From 09186e31ae4409f50c1437911a413c36b381f3a4 Mon Sep 17 00:00:00 2001
From: Michael Sevakis <jethead71@rockbox.org>
Date: Fri, 16 Feb 2007 12:01:35 +0000
Subject: SWCODEC: Remove the last quirks when upsampling.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12336 a1c6a512-1295-4272-9138-f99709370657
---
 apps/dsp.c      | 13 +++++++++++--
 apps/playback.c | 26 ++++++++++----------------
 2 files changed, 21 insertions(+), 18 deletions(-)

diff --git a/apps/dsp.c b/apps/dsp.c
index e0fb4475da..02e231d800 100644
--- a/apps/dsp.c
+++ b/apps/dsp.c
@@ -253,9 +253,15 @@ static int downsample(int32_t **dst, int32_t **src, int count,
         if (pos < count)
             *d[j]++ = last_sample + FRACMUL((phase & 0xffff) << 15,
                 src[j][pos] - last_sample);
-        else  /* This is kinda nasty but works somewhat well for now */
-            *d[j]++ = src[j][count - 1];
+        else
+        {
+            /* No samples can be output here since were already passed the
+               end. Keep phase, save the last sample and return nothing. */
+            i = 0;
+            goto done;
+        }
     }
+
     phase += delta;
  
     while ((pos = phase >> 16) < count)
@@ -268,6 +274,7 @@ static int downsample(int32_t **dst, int32_t **src, int count,
     }
 
     /* Wrap phase accumulator back to start of next frame. */
+done:
     r->phase = phase - (count << 16);
     r->last_sample[0] = src[0][count - 1];
     r->last_sample[1] = src[1][count - 1];
@@ -768,6 +775,8 @@ int dsp_process(char *dst, const char *src[], int count)
         count -= samples;
         apply_gain(tmp, samples);
         samples = resample(tmp, samples);
+        if (samples <= 0)
+            break; /* I'm pretty sure we're downsampling here */
         if (dsp->crossfeed_enabled && dsp->stereo_mode != STEREO_MONO)
             apply_crossfeed(tmp, samples);
         if (dsp->eq_enabled)
diff --git a/apps/playback.c b/apps/playback.c
index caaaddec6e..aa21ccc39b 100644
--- a/apps/playback.c
+++ b/apps/playback.c
@@ -1047,14 +1047,8 @@ static bool voice_pcmbuf_insert_callback(
          * against resampling buffer overflows. */
         inp_count = dsp_input_count(out_count);
 
-        if (inp_count <= 0) 
-        {
-            DEBUGF("Error: dsp_input_count(%ld=dsp_output_count(%ld))=%ld<=0\n",
-                    out_count, count, inp_count);
-            /* If this happens, there are samples of codec data that don't
-             * become a number of pcm samples, and something is broken */
-            return false;
-        }
+        if (inp_count <= 0)
+            return true;
 
         /* Input size has grown, no error, just don't write more than length */
         if (inp_count > count)
@@ -1062,6 +1056,9 @@ static bool voice_pcmbuf_insert_callback(
 
         out_count = dsp_process(dest, src, inp_count);
 
+        if (out_count <= 0)
+            return true;
+
         if (playing)
         {
             pcmbuf_mix_voice(out_count);
@@ -1329,14 +1326,8 @@ static bool codec_pcmbuf_insert_callback(
          * against resampling buffer overflows. */
         inp_count = dsp_input_count(out_count);
 
-        if (inp_count <= 0) 
-        {
-            DEBUGF("Error: dsp_input_count(%ld=dsp_output_count(%ld))=%ld<=0\n",
-                    out_count, count, inp_count);
-            /* If this happens, there are samples of codec data that don't
-             * become a number of pcm samples, and something is broken */
-            return false;
-        }
+        if (inp_count <= 0)
+            return true;
 
         /* Input size has grown, no error, just don't write more than length */
         if (inp_count > count)
@@ -1344,6 +1335,9 @@ static bool codec_pcmbuf_insert_callback(
 
         out_count = dsp_process(dest, src, inp_count);
 
+        if (out_count <= 0)
+            return true;
+
         pcmbuf_write_complete(out_count);
 
 #ifdef PLAYBACK_VOICE
-- 
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