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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_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
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:821
ptrdiff_t size
Definition: opengl_enc.c:101
GLsizei GLsizei * length
Definition: opengl_enc.c:115
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
int n
Definition: avisynth_c.h:684
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_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
static AVCodec * c
FILE * out
Definition: movenc.c:54
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
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