codecs: Add Codec 2 mode 2400.

ASTERISK-26217 #close

Change-Id: I1e45d8084683fab5f2b272bf35f4a149cea8b8d6

2 files changed, 254 insertions, 0 deletions

diff --git a/codecs/codec_codec2.c b/codecs/codec_codec2.c
new file mode 100644
index 000000000..e446854c3
--- /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@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@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 000000000..f2f4c9723
--- /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;
+}