4 files changed, 63 insertions, 53 deletions

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 */