gstreamer/sys/wasapi/gstwasapiutil.h
Nirbheek Chauhan 4dbca8df09 wasapi: Try to use latency-time and buffer-time
So far, we have been completely discarding the values of latency-time
and buffer-time and trying to always open the device in the lowest
latency mode possible. However, sometimes this is a bad idea:

1. When we want to save power/CPU and don't want low latency
2. When the lowest latency setting causes glitches
3. Other audio-driver bugs

Now we will try to follow the user-set values of latency-time and
buffer-time in shared mode, and only latency-time in exclusive mode (we
have no control over the hardware buffer size, and there is no use in
setting GstAudioRingBuffer size to something larger).

The elements will still try to open the devices in the lowest latency
mode possible if you set the "low-latency" property to "true".

https://bugzilla.gnome.org/show_bug.cgi?id=793289
2018-02-08 14:29:58 +05:30

85 lines
3.1 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>
#include <mmdeviceapi.h>
#include <audioclient.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
/* 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 */
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,
gboolean capture, 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);
#endif /* __GST_WASAPI_UTIL_H__ */