Codec 2 README
--------------
-Codec 2 is an open source (LGPL licensed) speech codec for 3200 bit/s
-and below. For more information please see:
+Codec 2 is an open source (LGPL 2.1) low bit rate speech codec:
http://rowetel.com/codec2.html
-Also included is a FDMDV modem (README_fdmdv.txt), a coherent PSK
-modem and an API for embedding FreeDV in other programs (see example
-below). For more information on building Codec 2 see README.cmake
+Also included:
+
+ + FDMDV modem (README_fdmdv.txt) for HF channels
+ + Coherent PSK (cohpsk) for HF channels
+ + FSK modem (README_fsk.txt) for VHF channels
+ + an OFDM modem for HF channels (ofdm)
+ + the FreeDV API - a library for embedding FreeDV in other programs
SVN Repository
--------------
$ svn co https://svn.code.sf.net/p/freetel/code/codec2-dev codec2-dev
-There are 3rd party GIT mirrors of Codec2 and FreeDV. Use Git at your
-own risk.
+There are unauthorised 3rd party GIT mirrors of Codec 2.
GIT IS NOT SUPPORTED!!!
All patches, support questions etc, need to be against the SVN
repository above.
-Please do not email me (David), or the codec2-dev mailing list
-suggesting we change to Git. I get these emails every week. Really,
-I understand the arguments, but am content with SVN for now.
-
Quickstart
----------
-Refer to INSTALL for more general building and installing instructions.
-
-NOTE: You may not have the necessary speex libraries installed. Some
- distributions may have packages available
- i.e. sudo apt-get install speex-*
- alternatively, visit http://www.speex.org and follow the instructions
+Also see INSTALL for more general building and installing instructions.
1/ Listen to Codec 2:
Building Unit Tests
-------------------
-The unittests are no longer built by default. To build them:
+The unittests are no longer built by default. They require
+libsamplerate and Speex.
+
+To build them:
$ cd ~/codec2
$ rm -Rf build_linux && mkdir build_linux
$ cmake -DCMAKE_BUILD_TYPE=Debug ..
$ make
-
Directories
-----------
- asterisk & - unmaintained Asterisk support
- asterisk-11
cmake - cmake support files
octave - Octave scripts used to support development
script - shell scripts for playing and converting raw files
src - C source code for Codec 2, FDMDV modem, COHPSK modem, FreeDV API
raw - speech files in raw format (16 bits signed linear 8 kHz)
- stm32 - Support for the STM32F4 microcontroller and SM1000 FreeDV Adaptor
+ stm32 - STM32F4 microcontroller and SM1000 FreeDV Adaptor
unittest - unit test source code
wav - speech files in wave file format
+Octave Packages
+---------------
+
+To run the Octave scripts the following libraries are required:
+
+Package Name | Version | Installation directory
+--------------+---------+-----------------------
+ control *| 2.6.2 | /usr/share/octave/packages/control-2.6.2
+ general *| 1.3.4 | /usr/share/octave/packages/general-1.3.4
+ parallel *| 2.2.0 | /usr/share/octave/packages/parallel-2.2.0
+ plot *| 1.1.0 | /usr/share/octave/packages/plot-1.1.0
+ signal *| 1.2.2 | /usr/share/octave/packages/signal-1.2.2
+ specfun *| 1.1.0 | /usr/share/octave/packages/specfun-1.1.0
+
+These can be installed using your systems package management system or
+the Octave package management system. The version number of each
+package is not important.
+
+++ /dev/null
-#Asterisk 11 Codec 2 support
-===========================
-
-##Description
-These patches add Codec 2 2400 support to Asterisk 11.
-The following patches are provided:
-
-* asterisk-11.8.1-codec2.patch: plain Asterisk 11.
-* asterisk-11.8.1-opus-codec2.patch: Asterisk patched with Meetecho's Opus codec support.
-
-##Building
-Building and installing are integrated within Asterisk building environment. libcodec2 must be installed beforehand.
-
-##Credits
-I've followed the example of [asterisk-opus](https://github.com/meetecho/asterisk-opus), by [@meetecho](https://github.com/meetecho), to adapt Codec2 Asterisk 1.8 patch to version 11.
-
-Many thanks to the [Codec2](http://www.rowetel.com/blog/codec2.html) team for developing such great codec!
-
-Developed by [Antonio Eugenio Burriel](https://github.com/aeburriel)
+++ /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;
+++ /dev/null
-README for codec2/asterisk
-Asterisk Codec 2 support
-
-Test Configuration
-------------------
-
-Codec 2 is used to trunk calls between two Asterisk boxes:
-
- A - SIP phone - Asterisk A - Codec2 - Asterisk B - SIP Phone - B
-
-The two SIP phones are configured for mulaw.
-
-Building
----------
-
-Asterisk must be patched so that the core understand Codec 2 frames.
-
-1/ First install Codec 2:
-
- david@cool:~$ svn co https://freetel.svn.sourceforge.net/svnroot/freetel/codec2-dev codec2-dev
- david@cool:~/codec2-dev$ cd codec2-dev
- david@cool:~/codec2-dev$ ./configure && make && sudo make install
- david@bear:~/codec2-dev$ sudo ldconfig -v
- david@cool:~/codec2-dev$ cd ~
-
-2/ Then build Asterisk with Codec 2 support:
-
- david@cool:~$ tar xvzf asterisk-1.8.9.0.tar.gz
- david@cool:~/asterisk-1.8.9.0$ patch -p4 < ~/codec2-dev/asterisk/asterisk-codec2.patch
- david@cool:~/asterisk-1.8.9.0$ cp ~/codec2-dev/asterisk/codec_codec2.c .
- david@cool:~/asterisk-1.8.9.0$ cp ~/codec2-dev/asterisk/ex_codec2.h ./codecs
- david@cool:~/asterisk-1.8.9.0$ ./configure && make ASTLDFLAGS=-lcodec2
- david@cool:~/asterisk-1.8.9.0$ sudo make install
- david@cool:~/asterisk-1.8.9.0$ sudo make samples
-
-3/ Add this to the end of sip.conf on Asterisk A:
-
- [6013]
- type=friend
- context=default
- host=dynamic
- user=6013
- secret=6013
- canreinvite=no
- callerid=6013
- disallow=all
- allow=ulaw
-
- [potato]
- type=peer
- username=potato
- fromuser=potato
- secret=password
- context=default
- disallow=all
- dtmfmode=rfc2833
- callerid=server
- canreinvite=no
- host=cool
- allow=codec2
-
-3/ Add this to the end of sip.conf on Asterisk B:
-
- [6014]
- type=friend
- context=default
- host=dynamic
- user=6014
- secret=6014
- canreinvite=no
- callerid=6014
- disallow=all
- allow=ulaw
-
- [potato]
- type=peer
- username=potato
- fromuser=potato
- secret=password
- context=default
- disallow=all
- dtmfmode=rfc2833
- callerid=server
- canreinvite=no
- host=bear
- allow=codec2
-
-4/ Here is the [default] section of extensions.conf on Asterisk B:
-
- [default]
-
- exten => 6013,1,Dial(SIP/potato/6013)
- ;
- ; By default we include the demo. In a production system, you
- ; probably don't want to have the demo there.
- ;
- ;include => demo
-
-5/ After booting see if the codec2_codec2.so module is loaded with "core show translate"
-
-6/ To make a test call dial 6013 on the SIP phone connected to Asterisk B
-
-7/ If codec_codec2.so won't load and you see "can't find codec2_create" try:
-
- david@cool:~/asterisk-1.8.9.0$ touch codecs/codec_codec2.c
- david@cool:~/asterisk-1.8.9.0$ make ASTLDFLAGS=-lcodec2
- david@cool:~/asterisk-1.8.9.0$ sudo cp codecs/codec_codec2.so /usr/lib/asterisk/modules
- david@cool:~/asterisk-1.8.9.0$ sudo asterisk -vvvcn
-
+++ /dev/null
---- /home/david/asterisk-1.8.9.0-orig/include/asterisk/frame.h 2011-12-23 05:08:46.000000000 +1030
-+++ /home/david/asterisk-1.8.9.0-codec2/include/asterisk/frame.h 2012-03-27 13:16:55.623452431 +1030
-@@ -299,6 +299,7 @@
- #define AST_FORMAT_G719 (1ULL << 32)
- /*! SpeeX Wideband (16kHz) Free Compression */
- #define AST_FORMAT_SPEEX16 (1ULL << 33)
-+#define AST_FORMAT_CODEC2 (1ULL << 34)
- /*! Raw mu-law data (G.711) */
- #define AST_FORMAT_TESTLAW (1ULL << 47)
- /*! Reserved bit - do not use */
---- /home/david/asterisk-1.8.9.0-orig/main/frame.c 2010-06-18 02:53:43.000000000 +0930
-+++ /home/david/asterisk-1.8.9.0-codec2/main/frame.c 2012-03-28 15:16:16.975581316 +1030
-@@ -102,6 +102,7 @@
- { AST_FORMAT_ADPCM, "adpcm" , 8000, "ADPCM", 40, 10, 300, 10, 20 }, /*!< codec_adpcm.c */
- { AST_FORMAT_SLINEAR, "slin", 8000, "16 bit Signed Linear PCM", 160, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE }, /*!< Signed linear */
- { AST_FORMAT_LPC10, "lpc10", 8000, "LPC10", 7, 20, 20, 20, 20 }, /*!< codec_lpc10.c */
-+ { AST_FORMAT_CODEC2, "codec2", 8000, "Codec 2", 6, 20, 20, 20, 20 }, /*!< codec_codec2.c */
- { AST_FORMAT_G729A, "g729", 8000, "G.729A", 10, 10, 230, 10, 20, AST_SMOOTHER_FLAG_G729 }, /*!< Binary commercial distribution */
- { AST_FORMAT_SPEEX, "speex", 8000, "SpeeX", 10, 10, 60, 10, 20 }, /*!< codec_speex.c */
- { AST_FORMAT_SPEEX16, "speex16", 16000, "SpeeX 16khz", 10, 10, 60, 10, 20 }, /*!< codec_speex.c */
-@@ -1475,6 +1476,9 @@
- samples = 22 * 8;
- samples += (((char *)(f->data.ptr))[7] & 0x1) * 8;
- break;
-+ case AST_FORMAT_CODEC2:
-+ samples = 160 * (f->datalen / 6);
-+ break;
- case AST_FORMAT_ULAW:
- case AST_FORMAT_ALAW:
- case AST_FORMAT_TESTLAW:
-@@ -1519,6 +1523,9 @@
- case AST_FORMAT_GSM:
- len = (samples / 160) * 33;
- break;
-+ case AST_FORMAT_CODEC2:
-+ len = (samples / 160) * 6;
-+ break;
- case AST_FORMAT_G729A:
- len = samples / 8;
- break;
---- /home/david/asterisk-1.8.9.0-orig/main/channel.c 2011-12-17 10:21:13.000000000 +1030
-+++ /home/david/asterisk-1.8.9.0-codec2/main/channel.c 2012-03-28 15:05:22.395293391 +1030
-@@ -1075,6 +1075,7 @@
- /*! Ick, LPC10 sounds terrible, but at least we have code for it, if you're tacky enough
- to use it */
- AST_FORMAT_LPC10,
-+ AST_FORMAT_CODEC2,
- /*! G.729a is faster than 723 and slightly less expensive */
- AST_FORMAT_G729A,
- /*! Down to G.723.1 which is proprietary but at least designed for voice */
---- /home/david/asterisk-1.8.9.0-orig/main/rtp_engine.c 2011-12-30 01:43:03.000000000 +1030
-+++ /home/david/asterisk-1.8.9.0-codec2/main/rtp_engine.c 2012-03-28 16:42:02.880872891 +1030
-@@ -101,6 +101,7 @@
- {{1, AST_FORMAT_SLINEAR}, "audio", "L16", 8000},
- {{1, AST_FORMAT_SLINEAR16}, "audio", "L16", 16000},
- {{1, AST_FORMAT_LPC10}, "audio", "LPC", 8000},
-+ {{1, AST_FORMAT_CODEC2}, "audio", "CODEC2", 8000},
- {{1, AST_FORMAT_G729A}, "audio", "G729", 8000},
- {{1, AST_FORMAT_G729A}, "audio", "G729A", 8000},
- {{1, AST_FORMAT_G729A}, "audio", "G.729", 8000},
-@@ -178,6 +179,7 @@
- [117] = {1, AST_FORMAT_SPEEX16},
- [118] = {1, AST_FORMAT_SLINEAR16}, /* 16 Khz signed linear */
- [121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
-+ [121] = {1, AST_FORMAT_CODEC2},
- };
-
- int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)
+++ /dev/null
-/*
- * 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
- <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.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->codec2)
- 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",
- .srcfmt = AST_FORMAT_CODEC2,
- .dstfmt = AST_FORMAT_SLINEAR,
- .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",
- .srcfmt = AST_FORMAT_SLINEAR,
- .dstfmt = AST_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 ),
- .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;
-
- 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,
- );
+++ /dev/null
-/*! \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,
- .subclass.codec = AST_FORMAT_CODEC2,
- .datalen = sizeof(ex_codec2),
- .samples = CODEC2_SAMPLES,
- .mallocd = 0,
- .offset = 0,
- .src = __PRETTY_FUNCTION__,
- .data.ptr = ex_codec2,
- };
-
- return &f;
-}
+++ /dev/null
-#!/bin/sh
-# Create patch for Codec 2 support inside Asterisk
-
-ORIG=~/asterisk-1.8.9.0-orig
-CODEC2=~/asterisk-1.8.9.0-codec2
-diff -ruN $ORIG/include/asterisk/frame.h $CODEC2/include/asterisk/frame.h > asterisk-codec2.patch
-diff -ruN $ORIG/main/frame.c $CODEC2/main/frame.c >> asterisk-codec2.patch
-diff -ruN $ORIG/main/channel.c $CODEC2/main/channel.c >> asterisk-codec2.patch
-diff -ruN $ORIG/main/rtp_engine.c $CODEC2/main/rtp_engine.c >> asterisk-codec2.patch
-
-
projects / freetel-svn-tracking.git / commitdiff