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FFmpeg/libavcodec/speexdec.c

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/*
* 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 "codec_internal.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 *, uint32_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 */
uint32_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, uint32_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, uint32_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, uint32_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, &ltp_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, &ltp_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, &ltp_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, &ltp_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, &ltp_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, &ltp_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;
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) {
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 << s->mode)
return AVERROR_INVALIDDATA;
s->vbr = bytestream_get_le32(&buf);
s->frames_per_packet = bytestream_get_le32(&buf);
if (s->frames_per_packet <= 0 ||
s->frames_per_packet > 64 ||
s->frames_per_packet >= INT32_MAX / s->nb_channels / s->frame_size)
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->ch_layout.nb_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;
av_channel_layout_uninit(&avctx->ch_layout);
avctx->ch_layout.order = AV_CHANNEL_ORDER_UNSPEC;
avctx->ch_layout.nb_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, AVFrame *frame,
int *got_frame_ptr, AVPacket *avpkt)
{
SpeexContext *s = avctx->priv_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 = FFALIGN(s->frame_size * s->frames_per_packet, 4);
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->ch_layout.nb_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->ch_layout.nb_channels);
frame->nb_samples = s->frame_size * s->frames_per_packet;
*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 FFCodec ff_speex_decoder = {
.p.name = "speex",
.p.long_name = NULL_IF_CONFIG_SMALL("Speex"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_SPEEX,
.init = speex_decode_init,
FF_CODEC_DECODE_CB(speex_decode_frame),
.close = speex_decode_close,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
.priv_data_size = sizeof(SpeexContext),
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
};