Exemple #1
0
static void CALLBACK DSOUND_capture_timer(UINT timerID, UINT msg, DWORD_PTR user,
                                          DWORD_PTR dw1, DWORD_PTR dw2)
{
    DirectSoundCaptureDevice *device = (DirectSoundCaptureDevice*)user;
    UINT32 packet_frames, packet_bytes, avail_bytes, skip_bytes = 0;
    DWORD flags;
    BYTE *buf;
    HRESULT hr;

    if(!device->ref)
        return;

    EnterCriticalSection(&device->lock);

    if(!device->capture_buffer || device->state == STATE_STOPPED){
        LeaveCriticalSection(&device->lock);
        return;
    }

    if(device->state == STATE_STOPPING){
        device->state = STATE_STOPPED;
        LeaveCriticalSection(&device->lock);
        return;
    }

    if(device->state == STATE_STARTING)
        device->state = STATE_CAPTURING;

    hr = IAudioCaptureClient_GetBuffer(device->capture, &buf, &packet_frames,
            &flags, NULL, NULL);
    if(FAILED(hr)){
        LeaveCriticalSection(&device->lock);
        WARN("GetBuffer failed: %08x\n", hr);
        return;
    }

    packet_bytes = packet_frames * device->pwfx->nBlockAlign;
    if(packet_bytes > device->buflen){
        TRACE("audio glitch: dsound buffer too small for data\n");
        skip_bytes = packet_bytes - device->buflen;
        packet_bytes = device->buflen;
    }

    avail_bytes = device->buflen - device->write_pos_bytes;
    if(avail_bytes > packet_bytes)
        avail_bytes = packet_bytes;

    memcpy(device->buffer + device->write_pos_bytes, buf + skip_bytes, avail_bytes);
    capture_CheckNotify(device->capture_buffer, device->write_pos_bytes, avail_bytes);

    packet_bytes -= avail_bytes;
    if(packet_bytes > 0){
        if(device->capture_buffer->flags & DSCBSTART_LOOPING){
            memcpy(device->buffer, buf + skip_bytes + avail_bytes, packet_bytes);
            capture_CheckNotify(device->capture_buffer, 0, packet_bytes);
        }else{
            device->state = STATE_STOPPED;
            capture_CheckNotify(device->capture_buffer, 0, 0);
        }
    }

    device->write_pos_bytes += avail_bytes + packet_bytes;
    device->write_pos_bytes %= device->buflen;

    hr = IAudioCaptureClient_ReleaseBuffer(device->capture, packet_frames);
    if(FAILED(hr)){
        LeaveCriticalSection(&device->lock);
        WARN("ReleaseBuffer failed: %08x\n", hr);
        return;
    }

    LeaveCriticalSection(&device->lock);
}
Exemple #2
0
static void test_capture(IAudioClient *ac, HANDLE handle, WAVEFORMATEX *wfx)
{
    IAudioCaptureClient *acc;
    HRESULT hr;
    UINT32 frames = 0;
    BYTE *data = NULL;
    DWORD flags;
    UINT64 devpos, qpcpos;

    hr = IAudioClient_GetService(ac, &IID_IAudioCaptureClient, (void**)&acc);
    ok(hr == S_OK, "IAudioClient_GetService(IID_IAudioCaptureClient) returns %08x\n", hr);
    if (hr != S_OK)
        return;

    hr = IAudioCaptureClient_GetNextPacketSize(acc, NULL);
    ok(hr == E_POINTER, "IAudioCaptureClient_GetNextPacketSize(NULL) returns %08x\n", hr);

    ok(WaitForSingleObject(handle, 2000) == WAIT_OBJECT_0, "Waiting on event handle failed!\n");

    /* frames can be 0 if no data is available yet.. */
    hr = IAudioCaptureClient_GetNextPacketSize(acc, &frames);
    ok(hr == S_OK, "IAudioCaptureClient_GetNextPacketSize returns %08x\n", hr);

    data = (BYTE*)(DWORD_PTR)0xdeadbeef;
    hr = IAudioCaptureClient_GetBuffer(acc, &data, NULL, NULL, NULL, NULL);
    ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(data, NULL, NULL) returns %08x\n", hr);
    ok((DWORD_PTR)data == 0xdeadbeef, "data is reset to %p\n", data);

    frames = 0xdeadbeef;
    hr = IAudioCaptureClient_GetBuffer(acc, NULL, &frames, NULL, NULL, NULL);
    ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(NULL, &frames, NULL) returns %08x\n", hr);
    ok(frames == 0xdeadbeef, "frames is reset to %08x\n", frames);

    flags = 0xdeadbeef;
    hr = IAudioCaptureClient_GetBuffer(acc, NULL, NULL, &flags, NULL, NULL);
    ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(NULL, NULL, &flags) returns %08x\n", hr);
    ok(flags == 0xdeadbeef, "flags is reset to %08x\n", flags);

    hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, NULL, NULL, NULL);
    ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(&ata, &frames, NULL) returns %08x\n", hr);

    hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &devpos, &qpcpos);
    ok(hr == S_OK || hr == AUDCLNT_S_BUFFER_EMPTY, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr);
    if (hr == S_OK)
        ok(frames, "Amount of frames locked is 0!\n");
    else if (hr == AUDCLNT_S_BUFFER_EMPTY)
        ok(!frames, "Amount of frames locked with empty buffer is %u!\n", frames);
    else
        ok(0, "GetBuffer returned %08x\n", hr);
    trace("Device position is at %u, amount of frames locked: %u\n", (DWORD)devpos, frames);

    if (frames) {
        hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &devpos, &qpcpos);
        ok(hr == AUDCLNT_E_OUT_OF_ORDER, "Out of order IAudioCaptureClient_GetBuffer returns %08x\n", hr);
    }

    hr = IAudioCaptureClient_ReleaseBuffer(acc, frames);
    ok(hr == S_OK, "Releasing buffer returns %08x\n", hr);

    if (frames) {
        hr = IAudioCaptureClient_ReleaseBuffer(acc, frames);
        ok(hr == AUDCLNT_E_OUT_OF_ORDER, "Releasing buffer twice returns %08x\n", hr);
    }

    IUnknown_Release(acc);
}
static GstFlowReturn
gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
{
  GstWasapiSrc *self = GST_WASAPI_SRC (src);
  GstFlowReturn ret = GST_FLOW_OK;
  GstClock *clock;
  GstClockTime timestamp, duration = self->period_time;
  HRESULT hr;
  gint16 *samples = NULL;
  guint32 nsamples_read = 0, nsamples;
  DWORD flags = 0;
  guint64 devpos;

  GST_OBJECT_LOCK (self);
  clock = GST_ELEMENT_CLOCK (self);
  if (clock != NULL)
    gst_object_ref (clock);
  GST_OBJECT_UNLOCK (self);

  if (clock != NULL && GST_CLOCK_TIME_IS_VALID (self->next_time)) {
    GstClockID id;

    id = gst_clock_new_single_shot_id (clock, self->next_time);
    gst_clock_id_wait (id, NULL);
    gst_clock_id_unref (id);
  }

  do {
    hr = IAudioCaptureClient_GetBuffer (self->capture_client,
        (BYTE **) & samples, &nsamples_read, &flags, &devpos, NULL);
  }
  while (hr == AUDCLNT_S_BUFFER_EMPTY);

  if (hr != S_OK) {
    GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer () failed: %s",
        gst_wasapi_util_hresult_to_string (hr));
    ret = GST_FLOW_ERROR;
    goto beach;
  }

  if (flags != 0) {
    GST_WARNING_OBJECT (self, "devpos %" G_GUINT64_FORMAT ": flags=0x%08x",
        devpos, flags);
  }

  /* FIXME: Why do we get 1024 sometimes and not a multiple of
   *        samples_per_buffer? Shouldn't WASAPI provide a DISCONT
   *        flag if we read too slow?
   */
  nsamples = nsamples_read;
  g_assert (nsamples >= self->samples_per_buffer);
  if (nsamples > self->samples_per_buffer) {
    GST_WARNING_OBJECT (self,
        "devpos %" G_GUINT64_FORMAT ": got %d samples, expected %d, clipping!",
        devpos, nsamples, self->samples_per_buffer);

    nsamples = self->samples_per_buffer;
  }

  if (clock == NULL || clock == self->clock) {
    timestamp =
        gst_util_uint64_scale (devpos, GST_SECOND, self->client_clock_freq);
  } else {
    GstClockTime base_time;

    timestamp = gst_clock_get_time (clock);

    base_time = GST_ELEMENT_CAST (self)->base_time;
    if (timestamp > base_time)
      timestamp -= base_time;
    else
      timestamp = 0;

    if (timestamp > duration)
      timestamp -= duration;
    else
      timestamp = 0;
  }

  ret = gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (self),
      devpos,
      nsamples * sizeof (gint16), GST_PAD_CAPS (GST_BASE_SRC_PAD (self)), buf);

  if (ret == GST_FLOW_OK) {
    guint i;
    gint16 *dst;

    GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
    GST_BUFFER_TIMESTAMP (*buf) = timestamp;
    GST_BUFFER_DURATION (*buf) = duration;

    dst = (gint16 *) GST_BUFFER_DATA (*buf);
    for (i = 0; i < nsamples; i++) {
      *dst = *samples;

      samples += 2;
      dst++;
    }
  }

  hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read);
  if (hr != S_OK) {
    GST_ERROR_OBJECT (self, "IAudioCaptureClient::ReleaseBuffer () failed: %s",
        gst_wasapi_util_hresult_to_string (hr));
    ret = GST_FLOW_ERROR;
    goto beach;
  }

beach:
  if (clock != NULL)
    gst_object_unref (clock);

  return ret;
}
Exemple #4
0
static HRESULT DSOUND_capture_data(DirectSoundCaptureDevice *device)
{
    HRESULT hr;
    UINT32 packet_frames, packet_bytes, avail_bytes, skip_bytes = 0;
    DWORD flags;
    BYTE *buf;

    if(!device->capture_buffer || device->state == STATE_STOPPED)
        return S_FALSE;

    if(device->state == STATE_STOPPING){
        device->state = STATE_STOPPED;
        return S_FALSE;
    }

    if(device->state == STATE_STARTING)
        device->state = STATE_CAPTURING;

    hr = IAudioCaptureClient_GetBuffer(device->capture, &buf, &packet_frames,
            &flags, NULL, NULL);
    if(FAILED(hr)){
        WARN("GetBuffer failed: %08x\n", hr);
        return hr;
    }

    packet_bytes = packet_frames * device->pwfx->nBlockAlign;
    if(packet_bytes > device->buflen){
        TRACE("audio glitch: dsound buffer too small for data\n");
        skip_bytes = packet_bytes - device->buflen;
        packet_bytes = device->buflen;
    }

    avail_bytes = device->buflen - device->write_pos_bytes;
    if(avail_bytes > packet_bytes)
        avail_bytes = packet_bytes;

    memcpy(device->buffer + device->write_pos_bytes, buf + skip_bytes, avail_bytes);
    capture_CheckNotify(device->capture_buffer, device->write_pos_bytes, avail_bytes);

    packet_bytes -= avail_bytes;
    if(packet_bytes > 0){
        if(device->capture_buffer->flags & DSCBSTART_LOOPING){
            memcpy(device->buffer, buf + skip_bytes + avail_bytes, packet_bytes);
            capture_CheckNotify(device->capture_buffer, 0, packet_bytes);
        }else{
            device->state = STATE_STOPPED;
            capture_CheckNotify(device->capture_buffer, 0, 0);
        }
    }

    device->write_pos_bytes += avail_bytes + packet_bytes;
    device->write_pos_bytes %= device->buflen;

    hr = IAudioCaptureClient_ReleaseBuffer(device->capture, packet_frames);
    if(FAILED(hr)){
        WARN("ReleaseBuffer failed: %08x\n", hr);
        return hr;
    }

    return S_OK;
}
JNIEXPORT jint JNICALL
Java_org_jitsi_impl_neomedia_jmfext_media_protocol_wasapi_WASAPI_IAudioCaptureClient_1Read
    (JNIEnv *env, jclass clazz, jlong thiz, jbyteArray data, jint offset,
        jint length, jint srcSampleSize, jint srcChannels, jint dstSampleSize,
        jint dstChannels)
{
    HRESULT hr;
    IAudioCaptureClient *iAudioCaptureClient
        = (IAudioCaptureClient *) (intptr_t) thiz;
    BYTE *pData;
    UINT32 numFramesToRead;
    DWORD dwFlags;
    jint read;

    hr
        = IAudioCaptureClient_GetBuffer(
                iAudioCaptureClient,
                &pData,
                &numFramesToRead,
                &dwFlags,
                NULL,
                NULL);
    if (SUCCEEDED(hr))
    {
        UINT32 numFramesRead;
        jint dstFrameSize = dstSampleSize * dstChannels;

        if ((numFramesToRead == 0) || (hr == AUDCLNT_S_BUFFER_EMPTY))
            numFramesRead = 0;
        else
        {
            if (length < numFramesToRead * dstFrameSize)
            {
                numFramesRead = 0;
                WASAPI_throwNewHResultException(
                        env,
                        E_INVALIDARG,
                        __func__, __LINE__);
            }
            else
            {
                jbyte *data_
                    = (*env)->GetPrimitiveArrayCritical(env, data, NULL);

                if (data_)
                {
                    numFramesRead
                        = WASAPI_audiocopy(
                                pData, srcSampleSize, srcChannels,
                                data_ + offset, dstSampleSize, dstChannels,
                                numFramesToRead);
                    (*env)->ReleasePrimitiveArrayCritical(env, data, data_, 0);
                }
                else
                    numFramesRead = 0; /* An OutOfMemoryError has been thrown. */
            }
        }
        hr
            = IAudioCaptureClient_ReleaseBuffer(
                    iAudioCaptureClient,
                    numFramesRead);
        read = numFramesRead * dstFrameSize;
        if (FAILED(hr))
            WASAPI_throwNewHResultException(env, hr, __func__, __LINE__);
    }
    else
    {
        read = 0;
        WASAPI_throwNewHResultException(env, hr, __func__, __LINE__);
    }
    return read;
}
Exemple #6
0
static void test_capture(IAudioClient *ac, HANDLE handle, WAVEFORMATEX *wfx)
{
    IAudioCaptureClient *acc;
    HRESULT hr;
    UINT32 frames, next, pad, sum = 0;
    BYTE *data;
    DWORD flags;
    UINT64 pos, qpc;
    REFERENCE_TIME period;

    hr = IAudioClient_GetService(ac, &IID_IAudioCaptureClient, (void**)&acc);
    ok(hr == S_OK, "IAudioClient_GetService(IID_IAudioCaptureClient) returns %08x\n", hr);
    if (hr != S_OK)
        return;

    frames = 0xabadcafe;
    data = (void*)0xdeadf00d;
    flags = 0xabadcafe;
    pos = qpc = 0xdeadbeef;
    hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc);
    ok(hr == AUDCLNT_S_BUFFER_EMPTY, "Initial IAudioCaptureClient_GetBuffer returns %08x\n", hr);

    /* should be empty right after start. Otherwise consume one packet */
    if(hr == S_OK){
        hr = IAudioCaptureClient_ReleaseBuffer(acc, frames);
        ok(hr == S_OK, "Releasing buffer returns %08x\n", hr);
        sum += frames;

        frames = 0xabadcafe;
        data = (void*)0xdeadf00d;
        flags = 0xabadcafe;
        pos = qpc = 0xdeadbeef;
        hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc);
        ok(hr == AUDCLNT_S_BUFFER_EMPTY, "Initial IAudioCaptureClient_GetBuffer returns %08x\n", hr);
    }

    if(hr == AUDCLNT_S_BUFFER_EMPTY){
        ok(!frames, "frames changed to %u\n", frames);
        ok(data == (void*)0xdeadf00d, "data changed to %p\n", data);
        ok(flags == 0xabadcafe, "flags changed to %x\n", flags);
        ok(pos == 0xdeadbeef, "position changed to %u\n", (UINT)pos);
        ok(qpc == 0xdeadbeef, "timer changed to %u\n", (UINT)qpc);

        /* GetNextPacketSize yields 0 if no data is yet available
         * it is not constantly period_size * SamplesPerSec */
        hr = IAudioCaptureClient_GetNextPacketSize(acc, &next);
        ok(hr == S_OK, "IAudioCaptureClient_GetNextPacketSize returns %08x\n", hr);
        ok(!next, "GetNextPacketSize %u\n", next);
    }

    hr = IAudioCaptureClient_ReleaseBuffer(acc, frames);
    ok(hr == S_OK, "Releasing buffer returns %08x\n", hr);
    sum += frames;

    ok(ResetEvent(handle), "ResetEvent\n");

    hr = IAudioCaptureClient_GetNextPacketSize(acc, &next);
    ok(hr == S_OK, "IAudioCaptureClient_GetNextPacketSize returns %08x\n", hr);

    hr = IAudioClient_GetCurrentPadding(ac, &pad);
    ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr);
    ok(next == pad, "GetNextPacketSize %u vs. GCP %u\n", next, pad);
    /* later GCP will grow, while GNPS is 0 or period size */

    hr = IAudioCaptureClient_GetNextPacketSize(acc, NULL);
    ok(hr == E_POINTER, "IAudioCaptureClient_GetNextPacketSize(NULL) returns %08x\n", hr);

    data = (void*)0xdeadf00d;
    frames = 0xdeadbeef;
    flags = 0xabadcafe;
    hr = IAudioCaptureClient_GetBuffer(acc, &data, NULL, NULL, NULL, NULL);
    ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(data, NULL, NULL) returns %08x\n", hr);

    hr = IAudioCaptureClient_GetBuffer(acc, NULL, &frames, NULL, NULL, NULL);
    ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(NULL, &frames, NULL) returns %08x\n", hr);

    hr = IAudioCaptureClient_GetBuffer(acc, NULL, NULL, &flags, NULL, NULL);
    ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(NULL, NULL, &flags) returns %08x\n", hr);

    hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, NULL, NULL, NULL);
    ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(&ata, &frames, NULL) returns %08x\n", hr);
    ok((DWORD_PTR)data == 0xdeadf00d, "data is reset to %p\n", data);
    ok(frames == 0xdeadbeef, "frames is reset to %08x\n", frames);
    ok(flags == 0xabadcafe, "flags is reset to %08x\n", flags);

    hr = IAudioClient_GetDevicePeriod(ac, &period, NULL);
    ok(hr == S_OK, "GetDevicePeriod failed: %08x\n", hr);
    period = MulDiv(period, wfx->nSamplesPerSec, 10000000); /* as in render.c */

    ok(WaitForSingleObject(handle, 1000) == WAIT_OBJECT_0, "Waiting on event handle failed!\n");

    data = (void*)0xdeadf00d;
    hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc);
    ok(hr == S_OK || hr == AUDCLNT_S_BUFFER_EMPTY, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr);
    if (hr == S_OK){
        ok(frames, "Amount of frames locked is 0!\n");
        /* broken: some w7 machines return pad == 0 and DATA_DISCONTINUITY here,
         * AUDCLNT_S_BUFFER_EMPTY above, yet pos == 1-2 * period rather than 0 */
        ok(pos == sum || broken(pos == period || pos == 2*period),
           "Position %u expected %u\n", (UINT)pos, sum);
        sum = pos;
    }else if (hr == AUDCLNT_S_BUFFER_EMPTY){
        ok(!frames, "Amount of frames locked with empty buffer is %u!\n", frames);
        ok(data == (void*)0xdeadf00d, "No data changed to %p\n", data);
    }

    trace("Wait'ed position %d pad %u flags %x, amount of frames locked: %u\n",
          hr==S_OK ? (UINT)pos : -1, pad, flags, frames);

    hr = IAudioCaptureClient_GetNextPacketSize(acc, &next);
    ok(hr == S_OK, "IAudioCaptureClient_GetNextPacketSize returns %08x\n", hr);
    ok(next == frames, "GetNextPacketSize %u vs. GetBuffer %u\n", next, frames);

    hr = IAudioCaptureClient_ReleaseBuffer(acc, frames);
    ok(hr == S_OK, "Releasing buffer returns %08x\n", hr);

    hr = IAudioCaptureClient_ReleaseBuffer(acc, 0);
    ok(hr == S_OK, "Releasing 0 returns %08x\n", hr);

    hr = IAudioCaptureClient_GetNextPacketSize(acc, &next);
    ok(hr == S_OK, "IAudioCaptureClient_GetNextPacketSize returns %08x\n", hr);

    if (frames) {
        hr = IAudioCaptureClient_ReleaseBuffer(acc, frames);
        ok(hr == AUDCLNT_E_OUT_OF_ORDER, "Releasing buffer twice returns %08x\n", hr);
        sum += frames;
    }

    Sleep(350); /* for sure there's data now */

    hr = IAudioClient_GetCurrentPadding(ac, &pad);
    ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr);

    /** GetNextPacketSize
     * returns either 0 or one period worth of frames
     * whereas GetCurrentPadding grows when input is not consumed. */
    hr = IAudioCaptureClient_GetNextPacketSize(acc, &next);
    ok(hr == S_OK, "IAudioCaptureClient_GetNextPacketSize returns %08x\n", hr);
    ok(next <  pad, "GetNextPacketSize %u vs. GCP %u\n", next, pad);

    hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc);
    ok(hr == S_OK, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr);
    ok(next == frames, "GetNextPacketSize %u vs. GetBuffer %u\n", next, frames);

    if(hr == S_OK){
        UINT32 frames2 = frames;
        UINT64 pos2, qpc2;
        ok(frames, "Amount of frames locked is 0!\n");
        ok(pos == sum, "Position %u expected %u\n", (UINT)pos, sum);

        hr = IAudioCaptureClient_ReleaseBuffer(acc, 0);
        ok(hr == S_OK, "Releasing 0 returns %08x\n", hr);

        /* GCP did not decrement, no data consumed */
        hr = IAudioClient_GetCurrentPadding(ac, &frames);
        ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr);
        ok(frames == pad || frames == pad + next /* concurrent feeder */,
           "GCP %u past ReleaseBuffer(0) initially %u\n", frames, pad);

        /* should re-get the same data */
        hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos2, &qpc2);
        ok(hr == S_OK, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr);
        ok(frames2 == frames, "GetBuffer after ReleaseBuffer(0) %u/%u\n", frames2, frames);
        ok(pos2 == pos, "Position after ReleaseBuffer(0) %u/%u\n", (UINT)pos2, (UINT)pos);
        todo_wine ok(qpc2 == qpc, "HPC after ReleaseBuffer(0) %u vs. %u\n", (UINT)qpc2, (UINT)qpc);
    }

    /* trace after the GCP test because log output to MS-DOS console disturbs timing */
    trace("Sleep.1 position %d pad %u flags %x, amount of frames locked: %u\n",
          hr==S_OK ? (UINT)pos : -1, pad, flags, frames);

    if(hr == S_OK){
        UINT32 frames2 = 0xabadcafe;
        BYTE *data2 = (void*)0xdeadf00d;
        flags = 0xabadcafe;

        ok(pos == sum, "Position %u expected %u\n", (UINT)pos, sum);

        pos = qpc = 0xdeadbeef;
        hr = IAudioCaptureClient_GetBuffer(acc, &data2, &frames2, &flags, &pos, &qpc);
        ok(hr == AUDCLNT_E_OUT_OF_ORDER, "Out of order IAudioCaptureClient_GetBuffer returns %08x\n", hr);
        ok(frames2 == 0xabadcafe, "Out of order frames changed to %x\n", frames2);
        ok(data2 == (void*)0xdeadf00d, "Out of order data changed to %p\n", data2);
        ok(flags == 0xabadcafe, "Out of order flags changed to %x\n", flags);
        ok(pos == 0xdeadbeef, "Out of order position changed to %x\n", (UINT)pos);
        ok(qpc == 0xdeadbeef, "Out of order timer changed to %x\n", (UINT)qpc);

        hr = IAudioCaptureClient_ReleaseBuffer(acc, frames+1);
        ok(hr == AUDCLNT_E_INVALID_SIZE, "Releasing buffer+1 returns %08x\n", hr);

        hr = IAudioCaptureClient_ReleaseBuffer(acc, 1);
        ok(hr == AUDCLNT_E_INVALID_SIZE, "Releasing 1 returns %08x\n", hr);

        hr = IAudioClient_Reset(ac);
        ok(hr == AUDCLNT_E_NOT_STOPPED, "Reset failed: %08x\n", hr);
    }

    hr = IAudioCaptureClient_ReleaseBuffer(acc, frames);
    ok(hr == S_OK, "Releasing buffer returns %08x\n", hr);

    if (frames) {
        sum += frames;
        hr = IAudioCaptureClient_ReleaseBuffer(acc, frames);
        ok(hr == AUDCLNT_E_OUT_OF_ORDER, "Releasing buffer twice returns %08x\n", hr);
    }

    frames = period;
    ok(next == frames, "GetNextPacketSize %u vs. GetDevicePeriod %u\n", next, frames);

    /* GetBufferSize is not a multiple of the period size! */
    hr = IAudioClient_GetBufferSize(ac, &next);
    ok(hr == S_OK, "GetBufferSize failed: %08x\n", hr);
    trace("GetBufferSize %u period size %u\n", next, frames);

    Sleep(400); /* overrun */

    hr = IAudioClient_GetCurrentPadding(ac, &pad);
    ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr);

    hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc);
    ok(hr == S_OK, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr);

    trace("Overrun position %d pad %u flags %x, amount of frames locked: %u\n",
          hr==S_OK ? (UINT)pos : -1, pad, flags, frames);

    if(hr == S_OK){
        /* The discontinuity is reported here, but is this an old or new packet? */
        todo_wine ok(flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY, "expect DISCONTINUITY %x\n", flags);
        ok(pad == next, "GCP %u vs. BufferSize %u\n", (UINT32)pad, next);

        /* Native's position is one period further than what we read.
         * Perhaps that's precisely the meaning of DATA_DISCONTINUITY:
         * signal when the position jump left a gap. */
        todo_wine ok(pos == sum + frames, "Position %u gap %d\n",
                     (UINT)pos, (UINT)pos - sum);
        if(flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY)
            sum = pos;
    }

    hr = IAudioCaptureClient_ReleaseBuffer(acc, frames);
    ok(hr == S_OK, "Releasing buffer returns %08x\n", hr);
    sum += frames;

    hr = IAudioClient_GetCurrentPadding(ac, &pad);
    ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr);

    hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc);
    ok(hr == S_OK, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr);

    trace("Cont'ed position %d pad %u flags %x, amount of frames locked: %u\n",
          hr==S_OK ? (UINT)pos : -1, pad, flags, frames);

    if(hr == S_OK){
        ok(pos == sum, "Position %u expected %u\n", (UINT)pos, sum);
        ok(!flags, "flags %u\n", flags);

        hr = IAudioCaptureClient_ReleaseBuffer(acc, frames);
        ok(hr == S_OK, "Releasing buffer returns %08x\n", hr);
        sum += frames;
    }

    hr = IAudioClient_Stop(ac);
    ok(hr == S_OK, "Stop on a started stream returns %08x\n", hr);

    hr = IAudioClient_Start(ac);
    ok(hr == S_OK, "Start on a stopped stream returns %08x\n", hr);

    hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc);
    ok(hr == S_OK, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr);

    hr = IAudioClient_GetCurrentPadding(ac, &pad);
    ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr);

    trace("Restart position %d pad %u flags %x, amount of frames locked: %u\n",
          hr==S_OK ? (UINT)pos : -1, pad, flags, frames);
    ok(pad > sum, "restarted GCP %u\n", pad); /* GCP is still near buffer size */

    if(frames){
        ok(pos == sum, "Position %u expected %u\n", (UINT)pos, sum);
        ok(!flags, "flags %u\n", flags);

        hr = IAudioCaptureClient_ReleaseBuffer(acc, frames);
        ok(hr == S_OK, "Releasing buffer returns %08x\n", hr);
        sum += frames;
    }

    hr = IAudioClient_Stop(ac);
    ok(hr == S_OK, "Stop on a started stream returns %08x\n", hr);

    hr = IAudioClient_Reset(ac);
    ok(hr == S_OK, "Reset on a stopped stream returns %08x\n", hr);
    sum += pad - frames;

    hr = IAudioClient_Start(ac);
    ok(hr == S_OK, "Start on a stopped stream returns %08x\n", hr);

    hr = IAudioClient_GetCurrentPadding(ac, &pad);
    ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr);

    flags = 0xabadcafe;
    hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc);
    ok(hr == AUDCLNT_S_BUFFER_EMPTY || /*PulseAudio*/hr == S_OK,
       "Initial IAudioCaptureClient_GetBuffer returns %08x\n", hr);

    trace("Reset   position %d pad %u flags %x, amount of frames locked: %u\n",
          hr==S_OK ? (UINT)pos : -1, pad, flags, frames);

    if(hr == S_OK){
        /* Only PulseAudio goes here; despite snd_pcm_drop it manages
         * to fill GetBufferSize with a single snd_pcm_read */
        trace("Test marked todo: only PulseAudio gets here\n");
        todo_wine ok(flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY, "expect DISCONTINUITY %x\n", flags);
        /* Reset zeroes padding, not the position */
        ok(pos >= sum, "Position %u last %u\n", (UINT)pos, sum);
        /*sum = pos; check after next GetBuffer */

        hr = IAudioCaptureClient_ReleaseBuffer(acc, frames);
        ok(hr == S_OK, "Releasing buffer returns %08x\n", hr);
        sum += frames;
    }
    else if(hr == AUDCLNT_S_BUFFER_EMPTY){
        ok(!pad, "resetted GCP %u\n", pad);
        Sleep(180);
    }

    hr = IAudioClient_GetCurrentPadding(ac, &pad);
    ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr);

    hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc);
    ok(hr == S_OK, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr);
    trace("Running position %d pad %u flags %x, amount of frames locked: %u\n",
          hr==S_OK ? (UINT)pos : -1, pad, flags, frames);

    if(hr == S_OK){
        /* Some w7 machines signal DATA_DISCONTINUITY here following the
         * previous AUDCLNT_S_BUFFER_EMPTY, others not.  What logic? */
        ok(pos >= sum, "Position %u gap %d\n", (UINT)pos, (UINT)pos - sum);
        IAudioCaptureClient_ReleaseBuffer(acc, frames);
    }

    IAudioCaptureClient_Release(acc);
}