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, 20 insertions, 0 deletions

diff --git a/libavcodec/fmtconvert.c b/libavcodec/fmtconvert.c
index 0e8aa5e909..c03117c2cd 100644
--- a/libavcodec/fmtconvert.c
+++ b/libavcodec/fmtconvert.c
@@ -56,11 +56,31 @@ static void float_to_int16_interleave_c(int16_t *dst, const float **src,
     }
 }
 
+void ff_float_interleave_c(float *dst, const float **src, unsigned int len,
+                           int channels)
+{
+    int j, c;
+    unsigned int i;
+    if (channels == 2) {
+        for (i = 0; i < len; i++) {
+            dst[2*i]   = src[0][i];
+            dst[2*i+1] = src[1][i];
+        }
+    } else if (channels == 1 && len < INT_MAX / sizeof(float)) {
+        memcpy(dst, src[0], len * sizeof(float));
+    } else {
+        for (c = 0; c < channels; c++)
+            for (i = 0, j = c; i < len; i++, j += channels)
+                dst[j] = src[c][i];
+    }
+}
+
 av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
 {
     c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_c;
     c->float_to_int16             = float_to_int16_c;
     c->float_to_int16_interleave  = float_to_int16_interleave_c;
+    c->float_interleave           = ff_float_interleave_c;
 
     if (ARCH_ARM) ff_fmt_convert_init_arm(c, avctx);
     if (HAVE_ALTIVEC) ff_fmt_convert_init_altivec(c, avctx);