making act segmenter

1 files changed, 366 insertions, 0 deletions

diff --git a/ffmpeg/libavcodec/truespeech.c b/ffmpeg/libavcodec/truespeech.c
new file mode 100644
index 0000000..2eb218c
--- /dev/null
+++ b/ffmpeg/libavcodec/truespeech.c
@@ -0,0 +1,366 @@
+/*
+ * DSP Group TrueSpeech compatible decoder
+ * Copyright (c) 2005 Konstantin Shishkov
+ *
+ * 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/channel_layout.h"
+#include "libavutil/intreadwrite.h"
+#include "avcodec.h"
+#include "dsputil.h"
+#include "get_bits.h"
+#include "internal.h"
+
+#include "truespeech_data.h"
+/**
+ * @file
+ * TrueSpeech decoder.
+ */
+
+/**
+ * TrueSpeech decoder context
+ */
+typedef struct {
+    DSPContext dsp;
+    /* input data */
+    DECLARE_ALIGNED(16, uint8_t, buffer)[32];
+    int16_t vector[8];  ///< input vector: 5/5/4/4/4/3/3/3
+    int offset1[2];     ///< 8-bit value, used in one copying offset
+    int offset2[4];     ///< 7-bit value, encodes offsets for copying and for two-point filter
+    int pulseoff[4];    ///< 4-bit offset of pulse values block
+    int pulsepos[4];    ///< 27-bit variable, encodes 7 pulse positions
+    int pulseval[4];    ///< 7x2-bit pulse values
+    int flag;           ///< 1-bit flag, shows how to choose filters
+    /* temporary data */
+    int filtbuf[146];   // some big vector used for storing filters
+    int prevfilt[8];    // filter from previous frame
+    int16_t tmp1[8];    // coefficients for adding to out
+    int16_t tmp2[8];    // coefficients for adding to out
+    int16_t tmp3[8];    // coefficients for adding to out
+    int16_t cvector[8]; // correlated input vector
+    int filtval;        // gain value for one function
+    int16_t newvec[60]; // tmp vector
+    int16_t filters[32]; // filters for every subframe
+} TSContext;
+
+static av_cold int truespeech_decode_init(AVCodecContext * avctx)
+{
+    TSContext *c = avctx->priv_data;
+
+    if (avctx->channels != 1) {
+        avpriv_request_sample(avctx, "Channel count %d", avctx->channels);
+        return AVERROR_PATCHWELCOME;
+    }
+
+    avctx->channel_layout = AV_CH_LAYOUT_MONO;
+    avctx->sample_fmt     = AV_SAMPLE_FMT_S16;
+
+    ff_dsputil_init(&c->dsp, avctx);
+
+    return 0;
+}
+
+static void truespeech_read_frame(TSContext *dec, const uint8_t *input)
+{
+    GetBitContext gb;
+
+    dec->dsp.bswap_buf((uint32_t *)dec->buffer, (const uint32_t *)input, 8);
+    init_get_bits(&gb, dec->buffer, 32 * 8);
+
+    dec->vector[7] = ts_codebook[7][get_bits(&gb, 3)];
+    dec->vector[6] = ts_codebook[6][get_bits(&gb, 3)];
+    dec->vector[5] = ts_codebook[5][get_bits(&gb, 3)];
+    dec->vector[4] = ts_codebook[4][get_bits(&gb, 4)];
+    dec->vector[3] = ts_codebook[3][get_bits(&gb, 4)];
+    dec->vector[2] = ts_codebook[2][get_bits(&gb, 4)];
+    dec->vector[1] = ts_codebook[1][get_bits(&gb, 5)];
+    dec->vector[0] = ts_codebook[0][get_bits(&gb, 5)];
+    dec->flag      = get_bits1(&gb);
+
+    dec->offset1[0] = get_bits(&gb, 4) << 4;
+    dec->offset2[3] = get_bits(&gb, 7);
+    dec->offset2[2] = get_bits(&gb, 7);
+    dec->offset2[1] = get_bits(&gb, 7);
+    dec->offset2[0] = get_bits(&gb, 7);
+
+    dec->offset1[1]  = get_bits(&gb, 4);
+    dec->pulseval[1] = get_bits(&gb, 14);
+    dec->pulseval[0] = get_bits(&gb, 14);
+
+    dec->offset1[1] |= get_bits(&gb, 4) << 4;
+    dec->pulseval[3] = get_bits(&gb, 14);
+    dec->pulseval[2] = get_bits(&gb, 14);
+
+    dec->offset1[0] |= get_bits1(&gb);
+    dec->pulsepos[0] = get_bits_long(&gb, 27);
+    dec->pulseoff[0] = get_bits(&gb, 4);
+
+    dec->offset1[0] |= get_bits1(&gb) << 1;
+    dec->pulsepos[1] = get_bits_long(&gb, 27);
+    dec->pulseoff[1] = get_bits(&gb, 4);
+
+    dec->offset1[0] |= get_bits1(&gb) << 2;
+    dec->pulsepos[2] = get_bits_long(&gb, 27);
+    dec->pulseoff[2] = get_bits(&gb, 4);
+
+    dec->offset1[0] |= get_bits1(&gb) << 3;
+    dec->pulsepos[3] = get_bits_long(&gb, 27);
+    dec->pulseoff[3] = get_bits(&gb, 4);
+}
+
+static void truespeech_correlate_filter(TSContext *dec)
+{
+    int16_t tmp[8];
+    int i, j;
+
+    for(i = 0; i < 8; i++){
+        if(i > 0){
+            memcpy(tmp, dec->cvector, i * sizeof(*tmp));
+            for(j = 0; j < i; j++)
+                dec->cvector[j] = ((tmp[i - j - 1] * dec->vector[i]) +
+                                   (dec->cvector[j] << 15) + 0x4000) >> 15;
+        }
+        dec->cvector[i] = (8 - dec->vector[i]) >> 3;
+    }
+    for(i = 0; i < 8; i++)
+        dec->cvector[i] = (dec->cvector[i] * ts_decay_994_1000[i]) >> 15;
+
+    dec->filtval = dec->vector[0];
+}
+
+static void truespeech_filters_merge(TSContext *dec)
+{
+    int i;
+
+    if(!dec->flag){
+        for(i = 0; i < 8; i++){
+            dec->filters[i + 0] = dec->prevfilt[i];
+            dec->filters[i + 8] = dec->prevfilt[i];
+        }
+    }else{
+        for(i = 0; i < 8; i++){
+            dec->filters[i + 0]=(dec->cvector[i] * 21846 + dec->prevfilt[i] * 10923 + 16384) >> 15;
+            dec->filters[i + 8]=(dec->cvector[i] * 10923 + dec->prevfilt[i] * 21846 + 16384) >> 15;
+        }
+    }
+    for(i = 0; i < 8; i++){
+        dec->filters[i + 16] = dec->cvector[i];
+        dec->filters[i + 24] = dec->cvector[i];
+    }
+}
+
+static void truespeech_apply_twopoint_filter(TSContext *dec, int quart)
+{
+    int16_t tmp[146 + 60], *ptr0, *ptr1;
+    const int16_t *filter;
+    int i, t, off;
+
+    t = dec->offset2[quart];
+    if(t == 127){
+        memset(dec->newvec, 0, 60 * sizeof(*dec->newvec));
+        return;
+    }
+    for(i = 0; i < 146; i++)
+        tmp[i] = dec->filtbuf[i];
+    off = (t / 25) + dec->offset1[quart >> 1] + 18;
+    off = av_clip(off, 0, 145);
+    ptr0 = tmp + 145 - off;
+    ptr1 = tmp + 146;
+    filter = ts_order2_coeffs + (t % 25) * 2;
+    for(i = 0; i < 60; i++){
+        t = (ptr0[0] * filter[0] + ptr0[1] * filter[1] + 0x2000) >> 14;
+        ptr0++;
+        dec->newvec[i] = t;
+        ptr1[i] = t;
+    }
+}
+
+static void truespeech_place_pulses(TSContext *dec, int16_t *out, int quart)
+{
+    int16_t tmp[7];
+    int i, j, t;
+    const int16_t *ptr1;
+    int16_t *ptr2;
+    int coef;
+
+    memset(out, 0, 60 * sizeof(*out));
+    for(i = 0; i < 7; i++) {
+        t = dec->pulseval[quart] & 3;
+        dec->pulseval[quart] >>= 2;
+        tmp[6 - i] = ts_pulse_scales[dec->pulseoff[quart] * 4 + t];
+    }
+
+    coef = dec->pulsepos[quart] >> 15;
+    ptr1 = ts_pulse_values + 30;
+    ptr2 = tmp;
+    for(i = 0, j = 3; (i < 30) && (j > 0); i++){
+        t = *ptr1++;
+        if(coef >= t)
+            coef -= t;
+        else{
+            out[i] = *ptr2++;
+            ptr1 += 30;
+            j--;
+        }
+    }
+    coef = dec->pulsepos[quart] & 0x7FFF;
+    ptr1 = ts_pulse_values;
+    for(i = 30, j = 4; (i < 60) && (j > 0); i++){
+        t = *ptr1++;
+        if(coef >= t)
+            coef -= t;
+        else{
+            out[i] = *ptr2++;
+            ptr1 += 30;
+            j--;
+        }
+    }
+
+}
+
+static void truespeech_update_filters(TSContext *dec, int16_t *out, int quart)
+{
+    int i;
+
+    memmove(dec->filtbuf, &dec->filtbuf[60], 86 * sizeof(*dec->filtbuf));
+    for(i = 0; i < 60; i++){
+        dec->filtbuf[i + 86] = out[i] + dec->newvec[i] - (dec->newvec[i] >> 3);
+        out[i] += dec->newvec[i];
+    }
+}
+
+static void truespeech_synth(TSContext *dec, int16_t *out, int quart)
+{
+    int i,k;
+    int t[8];
+    int16_t *ptr0, *ptr1;
+
+    ptr0 = dec->tmp1;
+    ptr1 = dec->filters + quart * 8;
+    for(i = 0; i < 60; i++){
+        int sum = 0;
+        for(k = 0; k < 8; k++)
+            sum += ptr0[k] * ptr1[k];
+        sum = (sum + (out[i] << 12) + 0x800) >> 12;
+        out[i] = av_clip(sum, -0x7FFE, 0x7FFE);
+        for(k = 7; k > 0; k--)
+            ptr0[k] = ptr0[k - 1];
+        ptr0[0] = out[i];
+    }
+
+    for(i = 0; i < 8; i++)
+        t[i] = (ts_decay_35_64[i] * ptr1[i]) >> 15;
+
+    ptr0 = dec->tmp2;
+    for(i = 0; i < 60; i++){
+        int sum = 0;
+        for(k = 0; k < 8; k++)
+            sum += ptr0[k] * t[k];
+        for(k = 7; k > 0; k--)
+            ptr0[k] = ptr0[k - 1];
+        ptr0[0] = out[i];
+        out[i] = ((out[i] << 12) - sum) >> 12;
+    }
+
+    for(i = 0; i < 8; i++)
+        t[i] = (ts_decay_3_4[i] * ptr1[i]) >> 15;
+
+    ptr0 = dec->tmp3;
+    for(i = 0; i < 60; i++){
+        int sum = out[i] << 12;
+        for(k = 0; k < 8; k++)
+            sum += ptr0[k] * t[k];
+        for(k = 7; k > 0; k--)
+            ptr0[k] = ptr0[k - 1];
+        ptr0[0] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
+
+        sum = ((ptr0[1] * (dec->filtval - (dec->filtval >> 2))) >> 4) + sum;
+        sum = sum - (sum >> 3);
+        out[i] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
+    }
+}
+
+static void truespeech_save_prevvec(TSContext *c)
+{
+    int i;
+
+    for(i = 0; i < 8; i++)
+        c->prevfilt[i] = c->cvector[i];
+}
+
+static int truespeech_decode_frame(AVCodecContext *avctx, void *data,
+                                   int *got_frame_ptr, AVPacket *avpkt)
+{
+    AVFrame *frame     = data;
+    const uint8_t *buf = avpkt->data;
+    int buf_size = avpkt->size;
+    TSContext *c = avctx->priv_data;
+
+    int i, j;
+    int16_t *samples;
+    int iterations, ret;
+
+    iterations = buf_size / 32;
+
+    if (!iterations) {
+        av_log(avctx, AV_LOG_ERROR,
+               "Too small input buffer (%d bytes), need at least 32 bytes\n", buf_size);
+        return -1;
+    }
+
+    /* get output buffer */
+    frame->nb_samples = iterations * 240;
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+        return ret;
+    samples = (int16_t *)frame->data[0];
+
+    memset(samples, 0, iterations * 240 * sizeof(*samples));
+
+    for(j = 0; j < iterations; j++) {
+        truespeech_read_frame(c, buf);
+        buf += 32;
+
+        truespeech_correlate_filter(c);
+        truespeech_filters_merge(c);
+
+        for(i = 0; i < 4; i++) {
+            truespeech_apply_twopoint_filter(c, i);
+            truespeech_place_pulses  (c, samples, i);
+            truespeech_update_filters(c, samples, i);
+            truespeech_synth         (c, samples, i);
+            samples += 60;
+        }
+
+        truespeech_save_prevvec(c);
+    }
+
+    *got_frame_ptr = 1;
+
+    return buf_size;
+}
+
+AVCodec ff_truespeech_decoder = {
+    .name           = "truespeech",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_TRUESPEECH,
+    .priv_data_size = sizeof(TSContext),
+    .init           = truespeech_decode_init,
+    .decode         = truespeech_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"),
+};