mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 15:51:11 +00:00
139 lines
4.9 KiB
C
139 lines
4.9 KiB
C
/*
|
|
* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.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.
|
|
*/
|
|
|
|
#ifndef __GST_WASAPI_UTIL_H__
|
|
#define __GST_WASAPI_UTIL_H__
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/audio/audio.h>
|
|
#include <gst/audio/gstaudiosrc.h>
|
|
#include <gst/audio/gstaudiosink.h>
|
|
|
|
#include <mmdeviceapi.h>
|
|
#include <audioclient.h>
|
|
|
|
#include "gstaudioclient3.h"
|
|
#include "gstmmdeviceenumerator.h"
|
|
|
|
/* Static Caps shared between source, sink, and device provider */
|
|
#define GST_WASAPI_STATIC_CAPS "audio/x-raw, " \
|
|
"format = (string) " GST_AUDIO_FORMATS_ALL ", " \
|
|
"layout = (string) interleaved, " \
|
|
"rate = " GST_AUDIO_RATE_RANGE ", " \
|
|
"channels = " GST_AUDIO_CHANNELS_RANGE
|
|
|
|
/* Standard error path */
|
|
#define HR_FAILED_AND(hr,func,and) \
|
|
G_STMT_START { \
|
|
if (FAILED (hr)) { \
|
|
gchar *msg = gst_wasapi_util_hresult_to_string (hr); \
|
|
GST_ERROR_OBJECT (self, #func " failed (%x): %s", (guint) hr, msg); \
|
|
g_free (msg); \
|
|
and; \
|
|
} \
|
|
} G_STMT_END
|
|
|
|
#define HR_FAILED_RET(hr,func,ret) HR_FAILED_AND(hr,func,return ret)
|
|
|
|
#define HR_FAILED_GOTO(hr,func,where) HR_FAILED_AND(hr,func,res = FALSE; goto where)
|
|
|
|
#define HR_FAILED_ELEMENT_ERROR_AND(hr,func,el,and) \
|
|
G_STMT_START { \
|
|
if (FAILED (hr)) { \
|
|
gchar *msg = gst_wasapi_util_hresult_to_string (hr); \
|
|
GST_ERROR_OBJECT (el, #func " failed (%x): %s", (guint) hr, msg); \
|
|
if (GST_IS_AUDIO_SRC (el)) \
|
|
GST_ELEMENT_ERROR(el, RESOURCE, READ, \
|
|
(#func " failed (%x): %s", (guint) hr, msg), (NULL)); \
|
|
else \
|
|
GST_ELEMENT_ERROR(el, RESOURCE, WRITE, \
|
|
(#func " failed (%x): %s", (guint) hr, msg), (NULL)); \
|
|
g_free (msg); \
|
|
and; \
|
|
} \
|
|
} G_STMT_END
|
|
|
|
#define HR_FAILED_ELEMENT_ERROR_RET(hr,func,el,ret) \
|
|
HR_FAILED_ELEMENT_ERROR_AND(hr,func,el,return ret)
|
|
|
|
|
|
/* Device role enum property */
|
|
typedef enum
|
|
{
|
|
GST_WASAPI_DEVICE_ROLE_CONSOLE,
|
|
GST_WASAPI_DEVICE_ROLE_MULTIMEDIA,
|
|
GST_WASAPI_DEVICE_ROLE_COMMS
|
|
} GstWasapiDeviceRole;
|
|
#define GST_WASAPI_DEVICE_TYPE_ROLE (gst_wasapi_device_role_get_type())
|
|
GType gst_wasapi_device_role_get_type (void);
|
|
|
|
/* Utilities */
|
|
|
|
gboolean gst_wasapi_util_have_audioclient3 (void);
|
|
|
|
gint gst_wasapi_device_role_to_erole (gint role);
|
|
|
|
gint gst_wasapi_erole_to_device_role (gint erole);
|
|
|
|
gchar *gst_wasapi_util_hresult_to_string (HRESULT hr);
|
|
|
|
gboolean gst_wasapi_util_get_devices (GstMMDeviceEnumerator * enumerator,
|
|
gboolean active,
|
|
GList ** devices);
|
|
|
|
gboolean gst_wasapi_util_get_device (GstMMDeviceEnumerator * enumerator,
|
|
gint data_flow, gint role, const wchar_t * device_strid,
|
|
IMMDevice ** ret_device);
|
|
|
|
gboolean gst_wasapi_util_get_audio_client (GstElement * self,
|
|
IMMDevice * device, IAudioClient ** ret_client);
|
|
|
|
gboolean gst_wasapi_util_get_device_format (GstElement * element,
|
|
gint device_mode, IMMDevice * device, IAudioClient * client,
|
|
WAVEFORMATEX ** ret_format);
|
|
|
|
gboolean gst_wasapi_util_get_render_client (GstElement * element,
|
|
IAudioClient * client, IAudioRenderClient ** ret_render_client);
|
|
|
|
gboolean gst_wasapi_util_get_capture_client (GstElement * element,
|
|
IAudioClient * client, IAudioCaptureClient ** ret_capture_client);
|
|
|
|
gboolean gst_wasapi_util_get_clock (GstElement * element,
|
|
IAudioClient * client, IAudioClock ** ret_clock);
|
|
|
|
gboolean gst_wasapi_util_parse_waveformatex (WAVEFORMATEXTENSIBLE * format,
|
|
GstCaps * template_caps, GstCaps ** out_caps,
|
|
GstAudioChannelPosition ** out_positions);
|
|
|
|
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);
|
|
|
|
gboolean gst_wasapi_util_initialize_audioclient (GstElement * element,
|
|
GstAudioRingBufferSpec * spec, IAudioClient * client,
|
|
WAVEFORMATEX * format, guint sharemode, gboolean low_latency,
|
|
gboolean loopback, guint * ret_devicep_frames);
|
|
|
|
gboolean gst_wasapi_util_initialize_audioclient3 (GstElement * element,
|
|
GstAudioRingBufferSpec * spec, IAudioClient3 * client,
|
|
WAVEFORMATEX * format, gboolean low_latency, gboolean loopback,
|
|
guint * ret_devicep_frames);
|
|
|
|
#endif /* __GST_WASAPI_UTIL_H__ */
|