Merge remote branch 'qatar/master'

* qatar/master: (32 commits)
  10-bit H.264 x86 chroma v loopfilter asm
  Port SMPTE S302M audio decoder from FFmbc 0.3. [Copyright headers corrected]
  Fix crash of interlaced MPEG2 decoding
  h264pred: fix one more aliasing violation.
  doc/APIchanges: fill in missing hashes and dates.
  flacenc: use proper initializers for AVOption default values.
  lavc: deprecate named constants for deprecated antialias_algo.
  aac: workaround for compilation on cygwin
  swscale: extend YUV422p support to 10bits depth
  tiff: add support for inverted FillOrder for uncompressed data
  Remove unused softfloat implementation.
  h264pred: fix aliasing violations.
  rotozoom: Eliminate French variable name.
  rotozoom: Check return value of fread().
  rotozoom: Return an error value instead of calling exit().
  rotozoom: Make init_demo() return int and check for errors on invocation.
  rotozoom: Drop silly UINT8 typedef.
  rotozoom: Drop some unnecessary parentheses.
  rotozoom: K&R coding style cosmetics
  rtsp: Only do keepalive using GET_PARAMETER if the server supports it
  ...

Conflicts:
	Changelog
	cmdutils.c
	doc/APIchanges
	doc/general.texi
	ffmpeg.c
	ffplay.c
	libavcodec/h264pred_template.c
	libavcodec/resample.c
	libavutil/pixfmt.h
	libavutil/softfloat.c
	libavutil/softfloat.h
	tests/rotozoom.c

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

1 files changed, 141 insertions, 0 deletions

diff --git a/libavcodec/s302m.c b/libavcodec/s302m.c
new file mode 100644
index 0000000000..dd0ec2ee19
--- /dev/null
+++ b/libavcodec/s302m.c
@@ -0,0 +1,141 @@
+/*
+ * SMPTE 302M decoder
+ * Copyright (c) 2008 Laurent Aimar <fenrir@videolan.org>
+ * Copyright (c) 2009 Baptiste Coudurier <baptiste.coudurier@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/intreadwrite.h"
+#include "avcodec.h"
+
+#define AES3_HEADER_LEN 4
+
+static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf,
+                                    int buf_size)
+{
+    uint32_t h;
+    int frame_size, channels, id, bits;
+
+    if (buf_size <= AES3_HEADER_LEN) {
+        av_log(avctx, AV_LOG_ERROR, "frame is too short\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /*
+     * AES3 header :
+     * size:            16
+     * number channels   2
+     * channel_id        8
+     * bits per samples  2
+     * alignments        4
+     */
+
+    h = AV_RB32(buf);
+    frame_size =  (h >> 16) & 0xffff;
+    channels   = ((h >> 14) & 0x0003) * 2 +  2;
+    id         =  (h >>  6) & 0x00ff;
+    bits       = ((h >>  4) & 0x0003) * 4 + 16;
+
+    if (AES3_HEADER_LEN + frame_size != buf_size || bits > 24) {
+        av_log(avctx, AV_LOG_ERROR, "frame has invalid header\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* Set output properties */
+    avctx->bits_per_coded_sample = bits;
+    if (bits > 16)
+        avctx->sample_fmt = SAMPLE_FMT_S32;
+    else
+        avctx->sample_fmt = SAMPLE_FMT_S16;
+
+    avctx->channels    = channels;
+    avctx->sample_rate = 48000;
+    avctx->bit_rate    = 48000 * avctx->channels * (avctx->bits_per_coded_sample + 4) +
+                         32 * (48000 / (buf_size * 8 /
+                                        (avctx->channels *
+                                         (avctx->bits_per_coded_sample + 4))));
+
+    return frame_size;
+}
+
+static int s302m_decode_frame(AVCodecContext *avctx, void *data,
+                              int *data_size, AVPacket *avpkt)
+{
+    const uint8_t *buf = avpkt->data;
+    int buf_size       = avpkt->size;
+
+    int frame_size = s302m_parse_frame_header(avctx, buf, buf_size);
+    if (frame_size < 0)
+        return frame_size;
+
+    buf_size -= AES3_HEADER_LEN;
+    buf      += AES3_HEADER_LEN;
+
+    if (*data_size < 4 * buf_size * 8 / (avctx->bits_per_coded_sample + 4))
+        return -1;
+
+    if (avctx->bits_per_coded_sample == 24) {
+        uint32_t *o = data;
+        for (; buf_size > 6; buf_size -= 7) {
+            *o++ = (av_reverse[buf[2]]        << 24) |
+                   (av_reverse[buf[1]]        << 16) |
+                   (av_reverse[buf[0]]        <<  8);
+            *o++ = (av_reverse[buf[6] & 0xf0] << 28) |
+                   (av_reverse[buf[5]]        << 20) |
+                   (av_reverse[buf[4]]        << 12) |
+                   (av_reverse[buf[3] & 0x0f] <<  8);
+            buf += 7;
+        }
+        *data_size = (uint8_t*) o - (uint8_t*) data;
+    } else if (avctx->bits_per_coded_sample == 20) {
+        uint32_t *o = data;
+        for (; buf_size > 5; buf_size -= 6) {
+            *o++ = (av_reverse[buf[2] & 0xf0] << 28) |
+                   (av_reverse[buf[1]]        << 20) |
+                   (av_reverse[buf[0]]        << 12);
+            *o++ = (av_reverse[buf[5] & 0xf0] << 28) |
+                   (av_reverse[buf[4]]        << 20) |
+                   (av_reverse[buf[3]]        << 12);
+            buf += 6;
+        }
+        *data_size = (uint8_t*) o - (uint8_t*) data;
+    } else {
+        uint16_t *o = data;
+        for (; buf_size > 4; buf_size -= 5) {
+            *o++ = (av_reverse[buf[1]]        <<  8) |
+                    av_reverse[buf[0]];
+            *o++ = (av_reverse[buf[4] & 0xf0] << 12) |
+                   (av_reverse[buf[3]]        <<  4) |
+                    av_reverse[buf[2] & 0x0f];
+            buf += 5;
+        }
+        *data_size = (uint8_t*) o - (uint8_t*) data;
+    }
+
+    return buf - avpkt->data;
+}
+
+
+AVCodec ff_s302m_decoder = {
+    .name           = "s302m",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_S302M,
+    .priv_data_size = 0,
+    .decode         = s302m_decode_frame,
+    .long_name      = NULL_IF_CONFIG_SMALL("SMPTE 302M"),
+};