* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+
/**
* @file qcelpdec.c
* QCELP decoder
* @author Reynaldo H. Verdejo Pinochet
+ * @remark FFmpeg merging spearheaded by Kenan Gillet
+ * @remark Development mentored by Benjamin Larson
*/
#include <stddef.h>
#include "avcodec.h"
+#include "internal.h"
#include "bitstream.h"
-#include "qcelp.h"
#include "qcelpdata.h"
#include "celp_math.h"
#undef NDEBUG
#include <assert.h>
-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;
+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 */
- for (i = 0; i < length; i++)
+ 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;
+} QCELPContext;
+
+/**
+ * Reconstructs LPC coefficients from the line spectral pair frequencies.
+ *
+ * TIA/EIA/IS-733 2.4.3.3.5
+ */
+void ff_qcelp_lspf2lpc(const float *lspf, float *lpc);
+
+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.
+ *
+ * TIA/EIA/IS-733 2.4.9
+ */
+static av_cold int qcelp_decode_init(AVCodecContext *avctx)
+{
+ QCELPContext *q = avctx->priv_data;
+ int i;
+
+ avctx->sample_fmt = SAMPLE_FMT_FLT;
+
+ 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.
+ *
+ * @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.
+ weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
+ }else
+ {
+ q->octave_count = 0;
+
+ tmp_lspf = 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;
+}
+
+/**
+ * Converts 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 == 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;
+}
+
+/**
+ * Computes 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;
+ }
+}
+
+/**
+ * Apply generic gain control.
+ *
+ * @param v_out output vector
+ * @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
+ */
+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];
+ }
+}
+
/**
* 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]) {
+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]) {
+ 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++)
+ 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
+ }else
*v_out = *v_lag;
*v_out = *v_in + gain[i] * *v_out;
v_lag++;
v_out++;
}
- } else {
+ }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;
}
-static int buf_size2framerate(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;
+/**
+ * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
+ * TIA/EIA/IS-733 2.4.5.2
+ *
+ * @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 == 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 = q->erasure_count < 3 ? 0.9 - 0.3 * (q->erasure_count - 1) : 0.0;
+ 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));
+ }
+}
+
+/**
+ * Interpolates LSP frequencies and computes LPC coefficients
+ * for a given bitrate & pitch subframe.
+ *
+ * TIA/EIA/IS-733 2.4.3.3.4
+ *
+ * @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)
+{
+ 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)
+ {
+ weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
+ weight, 1.0 - weight, 10);
+ ff_qcelp_lspf2lpc(interpolated_lspf, lpc);
+ }else if(q->bitrate >= RATE_QUARTER ||
+ (q->bitrate == I_F_Q && !subframe_num))
+ ff_qcelp_lspf2lpc(curr_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 -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 int 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: the decoder should not handle SILENCE frames as I_F_Q frames
+ ff_log_missing_feature(avctx, "Blank frame", 1);
+ bitrate = I_F_Q;
+ }
+ 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 int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
+ const uint8_t *buf, 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)
+ {
+ 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;
+ }
+ 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
+
+ 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->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
+ q->prev_bitrate = q->bitrate;
+
+ *data_size = 160 * sizeof(*outbuffer);
+
+ return *data_size;
+}
+
+AVCodec qcelp_decoder =
+{
+ .name = "qcelp",
+ .type = CODEC_TYPE_AUDIO,
+ .id = CODEC_ID_QCELP,
+ .init = qcelp_decode_init,
+ .decode = qcelp_decode_frame,
+ .priv_data_size = sizeof(QCELPContext),
+ .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
+};