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Diffstat (limited to 'libavcodec/ffwavesynth.c')
| -rw-r--r-- | libavcodec/ffwavesynth.c | 484 |
1 files changed, 484 insertions, 0 deletions
diff --git a/libavcodec/ffwavesynth.c b/libavcodec/ffwavesynth.c new file mode 100644 index 0000000000..4f392f2f3c --- /dev/null +++ b/libavcodec/ffwavesynth.c @@ -0,0 +1,484 @@ +/* + * 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"), +}; |