/*

 * QEMU DirectSound audio driver

 *

 * Copyright (c) 2005 Vassili Karpov (malc)

 *

 * Permission is hereby granted, free of charge, to any person obtaining a copy

 * of this software and associated documentation files (the "Software"), to deal

 * in the Software without restriction, including without limitation the rights

 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell

 * copies of the Software, and to permit persons to whom the Software is

 * furnished to do so, subject to the following conditions:

 *

 * The above copyright notice and this permission notice shall be included in

 * all copies or substantial portions of the Software.

 *

 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR

 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,

 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL

 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER

 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,

 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN

 * THE SOFTWARE.

 */

/*

 * SEAL 1.07 by Carlos 'pel' Hasan was used as documentation

 */

#include "qemu-common.h"

#include "audio.h"

#define AUDIO_CAP "dsound"

#include "audio_int.h"

#define WIN32_LEAN_AND_MEAN

#include <windows.h>

#include <mmsystem.h>

#include <objbase.h>

#include <dsound.h>

/* #define DEBUG_DSOUND */

static struct {

    int lock_retries;

    int restore_retries;

    int getstatus_retries;

    int set_primary;

    int bufsize_in;

    int bufsize_out;

    struct audsettings settings;

    int latency_millis;

} conf = {

    1,

    1,

    1,

    0,

    16384,

    16384,

    {

        44100,

        2,

        AUD_FMT_S16

    },

    10

};

typedef struct {

    LPDIRECTSOUND dsound;

    LPDIRECTSOUNDCAPTURE dsound_capture;

    LPDIRECTSOUNDBUFFER dsound_primary_buffer;

    struct audsettings settings;

} dsound;

static dsound glob_dsound;

typedef struct {

    HWVoiceOut hw;

    LPDIRECTSOUNDBUFFER dsound_buffer;

    DWORD old_pos;

    int first_time;

#ifdef DEBUG_DSOUND

    DWORD old_ppos;

    DWORD played;

    DWORD mixed;

#endif

} DSoundVoiceOut;

typedef struct {

    HWVoiceIn hw;

    int first_time;

    LPDIRECTSOUNDCAPTUREBUFFER dsound_capture_buffer;

} DSoundVoiceIn;

static void dsound_log_hresult (HRESULT hr)

{

    const char *str = "BUG";

    switch (hr) {

    case DS_OK:

        str = "The method succeeded";

        break;

#ifdef DS_NO_VIRTUALIZATION

    case DS_NO_VIRTUALIZATION:

        str = "The buffer was created, but another 3D algorithm was substituted";

        break;

#endif

#ifdef DS_INCOMPLETE

    case DS_INCOMPLETE:

        str = "The method succeeded, but not all the optional effects were obtained";

        break;

#endif

#ifdef DSERR_ACCESSDENIED

    case DSERR_ACCESSDENIED:

        str = "The request failed because access was denied";

        break;

#endif

#ifdef DSERR_ALLOCATED

    case DSERR_ALLOCATED:

        str = "The request failed because resources, such as a priority level, were already in use by another caller";

        break;

#endif

#ifdef DSERR_ALREADYINITIALIZED

    case DSERR_ALREADYINITIALIZED:

        str = "The object is already initialized";

        break;

#endif

#ifdef DSERR_BADFORMAT

    case DSERR_BADFORMAT:

        str = "The specified wave format is not supported";

        break;

#endif

#ifdef DSERR_BADSENDBUFFERGUID

    case DSERR_BADSENDBUFFERGUID:

        str = "The GUID specified in an audiopath file does not match a valid mix-in buffer";

        break;

#endif

#ifdef DSERR_BUFFERLOST

    case DSERR_BUFFERLOST:

        str = "The buffer memory has been lost and must be restored";

        break;

#endif

#ifdef DSERR_BUFFERTOOSMALL

    case DSERR_BUFFERTOOSMALL:

        str = "The buffer size is not great enough to enable effects processing";

        break;

#endif

#ifdef DSERR_CONTROLUNAVAIL

    case DSERR_CONTROLUNAVAIL:

        str = "The buffer control (volume, pan, and so on) requested by the caller is not available. Controls must be specified when the buffer is created, using the dwFlags member of DSBUFFERDESC";

        break;

#endif

#ifdef DSERR_DS8_REQUIRED

    case DSERR_DS8_REQUIRED:

        str = "A DirectSound object of class CLSID_DirectSound8 or later is required for the requested functionality. For more information, see IDirectSound8 Interface";

        break;

#endif

#ifdef DSERR_FXUNAVAILABLE

    case DSERR_FXUNAVAILABLE:

        str = "The effects requested could not be found on the system, or they are in the wrong order or in the wrong location; for example, an effect expected in hardware was found in software";

        break;

#endif

#ifdef DSERR_GENERIC

    case DSERR_GENERIC :

        str = "An undetermined error occurred inside the DirectSound subsystem";

        break;

#endif

#ifdef DSERR_INVALIDCALL

    case DSERR_INVALIDCALL:

        str = "This function is not valid for the current state of this object";

        break;

#endif

#ifdef DSERR_INVALIDPARAM

    case DSERR_INVALIDPARAM:

        str = "An invalid parameter was passed to the returning function";

        break;

#endif

#ifdef DSERR_NOAGGREGATION

    case DSERR_NOAGGREGATION:

        str = "The object does not support aggregation";

        break;

#endif

#ifdef DSERR_NODRIVER

    case DSERR_NODRIVER:

        str = "No sound driver is available for use, or the given GUID is not a valid DirectSound device ID";

        break;

#endif

#ifdef DSERR_NOINTERFACE

    case DSERR_NOINTERFACE:

        str = "The requested COM interface is not available";

        break;

#endif

#ifdef DSERR_OBJECTNOTFOUND

    case DSERR_OBJECTNOTFOUND:

        str = "The requested object was not found";

        break;

#endif

#ifdef DSERR_OTHERAPPHASPRIO

    case DSERR_OTHERAPPHASPRIO:

        str = "Another application has a higher priority level, preventing this call from succeeding";

        break;

#endif

#ifdef DSERR_OUTOFMEMORY

    case DSERR_OUTOFMEMORY:

        str = "The DirectSound subsystem could not allocate sufficient memory to complete the caller's request";

        break;

#endif

#ifdef DSERR_PRIOLEVELNEEDED

    case DSERR_PRIOLEVELNEEDED:

        str = "A cooperative level of DSSCL_PRIORITY or higher is required";

        break;

#endif

#ifdef DSERR_SENDLOOP

    case DSERR_SENDLOOP:

        str = "A circular loop of send effects was detected";

        break;

#endif

#ifdef DSERR_UNINITIALIZED

    case DSERR_UNINITIALIZED:

        str = "The Initialize method has not been called or has not been called successfully before other methods were called";

        break;

#endif

#ifdef DSERR_UNSUPPORTED

    case DSERR_UNSUPPORTED:

        str = "The function called is not supported at this time";

        break;

#endif

    default:

        AUD_log (AUDIO_CAP, "Reason: Unknown (HRESULT %#lx)\n", hr);

        return;

    }

    AUD_log (AUDIO_CAP, "Reason: %s\n", str);

}

static void GCC_FMT_ATTR (2, 3) dsound_logerr (

    HRESULT hr,

    const char *fmt,

    ...

    )

{

    va_list ap;

    va_start (ap, fmt);

    AUD_vlog (AUDIO_CAP, fmt, ap);

    va_end (ap);

    dsound_log_hresult (hr);

}

static void GCC_FMT_ATTR (3, 4) dsound_logerr2 (

    HRESULT hr,

    const char *typ,

    const char *fmt,

    ...

    )

{

    va_list ap;

    AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);

    va_start (ap, fmt);

    AUD_vlog (AUDIO_CAP, fmt, ap);

    va_end (ap);

    dsound_log_hresult (hr);

}

static DWORD millis_to_bytes (struct audio_pcm_info *info, DWORD millis)

{

    return (millis * info->bytes_per_second) / 1000;

}

#ifdef DEBUG_DSOUND

static void print_wave_format (WAVEFORMATEX *wfx)

{

    dolog ("tag             = %d\n", wfx->wFormatTag);

    dolog ("nChannels       = %d\n", wfx->nChannels);

    dolog ("nSamplesPerSec  = %ld\n", wfx->nSamplesPerSec);

    dolog ("nAvgBytesPerSec = %ld\n", wfx->nAvgBytesPerSec);

    dolog ("nBlockAlign     = %d\n", wfx->nBlockAlign);

    dolog ("wBitsPerSample  = %d\n", wfx->wBitsPerSample);

    dolog ("cbSize          = %d\n", wfx->cbSize);

}

#endif

static int dsound_restore_out (LPDIRECTSOUNDBUFFER dsb)

{

    HRESULT hr;

    int i;

    for (i = 0; i < conf.restore_retries; ++i) {

        hr = IDirectSoundBuffer_Restore (dsb);

        switch (hr) {

        case DS_OK:

            return 0;

        case DSERR_BUFFERLOST:

            continue;

        default:

            dsound_logerr (hr, "Could not restore playback buffer\n");

            return -1;

        }

    }

    dolog ("%d attempts to restore playback buffer failed\n", i);

    return -1;

}

static int waveformat_from_audio_settings (WAVEFORMATEX *wfx,

                                           struct audsettings *as)

{

    memset (wfx, 0, sizeof (*wfx));

    wfx->wFormatTag = WAVE_FORMAT_PCM;

    wfx->nChannels = as->nchannels;

    wfx->nSamplesPerSec = as->freq;

    wfx->nAvgBytesPerSec = as->freq << (as->nchannels == 2);

    wfx->nBlockAlign = 1 << (as->nchannels == 2);

    wfx->cbSize = 0;

    switch (as->fmt) {

    case AUD_FMT_S8:

    case AUD_FMT_U8:

        wfx->wBitsPerSample = 8;

        break;

    case AUD_FMT_S16:

    case AUD_FMT_U16:

        wfx->wBitsPerSample = 16;

        wfx->nAvgBytesPerSec <<= 1;

        wfx->nBlockAlign <<= 1;

        break;

    case AUD_FMT_S32:

    case AUD_FMT_U32:

        wfx->wBitsPerSample = 32;

        wfx->nAvgBytesPerSec <<= 2;

        wfx->nBlockAlign <<= 2;

        break;

    default:

        dolog ("Internal logic error: Bad audio format %d\n", as->freq);

        return -1;

    }

    return 0;

}

static int waveformat_to_audio_settings (WAVEFORMATEX *wfx,

                                         struct audsettings *as)

{

    if (wfx->wFormatTag != WAVE_FORMAT_PCM) {

        dolog ("Invalid wave format, tag is not PCM, but %d\n",

               wfx->wFormatTag);

        return -1;

    }

    if (!wfx->nSamplesPerSec) {

        dolog ("Invalid wave format, frequency is zero\n");

        return -1;

    }

    as->freq = wfx->nSamplesPerSec;

    switch (wfx->nChannels) {

    case 1:

        as->nchannels = 1;

        break;

    case 2:

        as->nchannels = 2;

        break;

    default:

        dolog (

            "Invalid wave format, number of channels is not 1 or 2, but %d\n",

            wfx->nChannels

            );

        return -1;

    }

    switch (wfx->wBitsPerSample) {

    case 8:

        as->fmt = AUD_FMT_U8;

        break;

    case 16:

        as->fmt = AUD_FMT_S16;

        break;

    case 32:

        as->fmt = AUD_FMT_S32;

        break;

    default:

        dolog ("Invalid wave format, bits per sample is not "

               "8, 16 or 32, but %d\n",

               wfx->wBitsPerSample);

        return -1;

    }

    return 0;

}

#include "dsound_template.h"

#define DSBTYPE_IN

#include "dsound_template.h"

#undef DSBTYPE_IN

static int dsound_get_status_out (LPDIRECTSOUNDBUFFER dsb, DWORD *statusp)

{

    HRESULT hr;

    int i;

    for (i = 0; i < conf.getstatus_retries; ++i) {

        hr = IDirectSoundBuffer_GetStatus (dsb, statusp);

        if (FAILED (hr)) {

            dsound_logerr (hr, "Could not get playback buffer status\n");

            return -1;

        }

        if (*statusp & DSERR_BUFFERLOST) {

            if (dsound_restore_out (dsb)) {

                return -1;

            }

            continue;

        }

        break;

    }

    return 0;

}

static int dsound_get_status_in (LPDIRECTSOUNDCAPTUREBUFFER dscb,

                                 DWORD *statusp)

{

    HRESULT hr;

    hr = IDirectSoundCaptureBuffer_GetStatus (dscb, statusp);

    if (FAILED (hr)) {

        dsound_logerr (hr, "Could not get capture buffer status\n");

        return -1;

    }

    return 0;

}

static void dsound_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)

{

    int src_len1 = dst_len;

    int src_len2 = 0;

    int pos = hw->rpos + dst_len;

    struct st_sample *src1 = hw->mix_buf + hw->rpos;

    struct st_sample *src2 = NULL;

    if (pos > hw->samples) {

        src_len1 = hw->samples - hw->rpos;

        src2 = hw->mix_buf;

        src_len2 = dst_len - src_len1;

        pos = src_len2;

    }

    if (src_len1) {

        hw->clip (dst, src1, src_len1);

    }

    if (src_len2) {

        dst = advance (dst, src_len1 << hw->info.shift);

        hw->clip (dst, src2, src_len2);

    }

    hw->rpos = pos % hw->samples;

}

static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb)

{

    int err;

    LPVOID p1, p2;

    DWORD blen1, blen2, len1, len2;

    err = dsound_lock_out (

        dsb,

        &hw->info,

        0,

        hw->samples << hw->info.shift,

        &p1, &p2,

        &blen1, &blen2,

        1

        );

    if (err) {

        return;

    }

    len1 = blen1 >> hw->info.shift;

    len2 = blen2 >> hw->info.shift;

#ifdef DEBUG_DSOUND

    dolog ("clear %p,%ld,%ld %p,%ld,%ld\n",

           p1, blen1, len1,

           p2, blen2, len2);

#endif

    if (p1 && len1) {

        audio_pcm_info_clear_buf (&hw->info, p1, len1);

    }

    if (p2 && len2) {

        audio_pcm_info_clear_buf (&hw->info, p2, len2);

    }

    dsound_unlock_out (dsb, p1, p2, blen1, blen2);

}

static void dsound_close (dsound *s)

{

    HRESULT hr;

    if (s->dsound_primary_buffer) {

        hr = IDirectSoundBuffer_Release (s->dsound_primary_buffer);

        if (FAILED (hr)) {

            dsound_logerr (hr, "Could not release primary buffer\n");

        }

        s->dsound_primary_buffer = NULL;

    }

}

static int dsound_open (dsound *s)

{