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>

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 */