[FFmpeg-cvslog] dsputil: Split audio operations off into a separate context
Diego Biurrun
git at videolan.org
Sun Jun 22 18:14:34 CEST 2014
ffmpeg | branch: master | Diego Biurrun <diego at biurrun.de> | Thu Jan 16 17:30:19 2014 +0100| [9a9e2f1c8aa4539a261625145e5c1f46a8106ac2] | committer: Diego Biurrun
dsputil: Split audio operations off into a separate context
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=9a9e2f1c8aa4539a261625145e5c1f46a8106ac2
---
configure | 9 +-
libavcodec/Makefile | 1 +
libavcodec/ac3enc.c | 2 +
libavcodec/ac3enc.h | 2 +
libavcodec/ac3enc_fixed.c | 6 +-
libavcodec/ac3enc_float.c | 6 +-
libavcodec/ac3enc_template.c | 9 +-
libavcodec/acelp_pitch_delay.c | 5 +-
libavcodec/acelp_pitch_delay.h | 7 +-
libavcodec/arm/Makefile | 5 +-
libavcodec/arm/audiodsp_arm.h | 26 +++++
libavcodec/arm/audiodsp_init_arm.c | 33 +++++++
libavcodec/arm/audiodsp_init_neon.c | 41 ++++++++
libavcodec/arm/audiodsp_neon.S | 64 ++++++++++++
libavcodec/arm/dsputil_init_neon.c | 12 ---
libavcodec/arm/dsputil_neon.S | 42 --------
libavcodec/audiodsp.c | 118 ++++++++++++++++++++++
libavcodec/audiodsp.h | 59 +++++++++++
libavcodec/cook.c | 11 ++-
libavcodec/dsputil.c | 85 ----------------
libavcodec/dsputil.h | 29 ------
libavcodec/ppc/Makefile | 2 +-
libavcodec/ppc/{int_altivec.c => audiodsp.c} | 18 +++-
libavcodec/ppc/dsputil_altivec.h | 1 -
libavcodec/ppc/dsputil_ppc.c | 2 +-
libavcodec/takdec.c | 14 +--
libavcodec/x86/Makefile | 3 +
libavcodec/x86/audiodsp.asm | 137 ++++++++++++++++++++++++++
libavcodec/x86/audiodsp.h | 25 +++++
libavcodec/x86/audiodsp_init.c | 66 +++++++++++++
libavcodec/x86/audiodsp_mmx.c | 58 +++++++++++
libavcodec/x86/dsputil.asm | 113 ---------------------
libavcodec/x86/dsputil_init.c | 50 ----------
libavcodec/x86/dsputil_mmx.c | 34 -------
libavcodec/x86/dsputil_x86.h | 3 -
35 files changed, 694 insertions(+), 404 deletions(-)
diff --git a/configure b/configure
index c271994..a65ccd8 100755
--- a/configure
+++ b/configure
@@ -1529,6 +1529,7 @@ CONFIG_EXTRA="
aandcttables
ac3dsp
audio_frame_queue
+ audiodsp
blockdsp
cabac
dsputil
@@ -1713,8 +1714,8 @@ aac_decoder_select="mdct sinewin"
aac_encoder_select="audio_frame_queue mdct sinewin"
aac_latm_decoder_select="aac_decoder aac_latm_parser"
ac3_decoder_select="mdct ac3dsp ac3_parser dsputil"
-ac3_encoder_select="mdct ac3dsp dsputil"
-ac3_fixed_encoder_select="mdct ac3dsp dsputil"
+ac3_encoder_select="ac3dsp audiodsp dsputil mdct"
+ac3_fixed_encoder_select="ac3dsp audiodsp dsputil mdct"
aic_decoder_select="dsputil golomb"
alac_encoder_select="lpc"
als_decoder_select="dsputil"
@@ -1735,7 +1736,7 @@ binkaudio_rdft_decoder_select="mdct rdft sinewin"
cavs_decoder_select="blockdsp dsputil golomb h264chroma qpeldsp videodsp"
cllc_decoder_select="dsputil"
comfortnoise_encoder_select="lpc"
-cook_decoder_select="dsputil mdct sinewin"
+cook_decoder_select="audiodsp mdct sinewin"
cscd_decoder_select="lzo"
cscd_decoder_suggest="zlib"
dca_decoder_select="mdct"
@@ -1849,7 +1850,7 @@ svq1_decoder_select="hpeldsp"
svq1_encoder_select="aandcttables dsputil hpeldsp mpegvideoenc"
svq3_decoder_select="h264_decoder hpeldsp tpeldsp"
svq3_decoder_suggest="zlib"
-tak_decoder_select="dsputil"
+tak_decoder_select="audiodsp"
theora_decoder_select="vp3_decoder"
thp_decoder_select="mjpeg_decoder"
tiff_decoder_suggest="zlib"
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index c591545..c2f7532 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -28,6 +28,7 @@ OBJS = allcodecs.o \
OBJS-$(CONFIG_AANDCTTABLES) += aandcttab.o
OBJS-$(CONFIG_AC3DSP) += ac3dsp.o
OBJS-$(CONFIG_AUDIO_FRAME_QUEUE) += audio_frame_queue.o
+OBJS-$(CONFIG_AUDIODSP) += audiodsp.o
OBJS-$(CONFIG_BLOCKDSP) += blockdsp.o
OBJS-$(CONFIG_CABAC) += cabac.o
OBJS-$(CONFIG_DCT) += dct.o dct32_fixed.o dct32_float.o
diff --git a/libavcodec/ac3enc.c b/libavcodec/ac3enc.c
index cc8df47..c6dc141 100644
--- a/libavcodec/ac3enc.c
+++ b/libavcodec/ac3enc.c
@@ -37,6 +37,7 @@
#include "libavutil/opt.h"
#include "avcodec.h"
#include "put_bits.h"
+#include "audiodsp.h"
#include "ac3dsp.h"
#include "ac3.h"
#include "fft.h"
@@ -2480,6 +2481,7 @@ av_cold int ff_ac3_encode_init(AVCodecContext *avctx)
if (ret)
goto init_fail;
+ ff_audiodsp_init(&s->adsp);
ff_dsputil_init(&s->dsp, avctx);
ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
diff --git a/libavcodec/ac3enc.h b/libavcodec/ac3enc.h
index 90bbf2a..b8e8768 100644
--- a/libavcodec/ac3enc.h
+++ b/libavcodec/ac3enc.h
@@ -39,6 +39,7 @@
#include "fft.h"
#include "mathops.h"
#include "put_bits.h"
+#include "audiodsp.h"
#ifndef CONFIG_AC3ENC_FLOAT
#define CONFIG_AC3ENC_FLOAT 0
@@ -162,6 +163,7 @@ typedef struct AC3EncodeContext {
AVCodecContext *avctx; ///< parent AVCodecContext
PutBitContext pb; ///< bitstream writer context
DSPContext dsp;
+ AudioDSPContext adsp;
AVFloatDSPContext fdsp;
AC3DSPContext ac3dsp; ///< AC-3 optimized functions
FFTContext mdct; ///< FFT context for MDCT calculation
diff --git a/libavcodec/ac3enc_fixed.c b/libavcodec/ac3enc_fixed.c
index f76d2ad..2bb82ef 100644
--- a/libavcodec/ac3enc_fixed.c
+++ b/libavcodec/ac3enc_fixed.c
@@ -29,6 +29,7 @@
#define FFT_FLOAT 0
#undef CONFIG_AC3ENC_FLOAT
#include "internal.h"
+#include "audiodsp.h"
#include "ac3enc.h"
#include "eac3enc.h"
@@ -100,9 +101,10 @@ static void scale_coefficients(AC3EncodeContext *s)
/*
* Clip MDCT coefficients to allowable range.
*/
-static void clip_coefficients(DSPContext *dsp, int32_t *coef, unsigned int len)
+static void clip_coefficients(AudioDSPContext *adsp, int32_t *coef,
+ unsigned int len)
{
- dsp->vector_clip_int32(coef, coef, COEF_MIN, COEF_MAX, len);
+ adsp->vector_clip_int32(coef, coef, COEF_MIN, COEF_MAX, len);
}
diff --git a/libavcodec/ac3enc_float.c b/libavcodec/ac3enc_float.c
index 6b6290f..d106d1b 100644
--- a/libavcodec/ac3enc_float.c
+++ b/libavcodec/ac3enc_float.c
@@ -28,6 +28,7 @@
#define CONFIG_AC3ENC_FLOAT 1
#include "internal.h"
+#include "audiodsp.h"
#include "ac3enc.h"
#include "eac3enc.h"
#include "kbdwin.h"
@@ -107,9 +108,10 @@ static void scale_coefficients(AC3EncodeContext *s)
/*
* Clip MDCT coefficients to allowable range.
*/
-static void clip_coefficients(DSPContext *dsp, float *coef, unsigned int len)
+static void clip_coefficients(AudioDSPContext *adsp, float *coef,
+ unsigned int len)
{
- dsp->vector_clipf(coef, coef, COEF_MIN, COEF_MAX, len);
+ adsp->vector_clipf(coef, coef, COEF_MIN, COEF_MAX, len);
}
diff --git a/libavcodec/ac3enc_template.c b/libavcodec/ac3enc_template.c
index ad296e1..79b4946 100644
--- a/libavcodec/ac3enc_template.c
+++ b/libavcodec/ac3enc_template.c
@@ -30,6 +30,8 @@
#include "libavutil/attributes.h"
#include "libavutil/internal.h"
+
+#include "audiodsp.h"
#include "internal.h"
#include "ac3enc.h"
#include "eac3enc.h"
@@ -40,7 +42,8 @@ static void scale_coefficients(AC3EncodeContext *s);
static int normalize_samples(AC3EncodeContext *s);
-static void clip_coefficients(DSPContext *dsp, CoefType *coef, unsigned int len);
+static void clip_coefficients(AudioDSPContext *adsp, CoefType *coef,
+ unsigned int len);
static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl);
@@ -161,7 +164,7 @@ static void apply_channel_coupling(AC3EncodeContext *s)
}
/* coefficients must be clipped in order to be encoded */
- clip_coefficients(&s->dsp, cpl_coef, num_cpl_coefs);
+ clip_coefficients(&s->adsp, cpl_coef, num_cpl_coefs);
}
/* calculate energy in each band in coupling channel and each fbw channel */
@@ -412,7 +415,7 @@ int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt,
if (s->fixed_point)
scale_coefficients(s);
- clip_coefficients(&s->dsp, s->blocks[0].mdct_coef[1],
+ clip_coefficients(&s->adsp, s->blocks[0].mdct_coef[1],
AC3_MAX_COEFS * s->num_blocks * s->channels);
s->cpl_on = s->cpl_enabled;
diff --git a/libavcodec/acelp_pitch_delay.c b/libavcodec/acelp_pitch_delay.c
index ab09bdb..1965772 100644
--- a/libavcodec/acelp_pitch_delay.c
+++ b/libavcodec/acelp_pitch_delay.c
@@ -26,6 +26,7 @@
#include "avcodec.h"
#include "acelp_pitch_delay.h"
#include "celp_math.h"
+#include "audiodsp.h"
int ff_acelp_decode_8bit_to_1st_delay3(int ac_index)
{
@@ -90,7 +91,7 @@ void ff_acelp_update_past_gain(
}
int16_t ff_acelp_decode_gain_code(
- DSPContext *dsp,
+ AudioDSPContext *adsp,
int gain_corr_factor,
const int16_t* fc_v,
int mr_energy,
@@ -107,7 +108,7 @@ int16_t ff_acelp_decode_gain_code(
mr_energy += quant_energy[i] * ma_prediction_coeff[i];
mr_energy = gain_corr_factor * exp(M_LN10 / (20 << 23) * mr_energy) /
- sqrt(dsp->scalarproduct_int16(fc_v, fc_v, subframe_size));
+ sqrt(adsp->scalarproduct_int16(fc_v, fc_v, subframe_size));
return mr_energy >> 12;
}
diff --git a/libavcodec/acelp_pitch_delay.h b/libavcodec/acelp_pitch_delay.h
index e5410bb..7b5b33d 100644
--- a/libavcodec/acelp_pitch_delay.h
+++ b/libavcodec/acelp_pitch_delay.h
@@ -24,7 +24,8 @@
#define AVCODEC_ACELP_PITCH_DELAY_H
#include <stdint.h>
-#include "dsputil.h"
+
+#include "audiodsp.h"
#define PITCH_DELAY_MIN 20
#define PITCH_DELAY_MAX 143
@@ -139,7 +140,7 @@ void ff_acelp_update_past_gain(
/**
* @brief Decode the adaptive codebook gain and add
* correction (4.1.5 and 3.9.1 of G.729).
- * @param dsp initialized dsputil context
+ * @param adsp initialized audio DSP context
* @param gain_corr_factor gain correction factor (2.13)
* @param fc_v fixed-codebook vector (2.13)
* @param mr_energy mean innovation energy and fixed-point correction (7.13)
@@ -208,7 +209,7 @@ void ff_acelp_update_past_gain(
* @remark The routine is used in G.729 and AMR (all modes).
*/
int16_t ff_acelp_decode_gain_code(
- DSPContext *dsp,
+ AudioDSPContext *adsp,
int gain_corr_factor,
const int16_t* fc_v,
int mr_energy,
diff --git a/libavcodec/arm/Makefile b/libavcodec/arm/Makefile
index 381e997..eb92a8c 100644
--- a/libavcodec/arm/Makefile
+++ b/libavcodec/arm/Makefile
@@ -4,6 +4,7 @@ OBJS += arm/fmtconvert_init_arm.o
OBJS-$(CONFIG_AC3DSP) += arm/ac3dsp_init_arm.o \
arm/ac3dsp_arm.o
+OBJS-$(CONFIG_AUDIODSP) += arm/audiodsp_init_arm.o
OBJS-$(CONFIG_BLOCKDSP) += arm/blockdsp_init_arm.o
OBJS-$(CONFIG_DSPUTIL) += arm/dsputil_init_arm.o \
arm/dsputil_arm.o \
@@ -77,11 +78,13 @@ VFP-OBJS-$(CONFIG_DCA_DECODER) += arm/dcadsp_vfp.o \
NEON-OBJS += arm/fmtconvert_neon.o
NEON-OBJS-$(CONFIG_AC3DSP) += arm/ac3dsp_neon.o
+NEON-OBJS-$(CONFIG_AUDIODSP) += arm/audiodsp_init_neon.o \
+ arm/audiodsp_neon.o \
+ arm/int_neon.o
NEON-OBJS-$(CONFIG_BLOCKDSP) += arm/blockdsp_init_neon.o \
arm/blockdsp_neon.o
NEON-OBJS-$(CONFIG_DSPUTIL) += arm/dsputil_init_neon.o \
arm/dsputil_neon.o \
- arm/int_neon.o \
arm/simple_idct_neon.o
NEON-OBJS-$(CONFIG_FFT) += arm/fft_neon.o \
arm/fft_fixed_neon.o
diff --git a/libavcodec/arm/audiodsp_arm.h b/libavcodec/arm/audiodsp_arm.h
new file mode 100644
index 0000000..e97e804
--- /dev/null
+++ b/libavcodec/arm/audiodsp_arm.h
@@ -0,0 +1,26 @@
+/*
+ * 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_AUDIODSP_ARM_H
+#define AVCODEC_ARM_AUDIODSP_ARM_H
+
+#include "libavcodec/audiodsp.h"
+
+void ff_audiodsp_init_neon(AudioDSPContext *c);
+
+#endif /* AVCODEC_ARM_AUDIODSP_ARM_H */
diff --git a/libavcodec/arm/audiodsp_init_arm.c b/libavcodec/arm/audiodsp_init_arm.c
new file mode 100644
index 0000000..ea9ec3c
--- /dev/null
+++ b/libavcodec/arm/audiodsp_init_arm.c
@@ -0,0 +1,33 @@
+/*
+ * ARM optimized audio functions
+ *
+ * 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 "libavutil/attributes.h"
+#include "libavutil/cpu.h"
+#include "libavutil/arm/cpu.h"
+#include "libavcodec/audiodsp.h"
+#include "audiodsp_arm.h"
+
+av_cold void ff_audiodsp_init_arm(AudioDSPContext *c)
+{
+ int cpu_flags = av_get_cpu_flags();
+
+ if (have_neon(cpu_flags))
+ ff_audiodsp_init_neon(c);
+}
diff --git a/libavcodec/arm/audiodsp_init_neon.c b/libavcodec/arm/audiodsp_init_neon.c
new file mode 100644
index 0000000..af53272
--- /dev/null
+++ b/libavcodec/arm/audiodsp_init_neon.c
@@ -0,0 +1,41 @@
+/*
+ * ARM NEON optimised audio functions
+ * Copyright (c) 2008 Mans Rullgard <mans at 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 <stdint.h>
+
+#include "libavutil/attributes.h"
+#include "libavcodec/audiodsp.h"
+#include "audiodsp_arm.h"
+
+void ff_vector_clipf_neon(float *dst, const float *src, float min, float max,
+ int len);
+void ff_vector_clip_int32_neon(int32_t *dst, const int32_t *src, int32_t min,
+ int32_t max, unsigned int len);
+
+int32_t ff_scalarproduct_int16_neon(const int16_t *v1, const int16_t *v2, int len);
+
+av_cold void ff_audiodsp_init_neon(AudioDSPContext *c)
+{
+ c->vector_clip_int32 = ff_vector_clip_int32_neon;
+ c->vector_clipf = ff_vector_clipf_neon;
+
+ c->scalarproduct_int16 = ff_scalarproduct_int16_neon;
+}
diff --git a/libavcodec/arm/audiodsp_neon.S b/libavcodec/arm/audiodsp_neon.S
new file mode 100644
index 0000000..dfb998d
--- /dev/null
+++ b/libavcodec/arm/audiodsp_neon.S
@@ -0,0 +1,64 @@
+/*
+ * ARM NEON optimised audio functions
+ * Copyright (c) 2008 Mans Rullgard <mans at 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 "libavutil/arm/asm.S"
+
+function ff_vector_clipf_neon, export=1
+VFP vdup.32 q1, d0[1]
+VFP vdup.32 q0, d0[0]
+NOVFP vdup.32 q0, r2
+NOVFP vdup.32 q1, r3
+NOVFP ldr r2, [sp]
+ vld1.f32 {q2},[r1,:128]!
+ vmin.f32 q10, q2, q1
+ vld1.f32 {q3},[r1,:128]!
+ vmin.f32 q11, q3, q1
+1: vmax.f32 q8, q10, q0
+ vmax.f32 q9, q11, q0
+ subs r2, r2, #8
+ beq 2f
+ vld1.f32 {q2},[r1,:128]!
+ vmin.f32 q10, q2, q1
+ vld1.f32 {q3},[r1,:128]!
+ vmin.f32 q11, q3, q1
+ vst1.f32 {q8},[r0,:128]!
+ vst1.f32 {q9},[r0,:128]!
+ b 1b
+2: vst1.f32 {q8},[r0,:128]!
+ vst1.f32 {q9},[r0,:128]!
+ bx lr
+endfunc
+
+function ff_vector_clip_int32_neon, export=1
+ vdup.32 q0, r2
+ vdup.32 q1, r3
+ ldr r2, [sp]
+1:
+ vld1.32 {q2-q3}, [r1,:128]!
+ vmin.s32 q2, q2, q1
+ vmin.s32 q3, q3, q1
+ vmax.s32 q2, q2, q0
+ vmax.s32 q3, q3, q0
+ vst1.32 {q2-q3}, [r0,:128]!
+ subs r2, r2, #8
+ bgt 1b
+ bx lr
+endfunc
diff --git a/libavcodec/arm/dsputil_init_neon.c b/libavcodec/arm/dsputil_init_neon.c
index 6863e05..9d4c76c 100644
--- a/libavcodec/arm/dsputil_init_neon.c
+++ b/libavcodec/arm/dsputil_init_neon.c
@@ -34,13 +34,6 @@ void ff_add_pixels_clamped_neon(const int16_t *, uint8_t *, int);
void ff_put_pixels_clamped_neon(const int16_t *, uint8_t *, int);
void ff_put_signed_pixels_clamped_neon(const int16_t *, uint8_t *, int);
-void ff_vector_clipf_neon(float *dst, const float *src, float min, float max,
- int len);
-void ff_vector_clip_int32_neon(int32_t *dst, const int32_t *src, int32_t min,
- int32_t max, unsigned int len);
-
-int32_t ff_scalarproduct_int16_neon(const int16_t *v1, const int16_t *v2, int len);
-
av_cold void ff_dsputil_init_neon(DSPContext *c, AVCodecContext *avctx,
unsigned high_bit_depth)
{
@@ -57,9 +50,4 @@ av_cold void ff_dsputil_init_neon(DSPContext *c, AVCodecContext *avctx,
c->add_pixels_clamped = ff_add_pixels_clamped_neon;
c->put_pixels_clamped = ff_put_pixels_clamped_neon;
c->put_signed_pixels_clamped = ff_put_signed_pixels_clamped_neon;
-
- c->vector_clipf = ff_vector_clipf_neon;
- c->vector_clip_int32 = ff_vector_clip_int32_neon;
-
- c->scalarproduct_int16 = ff_scalarproduct_int16_neon;
}
diff --git a/libavcodec/arm/dsputil_neon.S b/libavcodec/arm/dsputil_neon.S
index d494ec7..ed6f218 100644
--- a/libavcodec/arm/dsputil_neon.S
+++ b/libavcodec/arm/dsputil_neon.S
@@ -126,45 +126,3 @@ function ff_add_pixels_clamped_neon, export=1
vst1.8 {d6}, [r3,:64], r2
bx lr
endfunc
-
-function ff_vector_clipf_neon, export=1
-VFP vdup.32 q1, d0[1]
-VFP vdup.32 q0, d0[0]
-NOVFP vdup.32 q0, r2
-NOVFP vdup.32 q1, r3
-NOVFP ldr r2, [sp]
- vld1.f32 {q2},[r1,:128]!
- vmin.f32 q10, q2, q1
- vld1.f32 {q3},[r1,:128]!
- vmin.f32 q11, q3, q1
-1: vmax.f32 q8, q10, q0
- vmax.f32 q9, q11, q0
- subs r2, r2, #8
- beq 2f
- vld1.f32 {q2},[r1,:128]!
- vmin.f32 q10, q2, q1
- vld1.f32 {q3},[r1,:128]!
- vmin.f32 q11, q3, q1
- vst1.f32 {q8},[r0,:128]!
- vst1.f32 {q9},[r0,:128]!
- b 1b
-2: vst1.f32 {q8},[r0,:128]!
- vst1.f32 {q9},[r0,:128]!
- bx lr
-endfunc
-
-function ff_vector_clip_int32_neon, export=1
- vdup.32 q0, r2
- vdup.32 q1, r3
- ldr r2, [sp]
-1:
- vld1.32 {q2-q3}, [r1,:128]!
- vmin.s32 q2, q2, q1
- vmin.s32 q3, q3, q1
- vmax.s32 q2, q2, q0
- vmax.s32 q3, q3, q0
- vst1.32 {q2-q3}, [r0,:128]!
- subs r2, r2, #8
- bgt 1b
- bx lr
-endfunc
diff --git a/libavcodec/audiodsp.c b/libavcodec/audiodsp.c
new file mode 100644
index 0000000..f7e6167
--- /dev/null
+++ b/libavcodec/audiodsp.c
@@ -0,0 +1,118 @@
+/*
+ * 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 "libavutil/attributes.h"
+#include "libavutil/common.h"
+#include "audiodsp.h"
+
+static inline uint32_t clipf_c_one(uint32_t a, uint32_t mini,
+ uint32_t maxi, uint32_t maxisign)
+{
+ if (a > mini)
+ return mini;
+ else if ((a ^ (1U << 31)) > maxisign)
+ return maxi;
+ else
+ return a;
+}
+
+static void vector_clipf_c_opposite_sign(float *dst, const float *src,
+ float *min, float *max, int len)
+{
+ int i;
+ uint32_t mini = *(uint32_t *) min;
+ uint32_t maxi = *(uint32_t *) max;
+ uint32_t maxisign = maxi ^ (1U << 31);
+ uint32_t *dsti = (uint32_t *) dst;
+ const uint32_t *srci = (const uint32_t *) src;
+
+ for (i = 0; i < len; i += 8) {
+ dsti[i + 0] = clipf_c_one(srci[i + 0], mini, maxi, maxisign);
+ dsti[i + 1] = clipf_c_one(srci[i + 1], mini, maxi, maxisign);
+ dsti[i + 2] = clipf_c_one(srci[i + 2], mini, maxi, maxisign);
+ dsti[i + 3] = clipf_c_one(srci[i + 3], mini, maxi, maxisign);
+ dsti[i + 4] = clipf_c_one(srci[i + 4], mini, maxi, maxisign);
+ dsti[i + 5] = clipf_c_one(srci[i + 5], mini, maxi, maxisign);
+ dsti[i + 6] = clipf_c_one(srci[i + 6], mini, maxi, maxisign);
+ dsti[i + 7] = clipf_c_one(srci[i + 7], mini, maxi, maxisign);
+ }
+}
+
+static void vector_clipf_c(float *dst, const float *src,
+ float min, float max, int len)
+{
+ int i;
+
+ if (min < 0 && max > 0) {
+ vector_clipf_c_opposite_sign(dst, src, &min, &max, len);
+ } else {
+ for (i = 0; i < len; i += 8) {
+ dst[i] = av_clipf(src[i], min, max);
+ dst[i + 1] = av_clipf(src[i + 1], min, max);
+ dst[i + 2] = av_clipf(src[i + 2], min, max);
+ dst[i + 3] = av_clipf(src[i + 3], min, max);
+ dst[i + 4] = av_clipf(src[i + 4], min, max);
+ dst[i + 5] = av_clipf(src[i + 5], min, max);
+ dst[i + 6] = av_clipf(src[i + 6], min, max);
+ dst[i + 7] = av_clipf(src[i + 7], min, max);
+ }
+ }
+}
+
+static int32_t scalarproduct_int16_c(const int16_t *v1, const int16_t *v2,
+ int order)
+{
+ int res = 0;
+
+ while (order--)
+ res += *v1++ **v2++;
+
+ return res;
+}
+
+static void vector_clip_int32_c(int32_t *dst, const int32_t *src, int32_t min,
+ int32_t max, unsigned int len)
+{
+ do {
+ *dst++ = av_clip(*src++, min, max);
+ *dst++ = av_clip(*src++, min, max);
+ *dst++ = av_clip(*src++, min, max);
+ *dst++ = av_clip(*src++, min, max);
+ *dst++ = av_clip(*src++, min, max);
+ *dst++ = av_clip(*src++, min, max);
+ *dst++ = av_clip(*src++, min, max);
+ *dst++ = av_clip(*src++, min, max);
+ len -= 8;
+ } while (len > 0);
+}
+
+av_cold void ff_audiodsp_init(AudioDSPContext *c)
+{
+ c->scalarproduct_int16 = scalarproduct_int16_c;
+ c->vector_clip_int32 = vector_clip_int32_c;
+ c->vector_clipf = vector_clipf_c;
+
+ if (ARCH_ARM)
+ ff_audiodsp_init_arm(c);
+ if (ARCH_PPC)
+ ff_audiodsp_init_ppc(c);
+ if (ARCH_X86)
+ ff_audiodsp_init_x86(c);
+}
diff --git a/libavcodec/audiodsp.h b/libavcodec/audiodsp.h
new file mode 100644
index 0000000..58205a1
--- /dev/null
+++ b/libavcodec/audiodsp.h
@@ -0,0 +1,59 @@
+/*
+ * 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_AUDIODSP_H
+#define AVCODEC_AUDIODSP_H
+
+#include <stdint.h>
+
+typedef struct AudioDSPContext {
+ /**
+ * Calculate scalar product of two vectors.
+ * @param len length of vectors, should be multiple of 16
+ */
+ int32_t (*scalarproduct_int16)(const int16_t *v1,
+ const int16_t *v2 /* align 16 */, int len);
+
+ /**
+ * Clip each element in an array of int32_t to a given minimum and
+ * maximum value.
+ * @param dst destination array
+ * constraints: 16-byte aligned
+ * @param src source array
+ * constraints: 16-byte aligned
+ * @param min minimum value
+ * constraints: must be in the range [-(1 << 24), 1 << 24]
+ * @param max maximum value
+ * constraints: must be in the range [-(1 << 24), 1 << 24]
+ * @param len number of elements in the array
+ * constraints: multiple of 32 greater than zero
+ */
+ void (*vector_clip_int32)(int32_t *dst, const int32_t *src, int32_t min,
+ int32_t max, unsigned int len);
+ /* assume len is a multiple of 8, and arrays are 16-byte aligned */
+ void (*vector_clipf)(float *dst /* align 16 */,
+ const float *src /* align 16 */,
+ float min, float max, int len /* align 16 */);
+} AudioDSPContext;
+
+void ff_audiodsp_init(AudioDSPContext *c);
+void ff_audiodsp_init_arm(AudioDSPContext *c);
+void ff_audiodsp_init_ppc(AudioDSPContext *c);
+void ff_audiodsp_init_x86(AudioDSPContext *c);
+
+#endif /* AVCODEC_AUDIODSP_H */
diff --git a/libavcodec/cook.c b/libavcodec/cook.c
index 190d28c..2d77899 100644
--- a/libavcodec/cook.c
+++ b/libavcodec/cook.c
@@ -44,9 +44,10 @@
#include "libavutil/channel_layout.h"
#include "libavutil/lfg.h"
+
+#include "audiodsp.h"
#include "avcodec.h"
#include "get_bits.h"
-#include "dsputil.h"
#include "bytestream.h"
#include "fft.h"
#include "internal.h"
@@ -122,7 +123,7 @@ typedef struct cook {
void (*saturate_output)(struct cook *q, float *out);
AVCodecContext* avctx;
- DSPContext dsp;
+ AudioDSPContext adsp;
GetBitContext gb;
/* stream data */
int num_vectors;
@@ -865,8 +866,8 @@ static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
*/
static void saturate_output_float(COOKContext *q, float *out)
{
- q->dsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
- -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
+ q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
+ -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
}
@@ -1065,7 +1066,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
/* Initialize RNG. */
av_lfg_init(&q->random_state, 0);
- ff_dsputil_init(&q->dsp, avctx);
+ ff_audiodsp_init(&q->adsp);
while (edata_ptr < edata_ptr_end) {
/* 8 for mono, 16 for stereo, ? for multichannel
diff --git a/libavcodec/dsputil.c b/libavcodec/dsputil.c
index 8f5ddd0..27e58a5 100644
--- a/libavcodec/dsputil.c
+++ b/libavcodec/dsputil.c
@@ -1267,87 +1267,6 @@ WRAPPER8_16_SQ(quant_psnr8x8_c, quant_psnr16_c)
WRAPPER8_16_SQ(rd8x8_c, rd16_c)
WRAPPER8_16_SQ(bit8x8_c, bit16_c)
-static inline uint32_t clipf_c_one(uint32_t a, uint32_t mini,
- uint32_t maxi, uint32_t maxisign)
-{
- if (a > mini)
- return mini;
- else if ((a ^ (1U << 31)) > maxisign)
- return maxi;
- else
- return a;
-}
-
-static void vector_clipf_c_opposite_sign(float *dst, const float *src,
- float *min, float *max, int len)
-{
- int i;
- uint32_t mini = *(uint32_t *) min;
- uint32_t maxi = *(uint32_t *) max;
- uint32_t maxisign = maxi ^ (1U << 31);
- uint32_t *dsti = (uint32_t *) dst;
- const uint32_t *srci = (const uint32_t *) src;
-
- for (i = 0; i < len; i += 8) {
- dsti[i + 0] = clipf_c_one(srci[i + 0], mini, maxi, maxisign);
- dsti[i + 1] = clipf_c_one(srci[i + 1], mini, maxi, maxisign);
- dsti[i + 2] = clipf_c_one(srci[i + 2], mini, maxi, maxisign);
- dsti[i + 3] = clipf_c_one(srci[i + 3], mini, maxi, maxisign);
- dsti[i + 4] = clipf_c_one(srci[i + 4], mini, maxi, maxisign);
- dsti[i + 5] = clipf_c_one(srci[i + 5], mini, maxi, maxisign);
- dsti[i + 6] = clipf_c_one(srci[i + 6], mini, maxi, maxisign);
- dsti[i + 7] = clipf_c_one(srci[i + 7], mini, maxi, maxisign);
- }
-}
-
-static void vector_clipf_c(float *dst, const float *src,
- float min, float max, int len)
-{
- int i;
-
- if (min < 0 && max > 0) {
- vector_clipf_c_opposite_sign(dst, src, &min, &max, len);
- } else {
- for (i = 0; i < len; i += 8) {
- dst[i] = av_clipf(src[i], min, max);
- dst[i + 1] = av_clipf(src[i + 1], min, max);
- dst[i + 2] = av_clipf(src[i + 2], min, max);
- dst[i + 3] = av_clipf(src[i + 3], min, max);
- dst[i + 4] = av_clipf(src[i + 4], min, max);
- dst[i + 5] = av_clipf(src[i + 5], min, max);
- dst[i + 6] = av_clipf(src[i + 6], min, max);
- dst[i + 7] = av_clipf(src[i + 7], min, max);
- }
- }
-}
-
-static int32_t scalarproduct_int16_c(const int16_t *v1, const int16_t *v2,
- int order)
-{
- int res = 0;
-
- while (order--)
- res += *v1++ **v2++;
-
- return res;
-}
-
-static void vector_clip_int32_c(int32_t *dst, const int32_t *src, int32_t min,
- int32_t max, unsigned int len)
-{
- do {
- *dst++ = av_clip(*src++, min, max);
- *dst++ = av_clip(*src++, min, max);
- *dst++ = av_clip(*src++, min, max);
- *dst++ = av_clip(*src++, min, max);
- *dst++ = av_clip(*src++, min, max);
- *dst++ = av_clip(*src++, min, max);
- *dst++ = av_clip(*src++, min, max);
- *dst++ = av_clip(*src++, min, max);
- len -= 8;
- } while (len > 0);
-}
-
static void jref_idct_put(uint8_t *dest, int line_size, int16_t *block)
{
ff_j_rev_dct(block);
@@ -1502,10 +1421,6 @@ av_cold void ff_dsputil_init(DSPContext *c, AVCodecContext *avctx)
c->try_8x8basis = try_8x8basis_c;
c->add_8x8basis = add_8x8basis_c;
- c->scalarproduct_int16 = scalarproduct_int16_c;
- c->vector_clip_int32 = vector_clip_int32_c;
- c->vector_clipf = vector_clipf_c;
-
c->shrink[0] = av_image_copy_plane;
c->shrink[1] = ff_shrink22;
c->shrink[2] = ff_shrink44;
diff --git a/libavcodec/dsputil.h b/libavcodec/dsputil.h
index 1aad789..24a6f12 100644
--- a/libavcodec/dsputil.h
+++ b/libavcodec/dsputil.h
@@ -125,11 +125,6 @@ typedef struct DSPContext {
void (*bswap_buf)(uint32_t *dst, const uint32_t *src, int w);
void (*bswap16_buf)(uint16_t *dst, const uint16_t *src, int len);
- /* assume len is a multiple of 8, and arrays are 16-byte aligned */
- void (*vector_clipf)(float *dst /* align 16 */,
- const float *src /* align 16 */,
- float min, float max, int len /* align 16 */);
-
/* (I)DCT */
void (*fdct)(int16_t *block /* align 16 */);
void (*fdct248)(int16_t *block /* align 16 */);
@@ -189,30 +184,6 @@ typedef struct DSPContext {
void (*shrink[4])(uint8_t *dst, int dst_wrap, const uint8_t *src,
int src_wrap, int width, int height);
-
- /**
- * Calculate scalar product of two vectors.
- * @param len length of vectors, should be multiple of 16
- */
- int32_t (*scalarproduct_int16)(const int16_t *v1,
- const int16_t *v2 /* align 16 */, int len);
-
- /**
- * Clip each element in an array of int32_t to a given minimum and
- * maximum value.
- * @param dst destination array
- * constraints: 16-byte aligned
- * @param src source array
- * constraints: 16-byte aligned
- * @param min minimum value
- * constraints: must be in the range [-(1 << 24), 1 << 24]
- * @param max maximum value
- * constraints: must be in the range [-(1 << 24), 1 << 24]
- * @param len number of elements in the array
- * constraints: multiple of 32 greater than zero
- */
- void (*vector_clip_int32)(int32_t *dst, const int32_t *src, int32_t min,
- int32_t max, unsigned int len);
} DSPContext;
void ff_dsputil_static_init(void);
diff --git a/libavcodec/ppc/Makefile b/libavcodec/ppc/Makefile
index bd78f8e..8a4a789 100644
--- a/libavcodec/ppc/Makefile
+++ b/libavcodec/ppc/Makefile
@@ -1,5 +1,6 @@
OBJS += ppc/fmtconvert_altivec.o \
+OBJS-$(CONFIG_AUDIODSP) += ppc/audiodsp.o
OBJS-$(CONFIG_BLOCKDSP) += ppc/blockdsp.o
OBJS-$(CONFIG_DSPUTIL) += ppc/dsputil_ppc.o
OBJS-$(CONFIG_FFT) += ppc/fft_altivec.o
@@ -24,7 +25,6 @@ ALTIVEC-OBJS-$(CONFIG_DSPUTIL) += ppc/dsputil_altivec.o \
ppc/fdct_altivec.o \
ppc/gmc_altivec.o \
ppc/idct_altivec.o \
- ppc/int_altivec.o \
FFT-OBJS-$(HAVE_GNU_AS) += ppc/fft_altivec_s.o
ALTIVEC-OBJS-$(CONFIG_FFT) += $(FFT-OBJS-yes)
diff --git a/libavcodec/ppc/int_altivec.c b/libavcodec/ppc/audiodsp.c
similarity index 83%
rename from libavcodec/ppc/int_altivec.c
rename to libavcodec/ppc/audiodsp.c
index d76d34a..36506ce 100644
--- a/libavcodec/ppc/int_altivec.c
+++ b/libavcodec/ppc/audiodsp.c
@@ -20,7 +20,7 @@
/**
* @file
- * miscellaneous integer operations
+ * miscellaneous audio operations
*/
#include "config.h"
@@ -29,10 +29,13 @@
#endif
#include "libavutil/attributes.h"
+#include "libavutil/cpu.h"
+#include "libavutil/ppc/cpu.h"
#include "libavutil/ppc/types_altivec.h"
#include "libavutil/ppc/util_altivec.h"
-#include "libavcodec/dsputil.h"
-#include "dsputil_altivec.h"
+#include "libavcodec/audiodsp.h"
+
+#if HAVE_ALTIVEC
static int32_t scalarproduct_int16_altivec(const int16_t *v1, const int16_t *v2,
int order)
@@ -56,7 +59,14 @@ static int32_t scalarproduct_int16_altivec(const int16_t *v1, const int16_t *v2,
return ires;
}
-av_cold void ff_int_init_altivec(DSPContext *c, AVCodecContext *avctx)
+#endif /* HAVE_ALTIVEC */
+
+av_cold void ff_audiodsp_init_ppc(AudioDSPContext *c)
{
+#if HAVE_ALTIVEC
+ if (!PPC_ALTIVEC(av_get_cpu_flags()))
+ return;
+
c->scalarproduct_int16 = scalarproduct_int16_altivec;
+#endif /* HAVE_ALTIVEC */
}
diff --git a/libavcodec/ppc/dsputil_altivec.h b/libavcodec/ppc/dsputil_altivec.h
index 7833b4b..2ad4910 100644
--- a/libavcodec/ppc/dsputil_altivec.h
+++ b/libavcodec/ppc/dsputil_altivec.h
@@ -35,6 +35,5 @@ void ff_idct_add_altivec(uint8_t *dest, int line_size, int16_t *block);
void ff_dsputil_init_altivec(DSPContext *c, AVCodecContext *avctx,
unsigned high_bit_depth);
-void ff_int_init_altivec(DSPContext *c, AVCodecContext *avctx);
#endif /* AVCODEC_PPC_DSPUTIL_ALTIVEC_H */
diff --git a/libavcodec/ppc/dsputil_ppc.c b/libavcodec/ppc/dsputil_ppc.c
index b92fbf0..fb1ee4a 100644
--- a/libavcodec/ppc/dsputil_ppc.c
+++ b/libavcodec/ppc/dsputil_ppc.c
@@ -34,7 +34,7 @@ av_cold void ff_dsputil_init_ppc(DSPContext *c, AVCodecContext *avctx,
{
if (PPC_ALTIVEC(av_get_cpu_flags())) {
ff_dsputil_init_altivec(c, avctx, high_bit_depth);
- ff_int_init_altivec(c, avctx);
+
c->gmc1 = ff_gmc1_altivec;
if (!high_bit_depth) {
diff --git a/libavcodec/takdec.c b/libavcodec/takdec.c
index 0d2dcbb..b0e84ea 100644
--- a/libavcodec/takdec.c
+++ b/libavcodec/takdec.c
@@ -28,8 +28,8 @@
#include "libavutil/internal.h"
#include "libavutil/samplefmt.h"
#include "tak.h"
+#include "audiodsp.h"
#include "avcodec.h"
-#include "dsputil.h"
#include "internal.h"
#include "unary.h"
@@ -45,7 +45,7 @@ typedef struct MCDParam {
typedef struct TAKDecContext {
AVCodecContext *avctx; // parent AVCodecContext
- DSPContext dsp;
+ AudioDSPContext adsp;
TAKStreamInfo ti;
GetBitContext gb; // bitstream reader initialized to start at the current frame
@@ -172,7 +172,7 @@ static av_cold int tak_decode_init(AVCodecContext *avctx)
{
TAKDecContext *s = avctx->priv_data;
- ff_dsputil_init(&s->dsp, avctx);
+ ff_audiodsp_init(&s->adsp);
s->avctx = avctx;
@@ -484,8 +484,8 @@ static int decode_subframe(TAKDecContext *s, int32_t *decoded,
for (i = 0; i < subframe_size - filter_order; i++) {
int v = 1 << (filter_quant - 1);
- v += s->dsp.scalarproduct_int16(&s->residues[i], filter,
- FFALIGN(filter_order, 16));
+ v += s->adsp.scalarproduct_int16(&s->residues[i], filter,
+ FFALIGN(filter_order, 16));
v = (av_clip(v >> filter_quant, -8192, 8191) << dshift) - *decoded;
*decoded++ = v;
@@ -654,8 +654,8 @@ static int decorrelate(TAKDecContext *s, int c1, int c2, int length)
for (i = 0; i < length2; i++) {
int v = 1 << 9;
- v += s->dsp.scalarproduct_int16(&s->residues[i], filter,
- FFALIGN(filter_order, 16));
+ v += s->adsp.scalarproduct_int16(&s->residues[i], filter,
+ FFALIGN(filter_order, 16));
p1[i] = (av_clip(v >> 10, -8192, 8191) << dshift) - p1[i];
}
diff --git a/libavcodec/x86/Makefile b/libavcodec/x86/Makefile
index 222a0ff..483c850 100644
--- a/libavcodec/x86/Makefile
+++ b/libavcodec/x86/Makefile
@@ -2,6 +2,7 @@ OBJS += x86/constants.o \
x86/fmtconvert_init.o \
OBJS-$(CONFIG_AC3DSP) += x86/ac3dsp_init.o
+OBJS-$(CONFIG_AUDIODSP) += x86/audiodsp_init.o
OBJS-$(CONFIG_DCT) += x86/dct_init.o
OBJS-$(CONFIG_DSPUTIL) += x86/dsputil_init.o
OBJS-$(CONFIG_ENCODERS) += x86/dsputilenc_mmx.o \
@@ -44,6 +45,7 @@ OBJS-$(CONFIG_VP7_DECODER) += x86/vp8dsp_init.o
OBJS-$(CONFIG_VP8_DECODER) += x86/vp8dsp_init.o
OBJS-$(CONFIG_VP9_DECODER) += x86/vp9dsp_init.o
+MMX-OBJS-$(CONFIG_AUDIODSP) += x86/audiodsp_mmx.o
MMX-OBJS-$(CONFIG_BLOCKDSP) += x86/blockdsp_mmx.o
MMX-OBJS-$(CONFIG_DSPUTIL) += x86/dsputil_mmx.o \
x86/idct_mmx_xvid.o \
@@ -61,6 +63,7 @@ YASM-OBJS += x86/deinterlace.o \
x86/fmtconvert.o \
YASM-OBJS-$(CONFIG_AC3DSP) += x86/ac3dsp.o
+YASM-OBJS-$(CONFIG_AUDIODSP) += x86/audiodsp.o
YASM-OBJS-$(CONFIG_DCT) += x86/dct32.o
YASM-OBJS-$(CONFIG_DNXHD_ENCODER) += x86/dnxhdenc.o
YASM-OBJS-$(CONFIG_DSPUTIL) += x86/dsputil.o
diff --git a/libavcodec/x86/audiodsp.asm b/libavcodec/x86/audiodsp.asm
new file mode 100644
index 0000000..f2e831d
--- /dev/null
+++ b/libavcodec/x86/audiodsp.asm
@@ -0,0 +1,137 @@
+;******************************************************************************
+;* optimized audio functions
+;* Copyright (c) 2008 Loren Merritt
+;*
+;* 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 "libavutil/x86/x86util.asm"
+
+SECTION_TEXT
+
+%macro SCALARPRODUCT 0
+; int ff_scalarproduct_int16(int16_t *v1, int16_t *v2, int order)
+cglobal scalarproduct_int16, 3,3,3, v1, v2, order
+ shl orderq, 1
+ add v1q, orderq
+ add v2q, orderq
+ neg orderq
+ pxor m2, m2
+.loop:
+ movu m0, [v1q + orderq]
+ movu m1, [v1q + orderq + mmsize]
+ pmaddwd m0, [v2q + orderq]
+ pmaddwd m1, [v2q + orderq + mmsize]
+ paddd m2, m0
+ paddd m2, m1
+ add orderq, mmsize*2
+ jl .loop
+%if mmsize == 16
+ movhlps m0, m2
+ paddd m2, m0
+ pshuflw m0, m2, 0x4e
+%else
+ pshufw m0, m2, 0x4e
+%endif
+ paddd m2, m0
+ movd eax, m2
+ RET
+%endmacro
+
+INIT_MMX mmxext
+SCALARPRODUCT
+INIT_XMM sse2
+SCALARPRODUCT
+
+
+;-----------------------------------------------------------------------------
+; void ff_vector_clip_int32(int32_t *dst, const int32_t *src, int32_t min,
+; int32_t max, unsigned int len)
+;-----------------------------------------------------------------------------
+
+; %1 = number of xmm registers used
+; %2 = number of inline load/process/store loops per asm loop
+; %3 = process 4*mmsize (%3=0) or 8*mmsize (%3=1) bytes per loop
+; %4 = CLIPD function takes min/max as float instead of int (CLIPD_SSE2)
+; %5 = suffix
+%macro VECTOR_CLIP_INT32 4-5
+cglobal vector_clip_int32%5, 5,5,%1, dst, src, min, max, len
+%if %4
+ cvtsi2ss m4, minm
+ cvtsi2ss m5, maxm
+%else
+ movd m4, minm
+ movd m5, maxm
+%endif
+ SPLATD m4
+ SPLATD m5
+.loop:
+%assign %%i 1
+%rep %2
+ mova m0, [srcq+mmsize*0*%%i]
+ mova m1, [srcq+mmsize*1*%%i]
+ mova m2, [srcq+mmsize*2*%%i]
+ mova m3, [srcq+mmsize*3*%%i]
+%if %3
+ mova m7, [srcq+mmsize*4*%%i]
+ mova m8, [srcq+mmsize*5*%%i]
+ mova m9, [srcq+mmsize*6*%%i]
+ mova m10, [srcq+mmsize*7*%%i]
+%endif
+ CLIPD m0, m4, m5, m6
+ CLIPD m1, m4, m5, m6
+ CLIPD m2, m4, m5, m6
+ CLIPD m3, m4, m5, m6
+%if %3
+ CLIPD m7, m4, m5, m6
+ CLIPD m8, m4, m5, m6
+ CLIPD m9, m4, m5, m6
+ CLIPD m10, m4, m5, m6
+%endif
+ mova [dstq+mmsize*0*%%i], m0
+ mova [dstq+mmsize*1*%%i], m1
+ mova [dstq+mmsize*2*%%i], m2
+ mova [dstq+mmsize*3*%%i], m3
+%if %3
+ mova [dstq+mmsize*4*%%i], m7
+ mova [dstq+mmsize*5*%%i], m8
+ mova [dstq+mmsize*6*%%i], m9
+ mova [dstq+mmsize*7*%%i], m10
+%endif
+%assign %%i %%i+1
+%endrep
+ add srcq, mmsize*4*(%2+%3)
+ add dstq, mmsize*4*(%2+%3)
+ sub lend, mmsize*(%2+%3)
+ jg .loop
+ REP_RET
+%endmacro
+
+INIT_MMX mmx
+%define CLIPD CLIPD_MMX
+VECTOR_CLIP_INT32 0, 1, 0, 0
+INIT_XMM sse2
+VECTOR_CLIP_INT32 6, 1, 0, 0, _int
+%define CLIPD CLIPD_SSE2
+VECTOR_CLIP_INT32 6, 2, 0, 1
+INIT_XMM sse4
+%define CLIPD CLIPD_SSE41
+%ifdef m8
+VECTOR_CLIP_INT32 11, 1, 1, 0
+%else
+VECTOR_CLIP_INT32 6, 1, 0, 0
+%endif
diff --git a/libavcodec/x86/audiodsp.h b/libavcodec/x86/audiodsp.h
new file mode 100644
index 0000000..321056b
--- /dev/null
+++ b/libavcodec/x86/audiodsp.h
@@ -0,0 +1,25 @@
+/*
+ * 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_X86_AUDIODSP_H
+#define AVCODEC_X86_AUDIODSP_H
+
+void ff_vector_clipf_sse(float *dst, const float *src,
+ float min, float max, int len);
+
+#endif /* AVCODEC_X86_AUDIODSP_H */
diff --git a/libavcodec/x86/audiodsp_init.c b/libavcodec/x86/audiodsp_init.c
new file mode 100644
index 0000000..743f5a3
--- /dev/null
+++ b/libavcodec/x86/audiodsp_init.c
@@ -0,0 +1,66 @@
+/*
+ * 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 "config.h"
+#include "libavutil/attributes.h"
+#include "libavutil/cpu.h"
+#include "libavutil/x86/asm.h"
+#include "libavutil/x86/cpu.h"
+#include "libavcodec/audiodsp.h"
+#include "audiodsp.h"
+
+int32_t ff_scalarproduct_int16_mmxext(const int16_t *v1, const int16_t *v2,
+ int order);
+int32_t ff_scalarproduct_int16_sse2(const int16_t *v1, const int16_t *v2,
+ int order);
+
+void ff_vector_clip_int32_mmx(int32_t *dst, const int32_t *src,
+ int32_t min, int32_t max, unsigned int len);
+void ff_vector_clip_int32_sse2(int32_t *dst, const int32_t *src,
+ int32_t min, int32_t max, unsigned int len);
+void ff_vector_clip_int32_int_sse2(int32_t *dst, const int32_t *src,
+ int32_t min, int32_t max, unsigned int len);
+void ff_vector_clip_int32_sse4(int32_t *dst, const int32_t *src,
+ int32_t min, int32_t max, unsigned int len);
+
+av_cold void ff_audiodsp_init_x86(AudioDSPContext *c)
+{
+ int cpu_flags = av_get_cpu_flags();
+
+ if (EXTERNAL_MMX(cpu_flags))
+ c->vector_clip_int32 = ff_vector_clip_int32_mmx;
+
+ if (EXTERNAL_MMXEXT(cpu_flags))
+ c->scalarproduct_int16 = ff_scalarproduct_int16_mmxext;
+
+ if (INLINE_SSE(cpu_flags))
+ c->vector_clipf = ff_vector_clipf_sse;
+
+ if (EXTERNAL_SSE2(cpu_flags)) {
+ c->scalarproduct_int16 = ff_scalarproduct_int16_sse2;
+ if (cpu_flags & AV_CPU_FLAG_ATOM)
+ c->vector_clip_int32 = ff_vector_clip_int32_int_sse2;
+ else
+ c->vector_clip_int32 = ff_vector_clip_int32_sse2;
+ }
+
+ if (EXTERNAL_SSE4(cpu_flags))
+ c->vector_clip_int32 = ff_vector_clip_int32_sse4;
+}
diff --git a/libavcodec/x86/audiodsp_mmx.c b/libavcodec/x86/audiodsp_mmx.c
new file mode 100644
index 0000000..cb55059
--- /dev/null
+++ b/libavcodec/x86/audiodsp_mmx.c
@@ -0,0 +1,58 @@
+/*
+ * 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 "config.h"
+#include "libavutil/x86/asm.h"
+#include "audiodsp.h"
+
+#if HAVE_INLINE_ASM
+
+void ff_vector_clipf_sse(float *dst, const float *src,
+ float min, float max, int len)
+{
+ x86_reg i = (len - 16) * 4;
+ __asm__ volatile (
+ "movss %3, %%xmm4 \n\t"
+ "movss %4, %%xmm5 \n\t"
+ "shufps $0, %%xmm4, %%xmm4 \n\t"
+ "shufps $0, %%xmm5, %%xmm5 \n\t"
+ "1: \n\t"
+ "movaps (%2, %0), %%xmm0 \n\t" // 3/1 on intel
+ "movaps 16(%2, %0), %%xmm1 \n\t"
+ "movaps 32(%2, %0), %%xmm2 \n\t"
+ "movaps 48(%2, %0), %%xmm3 \n\t"
+ "maxps %%xmm4, %%xmm0 \n\t"
+ "maxps %%xmm4, %%xmm1 \n\t"
+ "maxps %%xmm4, %%xmm2 \n\t"
+ "maxps %%xmm4, %%xmm3 \n\t"
+ "minps %%xmm5, %%xmm0 \n\t"
+ "minps %%xmm5, %%xmm1 \n\t"
+ "minps %%xmm5, %%xmm2 \n\t"
+ "minps %%xmm5, %%xmm3 \n\t"
+ "movaps %%xmm0, (%1, %0) \n\t"
+ "movaps %%xmm1, 16(%1, %0) \n\t"
+ "movaps %%xmm2, 32(%1, %0) \n\t"
+ "movaps %%xmm3, 48(%1, %0) \n\t"
+ "sub $64, %0 \n\t"
+ "jge 1b \n\t"
+ : "+&r" (i)
+ : "r" (dst), "r" (src), "m" (min), "m" (max)
+ : "memory");
+}
+
+#endif /* HAVE_INLINE_ASM */
diff --git a/libavcodec/x86/dsputil.asm b/libavcodec/x86/dsputil.asm
index b5d6d3c..8f5a14d 100644
--- a/libavcodec/x86/dsputil.asm
+++ b/libavcodec/x86/dsputil.asm
@@ -26,119 +26,6 @@ pb_bswap32: db 3, 2, 1, 0, 7, 6, 5, 4, 11, 10, 9, 8, 15, 14, 13, 12
SECTION_TEXT
-%macro SCALARPRODUCT 0
-; int ff_scalarproduct_int16(int16_t *v1, int16_t *v2, int order)
-cglobal scalarproduct_int16, 3,3,3, v1, v2, order
- shl orderq, 1
- add v1q, orderq
- add v2q, orderq
- neg orderq
- pxor m2, m2
-.loop:
- movu m0, [v1q + orderq]
- movu m1, [v1q + orderq + mmsize]
- pmaddwd m0, [v2q + orderq]
- pmaddwd m1, [v2q + orderq + mmsize]
- paddd m2, m0
- paddd m2, m1
- add orderq, mmsize*2
- jl .loop
-%if mmsize == 16
- movhlps m0, m2
- paddd m2, m0
- pshuflw m0, m2, 0x4e
-%else
- pshufw m0, m2, 0x4e
-%endif
- paddd m2, m0
- movd eax, m2
- RET
-%endmacro
-
-INIT_MMX mmxext
-SCALARPRODUCT
-INIT_XMM sse2
-SCALARPRODUCT
-
-
-;-----------------------------------------------------------------------------
-; void ff_vector_clip_int32(int32_t *dst, const int32_t *src, int32_t min,
-; int32_t max, unsigned int len)
-;-----------------------------------------------------------------------------
-
-; %1 = number of xmm registers used
-; %2 = number of inline load/process/store loops per asm loop
-; %3 = process 4*mmsize (%3=0) or 8*mmsize (%3=1) bytes per loop
-; %4 = CLIPD function takes min/max as float instead of int (CLIPD_SSE2)
-; %5 = suffix
-%macro VECTOR_CLIP_INT32 4-5
-cglobal vector_clip_int32%5, 5,5,%1, dst, src, min, max, len
-%if %4
- cvtsi2ss m4, minm
- cvtsi2ss m5, maxm
-%else
- movd m4, minm
- movd m5, maxm
-%endif
- SPLATD m4
- SPLATD m5
-.loop:
-%assign %%i 1
-%rep %2
- mova m0, [srcq+mmsize*0*%%i]
- mova m1, [srcq+mmsize*1*%%i]
- mova m2, [srcq+mmsize*2*%%i]
- mova m3, [srcq+mmsize*3*%%i]
-%if %3
- mova m7, [srcq+mmsize*4*%%i]
- mova m8, [srcq+mmsize*5*%%i]
- mova m9, [srcq+mmsize*6*%%i]
- mova m10, [srcq+mmsize*7*%%i]
-%endif
- CLIPD m0, m4, m5, m6
- CLIPD m1, m4, m5, m6
- CLIPD m2, m4, m5, m6
- CLIPD m3, m4, m5, m6
-%if %3
- CLIPD m7, m4, m5, m6
- CLIPD m8, m4, m5, m6
- CLIPD m9, m4, m5, m6
- CLIPD m10, m4, m5, m6
-%endif
- mova [dstq+mmsize*0*%%i], m0
- mova [dstq+mmsize*1*%%i], m1
- mova [dstq+mmsize*2*%%i], m2
- mova [dstq+mmsize*3*%%i], m3
-%if %3
- mova [dstq+mmsize*4*%%i], m7
- mova [dstq+mmsize*5*%%i], m8
- mova [dstq+mmsize*6*%%i], m9
- mova [dstq+mmsize*7*%%i], m10
-%endif
-%assign %%i %%i+1
-%endrep
- add srcq, mmsize*4*(%2+%3)
- add dstq, mmsize*4*(%2+%3)
- sub lend, mmsize*(%2+%3)
- jg .loop
- REP_RET
-%endmacro
-
-INIT_MMX mmx
-%define CLIPD CLIPD_MMX
-VECTOR_CLIP_INT32 0, 1, 0, 0
-INIT_XMM sse2
-VECTOR_CLIP_INT32 6, 1, 0, 0, _int
-%define CLIPD CLIPD_SSE2
-VECTOR_CLIP_INT32 6, 2, 0, 1
-INIT_XMM sse4
-%define CLIPD CLIPD_SSE41
-%ifdef m8
-VECTOR_CLIP_INT32 11, 1, 1, 0
-%else
-VECTOR_CLIP_INT32 6, 1, 0, 0
-%endif
-
; %1 = aligned/unaligned
%macro BSWAP_LOOPS 1
mov r3, r2
diff --git a/libavcodec/x86/dsputil_init.c b/libavcodec/x86/dsputil_init.c
index a19b83d..646435d 100644
--- a/libavcodec/x86/dsputil_init.c
+++ b/libavcodec/x86/dsputil_init.c
@@ -26,23 +26,9 @@
#include "dsputil_x86.h"
#include "idct_xvid.h"
-int32_t ff_scalarproduct_int16_mmxext(const int16_t *v1, const int16_t *v2,
- int order);
-int32_t ff_scalarproduct_int16_sse2(const int16_t *v1, const int16_t *v2,
- int order);
-
void ff_bswap32_buf_ssse3(uint32_t *dst, const uint32_t *src, int w);
void ff_bswap32_buf_sse2(uint32_t *dst, const uint32_t *src, int w);
-void ff_vector_clip_int32_mmx(int32_t *dst, const int32_t *src,
- int32_t min, int32_t max, unsigned int len);
-void ff_vector_clip_int32_sse2(int32_t *dst, const int32_t *src,
- int32_t min, int32_t max, unsigned int len);
-void ff_vector_clip_int32_int_sse2(int32_t *dst, const int32_t *src,
- int32_t min, int32_t max, unsigned int len);
-void ff_vector_clip_int32_sse4(int32_t *dst, const int32_t *src,
- int32_t min, int32_t max, unsigned int len);
-
static av_cold void dsputil_init_mmx(DSPContext *c, AVCodecContext *avctx,
int cpu_flags, unsigned high_bit_depth)
{
@@ -72,10 +58,6 @@ static av_cold void dsputil_init_mmx(DSPContext *c, AVCodecContext *avctx,
c->gmc = ff_gmc_mmx;
#endif /* HAVE_MMX_INLINE */
-
-#if HAVE_MMX_EXTERNAL
- c->vector_clip_int32 = ff_vector_clip_int32_mmx;
-#endif /* HAVE_MMX_EXTERNAL */
}
static av_cold void dsputil_init_mmxext(DSPContext *c, AVCodecContext *avctx,
@@ -88,18 +70,6 @@ static av_cold void dsputil_init_mmxext(DSPContext *c, AVCodecContext *avctx,
c->idct = ff_idct_xvid_mmxext;
}
#endif /* HAVE_MMXEXT_INLINE */
-
-#if HAVE_MMXEXT_EXTERNAL
- c->scalarproduct_int16 = ff_scalarproduct_int16_mmxext;
-#endif /* HAVE_MMXEXT_EXTERNAL */
-}
-
-static av_cold void dsputil_init_sse(DSPContext *c, AVCodecContext *avctx,
- int cpu_flags, unsigned high_bit_depth)
-{
-#if HAVE_SSE_INLINE
- c->vector_clipf = ff_vector_clipf_sse;
-#endif /* HAVE_SSE_INLINE */
}
static av_cold void dsputil_init_sse2(DSPContext *c, AVCodecContext *avctx,
@@ -115,12 +85,6 @@ static av_cold void dsputil_init_sse2(DSPContext *c, AVCodecContext *avctx,
#endif /* HAVE_SSE2_INLINE */
#if HAVE_SSE2_EXTERNAL
- c->scalarproduct_int16 = ff_scalarproduct_int16_sse2;
- if (cpu_flags & AV_CPU_FLAG_ATOM) {
- c->vector_clip_int32 = ff_vector_clip_int32_int_sse2;
- } else {
- c->vector_clip_int32 = ff_vector_clip_int32_sse2;
- }
c->bswap_buf = ff_bswap32_buf_sse2;
#endif /* HAVE_SSE2_EXTERNAL */
}
@@ -133,14 +97,6 @@ static av_cold void dsputil_init_ssse3(DSPContext *c, AVCodecContext *avctx,
#endif /* HAVE_SSSE3_EXTERNAL */
}
-static av_cold void dsputil_init_sse4(DSPContext *c, AVCodecContext *avctx,
- int cpu_flags, unsigned high_bit_depth)
-{
-#if HAVE_SSE4_EXTERNAL
- c->vector_clip_int32 = ff_vector_clip_int32_sse4;
-#endif /* HAVE_SSE4_EXTERNAL */
-}
-
av_cold void ff_dsputil_init_x86(DSPContext *c, AVCodecContext *avctx,
unsigned high_bit_depth)
{
@@ -152,18 +108,12 @@ av_cold void ff_dsputil_init_x86(DSPContext *c, AVCodecContext *avctx,
if (X86_MMXEXT(cpu_flags))
dsputil_init_mmxext(c, avctx, cpu_flags, high_bit_depth);
- if (X86_SSE(cpu_flags))
- dsputil_init_sse(c, avctx, cpu_flags, high_bit_depth);
-
if (X86_SSE2(cpu_flags))
dsputil_init_sse2(c, avctx, cpu_flags, high_bit_depth);
if (EXTERNAL_SSSE3(cpu_flags))
dsputil_init_ssse3(c, avctx, cpu_flags, high_bit_depth);
- if (EXTERNAL_SSE4(cpu_flags))
- dsputil_init_sse4(c, avctx, cpu_flags, high_bit_depth);
-
if (CONFIG_ENCODERS)
ff_dsputilenc_init_mmx(c, avctx, high_bit_depth);
}
diff --git a/libavcodec/x86/dsputil_mmx.c b/libavcodec/x86/dsputil_mmx.c
index fd74efe..fe43804 100644
--- a/libavcodec/x86/dsputil_mmx.c
+++ b/libavcodec/x86/dsputil_mmx.c
@@ -25,7 +25,6 @@
#include "config.h"
#include "libavutil/cpu.h"
#include "libavutil/x86/asm.h"
-#include "constants.h"
#include "dsputil_x86.h"
#include "inline_asm.h"
@@ -375,37 +374,4 @@ void ff_gmc_mmx(uint8_t *dst, uint8_t *src,
}
}
-void ff_vector_clipf_sse(float *dst, const float *src,
- float min, float max, int len)
-{
- x86_reg i = (len - 16) * 4;
- __asm__ volatile (
- "movss %3, %%xmm4 \n\t"
- "movss %4, %%xmm5 \n\t"
- "shufps $0, %%xmm4, %%xmm4 \n\t"
- "shufps $0, %%xmm5, %%xmm5 \n\t"
- "1: \n\t"
- "movaps (%2, %0), %%xmm0 \n\t" // 3/1 on intel
- "movaps 16(%2, %0), %%xmm1 \n\t"
- "movaps 32(%2, %0), %%xmm2 \n\t"
- "movaps 48(%2, %0), %%xmm3 \n\t"
- "maxps %%xmm4, %%xmm0 \n\t"
- "maxps %%xmm4, %%xmm1 \n\t"
- "maxps %%xmm4, %%xmm2 \n\t"
- "maxps %%xmm4, %%xmm3 \n\t"
- "minps %%xmm5, %%xmm0 \n\t"
- "minps %%xmm5, %%xmm1 \n\t"
- "minps %%xmm5, %%xmm2 \n\t"
- "minps %%xmm5, %%xmm3 \n\t"
- "movaps %%xmm0, (%1, %0) \n\t"
- "movaps %%xmm1, 16(%1, %0) \n\t"
- "movaps %%xmm2, 32(%1, %0) \n\t"
- "movaps %%xmm3, 48(%1, %0) \n\t"
- "sub $64, %0 \n\t"
- "jge 1b \n\t"
- : "+&r" (i)
- : "r" (dst), "r" (src), "m" (min), "m" (max)
- : "memory");
-}
-
#endif /* HAVE_INLINE_ASM */
diff --git a/libavcodec/x86/dsputil_x86.h b/libavcodec/x86/dsputil_x86.h
index e99b6b7..eeb9ca6 100644
--- a/libavcodec/x86/dsputil_x86.h
+++ b/libavcodec/x86/dsputil_x86.h
@@ -46,7 +46,4 @@ void ff_gmc_mmx(uint8_t *dst, uint8_t *src,
int dxx, int dxy, int dyx, int dyy,
int shift, int r, int width, int height);
-void ff_vector_clipf_sse(float *dst, const float *src,
- float min, float max, int len);
-
#endif /* AVCODEC_X86_DSPUTIL_X86_H */
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