--- /dev/null
+/*
+ * Copyright (c) 2011 Justin Ruggles
+ *
+ * This file is part of Libav.
+ *
+ * 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.
+ *
+ * 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/mathematics.h"
+#include "adx.h"
+
+void ff_adx_calculate_coeffs(int cutoff, int sample_rate, int bits, int *coeff)
+{
+ double a, b, c;
+
+ a = M_SQRT2 - cos(2.0 * M_PI * cutoff / sample_rate);
+ b = M_SQRT2 - 1.0;
+ c = (a - sqrt((a + b) * (a - b))) / b;
+
+ coeff[0] = lrintf(c * 2.0 * (1 << bits));
+ coeff[1] = lrintf(-(c * c) * (1 << bits));
+}
int header_parsed;
unsigned char dec_temp[18*2];
int in_temp;
+ int cutoff;
+ int coeff[2];
} ADXContext;
#define COEFF_BITS 12
-#define COEFF1 0x1CA6
-#define COEFF2 0x0CD4
+
+/**
+ * Calculate LPC coefficients based on cutoff frequency and sample rate.
+ *
+ * @param cutoff cutoff frequency
+ * @param sample_rate sample rate
+ * @param bits number of bits used to quantize coefficients
+ * @param[out] coeff 2 quantized LPC coefficients
+ */
+void ff_adx_calculate_coeffs(int cutoff, int sample_rate, int bits, int *coeff);
#endif /* AVCODEC_ADX_H */
s2 = prev->s2;
for (i = 0; i < 32; i++) {
d = get_sbits(&gb, 4);
- s0 = ((d << COEFF_BITS) * scale + COEFF1 * s1 - COEFF2 * s2) >> COEFF_BITS;
+ s0 = ((d << COEFF_BITS) * scale + c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS;
s2 = s1;
s1 = av_clip_int16(s0);
*out = s1;
int bufsize)
{
ADXContext *c = avctx->priv_data;
- int offset;
+ int offset, cutoff;
if (AV_RB16(buf) != 0x8000)
return AVERROR_INVALIDDATA;
return AVERROR_INVALIDDATA;
avctx->bit_rate = avctx->sample_rate * avctx->channels * 18 * 8 / 32;
+ cutoff = AV_RB16(buf + 16);
+ ff_adx_calculate_coeffs(cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
+
return offset;
}
/* 18 bytes <-> 32 samples */
-static void adx_encode(unsigned char *adx,const short *wav,
+static void adx_encode(ADXContext *c, unsigned char *adx, const short *wav,
ADXChannelState *prev)
{
int scale;
s2 = prev->s2;
for(i=0;i<32;i++) {
s0 = wav[i];
- d = ((s0 << COEFF_BITS) - COEFF1 * s1 + COEFF2 * s2) >> COEFF_BITS;
+ d = ((s0 << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
data[i]=d;
if (max<d) max=d;
if (min>d) min=d;
} adxhdr; /* big endian */
/* offset-6 "(c)CRI" */
#endif
+ ADXContext *c = avctx->priv_data;
+
AV_WB32(buf+0x00,0x80000000|0x20);
AV_WB32(buf+0x04,0x03120400|avctx->channels);
AV_WB32(buf+0x08,avctx->sample_rate);
AV_WB32(buf+0x0c,0); /* FIXME: set after */
- AV_WB32(buf+0x10,0x01040300);
- AV_WB32(buf+0x14,0x00000000);
- AV_WB32(buf+0x18,0x00000000);
- memcpy(buf+0x1c,"\0\0(c)CRI",8);
+ AV_WB16(buf + 0x10, c->cutoff);
+ AV_WB32(buf + 0x12, 0x03000000);
+ AV_WB32(buf + 0x16, 0x00000000);
+ AV_WB32(buf + 0x1a, 0x00000000);
+ memcpy (buf + 0x1e, "(c)CRI", 6);
return 0x20+4;
}
static av_cold int adx_encode_init(AVCodecContext *avctx)
{
+ ADXContext *c = avctx->priv_data;
+
if (avctx->channels > 2)
return -1; /* only stereo or mono =) */
avctx->frame_size = 32;
// avctx->bit_rate = avctx->sample_rate*avctx->channels*18*8/32;
+ /* the cutoff can be adjusted, but this seems to work pretty well */
+ c->cutoff = 500;
+ ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
+
av_log(avctx, AV_LOG_DEBUG, "adx encode init\n");
return 0;
if (avctx->channels==1) {
while(rest>=32) {
- adx_encode(dst,samples,c->prev);
+ adx_encode(c, dst, samples, c->prev);
dst+=18;
samples+=32;
rest-=32;
tmpbuf[i+32] = samples[i*2+1];
}
- adx_encode(dst,tmpbuf,c->prev);
- adx_encode(dst+18,tmpbuf+32,c->prev+1);
+ adx_encode(c, dst, tmpbuf, c->prev);
+ adx_encode(c, dst + 18, tmpbuf + 32, c->prev + 1);
dst+=18*2;
samples+=32*2;
rest-=32*2;