lavc: add ffwavesynth pseudo-codec.

6 files changed, 499 insertions, 1 deletions

diff --git a/doc/decoders.texi b/doc/decoders.texi
index e7baadafc2..87ad4eea26 100644
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -48,3 +48,16 @@ top-field-first is assumed
 @end table
 
 @c man end VIDEO DECODERS
+
+@chapter Audio Decoders
+@c man begin AUDIO DECODERS
+
+@section ffwavesynth
+
+Internal wave synthetizer.
+
+This decoder generates wave patterns according to predefined sequences. Its
+use is purely internal and the format of the data it accepts is not publicly
+documented.
+
+@c man end AUDIO DECODERS
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 58b6431f70..9289494e05 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -156,6 +156,7 @@ OBJS-$(CONFIG_FFV1_DECODER)            += ffv1.o rangecoder.o
 OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1.o rangecoder.o
 OBJS-$(CONFIG_FFVHUFF_DECODER)         += huffyuv.o
 OBJS-$(CONFIG_FFVHUFF_ENCODER)         += huffyuv.o
+OBJS-$(CONFIG_FFWAVESYNTH_DECODER)     += ffwavesynth.o
 OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLAC_ENCODER)            += flacenc.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLASHSV_DECODER)         += flashsv.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 4c85056af8..3994f92d0b 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -265,6 +265,7 @@ void avcodec_register_all(void)
     REGISTER_ENCDEC  (DCA, dca);
     REGISTER_DECODER (DSICINAUDIO, dsicinaudio);
     REGISTER_ENCDEC  (EAC3, eac3);
+    REGISTER_DECODER (FFWAVESYNTH, ffwavesynth);
     REGISTER_ENCDEC  (FLAC, flac);
     REGISTER_ENCDEC  (G723_1, g723_1);
     REGISTER_DECODER (G729, g729);
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 5a0e0ed380..6d4489c2a4 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -403,6 +403,7 @@ enum CodecID {
     CODEC_ID_BMV_AUDIO,
     CODEC_ID_G729 = 0x15800,
     CODEC_ID_G723_1= 0x15801,
+    CODEC_ID_FFWAVESYNTH = MKBETAG('F','F','W','S'),
     CODEC_ID_8SVX_RAW   = MKBETAG('8','S','V','X'),
 
     /* subtitle codecs */
diff --git a/libavcodec/ffwavesynth.c b/libavcodec/ffwavesynth.c
new file mode 100644
index 0000000000..d18dd91452
--- /dev/null
+++ b/libavcodec/ffwavesynth.c
@@ -0,0 +1,482 @@
+/*
+ * 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"
+
+#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 = 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"),
+};
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 84d4dadfa1..a71e02e75e 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -21,7 +21,7 @@
 #define AVCODEC_VERSION_H
 
 #define LIBAVCODEC_VERSION_MAJOR 53
-#define LIBAVCODEC_VERSION_MINOR 45
+#define LIBAVCODEC_VERSION_MINOR 46
 #define LIBAVCODEC_VERSION_MICRO  0
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \