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- /*
- * 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_DECLINE;
- }
- return AST_MODULE_LOAD_SUCCESS;
- }
- AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Codec 2 Coder/Decoder");
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