* Copyright (c) 2007 Benjamin Zores <ben@geexbox.org>
* based upon libdemac from Dave Chapman.
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define ALT_BITSTREAM_READER_LE
#include "avcodec.h"
#include "dsputil.h"
-#include "bitstream.h"
+#include "get_bits.h"
#include "bytestream.h"
+#include "libavutil/audioconvert.h"
/**
- * @file apedec.c
+ * @file
* Monkey's Audio lossless audio decoder
*/
} APEContext;
// TODO: dsputilize
-static inline void vector_add(int16_t * v1, int16_t * v2, int order)
-{
- while (order--)
- *v1++ += *v2++;
-}
-
-// TODO: dsputilize
-static inline void vector_sub(int16_t * v1, int16_t * v2, int order)
-{
- while (order--)
- *v1++ -= *v2++;
-}
-
-// TODO: dsputilize
-static inline int32_t scalarproduct(int16_t * v1, int16_t * v2, int order)
-{
- int res = 0;
-
- while (order--)
- res += *v1++ * *v2++;
-
- return res;
-}
static av_cold int ape_decode_init(AVCodecContext * avctx)
{
av_log(avctx, AV_LOG_ERROR, "Incorrect extradata\n");
return -1;
}
- if (avctx->bits_per_sample != 16) {
+ if (avctx->bits_per_coded_sample != 16) {
av_log(avctx, AV_LOG_ERROR, "Only 16-bit samples are supported\n");
return -1;
}
}
dsputil_init(&s->dsp, avctx);
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
return 0;
}
for (i = 0; i < APE_FILTER_LEVELS; i++)
av_freep(&s->filterbuf[i]);
+ av_freep(&s->data);
return 0;
}
/**
- * @defgroup rangecoder APE range decoder
+ * @name APE range decoding functions
* @{
*/
/**
* Calculate culmulative frequency for next symbol. Does NO update!
+ * @param ctx decoder context
* @param tot_f is the total frequency or (code_value)1<<shift
* @return the culmulative frequency
*/
/**
* Decode value with given size in bits
+ * @param ctx decoder context
* @param shift number of bits to decode
*/
static inline int range_decode_culshift(APEContext * ctx, int shift)
/**
* Update decoding state
+ * @param ctx decoder context
* @param sy_f the interval length (frequency of the symbol)
* @param lt_f the lower end (frequency sum of < symbols)
*/
/**
* Decode symbol
+ * @param ctx decoder context
* @param counts probability range start position
- * @param count_diffs probability range widths
+ * @param counts_diff probability range widths
*/
static inline int range_get_symbol(APEContext * ctx,
const uint16_t counts[],
static inline void update_rice(APERice *rice, int x)
{
+ int lim = rice->k ? (1 << (rice->k + 4)) : 0;
rice->ksum += ((x + 1) / 2) - ((rice->ksum + 16) >> 5);
- if (rice->k == 0)
- rice->k = 1;
- else if (rice->ksum < (1 << (rice->k + 4)))
+ if (rice->ksum < lim)
rice->k--;
else if (rice->ksum >= (1 << (rice->k + 5)))
rice->k++;
{
int x, overflow;
- if (ctx->fileversion < 3980) {
+ if (ctx->fileversion < 3990) {
int tmpk;
overflow = range_get_symbol(ctx, counts_3970, counts_diff_3970);
overflow |= range_decode_bits(ctx, 16);
}
- base = range_decode_culfreq(ctx, pivot);
- range_decode_update(ctx, 1, base);
+ if (pivot < 0x10000) {
+ base = range_decode_culfreq(ctx, pivot);
+ range_decode_update(ctx, 1, base);
+ } else {
+ int base_hi = pivot, base_lo;
+ int bbits = 0;
+
+ while (base_hi & ~0xFFFF) {
+ base_hi >>= 1;
+ bbits++;
+ }
+ base_hi = range_decode_culfreq(ctx, base_hi + 1);
+ range_decode_update(ctx, 1, base_hi);
+ base_lo = range_decode_culfreq(ctx, 1 << bbits);
+ range_decode_update(ctx, 1, base_lo);
+
+ base = (base_hi << bbits) + base_lo;
+ }
x = base + overflow * pivot;
}
return (x < 0) - (x > 0);
}
-static int predictor_update_filter(APEPredictor *p, const int decoded, const int filter, const int delayA, const int delayB, const int adaptA, const int adaptB)
+static av_always_inline int predictor_update_filter(APEPredictor *p, const int decoded, const int filter, const int delayA, const int delayB, const int adaptA, const int adaptB)
{
- int32_t predictionA, predictionB;
+ int32_t predictionA, predictionB, sign;
p->buf[delayA] = p->lastA[filter];
p->buf[adaptA] = APESIGN(p->buf[delayA]);
p->lastA[filter] = decoded + ((predictionA + (predictionB >> 1)) >> 10);
p->filterA[filter] = p->lastA[filter] + ((p->filterA[filter] * 31) >> 5);
- if (!decoded) // no need updating filter coefficients
- return p->filterA[filter];
-
- if (decoded > 0) {
- p->coeffsA[filter][0] -= p->buf[adaptA ];
- p->coeffsA[filter][1] -= p->buf[adaptA - 1];
- p->coeffsA[filter][2] -= p->buf[adaptA - 2];
- p->coeffsA[filter][3] -= p->buf[adaptA - 3];
+ sign = APESIGN(decoded);
+ p->coeffsA[filter][0] += p->buf[adaptA ] * sign;
+ p->coeffsA[filter][1] += p->buf[adaptA - 1] * sign;
+ p->coeffsA[filter][2] += p->buf[adaptA - 2] * sign;
+ p->coeffsA[filter][3] += p->buf[adaptA - 3] * sign;
+ p->coeffsB[filter][0] += p->buf[adaptB ] * sign;
+ p->coeffsB[filter][1] += p->buf[adaptB - 1] * sign;
+ p->coeffsB[filter][2] += p->buf[adaptB - 2] * sign;
+ p->coeffsB[filter][3] += p->buf[adaptB - 3] * sign;
+ p->coeffsB[filter][4] += p->buf[adaptB - 4] * sign;
- p->coeffsB[filter][0] -= p->buf[adaptB ];
- p->coeffsB[filter][1] -= p->buf[adaptB - 1];
- p->coeffsB[filter][2] -= p->buf[adaptB - 2];
- p->coeffsB[filter][3] -= p->buf[adaptB - 3];
- p->coeffsB[filter][4] -= p->buf[adaptB - 4];
- } else {
- p->coeffsA[filter][0] += p->buf[adaptA ];
- p->coeffsA[filter][1] += p->buf[adaptA - 1];
- p->coeffsA[filter][2] += p->buf[adaptA - 2];
- p->coeffsA[filter][3] += p->buf[adaptA - 3];
-
- p->coeffsB[filter][0] += p->buf[adaptB ];
- p->coeffsB[filter][1] += p->buf[adaptB - 1];
- p->coeffsB[filter][2] += p->buf[adaptB - 2];
- p->coeffsB[filter][3] += p->buf[adaptB - 3];
- p->coeffsB[filter][4] += p->buf[adaptB - 4];
- }
return p->filterA[filter];
}
static void predictor_decode_stereo(APEContext * ctx, int count)
{
- int32_t predictionA, predictionB;
APEPredictor *p = &ctx->predictor;
int32_t *decoded0 = ctx->decoded0;
int32_t *decoded1 = ctx->decoded1;
while (count--) {
/* Predictor Y */
- predictionA = predictor_update_filter(p, *decoded0, 0, YDELAYA, YDELAYB, YADAPTCOEFFSA, YADAPTCOEFFSB);
- predictionB = predictor_update_filter(p, *decoded1, 1, XDELAYA, XDELAYB, XADAPTCOEFFSA, XADAPTCOEFFSB);
- *(decoded0++) = predictionA;
- *(decoded1++) = predictionB;
+ *decoded0 = predictor_update_filter(p, *decoded0, 0, YDELAYA, YDELAYB, YADAPTCOEFFSA, YADAPTCOEFFSB);
+ decoded0++;
+ *decoded1 = predictor_update_filter(p, *decoded1, 1, XDELAYA, XDELAYB, XADAPTCOEFFSA, XADAPTCOEFFSB);
+ decoded1++;
/* Combined */
p->buf++;
{
APEPredictor *p = &ctx->predictor;
int32_t *decoded0 = ctx->decoded0;
- int32_t predictionA, currentA, A;
+ int32_t predictionA, currentA, A, sign;
currentA = p->lastA[0];
p->buf[YADAPTCOEFFSA] = APESIGN(p->buf[YDELAYA ]);
p->buf[YADAPTCOEFFSA - 1] = APESIGN(p->buf[YDELAYA - 1]);
- if (A > 0) {
- p->coeffsA[0][0] -= p->buf[YADAPTCOEFFSA ];
- p->coeffsA[0][1] -= p->buf[YADAPTCOEFFSA - 1];
- p->coeffsA[0][2] -= p->buf[YADAPTCOEFFSA - 2];
- p->coeffsA[0][3] -= p->buf[YADAPTCOEFFSA - 3];
- } else if (A < 0) {
- p->coeffsA[0][0] += p->buf[YADAPTCOEFFSA ];
- p->coeffsA[0][1] += p->buf[YADAPTCOEFFSA - 1];
- p->coeffsA[0][2] += p->buf[YADAPTCOEFFSA - 2];
- p->coeffsA[0][3] += p->buf[YADAPTCOEFFSA - 3];
- }
+ sign = APESIGN(A);
+ p->coeffsA[0][0] += p->buf[YADAPTCOEFFSA ] * sign;
+ p->coeffsA[0][1] += p->buf[YADAPTCOEFFSA - 1] * sign;
+ p->coeffsA[0][2] += p->buf[YADAPTCOEFFSA - 2] * sign;
+ p->coeffsA[0][3] += p->buf[YADAPTCOEFFSA - 3] * sign;
p->buf++;
do_init_filter(&f[1], buf + order * 3 + HISTORY_SIZE, order);
}
-static inline void do_apply_filter(int version, APEFilter *f, int32_t *data, int count, int order, int fracbits)
+static void do_apply_filter(APEContext * ctx, int version, APEFilter *f, int32_t *data, int count, int order, int fracbits)
{
int res;
int absres;
while (count--) {
/* round fixedpoint scalar product */
- res = (scalarproduct(f->delay - order, f->coeffs, order) + (1 << (fracbits - 1))) >> fracbits;
-
- if (*data < 0)
- vector_add(f->coeffs, f->adaptcoeffs - order, order);
- else if (*data > 0)
- vector_sub(f->coeffs, f->adaptcoeffs - order, order);
-
+ res = ctx->dsp.scalarproduct_and_madd_int16(f->coeffs, f->delay - order, f->adaptcoeffs - order, order, APESIGN(*data));
+ res = (res + (1 << (fracbits - 1))) >> fracbits;
res += *data;
-
*data++ = res;
/* Update the output history */
/* Version 3.98 and later files */
/* Update the adaption coefficients */
- absres = (res < 0 ? -res : res);
-
- if (absres > (f->avg * 3))
- *f->adaptcoeffs = ((res >> 25) & 64) - 32;
- else if (absres > (f->avg * 4) / 3)
- *f->adaptcoeffs = ((res >> 26) & 32) - 16;
- else if (absres > 0)
- *f->adaptcoeffs = ((res >> 27) & 16) - 8;
+ absres = FFABS(res);
+ if (absres)
+ *f->adaptcoeffs = ((res & (1<<31)) - (1<<30)) >> (25 + (absres <= f->avg*3) + (absres <= f->avg*4/3));
else
*f->adaptcoeffs = 0;
int32_t * data0, int32_t * data1,
int count, int order, int fracbits)
{
- do_apply_filter(ctx->fileversion, &f[0], data0, count, order, fracbits);
+ do_apply_filter(ctx, ctx->fileversion, &f[0], data0, count, order, fracbits);
if (data1)
- do_apply_filter(ctx->fileversion, &f[1], data1, count, order, fracbits);
+ do_apply_filter(ctx, ctx->fileversion, &f[1], data1, count, order, fracbits);
}
static void ape_apply_filters(APEContext * ctx, int32_t * decoded0,
static int ape_decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
- const uint8_t * buf, int buf_size)
+ AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
APEContext *s = avctx->priv_data;
int16_t *samples = data;
int nblocks;
ape_unpack_mono(s, blockstodecode);
else
ape_unpack_stereo(s, blockstodecode);
+ emms_c();
if(s->error || s->ptr > s->data_end){
s->samples=0;
return bytes_used;
}
-AVCodec ape_decoder = {
+static void ape_flush(AVCodecContext *avctx)
+{
+ APEContext *s = avctx->priv_data;
+ s->samples= 0;
+}
+
+AVCodec ff_ape_decoder = {
"ape",
- CODEC_TYPE_AUDIO,
+ AVMEDIA_TYPE_AUDIO,
CODEC_ID_APE,
sizeof(APEContext),
ape_decode_init,
NULL,
ape_decode_close,
ape_decode_frame,
- .long_name = "Monkey's Audio",
+ .capabilities = CODEC_CAP_SUBFRAMES,
+ .flush = ape_flush,
+ .long_name = NULL_IF_CONFIG_SMALL("Monkey's Audio"),
};
projects / ffmpeg / blobdiff