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mpv/audio/out/ao_wasapi.c

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/*
* 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/threads.h"
#include "osdep/timer.h"
#include "osdep/io.h"
#include "misc/dispatch.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_ns)
{
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_ns = diff * 1e9 / 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_ns -= qpc_diff * 100.0; // convert to ns
} else {
MP_VERBOSE(state, "Insane qpc delay correction of %g seconds. "
"Ignoring it.\n", qpc_diff / 10000000.0);
}
if (sample_count > 0 && *delay_ns <= 0) {
MP_WARN(state, "Under-run: Device delay: %g ns\n", *delay_ns);
} else {
MP_TRACE(state, "Device delay: %g ns\n", *delay_ns);
}
return S_OK;
exit_label:
MP_ERR(state, "Error getting device delay: %s\n", mp_HRESULT_to_str(hr));
return hr;
}
static bool thread_feed(struct ao *ao)
{
struct wasapi_state *state = ao->priv;
HRESULT hr;
UINT32 frame_count = state->bufferFrameCount;
UINT32 padding;
hr = IAudioClient_GetCurrentPadding(state->pAudioClient, &padding);
EXIT_ON_ERROR(hr);
bool refill = false;
if (state->share_mode == AUDCLNT_SHAREMODE_SHARED) {
// Return if there's nothing to do.
if (frame_count <= padding)
return false;
// In shared mode, there is only one buffer of size bufferFrameCount.
// We must therefore take care not to overwrite the samples that have
// yet to play.
frame_count -= padding;
} else if (padding >= 2 * frame_count) {
// In exclusive mode, we exchange entire buffers of size
// bufferFrameCount with the device. If there are already two such
// full buffers waiting to play, there is no work to do.
return false;
} else if (padding < frame_count) {
// If there is not at least one full buffer of audio queued to play in
// exclusive mode, call this function again immediately to try and catch
// up and avoid a cascade of under-runs. WASAPI doesn't seem to be smart
// enough to send more feed events when it gets behind.
refill = true;
}
MP_TRACE(ao, "Frame to fill: %"PRIu32". Padding: %"PRIu32"\n",
frame_count, padding);
double delay_ns;
hr = get_device_delay(state, &delay_ns);
EXIT_ON_ERROR(hr);
// add the buffer delay
delay_ns += frame_count * 1e9 / 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_converted(ao, &state->convert_format,
(void **)data, frame_count,
mp_time_ns() + (int64_t)llrint(delay_ns));
// note, we can't use ao_read_data return value here since we already
// committed 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 refill;
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 false;
}
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);
}
static void thread_resume(struct ao *ao)
{
struct wasapi_state *state = ao->priv;
MP_DBG(state, "Thread Resume\n");
thread_reset(ao);
thread_feed(ao);
HRESULT hr = IAudioClient_Start(state->pAudioClient);
if (FAILED(hr)) {
MP_ERR(state, "IAudioClient_Start returned %s\n",
mp_HRESULT_to_str(hr));
}
}
static void set_state_and_wakeup_thread(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 thread_process_dispatch(void *ptr)
{
set_state_and_wakeup_thread(ptr, WASAPI_THREAD_DISPATCH);
}
static DWORD __stdcall AudioThread(void *lpParameter)
{
struct ao *ao = lpParameter;
struct wasapi_state *state = ao->priv;
mp_thread_set_name("ao/wasapi");
CoInitializeEx(NULL, COINIT_APARTMENTTHREADED);
state->init_ok = wasapi_thread_init(ao);
SetEvent(state->hInitDone);
if (!state->init_ok)
goto exit_label;
MP_DBG(ao, "Entering dispatch loop\n");
while (true) {
if (WaitForSingleObject(state->hWake, INFINITE) != WAIT_OBJECT_0)
MP_ERR(ao, "Unexpected return value from WaitForSingleObject\n");
int thread_state = atomic_load(&state->thread_state);
switch (thread_state) {
case WASAPI_THREAD_FEED:
// fill twice on under-full buffer (see comment in thread_feed)
if (thread_feed(ao) && thread_feed(ao))
MP_ERR(ao, "Unable to fill buffer fast enough\n");
break;
case WASAPI_THREAD_DISPATCH:
mp_dispatch_queue_process(state->dispatch, 0);
break;
case WASAPI_THREAD_RESET:
thread_reset(ao);
break;
case WASAPI_THREAD_RESUME:
thread_resume(ao);
break;
case WASAPI_THREAD_SHUTDOWN:
thread_reset(ao);
goto exit_label;
default:
MP_ERR(ao, "Unhandled thread state: %d\n", thread_state);
}
// the default is to feed unless something else is requested
atomic_compare_exchange_strong(&state->thread_state, &thread_state,
WASAPI_THREAD_FEED);
}
exit_label:
wasapi_thread_uninit(ao);
CoUninitialize();
MP_DBG(ao, "Thread return\n");
return 0;
}
static void uninit(struct ao *ao)
{
MP_DBG(ao, "Uninit wasapi\n");
struct wasapi_state *state = ao->priv;
if (state->hWake)
set_state_and_wakeup_thread(ao, WASAPI_THREAD_SHUTDOWN);
if (state->hAudioThread &&
WaitForSingleObject(state->hAudioThread, INFINITE) != WAIT_OBJECT_0)
{
MP_ERR(ao, "Unexpected return value from WaitForSingleObject "
"while waiting for audio thread to terminate\n");
}
SAFE_DESTROY(state->hInitDone, CloseHandle(state->hInitDone));
SAFE_DESTROY(state->hWake, CloseHandle(state->hWake));
SAFE_DESTROY(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_MULTITHREADED);
struct wasapi_state *state = ao->priv;
state->log = ao->log;
state->opt_exclusive |= ao->init_flags & AO_INIT_EXCLUSIVE;
#if !HAVE_UWP
state->deviceID = wasapi_find_deviceID(ao);
if (!state->deviceID) {
uninit(ao);
return -1;
}
#endif
if (state->deviceID)
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_FATAL(ao, "Error creating events\n");
uninit(ao);
return -1;
}
state->dispatch = mp_dispatch_create(state);
mp_dispatch_set_wakeup_fn(state->dispatch, thread_process_dispatch, ao);
state->init_ok = false;
state->hAudioThread = CreateThread(NULL, 0, &AudioThread, ao, 0, NULL);
if (!state->hAudioThread) {
MP_FATAL(ao, "Failed to create audio thread\n");
uninit(ao);
return -1;
}
WaitForSingleObject(state->hInitDone, INFINITE); // wait on init complete
SAFE_DESTROY(state->hInitDone,CloseHandle(state->hInitDone));
if (!state->init_ok) {
if (!ao->probing)
MP_FATAL(ao, "Received failure from audio thread\n");
uninit(ao);
return -1;
}
MP_DBG(ao, "Init wasapi done\n");
return 0;
}
static int thread_control_exclusive(struct ao *ao, enum aocontrol cmd, void *arg)
{
struct wasapi_state *state = ao->priv;
if (!state->pEndpointVolume)
return CONTROL_UNKNOWN;
switch (cmd) {
case AOCONTROL_GET_VOLUME:
case AOCONTROL_SET_VOLUME:
if (!(state->vol_hw_support & ENDPOINT_HARDWARE_SUPPORT_VOLUME))
return CONTROL_FALSE;
break;
case AOCONTROL_GET_MUTE:
case AOCONTROL_SET_MUTE:
if (!(state->vol_hw_support & ENDPOINT_HARDWARE_SUPPORT_MUTE))
return CONTROL_FALSE;
break;
}
float volume;
BOOL mute;
switch (cmd) {
case AOCONTROL_GET_VOLUME:
IAudioEndpointVolume_GetMasterVolumeLevelScalar(
state->pEndpointVolume, &volume);
*(float *)arg = volume * 100.f;
return CONTROL_OK;
case AOCONTROL_SET_VOLUME:
volume = (*(float *)arg) / 100.f;
IAudioEndpointVolume_SetMasterVolumeLevelScalar(
state->pEndpointVolume, volume, NULL);
return CONTROL_OK;
case AOCONTROL_GET_MUTE:
IAudioEndpointVolume_GetMute(state->pEndpointVolume, &mute);
*(bool *)arg = mute;
return CONTROL_OK;
case AOCONTROL_SET_MUTE:
mute = *(bool *)arg;
IAudioEndpointVolume_SetMute(state->pEndpointVolume, mute, NULL);
return CONTROL_OK;
}
return CONTROL_UNKNOWN;
}
static int thread_control_shared(struct ao *ao, enum aocontrol cmd, void *arg)
{
struct wasapi_state *state = ao->priv;
if (!state->pAudioVolume)
return CONTROL_UNKNOWN;
float volume;
BOOL mute;
switch(cmd) {
case AOCONTROL_GET_VOLUME:
ISimpleAudioVolume_GetMasterVolume(state->pAudioVolume, &volume);
*(float *)arg = volume * 100.f;
return CONTROL_OK;
case AOCONTROL_SET_VOLUME:
volume = (*(float *)arg) / 100.f;
ISimpleAudioVolume_SetMasterVolume(state->pAudioVolume, volume, NULL);
return CONTROL_OK;
case AOCONTROL_GET_MUTE:
ISimpleAudioVolume_GetMute(state->pAudioVolume, &mute);
*(bool *)arg = mute;
return CONTROL_OK;
case AOCONTROL_SET_MUTE:
mute = *(bool *)arg;
ISimpleAudioVolume_SetMute(state->pAudioVolume, mute, NULL);
return CONTROL_OK;
}
return CONTROL_UNKNOWN;
}
static int thread_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->pSessionControl)
return CONTROL_FALSE;
wchar_t *title = mp_from_utf8(NULL, (const char *)arg);
HRESULT hr = IAudioSessionControl_SetDisplayName(state->pSessionControl,
title,NULL);
talloc_free(title);
if (SUCCEEDED(hr))
return CONTROL_OK;
MP_WARN(ao, "Error setting audio session name: %s\n",
mp_HRESULT_to_str(hr));
assert(ao->client_name);
if (!ao->client_name)
return CONTROL_ERROR;
// Fallback to client name
title = mp_from_utf8(NULL, ao->client_name);
IAudioSessionControl_SetDisplayName(state->pSessionControl,
title, NULL);
talloc_free(title);
return CONTROL_ERROR;
}
return state->share_mode == AUDCLNT_SHAREMODE_EXCLUSIVE ?
thread_control_exclusive(ao, cmd, arg) :
thread_control_shared(ao, cmd, arg);
}
static void run_control(void *p)
{
void **pp = p;
struct ao *ao = pp[0];
enum aocontrol cmd = *(enum aocontrol *)pp[1];
void *arg = pp[2];
*(int *)pp[3] = thread_control(ao, cmd, arg);
}
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
struct wasapi_state *state = ao->priv;
int ret;
void *p[] = {ao, &cmd, arg, &ret};
mp_dispatch_run(state->dispatch, run_control, p);
return ret;
}
static void audio_reset(struct ao *ao)
{
set_state_and_wakeup_thread(ao, WASAPI_THREAD_RESET);
}
static void audio_resume(struct ao *ao)
{
set_state_and_wakeup_thread(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_MULTITHREADED);
HRESULT hr = wasapi_change_init(ao, true);
EXIT_ON_ERROR(hr);
return 0;
exit_label:
MP_FATAL(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,
.start = audio_resume,
.list_devs = wasapi_list_devs,
.hotplug_init = hotplug_init,
.hotplug_uninit = hotplug_uninit,
.priv_size = sizeof(wasapi_state),
};
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