From 32ac63ee10ca5daa149344a75d736c1b98177392 Mon Sep 17 00:00:00 2001
From: "Ronald S. Bultje" <rsbultje@gmail.com>
Date: Tue, 10 May 2011 11:29:08 -0400
Subject: mdec.c: fix overread.

---
 libavcodec/mdec.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'libavcodec')

diff --git a/libavcodec/mdec.c b/libavcodec/mdec.c
index 545b919411..9b6e6b6dd9 100644
--- a/libavcodec/mdec.c
+++ b/libavcodec/mdec.c
@@ -125,7 +125,8 @@ static inline int decode_mb(MDECContext *a, DCTELEM block[6][64]){
     a->dsp.clear_blocks(block[0]);
 
     for(i=0; i<6; i++){
-        if( mdec_decode_block_intra(a, block[ block_index[i] ], block_index[i]) < 0)
+        if( mdec_decode_block_intra(a, block[ block_index[i] ], block_index[i]) < 0 ||
+            get_bits_left(&a->gb) < 0)
             return -1;
     }
     return 0;
-- 
cgit v1.2.3


From a64c58a24023b3e8c9b30bfb6908150b57117a3b Mon Sep 17 00:00:00 2001
From: Alexander Strange <astrange@ithinksw.com>
Date: Tue, 10 May 2011 11:29:09 -0400
Subject: mdec: enable frame-level multithreading.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
---
 libavcodec/mdec.c | 20 +++++++++++++++++---
 1 file changed, 17 insertions(+), 3 deletions(-)

(limited to 'libavcodec')

diff --git a/libavcodec/mdec.c b/libavcodec/mdec.c
index 9b6e6b6dd9..02b69d045a 100644
--- a/libavcodec/mdec.c
+++ b/libavcodec/mdec.c
@@ -31,6 +31,7 @@
 #include "dsputil.h"
 #include "mpegvideo.h"
 #include "mpeg12.h"
+#include "thread.h"
 
 typedef struct MDECContext{
     AVCodecContext *avctx;
@@ -163,10 +164,10 @@ static int decode_frame(AVCodecContext *avctx,
     int i;
 
     if(p->data[0])
-        avctx->release_buffer(avctx, p);
+        ff_thread_release_buffer(avctx, p);
 
     p->reference= 0;
-    if(avctx->get_buffer(avctx, p) < 0){
+    if(ff_thread_get_buffer(avctx, p) < 0){
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return -1;
     }
@@ -239,6 +240,18 @@ static av_cold int decode_init(AVCodecContext *avctx){
     return 0;
 }
 
+static av_cold int decode_init_thread_copy(AVCodecContext *avctx){
+    MDECContext * const a = avctx->priv_data;
+    AVFrame *p = (AVFrame*)&a->picture;
+
+    avctx->coded_frame = p;
+    a->avctx= avctx;
+
+    p->qscale_table = av_mallocz( a->mb_width);
+
+    return 0;
+}
+
 static av_cold int decode_end(AVCodecContext *avctx){
     MDECContext * const a = avctx->priv_data;
 
@@ -260,7 +273,8 @@ AVCodec ff_mdec_decoder = {
     NULL,
     decode_end,
     decode_frame,
-    CODEC_CAP_DR1,
+    CODEC_CAP_DR1 | CODEC_CAP_FRAME_THREADS,
     .long_name= NULL_IF_CONFIG_SMALL("Sony PlayStation MDEC (Motion DECoder)"),
+    .init_thread_copy= ONLY_IF_THREADS_ENABLED(decode_init_thread_copy)
 };
 
-- 
cgit v1.2.3


From 8d44cd2cd82e27e6b051fe0606dece3b0bec0bcd Mon Sep 17 00:00:00 2001
From: "Ronald S. Bultje" <rsbultje@gmail.com>
Date: Tue, 17 May 2011 10:26:30 -0400
Subject: h264: copy pixel_shift between slice threading contexts.

Fixes "make THREADS=2 THREAD_TYPE=2
fate-h264-conformance-frext-pph10i3_panasonic_a".
---
 libavcodec/h264.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'libavcodec')

diff --git a/libavcodec/h264.c b/libavcodec/h264.c
index 616d0a4804..6b262bc992 100644
--- a/libavcodec/h264.c
+++ b/libavcodec/h264.c
@@ -1965,6 +1965,7 @@ static int decode_slice_header(H264Context *h, H264Context *h0){
             c->h264dsp = h->h264dsp;
             c->sps = h->sps;
             c->pps = h->pps;
+            c->pixel_shift = h->pixel_shift;
             init_scan_tables(c);
             clone_tables(c, h, i);
         }
-- 
cgit v1.2.3


From 508a24f8dc63e74bd9917e6f0c4cdbb744741ef0 Mon Sep 17 00:00:00 2001
From: "Ronald S. Bultje" <rsbultje@gmail.com>
Date: Tue, 17 May 2011 10:26:29 -0400
Subject: mpeg12: add slice-threading checks to slice-threading initializers.

Fixes "make THREADS=2 THREAD_TYPE=1 fate-mpeg2-field-enc".
---
 libavcodec/mpeg12.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'libavcodec')

diff --git a/libavcodec/mpeg12.c b/libavcodec/mpeg12.c
index ffe6b94c05..88ed5332ae 100644
--- a/libavcodec/mpeg12.c
+++ b/libavcodec/mpeg12.c
@@ -2262,7 +2262,7 @@ static int decode_chunks(AVCodecContext *avctx,
         buf_ptr = ff_find_start_code(buf_ptr,buf_end, &start_code);
         if (start_code > 0x1ff){
             if(s2->pict_type != AV_PICTURE_TYPE_B || avctx->skip_frame <= AVDISCARD_DEFAULT){
-                if(avctx->thread_count > 1){
+                if((avctx->active_thread_type & FF_THREAD_SLICE) && avctx->thread_count > 1){
                     int i;
 
                     avctx->execute(avctx, slice_decode_thread,  &s2->thread_context[0], NULL, s->slice_count, sizeof(void*));
@@ -2430,7 +2430,7 @@ static int decode_chunks(AVCodecContext *avctx,
                     break;
                 }
 
-                if(avctx->thread_count > 1){
+                if((avctx->active_thread_type & FF_THREAD_SLICE) && avctx->thread_count > 1){
                     int threshold= (s2->mb_height*s->slice_count + avctx->thread_count/2) / avctx->thread_count;
                     if(threshold <= mb_y){
                         MpegEncContext *thread_context= s2->thread_context[s->slice_count];
@@ -2541,7 +2541,7 @@ AVCodec ff_mpegvideo_decoder = {
 
 #if CONFIG_MPEG_XVMC_DECODER
 static av_cold int mpeg_mc_decode_init(AVCodecContext *avctx){
-    if( avctx->thread_count > 1)
+    if((avctx->active_thread_type & FF_THREAD_SLICE) && avctx->thread_count > 1)
         return -1;
     if( !(avctx->slice_flags & SLICE_FLAG_CODED_ORDER) )
         return -1;
-- 
cgit v1.2.3


From dede36bd024fb23da244dbfab32b7ef18ca51c82 Mon Sep 17 00:00:00 2001
From: Mans Rullgard <mans@mansr.com>
Date: Tue, 17 May 2011 20:39:09 +0100
Subject: mpegaudio: sanitise compute_antialias_* names

This makes the compute_antialias functions use the same naming
convention as everything else.

Signed-off-by: Mans Rullgard <mans@mansr.com>
---
 libavcodec/mpegaudiodec.c | 9 +++------
 1 file changed, 3 insertions(+), 6 deletions(-)

(limited to 'libavcodec')

diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c
index 77ecb44c9e..4802a04bc8 100644
--- a/libavcodec/mpegaudiodec.c
+++ b/libavcodec/mpegaudiodec.c
@@ -41,7 +41,6 @@
 
 #if CONFIG_FLOAT
 #   define SHR(a,b)       ((a)*(1.0f/(1<<(b))))
-#   define compute_antialias compute_antialias_float
 #   define FIXR_OLD(a)    ((int)((a) * FRAC_ONE + 0.5))
 #   define FIXR(x)        ((float)(x))
 #   define FIXHR(x)       ((float)(x))
@@ -51,7 +50,6 @@
 #   define OUT_FMT AV_SAMPLE_FMT_FLT
 #else
 #   define SHR(a,b)       ((a)>>(b))
-#   define compute_antialias compute_antialias_integer
 /* WARNING: only correct for posititive numbers */
 #   define FIXR_OLD(a)    ((int)((a) * FRAC_ONE + 0.5))
 #   define FIXR(a)        ((int)((a) * FRAC_ONE + 0.5))
@@ -69,7 +67,7 @@
 #include "mpegaudiodata.h"
 #include "mpegaudiodectab.h"
 
-static void compute_antialias(MPADecodeContext *s, GranuleDef *g);
+static void RENAME(compute_antialias)(MPADecodeContext *s, GranuleDef *g);
 static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
                                int *dither_state, OUT_INT *samples, int incr);
 
@@ -1480,8 +1478,7 @@ static void compute_stereo(MPADecodeContext *s,
 }
 
 #if !CONFIG_FLOAT
-static void compute_antialias_integer(MPADecodeContext *s,
-                              GranuleDef *g)
+static void compute_antialias_fixed(MPADecodeContext *s, GranuleDef *g)
 {
     int32_t *ptr, *csa;
     int n, i;
@@ -1848,7 +1845,7 @@ static int mp_decode_layer3(MPADecodeContext *s)
             g = &s->granules[ch][gr];
 
             reorder_block(s, g);
-            compute_antialias(s, g);
+            RENAME(compute_antialias)(s, g);
             compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
         }
     } /* gr */
-- 
cgit v1.2.3


From a06bf6368be2acb760a10289645eba1f65bf967b Mon Sep 17 00:00:00 2001
From: Mans Rullgard <mans@mansr.com>
Date: Mon, 16 May 2011 16:37:38 +0100
Subject: mpegaudiodec: remove decode_end() function

This function is not needed since 721d6f2dc5 removed the DCT table
allocations for the configuration used here.

Signed-off-by: Mans Rullgard <mans@mansr.com>
---
 libavcodec/mpegaudiodec_float.c | 15 ++++-----------
 1 file changed, 4 insertions(+), 11 deletions(-)

(limited to 'libavcodec')

diff --git a/libavcodec/mpegaudiodec_float.c b/libavcodec/mpegaudiodec_float.c
index e9041fcb63..0ef85d19c1 100644
--- a/libavcodec/mpegaudiodec_float.c
+++ b/libavcodec/mpegaudiodec_float.c
@@ -80,13 +80,6 @@ static void compute_antialias_float(MPADecodeContext *s,
     }
 }
 
-static av_cold int decode_end(AVCodecContext * avctx)
-{
-    MPADecodeContext *s = avctx->priv_data;
-    ff_dct_end(&s->dct);
-    return 0;
-}
-
 #if CONFIG_MP1FLOAT_DECODER
 AVCodec ff_mp1float_decoder =
 {
@@ -96,7 +89,7 @@ AVCodec ff_mp1float_decoder =
     sizeof(MPADecodeContext),
     decode_init,
     NULL,
-    decode_end,
+    .close = NULL,
     decode_frame,
     CODEC_CAP_PARSE_ONLY,
     .flush= flush,
@@ -112,7 +105,7 @@ AVCodec ff_mp2float_decoder =
     sizeof(MPADecodeContext),
     decode_init,
     NULL,
-    decode_end,
+    .close = NULL,
     decode_frame,
     CODEC_CAP_PARSE_ONLY,
     .flush= flush,
@@ -128,7 +121,7 @@ AVCodec ff_mp3float_decoder =
     sizeof(MPADecodeContext),
     decode_init,
     NULL,
-    decode_end,
+    .close = NULL,
     decode_frame,
     CODEC_CAP_PARSE_ONLY,
     .flush= flush,
@@ -144,7 +137,7 @@ AVCodec ff_mp3adufloat_decoder =
     sizeof(MPADecodeContext),
     decode_init,
     NULL,
-    decode_end,
+    .close = NULL,
     decode_frame_adu,
     CODEC_CAP_PARSE_ONLY,
     .flush= flush,
-- 
cgit v1.2.3


From 32f8fb8ecf8178b9c9ec8d7152f1fdd8537f7f3a Mon Sep 17 00:00:00 2001
From: Justin Ruggles <justin.ruggles@gmail.com>
Date: Sun, 24 Apr 2011 17:50:17 -0400
Subject: Add float_interleave() to FmtConvertContext with x86-optimized
 versions.

Partially based on patches by clsid2 in ffdshow-tryout.
ff_float_interleave6() x86 improvements by Loren Merrit.
---
 libavcodec/fmtconvert.c         |  20 ++++++
 libavcodec/fmtconvert.h         |   9 +++
 libavcodec/x86/fmtconvert.asm   | 141 ++++++++++++++++++++++++++++++++++++++++
 libavcodec/x86/fmtconvert_mmx.c |  30 +++++++++
 4 files changed, 200 insertions(+)

(limited to 'libavcodec')

diff --git a/libavcodec/fmtconvert.c b/libavcodec/fmtconvert.c
index e9707555af..58fece70b2 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);
diff --git a/libavcodec/fmtconvert.h b/libavcodec/fmtconvert.h
index e0afee47e1..d7741135b7 100644
--- a/libavcodec/fmtconvert.h
+++ b/libavcodec/fmtconvert.h
@@ -68,8 +68,17 @@ typedef struct FmtConvertContext {
      */
     void (*float_to_int16_interleave)(int16_t *dst, const float **src,
                                       long len, int channels);
+
+    /**
+     * Convert an array of interleaved float to multiple arrays of float.
+     */
+    void (*float_interleave)(float *dst, const float **src, unsigned int len,
+                             int channels);
 } FmtConvertContext;
 
+void ff_float_interleave_c(float *dst, const float **src, unsigned int len,
+                           int channels);
+
 void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx);
 
 void ff_fmt_convert_init_arm(FmtConvertContext *c, AVCodecContext *avctx);
diff --git a/libavcodec/x86/fmtconvert.asm b/libavcodec/x86/fmtconvert.asm
index 5cd8f6c596..e023b48322 100644
--- a/libavcodec/x86/fmtconvert.asm
+++ b/libavcodec/x86/fmtconvert.asm
@@ -20,6 +20,7 @@
 ;******************************************************************************
 
 %include "x86inc.asm"
+%include "x86util.asm"
 
 section .text align=16
 
@@ -89,3 +90,143 @@ FLOAT_TO_INT16_INTERLEAVE6 3dnow
 %undef pswapd
 FLOAT_TO_INT16_INTERLEAVE6 3dn2
 %undef cvtps2pi
+
+;-----------------------------------------------------------------------------
+; void ff_float_interleave6(float *dst, const float **src, unsigned int len);
+;-----------------------------------------------------------------------------
+
+%macro BUTTERFLYPS 3
+    movaps    m%3, m%1
+    unpcklps  m%1, m%2
+    unpckhps  m%3, m%2
+    SWAP %2, %3
+%endmacro
+
+%macro FLOAT_INTERLEAVE6 2
+cglobal float_interleave6_%1, 2,7,%2, dst, src, src1, src2, src3, src4, src5
+%ifdef ARCH_X86_64
+    %define lend r10d
+    mov     lend, r2d
+%else
+    %define lend dword r2m
+%endif
+    mov    src1q, [srcq+1*gprsize]
+    mov    src2q, [srcq+2*gprsize]
+    mov    src3q, [srcq+3*gprsize]
+    mov    src4q, [srcq+4*gprsize]
+    mov    src5q, [srcq+5*gprsize]
+    mov     srcq, [srcq]
+    sub    src1q, srcq
+    sub    src2q, srcq
+    sub    src3q, srcq
+    sub    src4q, srcq
+    sub    src5q, srcq
+.loop:
+%ifidn %1, sse
+    movaps    m0, [srcq]
+    movaps    m1, [srcq+src1q]
+    movaps    m2, [srcq+src2q]
+    movaps    m3, [srcq+src3q]
+    movaps    m4, [srcq+src4q]
+    movaps    m5, [srcq+src5q]
+
+    BUTTERFLYPS 0, 1, 6
+    BUTTERFLYPS 2, 3, 6
+    BUTTERFLYPS 4, 5, 6
+
+    movaps    m6, m4
+    shufps    m4, m0, 0xe4
+    movlhps   m0, m2
+    movhlps   m6, m2
+    movaps [dstq   ], m0
+    movaps [dstq+16], m4
+    movaps [dstq+32], m6
+
+    movaps    m6, m5
+    shufps    m5, m1, 0xe4
+    movlhps   m1, m3
+    movhlps   m6, m3
+    movaps [dstq+48], m1
+    movaps [dstq+64], m5
+    movaps [dstq+80], m6
+%else ; mmx
+    movq       m0, [srcq]
+    movq       m1, [srcq+src1q]
+    movq       m2, [srcq+src2q]
+    movq       m3, [srcq+src3q]
+    movq       m4, [srcq+src4q]
+    movq       m5, [srcq+src5q]
+
+    SBUTTERFLY dq, 0, 1, 6
+    SBUTTERFLY dq, 2, 3, 6
+    SBUTTERFLY dq, 4, 5, 6
+    movq [dstq   ], m0
+    movq [dstq+ 8], m2
+    movq [dstq+16], m4
+    movq [dstq+24], m1
+    movq [dstq+32], m3
+    movq [dstq+40], m5
+%endif
+    add      srcq, mmsize
+    add      dstq, mmsize*6
+    sub      lend, mmsize/4
+    jg .loop
+%ifidn %1, mmx
+    emms
+%endif
+    REP_RET
+%endmacro
+
+INIT_MMX
+FLOAT_INTERLEAVE6 mmx, 0
+INIT_XMM
+FLOAT_INTERLEAVE6 sse, 7
+
+;-----------------------------------------------------------------------------
+; void ff_float_interleave2(float *dst, const float **src, unsigned int len);
+;-----------------------------------------------------------------------------
+
+%macro FLOAT_INTERLEAVE2 2
+cglobal float_interleave2_%1, 3,4,%2, dst, src, len, src1
+    mov     src1q, [srcq+gprsize]
+    mov      srcq, [srcq        ]
+    sub     src1q, srcq
+.loop
+    MOVPS      m0, [srcq             ]
+    MOVPS      m1, [srcq+src1q       ]
+    MOVPS      m3, [srcq      +mmsize]
+    MOVPS      m4, [srcq+src1q+mmsize]
+
+    MOVPS      m2, m0
+    PUNPCKLDQ  m0, m1
+    PUNPCKHDQ  m2, m1
+
+    MOVPS      m1, m3
+    PUNPCKLDQ  m3, m4
+    PUNPCKHDQ  m1, m4
+
+    MOVPS [dstq         ], m0
+    MOVPS [dstq+1*mmsize], m2
+    MOVPS [dstq+2*mmsize], m3
+    MOVPS [dstq+3*mmsize], m1
+
+    add      srcq, mmsize*2
+    add      dstq, mmsize*4
+    sub      lend, mmsize/2
+    jg .loop
+%ifidn %1, mmx
+    emms
+%endif
+    REP_RET
+%endmacro
+
+INIT_MMX
+%define MOVPS     movq
+%define PUNPCKLDQ punpckldq
+%define PUNPCKHDQ punpckhdq
+FLOAT_INTERLEAVE2 mmx, 0
+INIT_XMM
+%define MOVPS     movaps
+%define PUNPCKLDQ unpcklps
+%define PUNPCKHDQ unpckhps
+FLOAT_INTERLEAVE2 sse, 5
diff --git a/libavcodec/x86/fmtconvert_mmx.c b/libavcodec/x86/fmtconvert_mmx.c
index 847bd80fcd..61a4272a69 100644
--- a/libavcodec/x86/fmtconvert_mmx.c
+++ b/libavcodec/x86/fmtconvert_mmx.c
@@ -235,11 +235,40 @@ static void float_to_int16_interleave_3dn2(int16_t *dst, const float **src, long
         float_to_int16_interleave_3dnow(dst, src, len, channels);
 }
 
+void ff_float_interleave2_mmx(float *dst, const float **src, unsigned int len);
+void ff_float_interleave2_sse(float *dst, const float **src, unsigned int len);
+
+void ff_float_interleave6_mmx(float *dst, const float **src, unsigned int len);
+void ff_float_interleave6_sse(float *dst, const float **src, unsigned int len);
+
+static void float_interleave_mmx(float *dst, const float **src,
+                                 unsigned int len, int channels)
+{
+    if (channels == 2) {
+        ff_float_interleave2_mmx(dst, src, len);
+    } else if (channels == 6)
+        ff_float_interleave6_mmx(dst, src, len);
+    else
+        ff_float_interleave_c(dst, src, len, channels);
+}
+
+static void float_interleave_sse(float *dst, const float **src,
+                                 unsigned int len, int channels)
+{
+    if (channels == 2) {
+        ff_float_interleave2_sse(dst, src, len);
+    } else if (channels == 6)
+        ff_float_interleave6_sse(dst, src, len);
+    else
+        ff_float_interleave_c(dst, src, len, channels);
+}
+
 void ff_fmt_convert_init_x86(FmtConvertContext *c, AVCodecContext *avctx)
 {
     int mm_flags = av_get_cpu_flags();
 
     if (mm_flags & AV_CPU_FLAG_MMX) {
+        c->float_interleave = float_interleave_mmx;
 
         if(mm_flags & AV_CPU_FLAG_3DNOW){
             if(!(avctx->flags & CODEC_FLAG_BITEXACT)){
@@ -256,6 +285,7 @@ void ff_fmt_convert_init_x86(FmtConvertContext *c, AVCodecContext *avctx)
             c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_sse;
             c->float_to_int16 = float_to_int16_sse;
             c->float_to_int16_interleave = float_to_int16_interleave_sse;
+            c->float_interleave = float_interleave_sse;
         }
         if(mm_flags & AV_CPU_FLAG_SSE2){
             c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_sse2;
-- 
cgit v1.2.3


From 64150ff014708b4b00cb5d76237e9e908ac0fbfc Mon Sep 17 00:00:00 2001
From: Justin Ruggles <justin.ruggles@gmail.com>
Date: Tue, 19 Apr 2011 19:47:01 -0400
Subject: Add request_sample_fmt field to AVCodecContext.

This will allow audio decoders to support output of different sample formats
as a runtime option.
---
 libavcodec/avcodec.h | 7 +++++++
 libavcodec/options.c | 6 ++++++
 libavcodec/version.h | 2 +-
 3 files changed, 14 insertions(+), 1 deletion(-)

(limited to 'libavcodec')

diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 2eb218ba4f..e067ee0273 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -2836,6 +2836,13 @@ typedef struct AVCodecContext {
      * - decoding: Set by libavcodec.
      */
     enum AVAudioServiceType audio_service_type;
+
+    /**
+     * Used to request a sample format from the decoder.
+     * - encoding: unused.
+     * - decoding: Set by user.
+     */
+    enum AVSampleFormat request_sample_fmt;
 } AVCodecContext;
 
 /**
diff --git a/libavcodec/options.c b/libavcodec/options.c
index 9c714fb73e..8f9aec4ac2 100644
--- a/libavcodec/options.c
+++ b/libavcodec/options.c
@@ -441,6 +441,12 @@ static const AVOption options[]={
 {"em", "Emergency",          0, FF_OPT_TYPE_CONST, {.dbl = AV_AUDIO_SERVICE_TYPE_EMERGENCY },         INT_MIN, INT_MAX, A|E, "audio_service_type"},
 {"vo", "Voice Over",         0, FF_OPT_TYPE_CONST, {.dbl = AV_AUDIO_SERVICE_TYPE_VOICE_OVER },        INT_MIN, INT_MAX, A|E, "audio_service_type"},
 {"ka", "Karaoke",            0, FF_OPT_TYPE_CONST, {.dbl = AV_AUDIO_SERVICE_TYPE_KARAOKE },           INT_MIN, INT_MAX, A|E, "audio_service_type"},
+{"request_sample_fmt", NULL, OFFSET(request_sample_fmt), FF_OPT_TYPE_INT, {.dbl = AV_SAMPLE_FMT_NONE }, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NB-1, A|D, "request_sample_fmt"},
+{"u8" , "8-bit unsigned integer", 0, FF_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_U8  }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
+{"s16", "16-bit signed integer",  0, FF_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_S16 }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
+{"s32", "32-bit signed integer",  0, FF_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_S32 }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
+{"flt", "32-bit float",           0, FF_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_FLT }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
+{"dbl", "64-bit double",          0, FF_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_DBL }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
 {NULL},
 };
 
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 1b454b8bd6..0def0d1ff9 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -21,7 +21,7 @@
 #define AVCODEC_VERSION_H
 
 #define LIBAVCODEC_VERSION_MAJOR 53
-#define LIBAVCODEC_VERSION_MINOR  3
+#define LIBAVCODEC_VERSION_MINOR  4
 #define LIBAVCODEC_VERSION_MICRO  0
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
-- 
cgit v1.2.3


From 9aa8193a234ccb6a79cba5cc550531f62ffb0a17 Mon Sep 17 00:00:00 2001
From: Justin Ruggles <justin.ruggles@gmail.com>
Date: Fri, 22 Apr 2011 21:30:19 -0400
Subject: Add floating-point sample format support to the ac3, eac3, dca, aac,
 and vorbis decoders.

Based on patches by clsid2 in ffdshow-tryout.
---
 libavcodec/aacdec.c    | 34 ++++++++++++++++++++++++----------
 libavcodec/aacsbr.c    | 11 ++++++++---
 libavcodec/aacsbr.h    |  2 +-
 libavcodec/ac3dec.c    | 32 ++++++++++++++++++++++++++------
 libavcodec/dca.c       | 34 +++++++++++++++++++++++++++-------
 libavcodec/vorbisdec.c | 23 ++++++++++++++++++-----
 6 files changed, 104 insertions(+), 32 deletions(-)

(limited to 'libavcodec')

diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 5f9dd834a0..f2d50f4aba 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -186,7 +186,7 @@ static av_cold int che_configure(AACContext *ac,
     if (che_pos[type][id]) {
         if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
             return AVERROR(ENOMEM);
-        ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
+        ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
         if (type != TYPE_CCE) {
             ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
             if (type == TYPE_CPE ||
@@ -546,6 +546,7 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
 static av_cold int aac_decode_init(AVCodecContext *avctx)
 {
     AACContext *ac = avctx->priv_data;
+    float output_scale_factor;
 
     ac->avctx = avctx;
     ac->m4ac.sample_rate = avctx->sample_rate;
@@ -557,7 +558,13 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
             return -1;
     }
 
-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+    if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+        output_scale_factor = 1.0 / 32768.0;
+    } else {
+        avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+        output_scale_factor = 1.0;
+    }
 
     AAC_INIT_VLC_STATIC( 0, 304);
     AAC_INIT_VLC_STATIC( 1, 270);
@@ -585,9 +592,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
                     ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
                     352);
 
-    ff_mdct_init(&ac->mdct,       11, 1, 1.0/1024.0);
-    ff_mdct_init(&ac->mdct_small,  8, 1, 1.0/128.0);
-    ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0);
+    ff_mdct_init(&ac->mdct,       11, 1, output_scale_factor/1024.0);
+    ff_mdct_init(&ac->mdct_small,  8, 1, output_scale_factor/128.0);
+    ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0/output_scale_factor);
     // window initialization
     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
     ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
@@ -2169,7 +2176,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
         avctx->frame_size = samples;
     }
 
-    data_size_tmp = samples * avctx->channels * sizeof(int16_t);
+    data_size_tmp = samples * avctx->channels *
+                    (av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
     if (*data_size < data_size_tmp) {
         av_log(avctx, AV_LOG_ERROR,
                "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
@@ -2178,8 +2186,14 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
     }
     *data_size = data_size_tmp;
 
-    if (samples)
-        ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
+    if (samples) {
+        if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
+            ac->fmt_conv.float_interleave(data, (const float **)ac->output_data,
+                                          samples, avctx->channels);
+        else
+            ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data,
+                                                   samples, avctx->channels);
+    }
 
     if (ac->output_configured)
         ac->output_configured = OC_LOCKED;
@@ -2497,7 +2511,7 @@ AVCodec ff_aac_decoder = {
     aac_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
     .sample_fmts = (const enum AVSampleFormat[]) {
-        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
     },
     .channel_layouts = aac_channel_layout,
 };
@@ -2517,7 +2531,7 @@ AVCodec ff_aac_latm_decoder = {
     .decode = latm_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
     .sample_fmts = (const enum AVSampleFormat[]) {
-        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
     },
     .channel_layouts = aac_channel_layout,
 };
diff --git a/libavcodec/aacsbr.c b/libavcodec/aacsbr.c
index 7a217abfda..81b0b4c001 100644
--- a/libavcodec/aacsbr.c
+++ b/libavcodec/aacsbr.c
@@ -126,14 +126,19 @@ av_cold void ff_aac_sbr_init(void)
     ff_ps_init();
 }
 
-av_cold void ff_aac_sbr_ctx_init(SpectralBandReplication *sbr)
+av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
 {
+    float mdct_scale;
     sbr->kx[0] = sbr->kx[1] = 32; //Typo in spec, kx' inits to 32
     sbr->data[0].e_a[1] = sbr->data[1].e_a[1] = -1;
     sbr->data[0].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
     sbr->data[1].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
-    ff_mdct_init(&sbr->mdct, 7, 1, 1.0/64);
-    ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0);
+    /* SBR requires samples to be scaled to +/-32768.0 to work correctly.
+     * mdct scale factors are adjusted to scale up from +/-1.0 at analysis
+     * and scale back down at synthesis. */
+    mdct_scale = ac->avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? 32768.0f : 1.0f;
+    ff_mdct_init(&sbr->mdct,     7, 1, 1.0 / (64 * mdct_scale));
+    ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0 * mdct_scale);
     ff_ps_ctx_init(&sbr->ps);
 }
 
diff --git a/libavcodec/aacsbr.h b/libavcodec/aacsbr.h
index dca83305ad..153070d3f2 100644
--- a/libavcodec/aacsbr.h
+++ b/libavcodec/aacsbr.h
@@ -36,7 +36,7 @@
 /** Initialize SBR. */
 av_cold void ff_aac_sbr_init(void);
 /** Initialize one SBR context. */
-av_cold void ff_aac_sbr_ctx_init(SpectralBandReplication *sbr);
+av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr);
 /** Close one SBR context. */
 av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr);
 /** Decode one SBR element. */
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c
index 015ebaebec..2966c33b25 100644
--- a/libavcodec/ac3dec.c
+++ b/libavcodec/ac3dec.c
@@ -189,7 +189,13 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
     av_lfg_init(&s->dith_state, 0);
 
     /* set scale value for float to int16 conversion */
-    s->mul_bias = 32767.0f;
+    if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+        s->mul_bias = 1.0f;
+        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+    } else {
+        s->mul_bias = 32767.0f;
+        avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+    }
 
     /* allow downmixing to stereo or mono */
     if (avctx->channels > 0 && avctx->request_channels > 0 &&
@@ -204,7 +210,6 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
         if (!s->input_buffer)
             return AVERROR(ENOMEM);
 
-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
     return 0;
 }
 
@@ -1299,7 +1304,8 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
     AC3DecodeContext *s = avctx->priv_data;
-    int16_t *out_samples = (int16_t *)data;
+    float   *out_samples_flt = data;
+    int16_t *out_samples_s16 = data;
     int blk, ch, err;
     const uint8_t *channel_map;
     const float *output[AC3_MAX_CHANNELS];
@@ -1405,10 +1411,18 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
             av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
             err = 1;
         }
-        s->fmt_conv.float_to_int16_interleave(out_samples, output, 256, s->out_channels);
-        out_samples += 256 * s->out_channels;
+        if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+            s->fmt_conv.float_interleave(out_samples_flt, output, 256,
+                                         s->out_channels);
+            out_samples_flt += 256 * s->out_channels;
+        } else {
+            s->fmt_conv.float_to_int16_interleave(out_samples_s16, output, 256,
+                                                  s->out_channels);
+            out_samples_s16 += 256 * s->out_channels;
+        }
     }
-    *data_size = s->num_blocks * 256 * avctx->channels * sizeof (int16_t);
+    *data_size = s->num_blocks * 256 * avctx->channels *
+                 (av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
     return FFMIN(buf_size, s->frame_size);
 }
 
@@ -1435,6 +1449,9 @@ AVCodec ff_ac3_decoder = {
     .close = ac3_decode_end,
     .decode = ac3_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+    .sample_fmts = (const enum AVSampleFormat[]) {
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+    },
 };
 
 #if CONFIG_EAC3_DECODER
@@ -1447,5 +1464,8 @@ AVCodec ff_eac3_decoder = {
     .close = ac3_decode_end,
     .decode = ac3_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
+    .sample_fmts = (const enum AVSampleFormat[]) {
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+    },
 };
 #endif
diff --git a/libavcodec/dca.c b/libavcodec/dca.c
index f1cd64e9ed..dbadebacff 100644
--- a/libavcodec/dca.c
+++ b/libavcodec/dca.c
@@ -1626,7 +1626,9 @@ static int dca_decode_frame(AVCodecContext * avctx,
     int lfe_samples;
     int num_core_channels = 0;
     int i;
-    int16_t *samples = data;
+    float   *samples_flt = data;
+    int16_t *samples_s16 = data;
+    int out_size;
     DCAContext *s = avctx->priv_data;
     int channels;
     int core_ss_end;
@@ -1812,9 +1814,11 @@ static int dca_decode_frame(AVCodecContext * avctx,
         return -1;
     }
 
-    if (*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
+    out_size = 256 / 8 * s->sample_blocks * channels *
+               (av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
+    if (*data_size < out_size)
         return -1;
-    *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
+    *data_size = out_size;
 
     /* filter to get final output */
     for (i = 0; i < (s->sample_blocks / 8); i++) {
@@ -1833,8 +1837,16 @@ static int dca_decode_frame(AVCodecContext * avctx,
             }
         }
 
-        s->fmt_conv.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
-        samples += 256 * channels;
+        if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+            s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,
+                                         channels);
+            samples_flt += 256 * channels;
+        } else {
+            s->fmt_conv.float_to_int16_interleave(samples_s16,
+                                                  s->samples_chanptr, 256,
+                                                  channels);
+            samples_s16 += 256 * channels;
+        }
     }
 
     /* update lfe history */
@@ -1870,9 +1882,14 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
 
     for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
         s->samples_chanptr[i] = s->samples + i * 256;
-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
 
-    s->scale_bias = 1.0;
+    if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+        s->scale_bias = 1.0 / 32768.0;
+    } else {
+        avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+        s->scale_bias = 1.0;
+    }
 
     /* allow downmixing to stereo */
     if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
@@ -1909,5 +1926,8 @@ AVCodec ff_dca_decoder = {
     .close = dca_decode_end,
     .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
     .capabilities = CODEC_CAP_CHANNEL_CONF,
+    .sample_fmts = (const enum AVSampleFormat[]) {
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+    },
     .profiles = NULL_IF_CONFIG_SMALL(profiles),
 };
diff --git a/libavcodec/vorbisdec.c b/libavcodec/vorbisdec.c
index 7443e98a64..f6ec74f4a1 100644
--- a/libavcodec/vorbisdec.c
+++ b/libavcodec/vorbisdec.c
@@ -979,7 +979,13 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
     dsputil_init(&vc->dsp, avccontext);
     ff_fmt_convert_init(&vc->fmt_conv, avccontext);
 
-    vc->scale_bias = 32768.0f;
+    if (avccontext->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+        avccontext->sample_fmt = AV_SAMPLE_FMT_FLT;
+        vc->scale_bias = 1.0f;
+    } else {
+        avccontext->sample_fmt = AV_SAMPLE_FMT_S16;
+        vc->scale_bias = 32768.0f;
+    }
 
     if (!headers_len) {
         av_log(avccontext, AV_LOG_ERROR, "Extradata missing.\n");
@@ -1024,7 +1030,6 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
     avccontext->channels    = vc->audio_channels;
     avccontext->sample_rate = vc->audio_samplerate;
     avccontext->frame_size  = FFMIN(vc->blocksize[0], vc->blocksize[1]) >> 2;
-    avccontext->sample_fmt  = AV_SAMPLE_FMT_S16;
 
     return 0 ;
 }
@@ -1634,9 +1639,14 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
                               len * ff_vorbis_channel_layout_offsets[vc->audio_channels - 1][i];
     }
 
-    vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len,
-                                           vc->audio_channels);
-    *data_size = len * 2 * vc->audio_channels;
+    if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT)
+        vc->fmt_conv.float_interleave(data, channel_ptrs, len, vc->audio_channels);
+    else
+        vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len,
+                                               vc->audio_channels);
+
+    *data_size = len * vc->audio_channels *
+                 (av_get_bits_per_sample_fmt(avccontext->sample_fmt) / 8);
 
     return buf_size ;
 }
@@ -1663,5 +1673,8 @@ AVCodec ff_vorbis_decoder = {
     vorbis_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
     .channel_layouts = ff_vorbis_channel_layouts,
+    .sample_fmts = (const enum AVSampleFormat[]) {
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+    },
 };
 
-- 
cgit v1.2.3