Make the targa encoder return sane error values rather than -1.
[ffmpeg.git] / libavcodec / qcelpdec.c
index 14bd43c..0441e1f 100644 (file)
  * 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
+ * @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 "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>
 
-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 */
+
+    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->sample_fmt = SAMPLE_FMT_FLT;
+
+    for(i=0; i<10; i++)
+        q->prev_lspf[i] = (i+1)/11.;
+
+    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.;
+        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 < length; i++)
-        out[i] = weight_coeff_a * in_a[i]
-               + weight_coeff_b * in_b[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);
 }
 
 /**
@@ -63,29 +444,33 @@ static void weighted_vector_sumf(float *out,
  * @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;
@@ -93,74 +478,374 @@ static const float *do_pitchfilter(float memory[303],
                 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;
 }
 
 /**
- * Interpolates LSP frequencies and computes LPC coefficients
- * for a given framerate & pitch subframe.
+ * 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 (q->framerate >= RATE_QUARTER) {
+    if(q->bitrate >= RATE_QUARTER)
         weight = 0.25 * (subframe_num + 1);
-    } else if (q->framerate == RATE_OCTAVE && !subframe_num) {
+    else if(q->bitrate == RATE_OCTAVE && !subframe_num)
         weight = 0.625;
-    } else {
+    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;
     }
 
-    if (weight != 1.0) {
-        weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf, weight, 1.0 - weight, 10);
-        lspf2lpc(q, interpolated_lspf, lpc);
-    } else if (q->framerate >= RATE_QUARTER || (q->framerate == I_F_Q && !subframe_num))
-        lspf2lpc(q, curr_lspf, lpc);
+    return I_F_Q;
 }
 
-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;
+/**
+ * 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 -1;
+    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,
+        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)
+    {
+        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;
+
+    *data_size = 160 * sizeof(*outbuffer);
+
+    return *data_size;
 }
 
 AVCodec qcelp_decoder =
 {
     .name   = "qcelp",
-    .type   = CODEC_TYPE_AUDIO,
+    .type   = AVMEDIA_TYPE_AUDIO,
     .id     = CODEC_ID_QCELP,
     .init   = qcelp_decode_init,
     .decode = qcelp_decode_frame,