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ffmpeg.wasm-core
Commit a6191d09 authored by Daniil Cherednik's avatar Daniil Cherednik Committed by Rostislav Pehlivanov
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dcaenc: Reverse data layout to prevent data copies during Huffman encoding introduction 

Reviewed-by: 's avatarRostislav Pehlivanov <atomnuker@gmail.com>
parent e7dec52d
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Showing with 43 additions and 45 deletions
libavcodec/dcaenc.c
View file @ a6191d09
......@@ -61,15 +61,15 @@ typedef struct DCAEncContext {
int32_t lfe_peak_cb;
const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
int32_t history[512][MAX_CHANNELS]; /* This is a circular buffer */
int32_t subband[SUBBAND_SAMPLES][DCAENC_SUBBANDS][MAX_CHANNELS];
int32_t quantized[SUBBAND_SAMPLES][DCAENC_SUBBANDS][MAX_CHANNELS];
int32_t peak_cb[DCAENC_SUBBANDS][MAX_CHANNELS];
int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
int32_t subband[MAX_CHANNELS][DCAENC_SUBBANDS][SUBBAND_SAMPLES];
int32_t quantized[MAX_CHANNELS][DCAENC_SUBBANDS][SUBBAND_SAMPLES];
int32_t peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS];
int32_t downsampled_lfe[DCA_LFE_SAMPLES];
int32_t masking_curve_cb[SUBSUBFRAMES][256];
int abits[DCAENC_SUBBANDS][MAX_CHANNELS];
int scale_factor[DCAENC_SUBBANDS][MAX_CHANNELS];
softfloat quant[DCAENC_SUBBANDS][MAX_CHANNELS];
int abits[MAX_CHANNELS][DCAENC_SUBBANDS];
int scale_factor[MAX_CHANNELS][DCAENC_SUBBANDS];
softfloat quant[MAX_CHANNELS][DCAENC_SUBBANDS];
int32_t eff_masking_curve_cb[256];
int32_t band_masking_cb[32];
int32_t worst_quantization_noise;
......@@ -259,8 +259,7 @@ static void subband_transform(DCAEncContext *c, const int32_t *input)
int hist_start = 0;
const int chi = c->channel_order_tab[ch];
for (i = 0; i < 512; i++)
hist[i] = c->history[i][ch];
memcpy(hist, &c->history[ch][0], 512 * sizeof(int32_t));
for (subs = 0; subs < SUBBAND_SAMPLES; subs++) {
int32_t accum[64];
......@@ -268,8 +267,7 @@ static void subband_transform(DCAEncContext *c, const int32_t *input)
int band;
/* Calculate the convolutions at once */
for (i = 0; i < 64; i++)
accum[i] = 0;
memset(accum, 0, 64 * sizeof(int32_t));
for (k = 0, i = hist_start, j = 0;
i < 512; k = (k + 1) & 63, i++, j++)
......@@ -289,12 +287,13 @@ static void subband_transform(DCAEncContext *c, const int32_t *input)
resp += mul32(accum[i], cos_t(s << 3)) >> 3;
}
c->subband[subs][band][ch] = ((band + 1) & 2) ? -resp : resp;
c->subband[ch][band][subs] = ((band + 1) & 2) ? -resp : resp;
}
/* Copy in 32 new samples from input */
for (i = 0; i < 32; i++)
hist[i + hist_start] = input[(subs * 32 + i) * c->channels + chi];
hist_start = (hist_start + 32) & 511;
}
}
......@@ -309,8 +308,7 @@ static void lfe_downsample(DCAEncContext *c, const int32_t *input)
int32_t accum;
int hist_start = 0;
for (i = 0; i < 512; i++)
hist[i] = c->history[i][c->channels - 1];
memcpy(hist, &c->history[c->channels - 1][0], 512 * sizeof(int32_t));
for (lfes = 0; lfes < DCA_LFE_SAMPLES; lfes++) {
/* Calculate the convolution */
......@@ -516,7 +514,7 @@ static void calc_masking(DCAEncContext *c, const int32_t *input)
const int chi = c->channel_order_tab[ch];
for (i = 0, k = 128 + 256 * ssf; k < 512; i++, k++)
data[i] = c->history[k][ch];
data[i] = c->history[ch][k];
for (k -= 512; i < 512; i++, k++)
data[i] = input[k * c->channels + chi];
adjust_jnd(c->samplerate_index, data, c->masking_curve_cb[ssf]);
......@@ -541,17 +539,17 @@ static void find_peaks(DCAEncContext *c)
{
int band, ch;
for (band = 0; band < 32; band++)
for (ch = 0; ch < c->fullband_channels; ch++) {
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < 32; band++) {
int sample;
int32_t m = 0;
for (sample = 0; sample < SUBBAND_SAMPLES; sample++) {
int32_t s = abs(c->subband[sample][band][ch]);
int32_t s = abs(c->subband[ch][band][sample]);
if (m < s)
m = s;
}
c->peak_cb[band][ch] = get_cb(m);
c->peak_cb[ch][band] = get_cb(m);
}
if (c->lfe_channel) {
......@@ -581,27 +579,27 @@ static int init_quantization_noise(DCAEncContext *c, int noise)
/* attempt to guess the bit distribution based on the prevoius frame */
for (ch = 0; ch < c->fullband_channels; ch++) {
for (band = 0; band < 32; band++) {
int snr_cb = c->peak_cb[band][ch] - c->band_masking_cb[band] - noise;
int snr_cb = c->peak_cb[ch][band] - c->band_masking_cb[band] - noise;
if (snr_cb >= 1312) {
c->abits[band][ch] = 26;
c->abits[ch][band] = 26;
ret |= USED_26ABITS;
} else if (snr_cb >= 222) {
c->abits[band][ch] = 8 + mul32(snr_cb - 222, 69000000);
c->abits[ch][band] = 8 + mul32(snr_cb - 222, 69000000);
ret |= USED_NABITS;
} else if (snr_cb >= 0) {
c->abits[band][ch] = 2 + mul32(snr_cb, 106000000);
c->abits[ch][band] = 2 + mul32(snr_cb, 106000000);
ret |= USED_NABITS;
} else {
c->abits[band][ch] = 1;
c->abits[ch][band] = 1;
ret |= USED_1ABITS;
}
}
}
for (band = 0; band < 32; band++)
for (ch = 0; ch < c->fullband_channels; ch++) {
c->consumed_bits += bit_consumption[c->abits[band][ch]];
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < 32; band++) {
c->consumed_bits += bit_consumption[c->abits[ch][band]];
}
return ret;
......@@ -653,7 +651,7 @@ static void shift_history(DCAEncContext *c, const int32_t *input)
for (ch = 0; ch < c->channels; ch++) {
const int chi = c->channel_order_tab[ch];
c->history[k][ch] = input[k * c->channels + chi];
c->history[ch][k] = input[k * c->channels + chi];
}
}
......@@ -702,11 +700,11 @@ static void calc_scales(DCAEncContext *c)
{
int band, ch;
for (band = 0; band < 32; band++)
for (ch = 0; ch < c->fullband_channels; ch++)
c->scale_factor[band][ch] = calc_one_scale(c->peak_cb[band][ch],
c->abits[band][ch],
&c->quant[band][ch]);
for (band = 0; band < 32; band++)
c->scale_factor[ch][band] = calc_one_scale(c->peak_cb[ch][band],
c->abits[ch][band],
&c->quant[ch][band]);
if (c->lfe_channel)
c->lfe_scale_factor = calc_one_scale(c->lfe_peak_cb, 11, &c->lfe_quant);
......@@ -716,10 +714,10 @@ static void quantize_all(DCAEncContext *c)
{
int sample, band, ch;
for (sample = 0; sample < SUBBAND_SAMPLES; sample++)
for (band = 0; band < 32; band++)
for (ch = 0; ch < c->fullband_channels; ch++)
c->quantized[sample][band][ch] = quantize_value(c->subband[sample][band][ch], c->quant[band][ch]);
for (band = 0; band < 32; band++)
for (sample = 0; sample < SUBBAND_SAMPLES; sample++)
c->quantized[ch][band][sample] = quantize_value(c->subband[ch][band][sample], c->quant[ch][band]);
}
static void put_frame_header(DCAEncContext *c)
......@@ -853,22 +851,22 @@ static void put_primary_audio_header(DCAEncContext *c)
static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch)
{
if (c->abits[band][ch] <= 7) {
if (c->abits[ch][band] <= 7) {
int sum, i, j;
for (i = 0; i < 8; i += 4) {
sum = 0;
for (j = 3; j >= 0; j--) {
sum *= ff_dca_quant_levels[c->abits[band][ch]];
sum += c->quantized[ss * 8 + i + j][band][ch];
sum += (ff_dca_quant_levels[c->abits[band][ch]] - 1) / 2;
sum *= ff_dca_quant_levels[c->abits[ch][band]];
sum += c->quantized[ch][band][ss * 8 + i + j];
sum += (ff_dca_quant_levels[c->abits[ch][band]] - 1) / 2;
}
put_bits(&c->pb, bit_consumption[c->abits[band][ch]] / 4, sum);
put_bits(&c->pb, bit_consumption[c->abits[ch][band]] / 4, sum);
}
} else {
int i;
for (i = 0; i < 8; i++) {
int bits = bit_consumption[c->abits[band][ch]] / 16;
put_sbits(&c->pb, bits, c->quantized[ss * 8 + i][band][ch]);
int bits = bit_consumption[c->abits[ch][band]] / 16;
put_sbits(&c->pb, bits, c->quantized[ch][band][ss * 8 + i]);
}
}
}
......@@ -892,7 +890,7 @@ static void put_subframe(DCAEncContext *c, int subframe)
/* Bit allocation index */
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < DCAENC_SUBBANDS; band++)
put_bits(&c->pb, 5, c->abits[band][ch]);
put_bits(&c->pb, 5, c->abits[ch][band]);
if (SUBSUBFRAMES > 1) {
/* Transition mode: none for each channel and subband */
......@@ -904,7 +902,7 @@ static void put_subframe(DCAEncContext *c, int subframe)
/* Scale factors */
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < DCAENC_SUBBANDS; band++)
put_bits(&c->pb, 7, c->scale_factor[band][ch]);
put_bits(&c->pb, 7, c->scale_factor[ch][band]);
/* Joint subband scale factor codebook select: not transmitted */
/* Scale factors for joint subband coding: not transmitted */
......
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