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gstreamer/subprojects/gst-plugins-bad/sys/wasapi/gstwasapiutil.c

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/*
* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* Copyright (C) 2018 Centricular Ltd.
* Author: Nirbheek Chauhan <nirbheek@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
/* Note: initguid.h can not be included in gstwasapiutil.h, otherwise a
* symbol redefinition error will be raised.
* initguid.h must be included in the C file before mmdeviceapi.h
* which is included in gstwasapiutil.h.
*/
#include <initguid.h>
#include "gstwasapiutil.h"
#include "gstwasapidevice.h"
GST_DEBUG_CATEGORY_EXTERN (gst_wasapi_debug);
#define GST_CAT_DEFAULT gst_wasapi_debug
#include <functiondiscoverykeys_devpkey.h>
/* __uuidof is only available in C++, so we hard-code the GUID values for all
* these. This is ok because these are ABI. MSYS2 provides these in C. */
#ifndef HAVE_AUDCLNT_GUIDS
const CLSID CLSID_MMDeviceEnumerator = { 0xbcde0395, 0xe52f, 0x467c,
{0x8e, 0x3d, 0xc4, 0x57, 0x92, 0x91, 0x69, 0x2e}
};
const IID IID_IMMDeviceEnumerator = { 0xa95664d2, 0x9614, 0x4f35,
{0xa7, 0x46, 0xde, 0x8d, 0xb6, 0x36, 0x17, 0xe6}
};
const IID IID_IMMEndpoint = { 0x1be09788, 0x6894, 0x4089,
{0x85, 0x86, 0x9a, 0x2a, 0x6c, 0x26, 0x5a, 0xc5}
};
const IID IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32,
{0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2}
};
const IID IID_IAudioClock = { 0xcd63314f, 0x3fba, 0x4a1b,
{0x81, 0x2c, 0xef, 0x96, 0x35, 0x87, 0x28, 0xe7}
};
const IID IID_IAudioCaptureClient = { 0xc8adbd64, 0xe71e, 0x48a0,
{0xa4, 0xde, 0x18, 0x5c, 0x39, 0x5c, 0xd3, 0x17}
};
const IID IID_IAudioRenderClient = { 0xf294acfc, 0x3146, 0x4483,
{0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2}
};
#endif
#ifndef HAVE_AUDCLNT3_GUID
const IID IID_IAudioClient3 = { 0x7ed4ee07, 0x8e67, 0x4cd4,
{0x8c, 0x1a, 0x2b, 0x7a, 0x59, 0x87, 0xad, 0x42}
};
#endif
/* Desktop only defines */
#ifndef KSAUDIO_SPEAKER_MONO
#define KSAUDIO_SPEAKER_MONO (SPEAKER_FRONT_CENTER)
#endif
#ifndef KSAUDIO_SPEAKER_1POINT1
#define KSAUDIO_SPEAKER_1POINT1 (SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY)
#endif
#ifndef KSAUDIO_SPEAKER_STEREO
#define KSAUDIO_SPEAKER_STEREO (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT)
#endif
#ifndef KSAUDIO_SPEAKER_2POINT1
#define KSAUDIO_SPEAKER_2POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY)
#endif
#ifndef KSAUDIO_SPEAKER_3POINT0
#define KSAUDIO_SPEAKER_3POINT0 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER)
#endif
#ifndef KSAUDIO_SPEAKER_3POINT1
#define KSAUDIO_SPEAKER_3POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY)
#endif
#ifndef KSAUDIO_SPEAKER_QUAD
#define KSAUDIO_SPEAKER_QUAD (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
#endif
#define KSAUDIO_SPEAKER_SURROUND (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER)
#ifndef KSAUDIO_SPEAKER_5POINT0
#define KSAUDIO_SPEAKER_5POINT0 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | \
SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
#endif
#define KSAUDIO_SPEAKER_5POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
#ifndef KSAUDIO_SPEAKER_7POINT0
#define KSAUDIO_SPEAKER_7POINT0 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | \
SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | \
SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
#endif
#ifndef KSAUDIO_SPEAKER_7POINT1
#define KSAUDIO_SPEAKER_7POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | \
SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER)
#endif
static DWORD default_ch_masks[] = {
0,
KSAUDIO_SPEAKER_MONO,
/* 2ch */
KSAUDIO_SPEAKER_STEREO,
/* 2.1ch */
/* KSAUDIO_SPEAKER_3POINT0 ? */
KSAUDIO_SPEAKER_2POINT1,
/* 4ch */
/* KSAUDIO_SPEAKER_3POINT1 or KSAUDIO_SPEAKER_SURROUND ? */
KSAUDIO_SPEAKER_QUAD,
/* 5ch */
KSAUDIO_SPEAKER_5POINT0,
/* 5.1ch */
KSAUDIO_SPEAKER_5POINT1,
/* 7ch */
KSAUDIO_SPEAKER_7POINT0,
/* 7.1ch */
KSAUDIO_SPEAKER_7POINT1,
};
/* *INDENT-OFF* */
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}
};
/* *INDENT-ON* */
static int windows_major_version = 0;
gboolean
gst_wasapi_util_have_audioclient3 (void)
{
if (windows_major_version > 0)
return windows_major_version == 10;
if (g_getenv ("GST_WASAPI_DISABLE_AUDIOCLIENT3") != NULL) {
windows_major_version = 6;
return FALSE;
}
/* https://msdn.microsoft.com/en-us/library/windows/desktop/ms724834(v=vs.85).aspx */
windows_major_version = 6;
if (g_win32_check_windows_version (10, 0, 0, G_WIN32_OS_ANY))
windows_major_version = 10;
return windows_major_version == 10;
}
GType
gst_wasapi_device_role_get_type (void)
{
static const GEnumValue values[] = {
{GST_WASAPI_DEVICE_ROLE_CONSOLE,
"Games, system notifications, voice commands", "console"},
{GST_WASAPI_DEVICE_ROLE_MULTIMEDIA, "Music, movies, recorded media",
"multimedia"},
{GST_WASAPI_DEVICE_ROLE_COMMS, "Voice communications", "comms"},
{0, NULL, NULL}
};
static GType id = 0;
if (g_once_init_enter ((gsize *) & id)) {
GType _id;
_id = g_enum_register_static ("GstWasapiDeviceRole", values);
g_once_init_leave ((gsize *) & id, _id);
}
return id;
}
gint
gst_wasapi_device_role_to_erole (gint role)
{
switch (role) {
case GST_WASAPI_DEVICE_ROLE_CONSOLE:
return eConsole;
case GST_WASAPI_DEVICE_ROLE_MULTIMEDIA:
return eMultimedia;
case GST_WASAPI_DEVICE_ROLE_COMMS:
return eCommunications;
default:
g_assert_not_reached ();
}
return -1;
}
gint
gst_wasapi_erole_to_device_role (gint erole)
{
switch (erole) {
case eConsole:
return GST_WASAPI_DEVICE_ROLE_CONSOLE;
case eMultimedia:
return GST_WASAPI_DEVICE_ROLE_MULTIMEDIA;
case eCommunications:
return GST_WASAPI_DEVICE_ROLE_COMMS;
default:
g_assert_not_reached ();
}
return -1;
}
static const gchar *
hresult_to_string_fallback (HRESULT hr)
{
const gchar *s = "unknown error";
switch (hr) {
case AUDCLNT_E_NOT_INITIALIZED:
s = "AUDCLNT_E_NOT_INITIALIZED";
break;
case AUDCLNT_E_ALREADY_INITIALIZED:
s = "AUDCLNT_E_ALREADY_INITIALIZED";
break;
case AUDCLNT_E_WRONG_ENDPOINT_TYPE:
s = "AUDCLNT_E_WRONG_ENDPOINT_TYPE";
break;
case AUDCLNT_E_DEVICE_INVALIDATED:
s = "AUDCLNT_E_DEVICE_INVALIDATED";
break;
case AUDCLNT_E_NOT_STOPPED:
s = "AUDCLNT_E_NOT_STOPPED";
break;
case AUDCLNT_E_BUFFER_TOO_LARGE:
s = "AUDCLNT_E_BUFFER_TOO_LARGE";
break;
case AUDCLNT_E_OUT_OF_ORDER:
s = "AUDCLNT_E_OUT_OF_ORDER";
break;
case AUDCLNT_E_UNSUPPORTED_FORMAT:
s = "AUDCLNT_E_UNSUPPORTED_FORMAT";
break;
case AUDCLNT_E_INVALID_DEVICE_PERIOD:
s = "AUDCLNT_E_INVALID_DEVICE_PERIOD";
break;
case AUDCLNT_E_INVALID_SIZE:
s = "AUDCLNT_E_INVALID_SIZE";
break;
case AUDCLNT_E_DEVICE_IN_USE:
s = "AUDCLNT_E_DEVICE_IN_USE";
break;
case AUDCLNT_E_BUFFER_OPERATION_PENDING:
s = "AUDCLNT_E_BUFFER_OPERATION_PENDING";
break;
case AUDCLNT_E_BUFFER_SIZE_ERROR:
s = "AUDCLNT_E_BUFFER_SIZE_ERROR";
break;
case AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED:
s = "AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED";
break;
case AUDCLNT_E_THREAD_NOT_REGISTERED:
s = "AUDCLNT_E_THREAD_NOT_REGISTERED";
break;
case AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED:
s = "AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED";
break;
case AUDCLNT_E_ENDPOINT_CREATE_FAILED:
s = "AUDCLNT_E_ENDPOINT_CREATE_FAILED";
break;
case AUDCLNT_E_SERVICE_NOT_RUNNING:
s = "AUDCLNT_E_SERVICE_NOT_RUNNING";
break;
case AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED:
s = "AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED";
break;
case AUDCLNT_E_EXCLUSIVE_MODE_ONLY:
s = "AUDCLNT_E_EXCLUSIVE_MODE_ONLY";
break;
case AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL:
s = "AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL";
break;
case AUDCLNT_E_EVENTHANDLE_NOT_SET:
s = "AUDCLNT_E_EVENTHANDLE_NOT_SET";
break;
case AUDCLNT_E_INCORRECT_BUFFER_SIZE:
s = "AUDCLNT_E_INCORRECT_BUFFER_SIZE";
break;
case AUDCLNT_E_CPUUSAGE_EXCEEDED:
s = "AUDCLNT_E_CPUUSAGE_EXCEEDED";
break;
case AUDCLNT_S_BUFFER_EMPTY:
s = "AUDCLNT_S_BUFFER_EMPTY";
break;
case AUDCLNT_S_THREAD_ALREADY_REGISTERED:
s = "AUDCLNT_S_THREAD_ALREADY_REGISTERED";
break;
case AUDCLNT_S_POSITION_STALLED:
s = "AUDCLNT_S_POSITION_STALLED";
break;
case E_POINTER:
s = "E_POINTER";
break;
case E_INVALIDARG:
s = "E_INVALIDARG";
break;
}
return s;
}
gchar *
gst_wasapi_util_hresult_to_string (HRESULT hr)
{
gchar *error_text = NULL;
error_text = g_win32_error_message ((gint) hr);
/* g_win32_error_message() seems to be returning empty string for
* AUDCLNT_* cases */
if (!error_text || strlen (error_text) == 0) {
g_free (error_text);
error_text = g_strdup (hresult_to_string_fallback (hr));
}
return error_text;
}
gboolean
gst_wasapi_util_get_devices (GstMMDeviceEnumerator * self,
gboolean active, GList ** devices)
{
gboolean res = FALSE;
static GstStaticCaps scaps = GST_STATIC_CAPS (GST_WASAPI_STATIC_CAPS);
DWORD dwStateMask = active ? DEVICE_STATE_ACTIVE : DEVICE_STATEMASK_ALL;
IMMDeviceCollection *device_collection = NULL;
IMMDeviceEnumerator *enum_handle = NULL;
const gchar *device_class, *element_name;
guint ii, count;
HRESULT hr;
*devices = NULL;
if (!self)
return FALSE;
enum_handle = gst_mm_device_enumerator_get_handle (self);
if (!enum_handle)
return FALSE;
hr = IMMDeviceEnumerator_EnumAudioEndpoints (enum_handle, eAll, dwStateMask,
&device_collection);
HR_FAILED_GOTO (hr, IMMDeviceEnumerator::EnumAudioEndpoints, err);
hr = IMMDeviceCollection_GetCount (device_collection, &count);
HR_FAILED_GOTO (hr, IMMDeviceCollection::GetCount, err);
/* Create a GList of GstDevices* to return */
for (ii = 0; ii < count; ii++) {
IMMDevice *item = NULL;
IMMEndpoint *endpoint = NULL;
IAudioClient *client = NULL;
IPropertyStore *prop_store = NULL;
WAVEFORMATEX *format = NULL;
gchar *description = NULL;
gchar *strid = NULL;
EDataFlow dataflow;
PROPVARIANT var;
wchar_t *wstrid;
GstDevice *device;
GstStructure *props;
GstCaps *caps;
gboolean parse_ret;
hr = IMMDeviceCollection_Item (device_collection, ii, &item);
if (hr != S_OK)
continue;
hr = IMMDevice_QueryInterface (item, &IID_IMMEndpoint, (void **) &endpoint);
if (hr != S_OK)
goto next;
hr = IMMEndpoint_GetDataFlow (endpoint, &dataflow);
if (hr != S_OK)
goto next;
if (dataflow == eRender) {
device_class = "Audio/Sink";
element_name = "wasapisink";
} else {
device_class = "Audio/Source";
element_name = "wasapisrc";
}
PropVariantInit (&var);
hr = IMMDevice_GetId (item, &wstrid);
if (hr != S_OK)
goto next;
strid = g_utf16_to_utf8 (wstrid, -1, NULL, NULL, NULL);
CoTaskMemFree (wstrid);
hr = IMMDevice_OpenPropertyStore (item, STGM_READ, &prop_store);
if (hr != S_OK)
goto next;
/* NOTE: More properties can be added as needed from here:
* https://msdn.microsoft.com/en-us/library/windows/desktop/dd370794(v=vs.85).aspx */
hr = IPropertyStore_GetValue (prop_store, &PKEY_Device_FriendlyName, &var);
if (hr != S_OK)
goto next;
description = g_utf16_to_utf8 (var.pwszVal, -1, NULL, NULL, NULL);
PropVariantClear (&var);
/* Get the audio client so we can fetch the mix format for shared mode
* to get the device format for exclusive mode (or something close to that)
* fetch PKEY_AudioEngine_DeviceFormat from the property store. */
hr = IMMDevice_Activate (item, &IID_IAudioClient, CLSCTX_ALL, NULL,
(void **) &client);
if (hr != S_OK) {
gchar *msg = gst_wasapi_util_hresult_to_string (hr);
GST_ERROR_OBJECT (self, "IMMDevice::Activate (IID_IAudioClient) failed"
"on %s: %s", strid, msg);
g_free (msg);
goto next;
}
hr = IAudioClient_GetMixFormat (client, &format);
if (hr != S_OK || format == NULL) {
gchar *msg = gst_wasapi_util_hresult_to_string (hr);
GST_ERROR_OBJECT (self, "GetMixFormat failed on %s: %s", strid, msg);
g_free (msg);
goto next;
}
parse_ret =
gst_wasapi_util_parse_waveformatex ((WAVEFORMATEXTENSIBLE *) format,
gst_static_caps_get (&scaps), &caps, NULL);
CoTaskMemFree (format);
if (!parse_ret)
goto next;
/* Set some useful properties */
props = gst_structure_new ("wasapi-proplist",
"device.api", G_TYPE_STRING, "wasapi",
"device.strid", G_TYPE_STRING, GST_STR_NULL (strid),
"wasapi.device.description", G_TYPE_STRING, description, NULL);
device = g_object_new (GST_TYPE_WASAPI_DEVICE, "device", strid,
"display-name", description, "caps", caps,
"device-class", device_class, "properties", props, NULL);
GST_WASAPI_DEVICE (device)->element = element_name;
gst_structure_free (props);
gst_caps_unref (caps);
*devices = g_list_prepend (*devices, device);
next:
PropVariantClear (&var);
if (prop_store)
IUnknown_Release (prop_store);
if (endpoint)
IUnknown_Release (endpoint);
if (client)
IUnknown_Release (client);
if (item)
IUnknown_Release (item);
if (description)
g_free (description);
if (strid)
g_free (strid);
}
res = TRUE;
err:
if (device_collection)
IUnknown_Release (device_collection);
return res;
}
gboolean
gst_wasapi_util_get_device_format (GstElement * self,
gint device_mode, IMMDevice * device, IAudioClient * client,
WAVEFORMATEX ** ret_format)
{
WAVEFORMATEX *format;
HRESULT hr;
*ret_format = NULL;
hr = IAudioClient_GetMixFormat (client, &format);
HR_FAILED_RET (hr, IAudioClient::GetMixFormat, FALSE);
/* WASAPI always accepts the format returned by GetMixFormat in shared mode */
if (device_mode == AUDCLNT_SHAREMODE_SHARED)
goto out;
/* WASAPI may or may not support this format in exclusive mode */
hr = IAudioClient_IsFormatSupported (client, AUDCLNT_SHAREMODE_EXCLUSIVE,
format, NULL);
if (hr == S_OK)
goto out;
CoTaskMemFree (format);
/* Open the device property store, and get the format that WASAPI has been
* using for sending data to the device */
{
PROPVARIANT var;
IPropertyStore *prop_store = NULL;
hr = IMMDevice_OpenPropertyStore (device, STGM_READ, &prop_store);
HR_FAILED_RET (hr, IMMDevice::OpenPropertyStore, FALSE);
hr = IPropertyStore_GetValue (prop_store, &PKEY_AudioEngine_DeviceFormat,
&var);
if (hr != S_OK) {
gchar *msg = gst_wasapi_util_hresult_to_string (hr);
GST_ERROR_OBJECT (self, "GetValue failed: %s", msg);
g_free (msg);
IUnknown_Release (prop_store);
return FALSE;
}
format = CoTaskMemAlloc (var.blob.cbSize);
memcpy (format, var.blob.pBlobData, var.blob.cbSize);
PropVariantClear (&var);
IUnknown_Release (prop_store);
}
/* WASAPI may or may not support this format in exclusive mode */
hr = IAudioClient_IsFormatSupported (client, AUDCLNT_SHAREMODE_EXCLUSIVE,
format, NULL);
if (hr == S_OK)
goto out;
GST_ERROR_OBJECT (self, "AudioEngine DeviceFormat not supported");
CoTaskMemFree (format);
return FALSE;
out:
*ret_format = format;
return TRUE;
}
gboolean
gst_wasapi_util_get_device (GstMMDeviceEnumerator * self,
gint data_flow, gint role, const wchar_t *device_strid,
IMMDevice ** ret_device)
{
gboolean res = FALSE;
HRESULT hr;
IMMDeviceEnumerator *enum_handle = NULL;
IMMDevice *device = NULL;
if (!self)
return FALSE;
enum_handle = gst_mm_device_enumerator_get_handle (self);
if (!enum_handle)
return FALSE;
if (!device_strid) {
hr = IMMDeviceEnumerator_GetDefaultAudioEndpoint (enum_handle, data_flow,
role, &device);
HR_FAILED_GOTO (hr, IMMDeviceEnumerator::GetDefaultAudioEndpoint, beach);
} else {
hr = IMMDeviceEnumerator_GetDevice (enum_handle, device_strid, &device);
if (hr != S_OK) {
gchar *msg = gst_wasapi_util_hresult_to_string (hr);
GST_ERROR_OBJECT (self, "IMMDeviceEnumerator::GetDevice (%S) failed"
": %s", device_strid, msg);
g_free (msg);
goto beach;
}
}
IUnknown_AddRef (device);
*ret_device = device;
res = TRUE;
beach:
if (device != NULL)
IUnknown_Release (device);
return res;
}
gboolean
gst_wasapi_util_get_audio_client (GstElement * self,
IMMDevice * device, IAudioClient ** ret_client)
{
IAudioClient *client = NULL;
gboolean res = FALSE;
HRESULT hr;
if (gst_wasapi_util_have_audioclient3 ())
hr = IMMDevice_Activate (device, &IID_IAudioClient3, CLSCTX_ALL, NULL,
(void **) &client);
else
hr = IMMDevice_Activate (device, &IID_IAudioClient, CLSCTX_ALL, NULL,
(void **) &client);
HR_FAILED_GOTO (hr, IMMDevice::Activate (IID_IAudioClient), beach);
IUnknown_AddRef (client);
*ret_client = client;
res = TRUE;
beach:
if (client != NULL)
IUnknown_Release (client);
return res;
}
gboolean
gst_wasapi_util_get_render_client (GstElement * self, IAudioClient * client,
IAudioRenderClient ** ret_render_client)
{
gboolean res = FALSE;
HRESULT hr;
IAudioRenderClient *render_client = NULL;
hr = IAudioClient_GetService (client, &IID_IAudioRenderClient,
(void **) &render_client);
HR_FAILED_GOTO (hr, IAudioClient::GetService, beach);
*ret_render_client = render_client;
res = TRUE;
beach:
return res;
}
gboolean
gst_wasapi_util_get_capture_client (GstElement * self, IAudioClient * client,
IAudioCaptureClient ** ret_capture_client)
{
gboolean res = FALSE;
HRESULT hr;
IAudioCaptureClient *capture_client = NULL;
hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
(void **) &capture_client);
HR_FAILED_GOTO (hr, IAudioClient::GetService, beach);
*ret_capture_client = capture_client;
res = TRUE;
beach:
return res;
}
gboolean
gst_wasapi_util_get_clock (GstElement * self, IAudioClient * client,
IAudioClock ** ret_clock)
{
gboolean res = FALSE;
HRESULT hr;
IAudioClock *clock = NULL;
hr = IAudioClient_GetService (client, &IID_IAudioClock, (void **) &clock);
HR_FAILED_GOTO (hr, IAudioClient::GetService, beach);
*ret_clock = clock;
res = TRUE;
beach:
return res;
}
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;
}
/* 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;
if (nChannels > 2 && !dwChannelMask) {
GST_WARNING ("Unknown channel mask value for %d channel stream", nChannels);
if (nChannels >= G_N_ELEMENTS (default_ch_masks)) {
GST_ERROR ("Too many channels %d", nChannels);
return 0;
}
dwChannelMask = default_ch_masks[nChannels];
}
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;
}
gboolean
gst_wasapi_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;
}
void
gst_wasapi_util_get_best_buffer_sizes (GstAudioRingBufferSpec * spec,
gboolean exclusive, REFERENCE_TIME default_period,
REFERENCE_TIME min_period, REFERENCE_TIME * ret_period,
REFERENCE_TIME * ret_buffer_duration)
{
REFERENCE_TIME use_period, use_buffer;
/* Figure out what integral device period to use as the base */
if (exclusive) {
/* Exclusive mode can run at multiples of either the minimum period or the
* default period; these are on the hardware ringbuffer */
if (spec->latency_time * 10 > default_period)
use_period = default_period;
else
use_period = min_period;
} else {
/* 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);
if (exclusive) {
/* Buffer duration is the same as the period in exclusive mode. The
* hardware is always writing out one buffer (of size *ret_period), and
* we're writing to the other one. */
use_buffer = use_period;
} else {
/* 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;
}
gboolean
gst_wasapi_util_initialize_audioclient (GstElement * self,
GstAudioRingBufferSpec * spec, IAudioClient * client,
WAVEFORMATEX * format, guint sharemode, gboolean low_latency,
gboolean loopback, guint * ret_devicep_frames)
{
REFERENCE_TIME default_period, min_period;
REFERENCE_TIME device_period, device_buffer_duration;
guint rate, stream_flags;
guint32 n_frames;
HRESULT hr;
hr = IAudioClient_GetDevicePeriod (client, &default_period, &min_period);
HR_FAILED_RET (hr, IAudioClient::GetDevicePeriod, FALSE);
GST_INFO_OBJECT (self, "wasapi default period: %" G_GINT64_FORMAT
", min period: %" G_GINT64_FORMAT, default_period, min_period);
rate = GST_AUDIO_INFO_RATE (&spec->info);
if (low_latency) {
if (sharemode == AUDCLNT_SHAREMODE_SHARED) {
device_period = default_period;
device_buffer_duration = 0;
} else {
device_period = min_period;
device_buffer_duration = min_period;
}
} else {
/* Clamp values to integral multiples of an appropriate period */
gst_wasapi_util_get_best_buffer_sizes (spec,
sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE, default_period,
min_period, &device_period, &device_buffer_duration);
}
stream_flags = AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
if (loopback)
stream_flags |= AUDCLNT_STREAMFLAGS_LOOPBACK;
hr = IAudioClient_Initialize (client, sharemode, stream_flags,
device_buffer_duration,
/* This must always be 0 in shared mode */
sharemode == AUDCLNT_SHAREMODE_SHARED ? 0 : device_period, format, NULL);
if (hr == AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED &&
sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE) {
GST_WARNING_OBJECT (self, "initialize failed due to unaligned period %i",
(int) device_period);
/* Calculate a new aligned period. First get the aligned buffer size. */
hr = IAudioClient_GetBufferSize (client, &n_frames);
HR_FAILED_RET (hr, IAudioClient::GetBufferSize, FALSE);
device_period = (GST_SECOND / 100) * n_frames / rate;
GST_WARNING_OBJECT (self, "trying to re-initialize with period %i "
"(%i frames, %i rate)", (int) device_period, n_frames, rate);
hr = IAudioClient_Initialize (client, sharemode, stream_flags,
device_period, device_period, format, NULL);
}
HR_FAILED_RET (hr, IAudioClient::Initialize, FALSE);
if (sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE) {
/* We use the device period for the segment size and that needs to match
* the buffer size exactly when we write into it */
hr = IAudioClient_GetBufferSize (client, &n_frames);
HR_FAILED_RET (hr, IAudioClient::GetBufferSize, FALSE);
*ret_devicep_frames = n_frames;
} else {
/* device_period can be a non-power-of-10 value so round while converting */
*ret_devicep_frames =
gst_util_uint64_scale_round (device_period, rate * 100, GST_SECOND);
}
return TRUE;
}
gboolean
gst_wasapi_util_initialize_audioclient3 (GstElement * self,
GstAudioRingBufferSpec * spec, IAudioClient3 * client,
WAVEFORMATEX * format, gboolean low_latency, gboolean loopback,
guint * ret_devicep_frames)
{
HRESULT hr;
gint stream_flags;
guint devicep_frames;
guint defaultp_frames, fundp_frames, minp_frames, maxp_frames;
WAVEFORMATEX *tmpf;
hr = IAudioClient3_GetSharedModeEnginePeriod (client, format,
&defaultp_frames, &fundp_frames, &minp_frames, &maxp_frames);
HR_FAILED_RET (hr, IAudioClient3::GetSharedModeEnginePeriod, FALSE);
GST_INFO_OBJECT (self, "Using IAudioClient3, default period %i frames, "
"fundamental period %i frames, minimum period %i frames, maximum period "
"%i frames", defaultp_frames, fundp_frames, minp_frames, maxp_frames);
if (low_latency)
devicep_frames = minp_frames;
else
/* Just pick the max period, because lower values can cause glitches
* https://bugzilla.gnome.org/show_bug.cgi?id=794497 */
devicep_frames = maxp_frames;
stream_flags = AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
if (loopback)
stream_flags |= AUDCLNT_STREAMFLAGS_LOOPBACK;
hr = IAudioClient3_InitializeSharedAudioStream (client, stream_flags,
devicep_frames, format, NULL);
HR_FAILED_RET (hr, IAudioClient3::InitializeSharedAudioStream, FALSE);
hr = IAudioClient3_GetCurrentSharedModeEnginePeriod (client, &tmpf,
&devicep_frames);
CoTaskMemFree (tmpf);
HR_FAILED_RET (hr, IAudioClient3::GetCurrentSharedModeEnginePeriod, FALSE);
*ret_devicep_frames = devicep_frames;
return TRUE;
}
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