35 for (n = 0; n <
len; n++)
36 s +=
MUL64(x[n-j], x[n-k]);
47 tmp +=
MUL64(
a[0], corr[1]);
48 tmp +=
MUL64(
a[1], corr[2]);
49 tmp +=
MUL64(
a[2], corr[3]);
50 tmp +=
MUL64(
a[3], corr[4]);
58 tmp +=
MUL64(corr[5], aa[0]);
59 tmp +=
MUL64(corr[6], aa[1]);
60 tmp +=
MUL64(corr[7], aa[2]);
61 tmp +=
MUL64(corr[8], aa[3]);
63 tmp +=
MUL64(corr[9], aa[4]);
64 tmp +=
MUL64(corr[10], aa[5]);
65 tmp +=
MUL64(corr[11], aa[6]);
67 tmp +=
MUL64(corr[12], aa[7]);
68 tmp +=
MUL64(corr[13], aa[8]);
70 tmp +=
MUL64(corr[14], aa[9]);
85 int64_t min_err = 1ll << 62;
109 int64_t signal_energy = 0;
110 int64_t error_energy = 0;
112 for (i = 0; i <
len; i++) {
116 error_energy +=
MUL64(error, error);
122 return signal_energy / error_energy;
139 shift_bits =
av_log2(max) - 11;
142 input_buffer[i] =
norm__(in[i], 7);
143 input_buffer2[i] =
norm__(in[i], shift_bits);
158 for (i = 0; i <
len; i++)
195 for (i = 0; i <
len; i++) {
198 delta = (int64_t)in[i] - ((int64_t)work_bufer[DCA_ADPCM_COEFFS + i] << 7);
204 work_bufer[DCA_ADPCM_COEFFS+i] += dequant_delta;
207 memcpy(next_hist, &work_bufer[len],
sizeof(
int32_t) * DCA_ADPCM_COEFFS);
av_cold int ff_dcaadpcm_init(DCAADPCMEncContext *s)
int32_t premultiplied_coeffs[10]
static int64_t ff_dcaadpcm_predict(int pred_vq_index, const int32_t *input)
static int64_t calc_corr(const int32_t *x, int len, int j, int k)
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)
static int32_t quantize_value(int32_t value, softfloat quant)
static void ff_dca_core_dequantize(int32_t *output, const int32_t *input, int32_t step_size, int32_t scale, int residual, int len)
#define DCA_ADPCM_VQCODEBOOK_SZ
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
static void error(const char *err)
static int64_t apply_filter(const int16_t a[DCA_ADPCM_COEFFS], const int64_t corr[15], const int32_t aa[10])
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 precalc(premultiplied_coeffs *data)
const int16_t ff_dca_adpcm_vb[DCA_ADPCM_VQCODEBOOK_SZ][DCA_ADPCM_COEFFS]
int ff_dcaadpcm_subband_analysis(const DCAADPCMEncContext *s, const int32_t *in, int len, int *diff)
static int64_t calc_prediction_gain(int pred_vq, const int32_t *in, int32_t *out, int len)
static av_always_inline int diff(const uint32_t a, const uint32_t b)
static int64_t find_best_filter(const DCAADPCMEncContext *s, const int32_t *in, int len)
static int32_t norm__(int64_t a, int bits)
av_cold void ff_dcaadpcm_free(DCAADPCMEncContext *s)