCO_DTMF); /* Must be type 121 */
++ /* Codec 2 */
++ add_static_payload(121, ast_format_set(&tmpfmt, AST_FORMAT_CODEC2, 0), 0);
+
+ return 0;
+ }
+diff -urN asterisk-11.8.1-orig/makeopts.in asterisk-11.8.1-codec2/makeopts.in
+--- asterisk-11.8.1-orig/makeopts.in 2013-04-11 21:59:35.000000000 +0200
++++ asterisk-11.8.1-codec2/makeopts.in 2014-04-01 10:28:49.424993462 +0200
+@@ -120,6 +120,9 @@
+ BLUETOOTH_INCLUDE=@BLUETOOTH_INCLUDE@
+ BLUETOOTH_LIB=@BLUETOOTH_LIB@
+
++CODEC2_INCLUDE=@CODEC2_INCLUDE@
++CODEC2_LIB=@CODEC2_LIB@
++
+ CURL_INCLUDE=@CURL_INCLUDE@
+ CURL_LIB=@CURL_LIB@
+
+diff -urN asterisk-11.8.1-orig/res/res_rtp_asterisk.c asterisk-11.8.1-codec2/res/res_rtp_asterisk.c
+--- asterisk-11.8.1-orig/res/res_rtp_asterisk.c 2014-02-27 22:39:30.000000000 +0100
++++ asterisk-11.8.1-codec2/res/res_rtp_asterisk.c 2014-04-01 10:19:18.727901747 +0200
+@@ -2738,6 +2738,8 @@
+ case AST_FORMAT_SIREN7:
+ case AST_FORMAT_SIREN14:
+ case AST_FORMAT_G719:
++ /* Codec 2 */
++ case AST_FORMAT_CODEC2:
+ /* these are all frame-based codecs and cannot be safely run through
+ a smoother */
+ break;
--- /dev/null
+diff -urN asterisk-11.8.1-opus/build_tools/menuselect-deps.in asterisk-11.8.1-opus-codec2/build_tools/menuselect-deps.in
+--- asterisk-11.8.1-opus/build_tools/menuselect-deps.in 2014-04-01 10:58:52.127106070 +0200
++++ asterisk-11.8.1-opus-codec2/build_tools/menuselect-deps.in 2014-04-01 10:50:21.490079146 +0200
+@@ -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 -urN asterisk-11.8.1-opus/codecs/codec_codec2.c asterisk-11.8.1-opus-codec2/codecs/codec_codec2.c
+--- asterisk-11.8.1-opus/codecs/codec_codec2.c 1970-01-01 01:00:00.000000000 +0100
++++ asterisk-11.8.1-opus-codec2/codecs/codec_codec2.c 2014-04-01 10:50:21.490079146 +0200
+@@ -0,0 +1,190 @@
++/*
++ * Codec 2 module for Asterisk.
++ *
++ * Credit: codec_gsm.c used as a starting point.
++ *
++ * Copyright (C) 2012 Ed W and David Rowe
++ *
++ * 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
++ *
++ * \ingroup codecs
++ */
++
++/*** MODULEINFO
++ <depend>codec2</depend>
++ <support_level>core</support_level>
++ ***/
++
++#include "asterisk.h"
++
++#include "asterisk/translate.h"
++#include "asterisk/config.h"
++#include "asterisk/module.h"
++#include "asterisk/utils.h"
++
++#include <codec2/codec2.h>
++
++#define BUFFER_SAMPLES 8000
++#define CODEC2_SAMPLES 160
++#define CODEC2_FRAME_LEN 6
++
++/* Sample frame data */
++
++#include "asterisk/slin.h"
++#include "ex_codec2.h"
++
++struct codec2_translator_pvt { /* both codec2tolin and codec2togsm */
++ struct CODEC2 *codec2;
++ int16_t buf[BUFFER_SAMPLES]; /* lintocodec2, temporary storage */
++};
++
++static int codec2_new(struct ast_trans_pvt *pvt)
++{
++ struct codec2_translator_pvt *tmp = pvt->pvt;
++
++ tmp->codec2 = codec2_create(CODEC2_MODE_2400);
++ if (!tmp)
++ 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;
++ int16_t *dst = pvt->outbuf.i16;
++ int flen = CODEC2_FRAME_LEN;
++
++ for (x=0; x < f->datalen; x += flen) {
++ unsigned char *src;
++ int len;
++ len = CODEC2_SAMPLES;
++ src = f->data.ptr + x;
++
++ codec2_decode(tmp->codec2, dst + pvt->samples, src);
++
++ pvt->samples += CODEC2_SAMPLES;
++ pvt->datalen += 2 * CODEC2_SAMPLES;
++ }
++ 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;
++
++ if (pvt->samples + f->samples > BUFFER_SAMPLES) {
++ ast_log(LOG_WARNING, "Out of buffer space\n");
++ return -1;
++ }
++ 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;
++ int datalen = 0;
++ int samples = 0;
++
++ /* We can't work on anything less than a frame in size */
++ if (pvt->samples < CODEC2_SAMPLES)
++ return NULL;
++ while (pvt->samples >= CODEC2_SAMPLES) {
++ /* Encode a frame of data */
++ codec2_encode(tmp->codec2, (unsigned char*)(pvt->outbuf.c + datalen), tmp->buf + samples);
++ datalen += CODEC2_FRAME_LEN;
++ samples += CODEC2_SAMPLES;
++ pvt->samples -= CODEC2_SAMPLES;
++ }
++
++ /* Move the data at the end of the buffer to the front */
++ if (pvt->samples)
++ memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2);
++
++ return ast_trans_frameout(pvt, datalen, samples);
++}
++
++static void codec2_destroy_stuff(struct ast_trans_pvt *pvt)
++{
++ struct codec2_translator_pvt *tmp = pvt->pvt;
++ if (tmp->codec2)
++ codec2_destroy(tmp->codec2);
++}
++
++static struct ast_translator codec2tolin = {
++ .name = "codec2tolin",
++ .newpvt = codec2_new,
++ .framein = codec2tolin_framein,
++ .destroy = codec2_destroy_stuff,
++ .sample = codec2_sample,
++ .buffer_samples = BUFFER_SAMPLES,
++ .buf_size = BUFFER_SAMPLES * 2,
++ .desc_size = sizeof (struct codec2_translator_pvt ),
++};
++
++static struct ast_translator lintocodec2 = {
++ .name = "lintocodec2",
++ .newpvt = codec2_new,
++ .framein = lintocodec2_framein,
++ .frameout = lintocodec2_frameout,
++ .destroy = codec2_destroy_stuff,
++ .sample = slin8_sample,
++ .desc_size = sizeof (struct codec2_translator_pvt ),
++ .buf_size = (BUFFER_SAMPLES * CODEC2_FRAME_LEN + CODEC2_SAMPLES - 1)/CODEC2_SAMPLES,
++};
++
++/*! \brief standard module glue */
++static int reload(void)
++{
++ return AST_MODULE_LOAD_SUCCESS;
++}
++
++static int unload_module(void)
++{
++ int res;
++
++ res = ast_unregister_translator(&lintocodec2);
++ if (!res)
++ res = ast_unregister_translator(&codec2tolin);
++
++ return res;
++}
++
++static int load_module(void)
++{
++ int res;
++
++ ast_format_set(&codec2tolin.src_format, AST_FORMAT_CODEC2, 0);
++ ast_format_set(&codec2tolin.dst_format, AST_FORMAT_SLINEAR, 0);
++
++ ast_format_set(&lintocodec2.src_format, AST_FORMAT_SLINEAR, 0);
++ ast_format_set(&lintocodec2.dst_format, AST_FORMAT_CODEC2, 0);
++
++ res = ast_register_translator(&codec2tolin);
++ if (!res)
++ res=ast_register_translator(&lintocodec2);
++ else
++ ast_unregister_translator(&codec2tolin);
++ if (res)
++ return AST_MODULE_LOAD_FAILURE;
++ return AST_MODULE_LOAD_SUCCESS;
++}
++
++AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Codec 2 Coder/Decoder",
++ .load = load_module,
++ .unload = unload_module,
++ .reload = reload,
++ );
+diff -urN asterisk-11.8.1-opus/codecs/ex_codec2.h asterisk-11.8.1-opus-codec2/codecs/ex_codec2.h
+--- asterisk-11.8.1-opus/codecs/ex_codec2.h 1970-01-01 01:00:00.000000000 +0100
++++ asterisk-11.8.1-opus-codec2/codecs/ex_codec2.h 2014-04-01 10:50:21.490079146 +0200
+@@ -0,0 +1,29 @@
++/*! \file
++ * \brief 8-bit raw data
++ *
++ * Copyright (C) 2012, 2012 Ed W and David Rowe
++ *
++ * Distributed under the terms of the GNU General Public License
++ *
++ */
++
++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,
++ };
++
++ ast_format_set(&f.subclass.format, AST_FORMAT_CODEC2, 0);
++
++ return &f;
++}
+diff -urN asterisk-11.8.1-opus/configure.ac asterisk-11.8.1-opus-codec2/configure.ac
+--- asterisk-11.8.1-opus/configure.ac 2014-04-01 10:58:52.143106352 +0200
++++ asterisk-11.8.1-opus-codec2/configure.ac 2014-04-01 10:50:21.494079217 +0200
+@@ -384,6 +384,7 @@
+ 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], [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])
+@@ -2117,6 +2118,8 @@
+
+ AST_EXT_LIB_CHECK([RADIUS], [radiusclient-ng], [rc_read_config], [radiusclient-ng.h])
+
++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 -urN asterisk-11.8.1-opus/include/asterisk/format.h asterisk-11.8.1-opus-codec2/include/asterisk/format.h
+--- asterisk-11.8.1-opus/include/asterisk/format.h 2014-04-01 10:58:52.143106352 +0200
++++ asterisk-11.8.1-opus-codec2/include/asterisk/format.h 2014-04-01 10:53:25.961351185 +0200
+@@ -103,6 +103,7 @@
+ AST_FORMAT_CELT = 29 + AST_FORMAT_TYPE_AUDIO,
+ /*! Opus */
+ AST_FORMAT_OPUS = 30 + AST_FORMAT_TYPE_AUDIO,
++ AST_FORMAT_CODEC2 = 31 + AST_FORMAT_TYPE_AUDIO,
+
+ /*! H.261 Video */
+ AST_FORMAT_H261 = 1 + AST_FORMAT_TYPE_VIDEO,
+diff -urN asterisk-11.8.1-opus/main/channel.c asterisk-11.8.1-opus-codec2/main/channel.c
+--- asterisk-11.8.1-opus/main/channel.c 2014-04-01 10:58:52.147106422 +0200
++++ asterisk-11.8.1-opus-codec2/main/channel.c 2014-04-01 10:50:21.498079289 +0200
+@@ -920,6 +920,8 @@
+ AST_FORMAT_SILK,
+ /*! CELT supports crazy high sample rates */
+ AST_FORMAT_CELT,
++ /* Codec 2 */
++ AST_FORMAT_CODEC2,
+ /*! Ick, LPC10 sounds terrible, but at least we have code for it, if you're tacky enough
+ to use it */
+ AST_FORMAT_LPC10,
+diff -urN asterisk-11.8.1-opus/main/format.c asterisk-11.8.1-opus-codec2/main/format.c
+--- asterisk-11.8.1-opus/main/format.c 2014-04-01 10:58:52.147106422 +0200
++++ asterisk-11.8.1-opus-codec2/main/format.c 2014-04-01 10:52:23.820250398 +0200
+@@ -433,6 +433,9 @@
+ /*! Opus audio (8kHz, 16kHz, 24kHz, 48Khz) */
+ case AST_FORMAT_OPUS:
+ return (1ULL << 34);
++ /*! Codec 2 (8KHz) */
++ case AST_FORMAT_CODEC2:
++ return (1ULL << 35);
+ /*! Raw mu-law data (G.711) */
+ case AST_FORMAT_TESTLAW:
+ return (1ULL << 47);
+@@ -541,6 +544,9 @@
+ /*! Opus audio (8kHz, 16kHz, 24kHz, 48Khz) */
+ case (1ULL << 34):
+ return ast_format_set(dst, AST_FORMAT_OPUS, 0);
++ /*! Codec 2 (8KHz) */
++ case (1ULL << 35):
++ return ast_format_set(dst, AST_FORMAT_CODEC2, 0);
+ /*! Raw mu-law data (G.711) */
+ case (1ULL << 47):
+ return ast_format_set(dst, AST_FORMAT_TESTLAW, 0);
+@@ -1090,6 +1096,8 @@
+ format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), "opus", 48000, "Opus Codec", 10, 20, 60, 20, 20, 0, 0); /*!< codec_opus.c */
+ /* VP8 (passthrough) */
+ format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), "vp8", 0, "VP8 Video", 0, 0, 0, 0 ,0 ,0, 0); /*!< Passthrough support, see format_h263.c */
++ /* Codec 2 */
++ format_list_add_static(ast_format_set(&tmpfmt, AST_FORMAT_CODEC2, 0), "codec2", 8000, "Codec 2", 6, 20, 20, 20, 20, 0, 0); /*!< codec_codec2.c */
+
+ return 0;
+ }
+diff -urN asterisk-11.8.1-opus/main/frame.c asterisk-11.8.1-opus-codec2/main/frame.c
+--- asterisk-11.8.1-opus/main/frame.c 2014-04-01 10:58:52.151106493 +0200
++++ asterisk-11.8.1-opus-codec2/main/frame.c 2014-04-01 10:50:21.502079360 +0200
+@@ -1121,6 +1121,10 @@
+ case AST_FORMAT_OPUS:
+ samples = opus_samples(f->data.ptr, f->datalen);
+ break;
++ /* Codec 2 */
++ case AST_FORMAT_CODEC2:
++ samples = 160 * (f->datalen / 6);
++ break;
+ default:
+ ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(&f->subclass.format));
+ }
+@@ -1172,6 +1176,10 @@
+ /* 48,000 samples per second at 64kbps is 8,000 bytes per second */
+ len = (int) samples / ((float) 48000 / 8000);
+ break;
++ /* Codec 2 */
++ case AST_FORMAT_CODEC2:
++ len = (samples / 160) * 6;
++ break;
+ default:
+ ast_log(LOG_WARNING, "Unable to calculate sample length for format %s\n", ast_getformatname(format));
+ }
+diff -urN asterisk-11.8.1-opus/main/rtp_engine.c asterisk-11.8.1-opus-codec2/main/rtp_engine.c
+--- asterisk-11.8.1-opus/main/rtp_engine.c 2014-04-01 10:58:52.151106493 +0200
++++ asterisk-11.8.1-opus-codec2/main/rtp_engine.c 2014-04-01 10:51:17.367071621 +0200
+@@ -2292,6 +2292,8 @@
+ /* Opus and VP8 */
+ set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0, "audio", "opus", 48000);
+ set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0, "video", "VP8", 90000);
++ /* Codec 2 */
++ set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_CODEC2, 0), 0, "audio", "CODEC2", 8000);
+
+ /* Define the static rtp payload mappings */
+ add_static_payload(0, ast_format_set(&tmpfmt, AST_FORMAT_ULAW, 0), 0);
+@@ -2336,6 +2338,8 @@
+ /* Opus and VP8 */
+ add_static_payload(100, ast_format_set(&tmpfmt, AST_FORMAT_VP8, 0), 0);
+ add_static_payload(107, ast_format_set(&tmpfmt, AST_FORMAT_OPUS, 0), 0);
++ /* Codec 2 */
++ add_static_payload(121, ast_format_set(&tmpfmt, AST_FORMAT_CODEC2, 0), 0);
+
+ return 0;
+ }
+diff -urN asterisk-11.8.1-opus/makeopts.in asterisk-11.8.1-opus-codec2/makeopts.in
+--- asterisk-11.8.1-opus/makeopts.in 2014-04-01 10:58:52.151106493 +0200
++++ asterisk-11.8.1-opus-codec2/makeopts.in 2014-04-01 10:50:21.538079999 +0200
+@@ -120,6 +120,9 @@
+ BLUETOOTH_INCLUDE=@BLUETOOTH_INCLUDE@
+ BLUETOOTH_LIB=@BLUETOOTH_LIB@
+
++CODEC2_INCLUDE=@CODEC2_INCLUDE@
++CODEC2_LIB=@CODEC2_LIB@
++
+ CURL_INCLUDE=@CURL_INCLUDE@
+ CURL_LIB=@CURL_LIB@
+
+diff -urN asterisk-11.8.1-opus/res/res_rtp_asterisk.c asterisk-11.8.1-opus-codec2/res/res_rtp_asterisk.c
+--- asterisk-11.8.1-opus/res/res_rtp_asterisk.c 2014-04-01 10:58:52.151106493 +0200
++++ asterisk-11.8.1-opus-codec2/res/res_rtp_asterisk.c 2014-04-01 10:53:02.292932085 +0200
+@@ -2780,6 +2780,8 @@
+ case AST_FORMAT_G719:
+ /* Opus */
+ case AST_FORMAT_OPUS:
++ /* Codec 2 */
++ case AST_FORMAT_CODEC2:
+ /* these are all frame-based codecs and cannot be safely run through
+ a smoother */
+ break;
projects / freetel-svn-tracking.git / commitdiff