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Merge changes from team/russell/codec_resample
This commit imports libresample for use in Asterisk. It also adds a new codec module, codec_resample. This module uses libresample to re-sample signed linear audio between 8 kHz and 16 kHz. It also provides an alternative for converting between 16 kHz G.722 and 8 kHz signed linear when using G.722, which will likely be useful as some people have complained about volume issues when the current codec_g722 converts to 8 kHz signed linear. But, to test this, you will have to disable the g722-to-slin and g722-to-slin16 translators in codec_g722.c. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
234
codecs/codec_resample.c
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234
codecs/codec_resample.c
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2007, Digium, Inc.
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*
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* Russell Bryant <russell@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*!
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* \file
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*
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* \brief Resample slinear audio
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*
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* \ingroup codecs
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*/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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/* These are for SHRT_MAX and FLT_MAX -- { */
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#ifdef __Darwin__
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#include <float.h>
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#else
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#include <values.h>
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#endif
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#include <limits.h>
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/* } */
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#include "asterisk/module.h"
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#include "asterisk/translate.h"
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#include "libresample.h"
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#include "slin_resample_ex.h"
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#define RESAMPLER_QUALITY 0
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#define OUTBUF_SIZE 8096
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struct slin16_to_slin8_pvt {
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void *resampler;
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double resample_factor;
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};
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struct slin8_to_slin16_pvt {
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void *resampler;
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double resample_factor;
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};
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static int slin16_to_slin8_new(struct ast_trans_pvt *pvt)
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{
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struct slin16_to_slin8_pvt *resamp_pvt = pvt->pvt;
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resamp_pvt->resample_factor = 0.5;
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if (!(resamp_pvt->resampler = resample_open(RESAMPLER_QUALITY, 0.5, 0.5)))
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return -1;
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return 0;
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}
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static int slin8_to_slin16_new(struct ast_trans_pvt *pvt)
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{
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struct slin8_to_slin16_pvt *resamp_pvt = pvt->pvt;
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resamp_pvt->resample_factor = 2.0;
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if (!(resamp_pvt->resampler = resample_open(RESAMPLER_QUALITY, 2.0, 2.0)))
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return -1;
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return 0;
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}
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static void slin16_to_slin8_destroy(struct ast_trans_pvt *pvt)
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{
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struct slin16_to_slin8_pvt *resamp_pvt = pvt->pvt;
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if (resamp_pvt->resampler)
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resample_close(resamp_pvt->resampler);
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}
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static void slin8_to_slin16_destroy(struct ast_trans_pvt *pvt)
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{
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struct slin8_to_slin16_pvt *resamp_pvt = pvt->pvt;
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if (resamp_pvt->resampler)
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resample_close(resamp_pvt->resampler);
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}
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static int resample_frame(struct ast_trans_pvt *pvt,
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void *resampler, double resample_factor, struct ast_frame *f)
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{
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int total_in_buf_used = 0;
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int total_out_buf_used = 0;
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int16_t *in_buf = (int16_t *) f->data;
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int16_t *out_buf = (int16_t *) pvt->outbuf + pvt->samples;
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float in_buf_f[f->samples];
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float out_buf_f[2048];
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int res = 0;
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int i;
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for (i = 0; i < f->samples; i++)
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in_buf_f[i] = in_buf[i] * (FLT_MAX / SHRT_MAX);
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while (total_in_buf_used < f->samples) {
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int in_buf_used, out_buf_used;
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out_buf_used = resample_process(resampler, resample_factor,
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&in_buf_f[total_in_buf_used], f->samples - total_in_buf_used,
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0, &in_buf_used,
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&out_buf_f[total_out_buf_used], ARRAY_LEN(out_buf_f) - total_out_buf_used);
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if (out_buf_used < 0)
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break;
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total_out_buf_used += out_buf_used;
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total_in_buf_used += in_buf_used;
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if (total_out_buf_used == ARRAY_LEN(out_buf_f)) {
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ast_log(LOG_ERROR, "Output buffer filled ... need to increase its size\n");
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res = -1;
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break;
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}
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}
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for (i = 0; i < total_out_buf_used; i++)
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out_buf[i] = out_buf_f[i] * (SHRT_MAX / FLT_MAX);
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pvt->samples += total_out_buf_used;
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pvt->datalen += (total_out_buf_used * sizeof(int16_t));
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return res;
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}
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static int slin16_to_slin8_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
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struct slin16_to_slin8_pvt *resamp_pvt = pvt->pvt;
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return resample_frame(pvt, resamp_pvt->resampler, resamp_pvt->resample_factor, f);
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}
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static int slin8_to_slin16_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
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struct slin8_to_slin16_pvt *resamp_pvt = pvt->pvt;
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return resample_frame(pvt, resamp_pvt->resampler, resamp_pvt->resample_factor, f);
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}
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static struct ast_frame *slin16_to_slin8_sample(void)
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{
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static struct ast_frame f = {
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.frametype = AST_FRAME_VOICE,
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.subclass = AST_FORMAT_SLINEAR16,
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.datalen = sizeof(slin16_slin8_ex),
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.samples = sizeof(slin16_slin8_ex) / sizeof(slin16_slin8_ex[0]),
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.src = __PRETTY_FUNCTION__,
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.data = slin16_slin8_ex,
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};
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return &f;
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}
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static struct ast_frame *slin8_to_slin16_sample(void)
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{
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static struct ast_frame f = {
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.frametype = AST_FRAME_VOICE,
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.subclass = AST_FORMAT_SLINEAR,
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.datalen = sizeof(slin8_slin16_ex),
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.samples = sizeof(slin8_slin16_ex) / sizeof(slin8_slin16_ex[0]),
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.src = __PRETTY_FUNCTION__,
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.data = slin8_slin16_ex,
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};
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return &f;
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}
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static struct ast_translator slin16_to_slin8 = {
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.name = "slin16_to_slin8",
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.srcfmt = AST_FORMAT_SLINEAR16,
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.dstfmt = AST_FORMAT_SLINEAR,
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.newpvt = slin16_to_slin8_new,
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.destroy = slin16_to_slin8_destroy,
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.framein = slin16_to_slin8_framein,
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.sample = slin16_to_slin8_sample,
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.desc_size = sizeof(struct slin16_to_slin8_pvt),
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.buffer_samples = (OUTBUF_SIZE / sizeof(int16_t)),
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.buf_size = OUTBUF_SIZE,
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};
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static struct ast_translator slin8_to_slin16 = {
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.name = "slin8_to_slin16",
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.srcfmt = AST_FORMAT_SLINEAR,
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.dstfmt = AST_FORMAT_SLINEAR16,
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.newpvt = slin8_to_slin16_new,
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.destroy = slin8_to_slin16_destroy,
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.framein = slin8_to_slin16_framein,
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.sample = slin8_to_slin16_sample,
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.desc_size = sizeof(struct slin8_to_slin16_pvt),
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.buffer_samples = OUTBUF_SIZE,
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.buf_size = OUTBUF_SIZE,
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};
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static int unload_module(void)
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{
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int res = 0;
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res |= ast_unregister_translator(&slin16_to_slin8);
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res |= ast_unregister_translator(&slin8_to_slin16);
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return res;
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}
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static int load_module(void)
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{
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int res = 0;
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res |= ast_register_translator(&slin16_to_slin8);
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res |= ast_register_translator(&slin8_to_slin16);
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return AST_MODULE_LOAD_SUCCESS;
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "SLIN Resampling Codec");
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