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>

1 files changed, 20 insertions, 13 deletions

diff --git a/libavcodec/vorbisdec.c b/libavcodec/vorbisdec.c
index 0457d8b454..03ecc38ed4 100644
--- a/libavcodec/vorbisdec.c
+++ b/libavcodec/vorbisdec.c
@@ -125,6 +125,7 @@ typedef struct {
 
 typedef struct vorbis_context_s {
     AVCodecContext *avccontext;
+    AVFrame frame;
     GetBitContext gb;
     DSPContext dsp;
     FmtConvertContext fmt_conv;
@@ -1037,6 +1038,9 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
     avccontext->sample_rate = vc->audio_samplerate;
     avccontext->frame_size  = FFMIN(vc->blocksize[0], vc->blocksize[1]) >> 2;
 
+    avcodec_get_frame_defaults(&vc->frame);
+    avccontext->coded_frame = &vc->frame;
+
     return 0;
 }
 
@@ -1609,16 +1613,15 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
 
 // Return the decoded audio packet through the standard api
 
-static int vorbis_decode_frame(AVCodecContext *avccontext,
-                               void *data, int *data_size,
-                               AVPacket *avpkt)
+static int vorbis_decode_frame(AVCodecContext *avccontext, void *data,
+                               int *got_frame_ptr, AVPacket *avpkt)
 {
     const uint8_t *buf = avpkt->data;
     int buf_size       = avpkt->size;
     vorbis_context *vc = avccontext->priv_data;
     GetBitContext *gb = &(vc->gb);
     const float *channel_ptrs[255];
-    int i, len, out_size;
+    int i, len, ret;
 
     av_dlog(NULL, "packet length %d \n", buf_size);
 
@@ -1629,18 +1632,18 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
 
     if (!vc->first_frame) {
         vc->first_frame = 1;
-        *data_size = 0;
+        *got_frame_ptr = 0;
         return buf_size;
     }
 
     av_dlog(NULL, "parsed %d bytes %d bits, returned %d samples (*ch*bits) \n",
             get_bits_count(gb) / 8, get_bits_count(gb) % 8, len);
 
-    out_size = len * vc->audio_channels *
-               av_get_bytes_per_sample(avccontext->sample_fmt);
-    if (*data_size < out_size) {
-        av_log(avccontext, AV_LOG_ERROR, "output buffer is too small\n");
-        return AVERROR(EINVAL);
+    /* get output buffer */
+    vc->frame.nb_samples = len;
+    if ((ret = avccontext->get_buffer(avccontext, &vc->frame)) < 0) {
+        av_log(avccontext, AV_LOG_ERROR, "get_buffer() failed\n");
+        return ret;
     }
 
     if (vc->audio_channels > 8) {
@@ -1653,12 +1656,15 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
     }
 
     if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT)
-        vc->fmt_conv.float_interleave(data, channel_ptrs, len, vc->audio_channels);
+        vc->fmt_conv.float_interleave((float *)vc->frame.data[0], channel_ptrs,
+                                      len, vc->audio_channels);
     else
-        vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len,
+        vc->fmt_conv.float_to_int16_interleave((int16_t *)vc->frame.data[0],
+                                               channel_ptrs, len,
                                                vc->audio_channels);
 
-    *data_size = out_size;
+    *got_frame_ptr   = 1;
+    *(AVFrame *)data = vc->frame;
 
     return buf_size;
 }
@@ -1682,6 +1688,7 @@ AVCodec ff_vorbis_decoder = {
     .init           = vorbis_decode_init,
     .close          = vorbis_decode_close,
     .decode         = vorbis_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
     .long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
     .channel_layouts = ff_vorbis_channel_layouts,
     .sample_fmts = (const enum AVSampleFormat[]) {