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>
This commit is contained in:
Michael Niedermayer
2011-12-03 02:08:55 +01:00
87 changed files with 2226 additions and 1307 deletions

View File

@@ -129,6 +129,7 @@ typedef struct APEPredictor {
/** Decoder context */
typedef struct APEContext {
AVCodecContext *avctx;
AVFrame frame;
DSPContext dsp;
int channels;
int samples; ///< samples left to decode in current frame
@@ -215,6 +216,10 @@ static av_cold int ape_decode_init(AVCodecContext *avctx)
dsputil_init(&s->dsp, avctx);
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
filter_alloc_fail:
ape_decode_close(avctx);
@@ -805,16 +810,15 @@ static void ape_unpack_stereo(APEContext *ctx, int count)
}
}
static int ape_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int ape_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
APEContext *s = avctx->priv_data;
int16_t *samples = data;
int i;
int blockstodecode, out_size;
int16_t *samples;
int i, ret;
int blockstodecode;
int bytes_used = 0;
/* this should never be negative, but bad things will happen if it is, so
@@ -826,7 +830,7 @@ static int ape_decode_frame(AVCodecContext *avctx,
void *tmp_data;
if (!buf_size) {
*data_size = 0;
*got_frame_ptr = 0;
return 0;
}
if (buf_size < 8) {
@@ -874,18 +878,19 @@ static int ape_decode_frame(AVCodecContext *avctx,
}
if (!s->data) {
*data_size = 0;
*got_frame_ptr = 0;
return buf_size;
}
blockstodecode = FFMIN(BLOCKS_PER_LOOP, s->samples);
out_size = blockstodecode * 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);
/* get output buffer */
s->frame.nb_samples = blockstodecode;
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];
s->error=0;
@@ -909,7 +914,9 @@ static int ape_decode_frame(AVCodecContext *avctx,
s->samples -= blockstodecode;
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return bytes_used;
}
@@ -927,7 +934,7 @@ AVCodec ff_ape_decoder = {
.init = ape_decode_init,
.close = ape_decode_close,
.decode = ape_decode_frame,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY | CODEC_CAP_DR1,
.flush = ape_flush,
.long_name = NULL_IF_CONFIG_SMALL("Monkey's Audio"),
};