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589ff8ca6d
Adding IMMDeviceEnumerator::RegisterEndpointNotificationCallback in order to support device monitoring. On OnDeviceAdded(), OnDeviceRemoved(), and OnDefaultDeviceChanged() callback, wasapi device provider implementation will enumerate devices again and will notify newly added and removed device via GstDeviceProvider API. As a bonus point, this IMMDeviceEnumerator abstraction object will spawn a dedicated internal COM thread, so various COM thread related issues of WASAPI plugin can be resolved by this commit. Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1649 Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1110 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2484>
139 lines
4.9 KiB
C
139 lines
4.9 KiB
C
/*
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* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.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_WASAPI_UTIL_H__
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#define __GST_WASAPI_UTIL_H__
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/gstaudiosrc.h>
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#include <gst/audio/gstaudiosink.h>
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#include <mmdeviceapi.h>
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#include <audioclient.h>
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#include "gstaudioclient3.h"
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#include "gstmmdeviceenumerator.h"
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/* Static Caps shared between source, sink, and device provider */
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#define GST_WASAPI_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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/* Standard error path */
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#define HR_FAILED_AND(hr,func,and) \
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G_STMT_START { \
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if (FAILED (hr)) { \
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gchar *msg = gst_wasapi_util_hresult_to_string (hr); \
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GST_ERROR_OBJECT (self, #func " failed (%x): %s", (guint) hr, msg); \
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g_free (msg); \
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and; \
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} \
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} G_STMT_END
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#define HR_FAILED_RET(hr,func,ret) HR_FAILED_AND(hr,func,return ret)
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#define HR_FAILED_GOTO(hr,func,where) HR_FAILED_AND(hr,func,res = FALSE; goto where)
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#define HR_FAILED_ELEMENT_ERROR_AND(hr,func,el,and) \
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G_STMT_START { \
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if (FAILED (hr)) { \
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gchar *msg = gst_wasapi_util_hresult_to_string (hr); \
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GST_ERROR_OBJECT (el, #func " failed (%x): %s", (guint) hr, msg); \
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if (GST_IS_AUDIO_SRC (el)) \
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GST_ELEMENT_ERROR(el, RESOURCE, READ, \
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(#func " failed (%x): %s", (guint) hr, msg), (NULL)); \
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else \
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GST_ELEMENT_ERROR(el, RESOURCE, WRITE, \
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(#func " failed (%x): %s", (guint) hr, msg), (NULL)); \
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g_free (msg); \
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and; \
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} \
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} G_STMT_END
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#define HR_FAILED_ELEMENT_ERROR_RET(hr,func,el,ret) \
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HR_FAILED_ELEMENT_ERROR_AND(hr,func,el,return ret)
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/* Device role enum property */
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typedef enum
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{
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GST_WASAPI_DEVICE_ROLE_CONSOLE,
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GST_WASAPI_DEVICE_ROLE_MULTIMEDIA,
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GST_WASAPI_DEVICE_ROLE_COMMS
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} GstWasapiDeviceRole;
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#define GST_WASAPI_DEVICE_TYPE_ROLE (gst_wasapi_device_role_get_type())
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GType gst_wasapi_device_role_get_type (void);
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/* Utilities */
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gboolean gst_wasapi_util_have_audioclient3 (void);
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gint gst_wasapi_device_role_to_erole (gint role);
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gint gst_wasapi_erole_to_device_role (gint erole);
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gchar *gst_wasapi_util_hresult_to_string (HRESULT hr);
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gboolean gst_wasapi_util_get_devices (GstMMDeviceEnumerator * enumerator,
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gboolean active,
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GList ** devices);
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gboolean gst_wasapi_util_get_device (GstMMDeviceEnumerator * enumerator,
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gint data_flow, gint role, const wchar_t * device_strid,
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IMMDevice ** ret_device);
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gboolean gst_wasapi_util_get_audio_client (GstElement * self,
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IMMDevice * device, IAudioClient ** ret_client);
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gboolean gst_wasapi_util_get_device_format (GstElement * element,
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gint device_mode, IMMDevice * device, IAudioClient * client,
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WAVEFORMATEX ** ret_format);
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gboolean gst_wasapi_util_get_render_client (GstElement * element,
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IAudioClient * client, IAudioRenderClient ** ret_render_client);
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gboolean gst_wasapi_util_get_capture_client (GstElement * element,
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IAudioClient * client, IAudioCaptureClient ** ret_capture_client);
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gboolean gst_wasapi_util_get_clock (GstElement * element,
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IAudioClient * client, IAudioClock ** ret_clock);
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gboolean gst_wasapi_util_parse_waveformatex (WAVEFORMATEXTENSIBLE * format,
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GstCaps * template_caps, GstCaps ** out_caps,
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GstAudioChannelPosition ** out_positions);
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void gst_wasapi_util_get_best_buffer_sizes (GstAudioRingBufferSpec * spec,
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gboolean exclusive, REFERENCE_TIME default_period,
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REFERENCE_TIME min_period, REFERENCE_TIME * ret_period,
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REFERENCE_TIME * ret_buffer_duration);
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gboolean gst_wasapi_util_initialize_audioclient (GstElement * element,
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GstAudioRingBufferSpec * spec, IAudioClient * client,
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WAVEFORMATEX * format, guint sharemode, gboolean low_latency,
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gboolean loopback, guint * ret_devicep_frames);
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gboolean gst_wasapi_util_initialize_audioclient3 (GstElement * element,
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GstAudioRingBufferSpec * spec, IAudioClient3 * client,
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WAVEFORMATEX * format, gboolean low_latency, gboolean loopback,
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guint * ret_devicep_frames);
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#endif /* __GST_WASAPI_UTIL_H__ */
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