1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-12 19:18:44 +02:00
FFmpeg/libavcodec/libspeexdec.c
Michael Niedermayer aedc908601 Merge remote-tracking branch 'qatar/master'
* qatar/master: (35 commits)
  flvdec: Do not call parse_keyframes_index with a NULL stream
  libspeexdec: include system headers before local headers
  libspeexdec: return meaningful error codes
  libspeexdec: cosmetics: reindent
  libspeexdec: decode one frame at a time.
  swscale: fix signed shift overflows in ff_yuv2rgb_c_init_tables()
  Move timefilter code from lavf to lavd.
  mov: add support for hdvd and pgapmetadata atoms
  mov: rename function _stik, some indentation cosmetics
  mov: rename function _int8 to remove ambiguity, some indentation cosmetics
  mov: parse the gnre atom
  mp3on4: check for allocation failures in decode_init_mp3on4()
  mp3on4: create a separate flush function for MP3onMP4.
  mp3on4: ensure that the frame channel count does not exceed the codec channel count.
  mp3on4: set channel layout
  mp3on4: fix the output channel order
  mp3on4: allocate temp buffer with av_malloc() instead of on the stack.
  mp3on4: copy MPADSPContext from first context to all contexts.
  fmtconvert: port float_to_int16_interleave() 2-channel x86 inline asm to yasm
  fmtconvert: port int32_to_float_fmul_scalar() x86 inline asm to yasm
  ...

Conflicts:
	libavcodec/arm/h264dsp_init_arm.c
	libavcodec/h264.c
	libavcodec/h264.h
	libavcodec/h264_cabac.c
	libavcodec/h264_cavlc.c
	libavcodec/h264_ps.c
	libavcodec/h264dsp_template.c
	libavcodec/h264idct_template.c
	libavcodec/h264pred.c
	libavcodec/h264pred_template.c
	libavcodec/x86/h264dsp_mmx.c
	libavdevice/Makefile
	libavdevice/jack_audio.c
	libavformat/Makefile
	libavformat/flvdec.c
	libavformat/flvenc.c
	libavutil/pixfmt.h
	libswscale/utils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-10-22 01:16:41 +02:00

169 lines
5.3 KiB
C

/*
* Copyright (C) 2008 David Conrad
*
* 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
*/
#include <speex/speex.h>
#include <speex/speex_header.h>
#include <speex/speex_stereo.h>
#include <speex/speex_callbacks.h>
#include "avcodec.h"
typedef struct {
SpeexBits bits;
SpeexStereoState stereo;
void *dec_state;
SpeexHeader *header;
int frame_size;
} LibSpeexContext;
static av_cold int libspeex_decode_init(AVCodecContext *avctx)
{
LibSpeexContext *s = avctx->priv_data;
const SpeexMode *mode;
// defaults in the case of a missing header
if (avctx->sample_rate <= 8000)
mode = &speex_nb_mode;
else if (avctx->sample_rate <= 16000)
mode = &speex_wb_mode;
else
mode = &speex_uwb_mode;
if (avctx->extradata_size >= 80)
s->header = speex_packet_to_header(avctx->extradata, avctx->extradata_size);
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
if (s->header) {
avctx->sample_rate = s->header->rate;
avctx->channels = s->header->nb_channels;
avctx->frame_size = s->frame_size = s->header->frame_size;
if (s->header->frames_per_packet)
avctx->frame_size *= s->header->frames_per_packet;
mode = speex_lib_get_mode(s->header->mode);
if (!mode) {
av_log(avctx, AV_LOG_ERROR, "Unknown Speex mode %d", s->header->mode);
return AVERROR_INVALIDDATA;
}
} else
av_log(avctx, AV_LOG_INFO, "Missing Speex header, assuming defaults.\n");
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "Only stereo and mono are supported.\n");
return AVERROR(EINVAL);
}
speex_bits_init(&s->bits);
s->dec_state = speex_decoder_init(mode);
if (!s->dec_state) {
av_log(avctx, AV_LOG_ERROR, "Error initializing libspeex decoder.\n");
return -1;
}
if (!s->header) {
speex_decoder_ctl(s->dec_state, SPEEX_GET_FRAME_SIZE, &s->frame_size);
}
if (avctx->channels == 2) {
SpeexCallback callback;
callback.callback_id = SPEEX_INBAND_STEREO;
callback.func = speex_std_stereo_request_handler;
callback.data = &s->stereo;
s->stereo = (SpeexStereoState)SPEEX_STEREO_STATE_INIT;
speex_decoder_ctl(s->dec_state, SPEEX_SET_HANDLER, &callback);
}
return 0;
}
static int libspeex_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
LibSpeexContext *s = avctx->priv_data;
int16_t *output = data;
int out_size, ret, consumed = 0;
/* check output buffer size */
out_size = s->frame_size * avctx->channels *
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 there is not enough data left for the smallest possible frame,
reset the libspeex buffer using the current packet, otherwise ignore
the current packet and keep decoding frames from the libspeex buffer. */
if (speex_bits_remaining(&s->bits) < 43) {
/* check for flush packet */
if (!buf || !buf_size) {
*data_size = 0;
return buf_size;
}
/* set new buffer */
speex_bits_read_from(&s->bits, buf, buf_size);
consumed = buf_size;
}
/* decode a single frame */
ret = speex_decode_int(s->dec_state, &s->bits, output);
if (ret <= -2) {
av_log(avctx, AV_LOG_ERROR, "Error decoding Speex frame.\n");
return AVERROR_INVALIDDATA;
}
if (avctx->channels == 2)
speex_decode_stereo_int(output, s->frame_size, &s->stereo);
*data_size = out_size;
return consumed;
}
static av_cold int libspeex_decode_close(AVCodecContext *avctx)
{
LibSpeexContext *s = avctx->priv_data;
speex_header_free(s->header);
speex_bits_destroy(&s->bits);
speex_decoder_destroy(s->dec_state);
return 0;
}
static av_cold void libspeex_decode_flush(AVCodecContext *avctx)
{
LibSpeexContext *s = avctx->priv_data;
speex_bits_reset(&s->bits);
}
AVCodec ff_libspeex_decoder = {
.name = "libspeex",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_SPEEX,
.priv_data_size = sizeof(LibSpeexContext),
.init = libspeex_decode_init,
.close = libspeex_decode_close,
.decode = libspeex_decode_frame,
.flush = libspeex_decode_flush,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY,
.long_name = NULL_IF_CONFIG_SMALL("libspeex Speex"),
};