Merge remote branch 'qatar/master'

[ffmpeg] / libavcodec / vorbisdec.c

diff --git a/libavcodec/vorbisdec.c b/libavcodec/vorbisdec.c
index 6c912f0019a21aeed17eed2c9b7335044797153e..161c54d6bcadf613073608c2885d0d9cd633562a 100644 (file)
--- a/libavcodec/vorbisdec.c
+++ b/libavcodec/vorbisdec.c
@@ -3,20 +3,20 @@
  * Vorbis I decoder
  * @author Denes Balatoni  ( dbalatoni programozo hu )
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -1021,7 +1021,9 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
     avccontext->channels    = vc->audio_channels;
     avccontext->sample_rate = vc->audio_samplerate;
     avccontext->frame_size  = FFMIN(vc->blocksize[0], vc->blocksize[1]) >> 2;
-    avccontext->sample_fmt  = AV_SAMPLE_FMT_S16;
+    avccontext->sample_fmt  =
+        avccontext->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
+        AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
 
     return 0 ;
 }
@@ -1148,7 +1150,7 @@ static int vorbis_floor1_decode(vorbis_context *vc,
     uint_fast16_t floor1_Y[258];
     uint_fast16_t floor1_Y_final[258];
     int floor1_flag[258];
-    unsigned class, cdim, cbits, csub, cval, offset, i, j;
+    unsigned partition_class, cdim, cbits, csub, cval, offset, i, j;
     int book, adx, ady, dy, off, predicted, err;
 
 
@@ -1164,20 +1166,20 @@ static int vorbis_floor1_decode(vorbis_context *vc,
 
     offset = 2;
     for (i = 0; i < vf->partitions; ++i) {
-        class = vf->partition_class[i];
-        cdim   = vf->class_dimensions[class];
-        cbits  = vf->class_subclasses[class];
+        partition_class = vf->partition_class[i];
+        cdim   = vf->class_dimensions[partition_class];
+        cbits  = vf->class_subclasses[partition_class];
         csub = (1 << cbits) - 1;
         cval = 0;
 
         AV_DEBUG("Cbits %u\n", cbits);
 
         if (cbits) // this reads all subclasses for this partition's class
-            cval = get_vlc2(gb, vc->codebooks[vf->class_masterbook[class]].vlc.table,
-                            vc->codebooks[vf->class_masterbook[class]].nb_bits, 3);
+            cval = get_vlc2(gb, vc->codebooks[vf->class_masterbook[partition_class]].vlc.table,
+                            vc->codebooks[vf->class_masterbook[partition_class]].nb_bits, 3);
 
         for (j = 0; j < cdim; ++j) {
-            book = vf->subclass_books[class][cval & csub];
+            book = vf->subclass_books[partition_class][cval & csub];
 
             AV_DEBUG("book %d Cbits %u cval %u  bits:%d\n",
                      book, cbits, cval, get_bits_count(gb));
@@ -1631,9 +1633,15 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
                               len * ff_vorbis_channel_layout_offsets[vc->audio_channels - 1][i];
     }
 
-    vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len,
-                                           vc->audio_channels);
-    *data_size = len * 2 * vc->audio_channels;
+    *data_size = len * vc->audio_channels;
+    if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT) {
+        float_interleave(data, channel_ptrs, len, vc->audio_channels);
+        *data_size *= sizeof(float);
+    } else {
+        vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len,
+                                               vc->audio_channels);
+        *data_size *= 2;
+    }
 
     return buf_size ;
 }