gstreamer/sys/wasapi/gstwasapiutil.h
Nirbheek Chauhan f62b7fd712 wasapi: Remove code that sets thread priority
This is now handled directly in gstaudiosrc/sink, and we were setting
it in the wrong thread anyway. prepare() is not the same thread as
sink_write() or src_read().
2018-09-11 01:00:21 +05:30

117 lines
4.1 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>
#ifdef _MSC_VER
#include <initguid.h>
#endif
#include <mmdeviceapi.h>
#include <audioclient.h>
#include "gstaudioclient3.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) \
do { \
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; \
} \
} while (0)
#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)
/* 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 (GstElement * element, gboolean active,
GList ** devices);
gboolean gst_wasapi_util_get_device_client (GstElement * element,
gint data_flow, gint role, const wchar_t * device_strid,
IMMDevice ** ret_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__ */