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kolibrios/contrib/sdk/sources/ffmpeg/ffmpeg-2.1/libavcodec/qcelpdec.c
Sergey Semyonov (Serge) ecf3e862ea ffmpeg-2.1.1: move directory 
git-svn-id: svn://kolibrios.org@6148 a494cfbc-eb01-0410-851d-a64ba20cac60
2016-02-05 22:14:10 +00:00

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/*
* QCELP decoder
* Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
*
* 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
*/
/**
* @file
* QCELP decoder
* @author Reynaldo H. Verdejo Pinochet
* @remark FFmpeg merging spearheaded by Kenan Gillet
* @remark Development mentored by Benjamin Larson
*/
#include <stddef.h>
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
#include "qcelpdata.h"
#include "celp_filters.h"
#include "acelp_filters.h"
#include "acelp_vectors.h"
#include "lsp.h"
typedef enum {
I_F_Q = -1, /**< insufficient frame quality */
SILENCE,
RATE_OCTAVE,
RATE_QUARTER,
RATE_HALF,
RATE_FULL
} qcelp_packet_rate;
typedef struct {
GetBitContext gb;
qcelp_packet_rate bitrate;
QCELPFrame frame; /**< unpacked data frame */
uint8_t erasure_count;
uint8_t octave_count; /**< count the consecutive RATE_OCTAVE frames */
float prev_lspf[10];
float predictor_lspf[10];/**< LSP predictor for RATE_OCTAVE and I_F_Q */
float pitch_synthesis_filter_mem[303];
float pitch_pre_filter_mem[303];
float rnd_fir_filter_mem[180];
float formant_mem[170];
float last_codebook_gain;
int prev_g1[2];
int prev_bitrate;
float pitch_gain[4];
uint8_t pitch_lag[4];
uint16_t first16bits;
uint8_t warned_buf_mismatch_bitrate;
/* postfilter */
float postfilter_synth_mem[10];
float postfilter_agc_mem;
float postfilter_tilt_mem;
} QCELPContext;
/**
* Initialize the speech codec according to the specification.
*
* TIA/EIA/IS-733 2.4.9
*/
static av_cold int qcelp_decode_init(AVCodecContext *avctx)
{
QCELPContext *q = avctx->priv_data;
int i;
avctx->channels = 1;
avctx->channel_layout = AV_CH_LAYOUT_MONO;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
for (i = 0; i < 10; i++)
q->prev_lspf[i] = (i + 1) / 11.0;
return 0;
}
/**
* Decode the 10 quantized LSP frequencies from the LSPV/LSP
* transmission codes of any bitrate and check for badly received packets.
*
* @param q the context
* @param lspf line spectral pair frequencies
*
* @return 0 on success, -1 if the packet is badly received
*
* TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
*/
static int decode_lspf(QCELPContext *q, float *lspf)
{
int i;
float tmp_lspf, smooth, erasure_coeff;
const float *predictors;
if (q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q) {
predictors = q->prev_bitrate != RATE_OCTAVE &&
q->prev_bitrate != I_F_Q ? q->prev_lspf
: q->predictor_lspf;
if (q->bitrate == RATE_OCTAVE) {
q->octave_count++;
for (i = 0; i < 10; i++) {
q->predictor_lspf[i] =
lspf[i] = (q->frame.lspv[i] ? QCELP_LSP_SPREAD_FACTOR
: -QCELP_LSP_SPREAD_FACTOR) +
predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR +
(i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR) / 11);
}
smooth = q->octave_count < 10 ? .875 : 0.1;
} else {
erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
av_assert2(q->bitrate == I_F_Q);
if (q->erasure_count > 1)
erasure_coeff *= q->erasure_count < 4 ? 0.9 : 0.7;
for (i = 0; i < 10; i++) {
q->predictor_lspf[i] =
lspf[i] = (i + 1) * (1 - erasure_coeff) / 11 +
erasure_coeff * predictors[i];
}
smooth = 0.125;
}
// Check the stability of the LSP frequencies.
lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
for (i = 1; i < 10; i++)
lspf[i] = FFMAX(lspf[i], lspf[i - 1] + QCELP_LSP_SPREAD_FACTOR);
lspf[9] = FFMIN(lspf[9], 1.0 - QCELP_LSP_SPREAD_FACTOR);
for (i = 9; i > 0; i--)
lspf[i - 1] = FFMIN(lspf[i - 1], lspf[i] - QCELP_LSP_SPREAD_FACTOR);
// Low-pass filter the LSP frequencies.
ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0 - smooth, 10);
} else {
q->octave_count = 0;
tmp_lspf = 0.0;
for (i = 0; i < 5; i++) {
lspf[2 * i + 0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
lspf[2 * i + 1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
}
// Check for badly received packets.
if (q->bitrate == RATE_QUARTER) {
if (lspf[9] <= .70 || lspf[9] >= .97)
return -1;
for (i = 3; i < 10; i++)
if (fabs(lspf[i] - lspf[i - 2]) < .08)
return -1;
} else {
if (lspf[9] <= .66 || lspf[9] >= .985)
return -1;
for (i = 4; i < 10; i++)
if (fabs(lspf[i] - lspf[i - 4]) < .0931)
return -1;
}
}
return 0;
}
/**
* Convert codebook transmission codes to GAIN and INDEX.
*
* @param q the context
* @param gain array holding the decoded gain
*
* TIA/EIA/IS-733 2.4.6.2
*/
static void decode_gain_and_index(QCELPContext *q, float *gain)
{
int i, subframes_count, g1[16];
float slope;
if (q->bitrate >= RATE_QUARTER) {
switch (q->bitrate) {
case RATE_FULL: subframes_count = 16; break;
case RATE_HALF: subframes_count = 4; break;
default: subframes_count = 5;
}
for (i = 0; i < subframes_count; i++) {
g1[i] = 4 * q->frame.cbgain[i];
if (q->bitrate == RATE_FULL && !((i + 1) & 3)) {
g1[i] += av_clip((g1[i - 1] + g1[i - 2] + g1[i - 3]) / 3 - 6, 0, 32);
}
gain[i] = qcelp_g12ga[g1[i]];
if (q->frame.cbsign[i]) {
gain[i] = -gain[i];
q->frame.cindex[i] = (q->frame.cindex[i] - 89) & 127;
}
}
q->prev_g1[0] = g1[i - 2];
q->prev_g1[1] = g1[i - 1];
q->last_codebook_gain = qcelp_g12ga[g1[i - 1]];
if (q->bitrate == RATE_QUARTER) {
// Provide smoothing of the unvoiced excitation energy.
gain[7] = gain[4];
gain[6] = 0.4 * gain[3] + 0.6 * gain[4];
gain[5] = gain[3];
gain[4] = 0.8 * gain[2] + 0.2 * gain[3];
gain[3] = 0.2 * gain[1] + 0.8 * gain[2];
gain[2] = gain[1];
gain[1] = 0.6 * gain[0] + 0.4 * gain[1];
}
} else if (q->bitrate != SILENCE) {
if (q->bitrate == RATE_OCTAVE) {
g1[0] = 2 * q->frame.cbgain[0] +
av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
subframes_count = 8;
} else {
av_assert2(q->bitrate == I_F_Q);
g1[0] = q->prev_g1[1];
switch (q->erasure_count) {
case 1 : break;
case 2 : g1[0] -= 1; break;
case 3 : g1[0] -= 2; break;
default: g1[0] -= 6;
}
if (g1[0] < 0)
g1[0] = 0;
subframes_count = 4;
}
// This interpolation is done to produce smoother background noise.
slope = 0.5 * (qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
for (i = 1; i <= subframes_count; i++)
gain[i - 1] = q->last_codebook_gain + slope * i;
q->last_codebook_gain = gain[i - 2];
q->prev_g1[0] = q->prev_g1[1];
q->prev_g1[1] = g1[0];
}
}
/**
* If the received packet is Rate 1/4 a further sanity check is made of the
* codebook gain.
*
* @param cbgain the unpacked cbgain array
* @return -1 if the sanity check fails, 0 otherwise
*
* TIA/EIA/IS-733 2.4.8.7.3
*/
static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
{
int i, diff, prev_diff = 0;
for (i = 1; i < 5; i++) {
diff = cbgain[i] - cbgain[i-1];
if (FFABS(diff) > 10)
return -1;
else if (FFABS(diff - prev_diff) > 12)
return -1;
prev_diff = diff;
}
return 0;
}
/**
* Compute the scaled codebook vector Cdn From INDEX and GAIN
* for all rates.
*
* The specification lacks some information here.
*
* TIA/EIA/IS-733 has an omission on the codebook index determination
* formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
* you have to subtract the decoded index parameter from the given scaled
* codebook vector index 'n' to get the desired circular codebook index, but
* it does not mention that you have to clamp 'n' to [0-9] in order to get
* RI-compliant results.
*
* The reason for this mistake seems to be the fact they forgot to mention you
* have to do these calculations per codebook subframe and adjust given
* equation values accordingly.
*
* @param q the context
* @param gain array holding the 4 pitch subframe gain values
* @param cdn_vector array for the generated scaled codebook vector
*/
static void compute_svector(QCELPContext *q, const float *gain,
float *cdn_vector)
{
int i, j, k;
uint16_t cbseed, cindex;
float *rnd, tmp_gain, fir_filter_value;
switch (q->bitrate) {
case RATE_FULL:
for (i = 0; i < 16; i++) {
tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
cindex = -q->frame.cindex[i];
for (j = 0; j < 10; j++)
*cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
}
break;
case RATE_HALF:
for (i = 0; i < 4; i++) {
tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
cindex = -q->frame.cindex[i];
for (j = 0; j < 40; j++)
*cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
}
break;
case RATE_QUARTER:
cbseed = (0x0003 & q->frame.lspv[4]) << 14 |
(0x003F & q->frame.lspv[3]) << 8 |
(0x0060 & q->frame.lspv[2]) << 1 |
(0x0007 & q->frame.lspv[1]) << 3 |
(0x0038 & q->frame.lspv[0]) >> 3;
rnd = q->rnd_fir_filter_mem + 20;
for (i = 0; i < 8; i++) {
tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
for (k = 0; k < 20; k++) {
cbseed = 521 * cbseed + 259;
*rnd = (int16_t) cbseed;
// FIR filter
fir_filter_value = 0.0;
for (j = 0; j < 10; j++)
fir_filter_value += qcelp_rnd_fir_coefs[j] *
(rnd[-j] + rnd[-20+j]);
fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
*cdn_vector++ = tmp_gain * fir_filter_value;
rnd++;
}
}
memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160,
20 * sizeof(float));
break;
case RATE_OCTAVE:
cbseed = q->first16bits;
for (i = 0; i < 8; i++) {
tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
for (j = 0; j < 20; j++) {
cbseed = 521 * cbseed + 259;
*cdn_vector++ = tmp_gain * (int16_t) cbseed;
}
}
break;
case I_F_Q:
cbseed = -44; // random codebook index
for (i = 0; i < 4; i++) {
tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
for (j = 0; j < 40; j++)
*cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
}
break;
case SILENCE:
memset(cdn_vector, 0, 160 * sizeof(float));
break;
}
}
/**
* Apply generic gain control.
*
* @param v_out output vector
* @param v_in gain-controlled vector
* @param v_ref vector to control gain of
*
* TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
*/
static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)
{
int i;
for (i = 0; i < 160; i += 40) {
float res = avpriv_scalarproduct_float_c(v_ref + i, v_ref + i, 40);
ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i, res, 40);
}
}
/**
* Apply filter in pitch-subframe steps.
*
* @param memory buffer for the previous state of the filter
* - must be able to contain 303 elements
* - the 143 first elements are from the previous state
* - the next 160 are for output
* @param v_in input filter vector
* @param gain per-subframe gain array, each element is between 0.0 and 2.0
* @param lag per-subframe lag array, each element is
* - between 16 and 143 if its corresponding pfrac is 0,
* - between 16 and 139 otherwise
* @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
* otherwise
*
* @return filter output vector
*/
static const float *do_pitchfilter(float memory[303], const float v_in[160],
const float gain[4], const uint8_t *lag,
const uint8_t pfrac[4])
{
int i, j;
float *v_lag, *v_out;
const float *v_len;
v_out = memory + 143; // Output vector starts at memory[143].
for (i = 0; i < 4; i++) {
if (gain[i]) {
v_lag = memory + 143 + 40 * i - lag[i];
for (v_len = v_in + 40; v_in < v_len; v_in++) {
if (pfrac[i]) { // If it is a fractional lag...
for (j = 0, *v_out = 0.0; j < 4; j++)
*v_out += qcelp_hammsinc_table[j] * (v_lag[j - 4] + v_lag[3 - j]);
} else
*v_out = *v_lag;
*v_out = *v_in + gain[i] * *v_out;
v_lag++;
v_out++;
}
} else {
memcpy(v_out, v_in, 40 * sizeof(float));
v_in += 40;
v_out += 40;
}
}
memmove(memory, memory + 160, 143 * sizeof(float));
return memory + 143;
}
/**
* Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
* TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
*
* @param q the context
* @param cdn_vector the scaled codebook vector
*/
static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
{
int i;
const float *v_synthesis_filtered, *v_pre_filtered;
if (q->bitrate >= RATE_HALF || q->bitrate == SILENCE ||
(q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) {
if (q->bitrate >= RATE_HALF) {
// Compute gain & lag for the whole frame.
for (i = 0; i < 4; i++) {
q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
q->pitch_lag[i] = q->frame.plag[i] + 16;
}
} else {
float max_pitch_gain;
if (q->bitrate == I_F_Q) {
if (q->erasure_count < 3)
max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
else
max_pitch_gain = 0.0;
} else {
av_assert2(q->bitrate == SILENCE);
max_pitch_gain = 1.0;
}
for (i = 0; i < 4; i++)
q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
}
// pitch synthesis filter
v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
cdn_vector, q->pitch_gain,
q->pitch_lag, q->frame.pfrac);
// pitch prefilter update
for (i = 0; i < 4; i++)
q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
v_synthesis_filtered,
q->pitch_gain, q->pitch_lag,
q->frame.pfrac);
apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
} else {
memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17, 143 * sizeof(float));
memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
memset(q->pitch_lag, 0, sizeof(q->pitch_lag));
}
}
/**
* Reconstruct LPC coefficients from the line spectral pair frequencies
* and perform bandwidth expansion.
*
* @param lspf line spectral pair frequencies
* @param lpc linear predictive coding coefficients
*
* @note: bandwidth_expansion_coeff could be precalculated into a table
* but it seems to be slower on x86
*
* TIA/EIA/IS-733 2.4.3.3.5
*/
static void lspf2lpc(const float *lspf, float *lpc)
{
double lsp[10];
double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
int i;
for (i = 0; i < 10; i++)
lsp[i] = cos(M_PI * lspf[i]);
ff_acelp_lspd2lpc(lsp, lpc, 5);
for (i = 0; i < 10; i++) {
lpc[i] *= bandwidth_expansion_coeff;
bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
}
}
/**
* Interpolate LSP frequencies and compute LPC coefficients
* for a given bitrate & pitch subframe.
*
* TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
*
* @param q the context
* @param curr_lspf LSP frequencies vector of the current frame
* @param lpc float vector for the resulting LPC
* @param subframe_num frame number in decoded stream
*/
static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
float *lpc, const int subframe_num)
{
float interpolated_lspf[10];
float weight;
if (q->bitrate >= RATE_QUARTER)
weight = 0.25 * (subframe_num + 1);
else if (q->bitrate == RATE_OCTAVE && !subframe_num)
weight = 0.625;
else
weight = 1.0;
if (weight != 1.0) {
ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
weight, 1.0 - weight, 10);
lspf2lpc(interpolated_lspf, lpc);
} else if (q->bitrate >= RATE_QUARTER ||
(q->bitrate == I_F_Q && !subframe_num))
lspf2lpc(curr_lspf, lpc);
else if (q->bitrate == SILENCE && !subframe_num)
lspf2lpc(q->prev_lspf, lpc);
}
static qcelp_packet_rate buf_size2bitrate(const int buf_size)
{
switch (buf_size) {
case 35: return RATE_FULL;
case 17: return RATE_HALF;
case 8: return RATE_QUARTER;
case 4: return RATE_OCTAVE;
case 1: return SILENCE;
}
return I_F_Q;
}
/**
* Determine the bitrate from the frame size and/or the first byte of the frame.
*
* @param avctx the AV codec context
* @param buf_size length of the buffer
* @param buf the bufffer
*
* @return the bitrate on success,
* I_F_Q if the bitrate cannot be satisfactorily determined
*
* TIA/EIA/IS-733 2.4.8.7.1
*/
static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx,
const int buf_size,
const uint8_t **buf)
{
qcelp_packet_rate bitrate;
if ((bitrate = buf_size2bitrate(buf_size)) >= 0) {
if (bitrate > **buf) {
QCELPContext *q = avctx->priv_data;
if (!q->warned_buf_mismatch_bitrate) {
av_log(avctx, AV_LOG_WARNING,
"Claimed bitrate and buffer size mismatch.\n");
q->warned_buf_mismatch_bitrate = 1;
}
bitrate = **buf;
} else if (bitrate < **buf) {
av_log(avctx, AV_LOG_ERROR,
"Buffer is too small for the claimed bitrate.\n");
return I_F_Q;
}
(*buf)++;
} else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) {
av_log(avctx, AV_LOG_WARNING,
"Bitrate byte is missing, guessing the bitrate from packet size.\n");
} else
return I_F_Q;
if (bitrate == SILENCE) {
// FIXME: Remove this warning when tested with samples.
avpriv_request_sample(avctx, "Blank frame handling");
}
return bitrate;
}
static void warn_insufficient_frame_quality(AVCodecContext *avctx,
const char *message)
{
av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n",
avctx->frame_number, message);
}
static void postfilter(QCELPContext *q, float *samples, float *lpc)
{
static const float pow_0_775[10] = {
0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
0.216676, 0.167924, 0.130141, 0.100859, 0.078166
}, pow_0_625[10] = {
0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
0.059605, 0.037253, 0.023283, 0.014552, 0.009095
};
float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
int n;
for (n = 0; n < 10; n++) {
lpc_s[n] = lpc[n] * pow_0_625[n];
lpc_p[n] = lpc[n] * pow_0_775[n];
}
ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
q->formant_mem + 10, 160, 10);
memcpy(pole_out, q->postfilter_synth_mem, sizeof(float) * 10);
ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
ff_adaptive_gain_control(samples, pole_out + 10,
avpriv_scalarproduct_float_c(q->formant_mem + 10,
q->formant_mem + 10,
160),
160, 0.9375, &q->postfilter_agc_mem);
}
static int qcelp_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
QCELPContext *q = avctx->priv_data;
AVFrame *frame = data;
float *outbuffer;
int i, ret;
float quantized_lspf[10], lpc[10];
float gain[16];
float *formant_mem;
/* get output buffer */
frame->nb_samples = 160;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
outbuffer = (float *)frame->data[0];
if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {
warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
goto erasure;
}
if (q->bitrate == RATE_OCTAVE &&
(q->first16bits = AV_RB16(buf)) == 0xFFFF) {
warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
goto erasure;
}
if (q->bitrate > SILENCE) {
const QCELPBitmap *bitmaps = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate] +
qcelp_unpacking_bitmaps_lengths[q->bitrate];
uint8_t *unpacked_data = (uint8_t *)&q->frame;
init_get_bits(&q->gb, buf, 8 * buf_size);
memset(&q->frame, 0, sizeof(QCELPFrame));
for (; bitmaps < bitmaps_end; bitmaps++)
unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
// Check for erasures/blanks on rates 1, 1/4 and 1/8.
if (q->frame.reserved) {
warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
goto erasure;
}
if (q->bitrate == RATE_QUARTER &&
codebook_sanity_check_for_rate_quarter(q->frame.cbgain)) {
warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
goto erasure;
}
if (q->bitrate >= RATE_HALF) {
for (i = 0; i < 4; i++) {
if (q->frame.pfrac[i] && q->frame.plag[i] >= 124) {
warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
goto erasure;
}
}
}
}
decode_gain_and_index(q, gain);
compute_svector(q, gain, outbuffer);
if (decode_lspf(q, quantized_lspf) < 0) {
warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
goto erasure;
}
apply_pitch_filters(q, outbuffer);
if (q->bitrate == I_F_Q) {
erasure:
q->bitrate = I_F_Q;
q->erasure_count++;
decode_gain_and_index(q, gain);
compute_svector(q, gain, outbuffer);
decode_lspf(q, quantized_lspf);
apply_pitch_filters(q, outbuffer);
} else
q->erasure_count = 0;
formant_mem = q->formant_mem + 10;
for (i = 0; i < 4; i++) {
interpolate_lpc(q, quantized_lspf, lpc, i);
ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40, 10);
formant_mem += 40;
}
// postfilter, as per TIA/EIA/IS-733 2.4.8.6
postfilter(q, outbuffer, lpc);
memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
q->prev_bitrate = q->bitrate;
*got_frame_ptr = 1;
return buf_size;
}
AVCodec ff_qcelp_decoder = {
.name = "qcelp",
.long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_QCELP,
.init = qcelp_decode_init,
.decode = qcelp_decode_frame,
.capabilities = CODEC_CAP_DR1,
.priv_data_size = sizeof(QCELPContext),
};
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