[FFmpeg-cvslog] avcodec/dcaenc: Initial implementation of ADPCM encoding for DCA encoder
Daniil Cherednik
git at videolan.org
Mon May 8 07:57:07 EEST 2017
ffmpeg | branch: master | Daniil Cherednik <dan.cherednik at gmail.com> | Mon Feb 20 23:22:51 2017 +0000| [b8c2b9c39279171f647d9c81f34ffa3d3ae93c47] | committer: Rostislav Pehlivanov
avcodec/dcaenc: Initial implementation of ADPCM encoding for DCA encoder
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b8c2b9c39279171f647d9c81f34ffa3d3ae93c47
---
libavcodec/Makefile | 3 +-
libavcodec/dca_core.c | 46 +++-------
libavcodec/dca_core.h | 25 ++++-
libavcodec/dcaadpcm.c | 228 ++++++++++++++++++++++++++++++++++++++++++++++
libavcodec/dcaadpcm.h | 54 +++++++++++
libavcodec/dcadata.c | 2 +-
libavcodec/dcadata.h | 5 +-
libavcodec/dcaenc.c | 247 ++++++++++++++++++++++++++++++++++++++++++--------
libavcodec/dcaenc.h | 11 +++
libavcodec/dcamath.h | 1 +
10 files changed, 546 insertions(+), 76 deletions(-)
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index b5c8cc1f98..44acc95394 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -244,7 +244,8 @@ OBJS-$(CONFIG_CYUV_DECODER) += cyuv.o
OBJS-$(CONFIG_DCA_DECODER) += dcadec.o dca.o dcadata.o dcahuff.o \
dca_core.o dca_exss.o dca_xll.o dca_lbr.o \
dcadsp.o dcadct.o synth_filter.o
-OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o dca.o dcadata.o dcahuff.o
+OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o dca.o dcadata.o dcahuff.o \
+ dcaadpcm.o
OBJS-$(CONFIG_DDS_DECODER) += dds.o
OBJS-$(CONFIG_DIRAC_DECODER) += diracdec.o dirac.o diracdsp.o diractab.o \
dirac_arith.o dirac_dwt.o dirac_vlc.o
diff --git a/libavcodec/dca_core.c b/libavcodec/dca_core.c
index d5e628e763..36040f6f9d 100644
--- a/libavcodec/dca_core.c
+++ b/libavcodec/dca_core.c
@@ -18,6 +18,7 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#include "dcaadpcm.h"
#include "dcadec.h"
#include "dcadata.h"
#include "dcahuff.h"
@@ -670,46 +671,21 @@ static inline int extract_audio(DCACoreDecoder *s, int32_t *audio, int abits, in
return 0;
}
-static inline void dequantize(int32_t *output, const int32_t *input,
- int32_t step_size, int32_t scale, int residual)
-{
- // Account for quantizer step size
- int64_t step_scale = (int64_t)step_size * scale;
- int n, shift = 0;
-
- // Limit scale factor resolution to 22 bits
- if (step_scale > (1 << 23)) {
- shift = av_log2(step_scale >> 23) + 1;
- step_scale >>= shift;
- }
-
- // Scale the samples
- if (residual) {
- for (n = 0; n < DCA_SUBBAND_SAMPLES; n++)
- output[n] += clip23(norm__(input[n] * step_scale, 22 - shift));
- } else {
- for (n = 0; n < DCA_SUBBAND_SAMPLES; n++)
- output[n] = clip23(norm__(input[n] * step_scale, 22 - shift));
- }
-}
-
static inline void inverse_adpcm(int32_t **subband_samples,
const int16_t *vq_index,
const int8_t *prediction_mode,
int sb_start, int sb_end,
int ofs, int len)
{
- int i, j, k;
+ int i, j;
for (i = sb_start; i < sb_end; i++) {
if (prediction_mode[i]) {
- const int16_t *coeff = ff_dca_adpcm_vb[vq_index[i]];
+ const int pred_id = vq_index[i];
int32_t *ptr = subband_samples[i] + ofs;
for (j = 0; j < len; j++) {
- int64_t err = 0;
- for (k = 0; k < DCA_ADPCM_COEFFS; k++)
- err += (int64_t)ptr[j - k - 1] * coeff[k];
- ptr[j] = clip23(ptr[j] + clip23(norm13(err)));
+ int32_t x = ff_dcaadpcm_predict(pred_id, ptr + j - DCA_ADPCM_COEFFS);
+ ptr[j] = clip23(ptr[j] + x);
}
}
}
@@ -817,8 +793,8 @@ static int parse_subframe_audio(DCACoreDecoder *s, int sf, enum HeaderType heade
scale = clip23(adj * scale >> 22);
}
- dequantize(s->subband_samples[ch][band] + ofs,
- audio, step_size, scale, 0);
+ ff_dca_core_dequantize(s->subband_samples[ch][band] + ofs,
+ audio, step_size, scale, 0, DCA_SUBBAND_SAMPLES);
}
}
@@ -1146,8 +1122,8 @@ static int parse_xbr_subframe(DCACoreDecoder *s, int xbr_base_ch, int xbr_nchann
else
scale = xbr_scale_factors[ch][band][1];
- dequantize(s->subband_samples[ch][band] + ofs,
- audio, step_size, scale, 1);
+ ff_dca_core_dequantize(s->subband_samples[ch][band] + ofs,
+ audio, step_size, scale, 1, DCA_SUBBAND_SAMPLES);
}
}
@@ -1326,8 +1302,8 @@ static int parse_x96_subframe_audio(DCACoreDecoder *s, int sf, int xch_base, int
// Get the scale factor
scale = s->scale_factors[ch][band >> 1][band & 1];
- dequantize(s->x96_subband_samples[ch][band] + ofs,
- audio, step_size, scale, 0);
+ ff_dca_core_dequantize(s->x96_subband_samples[ch][band] + ofs,
+ audio, step_size, scale, 0, DCA_SUBBAND_SAMPLES);
}
}
diff --git a/libavcodec/dca_core.h b/libavcodec/dca_core.h
index e84bdab18e..7dcfb13bc7 100644
--- a/libavcodec/dca_core.h
+++ b/libavcodec/dca_core.h
@@ -33,6 +33,7 @@
#include "dca_exss.h"
#include "dcadsp.h"
#include "dcadct.h"
+#include "dcamath.h"
#include "dcahuff.h"
#include "fft.h"
#include "synth_filter.h"
@@ -43,7 +44,6 @@
#define DCA_SUBFRAMES 16
#define DCA_SUBBAND_SAMPLES 8
#define DCA_PCMBLOCK_SAMPLES 32
-#define DCA_ADPCM_COEFFS 4
#define DCA_LFE_HISTORY 8
#define DCA_ABITS_MAX 26
@@ -195,6 +195,29 @@ static inline int ff_dca_core_map_spkr(DCACoreDecoder *core, int spkr)
return -1;
}
+static inline void ff_dca_core_dequantize(int32_t *output, const int32_t *input,
+ int32_t step_size, int32_t scale, int residual, int len)
+{
+ // Account for quantizer step size
+ int64_t step_scale = (int64_t)step_size * scale;
+ int n, shift = 0;
+
+ // Limit scale factor resolution to 22 bits
+ if (step_scale > (1 << 23)) {
+ shift = av_log2(step_scale >> 23) + 1;
+ step_scale >>= shift;
+ }
+
+ // Scale the samples
+ if (residual) {
+ for (n = 0; n < len; n++)
+ output[n] += clip23(norm__(input[n] * step_scale, 22 - shift));
+ } else {
+ for (n = 0; n < len; n++)
+ output[n] = clip23(norm__(input[n] * step_scale, 22 - shift));
+ }
+}
+
int ff_dca_core_parse(DCACoreDecoder *s, uint8_t *data, int size);
int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset);
int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth);
diff --git a/libavcodec/dcaadpcm.c b/libavcodec/dcaadpcm.c
new file mode 100644
index 0000000000..8742c7ccf6
--- /dev/null
+++ b/libavcodec/dcaadpcm.c
@@ -0,0 +1,228 @@
+/*
+ * DCA ADPCM engine
+ * Copyright (C) 2017 Daniil Cherednik
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+
+#include "dcaadpcm.h"
+#include "dcaenc.h"
+#include "dca_core.h"
+#include "mathops.h"
+
+typedef int32_t premultiplied_coeffs[10];
+
+//assume we have DCA_ADPCM_COEFFS values before x
+static inline int64_t calc_corr(const int32_t *x, int len, int j, int k)
+{
+ int n;
+ int64_t s = 0;
+ for (n = 0; n < len; n++)
+ s += MUL64(x[n-j], x[n-k]);
+ return s;
+}
+
+static inline int64_t apply_filter(const int16_t a[DCA_ADPCM_COEFFS], const int64_t corr[15], const int32_t aa[10])
+{
+ int64_t err = 0;
+ int64_t tmp = 0;
+
+ err = corr[0];
+
+ tmp += MUL64(a[0], corr[1]);
+ tmp += MUL64(a[1], corr[2]);
+ tmp += MUL64(a[2], corr[3]);
+ tmp += MUL64(a[3], corr[4]);
+
+ tmp = norm__(tmp, 13);
+ tmp += tmp;
+
+ err -= tmp;
+ tmp = 0;
+
+ tmp += MUL64(corr[5], aa[0]);
+ tmp += MUL64(corr[6], aa[1]);
+ tmp += MUL64(corr[7], aa[2]);
+ tmp += MUL64(corr[8], aa[3]);
+
+ tmp += MUL64(corr[9], aa[4]);
+ tmp += MUL64(corr[10], aa[5]);
+ tmp += MUL64(corr[11], aa[6]);
+
+ tmp += MUL64(corr[12], aa[7]);
+ tmp += MUL64(corr[13], aa[8]);
+
+ tmp += MUL64(corr[14], aa[9]);
+
+ tmp = norm__(tmp, 26);
+
+ err += tmp;
+
+ return llabs(err);
+}
+
+static int64_t find_best_filter(const DCAADPCMEncContext *s, const int32_t *in, int len)
+{
+ const premultiplied_coeffs *precalc_data = s->private_data;
+ int i, j, k = 0;
+ int vq;
+ int64_t err;
+ int64_t min_err = 1ll << 62;
+ int64_t corr[15];
+
+ for (i = 0; i <= DCA_ADPCM_COEFFS; i++)
+ for (j = i; j <= DCA_ADPCM_COEFFS; j++)
+ corr[k++] = calc_corr(in+4, len, i, j);
+
+ for (i = 0; i < DCA_ADPCM_VQCODEBOOK_SZ; i++) {
+ err = apply_filter(ff_dca_adpcm_vb[i], corr, *precalc_data);
+ if (err < min_err) {
+ min_err = err;
+ vq = i;
+ }
+ precalc_data++;
+ }
+
+ return vq;
+}
+
+static inline int64_t calc_prediction_gain(int pred_vq, const int32_t *in, int32_t *out, int len)
+{
+ int i;
+ int32_t error;
+
+ int64_t signal_energy = 0;
+ int64_t error_energy = 0;
+
+ for (i = 0; i < len; i++) {
+ error = in[DCA_ADPCM_COEFFS + i] - ff_dcaadpcm_predict(pred_vq, in + i);
+ out[i] = error;
+ signal_energy += MUL64(in[DCA_ADPCM_COEFFS + i], in[DCA_ADPCM_COEFFS + i]);
+ error_energy += MUL64(error, error);
+ }
+
+ if (!error_energy)
+ return -1;
+
+ return signal_energy / error_energy;
+}
+
+int ff_dcaadpcm_subband_analysis(const DCAADPCMEncContext *s, const int32_t *in, int len, int *diff)
+{
+ int pred_vq, i;
+ int32_t input_buffer[16 + DCA_ADPCM_COEFFS];
+ int32_t input_buffer2[16 + DCA_ADPCM_COEFFS];
+
+ int32_t max = 0;
+ int shift_bits;
+ uint64_t pg = 0;
+
+ for (i = 0; i < len + DCA_ADPCM_COEFFS; i++)
+ max |= FFABS(in[i]);
+
+ // normalize input to simplify apply_filter
+ shift_bits = av_log2(max) - 11;
+
+ for (i = 0; i < len + DCA_ADPCM_COEFFS; i++) {
+ input_buffer[i] = norm__(in[i], 7);
+ input_buffer2[i] = norm__(in[i], shift_bits);
+ }
+
+ pred_vq = find_best_filter(s, input_buffer2, len);
+
+ if (pred_vq < 0)
+ return -1;
+
+ pg = calc_prediction_gain(pred_vq, input_buffer, diff, len);
+
+ // Greater than 10db (10*log(10)) prediction gain to use ADPCM.
+ // TODO: Tune it.
+ if (pg < 10)
+ return -1;
+
+ for (i = 0; i < len; i++)
+ diff[i] <<= 7;
+
+ return pred_vq;
+}
+
+static void precalc(premultiplied_coeffs *data)
+{
+ int i, j, k;
+
+ for (i = 0; i < DCA_ADPCM_VQCODEBOOK_SZ; i++) {
+ int id = 0;
+ int32_t t = 0;
+ for (j = 0; j < DCA_ADPCM_COEFFS; j++) {
+ for (k = j; k < DCA_ADPCM_COEFFS; k++) {
+ t = (int32_t)ff_dca_adpcm_vb[i][j] * (int32_t)ff_dca_adpcm_vb[i][k];
+ if (j != k)
+ t *= 2;
+ (*data)[id++] = t;
+ }
+ }
+ data++;
+ }
+}
+
+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)
+{
+ int i;
+ int64_t delta;
+ int32_t dequant_delta;
+ int32_t work_bufer[16 + DCA_ADPCM_COEFFS];
+
+ memcpy(work_bufer, prev_hist, sizeof(int32_t) * DCA_ADPCM_COEFFS);
+
+ for (i = 0; i < len; i++) {
+ work_bufer[DCA_ADPCM_COEFFS + i] = ff_dcaadpcm_predict(pred_vq_index, &work_bufer[i]);
+
+ delta = (int64_t)in[i] - ((int64_t)work_bufer[DCA_ADPCM_COEFFS + i] << 7);
+
+ out[i] = quantize_value(av_clip64(delta, -peak, peak), quant);
+
+ ff_dca_core_dequantize(&dequant_delta, &out[i], step_size, scale_factor, 0, 1);
+
+ work_bufer[DCA_ADPCM_COEFFS+i] += dequant_delta;
+ }
+
+ memcpy(next_hist, &work_bufer[len], sizeof(int32_t) * DCA_ADPCM_COEFFS);
+
+ return 0;
+}
+
+av_cold int ff_dcaadpcm_init(DCAADPCMEncContext *s)
+{
+ if (!s)
+ return -1;
+
+ s->private_data = av_malloc(sizeof(premultiplied_coeffs) * DCA_ADPCM_VQCODEBOOK_SZ);
+ precalc(s->private_data);
+ return 0;
+}
+
+av_cold void ff_dcaadpcm_free(DCAADPCMEncContext *s)
+{
+ if (!s)
+ return;
+
+ av_freep(&s->private_data);
+}
diff --git a/libavcodec/dcaadpcm.h b/libavcodec/dcaadpcm.h
new file mode 100644
index 0000000000..23bfa79636
--- /dev/null
+++ b/libavcodec/dcaadpcm.h
@@ -0,0 +1,54 @@
+/*
+ * DCA ADPCM engine
+ * Copyright (C) 2017 Daniil Cherednik
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_DCAADPCM_H
+#define AVCODEC_DCAADPCM_H
+
+#include "dcamath.h"
+#include "dcadata.h"
+#include "dcaenc.h"
+
+typedef struct DCAADPCMEncContext {
+ void *private_data;
+} DCAADPCMEncContext;
+
+static inline int64_t ff_dcaadpcm_predict(int pred_vq_index, const int32_t *input)
+{
+ int i;
+ const int16_t *coeff = ff_dca_adpcm_vb[pred_vq_index];
+ int64_t pred = 0;
+ for (i = 0; i < DCA_ADPCM_COEFFS; i++)
+ pred += (int64_t)input[DCA_ADPCM_COEFFS - 1 - i] * coeff[i];
+
+ return clip23(norm13(pred));
+}
+
+int ff_dcaadpcm_subband_analysis(const DCAADPCMEncContext *s, const int32_t *input, int len, int *diff);
+
+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);
+
+av_cold int ff_dcaadpcm_init(DCAADPCMEncContext *s);
+av_cold void ff_dcaadpcm_free(DCAADPCMEncContext *s);
+
+#endif /* AVCODEC_DCAADPCM_H */
diff --git a/libavcodec/dcadata.c b/libavcodec/dcadata.c
index 193247b18b..eaef01875a 100644
--- a/libavcodec/dcadata.c
+++ b/libavcodec/dcadata.c
@@ -61,7 +61,7 @@ const uint8_t ff_dca_quant_index_group_size[DCA_CODE_BOOKS] = {
/* ADPCM data */
/* 16 bits signed fractional Q13 binary codes */
-const int16_t ff_dca_adpcm_vb[4096][4] = {
+const int16_t ff_dca_adpcm_vb[DCA_ADPCM_VQCODEBOOK_SZ][DCA_ADPCM_COEFFS] = {
{ 9928, -2618, -1093, -1263 },
{ 11077, -2876, -1747, -308 },
{ 10503, -1082, -1426, -1167 },
diff --git a/libavcodec/dcadata.h b/libavcodec/dcadata.h
index c838867bff..9dd6eba7f1 100644
--- a/libavcodec/dcadata.h
+++ b/libavcodec/dcadata.h
@@ -25,6 +25,9 @@
#include "dcahuff.h"
+#define DCA_ADPCM_COEFFS 4
+#define DCA_ADPCM_VQCODEBOOK_SZ 4096
+
extern const uint32_t ff_dca_bit_rates[32];
extern const uint8_t ff_dca_channels[16];
@@ -36,7 +39,7 @@ extern const uint8_t ff_dca_dmix_primary_nch[8];
extern const uint8_t ff_dca_quant_index_sel_nbits[DCA_CODE_BOOKS];
extern const uint8_t ff_dca_quant_index_group_size[DCA_CODE_BOOKS];
-extern const int16_t ff_dca_adpcm_vb[4096][4];
+extern const int16_t ff_dca_adpcm_vb[DCA_ADPCM_VQCODEBOOK_SZ][DCA_ADPCM_COEFFS];
extern const uint32_t ff_dca_scale_factor_quant6[64];
extern const uint32_t ff_dca_scale_factor_quant7[128];
diff --git a/libavcodec/dcaenc.c b/libavcodec/dcaenc.c
index 3c5c33cda2..3af0ec1e75 100644
--- a/libavcodec/dcaenc.c
+++ b/libavcodec/dcaenc.c
@@ -25,8 +25,12 @@
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/ffmath.h"
+#include "libavutil/opt.h"
#include "avcodec.h"
#include "dca.h"
+#include "dcaadpcm.h"
+#include "dcamath.h"
+#include "dca_core.h"
#include "dcadata.h"
#include "dcaenc.h"
#include "internal.h"
@@ -44,8 +48,15 @@
#define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)
#define AUBANDS 25
+typedef struct CompressionOptions {
+ int adpcm_mode;
+} CompressionOptions;
+
typedef struct DCAEncContext {
+ AVClass *class;
PutBitContext pb;
+ DCAADPCMEncContext adpcm_ctx;
+ CompressionOptions options;
int frame_size;
int frame_bits;
int fullband_channels;
@@ -61,10 +72,13 @@ typedef struct DCAEncContext {
int32_t lfe_peak_cb;
const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
+ int32_t prediction_mode[MAX_CHANNELS][DCAENC_SUBBANDS];
+ int32_t adpcm_history[MAX_CHANNELS][DCAENC_SUBBANDS][DCA_ADPCM_COEFFS * 2];
int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
- int32_t subband[MAX_CHANNELS][DCAENC_SUBBANDS][SUBBAND_SAMPLES];
+ int32_t *subband[MAX_CHANNELS][DCAENC_SUBBANDS];
int32_t quantized[MAX_CHANNELS][DCAENC_SUBBANDS][SUBBAND_SAMPLES];
int32_t peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS];
+ int32_t diff_peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS]; ///< expected peak of residual signal
int32_t downsampled_lfe[DCA_LFE_SAMPLES];
int32_t masking_curve_cb[SUBSUBFRAMES][256];
int32_t bit_allocation_sel[MAX_CHANNELS];
@@ -77,6 +91,7 @@ typedef struct DCAEncContext {
int32_t worst_quantization_noise;
int32_t worst_noise_ever;
int consumed_bits;
+ int consumed_adpcm_bits; ///< Number of bits to transmit ADPCM related info
} DCAEncContext;
static int32_t cos_table[2048];
@@ -107,18 +122,52 @@ static double gammafilter(int i, double f)
return 20 * log10(h);
}
+static int subband_bufer_alloc(DCAEncContext *c)
+{
+ int ch, band;
+ int32_t *bufer = av_calloc(MAX_CHANNELS * DCAENC_SUBBANDS *
+ (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS),
+ sizeof(int32_t));
+ if (!bufer)
+ return -1;
+
+ /* we need a place for DCA_ADPCM_COEFF samples from previous frame
+ * to calc prediction coefficients for each subband */
+ for (ch = 0; ch < MAX_CHANNELS; ch++) {
+ for (band = 0; band < DCAENC_SUBBANDS; band++) {
+ c->subband[ch][band] = bufer +
+ ch * DCAENC_SUBBANDS * (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS) +
+ band * (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS) + DCA_ADPCM_COEFFS;
+ }
+ }
+ return 0;
+}
+
+static void subband_bufer_free(DCAEncContext *c)
+{
+ int32_t *bufer = c->subband[0][0] - DCA_ADPCM_COEFFS;
+ av_freep(&bufer);
+}
+
static int encode_init(AVCodecContext *avctx)
{
DCAEncContext *c = avctx->priv_data;
uint64_t layout = avctx->channel_layout;
int i, j, min_frame_bits;
+ if (subband_bufer_alloc(c))
+ return AVERROR(ENOMEM);
+
c->fullband_channels = c->channels = avctx->channels;
c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
c->band_interpolation = band_interpolation[1];
c->band_spectrum = band_spectrum[1];
c->worst_quantization_noise = -2047;
c->worst_noise_ever = -2047;
+ c->consumed_adpcm_bits = 0;
+
+ if (ff_dcaadpcm_init(&c->adpcm_ctx))
+ return AVERROR(ENOMEM);
if (!layout) {
av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
@@ -150,6 +199,12 @@ static int encode_init(AVCodecContext *avctx)
}
/* 6 - no Huffman */
c->bit_allocation_sel[i] = 6;
+
+ for (j = 0; j < DCAENC_SUBBANDS; j++) {
+ /* -1 - no ADPCM */
+ c->prediction_mode[i][j] = -1;
+ memset(c->adpcm_history[i][j], 0, sizeof(int32_t)*DCA_ADPCM_COEFFS);
+ }
}
for (i = 0; i < 9; i++) {
@@ -238,6 +293,16 @@ static int encode_init(AVCodecContext *avctx)
return 0;
}
+static av_cold int encode_close(AVCodecContext *avctx)
+{
+ if (avctx->priv_data) {
+ DCAEncContext *c = avctx->priv_data;
+ subband_bufer_free(c);
+ ff_dcaadpcm_free(&c->adpcm_ctx);
+ }
+ return 0;
+}
+
static inline int32_t cos_t(int x)
{
return cos_table[x & 2047];
@@ -253,12 +318,6 @@ static inline int32_t half32(int32_t a)
return (a + 1) >> 1;
}
-static inline int32_t mul32(int32_t a, int32_t b)
-{
- int64_t r = (int64_t)a * b + 0x80000000ULL;
- return r >> 32;
-}
-
static void subband_transform(DCAEncContext *c, const int32_t *input)
{
int ch, subs, i, k, j;
@@ -545,31 +604,53 @@ static void calc_masking(DCAEncContext *c, const int32_t *input)
}
}
+static inline int32_t find_peak(const int32_t *in, int len) {
+ int sample;
+ int32_t m = 0;
+ for (sample = 0; sample < len; sample++) {
+ int32_t s = abs(in[sample]);
+ if (m < s) {
+ m = s;
+ }
+ }
+ return get_cb(m);
+}
+
static void find_peaks(DCAEncContext *c)
{
int band, ch;
- 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[ch][band][sample]);
- if (m < s)
- m = s;
- }
- c->peak_cb[ch][band] = get_cb(m);
+ c->peak_cb[ch][band] = find_peak(c->subband[ch][band], SUBBAND_SAMPLES);
}
+ }
if (c->lfe_channel) {
- int sample;
- int32_t m = 0;
+ c->lfe_peak_cb = find_peak(c->downsampled_lfe, DCA_LFE_SAMPLES);
+ }
+}
+
+static void adpcm_analysis(DCAEncContext *c)
+{
+ int ch, band;
+ int pred_vq_id;
+ int32_t *samples;
+ int32_t estimated_diff[SUBBAND_SAMPLES];
- for (sample = 0; sample < DCA_LFE_SAMPLES; sample++)
- if (m < abs(c->downsampled_lfe[sample]))
- m = abs(c->downsampled_lfe[sample]);
- c->lfe_peak_cb = get_cb(m);
+ c->consumed_adpcm_bits = 0;
+ for (ch = 0; ch < c->fullband_channels; ch++) {
+ for (band = 0; band < 32; band++) {
+ samples = c->subband[ch][band] - DCA_ADPCM_COEFFS;
+ pred_vq_id = ff_dcaadpcm_subband_analysis(&c->adpcm_ctx, samples, SUBBAND_SAMPLES, estimated_diff);
+ if (pred_vq_id >= 0) {
+ c->prediction_mode[ch][band] = pred_vq_id;
+ c->consumed_adpcm_bits += 12; //12 bits to transmit prediction vq index
+ c->diff_peak_cb[ch][band] = find_peak(estimated_diff, 16);
+ } else {
+ c->prediction_mode[ch][band] = -1;
+ }
+ }
}
}
@@ -578,13 +659,16 @@ static const int snr_fudge = 128;
#define USED_NABITS 2
#define USED_26ABITS 4
-static int32_t quantize_value(int32_t value, softfloat quant)
+static inline int32_t get_step_size(const DCAEncContext *c, int ch, int band)
{
- int32_t offset = 1 << (quant.e - 1);
+ int32_t step_size;
- value = mul32(value, quant.m) + offset;
- value = value >> quant.e;
- return value;
+ if (c->bitrate_index == 3)
+ step_size = ff_dca_lossless_quant[c->abits[ch][band]];
+ else
+ step_size = ff_dca_lossy_quant[c->abits[ch][band]];
+
+ return step_size;
}
static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant)
@@ -619,14 +703,40 @@ static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant)
return our_nscale;
}
-static void quantize_all(DCAEncContext *c)
+static inline void quantize_adpcm_subband(DCAEncContext *c, int ch, int band)
+{
+ int32_t step_size;
+ int32_t diff_peak_cb = c->diff_peak_cb[ch][band];
+ c->scale_factor[ch][band] = calc_one_scale(diff_peak_cb,
+ c->abits[ch][band],
+ &c->quant[ch][band]);
+
+ step_size = get_step_size(c, ch, band);
+ ff_dcaadpcm_do_real(c->prediction_mode[ch][band],
+ c->quant[ch][band], ff_dca_scale_factor_quant7[c->scale_factor[ch][band]], step_size,
+ c->adpcm_history[ch][band], c->subband[ch][band], c->adpcm_history[ch][band]+4, c->quantized[ch][band],
+ SUBBAND_SAMPLES, cb_to_level[-diff_peak_cb]);
+}
+
+static void quantize_adpcm(DCAEncContext *c)
+{
+ int band, ch;
+
+ for (ch = 0; ch < c->fullband_channels; ch++)
+ for (band = 0; band < 32; band++)
+ if (c->prediction_mode[ch][band] >= 0)
+ quantize_adpcm_subband(c, ch, band);
+}
+
+static void quantize_pcm(DCAEncContext *c)
{
int sample, band, ch;
for (ch = 0; ch < c->fullband_channels; 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]);
+ if (c->prediction_mode[ch][band] == -1)
+ 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 accumulate_huff_bit_consumption(int abits, int32_t *quantized, uint32_t *result)
@@ -710,6 +820,7 @@ static int init_quantization_noise(DCAEncContext *c, int noise)
uint32_t bits_counter = 0;
c->consumed_bits = 132 + 333 * c->fullband_channels;
+ c->consumed_bits += c->consumed_adpcm_bits;
if (c->lfe_channel)
c->consumed_bits += 72;
@@ -740,12 +851,15 @@ static int init_quantization_noise(DCAEncContext *c, int noise)
/* TODO: May be cache scaled values */
for (ch = 0; ch < c->fullband_channels; 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->prediction_mode[ch][band] == -1) {
+ c->scale_factor[ch][band] = calc_one_scale(c->peak_cb[ch][band],
+ c->abits[ch][band],
+ &c->quant[ch][band]);
+ }
}
}
- quantize_all(c);
+ quantize_adpcm(c);
+ quantize_pcm(c);
memset(huff_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * 7 * sizeof(uint32_t));
memset(clc_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * sizeof(uint32_t));
@@ -819,6 +933,41 @@ static void shift_history(DCAEncContext *c, const int32_t *input)
}
}
+static void fill_in_adpcm_bufer(DCAEncContext *c)
+{
+ int ch, band;
+ int32_t step_size;
+ /* We fill in ADPCM work buffer for subbands which hasn't been ADPCM coded
+ * in current frame - we need this data if subband of next frame is
+ * ADPCM
+ */
+ for (ch = 0; ch < c->channels; ch++) {
+ for (band = 0; band < 32; band++) {
+ int32_t *samples = c->subband[ch][band] - DCA_ADPCM_COEFFS;
+ if (c->prediction_mode[ch][band] == -1) {
+ step_size = get_step_size(c, ch, band);
+
+ ff_dca_core_dequantize(c->adpcm_history[ch][band],
+ c->quantized[ch][band]+12, step_size, ff_dca_scale_factor_quant7[c->scale_factor[ch][band]], 0, 4);
+ } else {
+ AV_COPY128U(c->adpcm_history[ch][band], c->adpcm_history[ch][band]+4);
+ }
+ /* Copy dequantized values for LPC analysis.
+ * It reduces artifacts in case of extreme quantization,
+ * example: in current frame abits is 1 and has no prediction flag,
+ * but end of this frame is sine like signal. In this case, if LPC analysis uses
+ * original values, likely LPC analysis returns good prediction gain, and sets prediction flag.
+ * But there are no proper value in decoder history, so likely result will be no good.
+ * Bitstream has "Predictor history flag switch", but this flag disables history for all subbands
+ */
+ samples[0] = c->adpcm_history[ch][band][0] << 7;
+ samples[1] = c->adpcm_history[ch][band][1] << 7;
+ samples[2] = c->adpcm_history[ch][band][2] << 7;
+ samples[3] = c->adpcm_history[ch][band][3] << 7;
+ }
+ }
+}
+
static void calc_lfe_scales(DCAEncContext *c)
{
if (c->lfe_channel)
@@ -1001,9 +1150,14 @@ static void put_subframe(DCAEncContext *c, int subframe)
/* Prediction mode: no ADPCM, in each channel and subband */
for (ch = 0; ch < c->fullband_channels; ch++)
for (band = 0; band < DCAENC_SUBBANDS; band++)
- put_bits(&c->pb, 1, 0);
+ put_bits(&c->pb, 1, !(c->prediction_mode[ch][band] == -1));
+
+ /* Prediction VQ address */
+ for (ch = 0; ch < c->fullband_channels; ch++)
+ for (band = 0; band < DCAENC_SUBBANDS; band++)
+ if (c->prediction_mode[ch][band] >= 0)
+ put_bits(&c->pb, 12, c->prediction_mode[ch][band]);
- /* Prediction VQ address: not transmitted */
/* Bit allocation index */
for (ch = 0; ch < c->fullband_channels; ch++) {
if (c->bit_allocation_sel[ch] == 6) {
@@ -1068,12 +1222,15 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
lfe_downsample(c, samples);
calc_masking(c, samples);
+ if (c->options.adpcm_mode)
+ adpcm_analysis(c);
find_peaks(c);
assign_bits(c);
calc_lfe_scales(c);
shift_history(c, samples);
init_put_bits(&c->pb, avpkt->data, avpkt->size);
+ fill_in_adpcm_bufer(c);
put_frame_header(c);
put_primary_audio_header(c);
for (i = 0; i < SUBFRAMES; i++)
@@ -1092,6 +1249,20 @@ static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
return 0;
}
+#define DCAENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
+
+static const AVOption options[] = {
+ { "dca_adpcm", "Use ADPCM encoding", offsetof(DCAEncContext, options.adpcm_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DCAENC_FLAGS },
+ { NULL },
+};
+
+static const AVClass dcaenc_class = {
+ .class_name = "DCA (DTS Coherent Acoustics)",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
static const AVCodecDefault defaults[] = {
{ "b", "1411200" },
{ NULL },
@@ -1104,6 +1275,7 @@ AVCodec ff_dca_encoder = {
.id = AV_CODEC_ID_DTS,
.priv_data_size = sizeof(DCAEncContext),
.init = encode_init,
+ .close = encode_close,
.encode2 = encode_frame,
.capabilities = AV_CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
@@ -1116,4 +1288,5 @@ AVCodec ff_dca_encoder = {
AV_CH_LAYOUT_5POINT1,
0 },
.defaults = defaults,
+ .priv_class = &dcaenc_class,
};
diff --git a/libavcodec/dcaenc.h b/libavcodec/dcaenc.h
index 06816c233d..63fdaf074e 100644
--- a/libavcodec/dcaenc.h
+++ b/libavcodec/dcaenc.h
@@ -24,6 +24,8 @@
#include <stdint.h>
+#include "dcamath.h"
+
typedef struct {
int32_t m;
int32_t e;
@@ -144,4 +146,13 @@ static const int8_t channel_reorder_nolfe[16][9] = {
{ 3, 2, 4, 0, 1, 5, 7, 6, -1 },
};
+static inline int32_t quantize_value(int32_t value, softfloat quant)
+{
+ int32_t offset = 1 << (quant.e - 1);
+
+ value = mul32(value, quant.m) + offset;
+ value = value >> quant.e;
+ return value;
+}
+
#endif /* AVCODEC_DCAENC_H */
diff --git a/libavcodec/dcamath.h b/libavcodec/dcamath.h
index e0d6f4fdaa..38fa9a6235 100644
--- a/libavcodec/dcamath.h
+++ b/libavcodec/dcamath.h
@@ -49,6 +49,7 @@ static inline int32_t mul17(int32_t a, int32_t b) { return mul__(a, b, 17); }
static inline int32_t mul22(int32_t a, int32_t b) { return mul__(a, b, 22); }
static inline int32_t mul23(int32_t a, int32_t b) { return mul__(a, b, 23); }
static inline int32_t mul31(int32_t a, int32_t b) { return mul__(a, b, 31); }
+static inline int32_t mul32(int32_t a, int32_t b) { return mul__(a, b, 32); }
static inline int32_t clip23(int32_t a) { return av_clip_intp2(a, 23); }
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