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https://github.com/libretro/RetroArch
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(PS3) don't use Griffin override for audio_flush anymore, but
do keep optimizations when we aren't using DSP audio plugins
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/* RetroArch - A frontend for libretro.
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* Copyright (C) 2010-2012 - Hans-Kristian Arntzen
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* Copyright (C) 2011-2012 - Daniel De Matteis
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*
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* RetroArch is free software: you can redistribute it and/or modify it under the terms
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* of the GNU General Public License as published by the Free Software Found-
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* ation, either version 3 of the License, or (at your option) any later version.
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*
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* RetroArch is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
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* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
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* PURPOSE. See the GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along with RetroArch.
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* If not, see <http://www.gnu.org/licenses/>.
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*/
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#ifndef _RARCH_FUNC_HOOKS_H
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#define _RARCH_FUNC_HOOKS_H
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/*============================================================
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PLAYSTATION3
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============================================================ */
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#ifdef __CELLOS_LV2__
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#define HAVE_GRIFFIN_OVERRIDE_AUDIO_FLUSH_FUNC 1
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static bool audio_flush(const int16_t *data, size_t samples)
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{
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const float *output_data = NULL;
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unsigned output_frames = 0;
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audio_convert_s16_to_float(g_extern.audio_data.data, data, samples);
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struct resampler_data src_data = {0};
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src_data.data_in = g_extern.audio_data.data;
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src_data.data_out = g_extern.audio_data.outsamples;
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src_data.input_frames = (samples / 2);
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src_data.ratio = g_extern.audio_data.src_ratio;
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if (g_extern.is_slowmotion)
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src_data.ratio *= g_settings.slowmotion_ratio;
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resampler_process(g_extern.audio_data.source, &src_data);
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output_data = g_extern.audio_data.outsamples;
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output_frames = src_data.output_frames;
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if (audio_write_func(output_data, output_frames * sizeof(float) * 2) < 0)
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return false;
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return true;
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}
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#endif
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#endif
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19
retroarch.c
19
retroarch.c
@ -250,10 +250,6 @@ static void video_frame(const void *data, unsigned width, unsigned height, size_
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g_extern.frame_cache.pitch = pitch;
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}
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#ifdef HAVE_GRIFFIN
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#include "console/griffin/rarch_func_hooks.h"
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#endif
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void rarch_render_cached_frame(void)
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{
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#ifdef HAVE_FFMPEG
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@ -275,7 +271,6 @@ void rarch_render_cached_frame(void)
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#endif
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}
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#ifndef HAVE_GRIFFIN_OVERRIDE_AUDIO_FLUSH_FUNC
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static bool audio_flush(const int16_t *data, size_t samples)
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{
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#ifdef HAVE_FFMPEG
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@ -298,24 +293,29 @@ static bool audio_flush(const int16_t *data, size_t samples)
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audio_convert_s16_to_float(g_extern.audio_data.data, data, samples);
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#ifdef HAVE_DYLIB
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rarch_dsp_output_t dsp_output = {0};
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dsp_output.should_resample = RARCH_TRUE;
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#ifdef HAVE_DYLIB
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rarch_dsp_input_t dsp_input = {0};
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dsp_input.samples = g_extern.audio_data.data;
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dsp_input.frames = samples / 2;
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if (g_extern.audio_data.dsp_plugin)
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g_extern.audio_data.dsp_plugin->process(g_extern.audio_data.dsp_handle, &dsp_output, &dsp_input);
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#endif
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if (dsp_output.should_resample)
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{
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#endif
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struct resampler_data src_data = {0};
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#ifdef HAVE_DYLIB
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src_data.data_in = dsp_output.samples ? dsp_output.samples : g_extern.audio_data.data;
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src_data.data_out = g_extern.audio_data.outsamples;
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src_data.input_frames = dsp_output.samples ? dsp_output.frames : (samples / 2);
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#else
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src_data.data_in = g_extern.audio_data.data;
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src_data.input_frames = (samples / 2);
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#endif
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src_data.data_out = g_extern.audio_data.outsamples;
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if (g_extern.audio_data.rate_control)
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readjust_audio_input_rate();
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@ -328,8 +328,8 @@ static bool audio_flush(const int16_t *data, size_t samples)
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output_data = g_extern.audio_data.outsamples;
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output_frames = src_data.output_frames;
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}
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#ifdef HAVE_DYLIB
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}
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else
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{
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output_data = dsp_output.samples;
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@ -370,7 +370,6 @@ static bool audio_flush(const int16_t *data, size_t samples)
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return true;
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}
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#endif
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static void audio_sample_rewind(int16_t left, int16_t right)
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{
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