mpv/audio/out/ao_wasapi.c

/*
 * This file is part of mpv.
 *
 * Original author: Jonathan Yong <10walls@gmail.com>
 *
 * mpv is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * mpv is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
 */

#include <math.h>
#include <inttypes.h>
#include <libavutil/mathematics.h>

#include "options/m_option.h"
#include "osdep/timer.h"
#include "osdep/io.h"
#include "ao_wasapi.h"

// naive av_rescale for unsigned
static UINT64 uint64_scale(UINT64 x, UINT64 num, UINT64 den)
{
    return (x / den) * num
        + ((x % den) * (num / den))
        + ((x % den) * (num % den)) / den;
}

static HRESULT get_device_delay(struct wasapi_state *state, double *delay_us) {
    UINT64 sample_count = atomic_load(&state->sample_count);
    UINT64 position, qpc_position;
    HRESULT hr;

    hr = IAudioClock_GetPosition(state->pAudioClock, &position, &qpc_position);
    EXIT_ON_ERROR(hr);
    // GetPosition succeeded, but the result may be
    // inaccurate due to the length of the call
    // http://msdn.microsoft.com/en-us/library/windows/desktop/dd370889%28v=vs.85%29.aspx
    if (hr == S_FALSE)
        MP_VERBOSE(state, "Possibly inaccurate device position.\n");

    // convert position to number of samples careful to avoid overflow
    UINT64 sample_position = uint64_scale(position,
                                          state->format.Format.nSamplesPerSec,
                                          state->clock_frequency);
    INT64 diff = sample_count - sample_position;
    *delay_us = diff * 1e6 / state->format.Format.nSamplesPerSec;

    // Correct for any delay in IAudioClock_GetPosition above.
    // This should normally be very small (<1 us), but just in case. . .
    LARGE_INTEGER qpc;
    QueryPerformanceCounter(&qpc);
    INT64 qpc_diff = av_rescale(qpc.QuadPart, 10000000, state->qpc_frequency.QuadPart)
                     - qpc_position;
    // ignore the above calculation if it yields more than 10 seconds (due to
    // possible overflow inside IAudioClock_GetPosition)
    if (qpc_diff < 10 * 10000000) {
        *delay_us -= qpc_diff / 10.0; // convert to us
    } else {
        MP_VERBOSE(state, "Insane qpc delay correction of %g seconds. "
                   "Ignoring it.\n", qpc_diff / 10000000.0);
    }

    MP_TRACE(state, "Device delay: %g us\n", *delay_us);

    return S_OK;
exit_label:
    MP_ERR(state, "Error getting device delay: %s\n", mp_HRESULT_to_str(hr));
    return hr;
}

static void thread_feed(struct ao *ao)
{
    struct wasapi_state *state = ao->priv;
    HRESULT hr;

    UINT32 frame_count = state->bufferFrameCount;

    if (state->share_mode == AUDCLNT_SHAREMODE_SHARED) {
        UINT32 padding = 0;
        hr = IAudioClient_GetCurrentPadding(state->pAudioClient, &padding);
        EXIT_ON_ERROR(hr);

        frame_count -= padding;
        MP_TRACE(ao, "Frame to fill: %"PRIu32". Padding: %"PRIu32"\n",
                 frame_count, padding);
    }
    double delay_us;
    hr = get_device_delay(state, &delay_us);
    EXIT_ON_ERROR(hr);
    // add the buffer delay
    delay_us += frame_count * 1e6 / state->format.Format.nSamplesPerSec;

    BYTE *pData;
    hr = IAudioRenderClient_GetBuffer(state->pRenderClient,
                                      frame_count, &pData);
    EXIT_ON_ERROR(hr);

    BYTE *data[1] = {pData};

    ao_read_data(ao, (void **)data, frame_count,
                 mp_time_us() + (int64_t)llrint(delay_us));

    // note, we can't use ao_read_data return value here since we already
    // commited to frame_count above in the GetBuffer call
    hr = IAudioRenderClient_ReleaseBuffer(state->pRenderClient,
                                          frame_count, 0);
    EXIT_ON_ERROR(hr);

    atomic_fetch_add(&state->sample_count, frame_count);

    return;
exit_label:
    MP_ERR(state, "Error feeding audio: %s\n", mp_HRESULT_to_str(hr));
    MP_VERBOSE(ao, "Requesting ao reload\n");
    ao_request_reload(ao);
    return;
}

static void thread_resume(struct ao *ao)
{
    struct wasapi_state *state = ao->priv;
    HRESULT hr;

    MP_DBG(state, "Thread Resume\n");
    UINT32 padding = 0;
    hr = IAudioClient_GetCurrentPadding(state->pAudioClient, &padding);
    if (FAILED(hr)) {
        MP_ERR(state, "IAudioClient_GetCurrentPadding returned %s\n",
               mp_HRESULT_to_str(hr));
    }

    // Fill the buffer before starting, but only if there is no audio queued to
    // play.  This prevents overfilling the buffer, which leads to problems in
    // exclusive mode
    if (padding < (UINT32) state->bufferFrameCount)
        thread_feed(ao);

    // start feeding next wakeup if something else hasn't been requested
    int expected = WASAPI_THREAD_RESUME;
    atomic_compare_exchange_strong(&state->thread_state, &expected,
                                   WASAPI_THREAD_FEED);
    hr = IAudioClient_Start(state->pAudioClient);
    if (FAILED(hr)) {
        MP_ERR(state, "IAudioClient_Start returned %s\n",
               mp_HRESULT_to_str(hr));
    }

    return;
}

static void thread_reset(struct ao *ao)
{
    struct wasapi_state *state = ao->priv;
    HRESULT hr;
    MP_DBG(state, "Thread Reset\n");
    hr = IAudioClient_Stop(state->pAudioClient);
    if (FAILED(hr))
        MP_ERR(state, "IAudioClient_Stop returned: %s\n", mp_HRESULT_to_str(hr));

    hr = IAudioClient_Reset(state->pAudioClient);
    if (FAILED(hr))
        MP_ERR(state, "IAudioClient_Reset returned: %s\n", mp_HRESULT_to_str(hr));

    atomic_store(&state->sample_count, 0);
    // start feeding next wakeup if something else hasn't been requested
    int expected = WASAPI_THREAD_RESET;
    atomic_compare_exchange_strong(&state->thread_state, &expected,
                                   WASAPI_THREAD_FEED);
    return;
}

static DWORD __stdcall AudioThread(void *lpParameter)
{
    struct ao *ao = lpParameter;
    struct wasapi_state *state = ao->priv;
    CoInitializeEx(NULL, COINIT_APARTMENTTHREADED);

    state->init_ret = wasapi_thread_init(ao);
    SetEvent(state->hInitDone);
    if (FAILED(state->init_ret))
        goto exit_label;

    MP_DBG(ao, "Entering dispatch loop\n");
    while (true) { // watch events
        HANDLE events[] = {state->hWake};
        switch (MsgWaitForMultipleObjects(MP_ARRAY_SIZE(events), events,
                                          FALSE, INFINITE,
                                          QS_POSTMESSAGE | QS_SENDMESSAGE)) {
        // AudioThread wakeup
        case WAIT_OBJECT_0:
            switch (atomic_load(&state->thread_state)) {
            case WASAPI_THREAD_FEED:
                thread_feed(ao);
                break;
            case WASAPI_THREAD_RESET:
                thread_reset(ao);
                break;
            case WASAPI_THREAD_RESUME:
                thread_reset(ao);
                thread_resume(ao);
                break;
            case WASAPI_THREAD_SHUTDOWN:
                thread_reset(ao);
                goto exit_label;
            default:
                MP_ERR(ao, "Unhandled thread state\n");
                goto exit_label;
            }
            break;
        // messages to dispatch (COM marshalling)
        case (WAIT_OBJECT_0 + MP_ARRAY_SIZE(events)):
            wasapi_dispatch(ao);
            break;
        default:
            MP_ERR(ao, "Unhandled thread event\n");
            goto exit_label;
        }
    }
exit_label:
    wasapi_thread_uninit(ao);

    CoUninitialize();
    MP_DBG(ao, "Thread return\n");
    return 0;
}

static void set_thread_state(struct ao *ao,
                             enum wasapi_thread_state thread_state)
{
    struct wasapi_state *state = ao->priv;
    atomic_store(&state->thread_state, thread_state);
    SetEvent(state->hWake);
}

static void uninit(struct ao *ao)
{
    MP_DBG(ao, "Uninit wasapi\n");
    struct wasapi_state *state = ao->priv;
    wasapi_release_proxies(state);
    if (state->hWake)
        set_thread_state(ao, WASAPI_THREAD_SHUTDOWN);

    // wait up to 10 seconds
    if (state->hAudioThread &&
        WaitForSingleObject(state->hAudioThread, 10000) == WAIT_TIMEOUT)
    {
        MP_ERR(ao, "Audio loop thread refuses to abort\n");
        return;
    }

    SAFE_RELEASE(state->hInitDone,   CloseHandle(state->hInitDone));
    SAFE_RELEASE(state->hWake,       CloseHandle(state->hWake));
    SAFE_RELEASE(state->hAudioThread,CloseHandle(state->hAudioThread));

    wasapi_change_uninit(ao);

    talloc_free(state->deviceID);

    CoUninitialize();
    MP_DBG(ao, "Uninit wasapi done\n");
}

static int init(struct ao *ao)
{
    MP_DBG(ao, "Init wasapi\n");
    CoInitializeEx(NULL, COINIT_APARTMENTTHREADED);

    struct wasapi_state *state = ao->priv;
    state->log = ao->log;

    state->deviceID = wasapi_find_deviceID(ao);
    if (!state->deviceID) {
        uninit(ao);
        return -1;
    }

    wasapi_change_init(ao, false);

    state->hInitDone = CreateEventW(NULL, FALSE, FALSE, NULL);
    state->hWake     = CreateEventW(NULL, FALSE, FALSE, NULL);
    if (!state->hInitDone || !state->hWake) {
        MP_ERR(ao, "Error creating events\n");
        uninit(ao);
        return -1;
    }

    state->init_ret = E_FAIL;
    state->hAudioThread = CreateThread(NULL, 0, &AudioThread, ao, 0, NULL);
    if (!state->hAudioThread) {
        MP_ERR(ao, "Failed to create audio thread\n");
        uninit(ao);
        return -1;
    }

    WaitForSingleObject(state->hInitDone, INFINITE); // wait on init complete
    SAFE_RELEASE(state->hInitDone,CloseHandle(state->hInitDone));
    if (FAILED(state->init_ret)) {
        if (!ao->probing)
            MP_ERR(ao, "Received failure from audio thread\n");
        uninit(ao);
        return -1;
    }

    wasapi_receive_proxies(state);
    MP_DBG(ao, "Init wasapi done\n");
    return 0;
}

static int control_exclusive(struct ao *ao, enum aocontrol cmd, void *arg)
{
    struct wasapi_state *state = ao->priv;

    switch (cmd) {
    case AOCONTROL_GET_VOLUME:
    case AOCONTROL_SET_VOLUME:
        if (!state->pEndpointVolumeProxy ||
            !(state->vol_hw_support & ENDPOINT_HARDWARE_SUPPORT_VOLUME)) {
            return CONTROL_FALSE;
        }

        float volume;
        switch (cmd) {
        case AOCONTROL_GET_VOLUME:
            IAudioEndpointVolume_GetMasterVolumeLevelScalar(
                state->pEndpointVolumeProxy,
                &volume);
            *(ao_control_vol_t *)arg = (ao_control_vol_t){
                .left  = 100.0f * volume,
                .right = 100.0f * volume,
            };
            return CONTROL_OK;
        case AOCONTROL_SET_VOLUME:
            volume = ((ao_control_vol_t *)arg)->left / 100.f;
            IAudioEndpointVolume_SetMasterVolumeLevelScalar(
                state->pEndpointVolumeProxy,
                volume, NULL);
            return CONTROL_OK;
        }
    case AOCONTROL_GET_MUTE:
    case AOCONTROL_SET_MUTE:
        if (!state->pEndpointVolumeProxy ||
            !(state->vol_hw_support & ENDPOINT_HARDWARE_SUPPORT_MUTE)) {
            return CONTROL_FALSE;
        }

        BOOL mute;
        switch (cmd) {
        case AOCONTROL_GET_MUTE:
            IAudioEndpointVolume_GetMute(state->pEndpointVolumeProxy,
                                         &mute);
            *(bool *)arg = mute;
            return CONTROL_OK;
        case AOCONTROL_SET_MUTE:
            mute = *(bool *)arg;
            IAudioEndpointVolume_SetMute(state->pEndpointVolumeProxy,
                                         mute, NULL);
            return CONTROL_OK;
        }
    case AOCONTROL_HAS_PER_APP_VOLUME:
        return CONTROL_FALSE;
    default:
        return CONTROL_UNKNOWN;
    }
}

static int control_shared(struct ao *ao, enum aocontrol cmd, void *arg)
{
    struct wasapi_state *state = ao->priv;
    if (!state->pAudioVolumeProxy)
        return CONTROL_UNKNOWN;

    float volume;
    BOOL mute;
    switch(cmd) {
    case AOCONTROL_GET_VOLUME:
        ISimpleAudioVolume_GetMasterVolume(state->pAudioVolumeProxy,
                                           &volume);
        *(ao_control_vol_t *)arg = (ao_control_vol_t){
            .left  = 100.0f * volume,
            .right = 100.0f * volume,
        };
        return CONTROL_OK;
    case AOCONTROL_SET_VOLUME:
        volume = ((ao_control_vol_t *)arg)->left / 100.f;
        ISimpleAudioVolume_SetMasterVolume(state->pAudioVolumeProxy,
                                           volume, NULL);
        return CONTROL_OK;
    case AOCONTROL_GET_MUTE:
        ISimpleAudioVolume_GetMute(state->pAudioVolumeProxy, &mute);
        *(bool *)arg = mute;
        return CONTROL_OK;
    case AOCONTROL_SET_MUTE:
        mute = *(bool *)arg;
        ISimpleAudioVolume_SetMute(state->pAudioVolumeProxy, mute, NULL);
        return CONTROL_OK;
    case AOCONTROL_HAS_PER_APP_VOLUME:
        return CONTROL_TRUE;
    default:
        return CONTROL_UNKNOWN;
    }
}

static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
    struct wasapi_state *state = ao->priv;

    // common to exclusive and shared
    switch (cmd) {
    case AOCONTROL_UPDATE_STREAM_TITLE:
        if (!state->pSessionControlProxy)
            return CONTROL_FALSE;

        wchar_t *title = mp_from_utf8(NULL, (char*)arg);
        wchar_t *tmp = NULL;
        // There is a weird race condition in the IAudioSessionControl itself --
        // it seems that *sometimes* the SetDisplayName does not take effect and
        // it still shows the old title. Use this loop to insist until it works.
        do {
            IAudioSessionControl_SetDisplayName(state->pSessionControlProxy,
                                                title, NULL);

            SAFE_RELEASE(tmp, CoTaskMemFree(tmp));
            IAudioSessionControl_GetDisplayName(state->pSessionControlProxy,
                                                &tmp);
        } while (lstrcmpW(title, tmp));
        SAFE_RELEASE(tmp, CoTaskMemFree(tmp));
        talloc_free(title);
        return CONTROL_OK;
    }

    return state->share_mode == AUDCLNT_SHAREMODE_EXCLUSIVE ?
        control_exclusive(ao, cmd, arg) : control_shared(ao, cmd, arg);
}

static void audio_reset(struct ao *ao)
{
    set_thread_state(ao, WASAPI_THREAD_RESET);
}

static void audio_resume(struct ao *ao)
{
    set_thread_state(ao, WASAPI_THREAD_RESUME);
}

static void hotplug_uninit(struct ao *ao)
{
    MP_DBG(ao, "Hotplug uninit\n");
    wasapi_change_uninit(ao);
    CoUninitialize();
}

static int hotplug_init(struct ao *ao)
{
    MP_DBG(ao, "Hotplug init\n");
    struct wasapi_state *state = ao->priv;
    state->log = ao->log;
    CoInitializeEx(NULL, COINIT_APARTMENTTHREADED);
    HRESULT hr = wasapi_change_init(ao, true);
    EXIT_ON_ERROR(hr);

    return 0;
    exit_label:
    MP_ERR(state, "Error setting up audio hotplug: %s\n", mp_HRESULT_to_str(hr));
    hotplug_uninit(ao);
    return -1;
}

#define OPT_BASE_STRUCT struct wasapi_state

const struct ao_driver audio_out_wasapi = {
    .description    = "Windows WASAPI audio output (event mode)",
    .name           = "wasapi",
    .init           = init,
    .uninit         = uninit,
    .control        = control,
    .reset          = audio_reset,
    .resume         = audio_resume,
    .list_devs      = wasapi_list_devs,
    .hotplug_init   = hotplug_init,
    .hotplug_uninit = hotplug_uninit,
    .priv_size      = sizeof(wasapi_state),
    .options        = (const struct m_option[]) {
        OPT_FLAG("exclusive", opt_exclusive, 0),
        OPT_STRING("device", opt_device, 0),
        {NULL},
    },
};