src/audio/SDL_audiocvt.c

/*
  Simple DirectMedia Layer
  Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>

  This software is provided 'as-is', without any express or implied
  warranty.  In no event will the authors be held liable for any damages
  arising from the use of this software.

  Permission is granted to anyone to use this software for any purpose,
  including commercial applications, and to alter it and redistribute it
  freely, subject to the following restrictions:

  1. The origin of this software must not be misrepresented; you must not
     claim that you wrote the original software. If you use this software
     in a product, an acknowledgment in the product documentation would be
     appreciated but is not required.
  2. Altered source versions must be plainly marked as such, and must not be
     misrepresented as being the original software.
  3. This notice may not be removed or altered from any source distribution.
*/
#include "../SDL_internal.h"

/* Functions for audio drivers to perform runtime conversion of audio format */

#include "SDL_audio.h"
#include "SDL_audio_c.h"

#include "SDL_loadso.h"
#include "SDL_assert.h"
#include "../SDL_dataqueue.h"

#ifdef HAVE_LIBSAMPLERATE
#include "samplerate.h"
#endif


/* Effectively mix right and left channels into a single channel */
static void SDLCALL
SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    float *dst = (float *) cvt->buf;
    const float *src = dst;
    int i;

    LOG_DEBUG_CONVERT("stereo", "mono");
    SDL_assert(format == AUDIO_F32SYS);

    for (i = cvt->len_cvt / 8; i; --i, src += 2) {
        *(dst++) = (float) ((((double) src[0]) + ((double) src[1])) * 0.5);
    }

    cvt->len_cvt /= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}


/* Discard top 4 channels */
static void SDLCALL
SDL_ConvertStrip(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    float *dst = (float *) cvt->buf;
    const float *src = dst;
    int i;

    LOG_DEBUG_CONVERT("6 channels", "stereo");
    SDL_assert(format == AUDIO_F32SYS);

    for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 2) {
        dst[0] = src[0];
        dst[1] = src[1];
    }

    cvt->len_cvt /= 3;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}


/* Discard top 2 channels of 6 */
static void SDLCALL
SDL_ConvertStrip_2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    float *dst = (float *) cvt->buf;
    const float *src = dst;
    int i;

    LOG_DEBUG_CONVERT("6 channels", "quad");
    SDL_assert(format == AUDIO_F32SYS);

    for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 4) {
        dst[0] = src[0];
        dst[1] = src[1];
        dst[2] = src[2];
        dst[3] = src[3];
    }

    cvt->len_cvt /= 6;
    cvt->len_cvt *= 4;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}

/* Duplicate a mono channel to both stereo channels */
static void SDLCALL
SDL_ConvertStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    const float *src = (const float *) (cvt->buf + cvt->len_cvt);
    float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
    int i;

    LOG_DEBUG_CONVERT("mono", "stereo");
    SDL_assert(format == AUDIO_F32SYS);

    for (i = cvt->len_cvt / sizeof (float); i; --i) {
        src--;
        dst -= 2;
        dst[0] = dst[1] = *src;
    }

    cvt->len_cvt *= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}


/* Duplicate a stereo channel to a pseudo-5.1 stream */
static void SDLCALL
SDL_ConvertSurround(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;
    float lf, rf, ce;
    const float *src = (const float *) (cvt->buf + cvt->len_cvt);
    float *dst = (float *) (cvt->buf + cvt->len_cvt * 3);

    LOG_DEBUG_CONVERT("stereo", "5.1");
    SDL_assert(format == AUDIO_F32SYS);

    for (i = cvt->len_cvt / 8; i; --i) {
        dst -= 6;
        src -= 2;
        lf = src[0];
        rf = src[1];
        ce = (lf * 0.5f) + (rf * 0.5f);
        dst[0] = src[0];
        dst[1] = src[1];
        dst[2] = lf - ce;
        dst[3] = rf - ce;
        dst[4] = dst[5] = ce;
    }

    cvt->len_cvt *= 3;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}


/* Duplicate a stereo channel to a pseudo-4.0 stream */
static void SDLCALL
SDL_ConvertSurround_4(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    const float *src = (const float *) (cvt->buf + cvt->len_cvt);
    float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
    float lf, rf, ce;
    int i;

    LOG_DEBUG_CONVERT("stereo", "quad");
    SDL_assert(format == AUDIO_F32SYS);

    for (i = cvt->len_cvt / 8; i; --i) {
        dst -= 4;
        src -= 2;
        lf = src[0];
        rf = src[1];
        ce = (lf / 2) + (rf / 2);
        dst[0] = src[0];
        dst[1] = src[1];
        dst[2] = lf - ce;
        dst[3] = rf - ce;
    }

    cvt->len_cvt *= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}


int
SDL_ConvertAudio(SDL_AudioCVT * cvt)
{
    /* !!! FIXME: (cvt) should be const; stack-copy it here. */
    /* !!! FIXME: (actually, we can't...len_cvt needs to be updated. Grr.) */

    /* Make sure there's data to convert */
    if (cvt->buf == NULL) {
        return SDL_SetError("No buffer allocated for conversion");
    }

    /* Return okay if no conversion is necessary */
    cvt->len_cvt = cvt->len;
    if (cvt->filters[0] == NULL) {
        return 0;
    }

    /* Set up the conversion and go! */
    cvt->filter_index = 0;
    cvt->filters[0] (cvt, cvt->src_format);
    return 0;
}

static void SDLCALL
SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
#if DEBUG_CONVERT
    printf("Converting byte order\n");
#endif

    switch (SDL_AUDIO_BITSIZE(format)) {
        #define CASESWAP(b) \
            case b: { \
                Uint##b *ptr = (Uint##b *) cvt->buf; \
                int i; \
                for (i = cvt->len_cvt / sizeof (*ptr); i; --i, ++ptr) { \
                    *ptr = SDL_Swap##b(*ptr); \
                } \
                break; \
            }

        CASESWAP(16);
        CASESWAP(32);
        CASESWAP(64);

        #undef CASESWAP

        default: SDL_assert(!"unhandled byteswap datatype!"); break;
    }

    if (cvt->filters[++cvt->filter_index]) {
        /* flip endian flag for data. */
        if (format & SDL_AUDIO_MASK_ENDIAN) {
            format &= ~SDL_AUDIO_MASK_ENDIAN;
        } else {
            format |= SDL_AUDIO_MASK_ENDIAN;
        }
        cvt->filters[cvt->filter_index](cvt, format);
    }
}


static int
SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
{
    int retval = 0;  /* 0 == no conversion necessary. */

    if ((SDL_AUDIO_ISBIGENDIAN(src_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
        cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
        retval = 1;  /* added a converter. */
    }

    if (!SDL_AUDIO_ISFLOAT(src_fmt)) {
        const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
        const Uint16 dst_bitsize = 32;
        SDL_AudioFilter filter = NULL;

        switch (src_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
            case AUDIO_S8: filter = SDL_Convert_S8_to_F32; break;
            case AUDIO_U8: filter = SDL_Convert_U8_to_F32; break;
            case AUDIO_S16: filter = SDL_Convert_S16_to_F32; break;
            case AUDIO_U16: filter = SDL_Convert_U16_to_F32; break;
            case AUDIO_S32: filter = SDL_Convert_S32_to_F32; break;
            default: SDL_assert(!"Unexpected audio format!"); break;
        }

        if (!filter) {
            return SDL_SetError("No conversion available for these formats");
        }

        cvt->filters[cvt->filter_index++] = filter;
        if (src_bitsize < dst_bitsize) {
            const int mult = (dst_bitsize / src_bitsize);
            cvt->len_mult *= mult;
            cvt->len_ratio *= mult;
        } else if (src_bitsize > dst_bitsize) {
            cvt->len_ratio /= (src_bitsize / dst_bitsize);
        }

        retval = 1;  /* added a converter. */
    }

    return retval;
}

static int
SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt)
{
    int retval = 0;  /* 0 == no conversion necessary. */

    if (!SDL_AUDIO_ISFLOAT(dst_fmt)) {
        const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
        const Uint16 src_bitsize = 32;
        SDL_AudioFilter filter = NULL;
        switch (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
            case AUDIO_S8: filter = SDL_Convert_F32_to_S8; break;
            case AUDIO_U8: filter = SDL_Convert_F32_to_U8; break;
            case AUDIO_S16: filter = SDL_Convert_F32_to_S16; break;
            case AUDIO_U16: filter = SDL_Convert_F32_to_U16; break;
            case AUDIO_S32: filter = SDL_Convert_F32_to_S32; break;
            default: SDL_assert(!"Unexpected audio format!"); break;
        }

        if (!filter) {
            return SDL_SetError("No conversion available for these formats");
        }

        cvt->filters[cvt->filter_index++] = filter;
        if (src_bitsize < dst_bitsize) {
            const int mult = (dst_bitsize / src_bitsize);
            cvt->len_mult *= mult;
            cvt->len_ratio *= mult;
        } else if (src_bitsize > dst_bitsize) {
            cvt->len_ratio /= (src_bitsize / dst_bitsize);
        }
        retval = 1;  /* added a converter. */
    }

    if ((SDL_AUDIO_ISBIGENDIAN(dst_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
        cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
        retval = 1;  /* added a converter. */
    }

    return retval;
}


/* !!! FIXME: We only have this macro salsa because SDL_AudioCVT doesn't store
   !!! FIXME:  channel info or integer sample rates, so we have to have
   !!! FIXME:  function entry points for each supported channel count and
   !!! FIXME:  multiple vs arbitrary. When we rev the ABI, remove this. */
#define RESAMPLER_FUNCS(chans) \
    static void SDLCALL \
    SDL_Upsample_Multiple_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
        SDL_assert(format == AUDIO_F32SYS); \
        SDL_Upsample_Multiple(cvt, chans); \
    } \
    static void SDLCALL \
    SDL_Upsample_Arbitrary_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
        SDL_assert(format == AUDIO_F32SYS); \
        SDL_Upsample_Arbitrary(cvt, chans); \
    }\
    static void SDLCALL \
    SDL_Downsample_Multiple_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
        SDL_assert(format == AUDIO_F32SYS); \
        SDL_Downsample_Multiple(cvt, chans); \
    } \
    static void SDLCALL \
    SDL_Downsample_Arbitrary_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
        SDL_assert(format == AUDIO_F32SYS); \
        SDL_Downsample_Arbitrary(cvt, chans); \
    }
RESAMPLER_FUNCS(1)
RESAMPLER_FUNCS(2)
RESAMPLER_FUNCS(4)
RESAMPLER_FUNCS(6)
RESAMPLER_FUNCS(8)
#undef RESAMPLER_FUNCS

static int
SDL_FindFrequencyMultiple(const int src_rate, const int dst_rate)
{
    int lo, hi;

    SDL_assert(src_rate != 0);
    SDL_assert(dst_rate != 0);
    SDL_assert(src_rate != dst_rate);

    if (src_rate < dst_rate) {
        lo = src_rate;
        hi = dst_rate;
    } else {
        lo = dst_rate;
        hi = src_rate;
    }

    if ((hi % lo) != 0)
        return 0;               /* not a multiple. */

    return hi / lo;
}

static SDL_AudioFilter
ChooseResampler(const int dst_channels, const int src_rate, const int dst_rate)
{
    const int upsample = (src_rate < dst_rate) ? 1 : 0;
    const int multiple = SDL_FindFrequencyMultiple(src_rate, dst_rate);
    SDL_AudioFilter filter = NULL;

    #define PICK_CHANNEL_FILTER(upordown, resampler) switch (dst_channels) { \
        case 1: filter = SDL_##upordown##_##resampler##_c1; break; \
        case 2: filter = SDL_##upordown##_##resampler##_c2; break; \
        case 4: filter = SDL_##upordown##_##resampler##_c4; break; \
        case 6: filter = SDL_##upordown##_##resampler##_c6; break; \
        case 8: filter = SDL_##upordown##_##resampler##_c8; break; \
        default: break; \
    }

    if (upsample) {
        if (multiple) {
            PICK_CHANNEL_FILTER(Upsample, Multiple);
        } else {
            PICK_CHANNEL_FILTER(Upsample, Arbitrary);
        }
    } else {
        if (multiple) {
            PICK_CHANNEL_FILTER(Downsample, Multiple);
        } else {
            PICK_CHANNEL_FILTER(Downsample, Arbitrary);
        }
    }

    #undef PICK_CHANNEL_FILTER

    return filter;
}

static int
SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, const int dst_channels,
                          const int src_rate, const int dst_rate)
{
    SDL_AudioFilter filter;

    if (src_rate == dst_rate) {
        return 0;  /* no conversion necessary. */
    }

    filter = ChooseResampler(dst_channels, src_rate, dst_rate);
    if (filter == NULL) {
        return SDL_SetError("No conversion available for these rates");
    }

    /* Update (cvt) with filter details... */
    cvt->filters[cvt->filter_index++] = filter;
    if (src_rate < dst_rate) {
        const double mult = ((double) dst_rate) / ((double) src_rate);
        cvt->len_mult *= (int) SDL_ceil(mult);
        cvt->len_ratio *= mult;
    } else {
        cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate);
    }

    return 1;               /* added a converter. */
}


/* Creates a set of audio filters to convert from one format to another.
   Returns -1 if the format conversion is not supported, 0 if there's
   no conversion needed, or 1 if the audio filter is set up.
*/

int
SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
                  SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate,
                  SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
{
    /* Sanity check target pointer */
    if (cvt == NULL) {
        return SDL_InvalidParamError("cvt");
    }

    /* Make sure we zero out the audio conversion before error checking */
    SDL_zerop(cvt);

    /* there are no unsigned types over 16 bits, so catch this up front. */
    if ((SDL_AUDIO_BITSIZE(src_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(src_fmt))) {
        return SDL_SetError("Invalid source format");
    }
    if ((SDL_AUDIO_BITSIZE(dst_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(dst_fmt))) {
        return SDL_SetError("Invalid destination format");
    }

    /* prevent possible divisions by zero, etc. */
    if ((src_channels == 0) || (dst_channels == 0)) {
        return SDL_SetError("Source or destination channels is zero");
    }
    if ((src_rate == 0) || (dst_rate == 0)) {
        return SDL_SetError("Source or destination rate is zero");
    }
#if DEBUG_CONVERT
    printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
           src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
#endif

    /* Start off with no conversion necessary */
    cvt->src_format = src_fmt;
    cvt->dst_format = dst_fmt;
    cvt->needed = 0;
    cvt->filter_index = 0;
    cvt->filters[0] = NULL;
    cvt->len_mult = 1;
    cvt->len_ratio = 1.0;
    cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);

    /* Type conversion goes like this now:
        - byteswap to CPU native format first if necessary.
        - convert to native Float32 if necessary.
        - resample and change channel count if necessary.
        - convert back to native format.
        - byteswap back to foreign format if necessary.

       The expectation is we can process data faster in float32
       (possibly with SIMD), and making several passes over the same
       buffer is likely to be CPU cache-friendly, avoiding the
       biggest performance hit in modern times. Previously we had
       (script-generated) custom converters for every data type and
       it was a bloat on SDL compile times and final library size. */

    /* see if we can skip float conversion entirely. */
    if (src_rate == dst_rate && src_channels == dst_channels) {
        if (src_fmt == dst_fmt) {
            return 0;
        }

        /* just a byteswap needed? */
        if ((src_fmt & ~SDL_AUDIO_MASK_ENDIAN) == (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN)) {
            cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
            cvt->needed = 1;
            return 1;
        }
    }

    /* Convert data types, if necessary. Updates (cvt). */
    if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
        return -1;              /* shouldn't happen, but just in case... */
    }

    /* Channel conversion */
    if (src_channels != dst_channels) {
        if ((src_channels == 1) && (dst_channels > 1)) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertStereo;
            cvt->len_mult *= 2;
            src_channels = 2;
            cvt->len_ratio *= 2;
        }
        if ((src_channels == 2) && (dst_channels == 6)) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertSurround;
            src_channels = 6;
            cvt->len_mult *= 3;
            cvt->len_ratio *= 3;
        }
        if ((src_channels == 2) && (dst_channels == 4)) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertSurround_4;
            src_channels = 4;
            cvt->len_mult *= 2;
            cvt->len_ratio *= 2;
        }
        while ((src_channels * 2) <= dst_channels) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertStereo;
            cvt->len_mult *= 2;
            src_channels *= 2;
            cvt->len_ratio *= 2;
        }
        if ((src_channels == 6) && (dst_channels <= 2)) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertStrip;
            src_channels = 2;
            cvt->len_ratio /= 3;
        }
        if ((src_channels == 6) && (dst_channels == 4)) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertStrip_2;
            src_channels = 4;
            cvt->len_ratio /= 2;
        }
        /* This assumes that 4 channel audio is in the format:
           Left {front/back} + Right {front/back}
           so converting to L/R stereo works properly.
         */
        while (((src_channels % 2) == 0) &&
               ((src_channels / 2) >= dst_channels)) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertMono;
            src_channels /= 2;
            cvt->len_ratio /= 2;
        }
        if (src_channels != dst_channels) {
            /* Uh oh.. */ ;
        }
    }

    /* Do rate conversion, if necessary. Updates (cvt). */
    if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) < 0) {
        return -1;              /* shouldn't happen, but just in case... */
    }

    /* Move to final data type. */
    if (SDL_BuildAudioTypeCVTFromFloat(cvt, dst_fmt) < 0) {
        return -1;              /* shouldn't happen, but just in case... */
    }

    cvt->needed = (cvt->filter_index != 0);
    return (cvt->needed);
}

typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen);
typedef void (*SDL_ResetAudioStreamResamplerFunc)(SDL_AudioStream *stream);
typedef void (*SDL_CleanupAudioStreamResamplerFunc)(SDL_AudioStream *stream);

struct SDL_AudioStream
{
    SDL_AudioCVT cvt_before_resampling;
    SDL_AudioCVT cvt_after_resampling;
    SDL_DataQueue *queue;
    Uint8 *work_buffer;
    int work_buffer_len;
    Uint8 *resample_buffer;
    int resample_buffer_len;
    int src_sample_frame_size;
    SDL_AudioFormat src_format;
    Uint8 src_channels;
    int src_rate;
    int dst_sample_frame_size;
    SDL_AudioFormat dst_format;
    Uint8 dst_channels;
    int dst_rate;
    double rate_incr;
    Uint8 pre_resample_channels;
    int packetlen;
    void *resampler_state;
    SDL_ResampleAudioStreamFunc resampler_func;
    SDL_ResetAudioStreamResamplerFunc reset_resampler_func;
    SDL_CleanupAudioStreamResamplerFunc cleanup_resampler_func;
};

#ifdef HAVE_LIBSAMPLERATE

typedef struct
{
    void *SRC_lib;

    SRC_STATE* (*src_new)(int converter_type, int channels, int *error);
    int (*src_process)(SRC_STATE *state, SRC_DATA *data);
    int (*src_reset)(SRC_STATE *state);
    SRC_STATE* (*src_delete)(SRC_STATE *state);
    const char* (*src_strerror)(int error);

    SRC_STATE *SRC_state;
} SDL_AudioStreamResamplerState_SRC;

static SDL_bool
LoadLibSampleRate(SDL_AudioStreamResamplerState_SRC *state)
{
#ifdef LIBSAMPLERATE_DYNAMIC
    state->SRC_lib = SDL_LoadObject(LIBSAMPLERATE_DYNAMIC);
    if (!state->SRC_lib) {
        return SDL_FALSE;
    }
#endif

    state->src_new = (SRC_STATE* (*)(int converter_type, int channels, int *error))SDL_LoadFunction(state->SRC_lib, "src_new");
    state->src_process = (int (*)(SRC_STATE *state, SRC_DATA *data))SDL_LoadFunction(state->SRC_lib, "src_process");
    state->src_reset = (int(*)(SRC_STATE *state))SDL_LoadFunction(state->SRC_lib, "src_reset");
    state->src_delete = (SRC_STATE* (*)(SRC_STATE *state))SDL_LoadFunction(state->SRC_lib, "src_delete");
    state->src_strerror = (const char* (*)(int error))SDL_LoadFunction(state->SRC_lib, "src_strerror");
    if (!state->src_new || !state->src_process || !state->src_reset || !state->src_delete || !state->src_strerror) {
        return SDL_FALSE;
    }
    return SDL_TRUE;
}

static int
SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
{
    SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC*)stream->resampler_state;
    SRC_DATA data;
    int result;

    data.data_in = inbuf;
    data.input_frames = inbuflen / ( sizeof(float) * stream->pre_resample_channels );
    data.input_frames_used = 0;

    data.data_out = outbuf;
    data.output_frames = outbuflen / (sizeof(float) * stream->pre_resample_channels);

    data.end_of_input = 0;
    data.src_ratio = stream->rate_incr;

    result = state->src_process(state->SRC_state, &data);
    if (result != 0) {
        SDL_SetError("src_process() failed: %s", state->src_strerror(result));
        return 0;
    }

    /* If this fails, we need to store them off somewhere */
    SDL_assert(data.input_frames_used == data.input_frames);

    return data.output_frames_gen * (sizeof(float) * stream->pre_resample_channels);
}

static void
SDL_ResetAudioStreamResampler_SRC(SDL_AudioStream *stream)
{
    SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC*)stream->resampler_state;
    state->src_reset(state->SRC_state);
}

static void
SDL_CleanupAudioStreamResampler_SRC(SDL_AudioStream *stream)
{
    SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC*)stream->resampler_state;
    if (state) {
        if (state->SRC_lib) {
            SDL_UnloadObject(state->SRC_lib);
        }
        state->src_delete(state->SRC_state);
        SDL_free(state);
    }

    stream->resampler_state = NULL;
    stream->resampler_func = NULL;
    stream->reset_resampler_func = NULL;
    stream->cleanup_resampler_func = NULL;
}

static SDL_bool
SetupLibSampleRateResampling(SDL_AudioStream *stream)
{
    int result;

    SDL_AudioStreamResamplerState_SRC *state = (SDL_AudioStreamResamplerState_SRC *)SDL_calloc(1, sizeof(*state));
    if (!state) {
        return SDL_FALSE;
    }

    if (!LoadLibSampleRate(state)) {
        SDL_free(state);
        return SDL_FALSE;
    }

    stream->resampler_state = state;
    stream->resampler_func = SDL_ResampleAudioStream_SRC;
    stream->reset_resampler_func = SDL_ResetAudioStreamResampler_SRC;
    stream->cleanup_resampler_func = SDL_CleanupAudioStreamResampler_SRC;

    state->SRC_state = state->src_new(SRC_SINC_FASTEST, stream->pre_resample_channels, &result);
    if (!state->SRC_state) {
        SDL_SetError("src_new() failed: %s", state->src_strerror(result));
        SDL_CleanupAudioStreamResampler_SRC(stream);
        return SDL_FALSE;
    }
    return SDL_TRUE;
}

#endif /* HAVE_LIBSAMPLERATE */

typedef struct
{
    SDL_bool resampler_seeded;
    float resampler_state[8];
} SDL_AudioStreamResamplerState;

static int
SDL_ResampleAudioStream(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
{
    /* !!! FIXME: this resampler sucks, but not much worse than our usual resampler.  :)  */  /* ... :( */
    SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
    const int chans = (int)stream->pre_resample_channels;
    const int framelen = chans * sizeof(float);
    const int total = (inbuflen / framelen);
    const int finalpos = total - chans;
    const double src_incr = 1.0 / stream->rate_incr;
    double idx = 0.0;
    float *dst = outbuf;
    float last_sample[SDL_arraysize(state->resampler_state)];
    int consumed = 0;
    int i;

    SDL_assert(chans <= SDL_arraysize(last_sample));
    SDL_assert((inbuflen % framelen) == 0);

    if (!state->resampler_seeded) {
        for (i = 0; i < chans; i++) {
            state->resampler_state[i] = inbuf[i];
        }
        state->resampler_seeded = SDL_TRUE;
    }

    for (i = 0; i < chans; i++) {
        last_sample[i] = state->resampler_state[i];
    }

    while (consumed < total) {
        const int pos = ((int)idx) * chans;
        const float *src = &inbuf[(pos >= finalpos) ? finalpos : pos];
        SDL_assert(dst < (outbuf + (outbuflen / framelen)));
        for (i = 0; i < chans; i++) {
            const float val = *(src++);
            *(dst++) = (val + last_sample[i]) * 0.5f;
            last_sample[i] = val;
        }
        consumed = pos + chans;
        idx += src_incr;
    }

    for (i = 0; i < chans; i++) {
        state->resampler_state[i] = last_sample[i];
    }

    return (int)((dst - outbuf) * sizeof(float));
}

static void
SDL_ResetAudioStreamResampler(SDL_AudioStream *stream)
{
    SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
    state->resampler_seeded = SDL_FALSE;
}

static void
SDL_CleanupAudioStreamResampler(SDL_AudioStream *stream)
{
    SDL_free(stream->resampler_state);
}

SDL_AudioStream *SDL_NewAudioStream(const SDL_AudioFormat src_format,
                                    const Uint8 src_channels,
                                    const int src_rate,
                                    const SDL_AudioFormat dst_format,
                                    const Uint8 dst_channels,
                                    const int dst_rate)
{
    const int packetlen = 4096;  /* !!! FIXME: good enough for now. */
    Uint8 pre_resample_channels;
    SDL_AudioStream *retval;

    retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
    if (!retval) {
        return NULL;
    }

    /* If increasing channels, do it after resampling, since we'd just
       do more work to resample duplicate channels. If we're decreasing, do
       it first so we resample the interpolated data instead of interpolating
       the resampled data (!!! FIXME: decide if that works in practice, though!). */
    pre_resample_channels = SDL_min(src_channels, dst_channels);

    retval->src_sample_frame_size = SDL_AUDIO_BITSIZE(src_format) * src_channels;
    retval->src_format = src_format;
    retval->src_channels = src_channels;
    retval->src_rate = src_rate;
    retval->dst_sample_frame_size = SDL_AUDIO_BITSIZE(dst_format) * dst_channels;
    retval->dst_format = dst_format;
    retval->dst_channels = dst_channels;
    retval->dst_rate = dst_rate;
    retval->pre_resample_channels = pre_resample_channels;
    retval->packetlen = packetlen;
    retval->rate_incr = ((double) dst_rate) / ((double) src_rate);

    /* Not resampling? It's an easy conversion (and maybe not even that!). */
    if (src_rate == dst_rate) {
        retval->cvt_before_resampling.needed = SDL_FALSE;
        retval->cvt_before_resampling.len_mult = 1;
        if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
            SDL_FreeAudioStream(retval);
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
        }
    } else {
        /* Don't resample at first. Just get us to Float32 format. */
        /* !!! FIXME: convert to int32 on devices without hardware float. */
        if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) < 0) {
            SDL_FreeAudioStream(retval);
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
        }

#ifdef HAVE_LIBSAMPLERATE
        SetupLibSampleRateResampling(retval);
#endif

        if (!retval->resampler_func) {
            retval->resampler_state = SDL_calloc(1, sizeof(SDL_AudioStreamResamplerState));
            if (!retval->resampler_state) {
                SDL_FreeAudioStream(retval);
                SDL_OutOfMemory();
                return NULL;
            }
            retval->resampler_func = SDL_ResampleAudioStream;
            retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
            retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
        }

        /* Convert us to the final format after resampling. */
        if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
            SDL_FreeAudioStream(retval);
            return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
        }
    }

    retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2);
    if (!retval->queue) {
        SDL_FreeAudioStream(retval);
        return NULL;  /* SDL_NewDataQueue should have called SDL_SetError. */
    }

    return retval;
}

static Uint8 *
EnsureBufferSize(Uint8 **buf, int *len, const int newlen)
{
    if (*len < newlen) {
        void *ptr = SDL_realloc(*buf, newlen);
        if (!ptr) {
            SDL_OutOfMemory();
            return NULL;
        }
        *buf = (Uint8 *) ptr;
        *len = newlen;
    }
    return *buf;
}

int
SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, const Uint32 _buflen)
{
    int buflen = (int) _buflen;

    if (!stream) {
        return SDL_InvalidParamError("stream");
    } else if (!buf) {
        return SDL_InvalidParamError("buf");
    } else if (buflen == 0) {
        return 0;  /* nothing to do. */
    } else if ((buflen % stream->src_sample_frame_size) != 0) {
        return SDL_SetError("Can't add partial sample frames");
    }

    if (stream->cvt_before_resampling.needed) {
        const int workbuflen = buflen * stream->cvt_before_resampling.len_mult;  /* will be "* 1" if not needed */
        Uint8 *workbuf = EnsureBufferSize(&stream->work_buffer, &stream->work_buffer_len, workbuflen);
        if (workbuf == NULL) {
            return -1;  /* probably out of memory. */
        }
        SDL_memcpy(workbuf, buf, buflen);
        stream->cvt_before_resampling.buf = workbuf;
        stream->cvt_before_resampling.len = buflen;
        if (SDL_ConvertAudio(&stream->cvt_before_resampling) == -1) {
            return -1;   /* uhoh! */
        }
        buf = workbuf;
        buflen = stream->cvt_before_resampling.len_cvt;
    }

    if (stream->dst_rate != stream->src_rate) {
        const int workbuflen = buflen * ((int) SDL_ceil(stream->rate_incr));
        float *workbuf = (float *) EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
        if (workbuf == NULL) {
            return -1;  /* probably out of memory. */
        }
        buflen = stream->resampler_func(stream, (float *) buf, buflen, workbuf, workbuflen);
        buf = workbuf;
    }

    if (stream->cvt_after_resampling.needed) {
        const int workbuflen = buflen * stream->cvt_before_resampling.len_mult;  /* will be "* 1" if not needed */
        Uint8 *workbuf;

        if (buf == stream->resample_buffer) {
            workbuf = EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
        } else {
            const int inplace = (buf == stream->work_buffer);
            workbuf = EnsureBufferSize(&stream->work_buffer, &stream->work_buffer_len, workbuflen);
            if (workbuf && !inplace) {
                SDL_memcpy(workbuf, buf, buflen);
            }
        }

        if (workbuf == NULL) {
            return -1;  /* probably out of memory. */
        }

        stream->cvt_after_resampling.buf = workbuf;
        stream->cvt_after_resampling.len = buflen;
        if (SDL_ConvertAudio(&stream->cvt_after_resampling) == -1) {
            return -1;   /* uhoh! */
        }
        buf = workbuf;
        buflen = stream->cvt_after_resampling.len_cvt;
    }

    return SDL_WriteToDataQueue(stream->queue, buf, buflen);
}

void
SDL_AudioStreamClear(SDL_AudioStream *stream)
{
    if (!stream) {
        SDL_InvalidParamError("stream");
    } else {
        SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
        stream->reset_resampler_func(stream);
    }
}


/* get converted/resampled data from the stream */
int
SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, const Uint32 len)
{
    if (!stream) {
        return SDL_InvalidParamError("stream");
    } else if (!buf) {
        return SDL_InvalidParamError("buf");
    } else if (len == 0) {
        return 0;  /* nothing to do. */
    } else if ((len % stream->dst_sample_frame_size) != 0) {
        return SDL_SetError("Can't request partial sample frames");
    }

    return (int) SDL_ReadFromDataQueue(stream->queue, buf, len);
}

/* number of converted/resampled bytes available */
int
SDL_AudioStreamAvailable(SDL_AudioStream *stream)
{
    return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
}

/* dispose of a stream */
void
SDL_FreeAudioStream(SDL_AudioStream *stream)
{
    if (stream) {
        if (stream->cleanup_resampler_func) {
            stream->cleanup_resampler_func(stream);
        }
        SDL_FreeDataQueue(stream->queue);
        SDL_free(stream->work_buffer);
        SDL_free(stream->resample_buffer);
        SDL_free(stream);
    }
}

/* vi: set ts=4 sw=4 expandtab: */