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@@ -0,0 +1,482 @@
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+/*
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+ * Wavesynth pseudo-codec
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+ * Copyright (c) 2011 Nicolas George
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+ *
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+ * This file is part of FFmpeg.
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+ *
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+ * FFmpeg is free software; you can redistribute it and/or
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+ * modify it under the terms of the GNU Lesser General Public
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+ * License as published by the Free Software Foundation; either
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+ * version 2.1 of the License, or (at your option) any later version.
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+ *
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+ * FFmpeg is distributed in the hope that it will be useful,
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+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
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+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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+ * Lesser General Public License for more details.
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+ *
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+ * You should have received a copy of the GNU Lesser General Public
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+ * License along with FFmpeg; if not, write to the Free Software
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+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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+ */
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+
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+#include "libavutil/intreadwrite.h"
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+#include "libavutil/log.h"
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+#include "avcodec.h"
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+
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+#define SIN_BITS 14
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+#define WS_MAX_CHANNELS 32
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+#define INF_TS 0x7FFFFFFFFFFFFFFF
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+
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+#define PINK_UNIT 128
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+
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+/*
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+ Format of the extradata and packets
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+
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+ THIS INFORMATION IS NOT PART OF THE PUBLIC API OR ABI.
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+ IT CAN CHANGE WITHOUT NOTIFICATION.
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+
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+ All numbers are in little endian.
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+
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+ The codec extradata define a set of intervals with uniform content.
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+ Overlapping intervals are added together.
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+
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+ extradata:
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+ uint32 number of intervals
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+ ... intervals
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+
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+ interval:
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+ int64 start timestamp; time_base must be 1/sample_rate;
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+ start timestamps must be in ascending order
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+ int64 end timestamp
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+ uint32 type
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+ uint32 channels mask
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+ ... additional information, depends on type
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+
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+ sine interval (type fourcc "SINE"):
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+ int32 start frequency, in 1/(1<<16) Hz
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+ int32 end frequency
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+ int32 start amplitude, 1<<16 is the full amplitude
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+ int32 end amplitude
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+ uint32 start phase, 0 is sin(0), 0x20000000 is sin(pi/2), etc.;
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+ n | (1<<31) means to match the phase of previous channel #n
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+
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+ pink noise interval (type fourcc "NOIS"):
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+ int32 start amplitude
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+ int32 end amplitude
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+
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+ The input packets encode the time and duration of the requested segment.
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+
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+ packet:
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+ int64 start timestamp
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+ int32 duration
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+
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+*/
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+
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+enum ws_interval_type {
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+ WS_SINE = MKTAG('S','I','N','E'),
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+ WS_NOISE = MKTAG('N','O','I','S'),
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+};
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+
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+struct ws_interval {
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+ int64_t ts_start, ts_end;
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+ uint64_t phi0, dphi0, ddphi;
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+ uint64_t amp0, damp;
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+ uint64_t phi, dphi, amp;
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+ uint32_t channels;
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+ enum ws_interval_type type;
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+ int next;
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+};
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+
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+struct wavesynth_context {
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+ int64_t cur_ts;
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+ int64_t next_ts;
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+ int32_t *sin;
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+ AVFrame frame;
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+ struct ws_interval *inter;
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+ uint32_t dither_state;
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+ uint32_t pink_state;
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+ int32_t pink_pool[PINK_UNIT];
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+ unsigned pink_need, pink_pos;
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+ int nb_inter;
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+ int cur_inter;
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+ int next_inter;
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+};
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+
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+#define LCG_A 1284865837
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+#define LCG_C 4150755663
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+#define LCG_AI 849225893 /* A*AI = 1 [mod 1<<32] */
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+
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+static uint32_t lcg_next(uint32_t *s)
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+{
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+ *s = *s * LCG_A + LCG_C;
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+ return *s;
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+}
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+
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+static void lcg_seek(uint32_t *s, int64_t dt)
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+{
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+ uint32_t a, c, t = *s;
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+
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+ if (dt >= 0) {
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+ a = LCG_A;
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+ c = LCG_C;
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+ } else { /* coefficients for a step backward */
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+ a = LCG_AI;
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+ c = (uint32_t)(LCG_AI * LCG_C);
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+ dt = -dt;
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+ }
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+ while (dt) {
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+ if (dt & 1)
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+ t = a * t + c;
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+ c *= a + 1; /* coefficients for a double step */
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+ a *= a;
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+ dt >>= 1;
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+ }
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+ *s = t;
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+}
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+
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+/* Emulate pink noise by summing white noise at the sampling frequency,
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+ * white noise at half the sampling frequency (each value taken twice),
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+ * etc., with a total of 8 octaves.
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+ * This is known as the Voss-McCartney algorithm. */
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+
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+static void pink_fill(struct wavesynth_context *ws)
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+{
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+ int32_t vt[7] = { 0 }, v = 0;
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+ int i, j;
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+
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+ ws->pink_pos = 0;
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+ if (!ws->pink_need)
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+ return;
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+ for (i = 0; i < PINK_UNIT; i++) {
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+ for (j = 0; j < 7; j++) {
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+ if ((i >> j) & 1)
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+ break;
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+ v -= vt[j];
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+ vt[j] = (int32_t)lcg_next(&ws->pink_state) >> 3;
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+ v += vt[j];
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+ }
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+ ws->pink_pool[i] = v + ((int32_t)lcg_next(&ws->pink_state) >> 3);
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+ }
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+ lcg_next(&ws->pink_state); /* so we use exactly 256 steps */
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+}
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+
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+/**
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+ * @return (1<<64) * a / b, without overflow, if a < b
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+ */
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+static uint64_t frac64(uint64_t a, uint64_t b)
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+{
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+ uint64_t r = 0;
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+ int i;
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+
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+ if (b < (uint64_t)1 << 32) { /* b small, use two 32-bits steps */
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+ a <<= 32;
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+ return ((a / b) << 32) | ((a % b) << 32) / b;
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+ }
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+ if (b < (uint64_t)1 << 48) { /* b medium, use four 16-bits steps */
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+ for (i = 0; i < 4; i++) {
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+ a <<= 16;
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+ r = (r << 16) | (a / b);
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+ a %= b;
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+ }
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+ return r;
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+ }
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+ for (i = 63; i >= 0; i--) {
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+ if (a >= (uint64_t)1 << 63 || a << 1 >= b) {
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+ r |= (uint64_t)1 << i;
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+ a = (a << 1) - b;
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+ } else {
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+ a <<= 1;
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+ }
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+ }
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+ return r;
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+}
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+
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+static uint64_t phi_at(struct ws_interval *in, int64_t ts)
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+{
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+ uint64_t dt = ts - in->ts_start;
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+ uint64_t dt2 = dt & 1 ? /* dt * (dt - 1) / 2 without overflow */
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+ dt * ((dt - 1) >> 1) : (dt >> 1) * (dt - 1);
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+ return in->phi0 + dt * in->dphi0 + dt2 * in->ddphi;
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+}
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+
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+static void wavesynth_seek(struct wavesynth_context *ws, int64_t ts)
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+{
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+ int *last, i;
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+ struct ws_interval *in;
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+
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+ last = &ws->cur_inter;
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+ for (i = 0; i < ws->nb_inter; i++) {
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+ in = &ws->inter[i];
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+ if (ts < in->ts_start)
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+ break;
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+ if (ts >= in->ts_end)
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+ continue;
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+ *last = i;
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+ last = &in->next;
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+ in->phi = phi_at(in, ts);
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+ in->dphi = in->dphi0 + (ts - in->ts_start) * in->ddphi;
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+ in->amp = in->amp0 + (ts - in->ts_start) * in->damp;
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+ }
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+ ws->next_inter = i;
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+ ws->next_ts = i < ws->nb_inter ? ws->inter[i].ts_start : INF_TS;
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+ *last = -1;
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+ lcg_seek(&ws->dither_state, ts - ws->cur_ts);
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+ if (ws->pink_need) {
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+ int64_t pink_ts_cur = (ws->cur_ts + PINK_UNIT - 1) & ~(PINK_UNIT - 1);
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+ int64_t pink_ts_next = ts & ~(PINK_UNIT - 1);
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+ int pos = ts & (PINK_UNIT - 1);
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+ lcg_seek(&ws->pink_state, (pink_ts_next - pink_ts_cur) << 1);
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+ if (pos) {
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+ pink_fill(ws);
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+ ws->pink_pos = pos;
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+ } else {
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+ ws->pink_pos = PINK_UNIT;
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+ }
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+ }
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+ ws->cur_ts = ts;
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+}
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+
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+static int wavesynth_parse_extradata(AVCodecContext *avc)
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+{
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+ struct wavesynth_context *ws = avc->priv_data;
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+ struct ws_interval *in;
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+ uint8_t *edata, *edata_end;
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+ int32_t f1, f2, a1, a2;
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+ uint32_t phi;
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+ int64_t dphi1, dphi2, dt, cur_ts = -0x8000000000000000;
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+ int i;
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+
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+ if (avc->extradata_size < 4)
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+ return AVERROR(EINVAL);
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+ edata = avc->extradata;
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+ edata_end = edata + avc->extradata_size;
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+ ws->nb_inter = AV_RL32(edata);
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+ edata += 4;
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+ if (ws->nb_inter < 0)
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+ return AVERROR(EINVAL);
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+ ws->inter = av_calloc(ws->nb_inter, sizeof(*ws->inter));
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+ if (!ws->inter)
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+ return AVERROR(ENOMEM);
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+ for (i = 0; i < ws->nb_inter; i++) {
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+ in = &ws->inter[i];
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+ if (edata_end - edata < 24)
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+ return AVERROR(EINVAL);
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+ in->ts_start = AV_RL64(edata + 0);
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+ in->ts_end = AV_RL64(edata + 8);
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+ in->type = AV_RL32(edata + 16);
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+ in->channels = AV_RL32(edata + 20);
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+ edata += 24;
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+ if (in->ts_start < cur_ts || in->ts_end <= in->ts_start)
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+ return AVERROR(EINVAL);
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+ cur_ts = in->ts_start;
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+ dt = in->ts_end - in->ts_start;
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+ switch (in->type) {
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+ case WS_SINE:
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+ if (edata_end - edata < 20)
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+ return AVERROR(EINVAL);
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+ f1 = AV_RL32(edata + 0);
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+ f2 = AV_RL32(edata + 4);
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+ a1 = AV_RL32(edata + 8);
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+ a2 = AV_RL32(edata + 12);
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+ phi = AV_RL32(edata + 16);
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+ edata += 20;
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+ dphi1 = frac64(f1, (int64_t)avc->sample_rate << 16);
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+ dphi2 = frac64(f2, (int64_t)avc->sample_rate << 16);
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+ in->dphi0 = dphi1;
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+ in->ddphi = (dphi2 - dphi1) / dt;
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+ if (phi & 0x80000000) {
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+ phi &= ~0x80000000;
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+ if (phi >= i)
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+ return AVERROR(EINVAL);
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+ in->phi0 = phi_at(&ws->inter[phi], in->ts_start);
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+ } else {
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+ in->phi0 = (uint64_t)phi << 33;
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+ }
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+ break;
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+ case WS_NOISE:
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+ if (edata_end - edata < 8)
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+ return AVERROR(EINVAL);
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+ a1 = AV_RL32(edata + 0);
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+ a2 = AV_RL32(edata + 4);
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+ edata += 8;
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+ break;
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+ default:
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+ return AVERROR(EINVAL);
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+ }
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+ in->amp0 = (int64_t)a1 << 32;
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+ in->damp = (((int64_t)a2 << 32) - ((int64_t)a1 << 32)) / dt;
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+ }
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+ if (edata != edata_end)
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+ return AVERROR(EINVAL);
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+ return 0;
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+}
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+
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+static av_cold int wavesynth_init(AVCodecContext *avc)
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+{
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+ struct wavesynth_context *ws = avc->priv_data;
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+ int i, r;
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+
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+ if (avc->channels > WS_MAX_CHANNELS) {
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+ av_log(avc, AV_LOG_ERROR,
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+ "This implementation is limited to %d channels.\n",
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+ WS_MAX_CHANNELS);
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+ return AVERROR(EINVAL);
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+ }
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+ r = wavesynth_parse_extradata(avc);
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+ if (r < 0) {
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+ av_log(avc, AV_LOG_ERROR, "Invalid intervals definitions.\n");
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+ goto fail;
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+ }
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+ ws->sin = av_malloc(sizeof(*ws->sin) << SIN_BITS);
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+ if (!ws->sin) {
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+ r = AVERROR(ENOMEM);
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+ goto fail;
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+ }
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+ for (i = 0; i < 1 << SIN_BITS; i++)
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+ ws->sin[i] = floor(32767 * sin(2 * M_PI * i / (1 << SIN_BITS)));
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+ ws->dither_state = MKTAG('D','I','T','H');
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+ for (i = 0; i < ws->nb_inter; i++)
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+ ws->pink_need += ws->inter[i].type == WS_NOISE;
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+ ws->pink_state = MKTAG('P','I','N','K');
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+ ws->pink_pos = PINK_UNIT;
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+ avcodec_get_frame_defaults(&ws->frame);
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+ avc->coded_frame = &ws->frame;
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+ wavesynth_seek(ws, 0);
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+ avc->sample_fmt = AV_SAMPLE_FMT_S16;
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+ return 0;
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+
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+fail:
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+ av_free(ws->inter);
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+ av_free(ws->sin);
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+ return r;
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+}
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+
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+static void wavesynth_synth_sample(struct wavesynth_context *ws, int64_t ts,
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+ int32_t *channels)
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+{
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+ int32_t amp, val, *cv;
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+ struct ws_interval *in;
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+ int i, *last, pink;
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+ uint32_t c, all_ch = 0;
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+
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+ i = ws->cur_inter;
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+ last = &ws->cur_inter;
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+ if (ws->pink_pos == PINK_UNIT)
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+ pink_fill(ws);
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+ pink = ws->pink_pool[ws->pink_pos++] >> 16;
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+ while (i >= 0) {
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+ in = &ws->inter[i];
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+ i = in->next;
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+ if (ts >= in->ts_end) {
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+ *last = i;
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+ continue;
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+ }
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+ last = &in->next;
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+ amp = in->amp >> 32;
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+ in->amp += in->damp;
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+ switch (in->type) {
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+ case WS_SINE:
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+ val = amp * ws->sin[in->phi >> (64 - SIN_BITS)];
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+ in->phi += in->dphi;
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+ in->dphi += in->ddphi;
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+ break;
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+ case WS_NOISE:
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+ val = amp * pink;
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+ break;
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+ default:
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+ val = 0;
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+ }
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+ all_ch |= in->channels;
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+ for (c = in->channels, cv = channels; c; c >>= 1, cv++)
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+ if (c & 1)
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+ *cv += val;
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+ }
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+ val = (int32_t)lcg_next(&ws->dither_state) >> 16;
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+ for (c = all_ch, cv = channels; c; c >>= 1, cv++)
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+ if (c & 1)
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|
+ *cv += val;
|
|
|
+}
|
|
|
+
|
|
|
+static void wavesynth_enter_intervals(struct wavesynth_context *ws, int64_t ts)
|
|
|
+{
|
|
|
+ int *last, i;
|
|
|
+ struct ws_interval *in;
|
|
|
+
|
|
|
+ last = &ws->cur_inter;
|
|
|
+ for (i = ws->cur_inter; i >= 0; i = ws->inter[i].next)
|
|
|
+ last = &ws->inter[i].next;
|
|
|
+ for (i = ws->next_inter; i < ws->nb_inter; i++) {
|
|
|
+ in = &ws->inter[i];
|
|
|
+ if (ts < in->ts_start)
|
|
|
+ break;
|
|
|
+ if (ts >= in->ts_end)
|
|
|
+ continue;
|
|
|
+ *last = i;
|
|
|
+ last = &in->next;
|
|
|
+ in->phi = in->phi0;
|
|
|
+ in->dphi = in->dphi0;
|
|
|
+ in->amp = in->amp0;
|
|
|
+ }
|
|
|
+ ws->next_inter = i;
|
|
|
+ ws->next_ts = i < ws->nb_inter ? ws->inter[i].ts_start : INF_TS;
|
|
|
+ *last = -1;
|
|
|
+}
|
|
|
+
|
|
|
+static int wavesynth_decode(AVCodecContext *avc, void *rframe, int *rgot_frame,
|
|
|
+ AVPacket *packet)
|
|
|
+{
|
|
|
+ struct wavesynth_context *ws = avc->priv_data;
|
|
|
+ int64_t ts;
|
|
|
+ int duration;
|
|
|
+ int s, c, r;
|
|
|
+ int16_t *pcm;
|
|
|
+ int32_t channels[WS_MAX_CHANNELS];
|
|
|
+
|
|
|
+ *rgot_frame = 0;
|
|
|
+ if (packet->size != 12)
|
|
|
+ return AVERROR_INVALIDDATA;
|
|
|
+ ts = AV_RL64(packet->data);
|
|
|
+ if (ts != ws->cur_ts)
|
|
|
+ wavesynth_seek(ws, ts);
|
|
|
+ duration = AV_RL32(packet->data + 8);
|
|
|
+ if (duration <= 0)
|
|
|
+ return AVERROR(EINVAL);
|
|
|
+ ws->frame.nb_samples = duration;
|
|
|
+ r = avc->get_buffer(avc, &ws->frame);
|
|
|
+ if (r < 0)
|
|
|
+ return r;
|
|
|
+ pcm = (int16_t *)ws->frame.data[0];
|
|
|
+ for (s = 0; s < duration; s++, ts++) {
|
|
|
+ memset(channels, 0, avc->channels * sizeof(*channels));
|
|
|
+ if (ts >= ws->next_ts)
|
|
|
+ wavesynth_enter_intervals(ws, ts);
|
|
|
+ wavesynth_synth_sample(ws, ts, channels);
|
|
|
+ for (c = 0; c < avc->channels; c++)
|
|
|
+ *(pcm++) = channels[c] >> 16;
|
|
|
+ }
|
|
|
+ ws->cur_ts += duration;
|
|
|
+ *rgot_frame = 1;
|
|
|
+ *(AVFrame *)rframe = ws->frame;
|
|
|
+ return packet->size;
|
|
|
+}
|
|
|
+
|
|
|
+static av_cold int wavesynth_close(AVCodecContext *avc)
|
|
|
+{
|
|
|
+ struct wavesynth_context *ws = avc->priv_data;
|
|
|
+
|
|
|
+ av_free(ws->sin);
|
|
|
+ av_free(ws->inter);
|
|
|
+ return 0;
|
|
|
+}
|
|
|
+
|
|
|
+AVCodec ff_ffwavesynth_decoder = {
|
|
|
+ .name = "wavesynth",
|
|
|
+ .type = AVMEDIA_TYPE_AUDIO,
|
|
|
+ .id = CODEC_ID_FFWAVESYNTH,
|
|
|
+ .priv_data_size = sizeof(struct wavesynth_context),
|
|
|
+ .init = wavesynth_init,
|
|
|
+ .close = wavesynth_close,
|
|
|
+ .decode = wavesynth_decode,
|
|
|
+ .capabilities = CODEC_CAP_DR1,
|
|
|
+ .long_name = NULL_IF_CONFIG_SMALL("Wave synthesis pseudo-codec"),
|
|
|
+};
|