Merge remote-tracking branch 'qatar/master'

* qatar/master: (24 commits)
  utils: Drop pointless '#if 1' preprocessor directive.
  ac3enc: remove empty ac3_float function that is never called
  ac3enc: split templated float vs. fixed functions into a separate file.
  ac3enc: dynamically allocate AC3EncodeContext fields windowed_samples and mdct
  ac3enc: use function pointer to choose between AC-3 and E-AC-3 header output functions.
  Roll back 4:4:4 H.264 for now Needs some ARM/PPC asm modifications.
  Fix SVQ3 after adding 4:4:4 H.264 support
  H.264: fix CODEC_FLAG_GRAY
  4:4:4 H.264 decoding support
  h264_parser: Fix whitespace after previous change.
  h264_parser: Fix behaviour when PARSER_FLAG_COMPLETE_FRAMES is set.
  wav: remove an invalid free().
  lavf: initialise reference_dts in av_estimate_timings_from_pts.
  h264: don't be so picky on decoding pps in extradata.
  avcodec.h: add or elaborate on some documentation comments.
  h264: change a few comments into error messages
  ac3dec: fix doxy-style for comment ("///>" should be "///<" instead).
  img2: add .dpx to the list of supported file extensions.
  ffv1: fix undefined behavior with insane widths.
  ARM: jrevdct_arm: simplify stack usage
  ...
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Changelog

@@ -10,6 +10,14 @@ version <next>:
 - libxvid aspect pickiness fixed
 - Frame multithreaded decoding
 - E-AC-3 audio encoder
+- ac3enc: add channel coupling support
+- floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
+- H264/MPEG frame-level multi-threading
+- All av_metadata_* functions renamed to av_dict_* and moved to libavutil
+
+
+version 0.7_beta2:
+
 - Lots of deprecated API cruft removed
 - fft and imdct optimizations for AVX (Sandy Bridge) processors
 - showinfo filter added

libavcodec/Makefile

@@ -63,8 +63,8 @@ OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \
 OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o
 OBJS-$(CONFIG_AC3_DECODER)             += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
 OBJS-$(CONFIG_AC3_ENCODER)             += ac3enc_combined.o ac3enc_fixed.o ac3enc_float.o ac3tab.o ac3.o kbdwin.o
-OBJS-$(CONFIG_AC3_FLOAT_ENCODER)       += ac3enc_float.o ac3tab.o ac3.o kbdwin.o
-OBJS-$(CONFIG_AC3_FIXED_ENCODER)       += ac3enc_fixed.o ac3tab.o ac3.o
+OBJS-$(CONFIG_AC3_FLOAT_ENCODER)       += ac3enc_float.o ac3tab.o ac3tab.o ac3.o kbdwin.o
+OBJS-$(CONFIG_AC3_FIXED_ENCODER)       += ac3enc_fixed.o ac3tab.o ac3tab.o ac3.o
 OBJS-$(CONFIG_ALAC_DECODER)            += alac.o
 OBJS-$(CONFIG_ALAC_ENCODER)            += alacenc.o
 OBJS-$(CONFIG_ALS_DECODER)             += alsdec.o bgmc.o mpeg4audio.o
@@ -128,8 +128,8 @@ OBJS-$(CONFIG_DVVIDEO_DECODER)         += dv.o dvdata.o
 OBJS-$(CONFIG_DVVIDEO_ENCODER)         += dv.o dvdata.o
 OBJS-$(CONFIG_DXA_DECODER)             += dxa.o
 OBJS-$(CONFIG_EAC3_DECODER)            += eac3dec.o eac3dec_data.o
-OBJS-$(CONFIG_EAC3_ENCODER)            += eac3enc.o ac3enc_float.o ac3tab.o \
-                                          ac3.o kbdwin.o
+OBJS-$(CONFIG_EAC3_ENCODER)            += eac3enc.o ac3enc.o ac3enc_float.o \
+                                          ac3tab.o ac3.o kbdwin.o
 OBJS-$(CONFIG_EACMV_DECODER)           += eacmv.o
 OBJS-$(CONFIG_EAMAD_DECODER)           += eamad.o eaidct.o mpeg12.o \
                                           mpeg12data.o mpegvideo.o  \

libavcodec/ac3dec.h

@@ -196,7 +196,7 @@ typedef struct {
 ///@}
 
 ///@defgroup arrays aligned arrays
-    DECLARE_ALIGNED(16, int,   fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];       ///> fixed-point transform coefficients
+    DECLARE_ALIGNED(16, int,   fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];       ///< fixed-point transform coefficients
     DECLARE_ALIGNED(32, float, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];   ///< transform coefficients
     DECLARE_ALIGNED(32, float, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];             ///< delay - added to the next block
     DECLARE_ALIGNED(32, float, window)[AC3_BLOCK_SIZE];                              ///< window coefficients

libavcodec/ac3enc.h

@@ -40,18 +40,28 @@
 #define CONFIG_AC3ENC_FLOAT 0
 #endif
 
+#define OFFSET(param) offsetof(AC3EncodeContext, options.param)
+#define AC3ENC_PARAM (AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM)
+
+#define AC3ENC_TYPE_AC3_FIXED   0
+#define AC3ENC_TYPE_AC3         1
+#define AC3ENC_TYPE_EAC3        2
+
 #if CONFIG_AC3ENC_FLOAT
+#define AC3_NAME(x) ff_ac3_float_ ## x
 #define MAC_COEF(d,a,b) ((d)+=(a)*(b))
 typedef float SampleType;
 typedef float CoefType;
 typedef float CoefSumType;
 #else
+#define AC3_NAME(x) ff_ac3_fixed_ ## x
 #define MAC_COEF(d,a,b) MAC64(d,a,b)
 typedef int16_t SampleType;
 typedef int32_t CoefType;
 typedef int64_t CoefSumType;
 #endif
 
+
 typedef struct AC3MDCTContext {
     const SampleType *window;           ///< MDCT window function
     FFTContext fft;                     ///< FFT context for MDCT calculation
@@ -128,10 +138,11 @@ typedef struct AC3EncodeContext {
     PutBitContext pb;                       ///< bitstream writer context
     DSPContext dsp;
     AC3DSPContext ac3dsp;                   ///< AC-3 optimized functions
-    AC3MDCTContext mdct;                    ///< MDCT context
+    AC3MDCTContext *mdct;                   ///< MDCT context
 
     AC3Block blocks[AC3_MAX_BLOCKS];        ///< per-block info
 
+    int fixed_point;                        ///< indicates if fixed-point encoder is being used
     int eac3;                               ///< indicates if this is E-AC-3 vs. AC-3
     int bitstream_id;                       ///< bitstream id                           (bsid)
     int bitstream_mode;                     ///< bitstream mode                         (bsmod)
@@ -189,6 +200,7 @@ typedef struct AC3EncodeContext {
     int frame_bits;                         ///< all frame bits except exponents and mantissas
     int exponent_bits;                      ///< number of bits used for exponents
 
+    SampleType *windowed_samples;
     SampleType **planar_samples;
     uint8_t *bap_buffer;
     uint8_t *bap1_buffer;
@@ -208,7 +220,74 @@ typedef struct AC3EncodeContext {
     uint8_t *ref_bap     [AC3_MAX_CHANNELS][AC3_MAX_BLOCKS]; ///< bit allocation pointers (bap)
     int ref_bap_set;                                         ///< indicates if ref_bap pointers have been set
 
-    DECLARE_ALIGNED(32, SampleType, windowed_samples)[AC3_WINDOW_SIZE];
+    /* fixed vs. float function pointers */
+    void (*mdct_end)(AC3MDCTContext *mdct);
+    int  (*mdct_init)(AVCodecContext *avctx, AC3MDCTContext *mdct, int nbits);
+    void (*apply_window)(DSPContext *dsp, SampleType *output,
+                         const SampleType *input, const SampleType *window,
+                         unsigned int len);
+    int  (*normalize_samples)(struct AC3EncodeContext *s);
+    void (*scale_coefficients)(struct AC3EncodeContext *s);
+
+    /* fixed vs. float templated function pointers */
+    void (*deinterleave_input_samples)(struct AC3EncodeContext *s,
+                                       const SampleType *samples);
+    void (*apply_mdct)(struct AC3EncodeContext *s);
+    void (*apply_channel_coupling)(struct AC3EncodeContext *s);
+    void (*compute_rematrixing_strategy)(struct AC3EncodeContext *s);
+
+    /* AC-3 vs. E-AC-3 function pointers */
+    void (*output_frame_header)(struct AC3EncodeContext *s);
 } AC3EncodeContext;
 
+
+extern const int64_t ff_ac3_channel_layouts[19];
+
+int ff_ac3_encode_init(AVCodecContext *avctx);
+
+int ff_ac3_encode_frame(AVCodecContext *avctx, unsigned char *frame,
+                        int buf_size, void *data);
+
+int ff_ac3_encode_close(AVCodecContext *avctx);
+
+
+/* prototypes for functions in ac3enc_fixed.c and ac3enc_float.c */
+
+void ff_ac3_fixed_mdct_end(AC3MDCTContext *mdct);
+void ff_ac3_float_mdct_end(AC3MDCTContext *mdct);
+
+int ff_ac3_fixed_mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
+                           int nbits);
+int ff_ac3_float_mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
+                           int nbits);
+
+void ff_ac3_fixed_apply_window(DSPContext *dsp, SampleType *output,
+                               const SampleType *input,
+                               const SampleType *window, unsigned int len);
+void ff_ac3_float_apply_window(DSPContext *dsp, SampleType *output,
+                               const SampleType *input,
+                               const SampleType *window, unsigned int len);
+
+int ff_ac3_fixed_normalize_samples(AC3EncodeContext *s);
+
+void ff_ac3_fixed_scale_coefficients(AC3EncodeContext *s);
+void ff_ac3_float_scale_coefficients(AC3EncodeContext *s);
+
+
+/* prototypes for functions in ac3enc_template.c */
+
+void ff_ac3_fixed_deinterleave_input_samples(AC3EncodeContext *s,
+                                             const SampleType *samples);
+void ff_ac3_float_deinterleave_input_samples(AC3EncodeContext *s,
+                                             const SampleType *samples);
+
+void ff_ac3_fixed_apply_mdct(AC3EncodeContext *s);
+void ff_ac3_float_apply_mdct(AC3EncodeContext *s);
+
+void ff_ac3_fixed_apply_channel_coupling(AC3EncodeContext *s);
+void ff_ac3_float_apply_channel_coupling(AC3EncodeContext *s);
+
+void ff_ac3_fixed_compute_rematrixing_strategy(AC3EncodeContext *s);
+void ff_ac3_float_compute_rematrixing_strategy(AC3EncodeContext *s);
+
 #endif /* AVCODEC_AC3ENC_H */

libavcodec/ac3enc_fixed.c

@@ -28,13 +28,20 @@
 
 #define CONFIG_FFT_FLOAT 0
 #undef CONFIG_AC3ENC_FLOAT
-#include "ac3enc.c"
+#include "ac3enc.h"
+
+#define AC3ENC_TYPE AC3ENC_TYPE_AC3_FIXED
+#include "ac3enc_opts_template.c"
+static AVClass ac3enc_class = { "Fixed-Point AC-3 Encoder", av_default_item_name,
+                                ac3fixed_options, LIBAVUTIL_VERSION_INT };
+
+#include "ac3enc_template.c"
 
 
 /**
  * Finalize MDCT and free allocated memory.
  */
-static av_cold void mdct_end(AC3MDCTContext *mdct)
+av_cold void AC3_NAME(mdct_end)(AC3MDCTContext *mdct)
 {
     ff_mdct_end(&mdct->fft);
 }
@@ -44,8 +51,8 @@ static av_cold void mdct_end(AC3MDCTContext *mdct)
  * Initialize MDCT tables.
  * @param nbits log2(MDCT size)
  */
-static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
-                             int nbits)
+av_cold int AC3_NAME(mdct_init)(AVCodecContext *avctx, AC3MDCTContext *mdct,
+                                int nbits)
 {
     int ret = ff_mdct_init(&mdct->fft, nbits, 0, -1.0);
     mdct->window = ff_ac3_window;
@@ -56,8 +63,9 @@ static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
 /**
  * Apply KBD window to input samples prior to MDCT.
  */
-static void apply_window(DSPContext *dsp, int16_t *output, const int16_t *input,
-                         const int16_t *window, unsigned int len)
+void AC3_NAME(apply_window)(DSPContext *dsp, int16_t *output,
+                            const int16_t *input, const int16_t *window,
+                            unsigned int len)
 {
     dsp->apply_window_int16(output, input, window, len);
 }
@@ -82,7 +90,7 @@ static int log2_tab(AC3EncodeContext *s, int16_t *src, int len)
  *
  * @return exponent shift
  */
-static int normalize_samples(AC3EncodeContext *s)
+int AC3_NAME(normalize_samples)(AC3EncodeContext *s)
 {
     int v = 14 - log2_tab(s, s->windowed_samples, AC3_WINDOW_SIZE);
     if (v > 0)
@@ -95,7 +103,7 @@ static int normalize_samples(AC3EncodeContext *s)
 /**
  * Scale MDCT coefficients to 25-bit signed fixed-point.
  */
-static void scale_coefficients(AC3EncodeContext *s)
+void AC3_NAME(scale_coefficients)(AC3EncodeContext *s)
 {
     int blk, ch;
 
@@ -109,14 +117,22 @@ static void scale_coefficients(AC3EncodeContext *s)
 }
 
 
+static av_cold int ac3_fixed_encode_init(AVCodecContext *avctx)
+{
+    AC3EncodeContext *s = avctx->priv_data;
+    s->fixed_point = 1;
+    return ff_ac3_encode_init(avctx);
+}
+
+
 AVCodec ff_ac3_fixed_encoder = {
     "ac3_fixed",
     AVMEDIA_TYPE_AUDIO,
     CODEC_ID_AC3,
     sizeof(AC3EncodeContext),
-    ac3_encode_init,
-    ac3_encode_frame,
-    ac3_encode_close,
+    ac3_fixed_encode_init,
+    ff_ac3_encode_frame,
+    ff_ac3_encode_close,
     NULL,
     .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),

libavcodec/ac3enc_float.c

@@ -27,14 +27,25 @@
  */
 
 #define CONFIG_AC3ENC_FLOAT 1
-#include "ac3enc.c"
+#include "ac3enc.h"
+#include "eac3enc.h"
 #include "kbdwin.h"
 
 
+#if CONFIG_AC3_ENCODER
+#define AC3ENC_TYPE AC3ENC_TYPE_AC3
+#include "ac3enc_opts_template.c"
+static AVClass ac3enc_class = { "AC-3 Encoder", av_default_item_name,
+                                ac3_options, LIBAVUTIL_VERSION_INT };
+#endif
+
+#include "ac3enc_template.c"
+
+
 /**
  * Finalize MDCT and free allocated memory.
  */
-static av_cold void mdct_end(AC3MDCTContext *mdct)
+av_cold void ff_ac3_float_mdct_end(AC3MDCTContext *mdct)
 {
     ff_mdct_end(&mdct->fft);
     av_freep(&mdct->window);
@@ -45,8 +56,8 @@ static av_cold void mdct_end(AC3MDCTContext *mdct)
  * Initialize MDCT tables.
  * @param nbits log2(MDCT size)
  */
-static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
-                             int nbits)
+av_cold int ff_ac3_float_mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
+                                   int nbits)
 {
     float *window;
     int i, n, n2;
@@ -71,27 +82,18 @@ static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
 /**
  * Apply KBD window to input samples prior to MDCT.
  */
-static void apply_window(DSPContext *dsp, float *output, const float *input,
-                         const float *window, unsigned int len)
+void ff_ac3_float_apply_window(DSPContext *dsp, float *output,
+                               const float *input, const float *window,
+                               unsigned int len)
 {
     dsp->vector_fmul(output, input, window, len);
 }
 
 
-/**
- * Normalize the input samples to use the maximum available precision.
- */
-static int normalize_samples(AC3EncodeContext *s)
-{
-    /* Normalization is not needed for floating-point samples, so just return 0 */
-    return 0;
-}
-
-
 /**
  * Scale MDCT coefficients from float to 24-bit fixed-point.
  */
-static void scale_coefficients(AC3EncodeContext *s)
+void ff_ac3_float_scale_coefficients(AC3EncodeContext *s)
 {
     int chan_size = AC3_MAX_COEFS * AC3_MAX_BLOCKS;
     s->ac3dsp.float_to_fixed24(s->fixed_coef_buffer + chan_size,
@@ -106,9 +108,9 @@ AVCodec ff_ac3_float_encoder = {
     AVMEDIA_TYPE_AUDIO,
     CODEC_ID_AC3,
     sizeof(AC3EncodeContext),
-    ac3_encode_init,
-    ac3_encode_frame,
-    ac3_encode_close,
+    ff_ac3_encode_init,
+    ff_ac3_encode_frame,
+    ff_ac3_encode_close,
     NULL,
     .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
@@ -116,19 +118,3 @@ AVCodec ff_ac3_float_encoder = {
     .channel_layouts = ff_ac3_channel_layouts,
 };
 #endif
-
-#if CONFIG_EAC3_ENCODER
-AVCodec ff_eac3_encoder = {
-    .name            = "eac3",
-    .type            = AVMEDIA_TYPE_AUDIO,
-    .id              = CODEC_ID_EAC3,
-    .priv_data_size  = sizeof(AC3EncodeContext),
-    .init            = ac3_encode_init,
-    .encode          = ac3_encode_frame,
-    .close           = ac3_encode_close,
-    .sample_fmts     = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
-    .long_name       = NULL_IF_CONFIG_SMALL("ATSC A/52 E-AC-3"),
-    .priv_class      = &eac3enc_class,
-    .channel_layouts = ff_ac3_channel_layouts,
-};
-#endif

libavcodec/ac3enc_opts_template.c

@@ -19,6 +19,9 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include "libavutil/opt.h"
+#include "ac3.h"
+
 #if AC3ENC_TYPE == AC3ENC_TYPE_AC3_FIXED
 static const AVOption ac3fixed_options[] = {
 #elif AC3ENC_TYPE == AC3ENC_TYPE_AC3

libavcodec/arm/Makefile

@@ -5,6 +5,9 @@ OBJS-$(CONFIG_DCA_DECODER)             += arm/dcadsp_init_arm.o         \
 
 ARMV6-OBJS-$(CONFIG_AC3DSP)            += arm/ac3dsp_armv6.o
 
+OBJS-$(CONFIG_MPEGAUDIODSP)            += arm/mpegaudiodsp_init_arm.o
+ARMV6-OBJS-$(CONFIG_MPEGAUDIODSP)      += arm/mpegaudiodsp_fixed_armv6.o
+
 OBJS-$(CONFIG_VP5_DECODER)             += arm/vp56dsp_init_arm.o
 OBJS-$(CONFIG_VP6_DECODER)             += arm/vp56dsp_init_arm.o
 OBJS-$(CONFIG_VP8_DECODER)             += arm/vp8dsp_init_arm.o

libavcodec/arm/jrevdct_arm.S

@@ -54,18 +54,13 @@
 #define FIX_M_1_961570560_ID   40
 #define FIX_M_2_562915447_ID   44
 #define FIX_0xFFFF_ID          48
-        .text
-        .align
 
 function ff_j_rev_dct_arm, export=1
-        stmdb   sp!, { r4 - r12, lr }   @ all callee saved regs
-
-        sub sp, sp, #4                  @ reserve some space on the stack
-        str r0, [ sp ]                  @ save the DCT pointer to the stack
+        push {r0, r4 - r11, lr}
 
         mov lr, r0                      @ lr = pointer to the current row
         mov r12, #8                     @ r12 = row-counter
-        adr r11, const_array            @ r11 = base pointer to the constants array
+        movrel r11, const_array         @ r11 = base pointer to the constants array
 row_loop:
         ldrsh r0, [lr, # 0]             @ r0 = 'd0'
         ldrsh r2, [lr, # 2]             @ r2 = 'd2'
@@ -102,7 +97,7 @@ row_loop:
         add r4, r6, r3, lsl #13             @ r4 = tmp11
         rsb r3, r6, r3, lsl #13             @ r3 = tmp12
 
-        stmdb   sp!, { r0, r2, r3, r4 } @ save on the stack tmp10, tmp13, tmp12, tmp11
+        push {r0, r2, r3, r4} @ save on the stack tmp10, tmp13, tmp12, tmp11
 
         ldrsh r3, [lr, #10]             @ r3 = 'd3'
         ldrsh r5, [lr, #12]             @ r5 = 'd5'
@@ -136,8 +131,8 @@ row_loop:
         add r3, r3, r4                  @ r3 = tmp2
         add r1, r1, r6                  @ r1 = tmp3
 
-        ldmia sp!, { r0, r2, r4, r6 } @ r0 = tmp10 / r2 = tmp13 / r4 = tmp12 / r6 = tmp11
-                                      @ r1 = tmp3  / r3 = tmp2  / r5 = tmp1  / r7 = tmp0
+        pop {r0, r2, r4, r6} @ r0 = tmp10 / r2 = tmp13 / r4 = tmp12 / r6 = tmp11
+                             @ r1 = tmp3  / r3 = tmp2  / r5 = tmp1  / r7 = tmp0
 
         @ Compute DESCALE(tmp10 + tmp3, CONST_BITS-PASS1_BITS)
         add r8, r0, r1
@@ -211,7 +206,7 @@ end_of_row_loop:
 
 start_column_loop:
         @ Start of column loop
-        ldr lr, [ sp ]
+        pop {lr}
         mov r12, #8
 column_loop:
         ldrsh r0, [lr, #( 0*8)]             @ r0 = 'd0'
@@ -245,7 +240,7 @@ column_loop:
         orrs r10, r9, r10
         beq empty_odd_column
 
-        stmdb   sp!, { r0, r2, r4, r6 } @ save on the stack tmp10, tmp13, tmp12, tmp11
+        push {r0, r2, r4, r6} @ save on the stack tmp10, tmp13, tmp12, tmp11
 
         add r0, r3, r5                  @ r0 = 'z2'
         add r2, r1, r7                  @ r2 = 'z1'
@@ -275,8 +270,8 @@ column_loop:
         add r3, r3, r4                  @ r3 = tmp2
         add r1, r1, r6                  @ r1 = tmp3
 
-        ldmia sp!, { r0, r2, r4, r6 } @ r0 = tmp10 / r2 = tmp13 / r4 = tmp11 / r6 = tmp12
-                                      @ r1 = tmp3  / r3 = tmp2  / r5 = tmp1  / r7 = tmp0
+        pop {r0, r2, r4, r6} @ r0 = tmp10 / r2 = tmp13 / r4 = tmp11 / r6 = tmp12
+                             @ r1 = tmp3  / r3 = tmp2  / r5 = tmp1  / r7 = tmp0
 
         @ Compute DESCALE(tmp10 + tmp3, CONST_BITS+PASS1_BITS+3)
         add r8, r0, r1
@@ -368,11 +363,10 @@ empty_odd_column:
 
 the_end:
         @ The end....
-        add sp, sp, #4
-        ldmia   sp!, { r4 - r12, pc }   @ restore callee saved regs and return
+        pop {r4 - r11, pc}
+endfunc
 
-const_array:
-        .align
+const const_array
         .word FIX_0_298631336
         .word FIX_0_541196100
         .word FIX_0_765366865
@@ -386,3 +380,4 @@ const_array:
         .word FIX_M_1_961570560
         .word FIX_M_2_562915447
         .word FIX_0xFFFF
+endconst

libavcodec/arm/mpegaudiodsp_fixed_armv6.S

+/*
+ * 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
+ */
+
+#include "asm.S"
+
+.macro  skip            args:vararg
+.endm
+
+.macro  sum8            lo,  hi,  w, p, t1, t2, t3, t4, rsb=skip, offs=0
+        ldr             \t1, [\w, #4*\offs]
+        ldr             \t2, [\p, #4]!
+        \rsb            \t1, \t1, #0
+  .irpc i, 135
+        ldr             \t3, [\w, #4*64*\i+4*\offs]
+        ldr             \t4, [\p, #4*64*\i]
+        smlal           \lo, \hi, \t1, \t2
+        \rsb            \t3, \t3, #0
+        ldr             \t1, [\w, #4*64*(\i+1)+4*\offs]
+        ldr             \t2, [\p, #4*64*(\i+1)]
+        smlal           \lo, \hi, \t3, \t4
+        \rsb            \t1, \t1, #0
+  .endr
+        ldr             \t3, [\w, #4*64*7+4*\offs]
+        ldr             \t4, [\p, #4*64*7]
+        smlal           \lo, \hi, \t1, \t2
+        \rsb            \t3, \t3, #0
+        smlal           \lo, \hi, \t3, \t4
+.endm
+
+.macro  round           rd,  lo,  hi
+        lsr             \rd, \lo, #24
+        bic             \lo, \lo, #0xff000000
+        orr             \rd, \rd, \hi, lsl #8
+        mov             \hi, #0
+        ssat            \rd, #16, \rd
+.endm
+
+function ff_mpadsp_apply_window_fixed_armv6, export=1
+        push            {r2,r4-r11,lr}
+
+        add             r4,  r0,  #4*512        @ synth_buf + 512
+    .rept 4
+        ldm             r0!, {r5-r12}
+        stm             r4!, {r5-r12}
+    .endr
+
+        ldr             r4,  [sp, #40]          @ incr
+        sub             r0,  r0,  #4*17         @ synth_buf + 16
+        ldr             r8,  [r2]               @ sum:low
+        add             r2,  r0,  #4*32         @ synth_buf + 48
+        rsb             r5,  r4,  r4,  lsl #5   @ 31 * incr
+        lsl             r4,  r4,  #1
+        asr             r9,  r8,  #31           @ sum:high
+        add             r5,  r3,  r5,  lsl #1   @ samples2
+        add             r6,  r1,  #4*32         @ w2
+        str             r4,  [sp, #40]
+
+        sum8            r8,  r9,  r1,  r0,  r10, r11, r12, lr
+        sum8            r8,  r9,  r1,  r2,  r10, r11, r12, lr, rsb, 32
+        round           r10, r8,  r9
+        strh            r10, [r3], r4
+
+        mov             lr,  #15
+1:
+        ldr             r12, [r0, #4]!
+        ldr             r11, [r6, #-4]!
+        ldr             r10, [r1, #4]!
+  .irpc i, 0246
+    .if \i
+        ldr             r11, [r6, #4*64*\i]
+        ldr             r10, [r1, #4*64*\i]
+    .endif
+        rsb             r11, r11, #0
+        smlal           r8,  r9,  r10, r12
+        ldr             r10, [r0, #4*64*(\i+1)]
+    .ifeq \i
+        smull           r4, r7, r11, r12
+    .else
+        smlal           r4, r7, r11, r12
+    .endif
+        ldr             r11, [r6, #4*64*(\i+1)]
+        ldr             r12, [r1, #4*64*(\i+1)]
+        rsb             r11, r11, #0
+        smlal           r8,  r9,  r12, r10
+    .iflt \i-6
+        ldr             r12, [r0, #4*64*(\i+2)]
+    .else
+        ldr             r12, [r2, #-4]!
+    .endif
+        smlal           r4,  r7,  r11, r10
+  .endr
+  .irpc i, 0246
+        ldr             r10, [r1, #4*64*\i+4*32]
+        rsb             r12, r12, #0
+        ldr             r11, [r6, #4*64*\i+4*32]
+        smlal           r8,  r9,  r10, r12
+        ldr             r10, [r2, #4*64*(\i+1)]
+        smlal           r4,  r7,  r11, r12
+        ldr             r12, [r1, #4*64*(\i+1)+4*32]
+        rsb             r10, r10, #0
+        ldr             r11, [r6, #4*64*(\i+1)+4*32]
+        smlal           r8,  r9,  r12, r10
+    .iflt \i-6
+        ldr             r12, [r2, #4*64*(\i+2)]
+    .else
+        ldr             r12, [sp, #40]
+    .endif
+        smlal           r4,  r7,  r11, r10
+  .endr
+        round           r10, r8,  r9
+        adds            r8,  r8,  r4
+        adc             r9,  r9,  r7
+        strh            r10, [r3], r12
+        round           r11, r8,  r9
+        subs            lr,  lr,  #1
+        strh            r11, [r5], -r12
+        bgt             1b
+
+        sum8            r8,  r9,  r1,  r0,  r10, r11, r12, lr, rsb, 33
+        pop             {r4}
+        round           r10, r8,  r9
+        str             r8,  [r4]
+        strh            r10, [r3]
+
+        pop             {r4-r11,pc}
+endfunc

libavcodec/arm/mpegaudiodsp_init_arm.c

+/*
+ * Copyright (c) 2011 Mans Rullgard
+ *
+ * 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
+ */
+
+#include <stdint.h>
+#include "libavcodec/mpegaudiodsp.h"
+#include "config.h"
+
+void ff_mpadsp_apply_window_fixed_armv6(int32_t *synth_buf, int32_t *window,
+                                        int *dither, int16_t *out, int incr);
+
+void ff_mpadsp_init_arm(MPADSPContext *s)
+{
+    if (HAVE_ARMV6) {
+        s->apply_window_fixed = ff_mpadsp_apply_window_fixed_armv6;
+    }
+}

libavcodec/eac3enc.c

@@ -28,6 +28,13 @@
 #include "ac3enc.h"
 #include "eac3enc.h"
 
+
+#define AC3ENC_TYPE AC3ENC_TYPE_EAC3
+#include "ac3enc_opts_template.c"
+static AVClass eac3enc_class = { "E-AC-3 Encoder", av_default_item_name,
+                                 eac3_options, LIBAVUTIL_VERSION_INT };
+
+
 void ff_eac3_set_cpl_states(AC3EncodeContext *s)
 {
     int ch, blk;
@@ -129,3 +136,20 @@ void ff_eac3_output_frame_header(AC3EncodeContext *s)
     /* block start info */
     put_bits(&s->pb, 1, 0);
 }
+
+
+#if CONFIG_EAC3_ENCODER
+AVCodec ff_eac3_encoder = {
+    .name            = "eac3",
+    .type            = AVMEDIA_TYPE_AUDIO,
+    .id              = CODEC_ID_EAC3,
+    .priv_data_size  = sizeof(AC3EncodeContext),
+    .init            = ff_ac3_encode_init,
+    .encode          = ff_ac3_encode_frame,
+    .close           = ff_ac3_encode_close,
+    .sample_fmts     = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
+    .long_name       = NULL_IF_CONFIG_SMALL("ATSC A/52 E-AC-3"),
+    .priv_class      = &eac3enc_class,
+    .channel_layouts = ff_ac3_channel_layouts,
+};
+#endif

libavcodec/h264.c

@@ -995,7 +995,7 @@ int ff_h264_decode_extradata(H264Context *h)
         cnt = *(p++); // Number of pps
         for (i = 0; i < cnt; i++) {
             nalsize = AV_RB16(p) + 2;
-            if(decode_nal_units(h, p, nalsize) < 0) {
+            if (decode_nal_units(h, p, nalsize) < 0) {
                 av_log(avctx, AV_LOG_ERROR, "Decoding pps %d from avcC failed\n", i);
                 return -1;
             }
@@ -2351,8 +2351,8 @@ static int decode_slice_header(H264Context *h, H264Context *h0){
         MPV_common_end(s);
     }
     if (!s->context_initialized) {
-        if(h != h0){
-            av_log(h->s.avctx, AV_LOG_ERROR, "we cant (re-)initialize context during parallel decoding\n");
+        if (h != h0) {
+            av_log(h->s.avctx, AV_LOG_ERROR, "Cannot (re-)initialize context during parallel decoding.\n");
             return -1;
         }
 
@@ -2398,8 +2398,8 @@ static int decode_slice_header(H264Context *h, H264Context *h0){
 
         s->avctx->hwaccel = ff_find_hwaccel(s->avctx->codec->id, s->avctx->pix_fmt);
 
-        if (MPV_common_init(s) < 0){
-            av_log(h->s.avctx, AV_LOG_ERROR, "MPV_common_init() failed\n");
+        if (MPV_common_init(s) < 0) {
+            av_log(h->s.avctx, AV_LOG_ERROR, "MPV_common_init() failed.\n");
             return -1;
         }
         s->first_field = 0;
@@ -2409,8 +2409,8 @@ static int decode_slice_header(H264Context *h, H264Context *h0){
         ff_h264_alloc_tables(h);
 
         if (!HAVE_THREADS || !(s->avctx->active_thread_type&FF_THREAD_SLICE)) {
-            if (context_init(h) < 0){
-                av_log(h->s.avctx, AV_LOG_ERROR, "context_init() failed\n");
+            if (context_init(h) < 0) {
+                av_log(h->s.avctx, AV_LOG_ERROR, "context_init() failed.\n");
                 return -1;
             }
         } else {
@@ -2428,8 +2428,8 @@ static int decode_slice_header(H264Context *h, H264Context *h0){
             }
 
             for(i = 0; i < s->avctx->thread_count; i++)
-                if(context_init(h->thread_context[i]) < 0){
-                    av_log(h->s.avctx, AV_LOG_ERROR, "context_init() failed\n");
+                if (context_init(h->thread_context[i]) < 0) {
+                    av_log(h->s.avctx, AV_LOG_ERROR, "context_init() failed.\n");
                     return -1;
                 }
         }
@@ -2737,8 +2737,8 @@ static int decode_slice_header(H264Context *h, H264Context *h0){
                 av_log(s->avctx, AV_LOG_INFO, "Cannot parallelize deblocking type 1, decoding such frames in sequential order\n");
                 h0->single_decode_warning = 1;
             }
-            if(h != h0){
-                av_log(h->s.avctx, AV_LOG_ERROR, "deblocking switched inside frame\n");
+            if (h != h0) {
+                av_log(h->s.avctx, AV_LOG_ERROR, "Deblocking switched inside frame.\n");
                 return 1;
             }
         }

libavcodec/mpegaudiodsp.c

@@ -35,6 +35,7 @@ void ff_mpadsp_init(MPADSPContext *s)
     s->dct32_float = dct.dct32;
     s->dct32_fixed = ff_dct32_fixed;
 
+    if (ARCH_ARM)     ff_mpadsp_init_arm(s);
     if (HAVE_MMX)     ff_mpadsp_init_mmx(s);
     if (HAVE_ALTIVEC) ff_mpadsp_init_altivec(s);
 }

libavcodec/mpegaudiodsp.h

@@ -47,6 +47,7 @@ void ff_mpa_synth_filter_float(MPADSPContext *s,
                                float *samples, int incr,
                                float *sb_samples);
 
+void ff_mpadsp_init_arm(MPADSPContext *s);
 void ff_mpadsp_init_mmx(MPADSPContext *s);
 void ff_mpadsp_init_altivec(MPADSPContext *s);
 

libavformat/utils.c

@@ -1605,14 +1605,12 @@ int64_t av_gen_search(AVFormatContext *s, int stream_index, int64_t target_ts, i
 
     pos = (flags & AVSEEK_FLAG_BACKWARD) ? pos_min : pos_max;
     ts  = (flags & AVSEEK_FLAG_BACKWARD) ?  ts_min :  ts_max;
-#if 1
     pos_min = pos;
     ts_min = read_timestamp(s, stream_index, &pos_min, INT64_MAX);
     pos_min++;
     ts_max = read_timestamp(s, stream_index, &pos_min, INT64_MAX);
     av_dlog(s, "pos=0x%"PRIx64" %"PRId64"<=%"PRId64"<=%"PRId64"\n",
             pos, ts_min, target_ts, ts_max);
-#endif
     *ts_ret= ts;
     return pos;
 }