qemu/audio/dsoundaudio.c

/*
 * 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"

#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 = {
    .lock_retries       = 1,
    .restore_retries    = 1,
    .getstatus_retries  = 1,
    .set_primary        = 0,
    .bufsize_in         = 16384,
    .bufsize_out        = 16384,
    .settings.freq      = 44100,
    .settings.nchannels = 2,
    .settings.fmt       = AUD_FMT_S16,
    .latency_millis     = 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)
{
    int err;
    HRESULT hr;
    WAVEFORMATEX wfx;
    DSBUFFERDESC dsbd;
    HWND hwnd;

    hwnd = GetForegroundWindow ();
    hr = IDirectSound_SetCooperativeLevel (
        s->dsound,
        hwnd,
        DSSCL_PRIORITY
        );

    if (FAILED (hr)) {
        dsound_logerr (hr, "Could not set cooperative level for window %p\n",
                       hwnd);
        return -1;
    }

    if (!conf.set_primary) {
        return 0;
    }

    err = waveformat_from_audio_settings (&wfx, &conf.settings);
    if (err) {
        return -1;
    }

    memset (&dsbd, 0, sizeof (dsbd));
    dsbd.dwSize = sizeof (dsbd);
    dsbd.dwFlags = DSBCAPS_PRIMARYBUFFER;
    dsbd.dwBufferBytes = 0;
    dsbd.lpwfxFormat = NULL;

    hr = IDirectSound_CreateSoundBuffer (
        s->dsound,
        &dsbd,
        &s->dsound_primary_buffer,
        NULL
        );
    if (FAILED (hr)) {
        dsound_logerr (hr, "Could not create primary playback buffer\n");
        return -1;
    }

    hr = IDirectSoundBuffer_SetFormat (s->dsound_primary_buffer, &wfx);
    if (FAILED (hr)) {
        dsound_logerr (hr, "Could not set primary playback buffer format\n");
    }

    hr = IDirectSoundBuffer_GetFormat (
        s->dsound_primary_buffer,
        &wfx,
        sizeof (wfx),
        NULL
        );
    if (FAILED (hr)) {
        dsound_logerr (hr, "Could not get primary playback buffer format\n");
        goto fail0;
    }

#ifdef DEBUG_DSOUND
    dolog ("Primary\n");
    print_wave_format (&wfx);
#endif

    err = waveformat_to_audio_settings (&wfx, &s->settings);
    if (err) {
        goto fail0;
    }

    return 0;

 fail0:
    dsound_close (s);
    return -1;
}

static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
    HRESULT hr;
    DWORD status;
    DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
    LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;

    if (!dsb) {
        dolog ("Attempt to control voice without a buffer\n");
        return 0;
    }

    switch (cmd) {
    case VOICE_ENABLE:
        if (dsound_get_status_out (dsb, &status)) {
            return -1;
        }

        if (status & DSBSTATUS_PLAYING) {
            dolog ("warning: Voice is already playing\n");
            return 0;
        }

        dsound_clear_sample (hw, dsb);

        hr = IDirectSoundBuffer_Play (dsb, 0, 0, DSBPLAY_LOOPING);
        if (FAILED (hr)) {
            dsound_logerr (hr, "Could not start playing buffer\n");
            return -1;
        }
        break;

    case VOICE_DISABLE:
        if (dsound_get_status_out (dsb, &status)) {
            return -1;
        }

        if (status & DSBSTATUS_PLAYING) {
            hr = IDirectSoundBuffer_Stop (dsb);
            if (FAILED (hr)) {
                dsound_logerr (hr, "Could not stop playing buffer\n");
                return -1;
            }
        }
        else {
            dolog ("warning: Voice is not playing\n");
        }
        break;
    }
    return 0;
}

static int dsound_write (SWVoiceOut *sw, void *buf, int len)
{
    return audio_pcm_sw_write (sw, buf, len);
}

static int dsound_run_out (HWVoiceOut *hw)
{
    int err;
    HRESULT hr;
    DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
    LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
    int live, len, hwshift;
    DWORD blen1, blen2;
    DWORD len1, len2;
    DWORD decr;
    DWORD wpos, ppos, old_pos;
    LPVOID p1, p2;
    int bufsize;

    if (!dsb) {
        dolog ("Attempt to run empty with playback buffer\n");
        return 0;
    }

    hwshift = hw->info.shift;
    bufsize = hw->samples << hwshift;

    live = audio_pcm_hw_get_live_out (hw);

    hr = IDirectSoundBuffer_GetCurrentPosition (
        dsb,
        &ppos,
        ds->first_time ? &wpos : NULL
        );
    if (FAILED (hr)) {
        dsound_logerr (hr, "Could not get playback buffer position\n");
        return 0;
    }

    len = live << hwshift;

    if (ds->first_time) {
        if (conf.latency_millis) {
            DWORD cur_blat;

            cur_blat = audio_ring_dist (wpos, ppos, bufsize);
            ds->first_time = 0;
            old_pos = wpos;
            old_pos +=
                millis_to_bytes (&hw->info, conf.latency_millis) - cur_blat;
            old_pos %= bufsize;
            old_pos &= ~hw->info.align;
        }
        else {
            old_pos = wpos;
        }
#ifdef DEBUG_DSOUND
        ds->played = 0;
        ds->mixed = 0;
#endif
    }
    else {
        if (ds->old_pos == ppos) {
#ifdef DEBUG_DSOUND
            dolog ("old_pos == ppos\n");
#endif
            return 0;
        }

#ifdef DEBUG_DSOUND
        ds->played += audio_ring_dist (ds->old_pos, ppos, hw->bufsize);
#endif
        old_pos = ds->old_pos;
    }

    if ((old_pos < ppos) && ((old_pos + len) > ppos)) {
        len = ppos - old_pos;
    }
    else {
        if ((old_pos > ppos) && ((old_pos + len) > (ppos + bufsize))) {
            len = bufsize - old_pos + ppos;
        }
    }

    if (audio_bug (AUDIO_FUNC, len < 0 || len > bufsize)) {
        dolog ("len=%d bufsize=%d old_pos=%ld ppos=%ld\n",
               len, bufsize, old_pos, ppos);
        return 0;
    }

    len &= ~hw->info.align;
    if (!len) {
        return 0;
    }

#ifdef DEBUG_DSOUND
    ds->old_ppos = ppos;
#endif
    err = dsound_lock_out (
        dsb,
        &hw->info,
        old_pos,
        len,
        &p1, &p2,
        &blen1, &blen2,
        0
        );
    if (err) {
        return 0;
    }

    len1 = blen1 >> hwshift;
    len2 = blen2 >> hwshift;
    decr = len1 + len2;

    if (p1 && len1) {
        dsound_write_sample (hw, p1, len1);
    }

    if (p2 && len2) {
        dsound_write_sample (hw, p2, len2);
    }

    dsound_unlock_out (dsb, p1, p2, blen1, blen2);
    ds->old_pos = (old_pos + (decr << hwshift)) % bufsize;

#ifdef DEBUG_DSOUND
    ds->mixed += decr << hwshift;

    dolog ("played %lu mixed %lu diff %ld sec %f\n",
           ds->played,
           ds->mixed,
           ds->mixed - ds->played,
           abs (ds->mixed - ds->played) / (double) hw->info.bytes_per_second);
#endif
    return decr;
}

static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
    HRESULT hr;
    DWORD status;
    DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
    LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;

    if (!dscb) {
        dolog ("Attempt to control capture voice without a buffer\n");
        return -1;
    }

    switch (cmd) {
    case VOICE_ENABLE:
        if (dsound_get_status_in (dscb, &status)) {
            return -1;
        }

        if (status & DSCBSTATUS_CAPTURING) {
            dolog ("warning: Voice is already capturing\n");
            return 0;
        }

        /* clear ?? */

        hr = IDirectSoundCaptureBuffer_Start (dscb, DSCBSTART_LOOPING);
        if (FAILED (hr)) {
            dsound_logerr (hr, "Could not start capturing\n");
            return -1;
        }
        break;

    case VOICE_DISABLE:
        if (dsound_get_status_in (dscb, &status)) {
            return -1;
        }

        if (status & DSCBSTATUS_CAPTURING) {
            hr = IDirectSoundCaptureBuffer_Stop (dscb);
            if (FAILED (hr)) {
                dsound_logerr (hr, "Could not stop capturing\n");
                return -1;
            }
        }
        else {
            dolog ("warning: Voice is not capturing\n");
        }
        break;
    }
    return 0;
}

static int dsound_read (SWVoiceIn *sw, void *buf, int len)
{
    return audio_pcm_sw_read (sw, buf, len);
}

static int dsound_run_in (HWVoiceIn *hw)
{
    int err;
    HRESULT hr;
    DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
    LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
    int live, len, dead;
    DWORD blen1, blen2;
    DWORD len1, len2;
    DWORD decr;
    DWORD cpos, rpos;
    LPVOID p1, p2;
    int hwshift;

    if (!dscb) {
        dolog ("Attempt to run without capture buffer\n");
        return 0;
    }

    hwshift = hw->info.shift;

    live = audio_pcm_hw_get_live_in (hw);
    dead = hw->samples - live;
    if (!dead) {
        return 0;
    }

    hr = IDirectSoundCaptureBuffer_GetCurrentPosition (
        dscb,
        &cpos,
        ds->first_time ? &rpos : NULL
        );
    if (FAILED (hr)) {
        dsound_logerr (hr, "Could not get capture buffer position\n");
        return 0;
    }

    if (ds->first_time) {
        ds->first_time = 0;
        if (rpos & hw->info.align) {
            ldebug ("warning: Misaligned capture read position %ld(%d)\n",
                    rpos, hw->info.align);
        }
        hw->wpos = rpos >> hwshift;
    }

    if (cpos & hw->info.align) {
        ldebug ("warning: Misaligned capture position %ld(%d)\n",
                cpos, hw->info.align);
    }
    cpos >>= hwshift;

    len = audio_ring_dist (cpos, hw->wpos, hw->samples);
    if (!len) {
        return 0;
    }
    len = audio_MIN (len, dead);

    err = dsound_lock_in (
        dscb,
        &hw->info,
        hw->wpos << hwshift,
        len << hwshift,
        &p1,
        &p2,
        &blen1,
        &blen2,
        0
        );
    if (err) {
        return 0;
    }

    len1 = blen1 >> hwshift;
    len2 = blen2 >> hwshift;
    decr = len1 + len2;

    if (p1 && len1) {
        hw->conv (hw->conv_buf + hw->wpos, p1, len1, &nominal_volume);
    }

    if (p2 && len2) {
        hw->conv (hw->conv_buf, p2, len2, &nominal_volume);
    }

    dsound_unlock_in (dscb, p1, p2, blen1, blen2);
    hw->wpos = (hw->wpos + decr) % hw->samples;
    return decr;
}

static void dsound_audio_fini (void *opaque)
{
    HRESULT hr;
    dsound *s = opaque;

    if (!s->dsound) {
        return;
    }

    hr = IDirectSound_Release (s->dsound);
    if (FAILED (hr)) {
        dsound_logerr (hr, "Could not release DirectSound\n");
    }
    s->dsound = NULL;

    if (!s->dsound_capture) {
        return;
    }

    hr = IDirectSoundCapture_Release (s->dsound_capture);
    if (FAILED (hr)) {
        dsound_logerr (hr, "Could not release DirectSoundCapture\n");
    }
    s->dsound_capture = NULL;
}

static void *dsound_audio_init (void)
{
    int err;
    HRESULT hr;
    dsound *s = &glob_dsound;

    hr = CoInitialize (NULL);
    if (FAILED (hr)) {
        dsound_logerr (hr, "Could not initialize COM\n");
        return NULL;
    }

    hr = CoCreateInstance (
        &CLSID_DirectSound,
        NULL,
        CLSCTX_ALL,
        &IID_IDirectSound,
        (void **) &s->dsound
        );
    if (FAILED (hr)) {
        dsound_logerr (hr, "Could not create DirectSound instance\n");
        return NULL;
    }

    hr = IDirectSound_Initialize (s->dsound, NULL);
    if (FAILED (hr)) {
        dsound_logerr (hr, "Could not initialize DirectSound\n");

        hr = IDirectSound_Release (s->dsound);
        if (FAILED (hr)) {
            dsound_logerr (hr, "Could not release DirectSound\n");
        }
        s->dsound = NULL;
        return NULL;
    }

    hr = CoCreateInstance (
        &CLSID_DirectSoundCapture,
        NULL,
        CLSCTX_ALL,
        &IID_IDirectSoundCapture,
        (void **) &s->dsound_capture
        );
    if (FAILED (hr)) {
        dsound_logerr (hr, "Could not create DirectSoundCapture instance\n");
    }
    else {
        hr = IDirectSoundCapture_Initialize (s->dsound_capture, NULL);
        if (FAILED (hr)) {
            dsound_logerr (hr, "Could not initialize DirectSoundCapture\n");

            hr = IDirectSoundCapture_Release (s->dsound_capture);
            if (FAILED (hr)) {
                dsound_logerr (hr, "Could not release DirectSoundCapture\n");
            }
            s->dsound_capture = NULL;
        }
    }

    err = dsound_open (s);
    if (err) {
        dsound_audio_fini (s);
        return NULL;
    }

    return s;
}

static struct audio_option dsound_options[] = {
    {
        .name  = "LOCK_RETRIES",
        .tag   = AUD_OPT_INT,
        .valp  = &conf.lock_retries,
        .descr = "Number of times to attempt locking the buffer"
    },
    {
        .name  = "RESTOURE_RETRIES",
        .tag   = AUD_OPT_INT,
        .valp  = &conf.restore_retries,
        .descr = "Number of times to attempt restoring the buffer"
    },
    {
        .name  = "GETSTATUS_RETRIES",
        .tag   = AUD_OPT_INT,
        .valp  = &conf.getstatus_retries,
        .descr = "Number of times to attempt getting status of the buffer"
    },
    {
        .name  = "SET_PRIMARY",
        .tag   = AUD_OPT_BOOL,
        .valp  = &conf.set_primary,
        .descr = "Set the parameters of primary buffer"
    },
    {
        .name  = "LATENCY_MILLIS",
        .tag   = AUD_OPT_INT,
        .valp  = &conf.latency_millis,
        .descr = "(undocumented)"
    },
    {
        .name  = "PRIMARY_FREQ",
        .tag   = AUD_OPT_INT,
        .valp  = &conf.settings.freq,
        .descr = "Primary buffer frequency"
    },
    {
        .name  = "PRIMARY_CHANNELS",
        .tag   = AUD_OPT_INT,
        .valp  = &conf.settings.nchannels,
        .descr = "Primary buffer number of channels (1 - mono, 2 - stereo)"
    },
    {
        .name  = "PRIMARY_FMT",
        .tag   = AUD_OPT_FMT,
        .valp  = &conf.settings.fmt,
        .descr = "Primary buffer format"
    },
    {
        .name  = "BUFSIZE_OUT",
        .tag   = AUD_OPT_INT,
        .valp  = &conf.bufsize_out,
        .descr = "(undocumented)"
    },
    {
        .name  = "BUFSIZE_IN",
        .tag   = AUD_OPT_INT,
        .valp  = &conf.bufsize_in,
        .descr = "(undocumented)"
    },
    { /* End of list */ }
};

static struct audio_pcm_ops dsound_pcm_ops = {
    .init_out = dsound_init_out,
    .fini_out = dsound_fini_out,
    .run_out  = dsound_run_out,
    .write    = dsound_write,
    .ctl_out  = dsound_ctl_out,

    .init_in  = dsound_init_in,
    .fini_in  = dsound_fini_in,
    .run_in   = dsound_run_in,
    .read     = dsound_read,
    .ctl_in   = dsound_ctl_in
};

struct audio_driver dsound_audio_driver = {
    .name           = "dsound",
    .descr          = "DirectSound http://wikipedia.org/wiki/DirectSound",
    .options        = dsound_options,
    .init           = dsound_audio_init,
    .fini           = dsound_audio_fini,
    .pcm_ops        = &dsound_pcm_ops,
    .can_be_default = 1,
    .max_voices_out = INT_MAX,
    .max_voices_in  = 1,
    .voice_size_out = sizeof (DSoundVoiceOut),
    .voice_size_in  = sizeof (DSoundVoiceIn)
};