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Code Issues Releases Activity

atrac1: use optimized float_interleave() function for stereo interleaving

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This commit is contained in:
Justin Ruggles 2011-10-14 15:25:01 -04:00
parent 96b5702efe
commit 21dcecc310
1 changed files with 19 additions and 7 deletions
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26
libavcodec/atrac1.c
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@ -36,6 +36,7 @@
#include "get_bits.h" #include "get_bits.h"
#include "dsputil.h" #include "dsputil.h"
#include "fft.h" #include "fft.h"
#include "fmtconvert.h"
#include "sinewin.h" #include "sinewin.h"
#include "atrac.h" #include "atrac.h"
@ -78,10 +79,11 @@ typedef struct {
DECLARE_ALIGNED(32, float, mid)[256]; DECLARE_ALIGNED(32, float, mid)[256];
DECLARE_ALIGNED(32, float, high)[512]; DECLARE_ALIGNED(32, float, high)[512];
float* bands[3]; float* bands[3];
DECLARE_ALIGNED(32, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]; float *out_samples[AT1_MAX_CHANNELS];
FFTContext mdct_ctx[3]; FFTContext mdct_ctx[3];
int channels; int channels;
DSPContext dsp; DSPContext dsp;
FmtConvertContext fmt_conv;
} AT1Ctx; } AT1Ctx;
/** size of the transform in samples in the long mode for each QMF band */ /** size of the transform in samples in the long mode for each QMF band */
@ -276,7 +278,7 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
const uint8_t *buf = avpkt->data; const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size; int buf_size = avpkt->size;
AT1Ctx *q = avctx->priv_data; AT1Ctx *q = avctx->priv_data;
int ch, ret, i, out_size; int ch, ret, out_size;
GetBitContext gb; GetBitContext gb;
float* samples = data; float* samples = data;
@ -313,12 +315,10 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
at1_subband_synthesis(q, su, q->channels == 1 ? samples : q->out_samples[ch]); at1_subband_synthesis(q, su, q->channels == 1 ? samples : q->out_samples[ch]);
} }
/* interleave; FIXME, should create/use a DSP function */ /* interleave */
if (q->channels == 2) { if (q->channels == 2) {
for (i = 0; i < AT1_SU_SAMPLES; i++) { q->fmt_conv.float_interleave(samples, (const float **)q->out_samples,
samples[i * 2] = q->out_samples[0][i]; AT1_SU_SAMPLES, 2);
samples[i * 2 + 1] = q->out_samples[1][i];
}
} }
*data_size = out_size; *data_size = out_size;
@ -339,6 +339,15 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
} }
q->channels = avctx->channels; q->channels = avctx->channels;
if (avctx->channels == 2) {
q->out_samples[0] = av_malloc(2 * AT1_SU_SAMPLES * sizeof(*q->out_samples[0]));
q->out_samples[1] = q->out_samples[0] + AT1_SU_SAMPLES;
if (!q->out_samples[0]) {
av_freep(&q->out_samples[0]);
return AVERROR(ENOMEM);
}
}
/* Init the mdct transforms */ /* Init the mdct transforms */
ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15)); ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15)); ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
@ -349,6 +358,7 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
atrac_generate_tables(); atrac_generate_tables();
dsputil_init(&q->dsp, avctx); dsputil_init(&q->dsp, avctx);
ff_fmt_convert_init(&q->fmt_conv, avctx);
q->bands[0] = q->low; q->bands[0] = q->low;
q->bands[1] = q->mid; q->bands[1] = q->mid;
@ -367,6 +377,8 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
static av_cold int atrac1_decode_end(AVCodecContext * avctx) { static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
AT1Ctx *q = avctx->priv_data; AT1Ctx *q = avctx->priv_data;
av_freep(&q->out_samples[0]);
ff_mdct_end(&q->mdct_ctx[0]); ff_mdct_end(&q->mdct_ctx[0]);
ff_mdct_end(&q->mdct_ctx[1]); ff_mdct_end(&q->mdct_ctx[1]);
ff_mdct_end(&q->mdct_ctx[2]); ff_mdct_end(&q->mdct_ctx[2]);