Fundamentally rewrite much of the audio DSP.

Creates a standard buffer passing, local data passing and messaging
system for processing stages. Stages can be moved to their own source
files to reduce clutter and ease assimilation of new ones. dsp.c
becomes dsp_core.c which supports an engine and framework for effects.

Formats and change notifications are passed along with the buffer so
that they arrive at the correct time at each stage in the chain
regardless of the internal delays of a particular one.

Removes restrictions on the number of samples that can be processed at
a time and it pays attention to destination buffer size restrictions
without having to limit input count, which also allows pcmbuf to
remain fuller and safely set its own buffer limits as it sees fit.
There is no longer a need to query input/output counts given a certain
number of input samples; just give it the sizes of the source and
destination buffers.

Works in harmony with stages that are not deterministic in terms of
sample input/output ratio (like both resamplers but most notably
the timestretch). As a result it fixes quirks with timestretch hanging
up with certain settings and it now operates properly throughout its
full settings range.
Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734
Reviewed-on: http://gerrit.rockbox.org/200
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>

1 files changed, 24 insertions, 27 deletions

diff --git a/apps/plugins/test_codec.c b/apps/plugins/test_codec.c
index dafcf35..920be54 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 */