... and add wasapi2enumerator debug category Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9468>
137 lines
4.2 KiB
C++
137 lines
4.2 KiB
C++
/* GStreamer
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* Copyright (C) 2020 Seungha Yang <seungha@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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#ifndef __GST_WASAPI2_UTIL_H__
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#define __GST_WASAPI2_UTIL_H__
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include <windows.h>
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#include <initguid.h>
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#include <audioclient.h>
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#include <endpointvolume.h>
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#include <string.h>
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#include <mmdeviceapi.h>
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G_BEGIN_DECLS
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/* Static Caps shared between source, sink, and device provider */
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#define GST_WASAPI2_STATIC_CAPS "audio/x-raw, " \
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"format = (string) " GST_AUDIO_FORMATS_ALL ", " \
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"layout = (string) interleaved, " \
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"rate = " GST_AUDIO_RATE_RANGE ", " \
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"channels = " GST_AUDIO_CHANNELS_RANGE
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#define GST_WASAPI2_CLEAR_COM(obj) G_STMT_START { \
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if (obj) { \
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(obj)->Release (); \
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(obj) = NULL; \
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} \
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} G_STMT_END
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gboolean _gst_wasapi2_result (HRESULT hr,
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GstDebugCategory * cat,
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const gchar * file,
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const gchar * function,
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gint line);
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#ifndef GST_DISABLE_GST_DEBUG
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#define gst_wasapi2_result(result) \
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_gst_wasapi2_result (result, GST_CAT_DEFAULT, __FILE__, GST_FUNCTION, __LINE__)
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#else
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#define gst_wasapi2_result(result) \
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_gst_wasapi2_result (result, NULL, __FILE__, GST_FUNCTION, __LINE__)
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#endif
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typedef enum
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{
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GST_WASAPI2_ENDPOINT_CLASS_CAPTURE = 0,
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GST_WASAPI2_ENDPOINT_CLASS_RENDER,
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GST_WASAPI2_ENDPOINT_CLASS_LOOPBACK_CAPTURE,
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GST_WASAPI2_ENDPOINT_CLASS_INCLUDE_PROCESS_LOOPBACK_CAPTURE,
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GST_WASAPI2_ENDPOINT_CLASS_EXCLUDE_PROCESS_LOOPBACK_CAPTURE,
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} GstWasapi2EndpointClass;
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static inline gboolean
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gst_wasapi2_is_loopback_class (GstWasapi2EndpointClass device_class)
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{
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switch (device_class) {
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case GST_WASAPI2_ENDPOINT_CLASS_LOOPBACK_CAPTURE:
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return TRUE;
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default:
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break;
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}
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return FALSE;
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}
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static inline gboolean
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gst_wasapi2_is_process_loopback_class (GstWasapi2EndpointClass device_class)
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{
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switch (device_class) {
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case GST_WASAPI2_ENDPOINT_CLASS_INCLUDE_PROCESS_LOOPBACK_CAPTURE:
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case GST_WASAPI2_ENDPOINT_CLASS_EXCLUDE_PROCESS_LOOPBACK_CAPTURE:
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return TRUE;
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default:
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break;
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}
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return FALSE;
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}
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guint64 gst_wasapi2_util_waveformatex_to_channel_mask (WAVEFORMATEX * format,
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GstAudioChannelPosition ** out_position);
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const gchar * gst_wasapi2_util_waveformatex_to_audio_format (WAVEFORMATEX * format);
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gboolean gst_wasapi2_util_parse_waveformatex (WAVEFORMATEX * format,
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GstCaps * template_caps,
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GstCaps ** out_caps,
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GstAudioChannelPosition ** out_positions);
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gchar * gst_wasapi2_util_get_error_message (HRESULT hr);
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gboolean gst_wasapi2_can_automatic_stream_routing (void);
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gboolean gst_wasapi2_can_process_loopback (void);
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WAVEFORMATEX * gst_wasapi2_get_default_mix_format (void);
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const wchar_t * gst_wasapi2_get_default_device_id_wide (EDataFlow flow);
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const char * gst_wasapi2_get_default_device_id (EDataFlow flow);
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const gchar * gst_wasapi2_data_flow_to_string (EDataFlow flow);
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const gchar * gst_wasapi2_role_to_string (ERole role);
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G_END_DECLS
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#ifdef __cplusplus
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#include <mutex>
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#define GST_WASAPI2_CALL_ONCE_BEGIN \
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static std::once_flag __once_flag; \
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std::call_once (__once_flag, [&]()
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#define GST_WASAPI2_CALL_ONCE_END )
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#endif
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#endif /* __GST_WASAPI_UTIL_H__ */
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