1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
FFmpeg/libavcodec/libgsm.c
Michael Niedermayer e4de71677f Merge remote-tracking branch 'qatar/master'
* qatar/master:
  aac_latm: reconfigure decoder on audio specific config changes
  latmdec: fix audio specific config parsing
  Add avcodec_decode_audio4().
  avcodec: change number of plane pointers from 4 to 8 at next major bump.
  Update developers documentation with coding conventions.
  svq1dec: avoid undefined get_bits(0) call
  ARM: h264dsp_neon cosmetics
  ARM: make some NEON macros reusable
  Do not memcpy raw video frames when using null muxer
  fate: update asf seektest
  vp8: flush buffers on size changes.
  doc: improve general documentation for MacOSX
  asf: use packet dts as approximation of pts
  asf: do not call av_read_frame
  rtsp: Initialize the media_type_mask in the rtp guessing demuxer
  Cleaned up alacenc.c

Conflicts:
	doc/APIchanges
	doc/developer.texi
	libavcodec/8svx.c
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/nellymoserdec.c
	libavcodec/tta.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/wmadec.c
	libavformat/asfdec.c
	tests/ref/seek/lavf_asf

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-03 03:00:30 +01:00

245 lines
7.4 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"),
};
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 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;
}
}
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;
gsm_destroy(s->state);
s->state = gsm_create();
}
AVCodec ff_libgsm_decoder = {
.name = "libgsm",
.type = AVMEDIA_TYPE_AUDIO,
.id = 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"),
};
AVCodec ff_libgsm_ms_decoder = {
.name = "libgsm_ms",
.type = AVMEDIA_TYPE_AUDIO,
.id = 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"),
};