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, 12 deletions

diff --git a/libavcodec/dca.c b/libavcodec/dca.c
index edafb967b4..7c43b922b1 100644
--- a/libavcodec/dca.c
+++ b/libavcodec/dca.c
@@ -261,6 +261,7 @@ static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int id
 
 typedef struct {
     AVCodecContext *avctx;
+    AVFrame frame;
     /* Frame header */
     int frame_type;             ///< type of the current frame
     int samples_deficit;        ///< deficit sample count
@@ -1634,9 +1635,8 @@ static void dca_exss_parse_header(DCAContext *s)
  * Main frame decoding function
  * FIXME add arguments
  */
-static int dca_decode_frame(AVCodecContext * avctx,
-                            void *data, int *data_size,
-                            AVPacket *avpkt)
+static int dca_decode_frame(AVCodecContext *avctx, void *data,
+                            int *got_frame_ptr, AVPacket *avpkt)
 {
     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
@@ -1644,9 +1644,8 @@ static int dca_decode_frame(AVCodecContext * avctx,
     int lfe_samples;
     int num_core_channels = 0;
     int i, ret;
-    float   *samples_flt = data;
-    int16_t *samples_s16 = data;
-    int out_size;
+    float   *samples_flt;
+    int16_t *samples_s16;
     DCAContext *s = avctx->priv_data;
     int channels;
     int core_ss_end;
@@ -1832,11 +1831,14 @@ static int dca_decode_frame(AVCodecContext * avctx,
         avctx->channels = channels;
     }
 
-    out_size = 256 / 8 * s->sample_blocks * channels *
-               av_get_bytes_per_sample(avctx->sample_fmt);
-    if (*data_size < out_size)
-        return AVERROR(EINVAL);
-    *data_size = out_size;
+    /* get output buffer */
+    s->frame.nb_samples = 256 * (s->sample_blocks / 8);
+    if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+        return ret;
+    }
+    samples_flt = (float   *)s->frame.data[0];
+    samples_s16 = (int16_t *)s->frame.data[0];
 
     /* filter to get final output */
     for (i = 0; i < (s->sample_blocks / 8); i++) {
@@ -1870,6 +1872,9 @@ static int dca_decode_frame(AVCodecContext * avctx,
         s->lfe_data[i] = s->lfe_data[i + lfe_samples];
     }
 
+    *got_frame_ptr   = 1;
+    *(AVFrame *)data = s->frame;
+
     return buf_size;
 }
 
@@ -1912,6 +1917,9 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
         avctx->channels = avctx->request_channels;
     }
 
+    avcodec_get_frame_defaults(&s->frame);
+    avctx->coded_frame = &s->frame;
+
     return 0;
 }
 
@@ -1940,7 +1948,7 @@ AVCodec ff_dca_decoder = {
     .decode = dca_decode_frame,
     .close = dca_decode_end,
     .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
-    .capabilities = CODEC_CAP_CHANNEL_CONF,
+    .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
     .sample_fmts = (const enum AVSampleFormat[]) {
         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
     },