gstreamer/subprojects/gst-plugins-bad/sys/wasapi2/gstwasapi2src.c

481 lines
14 KiB
C

/*
* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* 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-wasapi2src
* @title: wasapi2src
*
* Provides audio capture from the Windows Audio Session API available with
* Windows 10.
*
* ## Example pipelines
* |[
* gst-launch-1.0 -v wasapi2src ! fakesink
* ]| Capture from the default audio device and render to fakesink.
*
* |[
* gst-launch-1.0 -v wasapi2src low-latency=true ! fakesink
* ]| Capture from the default audio device with the minimum possible latency and render to fakesink.
*
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include "gstwasapi2src.h"
#include "gstwasapi2util.h"
#include "gstwasapi2ringbuffer.h"
GST_DEBUG_CATEGORY_STATIC (gst_wasapi2_src_debug);
#define GST_CAT_DEFAULT gst_wasapi2_src_debug
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_WASAPI2_STATIC_CAPS));
#define DEFAULT_LOW_LATENCY FALSE
#define DEFAULT_MUTE FALSE
#define DEFAULT_VOLUME 1.0
#define DEFAULT_LOOPBACK FALSE
enum
{
PROP_0,
PROP_DEVICE,
PROP_LOW_LATENCY,
PROP_MUTE,
PROP_VOLUME,
PROP_DISPATCHER,
PROP_LOOPBACK,
};
struct _GstWasapi2Src
{
GstAudioBaseSrc parent;
/* properties */
gchar *device_id;
gboolean low_latency;
gboolean mute;
gdouble volume;
gpointer dispatcher;
gboolean loopback;
gboolean mute_changed;
gboolean volume_changed;
};
static void gst_wasapi2_src_finalize (GObject * object);
static void gst_wasapi2_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_wasapi2_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_wasapi2_src_change_state (GstElement *
element, GstStateChange transition);
static GstCaps *gst_wasapi2_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter);
static GstAudioRingBuffer *gst_wasapi2_src_create_ringbuffer (GstAudioBaseSrc *
src);
static void gst_wasapi2_src_set_mute (GstWasapi2Src * self, gboolean mute);
static gboolean gst_wasapi2_src_get_mute (GstWasapi2Src * self);
static void gst_wasapi2_src_set_volume (GstWasapi2Src * self, gdouble volume);
static gdouble gst_wasapi2_src_get_volume (GstWasapi2Src * self);
#define gst_wasapi2_src_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstWasapi2Src, gst_wasapi2_src,
GST_TYPE_AUDIO_BASE_SRC,
G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL));
static void
gst_wasapi2_src_class_init (GstWasapi2SrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseSrcClass *basesrc_class = GST_BASE_SRC_CLASS (klass);
GstAudioBaseSrcClass *audiobasesrc_class = GST_AUDIO_BASE_SRC_CLASS (klass);
gobject_class->finalize = gst_wasapi2_src_finalize;
gobject_class->set_property = gst_wasapi2_src_set_property;
gobject_class->get_property = gst_wasapi2_src_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));
/**
* GstWasapi2Src: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));
/**
* GstWasapi2Src:loopback:
*
* Open render device for loopback recording
*
* Since: 1.20
*/
g_object_class_install_property (gobject_class, PROP_LOOPBACK,
g_param_spec_boolean ("loopback", "Loopback recording",
"Open render device for loopback recording", DEFAULT_LOOPBACK,
GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (element_class, &src_template);
gst_element_class_set_static_metadata (element_class, "Wasapi2Src",
"Source/Audio/Hardware",
"Stream audio from 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>");
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_wasapi2_src_change_state);
basesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi2_src_get_caps);
audiobasesrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_wasapi2_src_create_ringbuffer);
GST_DEBUG_CATEGORY_INIT (gst_wasapi2_src_debug, "wasapi2src",
0, "Windows audio session API source");
}
static void
gst_wasapi2_src_init (GstWasapi2Src * self)
{
self->mute = DEFAULT_MUTE;
self->volume = DEFAULT_VOLUME;
self->low_latency = DEFAULT_LOW_LATENCY;
self->loopback = DEFAULT_LOOPBACK;
}
static void
gst_wasapi2_src_finalize (GObject * object)
{
GstWasapi2Src *self = GST_WASAPI2_SRC (object);
g_free (self->device_id);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_wasapi2_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWasapi2Src *self = GST_WASAPI2_SRC (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_src_set_mute (self, g_value_get_boolean (value));
break;
case PROP_VOLUME:
gst_wasapi2_src_set_volume (self, g_value_get_double (value));
break;
case PROP_DISPATCHER:
self->dispatcher = g_value_get_pointer (value);
break;
case PROP_LOOPBACK:
self->loopback = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_wasapi2_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWasapi2Src *self = GST_WASAPI2_SRC (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_src_get_mute (self));
break;
case PROP_VOLUME:
g_value_set_double (value, gst_wasapi2_src_get_volume (self));
break;
case PROP_LOOPBACK:
g_value_set_boolean (value, self->loopback);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_wasapi2_src_change_state (GstElement * element, GstStateChange transition)
{
GstWasapi2Src *self = GST_WASAPI2_SRC (element);
GstAudioBaseSrc *asrc = GST_AUDIO_BASE_SRC_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 (asrc->ringbuffer) {
GstWasapi2RingBuffer *ringbuffer =
GST_WASAPI2_RING_BUFFER (asrc->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_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
{
GstAudioBaseSrc *asrc = GST_AUDIO_BASE_SRC_CAST (bsrc);
GstCaps *caps = NULL;
GST_OBJECT_LOCK (bsrc);
if (asrc->ringbuffer) {
GstWasapi2RingBuffer *ringbuffer =
GST_WASAPI2_RING_BUFFER (asrc->ringbuffer);
gst_object_ref (ringbuffer);
GST_OBJECT_UNLOCK (bsrc);
/* 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);
} else {
GST_OBJECT_UNLOCK (bsrc);
}
if (!caps)
caps = gst_pad_get_pad_template_caps (bsrc->srcpad);
if (filter) {
GstCaps *filtered =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = filtered;
}
GST_DEBUG_OBJECT (bsrc, "returning caps %" GST_PTR_FORMAT, caps);
return caps;
}
static GstAudioRingBuffer *
gst_wasapi2_src_create_ringbuffer (GstAudioBaseSrc * src)
{
GstWasapi2Src *self = GST_WASAPI2_SRC (src);
GstAudioRingBuffer *ringbuffer;
gchar *name;
GstWasapi2ClientDeviceClass device_class =
GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE;
if (self->loopback)
device_class = GST_WASAPI2_CLIENT_DEVICE_CLASS_LOOPBACK_CAPTURE;
name = g_strdup_printf ("%s-ringbuffer", GST_OBJECT_NAME (src));
ringbuffer =
gst_wasapi2_ring_buffer_new (device_class,
self->low_latency, self->device_id, self->dispatcher, name);
g_free (name);
return ringbuffer;
}
static void
gst_wasapi2_src_set_mute (GstWasapi2Src * self, gboolean mute)
{
GstAudioBaseSrc *bsrc = GST_AUDIO_BASE_SRC_CAST (self);
HRESULT hr;
GST_OBJECT_LOCK (self);
self->mute = mute;
self->mute_changed = TRUE;
if (bsrc->ringbuffer) {
GstWasapi2RingBuffer *ringbuffer =
GST_WASAPI2_RING_BUFFER (bsrc->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;
}
}
GST_OBJECT_UNLOCK (self);
}
static gboolean
gst_wasapi2_src_get_mute (GstWasapi2Src * self)
{
GstAudioBaseSrc *bsrc = GST_AUDIO_BASE_SRC_CAST (self);
gboolean mute;
HRESULT hr;
GST_OBJECT_LOCK (self);
mute = self->mute;
if (bsrc->ringbuffer) {
GstWasapi2RingBuffer *ringbuffer =
GST_WASAPI2_RING_BUFFER (bsrc->ringbuffer);
hr = gst_wasapi2_ring_buffer_get_mute (ringbuffer, &mute);
if (FAILED (hr)) {
GST_INFO_OBJECT (self, "Couldn't get mute");
} else {
self->mute = mute;
}
}
GST_OBJECT_UNLOCK (self);
return mute;
}
static void
gst_wasapi2_src_set_volume (GstWasapi2Src * self, gdouble volume)
{
GstAudioBaseSrc *bsrc = GST_AUDIO_BASE_SRC_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;
if (bsrc->ringbuffer) {
GstWasapi2RingBuffer *ringbuffer =
GST_WASAPI2_RING_BUFFER (bsrc->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;
}
}
GST_OBJECT_UNLOCK (self);
}
static gdouble
gst_wasapi2_src_get_volume (GstWasapi2Src * self)
{
GstAudioBaseSrc *bsrc = GST_AUDIO_BASE_SRC_CAST (self);
gfloat volume;
HRESULT hr;
GST_OBJECT_LOCK (self);
volume = (gfloat) self->volume;
if (bsrc->ringbuffer) {
GstWasapi2RingBuffer *ringbuffer =
GST_WASAPI2_RING_BUFFER (bsrc->ringbuffer);
hr = gst_wasapi2_ring_buffer_get_volume (ringbuffer, &volume);
if (FAILED (hr)) {
GST_INFO_OBJECT (self, "Couldn't set volume");
} else {
self->volume = volume;
}
}
GST_OBJECT_UNLOCK (self);
volume = MAX (0.0, volume);
volume = MIN (1.0, volume);
return volume;
}