* QCELP decoder
* Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file libavcodec/qcelpdec.c
+ * @file
* QCELP decoder
* @author Reynaldo H. Verdejo Pinochet
- * @remark FFmpeg merging spearheaded by Kenan Gillet
+ * @remark Libav merging spearheaded by Kenan Gillet
* @remark Development mentored by Benjamin Larson
*/
#include "avcodec.h"
#include "internal.h"
-#include "bitstream.h"
+#include "get_bits.h"
#include "qcelpdata.h"
#include "celp_math.h"
#include "celp_filters.h"
+#include "acelp_filters.h"
#include "acelp_vectors.h"
+#include "lsp.h"
#undef NDEBUG
#include <assert.h>
typedef enum
{
- I_F_Q = -1, /*!< insufficient frame quality */
+ I_F_Q = -1, /**< insufficient frame quality */
SILENCE,
RATE_OCTAVE,
RATE_QUARTER,
typedef struct
{
+ AVFrame avframe;
GetBitContext gb;
qcelp_packet_rate bitrate;
- QCELPFrame frame; /*!< unpacked data frame */
+ QCELPFrame frame; /**< unpacked data frame */
uint8_t erasure_count;
- uint8_t octave_count; /*!< count the consecutive RATE_OCTAVE frames */
+ 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 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];
uint8_t pitch_lag[4];
uint16_t first16bits;
uint8_t warned_buf_mismatch_bitrate;
-} QCELPContext;
-/**
- * Reconstructs LPC coefficients from the line spectral pair frequencies.
- *
- * TIA/EIA/IS-733 2.4.3.3.5
- */
-void ff_celp_lspf2lpc(const double *lspf, float *lpc);
+ /* postfilter */
+ float postfilter_synth_mem[10];
+ float postfilter_agc_mem;
+ float postfilter_tilt_mem;
+} QCELPContext;
/**
* Initialize the speech codec according to the specification.
QCELPContext *q = avctx->priv_data;
int i;
- avctx->sample_fmt = SAMPLE_FMT_FLT;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
for(i=0; i<10; i++)
q->prev_lspf[i] = (i+1)/11.;
+ avcodec_get_frame_defaults(&q->avframe);
+ avctx->coded_frame = &q->avframe;
+
return 0;
}
/**
- * Decodes the 10 quantized LSP frequencies from the LSPV/LSP
- * transmission codes of any bitrate and checks for badly received packets.
+ * 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
float tmp_lspf, smooth, erasure_coeff;
const float *predictors;
- if(q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q)
- {
+ 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)
- {
+ if (q->bitrate == RATE_OCTAVE) {
q->octave_count++;
- for(i=0; i<10; i++)
- {
+ for (i=0; i<10; i++) {
q->predictor_lspf[i] =
lspf[i] = (q->frame.lspv[i] ? QCELP_LSP_SPREAD_FACTOR
: -QCELP_LSP_SPREAD_FACTOR)
+ (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11);
}
smooth = (q->octave_count < 10 ? .875 : 0.1);
- }else
- {
+ } else {
erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
assert(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++)
- {
+ for(i = 0; i < 10; i++) {
q->predictor_lspf[i] =
lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11
+ erasure_coeff * predictors[i];
// Low-pass filter the LSP frequencies.
ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
- }else
- {
+ } else {
q->octave_count = 0;
tmp_lspf = 0.;
- for(i=0; i<5 ; i++)
- {
+ 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 (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
- {
+ } else {
if(lspf[9] <= .66 || lspf[9] >= .985)
return -1;
for(i=4; i<10; i++)
}
/**
- * Converts codebook transmission codes to GAIN and INDEX.
+ * Convert codebook transmission codes to GAIN and INDEX.
*
* @param q the context
* @param gain array holding the decoded gain
int i, subframes_count, g1[16];
float slope;
- if(q->bitrate >= RATE_QUARTER)
- {
- switch(q->bitrate)
- {
+ 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++)
- {
+ for(i = 0; i < subframes_count; i++) {
g1[i] = 4 * q->frame.cbgain[i];
- if(q->bitrate == RATE_FULL && !((i+1) & 3))
- {
+ 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])
- {
+ if (q->frame.cbsign[i]) {
gain[i] = -gain[i];
q->frame.cindex[i] = (q->frame.cindex[i]-89) & 127;
}
q->prev_g1[1] = g1[i-1];
q->last_codebook_gain = qcelp_g12ga[g1[i-1]];
- if(q->bitrate == RATE_QUARTER)
- {
+ 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[2] = gain[1];
gain[1] = 0.6*gain[0] + 0.4*gain[1];
}
- }else
- {
- if(q->bitrate == RATE_OCTAVE)
- {
+ } 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
- {
+ } else {
assert(q->bitrate == I_F_Q);
g1[0] = q->prev_g1[1];
- switch(q->erasure_count)
- {
+ switch (q->erasure_count) {
case 1 : break;
case 2 : g1[0] -= 1; break;
case 3 : g1[0] -= 2; break;
{
int i, diff, prev_diff=0;
- for(i=1; i<5; i++)
- {
+ for(i=1; i<5; i++) {
diff = cbgain[i] - cbgain[i-1];
if(FFABS(diff) > 10)
return -1;
}
/**
- * Computes the scaled codebook vector Cdn From INDEX and GAIN
+ * Compute the scaled codebook vector Cdn From INDEX and GAIN
* for all rates.
*
* The specification lacks some information here.
uint16_t cbseed, cindex;
float *rnd, tmp_gain, fir_filter_value;
- switch(q->bitrate)
- {
+ switch (q->bitrate) {
case RATE_FULL:
- for(i=0; i<16; i++)
- {
+ 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++)
}
break;
case RATE_HALF:
- for(i=0; i<4; i++)
- {
+ 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++)
(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++)
- {
+ for (i = 0; i < 8; i++) {
tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
- for(k=0; k<20; k++)
- {
+ for (k = 0; k < 20; k++) {
cbseed = 521 * cbseed + 259;
*rnd = (int16_t)cbseed;
break;
case RATE_OCTAVE:
cbseed = q->first16bits;
- for(i=0; i<8; i++)
- {
+ for (i = 0; i < 8; i++) {
tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
- for(j=0; j<20; j++)
- {
+ 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++)
- {
+ 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];
* @param v_in gain-controlled vector
* @param v_ref vector to control gain of
*
- * FIXME: If v_ref is a zero vector, it energy is zero
- * and the behavior of the gain control is
- * undefined in the specs.
- *
- * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
+ * 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, j, len;
- float scalefactor;
-
- for(i=0, j=0; i<4; i++)
- {
- scalefactor = ff_dot_productf(v_in + j, v_in + j, 40);
- if(scalefactor)
- scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40)
- / scalefactor);
- else
- ff_log_missing_feature(NULL, "Zero energy for gain control", 1);
- for(len=j+40; j<len; j++)
- v_out[j] = scalefactor * v_in[j];
- }
+ int i;
+
+ for (i = 0; i < 160; i += 40)
+ ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i,
+ ff_dot_productf(v_ref + i,
+ v_ref + i, 40),
+ 40);
}
/**
v_out = memory + 143; // Output vector starts at memory[143].
- for(i=0; i<4; i++)
- {
- if(gain[i])
- {
+ 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 (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.; j<4; j++)
*v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
}else
v_lag++;
v_out++;
}
- }else
- {
+ } else {
memcpy(v_out, v_in, 40 * sizeof(float));
v_in += 40;
v_out += 40;
if(q->bitrate >= RATE_HALF ||
q->bitrate == SILENCE ||
- (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF)))
- {
+ (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) {
- if(q->bitrate >= RATE_HALF)
- {
+ if(q->bitrate >= RATE_HALF) {
// Compute gain & lag for the whole frame.
- for(i=0; i<4; i++)
- {
+ 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
- {
+ } else {
float max_pitch_gain;
- if (q->bitrate == I_F_Q)
- {
+ 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
- {
+ } else {
assert(q->bitrate == SILENCE);
max_pitch_gain = 1.0;
}
q->frame.pfrac);
apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
- }else
- {
+ } 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));
}
/**
- * Reconstructs LPC coefficients from the line spectral pair frequencies
- * and performs bandwidth expansion.
+ * 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: bandwith_expansion_coeff could be precalculated into a table
+ * @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 lsf[10];
- double bandwith_expansion_coeff = QCELP_BANDWITH_EXPANSION_COEFF;
+ double lsp[10];
+ double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
int i;
for (i=0; i<10; i++)
- lsf[i] = cos(M_PI * lspf[i]);
+ lsp[i] = cos(M_PI * lspf[i]);
- ff_celp_lspf2lpc(lsf, lpc);
+ ff_acelp_lspd2lpc(lsp, lpc, 5);
- for (i=0; i<10; i++)
- {
- lpc[i] *= bandwith_expansion_coeff;
- bandwith_expansion_coeff *= QCELP_BANDWITH_EXPANSION_COEFF;
+ for (i = 0; i < 10; i++) {
+ lpc[i] *= bandwidth_expansion_coeff;
+ bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
}
}
/**
- * Interpolates LSP frequencies and computes LPC coefficients
+ * 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 lpc float vector for the resulting LPC
* @param subframe_num frame number in decoded stream
*/
-void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc,
- const int subframe_num)
+static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
+ float *lpc, const int subframe_num)
{
float interpolated_lspf[10];
float weight;
else
weight = 1.0;
- if(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);
static qcelp_packet_rate buf_size2bitrate(const int buf_size)
{
- switch(buf_size)
- {
+ switch (buf_size) {
case 35: return RATE_FULL;
case 17: return RATE_HALF;
case 8: return RATE_QUARTER;
{
qcelp_packet_rate bitrate;
- if((bitrate = buf_size2bitrate(buf_size)) >= 0)
- {
- if(bitrate > **buf)
- {
+ if ((bitrate = buf_size2bitrate(buf_size)) >= 0) {
+ if (bitrate > **buf) {
QCELPContext *q = avctx->priv_data;
- if (!q->warned_buf_mismatch_bitrate)
- {
+ 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)
- {
+ } 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)
- {
+ } 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)
- {
+ if (bitrate == SILENCE) {
//FIXME: Remove experimental warning when tested with samples.
- ff_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
+ av_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
}
return bitrate;
}
message);
}
-static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
- const uint8_t *buf, int buf_size)
+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,
+ ff_dot_productf(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;
- float *outbuffer = data;
- int i;
+ float *outbuffer;
+ int i, ret;
float quantized_lspf[10], lpc[10];
float gain[16];
float *formant_mem;
- if((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q)
- {
+ /* get output buffer */
+ q->avframe.nb_samples = 160;
+ if ((ret = avctx->get_buffer(avctx, &q->avframe)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ outbuffer = (float *)q->avframe.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)
- {
+ (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)
- {
+ 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];
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)
- {
+ 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))
- {
+ 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)
- {
+ 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)
- {
+ 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)
- {
+ if (q->bitrate == I_F_Q) {
erasure:
q->bitrate = I_F_Q;
q->erasure_count++;
q->erasure_count = 0;
formant_mem = q->formant_mem + 10;
- for(i=0; i<4; i++)
- {
+ 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;
}
- memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
- // FIXME: postfilter and final gain control should be here.
- // TIA/EIA/IS-733 2.4.8.6
+ // postfilter, as per TIA/EIA/IS-733 2.4.8.6
+ postfilter(q, outbuffer, lpc);
- formant_mem = q->formant_mem + 10;
- for(i=0; i<160; i++)
- *outbuffer++ = av_clipf(*formant_mem++, QCELP_CLIP_LOWER_BOUND,
- QCELP_CLIP_UPPER_BOUND);
+ 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;
- *data_size = 160 * sizeof(*outbuffer);
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = q->avframe;
- return *data_size;
+ return buf_size;
}
-AVCodec qcelp_decoder =
+AVCodec ff_qcelp_decoder =
{
.name = "qcelp",
- .type = CODEC_TYPE_AUDIO,
+ .type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_QCELP,
.init = qcelp_decode_init,
.decode = qcelp_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
.priv_data_size = sizeof(QCELPContext),
.long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
};
projects / ffmpeg / blobdiff