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gstreamer/sys/wasapi2/gstwasapi2client.cpp
Seungha Yang 2e1bec7ff6 wasapi2: Fallback to IAudioClient interface if IAudioClient3 API is unavailable 
When default device is selected, IAudioClient3 API doesn't look like
available.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1332>
2020-06-11 12:40:44 +00:00

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/*
* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* Copyright (C) 2013 Collabora Ltd.
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
* Copyright (C) 2018 Centricular Ltd.
* Author: Nirbheek Chauhan <nirbheek@centricular.com>
* Copyright (C) 2020 Seungha Yang <seungha@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "AsyncOperations.h"
#include "gstwasapi2client.h"
#include "gstwasapi2util.h"
#include <initguid.h>
#include <windows.foundation.h>
#include <windows.ui.core.h>
#include <wrl.h>
#include <wrl/wrappers/corewrappers.h>
#include <audioclient.h>
#include <mmdeviceapi.h>
#include <string.h>
#include <string>
#include <locale>
#include <codecvt>
using namespace ABI::Windows::ApplicationModel::Core;
using namespace ABI::Windows::Foundation;
using namespace ABI::Windows::Foundation::Collections;
using namespace ABI::Windows::UI::Core;
using namespace ABI::Windows::Media::Devices;
using namespace ABI::Windows::Devices::Enumeration;
using namespace Microsoft::WRL;
using namespace Microsoft::WRL::Wrappers;
G_BEGIN_DECLS
GST_DEBUG_CATEGORY_EXTERN (gst_wasapi2_client_debug);
#define GST_CAT_DEFAULT gst_wasapi2_client_debug
G_END_DECLS
static void
gst_wasapi2_client_on_device_activated (GstWasapi2Client * client,
IAudioClient3 * audio_client);
class GstWasapiDeviceActivator
: public RuntimeClass<RuntimeClassFlags<ClassicCom>, FtmBase,
IActivateAudioInterfaceCompletionHandler>
{
public:
GstWasapiDeviceActivator ()
: listener_(nullptr)
{
}
~GstWasapiDeviceActivator ()
{
}
HRESULT
RuntimeClassInitialize (GstWasapi2Client * listener)
{
if (!listener)
return E_INVALIDARG;
listener_ = listener;
findCoreDispatcher ();
return S_OK;
}
STDMETHOD(ActivateCompleted)
(IActivateAudioInterfaceAsyncOperation *async_op)
{
ComPtr<IAudioClient3> audio_client;
HRESULT hr = S_OK;
HRESULT hr_async_op = S_OK;
ComPtr<IUnknown> audio_interface;
if (!listener_) {
GST_WARNING ("No listener was configured");
return S_OK;
}
GST_INFO_OBJECT (listener_, "AsyncOperation done");
hr = async_op->GetActivateResult(&hr_async_op, &audio_interface);
if (!gst_wasapi2_result (hr)) {
GST_WARNING_OBJECT (listener_, "Failed to get activate result, hr: 0x%x", hr);
goto done;
}
if (!gst_wasapi2_result (hr_async_op)) {
GST_WARNING_OBJECT (listener_, "Failed to activate device");
goto done;
}
hr = audio_interface.As (&audio_client);
if (!gst_wasapi2_result (hr)) {
GST_ERROR_OBJECT (listener_, "Failed to get IAudioClient3 interface");
goto done;
}
done:
/* Should call this method anyway, listener will wait this event */
gst_wasapi2_client_on_device_activated (listener_, audio_client.Get());
/* return S_OK anyway, but listener can know it's succeeded or not
* by passed IAudioClient handle via gst_wasapi2_client_on_device_activated
*/
return S_OK;
}
HRESULT
ActivateDeviceAsync(const std::wstring &device_id)
{
return runOnUIThread (INFINITE,
[this, device_id] {
ComPtr<IActivateAudioInterfaceAsyncOperation> async_op;
HRESULT hr = S_OK;
hr = ActivateAudioInterfaceAsync (device_id.c_str (),
__uuidof(IAudioClient3), nullptr, this, &async_op);
/* for debugging */
gst_wasapi2_result (hr);
return hr;
});
}
/* Try to find ICoreDispatcher of main UI so that active audio
* interface on main UI thread */
void findCoreDispatcher(void)
{
HRESULT hr;
HStringReference hstr_core_app =
HStringReference(RuntimeClass_Windows_ApplicationModel_Core_CoreApplication);
ComPtr<ICoreApplication> core_app;
ComPtr<ICoreApplicationView> core_app_view;
ComPtr<ICoreWindow> core_window;
hr = GetActivationFactory (hstr_core_app.Get(), &core_app);
if (!gst_wasapi2_result (hr))
return;
hr = core_app->GetCurrentView (&core_app_view);
if (!gst_wasapi2_result (hr))
return;
hr = core_app_view->get_CoreWindow (&core_window);
if (!gst_wasapi2_result (hr))
return;
hr = core_window->get_Dispatcher (&dispatcher_);
if (!gst_wasapi2_result (hr))
return;
GST_DEBUG ("Main UI dispatcher is available");
}
template <typename CB>
HRESULT
runOnUIThread (DWORD timeout, CB && cb)
{
ComPtr<IAsyncAction> async_action;
HRESULT hr;
HRESULT hr_cb;
boolean can_now;
DWORD wait_ret;
if (!dispatcher_)
return cb();
hr = dispatcher_->get_HasThreadAccess (&can_now);
if (FAILED (hr))
return hr;
if (can_now)
return cb ();
Event event (CreateEventEx (NULL, NULL, CREATE_EVENT_MANUAL_RESET,
EVENT_ALL_ACCESS));
if (!event.IsValid())
return E_FAIL;
auto handler =
Callback<Implements<RuntimeClassFlags<ClassicCom>,
IDispatchedHandler, FtmBase>>([&hr_cb, &cb, &event] {
hr_cb = cb ();
SetEvent (event.Get());
return S_OK;
});
hr = dispatcher_->RunAsync (CoreDispatcherPriority_Normal,
handler.Get(), &async_action);
if (FAILED (hr))
return hr;
wait_ret = WaitForSingleObject (event.Get(), timeout);
if (wait_ret != WAIT_OBJECT_0)
return E_FAIL;
return hr;
}
private:
GstWasapi2Client * listener_;
ComPtr<ICoreDispatcher> dispatcher_;
};
struct _GstWasapi2Client
{
GstObject parent;
GstWasapi2ClientDeviceClass device_class;
gboolean low_latency;
gchar *device_id;
gchar *device_name;
gint device_index;
IAudioClient3 *audio_client;
IAudioCaptureClient *audio_capture_client;
IAudioRenderClient *audio_render_client;
ISimpleAudioVolume *audio_volume;
WAVEFORMATEX *mix_format;
GstCaps *supported_caps;
HANDLE event_handle;
HANDLE cancellable;
gboolean opened;
gboolean running;
guint32 device_period;
guint32 buffer_frame_count;
GstAudioChannelPosition *positions;
/* Used for capture mode */
GstAdapter *adapter;
GThread *thread;
GMutex lock;
GCond cond;
GMainContext *context;
GMainLoop *loop;
/* To wait ActivateCompleted event */
GMutex init_lock;
GCond init_cond;
gboolean init_done;
};
enum
{
PROP_0,
PROP_DEVICE,
PROP_DEVICE_NAME,
PROP_DEVICE_INDEX,
PROP_DEVICE_CLASS,
PROP_LOW_LATENCY,
};
#define DEFAULT_DEVICE_INDEX -1
#define DEFAULT_DEVICE_CLASS GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE
#define DEFAULT_LOW_LATENCY FALSE
GType
gst_wasapi2_client_device_class_get_type (void)
{
static volatile GType class_type = 0;
static const GEnumValue types[] = {
{GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE, "Capture", "capture"},
{GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER, "Render", "render"},
{0, NULL, NULL}
};
if (g_once_init_enter (&class_type)) {
GType gtype = g_enum_register_static ("GstWasapi2ClientDeviceClass", types);
g_once_init_leave (&class_type, gtype);
}
return class_type;
}
static void gst_wasapi2_client_constructed (GObject * object);
static void gst_wasapi2_client_dispose (GObject * object);
static void gst_wasapi2_client_finalize (GObject * object);
static void gst_wasapi2_client_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_wasapi2_client_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static gpointer gst_wasapi2_client_thread_func (GstWasapi2Client * self);
static gboolean
gst_wasapi2_client_main_loop_running_cb (GstWasapi2Client * self);
#define gst_wasapi2_client_parent_class parent_class
G_DEFINE_TYPE (GstWasapi2Client,
gst_wasapi2_client, GST_TYPE_OBJECT);
static void
gst_wasapi2_client_class_init (GstWasapi2ClientClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GParamFlags param_flags =
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY |
G_PARAM_STATIC_STRINGS);
gobject_class->constructed = gst_wasapi2_client_constructed;
gobject_class->dispose = gst_wasapi2_client_dispose;
gobject_class->finalize = gst_wasapi2_client_finalize;
gobject_class->get_property = gst_wasapi2_client_get_property;
gobject_class->set_property = gst_wasapi2_client_set_property;
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device",
"WASAPI playback device as a GUID string", NULL, param_flags));
g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
g_param_spec_string ("device-name", "Device Name",
"The human-readable device name", NULL, param_flags));
g_object_class_install_property (gobject_class, PROP_DEVICE_INDEX,
g_param_spec_int ("device-index", "Device Index",
"The zero-based device index", -1, G_MAXINT, DEFAULT_DEVICE_INDEX,
param_flags));
g_object_class_install_property (gobject_class, PROP_DEVICE_CLASS,
g_param_spec_enum ("device-class", "Device Class",
"Device class", GST_TYPE_WASAPI2_CLIENT_DEVICE_CLASS,
DEFAULT_DEVICE_CLASS, param_flags));
g_object_class_install_property (gobject_class, PROP_LOW_LATENCY,
g_param_spec_boolean ("low-latency", "Low latency",
"Optimize all settings for lowest latency. Always safe to enable.",
DEFAULT_LOW_LATENCY, param_flags));
}
static void
gst_wasapi2_client_init (GstWasapi2Client * self)
{
self->device_index = DEFAULT_DEVICE_INDEX;
self->device_class = DEFAULT_DEVICE_CLASS;
self->low_latency = DEFAULT_LOW_LATENCY;
self->adapter = gst_adapter_new ();
self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
self->cancellable = CreateEvent (NULL, TRUE, FALSE, NULL);
g_mutex_init (&self->lock);
g_cond_init (&self->cond);
g_mutex_init (&self->init_lock);
g_cond_init (&self->init_cond);
self->context = g_main_context_new ();
self->loop = g_main_loop_new (self->context, FALSE);
}
static void
gst_wasapi2_client_constructed (GObject * object)
{
GstWasapi2Client *self = GST_WASAPI2_CLIENT (object);
/* Create a new thread to ensure that COM thread can be MTA thread.
* We cannot ensure whether CoInitializeEx() was called outside of here for
* this thread or not. If it was called with non-COINIT_MULTITHREADED option,
* we cannot update it */
g_mutex_lock (&self->lock);
self->thread = g_thread_new ("GstWasapi2ClientWinRT",
(GThreadFunc) gst_wasapi2_client_thread_func, self);
while (!self->loop || !g_main_loop_is_running (self->loop))
g_cond_wait (&self->cond, &self->lock);
g_mutex_unlock (&self->lock);
G_OBJECT_CLASS (parent_class)->constructed (object);
}
static void
gst_wasapi2_client_dispose (GObject * object)
{
GstWasapi2Client *self = GST_WASAPI2_CLIENT (object);
GST_DEBUG_OBJECT (self, "dispose");
gst_clear_caps (&self->supported_caps);
if (self->loop) {
g_main_loop_quit (self->loop);
g_thread_join (self->thread);
g_main_context_unref (self->context);
g_main_loop_unref (self->loop);
self->thread = NULL;
self->context = NULL;
self->loop = NULL;
}
g_clear_object (&self->adapter);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_wasapi2_client_finalize (GObject * object)
{
GstWasapi2Client *self = GST_WASAPI2_CLIENT (object);
g_free (self->device_id);
g_free (self->device_name);
g_free (self->positions);
CoTaskMemFree (self->mix_format);
CloseHandle (self->event_handle);
CloseHandle (self->cancellable);
g_mutex_clear (&self->lock);
g_cond_clear (&self->cond);
g_mutex_clear (&self->init_lock);
g_cond_clear (&self->init_cond);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_wasapi2_client_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWasapi2Client *self = GST_WASAPI2_CLIENT (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, self->device_id);
break;
case PROP_DEVICE_NAME:
g_value_set_string (value, self->device_name);
break;
case PROP_DEVICE_INDEX:
g_value_set_int (value, self->device_index);
break;
case PROP_DEVICE_CLASS:
g_value_set_enum (value, self->device_class);
break;
case PROP_LOW_LATENCY:
g_value_set_boolean (value, self->low_latency);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_wasapi2_client_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWasapi2Client *self = GST_WASAPI2_CLIENT (object);
switch (prop_id) {
case PROP_DEVICE:
g_free (self->device_id);
self->device_id = g_value_dup_string (value);
break;
case PROP_DEVICE_NAME:
g_free (self->device_name);
self->device_name = g_value_dup_string (value);
break;
case PROP_DEVICE_INDEX:
self->device_index = g_value_get_int (value);
break;
case PROP_DEVICE_CLASS:
self->device_class =
(GstWasapi2ClientDeviceClass) g_value_get_enum (value);
break;
case PROP_LOW_LATENCY:
self->low_latency = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_wasapi2_client_main_loop_running_cb (GstWasapi2Client * self)
{
GST_DEBUG_OBJECT (self, "Main loop running now");
g_mutex_lock (&self->lock);
g_cond_signal (&self->cond);
g_mutex_unlock (&self->lock);
return G_SOURCE_REMOVE;
}
static void
gst_wasapi2_client_on_device_activated (GstWasapi2Client * self,
IAudioClient3 * audio_client)
{
GST_INFO_OBJECT (self, "Device activated");
g_mutex_lock (&self->init_lock);
if (audio_client) {
audio_client->AddRef();
self->audio_client = audio_client;
}
self->init_done = TRUE;
g_cond_broadcast (&self->init_cond);
g_mutex_unlock (&self->init_lock);
}
static std::string
convert_wstring_to_string (const std::wstring &wstr)
{
std::wstring_convert<std::codecvt_utf8<wchar_t>, wchar_t> converter;
return converter.to_bytes (wstr.c_str());
}
static std::string
convert_hstring_to_string (HString * hstr)
{
const wchar_t *raw_hstr;
if (!hstr)
return std::string();
raw_hstr = hstr->GetRawBuffer (nullptr);
if (!raw_hstr)
return std::string();
return convert_wstring_to_string (std::wstring (raw_hstr));
}
static std::wstring
gst_wasapi2_client_get_default_device_id (GstWasapi2Client * self)
{
HRESULT hr;
PWSTR default_device_id_wstr = nullptr;
if (self->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE)
hr = StringFromIID (DEVINTERFACE_AUDIO_CAPTURE, &default_device_id_wstr);
else
hr = StringFromIID (DEVINTERFACE_AUDIO_RENDER, &default_device_id_wstr);
if (!gst_wasapi2_result (hr))
return std::wstring();
std::wstring ret = std::wstring (default_device_id_wstr);
CoTaskMemFree (default_device_id_wstr);
return ret;
}
static void
gst_wasapi2_client_thread_func_internal (GstWasapi2Client * self)
{
HRESULT hr;
GSource *source;
ComPtr<GstWasapiDeviceActivator> activator;
ComPtr<IDeviceInformationStatics> device_info_static;
ComPtr<IAsyncOperation<DeviceInformationCollection*>> async_op;
ComPtr<IVectorView<DeviceInformation*>> device_list;
HStringReference hstr_device_info =
HStringReference(RuntimeClass_Windows_Devices_Enumeration_DeviceInformation);
DeviceClass device_class;
unsigned int count = 0;
gint device_index = 0;
std::wstring default_device_id_wstring;
std::string default_device_id;
std::wstring target_device_id_wstring;
std::string target_device_id;
std::string target_device_name;
gboolean use_default_device = FALSE;
g_main_context_push_thread_default (self->context);
GST_INFO_OBJECT (self,
"requested device info, device-class: %s, device: %s, device-index: %d",
self->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE ? "capture" :
"render", GST_STR_NULL (self->device_id), self->device_index);
hr = MakeAndInitialize<GstWasapiDeviceActivator> (&activator, self);
if (!gst_wasapi2_result (hr))
goto run_loop;
if (self->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE) {
device_class = DeviceClass::DeviceClass_AudioCapture;
} else {
device_class = DeviceClass::DeviceClass_AudioRender;
}
default_device_id_wstring = gst_wasapi2_client_get_default_device_id (self);
if (default_device_id_wstring.empty ()) {
GST_WARNING_OBJECT (self, "Couldn't get default device id");
goto run_loop;
}
default_device_id = convert_wstring_to_string (default_device_id_wstring);
GST_DEBUG_OBJECT (self, "Default device id: %s", default_device_id.c_str ());
/* When
* 1) default device was requested or
* 2) no explicitly requested device or
* 3) requested device string id is null but device index is zero
* will use default device
*
* Note that default device is much preferred
* See https://docs.microsoft.com/en-us/windows/win32/coreaudio/automatic-stream-routing
*/
if (self->device_id &&
g_ascii_strcasecmp (self->device_id, default_device_id.c_str()) == 0) {
GST_DEBUG_OBJECT (self, "Default device was requested");
use_default_device = TRUE;
} else if (self->device_index < 0 && !self->device_id) {
GST_DEBUG_OBJECT (self,
"No device was explicitly requested, use default device");
use_default_device = TRUE;
} else if (!self->device_id && self->device_index == 0) {
GST_DEBUG_OBJECT (self, "device-index == zero means default device");
use_default_device = TRUE;
}
if (use_default_device) {
target_device_id_wstring = default_device_id_wstring;
target_device_id = default_device_id;
if (self->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE)
target_device_name = "Default Audio Capture Device";
else
target_device_name = "Default Audio Render Device";
goto activate;
}
hr = GetActivationFactory (hstr_device_info.Get(), &device_info_static);
if (!gst_wasapi2_result (hr))
goto run_loop;
hr = device_info_static->FindAllAsyncDeviceClass (device_class, &async_op);
device_info_static.Reset ();
if (!gst_wasapi2_result (hr))
goto run_loop;
hr = SyncWait<DeviceInformationCollection*>(async_op.Get ());
if (!gst_wasapi2_result (hr))
goto run_loop;
hr = async_op->GetResults (&device_list);
async_op.Reset ();
if (!gst_wasapi2_result (hr))
goto run_loop;
hr = device_list->get_Size (&count);
if (!gst_wasapi2_result (hr))
goto run_loop;
if (count == 0) {
GST_WARNING_OBJECT (self, "No available device");
goto run_loop;
}
/* device_index 0 will be assigned for default device
* so the number of available device is count + 1 (for default device) */
if (self->device_index >= 0 && self->device_index > (gint) count) {
GST_WARNING_OBJECT (self, "Device index %d is unavailable");
goto run_loop;
}
GST_DEBUG_OBJECT (self, "Available device count: %d", count);
/* zero is for default device */
device_index = 1;
for (unsigned int i = 0; i < count; i++) {
ComPtr<IDeviceInformation> device_info;
HString id;
HString name;
boolean b_value;
std::string cur_device_id;
std::string cur_device_name;
hr = device_list->GetAt (i, &device_info);
if (!gst_wasapi2_result (hr))
continue;
hr = device_info->get_IsEnabled (&b_value);
if (!gst_wasapi2_result (hr))
continue;
/* select only enabled device */
if (!b_value) {
GST_DEBUG_OBJECT (self, "Device index %d is disabled", i);
continue;
}
/* To ensure device id and device name are available,
* will query this later again once target device is determined */
hr = device_info->get_Id (id.GetAddressOf());
if (!gst_wasapi2_result (hr))
continue;
if (!id.IsValid()) {
GST_WARNING_OBJECT (self, "Device index %d has invalid id", i);
continue;
}
hr = device_info->get_Name (name.GetAddressOf());
if (!gst_wasapi2_result (hr))
continue;
if (!name.IsValid ()) {
GST_WARNING_OBJECT (self, "Device index %d has invalid name", i);
continue;
}
cur_device_id = convert_hstring_to_string (&id);
if (cur_device_id.empty ()) {
GST_WARNING_OBJECT (self, "Device index %d has empty id", i);
continue;
}
cur_device_name = convert_hstring_to_string (&name);
if (cur_device_name.empty ()) {
GST_WARNING_OBJECT (self, "Device index %d has empty device name", i);
continue;
}
GST_DEBUG_OBJECT (self, "device [%d] id: %s, name: %s",
device_index, cur_device_id.c_str(), cur_device_name.c_str());
if (self->device_id &&
g_ascii_strcasecmp (self->device_id, cur_device_id.c_str ()) == 0) {
GST_INFO_OBJECT (self,
"Device index %d has matching device id %s", device_index,
cur_device_id.c_str ());
target_device_id_wstring = id.GetRawBuffer (nullptr);
target_device_id = cur_device_id;
target_device_name = cur_device_name;
break;
}
if (self->device_index >= 0 && self->device_index == device_index) {
GST_INFO_OBJECT (self, "Select device index %d, device id %s",
device_index, cur_device_id.c_str ());
target_device_id_wstring = id.GetRawBuffer (nullptr);
target_device_id = cur_device_id;
target_device_name = cur_device_name;
break;
}
/* count only available devices */
device_index++;
}
if (target_device_id_wstring.empty ()) {
GST_WARNING_OBJECT (self, "Couldn't find target device");
goto run_loop;
}
activate:
/* fill device id and name */
g_free (self->device_id);
self->device_id = g_strdup (target_device_id.c_str());
g_free (self->device_name);
self->device_name = g_strdup (target_device_name.c_str ());
self->device_index = device_index;
hr = activator->ActivateDeviceAsync (target_device_id_wstring);
if (!gst_wasapi2_result (hr)) {
GST_WARNING_OBJECT (self, "Failed to activate device");
goto run_loop;
}
/* Wait ActivateCompleted event */
GST_DEBUG_OBJECT (self, "Wait device activation");
g_mutex_lock (&self->init_lock);
while (!self->init_done)
g_cond_wait (&self->init_cond, &self->init_lock);
g_mutex_unlock (&self->init_lock);
GST_DEBUG_OBJECT (self, "Done device activation");
run_loop:
source = g_idle_source_new ();
g_source_set_callback (source,
(GSourceFunc) gst_wasapi2_client_main_loop_running_cb, self, NULL);
g_source_attach (source, self->context);
g_source_unref (source);
GST_DEBUG_OBJECT (self, "Starting main loop");
g_main_loop_run (self->loop);
GST_DEBUG_OBJECT (self, "Stopped main loop");
g_main_context_pop_thread_default (self->context);
gst_wasapi2_client_stop (self);
if (self->audio_volume) {
self->audio_volume->Release ();
self->audio_volume = NULL;
}
if (self->audio_render_client) {
self->audio_render_client->Release ();
self->audio_render_client = NULL;
}
if (self->audio_capture_client) {
self->audio_capture_client->Release ();
self->audio_capture_client = NULL;
}
if (self->audio_client) {
self->audio_client->Release ();
self->audio_client = NULL;
}
GST_DEBUG_OBJECT (self, "Exit thread function");
return;
}
static gpointer
gst_wasapi2_client_thread_func (GstWasapi2Client * self)
{
RoInitializeWrapper initialize (RO_INIT_MULTITHREADED);
/* Wrap thread function so that ensure everything happens inside of
* RoInitializeWrapper */
gst_wasapi2_client_thread_func_internal (self);
return NULL;
}
static const gchar *
gst_waveformatex_to_audio_format (WAVEFORMATEXTENSIBLE * format)
{
const gchar *fmt_str = NULL;
GstAudioFormat fmt = GST_AUDIO_FORMAT_UNKNOWN;
if (format->Format.wFormatTag == WAVE_FORMAT_PCM) {
fmt = gst_audio_format_build_integer (TRUE, G_LITTLE_ENDIAN,
format->Format.wBitsPerSample, format->Format.wBitsPerSample);
} else if (format->Format.wFormatTag == WAVE_FORMAT_IEEE_FLOAT) {
if (format->Format.wBitsPerSample == 32)
fmt = GST_AUDIO_FORMAT_F32LE;
else if (format->Format.wBitsPerSample == 64)
fmt = GST_AUDIO_FORMAT_F64LE;
} else if (format->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
if (IsEqualGUID (format->SubFormat, KSDATAFORMAT_SUBTYPE_PCM)) {
fmt = gst_audio_format_build_integer (TRUE, G_LITTLE_ENDIAN,
format->Format.wBitsPerSample, format->Samples.wValidBitsPerSample);
} else if (IsEqualGUID (format->SubFormat,
KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)) {
if (format->Format.wBitsPerSample == 32
&& format->Samples.wValidBitsPerSample == 32)
fmt = GST_AUDIO_FORMAT_F32LE;
else if (format->Format.wBitsPerSample == 64 &&
format->Samples.wValidBitsPerSample == 64)
fmt = GST_AUDIO_FORMAT_F64LE;
}
}
if (fmt != GST_AUDIO_FORMAT_UNKNOWN)
fmt_str = gst_audio_format_to_string (fmt);
return fmt_str;
}
static void
gst_wasapi_util_channel_position_all_none (guint channels,
GstAudioChannelPosition * position)
{
int ii;
for (ii = 0; ii < channels; ii++)
position[ii] = GST_AUDIO_CHANNEL_POSITION_NONE;
}
static struct
{
guint64 wasapi_pos;
GstAudioChannelPosition gst_pos;
} wasapi_to_gst_pos[] = {
{SPEAKER_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT},
{SPEAKER_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
{SPEAKER_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER},
{SPEAKER_LOW_FREQUENCY, GST_AUDIO_CHANNEL_POSITION_LFE1},
{SPEAKER_BACK_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT},
{SPEAKER_BACK_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
{SPEAKER_FRONT_LEFT_OF_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER},
{SPEAKER_FRONT_RIGHT_OF_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER},
{SPEAKER_BACK_CENTER, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
/* Enum values diverge from this point onwards */
{SPEAKER_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT},
{SPEAKER_SIDE_RIGHT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT},
{SPEAKER_TOP_CENTER, GST_AUDIO_CHANNEL_POSITION_TOP_CENTER},
{SPEAKER_TOP_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT},
{SPEAKER_TOP_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER},
{SPEAKER_TOP_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT},
{SPEAKER_TOP_BACK_LEFT, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT},
{SPEAKER_TOP_BACK_CENTER, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER},
{SPEAKER_TOP_BACK_RIGHT, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT}
};
/* Parse WAVEFORMATEX to get the gstreamer channel mask, and the wasapi channel
* positions so GstAudioRingbuffer can reorder the audio data to match the
* gstreamer channel order. */
static guint64
gst_wasapi_util_waveformatex_to_channel_mask (WAVEFORMATEXTENSIBLE * format,
GstAudioChannelPosition ** out_position)
{
int ii, ch;
guint64 mask = 0;
WORD nChannels = format->Format.nChannels;
DWORD dwChannelMask = format->dwChannelMask;
GstAudioChannelPosition *pos = NULL;
pos = g_new (GstAudioChannelPosition, nChannels);
gst_wasapi_util_channel_position_all_none (nChannels, pos);
/* Too many channels, have to assume that they are all non-positional */
if (nChannels > G_N_ELEMENTS (wasapi_to_gst_pos)) {
GST_INFO ("Got too many (%i) channels, assuming non-positional", nChannels);
goto out;
}
/* Too many bits in the channel mask, and the bits don't match nChannels */
if (dwChannelMask >> (G_N_ELEMENTS (wasapi_to_gst_pos) + 1) != 0) {
GST_WARNING ("Too many bits in channel mask (%lu), assuming "
"non-positional", dwChannelMask);
goto out;
}
/* Map WASAPI's channel mask to Gstreamer's channel mask and positions.
* If the no. of bits in the mask > nChannels, we will ignore the extra. */
for (ii = 0, ch = 0; ii < G_N_ELEMENTS (wasapi_to_gst_pos) && ch < nChannels;
ii++) {
if (!(dwChannelMask & wasapi_to_gst_pos[ii].wasapi_pos))
/* no match, try next */
continue;
mask |= G_GUINT64_CONSTANT (1) << wasapi_to_gst_pos[ii].gst_pos;
pos[ch++] = wasapi_to_gst_pos[ii].gst_pos;
}
/* XXX: Warn if some channel masks couldn't be mapped? */
GST_DEBUG ("Converted WASAPI mask 0x%" G_GINT64_MODIFIER "x -> 0x%"
G_GINT64_MODIFIER "x", (guint64) dwChannelMask, (guint64) mask);
out:
if (out_position)
*out_position = pos;
return mask;
}
static gboolean
gst_wasapi2_util_parse_waveformatex (WAVEFORMATEXTENSIBLE * format,
GstCaps * template_caps, GstCaps ** out_caps,
GstAudioChannelPosition ** out_positions)
{
int ii;
const gchar *afmt;
guint64 channel_mask;
*out_caps = NULL;
/* TODO: handle SPDIF and other encoded formats */
/* 1 or 2 channels <= 16 bits sample size OR
* 1 or 2 channels > 16 bits sample size or >2 channels */
if (format->Format.wFormatTag != WAVE_FORMAT_PCM &&
format->Format.wFormatTag != WAVE_FORMAT_IEEE_FLOAT &&
format->Format.wFormatTag != WAVE_FORMAT_EXTENSIBLE)
/* Unhandled format tag */
return FALSE;
/* WASAPI can only tell us one canonical mix format that it will accept. The
* alternative is calling IsFormatSupported on all combinations of formats.
* Instead, it's simpler and faster to require conversion inside gstreamer */
afmt = gst_waveformatex_to_audio_format (format);
if (afmt == NULL)
return FALSE;
*out_caps = gst_caps_copy (template_caps);
/* This will always return something that might be usable */
channel_mask =
gst_wasapi_util_waveformatex_to_channel_mask (format, out_positions);
for (ii = 0; ii < gst_caps_get_size (*out_caps); ii++) {
GstStructure *s = gst_caps_get_structure (*out_caps, ii);
gst_structure_set (s,
"format", G_TYPE_STRING, afmt,
"channels", G_TYPE_INT, format->Format.nChannels,
"rate", G_TYPE_INT, format->Format.nSamplesPerSec, NULL);
if (channel_mask) {
gst_structure_set (s,
"channel-mask", GST_TYPE_BITMASK, channel_mask, NULL);
}
}
return TRUE;
}
GstCaps *
gst_wasapi2_client_get_caps (GstWasapi2Client * client)
{
WAVEFORMATEX *format = NULL;
static GstStaticCaps static_caps = GST_STATIC_CAPS (GST_WASAPI2_STATIC_CAPS);
GstCaps *scaps;
HRESULT hr;
g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), NULL);
if (client->supported_caps)
return gst_caps_ref (client->supported_caps);
if (!client->audio_client) {
GST_WARNING_OBJECT (client, "IAudioClient3 wasn't configured");
return NULL;
}
CoTaskMemFree (client->mix_format);
client->mix_format = nullptr;
g_clear_pointer (&client->positions, g_free);
hr = client->audio_client->GetMixFormat (&format);
if (!gst_wasapi2_result (hr))
return NULL;
scaps = gst_static_caps_get (&static_caps);
gst_wasapi2_util_parse_waveformatex ((WAVEFORMATEXTENSIBLE *) format,
scaps, &client->supported_caps, &client->positions);
gst_caps_unref (scaps);
client->mix_format = format;
if (!client->supported_caps) {
GST_ERROR_OBJECT (client, "No caps from subclass");
return NULL;
}
return gst_caps_ref (client->supported_caps);
}
static gboolean
gst_wasapi2_client_initialize_audio_client3 (GstWasapi2Client * self)
{
HRESULT hr;
UINT32 default_period, fundamental_period, min_period, max_period;
DWORD stream_flags = AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
WAVEFORMATEX *format = NULL;
UINT32 period;
gboolean ret = FALSE;
IAudioClient3 *audio_client = self->audio_client;
hr = audio_client->GetSharedModeEnginePeriod (self->mix_format,
&default_period, &fundamental_period, &min_period, &max_period);
if (!gst_wasapi2_result (hr))
goto done;
GST_INFO_OBJECT (self, "Using IAudioClient3, default period %d frames, "
"fundamental period %d frames, minimum period %d frames, maximum period "
"%d frames", default_period, fundamental_period, min_period, max_period);
hr = audio_client->InitializeSharedAudioStream (stream_flags, min_period,
self->mix_format, nullptr);
if (!gst_wasapi2_result (hr)) {
GST_WARNING_OBJECT (self, "Failed to initialize IAudioClient3");
goto done;
}
/* query period again to be ensured */
hr = audio_client->GetCurrentSharedModeEnginePeriod (&format, &period);
if (!gst_wasapi2_result (hr)) {
GST_WARNING_OBJECT (self, "Failed to get current period");
goto done;
}
self->device_period = period;
ret = TRUE;
done:
CoTaskMemFree (format);
return ret;
}
static void
gst_wasapi2_util_get_best_buffer_sizes (GstAudioRingBufferSpec * spec,
REFERENCE_TIME default_period, REFERENCE_TIME min_period,
REFERENCE_TIME * ret_period, REFERENCE_TIME * ret_buffer_duration)
{
REFERENCE_TIME use_period, use_buffer;
/* Shared mode always runs at the default period, so if we want a larger
* period (for lower CPU usage), we do it as a multiple of that */
use_period = default_period;
/* Ensure that the period (latency_time) used is an integral multiple of
* either the default period or the minimum period */
use_period = use_period * MAX ((spec->latency_time * 10) / use_period, 1);
/* Ask WASAPI to create a software ringbuffer of at least this size; it may
* be larger so the actual buffer time may be different, which is why after
* initialization we read the buffer duration actually in-use and set
* segsize/segtotal from that. */
use_buffer = spec->buffer_time * 10;
/* Has to be at least twice the period */
if (use_buffer < 2 * use_period)
use_buffer = 2 * use_period;
*ret_period = use_period;
*ret_buffer_duration = use_buffer;
}
static gboolean
gst_wasapi2_client_initialize_audio_client (GstWasapi2Client * self,
GstAudioRingBufferSpec * spec)
{
REFERENCE_TIME default_period, min_period;
REFERENCE_TIME device_period, device_buffer_duration;
guint rate;
DWORD stream_flags = AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
HRESULT hr;
IAudioClient3 *audio_client = self->audio_client;
hr = audio_client->GetDevicePeriod (&default_period, &min_period);
if (!gst_wasapi2_result (hr)) {
GST_WARNING_OBJECT (self, "Couldn't get device period info");
return FALSE;
}
GST_INFO_OBJECT (self, "wasapi2 default period: %" G_GINT64_FORMAT
", min period: %" G_GINT64_FORMAT, default_period, min_period);
rate = GST_AUDIO_INFO_RATE (&spec->info);
if (self->low_latency) {
device_period = default_period;
/* this should be same as hnsPeriodicity
* when AUDCLNT_STREAMFLAGS_EVENTCALLBACK is used
* And in case of shared mode, hnsPeriodicity should be zero, so
* this value should be zero as well */
device_buffer_duration = 0;
} else {
/* Clamp values to integral multiples of an appropriate period */
gst_wasapi2_util_get_best_buffer_sizes (spec,
default_period, min_period, &device_period, &device_buffer_duration);
}
hr = audio_client->Initialize (AUDCLNT_SHAREMODE_SHARED, stream_flags,
device_buffer_duration,
/* This must always be 0 in shared mode */
0,
self->mix_format, nullptr);
if (!gst_wasapi2_result (hr)) {
GST_WARNING_OBJECT (self, "Couldn't initialize audioclient");
return FALSE;
}
/* device_period can be a non-power-of-10 value so round while converting */
self->device_period =
gst_util_uint64_scale_round (device_period, rate * 100, GST_SECOND);
return TRUE;
}
gboolean
gst_wasapi2_client_open (GstWasapi2Client * client, GstAudioRingBufferSpec * spec,
GstAudioRingBuffer * buf)
{
HRESULT hr;
REFERENCE_TIME latency_rt;
guint bpf, rate;
IAudioClient3 *audio_client;
ComPtr<ISimpleAudioVolume> audio_volume;
gboolean initialized = FALSE;
g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), FALSE);
/* FIXME: Once IAudioClient3 was initialized, we may need to re-open
* IAudioClient3 in order to handle audio format change */
if (client->opened) {
GST_INFO_OBJECT (client, "IAudioClient3 object is initialized already");
return TRUE;
}
audio_client = client->audio_client;
if (!audio_client) {
GST_ERROR_OBJECT (client, "IAudioClient3 object wasn't configured");
return FALSE;
}
if (!client->mix_format) {
GST_ERROR_OBJECT (client, "Unknown mix format");
return FALSE;
}
/* Only use audioclient3 when low-latency is requested because otherwise
* very slow machines and VMs with 1 CPU allocated will get glitches:
* https://bugzilla.gnome.org/show_bug.cgi?id=794497 */
if (client->low_latency)
initialized = gst_wasapi2_client_initialize_audio_client3 (client);
/* Try again if IAudioClinet3 API is unavailable.
* NOTE: IAudioClinet3:: methods might not be available for default device
* NOTE: The default device is a special device which is needed for supporting
* automatic stream routing
* https://docs.microsoft.com/en-us/windows/win32/coreaudio/automatic-stream-routing
*/
if (!initialized)
initialized = gst_wasapi2_client_initialize_audio_client (client, spec);
if (!initialized) {
GST_ERROR_OBJECT (client, "Failed to initialize audioclient");
return FALSE;
}
bpf = GST_AUDIO_INFO_BPF (&spec->info);
rate = GST_AUDIO_INFO_RATE (&spec->info);
/* Total size in frames of the allocated buffer that we will read from */
hr = audio_client->GetBufferSize (&client->buffer_frame_count);
if (!gst_wasapi2_result (hr)) {
return FALSE;
}
GST_INFO_OBJECT (client, "buffer size is %i frames, device period is %i "
"frames, bpf is %i bytes, rate is %i Hz", client->buffer_frame_count,
client->device_period, bpf, rate);
/* Actual latency-time/buffer-time will be different now */
spec->segsize = client->device_period * bpf;
/* We need a minimum of 2 segments to ensure glitch-free playback */
spec->segtotal = MAX (client->buffer_frame_count * bpf / spec->segsize, 2);
GST_INFO_OBJECT (client, "segsize is %i, segtotal is %i", spec->segsize,
spec->segtotal);
/* Get WASAPI latency for logging */
hr = audio_client->GetStreamLatency (&latency_rt);
if (!gst_wasapi2_result (hr)) {
return FALSE;
}
GST_INFO_OBJECT (client, "wasapi2 stream latency: %" G_GINT64_FORMAT " (%"
G_GINT64_FORMAT " ms)", latency_rt, latency_rt / 10000);
/* Set the event handler which will trigger read/write */
hr = audio_client->SetEventHandle (client->event_handle);
if (!gst_wasapi2_result (hr))
return FALSE;
if (client->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER) {
ComPtr<IAudioRenderClient> render_client;
hr = audio_client->GetService (IID_PPV_ARGS (&render_client));
if (!gst_wasapi2_result (hr))
return FALSE;
client->audio_render_client = render_client.Detach ();
} else {
ComPtr<IAudioCaptureClient> capture_client;
hr = audio_client->GetService (IID_PPV_ARGS (&capture_client));
if (!gst_wasapi2_result (hr))
return FALSE;
client->audio_capture_client = capture_client.Detach ();
}
hr = audio_client->GetService (IID_PPV_ARGS (&audio_volume));
if (!gst_wasapi2_result (hr))
return FALSE;
client->audio_volume = audio_volume.Detach ();
gst_audio_ring_buffer_set_channel_positions (buf, client->positions);
client->opened = TRUE;
return TRUE;
}
/* Get the empty space in the buffer that we have to write to */
static gint
gst_wasapi2_client_get_can_frames (GstWasapi2Client * self)
{
HRESULT hr;
UINT32 n_frames_padding;
IAudioClient3 *audio_client = self->audio_client;
if (!audio_client) {
GST_WARNING_OBJECT (self, "IAudioClient3 wasn't configured");
return -1;
}
/* Frames the card hasn't rendered yet */
hr = audio_client->GetCurrentPadding (&n_frames_padding);
if (!gst_wasapi2_result (hr))
return -1;
GST_LOG_OBJECT (self, "%d unread frames (padding)", n_frames_padding);
/* We can write out these many frames */
return self->buffer_frame_count - n_frames_padding;
}
gboolean
gst_wasapi2_client_start (GstWasapi2Client * client)
{
HRESULT hr;
IAudioClient3 *audio_client;
WAVEFORMATEX *mix_format;
g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), FALSE);
audio_client = client->audio_client;
mix_format = client->mix_format;
if (!audio_client) {
GST_ERROR_OBJECT (client, "IAudioClient3 object wasn't configured");
return FALSE;
}
if (!mix_format) {
GST_ERROR_OBJECT (client, "Unknown MixFormat");
return FALSE;
}
if (client->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE &&
!client->audio_capture_client) {
GST_ERROR_OBJECT (client, "IAudioCaptureClient wasn't configured");
return FALSE;
}
if (client->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER &&
!client->audio_render_client) {
GST_ERROR_OBJECT (client, "IAudioRenderClient wasn't configured");
return FALSE;
}
ResetEvent (client->cancellable);
if (client->running) {
GST_WARNING_OBJECT (client, "IAudioClient3 is running already");
return TRUE;
}
/* To avoid start-up glitches, before starting the streaming, we fill the
* buffer with silence as recommended by the documentation:
* https://msdn.microsoft.com/en-us/library/windows/desktop/dd370879%28v=vs.85%29.aspx */
if (client->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER) {
IAudioRenderClient *render_client = client->audio_render_client;
gint n_frames, len;
BYTE *dst = NULL;
n_frames = gst_wasapi2_client_get_can_frames (client);
if (n_frames < 1) {
GST_ERROR_OBJECT (client,
"should have more than %i frames to write", n_frames);
return FALSE;
}
len = n_frames * mix_format->nBlockAlign;
hr = render_client->GetBuffer (n_frames, &dst);
if (!gst_wasapi2_result (hr)) {
GST_ERROR_OBJECT (client, "Couldn't get buffer");
return FALSE;
}
GST_DEBUG_OBJECT (client, "pre-wrote %i bytes of silence", len);
hr = render_client->ReleaseBuffer (n_frames, AUDCLNT_BUFFERFLAGS_SILENT);
if (!gst_wasapi2_result (hr)) {
GST_ERROR_OBJECT (client, "Couldn't release buffer");
return FALSE;
}
}
hr = audio_client->Start ();
client->running = gst_wasapi2_result (hr);
gst_adapter_clear (client->adapter);
return client->running;
}
gboolean
gst_wasapi2_client_stop (GstWasapi2Client * client)
{
HRESULT hr;
IAudioClient3 *audio_client;
g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), FALSE);
audio_client = client->audio_client;
if (!client->running) {
GST_DEBUG_OBJECT (client, "We are not running now");
return TRUE;
}
if (!client->audio_client) {
GST_ERROR_OBJECT (client, "IAudioClient3 object wasn't configured");
return FALSE;
}
client->running = FALSE;
SetEvent (client->cancellable);
hr = audio_client->Stop ();
if (!gst_wasapi2_result (hr))
return FALSE;
/* reset state for reuse case */
hr = audio_client->Reset ();
return gst_wasapi2_result (hr);
}
gint
gst_wasapi2_client_read (GstWasapi2Client * client, gpointer data, guint length)
{
IAudioCaptureClient *capture_client;
WAVEFORMATEX *mix_format;
HRESULT hr;
BYTE *from = NULL;
guint wanted = length;
guint bpf;
DWORD flags;
g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), FALSE);
g_return_val_if_fail (client->audio_capture_client != NULL, -1);
g_return_val_if_fail (client->mix_format != NULL, -1);
capture_client = client->audio_capture_client;
mix_format = client->mix_format;
if (!client->running) {
GST_ERROR_OBJECT (client, "client is not running now");
return -1;
}
/* If we've accumulated enough data, return it immediately */
if (gst_adapter_available (client->adapter) >= wanted) {
memcpy (data, gst_adapter_map (client->adapter, wanted), wanted);
gst_adapter_flush (client->adapter, wanted);
GST_DEBUG_OBJECT (client, "Adapter has enough data, returning %i", wanted);
return wanted;
}
bpf = mix_format->nBlockAlign;
while (wanted > 0) {
DWORD dwWaitResult;
guint got_frames, avail_frames, n_frames, want_frames, read_len;
HANDLE event_handle[2];
event_handle[0] = client->event_handle;
event_handle[1] = client->cancellable;
/* Wait for data to become available */
dwWaitResult = WaitForMultipleObjects (2, event_handle, FALSE, INFINITE);
if (dwWaitResult != WAIT_OBJECT_0 && dwWaitResult != WAIT_OBJECT_0 + 1) {
GST_ERROR_OBJECT (client, "Error waiting for event handle: %x",
(guint) dwWaitResult);
return -1;
}
if (!client->running) {
GST_DEBUG_OBJECT (client, "Cancelled");
return -1;
}
hr = capture_client->GetBuffer (&from, &got_frames, &flags, nullptr,
nullptr);
if (!gst_wasapi2_result (hr)) {
if (hr == AUDCLNT_S_BUFFER_EMPTY) {
GST_INFO_OBJECT (client, "Client buffer is empty, retry");
return 0;
}
GST_ERROR_OBJECT (client, "Couldn't get buffer from capture client");
return -1;
}
if (got_frames == 0) {
GST_DEBUG_OBJECT (client, "No buffer to read");
capture_client->ReleaseBuffer (got_frames);
return 0;
}
if (G_UNLIKELY (flags != 0)) {
/* https://docs.microsoft.com/en-us/windows/win32/api/audioclient/ne-audioclient-_audclnt_bufferflags */
if (flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY)
GST_DEBUG_OBJECT (client, "WASAPI reported discontinuity (glitch?)");
if (flags & AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR)
GST_DEBUG_OBJECT (client, "WASAPI reported a timestamp error");
}
/* Copy all the frames we got into the adapter, and then extract at most
* @wanted size of frames from it. This helps when ::GetBuffer returns more
* data than we can handle right now. */
{
GstBuffer *tmp = gst_buffer_new_allocate (NULL, got_frames * bpf, NULL);
/* If flags has AUDCLNT_BUFFERFLAGS_SILENT, we will ignore the actual
* data and write out silence, see:
* https://docs.microsoft.com/en-us/windows/win32/api/audioclient/ne-audioclient-_audclnt_bufferflags */
if (flags & AUDCLNT_BUFFERFLAGS_SILENT)
memset (from, 0, got_frames * bpf);
gst_buffer_fill (tmp, 0, from, got_frames * bpf);
gst_adapter_push (client->adapter, tmp);
}
/* Release all captured buffers; we copied them above */
hr = capture_client->ReleaseBuffer (got_frames);
from = NULL;
if (!gst_wasapi2_result (hr)) {
GST_ERROR_OBJECT (client, "Failed to release buffer");
return -1;
}
want_frames = wanted / bpf;
avail_frames = gst_adapter_available (client->adapter) / bpf;
/* Only copy data that will fit into the allocated buffer of size @length */
n_frames = MIN (avail_frames, want_frames);
read_len = n_frames * bpf;
if (read_len == 0) {
GST_WARNING_OBJECT (client, "No data to read");
return 0;
}
GST_LOG_OBJECT (client, "frames captured: %d (%d bytes), "
"can read: %d (%d bytes), will read: %d (%d bytes), "
"adapter has: %d (%d bytes)", got_frames, got_frames * bpf, want_frames,
wanted, n_frames, read_len, avail_frames, avail_frames * bpf);
memcpy (data, gst_adapter_map (client->adapter, read_len), read_len);
gst_adapter_flush (client->adapter, read_len);
wanted -= read_len;
}
return length;
}
gint
gst_wasapi2_client_write (GstWasapi2Client * client, gpointer data,
guint length)
{
IAudioRenderClient *render_client;
WAVEFORMATEX *mix_format;
HRESULT hr;
BYTE *dst = nullptr;
DWORD dwWaitResult;
guint can_frames, have_frames, n_frames, write_len = 0;
g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), -1);
g_return_val_if_fail (client->audio_render_client != NULL, -1);
g_return_val_if_fail (client->mix_format != NULL, -1);
if (!client->running) {
GST_WARNING_OBJECT (client, "client is not running now");
return -1;
}
render_client = client->audio_render_client;
mix_format = client->mix_format;
/* We have N frames to be written out */
have_frames = length / (mix_format->nBlockAlign);
/* In shared mode we can write parts of the buffer, so only wait
* in case we can't write anything */
can_frames = gst_wasapi2_client_get_can_frames (client);
if (can_frames < 0) {
GST_ERROR_OBJECT (client, "Error getting frames to write to");
return -1;
}
if (can_frames == 0) {
HANDLE event_handle[2];
event_handle[0] = client->event_handle;
event_handle[1] = client->cancellable;
dwWaitResult = WaitForMultipleObjects (2, event_handle, FALSE, INFINITE);
if (dwWaitResult != WAIT_OBJECT_0 && dwWaitResult != WAIT_OBJECT_0 + 1) {
GST_ERROR_OBJECT (client, "Error waiting for event handle: %x",
(guint) dwWaitResult);
return -1;
}
if (!client->running) {
GST_DEBUG_OBJECT (client, "Cancelled");
return -1;
}
can_frames = gst_wasapi2_client_get_can_frames (client);
if (can_frames < 0) {
GST_ERROR_OBJECT (client, "Error getting frames to write to");
return -1;
}
}
/* We will write out these many frames, and this much length */
n_frames = MIN (can_frames, have_frames);
write_len = n_frames * mix_format->nBlockAlign;
GST_LOG_OBJECT (client, "total: %d, have_frames: %d (%d bytes), "
"can_frames: %d, will write: %d (%d bytes)", client->buffer_frame_count,
have_frames, length, can_frames, n_frames, write_len);
hr = render_client->GetBuffer (n_frames, &dst);
if (!gst_wasapi2_result (hr)) {
GST_ERROR_OBJECT (client, "Couldn't get buffer from client");
return -1;
}
memcpy (dst, data, write_len);
hr = render_client->ReleaseBuffer (n_frames, 0);
return write_len;
}
guint
gst_wasapi2_client_delay (GstWasapi2Client * client)
{
HRESULT hr;
UINT32 delay;
IAudioClient3 *audio_client;
g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), 0);
audio_client = client->audio_client;
if (!audio_client) {
GST_WARNING_OBJECT (client, "IAudioClient3 wasn't configured");
return 0;
}
hr = audio_client->GetCurrentPadding (&delay);
if (!gst_wasapi2_result (hr))
return 0;
return delay;
}
gboolean
gst_wasapi2_client_set_mute (GstWasapi2Client * client, gboolean mute)
{
HRESULT hr;
ISimpleAudioVolume *audio_volume;
g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), FALSE);
audio_volume = client->audio_volume;
if (!audio_volume) {
GST_WARNING_OBJECT (client, "ISimpleAudioVolume object wasn't configured");
return FALSE;
}
hr = audio_volume->SetMute (mute, nullptr);
GST_DEBUG_OBJECT (client, "Set mute %s, hr: 0x%x",
mute ? "enabled" : "disabled", (gint) hr);
return gst_wasapi2_result (hr);
}
gboolean
gst_wasapi2_client_get_mute (GstWasapi2Client * client, gboolean * mute)
{
HRESULT hr;
ISimpleAudioVolume *audio_volume;
BOOL current_mute = FALSE;
g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), FALSE);
g_return_val_if_fail (mute != NULL, FALSE);
audio_volume = client->audio_volume;
if (!audio_volume) {
GST_WARNING_OBJECT (client, "ISimpleAudioVolume object wasn't configured");
return FALSE;
}
hr = audio_volume->GetMute (&current_mute);
if (!gst_wasapi2_result (hr))
return FALSE;
*mute = (gboolean) current_mute;
return TRUE;
}
gboolean
gst_wasapi2_client_set_volume (GstWasapi2Client * client, gfloat volume)
{
HRESULT hr;
ISimpleAudioVolume *audio_volume;
g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), FALSE);
g_return_val_if_fail (volume >= 0 && volume <= 1.0, FALSE);
audio_volume = client->audio_volume;
if (!audio_volume) {
GST_WARNING_OBJECT (client, "ISimpleAudioVolume object wasn't configured");
return FALSE;
}
hr = audio_volume->SetMasterVolume (volume, nullptr);
GST_DEBUG_OBJECT (client, "Set volume %.2f hr: 0x%x", volume, (gint) hr);
return gst_wasapi2_result (hr);
}
gboolean
gst_wasapi2_client_get_volume (GstWasapi2Client * client, gfloat * volume)
{
HRESULT hr;
ISimpleAudioVolume *audio_volume;
float current_volume = FALSE;
g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), FALSE);
g_return_val_if_fail (volume != NULL, FALSE);
audio_volume = client->audio_volume;
if (!audio_volume) {
GST_WARNING_OBJECT (client, "ISimpleAudioVolume object wasn't configured");
return FALSE;
}
hr = audio_volume->GetMasterVolume (&current_volume);
if (!gst_wasapi2_result (hr))
return FALSE;
*volume = current_volume;
return TRUE;
}
GstWasapi2Client *
gst_wasapi2_client_new (GstWasapi2ClientDeviceClass device_class,
gboolean low_latency, gint device_index, const gchar * device_id)
{
GstWasapi2Client *self;
self = (GstWasapi2Client *) g_object_new (GST_TYPE_WASAPI2_CLIENT,
"device-class", device_class, "low-latency", low_latency,
"device-index", device_index, "device", device_id, NULL);
if (!self->audio_client) {
gst_object_unref (self);
return NULL;
}
gst_object_ref_sink (self);
return self;
}
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