Merge remote-tracking branch 'qatar/master'

* qatar/master: (34 commits)
  dpcm: return error if packet is too small
  dpcm: use smaller data types for static tables
  dpcm: use sol_table_16 directly instead of through the DPCMContext.
  dpcm: replace short with int16_t
  dpcm: check to make sure channels is 1 or 2.
  dpcm: misc pretty-printing
  dpcm: remove unnecessary variable by using bytestream functions.
  dpcm: move codec-specific variable declarations to their corresponding decoding blocks.
  dpcm: consistently use the variable name 'n' for the next input byte.
  dpcm: output AV_SAMPLE_FMT_U8 for Sol DPCM subcodecs 1 and 2.
  dpcm: calculate and check actual output data size prior to decoding.
  dpcm: factor out the stereo flag calculation
  dpcm: cosmetics: rename channel_number to ch
  avserver: Fix a bug where the socket is IPv4, but IPv6 is autoselected for the loopback address.
  lavf: Avoid using av_malloc(0) in av_dump_format
  dxva2_h264: pass the correct 8x8 scaling lists
  dca: NEON optimised high freq VQ decoding
  avcodec: reject audio packets with NULL data and non-zero size
  dxva: Add ability to enable workaround for older ATI cards
  latmenc: Set latmBufferFullness to largest value to indicate it is not used
  ...

Conflicts:
	libavcodec/dxva2_h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>

ffserver.c

@@ -522,6 +522,7 @@ static int socket_open_listen(struct sockaddr_in *my_addr)
     tmp = 1;
     setsockopt(server_fd, SOL_SOCKET, SO_REUSEADDR, &tmp, sizeof(tmp));
 
+    my_addr->sin_family = AF_INET;
     if (bind (server_fd, (struct sockaddr *) my_addr, sizeof (*my_addr)) < 0) {
         char bindmsg[32];
         snprintf(bindmsg, sizeof(bindmsg), "bind(port %d)", ntohs(my_addr->sin_port));

libavcodec/adpcm_data.c

@@ -38,14 +38,14 @@ const int8_t ff_adpcm_index_table[16] = {
  * this table, but such deviations are negligible:
  */
 const int16_t ff_adpcm_step_table[89] = {
-    7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
-    19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
-    50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
-    130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
-    337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
-    876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
-    2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
-    5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
+        7,     8,     9,    10,    11,    12,    13,    14,    16,    17,
+       19,    21,    23,    25,    28,    31,    34,    37,    41,    45,
+       50,    55,    60,    66,    73,    80,    88,    97,   107,   118,
+      130,   143,   157,   173,   190,   209,   230,   253,   279,   307,
+      337,   371,   408,   449,   494,   544,   598,   658,   724,   796,
+      876,   963,  1060,  1166,  1282,  1411,  1552,  1707,  1878,  2066,
+     2272,  2499,  2749,  3024,  3327,  3660,  4026,  4428,  4871,  5358,
+     5894,  6484,  7132,  7845,  8630,  9493, 10442, 11487, 12635, 13899,
     15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
 };
 
@@ -53,18 +53,18 @@ const int16_t ff_adpcm_step_table[89] = {
 /* ff_adpcm_AdaptationTable[], ff_adpcm_AdaptCoeff1[], and
    ff_adpcm_AdaptCoeff2[] are from libsndfile */
 const int16_t ff_adpcm_AdaptationTable[] = {
-        230, 230, 230, 230, 307, 409, 512, 614,
-        768, 614, 512, 409, 307, 230, 230, 230
+    230, 230, 230, 230, 307, 409, 512, 614,
+    768, 614, 512, 409, 307, 230, 230, 230
 };
 
 /** Divided by 4 to fit in 8-bit integers */
 const uint8_t ff_adpcm_AdaptCoeff1[] = {
-        64, 128, 0, 48, 60, 115, 98
+    64, 128, 0, 48, 60, 115, 98
 };
 
 /** Divided by 4 to fit in 8-bit integers */
 const int8_t ff_adpcm_AdaptCoeff2[] = {
-        0, -64, 0, 16, 0, -52, -58
+    0, -64, 0, 16, 0, -52, -58
 };
 
 const int16_t ff_adpcm_yamaha_indexscale[] = {
@@ -73,6 +73,6 @@ const int16_t ff_adpcm_yamaha_indexscale[] = {
 };
 
 const int8_t ff_adpcm_yamaha_difflookup[] = {
-    1, 3, 5, 7, 9, 11, 13, 15,
+     1,  3,  5,  7,  9,  11,  13,  15,
     -1, -3, -5, -7, -9, -11, -13, -15
 };

libavcodec/adpcmenc.c

@@ -32,13 +32,7 @@
  * Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
  *   by Mike Melanson (melanson@pcisys.net)
  *
- * Reference documents:
- * http://www.pcisys.net/~melanson/codecs/simpleaudio.html
- * http://www.geocities.com/SiliconValley/8682/aud3.txt
- * http://openquicktime.sourceforge.net/plugins.htm
- * XAnim sources (xa_codec.c) http://www.rasnaimaging.com/people/lapus/download.html
- * http://www.cs.ucla.edu/~leec/mediabench/applications.html
- * SoX source code http://home.sprynet.com/~cbagwell/sox.html
+ * See ADPCM decoder reference documents for codec information.
  */
 
 typedef struct TrellisPath {

libavcodec/arm/dca.h

+/*
+ * Copyright (c) 2011 Mans Rullgard <mans@mansr.com>
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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,
+ * 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
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_ARM_DCA_H
+#define AVCODEC_ARM_DCA_H
+
+#include <stdint.h>
+#include "config.h"
+
+#if HAVE_NEON && HAVE_INLINE_ASM
+
+#define int8x8_fmul_int32 int8x8_fmul_int32
+static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
+{
+    __asm__ ("vcvt.f32.s32 %2,  %2,  #4         \n"
+             "vld1.8       {d0},     [%1,:64]   \n"
+             "vmovl.s8     q0,  d0              \n"
+             "vmovl.s16    q1,  d1              \n"
+             "vmovl.s16    q0,  d0              \n"
+             "vcvt.f32.s32 q0,  q0              \n"
+             "vcvt.f32.s32 q1,  q1              \n"
+             "vmul.f32     q0,  q0,  %y2        \n"
+             "vmul.f32     q1,  q1,  %y2        \n"
+             "vst1.32      {q0-q1},  [%m0,:128] \n"
+             : "=Um"(*(float (*)[8])dst)
+             : "r"(src), "x"(scale)
+             : "d0", "d1", "d2", "d3");
+}
+
+#endif
+
+#endif /* AVCODEC_ARM_DCA_H */

libavcodec/dca.c

@@ -42,6 +42,10 @@
 #include "dcadsp.h"
 #include "fmtconvert.h"
 
+#if ARCH_ARM
+#   include "arm/dca.h"
+#endif
+
 //#define TRACE
 
 #define DCA_PRIM_CHANNELS_MAX (7)
@@ -320,7 +324,7 @@ typedef struct {
     int lfe_scale_factor;
 
     /* Subband samples history (for ADPCM) */
-    float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
+    DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
     DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
     DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
     int hist_index[DCA_PRIM_CHANNELS_MAX];
@@ -1057,6 +1061,16 @@ static int decode_blockcode(int code, int levels, int *values)
 static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
 static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
 
+#ifndef int8x8_fmul_int32
+static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
+{
+    float fscale = scale / 16.0;
+    int i;
+    for (i = 0; i < 8; i++)
+        dst[i] = src[i] * fscale;
+}
+#endif
+
 static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
 {
     int k, l;
@@ -1161,19 +1175,16 @@ static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
         for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {
             /* 1 vector -> 32 samples but we only need the 8 samples
              * for this subsubframe. */
-            int m;
+            int hfvq = s->high_freq_vq[k][l];
 
             if (!s->debug_flag & 0x01) {
                 av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");
                 s->debug_flag |= 0x01;
             }
 
-            for (m = 0; m < 8; m++) {
-                subband_samples[k][l][m] =
-                    high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 +
-                                                        m]
-                    * (float) s->scale_factor[k][l][0] / 16.0;
-            }
+            int8x8_fmul_int32(subband_samples[k][l],
+                              &high_freq_vq[hfvq][subsubframe * 8],
+                              s->scale_factor[k][l][0]);
         }
     }
 

libavcodec/dcadata.h

@@ -4224,7 +4224,7 @@ static const float lossless_quant_d[32] = {
 
 /* Vector quantization tables */
 
-static const int8_t high_freq_vq[1024][32] =
+DECLARE_ALIGNED(8, static const int8_t, high_freq_vq)[1024][32] =
 {
   {  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,
      0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0,  0 },

libavcodec/dxva2_h264.c

@@ -162,18 +162,18 @@ static void fill_scaling_lists(struct dxva_context *ctx, const H264Context *h, D
             for (j = 0; j < 16; j++)
                 qm->bScalingLists4x4[i][j] = h->pps.scaling_matrix4[i][j];
 
-        for (j = 0; j < 64; j++) {
-            qm->bScalingLists8x8[0][j] = h->pps.scaling_matrix8[0][j];
-            qm->bScalingLists8x8[1][j] = h->pps.scaling_matrix8[3][j];
+        for (i = 0; i < 64; i++) {
+            qm->bScalingLists8x8[0][i] = h->pps.scaling_matrix8[0][i];
+            qm->bScalingLists8x8[1][i] = h->pps.scaling_matrix8[3][i];
         }
     } else {
         for (i = 0; i < 6; i++)
             for (j = 0; j < 16; j++)
                 qm->bScalingLists4x4[i][j] = h->pps.scaling_matrix4[i][zigzag_scan[j]];
 
-        for (j = 0; j < 64; j++) {
-            qm->bScalingLists8x8[0][j] = h->pps.scaling_matrix8[0][ff_zigzag_direct[j]];
-            qm->bScalingLists8x8[1][j] = h->pps.scaling_matrix8[3][ff_zigzag_direct[j]];
+        for (i = 0; i < 64; i++) {
+            qm->bScalingLists8x8[0][i] = h->pps.scaling_matrix8[0][ff_zigzag_direct[i]];
+            qm->bScalingLists8x8[1][i] = h->pps.scaling_matrix8[3][ff_zigzag_direct[i]];
         }
     }
 }

libavcodec/proresdec_lgpl.c

@@ -427,13 +427,13 @@ static inline void decode_ac_coeffs(GetBitContext *gb, DCTELEM *out,
         lev_cb_index = lev_to_cb_index[FFMIN(level, 9)];
 
         bits_left = get_bits_left(gb);
-        if (bits_left <= 8 && !show_bits(gb, bits_left))
+        if (bits_left <= 0 || (bits_left <= 8 && !show_bits(gb, bits_left)))
             return;
 
         run = decode_vlc_codeword(gb, ac_codebook[run_cb_index]);
 
         bits_left = get_bits_left(gb);
-        if (bits_left <= 8 && !show_bits(gb, bits_left))
+        if (bits_left <= 0 || (bits_left <= 8 && !show_bits(gb, bits_left)))
             return;
 
         level = decode_vlc_codeword(gb, ac_codebook[lev_cb_index]) + 1;

libavcodec/utils.c

@@ -823,6 +823,11 @@ int attribute_align_arg avcodec_decode_audio3(AVCodecContext *avctx, int16_t *sa
 
     avctx->pkt = avpkt;
 
+    if (!avpkt->data && avpkt->size) {
+        av_log(avctx, AV_LOG_ERROR, "invalid packet: NULL data, size != 0\n");
+        return AVERROR(EINVAL);
+    }
+
     if((avctx->codec->capabilities & CODEC_CAP_DELAY) || avpkt->size){
         //FIXME remove the check below _after_ ensuring that all audio check that the available space is enough
         if(*frame_size_ptr < AVCODEC_MAX_AUDIO_FRAME_SIZE){

libavformat/latmenc.c

@@ -120,7 +120,7 @@ static int latm_write_frame_header(AVFormatContext *s, PutBitContext *bs)
         }
 
         put_bits(bs, 3, 0); /* frameLengthType */
-        put_bits(bs, 8, 0); /* latmBufferFullness */
+        put_bits(bs, 8, 0xff); /* latmBufferFullness */
 
         put_bits(bs, 1, 0); /* otherDataPresent */
         put_bits(bs, 1, 0); /* crcCheckPresent */

libavformat/mpeg.c

@@ -49,6 +49,10 @@ static int check_pes(uint8_t *p, uint8_t *end){
     return pes1||pes2;
 }
 
+static int check_pack_header(const uint8_t *buf) {
+    return (buf[1] & 0xC0) == 0x40 || (buf[1] & 0xF0) == 0x20;
+}
+
 static int mpegps_probe(AVProbeData *p)
 {
     uint32_t code= -1;
@@ -61,9 +65,10 @@ static int mpegps_probe(AVProbeData *p)
         if ((code & 0xffffff00) == 0x100) {
             int len= p->buf[i+1] << 8 | p->buf[i+2];
             int pes= check_pes(p->buf+i, p->buf+p->buf_size);
+            int pack = check_pack_header(p->buf+i);
 
             if(code == SYSTEM_HEADER_START_CODE) sys++;
-            else if(code == PACK_START_CODE)     pspack++;
+            else if(code == PACK_START_CODE && pack) pspack++;
             else if((code & 0xf0) == VIDEO_ID &&  pes) vid++;
             // skip pes payload to avoid start code emulation for private
             // and audio streams

libavformat/utils.c

@@ -3535,7 +3535,7 @@ void av_dump_format(AVFormatContext *ic,
                     int is_output)
 {
     int i;
-    uint8_t *printed = av_mallocz(ic->nb_streams);
+    uint8_t *printed = ic->nb_streams ? av_mallocz(ic->nb_streams) : NULL;
     if (ic->nb_streams && !printed)
         return;