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FFmpeg/libavcodec/speexdec.c
Andreas Rheinhardt 4243da4ff4 avcodec/codec_internal: Use union for FFCodec decode/encode callbacks 
This is possible, because every given FFCodec has to implement
exactly one of these. Doing so decreases sizeof(FFCodec) and
therefore decreases the size of the binary.
Notice that in case of position-independent code the decrease
is in .data.rel.ro, so that this translates to decreased
memory consumption.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-04-05 20:02:37 +02:00

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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_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
};
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