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-rw-r--r--libavcodec/dca.h8
-rw-r--r--libavcodec/dcadec.c111
-rw-r--r--libavcodec/dcadsp.c34
-rw-r--r--libavcodec/dcadsp.h6
-rw-r--r--libavcodec/fmtconvert.c9
-rw-r--r--libavcodec/fmtconvert.h10
-rw-r--r--tests/fate/audio.mak2
7 files changed, 122 insertions, 58 deletions
diff --git a/libavcodec/dca.h b/libavcodec/dca.h
index 6548d75634..a85470d106 100644
--- a/libavcodec/dca.h
+++ b/libavcodec/dca.h
@@ -138,8 +138,8 @@ 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
@@ -147,10 +147,10 @@ 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
diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c
index aca6ed325f..399b1e5d32 100644
--- a/libavcodec/dcadec.c
+++ b/libavcodec/dcadec.c
@@ -226,7 +226,7 @@ static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
{
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 };
@@ -265,7 +265,7 @@ 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++)
@@ -790,10 +790,7 @@ 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
@@ -801,13 +798,12 @@ 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;
@@ -818,27 +814,25 @@ 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) {
@@ -851,7 +845,7 @@ 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");
@@ -860,20 +854,18 @@ 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;
/*
@@ -883,25 +875,25 @@ 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);
}
}
@@ -921,11 +913,12 @@ 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]);
+
}
}
@@ -945,6 +938,8 @@ 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)
@@ -953,21 +948,31 @@ 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);
}
diff --git a/libavcodec/dcadsp.c b/libavcodec/dcadsp.c
index 9105a4cb4b..c333f4a44c 100644
--- a/libavcodec/dcadsp.c
+++ b/libavcodec/dcadsp.c
@@ -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);
diff --git a/libavcodec/dcadsp.h b/libavcodec/dcadsp.h
index 0669128cbc..eac9de977c 100644
--- a/libavcodec/dcadsp.h
+++ b/libavcodec/dcadsp.h
@@ -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, uint64_t scale);
} DCADSPContext;
void ff_dcadsp_init(DCADSPContext *s);
diff --git a/libavcodec/fmtconvert.c b/libavcodec/fmtconvert.c
index 5e29bfa20a..f94d438ba2 100644
--- a/libavcodec/fmtconvert.c
+++ b/libavcodec/fmtconvert.c
@@ -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;
diff --git a/libavcodec/fmtconvert.h b/libavcodec/fmtconvert.h
index dd603f9967..b2c2356778 100644
--- a/libavcodec/fmtconvert.h
+++ b/libavcodec/fmtconvert.h
@@ -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,
diff --git a/tests/fate/audio.mak b/tests/fate/audio.mak
index bc107c5017..cf11e9d3c3 100644
--- a/tests/fate/audio.mak
+++ b/tests/fate/audio.mak
@@ -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 = $(SAMPLES)/dts/master_audio_7.1_24bit_2.pcm
FATE_SAMPLES_AVCONV-$(CONFIG_DCA_DECODER) += $(FATE_DCA-yes)
fate-dca: $(FATE_DCA-yes)
generated by cgit v1.2.3 (git 2.39.1) at 2024-05-17 02:34:45 +0000