[FFmpeg-cvslog] Remove the add bias hack for the C version of DSPContext.float_to_int16_*() .
Justin Ruggles
git
Fri Jan 28 03:43:33 CET 2011
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Thu Jan 27 15:20:43 2011 +0000| [733dbe7d18c267728ef03762d83e9e1b086668cd] | committer: Michael Niedermayer
Remove the add bias hack for the C version of DSPContext.float_to_int16_*().
(cherry picked from commit 9d06d7bce3babb82ed650c13ed13a57f6f626a71)
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=733dbe7d18c267728ef03762d83e9e1b086668cd
---
libavcodec/aac.h | 1 -
libavcodec/aacdec.c | 42 +++++++++++++++------------------------
libavcodec/aacsbr.c | 13 ++---------
libavcodec/ac3dec.c | 13 +----------
libavcodec/ac3dec.h | 1 -
libavcodec/binkaudio.c | 5 ----
libavcodec/dca.c | 47 +++++++++++++++++--------------------------
libavcodec/dsputil.c | 9 +-------
libavcodec/dsputil.h | 3 +-
libavcodec/nellymoserdec.c | 9 +-------
libavcodec/vorbis_dec.c | 38 ++++++++---------------------------
libavcodec/wmadec.c | 21 +-----------------
12 files changed, 54 insertions(+), 148 deletions(-)
diff --git a/libavcodec/aac.h b/libavcodec/aac.h
index b40d2c0..714e314 100644
--- a/libavcodec/aac.h
+++ b/libavcodec/aac.h
@@ -276,7 +276,6 @@ typedef struct {
* @{
*/
float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
- float add_bias; ///< offset for dsp.float_to_int16
float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
/** @} */
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index fd65b1c..fddec17 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -566,18 +566,10 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ac->random_state = 0x1f2e3d4c;
// -1024 - Compensate wrong IMDCT method.
- // 32768 - Required to scale values to the correct range for the bias method
- // for float to int16 conversion.
-
- if (ac->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
- ac->add_bias = 385.0f;
- ac->sf_scale = 1. / (-1024. * 32768.);
- ac->sf_offset = 0;
- } else {
- ac->add_bias = 0.0f;
+ // 60 - Required to scale values to the correct range [-32768,32767]
+ // for float to int16 conversion. (1 << (60 / 4)) == 32768
ac->sf_scale = 1. / -1024.;
ac->sf_offset = 60;
- }
ff_aac_tableinit();
@@ -1701,7 +1693,7 @@ static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
/**
* Conduct IMDCT and windowing.
*/
-static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce, float bias)
+static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
float *in = sce->coeffs;
@@ -1729,29 +1721,29 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce, float
*/
if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
(ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
- ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, bias, 512);
+ ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, 0, 512);
} else {
for (i = 0; i < 448; i++)
- out[i] = saved[i] + bias;
+ out[i] = saved[i];
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
- ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, bias, 64);
- ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, bias, 64);
- ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, bias, 64);
- ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, bias, 64);
- ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, bias, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 0, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 0, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 0, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 0, 64);
+ ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 0, 64);
memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
} else {
- ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, bias, 64);
+ ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 0, 64);
for (i = 576; i < 1024; i++)
- out[i] = buf[i-512] + bias;
+ out[i] = buf[i-512];
}
}
// buffer update
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
for (i = 0; i < 64; i++)
- saved[i] = temp[64 + i] - bias;
+ saved[i] = temp[64 + i];
ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64);
ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64);
ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64);
@@ -1811,13 +1803,12 @@ static void apply_independent_coupling(AACContext *ac,
{
int i;
const float gain = cce->coup.gain[index][0];
- const float bias = ac->add_bias;
const float *src = cce->ch[0].ret;
float *dest = target->ret;
const int len = 1024 << (ac->m4ac.sbr == 1);
for (i = 0; i < len; i++)
- dest[i] += gain * (src[i] - bias);
+ dest[i] += gain * src[i];
}
/**
@@ -1861,7 +1852,6 @@ static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
static void spectral_to_sample(AACContext *ac)
{
int i, type;
- float imdct_bias = (ac->m4ac.sbr <= 0) ? ac->add_bias : 0.0f;
for (type = 3; type >= 0; type--) {
for (i = 0; i < MAX_ELEM_ID; i++) {
ChannelElement *che = ac->che[type][i];
@@ -1875,9 +1865,9 @@ static void spectral_to_sample(AACContext *ac)
if (type <= TYPE_CPE)
apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
- imdct_and_windowing(ac, &che->ch[0], imdct_bias);
+ imdct_and_windowing(ac, &che->ch[0]);
if (type == TYPE_CPE) {
- imdct_and_windowing(ac, &che->ch[1], imdct_bias);
+ imdct_and_windowing(ac, &che->ch[1]);
}
if (ac->m4ac.sbr > 0) {
ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
diff --git a/libavcodec/aacsbr.c b/libavcodec/aacsbr.c
index 050305a..9b10bf2 100644
--- a/libavcodec/aacsbr.c
+++ b/libavcodec/aacsbr.c
@@ -1175,12 +1175,10 @@ static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct, const float *in,
static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
float *out, float X[2][38][64],
float mdct_buf[2][64],
- float *v0, int *v_off, const unsigned int div,
- float bias, float scale)
+ float *v0, int *v_off, const unsigned int div)
{
int i, n;
const float *sbr_qmf_window = div ? sbr_qmf_window_ds : sbr_qmf_window_us;
- int scale_and_bias = scale != 1.0f || bias != 0.0f;
float *v;
for (i = 0; i < 32; i++) {
if (*v_off == 0) {
@@ -1222,9 +1220,6 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
dsp->vector_fmul_add(out, v + ( 960 >> div), sbr_qmf_window + (448 >> div), out , 64 >> div);
dsp->vector_fmul_add(out, v + (1024 >> div), sbr_qmf_window + (512 >> div), out , 64 >> div);
dsp->vector_fmul_add(out, v + (1216 >> div), sbr_qmf_window + (576 >> div), out , 64 >> div);
- if (scale_and_bias)
- for (n = 0; n < 64 >> div; n++)
- out[n] = out[n] * scale + bias;
out += 64 >> div;
}
}
@@ -1760,12 +1755,10 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, L, sbr->X[0], sbr->qmf_filter_scratch,
sbr->data[0].synthesis_filterbank_samples,
&sbr->data[0].synthesis_filterbank_samples_offset,
- downsampled,
- ac->add_bias, -1024 * ac->sf_scale);
+ downsampled);
if (nch == 2)
sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, R, sbr->X[1], sbr->qmf_filter_scratch,
sbr->data[1].synthesis_filterbank_samples,
&sbr->data[1].synthesis_filterbank_samples_offset,
- downsampled,
- ac->add_bias, -1024 * ac->sf_scale);
+ downsampled);
}
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c
index a1e731d..2f9bc26 100644
--- a/libavcodec/ac3dec.c
+++ b/libavcodec/ac3dec.c
@@ -196,13 +196,7 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
av_lfg_init(&s->dith_state, 0);
/* set bias values for float to int16 conversion */
- if(s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
- s->add_bias = 385.0f;
- s->mul_bias = 1.0f;
- } else {
- s->add_bias = 0.0f;
s->mul_bias = 32767.0f;
- }
/* allow downmixing to stereo or mono */
if (avctx->channels > 0 && avctx->request_channels > 0 &&
@@ -626,9 +620,6 @@ static void do_rematrixing(AC3DecodeContext *s)
static inline void do_imdct(AC3DecodeContext *s, int channels)
{
int ch;
- float add_bias = s->add_bias;
- if(s->out_channels==1 && channels>1)
- add_bias *= LEVEL_MINUS_3DB; // compensate for the gain in downmix
for (ch=1; ch<=channels; ch++) {
if (s->block_switch[ch]) {
@@ -637,13 +628,13 @@ static inline void do_imdct(AC3DecodeContext *s, int channels)
for(i=0; i<128; i++)
x[i] = s->transform_coeffs[ch][2*i];
ff_imdct_half(&s->imdct_256, s->tmp_output, x);
- s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, add_bias, 128);
+ s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 0, 128);
for(i=0; i<128; i++)
x[i] = s->transform_coeffs[ch][2*i+1];
ff_imdct_half(&s->imdct_256, s->delay[ch-1], x);
} else {
ff_imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
- s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, add_bias, 128);
+ s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 0, 128);
memcpy(s->delay[ch-1], s->tmp_output+128, 128*sizeof(float));
}
}
diff --git a/libavcodec/ac3dec.h b/libavcodec/ac3dec.h
index 0707769..55520cd 100644
--- a/libavcodec/ac3dec.h
+++ b/libavcodec/ac3dec.h
@@ -190,7 +190,6 @@ typedef struct {
///@defgroup opt optimization
DSPContext dsp; ///< for optimization
- float add_bias; ///< offset for float_to_int16 conversion
float mul_bias; ///< scaling for float_to_int16 conversion
///@}
diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c
index 7f57982..ae2f6c8 100644
--- a/libavcodec/binkaudio.c
+++ b/libavcodec/binkaudio.c
@@ -222,11 +222,6 @@ static void decode_block(BinkAudioContext *s, short *out, int use_dct)
ff_rdft_calc(&s->trans.rdft, coeffs);
}
- if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
- for (i = 0; i < s->channels; i++)
- for (j = 0; j < s->frame_len; j++)
- s->coeffs_ptr[i][j] = 385.0 + s->coeffs_ptr[i][j]*(1.0/32767.0);
- }
s->dsp.float_to_int16_interleave(out, (const float **)s->coeffs_ptr, s->frame_len, s->channels);
if (!s->first) {
diff --git a/libavcodec/dca.c b/libavcodec/dca.c
index c1c70cd..aa71411 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;
diff --git a/libavcodec/dsputil.c b/libavcodec/dsputil.c
index 2ed0052..03a5eee 100644
--- a/libavcodec/dsputil.c
+++ b/libavcodec/dsputil.c
@@ -3910,14 +3910,7 @@ static void vector_clipf_c(float *dst, const float *src, float min, float max, i
}
static av_always_inline int float_to_int16_one(const float *src){
- int_fast32_t tmp = *(const int32_t*)src;
- if(tmp & 0xf0000){
- tmp = (0x43c0ffff - tmp)>>31;
- // is this faster on some gcc/cpu combinations?
-// if(tmp > 0x43c0ffff) tmp = 0xFFFF;
-// else tmp = 0;
- }
- return tmp - 0x8000;
+ return av_clip_int16(lrintf(*src));
}
void ff_float_to_int16_c(int16_t *dst, const float *src, long len){
diff --git a/libavcodec/dsputil.h b/libavcodec/dsputil.h
index baa68be..eabecf0 100644
--- a/libavcodec/dsputil.h
+++ b/libavcodec/dsputil.h
@@ -435,8 +435,7 @@ typedef struct DSPContext {
*/
void (*butterflies_float)(float *restrict v1, float *restrict v2, int len);
- /* C version: convert floats from the range [384.0,386.0] to ints in [-32768,32767]
- * simd versions: convert floats from [-32768.0,32767.0] without rescaling and arrays are 16byte aligned */
+ /* convert floats from [-32768.0,32767.0] without rescaling and arrays are 16byte aligned */
void (*float_to_int16)(int16_t *dst, const float *src, long len);
void (*float_to_int16_interleave)(int16_t *dst, const float **src, long len, int channels);
diff --git a/libavcodec/nellymoserdec.c b/libavcodec/nellymoserdec.c
index 89dde91..e70d062 100644
--- a/libavcodec/nellymoserdec.c
+++ b/libavcodec/nellymoserdec.c
@@ -49,7 +49,6 @@ typedef struct NellyMoserDecodeContext {
float state[128];
AVLFG random_state;
GetBitContext gb;
- int add_bias;
float scale_bias;
DSPContext dsp;
FFTContext imdct_ctx;
@@ -65,7 +64,7 @@ static void overlap_and_window(NellyMoserDecodeContext *s, float *state, float *
while (bot < NELLY_BUF_LEN) {
audio[bot] = a_in [bot]*ff_sine_128[bot]
- +state[bot]*ff_sine_128[top] + s->add_bias;
+ +state[bot]*ff_sine_128[top];
bot++;
top--;
@@ -136,13 +135,7 @@ static av_cold int decode_init(AVCodecContext * avctx) {
dsputil_init(&s->dsp, avctx);
- if(s->dsp.float_to_int16 == ff_float_to_int16_c) {
- s->add_bias = 385;
- s->scale_bias = 1.0/(8*32768);
- } else {
- s->add_bias = 0;
s->scale_bias = 1.0/(1*8);
- }
/* Generate overlap window */
if (!ff_sine_128[127])
diff --git a/libavcodec/vorbis_dec.c b/libavcodec/vorbis_dec.c
index cdb485a..69b784c 100644
--- a/libavcodec/vorbis_dec.c
+++ b/libavcodec/vorbis_dec.c
@@ -153,8 +153,7 @@ typedef struct vorbis_context_s {
float *channel_residues;
float *channel_floors;
float *saved;
- uint_fast32_t add_bias; // for float->int conversion
- uint_fast32_t exp_bias;
+ float scale_bias; // for float->int conversion
} vorbis_context;
/* Helper functions */
@@ -932,8 +931,8 @@ static int vorbis_parse_id_hdr(vorbis_context *vc)
vc->saved = av_mallocz((vc->blocksize[1] / 4) * vc->audio_channels * sizeof(float));
vc->previous_window = 0;
- ff_mdct_init(&vc->mdct[0], bl0, 1, vc->exp_bias ? -(1 << 15) : -1.0);
- ff_mdct_init(&vc->mdct[1], bl1, 1, vc->exp_bias ? -(1 << 15) : -1.0);
+ ff_mdct_init(&vc->mdct[0], bl0, 1, -vc->scale_bias);
+ ff_mdct_init(&vc->mdct[1], bl1, 1, -vc->scale_bias);
AV_DEBUG(" vorbis version %d \n audio_channels %d \n audio_samplerate %d \n bitrate_max %d \n bitrate_nom %d \n bitrate_min %d \n blk_0 %d blk_1 %d \n ",
vc->version, vc->audio_channels, vc->audio_samplerate, vc->bitrate_maximum, vc->bitrate_nominal, vc->bitrate_minimum, vc->blocksize[0], vc->blocksize[1]);
@@ -963,13 +962,7 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
vc->avccontext = avccontext;
dsputil_init(&vc->dsp, avccontext);
- if (vc->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
- vc->add_bias = 385;
- vc->exp_bias = 0;
- } else {
- vc->add_bias = 0;
- vc->exp_bias = 15 << 23;
- }
+ vc->scale_bias = 32768.0f;
if (!headers_len) {
av_log(avccontext, AV_LOG_ERROR, "Extradata missing.\n");
@@ -1453,18 +1446,6 @@ void vorbis_inverse_coupling(float *mag, float *ang, int blocksize)
}
}
-static void copy_normalize(float *dst, float *src, int len, int exp_bias,
- float add_bias)
-{
- int i;
- if (exp_bias) {
- memcpy(dst, src, len * sizeof(float));
- } else {
- for (i = 0; i < len; i++)
- dst[i] = src[i] + add_bias;
- }
-}
-
// Decode the audio packet using the functions above
static int vorbis_parse_audio_packet(vorbis_context *vc)
@@ -1484,7 +1465,6 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
uint_fast8_t res_chan[255];
uint_fast8_t res_num = 0;
int_fast16_t retlen = 0;
- float fadd_bias = vc->add_bias;
if (get_bits1(gb)) {
av_log(vc->avccontext, AV_LOG_ERROR, "Not a Vorbis I audio packet.\n");
@@ -1595,13 +1575,13 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
const float *win = vc->win[blockflag & previous_window];
if (blockflag == previous_window) {
- vc->dsp.vector_fmul_window(ret, saved, buf, win, fadd_bias, blocksize / 4);
+ vc->dsp.vector_fmul_window(ret, saved, buf, win, 0, blocksize / 4);
} else if (blockflag > previous_window) {
- vc->dsp.vector_fmul_window(ret, saved, buf, win, fadd_bias, bs0 / 4);
- copy_normalize(ret+bs0/2, buf+bs0/4, (bs1-bs0)/4, vc->exp_bias, fadd_bias);
+ vc->dsp.vector_fmul_window(ret, saved, buf, win, 0, bs0 / 4);
+ memcpy(ret+bs0/2, buf+bs0/4, ((bs1-bs0)/4) * sizeof(float));
} else {
- copy_normalize(ret, saved, (bs1 - bs0) / 4, vc->exp_bias, fadd_bias);
- vc->dsp.vector_fmul_window(ret + (bs1 - bs0) / 4, saved + (bs1 - bs0) / 4, buf, win, fadd_bias, bs0 / 4);
+ memcpy(ret, saved, ((bs1 - bs0) / 4) * sizeof(float));
+ vc->dsp.vector_fmul_window(ret + (bs1 - bs0) / 4, saved + (bs1 - bs0) / 4, buf, win, 0, bs0 / 4);
}
memcpy(saved, buf + blocksize / 4, blocksize / 4 * sizeof(float));
}
diff --git a/libavcodec/wmadec.c b/libavcodec/wmadec.c
index b720ea5..03d7bd1 100644
--- a/libavcodec/wmadec.c
+++ b/libavcodec/wmadec.c
@@ -768,9 +768,8 @@ next:
/* decode a frame of frame_len samples */
static int wma_decode_frame(WMACodecContext *s, int16_t *samples)
{
- int ret, i, n, ch, incr;
- int16_t *ptr;
- float *iptr;
+ int ret, n, ch, incr;
+ const float *output[MAX_CHANNELS];
#ifdef TRACE
tprintf(s->avctx, "***decode_frame: %d size=%d\n", s->frame_count++, s->frame_len);
@@ -790,21 +789,6 @@ static int wma_decode_frame(WMACodecContext *s, int16_t *samples)
/* convert frame to integer */
n = s->frame_len;
incr = s->nb_channels;
- if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
- for(ch = 0; ch < s->nb_channels; ch++) {
- ptr = samples + ch;
- iptr = s->frame_out[ch];
-
- for(i=0;i<n;i++) {
- *ptr = av_clip_int16(lrintf(*iptr++));
- ptr += incr;
- }
- /* prepare for next block */
- memmove(&s->frame_out[ch][0], &s->frame_out[ch][s->frame_len],
- s->frame_len * sizeof(float));
- }
- } else {
- const float *output[MAX_CHANNELS];
for (ch = 0; ch < MAX_CHANNELS; ch++)
output[ch] = s->frame_out[ch];
s->dsp.float_to_int16_interleave(samples, output, n, incr);
@@ -812,7 +796,6 @@ static int wma_decode_frame(WMACodecContext *s, int16_t *samples)
/* prepare for next block */
memmove(&s->frame_out[ch][0], &s->frame_out[ch][n], n * sizeof(float));
}
- }
#ifdef TRACE
dump_shorts(s, "samples", samples, n * s->nb_channels);
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