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Merge remote-tracking branch 'qatar/master'

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* 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>
This commit is contained in:
Michael Niedermayer
2011-12-03 02:08:55 +01:00
parent 12804348f5 d268b79e34
commit e4de71677f
87 changed files with 2226 additions and 1307 deletions
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31
libavcodec/ra144dec.c
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@@ -38,6 +38,10 @@ static av_cold int ra144_decode_init(AVCodecContext * avctx)
ractx->lpc_coef[1] = ractx->lpc_tables[1];
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&ractx->frame);
avctx->coded_frame = &ractx->frame;
return 0;
}
@@ -54,8 +58,8 @@ static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs,
}
/** Uncompress one block (20 bytes -> 160*2 bytes). */
static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
int *data_size, AVPacket *avpkt)
static int ra144_decode_frame(AVCodecContext * avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
@@ -64,23 +68,25 @@ static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
uint16_t block_coefs[NBLOCKS][LPC_ORDER]; // LPC coefficients of each sub-block
unsigned int lpc_refl[LPC_ORDER]; // LPC reflection coefficients of the frame
int i, j;
int out_size;
int16_t *data = vdata;
int ret;
int16_t *samples;
unsigned int energy;
RA144Context *ractx = avctx->priv_data;
GetBitContext gb;
out_size = NBLOCKS * BLOCKSIZE * 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);
/* get output buffer */
ractx->frame.nb_samples = NBLOCKS * BLOCKSIZE;
if ((ret = avctx->get_buffer(avctx, &ractx->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)ractx->frame.data[0];
if(buf_size < FRAMESIZE) {
av_log(avctx, AV_LOG_ERROR,
"Frame too small (%d bytes). Truncated file?\n", buf_size);
*data_size = 0;
*got_frame_ptr = 0;
return buf_size;
}
init_get_bits(&gb, buf, FRAMESIZE * 8);
@@ -106,7 +112,7 @@ static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb);
for (j=0; j < BLOCKSIZE; j++)
*data++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2);
*samples++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2);
}
ractx->old_energy = energy;
@@ -114,7 +120,9 @@ static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = ractx->frame;
return FRAMESIZE;
}
@@ -125,5 +133,6 @@ AVCodec ff_ra_144_decoder = {
.priv_data_size = sizeof(RA144Context),
.init = ra144_decode_init,
.decode = ra144_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
};