diff options
author | Paul B Mahol <onemda@gmail.com> | 2021-09-13 17:00:38 +0200 |
---|---|---|
committer | Paul B Mahol <onemda@gmail.com> | 2021-10-10 17:44:41 +0200 |
commit | 148ada5577262c6c18ae97604df8fe1c18b096e2 (patch) | |
tree | 5aab26e0299c8f989a24a5495294c6c8069969c9 /libavcodec/speexdec.c | |
parent | 5274f2f7f8c5e40d18b84055fbb232752bd24f2f (diff) |
avcodec: add native Speex decoder
Diffstat (limited to 'libavcodec/speexdec.c')
-rw-r--r-- | libavcodec/speexdec.c | 1590 |
1 files changed, 1590 insertions, 0 deletions
diff --git a/libavcodec/speexdec.c b/libavcodec/speexdec.c new file mode 100644 index 0000000000..35270e6723 --- /dev/null +++ b/libavcodec/speexdec.c @@ -0,0 +1,1590 @@ +/* + * Copyright 2002-2008 Xiph.org Foundation + * Copyright 2002-2008 Jean-Marc Valin + * Copyright 2005-2007 Analog Devices Inc. + * Copyright 2005-2008 Commonwealth Scientific and Industrial Research Organisation (CSIRO) + * Copyright 1993, 2002, 2006 David Rowe + * Copyright 2003 EpicGames + * Copyright 1992-1994 Jutta Degener, Carsten Bormann + + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + + * - Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. + + * - Redistributions in binary form must reproduce the above copyright + * notice, this list of conditions and the following disclaimer in the + * documentation and/or other materials provided with the distribution. + + * - Neither the name of the Xiph.org Foundation nor the names of its + * contributors may be used to endorse or promote products derived from + * this software without specific prior written permission. + + * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + * + * 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/avassert.h" +#include "libavutil/float_dsp.h" +#include "avcodec.h" +#include "bytestream.h" +#include "get_bits.h" +#include "internal.h" +#include "speexdata.h" + +#define SPEEX_NB_MODES 3 +#define SPEEX_INBAND_STEREO 9 + +#define QMF_ORDER 64 +#define NB_ORDER 10 +#define NB_FRAME_SIZE 160 +#define NB_SUBMODES 9 +#define NB_SUBMODE_BITS 4 +#define SB_SUBMODE_BITS 3 + +#define NB_SUBFRAME_SIZE 40 +#define NB_NB_SUBFRAMES 4 +#define NB_PITCH_START 17 +#define NB_PITCH_END 144 + +#define NB_DEC_BUFFER (NB_FRAME_SIZE + 2 * NB_PITCH_END + NB_SUBFRAME_SIZE + 12) + +#define SPEEX_MEMSET(dst, c, n) (memset((dst), (c), (n) * sizeof(*(dst)))) +#define SPEEX_COPY(dst, src, n) (memcpy((dst), (src), (n) * sizeof(*(dst)))) + +#define LSP_LINEAR(i) (.25f * (i) + .25f) +#define LSP_LINEAR_HIGH(i) (.3125f * (i) + .75f) +#define LSP_DIV_256(x) (0.00390625f * (x)) +#define LSP_DIV_512(x) (0.001953125f * (x)) +#define LSP_DIV_1024(x) (0.0009765625f * (x)) + +typedef struct LtpParams { + const int8_t *gain_cdbk; + int gain_bits; + int pitch_bits; +} LtpParam; + +static const LtpParam ltp_params_vlbr = { gain_cdbk_lbr, 5, 0 }; +static const LtpParam ltp_params_lbr = { gain_cdbk_lbr, 5, 7 }; +static const LtpParam ltp_params_med = { gain_cdbk_lbr, 5, 7 }; +static const LtpParam ltp_params_nb = { gain_cdbk_nb, 7, 7 }; + +typedef struct SplitCodebookParams { + int subvect_size; + int nb_subvect; + const signed char *shape_cb; + int shape_bits; + int have_sign; +} SplitCodebookParams; + +static const SplitCodebookParams split_cb_nb_ulbr = { 20, 2, exc_20_32_table, 5, 0 }; +static const SplitCodebookParams split_cb_nb_vlbr = { 10, 4, exc_10_16_table, 4, 0 }; +static const SplitCodebookParams split_cb_nb_lbr = { 10, 4, exc_10_32_table, 5, 0 }; +static const SplitCodebookParams split_cb_nb_med = { 8, 5, exc_8_128_table, 7, 0 }; +static const SplitCodebookParams split_cb_nb = { 5, 8, exc_5_64_table, 6, 0 }; +static const SplitCodebookParams split_cb_sb = { 5, 8, exc_5_256_table, 8, 0 }; +static const SplitCodebookParams split_cb_high = { 8, 5, hexc_table, 7, 1 }; +static const SplitCodebookParams split_cb_high_lbr= { 10, 4, hexc_10_32_table,5, 0 }; + +/** Quantizes LSPs */ +typedef void (*lsp_quant_func)(float *, float *, int, GetBitContext *); + +/** Decodes quantized LSPs */ +typedef void (*lsp_unquant_func)(float *, int, GetBitContext *); + +/** Long-term predictor quantization */ +typedef int (*ltp_quant_func)(float *, float *, float *, + float *, float *, float *, + const void *, int, int, float, int, int, + GetBitContext *, char *, float *, + float *, int, int, int, float *); + +/** Long-term un-quantize */ +typedef void (*ltp_unquant_func)(float *, float *, int, int, + float, const void *, int, int *, + float *, GetBitContext *, int, int, + float, int); + +/** Innovation quantization function */ +typedef void (*innovation_quant_func)(float *, float *, + float *, float *, const void *, + int, int, float *, float *, + GetBitContext *, char *, int, int); + +/** Innovation unquantization function */ +typedef void (*innovation_unquant_func)(float *, const void *, int, + GetBitContext *, int32_t *); + +typedef struct SpeexSubmode { + int lbr_pitch; /**< Set to -1 for "normal" modes, otherwise encode pitch using + a global pitch and allowing a +- lbr_pitch variation (for + low not-rates)*/ + int forced_pitch_gain; /**< Use the same (forced) pitch gain for all + sub-frames */ + int have_subframe_gain; /**< Number of bits to use as sub-frame innovation + gain */ + int double_codebook; /**< Apply innovation quantization twice for higher + quality (and higher bit-rate)*/ + lsp_unquant_func lsp_unquant; /**< LSP unquantization function */ + + ltp_unquant_func ltp_unquant; /**< Long-term predictor (pitch) un-quantizer */ + const void *LtpParam; /**< Pitch parameters (options) */ + + innovation_unquant_func innovation_unquant; /**< Innovation un-quantization */ + const void *innovation_params; /**< Innovation quantization parameters*/ + + float comb_gain; /**< Gain of enhancer comb filter */ +} SpeexSubmode; + +typedef struct SpeexMode { + int modeID; /** ID of the mode */ + int (*decode)(AVCodecContext *avctx, void *dec, GetBitContext *gb, float *out); + int frame_size; /**< Size of frames used for decoding */ + int subframe_size; /**< Size of sub-frames used for decoding */ + int lpc_size; /**< Order of LPC filter */ + float folding_gain; /**< Folding gain */ + const SpeexSubmode *submodes[NB_SUBMODES]; /**< Sub-mode data for the mode */ + int default_submode; /**< Default sub-mode to use when decoding */ +} SpeexMode; + +typedef struct DecoderState { + const SpeexMode *mode; + int modeID; /** ID of the decoder mode */ + int first; /** Is first frame */ + int full_frame_size; /**< Length of full-band frames */ + int is_wideband; /**< If wideband is present */ + int count_lost; /**< Was the last frame lost? */ + int frame_size; /**< Length of high-band frames */ + int subframe_size; /**< Length of high-band sub-frames */ + int nb_subframes; /**< Number of high-band sub-frames */ + int lpc_size; /**< Order of high-band LPC analysis */ + float last_ol_gain; /**< Open-loop gain for previous frame */ + float *innov_save; /** If non-NULL, innovation is copied here */ + + /* This is used in packet loss concealment */ + int last_pitch; /**< Pitch of last correctly decoded frame */ + float last_pitch_gain; /**< Pitch gain of last correctly decoded frame */ + int32_t seed; /** Seed used for random number generation */ + + int encode_submode; + const SpeexSubmode *const *submodes; /**< Sub-mode data */ + int submodeID; /**< Activated sub-mode */ + int lpc_enh_enabled; /**< 1 when LPC enhancer is on, 0 otherwise */ + + /* Vocoder data */ + float voc_m1; + float voc_m2; + float voc_mean; + int voc_offset; + + int dtx_enabled; + int highpass_enabled; /**< Is the input filter enabled */ + + float *exc; /**< Start of excitation frame */ + float mem_hp[2]; /**< High-pass filter memory */ + float exc_buf[NB_DEC_BUFFER]; /**< Excitation buffer */ + float old_qlsp[NB_ORDER]; /**< Quantized LSPs for previous frame */ + float interp_qlpc[NB_ORDER]; /**< Interpolated quantized LPCs */ + float mem_sp[NB_ORDER]; /**< Filter memory for synthesis signal */ + float g0_mem[QMF_ORDER]; + float g1_mem[QMF_ORDER]; + float pi_gain[NB_NB_SUBFRAMES]; /**< Gain of LPC filter at theta=pi (fe/2) */ + float exc_rms[NB_NB_SUBFRAMES]; /**< RMS of excitation per subframe */ +} DecoderState; + +/* Default handler for user callbacks: skip it */ +static int speex_default_user_handler(GetBitContext *gb, void *state, void *data) +{ + const int req_size = get_bits(gb, 4); + skip_bits_long(gb, 5 + 8 * req_size); + return 0; +} + +typedef struct StereoState { + float balance; /**< Left/right balance info */ + float e_ratio; /**< Ratio of energies: E(left+right)/[E(left)+E(right)] */ + float smooth_left; /**< Smoothed left channel gain */ + float smooth_right; /**< Smoothed right channel gain */ +} StereoState; + +typedef struct SpeexContext { + AVClass *class; + GetBitContext gb; + + int32_t version_id; /**< Version for Speex (for checking compatibility) */ + int32_t rate; /**< Sampling rate used */ + int32_t mode; /**< Mode used (0 for narrowband, 1 for wideband) */ + int32_t bitstream_version; /**< Version ID of the bit-stream */ + int32_t nb_channels; /**< Number of channels decoded */ + int32_t bitrate; /**< Bit-rate used */ + int32_t frame_size; /**< Size of frames */ + int32_t vbr; /**< 1 for a VBR decoding, 0 otherwise */ + int32_t frames_per_packet; /**< Number of frames stored per Ogg packet */ + int32_t extra_headers; /**< Number of additional headers after the comments */ + + int pkt_size; + + StereoState stereo; + DecoderState st[SPEEX_NB_MODES]; + + AVFloatDSPContext *fdsp; +} SpeexContext; + +static void lsp_unquant_lbr(float *lsp, int order, GetBitContext *gb) +{ + int id; + + for (int i = 0; i < order; i++) + lsp[i] = LSP_LINEAR(i); + + id = get_bits(gb, 6); + for (int i = 0; i < 10; i++) + lsp[i] += LSP_DIV_256(cdbk_nb[id * 10 + i]); + + id = get_bits(gb, 6); + for (int i = 0; i < 5; i++) + lsp[i] += LSP_DIV_512(cdbk_nb_low1[id * 5 + i]); + + id = get_bits(gb, 6); + for (int i = 0; i < 5; i++) + lsp[i + 5] += LSP_DIV_512(cdbk_nb_high1[id * 5 + i]); +} + +static void forced_pitch_unquant(float *exc, float *exc_out, int start, int end, + float pitch_coef, const void *par, int nsf, + int *pitch_val, float *gain_val, GetBitContext *gb, int count_lost, + int subframe_offset, float last_pitch_gain, int cdbk_offset) +{ + av_assert0(!isnan(pitch_coef)); + pitch_coef = fminf(pitch_coef, .99f); + for (int i = 0; i < nsf; i++) { + exc_out[i] = exc[i - start] * pitch_coef; + exc[i] = exc_out[i]; + } + pitch_val[0] = start; + gain_val[0] = gain_val[2] = 0.f; + gain_val[1] = pitch_coef; +} + +static inline float speex_rand(float std, int32_t *seed) +{ + const uint32_t jflone = 0x3f800000; + const uint32_t jflmsk = 0x007fffff; + float fran; + uint32_t ran; + seed[0] = 1664525 * seed[0] + 1013904223; + ran = jflone | (jflmsk & seed[0]); + fran = av_int2float(ran); + fran -= 1.5f; + fran *= std; + return fran; +} + +static void noise_codebook_unquant(float *exc, const void *par, int nsf, + GetBitContext *gb, int32_t *seed) +{ + for (int i = 0; i < nsf; i++) + exc[i] = speex_rand(1.f, seed); +} + +static void split_cb_shape_sign_unquant(float *exc, const void *par, int nsf, + GetBitContext *gb, int32_t *seed) +{ + int subvect_size, nb_subvect, have_sign, shape_bits; + const SplitCodebookParams *params; + const signed char *shape_cb; + int signs[10], ind[10]; + + params = par; + subvect_size = params->subvect_size; + nb_subvect = params->nb_subvect; + + shape_cb = params->shape_cb; + have_sign = params->have_sign; + shape_bits = params->shape_bits; + + /* Decode codewords and gains */ + for (int i = 0; i < nb_subvect; i++) { + signs[i] = have_sign ? get_bits1(gb) : 0; + ind[i] = get_bitsz(gb, shape_bits); + } + /* Compute decoded excitation */ + for (int i = 0; i < nb_subvect; i++) { + const float s = signs[i] ? -1.f : 1.f; + + for (int j = 0; j < subvect_size; j++) + exc[subvect_size * i + j] += s * 0.03125f * shape_cb[ind[i] * subvect_size + j]; + } +} + +#define SUBMODE(x) st->submodes[st->submodeID]->x + +#define gain_3tap_to_1tap(g) (FFABS(g[1]) + (g[0] > 0.f ? g[0] : -.5f * g[0]) + (g[2] > 0.f ? g[2] : -.5f * g[2])) + +static void +pitch_unquant_3tap(float *exc, float *exc_out, int start, int end, float pitch_coef, + const void *par, int nsf, int *pitch_val, float *gain_val, GetBitContext *gb, + int count_lost, int subframe_offset, float last_pitch_gain, int cdbk_offset) +{ + int pitch, gain_index, gain_cdbk_size; + const int8_t *gain_cdbk; + const LtpParam *params; + float gain[3]; + + params = (const LtpParam *)par; + gain_cdbk_size = 1 << params->gain_bits; + gain_cdbk = params->gain_cdbk + 4 * gain_cdbk_size * cdbk_offset; + + pitch = get_bitsz(gb, params->pitch_bits); + pitch += start; + gain_index = get_bitsz(gb, params->gain_bits); + gain[0] = 0.015625f * gain_cdbk[gain_index * 4] + .5f; + gain[1] = 0.015625f * gain_cdbk[gain_index * 4 + 1] + .5f; + gain[2] = 0.015625f * gain_cdbk[gain_index * 4 + 2] + .5f; + + if (count_lost && pitch > subframe_offset) { + float tmp = count_lost < 4 ? last_pitch_gain : 0.5f * last_pitch_gain; + float gain_sum; + + tmp = fminf(tmp, .95f); + gain_sum = gain_3tap_to_1tap(gain); + + if (gain_sum > tmp && gain_sum > 0.f) { + float fact = tmp / gain_sum; + for (int i = 0; i < 3; i++) + gain[i] *= fact; + } + } + + pitch_val[0] = pitch; + gain_val[0] = gain[0]; + gain_val[1] = gain[1]; + gain_val[2] = gain[2]; + SPEEX_MEMSET(exc_out, 0, nsf); + + for (int i = 0; i < 3; i++) { + int tmp1, tmp3; + int pp = pitch + 1 - i; + tmp1 = nsf; + if (tmp1 > pp) + tmp1 = pp; + for (int j = 0; j < tmp1; j++) + exc_out[j] += gain[2 - i] * exc[j - pp]; + tmp3 = nsf; + if (tmp3 > pp + pitch) + tmp3 = pp + pitch; + for (int j = tmp1; j < tmp3; j++) + exc_out[j] += gain[2 - i] * exc[j - pp - pitch]; + } +} + +static void lsp_unquant_nb(float *lsp, int order, GetBitContext *gb) +{ + int id; + + for (int i = 0; i < order; i++) + lsp[i] = LSP_LINEAR(i); + + id = get_bits(gb, 6); + for (int i = 0; i < 10; i++) + lsp[i] += LSP_DIV_256(cdbk_nb[id * 10 + i]); + + id = get_bits(gb, 6); + for (int i = 0; i < 5; i++) + lsp[i] += LSP_DIV_512(cdbk_nb_low1[id * 5 + i]); + + id = get_bits(gb, 6); + for (int i = 0; i < 5; i++) + lsp[i] += LSP_DIV_1024(cdbk_nb_low2[id * 5 + i]); + + id = get_bits(gb, 6); + for (int i = 0; i < 5; i++) + lsp[i + 5] += LSP_DIV_512(cdbk_nb_high1[id * 5 + i]); + + id = get_bits(gb, 6); + for (int i = 0; i < 5; i++) + lsp[i + 5] += LSP_DIV_1024(cdbk_nb_high2[id * 5 + i]); +} + +static void lsp_unquant_high(float *lsp, int order, GetBitContext *gb) +{ + int id; + + for (int i = 0; i < order; i++) + lsp[i] = LSP_LINEAR_HIGH(i); + + id = get_bits(gb, 6); + for (int i = 0; i < order; i++) + lsp[i] += LSP_DIV_256(high_lsp_cdbk[id * order + i]); + + id = get_bits(gb, 6); + for (int i = 0; i < order; i++) + lsp[i] += LSP_DIV_512(high_lsp_cdbk2[id * order + i]); +} + +/* 2150 bps "vocoder-like" mode for comfort noise */ +static const SpeexSubmode nb_submode1 = { + 0, 1, 0, 0, lsp_unquant_lbr, forced_pitch_unquant, NULL, + noise_codebook_unquant, NULL, -1.f +}; + +/* 5.95 kbps very low bit-rate mode */ +static const SpeexSubmode nb_submode2 = { + 0, 0, 0, 0, lsp_unquant_lbr, pitch_unquant_3tap, <p_params_vlbr, + split_cb_shape_sign_unquant, &split_cb_nb_vlbr, .6f +}; + +/* 8 kbps low bit-rate mode */ +static const SpeexSubmode nb_submode3 = { + -1, 0, 1, 0, lsp_unquant_lbr, pitch_unquant_3tap, <p_params_lbr, + split_cb_shape_sign_unquant, &split_cb_nb_lbr, .55f +}; + +/* 11 kbps medium bit-rate mode */ +static const SpeexSubmode nb_submode4 = { + -1, 0, 1, 0, lsp_unquant_lbr, pitch_unquant_3tap, <p_params_med, + split_cb_shape_sign_unquant, &split_cb_nb_med, .45f +}; + +/* 15 kbps high bit-rate mode */ +static const SpeexSubmode nb_submode5 = { + -1, 0, 3, 0, lsp_unquant_nb, pitch_unquant_3tap, <p_params_nb, + split_cb_shape_sign_unquant, &split_cb_nb, .25f +}; + +/* 18.2 high bit-rate mode */ +static const SpeexSubmode nb_submode6 = { + -1, 0, 3, 0, lsp_unquant_nb, pitch_unquant_3tap, <p_params_nb, + split_cb_shape_sign_unquant, &split_cb_sb, .15f +}; + +/* 24.6 kbps high bit-rate mode */ +static const SpeexSubmode nb_submode7 = { + -1, 0, 3, 1, lsp_unquant_nb, pitch_unquant_3tap, <p_params_nb, + split_cb_shape_sign_unquant, &split_cb_nb, 0.05f +}; + +/* 3.95 kbps very low bit-rate mode */ +static const SpeexSubmode nb_submode8 = { + 0, 1, 0, 0, lsp_unquant_lbr, forced_pitch_unquant, NULL, + split_cb_shape_sign_unquant, &split_cb_nb_ulbr, .5f +}; + +static const SpeexSubmode wb_submode1 = { + 0, 0, 1, 0, lsp_unquant_high, NULL, NULL, + NULL, NULL, -1.f +}; + +static const SpeexSubmode wb_submode2 = { + 0, 0, 1, 0, lsp_unquant_high, NULL, NULL, + split_cb_shape_sign_unquant, &split_cb_high_lbr, -1.f +}; + +static const SpeexSubmode wb_submode3 = { + 0, 0, 1, 0, lsp_unquant_high, NULL, NULL, + split_cb_shape_sign_unquant, &split_cb_high, -1.f +}; + +static const SpeexSubmode wb_submode4 = { + 0, 0, 1, 1, lsp_unquant_high, NULL, NULL, + split_cb_shape_sign_unquant, &split_cb_high, -1.f +}; + +static int nb_decode(AVCodecContext *, void *, GetBitContext *, float *); +static int sb_decode(AVCodecContext *, void *, GetBitContext *, float *); + +static const SpeexMode speex_modes[SPEEX_NB_MODES] = { + { + .modeID = 0, + .decode = nb_decode, + .frame_size = NB_FRAME_SIZE, + .subframe_size = NB_SUBFRAME_SIZE, + .lpc_size = NB_ORDER, + .submodes = { + NULL, &nb_submode1, &nb_submode2, &nb_submode3, &nb_submode4, + &nb_submode5, &nb_submode6, &nb_submode7, &nb_submode8 + }, + .default_submode = 5, + }, + { + .modeID = 1, + .decode = sb_decode, + .frame_size = NB_FRAME_SIZE, + .subframe_size = NB_SUBFRAME_SIZE, + .lpc_size = 8, + .folding_gain = 0.9f, + .submodes = { + NULL, &wb_submode1, &wb_submode2, &wb_submode3, &wb_submode4 + }, + .default_submode = 3, + }, + { + .modeID = 2, + .decode = sb_decode, + .frame_size = 320, + .subframe_size = 80, + .lpc_size = 8, + .folding_gain = 0.7f, + .submodes = { + NULL, &wb_submode1 + }, + .default_submode = 1, + }, +}; + +static float compute_rms(const float *x, int len) +{ + float sum = 0.f; + + for (int i = 0; i < len; i++) + sum += x[i] * x[i]; + + av_assert0(len > 0); + return sqrtf(.1f + sum / len); +} + +static void bw_lpc(float gamma, const float *lpc_in, + float *lpc_out, int order) +{ + float tmp = gamma; + + for (int i = 0; i < order; i++) { + lpc_out[i] = tmp * lpc_in[i]; + tmp *= gamma; + } +} + +static void iir_mem(const float *x, const float *den, + float *y, int N, int ord, float *mem) +{ + for (int i = 0; i < N; i++) { + float yi = x[i] + mem[0]; + float nyi = -yi; + for (int j = 0; j < ord - 1; j++) + mem[j] = mem[j + 1] + den[j] * nyi; + mem[ord - 1] = den[ord - 1] * nyi; + y[i] = yi; + } +} + +static void highpass(const float *x, float *y, int len, float *mem, int wide) +{ + static const float Pcoef[2][3] = {{ 1.00000f, -1.92683f, 0.93071f }, { 1.00000f, -1.97226f, 0.97332f } }; + static const float Zcoef[2][3] = {{ 0.96446f, -1.92879f, 0.96446f }, { 0.98645f, -1.97277f, 0.98645f } }; + const float *den, *num; + + den = Pcoef[wide]; + num = Zcoef[wide]; + for (int i = 0; i < len; i++) { + float yi = num[0] * x[i] + mem[0]; + mem[0] = mem[1] + num[1] * x[i] + -den[1] * yi; + mem[1] = num[2] * x[i] + -den[2] * yi; + y[i] = yi; + } +} + +#define median3(a, b, c) \ + ((a) < (b) ? ((b) < (c) ? (b) : ((a) < (c) ? (c) : (a))) \ + : ((c) < (b) ? (b) : ((c) < (a) ? (c) : (a)))) + +static int speex_std_stereo(GetBitContext *gb, void *state, void *data) +{ + StereoState *stereo = data; + float sign = get_bits1(gb) ? -1.f : 1.f; + + stereo->balance = exp(sign * .25f * get_bits(gb, 5)); + stereo->e_ratio = e_ratio_quant[get_bits(gb, 2)]; + + return 0; +} + +static int speex_inband_handler(GetBitContext *gb, void *state, StereoState *stereo) +{ + int id = get_bits(gb, 4); + + if (id == SPEEX_INBAND_STEREO) { + return speex_std_stereo(gb, state, stereo); + } else { + int adv; + + if (id < 2) + adv = 1; + else if (id < 8) + adv = 4; + else if (id < 10) + adv = 8; + else if (id < 12) + adv = 16; + else if (id < 14) + adv = 32; + else + adv = 64; + skip_bits_long(gb, adv); + } + return 0; +} + +static void sanitize_values(float *vec, float min_val, float max_val, int len) +{ + for (int i = 0; i < len; i++) { + if (!isnormal(vec[i]) || fabsf(vec[i]) < 1e-8f) + vec[i] = 0.f; + else + vec[i] = av_clipf(vec[i], min_val, max_val); + } +} + +static void signal_mul(const float *x, float *y, float scale, int len) +{ + for (int i = 0; i < len; i++) + y[i] = scale * x[i]; +} + +static float inner_prod(const float *x, const float *y, int len) +{ + float sum = 0.f; + + for (int i = 0; i < len; i += 8) { + float part = 0.f; + part += x[i + 0] * y[i + 0]; + part += x[i + 1] * y[i + 1]; + part += x[i + 2] * y[i + 2]; + part += x[i + 3] * y[i + 3]; + part += x[i + 4] * y[i + 4]; + part += x[i + 5] * y[i + 5]; + part += x[i + 6] * y[i + 6]; + part += x[i + 7] * y[i + 7]; + sum += part; + } + + return sum; +} + +static int interp_pitch(const float *exc, float *interp, int pitch, int len) +{ + float corr[4][7], maxcorr; + int maxi, maxj; + + for (int i = 0; i < 7; i++) + corr[0][i] = inner_prod(exc, exc - pitch - 3 + i, len); + for (int i = 0; i < 3; i++) { + for (int j = 0; j < 7; j++) { + int i1, i2; + float tmp = 0.f; + + i1 = 3 - j; + if (i1 < 0) + i1 = 0; + i2 = 10 - j; + if (i2 > 7) + i2 = 7; + for (int k = i1; k < i2; k++) + tmp += shift_filt[i][k] * corr[0][j + k - 3]; + corr[i + 1][j] = tmp; + } + } + maxi = maxj = 0; + maxcorr = corr[0][0]; + for (int i = 0; i < 4; i++) { + for (int j = 0; j < 7; j++) { + if (corr[i][j] > maxcorr) { + maxcorr = corr[i][j]; + maxi = i; + maxj = j; + } + } + } + for (int i = 0; i < len; i++) { + float tmp = 0.f; + if (maxi > 0.f) { + for (int k = 0; k < 7; k++) + tmp += exc[i - (pitch - maxj + 3) + k - 3] * shift_filt[maxi - 1][k]; + } else { + tmp = exc[i - (pitch - maxj + 3)]; + } + interp[i] = tmp; + } + return pitch - maxj + 3; +} + +static void multicomb(const float *exc, float *new_exc, float *ak, int p, int nsf, + int pitch, int max_pitch, float comb_gain) +{ + float old_ener, new_ener; + float iexc0_mag, iexc1_mag, exc_mag; + float iexc[4 * NB_SUBFRAME_SIZE]; + float corr0, corr1, gain0, gain1; + float pgain1, pgain2; + float c1, c2, g1, g2; + float ngain, gg1, gg2; + int corr_pitch = pitch; + + interp_pitch(exc, iexc, corr_pitch, 80); + if (corr_pitch > max_pitch) + interp_pitch(exc, iexc + nsf, 2 * corr_pitch, 80); + else + interp_pitch(exc, iexc + nsf, -corr_pitch, 80); + + iexc0_mag = sqrtf(1000.f + inner_prod(iexc, iexc, nsf)); + iexc1_mag = sqrtf(1000.f + inner_prod(iexc + nsf, iexc + nsf, nsf)); + exc_mag = sqrtf(1.f + inner_prod(exc, exc, nsf)); + corr0 = inner_prod(iexc, exc, nsf); + corr1 = inner_prod(iexc + nsf, exc, nsf); + if (corr0 > iexc0_mag * exc_mag) + pgain1 = 1.f; + else + pgain1 = (corr0 / exc_mag) / iexc0_mag; + if (corr1 > iexc1_mag * exc_mag) + pgain2 = 1.f; + else + pgain2 = (corr1 / exc_mag) / iexc1_mag; + gg1 = exc_mag / iexc0_mag; + gg2 = exc_mag / iexc1_mag; + if (comb_gain > 0.f) { + c1 = .4f * comb_gain + .07f; + c2 = .5f + 1.72f * (c1 - .07f); + } else { + c1 = c2 = 0.f; + } + g1 = 1.f - c2 * pgain1 * pgain1; + g2 = 1.f - c2 * pgain2 * pgain2; + g1 = fmaxf(g1, c1); + g2 = fmaxf(g2, c1); + g1 = c1 / g1; + g2 = c1 / g2; + + if (corr_pitch > max_pitch) { + gain0 = .7f * g1 * gg1; + gain1 = .3f * g2 * gg2; + } else { + gain0 = .6f * g1 * gg1; + gain1 = .6f * g2 * gg2; + } + for (int i = 0; i < nsf; i++) + new_exc[i] = exc[i] + (gain0 * iexc[i]) + (gain1 * iexc[i + nsf]); + new_ener = compute_rms(new_exc, nsf); + old_ener = compute_rms(exc, nsf); + + old_ener = fmaxf(old_ener, 1.f); + new_ener = fmaxf(new_ener, 1.f); + old_ener = fminf(old_ener, new_ener); + ngain = old_ener / new_ener; + + for (int i = 0; i < nsf; i++) + new_exc[i] *= ngain; +} + +static void lsp_interpolate(const float *old_lsp, const float *new_lsp, + float *lsp, int len, int subframe, + int nb_subframes, float margin) +{ + const float tmp = (1.f + subframe) / nb_subframes; + + for (int i = 0; i < len; i++) { + lsp[i] = (1.f - tmp) * old_lsp[i] + tmp * new_lsp[i]; + lsp[i] = av_clipf(lsp[i], margin, M_PI - margin); + } + for (int i = 1; i < len - 1; i++) { + lsp[i] = fmaxf(lsp[i], lsp[i - 1] + margin); + if (lsp[i] > lsp[i + 1] - margin) + lsp[i] = .5f * (lsp[i] + lsp[i + 1] - margin); + } +} + +static void lsp_to_lpc(const float *freq, float *ak, int lpcrdr) +{ + float xout1, xout2, xin1, xin2; + float *pw, *n0; + float Wp[4 * NB_ORDER + 2] = { 0 }; + float x_freq[NB_ORDER]; + const int m = lpcrdr >> 1; + + pw = Wp; + + xin1 = xin2 = 1.f; + + for (int i = 0; i < lpcrdr; i++) + x_freq[i] = -cosf(freq[i]); + + /* reconstruct P(z) and Q(z) by cascading second order + * polynomials in form 1 - 2xz(-1) +z(-2), where x is the + * LSP coefficient + */ + for (int j = 0; j <= lpcrdr; j++) { + int i2 = 0; + for (int i = 0; i < m; i++, i2 += 2) { + n0 = pw + (i * 4); + xout1 = xin1 + 2.f * x_freq[i2 ] * n0[0] + n0[1]; + xout2 = xin2 + 2.f * x_freq[i2 + 1] * n0[2] + n0[3]; + n0[1] = n0[0]; + n0[3] = n0[2]; + n0[0] = xin1; + n0[2] = xin2; + xin1 = xout1; + xin2 = xout2; + } + xout1 = xin1 + n0[4]; + xout2 = xin2 - n0[5]; + if (j > 0) + ak[j - 1] = (xout1 + xout2) * 0.5f; + n0[4] = xin1; + n0[5] = xin2; + + xin1 = 0.f; + xin2 = 0.f; + } +} + +static int nb_decode(AVCodecContext *avctx, void *ptr_st, + GetBitContext *gb, float *out) +{ + DecoderState *st = ptr_st; + float ol_gain = 0, ol_pitch_coef = 0, best_pitch_gain = 0, pitch_average = 0; + int m, pitch, wideband, ol_pitch = 0, best_pitch = 40; + SpeexContext *s = avctx->priv_data; + float innov[NB_SUBFRAME_SIZE]; + float exc32[NB_SUBFRAME_SIZE]; + float interp_qlsp[NB_ORDER]; + float qlsp[NB_ORDER]; + float ak[NB_ORDER]; + float pitch_gain[3] = { 0 }; + + st->exc = st->exc_buf + 2 * NB_PITCH_END + NB_SUBFRAME_SIZE + 6; + + if (st->encode_submode) { + do { /* Search for next narrowband block (handle requests, skip wideband blocks) */ + if (get_bits_left(gb) < 5) + return AVERROR_INVALIDDATA; + wideband = get_bits1(gb); + if (wideband) /* Skip wideband block (for compatibility) */ { + int submode, advance; + + submode = get_bits(gb, SB_SUBMODE_BITS); + advance = wb_skip_table[submode]; + advance -= SB_SUBMODE_BITS + 1; + if (advance < 0) + return AVERROR_INVALIDDATA; + skip_bits_long(gb, advance); + + if (get_bits_left(gb) < 5) + return AVERROR_INVALIDDATA; + wideband = get_bits1(gb); + if (wideband) { + submode = get_bits(gb, SB_SUBMODE_BITS); + advance = wb_skip_table[submode]; + advance -= SB_SUBMODE_BITS + 1; + if (advance < 0) + return AVERROR_INVALIDDATA; + skip_bits_long(gb, advance); + wideband = get_bits1(gb); + if (wideband) { + av_log(avctx, AV_LOG_ERROR, "more than two wideband layers found\n"); + return AVERROR_INVALIDDATA; + } + } + } + if (get_bits_left(gb) < 4) + return AVERROR_INVALIDDATA; + m = get_bits(gb, 4); + if (m == 15) /* We found a terminator */ { + return AVERROR_INVALIDDATA; + } else if (m == 14) /* Speex in-band request */ { + int ret = speex_inband_handler(gb, st, &s->stereo); + if (ret) + return ret; + } else if (m == 13) /* User in-band request */ { + int ret = speex_default_user_handler(gb, st, NULL); + if (ret) + return ret; + } else if (m > 8) /* Invalid mode */ { + return AVERROR_INVALIDDATA; + } + } while (m > 8); + + st->submodeID = m; /* Get the sub-mode that was used */ + } + + /* Shift all buffers by one frame */ + memmove(st->exc_buf, st->exc_buf + NB_FRAME_SIZE, (2 * NB_PITCH_END + NB_SUBFRAME_SIZE + 12) * sizeof(float)); + + /* If null mode (no transmission), just set a couple things to zero */ + if (st->submodes[st->submodeID] == NULL) { + float lpc[NB_ORDER]; + float innov_gain = 0.f; + + bw_lpc(0.93f, st->interp_qlpc, lpc, NB_ORDER); + innov_gain = compute_rms(st->exc, NB_FRAME_SIZE); + for (int i = 0; i < NB_FRAME_SIZE; i++) + st->exc[i] = speex_rand(innov_gain, &st->seed); + + /* Final signal synthesis from excitation */ + iir_mem(st->exc, lpc, out, NB_FRAME_SIZE, NB_ORDER, st->mem_sp); + st->count_lost = 0; + + return 0; + } + + /* Unquantize LSPs */ + SUBMODE(lsp_unquant)(qlsp, NB_ORDER, gb); + + /* Damp memory if a frame was lost and the LSP changed too much */ + if (st->count_lost) { + float fact, lsp_dist = 0; + + for (int i = 0; i < NB_ORDER; i++) + lsp_dist = lsp_dist + FFABS(st->old_qlsp[i] - qlsp[i]); + fact = .6f * exp(-.2f * lsp_dist); + for (int i = 0; i < NB_ORDER; i++) + st->mem_sp[i] = fact * st->mem_sp[i]; + } + + /* Handle first frame and lost-packet case */ + if (st->first || st->count_lost) + memcpy(st->old_qlsp, qlsp, sizeof(st->old_qlsp)); + + /* Get open-loop pitch estimation for low bit-rate pitch coding */ + if (SUBMODE(lbr_pitch) != -1) + ol_pitch = NB_PITCH_START + get_bits(gb, 7); + + if (SUBMODE(forced_pitch_gain)) + ol_pitch_coef = 0.066667f * get_bits(gb, 4); + + /* Get global excitation gain */ + ol_gain = expf(get_bits(gb, 5) / 3.5f); + + if (st->submodeID == 1) + st->dtx_enabled = get_bits(gb, 4) == 15; + + if (st->submodeID > 1) + st->dtx_enabled = 0; + + for (int sub = 0; sub < NB_NB_SUBFRAMES; sub++) { /* Loop on subframes */ + float *exc, *innov_save = NULL, tmp, ener; + int pit_min, pit_max, offset, q_energy; + + offset = NB_SUBFRAME_SIZE * sub; /* Offset relative to start of frame */ + exc = st->exc + offset; /* Excitation */ + if (st->innov_save) /* Original signal */ + innov_save = st->innov_save + offset; + + SPEEX_MEMSET(exc, 0, NB_SUBFRAME_SIZE); /* Reset excitation */ + + /* Adaptive codebook contribution */ + av_assert0(SUBMODE(ltp_unquant)); + /* Handle pitch constraints if any */ + if (SUBMODE(lbr_pitch) != -1) { + int margin = SUBMODE(lbr_pitch); + + if (margin) { + pit_min = ol_pitch - margin + 1; + pit_min = FFMAX(pit_min, NB_PITCH_START); + pit_max = ol_pitch + margin; + pit_max = FFMIN(pit_max, NB_PITCH_START); + } else { + pit_min = pit_max = ol_pitch; + } + } else { + pit_min = NB_PITCH_START; + pit_max = NB_PITCH_END; + } + + SUBMODE(ltp_unquant)(exc, exc32, pit_min, pit_max, ol_pitch_coef, SUBMODE(LtpParam), + NB_SUBFRAME_SIZE, &pitch, pitch_gain, gb, st->count_lost, offset, + st->last_pitch_gain, 0); + + sanitize_values(exc32, -32000, 32000, NB_SUBFRAME_SIZE); + + tmp = gain_3tap_to_1tap(pitch_gain); + + pitch_average += tmp; + if ((tmp > best_pitch_gain && + FFABS(2 * best_pitch - pitch) >= 3 && + FFABS(3 * best_pitch - pitch) >= 4 && + FFABS(4 * best_pitch - pitch) >= 5) || + (tmp > .6f * best_pitch_gain && + (FFABS(best_pitch - 2 * pitch) < 3 || + FFABS(best_pitch - 3 * pitch) < 4 || + FFABS(best_pitch - 4 * pitch) < 5)) || + ((.67f * tmp) > best_pitch_gain && + (FFABS(2 * best_pitch - pitch) < 3 || + FFABS(3 * best_pitch - pitch) < 4 || + FFABS(4 * best_pitch - pitch) < 5))) { + best_pitch = pitch; + if (tmp > best_pitch_gain) + best_pitch_gain = tmp; + } + + memset(innov, 0, sizeof(innov)); + + /* Decode sub-frame gain correction */ + if (SUBMODE(have_subframe_gain) == 3) { + q_energy = get_bits(gb, 3); + ener = exc_gain_quant_scal3[q_energy] * ol_gain; + } else if (SUBMODE(have_subframe_gain) == 1) { + q_energy = get_bits1(gb); + ener = exc_gain_quant_scal1[q_energy] * ol_gain; + } else { + ener = ol_gain; + } + + av_assert0(SUBMODE(innovation_unquant)); + /* Fixed codebook contribution */ + SUBMODE(innovation_unquant)(innov, SUBMODE(innovation_params), NB_SUBFRAME_SIZE, gb, &st->seed); + /* De-normalize innovation and update excitation */ + + signal_mul(innov, innov, ener, NB_SUBFRAME_SIZE); + + /* Decode second codebook (only for some modes) */ + if (SUBMODE(double_codebook)) { + float innov2[NB_SUBFRAME_SIZE] = { 0 }; + + SUBMODE(innovation_unquant)(innov2, SUBMODE(innovation_params), NB_SUBFRAME_SIZE, gb, &st->seed); + signal_mul(innov2, innov2, 0.454545f * ener, NB_SUBFRAME_SIZE); + for (int i = 0; i < NB_SUBFRAME_SIZE; i++) + innov[i] += innov2[i]; + } + for (int i = 0; i < NB_SUBFRAME_SIZE; i++) + exc[i] = exc32[i] + innov[i]; + if (innov_save) + memcpy(innov_save, innov, sizeof(innov)); + + /* Vocoder mode */ + if (st->submodeID == 1) { + float g = ol_pitch_coef; + + g = av_clipf(1.5f * (g - .2f), 0.f, 1.f); + + SPEEX_MEMSET(exc, 0, NB_SUBFRAME_SIZE); + while (st->voc_offset < NB_SUBFRAME_SIZE) { + if (st->voc_offset >= 0) + exc[st->voc_offset] = sqrtf(2.f * ol_pitch) * (g * ol_gain); + st->voc_offset += ol_pitch; + } + st->voc_offset -= NB_SUBFRAME_SIZE; + + for (int i = 0; i < NB_SUBFRAME_SIZE; i++) { + float exci = exc[i]; + exc[i] = (.7f * exc[i] + .3f * st->voc_m1) + ((1.f - .85f * g) * innov[i]) - .15f * g * st->voc_m2; + st->voc_m1 = exci; + st->voc_m2 = innov[i]; + st->voc_mean = .8f * st->voc_mean + .2f * exc[i]; + exc[i] -= st->voc_mean; + } + } + } + + if (st->lpc_enh_enabled && SUBMODE(comb_gain) > 0 && !st->count_lost) { + multicomb(st->exc - NB_SUBFRAME_SIZE, out, st->interp_qlpc, NB_ORDER, + 2 * NB_SUBFRAME_SIZE, best_pitch, 40, SUBMODE(comb_gain)); + multicomb(st->exc + NB_SUBFRAME_SIZE, out + 2 * NB_SUBFRAME_SIZE, + st->interp_qlpc, NB_ORDER, 2 * NB_SUBFRAME_SIZE, best_pitch, 40, + SUBMODE(comb_gain)); + } else { + SPEEX_COPY(out, &st->exc[-NB_SUBFRAME_SIZE], NB_FRAME_SIZE); + } + + /* If the last packet was lost, re-scale the excitation to obtain the same + * energy as encoded in ol_gain */ + if (st->count_lost) { + float exc_ener, gain; + + exc_ener = compute_rms(st->exc, NB_FRAME_SIZE); + av_assert0(exc_ener + 1.f > 0.f); + gain = fminf(ol_gain / (exc_ener + 1.f), 2.f); + for (int i = 0; i < NB_FRAME_SIZE; i++) { + st->exc[i] *= gain; + out[i] = st->exc[i - NB_SUBFRAME_SIZE]; + } + } + + for (int sub = 0; sub < NB_NB_SUBFRAMES; sub++) { /* Loop on subframes */ + const int offset = NB_SUBFRAME_SIZE * sub; /* Offset relative to start of frame */ + float pi_g = 1.f, *sp = out + offset; /* Original signal */ + + lsp_interpolate(st->old_qlsp, qlsp, interp_qlsp, NB_ORDER, sub, NB_NB_SUBFRAMES, 0.002f); + lsp_to_lpc(interp_qlsp, ak, NB_ORDER); /* Compute interpolated LPCs (unquantized) */ + + for (int i = 0; i < NB_ORDER; i += 2) /* Compute analysis filter at w=pi */ + pi_g += ak[i + 1] - ak[i]; + st->pi_gain[sub] = pi_g; + st->exc_rms[sub] = compute_rms(st->exc + offset, NB_SUBFRAME_SIZE); + + iir_mem(sp, st->interp_qlpc, sp, NB_SUBFRAME_SIZE, NB_ORDER, st->mem_sp); + + memcpy(st->interp_qlpc, ak, sizeof(st->interp_qlpc)); + } + + if (st->highpass_enabled) + highpass(out, out, NB_FRAME_SIZE, st->mem_hp, st->is_wideband); + + /* Store the LSPs for interpolation in the next frame */ + memcpy(st->old_qlsp, qlsp, sizeof(st->old_qlsp)); + + st->count_lost = 0; + st->last_pitch = best_pitch; + st->last_pitch_gain = .25f * pitch_average; + st->last_ol_gain = ol_gain; + st->first = 0; + + return 0; +} + +static void qmf_synth(const float *x1, const float *x2, const float *a, float *y, int N, int M, float *mem1, float *mem2) +{ + const int M2 = M >> 1, N2 = N >> 1; + float xx1[352], xx2[352]; + + for (int i = 0; i < N2; i++) + xx1[i] = x1[N2-1-i]; + for (int i = 0; i < M2; i++) + xx1[N2+i] = mem1[2*i+1]; + for (int i = 0; i < N2; i++) + xx2[i] = x2[N2-1-i]; + for (int i = 0; i < M2; i++) + xx2[N2+i] = mem2[2*i+1]; + + for (int i = 0; i < N2; i += 2) { + float y0, y1, y2, y3; + float x10, x20; + + y0 = y1 = y2 = y3 = 0.f; + x10 = xx1[N2-2-i]; + x20 = xx2[N2-2-i]; + + for (int j = 0; j < M2; j += 2) { + float x11, x21; + float a0, a1; + + a0 = a[2*j]; + a1 = a[2*j+1]; + x11 = xx1[N2-1+j-i]; + x21 = xx2[N2-1+j-i]; + + y0 += a0 * (x11-x21); + y1 += a1 * (x11+x21); + y2 += a0 * (x10-x20); + y3 += a1 * (x10+x20); + a0 = a[2*j+2]; + a1 = a[2*j+3]; + x10 = xx1[N2+j-i]; + x20 = xx2[N2+j-i]; + + y0 += a0 * (x10-x20); + y1 += a1 * (x10+x20); + y2 += a0 * (x11-x21); + y3 += a1 * (x11+x21); + } + y[2 * i ] = 2.f * y0; + y[2 * i+1] = 2.f * y1; + y[2 * i+2] = 2.f * y2; + y[2 * i+3] = 2.f * y3; + } + + for (int i = 0; i < M2; i++) + mem1[2*i+1] = xx1[i]; + for (int i = 0; i < M2; i++) + mem2[2*i+1] = xx2[i]; +} + +static int sb_decode(AVCodecContext *avctx, void *ptr_st, + GetBitContext *gb, float *out) +{ + SpeexContext *s = avctx->priv_data; + DecoderState *st = ptr_st; + float low_pi_gain[NB_NB_SUBFRAMES]; + float low_exc_rms[NB_NB_SUBFRAMES]; + float interp_qlsp[NB_ORDER]; + int ret, wideband, dtx = 0; + float *low_innov_alias; + float qlsp[NB_ORDER]; + float ak[NB_ORDER]; + const SpeexMode *mode; + + mode = st->mode; + + if (st->modeID > 0) { + low_innov_alias = out + st->frame_size; + s->st[st->modeID - 1].innov_save = low_innov_alias; + ret = speex_modes[st->modeID - 1].decode(avctx, &s->st[st->modeID - 1], gb, out); + if (ret < 0) + return ret; + } + + if (st->encode_submode) { /* Check "wideband bit" */ + if (get_bits_left(gb) > 0) + wideband = show_bits1(gb); + else + wideband = 0; + if (wideband) { /* Regular wideband frame, read the submode */ + wideband = get_bits1(gb); + st->submodeID = get_bits(gb, SB_SUBMODE_BITS); + } else { /* Was a narrowband frame, set "null submode" */ + st->submodeID = 0; + } + if (st->submodeID != 0 && st->submodes[st->submodeID] == NULL) + return AVERROR_INVALIDDATA; + } + + /* If null mode (no transmission), just set a couple things to zero */ + if (st->submodes[st->submodeID] == NULL) { + if (dtx) { + //sb_decode_lost(st, out, 1); + return 0; + } + + for (int i = 0; i < st->frame_size; i++) + out[st->frame_size + i] = 1e-15f; + + st->first = 1; + + /* Final signal synthesis from excitation */ + iir_mem(out + st->frame_size, st->interp_qlpc, out + st->frame_size, st->frame_size, st->lpc_size, st->mem_sp); + + qmf_synth(out, out + st->frame_size, h0, out, st->full_frame_size, QMF_ORDER, st->g0_mem, st->g1_mem); + + return 0; + } + + memcpy(low_pi_gain, s->st[st->modeID - 1].pi_gain, sizeof(low_pi_gain)); + memcpy(low_exc_rms, s->st[st->modeID - 1].exc_rms, sizeof(low_exc_rms)); + + SUBMODE(lsp_unquant)(qlsp, st->lpc_size, gb); + + if (st->first) + memcpy(st->old_qlsp, qlsp, sizeof(st->old_qlsp)); + + for (int sub = 0; sub < st->nb_subframes; sub++) { + float filter_ratio, el, rl, rh; + float *innov_save = NULL, *sp; + float exc[80]; + int offset; + + offset = st->subframe_size * sub; + sp = out + st->frame_size + offset; + /* Pointer for saving innovation */ + if (st->innov_save) { + innov_save = st->innov_save + 2 * offset; + SPEEX_MEMSET(innov_save, 0, 2 * st->subframe_size); + } + + av_assert0(st->nb_subframes > 0); + lsp_interpolate(st->old_qlsp, qlsp, interp_qlsp, st->lpc_size, sub, st->nb_subframes, 0.05f); + lsp_to_lpc(interp_qlsp, ak, st->lpc_size); + + /* Calculate reponse ratio between the low and high filter in the middle + of the band (4000 Hz) */ + st->pi_gain[sub] = 1.f; + rh = 1.f; + for (int i = 0; i < st->lpc_size; i += 2) { + rh += ak[i + 1] - ak[i]; + st->pi_gain[sub] += ak[i] + ak[i + 1]; + } + + rl = low_pi_gain[sub]; + filter_ratio = (rl + .01f) / (rh + .01f); + + SPEEX_MEMSET(exc, 0, st->subframe_size); + if (!SUBMODE(innovation_unquant)) { + const int x = get_bits(gb, 5); + const float g = expf(.125f * (x - 10)) / filter_ratio; + + for (int i = 0; i < st->subframe_size; i += 2) { + exc[i ] = mode->folding_gain * low_innov_alias[offset + i ] * g; + exc[i + 1] = -mode->folding_gain * low_innov_alias[offset + i + 1] * g; + } + } else { + float gc, scale; + + el = low_exc_rms[sub]; + gc = 0.87360f * gc_quant_bound[get_bits(gb, 4)]; + + if (st->subframe_size == 80) + gc *= M_SQRT2; + + scale = (gc * el) / filter_ratio; + SUBMODE(innovation_unquant) + (exc, SUBMODE(innovation_params), st->subframe_size, + gb, &st->seed); + + signal_mul(exc, exc, scale, st->subframe_size); + if (SUBMODE(double_codebook)) { + float innov2[80]; + + SPEEX_MEMSET(innov2, 0, st->subframe_size); + SUBMODE(innovation_unquant)(innov2, SUBMODE(innovation_params), st->subframe_size, gb, &st->seed); + signal_mul(innov2, innov2, 0.4f * scale, st->subframe_size); + for (int i = 0; i < st->subframe_size; i++) + exc[i] += innov2[i]; + } + } + + if (st->innov_save) { + for (int i = 0; i < st->subframe_size; i++) + innov_save[2 * i] = exc[i]; + } + + iir_mem(st->exc_buf, st->interp_qlpc, sp, st->subframe_size, st->lpc_size, st->mem_sp); + memcpy(st->exc_buf, exc, sizeof(exc)); + memcpy(st->interp_qlpc, ak, sizeof(st->interp_qlpc)); + st->exc_rms[sub] = compute_rms(st->exc_buf, st->subframe_size); + } + + qmf_synth(out, out + st->frame_size, h0, out, st->full_frame_size, QMF_ORDER, st->g0_mem, st->g1_mem); + memcpy(st->old_qlsp, qlsp, sizeof(st->old_qlsp)); + + st->first = 0; + + return 0; +} + +static int decoder_init(SpeexContext *s, DecoderState *st, const SpeexMode *mode) +{ + st->mode = mode; + st->modeID = mode->modeID; + + st->first = 1; + st->encode_submode = 1; + st->is_wideband = st->modeID > 0; + st->innov_save = NULL; + + st->submodes = mode->submodes; + st->submodeID = mode->default_submode; + st->subframe_size = mode->subframe_size; + st->lpc_size = mode->lpc_size; + st->full_frame_size = (1 + (st->modeID > 0)) * mode->frame_size; + st->nb_subframes = mode->frame_size / mode->subframe_size; + st->frame_size = mode->frame_size; + + st->lpc_enh_enabled = 1; + + st->last_pitch = 40; + st->count_lost = 0; + st->seed = 1000; + st->last_ol_gain = 0; + + st->voc_m1 = st->voc_m2 = st->voc_mean = 0; + st->voc_offset = 0; + st->dtx_enabled = 0; + st->highpass_enabled = mode->modeID == 0; + + return 0; +} + +static int parse_speex_extradata(AVCodecContext *avctx, + const uint8_t *extradata, int extradata_size) +{ + SpeexContext *s = avctx->priv_data; + const uint8_t *buf = extradata; + + if (memcmp(buf, "Speex ", 8)) + return AVERROR_INVALIDDATA; + + buf += 28; + + s->version_id = bytestream_get_le32(&buf); + buf += 4; + s->rate = bytestream_get_le32(&buf); + if (s->rate <= 0) + return AVERROR_INVALIDDATA; + s->mode = bytestream_get_le32(&buf); + if (s->mode < 0 || s->mode >= SPEEX_NB_MODES) + return AVERROR_INVALIDDATA; + s->bitstream_version = bytestream_get_le32(&buf); + if (s->bitstream_version != 4) + return AVERROR_INVALIDDATA; + s->nb_channels = bytestream_get_le32(&buf); + if (s->nb_channels <= 0 || s->nb_channels > 2) + return AVERROR_INVALIDDATA; + s->bitrate = bytestream_get_le32(&buf); + s->frame_size = bytestream_get_le32(&buf); + if (s->frame_size < NB_FRAME_SIZE) + return AVERROR_INVALIDDATA; + s->vbr = bytestream_get_le32(&buf); + s->frames_per_packet = bytestream_get_le32(&buf); + if (s->frames_per_packet <= 0) + return AVERROR_INVALIDDATA; + s->extra_headers = bytestream_get_le32(&buf); + + return 0; +} + +static av_cold int speex_decode_init(AVCodecContext *avctx) +{ + SpeexContext *s = avctx->priv_data; + int ret; + + s->fdsp = avpriv_float_dsp_alloc(0); + if (!s->fdsp) + return AVERROR(ENOMEM); + + if (avctx->extradata && avctx->extradata_size >= 80) { + ret = parse_speex_extradata(avctx, avctx->extradata, avctx->extradata_size); + if (ret < 0) + return ret; + } else { + s->rate = avctx->sample_rate; + if (s->rate <= 0) + return AVERROR_INVALIDDATA; + + s->nb_channels = avctx->channels; + if (s->nb_channels <= 0) + return AVERROR_INVALIDDATA; + + switch (s->rate) { + case 8000: s->mode = 0; break; + case 16000: s->mode = 1; break; + case 32000: s->mode = 2; break; + default: s->mode = 2; + } + + s->frames_per_packet = 1; + s->frame_size = NB_FRAME_SIZE << s->mode; + } + + if (avctx->codec_tag == MKTAG('S', 'P', 'X', 'N')) { + int quality; + + if (!avctx->extradata || avctx->extradata && avctx->extradata_size < 47) { + av_log(avctx, AV_LOG_ERROR, "Missing or invalid extradata.\n"); + return AVERROR_INVALIDDATA; + } + + quality = avctx->extradata[37]; + if (quality > 10) { + av_log(avctx, AV_LOG_ERROR, "Unsupported quality mode %d.\n", quality); + return AVERROR_PATCHWELCOME; + } + + s->pkt_size = ((const uint8_t[]){ 5, 10, 15, 20, 20, 28, 28, 38, 38, 46, 62 })[quality]; + + s->mode = 0; + s->nb_channels = 1; + s->rate = avctx->sample_rate; + if (s->rate <= 0) + return AVERROR_INVALIDDATA; + s->frames_per_packet = 1; + s->frame_size = NB_FRAME_SIZE; + } + + if (s->bitrate > 0) + avctx->bit_rate = s->bitrate; + avctx->channels = s->nb_channels; + avctx->sample_rate = s->rate; + avctx->sample_fmt = AV_SAMPLE_FMT_FLT; + + for (int m = 0; m <= s->mode; m++) { + ret = decoder_init(s, &s->st[m], &speex_modes[m]); + if (ret < 0) + return ret; + } + + s->stereo.balance = 1.f; + s->stereo.e_ratio = .5f; + s->stereo.smooth_left = 1.f; + s->stereo.smooth_right = 1.f; + + return 0; +} + +static void speex_decode_stereo(float *data, int frame_size, StereoState *stereo) +{ + float balance, e_left, e_right, e_ratio; + + balance = stereo->balance; + e_ratio = stereo->e_ratio; + + /* These two are Q14, with max value just below 2. */ + e_right = 1.f / sqrtf(e_ratio * (1.f + balance)); + e_left = sqrtf(balance) * e_right; + + for (int i = frame_size - 1; i >= 0; i--) { + float tmp = data[i]; + stereo->smooth_left = stereo->smooth_left * 0.98f + e_left * 0.02f; + stereo->smooth_right = stereo->smooth_right * 0.98f + e_right * 0.02f; + data[2 * i ] = stereo->smooth_left * tmp; + data[2 * i + 1] = stereo->smooth_right * tmp; + } +} + +static int speex_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) +{ + SpeexContext *s = avctx->priv_data; + AVFrame *frame = data; + const float scale = 1.f / 32768.f; + int buf_size = avpkt->size; + float *dst; + int ret; + + if (s->pkt_size && avpkt->size == 62) + buf_size = s->pkt_size; + if ((ret = init_get_bits8(&s->gb, avpkt->data, buf_size)) < 0) + return ret; + + frame->nb_samples = s->frame_size * s->frames_per_packet; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) + return ret; + + dst = (float *)frame->extended_data[0]; + for (int i = 0; i < s->frames_per_packet; i++) { + ret = speex_modes[s->mode].decode(avctx, &s->st[s->mode], &s->gb, dst + i * s->frame_size); + if (ret < 0) + return ret; + if (avctx->channels == 2) + speex_decode_stereo(dst + i * s->frame_size, s->frame_size, &s->stereo); + } + + dst = (float *)frame->extended_data[0]; + s->fdsp->vector_fmul_scalar(dst, dst, scale, frame->nb_samples * frame->channels); + + *got_frame_ptr = 1; + + return buf_size; +} + +static av_cold int speex_decode_close(AVCodecContext *avctx) +{ + SpeexContext *s = avctx->priv_data; + av_freep(&s->fdsp); + return 0; +} + +const AVCodec ff_speex_decoder = { + .name = "speex", + .long_name = NULL_IF_CONFIG_SMALL("Speex"), + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_SPEEX, + .init = speex_decode_init, + .decode = speex_decode_frame, + .close = speex_decode_close, + .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, + .priv_data_size = sizeof(SpeexContext), + .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, +}; |