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avcodec: Implementation of AAC_fixed_decoder (PS-module)

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Add fixed point implementation.

Signed-off-by: Nedeljko Babic <nedeljko.babic@imgtec.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit is contained in:
Djordje Pesut 2015-07-20 13:36:19 +02:00 committed by Michael Niedermayer
parent 631496e057
commit 5fd81cf6f0
14 changed files with 571 additions and 350 deletions
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14
libavcodec/Makefile
View File

@ -123,12 +123,12 @@ OBJS-$(CONFIG_WMV2DSP) += wmv2dsp.o
OBJS-$(CONFIG_ZERO12V_DECODER) += 012v.o
OBJS-$(CONFIG_A64MULTI_ENCODER) += a64multienc.o elbg.o
OBJS-$(CONFIG_A64MULTI5_ENCODER) += a64multienc.o elbg.o
OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps.o \
OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps_float.o \
aacadtsdec.o mpeg4audio.o kbdwin.o \
sbrdsp.o aacpsdsp.o
OBJS-$(CONFIG_AAC_FIXED_DECODER) += aacdec_fixed.o aactab.o aacsbr_fixed.o \
sbrdsp.o aacpsdsp_float.o
OBJS-$(CONFIG_AAC_FIXED_DECODER) += aacdec_fixed.o aactab.o aacsbr_fixed.o aacps_fixed.o \
aacadtsdec.o mpeg4audio.o kbdwin.o \
sbrdsp_fixed.o
sbrdsp_fixed.o aacpsdsp_fixed.o
OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \
aacpsy.o aactab.o \
psymodel.o mpeg4audio.o kbdwin.o
@ -932,6 +932,7 @@ TOOLS = fourcc2pixfmt
HOSTPROGS = aac_tablegen \
aacps_tablegen \
aacps_fixed_tablegen \
aacsbr_tablegen \
aacsbr_fixed_tablegen \
cabac_tablegen \
@ -964,7 +965,7 @@ else
$(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=0
endif
GEN_HEADERS = cabac_tables.h cbrt_tables.h cbrt_fixed_tables.h aacps_tables.h aacsbr_tables.h \
GEN_HEADERS = cabac_tables.h cbrt_tables.h cbrt_fixed_tables.h aacps_tables.h aacps_fixed_tables.h aacsbr_tables.h \
aacsbr_fixed_tables.h aac_tables.h dsd_tables.h dv_tables.h \
sinewin_tables.h sinewin_fixed_tables.h mpegaudio_tables.h motionpixels_tables.h \
pcm_tables.h qdm2_tables.h
@ -976,7 +977,8 @@ $(GEN_HEADERS): $(SUBDIR)%_tables.h: $(SUBDIR)%_tablegen$(HOSTEXESUF)
ifdef CONFIG_HARDCODED_TABLES
$(SUBDIR)aacdec.o: $(SUBDIR)cbrt_tables.h
$(SUBDIR)aacdec_fixed.o: $(SUBDIR)cbrt_fixed_tables.h
$(SUBDIR)aacps.o: $(SUBDIR)aacps_tables.h
$(SUBDIR)aacps_float.o: $(SUBDIR)aacps_tables.h
$(SUBDIR)aacps_fixed.o: $(SUBDIR)aacps_fixed_tables.h
$(SUBDIR)aacsbr.o: $(SUBDIR)aacsbr_tables.h
$(SUBDIR)aacsbr_fixed.o: $(SUBDIR)aacsbr_fixed_tables.h
$(SUBDIR)aactab.o: $(SUBDIR)aac_tables.h

36
libavcodec/aac_defines.h
View File

@ -35,6 +35,7 @@
#define AAC_RENAME(x) x ## _fixed
#define AAC_RENAME_32(x) x ## _fixed_32
#define INTFLOAT int
#define INT64FLOAT int64_t
#define SHORTFLOAT int16_t
#define AAC_FLOAT SoftFloat
#define AAC_SIGNE int
@ -45,9 +46,33 @@
#define Q31(x) (int)((x)*2147483648.0 + 0.5)
#define RANGE15(x) x
#define GET_GAIN(x, y) (-(y) << (x)) + 1024
#define AAC_MUL16(x, y) (int)(((int64_t)(x) * (y) + 0x8000) >> 16)
#define AAC_MUL26(x, y) (int)(((int64_t)(x) * (y) + 0x2000000) >> 26)
#define AAC_MUL30(x, y) (int)(((int64_t)(x) * (y) + 0x20000000) >> 30)
#define AAC_MUL31(x, y) (int)(((int64_t)(x) * (y) + 0x40000000) >> 31)
#define AAC_MADD28(x, y, a, b) (int)((((int64_t)(x) * (y)) + \
((int64_t)(a) * (b)) + \
0x8000000) >> 28)
#define AAC_MADD30(x, y, a, b) (int)((((int64_t)(x) * (y)) + \
((int64_t)(a) * (b)) + \
0x20000000) >> 30)
#define AAC_MADD30_V8(x, y, a, b, c, d, e, f) (int)((((int64_t)(x) * (y)) + \
((int64_t)(a) * (b)) + \
((int64_t)(c) * (d)) + \
((int64_t)(e) * (f)) + \
0x20000000) >> 30)
#define AAC_MSUB30(x, y, a, b) (int)((((int64_t)(x) * (y)) - \
((int64_t)(a) * (b)) + \
0x20000000) >> 30)
#define AAC_MSUB30_V8(x, y, a, b, c, d, e, f) (int)((((int64_t)(x) * (y)) + \
((int64_t)(a) * (b)) - \
((int64_t)(c) * (d)) - \
((int64_t)(e) * (f)) + \
0x20000000) >> 30)
#define AAC_MSUB31_V3(x, y, z) (int)((((int64_t)(x) * (z)) - \
((int64_t)(y) * (z)) + \
0x40000000) >> 31)
#define AAC_HALF_SUM(x, y) (x) >> 1 + (y) >> 1
#define AAC_SRA_R(x, y) (int)(((x) + (1 << ((y) - 1))) >> (y))
#else
@ -58,6 +83,7 @@
#define AAC_RENAME(x) x
#define AAC_RENAME_32(x) x
#define INTFLOAT float
#define INT64FLOAT float
#define SHORTFLOAT float
#define AAC_FLOAT float
#define AAC_SIGNE unsigned
@ -68,9 +94,19 @@
#define Q31(x) x
#define RANGE15(x) (32768.0 * (x))
#define GET_GAIN(x, y) powf((x), -(y))
#define AAC_MUL16(x, y) ((x) * (y))
#define AAC_MUL26(x, y) ((x) * (y))
#define AAC_MUL30(x, y) ((x) * (y))
#define AAC_MUL31(x, y) ((x) * (y))
#define AAC_MADD28(x, y, a, b) ((x) * (y) + (a) * (b))
#define AAC_MADD30(x, y, a, b) ((x) * (y) + (a) * (b))
#define AAC_MADD30_V8(x, y, a, b, c, d, e, f) ((x) * (y) + (a) * (b) + \
(c) * (d) + (e) * (f))
#define AAC_MSUB30(x, y, a, b) ((x) * (y) - (a) * (b))
#define AAC_MSUB30_V8(x, y, a, b, c, d, e, f) ((x) * (y) + (a) * (b) - \
(c) * (d) - (e) * (f))
#define AAC_MSUB31_V3(x, y, z) ((x) - (y)) * (z)
#define AAC_HALF_SUM(x, y) ((x) + (y)) * 0.5f
#define AAC_SRA_R(x, y) (x)
#endif /* USE_FIXED */

255
libavcodec/aacps.c
View File

@ -17,16 +17,23 @@
* 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
*
* Note: Rounding-to-nearest used unless otherwise stated
*
*/
#include <stdint.h>
#include "libavutil/common.h"
#include "libavutil/internal.h"
#include "libavutil/mathematics.h"
#include "avcodec.h"
#include "get_bits.h"
#include "aacps.h"
#if USE_FIXED
#include "aacps_fixed_tablegen.h"
#else
#include "libavutil/internal.h"
#include "aacps_tablegen.h"
#endif /* USE_FIXED */
#include "aacpsdata.c"
#define PS_BASELINE 0 ///< Operate in Baseline PS mode
@ -148,7 +155,7 @@ static void ipdopd_reset(int8_t *ipd_hist, int8_t *opd_hist)
}
}
int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb_host, PSContext *ps, int bits_left)
int AAC_RENAME(ff_ps_read_data)(AVCodecContext *avctx, GetBitContext *gb_host, PSContext *ps, int bits_left)
{
int e;
int bit_count_start = get_bits_count(gb_host);
@ -302,35 +309,41 @@ err:
/** Split one subband into 2 subsubbands with a symmetric real filter.
* The filter must have its non-center even coefficients equal to zero. */
static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[8], int len, int reverse)
static void hybrid2_re(INTFLOAT (*in)[2], INTFLOAT (*out)[32][2], const INTFLOAT filter[8], int len, int reverse)
{
int i, j;
for (i = 0; i < len; i++, in++) {
float re_in = filter[6] * in[6][0]; //real inphase
float re_op = 0.0f; //real out of phase
float im_in = filter[6] * in[6][1]; //imag inphase
float im_op = 0.0f; //imag out of phase
INT64FLOAT re_in = AAC_MUL31(filter[6], in[6][0]); //real inphase
INT64FLOAT re_op = 0.0f; //real out of phase
INT64FLOAT im_in = AAC_MUL31(filter[6], in[6][1]); //imag inphase
INT64FLOAT im_op = 0.0f; //imag out of phase
for (j = 0; j < 6; j += 2) {
re_op += filter[j+1] * (in[j+1][0] + in[12-j-1][0]);
im_op += filter[j+1] * (in[j+1][1] + in[12-j-1][1]);
re_op += (INT64FLOAT)filter[j+1] * (in[j+1][0] + in[12-j-1][0]);
im_op += (INT64FLOAT)filter[j+1] * (in[j+1][1] + in[12-j-1][1]);
}
out[ reverse][i][0] = re_in + re_op;
out[ reverse][i][1] = im_in + im_op;
out[!reverse][i][0] = re_in - re_op;
out[!reverse][i][1] = im_in - im_op;
#if USE_FIXED
re_op = (re_op + 0x40000000) >> 31;
im_op = (im_op + 0x40000000) >> 31;
#endif /* USE_FIXED */
out[ reverse][i][0] = (INTFLOAT)(re_in + re_op);
out[ reverse][i][1] = (INTFLOAT)(im_in + im_op);
out[!reverse][i][0] = (INTFLOAT)(re_in - re_op);
out[!reverse][i][1] = (INTFLOAT)(im_in - im_op);
}
}
/** Split one subband into 6 subsubbands with a complex filter */
static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2],
TABLE_CONST float (*filter)[8][2], int len)
static void hybrid6_cx(PSDSPContext *dsp, INTFLOAT (*in)[2], INTFLOAT (*out)[32][2],
TABLE_CONST INTFLOAT (*filter)[8][2], int len)
{
int i;
int N = 8;
LOCAL_ALIGNED_16(float, temp, [8], [2]);
LOCAL_ALIGNED_16(INTFLOAT, temp, [8], [2]);
for (i = 0; i < len; i++, in++) {
dsp->hybrid_analysis(temp, in, (const float (*)[8][2]) filter, 1, N);
dsp->hybrid_analysis(temp, in, (const INTFLOAT (*)[8][2]) filter, 1, N);
out[0][i][0] = temp[6][0];
out[0][i][1] = temp[6][1];
out[1][i][0] = temp[7][0];
@ -347,18 +360,18 @@ static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2],
}
static void hybrid4_8_12_cx(PSDSPContext *dsp,
float (*in)[2], float (*out)[32][2],
TABLE_CONST float (*filter)[8][2], int N, int len)
INTFLOAT (*in)[2], INTFLOAT (*out)[32][2],
TABLE_CONST INTFLOAT (*filter)[8][2], int N, int len)
{
int i;
for (i = 0; i < len; i++, in++) {
dsp->hybrid_analysis(out[0] + i, in, (const float (*)[8][2]) filter, 32, N);
dsp->hybrid_analysis(out[0] + i, in, (const INTFLOAT (*)[8][2]) filter, 32, N);
}
}
static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
float in[5][44][2], float L[2][38][64],
static void hybrid_analysis(PSDSPContext *dsp, INTFLOAT out[91][32][2],
INTFLOAT in[5][44][2], INTFLOAT L[2][38][64],
int is34, int len)
{
int i, j;
@ -387,8 +400,8 @@ static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
}
}
static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
float in[91][32][2], int is34, int len)
static void hybrid_synthesis(PSDSPContext *dsp, INTFLOAT out[2][38][64],
INTFLOAT in[91][32][2], int is34, int len)
{
int i, n;
if (is34) {
@ -429,7 +442,7 @@ static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
}
/// All-pass filter decay slope
#define DECAY_SLOPE 0.05f
#define DECAY_SLOPE Q30(0.05f)
/// Number of frequency bands that can be addressed by the parameter index, b(k)
static const int NR_PAR_BANDS[] = { 20, 34 };
static const int NR_IPDOPD_BANDS[] = { 11, 17 };
@ -483,28 +496,43 @@ static void map_idx_34_to_20(int8_t *par_mapped, const int8_t *par, int full)
}
}
static void map_val_34_to_20(float par[PS_MAX_NR_IIDICC])
static void map_val_34_to_20(INTFLOAT par[PS_MAX_NR_IIDICC])
{
#if USE_FIXED
par[ 0] = (int)(((int64_t)(par[ 0] + (par[ 1]>>1)) * 1431655765 + \
0x40000000) >> 31);
par[ 1] = (int)(((int64_t)((par[ 1]>>1) + par[ 2]) * 1431655765 + \
0x40000000) >> 31);
par[ 2] = (int)(((int64_t)(par[ 3] + (par[ 4]>>1)) * 1431655765 + \
0x40000000) >> 31);
par[ 3] = (int)(((int64_t)((par[ 4]>>1) + par[ 5]) * 1431655765 + \
0x40000000) >> 31);
#else
par[ 0] = (2*par[ 0] + par[ 1]) * 0.33333333f;
par[ 1] = ( par[ 1] + 2*par[ 2]) * 0.33333333f;
par[ 2] = (2*par[ 3] + par[ 4]) * 0.33333333f;
par[ 3] = ( par[ 4] + 2*par[ 5]) * 0.33333333f;
par[ 4] = ( par[ 6] + par[ 7]) * 0.5f;
par[ 5] = ( par[ 8] + par[ 9]) * 0.5f;
#endif /* USE_FIXED */
par[ 4] = AAC_HALF_SUM(par[ 6], par[ 7]);
par[ 5] = AAC_HALF_SUM(par[ 8], par[ 9]);
par[ 6] = par[10];
par[ 7] = par[11];
par[ 8] = ( par[12] + par[13]) * 0.5f;
par[ 9] = ( par[14] + par[15]) * 0.5f;
par[ 8] = AAC_HALF_SUM(par[12], par[13]);
par[ 9] = AAC_HALF_SUM(par[14], par[15]);
par[10] = par[16];
par[11] = par[17];
par[12] = par[18];
par[13] = par[19];
par[14] = ( par[20] + par[21]) * 0.5f;
par[15] = ( par[22] + par[23]) * 0.5f;
par[16] = ( par[24] + par[25]) * 0.5f;
par[17] = ( par[26] + par[27]) * 0.5f;
par[14] = AAC_HALF_SUM(par[20], par[21]);
par[15] = AAC_HALF_SUM(par[22], par[23]);
par[16] = AAC_HALF_SUM(par[24], par[25]);
par[17] = AAC_HALF_SUM(par[26], par[27]);
#if USE_FIXED
par[18] = (((par[28]+2)>>2) + ((par[29]+2)>>2) + ((par[30]+2)>>2) + ((par[31]+2)>>2));
#else
par[18] = ( par[28] + par[29] + par[30] + par[31]) * 0.25f;
par[19] = ( par[32] + par[33]) * 0.5f;
#endif /* USE_FIXED */
par[19] = AAC_HALF_SUM(par[32], par[33]);
}
static void map_idx_10_to_34(int8_t *par_mapped, const int8_t *par, int full)
@ -589,7 +617,7 @@ static void map_idx_20_to_34(int8_t *par_mapped, const int8_t *par, int full)
par_mapped[ 0] = par[ 0];
}
static void map_val_20_to_34(float par[PS_MAX_NR_IIDICC])
static void map_val_20_to_34(INTFLOAT par[PS_MAX_NR_IIDICC])
{
par[33] = par[19];
par[32] = par[19];
@ -620,27 +648,29 @@ static void map_val_20_to_34(float par[PS_MAX_NR_IIDICC])
par[ 7] = par[ 4];
par[ 6] = par[ 4];
par[ 5] = par[ 3];
par[ 4] = (par[ 2] + par[ 3]) * 0.5f;
par[ 4] = AAC_HALF_SUM(par[ 2], par[ 3]);
par[ 3] = par[ 2];
par[ 2] = par[ 1];
par[ 1] = (par[ 0] + par[ 1]) * 0.5f;
par[ 1] = AAC_HALF_SUM(par[ 0], par[ 1]);
}
static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[32][2], int is34)
static void decorrelation(PSContext *ps, INTFLOAT (*out)[32][2], const INTFLOAT (*s)[32][2], int is34)
{
LOCAL_ALIGNED_16(float, power, [34], [PS_QMF_TIME_SLOTS]);
LOCAL_ALIGNED_16(float, transient_gain, [34], [PS_QMF_TIME_SLOTS]);
float *peak_decay_nrg = ps->peak_decay_nrg;
float *power_smooth = ps->power_smooth;
float *peak_decay_diff_smooth = ps->peak_decay_diff_smooth;
float (*delay)[PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2] = ps->delay;
float (*ap_delay)[PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2] = ps->ap_delay;
const int8_t *k_to_i = is34 ? k_to_i_34 : k_to_i_20;
const float peak_decay_factor = 0.76592833836465f;
LOCAL_ALIGNED_16(INTFLOAT, power, [34], [PS_QMF_TIME_SLOTS]);
LOCAL_ALIGNED_16(INTFLOAT, transient_gain, [34], [PS_QMF_TIME_SLOTS]);
INTFLOAT *peak_decay_nrg = ps->peak_decay_nrg;
INTFLOAT *power_smooth = ps->power_smooth;
INTFLOAT *peak_decay_diff_smooth = ps->peak_decay_diff_smooth;
INTFLOAT (*delay)[PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2] = ps->delay;
INTFLOAT (*ap_delay)[PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2] = ps->ap_delay;
#if !USE_FIXED
const float transient_impact = 1.5f;
const float a_smooth = 0.25f; ///< Smoothing coefficient
#endif /* USE_FIXED */
const int8_t *k_to_i = is34 ? k_to_i_34 : k_to_i_20;
int i, k, m, n;
int n0 = 0, nL = 32;
const INTFLOAT peak_decay_factor = Q31(0.76592833836465f);;
memset(power, 0, 34 * sizeof(*power));
@ -658,6 +688,33 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
}
//Transient detection
#if USE_FIXED
for (i = 0; i < NR_PAR_BANDS[is34]; i++) {
for (n = n0; n < nL; n++) {
int decayed_peak;
int denom;
decayed_peak = (int)(((int64_t)peak_decay_factor * \
peak_decay_nrg[i] + 0x40000000) >> 31);
peak_decay_nrg[i] = FFMAX(decayed_peak, power[i][n]);
power_smooth[i] += (power[i][n] - power_smooth[i] + 2) >> 2;
peak_decay_diff_smooth[i] += (peak_decay_nrg[i] - power[i][n] - \
peak_decay_diff_smooth[i] + 2) >> 2;
denom = peak_decay_diff_smooth[i] + (peak_decay_diff_smooth[i] >> 1);
if (denom > power_smooth[i]) {
int p = power_smooth[i];
while (denom < 0x40000000) {
denom <<= 1;
p <<= 1;
}
transient_gain[i][n] = p / (denom >> 16);
}
else {
transient_gain[i][n] = 1 << 16;
}
}
}
#else
for (i = 0; i < NR_PAR_BANDS[is34]; i++) {
for (n = n0; n < nL; n++) {
float decayed_peak = peak_decay_factor * peak_decay_nrg[i];
@ -671,6 +728,7 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
}
}
#endif /* USE_FIXED */
//Decorrelation and transient reduction
// PS_AP_LINKS - 1
// -----
@ -681,8 +739,22 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
//d[k][z] (out) = transient_gain_mapped[k][z] * H[k][z] * s[k][z]
for (k = 0; k < NR_ALLPASS_BANDS[is34]; k++) {
int b = k_to_i[k];
#if USE_FIXED
int g_decay_slope;
if (k - DECAY_CUTOFF[is34] <= 0) {
g_decay_slope = 1 << 30;
}
else if (k - DECAY_CUTOFF[is34] >= 20) {
g_decay_slope = 0;
}
else {
g_decay_slope = (1 << 30) - DECAY_SLOPE * (k - DECAY_CUTOFF[is34]);
}
#else
float g_decay_slope = 1.f - DECAY_SLOPE * (k - DECAY_CUTOFF[is34]);
g_decay_slope = av_clipf(g_decay_slope, 0.f, 1.f);
#endif /* USE_FIXED */
memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
for (m = 0; m < PS_AP_LINKS; m++) {
@ -690,7 +762,7 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
}
ps->dsp.decorrelate(out[k], delay[k] + PS_MAX_DELAY - 2, ap_delay[k],
phi_fract[is34][k],
(const float (*)[2]) Q_fract_allpass[is34][k],
(const INTFLOAT (*)[2]) Q_fract_allpass[is34][k],
transient_gain[b], g_decay_slope, nL - n0);
}
for (; k < SHORT_DELAY_BAND[is34]; k++) {
@ -749,14 +821,14 @@ static void remap20(int8_t (**p_par_mapped)[PS_MAX_NR_IIDICC],
}
}
static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2], int is34)
static void stereo_processing(PSContext *ps, INTFLOAT (*l)[32][2], INTFLOAT (*r)[32][2], int is34)
{
int e, b, k;
float (*H11)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H11;
float (*H12)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H12;
float (*H21)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H21;
float (*H22)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H22;
INTFLOAT (*H11)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H11;
INTFLOAT (*H12)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H12;
INTFLOAT (*H21)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H21;
INTFLOAT (*H22)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H22;
int8_t *opd_hist = ps->opd_hist;
int8_t *ipd_hist = ps->ipd_hist;
int8_t iid_mapped_buf[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC];
@ -768,7 +840,7 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
int8_t (*ipd_mapped)[PS_MAX_NR_IIDICC] = ipd_mapped_buf;
int8_t (*opd_mapped)[PS_MAX_NR_IIDICC] = opd_mapped_buf;
const int8_t *k_to_i = is34 ? k_to_i_34 : k_to_i_20;
TABLE_CONST float (*H_LUT)[8][4] = (PS_BASELINE || ps->icc_mode < 3) ? HA : HB;
TABLE_CONST INTFLOAT (*H_LUT)[8][4] = (PS_BASELINE || ps->icc_mode < 3) ? HA : HB;
//Remapping
if (ps->num_env_old) {
@ -823,7 +895,7 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
//Mixing
for (e = 0; e < ps->num_env; e++) {
for (b = 0; b < NR_PAR_BANDS[is34]; b++) {
float h11, h12, h21, h22;
INTFLOAT h11, h12, h21, h22;
h11 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][0];
h12 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][1];
h21 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][2];
@ -832,27 +904,27 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
if (!PS_BASELINE && ps->enable_ipdopd && b < NR_IPDOPD_BANDS[is34]) {
//The spec say says to only run this smoother when enable_ipdopd
//is set but the reference decoder appears to run it constantly
float h11i, h12i, h21i, h22i;
float ipd_adj_re, ipd_adj_im;
INTFLOAT h11i, h12i, h21i, h22i;
INTFLOAT ipd_adj_re, ipd_adj_im;
int opd_idx = opd_hist[b] * 8 + opd_mapped[e][b];
int ipd_idx = ipd_hist[b] * 8 + ipd_mapped[e][b];
float opd_re = pd_re_smooth[opd_idx];
float opd_im = pd_im_smooth[opd_idx];
float ipd_re = pd_re_smooth[ipd_idx];
float ipd_im = pd_im_smooth[ipd_idx];
INTFLOAT opd_re = pd_re_smooth[opd_idx];
INTFLOAT opd_im = pd_im_smooth[opd_idx];
INTFLOAT ipd_re = pd_re_smooth[ipd_idx];
INTFLOAT ipd_im = pd_im_smooth[ipd_idx];
opd_hist[b] = opd_idx & 0x3F;
ipd_hist[b] = ipd_idx & 0x3F;
ipd_adj_re = opd_re*ipd_re + opd_im*ipd_im;
ipd_adj_im = opd_im*ipd_re - opd_re*ipd_im;
h11i = h11 * opd_im;
h11 = h11 * opd_re;
h12i = h12 * ipd_adj_im;
h12 = h12 * ipd_adj_re;
h21i = h21 * opd_im;
h21 = h21 * opd_re;
h22i = h22 * ipd_adj_im;
h22 = h22 * ipd_adj_re;
ipd_adj_re = AAC_MADD30(opd_re, ipd_re, opd_im, ipd_im);
ipd_adj_im = AAC_MSUB30(opd_im, ipd_re, opd_re, ipd_im);
h11i = AAC_MUL30(h11, opd_im);
h11 = AAC_MUL30(h11, opd_re);
h12i = AAC_MUL30(h12, ipd_adj_im);
h12 = AAC_MUL30(h12, ipd_adj_re);
h21i = AAC_MUL30(h21, opd_im);
h21 = AAC_MUL30(h21, opd_re);
h22i = AAC_MUL30(h22, ipd_adj_im);
h22 = AAC_MUL30(h22, ipd_adj_re);
H11[1][e+1][b] = h11i;
H12[1][e+1][b] = h12i;
H21[1][e+1][b] = h21i;
@ -864,11 +936,14 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
H22[0][e+1][b] = h22;
}
for (k = 0; k < NR_BANDS[is34]; k++) {
float h[2][4];
float h_step[2][4];
INTFLOAT h[2][4];
INTFLOAT h_step[2][4];
int start = ps->border_position[e];
int stop = ps->border_position[e+1];
float width = 1.f / (stop - start);
INTFLOAT width = Q30(1.f) / (stop - start);
#if USE_FIXED
width <<= 1;
#endif
b = k_to_i[k];
h[0][0] = H11[0][e][b];
h[0][1] = H12[0][e][b];
@ -889,15 +964,15 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
}
}
//Interpolation
h_step[0][0] = (H11[0][e+1][b] - h[0][0]) * width;
h_step[0][1] = (H12[0][e+1][b] - h[0][1]) * width;
h_step[0][2] = (H21[0][e+1][b] - h[0][2]) * width;
h_step[0][3] = (H22[0][e+1][b] - h[0][3]) * width;
h_step[0][0] = AAC_MSUB31_V3(H11[0][e+1][b], h[0][0], width);
h_step[0][1] = AAC_MSUB31_V3(H12[0][e+1][b], h[0][1], width);
h_step[0][2] = AAC_MSUB31_V3(H21[0][e+1][b], h[0][2], width);
h_step[0][3] = AAC_MSUB31_V3(H22[0][e+1][b], h[0][3], width);
if (!PS_BASELINE && ps->enable_ipdopd) {
h_step[1][0] = (H11[1][e+1][b] - h[1][0]) * width;
h_step[1][1] = (H12[1][e+1][b] - h[1][1]) * width;
h_step[1][2] = (H21[1][e+1][b] - h[1][2]) * width;
h_step[1][3] = (H22[1][e+1][b] - h[1][3]) * width;
h_step[1][0] = AAC_MSUB31_V3(H11[1][e+1][b], h[1][0], width);
h_step[1][1] = AAC_MSUB31_V3(H12[1][e+1][b], h[1][1], width);
h_step[1][2] = AAC_MSUB31_V3(H21[1][e+1][b], h[1][2], width);
h_step[1][3] = AAC_MSUB31_V3(H22[1][e+1][b], h[1][3], width);
}
ps->dsp.stereo_interpolate[!PS_BASELINE && ps->enable_ipdopd](
l[k] + start + 1, r[k] + start + 1,
@ -906,10 +981,10 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
}
}
int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top)
int AAC_RENAME(ff_ps_apply)(AVCodecContext *avctx, PSContext *ps, INTFLOAT L[2][38][64], INTFLOAT R[2][38][64], int top)
{
float (*Lbuf)[32][2] = ps->Lbuf;
float (*Rbuf)[32][2] = ps->Rbuf;
INTFLOAT (*Lbuf)[32][2] = ps->Lbuf;
INTFLOAT (*Rbuf)[32][2] = ps->Rbuf;
const int len = 32;
int is34 = ps->is34bands;
@ -919,7 +994,7 @@ int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float
memset(ps->ap_delay + top, 0, (NR_ALLPASS_BANDS[is34] - top)*sizeof(ps->ap_delay[0]));
hybrid_analysis(&ps->dsp, Lbuf, ps->in_buf, L, is34, len);
decorrelation(ps, Rbuf, (const float (*)[32][2]) Lbuf, is34);
decorrelation(ps, Rbuf, (const INTFLOAT (*)[32][2]) Lbuf, is34);
stereo_processing(ps, Lbuf, Rbuf, is34);
hybrid_synthesis(&ps->dsp, L, Lbuf, is34, len);
hybrid_synthesis(&ps->dsp, R, Rbuf, is34, len);
@ -936,7 +1011,7 @@ int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float
#define PS_VLC_ROW(name) \
{ name ## _codes, name ## _bits, sizeof(name ## _codes), sizeof(name ## _codes[0]) }
av_cold void ff_ps_init(void) {
av_cold void AAC_RENAME(ff_ps_init)(void) {
// Syntax initialization
static const struct {
const void *ps_codes, *ps_bits;
@ -968,7 +1043,7 @@ av_cold void ff_ps_init(void) {
ps_tableinit();
}
av_cold void ff_ps_ctx_init(PSContext *ps)
av_cold void AAC_RENAME(ff_ps_ctx_init)(PSContext *ps)
{
ff_psdsp_init(&ps->dsp);
AAC_RENAME(ff_psdsp_init)(&ps->dsp);
}

32
libavcodec/aacps.h
View File

@ -61,26 +61,26 @@ typedef struct PSContext {
int is34bands;
int is34bands_old;
DECLARE_ALIGNED(16, float, in_buf)[5][44][2];
DECLARE_ALIGNED(16, float, delay)[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
DECLARE_ALIGNED(16, float, ap_delay)[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
DECLARE_ALIGNED(16, float, peak_decay_nrg)[34];
DECLARE_ALIGNED(16, float, power_smooth)[34];
DECLARE_ALIGNED(16, float, peak_decay_diff_smooth)[34];
DECLARE_ALIGNED(16, float, H11)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
DECLARE_ALIGNED(16, float, H12)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
DECLARE_ALIGNED(16, float, H21)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
DECLARE_ALIGNED(16, float, H22)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
DECLARE_ALIGNED(16, float, Lbuf)[91][32][2];
DECLARE_ALIGNED(16, float, Rbuf)[91][32][2];
DECLARE_ALIGNED(16, INTFLOAT, in_buf)[5][44][2];
DECLARE_ALIGNED(16, INTFLOAT, delay)[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
DECLARE_ALIGNED(16, INTFLOAT, ap_delay)[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
DECLARE_ALIGNED(16, INTFLOAT, peak_decay_nrg)[34];
DECLARE_ALIGNED(16, INTFLOAT, power_smooth)[34];
DECLARE_ALIGNED(16, INTFLOAT, peak_decay_diff_smooth)[34];
DECLARE_ALIGNED(16, INTFLOAT, H11)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
DECLARE_ALIGNED(16, INTFLOAT, H12)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
DECLARE_ALIGNED(16, INTFLOAT, H21)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
DECLARE_ALIGNED(16, INTFLOAT, H22)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
DECLARE_ALIGNED(16, INTFLOAT, Lbuf)[91][32][2];
DECLARE_ALIGNED(16, INTFLOAT, Rbuf)[91][32][2];
int8_t opd_hist[PS_MAX_NR_IIDICC];
int8_t ipd_hist[PS_MAX_NR_IIDICC];
PSDSPContext dsp;
} PSContext;
void ff_ps_init(void);
void ff_ps_ctx_init(PSContext *ps);
int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, int bits_left);
int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top);
void AAC_RENAME(ff_ps_init)(void);
void AAC_RENAME(ff_ps_ctx_init)(PSContext *ps);
int AAC_RENAME(ff_ps_read_data)(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, int bits_left);
int AAC_RENAME(ff_ps_apply)(AVCodecContext *avctx, PSContext *ps, INTFLOAT L[2][38][64], INTFLOAT R[2][38][64], int top);
#endif /* AVCODEC_PS_H */

24
libavcodec/aacps_fixed.c Normal file
View File

@ -0,0 +1,24 @@
/*
* MPEG-4 Parametric Stereo decoding functions
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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
*/
#define USE_FIXED 1
#include "aacps.c"

2
libavcodec/aacps_fixed_tablegen.h
View File

@ -288,7 +288,7 @@ static void ps_tableinit(void)
int im_smooth = pd0_im + pd1_im + pd2_im;
SoftFloat pd_mag = av_int2sf(((ipdopd_cos[(pd0-pd1)&7]+8)>>4) + ((ipdopd_cos[(pd0-pd2)&7]+4)>>3) +
((ipdopd_cos[(pd1-pd2)&7]+2)>>2) + 0x15000000, 2);
((ipdopd_cos[(pd1-pd2)&7]+2)>>2) + 0x15000000, 28);
pd_mag = av_div_sf(FLOAT_1, av_sqrt_sf(pd_mag));
shift = 30 - pd_mag.exp;
round = 1 << (shift-1);

24
libavcodec/aacps_float.c Normal file
View File

@ -0,0 +1,24 @@
/*
* MPEG-4 Parametric Stereo decoding functions
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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
*/
#define USE_FIXED 0
#include "aacps.c"

6
libavcodec/aacpsdata.c
View File

@ -157,7 +157,7 @@ static const int8_t k_to_i_34[] = {
33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33
};
static const float g1_Q2[] = {
0.0f, 0.01899487526049f, 0.0f, -0.07293139167538f,
0.0f, 0.30596630545168f, 0.5f
static const INTFLOAT g1_Q2[] = {
Q31(0.0f), Q31(0.01899487526049f), Q31(0.0f), Q31(-0.07293139167538f),
Q31(0.0f), Q31(0.30596630545168f), Q31(0.5f)
};

216
libavcodec/aacpsdsp.c
View File

@ -1,216 +0,0 @@
/*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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 "config.h"
#include "libavutil/attributes.h"
#include "aacpsdsp.h"
static void ps_add_squares_c(float *dst, const float (*src)[2], int n)
{
int i;
for (i = 0; i < n; i++)
dst[i] += src[i][0] * src[i][0] + src[i][1] * src[i][1];
}
static void ps_mul_pair_single_c(float (*dst)[2], float (*src0)[2], float *src1,
int n)
{
int i;
for (i = 0; i < n; i++) {
dst[i][0] = src0[i][0] * src1[i];
dst[i][1] = src0[i][1] * src1[i];
}
}
static void ps_hybrid_analysis_c(float (*out)[2], float (*in)[2],
const float (*filter)[8][2],
int stride, int n)
{
int i, j;
for (i = 0; i < n; i++) {
float sum_re = filter[i][6][0] * in[6][0];
float sum_im = filter[i][6][0] * in[6][1];
for (j = 0; j < 6; j++) {
float in0_re = in[j][0];
float in0_im = in[j][1];
float in1_re = in[12-j][0];
float in1_im = in[12-j][1];
sum_re += filter[i][j][0] * (in0_re + in1_re) -
filter[i][j][1] * (in0_im - in1_im);
sum_im += filter[i][j][0] * (in0_im + in1_im) +
filter[i][j][1] * (in0_re - in1_re);
}
out[i * stride][0] = sum_re;
out[i * stride][1] = sum_im;
}
}
static void ps_hybrid_analysis_ileave_c(float (*out)[32][2], float L[2][38][64],
int i, int len)
{
int j;
for (; i < 64; i++) {
for (j = 0; j < len; j++) {
out[i][j][0] = L[0][j][i];
out[i][j][1] = L[1][j][i];
}
}
}
static void ps_hybrid_synthesis_deint_c(float out[2][38][64],
float (*in)[32][2],
int i, int len)
{
int n;
for (; i < 64; i++) {
for (n = 0; n < len; n++) {
out[0][n][i] = in[i][n][0];
out[1][n][i] = in[i][n][1];
}
}
}
static void ps_decorrelate_c(float (*out)[2], float (*delay)[2],
float (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2],
const float phi_fract[2], const float (*Q_fract)[2],
const float *transient_gain,
float g_decay_slope,
int len)
{
static const float a[] = { 0.65143905753106f,
0.56471812200776f,
0.48954165955695f };
float ag[PS_AP_LINKS];
int m, n;
for (m = 0; m < PS_AP_LINKS; m++)
ag[m] = a[m] * g_decay_slope;
for (n = 0; n < len; n++) {
float in_re = delay[n][0] * phi_fract[0] - delay[n][1] * phi_fract[1];
float in_im = delay[n][0] * phi_fract[1] + delay[n][1] * phi_fract[0];
for (m = 0; m < PS_AP_LINKS; m++) {
float a_re = ag[m] * in_re;
float a_im = ag[m] * in_im;
float link_delay_re = ap_delay[m][n+2-m][0];
float link_delay_im = ap_delay[m][n+2-m][1];
float fractional_delay_re = Q_fract[m][0];
float fractional_delay_im = Q_fract[m][1];
float apd_re = in_re;
float apd_im = in_im;
in_re = link_delay_re * fractional_delay_re -
link_delay_im * fractional_delay_im - a_re;
in_im = link_delay_re * fractional_delay_im +
link_delay_im * fractional_delay_re - a_im;
ap_delay[m][n+5][0] = apd_re + ag[m] * in_re;
ap_delay[m][n+5][1] = apd_im + ag[m] * in_im;
}
out[n][0] = transient_gain[n] * in_re;
out[n][1] = transient_gain[n] * in_im;
}
}
static void ps_stereo_interpolate_c(float (*l)[2], float (*r)[2],
float h[2][4], float h_step[2][4],
int len)
{
float h0 = h[0][0];
float h1 = h[0][1];
float h2 = h[0][2];
float h3 = h[0][3];
float hs0 = h_step[0][0];
float hs1 = h_step[0][1];
float hs2 = h_step[0][2];
float hs3 = h_step[0][3];
int n;
for (n = 0; n < len; n++) {
//l is s, r is d
float l_re = l[n][0];
float l_im = l[n][1];
float r_re = r[n][0];
float r_im = r[n][1];
h0 += hs0;
h1 += hs1;
h2 += hs2;
h3 += hs3;
l[n][0] = h0 * l_re + h2 * r_re;
l[n][1] = h0 * l_im + h2 * r_im;
r[n][0] = h1 * l_re + h3 * r_re;
r[n][1] = h1 * l_im + h3 * r_im;
}
}
static void ps_stereo_interpolate_ipdopd_c(float (*l)[2], float (*r)[2],
float h[2][4], float h_step[2][4],
int len)
{
float h00 = h[0][0], h10 = h[1][0];
float h01 = h[0][1], h11 = h[1][1];
float h02 = h[0][2], h12 = h[1][2];
float h03 = h[0][3], h13 = h[1][3];
float hs00 = h_step[0][0], hs10 = h_step[1][0];
float hs01 = h_step[0][1], hs11 = h_step[1][1];
float hs02 = h_step[0][2], hs12 = h_step[1][2];
float hs03 = h_step[0][3], hs13 = h_step[1][3];
int n;
for (n = 0; n < len; n++) {
//l is s, r is d
float l_re = l[n][0];
float l_im = l[n][1];
float r_re = r[n][0];
float r_im = r[n][1];
h00 += hs00;
h01 += hs01;
h02 += hs02;
h03 += hs03;
h10 += hs10;
h11 += hs11;
h12 += hs12;
h13 += hs13;
l[n][0] = h00 * l_re + h02 * r_re - h10 * l_im - h12 * r_im;
l[n][1] = h00 * l_im + h02 * r_im + h10 * l_re + h12 * r_re;
r[n][0] = h01 * l_re + h03 * r_re - h11 * l_im - h13 * r_im;
r[n][1] = h01 * l_im + h03 * r_im + h11 * l_re + h13 * r_re;
}
}
av_cold void ff_psdsp_init(PSDSPContext *s)
{
s->add_squares = ps_add_squares_c;
s->mul_pair_single = ps_mul_pair_single_c;
s->hybrid_analysis = ps_hybrid_analysis_c;
s->hybrid_analysis_ileave = ps_hybrid_analysis_ileave_c;
s->hybrid_synthesis_deint = ps_hybrid_synthesis_deint_c;
s->decorrelate = ps_decorrelate_c;
s->stereo_interpolate[0] = ps_stereo_interpolate_c;
s->stereo_interpolate[1] = ps_stereo_interpolate_ipdopd_c;
if (ARCH_ARM)
ff_psdsp_init_arm(s);
if (ARCH_MIPS)
ff_psdsp_init_mips(s);
}

30
libavcodec/aacpsdsp.h
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@ -21,33 +21,35 @@
#ifndef LIBAVCODEC_AACPSDSP_H
#define LIBAVCODEC_AACPSDSP_H
#include "aac_defines.h"
#define PS_QMF_TIME_SLOTS 32
#define PS_AP_LINKS 3
#define PS_MAX_AP_DELAY 5
typedef struct PSDSPContext {
void (*add_squares)(float *dst, const float (*src)[2], int n);
void (*mul_pair_single)(float (*dst)[2], float (*src0)[2], float *src1,
void (*add_squares)(INTFLOAT *dst, const INTFLOAT (*src)[2], int n);
void (*mul_pair_single)(INTFLOAT (*dst)[2], INTFLOAT (*src0)[2], INTFLOAT *src1,
int n);
void (*hybrid_analysis)(float (*out)[2], float (*in)[2],
const float (*filter)[8][2],
void (*hybrid_analysis)(INTFLOAT (*out)[2], INTFLOAT (*in)[2],
const INTFLOAT (*filter)[8][2],
int stride, int n);
void (*hybrid_analysis_ileave)(float (*out)[32][2], float L[2][38][64],
void (*hybrid_analysis_ileave)(INTFLOAT (*out)[32][2], INTFLOAT L[2][38][64],
int i, int len);
void (*hybrid_synthesis_deint)(float out[2][38][64], float (*in)[32][2],
void (*hybrid_synthesis_deint)(INTFLOAT out[2][38][64], INTFLOAT (*in)[32][2],
int i, int len);
void (*decorrelate)(float (*out)[2], float (*delay)[2],
float (*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2],
const float phi_fract[2], const float (*Q_fract)[2],
const float *transient_gain,
float g_decay_slope,
void (*decorrelate)(INTFLOAT (*out)[2], INTFLOAT (*delay)[2],
INTFLOAT (*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2],
const INTFLOAT phi_fract[2], const INTFLOAT (*Q_fract)[2],
const INTFLOAT *transient_gain,
INTFLOAT g_decay_slope,
int len);
void (*stereo_interpolate[2])(float (*l)[2], float (*r)[2],
float h[2][4], float h_step[2][4],
void (*stereo_interpolate[2])(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
INTFLOAT h[2][4], INTFLOAT h_step[2][4],
int len);
} PSDSPContext;
void ff_psdsp_init(PSDSPContext *s);
void AAC_RENAME(ff_psdsp_init)(PSDSPContext *s);
void ff_psdsp_init_arm(PSDSPContext *s);
void ff_psdsp_init_mips(PSDSPContext *s);

23
libavcodec/aacpsdsp_fixed.c Normal file
View File

@ -0,0 +1,23 @@
/*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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
*/
#define USE_FIXED 1
#include "aacpsdsp_template.c"

23
libavcodec/aacpsdsp_float.c Normal file
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@ -0,0 +1,23 @@
/*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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
*/
#define USE_FIXED 0
#include "aacpsdsp_template.c"

228
libavcodec/aacpsdsp_template.c Normal file
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@ -0,0 +1,228 @@
/*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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
*
* Note: Rounding-to-nearest used unless otherwise stated
*
*/
#include <stdint.h>
#include "config.h"
#include "libavutil/attributes.h"
#include "aacpsdsp.h"
static void ps_add_squares_c(INTFLOAT *dst, const INTFLOAT (*src)[2], int n)
{
int i;
for (i = 0; i < n; i++)
dst[i] += AAC_MADD28(src[i][0], src[i][0], src[i][1], src[i][1]);
}
static void ps_mul_pair_single_c(INTFLOAT (*dst)[2], INTFLOAT (*src0)[2], INTFLOAT *src1,
int n)
{
int i;
for (i = 0; i < n; i++) {
dst[i][0] = AAC_MUL16(src0[i][0], src1[i]);
dst[i][1] = AAC_MUL16(src0[i][1], src1[i]);
}
}
static void ps_hybrid_analysis_c(INTFLOAT (*out)[2], INTFLOAT (*in)[2],
const INTFLOAT (*filter)[8][2],
int stride, int n)
{
int i, j;
for (i = 0; i < n; i++) {
INT64FLOAT sum_re = (INT64FLOAT)filter[i][6][0] * in[6][0];
INT64FLOAT sum_im = (INT64FLOAT)filter[i][6][0] * in[6][1];
for (j = 0; j < 6; j++) {
INTFLOAT in0_re = in[j][0];
INTFLOAT in0_im = in[j][1];
INTFLOAT in1_re = in[12-j][0];
INTFLOAT in1_im = in[12-j][1];
sum_re += (INT64FLOAT)filter[i][j][0] * (in0_re + in1_re) -
(INT64FLOAT)filter[i][j][1] * (in0_im - in1_im);
sum_im += (INT64FLOAT)filter[i][j][0] * (in0_im + in1_im) +
(INT64FLOAT)filter[i][j][1] * (in0_re - in1_re);
}
#if USE_FIXED
out[i * stride][0] = (int)((sum_re + 0x40000000) >> 31);
out[i * stride][1] = (int)((sum_im + 0x40000000) >> 31);
#else
out[i * stride][0] = sum_re;
out[i * stride][1] = sum_im;
#endif /* USE_FIXED */
}
}
static void ps_hybrid_analysis_ileave_c(INTFLOAT (*out)[32][2], INTFLOAT L[2][38][64],
int i, int len)
{
int j;
for (; i < 64; i++) {
for (j = 0; j < len; j++) {
out[i][j][0] = L[0][j][i];
out[i][j][1] = L[1][j][i];
}
}
}
static void ps_hybrid_synthesis_deint_c(INTFLOAT out[2][38][64],
INTFLOAT (*in)[32][2],
int i, int len)
{
int n;
for (; i < 64; i++) {
for (n = 0; n < len; n++) {
out[0][n][i] = in[i][n][0];
out[1][n][i] = in[i][n][1];
}
}
}
static void ps_decorrelate_c(INTFLOAT (*out)[2], INTFLOAT (*delay)[2],
INTFLOAT (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2],
const INTFLOAT phi_fract[2], const INTFLOAT (*Q_fract)[2],
const INTFLOAT *transient_gain,
INTFLOAT g_decay_slope,
int len)
{
static const INTFLOAT a[] = { Q31(0.65143905753106f),
Q31(0.56471812200776f),
Q31(0.48954165955695f) };
INTFLOAT ag[PS_AP_LINKS];
int m, n;
for (m = 0; m < PS_AP_LINKS; m++)
ag[m] = AAC_MUL30(a[m], g_decay_slope);
for (n = 0; n < len; n++) {
INTFLOAT in_re = AAC_MSUB30(delay[n][0], phi_fract[0], delay[n][1], phi_fract[1]);
INTFLOAT in_im = AAC_MADD30(delay[n][0], phi_fract[1], delay[n][1], phi_fract[0]);
for (m = 0; m < PS_AP_LINKS; m++) {
INTFLOAT a_re = AAC_MUL31(ag[m], in_re);
INTFLOAT a_im = AAC_MUL31(ag[m], in_im);
INTFLOAT link_delay_re = ap_delay[m][n+2-m][0];
INTFLOAT link_delay_im = ap_delay[m][n+2-m][1];
INTFLOAT fractional_delay_re = Q_fract[m][0];
INTFLOAT fractional_delay_im = Q_fract[m][1];
INTFLOAT apd_re = in_re;
INTFLOAT apd_im = in_im;
in_re = AAC_MSUB30(link_delay_re, fractional_delay_re,
link_delay_im, fractional_delay_im);
in_re -= a_re;
in_im = AAC_MADD30(link_delay_re, fractional_delay_im,
link_delay_im, fractional_delay_re);
in_im -= a_im;
ap_delay[m][n+5][0] = apd_re + AAC_MUL31(ag[m], in_re);
ap_delay[m][n+5][1] = apd_im + AAC_MUL31(ag[m], in_im);
}
out[n][0] = AAC_MUL16(transient_gain[n], in_re);
out[n][1] = AAC_MUL16(transient_gain[n], in_im);
}
}
static void ps_stereo_interpolate_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
INTFLOAT h[2][4], INTFLOAT h_step[2][4],
int len)
{
INTFLOAT h0 = h[0][0];
INTFLOAT h1 = h[0][1];
INTFLOAT h2 = h[0][2];
INTFLOAT h3 = h[0][3];
INTFLOAT hs0 = h_step[0][0];
INTFLOAT hs1 = h_step[0][1];
INTFLOAT hs2 = h_step[0][2];
INTFLOAT hs3 = h_step[0][3];
int n;
for (n = 0; n < len; n++) {
//l is s, r is d
INTFLOAT l_re = l[n][0];
INTFLOAT l_im = l[n][1];
INTFLOAT r_re = r[n][0];
INTFLOAT r_im = r[n][1];
h0 += hs0;
h1 += hs1;
h2 += hs2;
h3 += hs3;
l[n][0] = AAC_MADD30(h0, l_re, h2, r_re);
l[n][1] = AAC_MADD30(h0, l_im, h2, r_im);
r[n][0] = AAC_MADD30(h1, l_re, h3, r_re);
r[n][1] = AAC_MADD30(h1, l_im, h3, r_im);
}
}
static void ps_stereo_interpolate_ipdopd_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
INTFLOAT h[2][4], INTFLOAT h_step[2][4],
int len)
{
INTFLOAT h00 = h[0][0], h10 = h[1][0];
INTFLOAT h01 = h[0][1], h11 = h[1][1];
INTFLOAT h02 = h[0][2], h12 = h[1][2];
INTFLOAT h03 = h[0][3], h13 = h[1][3];
INTFLOAT hs00 = h_step[0][0], hs10 = h_step[1][0];
INTFLOAT hs01 = h_step[0][1], hs11 = h_step[1][1];
INTFLOAT hs02 = h_step[0][2], hs12 = h_step[1][2];
INTFLOAT hs03 = h_step[0][3], hs13 = h_step[1][3];
int n;
for (n = 0; n < len; n++) {
//l is s, r is d
INTFLOAT l_re = l[n][0];
INTFLOAT l_im = l[n][1];
INTFLOAT r_re = r[n][0];
INTFLOAT r_im = r[n][1];
h00 += hs00;
h01 += hs01;
h02 += hs02;
h03 += hs03;
h10 += hs10;
h11 += hs11;
h12 += hs12;
h13 += hs13;
l[n][0] = AAC_MSUB30_V8(h00, l_re, h02, r_re, h10, l_im, h12, r_im);
l[n][1] = AAC_MADD30_V8(h00, l_im, h02, r_im, h10, l_re, h12, r_re);
r[n][0] = AAC_MSUB30_V8(h01, l_re, h03, r_re, h11, l_im, h13, r_im);
r[n][1] = AAC_MADD30_V8(h01, l_im, h03, r_im, h11, l_re, h13, r_re);
}
}
av_cold void AAC_RENAME(ff_psdsp_init)(PSDSPContext *s)
{
s->add_squares = ps_add_squares_c;
s->mul_pair_single = ps_mul_pair_single_c;
s->hybrid_analysis = ps_hybrid_analysis_c;
s->hybrid_analysis_ileave = ps_hybrid_analysis_ileave_c;
s->hybrid_synthesis_deint = ps_hybrid_synthesis_deint_c;
s->decorrelate = ps_decorrelate_c;
s->stereo_interpolate[0] = ps_stereo_interpolate_c;
s->stereo_interpolate[1] = ps_stereo_interpolate_ipdopd_c;
#if !USE_FIXED
if (ARCH_ARM)
ff_psdsp_init_arm(s);
if (ARCH_MIPS)
ff_psdsp_init_mips(s);
#endif /* !USE_FIXED */
}

8
libavcodec/aacsbr_template.c
View File

@ -64,7 +64,7 @@ av_cold void AAC_RENAME(ff_aac_sbr_init)(void)
aacsbr_tableinit();
ff_ps_init();
AAC_RENAME(ff_ps_init)();
}
/** Places SBR in pure upsampling mode. */
@ -91,7 +91,7 @@ av_cold void AAC_RENAME(ff_aac_sbr_ctx_init)(AACContext *ac, SpectralBandReplica
* and scale back down at synthesis. */
AAC_RENAME_32(ff_mdct_init)(&sbr->mdct, 7, 1, 1.0 / (64 * 32768.0));
AAC_RENAME_32(ff_mdct_init)(&sbr->mdct_ana, 7, 1, -2.0 * 32768.0);
ff_ps_ctx_init(&sbr->ps);
AAC_RENAME(ff_ps_ctx_init)(&sbr->ps);
AAC_RENAME(ff_sbrdsp_init)(&sbr->dsp);
aacsbr_func_ptr_init(&sbr->c);
}
@ -945,7 +945,7 @@ static void read_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
*num_bits_left = 0;
} else {
#if 1
*num_bits_left -= ff_ps_read_data(ac->avctx, gb, &sbr->ps, *num_bits_left);
*num_bits_left -= AAC_RENAME(ff_ps_read_data)(ac->avctx, gb, &sbr->ps, *num_bits_left);
ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
#else
avpriv_report_missing_feature(ac->avctx, "Parametric Stereo");
@ -1501,7 +1501,7 @@ void AAC_RENAME(ff_sbr_apply)(AACContext *ac, SpectralBandReplication *sbr, int
if (ac->oc[1].m4ac.ps == 1) {
if (sbr->ps.start) {
ff_ps_apply(ac->avctx, &sbr->ps, sbr->X[0], sbr->X[1], sbr->kx[1] + sbr->m[1]);
AAC_RENAME(ff_ps_apply)(ac->avctx, &sbr->ps, sbr->X[0], sbr->X[1], sbr->kx[1] + sbr->m[1]);
} else {
memcpy(sbr->X[1], sbr->X[0], sizeof(sbr->X[0]));
}