X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Faacps.c;h=1165d9be3f3a71aea9a43abe42410c449a098473;hb=9ca19971bcdd1af39680c15a0319d57505b9542b;hp=ea5a5d23317fcafa8d5c03a438c0e7d387df90d9;hpb=f929081f2e64d979fd8c58b17705e9554f99baf9;p=ffmpeg diff --git a/libavcodec/aacps.c b/libavcodec/aacps.c index ea5a5d23317..1165d9be3f3 100644 --- a/libavcodec/aacps.c +++ b/libavcodec/aacps.c @@ -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 #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]; + LOCAL_ALIGNED_16(INTFLOAT, h, [2], [4]); + LOCAL_ALIGNED_16(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); }