ffmpeg/libavcodec/libgsm.c

223 lines
6.9 KiB
C

/*
* Interface to libgsm for gsm encoding/decoding
* Copyright (c) 2005 Alban Bedel <albeu@free.fr>
* Copyright (c) 2006, 2007 Michel Bardiaux <mbardiaux@mediaxim.be>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Interface to libgsm for gsm encoding/decoding
*/
// The idiosyncrasies of GSM-in-WAV are explained at http://kbs.cs.tu-berlin.de/~jutta/toast.html
#include "avcodec.h"
#include <gsm/gsm.h>
// gsm.h misses some essential constants
#define GSM_BLOCK_SIZE 33
#define GSM_MS_BLOCK_SIZE 65
#define GSM_FRAME_SIZE 160
static av_cold int libgsm_encode_init(AVCodecContext *avctx) {
if (avctx->channels > 1) {
av_log(avctx, AV_LOG_ERROR, "Mono required for GSM, got %d channels\n",
avctx->channels);
return -1;
}
if (avctx->sample_rate != 8000) {
av_log(avctx, AV_LOG_ERROR, "Sample rate 8000Hz required for GSM, got %dHz\n",
avctx->sample_rate);
if (avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL)
return -1;
}
if (avctx->bit_rate != 13000 /* Official */ &&
avctx->bit_rate != 13200 /* Very common */ &&
avctx->bit_rate != 0 /* Unknown; a.o. mov does not set bitrate when decoding */ ) {
av_log(avctx, AV_LOG_ERROR, "Bitrate 13000bps required for GSM, got %dbps\n",
avctx->bit_rate);
if (avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL)
return -1;
}
avctx->priv_data = gsm_create();
switch(avctx->codec_id) {
case CODEC_ID_GSM:
avctx->frame_size = GSM_FRAME_SIZE;
avctx->block_align = GSM_BLOCK_SIZE;
break;
case CODEC_ID_GSM_MS: {
int one = 1;
gsm_option(avctx->priv_data, GSM_OPT_WAV49, &one);
avctx->frame_size = 2*GSM_FRAME_SIZE;
avctx->block_align = GSM_MS_BLOCK_SIZE;
}
}
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
return 0;
}
static av_cold int libgsm_encode_close(AVCodecContext *avctx) {
av_freep(&avctx->coded_frame);
gsm_destroy(avctx->priv_data);
avctx->priv_data = NULL;
return 0;
}
static int libgsm_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data) {
// we need a full block
if(buf_size < avctx->block_align) return 0;
switch(avctx->codec_id) {
case CODEC_ID_GSM:
gsm_encode(avctx->priv_data,data,frame);
break;
case CODEC_ID_GSM_MS:
gsm_encode(avctx->priv_data,data,frame);
gsm_encode(avctx->priv_data,((short*)data)+GSM_FRAME_SIZE,frame+32);
}
return avctx->block_align;
}
AVCodec ff_libgsm_encoder = {
.name = "libgsm",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_GSM,
.init = libgsm_encode_init,
.encode = libgsm_encode_frame,
.close = libgsm_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
};
AVCodec ff_libgsm_ms_encoder = {
.name = "libgsm_ms",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_GSM_MS,
.init = libgsm_encode_init,
.encode = libgsm_encode_frame,
.close = libgsm_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
};
static av_cold int libgsm_decode_init(AVCodecContext *avctx) {
if (avctx->channels > 1) {
av_log(avctx, AV_LOG_ERROR, "Mono required for GSM, got %d channels\n",
avctx->channels);
return -1;
}
if (!avctx->channels)
avctx->channels = 1;
if (!avctx->sample_rate)
avctx->sample_rate = 8000;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->priv_data = gsm_create();
switch(avctx->codec_id) {
case CODEC_ID_GSM:
avctx->frame_size = GSM_FRAME_SIZE;
avctx->block_align = GSM_BLOCK_SIZE;
break;
case CODEC_ID_GSM_MS: {
int one = 1;
gsm_option(avctx->priv_data, GSM_OPT_WAV49, &one);
avctx->frame_size = 2 * GSM_FRAME_SIZE;
avctx->block_align = GSM_MS_BLOCK_SIZE;
}
}
return 0;
}
static av_cold int libgsm_decode_close(AVCodecContext *avctx) {
gsm_destroy(avctx->priv_data);
avctx->priv_data = NULL;
return 0;
}
static int libgsm_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt) {
int i, ret;
struct gsm_state *s = avctx->priv_data;
uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int16_t *samples = data;
int out_size = avctx->frame_size * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
}
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
for (i = 0; i < avctx->frame_size / GSM_FRAME_SIZE; i++) {
if ((ret = gsm_decode(s, buf, samples)) < 0)
return -1;
buf += GSM_BLOCK_SIZE;
samples += GSM_FRAME_SIZE;
}
*data_size = out_size;
return avctx->block_align;
}
static void libgsm_flush(AVCodecContext *avctx) {
gsm_destroy(avctx->priv_data);
avctx->priv_data = gsm_create();
}
AVCodec ff_libgsm_decoder = {
.name = "libgsm",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_GSM,
.init = libgsm_decode_init,
.close = libgsm_decode_close,
.decode = libgsm_decode_frame,
.flush = libgsm_flush,
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
};
AVCodec ff_libgsm_ms_decoder = {
.name = "libgsm_ms",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_GSM_MS,
.init = libgsm_decode_init,
.close = libgsm_decode_close,
.decode = libgsm_decode_frame,
.flush = libgsm_flush,
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
};