40 -0.00001461907, -0.00009205479,-0.000056157569,0.00030117269,
41 0.0002422519, -0.00085293897,-0.0005205574, 0.0020340169,
42 0.00078333891, -0.0042153862, -0.00075614988, 0.0078402944,
43 -0.000061169922,-0.01344162, 0.0024626821, 0.021736089,
44 -0.007801671, -0.034090221, 0.01880949, 0.054326009,
45 -0.043596379, -0.099384367, 0.13207909, 0.46424159
55 for (i=0 ; i<64 ; i++)
60 for (i=0 ; i<24; i++) {
76 for (i = 0; i < 16; i++)
80 for (i = -15; i < 16; i++)
86 int num_samples,
float *
out)
88 float lev, gc_scale, gain_inc;
95 for (pos = 0; pos < num_samples; pos++)
96 out[pos] = in[pos] * gc_scale + prev[pos];
109 for (; pos < lastpos; pos++)
110 out[pos] = (in[pos] * gc_scale + prev[pos]) * lev;
113 for (; pos < lastpos + gctx->
loc_size; pos++) {
114 out[pos] = (in[pos] * gc_scale + prev[pos]) * lev;
119 for (; pos < num_samples; pos++)
120 out[pos] = in[pos] * gc_scale + prev[pos];
124 memcpy(prev, &in[num_samples], num_samples *
sizeof(
float));
128 float *delayBuf,
float *
temp)
133 memcpy(temp, delayBuf, 46*
sizeof(
float));
138 for(i=0; i<nIn; i+=2){
139 p3[2*i+0] = inlo[i ] + inhi[i ];
140 p3[2*i+1] = inlo[i ] - inhi[i ];
141 p3[2*i+2] = inlo[i+1] + inhi[i+1];
142 p3[2*i+3] = inlo[i+1] - inhi[i+1];
147 for (j = nIn; j != 0; j--) {
151 for (i = 0; i < 48; i += 2) {
153 s2 += p1[i+1] * qmf_window[i+1];
164 memcpy(delayBuf, temp + nIn*2, 46*
sizeof(
float));
void ff_atrac_iqmf(float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
Quadrature mirror synthesis filter.
int lev_code[7]
level at corresponding control point
static float qmf_window[48]
float ff_atrac_sf_table[64]
float gain_tab1[16]
gain compensation level table
int loc_code[7]
location of gain control points
Gain compensation context structure.
av_cold void ff_atrac_init_gain_compensation(AtracGCContext *gctx, int id2exp_offset, int loc_scale)
Initialize gain compensation context.
float gain_tab2[31]
gain compensation interpolation table
int loc_scale
scale of location code = 2^loc_scale samples
Gain control parameters for one subband.
Libavcodec external API header.
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
int loc_size
size of location code in samples
int id2exp_offset
offset for converting level index into level exponent
int num_points
number of gain control points
static const float qmf_48tap_half[24]
void ff_atrac_gain_compensation(AtracGCContext *gctx, float *in, float *prev, AtracGainInfo *gc_now, AtracGainInfo *gc_next, int num_samples, float *out)
Apply gain compensation and perform the MDCT overlapping part.
av_cold void ff_atrac_generate_tables(void)
Generate common tables.