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atrac3: use optimized float_interleave() function for stereo interleaving

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This commit is contained in:
Justin Ruggles 2011-10-14 16:47:03 -04:00
parent 8af33cb38a
commit 5e76b8bb76
1 changed files with 19 additions and 7 deletions
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26
libavcodec/atrac3.c
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@ -41,6 +41,7 @@
#include "dsputil.h" #include "dsputil.h"
#include "bytestream.h" #include "bytestream.h"
#include "fft.h" #include "fft.h"
#include "fmtconvert.h"
#include "atrac.h" #include "atrac.h"
#include "atrac3data.h" #include "atrac3data.h"
@ -107,7 +108,7 @@ typedef struct {
//@} //@}
//@{ //@{
/** data buffers */ /** data buffers */
float outSamples[2048]; float *outSamples[2];
uint8_t* decoded_bytes_buffer; uint8_t* decoded_bytes_buffer;
float tempBuf[1070]; float tempBuf[1070];
//@} //@}
@ -120,6 +121,7 @@ typedef struct {
//@} //@}
FFTContext mdct_ctx; FFTContext mdct_ctx;
FmtConvertContext fmt_conv;
} ATRAC3Context; } ATRAC3Context;
static DECLARE_ALIGNED(32, float, mdct_window)[512]; static DECLARE_ALIGNED(32, float, mdct_window)[512];
@ -221,6 +223,8 @@ static av_cold int atrac3_decode_close(AVCodecContext *avctx)
av_free(q->pUnits); av_free(q->pUnits);
av_free(q->decoded_bytes_buffer); av_free(q->decoded_bytes_buffer);
av_freep(&q->outSamples[0]);
ff_mdct_end(&q->mdct_ctx); ff_mdct_end(&q->mdct_ctx);
return 0; return 0;
@ -824,7 +828,7 @@ static int atrac3_decode_frame(AVCodecContext *avctx,
const uint8_t *buf = avpkt->data; const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size; int buf_size = avpkt->size;
ATRAC3Context *q = avctx->priv_data; ATRAC3Context *q = avctx->priv_data;
int result = 0, i; int result = 0;
const uint8_t* databuf; const uint8_t* databuf;
float *samples = data; float *samples = data;
@ -843,7 +847,7 @@ static int atrac3_decode_frame(AVCodecContext *avctx,
databuf = buf; databuf = buf;
} }
result = decodeFrame(q, databuf, q->channels == 2 ? q->outSamples : samples); result = decodeFrame(q, databuf, q->channels == 2 ? q->outSamples[0] : samples);
if (result != 0) { if (result != 0) {
av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n"); av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
@ -852,10 +856,8 @@ static int atrac3_decode_frame(AVCodecContext *avctx,
/* interleave */ /* interleave */
if (q->channels == 2) { if (q->channels == 2) {
for (i = 0; i < 1024; i++) { q->fmt_conv.float_interleave(samples, (const float **)q->outSamples,
samples[i*2] = q->outSamples[i]; 1024, 2);
samples[i*2+1] = q->outSamples[1024+i];
}
} }
*data_size = 1024 * q->channels * av_get_bytes_per_sample(avctx->sample_fmt); *data_size = 1024 * q->channels * av_get_bytes_per_sample(avctx->sample_fmt);
@ -1003,6 +1005,7 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
} }
dsputil_init(&dsp, avctx); dsputil_init(&dsp, avctx);
ff_fmt_convert_init(&q->fmt_conv, avctx);
q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels); q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
if (!q->pUnits) { if (!q->pUnits) {
@ -1010,6 +1013,15 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
return AVERROR(ENOMEM); return AVERROR(ENOMEM);
} }
if (avctx->channels > 1) {
q->outSamples[0] = av_mallocz(1024 * 2 * sizeof(*q->outSamples[0]));
q->outSamples[1] = q->outSamples[0] + 1024;
if (!q->outSamples[0]) {
atrac3_decode_close(avctx);
return AVERROR(ENOMEM);
}
}
avctx->sample_fmt = AV_SAMPLE_FMT_FLT; avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
return 0; return 0;
} }