Merge remote-tracking branch 'qatar/master'

* qatar/master:
  APIchanges: fill in date and commit for request_sample_fmt
  Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
  Add support for request_sample_format in ffmpeg and ffplay.
  Add APIchanges entry for request_sample_fmt.
  Add request_sample_fmt field to AVCodecContext.
  Add float_interleave() to FmtConvertContext with x86-optimized versions.
  Remove unused make variable SEEK_REFFILE
  fate: remove redundant aref and vref references
  fate: remove do_ffmpeg_nocheck function
  fate: do not collect -benchmark output
  mpegaudiodec: remove decode_end() function
  fate: run aref and vref as regular tests
  mpegaudio: sanitise compute_antialias_* names
  mpeg12: add slice-threading checks to slice-threading initializers.
  h264: copy pixel_shift between slice threading contexts.
  mdec: enable frame-level multithreading.
  mdec.c: fix overread.

Conflicts:
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/dca.c
	libavcodec/h264.c
	libavcodec/mdec.c
	libavcodec/mpeg12.c
	libavcodec/options.c
	libavcodec/version.h
	libavcodec/vorbisdec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 16 insertions, 11 deletions

diff --git a/libavcodec/vorbisdec.c b/libavcodec/vorbisdec.c
index f4b743e8ab..f93fff113f 100644
--- a/libavcodec/vorbisdec.c
+++ b/libavcodec/vorbisdec.c
@@ -979,7 +979,13 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
     dsputil_init(&vc->dsp, avccontext);
     ff_fmt_convert_init(&vc->fmt_conv, avccontext);
 
-    vc->scale_bias = 32768.0f;
+    if (avccontext->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+        avccontext->sample_fmt = AV_SAMPLE_FMT_FLT;
+        vc->scale_bias = 1.0f;
+    } else {
+        avccontext->sample_fmt = AV_SAMPLE_FMT_S16;
+        vc->scale_bias = 32768.0f;
+    }
 
     if (!headers_len) {
         av_log(avccontext, AV_LOG_ERROR, "Extradata missing.\n");
@@ -1024,9 +1030,6 @@ 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  =
-        avccontext->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
-        AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
 
     return 0 ;
 }
@@ -1636,15 +1639,14 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
                               len * ff_vorbis_channel_layout_offsets[vc->audio_channels - 1][i];
     }
 
-    *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 {
+    if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT)
+        vc->fmt_conv.float_interleave(data, channel_ptrs, len, vc->audio_channels);
+    else
         vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len,
                                                vc->audio_channels);
-        *data_size *= 2;
-    }
+
+    *data_size = len * vc->audio_channels *
+                 (av_get_bits_per_sample_fmt(avccontext->sample_fmt) / 8);
 
     return buf_size ;
 }
@@ -1671,5 +1673,8 @@ AVCodec ff_vorbis_decoder = {
     vorbis_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
     .channel_layouts = ff_vorbis_channel_layouts,
+    .sample_fmts = (const enum AVSampleFormat[]) {
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+    },
 };