* 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 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 <stddef.h>
#include "avcodec.h"
-#include "bitstream.h"
+#include "internal.h"
+#include "get_bits.h"
-#include "qcelp.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 struct {
+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,
- only use for RATE_OCTAVE and I_F_Q */
- float formant_mem[170];
- float last_codebook_gain;
- int prev_g1[2];
- int prev_bitrate;
- float prev_pitch_gain[4];
- uint8_t prev_pitch_lag[4];
- uint16_t first16bits;
+ 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;
-static void weighted_vector_sumf(float *out, const float *in_a,
- const float *in_b, float weight_coeff_a,
- float weight_coeff_b, int length)
-{
- int i;
-
- for(i=0; i<length; i++)
- out[i] = weight_coeff_a * in_a[i]
- + weight_coeff_b * in_b[i];
-}
-
/**
* 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.;
+ for(i=0; i<10; i++)
+ q->prev_lspf[i] = (i+1)/11.;
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
static int decode_lspf(QCELPContext *q, float *lspf)
{
int i;
- float tmp_lspf;
+ float tmp_lspf, smooth, erasure_coeff;
+ const float *predictors;
if(q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q)
{
- float smooth;
- const float *predictors = (q->prev_bitrate != RATE_OCTAVE &&
- q->prev_bitrate != I_F_Q ? q->prev_lspf
- : q->predictor_lspf);
+ predictors = (q->prev_bitrate != RATE_OCTAVE &&
+ q->prev_bitrate != I_F_Q ?
+ q->prev_lspf : q->predictor_lspf);
if(q->bitrate == RATE_OCTAVE)
{
smooth = (q->octave_count < 10 ? .875 : 0.1);
}else
{
- float erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
+ erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
assert(q->bitrate == I_F_Q);
lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));
// Low-pass filter the LSP frequencies.
- weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
+ ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
}else
{
q->octave_count = 0;
}
/**
- * 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) {
- case 1 : break;
- case 2 : g1[0] -= 1; break;
- case 3 : g1[0] -= 2; break;
- default: g1[0] -= 6;
+ 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)
+ 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++)
+ for(i=1; i<=subframes_count; i++)
gain[i-1] = q->last_codebook_gain + slope * i;
- q->last_codebook_gain = gain[i-2];
+ q->last_codebook_gain = gain[i-2];
q->prev_g1[0] = q->prev_g1[1];
q->prev_g1[1] = g1[0];
}
*/
static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
{
- int i, prev_diff=0;
+ int i, diff, prev_diff=0;
for(i=1; i<5; i++)
{
- int diff = cbgain[i] - cbgain[i-1];
+ diff = cbgain[i] - cbgain[i-1];
if(FFABS(diff) > 10)
return -1;
else if(FFABS(diff - prev_diff) > 12)
}
/**
- * 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.
* @param gain array holding the 4 pitch subframe gain values
* @param cdn_vector array for the generated scaled codebook vector
*/
-static void compute_svector(const QCELPContext *q, const float *gain,
+static void compute_svector(QCELPContext *q, const float *gain,
float *cdn_vector)
{
int i, j, k;
*cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
}
break;
+ case SILENCE:
+ memset(cdn_vector, 0, 160 * sizeof(float));
+ break;
}
}
* @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;
+ int i;
- 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
- av_log_missing_feature(NULL, "Zero energy for gain control", 1);
- for(len=j+40; j<len; j++)
- v_out[j] = scalefactor * v_in[j];
- }
+ 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);
}
/**
}
/**
- * Interpolates LSP frequencies and computes LPC coefficients
+ * 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
+ {
+ assert(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
+ * 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
*/
-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;
if(weight != 1.0)
{
- weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
- weight, 1.0 - weight, 10);
- qcelp_lspf2lpc(interpolated_lspf, lpc);
- }else if(q->bitrate >= RATE_QUARTER || (q->bitrate == I_F_Q && !subframe_num))
- qcelp_lspf2lpc(curr_lspf, lpc);
+ 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 int buf_size2bitrate(const int buf_size)
+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;
+ 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 -1;
+ 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 experimental warning when tested with samples.
+ av_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
+ }
+ return bitrate;
}
static void warn_insufficient_frame_quality(AVCodecContext *avctx,
message);
}
-static int qcelp_decode_frame(AVCodecContext *avctx,
- void *data,
- int *data_size,
- uint8_t *buf,
- const int buf_size) {
- QCELPContext *q = avctx->priv_data;
- float *outbuffer = data;
- int i;
- float quantized_lspf[10], lpc[10];
- float gain[16];
- float *formant_mem;
-
- if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {
+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 *data_size,
+ AVPacket *avpkt)
+{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+ QCELPContext *q = avctx->priv_data;
+ float *outbuffer = data;
+ int i;
+ float quantized_lspf[10], lpc[10];
+ float gain[16];
+ float *formant_mem;
+
+ 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) {
+ 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) {
+ 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];
memset(&q->frame, 0, sizeof(QCELPFrame));
- for (; bitmaps < bitmaps_end; bitmaps++)
+ 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) {
+ 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)) {
+ 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) {
+ 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++;
compute_svector(q, gain, outbuffer);
decode_lspf(q, quantized_lspf);
apply_pitch_filters(q, outbuffer);
- } else
+ }else
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);
+ 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;
return *data_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,
projects / ffmpeg / blobdiff