src/audio/SDL_audiocvt.c
author Ryan C. Gordon <icculus@icculus.org>
Wed, 05 Jul 2017 12:04:37 -0400
changeset 11128 9dda3f3e9794
parent 11097 62a0a6e9b48b
child 11319 86b1fde471c6
permissions -rw-r--r--
audio: trying to pacify static analysis.
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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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#include "SDL_cpuinfo.h"
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#ifdef __SSE3__
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#define HAVE_SSE3_INTRINSICS 1
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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) {  /* upsample */
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        float *target = (outbuf + chans);
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        dst = outbuf + (dest_samples * chans);
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        idx = (double) total;
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        if (chans == 1) {
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            const float final_sample = inbuf[finalpos];
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            float earlier_sample = inbuf[finalpos];
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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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                const float val = *(--src);
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                SDL_assert(pos >= 0.0);
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                *(--dst) = (val + earlier_sample) * 0.5f;
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                earlier_sample = val;
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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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            *(--dst) = (inbuf[0] + last_sample[0]) * 0.5f;
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            *last_sample = final_sample;
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        } else if (chans == 2) {
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            const float final_sample2 = inbuf[finalpos+1];
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            const float final_sample1 = inbuf[finalpos];
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            float earlier_sample2 = inbuf[finalpos];
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            float earlier_sample1 = inbuf[finalpos-1];
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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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                const float val2 = *(--src);
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                const float val1 = *(--src);
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                SDL_assert(pos >= 0.0);
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                *(--dst) = (val2 + earlier_sample2) * 0.5f;
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                *(--dst) = (val1 + earlier_sample1) * 0.5f;
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                earlier_sample2 = val2;
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                earlier_sample1 = val1;
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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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            *(--dst) = (inbuf[1] + last_sample[1]) * 0.5f;
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            *(--dst) = (inbuf[0] + last_sample[0]) * 0.5f;
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            last_sample[1] = final_sample2;
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            last_sample[0] = final_sample1;
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        } else {
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            const float *earlier_sample = &inbuf[finalpos];
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            float final_sample[8];
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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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        }
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        dst = (outbuf + (dest_samples * chans));
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    } else {  /* downsample */
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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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        if (chans == 1) {
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            float last = *last_sample;
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            while (dst < target) {
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                const int pos = ((int) idx) * chans;
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                const float val = inbuf[pos];
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                SDL_assert(pos <= finalpos);
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                *(dst++) = (val + last) * 0.5f;
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                last = val;
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                idx += src_incr;
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            }
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            *last_sample = last;
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        } else if (chans == 2) {
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            float last1 = last_sample[0];
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            float last2 = last_sample[1];
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            while (dst < target) {
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                const int pos = ((int) idx) * chans;
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                const float val1 = inbuf[pos];
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                const float val2 = inbuf[pos+1];
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                SDL_assert(pos <= finalpos);
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                *(dst++) = (val1 + last1) * 0.5f;
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                *(dst++) = (val2 + last2) * 0.5f;
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   343
                last1 = val1;
icculus@10840
   344
                last2 = val2;
icculus@10840
   345
                idx += src_incr;
icculus@10840
   346
            }
icculus@10840
   347
            last_sample[0] = last1;
icculus@10840
   348
            last_sample[1] = last2;
icculus@10840
   349
        } else {
icculus@10840
   350
            while (dst < target) {
icculus@10840
   351
                const int pos = ((int) idx) * chans;
icculus@10840
   352
                const float *src = &inbuf[pos];
icculus@10840
   353
                SDL_assert(pos <= finalpos);
icculus@10840
   354
                for (i = 0; i < chans; i++) {
icculus@10840
   355
                    const float val = *(src++);
icculus@10840
   356
                    *(dst++) = (val + last_sample[i]) * 0.5f;
icculus@10840
   357
                    last_sample[i] = val;
icculus@10840
   358
                }
icculus@10840
   359
                idx += src_incr;
icculus@10840
   360
            }
icculus@10833
   361
        }
icculus@10799
   362
    }
icculus@10799
   363
icculus@10833
   364
    return (int) ((dst - outbuf) * ((int) sizeof (float)));
icculus@10799
   365
}
icculus@10799
   366
icculus@10834
   367
/* We keep one special-case fast path around for an extremely common audio format. */
icculus@10834
   368
static int
icculus@10834
   369
SDL_ResampleAudioSimple_si16_c2(const double rate_incr,
icculus@10834
   370
                        Sint16 *last_sample, const Sint16 *inbuf,
icculus@10834
   371
                        const int inbuflen, Sint16 *outbuf, const int outbuflen)
icculus@10834
   372
{
icculus@10834
   373
    const int chans = 2;
icculus@10834
   374
    const int framelen = 4;  /* stereo 16 bit */
icculus@10834
   375
    const int total = (inbuflen / framelen);
icculus@10834
   376
    const int finalpos = (total * chans) - chans;
icculus@10834
   377
    const int dest_samples = (int)(((double)total) * rate_incr);
icculus@10834
   378
    const double src_incr = 1.0 / rate_incr;
icculus@10834
   379
    Sint16 *dst;
icculus@10834
   380
    double idx;
icculus@10834
   381
icculus@10834
   382
    SDL_assert((dest_samples * framelen) <= outbuflen);
icculus@10834
   383
    SDL_assert((inbuflen % framelen) == 0);
icculus@10834
   384
icculus@10834
   385
    if (rate_incr > 1.0) {
icculus@10834
   386
        Sint16 *target = (outbuf + chans);
icculus@10834
   387
        const Sint16 final_right = inbuf[finalpos+1];
icculus@10834
   388
        const Sint16 final_left = inbuf[finalpos];
icculus@10834
   389
        Sint16 earlier_right = inbuf[finalpos-1];
icculus@10834
   390
        Sint16 earlier_left = inbuf[finalpos-2];
icculus@10834
   391
        dst = outbuf + (dest_samples * chans);
icculus@10834
   392
        idx = (double) total;
icculus@10834
   393
icculus@10834
   394
        while (dst > target) {
icculus@10834
   395
            const int pos = ((int) idx) * chans;
icculus@10834
   396
            const Sint16 *src = &inbuf[pos];
icculus@10834
   397
            const Sint16 right = *(--src);
icculus@10834
   398
            const Sint16 left = *(--src);
icculus@10834
   399
            SDL_assert(pos >= 0.0);
icculus@10834
   400
            *(--dst) = (((Sint32) right) + ((Sint32) earlier_right)) >> 1;
icculus@10834
   401
            *(--dst) = (((Sint32) left) + ((Sint32) earlier_left)) >> 1;
icculus@10834
   402
            earlier_right = right;
icculus@10834
   403
            earlier_left = left;
icculus@10834
   404
            idx -= src_incr;
icculus@10834
   405
        }
icculus@10834
   406
icculus@10834
   407
        /* do last sample, interpolated against previous run's state. */
icculus@10834
   408
        *(--dst) = (((Sint32) inbuf[1]) + ((Sint32) last_sample[1])) >> 1;
icculus@10834
   409
        *(--dst) = (((Sint32) inbuf[0]) + ((Sint32) last_sample[0])) >> 1;
icculus@10834
   410
        last_sample[1] = final_right;
icculus@10834
   411
        last_sample[0] = final_left;
icculus@10834
   412
icculus@10841
   413
        dst = (outbuf + (dest_samples * chans));
icculus@10834
   414
    } else {
icculus@10834
   415
        Sint16 *target = (outbuf + (dest_samples * chans));
icculus@10834
   416
        dst = outbuf;
icculus@10834
   417
        idx = 0.0;
icculus@10834
   418
        while (dst < target) {
icculus@10834
   419
            const int pos = ((int) idx) * chans;
icculus@10834
   420
            const Sint16 *src = &inbuf[pos];
icculus@10834
   421
            const Sint16 left = *(src++);
icculus@10834
   422
            const Sint16 right = *(src++);
icculus@10834
   423
            SDL_assert(pos <= finalpos);
icculus@10834
   424
            *(dst++) = (((Sint32) left) + ((Sint32) last_sample[0])) >> 1;
icculus@10834
   425
            *(dst++) = (((Sint32) right) + ((Sint32) last_sample[1])) >> 1;
icculus@10834
   426
            last_sample[0] = left;
icculus@10834
   427
            last_sample[1] = right;
icculus@10834
   428
            idx += src_incr;
icculus@10834
   429
        }
icculus@10834
   430
    }
icculus@10834
   431
icculus@10834
   432
    return (int) ((dst - outbuf) * ((int) sizeof (Sint16)));
icculus@10834
   433
}
icculus@10834
   434
icculus@10834
   435
static void SDLCALL
icculus@10834
   436
SDL_ResampleCVT_si16_c2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
icculus@10834
   437
{
icculus@10834
   438
    const Sint16 *src = (const Sint16 *) cvt->buf;
icculus@10834
   439
    const int srclen = cvt->len_cvt;
icculus@10839
   440
    Sint16 *dst = (Sint16 *) cvt->buf;
icculus@10839
   441
    const int dstlen = (cvt->len * cvt->len_mult);
icculus@10925
   442
    Sint16 state[2];
icculus@10925
   443
icculus@10925
   444
    state[0] = src[0];
icculus@10925
   445
    state[1] = src[1];
icculus@10834
   446
icculus@10834
   447
    SDL_assert(format == AUDIO_S16SYS);
icculus@10834
   448
icculus@10834
   449
    cvt->len_cvt = SDL_ResampleAudioSimple_si16_c2(cvt->rate_incr, state, src, srclen, dst, dstlen);
icculus@10834
   450
    if (cvt->filters[++cvt->filter_index]) {
icculus@10834
   451
        cvt->filters[cvt->filter_index](cvt, format);
icculus@10834
   452
    }
icculus@10834
   453
}
icculus@10834
   454
slouken@0
   455
slouken@1895
   456
int
slouken@1895
   457
SDL_ConvertAudio(SDL_AudioCVT * cvt)
slouken@0
   458
{
icculus@3021
   459
    /* !!! FIXME: (cvt) should be const; stack-copy it here. */
icculus@3021
   460
    /* !!! FIXME: (actually, we can't...len_cvt needs to be updated. Grr.) */
icculus@3021
   461
slouken@1895
   462
    /* Make sure there's data to convert */
slouken@1895
   463
    if (cvt->buf == NULL) {
icculus@10575
   464
        return SDL_SetError("No buffer allocated for conversion");
slouken@1895
   465
    }
icculus@10575
   466
slouken@1895
   467
    /* Return okay if no conversion is necessary */
slouken@1895
   468
    cvt->len_cvt = cvt->len;
slouken@1895
   469
    if (cvt->filters[0] == NULL) {
icculus@10575
   470
        return 0;
slouken@1895
   471
    }
slouken@0
   472
slouken@1895
   473
    /* Set up the conversion and go! */
slouken@1895
   474
    cvt->filter_index = 0;
slouken@1895
   475
    cvt->filters[0] (cvt, cvt->src_format);
icculus@10575
   476
    return 0;
slouken@0
   477
}
slouken@0
   478
icculus@10575
   479
static void SDLCALL
icculus@10575
   480
SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format)
icculus@10575
   481
{
slouken@10579
   482
#if DEBUG_CONVERT
slouken@10579
   483
    printf("Converting byte order\n");
slouken@10579
   484
#endif
icculus@1982
   485
icculus@10575
   486
    switch (SDL_AUDIO_BITSIZE(format)) {
icculus@10575
   487
        #define CASESWAP(b) \
icculus@10575
   488
            case b: { \
icculus@10575
   489
                Uint##b *ptr = (Uint##b *) cvt->buf; \
icculus@10575
   490
                int i; \
icculus@10575
   491
                for (i = cvt->len_cvt / sizeof (*ptr); i; --i, ++ptr) { \
icculus@10575
   492
                    *ptr = SDL_Swap##b(*ptr); \
icculus@10575
   493
                } \
icculus@10575
   494
                break; \
icculus@10575
   495
            }
icculus@1982
   496
icculus@10575
   497
        CASESWAP(16);
icculus@10575
   498
        CASESWAP(32);
icculus@10575
   499
        CASESWAP(64);
icculus@10575
   500
icculus@10575
   501
        #undef CASESWAP
icculus@10575
   502
icculus@10575
   503
        default: SDL_assert(!"unhandled byteswap datatype!"); break;
icculus@10575
   504
    }
icculus@10575
   505
icculus@10575
   506
    if (cvt->filters[++cvt->filter_index]) {
icculus@10575
   507
        /* flip endian flag for data. */
icculus@10575
   508
        if (format & SDL_AUDIO_MASK_ENDIAN) {
icculus@10575
   509
            format &= ~SDL_AUDIO_MASK_ENDIAN;
icculus@10575
   510
        } else {
icculus@10575
   511
            format |= SDL_AUDIO_MASK_ENDIAN;
icculus@10575
   512
        }
icculus@10575
   513
        cvt->filters[cvt->filter_index](cvt, format);
icculus@10575
   514
    }
icculus@1982
   515
}
icculus@1982
   516
slouken@11096
   517
static int
slouken@11096
   518
SDL_AddAudioCVTFilter(SDL_AudioCVT *cvt, const SDL_AudioFilter filter)
slouken@11096
   519
{
slouken@11096
   520
    if (cvt->filter_index >= SDL_AUDIOCVT_MAX_FILTERS) {
slouken@11096
   521
        return SDL_SetError("Too many filters needed for conversion, exceeded maximum of %d", SDL_AUDIOCVT_MAX_FILTERS);
slouken@11096
   522
    }
slouken@11096
   523
    if (filter == NULL) {
slouken@11096
   524
        return SDL_SetError("Audio filter pointer is NULL");
slouken@11096
   525
    }
slouken@11096
   526
    cvt->filters[cvt->filter_index++] = filter;
slouken@11096
   527
    cvt->filters[cvt->filter_index] = NULL; /* Moving terminator */
slouken@11096
   528
    return 0;
slouken@11096
   529
}
icculus@1982
   530
icculus@1982
   531
static int
icculus@10575
   532
SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
icculus@1982
   533
{
icculus@10575
   534
    int retval = 0;  /* 0 == no conversion necessary. */
icculus@1982
   535
icculus@10575
   536
    if ((SDL_AUDIO_ISBIGENDIAN(src_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
slouken@11096
   537
        if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
slouken@11096
   538
            return -1;
slouken@11096
   539
        }
icculus@10575
   540
        retval = 1;  /* added a converter. */
icculus@10575
   541
    }
icculus@1982
   542
icculus@10575
   543
    if (!SDL_AUDIO_ISFLOAT(src_fmt)) {
icculus@10576
   544
        const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
icculus@10576
   545
        const Uint16 dst_bitsize = 32;
icculus@10575
   546
        SDL_AudioFilter filter = NULL;
icculus@10576
   547
icculus@10575
   548
        switch (src_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
icculus@10575
   549
            case AUDIO_S8: filter = SDL_Convert_S8_to_F32; break;
icculus@10575
   550
            case AUDIO_U8: filter = SDL_Convert_U8_to_F32; break;
icculus@10575
   551
            case AUDIO_S16: filter = SDL_Convert_S16_to_F32; break;
philipp@10591
   552
            case AUDIO_U16: filter = SDL_Convert_U16_to_F32; break;
icculus@10575
   553
            case AUDIO_S32: filter = SDL_Convert_S32_to_F32; break;
icculus@10575
   554
            default: SDL_assert(!"Unexpected audio format!"); break;
icculus@1982
   555
        }
icculus@1982
   556
icculus@10575
   557
        if (!filter) {
icculus@10575
   558
            return SDL_SetError("No conversion available for these formats");
icculus@10575
   559
        }
icculus@10575
   560
slouken@11096
   561
        if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
slouken@11096
   562
            return -1;
slouken@11096
   563
        }
icculus@1982
   564
        if (src_bitsize < dst_bitsize) {
icculus@1982
   565
            const int mult = (dst_bitsize / src_bitsize);
icculus@1982
   566
            cvt->len_mult *= mult;
icculus@1982
   567
            cvt->len_ratio *= mult;
icculus@1982
   568
        } else if (src_bitsize > dst_bitsize) {
icculus@1982
   569
            cvt->len_ratio /= (src_bitsize / dst_bitsize);
icculus@1982
   570
        }
icculus@10576
   571
icculus@10575
   572
        retval = 1;  /* added a converter. */
icculus@1982
   573
    }
icculus@1982
   574
icculus@10575
   575
    return retval;
icculus@1982
   576
}
icculus@1982
   577
icculus@10575
   578
static int
icculus@10575
   579
SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt)
icculus@10575
   580
{
icculus@10575
   581
    int retval = 0;  /* 0 == no conversion necessary. */
icculus@3021
   582
icculus@10575
   583
    if (!SDL_AUDIO_ISFLOAT(dst_fmt)) {
icculus@10577
   584
        const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
icculus@10577
   585
        const Uint16 src_bitsize = 32;
icculus@10575
   586
        SDL_AudioFilter filter = NULL;
icculus@10575
   587
        switch (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
icculus@10575
   588
            case AUDIO_S8: filter = SDL_Convert_F32_to_S8; break;
icculus@10575
   589
            case AUDIO_U8: filter = SDL_Convert_F32_to_U8; break;
icculus@10575
   590
            case AUDIO_S16: filter = SDL_Convert_F32_to_S16; break;
philipp@10591
   591
            case AUDIO_U16: filter = SDL_Convert_F32_to_U16; break;
icculus@10575
   592
            case AUDIO_S32: filter = SDL_Convert_F32_to_S32; break;
icculus@10575
   593
            default: SDL_assert(!"Unexpected audio format!"); break;
icculus@10575
   594
        }
slouken@2716
   595
icculus@10575
   596
        if (!filter) {
icculus@10575
   597
            return SDL_SetError("No conversion available for these formats");
icculus@10575
   598
        }
icculus@10575
   599
slouken@11096
   600
        if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
slouken@11096
   601
            return -1;
slouken@11096
   602
        }
icculus@10575
   603
        if (src_bitsize < dst_bitsize) {
icculus@10575
   604
            const int mult = (dst_bitsize / src_bitsize);
icculus@10575
   605
            cvt->len_mult *= mult;
icculus@10575
   606
            cvt->len_ratio *= mult;
icculus@10575
   607
        } else if (src_bitsize > dst_bitsize) {
icculus@10575
   608
            cvt->len_ratio /= (src_bitsize / dst_bitsize);
icculus@10575
   609
        }
icculus@10575
   610
        retval = 1;  /* added a converter. */
icculus@10575
   611
    }
icculus@10575
   612
icculus@10575
   613
    if ((SDL_AUDIO_ISBIGENDIAN(dst_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
slouken@11096
   614
        if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
slouken@11096
   615
            return -1;
slouken@11096
   616
        }
icculus@10575
   617
        retval = 1;  /* added a converter. */
icculus@10575
   618
    }
icculus@10575
   619
icculus@10575
   620
    return retval;
icculus@3021
   621
}
slouken@2716
   622
icculus@10799
   623
static void
icculus@10799
   624
SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format)
icculus@10799
   625
{
icculus@10799
   626
    const float *src = (const float *) cvt->buf;
icculus@10799
   627
    const int srclen = cvt->len_cvt;
icculus@10833
   628
    float *dst = (float *) cvt->buf;
icculus@10833
   629
    const int dstlen = (cvt->len * cvt->len_mult);
icculus@10804
   630
    float state[8];
icculus@10756
   631
icculus@10799
   632
    SDL_assert(format == AUDIO_F32SYS);
icculus@10799
   633
slouken@10805
   634
    SDL_memcpy(state, src, chans*sizeof(*src));
icculus@10799
   635
icculus@10804
   636
    cvt->len_cvt = SDL_ResampleAudioSimple(chans, cvt->rate_incr, state, src, srclen, dst, dstlen);
icculus@10799
   637
    if (cvt->filters[++cvt->filter_index]) {
icculus@10799
   638
        cvt->filters[cvt->filter_index](cvt, format);
icculus@10799
   639
    }
icculus@10799
   640
}
icculus@10799
   641
icculus@10799
   642
/* !!! FIXME: We only have this macro salsa because SDL_AudioCVT doesn't
icculus@10799
   643
   !!! FIXME:  store channel info, so we have to have function entry
icculus@10799
   644
   !!! FIXME:  points for each supported channel count and multiple
icculus@10799
   645
   !!! FIXME:  vs arbitrary. When we rev the ABI, clean this up. */
icculus@10756
   646
#define RESAMPLER_FUNCS(chans) \
icculus@10756
   647
    static void SDLCALL \
icculus@10799
   648
    SDL_ResampleCVT_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
icculus@10799
   649
        SDL_ResampleCVT(cvt, chans, format); \
icculus@10756
   650
    }
icculus@10756
   651
RESAMPLER_FUNCS(1)
icculus@10756
   652
RESAMPLER_FUNCS(2)
icculus@10756
   653
RESAMPLER_FUNCS(4)
icculus@10756
   654
RESAMPLER_FUNCS(6)
icculus@10756
   655
RESAMPLER_FUNCS(8)
icculus@10756
   656
#undef RESAMPLER_FUNCS
icculus@10756
   657
icculus@10799
   658
static SDL_AudioFilter
icculus@10799
   659
ChooseCVTResampler(const int dst_channels)
icculus@3021
   660
{
icculus@10799
   661
    switch (dst_channels) {
icculus@10799
   662
        case 1: return SDL_ResampleCVT_c1;
icculus@10799
   663
        case 2: return SDL_ResampleCVT_c2;
icculus@10799
   664
        case 4: return SDL_ResampleCVT_c4;
icculus@10799
   665
        case 6: return SDL_ResampleCVT_c6;
icculus@10799
   666
        case 8: return SDL_ResampleCVT_c8;
icculus@10799
   667
        default: break;
icculus@3021
   668
    }
slouken@2716
   669
icculus@10799
   670
    return NULL;
icculus@10756
   671
}
icculus@10575
   672
icculus@3021
   673
static int
icculus@10756
   674
SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, const int dst_channels,
icculus@10756
   675
                          const int src_rate, const int dst_rate)
icculus@3021
   676
{
icculus@10756
   677
    SDL_AudioFilter filter;
icculus@3021
   678
icculus@10756
   679
    if (src_rate == dst_rate) {
icculus@10756
   680
        return 0;  /* no conversion necessary. */
slouken@2716
   681
    }
slouken@2716
   682
icculus@10799
   683
    filter = ChooseCVTResampler(dst_channels);
icculus@10756
   684
    if (filter == NULL) {
icculus@10756
   685
        return SDL_SetError("No conversion available for these rates");
icculus@10756
   686
    }
icculus@10756
   687
icculus@10756
   688
    /* Update (cvt) with filter details... */
slouken@11096
   689
    if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
slouken@11096
   690
        return -1;
slouken@11096
   691
    }
icculus@10756
   692
    if (src_rate < dst_rate) {
icculus@10756
   693
        const double mult = ((double) dst_rate) / ((double) src_rate);
icculus@10756
   694
        cvt->len_mult *= (int) SDL_ceil(mult);
icculus@10756
   695
        cvt->len_ratio *= mult;
icculus@10756
   696
    } else {
icculus@10756
   697
        cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate);
icculus@10756
   698
    }
icculus@10756
   699
icculus@10756
   700
    return 1;               /* added a converter. */
slouken@2716
   701
}
icculus@1982
   702
icculus@11097
   703
static SDL_bool
icculus@11097
   704
SDL_SupportedAudioFormat(const SDL_AudioFormat fmt)
icculus@11097
   705
{
icculus@11097
   706
    switch (fmt) {
icculus@11097
   707
        case AUDIO_U8:
icculus@11097
   708
        case AUDIO_S8:
icculus@11097
   709
        case AUDIO_U16LSB:
icculus@11097
   710
        case AUDIO_S16LSB:
icculus@11097
   711
        case AUDIO_U16MSB:
icculus@11097
   712
        case AUDIO_S16MSB:
icculus@11097
   713
        case AUDIO_S32LSB:
icculus@11097
   714
        case AUDIO_S32MSB:
icculus@11097
   715
        case AUDIO_F32LSB:
icculus@11097
   716
        case AUDIO_F32MSB:
icculus@11097
   717
            return SDL_TRUE;  /* supported. */
icculus@11097
   718
icculus@11097
   719
        default:
icculus@11097
   720
            break;
icculus@11097
   721
    }
icculus@11097
   722
icculus@11097
   723
    return SDL_FALSE;  /* unsupported. */
icculus@11097
   724
}
icculus@11097
   725
icculus@11097
   726
static SDL_bool
icculus@11097
   727
SDL_SupportedChannelCount(const int channels)
icculus@11097
   728
{
icculus@11097
   729
    switch (channels) {
icculus@11097
   730
        case 1:  /* mono */
icculus@11097
   731
        case 2:  /* stereo */
icculus@11097
   732
        case 4:  /* quad */
icculus@11097
   733
        case 6:  /* 5.1 */
icculus@11097
   734
            return SDL_TRUE;  /* supported. */
icculus@11097
   735
icculus@11097
   736
        case 8:  /* !!! FIXME: 7.1 */
icculus@11097
   737
        default:
icculus@11097
   738
            break;
icculus@11097
   739
    }
icculus@11097
   740
icculus@11097
   741
    return SDL_FALSE;  /* unsupported. */
icculus@11097
   742
}
icculus@11097
   743
icculus@1982
   744
icculus@1982
   745
/* Creates a set of audio filters to convert from one format to another.
icculus@1982
   746
   Returns -1 if the format conversion is not supported, 0 if there's
icculus@1982
   747
   no conversion needed, or 1 if the audio filter is set up.
slouken@0
   748
*/
slouken@1895
   749
slouken@1895
   750
int
slouken@1895
   751
SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
icculus@1982
   752
                  SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate,
icculus@1982
   753
                  SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
slouken@0
   754
{
aschiffler@6819
   755
    /* Sanity check target pointer */
aschiffler@6819
   756
    if (cvt == NULL) {
icculus@7037
   757
        return SDL_InvalidParamError("cvt");
aschiffler@6819
   758
    }
slouken@7191
   759
slouken@10767
   760
    /* Make sure we zero out the audio conversion before error checking */
slouken@10767
   761
    SDL_zerop(cvt);
slouken@10767
   762
icculus@11097
   763
    if (!SDL_SupportedAudioFormat(src_fmt)) {
icculus@7037
   764
        return SDL_SetError("Invalid source format");
icculus@11097
   765
    } else if (!SDL_SupportedAudioFormat(dst_fmt)) {
icculus@7037
   766
        return SDL_SetError("Invalid destination format");
icculus@11097
   767
    } else if (!SDL_SupportedChannelCount(src_channels)) {
icculus@11097
   768
        return SDL_SetError("Invalid source channels");
icculus@11097
   769
    } else if (!SDL_SupportedChannelCount(dst_channels)) {
icculus@11097
   770
        return SDL_SetError("Invalid destination channels");
icculus@11097
   771
    } else if (src_rate == 0) {
icculus@11097
   772
        return SDL_SetError("Source rate is zero");
icculus@11097
   773
    } else if (dst_rate == 0) {
icculus@11097
   774
        return SDL_SetError("Destination rate is zero");
icculus@1982
   775
    }
icculus@3021
   776
slouken@10579
   777
#if DEBUG_CONVERT
icculus@1982
   778
    printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
slouken@1985
   779
           src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
slouken@1985
   780
#endif
icculus@1982
   781
slouken@1895
   782
    /* Start off with no conversion necessary */
icculus@1982
   783
    cvt->src_format = src_fmt;
icculus@1982
   784
    cvt->dst_format = dst_fmt;
slouken@1895
   785
    cvt->needed = 0;
slouken@1895
   786
    cvt->filter_index = 0;
slouken@1895
   787
    cvt->filters[0] = NULL;
slouken@1895
   788
    cvt->len_mult = 1;
slouken@1895
   789
    cvt->len_ratio = 1.0;
icculus@3021
   790
    cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
slouken@0
   791
icculus@10834
   792
    /* SDL now favors float32 as its preferred internal format, and considers
icculus@10834
   793
       everything else to be a degenerate case that we might have to make
icculus@10834
   794
       multiple passes over the data to convert to and from float32 as
icculus@10834
   795
       necessary. That being said, we keep one special case around for
icculus@10834
   796
       efficiency: stereo data in Sint16 format, in the native byte order,
icculus@10834
   797
       that only needs resampling. This is likely to be the most popular
icculus@10834
   798
       legacy format, that apps, hardware and the OS are likely to be able
icculus@10834
   799
       to process directly, so we handle this one case directly without
icculus@10834
   800
       unnecessary conversions. This means that apps on embedded devices
icculus@10834
   801
       without floating point hardware should consider aiming for this
icculus@10834
   802
       format as well. */
icculus@10834
   803
    if ((src_channels == 2) && (dst_channels == 2) && (src_fmt == AUDIO_S16SYS) && (dst_fmt == AUDIO_S16SYS) && (src_rate != dst_rate)) {
icculus@10834
   804
        cvt->needed = 1;
slouken@11096
   805
        if (SDL_AddAudioCVTFilter(cvt, SDL_ResampleCVT_si16_c2) < 0) {
slouken@11096
   806
            return -1;
slouken@11096
   807
        }
icculus@10834
   808
        if (src_rate < dst_rate) {
icculus@10834
   809
            const double mult = ((double) dst_rate) / ((double) src_rate);
icculus@10834
   810
            cvt->len_mult *= (int) SDL_ceil(mult);
icculus@10834
   811
            cvt->len_ratio *= mult;
icculus@10834
   812
        } else {
icculus@10834
   813
            cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate);
icculus@10834
   814
        }
icculus@10834
   815
        return 1;
icculus@10834
   816
    }
icculus@10834
   817
icculus@10575
   818
    /* Type conversion goes like this now:
icculus@10575
   819
        - byteswap to CPU native format first if necessary.
icculus@10575
   820
        - convert to native Float32 if necessary.
icculus@10575
   821
        - resample and change channel count if necessary.
icculus@10575
   822
        - convert back to native format.
icculus@10575
   823
        - byteswap back to foreign format if necessary.
icculus@10575
   824
icculus@10575
   825
       The expectation is we can process data faster in float32
icculus@10575
   826
       (possibly with SIMD), and making several passes over the same
icculus@10756
   827
       buffer is likely to be CPU cache-friendly, avoiding the
icculus@10575
   828
       biggest performance hit in modern times. Previously we had
icculus@10575
   829
       (script-generated) custom converters for every data type and
icculus@10575
   830
       it was a bloat on SDL compile times and final library size. */
icculus@10575
   831
slouken@10767
   832
    /* see if we can skip float conversion entirely. */
slouken@10767
   833
    if (src_rate == dst_rate && src_channels == dst_channels) {
slouken@10767
   834
        if (src_fmt == dst_fmt) {
slouken@10767
   835
            return 0;
slouken@10767
   836
        }
slouken@10767
   837
slouken@10767
   838
        /* just a byteswap needed? */
slouken@10767
   839
        if ((src_fmt & ~SDL_AUDIO_MASK_ENDIAN) == (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN)) {
slouken@11096
   840
            if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
slouken@11096
   841
                return -1;
slouken@11096
   842
            }
slouken@10767
   843
            cvt->needed = 1;
slouken@10767
   844
            return 1;
slouken@10767
   845
        }
icculus@10575
   846
    }
icculus@10575
   847
icculus@1982
   848
    /* Convert data types, if necessary. Updates (cvt). */
slouken@10767
   849
    if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
slouken@1985
   850
        return -1;              /* shouldn't happen, but just in case... */
icculus@3021
   851
    }
slouken@0
   852
icculus@1982
   853
    /* Channel conversion */
slouken@1895
   854
    if (src_channels != dst_channels) {
slouken@1895
   855
        if ((src_channels == 1) && (dst_channels > 1)) {
slouken@11096
   856
            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertMonoToStereo) < 0) {
slouken@11096
   857
                return -1;
slouken@11096
   858
            }
slouken@1895
   859
            cvt->len_mult *= 2;
slouken@1895
   860
            src_channels = 2;
slouken@1895
   861
            cvt->len_ratio *= 2;
slouken@1895
   862
        }
slouken@1895
   863
        if ((src_channels == 2) && (dst_channels == 6)) {
slouken@11096
   864
            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertStereoTo51) < 0) {
slouken@11096
   865
                return -1;
slouken@11096
   866
            }
slouken@1895
   867
            src_channels = 6;
slouken@1895
   868
            cvt->len_mult *= 3;
slouken@1895
   869
            cvt->len_ratio *= 3;
slouken@1895
   870
        }
slouken@1895
   871
        if ((src_channels == 2) && (dst_channels == 4)) {
slouken@11096
   872
            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertStereoToQuad) < 0) {
slouken@11096
   873
                return -1;
slouken@11096
   874
            }
slouken@1895
   875
            src_channels = 4;
slouken@1895
   876
            cvt->len_mult *= 2;
slouken@1895
   877
            cvt->len_ratio *= 2;
slouken@1895
   878
        }
slouken@1895
   879
        while ((src_channels * 2) <= dst_channels) {
slouken@11096
   880
            if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertMonoToStereo) < 0) {
slouken@11096
   881
                return -1;
slouken@11096
   882
            }
slouken@1895
   883
            cvt->len_mult *= 2;
slouken@1895
   884
            src_channels *= 2;
slouken@1895
   885
            cvt->len_ratio *= 2;
slouken@1895
   886
        }
slouken@1895
   887
        if ((src_channels == 6) && (dst_channels <= 2)) {
slouken@11096
   888
            if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51ToStereo) < 0) {
slouken@11096
   889
                return -1;
slouken@11096
   890
            }
slouken@1895
   891
            src_channels = 2;
slouken@1895
   892
            cvt->len_ratio /= 3;
slouken@1895
   893
        }
slouken@1895
   894
        if ((src_channels == 6) && (dst_channels == 4)) {
slouken@11096
   895
            if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51ToQuad) < 0) {
slouken@11096
   896
                return -1;
slouken@11096
   897
            }
slouken@1895
   898
            src_channels = 4;
slouken@1895
   899
            cvt->len_ratio /= 2;
slouken@1895
   900
        }
slouken@1895
   901
        /* This assumes that 4 channel audio is in the format:
slouken@1895
   902
           Left {front/back} + Right {front/back}
slouken@1895
   903
           so converting to L/R stereo works properly.
slouken@1895
   904
         */
slouken@1895
   905
        while (((src_channels % 2) == 0) &&
slouken@1895
   906
               ((src_channels / 2) >= dst_channels)) {
icculus@10832
   907
            SDL_AudioFilter filter = NULL;
icculus@10832
   908
icculus@10832
   909
            #if HAVE_SSE3_INTRINSICS
icculus@10832
   910
            if (SDL_HasSSE3()) {
icculus@10832
   911
                filter = SDL_ConvertStereoToMono_SSE3;
icculus@10832
   912
            }
icculus@10832
   913
            #endif
icculus@10832
   914
icculus@10832
   915
            if (!filter) {
icculus@10832
   916
                filter = SDL_ConvertStereoToMono;
icculus@10832
   917
            }
icculus@10832
   918
slouken@11096
   919
            if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
slouken@11096
   920
                return -1;
slouken@11096
   921
            }
icculus@10832
   922
slouken@1895
   923
            src_channels /= 2;
slouken@1895
   924
            cvt->len_ratio /= 2;
slouken@1895
   925
        }
slouken@1895
   926
        if (src_channels != dst_channels) {
slouken@1895
   927
            /* Uh oh.. */ ;
slouken@1895
   928
        }
slouken@1895
   929
    }
slouken@0
   930
icculus@3021
   931
    /* Do rate conversion, if necessary. Updates (cvt). */
slouken@10767
   932
    if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) < 0) {
icculus@3021
   933
        return -1;              /* shouldn't happen, but just in case... */
slouken@2716
   934
    }
slouken@2716
   935
icculus@10756
   936
    /* Move to final data type. */
slouken@10767
   937
    if (SDL_BuildAudioTypeCVTFromFloat(cvt, dst_fmt) < 0) {
icculus@10575
   938
        return -1;              /* shouldn't happen, but just in case... */
slouken@1895
   939
    }
icculus@10575
   940
icculus@10575
   941
    cvt->needed = (cvt->filter_index != 0);
slouken@1895
   942
    return (cvt->needed);
slouken@0
   943
}
slouken@1895
   944
icculus@10842
   945
typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const void *inbuf, const int inbuflen, void *outbuf, const int outbuflen);
slouken@10773
   946
typedef void (*SDL_ResetAudioStreamResamplerFunc)(SDL_AudioStream *stream);
slouken@10773
   947
typedef void (*SDL_CleanupAudioStreamResamplerFunc)(SDL_AudioStream *stream);
icculus@10757
   948
icculus@10757
   949
struct SDL_AudioStream
icculus@10757
   950
{
icculus@10757
   951
    SDL_AudioCVT cvt_before_resampling;
icculus@10757
   952
    SDL_AudioCVT cvt_after_resampling;
icculus@10757
   953
    SDL_DataQueue *queue;
icculus@10844
   954
    Uint8 *work_buffer_base;  /* maybe unaligned pointer from SDL_realloc(). */
icculus@10757
   955
    int work_buffer_len;
icculus@10757
   956
    int src_sample_frame_size;
icculus@10757
   957
    SDL_AudioFormat src_format;
icculus@10757
   958
    Uint8 src_channels;
icculus@10757
   959
    int src_rate;
icculus@10757
   960
    int dst_sample_frame_size;
icculus@10757
   961
    SDL_AudioFormat dst_format;
icculus@10757
   962
    Uint8 dst_channels;
icculus@10757
   963
    int dst_rate;
icculus@10757
   964
    double rate_incr;
icculus@10757
   965
    Uint8 pre_resample_channels;
slouken@10773
   966
    int packetlen;
slouken@10773
   967
    void *resampler_state;
slouken@10773
   968
    SDL_ResampleAudioStreamFunc resampler_func;
slouken@10773
   969
    SDL_ResetAudioStreamResamplerFunc reset_resampler_func;
slouken@10773
   970
    SDL_CleanupAudioStreamResamplerFunc cleanup_resampler_func;
slouken@10773
   971
};
slouken@10773
   972
icculus@10851
   973
static Uint8 *
icculus@10851
   974
EnsureStreamBufferSize(SDL_AudioStream *stream, const int newlen)
icculus@10851
   975
{
icculus@10851
   976
    Uint8 *ptr;
icculus@10851
   977
    size_t offset;
icculus@10851
   978
icculus@10851
   979
    if (stream->work_buffer_len >= newlen) {
icculus@10851
   980
        ptr = stream->work_buffer_base;
icculus@10851
   981
    } else {
icculus@10851
   982
        ptr = (Uint8 *) SDL_realloc(stream->work_buffer_base, newlen + 32);
icculus@10851
   983
        if (!ptr) {
icculus@10851
   984
            SDL_OutOfMemory();
icculus@10851
   985
            return NULL;
icculus@10851
   986
        }
icculus@10851
   987
        /* Make sure we're aligned to 16 bytes for SIMD code. */
icculus@10851
   988
        stream->work_buffer_base = ptr;
icculus@10851
   989
        stream->work_buffer_len = newlen;
icculus@10851
   990
    }
icculus@10851
   991
icculus@10851
   992
    offset = ((size_t) ptr) & 15;
icculus@10851
   993
    return offset ? ptr + (16 - offset) : ptr;
icculus@10851
   994
}
icculus@10851
   995
slouken@10777
   996
#ifdef HAVE_LIBSAMPLERATE_H
slouken@10773
   997
static int
icculus@10842
   998
SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen)
slouken@10773
   999
{
icculus@10842
  1000
    const float *inbuf = (const float *) _inbuf;
icculus@10842
  1001
    float *outbuf = (float *) _outbuf;
icculus@10799
  1002
    const int framelen = sizeof(float) * stream->pre_resample_channels;
icculus@10790
  1003
    SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
slouken@10773
  1004
    SRC_DATA data;
slouken@10773
  1005
    int result;
slouken@10773
  1006
icculus@10851
  1007
    if (inbuf == ((const float *) outbuf)) {  /* libsamplerate can't work in-place. */
icculus@10851
  1008
        Uint8 *ptr = EnsureStreamBufferSize(stream, inbuflen + outbuflen);
icculus@10851
  1009
        if (ptr == NULL) {
icculus@10851
  1010
            SDL_OutOfMemory();
icculus@10851
  1011
            return 0;
icculus@10851
  1012
        }
icculus@10851
  1013
        SDL_memcpy(ptr + outbuflen, ptr, inbuflen);
icculus@10851
  1014
        inbuf = (const float *) (ptr + outbuflen);
icculus@10851
  1015
        outbuf = (float *) ptr;
icculus@10851
  1016
    }
icculus@10851
  1017
slouken@10777
  1018
    data.data_in = (float *)inbuf; /* Older versions of libsamplerate had a non-const pointer, but didn't write to it */
icculus@10799
  1019
    data.input_frames = inbuflen / framelen;
slouken@10773
  1020
    data.input_frames_used = 0;
slouken@10773
  1021
slouken@10773
  1022
    data.data_out = outbuf;
icculus@10799
  1023
    data.output_frames = outbuflen / framelen;
slouken@10773
  1024
slouken@10773
  1025
    data.end_of_input = 0;
slouken@10773
  1026
    data.src_ratio = stream->rate_incr;
slouken@10773
  1027
icculus@10790
  1028
    result = SRC_src_process(state, &data);
slouken@10773
  1029
    if (result != 0) {
icculus@10790
  1030
        SDL_SetError("src_process() failed: %s", SRC_src_strerror(result));
slouken@10773
  1031
        return 0;
slouken@10773
  1032
    }
slouken@10773
  1033
slouken@10773
  1034
    /* If this fails, we need to store them off somewhere */
slouken@10773
  1035
    SDL_assert(data.input_frames_used == data.input_frames);
slouken@10773
  1036
slouken@10773
  1037
    return data.output_frames_gen * (sizeof(float) * stream->pre_resample_channels);
slouken@10773
  1038
}
slouken@10773
  1039
slouken@10773
  1040
static void
slouken@10773
  1041
SDL_ResetAudioStreamResampler_SRC(SDL_AudioStream *stream)
slouken@10773
  1042
{
icculus@10790
  1043
    SRC_src_reset((SRC_STATE *)stream->resampler_state);
slouken@10773
  1044
}
slouken@10773
  1045
slouken@10773
  1046
static void
slouken@10773
  1047
SDL_CleanupAudioStreamResampler_SRC(SDL_AudioStream *stream)
slouken@10773
  1048
{
icculus@10790
  1049
    SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
slouken@10773
  1050
    if (state) {
icculus@10790
  1051
        SRC_src_delete(state);
slouken@10773
  1052
    }
slouken@10773
  1053
slouken@10773
  1054
    stream->resampler_state = NULL;
slouken@10773
  1055
    stream->resampler_func = NULL;
slouken@10773
  1056
    stream->reset_resampler_func = NULL;
slouken@10773
  1057
    stream->cleanup_resampler_func = NULL;
slouken@10773
  1058
}
slouken@10773
  1059
slouken@10773
  1060
static SDL_bool
slouken@10773
  1061
SetupLibSampleRateResampling(SDL_AudioStream *stream)
slouken@10773
  1062
{
icculus@10790
  1063
    int result = 0;
icculus@10790
  1064
    SRC_STATE *state = NULL;
slouken@10773
  1065
icculus@10790
  1066
    if (SRC_available) {
icculus@10849
  1067
        state = SRC_src_new(SRC_converter, stream->pre_resample_channels, &result);
icculus@10790
  1068
        if (!state) {
icculus@10790
  1069
            SDL_SetError("src_new() failed: %s", SRC_src_strerror(result));
icculus@10790
  1070
        }
slouken@10773
  1071
    }
slouken@10773
  1072
icculus@10790
  1073
    if (!state) {
icculus@10790
  1074
        SDL_CleanupAudioStreamResampler_SRC(stream);
slouken@10773
  1075
        return SDL_FALSE;
slouken@10773
  1076
    }
slouken@10773
  1077
slouken@10773
  1078
    stream->resampler_state = state;
slouken@10773
  1079
    stream->resampler_func = SDL_ResampleAudioStream_SRC;
slouken@10773
  1080
    stream->reset_resampler_func = SDL_ResetAudioStreamResampler_SRC;
slouken@10773
  1081
    stream->cleanup_resampler_func = SDL_CleanupAudioStreamResampler_SRC;
slouken@10773
  1082
slouken@10773
  1083
    return SDL_TRUE;
slouken@10773
  1084
}
icculus@10790
  1085
#endif /* HAVE_LIBSAMPLERATE_H */
slouken@10773
  1086
slouken@10773
  1087
slouken@10773
  1088
typedef struct
slouken@10773
  1089
{
icculus@10757
  1090
    SDL_bool resampler_seeded;
icculus@10842
  1091
    union
icculus@10842
  1092
    {
icculus@10842
  1093
        float f[8];
icculus@10842
  1094
        Sint16 si16[2];
icculus@10842
  1095
    } resampler_state;
slouken@10773
  1096
} SDL_AudioStreamResamplerState;
slouken@10773
  1097
slouken@10773
  1098
static int
icculus@10842
  1099
SDL_ResampleAudioStream(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen)
slouken@10773
  1100
{
icculus@10842
  1101
    const float *inbuf = (const float *) _inbuf;
icculus@10842
  1102
    float *outbuf = (float *) _outbuf;
slouken@10773
  1103
    SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
slouken@10773
  1104
    const int chans = (int)stream->pre_resample_channels;
slouken@10773
  1105
icculus@10842
  1106
    SDL_assert(chans <= SDL_arraysize(state->resampler_state.f));
slouken@10773
  1107
slouken@10773
  1108
    if (!state->resampler_seeded) {
icculus@10842
  1109
        SDL_memcpy(state->resampler_state.f, inbuf, chans * sizeof (float));
slouken@10773
  1110
        state->resampler_seeded = SDL_TRUE;
slouken@10773
  1111
    }
slouken@10773
  1112
icculus@10842
  1113
    return SDL_ResampleAudioSimple(chans, stream->rate_incr, state->resampler_state.f, inbuf, inbuflen, outbuf, outbuflen);
icculus@10842
  1114
}
icculus@10842
  1115
icculus@10842
  1116
static int
icculus@10842
  1117
SDL_ResampleAudioStream_si16_c2(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen)
icculus@10842
  1118
{
icculus@10842
  1119
    const Sint16 *inbuf = (const Sint16 *) _inbuf;
icculus@10842
  1120
    Sint16 *outbuf = (Sint16 *) _outbuf;
icculus@10842
  1121
    SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
icculus@10842
  1122
icculus@10926
  1123
    SDL_assert(((int)stream->pre_resample_channels) <= SDL_arraysize(state->resampler_state.si16));
icculus@10842
  1124
icculus@10842
  1125
    if (!state->resampler_seeded) {
icculus@10842
  1126
        state->resampler_state.si16[0] = inbuf[0];
icculus@10842
  1127
        state->resampler_state.si16[1] = inbuf[1];
icculus@10842
  1128
        state->resampler_seeded = SDL_TRUE;
icculus@10842
  1129
    }
icculus@10842
  1130
icculus@10842
  1131
    return SDL_ResampleAudioSimple_si16_c2(stream->rate_incr, state->resampler_state.si16, inbuf, inbuflen, outbuf, outbuflen);
slouken@10773
  1132
}
slouken@10773
  1133
slouken@10773
  1134
static void
slouken@10773
  1135
SDL_ResetAudioStreamResampler(SDL_AudioStream *stream)
slouken@10773
  1136
{
slouken@10773
  1137
    SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
slouken@10773
  1138
    state->resampler_seeded = SDL_FALSE;
slouken@10773
  1139
}
slouken@10773
  1140
slouken@10773
  1141
static void
slouken@10773
  1142
SDL_CleanupAudioStreamResampler(SDL_AudioStream *stream)
slouken@10773
  1143
{
slouken@10773
  1144
    SDL_free(stream->resampler_state);
slouken@10773
  1145
}
icculus@10757
  1146
icculus@10789
  1147
SDL_AudioStream *
icculus@10789
  1148
SDL_NewAudioStream(const SDL_AudioFormat src_format,
icculus@10789
  1149
                   const Uint8 src_channels,
icculus@10789
  1150
                   const int src_rate,
icculus@10789
  1151
                   const SDL_AudioFormat dst_format,
icculus@10789
  1152
                   const Uint8 dst_channels,
icculus@10789
  1153
                   const int dst_rate)
icculus@10757
  1154
{
icculus@10757
  1155
    const int packetlen = 4096;  /* !!! FIXME: good enough for now. */
icculus@10757
  1156
    Uint8 pre_resample_channels;
icculus@10757
  1157
    SDL_AudioStream *retval;
icculus@10842
  1158
#ifndef HAVE_LIBSAMPLERATE_H
icculus@10842
  1159
    const SDL_bool SRC_available = SDL_FALSE;
icculus@10842
  1160
#endif
icculus@10757
  1161
icculus@10757
  1162
    retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
icculus@10757
  1163
    if (!retval) {
icculus@10757
  1164
        return NULL;
icculus@10757
  1165
    }
icculus@10757
  1166
icculus@10757
  1167
    /* If increasing channels, do it after resampling, since we'd just
icculus@10757
  1168
       do more work to resample duplicate channels. If we're decreasing, do
icculus@10757
  1169
       it first so we resample the interpolated data instead of interpolating
icculus@10757
  1170
       the resampled data (!!! FIXME: decide if that works in practice, though!). */
icculus@10757
  1171
    pre_resample_channels = SDL_min(src_channels, dst_channels);
icculus@10757
  1172
icculus@10883
  1173
    retval->src_sample_frame_size = (SDL_AUDIO_BITSIZE(src_format) / 8) * src_channels;
icculus@10757
  1174
    retval->src_format = src_format;
icculus@10757
  1175
    retval->src_channels = src_channels;
icculus@10757
  1176
    retval->src_rate = src_rate;
icculus@10883
  1177
    retval->dst_sample_frame_size = (SDL_AUDIO_BITSIZE(dst_format) / 8) * dst_channels;
icculus@10757
  1178
    retval->dst_format = dst_format;
icculus@10757
  1179
    retval->dst_channels = dst_channels;
icculus@10757
  1180
    retval->dst_rate = dst_rate;
icculus@10757
  1181
    retval->pre_resample_channels = pre_resample_channels;
icculus@10757
  1182
    retval->packetlen = packetlen;
icculus@10757
  1183
    retval->rate_incr = ((double) dst_rate) / ((double) src_rate);
icculus@10757
  1184
icculus@10757
  1185
    /* Not resampling? It's an easy conversion (and maybe not even that!). */
icculus@10757
  1186
    if (src_rate == dst_rate) {
icculus@10757
  1187
        retval->cvt_before_resampling.needed = SDL_FALSE;
slouken@10773
  1188
        if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
slouken@10773
  1189
            SDL_FreeAudioStream(retval);
icculus@10757
  1190
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
icculus@10757
  1191
        }
icculus@10842
  1192
    /* fast path special case for stereo Sint16 data that just needs resampling. */
icculus@10842
  1193
    } else if ((!SRC_available) && (src_channels == 2) && (dst_channels == 2) && (src_format == AUDIO_S16SYS) && (dst_format == AUDIO_S16SYS)) {
icculus@10842
  1194
        SDL_assert(src_rate != dst_rate);
icculus@10842
  1195
        retval->resampler_state = SDL_calloc(1, sizeof(SDL_AudioStreamResamplerState));
icculus@10842
  1196
        if (!retval->resampler_state) {
icculus@10842
  1197
            SDL_FreeAudioStream(retval);
icculus@10842
  1198
            SDL_OutOfMemory();
icculus@10842
  1199
            return NULL;
icculus@10842
  1200
        }
icculus@10842
  1201
        retval->resampler_func = SDL_ResampleAudioStream_si16_c2;
icculus@10842
  1202
        retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
icculus@10842
  1203
        retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
icculus@10757
  1204
    } else {
icculus@10757
  1205
        /* Don't resample at first. Just get us to Float32 format. */
icculus@10757
  1206
        /* !!! FIXME: convert to int32 on devices without hardware float. */
slouken@10773
  1207
        if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) < 0) {
slouken@10773
  1208
            SDL_FreeAudioStream(retval);
icculus@10757
  1209
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
icculus@10757
  1210
        }
icculus@10757
  1211
slouken@10777
  1212
#ifdef HAVE_LIBSAMPLERATE_H
slouken@10773
  1213
        SetupLibSampleRateResampling(retval);
slouken@10773
  1214
#endif
slouken@10773
  1215
slouken@10773
  1216
        if (!retval->resampler_func) {
slouken@10773
  1217
            retval->resampler_state = SDL_calloc(1, sizeof(SDL_AudioStreamResamplerState));
slouken@10773
  1218
            if (!retval->resampler_state) {
slouken@10773
  1219
                SDL_FreeAudioStream(retval);
slouken@10773
  1220
                SDL_OutOfMemory();
slouken@10773
  1221
                return NULL;
slouken@10773
  1222
            }
slouken@10773
  1223
            retval->resampler_func = SDL_ResampleAudioStream;
slouken@10773
  1224
            retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
slouken@10773
  1225
            retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
slouken@10773
  1226
        }
slouken@10773
  1227
icculus@10757
  1228
        /* Convert us to the final format after resampling. */
slouken@10773
  1229
        if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
slouken@10773
  1230
            SDL_FreeAudioStream(retval);
icculus@10757
  1231
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
icculus@10757
  1232
        }
icculus@10757
  1233
    }
icculus@10757
  1234
icculus@10757
  1235
    retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2);
icculus@10757
  1236
    if (!retval->queue) {
slouken@10773
  1237
        SDL_FreeAudioStream(retval);
icculus@10757
  1238
        return NULL;  /* SDL_NewDataQueue should have called SDL_SetError. */
icculus@10757
  1239
    }
icculus@10757
  1240
icculus@10757
  1241
    return retval;
icculus@10757
  1242
}
icculus@10757
  1243
icculus@10757
  1244
int
icculus@10757
  1245
SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, const Uint32 _buflen)
icculus@10757
  1246
{
icculus@10757
  1247
    int buflen = (int) _buflen;
icculus@10846
  1248
    const void *origbuf = buf;
icculus@10757
  1249
icculus@10844
  1250
    /* !!! FIXME: several converters can take advantage of SIMD, but only
icculus@10844
  1251
       !!! FIXME:  if the data is aligned to 16 bytes. EnsureStreamBufferSize()
icculus@10844
  1252
       !!! FIXME:  guarantees the buffer will align, but the
icculus@10844
  1253
       !!! FIXME:  converters will iterate over the data backwards if
icculus@10844
  1254
       !!! FIXME:  the output grows, and this means we won't align if buflen
icculus@10844
  1255
       !!! FIXME:  isn't a multiple of 16. In these cases, we should chop off
icculus@10844
  1256
       !!! FIXME:  a few samples at the end and convert them separately. */
icculus@10844
  1257
icculus@10757
  1258
    if (!stream) {
icculus@10757
  1259
        return SDL_InvalidParamError("stream");
icculus@10757
  1260
    } else if (!buf) {
icculus@10757
  1261
        return SDL_InvalidParamError("buf");
icculus@10757
  1262
    } else if (buflen == 0) {
icculus@10757
  1263
        return 0;  /* nothing to do. */
icculus@10757
  1264
    } else if ((buflen % stream->src_sample_frame_size) != 0) {
icculus@10757
  1265
        return SDL_SetError("Can't add partial sample frames");
icculus@10757
  1266
    }
icculus@10757
  1267
icculus@10757
  1268
    if (stream->cvt_before_resampling.needed) {
icculus@10757
  1269
        const int workbuflen = buflen * stream->cvt_before_resampling.len_mult;  /* will be "* 1" if not needed */
icculus@10844
  1270
        Uint8 *workbuf = EnsureStreamBufferSize(stream, workbuflen);
icculus@10757
  1271
        if (workbuf == NULL) {
icculus@10757
  1272
            return -1;  /* probably out of memory. */
icculus@10757
  1273
        }
icculus@10846
  1274
        SDL_assert(buf == origbuf);
icculus@10757
  1275
        SDL_memcpy(workbuf, buf, buflen);
icculus@10757
  1276
        stream->cvt_before_resampling.buf = workbuf;
icculus@10757
  1277
        stream->cvt_before_resampling.len = buflen;
icculus@10757
  1278
        if (SDL_ConvertAudio(&stream->cvt_before_resampling) == -1) {
icculus@10757
  1279
            return -1;   /* uhoh! */
icculus@10757
  1280
        }
icculus@10757
  1281
        buf = workbuf;
icculus@10757
  1282
        buflen = stream->cvt_before_resampling.len_cvt;
icculus@10757
  1283
    }
icculus@10757
  1284
icculus@10757
  1285
    if (stream->dst_rate != stream->src_rate) {
icculus@10757
  1286
        const int workbuflen = buflen * ((int) SDL_ceil(stream->rate_incr));
icculus@10844
  1287
        Uint8 *workbuf = EnsureStreamBufferSize(stream, workbuflen);
icculus@10757
  1288
        if (workbuf == NULL) {
icculus@10757
  1289
            return -1;  /* probably out of memory. */
icculus@10757
  1290
        }
icculus@10851
  1291
        /* don't SDL_memcpy(workbuf, buf, buflen) here; our resampler can work inplace or not,
icculus@10851
  1292
           libsamplerate needs buffers to be separate; either way, avoid a copy here if possible. */
icculus@10851
  1293
        if (buf != origbuf) {
icculus@10851
  1294
            buf = workbuf;  /* in case we realloc()'d the pointer. */
icculus@10843
  1295
        }
icculus@10851
  1296
        buflen = stream->resampler_func(stream, buf, buflen, workbuf, workbuflen);
icculus@10851
  1297
        buf = EnsureStreamBufferSize(stream, workbuflen);
icculus@10851
  1298
        SDL_assert(buf != NULL);  /* shouldn't be growing, just aligning. */
icculus@10757
  1299
    }
icculus@10757
  1300
icculus@10757
  1301
    if (stream->cvt_after_resampling.needed) {
icculus@10842
  1302
        const int workbuflen = buflen * stream->cvt_after_resampling.len_mult;  /* will be "* 1" if not needed */
icculus@10844
  1303
        Uint8 *workbuf = EnsureStreamBufferSize(stream, workbuflen);
icculus@10757
  1304
        if (workbuf == NULL) {
icculus@10757
  1305
            return -1;  /* probably out of memory. */
icculus@10757
  1306
        }
icculus@10846
  1307
        if (buf == origbuf) {  /* copy if we haven't before. */
icculus@11128
  1308
            SDL_memcpy(workbuf, origbuf, buflen);
icculus@10843
  1309
        }
icculus@10757
  1310
        stream->cvt_after_resampling.buf = workbuf;
icculus@10757
  1311
        stream->cvt_after_resampling.len = buflen;
icculus@10757
  1312
        if (SDL_ConvertAudio(&stream->cvt_after_resampling) == -1) {
icculus@10757
  1313
            return -1;   /* uhoh! */
icculus@10757
  1314
        }
icculus@10757
  1315
        buf = workbuf;
icculus@10757
  1316
        buflen = stream->cvt_after_resampling.len_cvt;
icculus@10757
  1317
    }
icculus@10757
  1318
icculus@10757
  1319
    return SDL_WriteToDataQueue(stream->queue, buf, buflen);
icculus@10757
  1320
}
icculus@10757
  1321
icculus@10757
  1322
void
icculus@10757
  1323
SDL_AudioStreamClear(SDL_AudioStream *stream)
icculus@10757
  1324
{
icculus@10757
  1325
    if (!stream) {
icculus@10757
  1326
        SDL_InvalidParamError("stream");
icculus@10757
  1327
    } else {
icculus@10757
  1328
        SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
icculus@10776
  1329
        if (stream->reset_resampler_func) {
icculus@10776
  1330
            stream->reset_resampler_func(stream);
icculus@10776
  1331
        }
icculus@10757
  1332
    }
icculus@10757
  1333
}
icculus@10757
  1334
icculus@10757
  1335
icculus@10757
  1336
/* get converted/resampled data from the stream */
icculus@10757
  1337
int
icculus@10764
  1338
SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, const Uint32 len)
icculus@10757
  1339
{
icculus@10757
  1340
    if (!stream) {
icculus@10757
  1341
        return SDL_InvalidParamError("stream");
icculus@10757
  1342
    } else if (!buf) {
icculus@10757
  1343
        return SDL_InvalidParamError("buf");
icculus@10757
  1344
    } else if (len == 0) {
icculus@10757
  1345
        return 0;  /* nothing to do. */
icculus@10757
  1346
    } else if ((len % stream->dst_sample_frame_size) != 0) {
icculus@10757
  1347
        return SDL_SetError("Can't request partial sample frames");
icculus@10757
  1348
    }
icculus@10757
  1349
icculus@10764
  1350
    return (int) SDL_ReadFromDataQueue(stream->queue, buf, len);
icculus@10757
  1351
}
icculus@10757
  1352
icculus@10757
  1353
/* number of converted/resampled bytes available */
icculus@10757
  1354
int
icculus@10757
  1355
SDL_AudioStreamAvailable(SDL_AudioStream *stream)
icculus@10757
  1356
{
icculus@10757
  1357
    return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
icculus@10757
  1358
}
icculus@10757
  1359
icculus@10757
  1360
/* dispose of a stream */
icculus@10757
  1361
void
icculus@10757
  1362
SDL_FreeAudioStream(SDL_AudioStream *stream)
icculus@10757
  1363
{
icculus@10757
  1364
    if (stream) {
slouken@10773
  1365
        if (stream->cleanup_resampler_func) {
slouken@10773
  1366
            stream->cleanup_resampler_func(stream);
slouken@10773
  1367
        }
icculus@10757
  1368
        SDL_FreeDataQueue(stream->queue);
icculus@10844
  1369
        SDL_free(stream->work_buffer_base);
icculus@10757
  1370
        SDL_free(stream);
icculus@10757
  1371
    }
icculus@10757
  1372
}
icculus@10757
  1373
icculus@10575
  1374
/* vi: set ts=4 sw=4 expandtab: */
slouken@2716
  1375