FFmpeg  4.0
gsmdec_template.c
Go to the documentation of this file.
1 /*
2  * gsm 06.10 decoder
3  * Copyright (c) 2010 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * GSM decoder
25  */
26 
27 #include "get_bits.h"
28 #include "gsm.h"
29 #include "gsmdec_data.h"
30 
31 static void apcm_dequant_add(GetBitContext *gb, int16_t *dst, const int *frame_bits)
32 {
33  int i, val;
34  int maxidx = get_bits(gb, 6);
35  const int16_t *tab = ff_gsm_dequant_tab[maxidx];
36  for (i = 0; i < 13; i++) {
37  val = get_bits(gb, frame_bits[i]);
38  dst[3 * i] += tab[ff_gsm_requant_tab[frame_bits[i]][val]];
39  }
40 }
41 
42 static inline int gsm_mult(int a, int b)
43 {
44  return (int)(a * (SUINT)b + (1 << 14)) >> 15;
45 }
46 
47 static void long_term_synth(int16_t *dst, int lag, int gain_idx)
48 {
49  int i;
50  const int16_t *src = dst - lag;
51  uint16_t gain = ff_gsm_long_term_gain_tab[gain_idx];
52  for (i = 0; i < 40; i++)
53  dst[i] = gsm_mult(gain, src[i]);
54 }
55 
56 static inline int decode_log_area(int coded, int factor, int offset)
57 {
58  coded <<= 10;
59  coded -= offset;
60  return gsm_mult(coded, factor) * 2;
61 }
62 
63 static av_noinline int get_rrp(int filtered)
64 {
65  int abs = FFABS(filtered);
66  if (abs < 11059) abs <<= 1;
67  else if (abs < 20070) abs += 11059;
68  else abs = (abs >> 2) + 26112;
69  return filtered < 0 ? -abs : abs;
70 }
71 
72 static int filter_value(int in, int rrp[8], int v[9])
73 {
74  int i;
75  for (i = 7; i >= 0; i--) {
76  in -= gsm_mult(rrp[i], v[i]);
77  v[i + 1] = v[i] + gsm_mult(rrp[i], in);
78  }
79  v[0] = in;
80  return in;
81 }
82 
83 static void short_term_synth(GSMContext *ctx, int16_t *dst, const int16_t *src)
84 {
85  int i;
86  int rrp[8];
87  int *lar = ctx->lar[ctx->lar_idx];
88  int *lar_prev = ctx->lar[ctx->lar_idx ^ 1];
89  for (i = 0; i < 8; i++)
90  rrp[i] = get_rrp((lar_prev[i] >> 2) + (lar_prev[i] >> 1) + (lar[i] >> 2));
91  for (i = 0; i < 13; i++)
92  dst[i] = filter_value(src[i], rrp, ctx->v);
93 
94  for (i = 0; i < 8; i++)
95  rrp[i] = get_rrp((lar_prev[i] >> 1) + (lar [i] >> 1));
96  for (i = 13; i < 27; i++)
97  dst[i] = filter_value(src[i], rrp, ctx->v);
98 
99  for (i = 0; i < 8; i++)
100  rrp[i] = get_rrp((lar_prev[i] >> 2) + (lar [i] >> 1) + (lar[i] >> 2));
101  for (i = 27; i < 40; i++)
102  dst[i] = filter_value(src[i], rrp, ctx->v);
103 
104  for (i = 0; i < 8; i++)
105  rrp[i] = get_rrp(lar[i]);
106  for (i = 40; i < 160; i++)
107  dst[i] = filter_value(src[i], rrp, ctx->v);
108 
109  ctx->lar_idx ^= 1;
110 }
111 
112 static int postprocess(int16_t *data, int msr)
113 {
114  int i;
115  for (i = 0; i < 160; i++) {
116  msr = av_clip_int16(data[i] + gsm_mult(msr, 28180));
117  data[i] = av_clip_int16(msr * 2) & ~7;
118  }
119  return msr;
120 }
121 
122 static int gsm_decode_block(AVCodecContext *avctx, int16_t *samples,
123  GetBitContext *gb, int mode)
124 {
125  GSMContext *ctx = avctx->priv_data;
126  int i;
127  int16_t *ref_dst = ctx->ref_buf + 120;
128  int *lar = ctx->lar[ctx->lar_idx];
129  lar[0] = decode_log_area(get_bits(gb, 6), 13107, 1 << 15);
130  lar[1] = decode_log_area(get_bits(gb, 6), 13107, 1 << 15);
131  lar[2] = decode_log_area(get_bits(gb, 5), 13107, (1 << 14) + 2048*2);
132  lar[3] = decode_log_area(get_bits(gb, 5), 13107, (1 << 14) - 2560*2);
133  lar[4] = decode_log_area(get_bits(gb, 4), 19223, (1 << 13) + 94*2);
134  lar[5] = decode_log_area(get_bits(gb, 4), 17476, (1 << 13) - 1792*2);
135  lar[6] = decode_log_area(get_bits(gb, 3), 31454, (1 << 12) - 341*2);
136  lar[7] = decode_log_area(get_bits(gb, 3), 29708, (1 << 12) - 1144*2);
137 
138  for (i = 0; i < 4; i++) {
139  int lag = get_bits(gb, 7);
140  int gain_idx = get_bits(gb, 2);
141  int offset = get_bits(gb, 2);
142  lag = av_clip(lag, 40, 120);
143  long_term_synth(ref_dst, lag, gain_idx);
144  apcm_dequant_add(gb, ref_dst + offset, ff_gsm_apcm_bits[mode][i]);
145  ref_dst += 40;
146  }
147  memcpy(ctx->ref_buf, ctx->ref_buf + 160, 120 * sizeof(*ctx->ref_buf));
148  short_term_synth(ctx, samples, ctx->ref_buf + 120);
149  // for optimal speed this could be merged with short_term_synth,
150  // not done yet because it is a bit ugly
151  ctx->msr = postprocess(samples, ctx->msr);
152  return 0;
153 }
const char const char void * val
Definition: avisynth_c.h:771
static void short_term_synth(GSMContext *ctx, int16_t *dst, const int16_t *src)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:269
const char * b
Definition: vf_curves.c:113
#define SUINT
#define src
Definition: vp8dsp.c:254
static av_noinline int get_rrp(int filtered)
static int gsm_mult(int a, int b)
int v[9]
Definition: gsmdec_data.h:34
static int decode_log_area(int coded, int factor, int offset)
const char data[16]
Definition: mxf.c:90
bitstream reader API header.
static int gsm_decode_block(AVCodecContext *avctx, int16_t *samples, GetBitContext *gb, int mode)
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
const int16_t ff_gsm_dequant_tab[64][8]
Definition: gsmdec_data.c:36
static void long_term_synth(int16_t *dst, int lag, int gain_idx)
static int filter_value(int in, int rrp[8], int v[9])
AVFormatContext * ctx
Definition: movenc.c:48
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
static int postprocess(int16_t *data, int msr)
main external API structure.
Definition: avcodec.h:1518
const uint16_t ff_gsm_long_term_gain_tab[4]
Definition: gsmdec_data.c:25
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)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
static void apcm_dequant_add(GetBitContext *gb, int16_t *dst, const int *frame_bits)
static const int factor[16]
Definition: vf_pp7.c:75
int16_t ref_buf[280]
Definition: gsmdec_data.h:33
int lar[2][8]
Definition: gsmdec_data.h:35
const int *const ff_gsm_apcm_bits[][4]
Definition: gsmdec_data.c:117
void * priv_data
Definition: avcodec.h:1545
static const struct twinvq_data tab
#define av_noinline
Definition: attributes.h:62
mode
Use these values in ebur128_init (or&#39;ed).
Definition: ebur128.h:83
const uint8_t ff_gsm_requant_tab[4][8]
Definition: gsmdec_data.c:29