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Diffstat (limited to 'ffmpeg/libavcodec/truespeech.c')
-rw-r--r--ffmpeg/libavcodec/truespeech.c366
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"),
+};
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