FFmpeg
acelp_filters.h
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1 /*
2  * various filters for ACELP-based codecs
3  *
4  * Copyright (c) 2008 Vladimir Voroshilov
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 #ifndef AVCODEC_ACELP_FILTERS_H
24 #define AVCODEC_ACELP_FILTERS_H
25 
26 #include <stdint.h>
27 
28 typedef struct ACELPFContext {
29  /**
30  * Floating point version of ff_acelp_interpolate()
31  */
32  void (*acelp_interpolatef)(float *out, const float *in,
33  const float *filter_coeffs, int precision,
34  int frac_pos, int filter_length, int length);
35 
36  /**
37  * Apply an order 2 rational transfer function in-place.
38  *
39  * @param out output buffer for filtered speech samples
40  * @param in input buffer containing speech data (may be the same as out)
41  * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
42  * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
43  * @param gain scale factor for final output
44  * @param mem intermediate values used by filter (should be 0 initially)
45  * @param n number of samples (should be a multiple of eight)
46  */
48  const float zero_coeffs[2],
49  const float pole_coeffs[2],
50  float gain,
51  float mem[2], int n);
52 
54 
55 /**
56  * Initialize ACELPFContext.
57  */
60 
61 /**
62  * low-pass Finite Impulse Response filter coefficients.
63  *
64  * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
65  * the coefficients are scaled by 2^15.
66  * This array only contains the right half of the filter.
67  * This filter is likely identical to the one used in G.729, though this
68  * could not be determined from the original comments with certainty.
69  */
70 extern const int16_t ff_acelp_interp_filter[61];
71 
72 /**
73  * Generic FIR interpolation routine.
74  * @param[out] out buffer for interpolated data
75  * @param in input data
76  * @param filter_coeffs interpolation filter coefficients (0.15)
77  * @param precision sub sample factor, that is the precision of the position
78  * @param frac_pos fractional part of position [0..precision-1]
79  * @param filter_length filter length
80  * @param length length of output
81  *
82  * filter_coeffs contains coefficients of the right half of the symmetric
83  * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
84  * See ff_acelp_interp_filter for an example.
85  */
86 void ff_acelp_interpolate(int16_t* out, const int16_t* in,
87  const int16_t* filter_coeffs, int precision,
88  int frac_pos, int filter_length, int length);
89 
90 /**
91  * Floating point version of ff_acelp_interpolate()
92  */
93 void ff_acelp_interpolatef(float *out, const float *in,
94  const float *filter_coeffs, int precision,
95  int frac_pos, int filter_length, int length);
96 
97 
98 /**
99  * high-pass filtering and upscaling (4.2.5 of G.729).
100  * @param[out] out output buffer for filtered speech data
101  * @param[in,out] hpf_f past filtered data from previous (2 items long)
102  * frames (-0x20000000 <= (14.13) < 0x20000000)
103  * @param in speech data to process
104  * @param length input data size
105  *
106  * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
107  * 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
108  *
109  * The filter has a cut-off frequency of 1/80 of the sampling freq
110  *
111  * @note Two items before the top of the in buffer must contain two items from the
112  * tail of the previous subframe.
113  *
114  * @remark It is safe to pass the same array in in and out parameters.
115  *
116  * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
117  * but constants differs in 5th sign after comma). Fortunately in
118  * fixed-point all coefficients are the same as in G.729. Thus this
119  * routine can be used for the fixed-point AMR decoder, too.
120  */
121 void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
122  const int16_t* in, int length);
123 
124 /**
125  * Apply an order 2 rational transfer function in-place.
126  *
127  * @param out output buffer for filtered speech samples
128  * @param in input buffer containing speech data (may be the same as out)
129  * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
130  * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
131  * @param gain scale factor for final output
132  * @param mem intermediate values used by filter (should be 0 initially)
133  * @param n number of samples
134  */
135 void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,
136  const float zero_coeffs[2],
137  const float pole_coeffs[2],
138  float gain,
139  float mem[2], int n);
140 
141 /**
142  * Apply tilt compensation filter, 1 - tilt * z-1.
143  *
144  * @param mem pointer to the filter's state (one single float)
145  * @param tilt tilt factor
146  * @param samples array where the filter is applied
147  * @param size the size of the samples array
148  */
149 void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);
150 
151 
152 #endif /* AVCODEC_ACELP_FILTERS_H */
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
Apply tilt compensation filter, 1 - tilt * z-1.
void ff_acelp_filter_init_mips(ACELPFContext *c)
void ff_acelp_apply_order_2_transfer_function(float *out, const float *in, const float zero_coeffs[2], const float pole_coeffs[2], float gain, float mem[2], int n)
Apply an order 2 rational transfer function in-place.
const int16_t ff_acelp_interp_filter[61]
low-pass Finite Impulse Response filter coefficients.
Definition: acelp_filters.c:30
void(* acelp_interpolatef)(float *out, const float *in, const float *filter_coeffs, int precision, int frac_pos, int filter_length, int length)
Floating point version of ff_acelp_interpolate()
Definition: acelp_filters.h:32
void ff_acelp_high_pass_filter(int16_t *out, int hpf_f[2], const int16_t *in, int length)
high-pass filtering and upscaling (4.2.5 of G.729).
Definition: acelp_filters.c:99
int mem
Definition: avisynth_c.h:916
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
ptrdiff_t size
Definition: opengl_enc.c:100
GLsizei GLsizei * length
Definition: opengl_enc.c:114
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
int n
Definition: avisynth_c.h:760
void(* acelp_apply_order_2_transfer_function)(float *out, const float *in, const float zero_coeffs[2], const float pole_coeffs[2], float gain, float mem[2], int n)
Apply an order 2 rational transfer function in-place.
Definition: acelp_filters.h:47
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
void ff_acelp_filter_init(ACELPFContext *c)
Initialize ACELPFContext.
void ff_acelp_interpolatef(float *out, const float *in, const float *filter_coeffs, int precision, int frac_pos, int filter_length, int length)
Floating point version of ff_acelp_interpolate()
Definition: acelp_filters.c:78
FILE * out
Definition: movenc.c:54
Filter the word “frame” indicates either a video frame or a group of audio samples
void ff_acelp_interpolate(int16_t *out, const int16_t *in, const int16_t *filter_coeffs, int precision, int frac_pos, int filter_length, int length)
Generic FIR interpolation routine.
Definition: acelp_filters.c:44