FFmpeg
dcaadpcm.h
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1 /*
2  * DCA ADPCM engine
3  * Copyright (C) 2017 Daniil Cherednik
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 #ifndef AVCODEC_DCAADPCM_H
23 #define AVCODEC_DCAADPCM_H
24 
25 #include "dcamath.h"
26 #include "dcadata.h"
27 #include "dcaenc.h"
28 
29 typedef struct DCAADPCMEncContext {
30  void *private_data;
31 } DCAADPCMEncContext;
32 
33 static inline int64_t ff_dcaadpcm_predict(int pred_vq_index, const int32_t *input)
34 {
35  int i;
36  const int16_t *coeff = ff_dca_adpcm_vb[pred_vq_index];
37  int64_t pred = 0;
38  for (i = 0; i < DCA_ADPCM_COEFFS; i++)
39  pred += (int64_t)input[DCA_ADPCM_COEFFS - 1 - i] * coeff[i];
40 
41  return clip23(norm13(pred));
42 }
43 
44 int ff_dcaadpcm_subband_analysis(const DCAADPCMEncContext *s, const int32_t *input, int len, int *diff);
45 
46 int ff_dcaadpcm_do_real(int pred_vq_index,
47  softfloat quant, int32_t scale_factor, int32_t step_size,
48  const int32_t *prev_hist, const int32_t *in, int32_t *next_hist, int32_t *out,
49  int len, int32_t peak);
50 
51 av_cold int ff_dcaadpcm_init(DCAADPCMEncContext *s);
52 av_cold void ff_dcaadpcm_free(DCAADPCMEncContext *s);
53 
54 #endif /* AVCODEC_DCAADPCM_H */
dcamath.h
ff_dcaadpcm_do_real
int ff_dcaadpcm_do_real(int pred_vq_index, softfloat quant, int32_t scale_factor, int32_t step_size, const int32_t *prev_hist, const int32_t *in, int32_t *next_hist, int32_t *out, int len, int32_t peak)
Definition: dcaadpcm.c:183
norm13
static int32_t norm13(int64_t a)
Definition: dcamath.h:40
DCAADPCMEncContext
Definition: dcaadpcm.h:29
out
FILE * out
Definition: movenc.c:54
DCAADPCMEncContext::private_data
void * private_data
Definition: dcaadpcm.h:30
dcadata.h
clip23
static int32_t clip23(int32_t a)
Definition: dcamath.h:54
av_cold
#define av_cold
Definition: attributes.h:90
ff_dcaadpcm_predict
static int64_t ff_dcaadpcm_predict(int pred_vq_index, const int32_t *input)
Definition: dcaadpcm.h:33
s
#define s(width, name)
Definition: cbs_vp9.c:257
DCA_ADPCM_COEFFS
#define DCA_ADPCM_COEFFS
Definition: dcadata.h:28
int32_t
int32_t
Definition: audio_convert.c:194
softfloat
Definition: dcaenc.h:29
dcaenc.h
input
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some input
Definition: filter_design.txt:172
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
int i
Definition: input.c:407
ff_dcaadpcm_free
av_cold void ff_dcaadpcm_free(DCAADPCMEncContext *s)
Definition: dcaadpcm.c:225
len
int len
Definition: vorbis_enc_data.h:452
ff_dca_adpcm_vb
const int16_t ff_dca_adpcm_vb[DCA_ADPCM_VQCODEBOOK_SZ][DCA_ADPCM_COEFFS]
Definition: dcadata.c:60
pred
static const float pred[4]
Definition: siprdata.h:259
quant
const uint8_t * quant
Definition: vorbis_enc_data.h:458
diff
static av_always_inline int diff(const uint32_t a, const uint32_t b)
Definition: vf_palettegen.c:136
coeff
static const double coeff[2][5]
Definition: vf_owdenoise.c:73
ff_dcaadpcm_init
av_cold int ff_dcaadpcm_init(DCAADPCMEncContext *s)
Definition: dcaadpcm.c:212
ff_dcaadpcm_subband_analysis
int ff_dcaadpcm_subband_analysis(const DCAADPCMEncContext *s, const int32_t *input, int len, int *diff)
Definition: dcaadpcm.c:125
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