gstreamer/subprojects/gst-plugins-bad/sys/wasapi2/gstwasapi2sink.c

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/*
* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* Copyright (C) 2013 Collabora Ltd.
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
* Copyright (C) 2018 Centricular Ltd.
* Author: Nirbheek Chauhan <nirbheek@centricular.com>
* Copyright (C) 2020 Seungha Yang <seungha@centricular.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.
*/
/**
* SECTION:element-wasapi2sink
* @title: wasapi2sink
*
* Provides audio playback using the Windows Audio Session API available with
* Windows 10.
*
* ## Example pipelines
* |[
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
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* gst-launch-1.0 -v audiotestsink samplesperbuffer=160 ! wasapi2sink
* ]| Generate 20 ms buffers and render to the default audio device.
*
* |[
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
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* gst-launch-1.0 -v audiotestsink samplesperbuffer=160 ! wasapi2sink low-latency=true
* ]| Same as above, but with the minimum possible latency
*
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include "gstwasapi2sink.h"
#include "gstwasapi2util.h"
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
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#include "gstwasapi2ringbuffer.h"
GST_DEBUG_CATEGORY_STATIC (gst_wasapi2_sink_debug);
#define GST_CAT_DEFAULT gst_wasapi2_sink_debug
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_WASAPI2_STATIC_CAPS));
#define DEFAULT_LOW_LATENCY FALSE
#define DEFAULT_MUTE FALSE
#define DEFAULT_VOLUME 1.0
enum
{
PROP_0,
PROP_DEVICE,
PROP_LOW_LATENCY,
PROP_MUTE,
PROP_VOLUME,
PROP_DISPATCHER,
};
struct _GstWasapi2Sink
{
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
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GstAudioBaseSink parent;
/* properties */
gchar *device_id;
gboolean low_latency;
gboolean mute;
gdouble volume;
gpointer dispatcher;
gboolean mute_changed;
gboolean volume_changed;
};
static void gst_wasapi2_sink_finalize (GObject * object);
static void gst_wasapi2_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_wasapi2_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
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static GstStateChangeReturn gst_wasapi2_sink_change_state (GstElement *
element, GstStateChange transition);
static GstCaps *gst_wasapi2_sink_get_caps (GstBaseSink * bsink,
GstCaps * filter);
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
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static GstAudioRingBuffer *gst_wasapi2_sink_create_ringbuffer (GstAudioBaseSink
* sink);
static void gst_wasapi2_sink_set_mute (GstWasapi2Sink * self, gboolean mute);
static gboolean gst_wasapi2_sink_get_mute (GstWasapi2Sink * self);
static void gst_wasapi2_sink_set_volume (GstWasapi2Sink * self, gdouble volume);
static gdouble gst_wasapi2_sink_get_volume (GstWasapi2Sink * self);
#define gst_wasapi2_sink_parent_class parent_class
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
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G_DEFINE_TYPE_WITH_CODE (GstWasapi2Sink, gst_wasapi2_sink,
GST_TYPE_AUDIO_BASE_SINK,
G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL));
static void
gst_wasapi2_sink_class_init (GstWasapi2SinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseSinkClass *basesink_class = GST_BASE_SINK_CLASS (klass);
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
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GstAudioBaseSinkClass *audiobasesink_class =
GST_AUDIO_BASE_SINK_CLASS (klass);
gobject_class->finalize = gst_wasapi2_sink_finalize;
gobject_class->set_property = gst_wasapi2_sink_set_property;
gobject_class->get_property = gst_wasapi2_sink_get_property;
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device",
"WASAPI playback device as a GUID string",
NULL, GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_LOW_LATENCY,
g_param_spec_boolean ("low-latency", "Low latency",
"Optimize all settings for lowest latency. Always safe to enable.",
DEFAULT_LOW_LATENCY, GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MUTE,
g_param_spec_boolean ("mute", "Mute", "Mute state of this stream",
DEFAULT_MUTE, GST_PARAM_MUTABLE_PLAYING | G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_VOLUME,
g_param_spec_double ("volume", "Volume", "Volume of this stream",
0.0, 1.0, DEFAULT_VOLUME,
GST_PARAM_MUTABLE_PLAYING | G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS));
/**
* GstWasapi2Sink:dispatcher:
*
* ICoreDispatcher COM object used for activating device from UI thread.
*
* Since: 1.18
*/
g_object_class_install_property (gobject_class, PROP_DISPATCHER,
g_param_spec_pointer ("dispatcher", "Dispatcher",
"ICoreDispatcher COM object to use. In order for application to ask "
"permission of audio device, device activation should be running "
"on UI thread via ICoreDispatcher. This element will increase "
"the reference count of given ICoreDispatcher and release it after "
"use. Therefore, caller does not need to consider additional "
"reference count management",
GST_PARAM_MUTABLE_READY | G_PARAM_WRITABLE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (element_class, &sink_template);
gst_element_class_set_static_metadata (element_class, "Wasapi2Sink",
"Sink/Audio/Hardware",
"Stream audio to an audio capture device through WASAPI",
"Nirbheek Chauhan <nirbheek@centricular.com>, "
"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>, "
"Seungha Yang <seungha@centricular.com>");
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
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element_class->change_state =
GST_DEBUG_FUNCPTR (gst_wasapi2_sink_change_state);
basesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_get_caps);
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
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audiobasesink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_wasapi2_sink_create_ringbuffer);
GST_DEBUG_CATEGORY_INIT (gst_wasapi2_sink_debug, "wasapi2sink",
0, "Windows audio session API sink");
}
static void
gst_wasapi2_sink_init (GstWasapi2Sink * self)
{
self->low_latency = DEFAULT_LOW_LATENCY;
self->mute = DEFAULT_MUTE;
self->volume = DEFAULT_VOLUME;
}
static void
gst_wasapi2_sink_finalize (GObject * object)
{
GstWasapi2Sink *self = GST_WASAPI2_SINK (object);
g_free (self->device_id);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_wasapi2_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWasapi2Sink *self = GST_WASAPI2_SINK (object);
switch (prop_id) {
case PROP_DEVICE:
g_free (self->device_id);
self->device_id = g_value_dup_string (value);
break;
case PROP_LOW_LATENCY:
self->low_latency = g_value_get_boolean (value);
break;
case PROP_MUTE:
gst_wasapi2_sink_set_mute (self, g_value_get_boolean (value));
break;
case PROP_VOLUME:
gst_wasapi2_sink_set_volume (self, g_value_get_double (value));
break;
case PROP_DISPATCHER:
self->dispatcher = g_value_get_pointer (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_wasapi2_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWasapi2Sink *self = GST_WASAPI2_SINK (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, self->device_id);
break;
case PROP_LOW_LATENCY:
g_value_set_boolean (value, self->low_latency);
break;
case PROP_MUTE:
g_value_set_boolean (value, gst_wasapi2_sink_get_mute (self));
break;
case PROP_VOLUME:
g_value_set_double (value, gst_wasapi2_sink_get_volume (self));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
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static GstStateChangeReturn
gst_wasapi2_sink_change_state (GstElement * element, GstStateChange transition)
{
GstWasapi2Sink *self = GST_WASAPI2_SINK (element);
GstAudioBaseSink *asink = GST_AUDIO_BASE_SINK_CAST (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* If we have pending volume/mute values to set, do here */
GST_OBJECT_LOCK (self);
if (asink->ringbuffer) {
GstWasapi2RingBuffer *ringbuffer =
GST_WASAPI2_RING_BUFFER (asink->ringbuffer);
if (self->volume_changed) {
gst_wasapi2_ring_buffer_set_volume (ringbuffer, self->volume);
self->volume_changed = FALSE;
}
if (self->mute_changed) {
gst_wasapi2_ring_buffer_set_mute (ringbuffer, self->mute);
self->mute_changed = FALSE;
}
}
GST_OBJECT_UNLOCK (self);
break;
default:
break;
}
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
}
static GstCaps *
gst_wasapi2_sink_get_caps (GstBaseSink * bsink, GstCaps * filter)
{
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
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GstAudioBaseSink *asink = GST_AUDIO_BASE_SINK_CAST (bsink);
GstCaps *caps = NULL;
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GST_OBJECT_LOCK (bsink);
if (asink->ringbuffer) {
GstWasapi2RingBuffer *ringbuffer =
GST_WASAPI2_RING_BUFFER (asink->ringbuffer);
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
gst_object_ref (ringbuffer);
GST_OBJECT_UNLOCK (bsink);
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
/* Get caps might be able to block if device is not activated yet */
caps = gst_wasapi2_ring_buffer_get_caps (ringbuffer);
gst_object_unref (ringbuffer);
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
} else {
GST_OBJECT_UNLOCK (bsink);
}
if (!caps)
caps = gst_pad_get_pad_template_caps (bsink->sinkpad);
if (filter) {
GstCaps *filtered =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = filtered;
}
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GST_DEBUG_OBJECT (bsink, "returning caps %" GST_PTR_FORMAT, caps);
return caps;
}
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
static GstAudioRingBuffer *
gst_wasapi2_sink_create_ringbuffer (GstAudioBaseSink * sink)
{
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GstWasapi2Sink *self = GST_WASAPI2_SINK (sink);
GstAudioRingBuffer *ringbuffer;
gchar *name;
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
name = g_strdup_printf ("%s-ringbuffer", GST_OBJECT_NAME (sink));
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
ringbuffer =
gst_wasapi2_ring_buffer_new (GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER,
self->low_latency, self->device_id, self->dispatcher, name);
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
g_free (name);
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
return ringbuffer;
}
static void
gst_wasapi2_sink_set_mute (GstWasapi2Sink * self, gboolean mute)
{
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GstAudioBaseSink *bsink = GST_AUDIO_BASE_SINK_CAST (self);
HRESULT hr;
GST_OBJECT_LOCK (self);
self->mute = mute;
self->mute_changed = TRUE;
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
if (bsink->ringbuffer) {
GstWasapi2RingBuffer *ringbuffer =
GST_WASAPI2_RING_BUFFER (bsink->ringbuffer);
hr = gst_wasapi2_ring_buffer_set_mute (ringbuffer, mute);
if (FAILED (hr)) {
GST_INFO_OBJECT (self, "Couldn't set mute");
} else {
self->mute_changed = FALSE;
}
}
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GST_OBJECT_UNLOCK (self);
}
static gboolean
gst_wasapi2_sink_get_mute (GstWasapi2Sink * self)
{
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GstAudioBaseSink *bsink = GST_AUDIO_BASE_SINK_CAST (self);
gboolean mute;
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
HRESULT hr;
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GST_OBJECT_LOCK (self);
mute = self->mute;
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
if (bsink->ringbuffer) {
GstWasapi2RingBuffer *ringbuffer =
GST_WASAPI2_RING_BUFFER (bsink->ringbuffer);
hr = gst_wasapi2_ring_buffer_get_mute (ringbuffer, &mute);
if (FAILED (hr)) {
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GST_INFO_OBJECT (self, "Couldn't get mute");
} else {
self->mute = mute;
}
}
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GST_OBJECT_UNLOCK (self);
return mute;
}
static void
gst_wasapi2_sink_set_volume (GstWasapi2Sink * self, gdouble volume)
{
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GstAudioBaseSink *bsink = GST_AUDIO_BASE_SINK_CAST (self);
HRESULT hr;
GST_OBJECT_LOCK (self);
self->volume = volume;
/* clip volume value */
self->volume = MAX (0.0, self->volume);
self->volume = MIN (1.0, self->volume);
self->volume_changed = TRUE;
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
if (bsink->ringbuffer) {
GstWasapi2RingBuffer *ringbuffer =
GST_WASAPI2_RING_BUFFER (bsink->ringbuffer);
hr = gst_wasapi2_ring_buffer_set_volume (ringbuffer, (gfloat) self->volume);
if (FAILED (hr)) {
GST_INFO_OBJECT (self, "Couldn't set volume");
} else {
self->volume_changed = FALSE;
}
}
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GST_OBJECT_UNLOCK (self);
}
static gdouble
gst_wasapi2_sink_get_volume (GstWasapi2Sink * self)
{
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GstAudioBaseSink *bsink = GST_AUDIO_BASE_SINK_CAST (self);
gfloat volume;
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
HRESULT hr;
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GST_OBJECT_LOCK (self);
volume = (gfloat) self->volume;
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
if (bsink->ringbuffer) {
GstWasapi2RingBuffer *ringbuffer =
GST_WASAPI2_RING_BUFFER (bsink->ringbuffer);
hr = gst_wasapi2_ring_buffer_get_volume (ringbuffer, &volume);
if (FAILED (hr)) {
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GST_INFO_OBJECT (self, "Couldn't set volume");
} else {
self->volume = volume;
}
}
wasapi2: Rewrite plugin and implement audioringbuffer subclass ... based on MediaFoundation work queue API. By this commit, wasapi2 plugin will make use of pull mode scheduling with audioringbuffer subclass. There are several drawbacks of audiosrc/audiosink subclassing (not audiobasesrc/audiobasesink) for WASAPI API, which are: * audiosrc/audiosink classes try to set high priority to read/write thread via MMCSS (Multimedia Class Scheduler Service) but it's not allowed in case of UWP application. In order to use MMCSS in UWP, application should use MediaFoundation work queue indirectly. Since audiosrc/audiosink scheduling model is not compatible with MediaFoundation's work queue model, audioringbuffer subclassing is required. * WASAPI capture device might report larger packet size than expected (i.e., larger frames we can read than expected frame size per period). Meanwhile, in any case, application should drain all packets at that moment. In order to handle the case, wasapi/wasapi2 plugins were making use of GstAdapter which is obviously sub-optimal because it requires additional memory allocation and copy. By implementing audioringbuffer subclassing, we can avoid such inefficiency. In this commit, all the device read/write operations will be moved to newly implemented wasapi2ringbuffer class and existing wasapi2client class will take care of device enumeration and activation parts only. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
2021-05-10 11:45:28 +00:00
GST_OBJECT_UNLOCK (self);
volume = MAX (0.0, volume);
volume = MIN (1.0, volume);
return volume;
}