2015-06-21 15:33:46 +00:00
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/*
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Simple DirectMedia Layer
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2016-01-02 18:10:34 +00:00
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Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
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2015-06-21 15:33:46 +00:00
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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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/* Functions for audio drivers to perform runtime conversion of audio format */
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#include "SDL_audio.h"
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#include "SDL_audio_c.h"
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#include "SDL_assert.h"
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/* Effectively mix right and left channels into a single channel */
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static void SDLCALL
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SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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2016-11-05 06:34:38 +00:00
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float *dst = (float *) cvt->buf;
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const float *src = dst;
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2015-06-21 15:33:46 +00:00
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int i;
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2016-11-05 06:34:38 +00:00
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LOG_DEBUG_CONVERT("stereo", "mono");
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SDL_assert(format == AUDIO_F32SYS);
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for (i = cvt->len_cvt / 8; i; --i, src += 2) {
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*(dst++) = (float) ((((double) src[0]) + ((double) src[1])) * 0.5);
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2015-06-21 15:33:46 +00:00
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}
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cvt->len_cvt /= 2;
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if (cvt->filters[++cvt->filter_index]) {
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cvt->filters[cvt->filter_index] (cvt, format);
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}
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}
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/* Discard top 4 channels */
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static void SDLCALL
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SDL_ConvertStrip(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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2016-11-05 06:34:38 +00:00
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float *dst = (float *) cvt->buf;
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const float *src = dst;
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2015-06-21 15:33:46 +00:00
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int i;
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2016-11-05 06:34:38 +00:00
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LOG_DEBUG_CONVERT("6 channels", "stereo");
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SDL_assert(format == AUDIO_F32SYS);
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2015-06-21 15:33:46 +00:00
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2016-11-05 06:34:38 +00:00
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for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 2) {
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dst[0] = src[0];
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dst[1] = src[1];
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2015-06-21 15:33:46 +00:00
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}
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cvt->len_cvt /= 3;
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if (cvt->filters[++cvt->filter_index]) {
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cvt->filters[cvt->filter_index] (cvt, format);
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}
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}
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/* Discard top 2 channels of 6 */
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static void SDLCALL
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SDL_ConvertStrip_2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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2016-11-05 06:34:38 +00:00
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float *dst = (float *) cvt->buf;
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const float *src = dst;
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2015-06-21 15:33:46 +00:00
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int i;
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2016-11-05 06:34:38 +00:00
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LOG_DEBUG_CONVERT("6 channels", "quad");
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SDL_assert(format == AUDIO_F32SYS);
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2015-06-21 15:33:46 +00:00
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2016-11-05 06:34:38 +00:00
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for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 4) {
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dst[0] = src[0];
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dst[1] = src[1];
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dst[2] = src[2];
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dst[3] = src[3];
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2015-06-21 15:33:46 +00:00
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}
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cvt->len_cvt /= 6;
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cvt->len_cvt *= 4;
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if (cvt->filters[++cvt->filter_index]) {
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cvt->filters[cvt->filter_index] (cvt, format);
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}
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}
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/* Duplicate a mono channel to both stereo channels */
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static void SDLCALL
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SDL_ConvertStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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2016-11-05 06:34:38 +00:00
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const float *src = (const float *) (cvt->buf + cvt->len_cvt);
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float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
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2015-06-21 15:33:46 +00:00
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int i;
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2016-11-05 06:34:38 +00:00
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LOG_DEBUG_CONVERT("mono", "stereo");
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SDL_assert(format == AUDIO_F32SYS);
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2015-06-21 15:33:46 +00:00
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2016-11-05 06:34:38 +00:00
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for (i = cvt->len_cvt / sizeof (float); i; --i) {
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src--;
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dst -= 2;
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dst[0] = dst[1] = *src;
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2015-06-21 15:33:46 +00:00
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}
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cvt->len_cvt *= 2;
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if (cvt->filters[++cvt->filter_index]) {
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cvt->filters[cvt->filter_index] (cvt, format);
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}
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}
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/* Duplicate a stereo channel to a pseudo-5.1 stream */
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static void SDLCALL
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SDL_ConvertSurround(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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int i;
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2016-11-05 06:34:38 +00:00
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float lf, rf, ce;
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const float *src = (const float *) (cvt->buf + cvt->len_cvt);
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float *dst = (float *) (cvt->buf + cvt->len_cvt * 3);
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LOG_DEBUG_CONVERT("stereo", "5.1");
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SDL_assert(format == AUDIO_F32SYS);
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for (i = cvt->len_cvt / 8; i; --i) {
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dst -= 6;
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src -= 2;
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lf = src[0];
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rf = src[1];
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ce = (lf * 0.5f) + (rf * 0.5f);
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dst[0] = src[0];
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dst[1] = src[1];
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dst[2] = lf - ce;
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dst[3] = rf - ce;
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dst[4] = dst[5] = ce;
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2015-06-21 15:33:46 +00:00
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}
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2016-11-05 06:34:38 +00:00
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2015-06-21 15:33:46 +00:00
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cvt->len_cvt *= 3;
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if (cvt->filters[++cvt->filter_index]) {
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cvt->filters[cvt->filter_index] (cvt, format);
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}
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}
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/* Duplicate a stereo channel to a pseudo-4.0 stream */
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static void SDLCALL
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SDL_ConvertSurround_4(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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2016-11-05 06:34:38 +00:00
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const float *src = (const float *) (cvt->buf + cvt->len_cvt);
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float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
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float lf, rf, ce;
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2015-06-21 15:33:46 +00:00
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int i;
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2016-11-05 06:34:38 +00:00
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LOG_DEBUG_CONVERT("stereo", "quad");
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SDL_assert(format == AUDIO_F32SYS);
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for (i = cvt->len_cvt / 8; i; --i) {
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dst -= 4;
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src -= 2;
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lf = src[0];
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rf = src[1];
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ce = (lf / 2) + (rf / 2);
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dst[0] = src[0];
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dst[1] = src[1];
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dst[2] = lf - ce;
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dst[3] = rf - ce;
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2015-06-21 15:33:46 +00:00
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}
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2016-11-05 06:34:38 +00:00
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2015-06-21 15:33:46 +00:00
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cvt->len_cvt *= 2;
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if (cvt->filters[++cvt->filter_index]) {
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cvt->filters[cvt->filter_index] (cvt, format);
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}
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}
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int
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SDL_ConvertAudio(SDL_AudioCVT * cvt)
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{
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/* !!! FIXME: (cvt) should be const; stack-copy it here. */
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/* !!! FIXME: (actually, we can't...len_cvt needs to be updated. Grr.) */
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/* Make sure there's data to convert */
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if (cvt->buf == NULL) {
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2016-11-05 06:34:38 +00:00
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return SDL_SetError("No buffer allocated for conversion");
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2015-06-21 15:33:46 +00:00
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}
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2016-11-05 06:34:38 +00:00
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2015-06-21 15:33:46 +00:00
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/* Return okay if no conversion is necessary */
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cvt->len_cvt = cvt->len;
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if (cvt->filters[0] == NULL) {
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2016-11-05 06:34:38 +00:00
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return 0;
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2015-06-21 15:33:46 +00:00
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}
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/* Set up the conversion and go! */
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cvt->filter_index = 0;
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cvt->filters[0] (cvt, cvt->src_format);
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2016-11-05 06:34:38 +00:00
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return 0;
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2015-06-21 15:33:46 +00:00
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}
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2016-11-05 06:34:38 +00:00
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static void SDLCALL
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SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format)
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2015-06-21 15:33:46 +00:00
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{
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2016-11-05 06:34:38 +00:00
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#if DEBUG_CONVERT
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fprintf(stderr, "Converting byte order\n");
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#endif
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switch (SDL_AUDIO_BITSIZE(format)) {
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#define CASESWAP(b) \
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case b: { \
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Uint##b *ptr = (Uint##b *) cvt->buf; \
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int i; \
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for (i = cvt->len_cvt / sizeof (*ptr); i; --i, ++ptr) { \
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*ptr = SDL_Swap##b(*ptr); \
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} \
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break; \
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}
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CASESWAP(16);
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CASESWAP(32);
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CASESWAP(64);
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#undef CASESWAP
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default: SDL_assert(!"unhandled byteswap datatype!"); break;
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}
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2015-06-21 15:33:46 +00:00
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2016-11-05 06:34:38 +00:00
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if (cvt->filters[++cvt->filter_index]) {
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/* flip endian flag for data. */
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if (format & SDL_AUDIO_MASK_ENDIAN) {
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format &= ~SDL_AUDIO_MASK_ENDIAN;
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} else {
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format |= SDL_AUDIO_MASK_ENDIAN;
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}
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cvt->filters[cvt->filter_index](cvt, format);
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}
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2015-06-21 15:33:46 +00:00
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}
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static int
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2016-11-05 06:34:38 +00:00
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SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
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2015-06-21 15:33:46 +00:00
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{
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2016-11-05 06:34:38 +00:00
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int retval = 0; /* 0 == no conversion necessary. */
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2015-06-21 15:33:46 +00:00
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2016-11-05 06:34:38 +00:00
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if ((SDL_AUDIO_ISBIGENDIAN(src_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
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cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
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retval = 1; /* added a converter. */
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}
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2015-06-21 15:33:46 +00:00
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2016-11-05 06:34:38 +00:00
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if (!SDL_AUDIO_ISFLOAT(src_fmt)) {
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SDL_AudioFilter filter = NULL;
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switch (src_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
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case AUDIO_S8: filter = SDL_Convert_S8_to_F32; break;
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case AUDIO_U8: filter = SDL_Convert_U8_to_F32; break;
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case AUDIO_S16: filter = SDL_Convert_S16_to_F32; break;
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case AUDIO_S32: filter = SDL_Convert_S32_to_F32; break;
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default: SDL_assert(!"Unexpected audio format!"); break;
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2015-06-21 15:33:46 +00:00
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}
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2016-11-05 06:34:38 +00:00
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if (!filter) {
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return SDL_SetError("No conversion available for these formats");
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}
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const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
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const Uint16 dst_bitsize = 32;
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2015-06-21 15:33:46 +00:00
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cvt->filters[cvt->filter_index++] = filter;
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if (src_bitsize < dst_bitsize) {
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const int mult = (dst_bitsize / src_bitsize);
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cvt->len_mult *= mult;
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cvt->len_ratio *= mult;
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} else if (src_bitsize > dst_bitsize) {
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cvt->len_ratio /= (src_bitsize / dst_bitsize);
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}
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2016-11-05 06:34:38 +00:00
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retval = 1; /* added a converter. */
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2015-06-21 15:33:46 +00:00
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}
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2016-11-05 06:34:38 +00:00
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return retval;
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2015-06-21 15:33:46 +00:00
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}
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2016-11-05 06:34:38 +00:00
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static int
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SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt)
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2015-06-21 15:33:46 +00:00
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{
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2016-11-05 06:34:38 +00:00
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int retval = 0; /* 0 == no conversion necessary. */
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if (!SDL_AUDIO_ISFLOAT(dst_fmt)) {
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SDL_AudioFilter filter = NULL;
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switch (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
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case AUDIO_S8: filter = SDL_Convert_F32_to_S8; break;
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case AUDIO_U8: filter = SDL_Convert_F32_to_U8; break;
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case AUDIO_S16: filter = SDL_Convert_F32_to_S16; break;
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case AUDIO_S32: filter = SDL_Convert_F32_to_S32; break;
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default: SDL_assert(!"Unexpected audio format!"); break;
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}
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if (!filter) {
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return SDL_SetError("No conversion available for these formats");
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}
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2015-06-21 15:33:46 +00:00
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2016-11-05 06:34:38 +00:00
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const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
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const Uint16 src_bitsize = 32;
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cvt->filters[cvt->filter_index++] = filter;
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if (src_bitsize < dst_bitsize) {
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const int mult = (dst_bitsize / src_bitsize);
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cvt->len_mult *= mult;
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cvt->len_ratio *= mult;
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} else if (src_bitsize > dst_bitsize) {
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|
|
cvt->len_ratio /= (src_bitsize / dst_bitsize);
|
|
|
|
}
|
|
|
|
retval = 1; /* added a converter. */
|
|
|
|
}
|
|
|
|
|
|
|
|
if ((SDL_AUDIO_ISBIGENDIAN(dst_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
|
|
|
|
cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
|
|
|
|
retval = 1; /* added a converter. */
|
|
|
|
}
|
|
|
|
|
|
|
|
return retval;
|
2015-06-21 15:33:46 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
SDL_FindFrequencyMultiple(const int src_rate, const int dst_rate)
|
|
|
|
{
|
|
|
|
int retval = 0;
|
|
|
|
|
|
|
|
/* If we only built with the arbitrary resamplers, ignore multiples. */
|
|
|
|
int lo, hi;
|
|
|
|
int div;
|
|
|
|
|
|
|
|
SDL_assert(src_rate != 0);
|
|
|
|
SDL_assert(dst_rate != 0);
|
|
|
|
SDL_assert(src_rate != dst_rate);
|
|
|
|
|
|
|
|
if (src_rate < dst_rate) {
|
|
|
|
lo = src_rate;
|
|
|
|
hi = dst_rate;
|
|
|
|
} else {
|
|
|
|
lo = dst_rate;
|
|
|
|
hi = src_rate;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* zero means "not a supported multiple" ... we only do 2x and 4x. */
|
|
|
|
if ((hi % lo) != 0)
|
|
|
|
return 0; /* not a multiple. */
|
|
|
|
|
|
|
|
div = hi / lo;
|
|
|
|
retval = ((div == 2) || (div == 4)) ? div : 0;
|
|
|
|
|
|
|
|
return retval;
|
|
|
|
}
|
|
|
|
|
2016-11-05 06:34:38 +00:00
|
|
|
#define RESAMPLER_FUNCS(chans) \
|
|
|
|
static void SDLCALL \
|
|
|
|
SDL_Upsample_Arbitrary_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
|
|
|
|
SDL_assert(format == AUDIO_F32SYS); \
|
|
|
|
SDL_Upsample_Arbitrary(cvt, chans); \
|
|
|
|
}\
|
|
|
|
static void SDLCALL \
|
|
|
|
SDL_Downsample_Arbitrary_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
|
|
|
|
SDL_assert(format == AUDIO_F32SYS); \
|
|
|
|
SDL_Downsample_Arbitrary(cvt, chans); \
|
|
|
|
} \
|
|
|
|
static void SDLCALL \
|
|
|
|
SDL_Upsample_x2_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
|
|
|
|
SDL_assert(format == AUDIO_F32SYS); \
|
|
|
|
SDL_Upsample_x2(cvt, chans); \
|
|
|
|
} \
|
|
|
|
static void SDLCALL \
|
|
|
|
SDL_Downsample_x2_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
|
|
|
|
SDL_assert(format == AUDIO_F32SYS); \
|
|
|
|
SDL_Downsample_Multiple(cvt, 2, chans); \
|
|
|
|
} \
|
|
|
|
static void SDLCALL \
|
|
|
|
SDL_Upsample_x4_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
|
|
|
|
SDL_assert(format == AUDIO_F32SYS); \
|
|
|
|
SDL_Upsample_x4(cvt, chans); \
|
|
|
|
} \
|
|
|
|
static void SDLCALL \
|
|
|
|
SDL_Downsample_x4_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
|
|
|
|
SDL_assert(format == AUDIO_F32SYS); \
|
|
|
|
SDL_Downsample_Multiple(cvt, 4, chans); \
|
|
|
|
}
|
|
|
|
RESAMPLER_FUNCS(1)
|
|
|
|
RESAMPLER_FUNCS(2)
|
|
|
|
RESAMPLER_FUNCS(4)
|
|
|
|
RESAMPLER_FUNCS(6)
|
|
|
|
RESAMPLER_FUNCS(8)
|
|
|
|
#undef RESAMPLER_FUNCS
|
|
|
|
|
2015-06-21 15:33:46 +00:00
|
|
|
static int
|
|
|
|
SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, int dst_channels,
|
|
|
|
int src_rate, int dst_rate)
|
|
|
|
{
|
|
|
|
if (src_rate != dst_rate) {
|
2016-11-05 06:34:38 +00:00
|
|
|
const int upsample = (src_rate < dst_rate) ? 1 : 0;
|
|
|
|
const int multiple = SDL_FindFrequencyMultiple(src_rate, dst_rate);
|
|
|
|
SDL_AudioFilter filter = NULL;
|
|
|
|
|
|
|
|
#define PICK_CHANNEL_FILTER(upordown, resampler) switch (dst_channels) { \
|
|
|
|
case 1: filter = SDL_##upordown##_##resampler##_c1; break; \
|
|
|
|
case 2: filter = SDL_##upordown##_##resampler##_c2; break; \
|
|
|
|
case 4: filter = SDL_##upordown##_##resampler##_c4; break; \
|
|
|
|
case 6: filter = SDL_##upordown##_##resampler##_c6; break; \
|
|
|
|
case 8: filter = SDL_##upordown##_##resampler##_c8; break; \
|
|
|
|
default: break; \
|
|
|
|
}
|
2015-06-21 15:33:46 +00:00
|
|
|
|
2016-11-05 06:34:38 +00:00
|
|
|
if (upsample) {
|
|
|
|
if (multiple == 0) {
|
|
|
|
PICK_CHANNEL_FILTER(Upsample, Arbitrary);
|
|
|
|
} else if (multiple == 2) {
|
|
|
|
PICK_CHANNEL_FILTER(Upsample, x2);
|
|
|
|
} else if (multiple == 4) {
|
|
|
|
PICK_CHANNEL_FILTER(Upsample, x4);
|
2015-06-21 15:33:46 +00:00
|
|
|
}
|
2016-11-05 06:34:38 +00:00
|
|
|
} else {
|
|
|
|
if (multiple == 0) {
|
|
|
|
PICK_CHANNEL_FILTER(Downsample, Arbitrary);
|
|
|
|
} else if (multiple == 2) {
|
|
|
|
PICK_CHANNEL_FILTER(Downsample, x2);
|
|
|
|
} else if (multiple == 4) {
|
|
|
|
PICK_CHANNEL_FILTER(Downsample, x4);
|
2015-06-21 15:33:46 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2016-11-05 06:34:38 +00:00
|
|
|
#undef PICK_CHANNEL_FILTER
|
|
|
|
|
|
|
|
if (filter == NULL) {
|
|
|
|
return SDL_SetError("No conversion available for these rates");
|
|
|
|
}
|
|
|
|
|
2015-06-21 15:33:46 +00:00
|
|
|
/* Update (cvt) with filter details... */
|
|
|
|
cvt->filters[cvt->filter_index++] = filter;
|
|
|
|
if (src_rate < dst_rate) {
|
|
|
|
const double mult = ((double) dst_rate) / ((double) src_rate);
|
|
|
|
cvt->len_mult *= (int) SDL_ceil(mult);
|
|
|
|
cvt->len_ratio *= mult;
|
|
|
|
} else {
|
|
|
|
cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate);
|
|
|
|
}
|
|
|
|
|
|
|
|
return 1; /* added a converter. */
|
|
|
|
}
|
|
|
|
|
|
|
|
return 0; /* no conversion necessary. */
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
/* Creates a set of audio filters to convert from one format to another.
|
|
|
|
Returns -1 if the format conversion is not supported, 0 if there's
|
|
|
|
no conversion needed, or 1 if the audio filter is set up.
|
|
|
|
*/
|
|
|
|
|
|
|
|
int
|
|
|
|
SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
|
|
|
|
SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate,
|
|
|
|
SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
|
|
|
|
{
|
|
|
|
/* Sanity check target pointer */
|
|
|
|
if (cvt == NULL) {
|
|
|
|
return SDL_InvalidParamError("cvt");
|
|
|
|
}
|
|
|
|
|
|
|
|
/* there are no unsigned types over 16 bits, so catch this up front. */
|
|
|
|
if ((SDL_AUDIO_BITSIZE(src_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(src_fmt))) {
|
|
|
|
return SDL_SetError("Invalid source format");
|
|
|
|
}
|
|
|
|
if ((SDL_AUDIO_BITSIZE(dst_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(dst_fmt))) {
|
|
|
|
return SDL_SetError("Invalid destination format");
|
|
|
|
}
|
|
|
|
|
|
|
|
/* prevent possible divisions by zero, etc. */
|
|
|
|
if ((src_channels == 0) || (dst_channels == 0)) {
|
|
|
|
return SDL_SetError("Source or destination channels is zero");
|
|
|
|
}
|
|
|
|
if ((src_rate == 0) || (dst_rate == 0)) {
|
|
|
|
return SDL_SetError("Source or destination rate is zero");
|
|
|
|
}
|
|
|
|
#ifdef DEBUG_CONVERT
|
|
|
|
printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
|
|
|
|
src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
|
|
|
|
#endif
|
|
|
|
|
|
|
|
/* Start off with no conversion necessary */
|
|
|
|
SDL_zerop(cvt);
|
|
|
|
cvt->src_format = src_fmt;
|
|
|
|
cvt->dst_format = dst_fmt;
|
|
|
|
cvt->needed = 0;
|
|
|
|
cvt->filter_index = 0;
|
|
|
|
cvt->filters[0] = NULL;
|
|
|
|
cvt->len_mult = 1;
|
|
|
|
cvt->len_ratio = 1.0;
|
|
|
|
cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
|
|
|
|
|
2016-11-05 06:34:38 +00:00
|
|
|
/* Type conversion goes like this now:
|
|
|
|
- byteswap to CPU native format first if necessary.
|
|
|
|
- convert to native Float32 if necessary.
|
|
|
|
- resample and change channel count if necessary.
|
|
|
|
- convert back to native format.
|
|
|
|
- byteswap back to foreign format if necessary.
|
|
|
|
|
|
|
|
The expectation is we can process data faster in float32
|
|
|
|
(possibly with SIMD), and making several passes over the same
|
|
|
|
buffer in is likely to be CPU cache-friendly, avoiding the
|
|
|
|
biggest performance hit in modern times. Previously we had
|
|
|
|
(script-generated) custom converters for every data type and
|
|
|
|
it was a bloat on SDL compile times and final library size. */
|
|
|
|
|
|
|
|
/* see if we can skip float conversion entirely (just a byteswap needed). */
|
|
|
|
if ((src_rate == dst_rate) && (src_channels == dst_channels) &&
|
|
|
|
((src_fmt != dst_fmt) &&
|
|
|
|
((src_fmt & ~SDL_AUDIO_MASK_ENDIAN) == (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN)))) {
|
|
|
|
cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
|
|
|
|
cvt->needed = 1;
|
|
|
|
return 1;
|
|
|
|
}
|
|
|
|
|
2015-06-21 15:33:46 +00:00
|
|
|
/* Convert data types, if necessary. Updates (cvt). */
|
2016-11-05 06:34:38 +00:00
|
|
|
if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) == -1) {
|
2015-06-21 15:33:46 +00:00
|
|
|
return -1; /* shouldn't happen, but just in case... */
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Channel conversion */
|
|
|
|
if (src_channels != dst_channels) {
|
|
|
|
if ((src_channels == 1) && (dst_channels > 1)) {
|
|
|
|
cvt->filters[cvt->filter_index++] = SDL_ConvertStereo;
|
|
|
|
cvt->len_mult *= 2;
|
|
|
|
src_channels = 2;
|
|
|
|
cvt->len_ratio *= 2;
|
|
|
|
}
|
|
|
|
if ((src_channels == 2) && (dst_channels == 6)) {
|
|
|
|
cvt->filters[cvt->filter_index++] = SDL_ConvertSurround;
|
|
|
|
src_channels = 6;
|
|
|
|
cvt->len_mult *= 3;
|
|
|
|
cvt->len_ratio *= 3;
|
|
|
|
}
|
|
|
|
if ((src_channels == 2) && (dst_channels == 4)) {
|
|
|
|
cvt->filters[cvt->filter_index++] = SDL_ConvertSurround_4;
|
|
|
|
src_channels = 4;
|
|
|
|
cvt->len_mult *= 2;
|
|
|
|
cvt->len_ratio *= 2;
|
|
|
|
}
|
|
|
|
while ((src_channels * 2) <= dst_channels) {
|
|
|
|
cvt->filters[cvt->filter_index++] = SDL_ConvertStereo;
|
|
|
|
cvt->len_mult *= 2;
|
|
|
|
src_channels *= 2;
|
|
|
|
cvt->len_ratio *= 2;
|
|
|
|
}
|
|
|
|
if ((src_channels == 6) && (dst_channels <= 2)) {
|
|
|
|
cvt->filters[cvt->filter_index++] = SDL_ConvertStrip;
|
|
|
|
src_channels = 2;
|
|
|
|
cvt->len_ratio /= 3;
|
|
|
|
}
|
|
|
|
if ((src_channels == 6) && (dst_channels == 4)) {
|
|
|
|
cvt->filters[cvt->filter_index++] = SDL_ConvertStrip_2;
|
|
|
|
src_channels = 4;
|
|
|
|
cvt->len_ratio /= 2;
|
|
|
|
}
|
|
|
|
/* This assumes that 4 channel audio is in the format:
|
|
|
|
Left {front/back} + Right {front/back}
|
|
|
|
so converting to L/R stereo works properly.
|
|
|
|
*/
|
|
|
|
while (((src_channels % 2) == 0) &&
|
|
|
|
((src_channels / 2) >= dst_channels)) {
|
|
|
|
cvt->filters[cvt->filter_index++] = SDL_ConvertMono;
|
|
|
|
src_channels /= 2;
|
|
|
|
cvt->len_ratio /= 2;
|
|
|
|
}
|
|
|
|
if (src_channels != dst_channels) {
|
|
|
|
/* Uh oh.. */ ;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Do rate conversion, if necessary. Updates (cvt). */
|
|
|
|
if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) ==
|
|
|
|
-1) {
|
|
|
|
return -1; /* shouldn't happen, but just in case... */
|
|
|
|
}
|
|
|
|
|
2016-11-05 06:34:38 +00:00
|
|
|
if (SDL_BuildAudioTypeCVTFromFloat(cvt, dst_fmt) == -1) {
|
|
|
|
return -1; /* shouldn't happen, but just in case... */
|
2015-06-21 15:33:46 +00:00
|
|
|
}
|
2016-11-05 06:34:38 +00:00
|
|
|
|
|
|
|
cvt->needed = (cvt->filter_index != 0);
|
2015-06-21 15:33:46 +00:00
|
|
|
return (cvt->needed);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* vi: set ts=4 sw=4 expandtab: */
|
2016-11-05 06:34:38 +00:00
|
|
|
|