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avcodec/dcaenc: Initial implementation of ADPCM encoding for DCA encoder

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This commit is contained in:
Daniil Cherednik
2017-02-20 23:22:51 +00:00
committed by Rostislav Pehlivanov
parent 5f928c5201
commit b8c2b9c392
10 changed files with 546 additions and 76 deletions
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247
libavcodec/dcaenc.c
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@@ -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);
}
}
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);
static void adpcm_analysis(DCAEncContext *c)
{
int ch, band;
int pred_vq_id;
int32_t *samples;
int32_t estimated_diff[SUBBAND_SAMPLES];
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,
};