FFmpeg  4.3
gsmdec_template.c
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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 }
b
#define b
Definition: input.c:41
data
const char data[16]
Definition: mxf.c:91
short_term_synth
static void short_term_synth(GSMContext *ctx, int16_t *dst, const int16_t *src)
Definition: gsmdec_template.c:83
samples
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new samples
Definition: fate.txt:139
gsmdec_data.h
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
GetBitContext
Definition: get_bits.h:61
tab
static const struct twinvq_data tab
Definition: twinvq_data.h:11135
val
static double val(void *priv, double ch)
Definition: aeval.c:76
av_noinline
#define av_noinline
Definition: attributes.h:72
get_rrp
static av_noinline int get_rrp(int filtered)
Definition: gsmdec_template.c:63
gsm_mult
static int gsm_mult(int a, int b)
Definition: gsmdec_template.c:42
gsm.h
ff_gsm_apcm_bits
const int *const ff_gsm_apcm_bits[][4]
Definition: gsmdec_data.c:117
ctx
AVFormatContext * ctx
Definition: movenc.c:48
get_bits.h
FFABS
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
src
#define src
Definition: vp8dsp.c:254
abs
#define abs(x)
Definition: cuda_runtime.h:35
ff_gsm_requant_tab
const uint8_t ff_gsm_requant_tab[4][8]
Definition: gsmdec_data.c:29
filter_value
static int filter_value(int in, int rrp[8], int v[9])
Definition: gsmdec_template.c:72
ff_gsm_long_term_gain_tab
const uint16_t ff_gsm_long_term_gain_tab[4]
Definition: gsmdec_data.c:25
a
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:41
postprocess
static int postprocess(int16_t *data, int msr)
Definition: gsmdec_template.c:112
offset
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
Definition: writing_filters.txt:86
in
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
Definition: audio_convert.c:326
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
gsm_decode_block
static int gsm_decode_block(AVCodecContext *avctx, int16_t *samples, GetBitContext *gb, int mode)
Definition: gsmdec_template.c:122
SUINT
#define SUINT
Definition: dct32_template.c:30
ff_gsm_dequant_tab
const int16_t ff_gsm_dequant_tab[64][8]
Definition: gsmdec_data.c:36
apcm_dequant_add
static void apcm_dequant_add(GetBitContext *gb, int16_t *dst, const int *frame_bits)
Definition: gsmdec_template.c:31
AVCodecContext
main external API structure.
Definition: avcodec.h:526
mode
mode
Definition: ebur128.h:83
factor
static const int factor[16]
Definition: vf_pp7.c:75
decode_log_area
static int decode_log_area(int coded, int factor, int offset)
Definition: gsmdec_template.c:56
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:553
long_term_synth
static void long_term_synth(int16_t *dst, int lag, int gain_idx)
Definition: gsmdec_template.c:47
GSMContext
Definition: gsmdec_data.h:28