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mpv/audio/out/ao_wasapi.c
nanahi 51e01e9772 ao_wasapi: fix player core lockup when avoiding premature buffer fills
6863eefc3d handled this situation by using
an atomic variable to express the state for which the wakeup is caused
by AO control, and the dispatch queue is only processed at this state.
However, this can cause permanent lockup of the player core when the
following happens:

- AO control sets the thread state to WASAPI_THREAD_DISPATCH, and
  sets the wakeup handle.
- WASAPI thread reads the WASAPI_THREAD_DISPATCH state and processes
  the dispatch queue.
- Another AO control happens. A dispatch item is enqueued, and the
  state stays at WASAPI_THREAD_DISPATCH.
- WASAPI thread resets the thread state to WASAPI_THREAD_FEED since
  the state has not changed.
- WaitForSingleObject() returns in the WASAPI thread, sees this state,
  and does not process the dispatch queue.
- The player core locks permanently because it is waiting for the dispatch
  to be processed.

This has been experimentally verified on a system under high contention:
The easiest way to trigger this lockup is to continuously hold down "i",
which rapidly issues AO get volume/mute controls.

To properly handle this, use separate handles for system and user wakeup
requests. Only feed audio when woke up by system and only process the
dispatch queue when woke up by user.

Fixes: 6863eefc3d
2024-04-27 00:59:09 +02:00

547 lines
17 KiB
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/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_pause(struct ao *ao)
{
struct wasapi_state *state = ao->priv;
MP_DBG(state, "Thread Pause\n");
HRESULT hr = IAudioClient_Stop(state->pAudioClient);
if (FAILED(hr))
MP_ERR(state, "IAudioClient_Stop returned: %s\n", mp_HRESULT_to_str(hr));
}
static void thread_unpause(struct ao *ao)
{
struct wasapi_state *state = ao->priv;
MP_DBG(state, "Thread Unpause\n");
HRESULT hr = IAudioClient_Start(state->pAudioClient);
if (FAILED(hr)) {
MP_ERR(state, "IAudioClient_Start returned %s\n",
mp_HRESULT_to_str(hr));
}
}
static void thread_reset(struct ao *ao)
{
struct wasapi_state *state = ao->priv;
HRESULT hr;
MP_DBG(state, "Thread Reset\n");
thread_pause(ao);
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);
thread_unpause(ao);
}
static void thread_wakeup(void *ptr)
{
struct ao *ao = ptr;
struct wasapi_state *state = ao->priv;
SetEvent(state->hUserWake);
}
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);
thread_wakeup(ao);
}
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) {
HANDLE handles[] = {state->hWake, state->hUserWake};
switch (WaitForMultipleObjects(MP_ARRAY_SIZE(handles), handles, FALSE, INFINITE)) {
case WAIT_OBJECT_0:
// 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");
continue;
case WAIT_OBJECT_0 + 1:
break;
default:
MP_ERR(ao, "Unexpected return value from WaitForMultipleObjects\n");
break;
}
mp_dispatch_queue_process(state->dispatch, 0);
int thread_state = atomic_load(&state->thread_state);
switch (thread_state) {
case WASAPI_THREAD_FEED:
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;
case WASAPI_THREAD_PAUSE:
thread_pause(ao);
break;
case WASAPI_THREAD_UNPAUSE:
thread_unpause(ao);
break;
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 && state->hUserWake)
set_thread_state(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->hUserWake, CloseHandle(state->hUserWake));
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);
state->hUserWake = CreateEventW(NULL, FALSE, FALSE, NULL);
if (!state->hInitDone || !state->hWake || !state->hUserWake) {
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_wakeup, 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_thread_state(ao, WASAPI_THREAD_RESET);
}
static void audio_resume(struct ao *ao)
{
set_thread_state(ao, WASAPI_THREAD_RESUME);
}
static bool audio_set_pause(struct ao *ao, bool paused)
{
set_thread_state(ao, paused ? WASAPI_THREAD_PAUSE : WASAPI_THREAD_UNPAUSE);
return true;
}
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,
.set_pause = audio_set_pause,
.list_devs = wasapi_list_devs,
.hotplug_init = hotplug_init,
.hotplug_uninit = hotplug_uninit,
.priv_size = sizeof(wasapi_state),
.options_prefix = "wasapi",
.options = (const struct m_option[]) {
{"exclusive-buffer", OPT_CHOICE(opt_exclusive_buffer,
{"default", 0}, {"min", -1}), M_RANGE(1, 2000000)},
{0}
},
};