Replace limiter with dynamic range compressor

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@22832 a1c6a512-1295-4272-9138-f99709370657

21 files changed, 568 insertions, 498 deletions

diff --git a/apps/dsp.c b/apps/dsp.c
index e7a6a9182a..a5ceecb048 100644
--- a/apps/dsp.c
+++ b/apps/dsp.c
@@ -137,6 +137,15 @@ struct eq_state
                                 /* 10ch */
 };
 
+struct compressor_menu
+{
+    int  threshold;     /* dB - from menu */
+    int  ratio;         /* from menu */
+    int  gain;          /* dB - from menu */
+    bool soft_knee;     /* 0 = hard knee, 1 = soft knee */     
+    int  release;       /* samples - from menu */
+};
+
 /* Include header with defines which functions are implemented in assembly
    code for the target */
 #include <dsp_asm.h>
@@ -171,7 +180,6 @@ struct dsp_config
     int32_t  tdspeed_percent; /* Speed% * PITCH_SPEED_PRECISION */
     bool tdspeed_active;  /* Timestretch is in use */
     int  frac_bits;
-    long limiter_preamp;     /* limiter amp gain in S7.24 format */
 #ifdef HAVE_SW_TONE_CONTROLS
     /* Filter struct for software bass/treble controls */
     struct eqfilter tone_filter;
@@ -187,7 +195,7 @@ struct dsp_config
     channels_process_fn_type     apply_crossfeed;
     channels_process_fn_type     eq_process;
     channels_process_fn_type     channels_process;
-    return_fn_type               limiter_process;
+    return_fn_type               compressor_process;
 };
 
 /* General DSP config */
@@ -253,58 +261,17 @@ static int32_t *resample_buf;
 #define RESAMPLE_BUF_LEFT_CHANNEL 0
 #define RESAMPLE_BUF_RIGHT_CHANNEL (sample_buf_count/2 * RESAMPLE_RATIO)
 
-/* limiter */
-/* MAX_COUNT is largest possible sample count in limiter_process.  This is
-   needed in case time stretch makes the count in dsp_process larger than
-   the limiter buffer. */
-#define MAX_COUNT  MAX(SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO / 2, LIMITER_BUFFER_SIZE)
-static int      count_adjust;
-static bool     limiter_buffer_active;
-static bool     limiter_buffer_full;
-static bool     limiter_buffer_emptying;
-static int32_t  limiter_buffer[2][LIMITER_BUFFER_SIZE] IBSS_ATTR;
-static int32_t  *start_lim_buf[2] IBSS_ATTR,
-                *end_lim_buf[2] IBSS_ATTR;
-static uint16_t lim_buf_peak[LIMITER_BUFFER_SIZE] IBSS_ATTR;
-static uint16_t *start_peak IBSS_ATTR,
-                *end_peak IBSS_ATTR;
-static uint16_t out_buf_peak[MAX_COUNT];
-static uint16_t *out_buf_peak_index IBSS_ATTR;
-static uint16_t release_peak IBSS_ATTR;
-static int32_t  in_samp IBSS_ATTR,
-                samp0 IBSS_ATTR;
-
-static void     reset_limiter_buffer(struct dsp_config *dsp);
-static int      limiter_buffer_count(bool buf_count);
-static int      limiter_process(int count, int32_t *buf[]);
-static uint16_t get_peak_value(int32_t sample);
-
- /* The clip_steps array essentially stores the results of fp_factor from
- * 0 to 12 dB, in 48 equal steps, in S3.28 format. */
-const  long     clip_steps[49] ICONST_ATTR = {      0x10000000,
-    0x10779AFA, 0x10F2B409, 0x1171654C, 0x11F3C9A0, 0x1279FCAD,
-    0x13041AE9, 0x139241A2, 0x14248EF9, 0x14BB21F9, 0x15561A92,
-    0x15F599A0, 0x1699C0F9, 0x1742B36B, 0x17F094CE, 0x18A38A01,
-    0x195BB8F9, 0x1A1948C5, 0x1ADC619B, 0x1BA52CDC, 0x1C73D51D,
-    0x1D488632, 0x1E236D3A, 0x1F04B8A1, 0x1FEC982C, 0x20DB3D0E,
-    0x21D0D9E2, 0x22CDA2BE, 0x23D1CD41, 0x24DD9099, 0x25F12590,
-    0x270CC693, 0x2830AFD3, 0x295D1F37, 0x2A925471, 0x2BD0911F,
-    0x2D1818B3, 0x2E6930AD, 0x2FC42095, 0x312931EC, 0x3298B072,
-    0x3412EA24, 0x35982F3A, 0x3728D22E, 0x38C52808, 0x3A6D8847,
-    0x3C224CD9, 0x3DE3D264, 0x3FB2783F};
-/* The gain_steps array essentially stores the results of fp_factor from
- * 0 to -12 dB, in 48 equal steps, in S3.28 format. */
-const  long     gain_steps[49] ICONST_ATTR = { 0x10000000,
-    0xF8BC9C0, 0xF1ADF94, 0xEAD2988, 0xE429058, 0xDDAFD68,
-    0xD765AC1, 0xD149309, 0xCB59186, 0xC594210, 0xBFF9112,
-    0xBA86B88, 0xB53BEF5, 0xB017965, 0xAB18964, 0xA63DDFE,
-    0xA1866BA, 0x9CF1397, 0x987D507, 0x9429BEE, 0x8FF599E,
-    0x8BDFFD3, 0x87E80B0, 0x840CEBE, 0x804DCE8, 0x7CA9E76,
-    0x792070E, 0x75B0AB0, 0x7259DB2, 0x6F1B4BF, 0x6BF44D5,
-    0x68E4342, 0x65EA5A0, 0x63061D6, 0x6036E15, 0x5D7C0D3,
-    0x5AD50CE, 0x5841505, 0x55C04B8, 0x535176A, 0x50F44D9,
-    0x4EA84FE, 0x4C6D00E, 0x4A41E78, 0x48268DF, 0x461A81C,
-    0x441D53E, 0x422E985, 0x404DE62};
+/* compressor */
+/* MAX_COUNT is largest possible sample count in compressor_process */
+#define MAX_COUNT (SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO / 2)
+static struct   compressor_menu c_menu;
+static int32_t  comp_rel_slope IBSS_ATTR;   /* S7.24 format */
+static int32_t  comp_makeup_gain IBSS_ATTR; /* S7.24 format */
+static int32_t  comp_curve[65] IBSS_ATTR;   /* S7.24 format */
+static int32_t  gain_buffer[MAX_COUNT] IBSS_ATTR;
+static int32_t  release_gain IBSS_ATTR;
+
+static int      compressor_process(int count, int32_t *buf[]);
 
 
 /* Clip sample to signed 16 bit range */
@@ -944,13 +911,6 @@ static void set_gain(struct dsp_config *dsp)
         dsp->data.gain = fp_mul(dsp->data.gain, eq_precut, 24);
     }
 
-    /* only preamp for the limiter if limiter is active and sample depth
-     * allows safe pre-amping (12 dB is OK with 29 or less frac bits) */
-    if ((dsp->limiter_preamp) && (dsp->frac_bits <= 29))
-    {
-        dsp->data.gain = fp_mul(dsp->data.gain, dsp->limiter_preamp, 24);
-    }
-
 #ifdef HAVE_SW_VOLUME_CONTROL
     if (global_settings.volume < SW_VOLUME_MAX ||
         global_settings.volume > SW_VOLUME_MIN)
@@ -1308,8 +1268,8 @@ int dsp_process(struct dsp_config *dsp, char *dst, const char *src[], int count)
             if (dsp->channels_process)
                 dsp->channels_process(chunk, t2);
             
-            if (dsp->limiter_process)
-                chunk = dsp->limiter_process(chunk, t2);
+            if (dsp->compressor_process)
+                chunk = dsp->compressor_process(chunk, t2);
 
             dsp->output_samples(chunk, &dsp->data, (const int32_t **)t2, (int16_t *)dst);
 
@@ -1358,15 +1318,6 @@ int dsp_output_count(struct dsp_config *dsp, int count)
     if (count > RESAMPLE_BUF_RIGHT_CHANNEL)
         count = RESAMPLE_BUF_RIGHT_CHANNEL;
         
-    /* If the limiter buffer is filling, some or all samples will
-     * be captured by it, so expect fewer samples coming out. */
-    if (limiter_buffer_active && !limiter_buffer_full)
-    {
-        int empty_space = limiter_buffer_count(false);
-        count_adjust = MIN(empty_space, count);
-        count -= count_adjust;
-    }
-
     return count;
 }
 
@@ -1375,13 +1326,6 @@ int dsp_output_count(struct dsp_config *dsp, int count)
  */
 int dsp_input_count(struct dsp_config *dsp, int count)
 {
-    /* If the limiter buffer is filling, the output count was
-     * adjusted downward.  This adjusts it back so that input
-     * count is not affected.
-     */
-    if (limiter_buffer_active && !limiter_buffer_full)
-        count += count_adjust;
-
     /* count is now the number of resampled input samples. Convert to
        original input samples. */
     if (dsp->resample)
@@ -1499,7 +1443,8 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value)
         dsp_update_functions(dsp);
         resampler_new_delta(dsp);
         tdspeed_setup(dsp);
-        reset_limiter_buffer(dsp);
+        if (dsp == &AUDIO_DSP)
+            release_gain = (1 << 24);
         break;
 
     case DSP_FLUSH:
@@ -1508,7 +1453,8 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value)
         resampler_new_delta(dsp);
         dither_init(dsp);
         tdspeed_setup(dsp);
-        reset_limiter_buffer(dsp);
+        if (dsp == &AUDIO_DSP)
+            release_gain = (1 << 24);
         break;
 
     case DSP_SET_TRACK_GAIN:
@@ -1588,369 +1534,257 @@ void dsp_set_replaygain(void)
     set_gain(&AUDIO_DSP);
 }
 
-/** RESET THE LIMITER BUFFER
- *  Force the limiter buffer to its initial state and discard
- *  any samples held there. */
-static void reset_limiter_buffer(struct dsp_config *dsp)
-{
-    if (dsp == &AUDIO_DSP)
+/** SET COMPRESSOR
+ *  Called by the menu system to configure the compressor process */
+void dsp_set_compressor(int c_threshold, int c_ratio, int c_gain,
+                        int c_knee, int c_release)
+{
+    bool changed = false;
+    bool active = (c_threshold < 0);
+    const int comp_ratio[] = {2, 4, 6, 10, 0};
+    int  new_ratio = comp_ratio[c_ratio];
+    bool new_knee = (c_knee == 1);
+    int  new_release = c_release * NATIVE_FREQUENCY / 1000;
+    
+    if (c_menu.threshold != c_threshold)
     {
-        int i;
-        logf("   reset_limiter_buffer");
-        for (i = 0; i < 2; i++)
-            start_lim_buf[i] = end_lim_buf[i] = limiter_buffer[i];
-        start_peak = end_peak = lim_buf_peak;
-        limiter_buffer_full = false;
-        limiter_buffer_emptying = false;
-        release_peak = 0;
+        changed = true;
+        c_menu.threshold = c_threshold;
+        logf("   Compressor Threshold: %d dB\tEnabled: %s",
+            c_menu.threshold, active ? "Yes" : "No");
     }
-}
 
-/** OPERATE THE LIMITER BUFFER
- *  Handle all samples entering or exiting the limiter buffer. */
-static inline int set_limiter_buffer(int count, int32_t *buf[])
-{
-    int32_t *in_buf[]  = {buf[0], buf[1]},
-            *out_buf[] = {buf[0], buf[1]};
-    int empty_space, i, out_count;
-    const long clip_max = AUDIO_DSP.data.clip_max;
-    const int ch = AUDIO_DSP.data.num_channels - 1;
-    out_buf_peak_index = out_buf_peak;
-
-    if (limiter_buffer_emptying)
-    /** EMPTY THE BUFFER
-     *  since the empty flag has been set, assume no inbound samples and
-        return all samples in the limiter buffer to the outbound buffer */
+    if (c_menu.ratio != new_ratio)
     {
-        count = limiter_buffer_count(true);
-        out_count = count;
-        logf("   Emptying limiter buffer: %d", count);
-        while (count-- > 0)
-        {
-            for (i = 0; i <= ch; i++)
-            {
-                /* move samples in limiter buffer to output buffer */
-                *out_buf[i]++ = *start_lim_buf[i]++;
-                if (start_lim_buf[i] == &limiter_buffer[i][LIMITER_BUFFER_SIZE])
-                    start_lim_buf[i] = limiter_buffer[i];
-                /* move limiter buffer peak values to output peak values */
-                if (i == 0)
-                {
-                    *out_buf_peak_index++ = *start_peak++;
-                    if (start_peak == &lim_buf_peak[LIMITER_BUFFER_SIZE])
-                        start_peak = lim_buf_peak;
-                }
-            }
-        }
-        limiter_buffer_full = false;
-        limiter_buffer_emptying = false;
+        changed = true;
+        c_menu.ratio = new_ratio;
+        if (c_menu.ratio)
+            logf("   Compressor Ratio: %d:1", c_menu.ratio);
+        else
+            logf("   Compressor Ratio: Limit");
     }
-    else    /* limiter buffer NOT emptying */
+    
+    if (c_menu.gain != c_gain)
+    {
+        changed = true;
+        c_menu.gain = c_gain;
+        if (c_menu.gain >= 0)
+            logf("   Compressor Makeup Gain: %d dB", c_menu.gain);
+        else
+            logf("   Compressor Makeup Gain: Auto");
+    }
+    
+    if (c_menu.soft_knee != new_knee)
     {
-        if (count <= 0) return 0;
+        changed = true;
+        c_menu.soft_knee = new_knee;
+        logf("   Compressor Knee: %s", c_menu.soft_knee==1?"Soft":"Hard");
+    }
+    
+    if (c_menu.release != new_release)
+    {
+        changed = true;
+        c_menu.release = new_release;
+        logf("   Compressor Release: %d", c_menu.release);
+    }
+
+    if (changed && active)
+    {
+        /* configure variables for compressor operation */
+        int i;
+        const int32_t db[] ={0x000000,   /* positive db equivalents in S15.16 format */
+         0x241FA4, 0x1E1A5E, 0x1A94C8, 0x181518, 0x1624EA, 0x148F82, 0x1338BD, 0x120FD2,
+         0x1109EB, 0x101FA4, 0x0F4BB6, 0x0E8A3C, 0x0DD840, 0x0D3377, 0x0C9A0E, 0x0C0A8C,
+         0x0B83BE, 0x0B04A5, 0x0A8C6C, 0x0A1A5E, 0x09ADE1, 0x094670, 0x08E398, 0x0884F6,
+         0x082A30, 0x07D2FA, 0x077F0F, 0x072E31, 0x06E02A, 0x0694C8, 0x064BDF, 0x060546,
+         0x05C0DA, 0x057E78, 0x053E03, 0x04FF5F, 0x04C273, 0x048726, 0x044D64, 0x041518,
+         0x03DE30, 0x03A89B, 0x037448, 0x03412A, 0x030F32, 0x02DE52, 0x02AE80, 0x027FB0,
+         0x0251D6, 0x0224EA, 0x01F8E2, 0x01CDB4, 0x01A359, 0x0179C9, 0x0150FC, 0x0128EB,
+         0x010190, 0x00DAE4, 0x00B4E1, 0x008F82, 0x006AC1, 0x004699, 0x002305};
         
-        empty_space = limiter_buffer_count(false);
+        struct curve_point
+        {
+            int32_t db;     /* S15.16 format */
+            int32_t offset; /* S15.16 format */
+        } db_curve[4];
         
-        if (empty_space > 0)
-        /** FILL BUFFER
-         *  use as many inbound samples as necessary to fill the buffer */
+        /** Set up the shape of the compression curve first as decibel values*/
+        /* db_curve[0] = bottom of knee
+                   [1] = threshold
+                   [2] = top of knee
+                   [3] = 0 db input */
+        db_curve[1].db = c_menu.threshold << 16;
+        db_curve[1].offset = 0;
+        if (c_menu.soft_knee)
         {
-            /* don't try to fill with more samples than available */
-            if (empty_space > count)
-                empty_space = count;
-            logf("   Filling limiter buffer: %d", empty_space);
-            while (empty_space-- > 0)
-            {
-                for (i = 0; i <= ch; i++)
-                {
-                    /* put inbound samples in the limiter buffer */
-                    in_samp = *in_buf[i]++;
-                    *end_lim_buf[i]++ = in_samp;
-                    if (end_lim_buf[i] == &limiter_buffer[i][LIMITER_BUFFER_SIZE])
-                        end_lim_buf[i] = limiter_buffer[i];
-                    if (in_samp < 0)       /* make positive for comparison */
-                        in_samp = -in_samp - 1;
-                    if (in_samp <= clip_max)
-                        in_samp = 0;       /* disregard if not clipped */
-                    if (i == 0)
-                        samp0 = in_samp;
-                    if (i == ch)
-                    {
-                        /* assign peak value for each inbound sample pair */
-                        *end_peak++ = ((samp0 > 0) || (in_samp > 0)) ?
-                            get_peak_value(MAX(samp0, in_samp)) : 0;
-                        if (end_peak == &lim_buf_peak[LIMITER_BUFFER_SIZE])
-                            end_peak = lim_buf_peak;
-                    }
-                }
-                count--;
-            }
-            /* after buffer fills, the remaining inbound samples are cycled */
+            /* bottom of knee is 3dB below the threshold for soft knee*/
+            db_curve[0].db = db_curve[1].db - (3 << 16);
+            db_curve[0].offset = 0;
+            /* top of knee is 3dB above the threshold for soft knee */
+            db_curve[2].db = db_curve[1].db + (3 << 16);
+            if (c_menu.ratio)
+                /* offset = -3db * (ratio - 1) / ratio */
+                db_curve[2].offset = (int32_t)((long long)(-3 << 16)
+                    * (c_menu.ratio - 1) / c_menu.ratio);
+            else
+                /* offset = -3db for hard limit */
+                db_curve[2].offset = (-3 << 16);
         }
-        
-        limiter_buffer_full = (end_lim_buf[0] == start_lim_buf[0]);
-        out_count = count;
-        
-        /** CYCLE BUFFER
-         *  return buffered samples and backfill limiter buffer with new ones.
-         *  The buffer is always full when cycling. */
-        while (count-- > 0)
+        else
         {
-            for (i = 0; i <= ch; i++)
-            {
-                /* copy incoming sample */
-                in_samp = *in_buf[i]++;
-                /* put limiter buffer sample into outbound buffer */
-                *out_buf[i]++ = *start_lim_buf[i]++;
-                /* put incoming sample on the end of the limiter buffer */
-                *end_lim_buf[i]++ = in_samp;
-                /* ring buffer pointer wrap */
-                if (start_lim_buf[i] == &limiter_buffer[i][LIMITER_BUFFER_SIZE])
-                    start_lim_buf[i] = limiter_buffer[i];
-                if (end_lim_buf[i] == &limiter_buffer[i][LIMITER_BUFFER_SIZE])
-                    end_lim_buf[i] = limiter_buffer[i];
-                if (in_samp < 0)       /* make positive for comparison */
-                    in_samp = -in_samp - 1;
-                if (in_samp <= clip_max)
-                    in_samp = 0;       /* disregard if not clipped */
-                if (i == 0)
-                {
-                    samp0 = in_samp;
-                    /* assign outgoing sample its associated peak value */
-                    *out_buf_peak_index++ = *start_peak++;
-                    if (start_peak == &lim_buf_peak[LIMITER_BUFFER_SIZE])
-                        start_peak = lim_buf_peak;
-                }
-                if (i == ch)
-                {
-                    /* assign peak value for each inbound sample pair */
-                    *end_peak++ = ((samp0 > 0) || (in_samp > 0)) ?
-                        get_peak_value(MAX(samp0, in_samp)) : 0;
-                    if (end_peak == &lim_buf_peak[LIMITER_BUFFER_SIZE])
-                        end_peak = lim_buf_peak;
-                }
-            }
+            /* bottom of knee is at the threshold for hard knee */
+            db_curve[0].db = c_menu.threshold << 16;
+            db_curve[0].offset = 0;
+            /* top of knee is at the threshold for hard knee */
+            db_curve[2].db = c_menu.threshold << 16;
+            db_curve[2].offset = 0;
         }
-    }   
-    
-    return out_count;
-}
-
-/** RETURN LIMITER BUFFER COUNT
- *  If argument is true, returns number of samples in the buffer,
- *  otherwise, returns empty space remaining */
-static int limiter_buffer_count(bool buf_count)
-{
-    int count;
-    if (limiter_buffer_full)
-        count = LIMITER_BUFFER_SIZE;
-    else if (end_lim_buf[0] >= start_lim_buf[0])
-        count = (end_lim_buf[0] - start_lim_buf[0]);
-    else
-        count = (end_lim_buf[0] - start_lim_buf[0]) + LIMITER_BUFFER_SIZE;
-    return buf_count ? count : (LIMITER_BUFFER_SIZE - count);
-}
-
-/** FLUSH THE LIMITER BUFFER
- *  Empties the limiter buffer into the buffer pointed to by the argument
- *  and returns the number of samples in that buffer */
-int dsp_flush_limiter_buffer(char *dest)
-{
-    if ((!limiter_buffer_active) || (limiter_buffer_count(true) <= 0))
-        return 0;
-    
-    logf("   dsp_flush_limiter_buffer");
-    int32_t flush_buf[2][LIMITER_BUFFER_SIZE];
-    int32_t *src[2] = {flush_buf[0], flush_buf[1]};
-
-    limiter_buffer_emptying = true;
-    int count = limiter_process(0, src);
-    AUDIO_DSP.output_samples(count, &AUDIO_DSP.data,
-        (const int32_t **)src, (int16_t *)dest);
-    return count;
-}
-
-/** GET PEAK VALUE
- *  Return a small value representing how much the sample is clipped.  This
- *  should only be called if a sample is actually clipped.  Sample is a
- *  positive value.
- */
-static uint16_t get_peak_value(int32_t sample)
-{
-    const int frac_bits = AUDIO_DSP.frac_bits;
-    int mid,
-        hi = 48,
-        lo = 0;
-    
-    /* shift sample into 28 frac bit range for comparison */
-    if (frac_bits > 28)
-        sample >>= (frac_bits - 28);
-    if (frac_bits < 28)
-        sample <<= (28 - frac_bits);
-    
-    /* if clipped out of range, return maximum value */
-    if (sample >= clip_steps[48])
-        return 48 * 90;
-    
-    /* find amount of sample clipping on the table */
-    do
-    {
-        mid = (hi + lo) / 2;
-        if (sample < clip_steps[mid])
-            hi = mid;
-        else if (sample > clip_steps[mid])
-            lo = mid;
+        /* 0db input is also max offset point (most compression) */
+        db_curve[3].db = 0;
+        if (c_menu.ratio)
+            /* offset = threshold * (ratio - 1) / ratio */
+            db_curve[3].offset = (int32_t)((long long)(c_menu.threshold << 16)
+                * (c_menu.ratio - 1) / c_menu.ratio);
         else
-            return mid * 90;
-    }
-    while (hi > (lo + 1));
-    
-    /* interpolate linearly between steps (less accurate but faster) */
-    return ((hi-1) * 90) + (((sample - clip_steps[hi-1]) * 90) /
-        (clip_steps[hi] - clip_steps[hi-1]));
-}
-
-/** SET LIMITER
- *  Called by the menu system to configure the limiter process */
-void dsp_set_limiter(int limiter_level)
-{
-    if (limiter_level > 0)
-    {
-        if (!limiter_buffer_active)
+            /* offset = threshold for hard limit */
+            db_curve[3].offset = (c_menu.threshold << 16);
+        
+        /* Now set up the comp_curve table with compression offsets in the form
+           of gain factors in S7.24 format */
+        comp_curve[0]  = (1 << 24);
+        for (i = 1; i < 64; i++)
         {
-            /* enable limiter process */
-            AUDIO_DSP.limiter_process = limiter_process;
-            limiter_buffer_active = true;
+            int32_t this_db = -db[i];
+            /* no compression below the knee */
+            if (this_db <= db_curve[0].db)
+                comp_curve[i] = (1 << 24);
+            
+            /* if soft knee and below top of knee, interpolate along soft knee slope */
+            else if (c_menu.soft_knee && (this_db <= db_curve[2].db))
+                comp_curve[i] = fp_factor(fp_mul(((this_db - db_curve[0].db) / 6),
+                    db_curve[2].offset, 16), 16) << 8;
+            
+            /* interpolate along ratio slope above the knee */
+            else
+                comp_curve[i] = fp_factor(fp_mul(fp_div((this_db - db_curve[1].db),
+                    -db_curve[1].db, 16), db_curve[3].offset, 16), 16) << 8;
         }
-        /* limiter preamp is a gain factor in S7.24 format */
-        long old_preamp = AUDIO_DSP.limiter_preamp;
-        long new_preamp = fp_factor((((long)limiter_level << 24) / 10), 24);
-        if (old_preamp != new_preamp)
+        comp_curve[64] = fp_factor(db_curve[3].offset, 16) << 8;
+
+        logf("\n   *** Compression Offsets ***");
+        for (i = 0; i <= 3; i++)
         {
-            AUDIO_DSP.limiter_preamp = new_preamp;
-            set_gain(&AUDIO_DSP);
-            logf("   Limiter enable: Yes\tLimiter amp: %.8f",
-                (float)AUDIO_DSP.limiter_preamp / (1 << 24));
+            logf("Curve[%d]: db: % .1f\toffset: % .4f", i, (float)db_curve[i].db / (1 << 16),
+                (float)db_curve[i].offset / (1 << 16));
         }
-    }
-    else
-    {
-        /* disable limiter process*/
-        if (limiter_buffer_active)
+        
+        logf("\nGain factors:");
+        for (i = 1; i <= 64; i++)
         {
-            AUDIO_DSP.limiter_preamp = (1 << 24);
-            set_gain(&AUDIO_DSP);
-            /* pcmbuf_flush_limiter_buffer(); */
-            limiter_buffer_active = false;
-            AUDIO_DSP.limiter_process = NULL;
-            reset_limiter_buffer(&AUDIO_DSP);
-            logf("   Limiter enable:  No\tLimiter amp: %.8f",
-                (float)AUDIO_DSP.limiter_preamp / (1 << 24));
+            debugf("%02d: %.6f  ", i, (float)comp_curve[i] / (1 << 24));
+            if (i % 4 == 0) debugf("\n");
         }
+        
+        /* if using auto peak, then makeup gain is max offset - .1dB headroom */
+        int32_t db_makeup = (c_menu.gain == -1) ?
+            -(db_curve[3].offset) - 0x199A : c_menu.gain << 16;
+        comp_makeup_gain = fp_factor(db_makeup, 16) << 8;
+        logf("Makeup gain:\t%.6f", (float)comp_makeup_gain / (1 << 24));
+
+        /* calculate per-sample gain change a rate of 10db over release time */
+        comp_rel_slope = 0xAF0BB2 / c_menu.release;
+        logf("Release slope:\t%.6f", (float)comp_rel_slope / (1 << 24));
+        
+        release_gain = (1 << 24);
     }
+    
+    /* enable/disable the compressor */
+    AUDIO_DSP.compressor_process = active ? compressor_process : NULL;
 }
 
-/** LIMITER PROCESS
- *  Checks pre-amplified signal for clipped samples and smoothly reduces gain
- *  around the clipped samples using a preset attack/release schedule.
+/** GET COMPRESSION GAIN
+ *  Returns the required gain factor in S7.24 format in order to compress the
+ *  sample in accordance with the compression curve.  Always 1 or less.
  */
-static int limiter_process(int count, int32_t *buf[])
+static inline int32_t get_compression_gain(int32_t sample)
 {
-    /* Limiter process passes through if limiter buffer isn't active, or the 
-     * sample depth is too large for safe pre-amping */
-    if ((!limiter_buffer_active) || (AUDIO_DSP.frac_bits > 29))
-        return count;
+    const int frac_bits = AUDIO_DSP.frac_bits;
     
-    count = set_limiter_buffer(count, buf);
+    /* sample must be positive */
+    if (sample < 0)
+        sample = -sample - 1;
+
+    /* shift sample into 22 frac bit range */
+    if (frac_bits > 22)
+        sample >>= (frac_bits - 22);
+    if (frac_bits < 22)
+        sample <<= (22 - frac_bits);
     
-    if (count <= 0)
-        return 0;
+    /* index is 6 MSB, rem is 16 LSB */
+    int index = sample >> 16;
+    int rem = (sample & 0xFFFF) << 8;
     
-    const int attack_slope = 15;    /* 15:1 ratio between attack and release */
-    const int buffer_count = limiter_buffer_count(true);
+    /* interpolate from the compression curve */
+    return comp_curve[index] + (int32_t)FRACMUL_SHL((comp_curve[index + 1] 
+        - comp_curve[index]), rem, 7);
+}
+
+/** COMPRESSOR PROCESS
+ *  Changes the gain of the samples according to the compressor curve
+ */
+static int compressor_process(int count, int32_t *buf[])
+{
+    const int num_chan = AUDIO_DSP.data.num_channels;
+    const int32_t fp_one = (1 << 24);
     
+    int32_t sample_gain,      /* S7.24 format */
+            this_gain;        /* S7.24 format */
     int i, ch;
-    uint16_t max_peak = 0,
-             gain_peak,
-             gain_rem;
-    long gain;
     
-    /* step through limiter buffer in reverse order, in order to find the
-     * appropriate max_peak for modifying the output buffer */
-    for (i = buffer_count - 1; i >= 0; i--)
-    {
-        const uint16_t peak_i = lim_buf_peak[(start_peak - lim_buf_peak + i) %
-            LIMITER_BUFFER_SIZE];
-        /* if no attack slope, nothing to do */
-        if ((peak_i == 0) && (max_peak == 0)) continue;
-        /* if new peak, start attack slope */
-        if (peak_i >= max_peak)
-        {
-            max_peak = peak_i;
-        }
-        /* keep sloping */
-        else
-        {
-            if (max_peak > attack_slope)
-                max_peak -= attack_slope;
-            else
-                max_peak = 0;
-        }
-    }
-    /* step through output buffer the same way, but this time modifying peak
-     * values to create a smooth attack slope. */
-    for (i = count - 1; i >= 0; i--)
-    {
-        /* if no attack slope, nothing to do */
-        if ((out_buf_peak[i] == 0) && (max_peak == 0)) continue;
-        /* if new peak, start attack slope */
-        if (out_buf_peak[i] >= max_peak)
-        {
-            max_peak = out_buf_peak[i];
-        }
-        /* keep sloping */
-        else
-        {
-            if (max_peak > attack_slope)
-                max_peak -= attack_slope;
-            else
-                max_peak = 0;
-            out_buf_peak[i] = max_peak;
-        }
-    }
-    /* Now step forward through the output buffer, and modify the peak values
+    /* Step forward through the output buffer, and modify the offset values
      * to establish a smooth, slow release slope.*/
     for (i = 0; i < count; i++)
     {
-        /* if no release slope, nothing to do */
-        if ((out_buf_peak[i] == 0) && (release_peak == 0)) continue;
-        /* if new peak, start release slope */
-        if (out_buf_peak[i] >= release_peak)
+        sample_gain = fp_one;
+        for (ch = 0; ch < num_chan; ch++)
         {
-            release_peak = out_buf_peak[i];
+            this_gain = get_compression_gain(buf[ch][i]);
+            if (this_gain < sample_gain)
+                sample_gain = this_gain;
         }
-        /* keep sloping */
+        /* if no release slope, only apply makeup gain */
+        if ((sample_gain == fp_one) && (release_gain == fp_one))
+            gain_buffer[i] = comp_makeup_gain;
         else
         {
-            release_peak--;
-            out_buf_peak[i] = release_peak;
+            /* if larger offset, start release slope */
+            if (sample_gain <= release_gain)
+                release_gain = sample_gain;
+            else    /* keep sloping */
+            {
+                if (release_gain < (fp_one - comp_rel_slope))
+                    release_gain += comp_rel_slope;
+                else
+                    release_gain = fp_one;
+            }
+            /* store offset with release and also apply makeup gain */
+            if ((release_gain == fp_one) && (comp_makeup_gain == fp_one))
+                gain_buffer[i] = fp_one;
+            else
+                gain_buffer[i] = FRACMUL_SHL(release_gain, comp_makeup_gain, 7);
         }
     }
-    /* Implement the limiter: adjust gain of the outbound samples by the gain
-     * amounts in the gain steps array corresponding to the peak values. */
+    
+    /* Implement the compressor: apply those gain factors to the output
+     * buffer samples */
+
     for (i = 0; i < count; i++)
     {
-        if (out_buf_peak[i] > 0)
+        if (gain_buffer[i] != fp_one)
         {
-            gain_peak = (out_buf_peak[i] + 1) / 90;
-            gain_rem  = (out_buf_peak[i] + 1) % 90;
-            gain = gain_steps[gain_peak];
-            if ((gain_peak < 48) && (gain_rem > 0))
-                gain -= gain_rem * ((gain_steps[gain_peak] -
-                    gain_steps[gain_peak + 1]) / 90);
-            for (ch = 0; ch < AUDIO_DSP.data.num_channels; ch++)
-                buf[ch][i] = FRACMUL_SHL(buf[ch][i], gain, 3);
+            for (ch = 0; ch < num_chan; ch++)
+                buf[ch][i] = FRACMUL_SHL(buf[ch][i], gain_buffer[i], 7);
         }
     }
     return count;
-}   
+}
diff --git a/apps/dsp.h b/apps/dsp.h
index b2d8493445..ab42d73b24 100644
--- a/apps/dsp.h
+++ b/apps/dsp.h
@@ -26,8 +26,7 @@
 #include <stdbool.h>
 
 #define NATIVE_FREQUENCY       44100
-#define LIMITER_BUFFER_SIZE      288    /* ~6.5 ms */
-#define MAX_LIMITER_GAIN          80    /*    8 dB */
+
 enum
 {
     STEREO_INTERLEAVED = 0,
@@ -82,7 +81,7 @@ int32_t sound_get_pitch(void);
 void dsp_set_timestretch(int32_t percent);
 int32_t dsp_get_timestretch(void);
 int dsp_callback(int msg, intptr_t param);
-int dsp_flush_limiter_buffer(char *dest);
-void dsp_set_limiter(int limiter_level);
+void dsp_set_compressor(int c_threshold, int c_ratio, int c_gain,
+                        int c_knee, int c_release);
 
 #endif
diff --git a/apps/fixedpoint.h b/apps/fixedpoint.h
index fd68fc162c..6670e597fb 100644
--- a/apps/fixedpoint.h
+++ b/apps/fixedpoint.h
@@ -61,7 +61,7 @@ long fp_sincos(unsigned long phase, long *cos);
 #define FP_NEGINF      -(0x7fffffff)
 
 /* fracbits in range 12 - 22 work well. Higher is better for
- * calculating dB, lower is better for calculating ratio.
+ * calculating dB, lower is better for calculating factor.
  */
 /* long fp_decibels(unsigned long factor, unsigned int fracbits); */
 long fp_factor(long decibels, unsigned int fracbits);
diff --git a/apps/lang/dansk.lang b/apps/lang/dansk.lang
index 1c005fb9ac..e4393e7dd3 100644
--- a/apps/lang/dansk.lang
+++ b/apps/lang/dansk.lang
@@ -12646,15 +12646,15 @@
   user: core
   <source>
     *: none
-    swcodec: "Limiter Preamp"
+    swcodec: "Compressor"
   </source>
   <dest>
     *: none
-    swcodec: "Begrænser forforstærkning"
+    swcodec: "Compressor"
   </dest>
   <voice>
     *: none
-    swcodec: "Begrænser for-forstærkning"
+    swcodec: "Compressor"
   </voice>
 </phrase>
 <phrase>
diff --git a/apps/lang/deutsch.lang b/apps/lang/deutsch.lang
index bdacebaa27..e7f78f800e 100644
--- a/apps/lang/deutsch.lang
+++ b/apps/lang/deutsch.lang
@@ -12641,15 +12641,15 @@
   user: core
   <source>
     *: none
-    swcodec: "Limiter Preamp"
+    swcodec: "Compressor"
   </source>
   <dest>
     *: none
-    swcodec: "Begrenzer-Vorverstärkung"
+    swcodec: "Compressor"
   </dest>
   <voice>
     *: none
-    swcodec: "Begrenzer-Vorverstärkung"
+    swcodec: "Compressor"
   </voice>
 </phrase>
 <phrase>
diff --git a/apps/lang/english.lang b/apps/lang/english.lang
index 08412d376d..72e0a4ce90 100644
--- a/apps/lang/english.lang
+++ b/apps/lang/english.lang
@@ -12707,15 +12707,15 @@
   user: core
   <source>
     *: none
-    swcodec: "Limiter Preamp"
+    swcodec: "Compressor"
   </source>
   <dest>
     *: none
-    swcodec: "Limiter Preamp"
+    swcodec: "Compressor"
   </dest>
   <voice>
     *: none
-    swcodec: "Limiter Preamp"
+    swcodec: "Compressor"
   </voice>
 </phrase>
 <phrase>
@@ -12769,3 +12769,224 @@
     radio: "Other"
   </voice>
 </phrase>
+<phrase>
+  id: LANG_COMPRESSOR_THRESHOLD
+  desc: in sound settings
+  user: core
+  <source>
+    *: none
+    swcodec: "Threshold"
+  </source>
+  <dest>
+    *: none
+    swcodec: "Threshold"
+  </dest>
+  <voice>
+    *: none
+    swcodec: "Threshold"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_COMPRESSOR_RATIO
+  desc: in sound settings
+  user: core
+  <source>
+    *: none
+    swcodec: "Ratio"
+  </source>
+  <dest>
+    *: none
+    swcodec: "Ratio"
+  </dest>
+  <voice>
+    *: none
+    swcodec: "Ratio"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_COMPRESSOR_RATIO_2
+  desc: in sound settings
+  user: core
+  <source>
+    *: none
+    swcodec: "2:1"
+  </source>
+  <dest>
+    *: none
+    swcodec: "2:1"
+  </dest>
+  <voice>
+    *: none
+    swcodec: "2 to 1"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_COMPRESSOR_RATIO_4
+  desc: in sound settings
+  user: core
+  <source>
+    *: none
+    swcodec: "4:1"
+  </source>
+  <dest>
+    *: none
+    swcodec: "4:1"
+  </dest>
+  <voice>
+    *: none
+    swcodec: "4 to 1"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_COMPRESSOR_RATIO_6
+  desc: in sound settings
+  user: core
+  <source>
+    *: none
+    swcodec: "6:1"
+  </source>
+  <dest>
+    *: none
+    swcodec: "6:1"
+  </dest>
+  <voice>
+    *: none
+    swcodec: "6 to 1"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_COMPRESSOR_RATIO_10
+  desc: in sound settings
+  user: core
+  <source>
+    *: none
+    swcodec: "10:1"
+  </source>
+  <dest>
+    *: none
+    swcodec: "10:1"
+  </dest>
+  <voice>
+    *: none
+    swcodec: "10 to 1"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_COMPRESSOR_RATIO_LIMIT
+  desc: in sound settings
+  user: core
+  <source>
+    *: none
+    swcodec: "Limit"
+  </source>
+  <dest>
+    *: none
+    swcodec: "Limit"
+  </dest>
+  <voice>
+    *: none
+    swcodec: "Limit"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_COMPRESSOR_GAIN
+  desc: in sound settings
+  user: core
+  <source>
+    *: none
+    swcodec: "Makeup Gain"
+  </source>
+  <dest>
+    *: none
+    swcodec: "Makeup Gain"
+  </dest>
+  <voice>
+    *: none
+    swcodec: "Makeup Gain"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_AUTO
+  desc: in sound settings
+  user: core
+  <source>
+    *: none
+    swcodec: "Auto"
+  </source>
+  <dest>
+    *: none
+    swcodec: "Auto"
+  </dest>
+  <voice>
+    *: none
+    swcodec: "Auto"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_COMPRESSOR_KNEE
+  desc: in sound settings
+  user: core
+  <source>
+    *: none
+    swcodec: "Knee"
+  </source>
+  <dest>
+    *: none
+    swcodec: "Knee"
+  </dest>
+  <voice>
+    *: none
+    swcodec: "Knee"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_COMPRESSOR_HARD_KNEE
+  desc: in sound settings
+  user: core
+  <source>
+    *: none
+    swcodec: "Hard Knee"
+  </source>
+  <dest>
+    *: none
+    swcodec: "Hard Knee"
+  </dest>
+  <voice>
+    *: none
+    swcodec: "Hard Knee"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_COMPRESSOR_SOFT_KNEE
+  desc: in sound settings
+  user: core
+  <source>
+    *: none
+    swcodec: "Soft Knee"
+  </source>
+  <dest>
+    *: none
+    swcodec: "Soft Knee"
+  </dest>
+  <voice>
+    *: none
+    swcodec: "Soft Knee"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_COMPRESSOR_RELEASE
+  desc: in sound settings
+  user: core
+  <source>
+    *: none
+    swcodec: "Release Time"
+  </source>
+  <dest>
+    *: none
+    swcodec: "Release Time"
+  </dest>
+  <voice>
+    *: none
+    swcodec: "Release Time"
+  </voice>
+</phrase>
diff --git a/apps/lang/francais.lang b/apps/lang/francais.lang
index 43fe7561f2..7d4af8ade7 100644
--- a/apps/lang/francais.lang
+++ b/apps/lang/francais.lang
@@ -12663,15 +12663,15 @@
   user: core
   <source>
     *: none
-    swcodec: "Limiter Preamp"
+    swcodec: "Compressor"
   </source>
   <dest>
     *: none
-    swcodec: "Limiteur préampli."
+    swcodec: "Compressor"
   </dest>
   <voice>
     *: none
-    swcodec: "Limiteur préampli"
+    swcodec: "Compressor"
   </voice>
 </phrase>
 <phrase>
diff --git a/apps/lang/italiano.lang b/apps/lang/italiano.lang
index 075520d2af..0727320c45 100644
--- a/apps/lang/italiano.lang
+++ b/apps/lang/italiano.lang
@@ -12636,15 +12636,15 @@
   user: core
   <source>
     *: none
-    swcodec: "Limiter Preamp"
+    swcodec: "Compressor"
   </source>
   <dest>
     *: none
-    swcodec: "Limitatore Preamp"
+    swcodec: "Compressor"
   </dest>
   <voice>
     *: none
-    swcodec: "Limitatore Preamp"
+    swcodec: "Compressor"
   </voice>
 </phrase>
 <phrase>
diff --git a/apps/lang/polski.lang b/apps/lang/polski.lang
index 5595526145..86017a4998 100644
--- a/apps/lang/polski.lang
+++ b/apps/lang/polski.lang
@@ -12644,15 +12644,15 @@
   user: core
   <source>
     *: none
-    swcodec: "Limiter Preamp"
+    swcodec: "Compressor"
   </source>
   <dest>
     *: none
-    swcodec: "Wzmacnianie cichych fragmentów"
+    swcodec: "Compressor"
   </dest>
   <voice>
     *: none
-    swcodec: "Wzmacnianie cichych fragmentów"
+    swcodec: "Compressor"
   </voice>
 </phrase>
 <phrase>
diff --git a/apps/lang/svenska.lang b/apps/lang/svenska.lang
index 7a1564bded..24c33e7d0b 100644
--- a/apps/lang/svenska.lang
+++ b/apps/lang/svenska.lang
@@ -12637,15 +12637,15 @@
   user: core
   <source>
     *: none
-    swcodec: "Limiter Preamp"
+    swcodec: "Compressor"
   </source>
   <dest>
     *: none
-    swcodec: "Limiter-förförstärkning"
+    swcodec: "Compressor"
   </dest>
   <voice>
     *: none
-    swcodec: "Limiter-förförstärkning"
+    swcodec: "Compressor"
   </voice>
 </phrase>
 <phrase>
diff --git a/apps/lang/tagalog.lang b/apps/lang/tagalog.lang
index 667a9d8c28..4600eab9f4 100644
--- a/apps/lang/tagalog.lang
+++ b/apps/lang/tagalog.lang
@@ -12445,15 +12445,15 @@
   user: core
   <source>
     *: none
-    swcodec: "Limiter Preamp"
+    swcodec: "Compressor"
   </source>
   <dest>
     *: none
-    swcodec: "Limiter Preamp"
+    swcodec: "Compressor"
   </dest>
   <voice>
     *: none
-    swcodec: "Limiter Preamp"
+    swcodec: "Compressor"
   </voice>
 </phrase>
 <phrase>
diff --git a/apps/lang/walon.lang b/apps/lang/walon.lang
index c645ad10a2..c7a06ff55a 100644
--- a/apps/lang/walon.lang
+++ b/apps/lang/walon.lang
@@ -12632,19 +12632,19 @@
   </voice>
 </phrase>
 <phrase>
-  id: LANG_LIMITER
+  id: LANG_COMPRESSOR
   desc: in sound settings
   user: core
   <source>
     *: none
-    swcodec: "Limiter Preamp"
+    swcodec: "Compressor"
   </source>
   <dest>
     *: none
-    swcodec: "Aschatrece Preamp"
+    swcodec: "Compressor"
   </dest>
   <voice>
     *: none
-    swcodec: "Aschatrece Preamp"
+    swcodec: "Compressor"
   </voice>
 </phrase>
diff --git a/apps/menus/sound_menu.c b/apps/menus/sound_menu.c
index 678d495d07..60315804a9 100644
--- a/apps/menus/sound_menu.c
+++ b/apps/menus/sound_menu.c
@@ -105,8 +105,21 @@ static int timestretch_callback(int action,const struct menu_item_ex *this_item)
                      &global_settings.timestretch_enabled, timestretch_callback);
     MENUITEM_SETTING(dithering_enabled,
                      &global_settings.dithering_enabled, lowlatency_callback);
-    MENUITEM_SETTING(limiter_level,
-                     &global_settings.limiter_level, lowlatency_callback);
+
+    /* compressor submenu */
+    MENUITEM_SETTING(compressor_threshold,
+                     &global_settings.compressor_threshold, lowlatency_callback);
+    MENUITEM_SETTING(compressor_ratio,
+                     &global_settings.compressor_ratio, lowlatency_callback);
+    MENUITEM_SETTING(compressor_gain,
+                     &global_settings.compressor_makeup_gain, lowlatency_callback);
+    MENUITEM_SETTING(compressor_knee,
+                     &global_settings.compressor_knee, lowlatency_callback);
+    MENUITEM_SETTING(compressor_release,
+                     &global_settings.compressor_release_time, lowlatency_callback);
+    MAKE_MENU(compressor_menu,ID2P(LANG_COMPRESSOR), NULL, Icon_NOICON,
+              &compressor_threshold, &compressor_ratio, &compressor_gain,
+              &compressor_knee, &compressor_release);
 #endif
 
 #if (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F)
@@ -140,7 +153,7 @@ MAKE_MENU(sound_settings, ID2P(LANG_SOUND_SETTINGS), NULL, Icon_Audio,
 #if CONFIG_CODEC == SWCODEC
           ,&crossfeed_menu, &equalizer_menu, &dithering_enabled
           ,&timestretch_enabled
-          ,&limiter_level
+          ,&compressor_menu
 #endif
 #if (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F)
          ,&loudness,&avc,&superbass,&mdb_enable,&mdb_strength
diff --git a/apps/pcmbuf.c b/apps/pcmbuf.c
index 66a4ed4128..319e3e8044 100644
--- a/apps/pcmbuf.c
+++ b/apps/pcmbuf.c
@@ -1170,30 +1170,6 @@ bool pcmbuf_is_crossfade_enabled(void)
  *  Commit any remaining samples in the PCM buffer for playback. */
 void pcmbuf_play_remainder(void)
 {
-    pcmbuf_flush_limiter_buffer();
-    
     if (audiobuffer_fillpos)
         pcmbuf_flush_fillpos();
 }
-
-/** FLUSH LIMITER BUFFER
- *  Empty the limiter buffer and commit its contents
- *  to the PCM buffer for playback. */
-void pcmbuf_flush_limiter_buffer(void)
-{
-    char *dest;
-    int out_count = LIMITER_BUFFER_SIZE;
-    
-    /* create room at the end of the PCM buffer for any
-       samples that may be held back in the limiter buffer */
-    while ((dest = pcmbuf_request_buffer(&out_count)) == NULL)
-    {
-        cancel_cpu_boost();
-        sleep(1);
-    }
-
-    /* flush the limiter buffer into the PCM buffer */
-    out_count = dsp_flush_limiter_buffer(dest);
-    if (out_count > 0)
-        pcmbuf_write_complete(out_count);
-}
diff --git a/apps/pcmbuf.h b/apps/pcmbuf.h
index 12a41a253c..4cb1c1acb7 100644
--- a/apps/pcmbuf.h
+++ b/apps/pcmbuf.h
@@ -76,6 +76,5 @@ void pcmbuf_beep(unsigned int frequency, size_t duration, int amplitude);
 int pcmbuf_used_descs(void);
 int pcmbuf_descs(void);
 void pcmbuf_play_remainder(void);
-void pcmbuf_flush_limiter_buffer(void);
 
 #endif
diff --git a/apps/plugin.c b/apps/plugin.c
index a497ad0456..aa7077edae 100644
--- a/apps/plugin.c
+++ b/apps/plugin.c
@@ -467,7 +467,6 @@ static const struct plugin_api rockbox_api = {
     dsp_process,
     dsp_input_count,
     dsp_output_count,
-    dsp_flush_limiter_buffer,
 #endif /* CONFIG_CODEC == SWCODEC */
 
     /* playback control */
diff --git a/apps/plugin.h b/apps/plugin.h
index 4981230df3..b4a6b4fc1b 100644
--- a/apps/plugin.h
+++ b/apps/plugin.h
@@ -133,12 +133,12 @@ void* plugin_get_buffer(size_t *buffer_size);
 #define PLUGIN_MAGIC 0x526F634B /* RocK */
 
 /* increase this every time the api struct changes */
-#define PLUGIN_API_VERSION 171
+#define PLUGIN_API_VERSION 172
 
 /* update this to latest version if a change to the api struct breaks
    backwards compatibility (and please take the opportunity to sort in any
    new function which are "waiting" at the end of the function table) */
-#define PLUGIN_MIN_API_VERSION 171
+#define PLUGIN_MIN_API_VERSION 172
 
 /* plugin return codes */
 enum plugin_status {
@@ -596,7 +596,6 @@ struct plugin_api {
                        const char *src[], int count);
     int (*dsp_input_count)(struct dsp_config *dsp, int count);
     int (*dsp_output_count)(struct dsp_config *dsp, int count);
-    int (*dsp_flush_limiter_buffer)(char *dest);
 #endif /* CONFIG_CODEC == SWCODC */
 
     /* playback control */
diff --git a/apps/plugins/test_codec.c b/apps/plugins/test_codec.c
index b77611689b..b41f053571 100644
--- a/apps/plugins/test_codec.c
+++ b/apps/plugins/test_codec.c
@@ -810,34 +810,10 @@ show_menu:
             
             rb->closedir(dir);
         }
-        /* process last samples */
-        if (use_dsp)
-            rb->dsp_flush_limiter_buffer(dspbuffer);
     } else {
         /* Just test the file */
         res = test_track(parameter);
 
-        /* process last samples */
-        if (use_dsp)
-        {
-            int channels = (wavinfo.stereomode == STEREO_MONO) ? 1 : 2;
-            int count = rb->dsp_flush_limiter_buffer(dspbuffer);
-            if (channels == 1)
-            {
-                unsigned char *s = dspbuffer, *d = dspbuffer;
-                int c = count;
-                while (c-- > 0)
-                {
-                    *d++ = *s++;
-                    *d++ = *s++;
-                    s++;
-                    s++;
-                }
-            }
-            if (wavinfo.fd >= 0)
-                rb->write(wavinfo.fd, dspbuffer, count * 2 * channels);
-        }
-
         /* Close WAV file (if there was one) */
         if (wavinfo.fd >= 0) {
             close_wav();
diff --git a/apps/settings.c b/apps/settings.c
index 4c16c6a0ec..ce1ee07054 100644
--- a/apps/settings.c
+++ b/apps/settings.c
@@ -956,7 +956,11 @@ void settings_apply(bool read_disk)
 
     dsp_dither_enable(global_settings.dithering_enabled);
     dsp_timestretch_enable(global_settings.timestretch_enabled);
-    dsp_set_limiter(global_settings.limiter_level);
+    dsp_set_compressor(global_settings.compressor_threshold,
+                       global_settings.compressor_ratio,
+                       global_settings.compressor_makeup_gain,
+                       global_settings.compressor_knee,
+                       global_settings.compressor_release_time);
 #endif
 
 #ifdef HAVE_SPDIF_POWER
diff --git a/apps/settings.h b/apps/settings.h
index e8ffe91471..ec92a81734 100644
--- a/apps/settings.h
+++ b/apps/settings.h
@@ -777,7 +777,11 @@ struct user_settings
 #endif
 
 #if CONFIG_CODEC == SWCODEC
-    int limiter_level;
+    int compressor_threshold;
+    int compressor_ratio;
+    int compressor_makeup_gain;
+    int compressor_knee;
+    int compressor_release_time;
 #endif
 
 };
diff --git a/apps/settings_list.c b/apps/settings_list.c
index 2a7bf9658f..2e3632b949 100644
--- a/apps/settings_list.c
+++ b/apps/settings_list.c
@@ -356,6 +356,34 @@ static void crossfeed_cross_set(int val)
                                   global_settings.crossfeed_hf_cutoff);
 }
 
+static void compressor_set(int val)
+{
+    (void)val;
+    dsp_set_compressor(global_settings.compressor_threshold,
+                       global_settings.compressor_ratio,
+                       global_settings.compressor_makeup_gain,
+                       global_settings.compressor_knee,
+                       global_settings.compressor_release_time);
+}
+
+static const char* auto_formatter(char *buffer, size_t buffer_size,
+                                int val, const char *unit)
+{
+    if (val == -1)
+        return str(LANG_AUTO);
+    else
+        snprintf(buffer, buffer_size, "%d %s", val, unit);
+    return buffer;
+}
+
+static int32_t auto_getlang(int value, int unit)
+{
+    if (value == -1)
+        return LANG_AUTO;
+    else
+        return TALK_ID(value, unit);
+}
+
 static const char* db_format(char* buffer, size_t buffer_size, int value,
                       const char* unit)
 {
@@ -1256,11 +1284,29 @@ const struct settings_list settings[] = {
     OFFON_SETTING(F_SOUNDSETTING, timestretch_enabled, LANG_TIMESTRETCH, false,
                   "timestretch enabled", dsp_timestretch_enable),
 
-    /* limiter */
-    INT_SETTING_NOWRAP(F_SOUNDSETTING, limiter_level,
-                       LANG_COMPRESSOR, 0,
-                       "limiter level", UNIT_DB, 0, MAX_LIMITER_GAIN,
-                       5, db_format, get_dec_talkid, dsp_set_limiter),
+    /* compressor */
+    INT_SETTING_NOWRAP(F_SOUNDSETTING, compressor_threshold,
+                       LANG_COMPRESSOR_THRESHOLD, 0,
+                       "compressor threshold", UNIT_DB, 0, -24,
+                       -3, formatter_unit_0_is_off, getlang_unit_0_is_off, compressor_set),
+    CHOICE_SETTING(F_SOUNDSETTING|F_NO_WRAP, compressor_ratio,
+                   LANG_COMPRESSOR_RATIO, 1, "compressor ratio",
+                   "2:1,4:1,6:1,10:1,limit", compressor_set, 5,
+                   ID2P(LANG_COMPRESSOR_RATIO_2), ID2P(LANG_COMPRESSOR_RATIO_4),
+                   ID2P(LANG_COMPRESSOR_RATIO_6), ID2P(LANG_COMPRESSOR_RATIO_10),
+                   ID2P(LANG_COMPRESSOR_RATIO_LIMIT)),
+    INT_SETTING_NOWRAP(F_SOUNDSETTING, compressor_makeup_gain,
+                       LANG_COMPRESSOR_GAIN, -1,
+                       "compressor makeup gain", UNIT_DB, -1, 20,
+                       1, auto_formatter, auto_getlang, compressor_set),
+    CHOICE_SETTING(F_SOUNDSETTING|F_NO_WRAP, compressor_knee,
+                   LANG_COMPRESSOR_KNEE, 1, "compressor knee",
+                   "hard knee,soft knee", compressor_set, 2,
+                   ID2P(LANG_COMPRESSOR_HARD_KNEE), ID2P(LANG_COMPRESSOR_SOFT_KNEE)),
+    INT_SETTING_NOWRAP(F_SOUNDSETTING, compressor_release_time,
+                       LANG_COMPRESSOR_RELEASE, 100,
+                       "compressor release time", UNIT_MS, 20, 200,
+                       10, NULL, NULL, compressor_set),
 #endif
 #ifdef HAVE_WM8758
     SOUND_SETTING(F_NO_WRAP, bass_cutoff, LANG_BASS_CUTOFF,