diff options
Diffstat (limited to 'ffmpeg1/libavcodec/truespeech.c')
| -rw-r--r-- | ffmpeg1/libavcodec/truespeech.c | 366 |
1 files changed, 0 insertions, 366 deletions
diff --git a/ffmpeg1/libavcodec/truespeech.c b/ffmpeg1/libavcodec/truespeech.c deleted file mode 100644 index 2eb218c..0000000 --- a/ffmpeg1/libavcodec/truespeech.c +++ /dev/null @@ -1,366 +0,0 @@ -/* - * 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"), -}; |
