diff options
author | Joshua Colp <jcolp@digium.com> | 2016-09-04 14:11:34 -0500 |
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committer | Gerrit Code Review <gerrit2@gerrit.digium.api> | 2016-09-04 14:11:34 -0500 |
commit | e34f299a9672f829ee8d6652d221700fe2c6c56d (patch) | |
tree | 06e7940a019b1177c7aadbe7810c9e34a6ea17f7 /codecs | |
parent | f87008f11adb6652f13fbca965a7b580ab0dac93 (diff) | |
parent | 2e79f52d7116e5529ab78972cee8081b6ffe6878 (diff) |
Merge "codecs: Add Codec 2 mode 2400."
Diffstat (limited to 'codecs')
-rw-r--r-- | codecs/codec_codec2.c | 222 | ||||
-rw-r--r-- | codecs/ex_codec2.h | 32 |
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; +} |