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Code Issues Releases Activity

atrac1: use planar sample format

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This commit is contained in:
Justin Ruggles 2012-08-26 01:10:51 -04:00
parent ac56ff9cc9
commit 23d53c5473
1 changed files with 4 additions and 25 deletions
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29
libavcodec/atrac1.c
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@ -36,7 +36,6 @@
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
#include "fmtconvert.h"
#include "sinewin.h"
#include "atrac.h"
@ -80,11 +79,9 @@ typedef struct {
DECLARE_ALIGNED(32, float, mid)[256];
DECLARE_ALIGNED(32, float, high)[512];
float* bands[3];
float *out_samples[AT1_MAX_CHANNELS];
FFTContext mdct_ctx[3];
int channels;
DSPContext dsp;
FmtConvertContext fmt_conv;
} AT1Ctx;
/** size of the transform in samples in the long mode for each QMF band */
@ -281,7 +278,6 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
AT1Ctx *q = avctx->priv_data;
int ch, ret;
GetBitContext gb;
float *samples;
if (buf_size < 212 * q->channels) {
@ -295,7 +291,6 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (float *)q->frame.data[0];
for (ch = 0; ch < q->channels; ch++) {
AT1SUCtx* su = &q->SUs[ch];
@ -314,13 +309,7 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
ret = at1_imdct_block(su, q);
if (ret < 0)
return ret;
at1_subband_synthesis(q, su, q->channels == 1 ? samples : q->out_samples[ch]);
}
/* interleave */
if (q->channels == 2) {
q->fmt_conv.float_interleave(samples, (const float **)q->out_samples,
AT1_SU_SAMPLES, 2);
at1_subband_synthesis(q, su, (float *)q->frame.extended_data[ch]);
}
*got_frame_ptr = 1;
@ -334,8 +323,6 @@ static av_cold int atrac1_decode_end(AVCodecContext * avctx)
{
AT1Ctx *q = avctx->priv_data;
av_freep(&q->out_samples[0]);
ff_mdct_end(&q->mdct_ctx[0]);
ff_mdct_end(&q->mdct_ctx[1]);
ff_mdct_end(&q->mdct_ctx[2]);
@ -349,7 +336,7 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
AT1Ctx *q = avctx->priv_data;
int ret;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
@ -358,15 +345,6 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
}
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 */
if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) ||
(ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) ||
@ -381,7 +359,6 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
ff_atrac_generate_tables();
ff_dsputil_init(&q->dsp, avctx);
ff_fmt_convert_init(&q->fmt_conv, avctx);
q->bands[0] = q->low;
q->bands[1] = q->mid;
@ -410,4 +387,6 @@ AVCodec ff_atrac1_decoder = {
.decode = atrac1_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};