dsoundaudio.c

xen虚拟机源代码安装包

/*
 * 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 "vl.h"

#define AUDIO_CAP "dsound"
#include "audio_int.h"

#include <windows.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;
    audsettings_t 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;
    audsettings_t 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, audsettings_t *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:
        wfx->wBitsPerSample = 8;
        break;

    case AUD_FMT_U8:
        wfx->wBitsPerSample = 8;
        break;

    case AUD_FMT_S16:
        wfx->wBitsPerSample = 16;
        wfx->nAvgBytesPerSec <<= 1;
        wfx->nBlockAlign <<= 1;
        break;

    case AUD_FMT_U16:
        wfx->wBitsPerSample = 16;
        wfx->nAvgBytesPerSec <<= 1;
        wfx->nBlockAlign <<= 1;
        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, audsettings_t *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;

    default:
        dolog ("Invalid wave format, bits per sample is not 8 or 16, 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;
    st_sample_t *src1 = hw->mix_buf + hw->rpos;
    st_sample_t *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;
    }