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mirror of https://github.com/mpv-player/mpv.git synced 2024-09-20 12:02:23 +02:00
mpv/audio/out/ao_wasapi.c
Kevin Mitchell 84a3c21beb ao_wasapi: replace laggy COM messaging with mp_dispatch_queue
A COM message loop is apparently totally inappropriate for a low latency
thread. It leads to audio glitches because the thread doesn't wake up fast
enough when it should. It also causes mysterious correlations between the vo
and ao thread (i.e., toggling fullscreen delays audio feed events). Instead use
an mp_dispatch_queue to set/get volume/mute/session display name from the audio
thread. This has the added benefit of obviating the need to marshal the
associated interfaces from the audio thread.
2016-02-26 15:43:51 -08:00

519 lines
16 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/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_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);
}
if (sample_count > 0 && *delay_us <= 0) {
MP_WARN(state, "Under-run: Device delay: %g us\n", *delay_us);
} else {
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 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_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 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_resume(struct ao *ao)
{
struct wasapi_state *state = ao->priv;
HRESULT hr;
MP_DBG(state, "Thread Resume\n");
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 void thread_wakeup(void *ptr)
{
struct ao *ao = ptr;
struct wasapi_state *state = ao->priv;
SetEvent(state->hWake);
}
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;
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) {
if (WaitForSingleObject(state->hWake, INFINITE) != WAIT_OBJECT_0)
MP_ERR(ao, "Unexpected return value from WaitForSingleObject\n");
mp_dispatch_queue_process(state->dispatch, 0);
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_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: %d\n", thread_state);
}
}
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_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->dispatch = mp_dispatch_create(state);
mp_dispatch_set_wakeup_fn(state->dispatch, thread_wakeup, ao);
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;
}
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;
switch (cmd) {
case AOCONTROL_GET_VOLUME:
case AOCONTROL_SET_VOLUME:
if (!state->pEndpointVolume ||
!(state->vol_hw_support & ENDPOINT_HARDWARE_SUPPORT_VOLUME)) {
return CONTROL_FALSE;
}
float volume;
switch (cmd) {
case AOCONTROL_GET_VOLUME:
IAudioEndpointVolume_GetMasterVolumeLevelScalar(
state->pEndpointVolume, &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->pEndpointVolume, volume, NULL);
return CONTROL_OK;
}
case AOCONTROL_GET_MUTE:
case AOCONTROL_SET_MUTE:
if (!state->pEndpointVolume ||
!(state->vol_hw_support & ENDPOINT_HARDWARE_SUPPORT_MUTE)) {
return CONTROL_FALSE;
}
BOOL mute;
switch (cmd) {
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;
}
case AOCONTROL_HAS_PER_APP_VOLUME:
return CONTROL_FALSE;
default:
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);
*(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->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;
case AOCONTROL_HAS_PER_APP_VOLUME:
return CONTROL_TRUE;
default:
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, (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->pSessionControl, title, NULL);
SAFE_RELEASE(tmp, CoTaskMemFree(tmp));
IAudioSessionControl_GetDisplayName(state->pSessionControl, &tmp);
} while (lstrcmpW(title, tmp));
SAFE_RELEASE(tmp, CoTaskMemFree(tmp));
talloc_free(title);
return CONTROL_OK;
}
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 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},
},
};