A typo fixed in the OKI ADPCM codec

The V.17 modem has been modified, so it will not clear the most recent
saved AGC value if long training is requested.
The T.38 gateway codec now has fillin implemented properly

libs/spandsp/src/oki_adpcm.c

@@ -26,8 +26,6 @@
  *
  * The actual OKI ADPCM encode and decode method is derived from freely
  * available code, whose exact origins seem uncertain.
- *
- * $Id: oki_adpcm.c,v 1.32 2009/02/10 13:06:46 steveu Exp $
  */
 
 /*! \file */
@@ -153,7 +151,7 @@ static const float cutoff_coeffs[] =
 
 static int16_t decode(oki_adpcm_state_t *s, uint8_t adpcm)
 {
-    int16_t e;
+    int16_t d;
     int16_t ss;
     int16_t linear;
 
@@ -170,20 +168,20 @@ static int16_t decode(oki_adpcm_state_t *s, uint8_t adpcm)
      */
 
     ss = step_size[s->step_index];
-    e = ss >> 3;
+    d = ss >> 3;
     if (adpcm & 0x01)
-        e += (ss >> 2);
+        d += (ss >> 2);
     /*endif*/
     if (adpcm & 0x02)
-        e += (ss >> 1);
+        d += (ss >> 1);
     /*endif*/
     if (adpcm & 0x04)
-        e += ss;
+        d += ss;
     /*endif*/
     if (adpcm & 0x08)
-        e = -e;
+        d = -d;
     /*endif*/
-    linear = s->last + e;
+    linear = s->last + d;
 
     /* Saturate the values to +/- 2^11 (supposed to be 12 bits) */
     if (linear > 2047)
@@ -206,32 +204,32 @@ static int16_t decode(oki_adpcm_state_t *s, uint8_t adpcm)
 
 static uint8_t encode(oki_adpcm_state_t *s, int16_t linear)
 {
-    int16_t e;
+    int16_t d;
     int16_t ss;
     uint8_t adpcm;
 
     ss = step_size[s->step_index];
-    e = (linear >> 4) - s->last;
+    d = (linear >> 4) - s->last;
     adpcm = (uint8_t) 0x00;
-    if (e < 0)
+    if (d < 0)
     {
         adpcm = (uint8_t) 0x08;
-        e = -e;
+        d = -d;
     }
     /*endif*/
-    if (e >= ss)
+    if (d >= ss)
     {
         adpcm |= (uint8_t) 0x04;
-        e -= ss;
+        d -= ss;
     }
     /*endif*/
-    if (e >= (ss >> 1))
+    if (d >= (ss >> 1))
     {
         adpcm |= (uint8_t) 0x02;
-        e -= ss;
+        d -= (ss >> 1);
     }
     /*endif*/
-    if (e >= (ss >> 2))
+    if (d >= (ss >> 2))
         adpcm |= (uint8_t) 0x01;
     /*endif*/
 

libs/spandsp/src/spandsp/private/t38_gateway.h

@@ -21,8 +21,6 @@
  * You should have received a copy of the GNU Lesser General Public
  * License along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- *
- * $Id: t38_gateway.h,v 1.5 2009/11/07 08:58:58 steveu Exp $
  */
 
 /*! \file */
@@ -35,7 +33,7 @@
 */
 typedef struct
 {
-    /*! Core T.38 IFP support */
+    /*! \brief Core T.38 IFP support */
     t38_core_state_t t38;
 
     /*! \brief TRUE if the NSF, NSC, and NSS are to be suppressed by altering
@@ -61,9 +59,10 @@ typedef struct
 {
     /*! \brief The FAX modem set for the audio side fo the gateway. */
     fax_modems_state_t modems;
-    /*! \brief The current receive signal handler. Actual receiving hop between this
+    /*! \brief The current receive signal handler. Actual receiving hops between this
                and a dummy receive routine. */
     span_rx_handler_t *base_rx_handler;
+    span_rx_fillin_handler_t *base_rx_fillin_handler;
 } t38_gateway_audio_state_t;
 
 /*!
@@ -144,6 +143,8 @@ typedef struct
     int supported_modems;
     /*! \brief TRUE if ECM FAX mode is allowed through the gateway. */
     int ecm_allowed;
+    /*! \brief Required time between T.38 transmissions, in ms. */
+    int ms_per_tx_chunk;
 
     /*! \brief TRUE if in image data modem is to use short training. This usually
                follows image_data_mode, but in ECM mode T.30 defines recovery

libs/spandsp/src/spandsp/private/v17rx.h

@@ -21,8 +21,6 @@
  * You should have received a copy of the GNU Lesser General Public
  * License along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- *
- * $Id: v17rx.h,v 1.2.4.1 2009/12/24 16:52:30 steveu Exp $
  */
 
 #if !defined(_SPANDSP_PRIVATE_V17RX_H_)
@@ -30,10 +28,11 @@
 
 /* Target length for the equalizer is about 63 taps, to deal with the worst stuff
    in V.56bis. */
+/*! The length of the equalizer buffer */
+#define V17_EQUALIZER_LEN           33
+
 /*! Samples before the target position in the equalizer buffer */
-#define V17_EQUALIZER_PRE_LEN       8
-/*! Samples after the target position in the equalizer buffer */
-#define V17_EQUALIZER_POST_LEN      8
+#define V17_EQUALIZER_PRE_LEN       16
 
 /*! The number of taps in the pulse shaping/bandpass filter */
 #define V17_RX_FILTER_STEPS         27
@@ -137,7 +136,7 @@ struct v17_rx_state_s
     int eq_step;
     /*! \brief Current write offset into the equalizer buffer. */
     int eq_put_step;
-    /*! \brief Symbol counter to the next equalizer update. */
+    /*! \brief Symbol count to the next equalizer update. */
     int eq_skip;
 
     /*! \brief The current half of the baud. */
@@ -152,11 +151,11 @@ struct v17_rx_state_s
     /*! \brief The current delta factor for updating the equalizer coefficients. */
     float eq_delta;
     /*! \brief The adaptive equalizer coefficients. */
-    complexi16_t eq_coeff[V17_EQUALIZER_PRE_LEN + 1 + V17_EQUALIZER_POST_LEN];
+    complexi16_t eq_coeff[V17_EQUALIZER_LEN];
     /*! \brief A saved set of adaptive equalizer coefficients for use after restarts. */
-    complexi16_t eq_coeff_save[V17_EQUALIZER_PRE_LEN + 1 + V17_EQUALIZER_POST_LEN];
+    complexi16_t eq_coeff_save[V17_EQUALIZER_LEN];
     /*! \brief The equalizer signal buffer. */
-    complexi16_t eq_buf[V17_EQUALIZER_PRE_LEN + 1 + V17_EQUALIZER_POST_LEN];
+    complexi16_t eq_buf[V17_EQUALIZER_LEN];
 
     /*! Low band edge filter for symbol sync. */
     int32_t symbol_sync_low[2];
@@ -175,11 +174,11 @@ struct v17_rx_state_s
     /*! \brief The current delta factor for updating the equalizer coefficients. */
     float eq_delta;
     /*! \brief The adaptive equalizer coefficients. */
-    complexf_t eq_coeff[V17_EQUALIZER_PRE_LEN + 1 + V17_EQUALIZER_POST_LEN];
+    complexf_t eq_coeff[V17_EQUALIZER_LEN];
     /*! \brief A saved set of adaptive equalizer coefficients for use after restarts. */
-    complexf_t eq_coeff_save[V17_EQUALIZER_PRE_LEN + 1 + V17_EQUALIZER_POST_LEN];
+    complexf_t eq_coeff_save[V17_EQUALIZER_LEN];
     /*! \brief The equalizer signal buffer. */
-    complexf_t eq_buf[V17_EQUALIZER_PRE_LEN + 1 + V17_EQUALIZER_POST_LEN];
+    complexf_t eq_buf[V17_EQUALIZER_LEN];
 
     /*! Low band edge filter for symbol sync. */
     float symbol_sync_low[2];

libs/spandsp/src/spandsp/v17rx.h

@@ -287,7 +287,7 @@ SPAN_DECLARE_NONSTD(int) v17_rx(v17_rx_state_t *s, const int16_t amp[], int len)
     \param len The number of samples to fake.
     \return The number of samples unprocessed.
 */
-SPAN_DECLARE(int) v17_rx_fillin(v17_rx_state_t *s, int len);
+SPAN_DECLARE_NONSTD(int) v17_rx_fillin(v17_rx_state_t *s, int len);
 
 /*! Get a snapshot of the current equalizer coefficients.
     \brief Get a snapshot of the current equalizer coefficients.

libs/spandsp/src/v17rx.c

@@ -22,8 +22,6 @@
  * You should have received a copy of the GNU Lesser General Public
  * License along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- *
- * $Id: v17rx.c,v 1.153.4.6 2009/12/28 12:20:46 steveu Exp $
  */
 
 /*! \file */
@@ -98,9 +96,6 @@
 /*! The 16 bit pattern used in the bridge section of the training sequence */
 #define V17_BRIDGE_WORD                 0x8880
 
-/*! The length of the equalizer buffer */
-#define V17_EQUALIZER_LEN    (V17_EQUALIZER_PRE_LEN + 1 + V17_EQUALIZER_POST_LEN)
-
 enum
 {
     TRAINING_STAGE_NORMAL_OPERATION = 0,
@@ -233,6 +228,7 @@ static void equalizer_restore(v17_rx_state_t *s)
 
     s->eq_put_step = RX_PULSESHAPER_COEFF_SETS*10/(3*2) - 1;
     s->eq_step = 0;
+    s->eq_skip = 0;
 }
 /*- End of function --------------------------------------------------------*/
 
@@ -253,6 +249,7 @@ static void equalizer_reset(v17_rx_state_t *s)
 
     s->eq_put_step = RX_PULSESHAPER_COEFF_SETS*10/(3*2) - 1;
     s->eq_step = 0;
+    s->eq_skip = 0;
 }
 /*- End of function --------------------------------------------------------*/
 
@@ -565,7 +562,7 @@ static __inline__ void symbol_sync(v17_rx_state_t *s)
         Passband Timing Recovery in an All-Digital Modem Receiver
         IEEE TRANSACTIONS ON COMMUNICATIONS, VOL. COM-26, NO. 5, MAY 1978 */
 
-    /* This is slightly rearranged for figure 3b of the Godard paper, as this saves a couple of
+    /* This is slightly rearranged from figure 3b of the Godard paper, as this saves a couple of
        maths operations */
 #if defined(SPANDSP_USE_FIXED_POINTx)
     /* TODO: The scalings used here need more thorough evaluation, to see if overflows are possible. */
@@ -579,12 +576,12 @@ static __inline__ void symbol_sync(v17_rx_state_t *s)
     s->symbol_sync_dc_filter[0] = v;
     /* A little integration will now filter away much of the HF noise */
     s->baud_phase -= p;
-    if (abs(s->baud_phase) > 100*FP_FACTOR)
+    v = labs(s->baud_phase);
+    if (v > 100*FP_FACTOR)
     {
-        if (s->baud_phase > 0)
-            i = (s->baud_phase > 1000*FP_FACTOR)  ?  15  :  1;
-        else
-            i = (s->baud_phase < -1000*FP_FACTOR)  ?  -15  :  -1;
+        i = (v > 1000*FP_FACTOR)  ?  15  :  1;
+        if (s->baud_phase < 0)
+            i = -i;
         //printf("v = %10.5f %5d - %f %f %d %d\n", v, i, p, s->baud_phase, s->total_baud_timing_correction);
         s->eq_put_step += i;
         s->total_baud_timing_correction += i;
@@ -600,12 +597,12 @@ static __inline__ void symbol_sync(v17_rx_state_t *s)
     s->symbol_sync_dc_filter[0] = v;
     /* A little integration will now filter away much of the HF noise */
     s->baud_phase -= p;
-    if (fabsf(s->baud_phase) > 100.0f)
+    v = fabsf(s->baud_phase);
+    if (v > 100.0f)
     {
-        if (s->baud_phase > 0.0f)
-            i = (s->baud_phase > 1000.0f)  ?  15  :  1;
-        else
-            i = (s->baud_phase < -1000.0f)  ?  -15  :  -1;
+        i = (v > 1000.0f)  ?  15  :  1;
+        if (s->baud_phase < 0.0f)
+            i = -i;
         //printf("v = %10.5f %5d - %f %f %d\n", v, i, p, s->baud_phase, s->total_baud_timing_correction);
         s->eq_put_step += i;
         s->total_baud_timing_correction += i;
@@ -732,18 +729,14 @@ static void process_half_baud(v17_rx_state_t *s, const complexf_t *sample)
         if (s->training_count == 100)
         {
             i = s->training_count;
-            /* Avoid the possibility of a divide by zero */
-            if (i)
-            {
-                j = i & 0xF;
-                ang = (s->angles[j] - s->start_angles[0])/i
-                    + (s->angles[j | 0x1] - s->start_angles[1])/i;
-                s->carrier_phase_rate += 3*(ang/20);
-                //span_log(&s->logging, SPAN_LOG_FLOW, "Angles %x, %x, %x, %x, dist %d\n", s->angles[j], s->start_angles[0], s->angles[j | 0x1], s->start_angles[1], i);
-
-                s->start_angles[0] = s->angles[j];
-                s->start_angles[1] = s->angles[j | 0x1];
-            }
+            j = i & 0xF;
+            ang = (s->angles[j] - s->start_angles[0])/i
+                + (s->angles[j | 0x1] - s->start_angles[1])/i;
+            s->carrier_phase_rate += 3*(ang/20);
+            //span_log(&s->logging, SPAN_LOG_FLOW, "Angles %x, %x, %x, %x, dist %d\n", s->angles[j], s->start_angles[0], s->angles[j | 0x1], s->start_angles[1], i);
+
+            s->start_angles[0] = s->angles[j];
+            s->start_angles[1] = s->angles[j | 0x1];
             //span_log(&s->logging, SPAN_LOG_FLOW, "%d %d %d %d %d\n", s->angles[s->training_count & 0xF], s->start_angles[0], s->angles[(s->training_count | 0x1) & 0xF], s->start_angles[1], s->training_count);
             span_log(&s->logging, SPAN_LOG_FLOW, "First coarse carrier frequency %7.2f (%d)\n", dds_frequencyf(s->carrier_phase_rate), s->training_count);
 
@@ -1219,7 +1212,7 @@ SPAN_DECLARE_NONSTD(int) v17_rx(v17_rx_state_t *s, const int16_t amp[], int len)
 }
 /*- End of function --------------------------------------------------------*/
 
-SPAN_DECLARE(int) v17_rx_fillin(v17_rx_state_t *s, int len)
+SPAN_DECLARE_NONSTD(int) v17_rx_fillin(v17_rx_state_t *s, int len)
 {
     int i;
 
@@ -1343,15 +1336,14 @@ SPAN_DECLARE(int) v17_rx_restart(v17_rx_state_t *s, int bit_rate, int short_trai
     s->distances[0] = 0;
     s->trellis_ptr = 14;
 
-    span_log(&s->logging, SPAN_LOG_FLOW, "Phase rates %f %f\n", dds_frequencyf(s->carrier_phase_rate), dds_frequencyf(s->carrier_phase_rate_save));
     s->carrier_phase = 0;
     power_meter_init(&(s->power), 4);
 
     if (s->short_train)
     {
         s->carrier_phase_rate = s->carrier_phase_rate_save;
-        s->agc_scaling = s->agc_scaling_save;
         equalizer_restore(s);
+        s->agc_scaling = s->agc_scaling_save;
         /* Don't allow any frequency correction at all, until we start to pull the phase in. */
 #if defined(SPANDSP_USE_FIXED_POINTx)
         s->carrier_track_i = 0;
@@ -1366,18 +1358,19 @@ SPAN_DECLARE(int) v17_rx_restart(v17_rx_state_t *s, int bit_rate, int short_trai
         s->carrier_phase_rate = dds_phase_ratef(CARRIER_NOMINAL_FREQ);
         equalizer_reset(s);
 #if defined(SPANDSP_USE_FIXED_POINTx)
-        s->agc_scaling_save = 0;
+        //s->agc_scaling_save = 0;
         s->agc_scaling = (float) FP_FACTOR*32768.0f*0.0017f/RX_PULSESHAPER_GAIN;
         s->carrier_track_i = 5000;
         s->carrier_track_p = 40000;
 #else
-        s->agc_scaling_save = 0.0f;
+        //s->agc_scaling_save = 0.0f;
         s->agc_scaling = 0.0017f/RX_PULSESHAPER_GAIN;
         s->carrier_track_i = 5000.0f;
         s->carrier_track_p = 40000.0f;
 #endif
     }
     s->last_sample = 0;
+    span_log(&s->logging, SPAN_LOG_FLOW, "Phase rates %f %f\n", dds_frequencyf(s->carrier_phase_rate), dds_frequencyf(s->carrier_phase_rate_save));
 
     /* Initialise the working data for symbol timing synchronisation */
 #if defined(SPANDSP_USE_FIXED_POINTx)
@@ -1432,8 +1425,13 @@ SPAN_DECLARE(v17_rx_state_t *) v17_rx_init(v17_rx_state_t *s, int bit_rate, put_
     s->short_train = FALSE;
     //s->scrambler_tap = 18 - 1;
     v17_rx_signal_cutoff(s, -45.5f);
-    s->agc_scaling = 0.0017f/RX_PULSESHAPER_GAIN;
+#if defined(SPANDSP_USE_FIXED_POINTx)
+    s->agc_scaling_save = 0;
+    s->agc_scaling = (float) FP_FACTOR*32768.0f*0.0017f/RX_PULSESHAPER_GAIN;
+#else
     s->agc_scaling_save = 0.0f;
+    s->agc_scaling = 0.0017f/RX_PULSESHAPER_GAIN;
+#endif
     s->carrier_phase_rate_save = dds_phase_ratef(CARRIER_NOMINAL_FREQ);
     v17_rx_restart(s, bit_rate, s->short_train);
     return s;

libs/spandsp/tests/oki_adpcm_tests.c

@@ -22,8 +22,6 @@
  * You should have received a copy of the GNU General Public License
  * along with this program; if not, write to the Free Software
  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- *
- * $Id: oki_adpcm_tests.c,v 1.37 2009/05/30 15:23:14 steveu Exp $
  */
 
 /*! \file */
@@ -85,24 +83,33 @@ int main(int argc, char *argv[])
     int hist_out;
     oki_adpcm_state_t *oki_enc_state;
     oki_adpcm_state_t *oki_dec_state;
+    oki_adpcm_state_t *oki_dec_state2;
     int xx;
     int total_pre_samples;
     int total_compressed_bytes;
     int total_post_samples;
+    int successive_08_bytes;
+    int successive_80_bytes;
+    int encoded_fd;
+    const char *encoded_file_name;
     const char *in_file_name;
     int log_encoded_data;
     int opt;
 
     bit_rate = 32000;
+    encoded_file_name = NULL;
     in_file_name = IN_FILE_NAME;
     log_encoded_data = FALSE;
-    while ((opt = getopt(argc, argv, "2i:l")) != -1)
+    while ((opt = getopt(argc, argv, "2d:i:l")) != -1)
     {
         switch (opt)
         {
         case '2':
             bit_rate = 24000;
             break;
+        case 'd':
+            encoded_file_name = optarg;
+            break;
         case 'i':
             in_file_name = optarg;
             break;
@@ -116,24 +123,39 @@ int main(int argc, char *argv[])
         }
     }
 
-    if ((inhandle = sf_open_telephony_read(in_file_name, 1)) == NULL)
+    if (encoded_file_name)
     {
-        fprintf(stderr, "    Cannot open audio file '%s'\n", in_file_name);
-        exit(2);
+        if ((encoded_fd = open(encoded_file_name, O_RDONLY)) < 0)
+        {
+            fprintf(stderr, "    Cannot open encoded file '%s'\n", encoded_file_name);
+            exit(2);
+        }
+    }
+    else
+    {
+        if ((inhandle = sf_open_telephony_read(in_file_name, 1)) == NULL)
+        {
+            fprintf(stderr, "    Cannot open audio file '%s'\n", in_file_name);
+            exit(2);
+        }
+        if ((oki_enc_state = oki_adpcm_init(NULL, bit_rate)) == NULL)
+        {
+            fprintf(stderr, "    Cannot create encoder\n");
+            exit(2);
+        }
     }
+
     if ((outhandle = sf_open_telephony_write(OUT_FILE_NAME, 1)) == NULL)
     {
         fprintf(stderr, "    Cannot create audio file '%s'\n", OUT_FILE_NAME);
         exit(2);
     }
-
-    if ((oki_enc_state = oki_adpcm_init(NULL, bit_rate)) == NULL)
+    if ((oki_dec_state = oki_adpcm_init(NULL, bit_rate)) == NULL)
     {
-        fprintf(stderr, "    Cannot create encoder\n");
+        fprintf(stderr, "    Cannot create decoder\n");
         exit(2);
     }
-        
-    if ((oki_dec_state = oki_adpcm_init(NULL, bit_rate)) == NULL)
+    if ((oki_dec_state2 = oki_adpcm_init(NULL, bit_rate)) == NULL)
     {
         fprintf(stderr, "    Cannot create decoder\n");
         exit(2);
@@ -151,71 +173,124 @@ int main(int argc, char *argv[])
     total_pre_samples = 0;
     total_compressed_bytes = 0;
     total_post_samples = 0;
-    while ((frames = sf_readf_short(inhandle, pre_amp, 159)))
+    if (encoded_file_name)
     {
-        total_pre_samples += frames;
-        oki_bytes = oki_adpcm_encode(oki_enc_state, oki_data, pre_amp, frames);
-        if (log_encoded_data)
-            write(1, oki_data, oki_bytes);
-        total_compressed_bytes += oki_bytes;
-        dec_frames = oki_adpcm_decode(oki_dec_state, post_amp, oki_data, oki_bytes);
-        total_post_samples += dec_frames;
-        for (i = 0;  i < frames;  i++)
-        {
-            history[hist_in++] = pre_amp[i];
-            if (hist_in >= HIST_LEN)
-                hist_in = 0;
-            pre_energy += (double) pre_amp[i] * (double) pre_amp[i];
-        }
-        for (i = 0;  i < dec_frames;  i++)
+        /* Decode a file of OKI ADPCM code to a linear wave file */
+        while ((oki_bytes = read(encoded_fd, oki_data, 80)) > 0)
         {
-            post_energy += (double) post_amp[i] * (double) post_amp[i];
-            xx = post_amp[i] - history[hist_out++];
-            if (hist_out >= HIST_LEN)
-                hist_out = 0;
-            diff_energy += (double) xx * (double) xx;
-            //post_amp[i] = xx;
+            total_compressed_bytes += oki_bytes;
+            dec_frames = oki_adpcm_decode(oki_dec_state, post_amp, oki_data, oki_bytes);
+            total_post_samples += dec_frames;
+            for (i = 0;  i < dec_frames;  i++)
+            {
+                post_energy += (double) post_amp[i] * (double) post_amp[i];
+                xx = post_amp[i] - history[hist_out++];
+                if (hist_out >= HIST_LEN)
+                    hist_out = 0;
+                diff_energy += (double) xx * (double) xx;
+            }
+            outframes = sf_writef_short(outhandle, post_amp, dec_frames);
         }
-        outframes = sf_writef_short(outhandle, post_amp, dec_frames);
-    }
-    if (sf_close(inhandle) != 0)
-    {
-        fprintf(stderr, "    Cannot close audio file '%s'\n", in_file_name);
-        exit(2);
+        close(encoded_fd);
     }
-    if (sf_close(outhandle) != 0)
+    else
     {
-        fprintf(stderr, "    Cannot close audio file '%s'\n", OUT_FILE_NAME);
-        exit(2);
-    }
-    oki_adpcm_release(oki_enc_state);
-    oki_adpcm_release(oki_dec_state);
+        /* Perform a linear wave file -> OKI ADPCM -> linear wave file cycle. Along the way
+           check the decoder resets on the sequence specified for this codec, and the gain
+           and worst case sample distortion. */
+        successive_08_bytes = 0;
+        successive_80_bytes = 0;
+        while ((frames = sf_readf_short(inhandle, pre_amp, 159)))
+        {
+            total_pre_samples += frames;
+            oki_bytes = oki_adpcm_encode(oki_enc_state, oki_data, pre_amp, frames);
+            if (log_encoded_data)
+                write(1, oki_data, oki_bytes);
+            total_compressed_bytes += oki_bytes;
+            /* Look for a condition which is defined as something which should cause a reset of
+               the decoder (48 samples of 0, 8, 0, 8, etc.), and verify that it really does. Use
+               a second decode, which we feed byte by byte, for this. */
+            for (i = 0;  i < oki_bytes;  i++)
+            {
+                oki_adpcm_decode(oki_dec_state2, post_amp, &oki_data[i], 1);
+                if (oki_data[i] == 0x08)
+                    successive_08_bytes++;
+                else
+                    successive_08_bytes = 0;
+                if (oki_data[i] == 0x80)
+                    successive_80_bytes++;
+                else
+                    successive_80_bytes = 0;
+                if (successive_08_bytes == 24  ||  successive_80_bytes == 24)
+                {
+                    if (oki_dec_state2->step_index != 0)
+                    {
+                        fprintf(stderr, "Decoder reset failure\n");
+                        exit(2);
+                    }
+                }
+            }
+            dec_frames = oki_adpcm_decode(oki_dec_state, post_amp, oki_data, oki_bytes);
+            total_post_samples += dec_frames;
+            for (i = 0;  i < frames;  i++)
+            {
+                history[hist_in++] = pre_amp[i];
+                if (hist_in >= HIST_LEN)
+                    hist_in = 0;
+                pre_energy += (double) pre_amp[i] * (double) pre_amp[i];
+            }
+            for (i = 0;  i < dec_frames;  i++)
+            {
+                post_energy += (double) post_amp[i] * (double) post_amp[i];
+                xx = post_amp[i] - history[hist_out++];
+                if (hist_out >= HIST_LEN)
+                    hist_out = 0;
+                diff_energy += (double) xx * (double) xx;
+                //post_amp[i] = xx;
+            }
+            outframes = sf_writef_short(outhandle, post_amp, dec_frames);
+        }
+        printf("Pre samples: %d\n", total_pre_samples);
+        printf("Compressed bytes: %d\n", total_compressed_bytes);
+        printf("Post samples: %d\n", total_post_samples);
+
+        printf("Output energy is %f%% of input energy.\n", 100.0*post_energy/pre_energy);
+        printf("Residual energy is %f%% of the total.\n", 100.0*diff_energy/post_energy);
+        if (bit_rate == 32000)
+        {
+            if (fabs(1.0 - post_energy/pre_energy) > 0.01
+                ||
+                fabs(diff_energy/post_energy) > 0.01)
+            {
+                printf("Tests failed.\n");
+                exit(2);
+            }
+        }
+        else
+        {
+            if (fabs(1.0 - post_energy/pre_energy) > 0.11
+                ||
+                fabs(diff_energy/post_energy) > 0.05)
+            {
+                printf("Tests failed.\n");
+                exit(2);
+            }
+        }
 
-    printf("Pre samples: %d\n", total_pre_samples);
-    printf("Compressed bytes: %d\n", total_compressed_bytes);
-    printf("Post samples: %d\n", total_post_samples);
 
-    printf("Output energy is %f%% of input energy.\n", 100.0*post_energy/pre_energy);
-    printf("Residual energy is %f%% of the total.\n", 100.0*diff_energy/post_energy);
-    if (bit_rate == 32000)
-    {
-        if (fabs(1.0 - post_energy/pre_energy) > 0.05
-            ||
-            fabs(diff_energy/post_energy) > 0.03)
+        oki_adpcm_release(oki_enc_state);
+        if (sf_close(inhandle) != 0)
         {
-            printf("Tests failed.\n");
+            fprintf(stderr, "    Cannot close audio file '%s'\n", in_file_name);
             exit(2);
         }
     }
-    else
+    oki_adpcm_release(oki_dec_state);
+    oki_adpcm_release(oki_dec_state2);
+    if (sf_close(outhandle) != 0)
     {
-        if (fabs(1.0 - post_energy/pre_energy) > 0.20
-            ||
-            fabs(diff_energy/post_energy) > 0.10)
-        {
-            printf("Tests failed.\n");
-            exit(2);
-        }
+        fprintf(stderr, "    Cannot close audio file '%s'\n", OUT_FILE_NAME);
+        exit(2);
     }
 
     printf("Tests passed.\n");