18 files changed, 181 insertions, 159 deletions

diff --git a/apps/codec_thread.c b/apps/codec_thread.c
index 39db741054..17ca980e41 100644
--- a/apps/codec_thread.c
+++ b/apps/codec_thread.c
@@ -213,49 +213,41 @@ void codec_thread_do_callback(void (*fn)(void), unsigned int *id)
 static void codec_pcmbuf_insert_callback(
         const void *ch1, const void *ch2, int count)
 {
-    const char *src[2] = { ch1, ch2 };
+    struct dsp_buffer src;
 
-    while (count > 0)
+    src.remcount  = count;
+    src.pin[0]    = ch1;
+    src.pin[1]    = ch2;
+    src.proc_mask = 0;
+
+    while (1)
     {
-        int out_count = dsp_output_count(ci.dsp, count);
-        int inp_count;
-        char *dest;
+        struct dsp_buffer dst;
+        dst.remcount = 0;
+        dst.bufcount = MAX(src.remcount, 1024); /* Arbitrary min request */
 
-        while (1)
+        while ((dst.p16out = pcmbuf_request_buffer(&dst.bufcount)) == NULL)
         {
-            if ((dest = pcmbuf_request_buffer(&out_count)) != NULL)
-                break;
-
             cancel_cpu_boost();
 
-            /* It will be awhile before space is available but we want
+            /* It may be awhile before space is available but we want
                "instant" response to any message */
             queue_wait_w_tmo(&codec_queue, NULL, HZ/20);
 
             if (!queue_empty(&codec_queue) &&
                 codec_check_queue__have_msg() < 0)
+            {
+                dsp_configure(ci.dsp, DSP_FLUSH, 0); /* Discontinuity */
                 return;
+            }
         }
 
-        /* Get the real input_size for output_size bytes, guarding
-         * against resampling buffer overflows. */
-        inp_count = dsp_input_count(ci.dsp, out_count);
-
-        if (inp_count <= 0)
-            return;
-
-        /* Input size has grown, no error, just don't write more than length */
-        if (inp_count > count)
-            inp_count = count;
+        dsp_process(ci.dsp, &src, &dst);
 
-        out_count = dsp_process(ci.dsp, dest, src, inp_count);
-
-        if (out_count <= 0)
-            return;
-
-        pcmbuf_write_complete(out_count, ci.id3->elapsed, ci.id3->offset);
-
-        count -= inp_count;
+        if (dst.remcount > 0)
+            pcmbuf_write_complete(dst.remcount, ci.id3->elapsed, ci.id3->offset);
+        else if (src.remcount <= 0)
+            break; /* No input remains and DSP purged */
     }
 }
 
@@ -352,10 +344,7 @@ static void codec_seek_complete_callback(void)
 
 static void codec_configure_callback(int setting, intptr_t value)
 {
-    if (!dsp_configure(ci.dsp, setting, value))
-    {
-        logf("Illegal key: %d", setting);
-    }
+    dsp_configure(ci.dsp, setting, value);
 }
 
 static enum codec_command_action
@@ -611,8 +600,7 @@ static void NORETURN_ATTR codec_thread(void)
 void codec_thread_init(void)
 {
     /* Init API */
-    ci.dsp              = (struct dsp_config *)dsp_configure(NULL, DSP_MYDSP,
-                                                             CODEC_IDX_AUDIO);
+    ci.dsp              = dsp_get_config(CODEC_IDX_AUDIO);
     ci.codec_get_buffer = codec_get_buffer_callback;
     ci.pcmbuf_insert    = codec_pcmbuf_insert_callback;
     ci.set_elapsed      = audio_codec_update_elapsed;
diff --git a/apps/codecs.c b/apps/codecs.c
index fafe4ac7a3..69204b7c4f 100644
--- a/apps/codecs.c
+++ b/apps/codecs.c
@@ -118,6 +118,7 @@ struct codec_api ci = {
 
     commit_dcache,
     commit_discard_dcache,
+    commit_discard_idcache,
 
     /* strings and memory */
     strcpy,
@@ -166,7 +167,6 @@ struct codec_api ci = {
     /* new stuff at the end, sort into place next time
        the API gets incompatible */
 
-    commit_discard_idcache,
 };
 
 void codec_get_full_path(char *path, const char *codec_root_fn)
diff --git a/apps/gui/bitmap/list-skinned.c b/apps/gui/bitmap/list-skinned.c
index 95430ae278..7d3620ed81 100644
--- a/apps/gui/bitmap/list-skinned.c
+++ b/apps/gui/bitmap/list-skinned.c
@@ -20,13 +20,13 @@
  ****************************************************************************/
 
 #include "config.h"
+#include "system.h"
 #include "lcd.h"
 #include "font.h"
 #include "button.h"
 #include "string.h"
 #include "settings.h"
 #include "kernel.h"
-#include "system.h"
 #include "file.h"
 
 #include "action.h"
diff --git a/apps/gui/bitmap/list.c b/apps/gui/bitmap/list.c
index 0581050091..49479c1cb3 100644
--- a/apps/gui/bitmap/list.c
+++ b/apps/gui/bitmap/list.c
@@ -22,13 +22,13 @@
 /* This file contains the code to draw the list widget on BITMAP LCDs. */
 
 #include "config.h"
+#include "system.h"
 #include "lcd.h"
 #include "font.h"
 #include "button.h"
 #include "string.h"
 #include "settings.h"
 #include "kernel.h"
-#include "system.h"
 #include "file.h"
 
 #include "action.h"
diff --git a/apps/gui/option_select.c b/apps/gui/option_select.c
index 7c3d34f024..ca206b86da 100644
--- a/apps/gui/option_select.c
+++ b/apps/gui/option_select.c
@@ -22,6 +22,7 @@
 #include <stdlib.h>
 #include "string-extra.h"
 #include "config.h"
+#include "system.h"
 #include "option_select.h"
 #include "kernel.h"
 #include "lang.h"
diff --git a/apps/main.c b/apps/main.c
index 631236852e..0fff0846a6 100644
--- a/apps/main.c
+++ b/apps/main.c
@@ -384,6 +384,9 @@ static void init(void)
     viewportmanager_init();
 
     storage_init();
+#if CONFIG_CODEC == SWCODEC
+    dsp_init();
+#endif
     settings_reset();
     settings_load(SETTINGS_ALL);
     settings_apply(true);
@@ -632,6 +635,10 @@ static void init(void)
         }
     }
 
+#if CONFIG_CODEC == SWCODEC
+    dsp_init();
+#endif
+
 #if defined(SETTINGS_RESET) || (CONFIG_KEYPAD == IPOD_4G_PAD) || \
     (CONFIG_KEYPAD == IRIVER_H10_PAD)
 #ifdef SETTINGS_RESET
diff --git a/apps/menus/eq_menu.c b/apps/menus/eq_menu.c
index 2cfb80f76a..60b2687050 100644
--- a/apps/menus/eq_menu.c
+++ b/apps/menus/eq_menu.c
@@ -70,13 +70,11 @@ const char* eq_precut_format(char* buffer, size_t buffer_size, int value, const
  */
 static void eq_apply(void)
 {
-    dsp_set_eq(global_settings.eq_enabled); 
+    dsp_eq_enable(global_settings.eq_enabled); 
     dsp_set_eq_precut(global_settings.eq_precut);    
     /* Update all bands */
-    for(int i = 0; i < 5; i++) {
-        dsp_set_eq_coefs(i, global_settings.eq_band_settings[i].cutoff,
-                         global_settings.eq_band_settings[i].q,
-                         global_settings.eq_band_settings[i].gain);
+    for(int i = 0; i < EQ_NUM_BANDS; i++) {
+        dsp_set_eq_coefs(i, &global_settings.eq_band_settings[i]);
     }
 }
 
@@ -580,9 +578,7 @@ bool eq_menu_graphical(void)
         /* Update the filter if the user changed something */
         if (has_changed) {
             dsp_set_eq_coefs(current_band,
-                             global_settings.eq_band_settings[current_band].cutoff,
-                             global_settings.eq_band_settings[current_band].q,
-                             global_settings.eq_band_settings[current_band].gain);
+                &global_settings.eq_band_settings[current_band]);
             has_changed = false;
         }
     }
diff --git a/apps/pcmbuf.c b/apps/pcmbuf.c
index d36883fc5b..9cedae0b67 100644
--- a/apps/pcmbuf.c
+++ b/apps/pcmbuf.c
@@ -47,16 +47,23 @@
    smaller math - must be < 65536 bytes */
 #define PCMBUF_CHUNK_SIZE    8192u
 
-/* Massive size is a nasty temp fix */
-#define PCMBUF_GUARD_SIZE    (1024u*12*((NATIVE_FREQUENCY+7999)/8000))
+/* Small guard buf to give decent space near end */
+#define PCMBUF_GUARD_SIZE    (PCMBUF_CHUNK_SIZE / 8)
 
 /* Mnemonics for common data commit thresholds */
 #define COMMIT_CHUNKS        PCMBUF_CHUNK_SIZE
 #define COMMIT_ALL_DATA      1u
 
- /* Size of the crossfade buffer where codec data is written to be faded
-    on commit */
-#define CROSSFADE_BUFSIZE    8192u
+/* Size of the crossfade buffer where codec data is written to be faded
+   on commit */
+#define CROSSFADE_BUFSIZE    PCMBUF_CHUNK_SIZE
+
+/* Maximum contiguous space that PCM buffer will allow (to avoid excessive
+   draining between inserts and observe low-latency mode) */
+#define PCMBUF_MAX_BUFFER    (PCMBUF_CHUNK_SIZE * 4)
+
+/* Forced buffer insert constraint can thus be from 1KB to 32KB using 8KB
+   chunks */
 
 /* Return data level in 1/4-second increments */
 #define DATA_LEVEL(quarter_secs) (NATIVE_FREQUENCY * (quarter_secs))
@@ -383,7 +390,11 @@ void * pcmbuf_request_buffer(int *count)
     /* If crossfade has begun, put the new track samples in crossfade_buffer */
     if (crossfade_status != CROSSFADE_INACTIVE && size > CROSSFADE_BUFSIZE)
         size = CROSSFADE_BUFSIZE;
-#endif
+    else
+#endif /* HAVE_CROSSFADE */
+
+    if (size > PCMBUF_MAX_BUFFER)
+        size = PCMBUF_MAX_BUFFER; /* constrain */
 
     enum channel_status status = mixer_channel_status(PCM_MIXER_CHAN_PLAYBACK);
     size_t remaining = pcmbuf_unplayed_bytes();
@@ -432,11 +443,22 @@ void * pcmbuf_request_buffer(int *count)
             pcmbuf_play_start();
     }
 
-    void *buf =
+    void *buf;
+
 #ifdef HAVE_CROSSFADE
-        crossfade_status != CROSSFADE_INACTIVE ? crossfade_buffer :
+    if (crossfade_status != CROSSFADE_INACTIVE)
+    {
+        buf = crossfade_buffer; /* always CROSSFADE_BUFSIZE */
+    }
+    else
 #endif
-        get_write_buffer(&size);
+    {
+        /* Give the maximum amount available if there's more */
+        if (size + PCMBUF_CHUNK_SIZE < freespace)
+            size = freespace - PCMBUF_CHUNK_SIZE;
+
+        buf = get_write_buffer(&size);
+    }
 
     *count = size / 4;
     return buf;
diff --git a/apps/plugin.c b/apps/plugin.c
index 5406610e23..129fd6954e 100644
--- a/apps/plugin.c
+++ b/apps/plugin.c
@@ -565,13 +565,15 @@ static const struct plugin_api rockbox_api = {
     audio_set_output_source,
     audio_set_input_source,
 #endif
-    dsp_set_crossfeed,
-    dsp_set_eq,
+    dsp_crossfeed_enable,
+    dsp_eq_enable,
     dsp_dither_enable,
+#ifdef HAVE_PITCHSCREEN
+    dsp_set_timestretch,
+#endif
     dsp_configure,
+    dsp_get_config,
     dsp_process,
-    dsp_input_count,
-    dsp_output_count,
 
     mixer_channel_status,
     mixer_channel_get_buffer,
@@ -584,7 +586,7 @@ static const struct plugin_api rockbox_api = {
 
     system_sound_play,
     keyclick_click,
-#endif
+#endif /* CONFIG_CODEC == SWCODEC */
     /* playback control */
     playlist_amount,
     playlist_resume,
diff --git a/apps/plugin.h b/apps/plugin.h
index 8123414ff0..3820c7ede6 100644
--- a/apps/plugin.h
+++ b/apps/plugin.h
@@ -153,12 +153,12 @@ 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 218
+#define PLUGIN_API_VERSION 219
 
 /* 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 218
+#define PLUGIN_MIN_API_VERSION 219
 
 /* plugin return codes */
 /* internal returns start at 0x100 to make exit(1..255) work */
@@ -680,15 +680,17 @@ struct plugin_api {
     void (*audio_set_output_source)(int monitor);
     void (*audio_set_input_source)(int source, unsigned flags);
 #endif
-    void (*dsp_set_crossfeed)(bool enable);
-    void (*dsp_set_eq)(bool enable);
+    void (*dsp_crossfeed_enable)(bool enable);
+    void (*dsp_eq_enable)(bool enable);
     void (*dsp_dither_enable)(bool enable);
-    intptr_t (*dsp_configure)(struct dsp_config *dsp, int setting,
-                              intptr_t value);
-    int (*dsp_process)(struct dsp_config *dsp, char *dest,
-                       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);
+#ifdef HAVE_PITCHSCREEN
+    void (*dsp_set_timestretch)(int32_t percent);
+#endif
+    intptr_t (*dsp_configure)(struct dsp_config *dsp,
+                              unsigned int setting, intptr_t value);
+    struct dsp_config * (*dsp_get_config)(enum dsp_ids id);
+    void (*dsp_process)(struct dsp_config *dsp, struct dsp_buffer *src,
+                        struct dsp_buffer *dst);
 
     enum channel_status (*mixer_channel_status)(enum pcm_mixer_channel channel);
     const void * (*mixer_channel_get_buffer)(enum pcm_mixer_channel channel,
diff --git a/apps/plugins/SOURCES b/apps/plugins/SOURCES
index db690a638a..e5f026c5b4 100644
--- a/apps/plugins/SOURCES
+++ b/apps/plugins/SOURCES
@@ -33,6 +33,7 @@ flipit.c
 shopper.c
 resistor.c
 
+test_codec.c
 
 #ifdef USB_ENABLE_HID
 remote_control.c
diff --git a/apps/plugins/mpegplayer/audio_thread.c b/apps/plugins/mpegplayer/audio_thread.c
index f976fd6007..b06727f759 100644
--- a/apps/plugins/mpegplayer/audio_thread.c
+++ b/apps/plugins/mpegplayer/audio_thread.c
@@ -36,6 +36,7 @@ struct audio_thread_data
     unsigned samplerate;    /* Current stream sample rate */
     int nchannels;          /* Number of audio channels */
     struct dsp_config *dsp; /* The DSP we're using */
+    struct dsp_buffer src;  /* Current audio data for DSP processing */
 };
 
 /* The audio thread is stolen from the core codec thread */
@@ -479,12 +480,13 @@ static void audio_thread(void)
     /* We need this here to init the EMAC for Coldfire targets */
     init_mad();
 
-    td.dsp = (struct dsp_config *)rb->dsp_configure(NULL, DSP_MYDSP,
-                                                    CODEC_IDX_AUDIO);
+    td.dsp = rb->dsp_get_config(CODEC_IDX_AUDIO);
 #ifdef HAVE_PITCHSCREEN
     rb->sound_set_pitch(PITCH_SPEED_100);
+    rb->dsp_set_timestretch(PITCH_SPEED_100);
 #endif
     rb->dsp_configure(td.dsp, DSP_RESET, 0);
+    rb->dsp_configure(td.dsp, DSP_FLUSH, 0);
     rb->dsp_configure(td.dsp, DSP_SET_SAMPLE_DEPTH, MAD_F_FRACBITS);
 
     goto message_wait;
@@ -631,43 +633,53 @@ static void audio_thread(void)
                                 STEREO_MONO : STEREO_NONINTERLEAVED);
         }
 
+        td.src.remcount  = synth.pcm.length;
+        td.src.pin[0]    = synth.pcm.samples[0];
+        td.src.pin[1]    = synth.pcm.samples[1];
+        td.src.proc_mask = 0;
+
         td.state  = TSTATE_RENDER_WAIT;
 
         /* Add a frame of audio to the pcm buffer. Maximum is 1152 samples. */
     render_wait:
-        if (synth.pcm.length > 0)
+        rb->yield();
+
+        while (1)
         {
-            const char *src[2] =
-                { (char *)synth.pcm.samples[0], (char *)synth.pcm.samples[1] };
-            int out_count = (synth.pcm.length * CLOCK_RATE
-                                + (td.samplerate - 1)) / td.samplerate;
-            unsigned char *out_buf;
-            ssize_t size = out_count*4;
+            struct dsp_buffer dst;
+            dst.remcount = 0;
+            dst.bufcount = MAX(td.src.remcount, 1024);
+
+            ssize_t size = dst.bufcount * 2 * sizeof(int16_t);
 
             /* Wait for required amount of free buffer space */
-            while ((out_buf = pcm_output_get_buffer(&size)) == NULL)
+            while ((dst.p16out = pcm_output_get_buffer(&size)) == NULL)
             {
                 /* Wait one frame */
-                int timeout = out_count*HZ / td.samplerate;
+                int timeout = dst.bufcount*HZ / td.samplerate;
                 str_get_msg_w_tmo(&audio_str, &td.ev, MAX(timeout, 1));
                 if (td.ev.id != SYS_TIMEOUT)
                     goto message_process;
             }
 
-            out_count = rb->dsp_process(td.dsp, out_buf, src, synth.pcm.length);
+            dst.bufcount = size / (2 * sizeof (int16_t));
+            rb->dsp_process(td.dsp, &td.src, &dst);
 
-            if (out_count <= 0)
-                break;
-
-            /* Make this data available to DMA */
-            pcm_output_commit_data(out_count*4, audio_queue.curr->time);
+            if (dst.remcount > 0)
+            {
+                /* Make this data available to DMA */
+                pcm_output_commit_data(dst.remcount * 2 * sizeof(int16_t),
+                                       audio_queue.curr->time);
 
-            /* As long as we're on this timestamp, the time is just
-               incremented by the number of samples */
-            audio_queue.curr->time += out_count;
+                /* As long as we're on this timestamp, the time is just
+                   incremented by the number of samples */
+                audio_queue.curr->time += dst.remcount;
+            }
+            else if (td.src.remcount <= 0)
+            {
+                break;
+            }
         }
-
-        rb->yield();
     } /* end decoding loop */
 }
 
diff --git a/apps/plugins/mpegplayer/mpeg_settings.c b/apps/plugins/mpegplayer/mpeg_settings.c
index 1c3f3c0b92..7f92fb7c69 100644
--- a/apps/plugins/mpegplayer/mpeg_settings.c
+++ b/apps/plugins/mpegplayer/mpeg_settings.c
@@ -457,13 +457,13 @@ static void sync_audio_setting(int setting, bool global)
         break;
 
     case MPEG_AUDIO_CROSSFEED:
-        rb->dsp_set_crossfeed((global || settings.crossfeed) ?
-                              rb->global_settings->crossfeed : false);
+        rb->dsp_crossfeed_enable((global || settings.crossfeed) ?
+                                 rb->global_settings->crossfeed : false);
         break;
 
     case MPEG_AUDIO_EQUALIZER:
-        rb->dsp_set_eq((global || settings.equalizer) ?
-                       rb->global_settings->eq_enabled : false);
+        rb->dsp_eq_enable((global || settings.equalizer) ?
+                          rb->global_settings->eq_enabled : false);
         break;
 
     case MPEG_AUDIO_DITHERING:
diff --git a/apps/plugins/test_codec.c b/apps/plugins/test_codec.c
index dafcf35710..920be54d56 100644
--- a/apps/plugins/test_codec.c
+++ b/apps/plugins/test_codec.c
@@ -164,6 +164,7 @@ static inline void int2le16(unsigned char* buf, int16_t x)
 
 static unsigned char *wavbuffer;
 static unsigned char *dspbuffer;
+static int dspbuffer_count;
 
 void init_wav(char* filename)
 {
@@ -215,34 +216,31 @@ static void* codec_get_buffer(size_t *size)
 
 static int process_dsp(const void *ch1, const void *ch2, int count)
 {
-    const char *src[2] = { ch1, ch2 };
-    int written_count = 0;
-    char *dest = dspbuffer;
-    
-    while (count > 0)
+    struct dsp_buffer src;
+    src.remcount = count;
+    src.pin[0] = ch1;
+    src.pin[1] = ch2;
+    src.proc_mask = 0;
+
+    struct dsp_buffer dst;
+    dst.remcount = 0;
+    dst.p16out = (int16_t *)dspbuffer;
+    dst.bufcount = dspbuffer_count;
+
+    while (1)
     {
-        int out_count = rb->dsp_output_count(ci.dsp, count);
-        
-        int inp_count = rb->dsp_input_count(ci.dsp, out_count);
-        
-        if (inp_count <= 0)
-            break;
-        
-        if (inp_count > count)
-            inp_count = count;
-        
-        out_count = rb->dsp_process(ci.dsp, dest, src, inp_count);
+        int old_remcount = dst.remcount;
+        rb->dsp_process(ci.dsp, &src, &dst);
         
-        if (out_count <= 0)
+        if (dst.bufcount <= 0 ||
+            (src.remcount <= 0 && dst.remcount <= old_remcount))
+        {
+            /* Dest is full or no input left and DSP purged */
             break;
-        
-        written_count += out_count;
-        dest += out_count * 4;
-        
-        count -= inp_count;
+        }
     }
     
-    return written_count;
+    return dst.remcount;
 }
 
 /* Null output */
@@ -502,7 +500,6 @@ static void configure(int setting, intptr_t value)
         rb->dsp_configure(ci.dsp, setting, value);
     switch(setting)
     {
-        case DSP_SWITCH_FREQUENCY:
         case DSP_SET_FREQUENCY:
             DEBUGF("samplerate=%d\n",(int)value);
             wavinfo.samplerate = use_dsp ? NATIVE_FREQUENCY : (int)value;
@@ -525,9 +522,7 @@ static void init_ci(void)
 {
     /* --- Our "fake" implementations of the codec API functions. --- */
 
-    ci.dsp = (struct dsp_config *)rb->dsp_configure(NULL, DSP_MYDSP,
-                                                    CODEC_IDX_AUDIO);
-
+    ci.dsp = rb->dsp_get_config(CODEC_IDX_AUDIO);
     ci.codec_get_buffer = codec_get_buffer;
 
     if (wavinfo.fd >= 0 || checksum) {
@@ -849,6 +844,8 @@ enum plugin_status plugin_start(const void* parameter)
 
     wavbuffer = rb->plugin_get_buffer(&buffer_size);
     dspbuffer = wavbuffer + buffer_size / 2;
+    dspbuffer_count = (buffer_size - (dspbuffer - wavbuffer)) /
+                        (2 * sizeof (int16_t));
 
     codec_mallocbuf = rb->plugin_get_audio_buffer(&audiosize);
     /* Align codec_mallocbuf to pointer size, tlsf wants that */
diff --git a/apps/settings.c b/apps/settings.c
index acc38c2388..49d239a2c1 100644
--- a/apps/settings.c
+++ b/apps/settings.c
@@ -979,20 +979,17 @@ void settings_apply(bool read_disk)
     audio_set_crossfade(global_settings.crossfade);
 #endif
     dsp_set_replaygain();
-    dsp_set_crossfeed(global_settings.crossfeed);
+    dsp_crossfeed_enable(global_settings.crossfeed);
     dsp_set_crossfeed_direct_gain(global_settings.crossfeed_direct_gain);
     dsp_set_crossfeed_cross_params(global_settings.crossfeed_cross_gain,
                                    global_settings.crossfeed_hf_attenuation,
                                    global_settings.crossfeed_hf_cutoff);
 
     /* Configure software equalizer, hardware eq is handled in audio_init() */
-    dsp_set_eq(global_settings.eq_enabled);
+    dsp_eq_enable(global_settings.eq_enabled);
     dsp_set_eq_precut(global_settings.eq_precut);
-
-    for(int i = 0; i < 5; i++) {
-        dsp_set_eq_coefs(i, global_settings.eq_band_settings[i].cutoff,
-                         global_settings.eq_band_settings[i].q,
-                         global_settings.eq_band_settings[i].gain);
+    for(int i = 0; i < EQ_NUM_BANDS; i++) {
+        dsp_set_eq_coefs(i, &global_settings.eq_band_settings[i]);
     }
 
     dsp_dither_enable(global_settings.dithering_enabled);
diff --git a/apps/settings.h b/apps/settings.h
index b312c1e784..4d94ca8ba8 100644
--- a/apps/settings.h
+++ b/apps/settings.h
@@ -32,6 +32,7 @@
 #include "button.h"
 #if CONFIG_CODEC == SWCODEC
 #include "audio.h"
+#include "dsp.h"
 #endif
 #include "rbpaths.h"
 
@@ -339,13 +340,7 @@ struct user_settings
     /* EQ */
     bool eq_enabled;            /* Enable equalizer */
     unsigned int eq_precut;     /* dB */
-
-    struct eq_band_setting
-    {
-        int cutoff;        /* Hz */
-        int q;
-        int gain;          /* +/- dB */
-    } eq_band_settings[5];
+    struct eq_band_setting eq_band_settings[EQ_NUM_BANDS]; /* for each band */
 
     /* Misc. swcodec */
     int  beep;              /* system beep volume when changing tracks etc. */
@@ -772,14 +767,7 @@ struct user_settings
 #endif
 
 #if CONFIG_CODEC == SWCODEC
-    struct compressor_settings
-    {
-        int threshold;
-        int makeup_gain;
-        int ratio;
-        int knee;
-        int release_time;
-    } compressor_settings;
+    struct compressor_settings compressor_settings;
 #endif
 
     int sleeptimer_duration; /* In minutes; 0=off */
diff --git a/apps/settings_list.c b/apps/settings_list.c
index af48d11c85..82cccd891f 100644
--- a/apps/settings_list.c
+++ b/apps/settings_list.c
@@ -1398,7 +1398,7 @@ const struct settings_list settings[] = {
 
     /* crossfeed */
     OFFON_SETTING(F_SOUNDSETTING, crossfeed, LANG_CROSSFEED, false,
-                  "crossfeed", dsp_set_crossfeed),
+                  "crossfeed", dsp_crossfeed_enable),
     INT_SETTING_NOWRAP(F_SOUNDSETTING, crossfeed_direct_gain,
                        LANG_CROSSFEED_DIRECT_GAIN, -15,
                        "crossfeed direct gain", UNIT_DB, -60, 0, 5,
diff --git a/apps/voice_thread.c b/apps/voice_thread.c
index 07a67256c4..cff703adfa 100644
--- a/apps/voice_thread.c
+++ b/apps/voice_thread.c
@@ -133,9 +133,8 @@ struct voice_thread_data
     SpeexBits bits;         /* Bit cursor */
     struct dsp_config *dsp; /* DSP used for voice output */
     struct voice_info vi;   /* Copy of clip data */
-    const char *src[2];     /* Current output buffer pointers */
     int lookahead;          /* Number of samples to drop at start of clip */
-    int count;              /* Count of samples remaining to send to PCM */
+    struct dsp_buffer src;  /* Speex output buffer/input to DSP */
 };
 
 /* Functions called in their repective state that return the next state to
@@ -264,9 +263,7 @@ void voice_wait(void)
  * setup the DSP parameters */
 static void voice_data_init(struct voice_thread_data *td)
 {
-    td->dsp = (struct dsp_config *)dsp_configure(NULL, DSP_MYDSP,
-                                                 CODEC_IDX_VOICE);
-
+    td->dsp = dsp_get_config(CODEC_IDX_VOICE);
     dsp_configure(td->dsp, DSP_RESET, 0);
     dsp_configure(td->dsp, DSP_SET_FREQUENCY, VOICE_SAMPLE_RATE);
     dsp_configure(td->dsp, DSP_SET_SAMPLE_DEPTH, VOICE_SAMPLE_DEPTH);
@@ -378,7 +375,8 @@ static enum voice_state voice_decode(struct voice_thread_data *td)
         else
         {
             /* If all clips are done and not playing, force pcm playback. */
-            voice_start_playback();
+            if (voice_unplayed_frames() > 0)
+                voice_start_playback();
             return VOICE_STATE_MESSAGE;
         }
     }
@@ -387,12 +385,14 @@ static enum voice_state voice_decode(struct voice_thread_data *td)
         yield();
 
         /* Output the decoded frame */
-        td->count = VOICE_FRAME_COUNT - td->lookahead;
-        td->src[0] = (const char *)&voice_output_buf[td->lookahead];
-        td->src[1] = NULL;
+        td->src.remcount  = VOICE_FRAME_COUNT - td->lookahead;
+        td->src.pin[0]    = &voice_output_buf[td->lookahead];
+        td->src.pin[1]    = NULL;
+        td->src.proc_mask = 0;
+
         td->lookahead -= MIN(VOICE_FRAME_COUNT, td->lookahead);
 
-        if (td->count > 0)
+        if (td->src.remcount > 0)
             return VOICE_STATE_BUFFER_INSERT;
     }
 
@@ -405,12 +405,21 @@ static enum voice_state voice_buffer_insert(struct voice_thread_data *td)
     if (!queue_empty(&voice_queue))
         return VOICE_STATE_MESSAGE;
 
-    char *dest = (char *)voice_buf_get();
+    struct dsp_buffer dst;
 
-    if (dest != NULL)
+    if ((dst.p16out = voice_buf_get()) != NULL)
     {
-        voice_buf_commit(dsp_process(td->dsp, dest, td->src, td->count));
-        return VOICE_STATE_DECODE;
+        dst.remcount = 0;
+        dst.bufcount = VOICE_PCM_FRAME_COUNT;
+
+        dsp_process(td->dsp, &td->src, &dst);
+
+        voice_buf_commit(dst.remcount);
+
+        /* Unless other effects are introduced to voice that have delays,
+           all output should have been purged to dst in one call */
+        return td->src.remcount > 0 ?
+            VOICE_STATE_BUFFER_INSERT : VOICE_STATE_DECODE;
     }
 
     sleep(0);