src/audio/SDL_audiocvt.c
author Ryan C. Gordon <icculus@icculus.org>
Fri, 20 Jan 2017 16:26:24 -0500
changeset 10830 92013fad89d1
parent 10817 efc103e60c5b
child 10831 fcbb4d7f2344
permissions -rw-r--r--
audio: removed conditional from simple resampler's inner loop.

We never seem to overflow the source buffer now; this might have been a
leftover from a bug that was covered by Vitaly's fixes?

Removing this conditional makes the resampler 10-20% faster. Left an
assert in there for debug builds, in case this still happens.
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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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/* 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++) = (float) ((((double) src[0]) + ((double) src[1])) * 0.5);
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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 = outbuf;
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    float *target = (dst + (dest_samples * chans));
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    double idx = 0.0;
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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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    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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    return (int) ((dst - outbuf) * (int)sizeof(float));
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}
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int
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SDL_ConvertAudio(SDL_AudioCVT * cvt)
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{
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    /* !!! FIXME: (cvt) should be const; stack-copy it here. */
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    /* !!! FIXME: (actually, we can't...len_cvt needs to be updated. Grr.) */
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    /* Make sure there's data to convert */
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    if (cvt->buf == NULL) {
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        return SDL_SetError("No buffer allocated for conversion");
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    }
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    /* Return okay if no conversion is necessary */
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    cvt->len_cvt = cvt->len;
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    if (cvt->filters[0] == NULL) {
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        return 0;
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    }
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    /* Set up the conversion and go! */
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    cvt->filter_index = 0;
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    cvt->filters[0] (cvt, cvt->src_format);
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    return 0;
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}
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static void SDLCALL
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SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format)
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{
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#if DEBUG_CONVERT
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    printf("Converting byte order\n");
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#endif
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    switch (SDL_AUDIO_BITSIZE(format)) {
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        #define CASESWAP(b) \
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            case b: { \
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                Uint##b *ptr = (Uint##b *) cvt->buf; \
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                int i; \
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                for (i = cvt->len_cvt / sizeof (*ptr); i; --i, ++ptr) { \
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                    *ptr = SDL_Swap##b(*ptr); \
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                } \
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                break; \
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            }
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        CASESWAP(16);
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        CASESWAP(32);
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        CASESWAP(64);
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        #undef CASESWAP
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        default: SDL_assert(!"unhandled byteswap datatype!"); break;
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    }
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    if (cvt->filters[++cvt->filter_index]) {
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        /* flip endian flag for data. */
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        if (format & SDL_AUDIO_MASK_ENDIAN) {
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            format &= ~SDL_AUDIO_MASK_ENDIAN;
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        } else {
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            format |= SDL_AUDIO_MASK_ENDIAN;
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        }
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        cvt->filters[cvt->filter_index](cvt, format);
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    }
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}
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static int
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SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
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{
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    int retval = 0;  /* 0 == no conversion necessary. */
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    if ((SDL_AUDIO_ISBIGENDIAN(src_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
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        cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
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        retval = 1;  /* added a converter. */
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    }
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    if (!SDL_AUDIO_ISFLOAT(src_fmt)) {
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        const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
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        const Uint16 dst_bitsize = 32;
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        SDL_AudioFilter filter = NULL;
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        switch (src_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
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            case AUDIO_S8: filter = SDL_Convert_S8_to_F32; break;
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            case AUDIO_U8: filter = SDL_Convert_U8_to_F32; break;
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            case AUDIO_S16: filter = SDL_Convert_S16_to_F32; break;
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            case AUDIO_U16: filter = SDL_Convert_U16_to_F32; break;
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            case AUDIO_S32: filter = SDL_Convert_S32_to_F32; break;
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            default: SDL_assert(!"Unexpected audio format!"); break;
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        }
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        if (!filter) {
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            return SDL_SetError("No conversion available for these formats");
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        }
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        cvt->filters[cvt->filter_index++] = filter;
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        if (src_bitsize < dst_bitsize) {
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            const int mult = (dst_bitsize / src_bitsize);
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            cvt->len_mult *= mult;
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            cvt->len_ratio *= mult;
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        } else if (src_bitsize > dst_bitsize) {
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            cvt->len_ratio /= (src_bitsize / dst_bitsize);
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        }
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        retval = 1;  /* added a converter. */
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    }
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    return retval;
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}
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static int
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SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt)
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{
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    int retval = 0;  /* 0 == no conversion necessary. */
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    if (!SDL_AUDIO_ISFLOAT(dst_fmt)) {
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        const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
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        const Uint16 src_bitsize = 32;
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        SDL_AudioFilter filter = NULL;
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        switch (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
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            case AUDIO_S8: filter = SDL_Convert_F32_to_S8; break;
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            case AUDIO_U8: filter = SDL_Convert_F32_to_U8; break;
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            case AUDIO_S16: filter = SDL_Convert_F32_to_S16; break;
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            case AUDIO_U16: filter = SDL_Convert_F32_to_U16; break;
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            case AUDIO_S32: filter = SDL_Convert_F32_to_S32; break;
icculus@10575
   348
            default: SDL_assert(!"Unexpected audio format!"); break;
icculus@10575
   349
        }
slouken@2716
   350
icculus@10575
   351
        if (!filter) {
icculus@10575
   352
            return SDL_SetError("No conversion available for these formats");
icculus@10575
   353
        }
icculus@10575
   354
icculus@10575
   355
        cvt->filters[cvt->filter_index++] = filter;
icculus@10575
   356
        if (src_bitsize < dst_bitsize) {
icculus@10575
   357
            const int mult = (dst_bitsize / src_bitsize);
icculus@10575
   358
            cvt->len_mult *= mult;
icculus@10575
   359
            cvt->len_ratio *= mult;
icculus@10575
   360
        } else if (src_bitsize > dst_bitsize) {
icculus@10575
   361
            cvt->len_ratio /= (src_bitsize / dst_bitsize);
icculus@10575
   362
        }
icculus@10575
   363
        retval = 1;  /* added a converter. */
icculus@10575
   364
    }
icculus@10575
   365
icculus@10575
   366
    if ((SDL_AUDIO_ISBIGENDIAN(dst_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
icculus@10575
   367
        cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
icculus@10575
   368
        retval = 1;  /* added a converter. */
icculus@10575
   369
    }
icculus@10575
   370
icculus@10575
   371
    return retval;
icculus@3021
   372
}
slouken@2716
   373
icculus@10799
   374
static void
icculus@10799
   375
SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format)
icculus@10799
   376
{
icculus@10799
   377
    const float *src = (const float *) cvt->buf;
icculus@10799
   378
    const int srclen = cvt->len_cvt;
icculus@10799
   379
    float *dst = (float *) (cvt->buf + srclen);
icculus@10799
   380
    const int dstlen = (cvt->len * cvt->len_mult) - srclen;
icculus@10804
   381
    float state[8];
icculus@10756
   382
icculus@10799
   383
    SDL_assert(format == AUDIO_F32SYS);
icculus@10799
   384
slouken@10805
   385
    SDL_memcpy(state, src, chans*sizeof(*src));
icculus@10799
   386
icculus@10804
   387
    cvt->len_cvt = SDL_ResampleAudioSimple(chans, cvt->rate_incr, state, src, srclen, dst, dstlen);
icculus@10799
   388
icculus@10799
   389
    SDL_memcpy(cvt->buf, dst, cvt->len_cvt);
icculus@10799
   390
    if (cvt->filters[++cvt->filter_index]) {
icculus@10799
   391
        cvt->filters[cvt->filter_index](cvt, format);
icculus@10799
   392
    }
icculus@10799
   393
}
icculus@10799
   394
icculus@10799
   395
/* !!! FIXME: We only have this macro salsa because SDL_AudioCVT doesn't
icculus@10799
   396
   !!! FIXME:  store channel info, so we have to have function entry
icculus@10799
   397
   !!! FIXME:  points for each supported channel count and multiple
icculus@10799
   398
   !!! FIXME:  vs arbitrary. When we rev the ABI, clean this up. */
icculus@10756
   399
#define RESAMPLER_FUNCS(chans) \
icculus@10756
   400
    static void SDLCALL \
icculus@10799
   401
    SDL_ResampleCVT_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
icculus@10799
   402
        SDL_ResampleCVT(cvt, chans, format); \
icculus@10756
   403
    }
icculus@10756
   404
RESAMPLER_FUNCS(1)
icculus@10756
   405
RESAMPLER_FUNCS(2)
icculus@10756
   406
RESAMPLER_FUNCS(4)
icculus@10756
   407
RESAMPLER_FUNCS(6)
icculus@10756
   408
RESAMPLER_FUNCS(8)
icculus@10756
   409
#undef RESAMPLER_FUNCS
icculus@10756
   410
icculus@10799
   411
static SDL_AudioFilter
icculus@10799
   412
ChooseCVTResampler(const int dst_channels)
icculus@3021
   413
{
icculus@10799
   414
    switch (dst_channels) {
icculus@10799
   415
        case 1: return SDL_ResampleCVT_c1;
icculus@10799
   416
        case 2: return SDL_ResampleCVT_c2;
icculus@10799
   417
        case 4: return SDL_ResampleCVT_c4;
icculus@10799
   418
        case 6: return SDL_ResampleCVT_c6;
icculus@10799
   419
        case 8: return SDL_ResampleCVT_c8;
icculus@10799
   420
        default: break;
icculus@3021
   421
    }
slouken@2716
   422
icculus@10799
   423
    return NULL;
icculus@10756
   424
}
icculus@10575
   425
icculus@3021
   426
static int
icculus@10756
   427
SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, const int dst_channels,
icculus@10756
   428
                          const int src_rate, const int dst_rate)
icculus@3021
   429
{
icculus@10756
   430
    SDL_AudioFilter filter;
icculus@3021
   431
icculus@10756
   432
    if (src_rate == dst_rate) {
icculus@10756
   433
        return 0;  /* no conversion necessary. */
slouken@2716
   434
    }
slouken@2716
   435
icculus@10799
   436
    filter = ChooseCVTResampler(dst_channels);
icculus@10756
   437
    if (filter == NULL) {
icculus@10756
   438
        return SDL_SetError("No conversion available for these rates");
icculus@10756
   439
    }
icculus@10756
   440
icculus@10756
   441
    /* Update (cvt) with filter details... */
icculus@10756
   442
    cvt->filters[cvt->filter_index++] = filter;
icculus@10756
   443
    if (src_rate < dst_rate) {
icculus@10756
   444
        const double mult = ((double) dst_rate) / ((double) src_rate);
icculus@10756
   445
        cvt->len_mult *= (int) SDL_ceil(mult);
icculus@10756
   446
        cvt->len_ratio *= mult;
icculus@10756
   447
    } else {
icculus@10756
   448
        cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate);
icculus@10756
   449
    }
icculus@10756
   450
icculus@10799
   451
    /* the buffer is big enough to hold the destination now, but
icculus@10799
   452
       we need it large enough to hold a separate scratch buffer. */
icculus@10799
   453
    cvt->len_mult *= 2;
icculus@10799
   454
icculus@10756
   455
    return 1;               /* added a converter. */
slouken@2716
   456
}
icculus@1982
   457
icculus@1982
   458
icculus@1982
   459
/* Creates a set of audio filters to convert from one format to another.
icculus@1982
   460
   Returns -1 if the format conversion is not supported, 0 if there's
icculus@1982
   461
   no conversion needed, or 1 if the audio filter is set up.
slouken@0
   462
*/
slouken@1895
   463
slouken@1895
   464
int
slouken@1895
   465
SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
icculus@1982
   466
                  SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate,
icculus@1982
   467
                  SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
slouken@0
   468
{
aschiffler@6819
   469
    /* Sanity check target pointer */
aschiffler@6819
   470
    if (cvt == NULL) {
icculus@7037
   471
        return SDL_InvalidParamError("cvt");
aschiffler@6819
   472
    }
slouken@7191
   473
slouken@10767
   474
    /* Make sure we zero out the audio conversion before error checking */
slouken@10767
   475
    SDL_zerop(cvt);
slouken@10767
   476
slouken@3491
   477
    /* there are no unsigned types over 16 bits, so catch this up front. */
icculus@1982
   478
    if ((SDL_AUDIO_BITSIZE(src_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(src_fmt))) {
icculus@7037
   479
        return SDL_SetError("Invalid source format");
icculus@1982
   480
    }
icculus@1982
   481
    if ((SDL_AUDIO_BITSIZE(dst_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(dst_fmt))) {
icculus@7037
   482
        return SDL_SetError("Invalid destination format");
icculus@1982
   483
    }
icculus@3021
   484
icculus@3021
   485
    /* prevent possible divisions by zero, etc. */
aschiffler@6819
   486
    if ((src_channels == 0) || (dst_channels == 0)) {
icculus@7037
   487
        return SDL_SetError("Source or destination channels is zero");
aschiffler@6819
   488
    }
icculus@3021
   489
    if ((src_rate == 0) || (dst_rate == 0)) {
icculus@7037
   490
        return SDL_SetError("Source or destination rate is zero");
icculus@3021
   491
    }
slouken@10579
   492
#if DEBUG_CONVERT
icculus@1982
   493
    printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
slouken@1985
   494
           src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
slouken@1985
   495
#endif
icculus@1982
   496
slouken@1895
   497
    /* Start off with no conversion necessary */
icculus@1982
   498
    cvt->src_format = src_fmt;
icculus@1982
   499
    cvt->dst_format = dst_fmt;
slouken@1895
   500
    cvt->needed = 0;
slouken@1895
   501
    cvt->filter_index = 0;
slouken@1895
   502
    cvt->filters[0] = NULL;
slouken@1895
   503
    cvt->len_mult = 1;
slouken@1895
   504
    cvt->len_ratio = 1.0;
icculus@3021
   505
    cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
slouken@0
   506
icculus@10575
   507
    /* Type conversion goes like this now:
icculus@10575
   508
        - byteswap to CPU native format first if necessary.
icculus@10575
   509
        - convert to native Float32 if necessary.
icculus@10575
   510
        - resample and change channel count if necessary.
icculus@10575
   511
        - convert back to native format.
icculus@10575
   512
        - byteswap back to foreign format if necessary.
icculus@10575
   513
icculus@10575
   514
       The expectation is we can process data faster in float32
icculus@10575
   515
       (possibly with SIMD), and making several passes over the same
icculus@10756
   516
       buffer is likely to be CPU cache-friendly, avoiding the
icculus@10575
   517
       biggest performance hit in modern times. Previously we had
icculus@10575
   518
       (script-generated) custom converters for every data type and
icculus@10575
   519
       it was a bloat on SDL compile times and final library size. */
icculus@10575
   520
slouken@10767
   521
    /* see if we can skip float conversion entirely. */
slouken@10767
   522
    if (src_rate == dst_rate && src_channels == dst_channels) {
slouken@10767
   523
        if (src_fmt == dst_fmt) {
slouken@10767
   524
            return 0;
slouken@10767
   525
        }
slouken@10767
   526
slouken@10767
   527
        /* just a byteswap needed? */
slouken@10767
   528
        if ((src_fmt & ~SDL_AUDIO_MASK_ENDIAN) == (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN)) {
slouken@10767
   529
            cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
slouken@10767
   530
            cvt->needed = 1;
slouken@10767
   531
            return 1;
slouken@10767
   532
        }
icculus@10575
   533
    }
icculus@10575
   534
icculus@1982
   535
    /* Convert data types, if necessary. Updates (cvt). */
slouken@10767
   536
    if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
slouken@1985
   537
        return -1;              /* shouldn't happen, but just in case... */
icculus@3021
   538
    }
slouken@0
   539
icculus@1982
   540
    /* Channel conversion */
slouken@1895
   541
    if (src_channels != dst_channels) {
slouken@1895
   542
        if ((src_channels == 1) && (dst_channels > 1)) {
icculus@10793
   543
            cvt->filters[cvt->filter_index++] = SDL_ConvertMonoToStereo;
slouken@1895
   544
            cvt->len_mult *= 2;
slouken@1895
   545
            src_channels = 2;
slouken@1895
   546
            cvt->len_ratio *= 2;
slouken@1895
   547
        }
slouken@1895
   548
        if ((src_channels == 2) && (dst_channels == 6)) {
icculus@10793
   549
            cvt->filters[cvt->filter_index++] = SDL_ConvertStereoTo51;
slouken@1895
   550
            src_channels = 6;
slouken@1895
   551
            cvt->len_mult *= 3;
slouken@1895
   552
            cvt->len_ratio *= 3;
slouken@1895
   553
        }
slouken@1895
   554
        if ((src_channels == 2) && (dst_channels == 4)) {
icculus@10793
   555
            cvt->filters[cvt->filter_index++] = SDL_ConvertStereoToQuad;
slouken@1895
   556
            src_channels = 4;
slouken@1895
   557
            cvt->len_mult *= 2;
slouken@1895
   558
            cvt->len_ratio *= 2;
slouken@1895
   559
        }
slouken@1895
   560
        while ((src_channels * 2) <= dst_channels) {
icculus@10793
   561
            cvt->filters[cvt->filter_index++] = SDL_ConvertMonoToStereo;
slouken@1895
   562
            cvt->len_mult *= 2;
slouken@1895
   563
            src_channels *= 2;
slouken@1895
   564
            cvt->len_ratio *= 2;
slouken@1895
   565
        }
slouken@1895
   566
        if ((src_channels == 6) && (dst_channels <= 2)) {
icculus@10793
   567
            cvt->filters[cvt->filter_index++] = SDL_Convert51ToStereo;
slouken@1895
   568
            src_channels = 2;
slouken@1895
   569
            cvt->len_ratio /= 3;
slouken@1895
   570
        }
slouken@1895
   571
        if ((src_channels == 6) && (dst_channels == 4)) {
icculus@10793
   572
            cvt->filters[cvt->filter_index++] = SDL_Convert51ToQuad;
slouken@1895
   573
            src_channels = 4;
slouken@1895
   574
            cvt->len_ratio /= 2;
slouken@1895
   575
        }
slouken@1895
   576
        /* This assumes that 4 channel audio is in the format:
slouken@1895
   577
           Left {front/back} + Right {front/back}
slouken@1895
   578
           so converting to L/R stereo works properly.
slouken@1895
   579
         */
slouken@1895
   580
        while (((src_channels % 2) == 0) &&
slouken@1895
   581
               ((src_channels / 2) >= dst_channels)) {
icculus@10793
   582
            cvt->filters[cvt->filter_index++] = SDL_ConvertStereoToMono;
slouken@1895
   583
            src_channels /= 2;
slouken@1895
   584
            cvt->len_ratio /= 2;
slouken@1895
   585
        }
slouken@1895
   586
        if (src_channels != dst_channels) {
slouken@1895
   587
            /* Uh oh.. */ ;
slouken@1895
   588
        }
slouken@1895
   589
    }
slouken@0
   590
icculus@3021
   591
    /* Do rate conversion, if necessary. Updates (cvt). */
slouken@10767
   592
    if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) < 0) {
icculus@3021
   593
        return -1;              /* shouldn't happen, but just in case... */
slouken@2716
   594
    }
slouken@2716
   595
icculus@10756
   596
    /* Move to final data type. */
slouken@10767
   597
    if (SDL_BuildAudioTypeCVTFromFloat(cvt, dst_fmt) < 0) {
icculus@10575
   598
        return -1;              /* shouldn't happen, but just in case... */
slouken@1895
   599
    }
icculus@10575
   600
icculus@10575
   601
    cvt->needed = (cvt->filter_index != 0);
slouken@1895
   602
    return (cvt->needed);
slouken@0
   603
}
slouken@1895
   604
slouken@10773
   605
typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen);
slouken@10773
   606
typedef void (*SDL_ResetAudioStreamResamplerFunc)(SDL_AudioStream *stream);
slouken@10773
   607
typedef void (*SDL_CleanupAudioStreamResamplerFunc)(SDL_AudioStream *stream);
icculus@10757
   608
icculus@10757
   609
struct SDL_AudioStream
icculus@10757
   610
{
icculus@10757
   611
    SDL_AudioCVT cvt_before_resampling;
icculus@10757
   612
    SDL_AudioCVT cvt_after_resampling;
icculus@10757
   613
    SDL_DataQueue *queue;
icculus@10757
   614
    Uint8 *work_buffer;
icculus@10757
   615
    int work_buffer_len;
icculus@10757
   616
    Uint8 *resample_buffer;
icculus@10757
   617
    int resample_buffer_len;
icculus@10757
   618
    int src_sample_frame_size;
icculus@10757
   619
    SDL_AudioFormat src_format;
icculus@10757
   620
    Uint8 src_channels;
icculus@10757
   621
    int src_rate;
icculus@10757
   622
    int dst_sample_frame_size;
icculus@10757
   623
    SDL_AudioFormat dst_format;
icculus@10757
   624
    Uint8 dst_channels;
icculus@10757
   625
    int dst_rate;
icculus@10757
   626
    double rate_incr;
icculus@10757
   627
    Uint8 pre_resample_channels;
slouken@10773
   628
    int packetlen;
slouken@10773
   629
    void *resampler_state;
slouken@10773
   630
    SDL_ResampleAudioStreamFunc resampler_func;
slouken@10773
   631
    SDL_ResetAudioStreamResamplerFunc reset_resampler_func;
slouken@10773
   632
    SDL_CleanupAudioStreamResamplerFunc cleanup_resampler_func;
slouken@10773
   633
};
slouken@10773
   634
slouken@10777
   635
#ifdef HAVE_LIBSAMPLERATE_H
slouken@10773
   636
static int
slouken@10773
   637
SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
slouken@10773
   638
{
icculus@10799
   639
    const int framelen = sizeof(float) * stream->pre_resample_channels;
icculus@10790
   640
    SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
slouken@10773
   641
    SRC_DATA data;
slouken@10773
   642
    int result;
slouken@10773
   643
slouken@10777
   644
    data.data_in = (float *)inbuf; /* Older versions of libsamplerate had a non-const pointer, but didn't write to it */
icculus@10799
   645
    data.input_frames = inbuflen / framelen;
slouken@10773
   646
    data.input_frames_used = 0;
slouken@10773
   647
slouken@10773
   648
    data.data_out = outbuf;
icculus@10799
   649
    data.output_frames = outbuflen / framelen;
slouken@10773
   650
slouken@10773
   651
    data.end_of_input = 0;
slouken@10773
   652
    data.src_ratio = stream->rate_incr;
slouken@10773
   653
icculus@10790
   654
    result = SRC_src_process(state, &data);
slouken@10773
   655
    if (result != 0) {
icculus@10790
   656
        SDL_SetError("src_process() failed: %s", SRC_src_strerror(result));
slouken@10773
   657
        return 0;
slouken@10773
   658
    }
slouken@10773
   659
slouken@10773
   660
    /* If this fails, we need to store them off somewhere */
slouken@10773
   661
    SDL_assert(data.input_frames_used == data.input_frames);
slouken@10773
   662
slouken@10773
   663
    return data.output_frames_gen * (sizeof(float) * stream->pre_resample_channels);
slouken@10773
   664
}
slouken@10773
   665
slouken@10773
   666
static void
slouken@10773
   667
SDL_ResetAudioStreamResampler_SRC(SDL_AudioStream *stream)
slouken@10773
   668
{
icculus@10790
   669
    SRC_src_reset((SRC_STATE *)stream->resampler_state);
slouken@10773
   670
}
slouken@10773
   671
slouken@10773
   672
static void
slouken@10773
   673
SDL_CleanupAudioStreamResampler_SRC(SDL_AudioStream *stream)
slouken@10773
   674
{
icculus@10790
   675
    SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
slouken@10773
   676
    if (state) {
icculus@10790
   677
        SRC_src_delete(state);
slouken@10773
   678
    }
slouken@10773
   679
slouken@10773
   680
    stream->resampler_state = NULL;
slouken@10773
   681
    stream->resampler_func = NULL;
slouken@10773
   682
    stream->reset_resampler_func = NULL;
slouken@10773
   683
    stream->cleanup_resampler_func = NULL;
slouken@10773
   684
}
slouken@10773
   685
slouken@10773
   686
static SDL_bool
slouken@10773
   687
SetupLibSampleRateResampling(SDL_AudioStream *stream)
slouken@10773
   688
{
icculus@10790
   689
    int result = 0;
icculus@10790
   690
    SRC_STATE *state = NULL;
slouken@10773
   691
icculus@10790
   692
    if (SRC_available) {
icculus@10790
   693
        state = SRC_src_new(SRC_SINC_FASTEST, stream->pre_resample_channels, &result);
icculus@10790
   694
        if (!state) {
icculus@10790
   695
            SDL_SetError("src_new() failed: %s", SRC_src_strerror(result));
icculus@10790
   696
        }
slouken@10773
   697
    }
slouken@10773
   698
icculus@10790
   699
    if (!state) {
icculus@10790
   700
        SDL_CleanupAudioStreamResampler_SRC(stream);
slouken@10773
   701
        return SDL_FALSE;
slouken@10773
   702
    }
slouken@10773
   703
slouken@10773
   704
    stream->resampler_state = state;
slouken@10773
   705
    stream->resampler_func = SDL_ResampleAudioStream_SRC;
slouken@10773
   706
    stream->reset_resampler_func = SDL_ResetAudioStreamResampler_SRC;
slouken@10773
   707
    stream->cleanup_resampler_func = SDL_CleanupAudioStreamResampler_SRC;
slouken@10773
   708
slouken@10773
   709
    return SDL_TRUE;
slouken@10773
   710
}
icculus@10790
   711
#endif /* HAVE_LIBSAMPLERATE_H */
slouken@10773
   712
slouken@10773
   713
slouken@10773
   714
typedef struct
slouken@10773
   715
{
icculus@10757
   716
    SDL_bool resampler_seeded;
icculus@10757
   717
    float resampler_state[8];
slouken@10773
   718
} SDL_AudioStreamResamplerState;
slouken@10773
   719
slouken@10773
   720
static int
slouken@10773
   721
SDL_ResampleAudioStream(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
slouken@10773
   722
{
slouken@10773
   723
    SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
slouken@10773
   724
    const int chans = (int)stream->pre_resample_channels;
slouken@10773
   725
icculus@10799
   726
    SDL_assert(chans <= SDL_arraysize(state->resampler_state));
slouken@10773
   727
slouken@10773
   728
    if (!state->resampler_seeded) {
icculus@10799
   729
        int i;
slouken@10773
   730
        for (i = 0; i < chans; i++) {
slouken@10773
   731
            state->resampler_state[i] = inbuf[i];
slouken@10773
   732
        }
slouken@10773
   733
        state->resampler_seeded = SDL_TRUE;
slouken@10773
   734
    }
slouken@10773
   735
icculus@10799
   736
    return SDL_ResampleAudioSimple(chans, stream->rate_incr, state->resampler_state, inbuf, inbuflen, outbuf, outbuflen);
slouken@10773
   737
}
slouken@10773
   738
slouken@10773
   739
static void
slouken@10773
   740
SDL_ResetAudioStreamResampler(SDL_AudioStream *stream)
slouken@10773
   741
{
slouken@10773
   742
    SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
slouken@10773
   743
    state->resampler_seeded = SDL_FALSE;
slouken@10773
   744
}
slouken@10773
   745
slouken@10773
   746
static void
slouken@10773
   747
SDL_CleanupAudioStreamResampler(SDL_AudioStream *stream)
slouken@10773
   748
{
slouken@10773
   749
    SDL_free(stream->resampler_state);
slouken@10773
   750
}
icculus@10757
   751
icculus@10789
   752
SDL_AudioStream *
icculus@10789
   753
SDL_NewAudioStream(const SDL_AudioFormat src_format,
icculus@10789
   754
                   const Uint8 src_channels,
icculus@10789
   755
                   const int src_rate,
icculus@10789
   756
                   const SDL_AudioFormat dst_format,
icculus@10789
   757
                   const Uint8 dst_channels,
icculus@10789
   758
                   const int dst_rate)
icculus@10757
   759
{
icculus@10757
   760
    const int packetlen = 4096;  /* !!! FIXME: good enough for now. */
icculus@10757
   761
    Uint8 pre_resample_channels;
icculus@10757
   762
    SDL_AudioStream *retval;
icculus@10757
   763
icculus@10757
   764
    retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
icculus@10757
   765
    if (!retval) {
icculus@10757
   766
        return NULL;
icculus@10757
   767
    }
icculus@10757
   768
icculus@10757
   769
    /* If increasing channels, do it after resampling, since we'd just
icculus@10757
   770
       do more work to resample duplicate channels. If we're decreasing, do
icculus@10757
   771
       it first so we resample the interpolated data instead of interpolating
icculus@10757
   772
       the resampled data (!!! FIXME: decide if that works in practice, though!). */
icculus@10757
   773
    pre_resample_channels = SDL_min(src_channels, dst_channels);
icculus@10757
   774
icculus@10757
   775
    retval->src_sample_frame_size = SDL_AUDIO_BITSIZE(src_format) * src_channels;
icculus@10757
   776
    retval->src_format = src_format;
icculus@10757
   777
    retval->src_channels = src_channels;
icculus@10757
   778
    retval->src_rate = src_rate;
icculus@10757
   779
    retval->dst_sample_frame_size = SDL_AUDIO_BITSIZE(dst_format) * dst_channels;
icculus@10757
   780
    retval->dst_format = dst_format;
icculus@10757
   781
    retval->dst_channels = dst_channels;
icculus@10757
   782
    retval->dst_rate = dst_rate;
icculus@10757
   783
    retval->pre_resample_channels = pre_resample_channels;
icculus@10757
   784
    retval->packetlen = packetlen;
icculus@10757
   785
    retval->rate_incr = ((double) dst_rate) / ((double) src_rate);
icculus@10757
   786
icculus@10757
   787
    /* Not resampling? It's an easy conversion (and maybe not even that!). */
icculus@10757
   788
    if (src_rate == dst_rate) {
icculus@10757
   789
        retval->cvt_before_resampling.needed = SDL_FALSE;
icculus@10757
   790
        retval->cvt_before_resampling.len_mult = 1;
slouken@10773
   791
        if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
slouken@10773
   792
            SDL_FreeAudioStream(retval);
icculus@10757
   793
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
icculus@10757
   794
        }
icculus@10757
   795
    } else {
icculus@10757
   796
        /* Don't resample at first. Just get us to Float32 format. */
icculus@10757
   797
        /* !!! FIXME: convert to int32 on devices without hardware float. */
slouken@10773
   798
        if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) < 0) {
slouken@10773
   799
            SDL_FreeAudioStream(retval);
icculus@10757
   800
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
icculus@10757
   801
        }
icculus@10757
   802
slouken@10777
   803
#ifdef HAVE_LIBSAMPLERATE_H
slouken@10773
   804
        SetupLibSampleRateResampling(retval);
slouken@10773
   805
#endif
slouken@10773
   806
slouken@10773
   807
        if (!retval->resampler_func) {
slouken@10773
   808
            retval->resampler_state = SDL_calloc(1, sizeof(SDL_AudioStreamResamplerState));
slouken@10773
   809
            if (!retval->resampler_state) {
slouken@10773
   810
                SDL_FreeAudioStream(retval);
slouken@10773
   811
                SDL_OutOfMemory();
slouken@10773
   812
                return NULL;
slouken@10773
   813
            }
slouken@10773
   814
            retval->resampler_func = SDL_ResampleAudioStream;
slouken@10773
   815
            retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
slouken@10773
   816
            retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
slouken@10773
   817
        }
slouken@10773
   818
icculus@10757
   819
        /* Convert us to the final format after resampling. */
slouken@10773
   820
        if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
slouken@10773
   821
            SDL_FreeAudioStream(retval);
icculus@10757
   822
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
icculus@10757
   823
        }
icculus@10757
   824
    }
icculus@10757
   825
icculus@10757
   826
    retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2);
icculus@10757
   827
    if (!retval->queue) {
slouken@10773
   828
        SDL_FreeAudioStream(retval);
icculus@10757
   829
        return NULL;  /* SDL_NewDataQueue should have called SDL_SetError. */
icculus@10757
   830
    }
icculus@10757
   831
icculus@10757
   832
    return retval;
icculus@10757
   833
}
icculus@10757
   834
icculus@10757
   835
static Uint8 *
icculus@10757
   836
EnsureBufferSize(Uint8 **buf, int *len, const int newlen)
icculus@10757
   837
{
icculus@10757
   838
    if (*len < newlen) {
icculus@10757
   839
        void *ptr = SDL_realloc(*buf, newlen);
icculus@10757
   840
        if (!ptr) {
icculus@10757
   841
            SDL_OutOfMemory();
icculus@10757
   842
            return NULL;
icculus@10757
   843
        }
icculus@10757
   844
        *buf = (Uint8 *) ptr;
icculus@10757
   845
        *len = newlen;
icculus@10757
   846
    }
icculus@10757
   847
    return *buf;
icculus@10757
   848
}
icculus@10757
   849
icculus@10757
   850
int
icculus@10757
   851
SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, const Uint32 _buflen)
icculus@10757
   852
{
icculus@10757
   853
    int buflen = (int) _buflen;
icculus@10757
   854
icculus@10757
   855
    if (!stream) {
icculus@10757
   856
        return SDL_InvalidParamError("stream");
icculus@10757
   857
    } else if (!buf) {
icculus@10757
   858
        return SDL_InvalidParamError("buf");
icculus@10757
   859
    } else if (buflen == 0) {
icculus@10757
   860
        return 0;  /* nothing to do. */
icculus@10757
   861
    } else if ((buflen % stream->src_sample_frame_size) != 0) {
icculus@10757
   862
        return SDL_SetError("Can't add partial sample frames");
icculus@10757
   863
    }
icculus@10757
   864
icculus@10757
   865
    if (stream->cvt_before_resampling.needed) {
icculus@10757
   866
        const int workbuflen = buflen * stream->cvt_before_resampling.len_mult;  /* will be "* 1" if not needed */
icculus@10757
   867
        Uint8 *workbuf = EnsureBufferSize(&stream->work_buffer, &stream->work_buffer_len, workbuflen);
icculus@10757
   868
        if (workbuf == NULL) {
icculus@10757
   869
            return -1;  /* probably out of memory. */
icculus@10757
   870
        }
icculus@10757
   871
        SDL_memcpy(workbuf, buf, buflen);
icculus@10757
   872
        stream->cvt_before_resampling.buf = workbuf;
icculus@10757
   873
        stream->cvt_before_resampling.len = buflen;
icculus@10757
   874
        if (SDL_ConvertAudio(&stream->cvt_before_resampling) == -1) {
icculus@10757
   875
            return -1;   /* uhoh! */
icculus@10757
   876
        }
icculus@10757
   877
        buf = workbuf;
icculus@10757
   878
        buflen = stream->cvt_before_resampling.len_cvt;
icculus@10757
   879
    }
icculus@10757
   880
icculus@10757
   881
    if (stream->dst_rate != stream->src_rate) {
icculus@10757
   882
        const int workbuflen = buflen * ((int) SDL_ceil(stream->rate_incr));
icculus@10757
   883
        float *workbuf = (float *) EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
icculus@10757
   884
        if (workbuf == NULL) {
icculus@10757
   885
            return -1;  /* probably out of memory. */
icculus@10757
   886
        }
slouken@10773
   887
        buflen = stream->resampler_func(stream, (float *) buf, buflen, workbuf, workbuflen);
icculus@10757
   888
        buf = workbuf;
icculus@10757
   889
    }
icculus@10757
   890
icculus@10757
   891
    if (stream->cvt_after_resampling.needed) {
icculus@10757
   892
        const int workbuflen = buflen * stream->cvt_before_resampling.len_mult;  /* will be "* 1" if not needed */
icculus@10757
   893
        Uint8 *workbuf;
icculus@10757
   894
icculus@10757
   895
        if (buf == stream->resample_buffer) {
icculus@10757
   896
            workbuf = EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
icculus@10757
   897
        } else {
icculus@10757
   898
            const int inplace = (buf == stream->work_buffer);
icculus@10757
   899
            workbuf = EnsureBufferSize(&stream->work_buffer, &stream->work_buffer_len, workbuflen);
icculus@10757
   900
            if (workbuf && !inplace) {
icculus@10757
   901
                SDL_memcpy(workbuf, buf, buflen);
icculus@10757
   902
            }
icculus@10757
   903
        }
icculus@10757
   904
icculus@10757
   905
        if (workbuf == NULL) {
icculus@10757
   906
            return -1;  /* probably out of memory. */
icculus@10757
   907
        }
icculus@10757
   908
icculus@10757
   909
        stream->cvt_after_resampling.buf = workbuf;
icculus@10757
   910
        stream->cvt_after_resampling.len = buflen;
icculus@10757
   911
        if (SDL_ConvertAudio(&stream->cvt_after_resampling) == -1) {
icculus@10757
   912
            return -1;   /* uhoh! */
icculus@10757
   913
        }
icculus@10757
   914
        buf = workbuf;
icculus@10757
   915
        buflen = stream->cvt_after_resampling.len_cvt;
icculus@10757
   916
    }
icculus@10757
   917
icculus@10757
   918
    return SDL_WriteToDataQueue(stream->queue, buf, buflen);
icculus@10757
   919
}
icculus@10757
   920
icculus@10757
   921
void
icculus@10757
   922
SDL_AudioStreamClear(SDL_AudioStream *stream)
icculus@10757
   923
{
icculus@10757
   924
    if (!stream) {
icculus@10757
   925
        SDL_InvalidParamError("stream");
icculus@10757
   926
    } else {
icculus@10757
   927
        SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
icculus@10776
   928
        if (stream->reset_resampler_func) {
icculus@10776
   929
            stream->reset_resampler_func(stream);
icculus@10776
   930
        }
icculus@10757
   931
    }
icculus@10757
   932
}
icculus@10757
   933
icculus@10757
   934
icculus@10757
   935
/* get converted/resampled data from the stream */
icculus@10757
   936
int
icculus@10764
   937
SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, const Uint32 len)
icculus@10757
   938
{
icculus@10757
   939
    if (!stream) {
icculus@10757
   940
        return SDL_InvalidParamError("stream");
icculus@10757
   941
    } else if (!buf) {
icculus@10757
   942
        return SDL_InvalidParamError("buf");
icculus@10757
   943
    } else if (len == 0) {
icculus@10757
   944
        return 0;  /* nothing to do. */
icculus@10757
   945
    } else if ((len % stream->dst_sample_frame_size) != 0) {
icculus@10757
   946
        return SDL_SetError("Can't request partial sample frames");
icculus@10757
   947
    }
icculus@10757
   948
icculus@10764
   949
    return (int) SDL_ReadFromDataQueue(stream->queue, buf, len);
icculus@10757
   950
}
icculus@10757
   951
icculus@10757
   952
/* number of converted/resampled bytes available */
icculus@10757
   953
int
icculus@10757
   954
SDL_AudioStreamAvailable(SDL_AudioStream *stream)
icculus@10757
   955
{
icculus@10757
   956
    return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
icculus@10757
   957
}
icculus@10757
   958
icculus@10757
   959
/* dispose of a stream */
icculus@10757
   960
void
icculus@10757
   961
SDL_FreeAudioStream(SDL_AudioStream *stream)
icculus@10757
   962
{
icculus@10757
   963
    if (stream) {
slouken@10773
   964
        if (stream->cleanup_resampler_func) {
slouken@10773
   965
            stream->cleanup_resampler_func(stream);
slouken@10773
   966
        }
icculus@10757
   967
        SDL_FreeDataQueue(stream->queue);
icculus@10757
   968
        SDL_free(stream->work_buffer);
icculus@10757
   969
        SDL_free(stream->resample_buffer);
icculus@10757
   970
        SDL_free(stream);
icculus@10757
   971
    }
icculus@10757
   972
}
icculus@10757
   973
icculus@10575
   974
/* vi: set ts=4 sw=4 expandtab: */
slouken@2716
   975