Merge remote branch 'qatar/master'

12 files changed, 36 insertions(+), 81 deletions(-)
yes thats 36 new lines in 14 commits

* qatar/master:
  ffmpeg: fix -aspect cli option
  Restructure video filter implementation in ffmpeg.c.
  ffplay: remove audio_write_get_buf_size() forward declaration
  lavfi: print key-frame and picture type information in ff_dlog_ref()
  mathops: remove ancient confusing comment
  cws2fws: Improve error message wording.
  tools: Check the return value of write().
  mpegaudio: move OUT_FMT macro to mpegaudiodec.c
  mpegaudio: remove OUT_MIN/MAX macros
  Add missing #includes to mp3_header_(de)compress bsf
  dct: fix indentation
  dct: bypass table allocation for DCT_II of size 32
  h264dsp_mmx: Add #ifdefs around some mmxext functions on x86_64.
  Remove unused header mpegaudio3.h.

Conflicts:
	ffmpeg.c
	libavcodec/mpegaudio.h
	libavcodec/mpegaudio3.h
	libavfilter/avfilter.c

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

ffmpeg.c

@@ -171,8 +171,6 @@ static int loop_output = AVFMT_NOOUTPUTLOOP;
 static int qp_hist = 0;
 #if CONFIG_AVFILTER
 static char *vfilters = NULL;
-#else
-static unsigned int sws_flags = SWS_BICUBIC;
 #endif
 
 static int intra_only = 0;
@@ -289,6 +287,7 @@ typedef struct AVOutputStream {
     int resample_pix_fmt;
 
     float frame_aspect_ratio;
+
     /* forced key frames */
     int64_t *forced_kf_pts;
     int forced_kf_count;
@@ -1642,7 +1641,7 @@ static int output_packet(AVInputStream *ist, int ist_index,
         }
 
 #if CONFIG_AVFILTER
-        if(ist->st->codec->codec_type == AVMEDIA_TYPE_VIDEO){
+        if(ist->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
             for(i=0;i<nb_ostreams;i++) {
                 ost = ost_table[i];
                 if (ost->input_video_filter && ost->source_index == ist_index) {
@@ -1786,7 +1785,7 @@ static int output_packet(AVInputStream *ist, int ist_index,
                     cont:
                     frame_available = (ist->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) &&
                                        ost->output_video_filter && avfilter_poll_frame(ost->output_video_filter->inputs[0]);
-                    if(ost->picref)
+                    if (ost->picref)
                         avfilter_unref_buffer(ost->picref);
                 }
 #endif
@@ -3537,8 +3536,8 @@ static void new_video_stream(AVFormatContext *oc, int file_idx)
         ost->frame_aspect_ratio = frame_aspect_ratio;
         frame_aspect_ratio = 0;
 #if CONFIG_AVFILTER
-        ost->avfilter= vfilters;
-        vfilters= NULL;
+        ost->avfilter = vfilters;
+        vfilters = NULL;
 #endif
     }
 

libavcodec/Makefile

@@ -669,7 +669,6 @@ SKIPHEADERS-$(CONFIG_LIBSCHROEDINGER)  += libschroedinger.h
 SKIPHEADERS-$(CONFIG_VAAPI)            += vaapi_internal.h
 SKIPHEADERS-$(CONFIG_VDPAU)            += vdpau.h
 SKIPHEADERS-$(CONFIG_XVMC)             += xvmc.h
-SKIPHEADERS                            += mpegaudio3.h
 
 EXAMPLES = api
 

libavcodec/dct.c

@@ -180,33 +180,36 @@ av_cold int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType inverse)
     int n = 1 << nbits;
     int i;
 
+    memset(s, 0, sizeof(*s));
+
     s->nbits    = nbits;
     s->inverse  = inverse;
 
-    ff_init_ff_cos_tabs(nbits+2);
+    if (inverse == DCT_II && nbits == 5) {
+        s->dct_calc = dct32_func;
+    } else {
+        ff_init_ff_cos_tabs(nbits+2);
 
-    s->costab = ff_cos_tabs[nbits+2];
+        s->costab = ff_cos_tabs[nbits+2];
 
-    s->csc2 = av_malloc(n/2 * sizeof(FFTSample));
+        s->csc2 = av_malloc(n/2 * sizeof(FFTSample));
 
-    if (ff_rdft_init(&s->rdft, nbits, inverse == DCT_III) < 0) {
-        av_free(s->csc2);
-        return -1;
-    }
+        if (ff_rdft_init(&s->rdft, nbits, inverse == DCT_III) < 0) {
+            av_free(s->csc2);
+            return -1;
+        }
 
-    for (i = 0; i < n/2; i++)
-        s->csc2[i] = 0.5 / sin((M_PI / (2*n) * (2*i + 1)));
+        for (i = 0; i < n/2; i++)
+            s->csc2[i] = 0.5 / sin((M_PI / (2*n) * (2*i + 1)));
 
-    switch(inverse) {
-    case DCT_I  : s->dct_calc = ff_dct_calc_I_c; break;
-    case DCT_II : s->dct_calc = ff_dct_calc_II_c ; break;
-    case DCT_III: s->dct_calc = ff_dct_calc_III_c; break;
-    case DST_I  : s->dct_calc = ff_dst_calc_I_c; break;
+        switch(inverse) {
+        case DCT_I  : s->dct_calc = ff_dct_calc_I_c; break;
+        case DCT_II : s->dct_calc = ff_dct_calc_II_c ; break;
+        case DCT_III: s->dct_calc = ff_dct_calc_III_c; break;
+        case DST_I  : s->dct_calc = ff_dst_calc_I_c; break;
+        }
     }
 
-    if (inverse == DCT_II && nbits == 5)
-        s->dct_calc = dct32_func;
-
     s->dct32 = dct32;
     if (HAVE_MMX)     ff_dct_init_mmx(s);
 

libavcodec/mathops.h

@@ -45,9 +45,6 @@
 #endif
 
 #ifndef MULH
-//gcc 3.4 creates an incredibly bloated mess out of this
-//#    define MULH(a,b) (((int64_t)(a) * (int64_t)(b))>>32)
-
 static av_always_inline int MULH(int a, int b){
     return ((int64_t)(a) * (int64_t)(b))>>32;
 }

libavcodec/mp3_header_compress_bsf.c

@@ -18,6 +18,7 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include "libavutil/intreadwrite.h"
 #include "avcodec.h"
 #include "mpegaudio.h"
 

libavcodec/mp3_header_decompress_bsf.c

@@ -18,6 +18,7 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include "libavutil/intreadwrite.h"
 #include "avcodec.h"
 #include "mpegaudio.h"
 #include "mpegaudiodata.h"

libavcodec/mpegaudio.h

@@ -35,8 +35,6 @@
 #include "dsputil.h"
 #include "dct.h"
 
-#define CONFIG_AUDIO_NONSHORT 0
-
 /* max frame size, in samples */
 #define MPA_FRAME_SIZE 1152
 
@@ -69,19 +67,9 @@
 
 #if CONFIG_FLOAT
 typedef float OUT_INT;
-#define OUT_FMT AV_SAMPLE_FMT_FLT
-#elif CONFIG_MPEGAUDIO_HP && CONFIG_AUDIO_NONSHORT
-typedef int32_t OUT_INT;
-#define OUT_MAX INT32_MAX
-#define OUT_MIN INT32_MIN
-#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 31)
-#define OUT_FMT AV_SAMPLE_FMT_S32
 #else
 typedef int16_t OUT_INT;
-#define OUT_MAX INT16_MAX
-#define OUT_MIN INT16_MIN
 #define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15)
-#define OUT_FMT AV_SAMPLE_FMT_S16
 #endif
 
 #if CONFIG_FLOAT
@@ -147,6 +135,9 @@ typedef struct MPADecodeContext {
     DECLARE_ALIGNED(16, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
     INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
     GranuleDef granules[2][2]; /* Used in Layer 3 */
+#ifdef DEBUG
+    int frame_count;
+#endif
     int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
     int dither_state;
     int error_recognition;

libavcodec/mpegaudio3.h

@@ -1,53 +0,0 @@
-/*
- * Copyright (c) 2007 Michael Niedermayer
- *
- * 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
- */
-
-/* layer 3 "granule" */
-typedef struct GranuleDef {
-    uint8_t scfsi;
-    int part2_3_length;
-    int big_values;
-    int global_gain;
-    int scalefac_compress;
-    uint8_t block_type;
-    uint8_t switch_point;
-    int table_select[3];
-    int subblock_gain[3];
-    uint8_t scalefac_scale;
-    uint8_t count1table_select;
-    int region_size[3]; /* number of huffman codes in each region */
-    int preflag;
-    int short_start, long_end; /* long/short band indexes */
-    uint8_t scale_factors[40];
-    int32_t sb_hybrid[SBLIMIT * 18]; /* 576 samples */
-} GranuleDef;
-
-void ff_mp3_init(void);
-
-/**
- * Compute huffman coded region sizes.
- */
-void ff_init_short_region(MPADecodeContext *s, GranuleDef *g);
-
-/**
- * Compute huffman coded region sizes.
- */
-void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2);
-
-void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g);

libavcodec/mpegaudiodec.c

@@ -47,6 +47,7 @@
 #   define MULH3(x, y, s) ((s)*(y)*(x))
 #   define MULLx(x, y, s) ((y)*(x))
 #   define RENAME(a) a ## _float
+#   define OUT_FMT AV_SAMPLE_FMT_FLT
 #else
 #   define SHR(a,b)       ((a)>>(b))
 #   define compute_antialias compute_antialias_integer
@@ -57,6 +58,7 @@
 #   define MULH3(x, y, s) MULH((s)*(x), y)
 #   define MULLx(x, y, s) MULL(x,y,s)
 #   define RENAME(a)      a
+#   define OUT_FMT AV_SAMPLE_FMT_S16
 #endif
 
 /****************/
@@ -490,7 +492,7 @@ static inline int round_sample(int64_t *sum)
     int sum1;
     sum1 = (int)((*sum) >> OUT_SHIFT);
     *sum &= (1<<OUT_SHIFT)-1;
-    return av_clip(sum1, OUT_MIN, OUT_MAX);
+    return av_clip_int16(sum1);
 }
 
 #   define MULS(ra, rb) MUL64(ra, rb)

libavcodec/x86/h264dsp_mmx.c

@@ -252,6 +252,7 @@ LF_IFUNC(v,  chroma_intra, depth,  avx)
 LF_FUNCS( uint8_t,  8)
 LF_FUNCS(uint16_t, 10)
 
+#if ARCH_X86_32
 LF_FUNC (v8, luma,             8, mmxext)
 static void ff_deblock_v_luma_8_mmxext(uint8_t *pix, int stride, int alpha, int beta, int8_t *tc0)
 {
@@ -266,6 +267,7 @@ static void ff_deblock_v_luma_intra_8_mmxext(uint8_t *pix, int stride, int alpha
     ff_deblock_v8_luma_intra_8_mmxext(pix+0, stride, alpha, beta);
     ff_deblock_v8_luma_intra_8_mmxext(pix+8, stride, alpha, beta);
 }
+#endif /* ARCH_X86_32 */
 
 LF_FUNC (v,  luma,            10, mmxext)
 LF_IFUNC(v,  luma_intra,      10, mmxext)