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mpv/audio/out/ao_wasapi.c
wm4 8a9b64329c Relicense some non-MPlayer source files to LGPL 2.1 or later
This covers source files which were added in mplayer2 and mpv times
only, and where all code is covered by LGPL relicensing agreements.

There are probably more files to which this applies, but I'm being
conservative here.

A file named ao_sdl.c exists in MPlayer too, but the mpv one is a
complete rewrite, and was added some time after the original ao_sdl.c
was removed. The same applies to vo_sdl.c, for which the SDL2 API is
radically different in addition (MPlayer supports SDL 1.2 only).

common.c contains only code written by me. But common.h is a strange
case: although it originally was named mp_common.h and exists in MPlayer
too, by now it contains only definitions written by uau and me. The
exceptions are the CONTROL_ defines - thus not changing the license of
common.h yet.

codec_tags.c contained once large tables generated from MPlayer's
codecs.conf, but all of these tables were removed.

From demux_playlist.c I'm removing a code fragment from someone who was
not asked; this probably could be done later (see commit 15dccc37).

misc.c is a bit complicated to reason about (it was split off mplayer.c
and thus contains random functions out of this file), but actually all
functions have been added post-MPlayer. Except get_relative_time(),
which was written by uau, but looks similar to 3 different versions of
something similar in each of the Unix/win32/OSX timer source files. I'm
not sure what that means in regards to copyright, so I've just moved it
into another still-GPL source file for now.

screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but
they're all gone.
2016-01-19 18:36:06 +01:00

501 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 "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);
// 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");
hr = S_OK;
}
EXIT_ON_ERROR(hr);
// 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 yeilds 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 (hr != S_OK) {
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 (hr != S_OK) {
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);
// we may get S_FALSE if the stream is already stopped
if (hr != S_OK && hr != S_FALSE)
MP_ERR(state, "IAudioClient_Stop returned: %s\n", mp_HRESULT_to_str(hr));
// we may get S_FALSE if the stream is already reset
hr = IAudioClient_Reset(state->pAudioClient);
if (hr != S_OK && hr != S_FALSE)
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 (state->init_ret != S_OK)
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 = 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 (state->init_ret != S_OK) {
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},
},
};