| /* |
| * 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 "bswapdsp.h" |
| #include "get_bits.h" |
| #include "internal.h" |
| |
| #include "truespeech_data.h" |
| /** |
| * @file |
| * TrueSpeech decoder. |
| */ |
| |
| /** |
| * TrueSpeech decoder context |
| */ |
| typedef struct TSContext { |
| BswapDSPContext bdsp; |
| /* 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_bswapdsp_init(&c->bdsp); |
| |
| return 0; |
| } |
| |
| static void truespeech_read_frame(TSContext *dec, const uint8_t *input) |
| { |
| GetBitContext gb; |
| |
| dec->bdsp.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", |
| .long_name = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"), |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = AV_CODEC_ID_TRUESPEECH, |
| .priv_data_size = sizeof(TSContext), |
| .init = truespeech_decode_init, |
| .decode = truespeech_decode_frame, |
| .capabilities = AV_CODEC_CAP_DR1, |
| }; |
