336 lines
11 KiB
C
336 lines
11 KiB
C
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/*
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Simple DirectMedia Layer
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Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
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This software is provided 'as-is', without any express or implied
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warranty. In no event will the authors be held liable for any damages
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arising from the use of this software.
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Permission is granted to anyone to use this software for any purpose,
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including commercial applications, and to alter it and redistribute it
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freely, subject to the following restrictions:
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1. The origin of this software must not be misrepresented; you must not
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claim that you wrote the original software. If you use this software
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in a product, an acknowledgment in the product documentation would be
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appreciated but is not required.
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2. Altered source versions must be plainly marked as such, and must not be
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misrepresented as being the original software.
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3. This notice may not be removed or altered from any source distribution.
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*/
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#include "SDL_internal.h"
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#include "SDL_sysaudio.h"
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#include "SDL_audioresample.h"
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/* SDL's resampler uses a "bandlimited interpolation" algorithm:
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https://ccrma.stanford.edu/~jos/resample/ */
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#include "SDL_audio_resampler_filter.h"
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/* For a given srcpos, `srcpos + frame` are sampled, where `-RESAMPLER_ZERO_CROSSINGS < frame <= RESAMPLER_ZERO_CROSSINGS`.
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* Note, when upsampling, it is also possible to start sampling from `srcpos = -1`. */
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#define RESAMPLER_MAX_PADDING_FRAMES (RESAMPLER_ZERO_CROSSINGS + 1)
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#define RESAMPLER_FILTER_INTERP_BITS (32 - RESAMPLER_BITS_PER_ZERO_CROSSING)
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#define RESAMPLER_FILTER_INTERP_RANGE (1 << RESAMPLER_FILTER_INTERP_BITS)
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#define RESAMPLER_SAMPLES_PER_FRAME (RESAMPLER_ZERO_CROSSINGS * 2)
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#define RESAMPLER_FULL_FILTER_SIZE (RESAMPLER_SAMPLES_PER_FRAME * (RESAMPLER_SAMPLES_PER_ZERO_CROSSING + 1))
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static void ResampleFrame_Scalar(const float *src, float *dst, const float *raw_filter, float interp, int chans)
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{
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int i, chan;
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float filter[RESAMPLER_SAMPLES_PER_FRAME];
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// Interpolate between the nearest two filters
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for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; i++) {
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filter[i] = (raw_filter[i] * (1.0f - interp)) + (raw_filter[i + RESAMPLER_SAMPLES_PER_FRAME] * interp);
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}
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if (chans == 2) {
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float out[2];
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out[0] = 0.0f;
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out[1] = 0.0f;
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for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; i++) {
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const float scale = filter[i];
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out[0] += src[i * 2 + 0] * scale;
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out[1] += src[i * 2 + 1] * scale;
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}
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dst[0] = out[0];
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dst[1] = out[1];
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return;
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}
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if (chans == 1) {
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float out = 0.0f;
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for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; i++) {
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out += src[i] * filter[i];
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}
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dst[0] = out;
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return;
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}
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for (chan = 0; chan < chans; chan++) {
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float f = 0.0f;
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for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; i++) {
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f += src[i * chans + chan] * filter[i];
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}
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dst[chan] = f;
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}
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}
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#ifdef SDL_SSE_INTRINSICS
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static void SDL_TARGETING("sse") ResampleFrame_SSE(const float *src, float *dst, const float *raw_filter, float interp, int chans)
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{
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#if RESAMPLER_SAMPLES_PER_FRAME != 10
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#error Invalid samples per frame
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#endif
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// Load the filter
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__m128 f0 = _mm_loadu_ps(raw_filter + 0);
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__m128 f1 = _mm_loadu_ps(raw_filter + 4);
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__m128 f2 = _mm_loadl_pi(_mm_setzero_ps(), (const __m64 *)(raw_filter + 8));
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__m128 g0 = _mm_loadu_ps(raw_filter + 10);
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__m128 g1 = _mm_loadu_ps(raw_filter + 14);
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__m128 g2 = _mm_loadl_pi(_mm_setzero_ps(), (const __m64 *)(raw_filter + 18));
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__m128 interp1 = _mm_set1_ps(interp);
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__m128 interp2 = _mm_sub_ps(_mm_set1_ps(1.0f), _mm_set1_ps(interp));
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// Linear interpolate the filter
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f0 = _mm_add_ps(_mm_mul_ps(f0, interp2), _mm_mul_ps(g0, interp1));
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f1 = _mm_add_ps(_mm_mul_ps(f1, interp2), _mm_mul_ps(g1, interp1));
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f2 = _mm_add_ps(_mm_mul_ps(f2, interp2), _mm_mul_ps(g2, interp1));
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if (chans == 2) {
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// Duplicate each of the filter elements
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g0 = _mm_unpackhi_ps(f0, f0);
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f0 = _mm_unpacklo_ps(f0, f0);
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g1 = _mm_unpackhi_ps(f1, f1);
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f1 = _mm_unpacklo_ps(f1, f1);
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f2 = _mm_unpacklo_ps(f2, f2);
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// Multiply the filter by the input
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f0 = _mm_mul_ps(f0, _mm_loadu_ps(src + 0));
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g0 = _mm_mul_ps(g0, _mm_loadu_ps(src + 4));
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f1 = _mm_mul_ps(f1, _mm_loadu_ps(src + 8));
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g1 = _mm_mul_ps(g1, _mm_loadu_ps(src + 12));
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f2 = _mm_mul_ps(f2, _mm_loadu_ps(src + 16));
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// Calculate the sum
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f0 = _mm_add_ps(_mm_add_ps(_mm_add_ps(f0, g0), _mm_add_ps(f1, g1)), f2);
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f0 = _mm_add_ps(f0, _mm_movehl_ps(f0, f0));
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// Store the result
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_mm_storel_pi((__m64 *)dst, f0);
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return;
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}
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if (chans == 1) {
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// Multiply the filter by the input
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f0 = _mm_mul_ps(f0, _mm_loadu_ps(src + 0));
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f1 = _mm_mul_ps(f1, _mm_loadu_ps(src + 4));
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f2 = _mm_mul_ps(f2, _mm_loadl_pi(_mm_setzero_ps(), (const __m64 *)(src + 8)));
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// Calculate the sum
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f0 = _mm_add_ps(f0, f1);
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f0 = _mm_add_ps(_mm_add_ps(f0, f2), _mm_movehl_ps(f0, f0));
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f0 = _mm_add_ss(f0, _mm_shuffle_ps(f0, f0, _MM_SHUFFLE(1, 1, 1, 1)));
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// Store the result
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_mm_store_ss(dst, f0);
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return;
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}
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float filter[RESAMPLER_SAMPLES_PER_FRAME];
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_mm_storeu_ps(filter + 0, f0);
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_mm_storeu_ps(filter + 4, f1);
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_mm_storel_pi((__m64 *)(filter + 8), f2);
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int i, chan = 0;
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for (; chan + 4 <= chans; chan += 4) {
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f0 = _mm_setzero_ps();
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for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; i++) {
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f0 = _mm_add_ps(f0, _mm_mul_ps(_mm_loadu_ps(&src[i * chans + chan]), _mm_load1_ps(&filter[i])));
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}
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_mm_storeu_ps(&dst[chan], f0);
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}
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for (; chan < chans; chan++) {
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f0 = _mm_setzero_ps();
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for (i = 0; i < RESAMPLER_SAMPLES_PER_FRAME; i++) {
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f0 = _mm_add_ss(f0, _mm_mul_ss(_mm_load_ss(&src[i * chans + chan]), _mm_load_ss(&filter[i])));
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}
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_mm_store_ss(&dst[chan], f0);
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}
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}
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#endif
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static void (*ResampleFrame)(const float *src, float *dst, const float *raw_filter, float interp, int chans);
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static float FullResamplerFilter[RESAMPLER_FULL_FILTER_SIZE];
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void SDL_SetupAudioResampler(void)
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{
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static SDL_bool setup = SDL_FALSE;
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if (setup) {
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return;
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}
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// Build a table combining the left and right wings, for faster access
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int i, j;
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for (i = 0; i < RESAMPLER_SAMPLES_PER_ZERO_CROSSING; ++i) {
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for (j = 0; j < RESAMPLER_ZERO_CROSSINGS; j++) {
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int lwing = (i * RESAMPLER_SAMPLES_PER_FRAME) + (RESAMPLER_ZERO_CROSSINGS - 1) - j;
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int rwing = (RESAMPLER_FULL_FILTER_SIZE - 1) - lwing;
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float value = ResamplerFilter[(i * RESAMPLER_ZERO_CROSSINGS) + j];
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FullResamplerFilter[lwing] = value;
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FullResamplerFilter[rwing] = value;
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}
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}
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for (i = 0; i < RESAMPLER_ZERO_CROSSINGS; ++i) {
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int rwing = i + RESAMPLER_ZERO_CROSSINGS;
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int lwing = (RESAMPLER_FULL_FILTER_SIZE - 1) - rwing;
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FullResamplerFilter[lwing] = 0.0f;
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FullResamplerFilter[rwing] = 0.0f;
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}
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ResampleFrame = ResampleFrame_Scalar;
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#ifdef SDL_SSE_INTRINSICS
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if (SDL_HasSSE()) {
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ResampleFrame = ResampleFrame_SSE;
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}
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#endif
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setup = SDL_TRUE;
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}
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Sint64 SDL_GetResampleRate(int src_rate, int dst_rate)
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{
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SDL_assert(src_rate > 0);
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SDL_assert(dst_rate > 0);
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Sint64 sample_rate = ((Sint64)src_rate << 32) / (Sint64)dst_rate;
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SDL_assert(sample_rate > 0);
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return sample_rate;
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}
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int SDL_GetResamplerHistoryFrames(void)
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{
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// Even if we aren't currently resampling, make sure to keep enough history in case we need to later.
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return RESAMPLER_MAX_PADDING_FRAMES;
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}
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int SDL_GetResamplerPaddingFrames(Sint64 resample_rate)
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{
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// This must always be <= SDL_GetResamplerHistoryFrames()
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return resample_rate ? RESAMPLER_MAX_PADDING_FRAMES : 0;
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}
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// These are not general purpose. They do not check for all possible underflow/overflow
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SDL_FORCE_INLINE Sint64 ResamplerAdd(Sint64 a, Sint64 b, Sint64 *ret)
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{
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if ((b > 0) && (a > SDL_MAX_SINT64 - b)) {
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return -1;
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}
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*ret = a + b;
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return 0;
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}
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SDL_FORCE_INLINE Sint64 ResamplerMul(Sint64 a, Sint64 b, Sint64 *ret)
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{
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if ((b > 0) && (a > SDL_MAX_SINT64 / b)) {
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return -1;
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}
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*ret = a * b;
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return 0;
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}
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Sint64 SDL_GetResamplerInputFrames(Sint64 output_frames, Sint64 resample_rate, Sint64 resample_offset)
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{
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// Calculate the index of the last input frame, then add 1.
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// ((((output_frames - 1) * resample_rate) + resample_offset) >> 32) + 1
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Sint64 output_offset;
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if (ResamplerMul(output_frames, resample_rate, &output_offset) ||
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ResamplerAdd(output_offset, -resample_rate + resample_offset + 0x100000000, &output_offset)) {
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output_offset = SDL_MAX_SINT64;
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}
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Sint64 input_frames = (Sint64)(Sint32)(output_offset >> 32);
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input_frames = SDL_max(input_frames, 0);
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return input_frames;
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}
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Sint64 SDL_GetResamplerOutputFrames(Sint64 input_frames, Sint64 resample_rate, Sint64 *inout_resample_offset)
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{
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Sint64 resample_offset = *inout_resample_offset;
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// input_offset = (input_frames << 32) - resample_offset;
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Sint64 input_offset;
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if (ResamplerMul(input_frames, 0x100000000, &input_offset) ||
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ResamplerAdd(input_offset, -resample_offset, &input_offset)) {
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input_offset = SDL_MAX_SINT64;
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}
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// output_frames = div_ceil(input_offset, resample_rate)
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Sint64 output_frames = (input_offset > 0) ? (((input_offset - 1) / resample_rate) + 1) : 0;
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*inout_resample_offset = (output_frames * resample_rate) - input_offset;
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return output_frames;
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}
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void SDL_ResampleAudio(int chans, const float *src, int inframes, float *dst, int outframes,
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Sint64 resample_rate, Sint64 *inout_resample_offset)
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{
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int i;
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Sint64 srcpos = *inout_resample_offset;
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SDL_assert(resample_rate > 0);
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for (i = 0; i < outframes; i++) {
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int srcindex = (int)(Sint32)(srcpos >> 32);
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Uint32 srcfraction = (Uint32)(srcpos & 0xFFFFFFFF);
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srcpos += resample_rate;
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SDL_assert(srcindex >= -1 && srcindex < inframes);
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const float *filter = &FullResamplerFilter[(srcfraction >> RESAMPLER_FILTER_INTERP_BITS) * RESAMPLER_SAMPLES_PER_FRAME];
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const float interp = (float)(srcfraction & (RESAMPLER_FILTER_INTERP_RANGE - 1)) * (1.0f / RESAMPLER_FILTER_INTERP_RANGE);
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const float *frame = &src[(srcindex - (RESAMPLER_ZERO_CROSSINGS - 1)) * chans];
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ResampleFrame(frame, dst, filter, interp, chans);
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dst += chans;
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}
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*inout_resample_offset = srcpos - ((Sint64)inframes << 32);
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}
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