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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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cb7958e710
Adding loopback capture mode for specified PID. Note that this feature requires Windows 10 build 20348 (Windows 11/Windows Server 2022 or later), and any process loopback related properties will not be exposed if OS does not support it. Example launch lines: * wasapi2src loopback-mode=include-process-tree loopback-target-pid=<PID> Captures audio generated by an application (specified by PID) and its child process * wasapi2src loopback-mode=exclude-process-tree loopback-target-pid=<PID> Captures desktop audio excluding PID and its child process Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1278 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3195>
610 lines
18 KiB
C
610 lines
18 KiB
C
/*
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* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
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* Copyright (C) 2018 Centricular Ltd.
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* Author: Nirbheek Chauhan <nirbheek@centricular.com>
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* Copyright (C) 2020 Seungha Yang <seungha@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-wasapi2src
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* @title: wasapi2src
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*
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* Provides audio capture from the Windows Audio Session API available with
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* Windows 10.
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*
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* ## Example pipelines
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* |[
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* gst-launch-1.0 -v wasapi2src ! fakesink
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* ]| Capture from the default audio device and render to fakesink.
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*
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* |[
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* gst-launch-1.0 -v wasapi2src low-latency=true ! fakesink
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* ]| Capture from the default audio device with the minimum possible latency and render to fakesink.
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#include "gstwasapi2src.h"
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#include "gstwasapi2util.h"
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#include "gstwasapi2ringbuffer.h"
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GST_DEBUG_CATEGORY_STATIC (gst_wasapi2_src_debug);
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#define GST_CAT_DEFAULT gst_wasapi2_src_debug
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (GST_WASAPI2_STATIC_CAPS));
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/**
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* GstWasapi2SrcLoopbackMode:
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*
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* Loopback capture mode
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*
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* Since: 1.22
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*/
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typedef enum
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{
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/**
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* GstWasapi2SrcLoopbackMode::default:
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*
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* Default loopback mode
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*
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* Since: 1.22
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*/
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GST_WASAPI2_SRC_LOOPBACK_DEFAULT,
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/**
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* GstWasapi2SrcLoopbackMode::include-process-tree:
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*
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* Captures only specified process and its child process
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*
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* Since: 1.22
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*/
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GST_WASAPI2_SRC_LOOPBACK_INCLUDE_PROCESS_TREE,
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/**
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* GstWasapi2SrcLoopbackMode::exclude-process-tree:
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*
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* Excludes specified process and its child process
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*
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* Since: 1.22
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*/
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GST_WASAPI2_SRC_LOOPBACK_EXCLUDE_PROCESS_TREE,
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} GstWasapi2SrcLoopbackMode;
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#define GST_TYPE_WASAPI2_SRC_LOOPBACK_MODE (gst_wasapi2_src_loopback_mode_get_type ())
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static GType
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gst_wasapi2_src_loopback_mode_get_type (void)
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{
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static GType loopback_type = 0;
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static const GEnumValue types[] = {
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{GST_WASAPI2_SRC_LOOPBACK_DEFAULT, "Default", "default"},
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{GST_WASAPI2_SRC_LOOPBACK_INCLUDE_PROCESS_TREE,
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"Include process and its child processes",
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"include-process-tree"},
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{GST_WASAPI2_SRC_LOOPBACK_EXCLUDE_PROCESS_TREE,
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"Exclude process and its child processes",
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"exclude-process-tree"},
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{0, NULL, NULL}
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};
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if (g_once_init_enter (&loopback_type)) {
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GType gtype = g_enum_register_static ("GstWasapi2SrcLoopbackMode", types);
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g_once_init_leave (&loopback_type, gtype);
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}
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return loopback_type;
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}
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#define DEFAULT_LOW_LATENCY FALSE
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#define DEFAULT_MUTE FALSE
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#define DEFAULT_VOLUME 1.0
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#define DEFAULT_LOOPBACK FALSE
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#define DEFAULT_LOOPBACK_MODE GST_WASAPI2_SRC_LOOPBACK_DEFAULT
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enum
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{
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PROP_0,
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PROP_DEVICE,
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PROP_LOW_LATENCY,
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PROP_MUTE,
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PROP_VOLUME,
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PROP_DISPATCHER,
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PROP_LOOPBACK,
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PROP_LOOPBACK_MODE,
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PROP_LOOPBACK_TARGET_PID,
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};
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struct _GstWasapi2Src
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{
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GstAudioBaseSrc parent;
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/* properties */
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gchar *device_id;
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gboolean low_latency;
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gboolean mute;
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gdouble volume;
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gpointer dispatcher;
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gboolean loopback;
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GstWasapi2SrcLoopbackMode loopback_mode;
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guint loopback_pid;
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gboolean mute_changed;
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gboolean volume_changed;
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};
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static void gst_wasapi2_src_finalize (GObject * object);
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static void gst_wasapi2_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_wasapi2_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn gst_wasapi2_src_change_state (GstElement *
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element, GstStateChange transition);
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static GstCaps *gst_wasapi2_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter);
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static GstAudioRingBuffer *gst_wasapi2_src_create_ringbuffer (GstAudioBaseSrc *
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src);
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static void gst_wasapi2_src_set_mute (GstWasapi2Src * self, gboolean mute);
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static gboolean gst_wasapi2_src_get_mute (GstWasapi2Src * self);
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static void gst_wasapi2_src_set_volume (GstWasapi2Src * self, gdouble volume);
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static gdouble gst_wasapi2_src_get_volume (GstWasapi2Src * self);
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#define gst_wasapi2_src_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstWasapi2Src, gst_wasapi2_src,
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GST_TYPE_AUDIO_BASE_SRC,
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G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL));
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static void
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gst_wasapi2_src_class_init (GstWasapi2SrcClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstBaseSrcClass *basesrc_class = GST_BASE_SRC_CLASS (klass);
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GstAudioBaseSrcClass *audiobasesrc_class = GST_AUDIO_BASE_SRC_CLASS (klass);
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gobject_class->finalize = gst_wasapi2_src_finalize;
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gobject_class->set_property = gst_wasapi2_src_set_property;
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gobject_class->get_property = gst_wasapi2_src_get_property;
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g_object_class_install_property (gobject_class, PROP_DEVICE,
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g_param_spec_string ("device", "Device",
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"WASAPI playback device as a GUID string",
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NULL, GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_LOW_LATENCY,
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g_param_spec_boolean ("low-latency", "Low latency",
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"Optimize all settings for lowest latency. Always safe to enable.",
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DEFAULT_LOW_LATENCY, GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_MUTE,
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g_param_spec_boolean ("mute", "Mute", "Mute state of this stream",
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DEFAULT_MUTE, GST_PARAM_MUTABLE_PLAYING | G_PARAM_READWRITE |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_VOLUME,
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g_param_spec_double ("volume", "Volume", "Volume of this stream",
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0.0, 1.0, DEFAULT_VOLUME,
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GST_PARAM_MUTABLE_PLAYING | G_PARAM_READWRITE |
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G_PARAM_STATIC_STRINGS));
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/**
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* GstWasapi2Src:dispatcher:
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*
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* ICoreDispatcher COM object used for activating device from UI thread.
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*
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* Since: 1.18
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*/
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g_object_class_install_property (gobject_class, PROP_DISPATCHER,
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g_param_spec_pointer ("dispatcher", "Dispatcher",
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"ICoreDispatcher COM object to use. In order for application to ask "
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"permission of audio device, device activation should be running "
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"on UI thread via ICoreDispatcher. This element will increase "
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"the reference count of given ICoreDispatcher and release it after "
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"use. Therefore, caller does not need to consider additional "
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"reference count management",
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GST_PARAM_MUTABLE_READY | G_PARAM_WRITABLE | G_PARAM_STATIC_STRINGS));
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/**
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* GstWasapi2Src:loopback:
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*
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* Open render device for loopback recording
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*
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* Since: 1.20
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*/
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g_object_class_install_property (gobject_class, PROP_LOOPBACK,
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g_param_spec_boolean ("loopback", "Loopback recording",
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"Open render device for loopback recording", DEFAULT_LOOPBACK,
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GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
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G_PARAM_STATIC_STRINGS));
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if (gst_wasapi2_can_process_loopback ()) {
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/**
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* GstWasapi2Src:loopback-mode:
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*
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* Loopback mode. "target-process-id" must be specified in case of
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* process loopback modes.
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*
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* This feature requires "Windows 10 build 20348"
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*
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* Since: 1.22
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*/
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g_object_class_install_property (gobject_class, PROP_LOOPBACK_MODE,
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g_param_spec_enum ("loopback-mode", "Loopback Mode",
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"Loopback mode to use", GST_TYPE_WASAPI2_SRC_LOOPBACK_MODE,
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DEFAULT_LOOPBACK_MODE,
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GST_PARAM_CONDITIONALLY_AVAILABLE | GST_PARAM_MUTABLE_READY |
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstWasapi2Src:loopback-target-pid:
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*
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* Target process id to be recorded or excluded depending on loopback mode
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*
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* This feature requires "Windows 10 build 20348"
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*
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* Since: 1.22
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*/
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g_object_class_install_property (gobject_class, PROP_LOOPBACK_TARGET_PID,
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g_param_spec_uint ("loopback-target-pid", "Loopback Target PID",
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"Process ID to be recorded or excluded for process loopback mode",
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0, G_MAXUINT32, 0,
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GST_PARAM_CONDITIONALLY_AVAILABLE | GST_PARAM_MUTABLE_READY |
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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gst_element_class_add_static_pad_template (element_class, &src_template);
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gst_element_class_set_static_metadata (element_class, "Wasapi2Src",
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"Source/Audio/Hardware",
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"Stream audio from an audio capture device through WASAPI",
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"Nirbheek Chauhan <nirbheek@centricular.com>, "
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"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>, "
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"Seungha Yang <seungha@centricular.com>");
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element_class->change_state =
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GST_DEBUG_FUNCPTR (gst_wasapi2_src_change_state);
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basesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi2_src_get_caps);
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audiobasesrc_class->create_ringbuffer =
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GST_DEBUG_FUNCPTR (gst_wasapi2_src_create_ringbuffer);
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GST_DEBUG_CATEGORY_INIT (gst_wasapi2_src_debug, "wasapi2src",
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0, "Windows audio session API source");
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if (gst_wasapi2_can_process_loopback ())
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gst_type_mark_as_plugin_api (GST_TYPE_WASAPI2_SRC_LOOPBACK_MODE, 0);
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}
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static void
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gst_wasapi2_src_init (GstWasapi2Src * self)
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{
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self->mute = DEFAULT_MUTE;
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self->volume = DEFAULT_VOLUME;
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self->low_latency = DEFAULT_LOW_LATENCY;
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self->loopback = DEFAULT_LOOPBACK;
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}
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static void
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gst_wasapi2_src_finalize (GObject * object)
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{
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GstWasapi2Src *self = GST_WASAPI2_SRC (object);
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g_free (self->device_id);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_wasapi2_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstWasapi2Src *self = GST_WASAPI2_SRC (object);
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switch (prop_id) {
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case PROP_DEVICE:
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g_free (self->device_id);
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self->device_id = g_value_dup_string (value);
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break;
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case PROP_LOW_LATENCY:
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self->low_latency = g_value_get_boolean (value);
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break;
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case PROP_MUTE:
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gst_wasapi2_src_set_mute (self, g_value_get_boolean (value));
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break;
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case PROP_VOLUME:
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gst_wasapi2_src_set_volume (self, g_value_get_double (value));
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break;
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case PROP_DISPATCHER:
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self->dispatcher = g_value_get_pointer (value);
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break;
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case PROP_LOOPBACK:
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self->loopback = g_value_get_boolean (value);
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break;
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case PROP_LOOPBACK_MODE:
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self->loopback_mode = g_value_get_enum (value);
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break;
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case PROP_LOOPBACK_TARGET_PID:
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self->loopback_pid = g_value_get_uint (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_wasapi2_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstWasapi2Src *self = GST_WASAPI2_SRC (object);
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switch (prop_id) {
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case PROP_DEVICE:
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g_value_set_string (value, self->device_id);
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break;
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case PROP_LOW_LATENCY:
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g_value_set_boolean (value, self->low_latency);
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break;
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case PROP_MUTE:
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g_value_set_boolean (value, gst_wasapi2_src_get_mute (self));
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break;
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case PROP_VOLUME:
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g_value_set_double (value, gst_wasapi2_src_get_volume (self));
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break;
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case PROP_LOOPBACK:
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g_value_set_boolean (value, self->loopback);
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break;
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case PROP_LOOPBACK_MODE:
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g_value_set_enum (value, self->loopback_mode);
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break;
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case PROP_LOOPBACK_TARGET_PID:
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g_value_set_uint (value, self->loopback_pid);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstStateChangeReturn
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gst_wasapi2_src_change_state (GstElement * element, GstStateChange transition)
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{
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GstWasapi2Src *self = GST_WASAPI2_SRC (element);
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GstAudioBaseSrc *asrc = GST_AUDIO_BASE_SRC_CAST (element);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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/* If we have pending volume/mute values to set, do here */
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GST_OBJECT_LOCK (self);
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if (asrc->ringbuffer) {
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GstWasapi2RingBuffer *ringbuffer =
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GST_WASAPI2_RING_BUFFER (asrc->ringbuffer);
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if (self->volume_changed) {
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gst_wasapi2_ring_buffer_set_volume (ringbuffer, self->volume);
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self->volume_changed = FALSE;
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}
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if (self->mute_changed) {
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gst_wasapi2_ring_buffer_set_mute (ringbuffer, self->mute);
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self->mute_changed = FALSE;
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}
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}
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GST_OBJECT_UNLOCK (self);
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break;
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default:
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break;
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}
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return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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}
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static GstCaps *
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gst_wasapi2_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
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{
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GstAudioBaseSrc *asrc = GST_AUDIO_BASE_SRC_CAST (bsrc);
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GstCaps *caps = NULL;
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GST_OBJECT_LOCK (bsrc);
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if (asrc->ringbuffer) {
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GstWasapi2RingBuffer *ringbuffer =
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GST_WASAPI2_RING_BUFFER (asrc->ringbuffer);
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gst_object_ref (ringbuffer);
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GST_OBJECT_UNLOCK (bsrc);
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/* Get caps might be able to block if device is not activated yet */
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caps = gst_wasapi2_ring_buffer_get_caps (ringbuffer);
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gst_object_unref (ringbuffer);
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} else {
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GST_OBJECT_UNLOCK (bsrc);
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}
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if (!caps)
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caps = gst_pad_get_pad_template_caps (bsrc->srcpad);
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if (filter) {
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GstCaps *filtered =
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gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (caps);
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caps = filtered;
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}
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GST_DEBUG_OBJECT (bsrc, "returning caps %" GST_PTR_FORMAT, caps);
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return caps;
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}
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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_pid) {
|
|
if (self->loopback_mode == GST_WASAPI2_SRC_LOOPBACK_INCLUDE_PROCESS_TREE) {
|
|
device_class =
|
|
GST_WASAPI2_CLIENT_DEVICE_CLASS_INCLUDE_PROCESS_LOOPBACK_CAPTURE;
|
|
} else if (self->loopback_mode ==
|
|
GST_WASAPI2_SRC_LOOPBACK_EXCLUDE_PROCESS_TREE) {
|
|
device_class =
|
|
GST_WASAPI2_CLIENT_DEVICE_CLASS_EXCLUDE_PROCESS_LOOPBACK_CAPTURE;
|
|
}
|
|
} else if (self->loopback) {
|
|
device_class = GST_WASAPI2_CLIENT_DEVICE_CLASS_LOOPBACK_CAPTURE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (self, "Device class %d", device_class);
|
|
|
|
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,
|
|
self->loopback_pid);
|
|
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;
|
|
}
|