* 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 Libav merging spearheaded by Kenan Gillet
+ * @remark Development mentored by Benjamin Larson
*/
#include <stddef.h>
+#include "libavutil/channel_layout.h"
+#include "libavutil/float_dsp.h"
#include "avcodec.h"
-#include "bitstream.h"
-
-#include "qcelp.h"
+#include "internal.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 */
+ 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;
+
+ 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++) {
+ 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 {
+ 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;
+ }
+ 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 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
+ * @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;
+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++)
+ 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.; j < 4; j++)
- *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
+ 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_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 {
+ 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, 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);
+ 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) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ 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),
+};