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-rw-r--r--libavcodec/ffwavesynth.c484
1 files changed, 484 insertions, 0 deletions
diff --git a/libavcodec/ffwavesynth.c b/libavcodec/ffwavesynth.c
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+++ b/libavcodec/ffwavesynth.c
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+/*
+ * Wavesynth pseudo-codec
+ * Copyright (c) 2011 Nicolas George
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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,
+ * 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
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/intreadwrite.h"
+#include "libavutil/log.h"
+#include "avcodec.h"
+#include "internal.h"
+
+
+#define SIN_BITS 14
+#define WS_MAX_CHANNELS 32
+#define INF_TS 0x7FFFFFFFFFFFFFFF
+
+#define PINK_UNIT 128
+
+/*
+ Format of the extradata and packets
+
+ THIS INFORMATION IS NOT PART OF THE PUBLIC API OR ABI.
+ IT CAN CHANGE WITHOUT NOTIFICATION.
+
+ All numbers are in little endian.
+
+ The codec extradata define a set of intervals with uniform content.
+ Overlapping intervals are added together.
+
+ extradata:
+ uint32 number of intervals
+ ... intervals
+
+ interval:
+ int64 start timestamp; time_base must be 1/sample_rate;
+ start timestamps must be in ascending order
+ int64 end timestamp
+ uint32 type
+ uint32 channels mask
+ ... additional information, depends on type
+
+ sine interval (type fourcc "SINE"):
+ int32 start frequency, in 1/(1<<16) Hz
+ int32 end frequency
+ int32 start amplitude, 1<<16 is the full amplitude
+ int32 end amplitude
+ uint32 start phase, 0 is sin(0), 0x20000000 is sin(pi/2), etc.;
+ n | (1<<31) means to match the phase of previous channel #n
+
+ pink noise interval (type fourcc "NOIS"):
+ int32 start amplitude
+ int32 end amplitude
+
+ The input packets encode the time and duration of the requested segment.
+
+ packet:
+ int64 start timestamp
+ int32 duration
+
+*/
+
+enum ws_interval_type {
+ WS_SINE = MKTAG('S','I','N','E'),
+ WS_NOISE = MKTAG('N','O','I','S'),
+};
+
+struct ws_interval {
+ int64_t ts_start, ts_end;
+ uint64_t phi0, dphi0, ddphi;
+ uint64_t amp0, damp;
+ uint64_t phi, dphi, amp;
+ uint32_t channels;
+ enum ws_interval_type type;
+ int next;
+};
+
+struct wavesynth_context {
+ int64_t cur_ts;
+ int64_t next_ts;
+ int32_t *sin;
+ AVFrame frame;
+ struct ws_interval *inter;
+ uint32_t dither_state;
+ uint32_t pink_state;
+ int32_t pink_pool[PINK_UNIT];
+ unsigned pink_need, pink_pos;
+ int nb_inter;
+ int cur_inter;
+ int next_inter;
+};
+
+#define LCG_A 1284865837
+#define LCG_C 4150755663
+#define LCG_AI 849225893 /* A*AI = 1 [mod 1<<32] */
+
+static uint32_t lcg_next(uint32_t *s)
+{
+ *s = *s * LCG_A + LCG_C;
+ return *s;
+}
+
+static void lcg_seek(uint32_t *s, int64_t dt)
+{
+ uint32_t a, c, t = *s;
+
+ if (dt >= 0) {
+ a = LCG_A;
+ c = LCG_C;
+ } else { /* coefficients for a step backward */
+ a = LCG_AI;
+ c = (uint32_t)(LCG_AI * LCG_C);
+ dt = -dt;
+ }
+ while (dt) {
+ if (dt & 1)
+ t = a * t + c;
+ c *= a + 1; /* coefficients for a double step */
+ a *= a;
+ dt >>= 1;
+ }
+ *s = t;
+}
+
+/* Emulate pink noise by summing white noise at the sampling frequency,
+ * white noise at half the sampling frequency (each value taken twice),
+ * etc., with a total of 8 octaves.
+ * This is known as the Voss-McCartney algorithm. */
+
+static void pink_fill(struct wavesynth_context *ws)
+{
+ int32_t vt[7] = { 0 }, v = 0;
+ int i, j;
+
+ ws->pink_pos = 0;
+ if (!ws->pink_need)
+ return;
+ for (i = 0; i < PINK_UNIT; i++) {
+ for (j = 0; j < 7; j++) {
+ if ((i >> j) & 1)
+ break;
+ v -= vt[j];
+ vt[j] = (int32_t)lcg_next(&ws->pink_state) >> 3;
+ v += vt[j];
+ }
+ ws->pink_pool[i] = v + ((int32_t)lcg_next(&ws->pink_state) >> 3);
+ }
+ lcg_next(&ws->pink_state); /* so we use exactly 256 steps */
+}
+
+/**
+ * @return (1<<64) * a / b, without overflow, if a < b
+ */
+static uint64_t frac64(uint64_t a, uint64_t b)
+{
+ uint64_t r = 0;
+ int i;
+
+ if (b < (uint64_t)1 << 32) { /* b small, use two 32-bits steps */
+ a <<= 32;
+ return ((a / b) << 32) | ((a % b) << 32) / b;
+ }
+ if (b < (uint64_t)1 << 48) { /* b medium, use four 16-bits steps */
+ for (i = 0; i < 4; i++) {
+ a <<= 16;
+ r = (r << 16) | (a / b);
+ a %= b;
+ }
+ return r;
+ }
+ for (i = 63; i >= 0; i--) {
+ if (a >= (uint64_t)1 << 63 || a << 1 >= b) {
+ r |= (uint64_t)1 << i;
+ a = (a << 1) - b;
+ } else {
+ a <<= 1;
+ }
+ }
+ return r;
+}
+
+static uint64_t phi_at(struct ws_interval *in, int64_t ts)
+{
+ uint64_t dt = ts - in->ts_start;
+ uint64_t dt2 = dt & 1 ? /* dt * (dt - 1) / 2 without overflow */
+ dt * ((dt - 1) >> 1) : (dt >> 1) * (dt - 1);
+ return in->phi0 + dt * in->dphi0 + dt2 * in->ddphi;
+}
+
+static void wavesynth_seek(struct wavesynth_context *ws, int64_t ts)
+{
+ int *last, i;
+ struct ws_interval *in;
+
+ last = &ws->cur_inter;
+ for (i = 0; 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 = phi_at(in, ts);
+ in->dphi = in->dphi0 + (ts - in->ts_start) * in->ddphi;
+ in->amp = in->amp0 + (ts - in->ts_start) * in->damp;
+ }
+ ws->next_inter = i;
+ ws->next_ts = i < ws->nb_inter ? ws->inter[i].ts_start : INF_TS;
+ *last = -1;
+ lcg_seek(&ws->dither_state, ts - ws->cur_ts);
+ if (ws->pink_need) {
+ int64_t pink_ts_cur = (ws->cur_ts + PINK_UNIT - 1) & ~(PINK_UNIT - 1);
+ int64_t pink_ts_next = ts & ~(PINK_UNIT - 1);
+ int pos = ts & (PINK_UNIT - 1);
+ lcg_seek(&ws->pink_state, (pink_ts_next - pink_ts_cur) << 1);
+ if (pos) {
+ pink_fill(ws);
+ ws->pink_pos = pos;
+ } else {
+ ws->pink_pos = PINK_UNIT;
+ }
+ }
+ ws->cur_ts = ts;
+}
+
+static int wavesynth_parse_extradata(AVCodecContext *avc)
+{
+ struct wavesynth_context *ws = avc->priv_data;
+ struct ws_interval *in;
+ uint8_t *edata, *edata_end;
+ int32_t f1, f2, a1, a2;
+ uint32_t phi;
+ int64_t dphi1, dphi2, dt, cur_ts = -0x8000000000000000;
+ int i;
+
+ if (avc->extradata_size < 4)
+ return AVERROR(EINVAL);
+ edata = avc->extradata;
+ edata_end = edata + avc->extradata_size;
+ ws->nb_inter = AV_RL32(edata);
+ edata += 4;
+ if (ws->nb_inter < 0)
+ return AVERROR(EINVAL);
+ ws->inter = av_calloc(ws->nb_inter, sizeof(*ws->inter));
+ if (!ws->inter)
+ return AVERROR(ENOMEM);
+ for (i = 0; i < ws->nb_inter; i++) {
+ in = &ws->inter[i];
+ if (edata_end - edata < 24)
+ return AVERROR(EINVAL);
+ in->ts_start = AV_RL64(edata + 0);
+ in->ts_end = AV_RL64(edata + 8);
+ in->type = AV_RL32(edata + 16);
+ in->channels = AV_RL32(edata + 20);
+ edata += 24;
+ if (in->ts_start < cur_ts || in->ts_end <= in->ts_start)
+ return AVERROR(EINVAL);
+ cur_ts = in->ts_start;
+ dt = in->ts_end - in->ts_start;
+ switch (in->type) {
+ case WS_SINE:
+ if (edata_end - edata < 20)
+ return AVERROR(EINVAL);
+ f1 = AV_RL32(edata + 0);
+ f2 = AV_RL32(edata + 4);
+ a1 = AV_RL32(edata + 8);
+ a2 = AV_RL32(edata + 12);
+ phi = AV_RL32(edata + 16);
+ edata += 20;
+ dphi1 = frac64(f1, (int64_t)avc->sample_rate << 16);
+ dphi2 = frac64(f2, (int64_t)avc->sample_rate << 16);
+ in->dphi0 = dphi1;
+ in->ddphi = (dphi2 - dphi1) / dt;
+ if (phi & 0x80000000) {
+ phi &= ~0x80000000;
+ if (phi >= i)
+ return AVERROR(EINVAL);
+ in->phi0 = phi_at(&ws->inter[phi], in->ts_start);
+ } else {
+ in->phi0 = (uint64_t)phi << 33;
+ }
+ break;
+ case WS_NOISE:
+ if (edata_end - edata < 8)
+ return AVERROR(EINVAL);
+ a1 = AV_RL32(edata + 0);
+ a2 = AV_RL32(edata + 4);
+ edata += 8;
+ break;
+ default:
+ return AVERROR(EINVAL);
+ }
+ in->amp0 = (int64_t)a1 << 32;
+ in->damp = (((int64_t)a2 << 32) - ((int64_t)a1 << 32)) / dt;
+ }
+ if (edata != edata_end)
+ return AVERROR(EINVAL);
+ return 0;
+}
+
+static av_cold int wavesynth_init(AVCodecContext *avc)
+{
+ struct wavesynth_context *ws = avc->priv_data;
+ int i, r;
+
+ if (avc->channels > WS_MAX_CHANNELS) {
+ av_log(avc, AV_LOG_ERROR,
+ "This implementation is limited to %d channels.\n",
+ WS_MAX_CHANNELS);
+ return AVERROR(EINVAL);
+ }
+ r = wavesynth_parse_extradata(avc);
+ if (r < 0) {
+ av_log(avc, AV_LOG_ERROR, "Invalid intervals definitions.\n");
+ goto fail;
+ }
+ ws->sin = av_malloc(sizeof(*ws->sin) << SIN_BITS);
+ if (!ws->sin) {
+ r = AVERROR(ENOMEM);
+ goto fail;
+ }
+ for (i = 0; i < 1 << SIN_BITS; i++)
+ ws->sin[i] = floor(32767 * sin(2 * M_PI * i / (1 << SIN_BITS)));
+ ws->dither_state = MKTAG('D','I','T','H');
+ for (i = 0; i < ws->nb_inter; i++)
+ ws->pink_need += ws->inter[i].type == WS_NOISE;
+ ws->pink_state = MKTAG('P','I','N','K');
+ ws->pink_pos = PINK_UNIT;
+ avcodec_get_frame_defaults(&ws->frame);
+ avc->coded_frame = &ws->frame;
+ wavesynth_seek(ws, 0);
+ avc->sample_fmt = AV_SAMPLE_FMT_S16;
+ return 0;
+
+fail:
+ av_free(ws->inter);
+ av_free(ws->sin);
+ return r;
+}
+
+static void wavesynth_synth_sample(struct wavesynth_context *ws, int64_t ts,
+ int32_t *channels)
+{
+ int32_t amp, val, *cv;
+ struct ws_interval *in;
+ int i, *last, pink;
+ uint32_t c, all_ch = 0;
+
+ i = ws->cur_inter;
+ last = &ws->cur_inter;
+ if (ws->pink_pos == PINK_UNIT)
+ pink_fill(ws);
+ pink = ws->pink_pool[ws->pink_pos++] >> 16;
+ while (i >= 0) {
+ in = &ws->inter[i];
+ i = in->next;
+ if (ts >= in->ts_end) {
+ *last = i;
+ continue;
+ }
+ last = &in->next;
+ amp = in->amp >> 32;
+ in->amp += in->damp;
+ switch (in->type) {
+ case WS_SINE:
+ val = amp * ws->sin[in->phi >> (64 - SIN_BITS)];
+ in->phi += in->dphi;
+ in->dphi += in->ddphi;
+ break;
+ case WS_NOISE:
+ val = amp * pink;
+ break;
+ default:
+ val = 0;
+ }
+ all_ch |= in->channels;
+ for (c = in->channels, cv = channels; c; c >>= 1, cv++)
+ if (c & 1)
+ *cv += val;
+ }
+ val = (int32_t)lcg_next(&ws->dither_state) >> 16;
+ for (c = all_ch, cv = channels; c; c >>= 1, cv++)
+ if (c & 1)
+ *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 = ff_get_buffer(avc, &ws->frame, 0);
+ 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 = AV_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"),
+};