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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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32
libavcodec/qdm2.c
View File

@@ -130,6 +130,8 @@ typedef struct {
* QDM2 decoder context
*/
typedef struct {
AVFrame frame;
/// Parameters from codec header, do not change during playback
int nb_channels; ///< number of channels
int channels; ///< number of channels
@@ -1876,6 +1878,9 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
// dump_context(s);
return 0;
}
@@ -1956,30 +1961,27 @@ static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
}
static int qdm2_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
QDM2Context *s = avctx->priv_data;
int16_t *out = data;
int i, out_size;
int16_t *out;
int i, ret;
if(!buf)
return 0;
if(buf_size < s->checksum_size)
return -1;
out_size = 16 * s->channels * s->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);
/* get output buffer */
s->frame.nb_samples = 16 * s->frame_size;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
buf_size, buf, s->checksum_size, data, *data_size);
out = (int16_t *)s->frame.data[0];
for (i = 0; i < 16; i++) {
if (qdm2_decode(s, buf, out) < 0)
@@ -1987,7 +1989,8 @@ static int qdm2_decode_frame(AVCodecContext *avctx,
out += s->channels * s->frame_size;
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return s->checksum_size;
}
@@ -2001,5 +2004,6 @@ AVCodec ff_qdm2_decoder =
.init = qdm2_decode_init,
.close = qdm2_decode_close,
.decode = qdm2_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
};