mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-22 16:26:39 +00:00
964 lines
30 KiB
C
964 lines
30 KiB
C
/*
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* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
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* Copyright (C) 2018 Centricular Ltd.
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* Author: Nirbheek Chauhan <nirbheek@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include <config.h>
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#endif
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/* Note: initguid.h can not be included in gstwasapiutil.h, otherwise a
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* symbol redefinition error will be raised.
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* initguid.h must be included in the C file before mmdeviceapi.h
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* which is included in gstwasapiutil.h.
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*/
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#ifdef _MSC_VER
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#include <initguid.h>
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#endif
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#include "gstwasapiutil.h"
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#include "gstwasapidevice.h"
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GST_DEBUG_CATEGORY_EXTERN (gst_wasapi_debug);
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#define GST_CAT_DEFAULT gst_wasapi_debug
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/* This was only added to MinGW in ~2015 and our Cerbero toolchain is too old */
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#if defined(_MSC_VER)
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#include <functiondiscoverykeys_devpkey.h>
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#elif !defined(PKEY_Device_FriendlyName)
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#include <initguid.h>
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#include <propkey.h>
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DEFINE_PROPERTYKEY (PKEY_Device_FriendlyName, 0xa45c254e, 0xdf1c, 0x4efd, 0x80,
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0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 14);
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DEFINE_PROPERTYKEY (PKEY_AudioEngine_DeviceFormat, 0xf19f064d, 0x82c, 0x4e27,
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0xbc, 0x73, 0x68, 0x82, 0xa1, 0xbb, 0x8e, 0x4c, 0);
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#endif
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/* __uuidof is only available in C++, so we hard-code the GUID values for all
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* these. This is ok because these are ABI. */
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const CLSID CLSID_MMDeviceEnumerator = { 0xbcde0395, 0xe52f, 0x467c,
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{0x8e, 0x3d, 0xc4, 0x57, 0x92, 0x91, 0x69, 0x2e}
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};
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const IID IID_IMMDeviceEnumerator = { 0xa95664d2, 0x9614, 0x4f35,
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{0xa7, 0x46, 0xde, 0x8d, 0xb6, 0x36, 0x17, 0xe6}
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};
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const IID IID_IMMEndpoint = { 0x1be09788, 0x6894, 0x4089,
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{0x85, 0x86, 0x9a, 0x2a, 0x6c, 0x26, 0x5a, 0xc5}
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};
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const IID IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32,
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{0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2}
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};
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const IID IID_IAudioClient3 = { 0x7ed4ee07, 0x8e67, 0x4cd4,
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{0x8c, 0x1a, 0x2b, 0x7a, 0x59, 0x87, 0xad, 0x42}
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};
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const IID IID_IAudioClock = { 0xcd63314f, 0x3fba, 0x4a1b,
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{0x81, 0x2c, 0xef, 0x96, 0x35, 0x87, 0x28, 0xe7}
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};
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const IID IID_IAudioCaptureClient = { 0xc8adbd64, 0xe71e, 0x48a0,
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{0xa4, 0xde, 0x18, 0x5c, 0x39, 0x5c, 0xd3, 0x17}
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};
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const IID IID_IAudioRenderClient = { 0xf294acfc, 0x3146, 0x4483,
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{0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2}
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};
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/* *INDENT-OFF* */
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static struct
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{
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guint64 wasapi_pos;
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GstAudioChannelPosition gst_pos;
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} wasapi_to_gst_pos[] = {
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{SPEAKER_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT},
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{SPEAKER_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{SPEAKER_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER},
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{SPEAKER_LOW_FREQUENCY, GST_AUDIO_CHANNEL_POSITION_LFE1},
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{SPEAKER_BACK_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT},
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{SPEAKER_BACK_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
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{SPEAKER_FRONT_LEFT_OF_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER},
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{SPEAKER_FRONT_RIGHT_OF_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER},
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{SPEAKER_BACK_CENTER, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
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/* Enum values diverge from this point onwards */
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{SPEAKER_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT},
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{SPEAKER_SIDE_RIGHT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT},
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{SPEAKER_TOP_CENTER, GST_AUDIO_CHANNEL_POSITION_TOP_CENTER},
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{SPEAKER_TOP_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT},
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{SPEAKER_TOP_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER},
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{SPEAKER_TOP_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT},
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{SPEAKER_TOP_BACK_LEFT, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT},
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{SPEAKER_TOP_BACK_CENTER, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER},
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{SPEAKER_TOP_BACK_RIGHT, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT}
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};
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/* *INDENT-ON* */
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static int windows_major_version = 0;
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gboolean
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gst_wasapi_util_have_audioclient3 (void)
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{
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if (windows_major_version > 0)
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return windows_major_version == 10;
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if (g_getenv ("GST_WASAPI_DISABLE_AUDIOCLIENT3") != NULL) {
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windows_major_version = 6;
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return FALSE;
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}
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/* https://msdn.microsoft.com/en-us/library/windows/desktop/ms724834(v=vs.85).aspx */
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windows_major_version = 6;
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if (g_win32_check_windows_version (10, 0, 0, G_WIN32_OS_ANY))
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windows_major_version = 10;
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return windows_major_version == 10;
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}
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GType
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gst_wasapi_device_role_get_type (void)
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{
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static const GEnumValue values[] = {
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{GST_WASAPI_DEVICE_ROLE_CONSOLE,
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"Games, system notifications, voice commands", "console"},
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{GST_WASAPI_DEVICE_ROLE_MULTIMEDIA, "Music, movies, recorded media",
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"multimedia"},
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{GST_WASAPI_DEVICE_ROLE_COMMS, "Voice communications", "comms"},
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{0, NULL, NULL}
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};
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static GType id = 0;
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if (g_once_init_enter ((gsize *) & id)) {
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GType _id;
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_id = g_enum_register_static ("GstWasapiDeviceRole", values);
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g_once_init_leave ((gsize *) & id, _id);
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}
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return id;
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}
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gint
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gst_wasapi_device_role_to_erole (gint role)
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{
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switch (role) {
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case GST_WASAPI_DEVICE_ROLE_CONSOLE:
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return eConsole;
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case GST_WASAPI_DEVICE_ROLE_MULTIMEDIA:
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return eMultimedia;
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case GST_WASAPI_DEVICE_ROLE_COMMS:
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return eCommunications;
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default:
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g_assert_not_reached ();
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}
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return -1;
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}
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gint
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gst_wasapi_erole_to_device_role (gint erole)
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{
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switch (erole) {
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case eConsole:
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return GST_WASAPI_DEVICE_ROLE_CONSOLE;
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case eMultimedia:
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return GST_WASAPI_DEVICE_ROLE_MULTIMEDIA;
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case eCommunications:
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return GST_WASAPI_DEVICE_ROLE_COMMS;
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default:
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g_assert_not_reached ();
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}
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return -1;
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}
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static const gchar *
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hresult_to_string_fallback (HRESULT hr)
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{
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const gchar *s = "unknown error";
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switch (hr) {
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case AUDCLNT_E_NOT_INITIALIZED:
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s = "AUDCLNT_E_NOT_INITIALIZED";
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break;
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case AUDCLNT_E_ALREADY_INITIALIZED:
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s = "AUDCLNT_E_ALREADY_INITIALIZED";
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break;
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case AUDCLNT_E_WRONG_ENDPOINT_TYPE:
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s = "AUDCLNT_E_WRONG_ENDPOINT_TYPE";
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break;
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case AUDCLNT_E_DEVICE_INVALIDATED:
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s = "AUDCLNT_E_DEVICE_INVALIDATED";
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break;
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case AUDCLNT_E_NOT_STOPPED:
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s = "AUDCLNT_E_NOT_STOPPED";
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break;
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case AUDCLNT_E_BUFFER_TOO_LARGE:
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s = "AUDCLNT_E_BUFFER_TOO_LARGE";
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break;
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case AUDCLNT_E_OUT_OF_ORDER:
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s = "AUDCLNT_E_OUT_OF_ORDER";
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break;
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case AUDCLNT_E_UNSUPPORTED_FORMAT:
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s = "AUDCLNT_E_UNSUPPORTED_FORMAT";
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break;
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case AUDCLNT_E_INVALID_DEVICE_PERIOD:
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s = "AUDCLNT_E_INVALID_DEVICE_PERIOD";
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break;
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case AUDCLNT_E_INVALID_SIZE:
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s = "AUDCLNT_E_INVALID_SIZE";
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break;
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case AUDCLNT_E_DEVICE_IN_USE:
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s = "AUDCLNT_E_DEVICE_IN_USE";
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break;
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case AUDCLNT_E_BUFFER_OPERATION_PENDING:
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s = "AUDCLNT_E_BUFFER_OPERATION_PENDING";
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break;
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case AUDCLNT_E_BUFFER_SIZE_ERROR:
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s = "AUDCLNT_E_BUFFER_SIZE_ERROR";
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break;
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case AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED:
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s = "AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED";
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break;
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case AUDCLNT_E_THREAD_NOT_REGISTERED:
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s = "AUDCLNT_E_THREAD_NOT_REGISTERED";
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break;
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case AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED:
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s = "AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED";
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break;
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case AUDCLNT_E_ENDPOINT_CREATE_FAILED:
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s = "AUDCLNT_E_ENDPOINT_CREATE_FAILED";
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break;
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case AUDCLNT_E_SERVICE_NOT_RUNNING:
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s = "AUDCLNT_E_SERVICE_NOT_RUNNING";
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break;
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case AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED:
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s = "AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED";
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break;
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case AUDCLNT_E_EXCLUSIVE_MODE_ONLY:
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s = "AUDCLNT_E_EXCLUSIVE_MODE_ONLY";
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break;
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case AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL:
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s = "AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL";
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break;
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case AUDCLNT_E_EVENTHANDLE_NOT_SET:
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s = "AUDCLNT_E_EVENTHANDLE_NOT_SET";
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break;
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case AUDCLNT_E_INCORRECT_BUFFER_SIZE:
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s = "AUDCLNT_E_INCORRECT_BUFFER_SIZE";
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break;
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case AUDCLNT_E_CPUUSAGE_EXCEEDED:
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s = "AUDCLNT_E_CPUUSAGE_EXCEEDED";
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break;
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case AUDCLNT_S_BUFFER_EMPTY:
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s = "AUDCLNT_S_BUFFER_EMPTY";
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break;
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case AUDCLNT_S_THREAD_ALREADY_REGISTERED:
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s = "AUDCLNT_S_THREAD_ALREADY_REGISTERED";
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break;
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case AUDCLNT_S_POSITION_STALLED:
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s = "AUDCLNT_S_POSITION_STALLED";
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break;
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case E_POINTER:
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s = "E_POINTER";
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break;
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case E_INVALIDARG:
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s = "E_INVALIDARG";
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break;
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}
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return s;
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}
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gchar *
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gst_wasapi_util_hresult_to_string (HRESULT hr)
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{
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gchar *error_text = NULL;
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error_text = g_win32_error_message ((gint) hr);
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/* g_win32_error_message() seems to be returning empty string for
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* AUDCLNT_* cases */
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if (!error_text || strlen (error_text) == 0) {
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g_free (error_text);
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error_text = g_strdup (hresult_to_string_fallback (hr));
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}
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return error_text;
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}
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gboolean
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gst_wasapi_util_get_devices (GstMMDeviceEnumerator * self,
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gboolean active, GList ** devices)
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{
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gboolean res = FALSE;
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static GstStaticCaps scaps = GST_STATIC_CAPS (GST_WASAPI_STATIC_CAPS);
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DWORD dwStateMask = active ? DEVICE_STATE_ACTIVE : DEVICE_STATEMASK_ALL;
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IMMDeviceCollection *device_collection = NULL;
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IMMDeviceEnumerator *enum_handle = NULL;
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const gchar *device_class, *element_name;
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guint ii, count;
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HRESULT hr;
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*devices = NULL;
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if (!self)
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return FALSE;
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enum_handle = gst_mm_device_enumerator_get_handle (self);
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if (!enum_handle)
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return FALSE;
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hr = IMMDeviceEnumerator_EnumAudioEndpoints (enum_handle, eAll, dwStateMask,
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&device_collection);
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HR_FAILED_GOTO (hr, IMMDeviceEnumerator::EnumAudioEndpoints, err);
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hr = IMMDeviceCollection_GetCount (device_collection, &count);
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HR_FAILED_GOTO (hr, IMMDeviceCollection::GetCount, err);
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/* Create a GList of GstDevices* to return */
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for (ii = 0; ii < count; ii++) {
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IMMDevice *item = NULL;
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IMMEndpoint *endpoint = NULL;
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IAudioClient *client = NULL;
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IPropertyStore *prop_store = NULL;
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WAVEFORMATEX *format = NULL;
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gchar *description = NULL;
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gchar *strid = NULL;
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EDataFlow dataflow;
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PROPVARIANT var;
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wchar_t *wstrid;
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GstDevice *device;
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GstStructure *props;
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GstCaps *caps;
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hr = IMMDeviceCollection_Item (device_collection, ii, &item);
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if (hr != S_OK)
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continue;
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|
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hr = IMMDevice_QueryInterface (item, &IID_IMMEndpoint, (void **) &endpoint);
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if (hr != S_OK)
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goto next;
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|
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hr = IMMEndpoint_GetDataFlow (endpoint, &dataflow);
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if (hr != S_OK)
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goto next;
|
|
|
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if (dataflow == eRender) {
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device_class = "Audio/Sink";
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element_name = "wasapisink";
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} else {
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device_class = "Audio/Source";
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element_name = "wasapisrc";
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}
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|
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PropVariantInit (&var);
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hr = IMMDevice_GetId (item, &wstrid);
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if (hr != S_OK)
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goto next;
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strid = g_utf16_to_utf8 (wstrid, -1, NULL, NULL, NULL);
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CoTaskMemFree (wstrid);
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|
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hr = IMMDevice_OpenPropertyStore (item, STGM_READ, &prop_store);
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if (hr != S_OK)
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goto next;
|
|
|
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/* NOTE: More properties can be added as needed from here:
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* https://msdn.microsoft.com/en-us/library/windows/desktop/dd370794(v=vs.85).aspx */
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hr = IPropertyStore_GetValue (prop_store, &PKEY_Device_FriendlyName, &var);
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if (hr != S_OK)
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goto next;
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description = g_utf16_to_utf8 (var.pwszVal, -1, NULL, NULL, NULL);
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PropVariantClear (&var);
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|
|
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/* Get the audio client so we can fetch the mix format for shared mode
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* to get the device format for exclusive mode (or something close to that)
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* fetch PKEY_AudioEngine_DeviceFormat from the property store. */
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hr = IMMDevice_Activate (item, &IID_IAudioClient, CLSCTX_ALL, NULL,
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(void **) &client);
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if (hr != S_OK) {
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gchar *msg = gst_wasapi_util_hresult_to_string (hr);
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GST_ERROR_OBJECT (self, "IMMDevice::Activate (IID_IAudioClient) failed"
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"on %s: %s", strid, msg);
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g_free (msg);
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goto next;
|
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}
|
|
|
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hr = IAudioClient_GetMixFormat (client, &format);
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if (hr != S_OK || format == NULL) {
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gchar *msg = gst_wasapi_util_hresult_to_string (hr);
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GST_ERROR_OBJECT (self, "GetMixFormat failed on %s: %s", strid, msg);
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g_free (msg);
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goto next;
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}
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|
|
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if (!gst_wasapi_util_parse_waveformatex ((WAVEFORMATEXTENSIBLE *) format,
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gst_static_caps_get (&scaps), &caps, NULL))
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goto next;
|
|
|
|
/* Set some useful properties */
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|
props = gst_structure_new ("wasapi-proplist",
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"device.api", G_TYPE_STRING, "wasapi",
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"device.strid", G_TYPE_STRING, GST_STR_NULL (strid),
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"wasapi.device.description", G_TYPE_STRING, description, NULL);
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|
|
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device = g_object_new (GST_TYPE_WASAPI_DEVICE, "device", strid,
|
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"display-name", description, "caps", caps,
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"device-class", device_class, "properties", props, NULL);
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|
GST_WASAPI_DEVICE (device)->element = element_name;
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|
|
|
gst_structure_free (props);
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gst_caps_unref (caps);
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*devices = g_list_prepend (*devices, device);
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|
|
|
next:
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PropVariantClear (&var);
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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 = malloc (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");
|
|
free (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;
|
|
|
|
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;
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WAVEFORMATEX *tmpf;
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|
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hr = IAudioClient3_GetSharedModeEnginePeriod (client, format,
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|
&defaultp_frames, &fundp_frames, &minp_frames, &maxp_frames);
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|
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)
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|
devicep_frames = minp_frames;
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|
else
|
|
/* Just pick the max period, because lower values can cause glitches
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|
* 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;
|
|
}
|