FFmpeg
aacpsdsp_init_arm.c
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1 /*
2  * Copyright (c) 2012 Mans Rullgard
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include "config.h"
22 
23 #include "libavutil/arm/cpu.h"
24 #include "libavutil/attributes.h"
25 #include "libavcodec/aacpsdsp.h"
26 
27 void ff_ps_add_squares_neon(float *dst, const float (*src)[2], int n);
28 void ff_ps_mul_pair_single_neon(float (*dst)[2], float (*src0)[2],
29  float *src1, int n);
30 void ff_ps_hybrid_analysis_neon(float (*out)[2], float (*in)[2],
31  const float (*filter)[8][2],
32  ptrdiff_t stride, int n);
33 void ff_ps_hybrid_analysis_ileave_neon(float (*out)[32][2], float L[2][38][64],
34  int i, int len);
35 void ff_ps_hybrid_synthesis_deint_neon(float out[2][38][64], float (*in)[32][2],
36  int i, int len);
37 void ff_ps_decorrelate_neon(float (*out)[2], float (*delay)[2],
38  float (*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2],
39  const float phi_fract[2], float (*Q_fract)[2],
40  const float *transient_gain, float g_decay_slope,
41  int len);
42 void ff_ps_stereo_interpolate_neon(float (*l)[2], float (*r)[2],
43  float h[2][4], float h_step[2][4],
44  int len);
45 
47 {
49 
50  if (have_neon(cpu_flags)) {
56  }
57 }
void(* mul_pair_single)(INTFLOAT(*dst)[2], INTFLOAT(*src0)[2], INTFLOAT *src1, int n)
Definition: aacpsdsp.h:34
av_cold void ff_psdsp_init_arm(PSDSPContext *s)
static atomic_int cpu_flags
Definition: cpu.c:50
#define src
Definition: vp8dsp.c:254
void(* hybrid_synthesis_deint)(INTFLOAT out[2][38][64], INTFLOAT(*in)[32][2], int i, int len)
Definition: aacpsdsp.h:41
Macro definitions for various function/variable attributes.
#define av_cold
Definition: attributes.h:82
#define PS_MAX_AP_DELAY
Definition: aacps.h:39
void(* add_squares)(INTFLOAT *dst, const INTFLOAT(*src)[2], int n)
Definition: aacpsdsp.h:33
void ff_ps_decorrelate_neon(float(*out)[2], float(*delay)[2], float(*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2], const float phi_fract[2], float(*Q_fract)[2], const float *transient_gain, float g_decay_slope, int len)
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
filter_frame For filters that do not use the this method is called when a frame is pushed to the filter s input It can be called at any time except in a reentrant way If the input frame is enough to produce then the filter should push the output frames on the output link immediately As an exception to the previous rule if the input frame is enough to produce several output frames then the filter needs output only at least one per link The additional frames can be left buffered in the filter
const char * r
Definition: vf_curves.c:114
void(* hybrid_analysis)(INTFLOAT(*out)[2], INTFLOAT(*in)[2], const INTFLOAT(*filter)[8][2], ptrdiff_t stride, int n)
Definition: aacpsdsp.h:36
static int phi_fract[2][50][2]
void(* stereo_interpolate[2])(INTFLOAT(*l)[2], INTFLOAT(*r)[2], INTFLOAT h[2][4], INTFLOAT h_step[2][4], int len)
Definition: aacpsdsp.h:49
#define have_neon(flags)
Definition: cpu.h:26
void ff_ps_hybrid_analysis_ileave_neon(float(*out)[32][2], float L[2][38][64], int i, int len)
#define s(width, name)
Definition: cbs_vp9.c:257
int n
Definition: avisynth_c.h:760
#define L(x)
Definition: vp56_arith.h:36
void ff_ps_mul_pair_single_neon(float(*dst)[2], float(*src0)[2], float *src1, int n)
void ff_ps_add_squares_neon(float *dst, const float(*src)[2], int n)
#define src1
Definition: h264pred.c:139
void ff_ps_stereo_interpolate_neon(float(*l)[2], float(*r)[2], float h[2][4], float h_step[2][4], int len)
void ff_ps_hybrid_synthesis_deint_neon(float out[2][38][64], float(*in)[32][2], int i, int len)
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);return NULL;}return ac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
#define src0
Definition: h264pred.c:138
int av_get_cpu_flags(void)
Return the flags which specify extensions supported by the CPU.
Definition: cpu.c:93
GLint GLenum GLboolean GLsizei stride
Definition: opengl_enc.c:104
#define PS_QMF_TIME_SLOTS
Definition: aacps.h:36
void ff_ps_hybrid_analysis_neon(float(*out)[2], float(*in)[2], const float(*filter)[8][2], ptrdiff_t stride, int n)
int len
FILE * out
Definition: movenc.c:54