2015-06-21 17:33:46 +02:00
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/*
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Simple DirectMedia Layer
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2017-01-02 03:33:28 +01:00
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Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
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2015-06-21 17:33:46 +02:00
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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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2017-01-06 11:16:26 +01:00
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#include "SDL_loadso.h"
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2015-06-21 17:33:46 +02:00
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#include "SDL_assert.h"
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2017-01-06 01:29:38 +01:00
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#include "../SDL_dataqueue.h"
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2015-06-21 17:33:46 +02:00
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2017-01-06 11:16:26 +01:00
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#ifdef HAVE_LIBSAMPLERATE
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#include "samplerate.h"
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#endif
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2015-06-21 17:33:46 +02:00
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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 07:34:38 +01:00
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float *dst = (float *) cvt->buf;
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const float *src = dst;
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2015-06-21 17:33:46 +02:00
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int i;
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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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 07:34:38 +01:00
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float *dst = (float *) cvt->buf;
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const float *src = dst;
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2015-06-21 17:33:46 +02:00
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int i;
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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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 07:34:38 +01:00
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float *dst = (float *) cvt->buf;
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const float *src = dst;
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2015-06-21 17:33:46 +02:00
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int i;
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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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 07:34:38 +01:00
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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 17:33:46 +02:00
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int i;
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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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 07:34:38 +01:00
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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 17:33:46 +02:00
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}
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2016-11-05 07:34:38 +01:00
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2015-06-21 17:33:46 +02:00
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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 07:34:38 +01:00
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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 17:33:46 +02:00
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int i;
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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}
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2016-11-05 07:34:38 +01:00
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2015-06-21 17:33:46 +02:00
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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 07:34:38 +01:00
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return SDL_SetError("No buffer allocated for conversion");
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2015-06-21 17:33:46 +02:00
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}
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2016-11-05 07:34:38 +01:00
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2015-06-21 17:33:46 +02:00
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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 07:34:38 +01:00
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return 0;
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2015-06-21 17:33:46 +02:00
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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 07:34:38 +01:00
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return 0;
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2015-06-21 17:33:46 +02:00
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}
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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{
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2016-11-05 09:52:28 +01:00
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#if DEBUG_CONVERT
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printf("Converting byte order\n");
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#endif
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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}
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static int
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2016-11-05 07:34:38 +01:00
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SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
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2015-06-21 17:33:46 +02:00
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{
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2016-11-05 07:34:38 +01:00
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int retval = 0; /* 0 == no conversion necessary. */
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2015-06-21 17:33:46 +02:00
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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2016-11-05 07:34:38 +01:00
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if (!SDL_AUDIO_ISFLOAT(src_fmt)) {
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2016-11-05 08:53:59 +01:00
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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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2016-11-05 07:34:38 +01:00
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SDL_AudioFilter filter = NULL;
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2016-11-05 08:53:59 +01:00
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2016-11-05 07:34:38 +01:00
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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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2016-11-07 21:10:01 +01:00
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case AUDIO_U16: filter = SDL_Convert_U16_to_F32; break;
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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}
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2016-11-05 07:34:38 +01:00
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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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2015-06-21 17:33:46 +02:00
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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 08:53:59 +01:00
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2016-11-05 07:34:38 +01:00
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retval = 1; /* added a converter. */
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2015-06-21 17:33:46 +02:00
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}
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2016-11-05 07:34:38 +01:00
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return retval;
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2015-06-21 17:33:46 +02:00
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}
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2016-11-05 07:34:38 +01:00
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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 17:33:46 +02:00
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{
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2016-11-05 07:34:38 +01:00
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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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2016-11-05 08:56:55 +01:00
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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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2016-11-05 07:34:38 +01:00
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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;
|
|
|
|
case AUDIO_S16: filter = SDL_Convert_F32_to_S16; break;
|
2016-11-07 21:10:01 +01:00
|
|
|
case AUDIO_U16: filter = SDL_Convert_F32_to_U16; break;
|
2016-11-05 07:34:38 +01:00
|
|
|
case AUDIO_S32: filter = SDL_Convert_F32_to_S32; break;
|
|
|
|
default: SDL_assert(!"Unexpected audio format!"); break;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (!filter) {
|
|
|
|
return SDL_SetError("No conversion available for these formats");
|
|
|
|
}
|
2015-06-21 17:33:46 +02:00
|
|
|
|
2016-11-05 07:34:38 +01:00
|
|
|
cvt->filters[cvt->filter_index++] = filter;
|
|
|
|
if (src_bitsize < dst_bitsize) {
|
|
|
|
const int mult = (dst_bitsize / src_bitsize);
|
|
|
|
cvt->len_mult *= mult;
|
|
|
|
cvt->len_ratio *= mult;
|
|
|
|
} else if (src_bitsize > dst_bitsize) {
|
|
|
|
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 17:33:46 +02:00
|
|
|
}
|
|
|
|
|
|
|
|
|
2017-01-06 01:12:20 +01:00
|
|
|
/* !!! FIXME: We only have this macro salsa because SDL_AudioCVT doesn't store
|
|
|
|
!!! FIXME: channel info or integer sample rates, so we have to have
|
|
|
|
!!! FIXME: function entry points for each supported channel count and
|
|
|
|
!!! FIXME: multiple vs arbitrary. When we rev the ABI, remove this. */
|
2016-11-05 07:34:38 +01:00
|
|
|
#define RESAMPLER_FUNCS(chans) \
|
|
|
|
static void SDLCALL \
|
2017-01-06 01:12:20 +01:00
|
|
|
SDL_Upsample_Multiple_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
|
2016-11-05 07:34:38 +01:00
|
|
|
SDL_assert(format == AUDIO_F32SYS); \
|
2017-01-06 01:12:20 +01:00
|
|
|
SDL_Upsample_Multiple(cvt, chans); \
|
2016-11-05 07:34:38 +01:00
|
|
|
} \
|
|
|
|
static void SDLCALL \
|
2017-01-06 01:12:20 +01:00
|
|
|
SDL_Upsample_Arbitrary_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
|
2016-11-05 07:34:38 +01:00
|
|
|
SDL_assert(format == AUDIO_F32SYS); \
|
2017-01-06 01:12:20 +01:00
|
|
|
SDL_Upsample_Arbitrary(cvt, chans); \
|
|
|
|
}\
|
2016-11-05 07:34:38 +01:00
|
|
|
static void SDLCALL \
|
2017-01-06 01:12:20 +01:00
|
|
|
SDL_Downsample_Multiple_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
|
2016-11-05 07:34:38 +01:00
|
|
|
SDL_assert(format == AUDIO_F32SYS); \
|
2017-01-06 01:12:20 +01:00
|
|
|
SDL_Downsample_Multiple(cvt, chans); \
|
2016-11-05 07:34:38 +01:00
|
|
|
} \
|
|
|
|
static void SDLCALL \
|
2017-01-06 01:12:20 +01:00
|
|
|
SDL_Downsample_Arbitrary_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
|
2016-11-05 07:34:38 +01:00
|
|
|
SDL_assert(format == AUDIO_F32SYS); \
|
2017-01-06 01:12:20 +01:00
|
|
|
SDL_Downsample_Arbitrary(cvt, chans); \
|
2016-11-05 07:34:38 +01:00
|
|
|
}
|
|
|
|
RESAMPLER_FUNCS(1)
|
|
|
|
RESAMPLER_FUNCS(2)
|
|
|
|
RESAMPLER_FUNCS(4)
|
|
|
|
RESAMPLER_FUNCS(6)
|
|
|
|
RESAMPLER_FUNCS(8)
|
|
|
|
#undef RESAMPLER_FUNCS
|
|
|
|
|
2015-06-21 17:33:46 +02:00
|
|
|
static int
|
2017-01-06 01:12:20 +01:00
|
|
|
SDL_FindFrequencyMultiple(const int src_rate, const int dst_rate)
|
2015-06-21 17:33:46 +02:00
|
|
|
{
|
2017-01-06 01:12:20 +01:00
|
|
|
int lo, hi;
|
2016-11-05 07:34:38 +01:00
|
|
|
|
2017-01-06 01:12:20 +01:00
|
|
|
SDL_assert(src_rate != 0);
|
|
|
|
SDL_assert(dst_rate != 0);
|
|
|
|
SDL_assert(src_rate != dst_rate);
|
2015-06-21 17:33:46 +02:00
|
|
|
|
2017-01-06 01:12:20 +01:00
|
|
|
if (src_rate < dst_rate) {
|
|
|
|
lo = src_rate;
|
|
|
|
hi = dst_rate;
|
|
|
|
} else {
|
|
|
|
lo = dst_rate;
|
|
|
|
hi = src_rate;
|
|
|
|
}
|
2015-06-21 17:33:46 +02:00
|
|
|
|
2017-01-06 01:12:20 +01:00
|
|
|
if ((hi % lo) != 0)
|
|
|
|
return 0; /* not a multiple. */
|
2016-11-05 07:34:38 +01:00
|
|
|
|
2017-01-06 01:12:20 +01:00
|
|
|
return hi / lo;
|
|
|
|
}
|
2016-11-05 07:34:38 +01:00
|
|
|
|
2017-01-06 01:12:20 +01:00
|
|
|
static SDL_AudioFilter
|
|
|
|
ChooseResampler(const int dst_channels, const int src_rate, const int dst_rate)
|
|
|
|
{
|
|
|
|
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; \
|
|
|
|
}
|
|
|
|
|
|
|
|
if (upsample) {
|
|
|
|
if (multiple) {
|
|
|
|
PICK_CHANNEL_FILTER(Upsample, Multiple);
|
|
|
|
} else {
|
|
|
|
PICK_CHANNEL_FILTER(Upsample, Arbitrary);
|
|
|
|
}
|
|
|
|
} else {
|
|
|
|
if (multiple) {
|
|
|
|
PICK_CHANNEL_FILTER(Downsample, Multiple);
|
2015-06-21 17:33:46 +02:00
|
|
|
} else {
|
2017-01-06 01:12:20 +01:00
|
|
|
PICK_CHANNEL_FILTER(Downsample, Arbitrary);
|
2015-06-21 17:33:46 +02:00
|
|
|
}
|
2017-01-06 01:12:20 +01:00
|
|
|
}
|
|
|
|
|
|
|
|
#undef PICK_CHANNEL_FILTER
|
|
|
|
|
|
|
|
return filter;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, const int dst_channels,
|
|
|
|
const int src_rate, const int dst_rate)
|
|
|
|
{
|
|
|
|
SDL_AudioFilter filter;
|
|
|
|
|
|
|
|
if (src_rate == dst_rate) {
|
|
|
|
return 0; /* no conversion necessary. */
|
|
|
|
}
|
|
|
|
|
|
|
|
filter = ChooseResampler(dst_channels, src_rate, dst_rate);
|
|
|
|
if (filter == NULL) {
|
|
|
|
return SDL_SetError("No conversion available for these rates");
|
|
|
|
}
|
2015-06-21 17:33:46 +02:00
|
|
|
|
2017-01-06 01:12:20 +01:00
|
|
|
/* 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);
|
2015-06-21 17:33:46 +02:00
|
|
|
}
|
|
|
|
|
2017-01-06 01:12:20 +01:00
|
|
|
return 1; /* added a converter. */
|
2015-06-21 17:33:46 +02:00
|
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
/* 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");
|
|
|
|
}
|
|
|
|
|
2017-01-06 08:53:46 +01:00
|
|
|
/* Make sure we zero out the audio conversion before error checking */
|
|
|
|
SDL_zerop(cvt);
|
|
|
|
|
2015-06-21 17:33:46 +02:00
|
|
|
/* 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");
|
|
|
|
}
|
2016-11-05 09:52:28 +01:00
|
|
|
#if DEBUG_CONVERT
|
2015-06-21 17:33:46 +02:00
|
|
|
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 */
|
|
|
|
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 07:34:38 +01:00
|
|
|
/* 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
|
2017-01-06 01:12:20 +01:00
|
|
|
buffer is likely to be CPU cache-friendly, avoiding the
|
2016-11-05 07:34:38 +01:00
|
|
|
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. */
|
|
|
|
|
2017-01-06 08:53:46 +01:00
|
|
|
/* see if we can skip float conversion entirely. */
|
|
|
|
if (src_rate == dst_rate && src_channels == dst_channels) {
|
|
|
|
if (src_fmt == dst_fmt) {
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* just a byteswap needed? */
|
|
|
|
if ((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;
|
|
|
|
}
|
2016-11-05 07:34:38 +01:00
|
|
|
}
|
|
|
|
|
2015-06-21 17:33:46 +02:00
|
|
|
/* Convert data types, if necessary. Updates (cvt). */
|
2017-01-06 08:53:46 +01:00
|
|
|
if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
|
2015-06-21 17:33:46 +02:00
|
|
|
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). */
|
2017-01-06 08:53:46 +01:00
|
|
|
if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) < 0) {
|
2015-06-21 17:33:46 +02:00
|
|
|
return -1; /* shouldn't happen, but just in case... */
|
|
|
|
}
|
|
|
|
|
2017-01-06 01:12:20 +01:00
|
|
|
/* Move to final data type. */
|
2017-01-06 08:53:46 +01:00
|
|
|
if (SDL_BuildAudioTypeCVTFromFloat(cvt, dst_fmt) < 0) {
|
2016-11-05 07:34:38 +01:00
|
|
|
return -1; /* shouldn't happen, but just in case... */
|
2015-06-21 17:33:46 +02:00
|
|
|
}
|
2016-11-05 07:34:38 +01:00
|
|
|
|
|
|
|
cvt->needed = (cvt->filter_index != 0);
|
2015-06-21 17:33:46 +02:00
|
|
|
return (cvt->needed);
|
|
|
|
}
|
|
|
|
|
2017-01-06 11:16:26 +01:00
|
|
|
typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen);
|
|
|
|
typedef void (*SDL_ResetAudioStreamResamplerFunc)(SDL_AudioStream *stream);
|
|
|
|
typedef void (*SDL_CleanupAudioStreamResamplerFunc)(SDL_AudioStream *stream);
|
2017-01-06 01:29:38 +01:00
|
|
|
|
|
|
|
struct SDL_AudioStream
|
|
|
|
{
|
|
|
|
SDL_AudioCVT cvt_before_resampling;
|
|
|
|
SDL_AudioCVT cvt_after_resampling;
|
|
|
|
SDL_DataQueue *queue;
|
|
|
|
Uint8 *work_buffer;
|
|
|
|
int work_buffer_len;
|
|
|
|
Uint8 *resample_buffer;
|
|
|
|
int resample_buffer_len;
|
|
|
|
int src_sample_frame_size;
|
|
|
|
SDL_AudioFormat src_format;
|
|
|
|
Uint8 src_channels;
|
|
|
|
int src_rate;
|
|
|
|
int dst_sample_frame_size;
|
|
|
|
SDL_AudioFormat dst_format;
|
|
|
|
Uint8 dst_channels;
|
|
|
|
int dst_rate;
|
|
|
|
double rate_incr;
|
|
|
|
Uint8 pre_resample_channels;
|
|
|
|
int packetlen;
|
2017-01-06 11:16:26 +01:00
|
|
|
void *resampler_state;
|
|
|
|
SDL_ResampleAudioStreamFunc resampler_func;
|
|
|
|
SDL_ResetAudioStreamResamplerFunc reset_resampler_func;
|
|
|
|
SDL_CleanupAudioStreamResamplerFunc cleanup_resampler_func;
|
2017-01-06 01:29:38 +01:00
|
|
|
};
|
|
|
|
|
2017-01-06 11:16:26 +01:00
|
|
|
#ifdef HAVE_LIBSAMPLERATE
|
|
|
|
|
|
|
|
typedef struct
|
|
|
|
{
|
|
|
|
void *SRC_lib;
|
|
|
|
|
|
|
|
SRC_STATE* (*src_new)(int converter_type, int channels, int *error);
|
|
|
|
int (*src_process)(SRC_STATE *state, SRC_DATA *data);
|
|
|
|
int (*src_reset)(SRC_STATE *state);
|
|
|
|
SRC_STATE* (*src_delete)(SRC_STATE *state);
|
|
|
|
const char* (*src_strerror)(int error);
|
|
|
|
|
|
|
|
SRC_STATE *SRC_state;
|
|
|
|
} SDL_AudioStreamResamplerState_SRC;
|
|
|
|
|
|
|
|
static SDL_bool
|
|
|
|
LoadLibSampleRate(SDL_AudioStreamResamplerState_SRC *state)
|
|
|
|
{
|
|
|
|
#ifdef LIBSAMPLERATE_DYNAMIC
|
|
|
|
state->SRC_lib = SDL_LoadObject(LIBSAMPLERATE_DYNAMIC);
|
|
|
|
if (!state->SRC_lib) {
|
|
|
|
return SDL_FALSE;
|
|
|
|
}
|
|
|
|
#endif
|
|
|
|
|
|
|
|
state->src_new = (SRC_STATE* (*)(int converter_type, int channels, int *error))SDL_LoadFunction(state->SRC_lib, "src_new");
|
|
|
|
state->src_process = (int (*)(SRC_STATE *state, SRC_DATA *data))SDL_LoadFunction(state->SRC_lib, "src_process");
|
|
|
|
state->src_reset = (int(*)(SRC_STATE *state))SDL_LoadFunction(state->SRC_lib, "src_reset");
|
|
|
|
state->src_delete = (SRC_STATE* (*)(SRC_STATE *state))SDL_LoadFunction(state->SRC_lib, "src_delete");
|
|
|
|
state->src_strerror = (const char* (*)(int error))SDL_LoadFunction(state->SRC_lib, "src_strerror");
|
|
|
|
if (!state->src_new || !state->src_process || !state->src_reset || !state->src_delete || !state->src_strerror) {
|
|
|
|
return SDL_FALSE;
|
|
|
|
}
|
|
|
|
return SDL_TRUE;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
|
|
|
|
{
|
|
|
|
SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC*)stream->resampler_state;
|
|
|
|
SRC_DATA data;
|
|
|
|
int result;
|
|
|
|
|
|
|
|
data.data_in = inbuf;
|
|
|
|
data.input_frames = inbuflen / ( sizeof(float) * stream->pre_resample_channels );
|
|
|
|
data.input_frames_used = 0;
|
|
|
|
|
|
|
|
data.data_out = outbuf;
|
|
|
|
data.output_frames = outbuflen / (sizeof(float) * stream->pre_resample_channels);
|
|
|
|
|
|
|
|
data.end_of_input = 0;
|
|
|
|
data.src_ratio = stream->rate_incr;
|
|
|
|
|
|
|
|
result = state->src_process(state->SRC_state, &data);
|
|
|
|
if (result != 0) {
|
|
|
|
SDL_SetError("src_process() failed: %s", state->src_strerror(result));
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* If this fails, we need to store them off somewhere */
|
|
|
|
SDL_assert(data.input_frames_used == data.input_frames);
|
|
|
|
|
|
|
|
return data.output_frames_gen * (sizeof(float) * stream->pre_resample_channels);
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
SDL_ResetAudioStreamResampler_SRC(SDL_AudioStream *stream)
|
|
|
|
{
|
|
|
|
SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC*)stream->resampler_state;
|
|
|
|
state->src_reset(state->SRC_state);
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
SDL_CleanupAudioStreamResampler_SRC(SDL_AudioStream *stream)
|
|
|
|
{
|
|
|
|
SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC*)stream->resampler_state;
|
|
|
|
if (state) {
|
|
|
|
if (state->SRC_lib) {
|
|
|
|
SDL_UnloadObject(state->SRC_lib);
|
|
|
|
}
|
|
|
|
state->src_delete(state->SRC_state);
|
|
|
|
SDL_free(state);
|
|
|
|
}
|
|
|
|
|
|
|
|
stream->resampler_state = NULL;
|
|
|
|
stream->resampler_func = NULL;
|
|
|
|
stream->reset_resampler_func = NULL;
|
|
|
|
stream->cleanup_resampler_func = NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
static SDL_bool
|
|
|
|
SetupLibSampleRateResampling(SDL_AudioStream *stream)
|
|
|
|
{
|
|
|
|
int result;
|
|
|
|
|
|
|
|
SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC *)SDL_calloc(1, sizeof(*state));
|
|
|
|
if (!state) {
|
|
|
|
return SDL_FALSE;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (!LoadLibSampleRate(state)) {
|
|
|
|
SDL_free(state);
|
|
|
|
return SDL_FALSE;
|
|
|
|
}
|
|
|
|
|
|
|
|
stream->resampler_state = state;
|
|
|
|
stream->resampler_func = SDL_ResampleAudioStream_SRC;
|
|
|
|
stream->reset_resampler_func = SDL_ResetAudioStreamResampler_SRC;
|
|
|
|
stream->cleanup_resampler_func = SDL_CleanupAudioStreamResampler_SRC;
|
|
|
|
|
|
|
|
state->SRC_state = state->src_new(SRC_SINC_FASTEST, stream->pre_resample_channels, &result);
|
|
|
|
if (!state->SRC_state) {
|
|
|
|
SDL_SetError("src_new() failed: %s", state->src_strerror(result));
|
|
|
|
SDL_CleanupAudioStreamResampler_SRC(stream);
|
|
|
|
return SDL_FALSE;
|
|
|
|
}
|
|
|
|
return SDL_TRUE;
|
|
|
|
}
|
|
|
|
|
|
|
|
#endif /* HAVE_LIBSAMPLERATE */
|
|
|
|
|
|
|
|
typedef struct
|
|
|
|
{
|
|
|
|
SDL_bool resampler_seeded;
|
|
|
|
float resampler_state[8];
|
|
|
|
} SDL_AudioStreamResamplerState;
|
|
|
|
|
|
|
|
static int
|
|
|
|
SDL_ResampleAudioStream(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
|
|
|
|
{
|
|
|
|
/* !!! FIXME: this resampler sucks, but not much worse than our usual resampler. :) */ /* ... :( */
|
|
|
|
SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
|
|
|
|
const int chans = (int)stream->pre_resample_channels;
|
|
|
|
const int framelen = chans * sizeof(float);
|
|
|
|
const int total = (inbuflen / framelen);
|
|
|
|
const int finalpos = total - chans;
|
|
|
|
const double src_incr = 1.0 / stream->rate_incr;
|
|
|
|
double idx = 0.0;
|
|
|
|
float *dst = outbuf;
|
|
|
|
float last_sample[SDL_arraysize(state->resampler_state)];
|
|
|
|
int consumed = 0;
|
|
|
|
int i;
|
|
|
|
|
|
|
|
SDL_assert(chans <= SDL_arraysize(last_sample));
|
|
|
|
SDL_assert((inbuflen % framelen) == 0);
|
|
|
|
|
|
|
|
if (!state->resampler_seeded) {
|
|
|
|
for (i = 0; i < chans; i++) {
|
|
|
|
state->resampler_state[i] = inbuf[i];
|
|
|
|
}
|
|
|
|
state->resampler_seeded = SDL_TRUE;
|
|
|
|
}
|
|
|
|
|
|
|
|
for (i = 0; i < chans; i++) {
|
|
|
|
last_sample[i] = state->resampler_state[i];
|
|
|
|
}
|
|
|
|
|
|
|
|
while (consumed < total) {
|
|
|
|
const int pos = ((int)idx) * chans;
|
|
|
|
const float *src = &inbuf[(pos >= finalpos) ? finalpos : pos];
|
|
|
|
SDL_assert(dst < (outbuf + (outbuflen / framelen)));
|
|
|
|
for (i = 0; i < chans; i++) {
|
|
|
|
const float val = *(src++);
|
|
|
|
*(dst++) = (val + last_sample[i]) * 0.5f;
|
|
|
|
last_sample[i] = val;
|
|
|
|
}
|
|
|
|
consumed = pos + chans;
|
|
|
|
idx += src_incr;
|
|
|
|
}
|
|
|
|
|
|
|
|
for (i = 0; i < chans; i++) {
|
|
|
|
state->resampler_state[i] = last_sample[i];
|
|
|
|
}
|
|
|
|
|
|
|
|
return (int)((dst - outbuf) * sizeof(float));
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
SDL_ResetAudioStreamResampler(SDL_AudioStream *stream)
|
|
|
|
{
|
|
|
|
SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
|
|
|
|
state->resampler_seeded = SDL_FALSE;
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
SDL_CleanupAudioStreamResampler(SDL_AudioStream *stream)
|
|
|
|
{
|
|
|
|
SDL_free(stream->resampler_state);
|
|
|
|
}
|
|
|
|
|
2017-01-06 01:29:38 +01:00
|
|
|
SDL_AudioStream *SDL_NewAudioStream(const SDL_AudioFormat src_format,
|
|
|
|
const Uint8 src_channels,
|
|
|
|
const int src_rate,
|
|
|
|
const SDL_AudioFormat dst_format,
|
|
|
|
const Uint8 dst_channels,
|
|
|
|
const int dst_rate)
|
|
|
|
{
|
|
|
|
const int packetlen = 4096; /* !!! FIXME: good enough for now. */
|
|
|
|
Uint8 pre_resample_channels;
|
|
|
|
SDL_AudioStream *retval;
|
|
|
|
|
|
|
|
retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
|
|
|
|
if (!retval) {
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* If increasing channels, do it after resampling, since we'd just
|
|
|
|
do more work to resample duplicate channels. If we're decreasing, do
|
|
|
|
it first so we resample the interpolated data instead of interpolating
|
|
|
|
the resampled data (!!! FIXME: decide if that works in practice, though!). */
|
|
|
|
pre_resample_channels = SDL_min(src_channels, dst_channels);
|
|
|
|
|
|
|
|
retval->src_sample_frame_size = SDL_AUDIO_BITSIZE(src_format) * src_channels;
|
|
|
|
retval->src_format = src_format;
|
|
|
|
retval->src_channels = src_channels;
|
|
|
|
retval->src_rate = src_rate;
|
|
|
|
retval->dst_sample_frame_size = SDL_AUDIO_BITSIZE(dst_format) * dst_channels;
|
|
|
|
retval->dst_format = dst_format;
|
|
|
|
retval->dst_channels = dst_channels;
|
|
|
|
retval->dst_rate = dst_rate;
|
|
|
|
retval->pre_resample_channels = pre_resample_channels;
|
|
|
|
retval->packetlen = packetlen;
|
|
|
|
retval->rate_incr = ((double) dst_rate) / ((double) src_rate);
|
|
|
|
|
|
|
|
/* Not resampling? It's an easy conversion (and maybe not even that!). */
|
|
|
|
if (src_rate == dst_rate) {
|
|
|
|
retval->cvt_before_resampling.needed = SDL_FALSE;
|
|
|
|
retval->cvt_before_resampling.len_mult = 1;
|
2017-01-06 11:16:26 +01:00
|
|
|
if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
|
|
|
|
SDL_FreeAudioStream(retval);
|
2017-01-06 01:29:38 +01:00
|
|
|
return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
|
|
|
|
}
|
|
|
|
} else {
|
|
|
|
/* Don't resample at first. Just get us to Float32 format. */
|
|
|
|
/* !!! FIXME: convert to int32 on devices without hardware float. */
|
2017-01-06 11:16:26 +01:00
|
|
|
if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) < 0) {
|
|
|
|
SDL_FreeAudioStream(retval);
|
2017-01-06 01:29:38 +01:00
|
|
|
return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
|
|
|
|
}
|
|
|
|
|
2017-01-06 11:16:26 +01:00
|
|
|
#ifdef HAVE_LIBSAMPLERATE
|
|
|
|
SetupLibSampleRateResampling(retval);
|
|
|
|
#endif
|
|
|
|
|
|
|
|
if (!retval->resampler_func) {
|
|
|
|
retval->resampler_state = SDL_calloc(1, sizeof(SDL_AudioStreamResamplerState));
|
|
|
|
if (!retval->resampler_state) {
|
|
|
|
SDL_FreeAudioStream(retval);
|
|
|
|
SDL_OutOfMemory();
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
retval->resampler_func = SDL_ResampleAudioStream;
|
|
|
|
retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
|
|
|
|
retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
|
|
|
|
}
|
|
|
|
|
2017-01-06 01:29:38 +01:00
|
|
|
/* Convert us to the final format after resampling. */
|
2017-01-06 11:16:26 +01:00
|
|
|
if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
|
|
|
|
SDL_FreeAudioStream(retval);
|
2017-01-06 01:29:38 +01:00
|
|
|
return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2);
|
|
|
|
if (!retval->queue) {
|
2017-01-06 11:16:26 +01:00
|
|
|
SDL_FreeAudioStream(retval);
|
2017-01-06 01:29:38 +01:00
|
|
|
return NULL; /* SDL_NewDataQueue should have called SDL_SetError. */
|
|
|
|
}
|
|
|
|
|
|
|
|
return retval;
|
|
|
|
}
|
|
|
|
|
|
|
|
static Uint8 *
|
|
|
|
EnsureBufferSize(Uint8 **buf, int *len, const int newlen)
|
|
|
|
{
|
|
|
|
if (*len < newlen) {
|
|
|
|
void *ptr = SDL_realloc(*buf, newlen);
|
|
|
|
if (!ptr) {
|
|
|
|
SDL_OutOfMemory();
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
*buf = (Uint8 *) ptr;
|
|
|
|
*len = newlen;
|
|
|
|
}
|
|
|
|
return *buf;
|
|
|
|
}
|
|
|
|
|
|
|
|
int
|
|
|
|
SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, const Uint32 _buflen)
|
|
|
|
{
|
|
|
|
int buflen = (int) _buflen;
|
|
|
|
|
|
|
|
if (!stream) {
|
|
|
|
return SDL_InvalidParamError("stream");
|
|
|
|
} else if (!buf) {
|
|
|
|
return SDL_InvalidParamError("buf");
|
|
|
|
} else if (buflen == 0) {
|
|
|
|
return 0; /* nothing to do. */
|
|
|
|
} else if ((buflen % stream->src_sample_frame_size) != 0) {
|
|
|
|
return SDL_SetError("Can't add partial sample frames");
|
|
|
|
}
|
|
|
|
|
|
|
|
if (stream->cvt_before_resampling.needed) {
|
|
|
|
const int workbuflen = buflen * stream->cvt_before_resampling.len_mult; /* will be "* 1" if not needed */
|
|
|
|
Uint8 *workbuf = EnsureBufferSize(&stream->work_buffer, &stream->work_buffer_len, workbuflen);
|
|
|
|
if (workbuf == NULL) {
|
|
|
|
return -1; /* probably out of memory. */
|
|
|
|
}
|
|
|
|
SDL_memcpy(workbuf, buf, buflen);
|
|
|
|
stream->cvt_before_resampling.buf = workbuf;
|
|
|
|
stream->cvt_before_resampling.len = buflen;
|
|
|
|
if (SDL_ConvertAudio(&stream->cvt_before_resampling) == -1) {
|
|
|
|
return -1; /* uhoh! */
|
|
|
|
}
|
|
|
|
buf = workbuf;
|
|
|
|
buflen = stream->cvt_before_resampling.len_cvt;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (stream->dst_rate != stream->src_rate) {
|
|
|
|
const int workbuflen = buflen * ((int) SDL_ceil(stream->rate_incr));
|
|
|
|
float *workbuf = (float *) EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
|
|
|
|
if (workbuf == NULL) {
|
|
|
|
return -1; /* probably out of memory. */
|
|
|
|
}
|
2017-01-06 11:16:26 +01:00
|
|
|
buflen = stream->resampler_func(stream, (float *) buf, buflen, workbuf, workbuflen);
|
2017-01-06 01:29:38 +01:00
|
|
|
buf = workbuf;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (stream->cvt_after_resampling.needed) {
|
|
|
|
const int workbuflen = buflen * stream->cvt_before_resampling.len_mult; /* will be "* 1" if not needed */
|
|
|
|
Uint8 *workbuf;
|
|
|
|
|
|
|
|
if (buf == stream->resample_buffer) {
|
|
|
|
workbuf = EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
|
|
|
|
} else {
|
|
|
|
const int inplace = (buf == stream->work_buffer);
|
|
|
|
workbuf = EnsureBufferSize(&stream->work_buffer, &stream->work_buffer_len, workbuflen);
|
|
|
|
if (workbuf && !inplace) {
|
|
|
|
SDL_memcpy(workbuf, buf, buflen);
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
if (workbuf == NULL) {
|
|
|
|
return -1; /* probably out of memory. */
|
|
|
|
}
|
|
|
|
|
|
|
|
stream->cvt_after_resampling.buf = workbuf;
|
|
|
|
stream->cvt_after_resampling.len = buflen;
|
|
|
|
if (SDL_ConvertAudio(&stream->cvt_after_resampling) == -1) {
|
|
|
|
return -1; /* uhoh! */
|
|
|
|
}
|
|
|
|
buf = workbuf;
|
|
|
|
buflen = stream->cvt_after_resampling.len_cvt;
|
|
|
|
}
|
|
|
|
|
|
|
|
return SDL_WriteToDataQueue(stream->queue, buf, buflen);
|
|
|
|
}
|
|
|
|
|
|
|
|
void
|
|
|
|
SDL_AudioStreamClear(SDL_AudioStream *stream)
|
|
|
|
{
|
|
|
|
if (!stream) {
|
|
|
|
SDL_InvalidParamError("stream");
|
|
|
|
} else {
|
|
|
|
SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
|
2017-01-07 03:23:51 +01:00
|
|
|
if (stream->reset_resampler_func) {
|
|
|
|
stream->reset_resampler_func(stream);
|
|
|
|
}
|
2017-01-06 01:29:38 +01:00
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
/* get converted/resampled data from the stream */
|
|
|
|
int
|
2017-01-06 07:02:58 +01:00
|
|
|
SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, const Uint32 len)
|
2017-01-06 01:29:38 +01:00
|
|
|
{
|
|
|
|
if (!stream) {
|
|
|
|
return SDL_InvalidParamError("stream");
|
|
|
|
} else if (!buf) {
|
|
|
|
return SDL_InvalidParamError("buf");
|
|
|
|
} else if (len == 0) {
|
|
|
|
return 0; /* nothing to do. */
|
|
|
|
} else if ((len % stream->dst_sample_frame_size) != 0) {
|
|
|
|
return SDL_SetError("Can't request partial sample frames");
|
|
|
|
}
|
|
|
|
|
2017-01-06 07:02:58 +01:00
|
|
|
return (int) SDL_ReadFromDataQueue(stream->queue, buf, len);
|
2017-01-06 01:29:38 +01:00
|
|
|
}
|
|
|
|
|
|
|
|
/* number of converted/resampled bytes available */
|
|
|
|
int
|
|
|
|
SDL_AudioStreamAvailable(SDL_AudioStream *stream)
|
|
|
|
{
|
|
|
|
return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* dispose of a stream */
|
|
|
|
void
|
|
|
|
SDL_FreeAudioStream(SDL_AudioStream *stream)
|
|
|
|
{
|
|
|
|
if (stream) {
|
2017-01-06 11:16:26 +01:00
|
|
|
if (stream->cleanup_resampler_func) {
|
|
|
|
stream->cleanup_resampler_func(stream);
|
|
|
|
}
|
2017-01-06 01:29:38 +01:00
|
|
|
SDL_FreeDataQueue(stream->queue);
|
|
|
|
SDL_free(stream->work_buffer);
|
|
|
|
SDL_free(stream->resample_buffer);
|
|
|
|
SDL_free(stream);
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2015-06-21 17:33:46 +02:00
|
|
|
/* vi: set ts=4 sw=4 expandtab: */
|
2016-11-05 07:34:38 +01:00
|
|
|
|