From a62893610eea17054ec278ea8fa7c5c73ddd6022 Mon Sep 17 00:00:00 2001
From: Twinaphex <autechre1024@hotmail.com>
Date: Sun, 24 Jun 2012 13:11:10 +0200
Subject: [PATCH] (PS3) don't use Griffin override for audio_flush anymore, but
 do keep optimizations when we aren't using DSP audio plugins

---
 console/griffin/rarch_func_hooks.h | 58 ------------------------------
 retroarch.c                        | 19 +++++-----
 2 files changed, 9 insertions(+), 68 deletions(-)
 delete mode 100644 console/griffin/rarch_func_hooks.h

diff --git a/console/griffin/rarch_func_hooks.h b/console/griffin/rarch_func_hooks.h
deleted file mode 100644
index dbae9dc3cd..0000000000
--- a/console/griffin/rarch_func_hooks.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*  RetroArch - A frontend for libretro.
- *  Copyright (C) 2010-2012 - Hans-Kristian Arntzen
- *  Copyright (C) 2011-2012 - Daniel De Matteis
- * 
- *  RetroArch is free software: you can redistribute it and/or modify it under the terms
- *  of the GNU General Public License as published by the Free Software Found-
- *  ation, either version 3 of the License, or (at your option) any later version.
- *
- *  RetroArch is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
- *  without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
- *  PURPOSE.  See the GNU General Public License for more details.
- *
- *  You should have received a copy of the GNU General Public License along with RetroArch.
- *  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#ifndef _RARCH_FUNC_HOOKS_H
-#define _RARCH_FUNC_HOOKS_H
-
-/*============================================================
-	PLAYSTATION3
-============================================================ */
-
-#ifdef __CELLOS_LV2__
-
-#define HAVE_GRIFFIN_OVERRIDE_AUDIO_FLUSH_FUNC 1
-
-static bool audio_flush(const int16_t *data, size_t samples)
-{
-   const float *output_data = NULL;
-   unsigned output_frames = 0;
-
-   audio_convert_s16_to_float(g_extern.audio_data.data, data, samples);
-
-   struct resampler_data src_data = {0};
-   src_data.data_in = g_extern.audio_data.data;
-   src_data.data_out = g_extern.audio_data.outsamples;
-   src_data.input_frames = (samples / 2);
-
-   src_data.ratio = g_extern.audio_data.src_ratio;
-   if (g_extern.is_slowmotion)
-      src_data.ratio *= g_settings.slowmotion_ratio;
-
-   resampler_process(g_extern.audio_data.source, &src_data);
-
-   output_data = g_extern.audio_data.outsamples;
-   output_frames = src_data.output_frames;
-
-   if (audio_write_func(output_data, output_frames * sizeof(float) * 2) < 0)
-	   return false;
-
-   return true;
-}
-
-#endif
-
-
-#endif
diff --git a/retroarch.c b/retroarch.c
index 7bab2ddaa0..47e92276b2 100644
--- a/retroarch.c
+++ b/retroarch.c
@@ -250,10 +250,6 @@ static void video_frame(const void *data, unsigned width, unsigned height, size_
    g_extern.frame_cache.pitch  = pitch;
 }
 
-#ifdef HAVE_GRIFFIN
-#include "console/griffin/rarch_func_hooks.h"
-#endif
-
 void rarch_render_cached_frame(void)
 {
 #ifdef HAVE_FFMPEG
@@ -275,7 +271,6 @@ void rarch_render_cached_frame(void)
 #endif
 }
 
-#ifndef HAVE_GRIFFIN_OVERRIDE_AUDIO_FLUSH_FUNC
 static bool audio_flush(const int16_t *data, size_t samples)
 {
 #ifdef HAVE_FFMPEG
@@ -298,24 +293,29 @@ static bool audio_flush(const int16_t *data, size_t samples)
 
    audio_convert_s16_to_float(g_extern.audio_data.data, data, samples);
 
+#ifdef HAVE_DYLIB
    rarch_dsp_output_t dsp_output = {0};
    dsp_output.should_resample = RARCH_TRUE;
 
-#ifdef HAVE_DYLIB
    rarch_dsp_input_t dsp_input = {0};
    dsp_input.samples = g_extern.audio_data.data;
    dsp_input.frames = samples / 2;
 
    if (g_extern.audio_data.dsp_plugin)
       g_extern.audio_data.dsp_plugin->process(g_extern.audio_data.dsp_handle, &dsp_output, &dsp_input);
-#endif
 
    if (dsp_output.should_resample)
    {
+#endif
       struct resampler_data src_data = {0};
+#ifdef HAVE_DYLIB
       src_data.data_in = dsp_output.samples ? dsp_output.samples : g_extern.audio_data.data;
-      src_data.data_out = g_extern.audio_data.outsamples;
       src_data.input_frames = dsp_output.samples ? dsp_output.frames : (samples / 2);
+#else
+      src_data.data_in = g_extern.audio_data.data;
+      src_data.input_frames = (samples / 2);
+#endif
+      src_data.data_out = g_extern.audio_data.outsamples;
 
       if (g_extern.audio_data.rate_control)
          readjust_audio_input_rate();
@@ -328,8 +328,8 @@ static bool audio_flush(const int16_t *data, size_t samples)
 
       output_data = g_extern.audio_data.outsamples;
       output_frames = src_data.output_frames;
-   }
 #ifdef HAVE_DYLIB
+   }
    else
    {
       output_data = dsp_output.samples;
@@ -370,7 +370,6 @@ static bool audio_flush(const int16_t *data, size_t samples)
 
    return true;
 }
-#endif
 
 static void audio_sample_rewind(int16_t left, int16_t right)
 {