Go to the documentation of this file.
43 #define MAX_CHANNELS 6
44 #define DCA_MAX_FRAME_SIZE 16384
45 #define DCA_HEADER_SIZE 13
46 #define DCA_LFE_SAMPLES 8
48 #define DCAENC_SUBBANDS 32
50 #define SUBSUBFRAMES 2
51 #define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)
54 #define COS_T(x) (c->cos_table[(x) & 2047])
115 double f1 =
f / 1000;
117 return -3.64 * pow(f1, -0.8)
118 + 6.8 *
exp(-0.6 * (f1 - 3.4) * (f1 - 3.4))
119 - 6.0 *
exp(-0.15 * (f1 - 8.7) * (f1 - 8.7))
120 - 0.0006 * (f1 * f1) * (f1 * f1);
129 return 20 * log10(
h);
145 c->subband[ch][band] = bufer +
155 if (
c->subband[0][0]) {
158 c->subband[0][0] =
NULL;
168 const uint8_t (**src_tablep)[2])
170 const uint8_t (*src_table)[2] = *src_tablep;
173 for (
unsigned i = 0;
i < count;
i++) {
174 unsigned dst_idx = src_table[
i][0];
176 dst[dst_idx][0] =
code >> (16 - src_table[
i][1]);
177 dst[dst_idx][1] = src_table[
i][1];
179 code += 1 << (16 - src_table[
i][1]);
181 *src_tablep += count;
207 int i, j, k, min_frame_bits;
214 c->fullband_channels =
c->channels =
layout.nb_channels;
215 c->lfe_channel = (
c->channels == 3 ||
c->channels == 6);
216 c->band_interpolation =
c->band_interpolation_tab[1];
217 c->band_spectrum =
c->band_spectrum_tab[1];
218 c->worst_quantization_noise = -2047;
219 c->worst_noise_ever = -2047;
220 c->consumed_adpcm_bits = 0;
225 switch (
layout.nb_channels) {
227 c->channel_config = 0;
230 c->channel_config = 2;
233 c->channel_config = 8;
236 c->channel_config = 9;
239 c->channel_config = 9;
245 if (
c->lfe_channel) {
246 c->fullband_channels--;
257 c->bit_allocation_sel[
i] = 6;
261 c->prediction_mode[
i][j] = -1;
266 for (
i = 0;
i < 9;
i++) {
272 c->samplerate_index =
i;
280 c->bitrate_index =
i;
282 min_frame_bits = 132 + (493 + 28 * 32) *
c->fullband_channels +
c->lfe_channel * 72;
286 c->frame_size = (
c->frame_bits + 7) / 8;
294 c->cos_table[0] = 0x7fffffff;
295 c->cos_table[512] = 0;
296 c->cos_table[1024] = -
c->cos_table[0];
297 for (
i = 1;
i < 512;
i++) {
299 c->cos_table[1024-
i] = -
c->cos_table[
i];
300 c->cos_table[1024+
i] = -
c->cos_table[
i];
301 c->cos_table[2048-
i] = +
c->cos_table[
i];
304 for (
i = 0;
i < 2048;
i++)
307 for (k = 0; k < 32; k++) {
308 for (j = 0; j < 8; j++) {
314 for (
i = 0;
i < 512;
i++) {
319 for (
i = 0;
i < 9;
i++) {
320 for (j = 0; j <
AUBANDS; j++) {
321 for (k = 0; k < 256; k++) {
329 for (
i = 0;
i < 256;
i++) {
331 c->cb_to_add[
i] = (
int32_t)(100 * log10(add));
333 for (j = 0; j < 8; j++) {
335 for (
i = 0;
i < 512;
i++) {
337 accum += reconst * cos(2 *
M_PI * (
i + 0.5 - 256) * (j + 0.5) / 512);
339 c->band_spectrum_tab[0][j] = (
int32_t)(200 * log10(accum));
341 for (j = 0; j < 8; j++) {
343 for (
i = 0;
i < 512;
i++) {
345 accum += reconst * cos(2 *
M_PI * (
i + 0.5 - 256) * (j + 0.5) / 512);
347 c->band_spectrum_tab[1][j] = (
int32_t)(200 * log10(accum));
366 int ch, subs,
i, k, j;
368 for (ch = 0; ch <
c->fullband_channels; ch++) {
372 const int chi =
c->channel_order_tab[ch];
374 memcpy(hist, &
c->history[ch][0], 512 *
sizeof(
int32_t));
382 memset(accum, 0, 64 *
sizeof(
int32_t));
384 for (k = 0,
i = hist_start, j = 0;
385 i < 512; k = (k + 1) & 63,
i++, j++)
386 accum[k] +=
mul32(hist[
i],
c->band_interpolation[j]);
387 for (
i = 0;
i < hist_start; k = (k + 1) & 63,
i++, j++)
388 accum[k] +=
mul32(hist[
i],
c->band_interpolation[j]);
390 for (k = 16; k < 32; k++)
391 accum[k] = accum[k] - accum[31 - k];
392 for (k = 32; k < 48; k++)
393 accum[k] = accum[k] + accum[95 - k];
395 for (band = 0; band < 32; band++) {
397 for (
i = 16;
i < 48;
i++) {
398 int s = (2 * band + 1) * (2 * (
i + 16) + 1);
402 c->subband[ch][band][subs] = ((band + 1) & 2) ? -resp : resp;
406 for (
i = 0;
i < 32;
i++)
407 hist[
i + hist_start] =
input[(subs * 32 +
i) *
c->channels + chi];
409 hist_start = (hist_start + 32) & 511;
417 const int lfech =
lfe_index[
c->channel_config];
423 memcpy(hist, &
c->history[
c->channels - 1][0], 512 *
sizeof(
int32_t));
429 for (
i = hist_start, j = 0;
i < 512;
i++, j++)
430 accum +=
mul32(hist[
i],
c->lfe_fir_64i[j]);
431 for (
i = 0;
i < hist_start;
i++, j++)
432 accum +=
mul32(hist[
i],
c->lfe_fir_64i[j]);
434 c->downsampled_lfe[lfes] = accum;
437 for (
i = 0;
i < 64;
i++)
438 hist[
i + hist_start] =
input[(lfes * 64 +
i) *
c->channels + lfech];
440 hist_start = (hist_start + 64) & 511;
447 for (
unsigned i = 0;
i < n;
i++)
453 uint8_t n, uint8_t sel)
455 for (
unsigned i = 0;
i < n;
i++)
461 uint8_t sel, uint8_t
table)
464 for (
unsigned i = 0;
i < n;
i++)
470 uint8_t n, uint8_t sel, uint8_t
table)
472 for (
unsigned i = 0;
i < n;
i++)
482 for (
i = 1024;
i > 0;
i >>= 1) {
483 if (
c->cb_to_level[
i + res] >= in)
496 return a +
c->cb_to_add[
a -
b];
506 for (
i = 0;
i < 512;
i++)
510 for (
i = 0;
i < 256;
i++) {
524 const int samplerate_index =
c->samplerate_index;
529 for (j = 0; j < 256; j++)
530 out_cb_unnorm[j] = -2047;
534 for (j = 0; j < 256; j++)
535 denom =
add_cb(
c, denom,
power[j] +
c->auf[samplerate_index][
i][j]);
536 for (j = 0; j < 256; j++)
537 out_cb_unnorm[j] =
add_cb(
c, out_cb_unnorm[j],
538 -denom +
c->auf[samplerate_index][
i][j]);
541 for (j = 0; j < 256; j++)
542 out_cb[j] =
add_cb(
c, out_cb[j], -out_cb_unnorm[j] - ca_cb - cs_cb);
555 for (
f = 0;
f < 4;
f++)
558 for (
f = 0;
f < 8;
f++)
559 walk(
c, band, band - 1, 8 * band - 4 +
f,
570 for (
f = 0;
f < 4;
f++)
573 for (
f = 0;
f < 8;
f++)
574 walk(
c, band, band + 1, 8 * band + 4 +
f,
585 if (value < c->band_masking_cb[band1])
586 c->band_masking_cb[band1] =
value;
591 int i, k, band, ch, ssf;
594 for (
i = 0;
i < 256;
i++)
596 c->masking_curve_cb[ssf][
i] = -2047;
599 for (ch = 0; ch <
c->fullband_channels; ch++) {
600 const int chi =
c->channel_order_tab[ch];
602 for (
i = 0, k = 128 + 256 * ssf; k < 512;
i++, k++)
603 data[
i] =
c->history[ch][k];
604 for (k -= 512;
i < 512;
i++, k++)
608 for (
i = 0;
i < 256;
i++) {
612 if (
c->masking_curve_cb[ssf][
i] < m)
613 m =
c->masking_curve_cb[ssf][
i];
614 c->eff_masking_curve_cb[
i] = m;
617 for (band = 0; band < 32; band++) {
618 c->band_masking_cb[band] = 2048;
640 for (ch = 0; ch <
c->fullband_channels; ch++) {
641 for (band = 0; band < 32; band++)
642 c->peak_cb[ch][band] =
find_peak(
c,
c->subband[ch][band],
657 c->consumed_adpcm_bits = 0;
658 for (ch = 0; ch <
c->fullband_channels; ch++) {
659 for (band = 0; band < 32; band++) {
663 if (pred_vq_id >= 0) {
664 c->prediction_mode[ch][band] = pred_vq_id;
665 c->consumed_adpcm_bits += 12;
666 c->diff_peak_cb[ch][band] =
find_peak(
c, estimated_diff, 16);
668 c->prediction_mode[ch][band] = -1;
675 #define USED_1ABITS 1
676 #define USED_26ABITS 4
682 if (
c->bitrate_index == 3)
694 int our_nscale, try_remove;
701 peak =
c->cb_to_level[-peak_cb];
703 for (try_remove = 64; try_remove > 0; try_remove >>= 1) {
710 our_nscale -= try_remove;
713 if (our_nscale >= 125)
726 int32_t diff_peak_cb =
c->diff_peak_cb[ch][band];
729 &
c->quant[ch][band]);
735 step_size,
c->adpcm_history[ch][band],
c->subband[ch][band],
736 c->adpcm_history[ch][band] + 4,
c->quantized[ch][band],
744 for (ch = 0; ch <
c->fullband_channels; ch++)
745 for (band = 0; band < 32; band++)
746 if (
c->prediction_mode[ch][band] >= 0)
754 for (ch = 0; ch <
c->fullband_channels; ch++) {
755 for (band = 0; band < 32; band++) {
756 if (
c->prediction_mode[ch][band] == -1) {
770 uint8_t sel,
id = abits - 1;
783 uint32_t t,
bits = 0;
788 if (vlc_bits[
i][0] == 0) {
795 best_sel_bits[
i] = vlc_bits[
i][0];
798 if (best_sel_bits[
i] > vlc_bits[
i][sel] && vlc_bits[
i][sel]) {
799 best_sel_bits[
i] = vlc_bits[
i][sel];
800 best_sel_id[
i] = sel;
805 t = best_sel_bits[
i] + 2;
806 if (t < clc_bits[
i]) {
807 res[
i] = best_sel_id[
i];
827 if (abits[
i] > 12 || abits[
i] == 0) {
850 uint32_t bits_counter = 0;
852 c->consumed_bits = 132 + 333 *
c->fullband_channels;
853 c->consumed_bits +=
c->consumed_adpcm_bits;
855 c->consumed_bits += 72;
858 for (ch = 0; ch <
c->fullband_channels; ch++) {
859 for (band = 0; band < 32; band++) {
860 int snr_cb =
c->peak_cb[ch][band] -
c->band_masking_cb[band] -
noise;
862 if (snr_cb >= 1312) {
863 c->abits[ch][band] = 26;
865 }
else if (snr_cb >= 222) {
866 c->abits[ch][band] = 8 +
mul32(snr_cb - 222, 69000000);
868 }
else if (snr_cb >= 0) {
869 c->abits[ch][band] = 2 +
mul32(snr_cb, 106000000);
871 }
else if (forbid_zero || snr_cb >= -140) {
872 c->abits[ch][band] = 1;
875 c->abits[ch][band] = 0;
880 &
c->bit_allocation_sel[ch]);
886 for (ch = 0; ch <
c->fullband_channels; ch++) {
887 for (band = 0; band < 32; band++) {
888 if (
c->prediction_mode[ch][band] == -1) {
891 &
c->quant[ch][band]);
900 for (ch = 0; ch <
c->fullband_channels; ch++) {
901 for (band = 0; band < 32; band++) {
904 c->quantized[ch][band],
905 huff_bit_count_accum[ch][
c->abits[ch][band] - 1]);
906 clc_bit_count_accum[ch][
c->abits[ch][band] - 1] +=
bit_consumption[
c->abits[ch][band]];
913 for (ch = 0; ch <
c->fullband_channels; ch++) {
915 clc_bit_count_accum[ch],
916 c->quant_index_sel[ch]);
919 c->consumed_bits += bits_counter;
932 low = high =
c->worst_quantization_noise;
933 if (
c->consumed_bits >
c->frame_bits) {
934 while (
c->consumed_bits >
c->frame_bits) {
944 while (
c->consumed_bits <=
c->frame_bits) {
954 for (down =
snr_fudge >> 1; down; down >>= 1) {
956 if (
c->consumed_bits <=
c->frame_bits)
961 c->worst_quantization_noise = high;
962 if (high >
c->worst_noise_ever)
963 c->worst_noise_ever = high;
970 for (k = 0; k < 512; k++)
971 for (ch = 0; ch <
c->channels; ch++) {
972 const int chi =
c->channel_order_tab[ch];
974 c->history[ch][k] =
input[k *
c->channels + chi];
986 for (ch = 0; ch <
c->channels; ch++) {
987 for (band = 0; band < 32; band++) {
989 if (
c->prediction_mode[ch][band] == -1) {
993 c->quantized[ch][band]+12, step_size,
996 AV_COPY128U(
c->adpcm_history[ch][band],
c->adpcm_history[ch][band]+4);
1006 samples[0] =
c->adpcm_history[ch][band][0] * (1 << 7);
1007 samples[1] =
c->adpcm_history[ch][band][1] * (1 << 7);
1008 samples[2] =
c->adpcm_history[ch][band][2] * (1 << 7);
1009 samples[3] =
c->adpcm_history[ch][band][3] * (1 << 7);
1110 put_bits(&
c->pb, 3,
c->fullband_channels - 1);
1113 for (ch = 0; ch <
c->fullband_channels; ch++)
1117 for (ch = 0; ch <
c->fullband_channels; ch++)
1121 for (ch = 0; ch <
c->fullband_channels; ch++)
1125 for (ch = 0; ch <
c->fullband_channels; ch++)
1129 for (ch = 0; ch <
c->fullband_channels; ch++)
1133 for (ch = 0; ch <
c->fullband_channels; ch++)
1134 put_bits(&
c->pb, 3,
c->bit_allocation_sel[ch]);
1138 for (ch = 0; ch <
c->fullband_channels; ch++)
1143 for (ch = 0; ch <
c->fullband_channels; ch++)
1152 int i, j, sum,
bits, sel;
1155 sel =
c->quant_index_sel[ch][
c->abits[ch][band] - 1];
1159 sel,
c->abits[ch][band] - 1);
1164 if (
c->abits[ch][band] <= 7) {
1165 for (
i = 0;
i < 8;
i += 4) {
1167 for (j = 3; j >= 0; j--) {
1169 sum +=
c->quantized[ch][band][
ss * 8 +
i + j];
1178 for (
i = 0;
i < 8;
i++) {
1186 int i, band,
ss, ch;
1195 for (ch = 0; ch <
c->fullband_channels; ch++)
1197 put_bits(&
c->pb, 1, !(
c->prediction_mode[ch][band] == -1));
1200 for (ch = 0; ch <
c->fullband_channels; ch++)
1202 if (
c->prediction_mode[ch][band] >= 0)
1203 put_bits(&
c->pb, 12,
c->prediction_mode[ch][band]);
1206 for (ch = 0; ch <
c->fullband_channels; ch++) {
1207 if (
c->bit_allocation_sel[ch] == 6) {
1213 c->bit_allocation_sel[ch]);
1219 for (ch = 0; ch <
c->fullband_channels; ch++)
1221 if (
c->abits[ch][band])
1226 for (ch = 0; ch <
c->fullband_channels; ch++)
1228 if (
c->abits[ch][band])
1229 put_bits(&
c->pb, 7,
c->scale_factor[ch][band]);
1239 if (
c->lfe_channel) {
1247 for (ch = 0; ch <
c->fullband_channels; ch++)
1249 if (
c->abits[ch][band])
1273 if (
c->options.adpcm_mode)
1290 *got_packet_ptr = 1;
1294 #define DCAENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1302 .
class_name =
"DCA (DTS Coherent Acoustics)",
static void lfe_downsample(DCAEncContext *c, const int32_t *input)
int frame_size
Number of samples per channel in an audio frame.
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
static uint32_t dca_vlc_calc_quant_bits(const int values[], uint8_t n, uint8_t sel, uint8_t table)
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
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)
int32_t * subband[MAX_CHANNELS][DCAENC_SUBBANDS]
#define AV_CHANNEL_LAYOUT_STEREO
int sample_rate
samples per second
static double cb(void *priv, double x, double y)
static const AVOption options[]
static void walk_band_low(DCAEncContext *c, int band, int channel, walk_band_t walk, int32_t *arg)
const uint32_t ff_dca_bit_rates[32]
#define AV_CH_LAYOUT_MONO
av_cold void ff_dcaadpcm_free(DCAADPCMEncContext *s)
static const softfloat scalefactor_inv[128]
static void put_sbits(PutBitContext *pb, int n, int32_t value)
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
static const uint16_t erb[]
#define AV_CHANNEL_LAYOUT_2_2
static const uint8_t lfe_index[7]
static void put_subframe(DCAEncContext *c, int subframe)
This structure describes decoded (raw) audio or video data.
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
CompressionOptions options
static int32_t get_step_size(DCAEncContext *c, int ch, int band)
const uint32_t ff_dca_lossy_quant[32]
static const uint16_t table[]
static void calc_lfe_scales(DCAEncContext *c)
int32_t adpcm_history[MAX_CHANNELS][DCAENC_SUBBANDS][DCA_ADPCM_COEFFS *2]
#define fc(width, name, range_min, range_max)
static void update_band_masking(DCAEncContext *c, int band1, int band2, int f, int32_t spectrum1, int32_t spectrum2, int channel, int32_t *arg)
static int calc_one_scale(DCAEncContext *c, int32_t peak_cb, int abits, softfloat *quant)
static int32_t quantize_value(int32_t value, softfloat quant)
const int32_t * band_interpolation
static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
static void put_frame_header(DCAEncContext *c)
DCAADPCMEncContext adpcm_ctx
av_cold int av_tx_init(AVTXContext **ctx, av_tx_fn *tx, enum AVTXType type, int inv, int len, const void *scale, uint64_t flags)
Initialize a transform context with the given configuration (i)MDCTs with an odd length are currently...
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
int32_t history[MAX_CHANNELS][512]
static void calc_masking(DCAEncContext *c, const int32_t *input)
static void dca_vlc_enc_quant(PutBitContext *pb, const int values[], uint8_t n, uint8_t sel, uint8_t table)
AVCodec p
The public AVCodec.
static void adpcm_analysis(DCAEncContext *c)
const float ff_dca_fir_32bands_nonperfect[512]
AVChannelLayout ch_layout
Audio channel layout.
const uint8_t ff_dca_quant_index_group_size[DCA_CODE_BOOKS]
static void accumulate_huff_bit_consumption(int abits, int32_t *quantized, uint32_t *result)
static int init_quantization_noise(DCAEncContext *c, int noise, int forbid_zero)
static double val(void *priv, double ch)
static av_always_inline float scale(float x, float s)
const uint32_t ff_dca_quant_levels[32]
#define ss(width, name, subs,...)
#define FF_CODEC_ENCODE_CB(func)
int32_t auf[9][AUBANDS][256]
#define AV_CH_LAYOUT_STEREO
static int put_bytes_left(const PutBitContext *s, int round_up)
static av_cold void create_enc_table(uint16_t dst[][2], unsigned count, const uint8_t(**src_tablep)[2])
static const int bit_consumption[27]
static void walk_band_high(DCAEncContext *c, int band, int channel, walk_band_t walk, int32_t *arg)
static void quantize_adpcm_subband(DCAEncContext *c, int ch, int band)
int32_t quantized[MAX_CHANNELS][DCAENC_SUBBANDS][SUBBAND_SAMPLES]
static const uint8_t quant[64]
static int ff_thread_once(char *control, void(*routine)(void))
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define AV_CODEC_CAP_EXPERIMENTAL
Codec is experimental and is thus avoided in favor of non experimental encoders.
const FFCodec ff_dca_encoder
void(* av_tx_fn)(AVTXContext *s, void *out, void *in, ptrdiff_t stride)
Function pointer to a function to perform the transform.
static void quantize_adpcm(DCAEncContext *c)
int abits[MAX_CHANNELS][DCAENC_SUBBANDS]
int32_t peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS]
const int32_t * band_spectrum
static double hom(double f)
static uint16_t bitalloc_table[DCA_NUM_BITALLOC_CODES][2]
int32_t eff_masking_curve_cb[256]
int32_t downsampled_lfe[DCA_LFE_SAMPLES]
int(* init)(AVBSFContext *ctx)
#define AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE
This encoder can reorder user opaque values from input AVFrames and return them with corresponding ou...
const int8_t ff_dca_bitalloc_offsets[DCA_CODE_BOOKS]
#define CODEC_OLD_CHANNEL_LAYOUTS(...)
static uint32_t set_best_abits_code(int abits[DCAENC_SUBBANDS], int bands, int32_t *res)
#define av_assert0(cond)
assert() equivalent, that is always enabled.
static av_cold void dcaenc_init_static_tables(void)
static const float bands[]
#define CODEC_LONG_NAME(str)
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
#define LIBAVUTIL_VERSION_INT
Describe the class of an AVClass context structure.
and forward the result(frame or status change) to the corresponding input. If nothing is possible
static void adjust_jnd(DCAEncContext *c, const int32_t in[512], int32_t out_cb[256])
#define LOCAL_ALIGNED_32(t, v,...)
const uint32_t ff_dca_lossless_quant[32]
static int32_t mul32(int32_t a, int32_t b)
const float ff_dca_lfe_fir_64[256]
#define AV_COPY128U(d, s)
int64_t bit_rate
the average bitrate
static uint16_t bitalloc_12_table[DCA_BITALLOC_12_COUNT][12+1][2]
const char * av_default_item_name(void *ptr)
Return the context name.
static const softfloat stepsize_inv[27]
#define AV_CH_LAYOUT_5POINT1
int32_t band_masking_cb[32]
static const FFCodecDefault defaults[]
int ff_dcaadpcm_subband_analysis(const DCAADPCMEncContext *s, const int32_t *in, int len, int *diff)
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
static av_cold int encode_close(AVCodecContext *avctx)
int32_t worst_quantization_noise
int32_t band_interpolation_tab[2][512]
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
const uint32_t ff_dca_scale_factor_quant7[128]
static void subband_bufer_free(DCAEncContext *c)
softfloat quant[MAX_CHANNELS][DCAENC_SUBBANDS]
An AVChannelLayout holds information about the channel layout of audio data.
void(* walk_band_t)(DCAEncContext *c, int band1, int band2, int f, int32_t spectrum1, int32_t spectrum2, int channel, int32_t *arg)
static int32_t add_cb(DCAEncContext *c, int32_t a, int32_t b)
#define DCA_BITALLOC_12_COUNT
static int encode_init(AVCodecContext *avctx)
static void fill_in_adpcm_bufer(DCAEncContext *c)
#define DCA_MAX_FRAME_SIZE
static void quantize_pcm(DCAEncContext *c)
int32_t masking_curve_cb[SUBSUBFRAMES][256]
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
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
static void put_primary_audio_header(DCAEncContext *c)
av_cold void av_tx_uninit(AVTXContext **ctx)
Frees a context and sets *ctx to NULL, does nothing when *ctx == NULL.
static const uint16_t(*[DCA_CODE_BOOKS][8] bitalloc_tables)[2]
int32_t quant_index_sel[MAX_CHANNELS][DCA_CODE_BOOKS]
static uint32_t dca_vlc_calc_alloc_bits(const int values[], uint8_t n, uint8_t sel)
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel layout
#define AV_CH_LAYOUT_5POINT0
static void find_peaks(DCAEncContext *c)
const uint8_t ff_dca_quant_index_sel_nbits[DCA_CODE_BOOKS]
#define i(width, name, range_min, range_max)
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 it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
AVSampleFormat
Audio sample formats.
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 default value
const char * name
Name of the codec implementation.
static int32_t norm__(int64_t a, int bits)
void * av_calloc(size_t nmemb, size_t size)
static const int8_t channel_reorder_nolfe[7][5]
static const int snr_fudge
#define FFSWAP(type, a, b)
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
static void subband_transform(DCAEncContext *c, const int32_t *input)
int consumed_adpcm_bits
Number of bits to transmit ADPCM related info.
static const int8_t channel_reorder_lfe[7][5]
static void ff_dca_core_dequantize(int32_t *output, const int32_t *input, int32_t step_size, int32_t scale, int residual, int len)
av_cold int ff_dcaadpcm_init(DCAADPCMEncContext *s)
main external API structure.
static float power(float r, float g, float b, float max)
static uint8_t * put_bits_ptr(PutBitContext *s)
Return the pointer to the byte where the bitstream writer will put the next bit.
static int noise(AVBSFContext *ctx, AVPacket *pkt)
int ff_get_encode_buffer(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int flags)
Get a buffer for a packet.
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return values
const uint8_t ff_dca_bitalloc_sizes[DCA_CODE_BOOKS]
Filter the word “frame” indicates either a video frame or a group of audio samples
const uint8_t ff_dca_vlc_src_tables[][2]
static int subband_bufer_alloc(DCAEncContext *c)
static void assign_bits(DCAEncContext *c)
static int32_t get_cb(DCAEncContext *c, int32_t in)
static const uint8_t bitstream_sfreq[]
static int32_t find_peak(DCAEncContext *c, const int32_t *in, int len)
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
#define AV_CHANNEL_LAYOUT_MONO
This structure stores compressed data.
const float ff_dca_fir_32bands_perfect[512]
static void shift_history(DCAEncContext *c, const int32_t *input)
static uint32_t set_best_code(uint32_t vlc_bits[DCA_CODE_BOOKS][7], uint32_t clc_bits[DCA_CODE_BOOKS], int32_t res[DCA_CODE_BOOKS])
static void dca_vlc_enc_alloc(PutBitContext *pb, const int values[], uint8_t n, uint8_t sel)
int32_t prediction_mode[MAX_CHANNELS][DCAENC_SUBBANDS]
static const double coeff[2][5]
#define AV_CHANNEL_LAYOUT_5POINT0
int32_t cb_to_level[2048]
static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch)
int32_t bit_allocation_sel[MAX_CHANNELS]
int32_t band_spectrum_tab[2][8]
#define AV_CHANNEL_LAYOUT_5POINT1
static void calc_power(DCAEncContext *c, const int32_t in[2 *256], int32_t power[256])
@ AV_SAMPLE_FMT_S32
signed 32 bits
int32_t diff_peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS]
expected peak of residual signal
static const AVClass dcaenc_class
int scale_factor[MAX_CHANNELS][DCAENC_SUBBANDS]
static double gammafilter(int i, double f)
const int8_t * channel_order_tab
channel reordering table, lfe and non lfe
#define DCA_NUM_BITALLOC_CODES