FFmpeg/libavcodec/libgsm.c
Michael Niedermayer 7a72695c05 Merge commit '36ef5369ee9b336febc2c270f8718cec4476cb85'
* commit '36ef5369ee9b336febc2c270f8718cec4476cb85':
  Replace all CODEC_ID_* with AV_CODEC_ID_*
  lavc: add AV prefix to codec ids.

Conflicts:
	doc/APIchanges
	doc/examples/decoding_encoding.c
	doc/examples/muxing.c
	ffmpeg.c
	ffprobe.c
	ffserver.c
	libavcodec/8svx.c
	libavcodec/avcodec.h
	libavcodec/dnxhd_parser.c
	libavcodec/dvdsubdec.c
	libavcodec/error_resilience.c
	libavcodec/h263dec.c
	libavcodec/libvorbisenc.c
	libavcodec/mjpeg_parser.c
	libavcodec/mjpegenc.c
	libavcodec/mpeg12.c
	libavcodec/mpeg4videodec.c
	libavcodec/mpegvideo.c
	libavcodec/mpegvideo_enc.c
	libavcodec/pcm.c
	libavcodec/r210dec.c
	libavcodec/utils.c
	libavcodec/v210dec.c
	libavcodec/version.h
	libavdevice/alsa-audio-dec.c
	libavdevice/bktr.c
	libavdevice/v4l2.c
	libavformat/asfdec.c
	libavformat/asfenc.c
	libavformat/avformat.h
	libavformat/avidec.c
	libavformat/caf.c
	libavformat/electronicarts.c
	libavformat/flacdec.c
	libavformat/flvdec.c
	libavformat/flvenc.c
	libavformat/framecrcenc.c
	libavformat/img2.c
	libavformat/img2dec.c
	libavformat/img2enc.c
	libavformat/ipmovie.c
	libavformat/isom.c
	libavformat/matroska.c
	libavformat/matroskadec.c
	libavformat/matroskaenc.c
	libavformat/mov.c
	libavformat/movenc.c
	libavformat/mp3dec.c
	libavformat/mpeg.c
	libavformat/mpegts.c
	libavformat/mxf.c
	libavformat/mxfdec.c
	libavformat/mxfenc.c
	libavformat/nsvdec.c
	libavformat/nut.c
	libavformat/oggenc.c
	libavformat/pmpdec.c
	libavformat/rawdec.c
	libavformat/rawenc.c
	libavformat/riff.c
	libavformat/sdp.c
	libavformat/utils.c
	libavformat/vocenc.c
	libavformat/wtv.c
	libavformat/xmv.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-08-07 22:45:46 +02:00

271 lines
8.1 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 <gsm/gsm.h>
#include "avcodec.h"
#include "internal.h"
#include "gsm.h"
static av_cold int libgsm_encode_close(AVCodecContext *avctx) {
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
gsm_destroy(avctx->priv_data);
avctx->priv_data = NULL;
return 0;
}
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();
if (!avctx->priv_data)
goto error;
switch(avctx->codec_id) {
case AV_CODEC_ID_GSM:
avctx->frame_size = GSM_FRAME_SIZE;
avctx->block_align = GSM_BLOCK_SIZE;
break;
case AV_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;
}
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame)
goto error;
#endif
return 0;
error:
libgsm_encode_close(avctx);
return -1;
}
static int libgsm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
int ret;
gsm_signal *samples = (gsm_signal *)frame->data[0];
struct gsm_state *state = avctx->priv_data;
if ((ret = ff_alloc_packet2(avctx, avpkt, avctx->block_align)))
return ret;
switch(avctx->codec_id) {
case AV_CODEC_ID_GSM:
gsm_encode(state, samples, avpkt->data);
break;
case AV_CODEC_ID_GSM_MS:
gsm_encode(state, samples, avpkt->data);
gsm_encode(state, samples + GSM_FRAME_SIZE, avpkt->data + 32);
}
*got_packet_ptr = 1;
return 0;
}
#if CONFIG_LIBGSM_ENCODER
AVCodec ff_libgsm_encoder = {
.name = "libgsm",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_GSM,
.init = libgsm_encode_init,
.encode2 = 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"),
};
#endif
#if CONFIG_LIBGSM_MS_ENCODER
AVCodec ff_libgsm_ms_encoder = {
.name = "libgsm_ms",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_GSM_MS,
.init = libgsm_encode_init,
.encode2 = 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"),
};
#endif
typedef struct LibGSMDecodeContext {
AVFrame frame;
struct gsm_state *state;
} LibGSMDecodeContext;
static av_cold int libgsm_decode_init(AVCodecContext *avctx) {
LibGSMDecodeContext *s = avctx->priv_data;
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;
s->state = gsm_create();
switch(avctx->codec_id) {
case AV_CODEC_ID_GSM:
avctx->frame_size = GSM_FRAME_SIZE;
avctx->block_align = GSM_BLOCK_SIZE;
break;
case AV_CODEC_ID_GSM_MS: {
int one = 1;
gsm_option(s->state, GSM_OPT_WAV49, &one);
avctx->frame_size = 2 * GSM_FRAME_SIZE;
avctx->block_align = GSM_MS_BLOCK_SIZE;
}
}
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
static av_cold int libgsm_decode_close(AVCodecContext *avctx) {
LibGSMDecodeContext *s = avctx->priv_data;
gsm_destroy(s->state);
s->state = NULL;
return 0;
}
static int libgsm_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
int i, ret;
LibGSMDecodeContext *s = avctx->priv_data;
uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int16_t *samples;
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
/* get output buffer */
s->frame.nb_samples = avctx->frame_size;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)s->frame.data[0];
for (i = 0; i < avctx->frame_size / GSM_FRAME_SIZE; i++) {
if ((ret = gsm_decode(s->state, buf, samples)) < 0)
return -1;
buf += GSM_BLOCK_SIZE;
samples += GSM_FRAME_SIZE;
}
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return avctx->block_align;
}
static void libgsm_flush(AVCodecContext *avctx) {
LibGSMDecodeContext *s = avctx->priv_data;
int one = 1;
gsm_destroy(s->state);
s->state = gsm_create();
if (avctx->codec_id == AV_CODEC_ID_GSM_MS)
gsm_option(s->state, GSM_OPT_WAV49, &one);
}
#if CONFIG_LIBGSM_DECODER
AVCodec ff_libgsm_decoder = {
.name = "libgsm",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_GSM,
.priv_data_size = sizeof(LibGSMDecodeContext),
.init = libgsm_decode_init,
.close = libgsm_decode_close,
.decode = libgsm_decode_frame,
.flush = libgsm_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
};
#endif
#if CONFIG_LIBGSM_MS_DECODER
AVCodec ff_libgsm_ms_decoder = {
.name = "libgsm_ms",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_GSM_MS,
.priv_data_size = sizeof(LibGSMDecodeContext),
.init = libgsm_decode_init,
.close = libgsm_decode_close,
.decode = libgsm_decode_frame,
.flush = libgsm_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
};
#endif