src/audio/SDL_audiocvt.c
author Ryan C. Gordon <icculus@icculus.org>
Sun, 22 Jan 2017 20:27:48 -0500
changeset 10834 336efe4fc373
parent 10833 86f6353f1aae
child 10835 0e9e7a128391
permissions -rw-r--r--
audio: Special case for resampling stereo AUDIO_S16SYS audio data.

This is a fairly common case, so we avoid the conversion to/from float here.
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/*
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  Simple DirectMedia Layer
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  Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
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  This software is provided 'as-is', without any express or implied
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  warranty.  In no event will the authors be held liable for any damages
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  arising from the use of this software.
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  Permission is granted to anyone to use this software for any purpose,
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  including commercial applications, and to alter it and redistribute it
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  freely, subject to the following restrictions:
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  1. The origin of this software must not be misrepresented; you must not
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     claim that you wrote the original software. If you use this software
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     in a product, an acknowledgment in the product documentation would be
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     appreciated but is not required.
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  2. Altered source versions must be plainly marked as such, and must not be
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     misrepresented as being the original software.
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  3. This notice may not be removed or altered from any source distribution.
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*/
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#include "../SDL_internal.h"
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/* 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_loadso.h"
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#include "SDL_assert.h"
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#include "../SDL_dataqueue.h"
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/* !!! FIXME: wire this up to the configure script, etc. */
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#include "SDL_cpuinfo.h"
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#define HAVE_SSE3_INTRINSICS 0
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#if HAVE_SSE3_INTRINSICS
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#include <pmmintrin.h>
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#endif
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#if HAVE_SSE3_INTRINSICS
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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_ConvertStereoToMono_SSE3(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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    float *dst = (float *) cvt->buf;
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    const float *src = dst;
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    int i = cvt->len_cvt / 8;
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    LOG_DEBUG_CONVERT("stereo", "mono (using SSE3)");
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    SDL_assert(format == AUDIO_F32SYS);
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    /* We can only do this if dst is aligned to 16 bytes; since src is the
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       same pointer and it moves by 2, it can't be forcibly aligned. */
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    if ((((size_t) dst) & 15) == 0) {
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        /* Aligned! Do SSE blocks as long as we have 16 bytes available. */
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        const __m128 divby2 = _mm_set1_ps(0.5f);
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        while (i >= 4) {   /* 4 * float32 */
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            _mm_store_ps(dst, _mm_mul_ps(_mm_hadd_ps(_mm_load_ps(src), _mm_load_ps(src+4)), divby2));
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            i -= 4; src += 8; dst += 4;
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        }
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    }
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    /* Finish off any leftovers with scalar operations. */
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    while (i) {
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        *dst = (src[0] + src[1]) * 0.5f;
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        dst++; i--; src += 2;
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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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#endif
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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_ConvertStereoToMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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    float *dst = (float *) cvt->buf;
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    const float *src = dst;
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    int i;
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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++) = (src[0] + src[1]) * 0.5f;
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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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/* Convert from 5.1 to stereo. Average left and right, discard subwoofer. */
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static void SDLCALL
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SDL_Convert51ToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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    float *dst = (float *) cvt->buf;
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    const float *src = dst;
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    int i;
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    LOG_DEBUG_CONVERT("5.1", "stereo");
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    SDL_assert(format == AUDIO_F32SYS);
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    /* this assumes FL+FR+FC+subwoof+BL+BR layout. */
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    for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 2) {
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        const double front_center = (double) src[2];
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        dst[0] = (float) ((src[0] + front_center + src[4]) / 3.0);  /* left */
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        dst[1] = (float) ((src[1] + front_center + src[5]) / 3.0);  /* right */
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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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/* Convert from 5.1 to quad */
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static void SDLCALL
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SDL_Convert51ToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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    float *dst = (float *) cvt->buf;
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    const float *src = dst;
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    int i;
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    LOG_DEBUG_CONVERT("5.1", "quad");
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    SDL_assert(format == AUDIO_F32SYS);
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    /* assumes quad is FL+FR+BL+BR layout and 5.1 is FL+FR+FC+subwoof+BL+BR */
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    for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 4) {
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        /* FIXME: this is a good candidate for SIMD. */
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        const double front_center = (double) src[2];
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        dst[0] = (float) ((src[0] + front_center) * 0.5);  /* FL */
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        dst[1] = (float) ((src[1] + front_center) * 0.5);  /* FR */
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        dst[2] = (float) ((src[4] + front_center) * 0.5);  /* BL */
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        dst[3] = (float) ((src[5] + front_center) * 0.5);  /* BR */
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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_ConvertMonoToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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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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    int i;
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    LOG_DEBUG_CONVERT("mono", "stereo");
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    SDL_assert(format == AUDIO_F32SYS);
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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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    }
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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_ConvertStereoTo51(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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    int i;
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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 + rf) * 0.5f;
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        dst[0] = lf + (lf - ce);  /* FL */
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        dst[1] = rf + (rf - ce);  /* FR */
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        dst[2] = ce;  /* FC */
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        dst[3] = ce;  /* !!! FIXME: wrong! This is the subwoofer. */
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        dst[4] = lf;  /* BL */
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        dst[5] = rf;  /* BR */
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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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/* Duplicate a stereo channel to a pseudo-4.0 stream */
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static void SDLCALL
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SDL_ConvertStereoToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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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;
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    int i;
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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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        dst[0] = lf;  /* FL */
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        dst[1] = rf;  /* FR */
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        dst[2] = lf;  /* BL */
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        dst[3] = rf;  /* BR */
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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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static int
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SDL_ResampleAudioSimple(const int chans, const double rate_incr,
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                        float *last_sample, const float *inbuf,
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                        const int inbuflen, float *outbuf, const int outbuflen)
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{
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    const int framelen = chans * (int)sizeof (float);
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    const int total = (inbuflen / framelen);
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    const int finalpos = (total * chans) - chans;
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    const int dest_samples = (int)(((double)total) * rate_incr);
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    const double src_incr = 1.0 / rate_incr;
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    float *dst;
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    double idx;
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    int i;
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    SDL_assert((dest_samples * framelen) <= outbuflen);
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    SDL_assert((inbuflen % framelen) == 0);
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    if (rate_incr > 1.0) {
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        float *target = (outbuf + chans);
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        const float *earlier_sample = &inbuf[finalpos];
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        float final_sample[8];
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        dst = outbuf + (dest_samples * chans);
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        idx = (double) total;
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        /* save this off so we can correctly maintain state between runs. */
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        SDL_memcpy(final_sample, &inbuf[finalpos], framelen);
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        while (dst > target) {
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            const int pos = ((int) idx) * chans;
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            const float *src = &inbuf[pos];
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            SDL_assert(pos >= 0.0);
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            for (i = chans - 1; i >= 0; i--) {
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                const float val = *(--src);
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                *(--dst) = (val + earlier_sample[i]) * 0.5f;
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            }
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            earlier_sample = src;
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            idx -= src_incr;
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        }
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        /* do last sample, interpolated against previous run's state. */
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        for (i = chans - 1; i >= 0; i--) {
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            const float val = inbuf[i];
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            *(--dst) = (val + last_sample[i]) * 0.5f;
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        }
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        SDL_memcpy(last_sample, final_sample, framelen);
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        dst = (outbuf + (dest_samples * chans)) - 1;
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    } else {
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        float *target = (outbuf + (dest_samples * chans));
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        dst = outbuf;
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        idx = 0.0;
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        while (dst < target) {
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            const int pos = ((int) idx) * chans;
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            const float *src = &inbuf[pos];
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            SDL_assert(pos <= finalpos);
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            for (i = 0; i < chans; i++) {
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                const float val = *(src++);
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                *(dst++) = (val + last_sample[i]) * 0.5f;
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                last_sample[i] = val;
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            }
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            idx += src_incr;
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        }
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    }
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    return (int) ((dst - outbuf) * ((int) sizeof (float)));
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}
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/* We keep one special-case fast path around for an extremely common audio format. */
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static int
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SDL_ResampleAudioSimple_si16_c2(const double rate_incr,
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                        Sint16 *last_sample, const Sint16 *inbuf,
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                        const int inbuflen, Sint16 *outbuf, const int outbuflen)
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{
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    const int chans = 2;
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    const int framelen = 4;  /* stereo 16 bit */
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    const int total = (inbuflen / framelen);
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    const int finalpos = (total * chans) - chans;
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    const int dest_samples = (int)(((double)total) * rate_incr);
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    const double src_incr = 1.0 / rate_incr;
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    Sint16 *dst;
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    double idx;
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    SDL_assert((dest_samples * framelen) <= outbuflen);
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    SDL_assert((inbuflen % framelen) == 0);
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    if (rate_incr > 1.0) {
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        Sint16 *target = (outbuf + chans);
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        const Sint16 final_right = inbuf[finalpos+1];
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        const Sint16 final_left = inbuf[finalpos];
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        Sint16 earlier_right = inbuf[finalpos-1];
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        Sint16 earlier_left = inbuf[finalpos-2];
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        dst = outbuf + (dest_samples * chans);
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        idx = (double) total;
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        while (dst > target) {
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            const int pos = ((int) idx) * chans;
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            const Sint16 *src = &inbuf[pos];
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            const Sint16 right = *(--src);
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            const Sint16 left = *(--src);
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            SDL_assert(pos >= 0.0);
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            *(--dst) = (((Sint32) right) + ((Sint32) earlier_right)) >> 1;
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            *(--dst) = (((Sint32) left) + ((Sint32) earlier_left)) >> 1;
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            earlier_right = right;
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            earlier_left = left;
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            idx -= src_incr;
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        }
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        /* do last sample, interpolated against previous run's state. */
icculus@10834
   345
        *(--dst) = (((Sint32) inbuf[1]) + ((Sint32) last_sample[1])) >> 1;
icculus@10834
   346
        *(--dst) = (((Sint32) inbuf[0]) + ((Sint32) last_sample[0])) >> 1;
icculus@10834
   347
        last_sample[1] = final_right;
icculus@10834
   348
        last_sample[0] = final_left;
icculus@10834
   349
icculus@10834
   350
        dst = (outbuf + (dest_samples * chans)) - 1;
icculus@10834
   351
    } else {
icculus@10834
   352
        Sint16 *target = (outbuf + (dest_samples * chans));
icculus@10834
   353
        dst = outbuf;
icculus@10834
   354
        idx = 0.0;
icculus@10834
   355
        while (dst < target) {
icculus@10834
   356
            const int pos = ((int) idx) * chans;
icculus@10834
   357
            const Sint16 *src = &inbuf[pos];
icculus@10834
   358
            const Sint16 left = *(src++);
icculus@10834
   359
            const Sint16 right = *(src++);
icculus@10834
   360
            SDL_assert(pos <= finalpos);
icculus@10834
   361
            *(dst++) = (((Sint32) left) + ((Sint32) last_sample[0])) >> 1;
icculus@10834
   362
            *(dst++) = (((Sint32) right) + ((Sint32) last_sample[1])) >> 1;
icculus@10834
   363
            last_sample[0] = left;
icculus@10834
   364
            last_sample[1] = right;
icculus@10834
   365
            idx += src_incr;
icculus@10834
   366
        }
icculus@10834
   367
    }
icculus@10834
   368
icculus@10834
   369
    return (int) ((dst - outbuf) * ((int) sizeof (Sint16)));
icculus@10834
   370
}
icculus@10834
   371
icculus@10834
   372
static void SDLCALL
icculus@10834
   373
SDL_ResampleCVT_si16_c2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
icculus@10834
   374
{
icculus@10834
   375
    const Sint16 *src = (const Sint16 *) cvt->buf;
icculus@10834
   376
    const int srclen = cvt->len_cvt;
icculus@10834
   377
    Sint16 *dst = (Sint16 *) (cvt->buf + srclen);
icculus@10834
   378
    const int dstlen = (cvt->len * cvt->len_mult) - srclen;
icculus@10834
   379
    Sint16 state[2] = { src[0], src[1] };
icculus@10834
   380
icculus@10834
   381
    SDL_assert(format == AUDIO_S16SYS);
icculus@10834
   382
icculus@10834
   383
    cvt->len_cvt = SDL_ResampleAudioSimple_si16_c2(cvt->rate_incr, state, src, srclen, dst, dstlen);
icculus@10834
   384
    if (cvt->filters[++cvt->filter_index]) {
icculus@10834
   385
        cvt->filters[cvt->filter_index](cvt, format);
icculus@10834
   386
    }
icculus@10834
   387
}
icculus@10834
   388
slouken@0
   389
slouken@1895
   390
int
slouken@1895
   391
SDL_ConvertAudio(SDL_AudioCVT * cvt)
slouken@0
   392
{
icculus@3021
   393
    /* !!! FIXME: (cvt) should be const; stack-copy it here. */
icculus@3021
   394
    /* !!! FIXME: (actually, we can't...len_cvt needs to be updated. Grr.) */
icculus@3021
   395
slouken@1895
   396
    /* Make sure there's data to convert */
slouken@1895
   397
    if (cvt->buf == NULL) {
icculus@10575
   398
        return SDL_SetError("No buffer allocated for conversion");
slouken@1895
   399
    }
icculus@10575
   400
slouken@1895
   401
    /* Return okay if no conversion is necessary */
slouken@1895
   402
    cvt->len_cvt = cvt->len;
slouken@1895
   403
    if (cvt->filters[0] == NULL) {
icculus@10575
   404
        return 0;
slouken@1895
   405
    }
slouken@0
   406
slouken@1895
   407
    /* Set up the conversion and go! */
slouken@1895
   408
    cvt->filter_index = 0;
slouken@1895
   409
    cvt->filters[0] (cvt, cvt->src_format);
icculus@10575
   410
    return 0;
slouken@0
   411
}
slouken@0
   412
icculus@10575
   413
static void SDLCALL
icculus@10575
   414
SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format)
icculus@10575
   415
{
slouken@10579
   416
#if DEBUG_CONVERT
slouken@10579
   417
    printf("Converting byte order\n");
slouken@10579
   418
#endif
icculus@1982
   419
icculus@10575
   420
    switch (SDL_AUDIO_BITSIZE(format)) {
icculus@10575
   421
        #define CASESWAP(b) \
icculus@10575
   422
            case b: { \
icculus@10575
   423
                Uint##b *ptr = (Uint##b *) cvt->buf; \
icculus@10575
   424
                int i; \
icculus@10575
   425
                for (i = cvt->len_cvt / sizeof (*ptr); i; --i, ++ptr) { \
icculus@10575
   426
                    *ptr = SDL_Swap##b(*ptr); \
icculus@10575
   427
                } \
icculus@10575
   428
                break; \
icculus@10575
   429
            }
icculus@1982
   430
icculus@10575
   431
        CASESWAP(16);
icculus@10575
   432
        CASESWAP(32);
icculus@10575
   433
        CASESWAP(64);
icculus@10575
   434
icculus@10575
   435
        #undef CASESWAP
icculus@10575
   436
icculus@10575
   437
        default: SDL_assert(!"unhandled byteswap datatype!"); break;
icculus@10575
   438
    }
icculus@10575
   439
icculus@10575
   440
    if (cvt->filters[++cvt->filter_index]) {
icculus@10575
   441
        /* flip endian flag for data. */
icculus@10575
   442
        if (format & SDL_AUDIO_MASK_ENDIAN) {
icculus@10575
   443
            format &= ~SDL_AUDIO_MASK_ENDIAN;
icculus@10575
   444
        } else {
icculus@10575
   445
            format |= SDL_AUDIO_MASK_ENDIAN;
icculus@10575
   446
        }
icculus@10575
   447
        cvt->filters[cvt->filter_index](cvt, format);
icculus@10575
   448
    }
icculus@1982
   449
}
icculus@1982
   450
icculus@1982
   451
icculus@1982
   452
static int
icculus@10575
   453
SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
icculus@1982
   454
{
icculus@10575
   455
    int retval = 0;  /* 0 == no conversion necessary. */
icculus@1982
   456
icculus@10575
   457
    if ((SDL_AUDIO_ISBIGENDIAN(src_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
icculus@10575
   458
        cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
icculus@10575
   459
        retval = 1;  /* added a converter. */
icculus@10575
   460
    }
icculus@1982
   461
icculus@10575
   462
    if (!SDL_AUDIO_ISFLOAT(src_fmt)) {
icculus@10576
   463
        const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
icculus@10576
   464
        const Uint16 dst_bitsize = 32;
icculus@10575
   465
        SDL_AudioFilter filter = NULL;
icculus@10576
   466
icculus@10575
   467
        switch (src_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
icculus@10575
   468
            case AUDIO_S8: filter = SDL_Convert_S8_to_F32; break;
icculus@10575
   469
            case AUDIO_U8: filter = SDL_Convert_U8_to_F32; break;
icculus@10575
   470
            case AUDIO_S16: filter = SDL_Convert_S16_to_F32; break;
philipp@10591
   471
            case AUDIO_U16: filter = SDL_Convert_U16_to_F32; break;
icculus@10575
   472
            case AUDIO_S32: filter = SDL_Convert_S32_to_F32; break;
icculus@10575
   473
            default: SDL_assert(!"Unexpected audio format!"); break;
icculus@1982
   474
        }
icculus@1982
   475
icculus@10575
   476
        if (!filter) {
icculus@10575
   477
            return SDL_SetError("No conversion available for these formats");
icculus@10575
   478
        }
icculus@10575
   479
icculus@1982
   480
        cvt->filters[cvt->filter_index++] = filter;
icculus@1982
   481
        if (src_bitsize < dst_bitsize) {
icculus@1982
   482
            const int mult = (dst_bitsize / src_bitsize);
icculus@1982
   483
            cvt->len_mult *= mult;
icculus@1982
   484
            cvt->len_ratio *= mult;
icculus@1982
   485
        } else if (src_bitsize > dst_bitsize) {
icculus@1982
   486
            cvt->len_ratio /= (src_bitsize / dst_bitsize);
icculus@1982
   487
        }
icculus@10576
   488
icculus@10575
   489
        retval = 1;  /* added a converter. */
icculus@1982
   490
    }
icculus@1982
   491
icculus@10575
   492
    return retval;
icculus@1982
   493
}
icculus@1982
   494
icculus@10575
   495
static int
icculus@10575
   496
SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt)
icculus@10575
   497
{
icculus@10575
   498
    int retval = 0;  /* 0 == no conversion necessary. */
icculus@3021
   499
icculus@10575
   500
    if (!SDL_AUDIO_ISFLOAT(dst_fmt)) {
icculus@10577
   501
        const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
icculus@10577
   502
        const Uint16 src_bitsize = 32;
icculus@10575
   503
        SDL_AudioFilter filter = NULL;
icculus@10575
   504
        switch (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
icculus@10575
   505
            case AUDIO_S8: filter = SDL_Convert_F32_to_S8; break;
icculus@10575
   506
            case AUDIO_U8: filter = SDL_Convert_F32_to_U8; break;
icculus@10575
   507
            case AUDIO_S16: filter = SDL_Convert_F32_to_S16; break;
philipp@10591
   508
            case AUDIO_U16: filter = SDL_Convert_F32_to_U16; break;
icculus@10575
   509
            case AUDIO_S32: filter = SDL_Convert_F32_to_S32; break;
icculus@10575
   510
            default: SDL_assert(!"Unexpected audio format!"); break;
icculus@10575
   511
        }
slouken@2716
   512
icculus@10575
   513
        if (!filter) {
icculus@10575
   514
            return SDL_SetError("No conversion available for these formats");
icculus@10575
   515
        }
icculus@10575
   516
icculus@10575
   517
        cvt->filters[cvt->filter_index++] = filter;
icculus@10575
   518
        if (src_bitsize < dst_bitsize) {
icculus@10575
   519
            const int mult = (dst_bitsize / src_bitsize);
icculus@10575
   520
            cvt->len_mult *= mult;
icculus@10575
   521
            cvt->len_ratio *= mult;
icculus@10575
   522
        } else if (src_bitsize > dst_bitsize) {
icculus@10575
   523
            cvt->len_ratio /= (src_bitsize / dst_bitsize);
icculus@10575
   524
        }
icculus@10575
   525
        retval = 1;  /* added a converter. */
icculus@10575
   526
    }
icculus@10575
   527
icculus@10575
   528
    if ((SDL_AUDIO_ISBIGENDIAN(dst_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
icculus@10575
   529
        cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
icculus@10575
   530
        retval = 1;  /* added a converter. */
icculus@10575
   531
    }
icculus@10575
   532
icculus@10575
   533
    return retval;
icculus@3021
   534
}
slouken@2716
   535
icculus@10799
   536
static void
icculus@10799
   537
SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format)
icculus@10799
   538
{
icculus@10799
   539
    const float *src = (const float *) cvt->buf;
icculus@10799
   540
    const int srclen = cvt->len_cvt;
icculus@10833
   541
    float *dst = (float *) cvt->buf;
icculus@10833
   542
    const int dstlen = (cvt->len * cvt->len_mult);
icculus@10804
   543
    float state[8];
icculus@10756
   544
icculus@10799
   545
    SDL_assert(format == AUDIO_F32SYS);
icculus@10799
   546
slouken@10805
   547
    SDL_memcpy(state, src, chans*sizeof(*src));
icculus@10799
   548
icculus@10804
   549
    cvt->len_cvt = SDL_ResampleAudioSimple(chans, cvt->rate_incr, state, src, srclen, dst, dstlen);
icculus@10799
   550
    if (cvt->filters[++cvt->filter_index]) {
icculus@10799
   551
        cvt->filters[cvt->filter_index](cvt, format);
icculus@10799
   552
    }
icculus@10799
   553
}
icculus@10799
   554
icculus@10799
   555
/* !!! FIXME: We only have this macro salsa because SDL_AudioCVT doesn't
icculus@10799
   556
   !!! FIXME:  store channel info, so we have to have function entry
icculus@10799
   557
   !!! FIXME:  points for each supported channel count and multiple
icculus@10799
   558
   !!! FIXME:  vs arbitrary. When we rev the ABI, clean this up. */
icculus@10756
   559
#define RESAMPLER_FUNCS(chans) \
icculus@10756
   560
    static void SDLCALL \
icculus@10799
   561
    SDL_ResampleCVT_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
icculus@10799
   562
        SDL_ResampleCVT(cvt, chans, format); \
icculus@10756
   563
    }
icculus@10756
   564
RESAMPLER_FUNCS(1)
icculus@10756
   565
RESAMPLER_FUNCS(2)
icculus@10756
   566
RESAMPLER_FUNCS(4)
icculus@10756
   567
RESAMPLER_FUNCS(6)
icculus@10756
   568
RESAMPLER_FUNCS(8)
icculus@10756
   569
#undef RESAMPLER_FUNCS
icculus@10756
   570
icculus@10799
   571
static SDL_AudioFilter
icculus@10799
   572
ChooseCVTResampler(const int dst_channels)
icculus@3021
   573
{
icculus@10799
   574
    switch (dst_channels) {
icculus@10799
   575
        case 1: return SDL_ResampleCVT_c1;
icculus@10799
   576
        case 2: return SDL_ResampleCVT_c2;
icculus@10799
   577
        case 4: return SDL_ResampleCVT_c4;
icculus@10799
   578
        case 6: return SDL_ResampleCVT_c6;
icculus@10799
   579
        case 8: return SDL_ResampleCVT_c8;
icculus@10799
   580
        default: break;
icculus@3021
   581
    }
slouken@2716
   582
icculus@10799
   583
    return NULL;
icculus@10756
   584
}
icculus@10575
   585
icculus@3021
   586
static int
icculus@10756
   587
SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, const int dst_channels,
icculus@10756
   588
                          const int src_rate, const int dst_rate)
icculus@3021
   589
{
icculus@10756
   590
    SDL_AudioFilter filter;
icculus@3021
   591
icculus@10756
   592
    if (src_rate == dst_rate) {
icculus@10756
   593
        return 0;  /* no conversion necessary. */
slouken@2716
   594
    }
slouken@2716
   595
icculus@10799
   596
    filter = ChooseCVTResampler(dst_channels);
icculus@10756
   597
    if (filter == NULL) {
icculus@10756
   598
        return SDL_SetError("No conversion available for these rates");
icculus@10756
   599
    }
icculus@10756
   600
icculus@10756
   601
    /* Update (cvt) with filter details... */
icculus@10756
   602
    cvt->filters[cvt->filter_index++] = filter;
icculus@10756
   603
    if (src_rate < dst_rate) {
icculus@10756
   604
        const double mult = ((double) dst_rate) / ((double) src_rate);
icculus@10756
   605
        cvt->len_mult *= (int) SDL_ceil(mult);
icculus@10756
   606
        cvt->len_ratio *= mult;
icculus@10756
   607
    } else {
icculus@10756
   608
        cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate);
icculus@10756
   609
    }
icculus@10756
   610
icculus@10756
   611
    return 1;               /* added a converter. */
slouken@2716
   612
}
icculus@1982
   613
icculus@1982
   614
icculus@1982
   615
/* Creates a set of audio filters to convert from one format to another.
icculus@1982
   616
   Returns -1 if the format conversion is not supported, 0 if there's
icculus@1982
   617
   no conversion needed, or 1 if the audio filter is set up.
slouken@0
   618
*/
slouken@1895
   619
slouken@1895
   620
int
slouken@1895
   621
SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
icculus@1982
   622
                  SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate,
icculus@1982
   623
                  SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
slouken@0
   624
{
aschiffler@6819
   625
    /* Sanity check target pointer */
aschiffler@6819
   626
    if (cvt == NULL) {
icculus@7037
   627
        return SDL_InvalidParamError("cvt");
aschiffler@6819
   628
    }
slouken@7191
   629
slouken@10767
   630
    /* Make sure we zero out the audio conversion before error checking */
slouken@10767
   631
    SDL_zerop(cvt);
slouken@10767
   632
slouken@3491
   633
    /* there are no unsigned types over 16 bits, so catch this up front. */
icculus@1982
   634
    if ((SDL_AUDIO_BITSIZE(src_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(src_fmt))) {
icculus@7037
   635
        return SDL_SetError("Invalid source format");
icculus@1982
   636
    }
icculus@1982
   637
    if ((SDL_AUDIO_BITSIZE(dst_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(dst_fmt))) {
icculus@7037
   638
        return SDL_SetError("Invalid destination format");
icculus@1982
   639
    }
icculus@3021
   640
icculus@3021
   641
    /* prevent possible divisions by zero, etc. */
aschiffler@6819
   642
    if ((src_channels == 0) || (dst_channels == 0)) {
icculus@7037
   643
        return SDL_SetError("Source or destination channels is zero");
aschiffler@6819
   644
    }
icculus@3021
   645
    if ((src_rate == 0) || (dst_rate == 0)) {
icculus@7037
   646
        return SDL_SetError("Source or destination rate is zero");
icculus@3021
   647
    }
slouken@10579
   648
#if DEBUG_CONVERT
icculus@1982
   649
    printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
slouken@1985
   650
           src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
slouken@1985
   651
#endif
icculus@1982
   652
slouken@1895
   653
    /* Start off with no conversion necessary */
icculus@1982
   654
    cvt->src_format = src_fmt;
icculus@1982
   655
    cvt->dst_format = dst_fmt;
slouken@1895
   656
    cvt->needed = 0;
slouken@1895
   657
    cvt->filter_index = 0;
slouken@1895
   658
    cvt->filters[0] = NULL;
slouken@1895
   659
    cvt->len_mult = 1;
slouken@1895
   660
    cvt->len_ratio = 1.0;
icculus@3021
   661
    cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
slouken@0
   662
icculus@10834
   663
    /* SDL now favors float32 as its preferred internal format, and considers
icculus@10834
   664
       everything else to be a degenerate case that we might have to make
icculus@10834
   665
       multiple passes over the data to convert to and from float32 as
icculus@10834
   666
       necessary. That being said, we keep one special case around for
icculus@10834
   667
       efficiency: stereo data in Sint16 format, in the native byte order,
icculus@10834
   668
       that only needs resampling. This is likely to be the most popular
icculus@10834
   669
       legacy format, that apps, hardware and the OS are likely to be able
icculus@10834
   670
       to process directly, so we handle this one case directly without
icculus@10834
   671
       unnecessary conversions. This means that apps on embedded devices
icculus@10834
   672
       without floating point hardware should consider aiming for this
icculus@10834
   673
       format as well. */
icculus@10834
   674
    if ((src_channels == 2) && (dst_channels == 2) && (src_fmt == AUDIO_S16SYS) && (dst_fmt == AUDIO_S16SYS) && (src_rate != dst_rate)) {
icculus@10834
   675
        cvt->needed = 1;
icculus@10834
   676
        cvt->filters[cvt->filter_index++] = SDL_ResampleCVT_si16_c2;
icculus@10834
   677
        if (src_rate < dst_rate) {
icculus@10834
   678
            const double mult = ((double) dst_rate) / ((double) src_rate);
icculus@10834
   679
            cvt->len_mult *= (int) SDL_ceil(mult);
icculus@10834
   680
            cvt->len_ratio *= mult;
icculus@10834
   681
        } else {
icculus@10834
   682
            cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate);
icculus@10834
   683
        }
icculus@10834
   684
        return 1;
icculus@10834
   685
    }
icculus@10834
   686
icculus@10575
   687
    /* Type conversion goes like this now:
icculus@10575
   688
        - byteswap to CPU native format first if necessary.
icculus@10575
   689
        - convert to native Float32 if necessary.
icculus@10575
   690
        - resample and change channel count if necessary.
icculus@10575
   691
        - convert back to native format.
icculus@10575
   692
        - byteswap back to foreign format if necessary.
icculus@10575
   693
icculus@10575
   694
       The expectation is we can process data faster in float32
icculus@10575
   695
       (possibly with SIMD), and making several passes over the same
icculus@10756
   696
       buffer is likely to be CPU cache-friendly, avoiding the
icculus@10575
   697
       biggest performance hit in modern times. Previously we had
icculus@10575
   698
       (script-generated) custom converters for every data type and
icculus@10575
   699
       it was a bloat on SDL compile times and final library size. */
icculus@10575
   700
slouken@10767
   701
    /* see if we can skip float conversion entirely. */
slouken@10767
   702
    if (src_rate == dst_rate && src_channels == dst_channels) {
slouken@10767
   703
        if (src_fmt == dst_fmt) {
slouken@10767
   704
            return 0;
slouken@10767
   705
        }
slouken@10767
   706
slouken@10767
   707
        /* just a byteswap needed? */
slouken@10767
   708
        if ((src_fmt & ~SDL_AUDIO_MASK_ENDIAN) == (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN)) {
slouken@10767
   709
            cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
slouken@10767
   710
            cvt->needed = 1;
slouken@10767
   711
            return 1;
slouken@10767
   712
        }
icculus@10575
   713
    }
icculus@10575
   714
icculus@1982
   715
    /* Convert data types, if necessary. Updates (cvt). */
slouken@10767
   716
    if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
slouken@1985
   717
        return -1;              /* shouldn't happen, but just in case... */
icculus@3021
   718
    }
slouken@0
   719
icculus@1982
   720
    /* Channel conversion */
slouken@1895
   721
    if (src_channels != dst_channels) {
slouken@1895
   722
        if ((src_channels == 1) && (dst_channels > 1)) {
icculus@10793
   723
            cvt->filters[cvt->filter_index++] = SDL_ConvertMonoToStereo;
slouken@1895
   724
            cvt->len_mult *= 2;
slouken@1895
   725
            src_channels = 2;
slouken@1895
   726
            cvt->len_ratio *= 2;
slouken@1895
   727
        }
slouken@1895
   728
        if ((src_channels == 2) && (dst_channels == 6)) {
icculus@10793
   729
            cvt->filters[cvt->filter_index++] = SDL_ConvertStereoTo51;
slouken@1895
   730
            src_channels = 6;
slouken@1895
   731
            cvt->len_mult *= 3;
slouken@1895
   732
            cvt->len_ratio *= 3;
slouken@1895
   733
        }
slouken@1895
   734
        if ((src_channels == 2) && (dst_channels == 4)) {
icculus@10793
   735
            cvt->filters[cvt->filter_index++] = SDL_ConvertStereoToQuad;
slouken@1895
   736
            src_channels = 4;
slouken@1895
   737
            cvt->len_mult *= 2;
slouken@1895
   738
            cvt->len_ratio *= 2;
slouken@1895
   739
        }
slouken@1895
   740
        while ((src_channels * 2) <= dst_channels) {
icculus@10793
   741
            cvt->filters[cvt->filter_index++] = SDL_ConvertMonoToStereo;
slouken@1895
   742
            cvt->len_mult *= 2;
slouken@1895
   743
            src_channels *= 2;
slouken@1895
   744
            cvt->len_ratio *= 2;
slouken@1895
   745
        }
slouken@1895
   746
        if ((src_channels == 6) && (dst_channels <= 2)) {
icculus@10793
   747
            cvt->filters[cvt->filter_index++] = SDL_Convert51ToStereo;
slouken@1895
   748
            src_channels = 2;
slouken@1895
   749
            cvt->len_ratio /= 3;
slouken@1895
   750
        }
slouken@1895
   751
        if ((src_channels == 6) && (dst_channels == 4)) {
icculus@10793
   752
            cvt->filters[cvt->filter_index++] = SDL_Convert51ToQuad;
slouken@1895
   753
            src_channels = 4;
slouken@1895
   754
            cvt->len_ratio /= 2;
slouken@1895
   755
        }
slouken@1895
   756
        /* This assumes that 4 channel audio is in the format:
slouken@1895
   757
           Left {front/back} + Right {front/back}
slouken@1895
   758
           so converting to L/R stereo works properly.
slouken@1895
   759
         */
slouken@1895
   760
        while (((src_channels % 2) == 0) &&
slouken@1895
   761
               ((src_channels / 2) >= dst_channels)) {
icculus@10832
   762
            SDL_AudioFilter filter = NULL;
icculus@10832
   763
icculus@10832
   764
            #if HAVE_SSE3_INTRINSICS
icculus@10832
   765
            if (SDL_HasSSE3()) {
icculus@10832
   766
                filter = SDL_ConvertStereoToMono_SSE3;
icculus@10832
   767
            }
icculus@10832
   768
            #endif
icculus@10832
   769
icculus@10832
   770
            if (!filter) {
icculus@10832
   771
                filter = SDL_ConvertStereoToMono;
icculus@10832
   772
            }
icculus@10832
   773
icculus@10832
   774
            cvt->filters[cvt->filter_index++] = filter;
icculus@10832
   775
slouken@1895
   776
            src_channels /= 2;
slouken@1895
   777
            cvt->len_ratio /= 2;
slouken@1895
   778
        }
slouken@1895
   779
        if (src_channels != dst_channels) {
slouken@1895
   780
            /* Uh oh.. */ ;
slouken@1895
   781
        }
slouken@1895
   782
    }
slouken@0
   783
icculus@3021
   784
    /* Do rate conversion, if necessary. Updates (cvt). */
slouken@10767
   785
    if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) < 0) {
icculus@3021
   786
        return -1;              /* shouldn't happen, but just in case... */
slouken@2716
   787
    }
slouken@2716
   788
icculus@10756
   789
    /* Move to final data type. */
slouken@10767
   790
    if (SDL_BuildAudioTypeCVTFromFloat(cvt, dst_fmt) < 0) {
icculus@10575
   791
        return -1;              /* shouldn't happen, but just in case... */
slouken@1895
   792
    }
icculus@10575
   793
icculus@10575
   794
    cvt->needed = (cvt->filter_index != 0);
slouken@1895
   795
    return (cvt->needed);
slouken@0
   796
}
slouken@1895
   797
slouken@10773
   798
typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen);
slouken@10773
   799
typedef void (*SDL_ResetAudioStreamResamplerFunc)(SDL_AudioStream *stream);
slouken@10773
   800
typedef void (*SDL_CleanupAudioStreamResamplerFunc)(SDL_AudioStream *stream);
icculus@10757
   801
icculus@10757
   802
struct SDL_AudioStream
icculus@10757
   803
{
icculus@10757
   804
    SDL_AudioCVT cvt_before_resampling;
icculus@10757
   805
    SDL_AudioCVT cvt_after_resampling;
icculus@10757
   806
    SDL_DataQueue *queue;
icculus@10757
   807
    Uint8 *work_buffer;
icculus@10757
   808
    int work_buffer_len;
icculus@10757
   809
    Uint8 *resample_buffer;
icculus@10757
   810
    int resample_buffer_len;
icculus@10757
   811
    int src_sample_frame_size;
icculus@10757
   812
    SDL_AudioFormat src_format;
icculus@10757
   813
    Uint8 src_channels;
icculus@10757
   814
    int src_rate;
icculus@10757
   815
    int dst_sample_frame_size;
icculus@10757
   816
    SDL_AudioFormat dst_format;
icculus@10757
   817
    Uint8 dst_channels;
icculus@10757
   818
    int dst_rate;
icculus@10757
   819
    double rate_incr;
icculus@10757
   820
    Uint8 pre_resample_channels;
slouken@10773
   821
    int packetlen;
slouken@10773
   822
    void *resampler_state;
slouken@10773
   823
    SDL_ResampleAudioStreamFunc resampler_func;
slouken@10773
   824
    SDL_ResetAudioStreamResamplerFunc reset_resampler_func;
slouken@10773
   825
    SDL_CleanupAudioStreamResamplerFunc cleanup_resampler_func;
slouken@10773
   826
};
slouken@10773
   827
slouken@10777
   828
#ifdef HAVE_LIBSAMPLERATE_H
slouken@10773
   829
static int
slouken@10773
   830
SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
slouken@10773
   831
{
icculus@10799
   832
    const int framelen = sizeof(float) * stream->pre_resample_channels;
icculus@10790
   833
    SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
slouken@10773
   834
    SRC_DATA data;
slouken@10773
   835
    int result;
slouken@10773
   836
slouken@10777
   837
    data.data_in = (float *)inbuf; /* Older versions of libsamplerate had a non-const pointer, but didn't write to it */
icculus@10799
   838
    data.input_frames = inbuflen / framelen;
slouken@10773
   839
    data.input_frames_used = 0;
slouken@10773
   840
slouken@10773
   841
    data.data_out = outbuf;
icculus@10799
   842
    data.output_frames = outbuflen / framelen;
slouken@10773
   843
slouken@10773
   844
    data.end_of_input = 0;
slouken@10773
   845
    data.src_ratio = stream->rate_incr;
slouken@10773
   846
icculus@10790
   847
    result = SRC_src_process(state, &data);
slouken@10773
   848
    if (result != 0) {
icculus@10790
   849
        SDL_SetError("src_process() failed: %s", SRC_src_strerror(result));
slouken@10773
   850
        return 0;
slouken@10773
   851
    }
slouken@10773
   852
slouken@10773
   853
    /* If this fails, we need to store them off somewhere */
slouken@10773
   854
    SDL_assert(data.input_frames_used == data.input_frames);
slouken@10773
   855
slouken@10773
   856
    return data.output_frames_gen * (sizeof(float) * stream->pre_resample_channels);
slouken@10773
   857
}
slouken@10773
   858
slouken@10773
   859
static void
slouken@10773
   860
SDL_ResetAudioStreamResampler_SRC(SDL_AudioStream *stream)
slouken@10773
   861
{
icculus@10790
   862
    SRC_src_reset((SRC_STATE *)stream->resampler_state);
slouken@10773
   863
}
slouken@10773
   864
slouken@10773
   865
static void
slouken@10773
   866
SDL_CleanupAudioStreamResampler_SRC(SDL_AudioStream *stream)
slouken@10773
   867
{
icculus@10790
   868
    SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
slouken@10773
   869
    if (state) {
icculus@10790
   870
        SRC_src_delete(state);
slouken@10773
   871
    }
slouken@10773
   872
slouken@10773
   873
    stream->resampler_state = NULL;
slouken@10773
   874
    stream->resampler_func = NULL;
slouken@10773
   875
    stream->reset_resampler_func = NULL;
slouken@10773
   876
    stream->cleanup_resampler_func = NULL;
slouken@10773
   877
}
slouken@10773
   878
slouken@10773
   879
static SDL_bool
slouken@10773
   880
SetupLibSampleRateResampling(SDL_AudioStream *stream)
slouken@10773
   881
{
icculus@10790
   882
    int result = 0;
icculus@10790
   883
    SRC_STATE *state = NULL;
slouken@10773
   884
icculus@10790
   885
    if (SRC_available) {
icculus@10790
   886
        state = SRC_src_new(SRC_SINC_FASTEST, stream->pre_resample_channels, &result);
icculus@10790
   887
        if (!state) {
icculus@10790
   888
            SDL_SetError("src_new() failed: %s", SRC_src_strerror(result));
icculus@10790
   889
        }
slouken@10773
   890
    }
slouken@10773
   891
icculus@10790
   892
    if (!state) {
icculus@10790
   893
        SDL_CleanupAudioStreamResampler_SRC(stream);
slouken@10773
   894
        return SDL_FALSE;
slouken@10773
   895
    }
slouken@10773
   896
slouken@10773
   897
    stream->resampler_state = state;
slouken@10773
   898
    stream->resampler_func = SDL_ResampleAudioStream_SRC;
slouken@10773
   899
    stream->reset_resampler_func = SDL_ResetAudioStreamResampler_SRC;
slouken@10773
   900
    stream->cleanup_resampler_func = SDL_CleanupAudioStreamResampler_SRC;
slouken@10773
   901
slouken@10773
   902
    return SDL_TRUE;
slouken@10773
   903
}
icculus@10790
   904
#endif /* HAVE_LIBSAMPLERATE_H */
slouken@10773
   905
slouken@10773
   906
slouken@10773
   907
typedef struct
slouken@10773
   908
{
icculus@10757
   909
    SDL_bool resampler_seeded;
icculus@10757
   910
    float resampler_state[8];
slouken@10773
   911
} SDL_AudioStreamResamplerState;
slouken@10773
   912
slouken@10773
   913
static int
slouken@10773
   914
SDL_ResampleAudioStream(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
slouken@10773
   915
{
slouken@10773
   916
    SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
slouken@10773
   917
    const int chans = (int)stream->pre_resample_channels;
slouken@10773
   918
icculus@10799
   919
    SDL_assert(chans <= SDL_arraysize(state->resampler_state));
slouken@10773
   920
slouken@10773
   921
    if (!state->resampler_seeded) {
icculus@10799
   922
        int i;
slouken@10773
   923
        for (i = 0; i < chans; i++) {
slouken@10773
   924
            state->resampler_state[i] = inbuf[i];
slouken@10773
   925
        }
slouken@10773
   926
        state->resampler_seeded = SDL_TRUE;
slouken@10773
   927
    }
slouken@10773
   928
icculus@10799
   929
    return SDL_ResampleAudioSimple(chans, stream->rate_incr, state->resampler_state, inbuf, inbuflen, outbuf, outbuflen);
slouken@10773
   930
}
slouken@10773
   931
slouken@10773
   932
static void
slouken@10773
   933
SDL_ResetAudioStreamResampler(SDL_AudioStream *stream)
slouken@10773
   934
{
slouken@10773
   935
    SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
slouken@10773
   936
    state->resampler_seeded = SDL_FALSE;
slouken@10773
   937
}
slouken@10773
   938
slouken@10773
   939
static void
slouken@10773
   940
SDL_CleanupAudioStreamResampler(SDL_AudioStream *stream)
slouken@10773
   941
{
slouken@10773
   942
    SDL_free(stream->resampler_state);
slouken@10773
   943
}
icculus@10757
   944
icculus@10789
   945
SDL_AudioStream *
icculus@10789
   946
SDL_NewAudioStream(const SDL_AudioFormat src_format,
icculus@10789
   947
                   const Uint8 src_channels,
icculus@10789
   948
                   const int src_rate,
icculus@10789
   949
                   const SDL_AudioFormat dst_format,
icculus@10789
   950
                   const Uint8 dst_channels,
icculus@10789
   951
                   const int dst_rate)
icculus@10757
   952
{
icculus@10757
   953
    const int packetlen = 4096;  /* !!! FIXME: good enough for now. */
icculus@10757
   954
    Uint8 pre_resample_channels;
icculus@10757
   955
    SDL_AudioStream *retval;
icculus@10757
   956
icculus@10757
   957
    retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
icculus@10757
   958
    if (!retval) {
icculus@10757
   959
        return NULL;
icculus@10757
   960
    }
icculus@10757
   961
icculus@10757
   962
    /* If increasing channels, do it after resampling, since we'd just
icculus@10757
   963
       do more work to resample duplicate channels. If we're decreasing, do
icculus@10757
   964
       it first so we resample the interpolated data instead of interpolating
icculus@10757
   965
       the resampled data (!!! FIXME: decide if that works in practice, though!). */
icculus@10757
   966
    pre_resample_channels = SDL_min(src_channels, dst_channels);
icculus@10757
   967
icculus@10757
   968
    retval->src_sample_frame_size = SDL_AUDIO_BITSIZE(src_format) * src_channels;
icculus@10757
   969
    retval->src_format = src_format;
icculus@10757
   970
    retval->src_channels = src_channels;
icculus@10757
   971
    retval->src_rate = src_rate;
icculus@10757
   972
    retval->dst_sample_frame_size = SDL_AUDIO_BITSIZE(dst_format) * dst_channels;
icculus@10757
   973
    retval->dst_format = dst_format;
icculus@10757
   974
    retval->dst_channels = dst_channels;
icculus@10757
   975
    retval->dst_rate = dst_rate;
icculus@10757
   976
    retval->pre_resample_channels = pre_resample_channels;
icculus@10757
   977
    retval->packetlen = packetlen;
icculus@10757
   978
    retval->rate_incr = ((double) dst_rate) / ((double) src_rate);
icculus@10757
   979
icculus@10757
   980
    /* Not resampling? It's an easy conversion (and maybe not even that!). */
icculus@10757
   981
    if (src_rate == dst_rate) {
icculus@10757
   982
        retval->cvt_before_resampling.needed = SDL_FALSE;
icculus@10757
   983
        retval->cvt_before_resampling.len_mult = 1;
slouken@10773
   984
        if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
slouken@10773
   985
            SDL_FreeAudioStream(retval);
icculus@10757
   986
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
icculus@10757
   987
        }
icculus@10757
   988
    } else {
icculus@10757
   989
        /* Don't resample at first. Just get us to Float32 format. */
icculus@10757
   990
        /* !!! FIXME: convert to int32 on devices without hardware float. */
slouken@10773
   991
        if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) < 0) {
slouken@10773
   992
            SDL_FreeAudioStream(retval);
icculus@10757
   993
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
icculus@10757
   994
        }
icculus@10757
   995
slouken@10777
   996
#ifdef HAVE_LIBSAMPLERATE_H
slouken@10773
   997
        SetupLibSampleRateResampling(retval);
slouken@10773
   998
#endif
slouken@10773
   999
slouken@10773
  1000
        if (!retval->resampler_func) {
slouken@10773
  1001
            retval->resampler_state = SDL_calloc(1, sizeof(SDL_AudioStreamResamplerState));
slouken@10773
  1002
            if (!retval->resampler_state) {
slouken@10773
  1003
                SDL_FreeAudioStream(retval);
slouken@10773
  1004
                SDL_OutOfMemory();
slouken@10773
  1005
                return NULL;
slouken@10773
  1006
            }
slouken@10773
  1007
            retval->resampler_func = SDL_ResampleAudioStream;
slouken@10773
  1008
            retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
slouken@10773
  1009
            retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
slouken@10773
  1010
        }
slouken@10773
  1011
icculus@10757
  1012
        /* Convert us to the final format after resampling. */
slouken@10773
  1013
        if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
slouken@10773
  1014
            SDL_FreeAudioStream(retval);
icculus@10757
  1015
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
icculus@10757
  1016
        }
icculus@10757
  1017
    }
icculus@10757
  1018
icculus@10757
  1019
    retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2);
icculus@10757
  1020
    if (!retval->queue) {
slouken@10773
  1021
        SDL_FreeAudioStream(retval);
icculus@10757
  1022
        return NULL;  /* SDL_NewDataQueue should have called SDL_SetError. */
icculus@10757
  1023
    }
icculus@10757
  1024
icculus@10757
  1025
    return retval;
icculus@10757
  1026
}
icculus@10757
  1027
icculus@10757
  1028
static Uint8 *
icculus@10757
  1029
EnsureBufferSize(Uint8 **buf, int *len, const int newlen)
icculus@10757
  1030
{
icculus@10757
  1031
    if (*len < newlen) {
icculus@10757
  1032
        void *ptr = SDL_realloc(*buf, newlen);
icculus@10757
  1033
        if (!ptr) {
icculus@10757
  1034
            SDL_OutOfMemory();
icculus@10757
  1035
            return NULL;
icculus@10757
  1036
        }
icculus@10757
  1037
        *buf = (Uint8 *) ptr;
icculus@10757
  1038
        *len = newlen;
icculus@10757
  1039
    }
icculus@10757
  1040
    return *buf;
icculus@10757
  1041
}
icculus@10757
  1042
icculus@10757
  1043
int
icculus@10757
  1044
SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, const Uint32 _buflen)
icculus@10757
  1045
{
icculus@10757
  1046
    int buflen = (int) _buflen;
icculus@10757
  1047
icculus@10757
  1048
    if (!stream) {
icculus@10757
  1049
        return SDL_InvalidParamError("stream");
icculus@10757
  1050
    } else if (!buf) {
icculus@10757
  1051
        return SDL_InvalidParamError("buf");
icculus@10757
  1052
    } else if (buflen == 0) {
icculus@10757
  1053
        return 0;  /* nothing to do. */
icculus@10757
  1054
    } else if ((buflen % stream->src_sample_frame_size) != 0) {
icculus@10757
  1055
        return SDL_SetError("Can't add partial sample frames");
icculus@10757
  1056
    }
icculus@10757
  1057
icculus@10757
  1058
    if (stream->cvt_before_resampling.needed) {
icculus@10757
  1059
        const int workbuflen = buflen * stream->cvt_before_resampling.len_mult;  /* will be "* 1" if not needed */
icculus@10757
  1060
        Uint8 *workbuf = EnsureBufferSize(&stream->work_buffer, &stream->work_buffer_len, workbuflen);
icculus@10757
  1061
        if (workbuf == NULL) {
icculus@10757
  1062
            return -1;  /* probably out of memory. */
icculus@10757
  1063
        }
icculus@10757
  1064
        SDL_memcpy(workbuf, buf, buflen);
icculus@10757
  1065
        stream->cvt_before_resampling.buf = workbuf;
icculus@10757
  1066
        stream->cvt_before_resampling.len = buflen;
icculus@10757
  1067
        if (SDL_ConvertAudio(&stream->cvt_before_resampling) == -1) {
icculus@10757
  1068
            return -1;   /* uhoh! */
icculus@10757
  1069
        }
icculus@10757
  1070
        buf = workbuf;
icculus@10757
  1071
        buflen = stream->cvt_before_resampling.len_cvt;
icculus@10757
  1072
    }
icculus@10757
  1073
icculus@10757
  1074
    if (stream->dst_rate != stream->src_rate) {
icculus@10757
  1075
        const int workbuflen = buflen * ((int) SDL_ceil(stream->rate_incr));
icculus@10757
  1076
        float *workbuf = (float *) EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
icculus@10757
  1077
        if (workbuf == NULL) {
icculus@10757
  1078
            return -1;  /* probably out of memory. */
icculus@10757
  1079
        }
slouken@10773
  1080
        buflen = stream->resampler_func(stream, (float *) buf, buflen, workbuf, workbuflen);
icculus@10757
  1081
        buf = workbuf;
icculus@10757
  1082
    }
icculus@10757
  1083
icculus@10757
  1084
    if (stream->cvt_after_resampling.needed) {
icculus@10757
  1085
        const int workbuflen = buflen * stream->cvt_before_resampling.len_mult;  /* will be "* 1" if not needed */
icculus@10757
  1086
        Uint8 *workbuf;
icculus@10757
  1087
icculus@10757
  1088
        if (buf == stream->resample_buffer) {
icculus@10757
  1089
            workbuf = EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
icculus@10757
  1090
        } else {
icculus@10757
  1091
            const int inplace = (buf == stream->work_buffer);
icculus@10757
  1092
            workbuf = EnsureBufferSize(&stream->work_buffer, &stream->work_buffer_len, workbuflen);
icculus@10757
  1093
            if (workbuf && !inplace) {
icculus@10757
  1094
                SDL_memcpy(workbuf, buf, buflen);
icculus@10757
  1095
            }
icculus@10757
  1096
        }
icculus@10757
  1097
icculus@10757
  1098
        if (workbuf == NULL) {
icculus@10757
  1099
            return -1;  /* probably out of memory. */
icculus@10757
  1100
        }
icculus@10757
  1101
icculus@10757
  1102
        stream->cvt_after_resampling.buf = workbuf;
icculus@10757
  1103
        stream->cvt_after_resampling.len = buflen;
icculus@10757
  1104
        if (SDL_ConvertAudio(&stream->cvt_after_resampling) == -1) {
icculus@10757
  1105
            return -1;   /* uhoh! */
icculus@10757
  1106
        }
icculus@10757
  1107
        buf = workbuf;
icculus@10757
  1108
        buflen = stream->cvt_after_resampling.len_cvt;
icculus@10757
  1109
    }
icculus@10757
  1110
icculus@10757
  1111
    return SDL_WriteToDataQueue(stream->queue, buf, buflen);
icculus@10757
  1112
}
icculus@10757
  1113
icculus@10757
  1114
void
icculus@10757
  1115
SDL_AudioStreamClear(SDL_AudioStream *stream)
icculus@10757
  1116
{
icculus@10757
  1117
    if (!stream) {
icculus@10757
  1118
        SDL_InvalidParamError("stream");
icculus@10757
  1119
    } else {
icculus@10757
  1120
        SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
icculus@10776
  1121
        if (stream->reset_resampler_func) {
icculus@10776
  1122
            stream->reset_resampler_func(stream);
icculus@10776
  1123
        }
icculus@10757
  1124
    }
icculus@10757
  1125
}
icculus@10757
  1126
icculus@10757
  1127
icculus@10757
  1128
/* get converted/resampled data from the stream */
icculus@10757
  1129
int
icculus@10764
  1130
SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, const Uint32 len)
icculus@10757
  1131
{
icculus@10757
  1132
    if (!stream) {
icculus@10757
  1133
        return SDL_InvalidParamError("stream");
icculus@10757
  1134
    } else if (!buf) {
icculus@10757
  1135
        return SDL_InvalidParamError("buf");
icculus@10757
  1136
    } else if (len == 0) {
icculus@10757
  1137
        return 0;  /* nothing to do. */
icculus@10757
  1138
    } else if ((len % stream->dst_sample_frame_size) != 0) {
icculus@10757
  1139
        return SDL_SetError("Can't request partial sample frames");
icculus@10757
  1140
    }
icculus@10757
  1141
icculus@10764
  1142
    return (int) SDL_ReadFromDataQueue(stream->queue, buf, len);
icculus@10757
  1143
}
icculus@10757
  1144
icculus@10757
  1145
/* number of converted/resampled bytes available */
icculus@10757
  1146
int
icculus@10757
  1147
SDL_AudioStreamAvailable(SDL_AudioStream *stream)
icculus@10757
  1148
{
icculus@10757
  1149
    return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
icculus@10757
  1150
}
icculus@10757
  1151
icculus@10757
  1152
/* dispose of a stream */
icculus@10757
  1153
void
icculus@10757
  1154
SDL_FreeAudioStream(SDL_AudioStream *stream)
icculus@10757
  1155
{
icculus@10757
  1156
    if (stream) {
slouken@10773
  1157
        if (stream->cleanup_resampler_func) {
slouken@10773
  1158
            stream->cleanup_resampler_func(stream);
slouken@10773
  1159
        }
icculus@10757
  1160
        SDL_FreeDataQueue(stream->queue);
icculus@10757
  1161
        SDL_free(stream->work_buffer);
icculus@10757
  1162
        SDL_free(stream->resample_buffer);
icculus@10757
  1163
        SDL_free(stream);
icculus@10757
  1164
    }
icculus@10757
  1165
}
icculus@10757
  1166
icculus@10575
  1167
/* vi: set ts=4 sw=4 expandtab: */
slouken@2716
  1168