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Merge commit 'aebf07075f4244caf591a3af71e5872fe314e87b'

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* commit 'aebf07075f4244caf591a3af71e5872fe314e87b':
  dca: change the core to work with integer coefficients.

Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
pull/172/head
Hendrik Leppkes 9 years ago
parent a51c2fcdc1 aebf07075f
commit af1238f863
7 changed files with 123 additions and 59 deletions
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  1. 8
      libavcodec/dca.h
  2. 111
      libavcodec/dcadec.c
  3. 34
      libavcodec/dcadsp.c
  4. 6
      libavcodec/dcadsp.h
  5. 9
      libavcodec/fmtconvert.c
  6. 10
      libavcodec/fmtconvert.h
  7. 4
      tests/fate/audio.mak

8
libavcodec/dca.h
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@ -140,8 +140,8 @@ typedef struct DCAAudioHeader {
int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book
int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select
int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
uint32_t scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
int subframes; ///< number of subframes
int total_channels; ///< number of channels including extensions
@ -149,10 +149,10 @@ typedef struct DCAAudioHeader {
} DCAAudioHeader;
typedef struct DCAChan {
DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_SUBBANDS][8];
DECLARE_ALIGNED(32, int32_t, subband_samples)[DCA_BLOCKS_MAX][DCA_SUBBANDS][8];
/* Subband samples history (for ADPCM) */
DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_SUBBANDS][4];
DECLARE_ALIGNED(32, int32_t, subband_samples_hist)[DCA_SUBBANDS][4];
int hist_index;
/* Half size is sufficient for core decoding, but for 96 kHz data

111
libavcodec/dcadec.c
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@ -214,7 +214,7 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
int xxch)
{
int i, j;
static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
static const uint8_t adj_table[4] = { 16, 18, 20, 23 };
static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
int hdr_pos = 0, hdr_size = 0;
@ -327,7 +327,7 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
/* Get scale factor adjustment */
for (j = 0; j < 11; j++)
for (i = base_channel; i < s->audio_header.prim_channels; i++)
s->audio_header.scalefactor_adj[i][j] = 1;
s->audio_header.scalefactor_adj[i][j] = 16;
for (j = 1; j < 11; j++)
for (i = base_channel; i < s->audio_header.prim_channels; i++)
@ -869,10 +869,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
{
int k, l;
int subsubframe = s->current_subsubframe;
const float *quant_step_table;
LOCAL_ALIGNED_16(int32_t, block, [SAMPLES_PER_SUBBAND * DCA_SUBBANDS]);
const uint32_t *quant_step_table;
/*
* Audio data
@ -880,13 +877,12 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
/* Select quantization step size table */
if (s->bit_rate_index == 0x1f)
quant_step_table = ff_dca_lossless_quant_d;
quant_step_table = ff_dca_lossless_quant;
else
quant_step_table = ff_dca_lossy_quant_d;
quant_step_table = ff_dca_lossy_quant;
for (k = base_channel; k < s->audio_header.prim_channels; k++) {
float (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index];
float rscale[DCA_SUBBANDS];
int32_t (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index];
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
@ -897,27 +893,25 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
/* Select the mid-tread linear quantizer */
int abits = s->dca_chan[k].bitalloc[l];
float quant_step_size = quant_step_table[abits];
/*
* Determine quantization index code book and its type
*/
/* Select quantization index code book */
int sel = s->audio_header.quant_index_huffman[k][abits];
uint32_t quant_step_size = quant_step_table[abits];
/*
* Extract bits from the bit stream
*/
if (!abits) {
rscale[l] = 0;
memset(block + SAMPLES_PER_SUBBAND * l, 0, SAMPLES_PER_SUBBAND * sizeof(block[0]));
} else {
if (!abits)
memset(subband_samples[l], 0, SAMPLES_PER_SUBBAND *
sizeof(subband_samples[l][0]));
else {
uint32_t rscale;
/* Deal with transients */
int sfi = s->dca_chan[k].transition_mode[l] &&
subsubframe >= s->dca_chan[k].transition_mode[l];
rscale[l] = quant_step_size * s->dca_chan[k].scale_factor[l][sfi] *
s->audio_header.scalefactor_adj[k][sel];
/* Determine quantization index code book and its type.
Select quantization index code book */
int sel = s->audio_header.quant_index_huffman[k][abits];
rscale = (s->dca_chan[k].scale_factor[l][sfi] *
s->audio_header.scalefactor_adj[k][sel] + 8) >> 4;
if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
if (abits <= 7) {
@ -930,7 +924,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
block_code1 = get_bits(&s->gb, size);
block_code2 = get_bits(&s->gb, size);
err = decode_blockcodes(block_code1, block_code2,
levels, block + SAMPLES_PER_SUBBAND * l);
levels, subband_samples[l]);
if (err) {
av_log(s->avctx, AV_LOG_ERROR,
"ERROR: block code look-up failed\n");
@ -939,20 +933,18 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
} else {
/* no coding */
for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
block[SAMPLES_PER_SUBBAND * l + m] = get_sbits(&s->gb, abits - 3);
subband_samples[l][m] = get_sbits(&s->gb, abits - 3);
}
} else {
/* Huffman coded */
for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
block[SAMPLES_PER_SUBBAND * l + m] = get_bitalloc(&s->gb,
&dca_smpl_bitalloc[abits], sel);
subband_samples[l][m] = get_bitalloc(&s->gb,
&dca_smpl_bitalloc[abits], sel);
}
s->dcadsp.dequantize(subband_samples[l], quant_step_size, rscale);
}
}
s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[0],
block, rscale, SAMPLES_PER_SUBBAND * s->audio_header.vq_start_subband[k]);
for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
int m;
/*
@ -962,25 +954,25 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
int n;
if (s->predictor_history)
subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
s->dca_chan[k].subband_samples_hist[l][3] +
ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] *
s->dca_chan[k].subband_samples_hist[l][2] +
ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] *
s->dca_chan[k].subband_samples_hist[l][1] +
ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] *
s->dca_chan[k].subband_samples_hist[l][0]) *
(1.0f / 8192);
(int64_t)s->dca_chan[k].subband_samples_hist[l][3] +
ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] *
(int64_t)s->dca_chan[k].subband_samples_hist[l][2] +
ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] *
(int64_t)s->dca_chan[k].subband_samples_hist[l][1] +
ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] *
(int64_t)s->dca_chan[k].subband_samples_hist[l][0]) +
(1 << 12) >> 13;
for (m = 1; m < SAMPLES_PER_SUBBAND; m++) {
float sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
subband_samples[l][m - 1];
int64_t sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
(int64_t)subband_samples[l][m - 1];
for (n = 2; n <= 4; n++)
if (m >= n)
sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
subband_samples[l][m - n];
(int64_t)subband_samples[l][m - n];
else if (s->predictor_history)
sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
s->dca_chan[k].subband_samples_hist[l][m - n + 4];
subband_samples[l][m] += sum * (1.0f / 8192);
(int64_t)s->dca_chan[k].subband_samples_hist[l][m - n + 4];
subband_samples[l][m] += (int32_t)(sum + (1 << 12) >> 13);
}
}
@ -1000,11 +992,12 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
s->debug_flag |= 0x01;
}
s->dcadsp.decode_hf(subband_samples, s->dca_chan[k].high_freq_vq,
ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND,
s->dca_chan[k].scale_factor,
s->audio_header.vq_start_subband[k],
s->audio_header.subband_activity[k]);
s->dcadsp.decode_hf_int(subband_samples, s->dca_chan[k].high_freq_vq,
ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND,
s->dca_chan[k].scale_factor,
s->audio_header.vq_start_subband[k],
s->audio_header.subband_activity[k]);
}
}
@ -1024,6 +1017,8 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
int k;
if (upsample) {
LOCAL_ALIGNED(32, float, samples, [64], [SAMPLES_PER_SUBBAND]);
if (!s->qmf64_table) {
s->qmf64_table = qmf64_precompute();
if (!s->qmf64_table)
@ -1032,21 +1027,31 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
/* 64 subbands QMF */
for (k = 0; k < s->audio_header.prim_channels; k++) {
float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index];
int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
s->dca_chan[k].subband_samples[block_index];
s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
64 * SAMPLES_PER_SUBBAND);
if (s->channel_order_tab[k] >= 0)
qmf_64_subbands(s, k, subband_samples,
qmf_64_subbands(s, k, samples,
s->samples_chanptr[s->channel_order_tab[k]],
/* Upsampling needs a factor 2 here. */
M_SQRT2 / 32768.0);
}
} else {
/* 32 subbands QMF */
LOCAL_ALIGNED(32, float, samples, [32], [SAMPLES_PER_SUBBAND]);
for (k = 0; k < s->audio_header.prim_channels; k++) {
float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index];
int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
s->dca_chan[k].subband_samples[block_index];
s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
32 * SAMPLES_PER_SUBBAND);
if (s->channel_order_tab[k] >= 0)
qmf_32_subbands(s, k, subband_samples,
qmf_32_subbands(s, k, samples,
s->samples_chanptr[s->channel_order_tab[k]],
M_SQRT1_2 / 32768.0);
}

34
libavcodec/dcadsp.c
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@ -25,6 +25,7 @@
#include "libavutil/intreadwrite.h"
#include "dcadsp.h"
#include "dcamath.h"
static void decode_hf_c(float dst[DCA_SUBBANDS][8],
const int32_t vq_num[DCA_SUBBANDS],
@ -44,6 +45,21 @@ static void decode_hf_c(float dst[DCA_SUBBANDS][8],
}
}
static void decode_hf_int_c(int32_t dst[DCA_SUBBANDS][8],
const int32_t vq_num[DCA_SUBBANDS],
const int8_t hf_vq[1024][32], intptr_t vq_offset,
int32_t scale[DCA_SUBBANDS][2],
intptr_t start, intptr_t end)
{
int i, j;
for (j = start; j < end; j++) {
const int8_t *ptr = &hf_vq[vq_num[j]][vq_offset];
for (i = 0; i < 8; i++)
dst[j][i] = ptr[i] * scale[j][0] + 8 >> 4;
}
}
static inline void dca_lfe_fir(float *out, const float *in, const float *coefs,
int decifactor)
{
@ -93,6 +109,22 @@ static void dca_qmf_32_subbands(float samples_in[32][8], int sb_act,
}
}
static void dequantize_c(int32_t *samples, uint32_t step_size, uint32_t scale)
{
int64_t step = (int64_t)step_size * scale;
int shift, i;
int32_t step_scale;
if (step > (1 << 23))
shift = av_log2(step >> 23) + 1;
else
shift = 0;
step_scale = (int32_t)(step >> shift);
for (i = 0; i < 8; i++)
samples[i] = dca_clip23(dca_norm((int64_t)samples[i] * step_scale, 22 - shift));
}
static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs)
{
dca_lfe_fir(out, in, coefs, 32);
@ -109,6 +141,8 @@ av_cold void ff_dcadsp_init(DCADSPContext *s)
s->lfe_fir[1] = dca_lfe_fir1_c;
s->qmf_32_subbands = dca_qmf_32_subbands;
s->decode_hf = decode_hf_c;
s->decode_hf_int = decode_hf_int_c;
s->dequantize = dequantize_c;
if (ARCH_AARCH64)
ff_dcadsp_init_aarch64(s);

6
libavcodec/dcadsp.h
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@ -37,6 +37,12 @@ typedef struct DCADSPContext {
const int8_t hf_vq[1024][32], intptr_t vq_offset,
int32_t scale[DCA_SUBBANDS][2],
intptr_t start, intptr_t end);
void (*decode_hf_int)(int32_t dst[DCA_SUBBANDS][8],
const int32_t vq_num[DCA_SUBBANDS],
const int8_t hf_vq[1024][32], intptr_t vq_offset,
int32_t scale[DCA_SUBBANDS][2],
intptr_t start, intptr_t end);
void (*dequantize)(int32_t *samples, uint32_t step_size, uint32_t scale);
} DCADSPContext;
void ff_dcadsp_init(DCADSPContext *s);

9
libavcodec/fmtconvert.c
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@ -32,6 +32,14 @@ static void int32_to_float_fmul_scalar_c(float *dst, const int32_t *src,
dst[i] = src[i] * mul;
}
static void int32_to_float_c(float *dst, const int32_t *src, intptr_t len)
{
int i;
for (i = 0; i < len; i++)
dst[i] = (float)src[i];
}
static void int32_to_float_fmul_array8_c(FmtConvertContext *c, float *dst,
const int32_t *src, const float *mul,
int len)
@ -43,6 +51,7 @@ static void int32_to_float_fmul_array8_c(FmtConvertContext *c, float *dst,
av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
{
c->int32_to_float = int32_to_float_c;
c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_c;
c->int32_to_float_fmul_array8 = int32_to_float_fmul_array8_c;

10
libavcodec/fmtconvert.h
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@ -37,6 +37,16 @@ typedef struct FmtConvertContext {
*/
void (*int32_to_float_fmul_scalar)(float *dst, const int32_t *src,
float mul, int len);
/**
* Convert an array of int32_t to float.
* @param dst destination array of float.
* constraints: 32-byte aligned
* @param src source array of int32_t.
* constraints: 32-byte aligned
* @param len number of elements to convert.
* constraints: multiple of 8
*/
void (*int32_to_float)(float *dst, const int32_t *src, intptr_t len);
/**
* Convert an array of int32_t to float and multiply by a float value from another array,

4
tests/fate/audio.mak
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@ -24,7 +24,7 @@ fate-dca-core: REF = $(SAMPLES)/dts/dts.pcm
FATE_DCA-$(CONFIG_DTS_DEMUXER) += fate-dca-xll
fate-dca-xll: CMD = pcm -disable_xll 0 -i $(TARGET_SAMPLES)/dts/master_audio_7.1_24bit.dts
fate-dca-xll: CMP = oneoff
fate-dca-xll: REF = $(SAMPLES)/dts/master_audio_7.1_24bit.pcm
fate-dca-xll: REF = $(SAMPLES)/dts/master_audio_7.1_24bit_2.pcm
FATE_SAMPLES_AUDIO-$(CONFIG_DCA_DECODER) += $(FATE_DCA-yes)
fate-dca: $(FATE_DCA-yes)
@ -39,7 +39,7 @@ fate-dss-sp: CMD = framecrc -i $(TARGET_SAMPLES)/dss/sp.dss -frames 30
FATE_SAMPLES_AUDIO-$(call DEMDEC, DTS, DCA) += fate-dts_es
fate-dts_es: CMD = pcm -i $(TARGET_SAMPLES)/dts/dts_es.dts
fate-dts_es: CMP = oneoff
fate-dts_es: REF = $(SAMPLES)/dts/dts_es.pcm
fate-dts_es: REF = $(SAMPLES)/dts/dts_es_2.pcm
FATE_SAMPLES_AUDIO-$(call DEMDEC, AVI, IMC) += fate-imc
fate-imc: CMD = pcm -i $(TARGET_SAMPLES)/imc/imc.avi

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