From 9d06d7bce3babb82ed650c13ed13a57f6f626a71 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Thu, 27 Jan 2011 15:20:43 +0000 Subject: Remove the add bias hack for the C version of DSPContext.float_to_int16_*(). --- libavcodec/dca.c | 47 +++++++++++++++++++---------------------------- 1 file changed, 19 insertions(+), 28 deletions(-) (limited to 'libavcodec/dca.c') diff --git a/libavcodec/dca.c b/libavcodec/dca.c index c1c70cde1b..aa71411efa 100644 --- a/libavcodec/dca.c +++ b/libavcodec/dca.c @@ -311,7 +311,6 @@ typedef struct { DECLARE_ALIGNED(16, float, raXin)[32]; int output; ///< type of output - float add_bias; ///< output bias float scale_bias; ///< output scale DECLARE_ALIGNED(16, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8]; @@ -868,7 +867,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index static void qmf_32_subbands(DCAContext * s, int chans, float samples_in[32][8], float *samples_out, - float scale, float bias) + float scale) { const float *prCoeff; int i; @@ -897,7 +896,7 @@ static void qmf_32_subbands(DCAContext * s, int chans, s->synth.synth_filter_float(&s->imdct, s->subband_fir_hist[chans], &s->hist_index[chans], s->subband_fir_noidea[chans], prCoeff, - samples_out, s->raXin, scale, bias); + samples_out, s->raXin, scale, 0); samples_out+= 32; } @@ -905,8 +904,7 @@ static void qmf_32_subbands(DCAContext * s, int chans, static void lfe_interpolation_fir(DCAContext *s, int decimation_select, int num_deci_sample, float *samples_in, - float *samples_out, float scale, - float bias) + float *samples_out, float scale) { /* samples_in: An array holding decimated samples. * Samples in current subframe starts from samples_in[0], @@ -931,7 +929,7 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select, /* Interpolation */ for (deciindex = 0; deciindex < num_deci_sample; deciindex++) { s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor, - scale, bias); + scale, 0); samples_in++; samples_out += 2 * decifactor; } @@ -939,19 +937,19 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select, /* downmixing routines */ #define MIX_REAR1(samples, si1, rs, coef) \ - samples[i] += (samples[si1] - add_bias) * coef[rs][0]; \ - samples[i+256] += (samples[si1] - add_bias) * coef[rs][1]; + samples[i] += samples[si1] * coef[rs][0]; \ + samples[i+256] += samples[si1] * coef[rs][1]; #define MIX_REAR2(samples, si1, si2, rs, coef) \ - samples[i] += (samples[si1] - add_bias) * coef[rs][0] + (samples[si2] - add_bias) * coef[rs+1][0]; \ - samples[i+256] += (samples[si1] - add_bias) * coef[rs][1] + (samples[si2] - add_bias) * coef[rs+1][1]; + samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \ + samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1]; #define MIX_FRONT3(samples, coef) \ - t = samples[i+c] - add_bias; \ - u = samples[i+l] - add_bias; \ - v = samples[i+r] - add_bias; \ - samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0] + add_bias; \ - samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1] + add_bias; + t = samples[i+c]; \ + u = samples[i+l]; \ + v = samples[i+r]; \ + samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \ + samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1]; #define DOWNMIX_TO_STEREO(op1, op2) \ for (i = 0; i < 256; i++){ \ @@ -961,7 +959,7 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select, static void dca_downmix(float *samples, int srcfmt, int downmix_coef[DCA_PRIM_CHANNELS_MAX][2], - const int8_t *channel_mapping, float add_bias) + const int8_t *channel_mapping) { int c,l,r,sl,sr,s; int i; @@ -1193,13 +1191,12 @@ static int dca_filter_channels(DCAContext * s, int block_index) /* static float pcm_to_double[8] = {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/ qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]], - M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ , - s->add_bias ); + M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ ); } /* Down mixing */ if (s->avctx->request_channels == 2 && s->prim_channels > 2) { - dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab, s->add_bias); + dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab); } /* Generate LFE samples for this subsubframe FIXME!!! */ @@ -1207,7 +1204,7 @@ static int dca_filter_channels(DCAContext * s, int block_index) lfe_interpolation_fir(s, s->lfe, 2 * s->lfe, s->lfe_data + 2 * s->lfe * (block_index + 4), &s->samples[256 * dca_lfe_index[s->amode]], - (1.0/256.0)*s->scale_bias, s->add_bias); + (1.0/256.0)*s->scale_bias); /* Outputs 20bits pcm samples */ } @@ -1798,8 +1795,8 @@ static int dca_decode_frame(AVCodecContext * avctx, float* rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256; int j; for(j = 0; j < 256; ++j) { - lt_chan[j] -= (back_chan[j] - s->add_bias) * M_SQRT1_2; - rt_chan[j] -= (back_chan[j] - s->add_bias) * M_SQRT1_2; + lt_chan[j] -= back_chan[j] * M_SQRT1_2; + rt_chan[j] -= back_chan[j] * M_SQRT1_2; } } @@ -1841,11 +1838,6 @@ static av_cold int dca_decode_init(AVCodecContext * avctx) s->samples_chanptr[i] = s->samples + i * 256; avctx->sample_fmt = AV_SAMPLE_FMT_S16; - if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) { - s->add_bias = 385.0f; - s->scale_bias = 1.0 / 32768.0; - } else { - s->add_bias = 0.0f; s->scale_bias = 1.0; /* allow downmixing to stereo */ @@ -1853,7 +1845,6 @@ static av_cold int dca_decode_init(AVCodecContext * avctx) avctx->request_channels == 2) { avctx->channels = avctx->request_channels; } - } return 0; -- cgit v1.2.3