mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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Disallow getting dispatcher pointer, since it doesn't seem to be useful and might not be safe. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1480>
580 lines
16 KiB
C
580 lines
16 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 ! fakesrc
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* ]| Capture from the default audio device and render to fakesrc.
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*
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* |[
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* gst-launch-1.0 -v wasapi2src low-latency=true ! fakesrc
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* ]| Capture from the default audio device with the minimum possible latency and render to fakesrc.
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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 "gstwasapi2client.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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#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 GST_WASAPI2_SRC_LOCK(s) g_mutex_lock(&(s)->lock)
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#define GST_WASAPI2_SRC_UNLOCK(s) g_mutex_unlock(&(s)->lock)
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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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};
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struct _GstWasapi2Src
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{
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GstAudioSrc parent;
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GstWasapi2Client *client;
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GstCaps *cached_caps;
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gboolean started;
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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 mute_changed;
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gboolean volume_changed;
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/* to protect audioclient from set/get property */
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GMutex lock;
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};
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static void gst_wasapi2_src_dispose (GObject * object);
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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 GstCaps *gst_wasapi2_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter);
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static gboolean gst_wasapi2_src_open (GstAudioSrc * asrc);
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static gboolean gst_wasapi2_src_close (GstAudioSrc * asrc);
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static gboolean gst_wasapi2_src_prepare (GstAudioSrc * asrc,
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GstAudioRingBufferSpec * spec);
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static gboolean gst_wasapi2_src_unprepare (GstAudioSrc * asrc);
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static guint gst_wasapi2_src_read (GstAudioSrc * asrc, gpointer data,
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guint length, GstClockTime * timestamp);
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static guint gst_wasapi2_src_delay (GstAudioSrc * asrc);
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static void gst_wasapi2_src_reset (GstAudioSrc * asrc);
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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, GST_TYPE_AUDIO_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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GstAudioSrcClass *audiosrc_class = GST_AUDIO_SRC_CLASS (klass);
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gobject_class->dispose = gst_wasapi2_src_dispose;
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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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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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basesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi2_src_get_caps);
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audiosrc_class->open = GST_DEBUG_FUNCPTR (gst_wasapi2_src_open);
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audiosrc_class->close = GST_DEBUG_FUNCPTR (gst_wasapi2_src_close);
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audiosrc_class->read = GST_DEBUG_FUNCPTR (gst_wasapi2_src_read);
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audiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi2_src_prepare);
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audiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi2_src_unprepare);
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audiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi2_src_delay);
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audiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi2_src_reset);
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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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}
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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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g_mutex_init (&self->lock);
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}
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static void
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gst_wasapi2_src_dispose (GObject * object)
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{
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GstWasapi2Src *self = GST_WASAPI2_SRC (object);
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GST_WASAPI2_SRC_LOCK (self);
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gst_clear_object (&self->client);
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gst_clear_caps (&self->cached_caps);
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GST_WASAPI2_SRC_UNLOCK (self);
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G_OBJECT_CLASS (parent_class)->dispose (object);
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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_mutex_clear (&self->lock);
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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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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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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 GstCaps *
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gst_wasapi2_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
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{
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GstWasapi2Src *self = GST_WASAPI2_SRC (bsrc);
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GstCaps *caps = NULL;
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/* In case of UWP, device activation might not be finished yet */
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if (self->client && !gst_wasapi2_client_ensure_activation (self->client)) {
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GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (NULL),
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("Failed to activate device"));
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return NULL;
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}
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if (self->client)
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caps = gst_wasapi2_client_get_caps (self->client);
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/* store one caps here so that we can return device caps even if
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* audioclient was closed due to unprepare() */
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if (!self->cached_caps && caps)
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self->cached_caps = gst_caps_ref (caps);
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if (!caps && self->cached_caps)
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caps = gst_caps_ref (self->cached_caps);
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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 (self, "returning caps %" GST_PTR_FORMAT, caps);
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return caps;
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}
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static gboolean
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gst_wasapi2_src_open_unlocked (GstAudioSrc * asrc)
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{
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GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
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self->client =
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gst_wasapi2_client_new (GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE,
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self->low_latency, -1, self->device_id, self->dispatcher);
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return ! !self->client;
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}
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static gboolean
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gst_wasapi2_src_open (GstAudioSrc * asrc)
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{
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GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
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gboolean ret;
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GST_DEBUG_OBJECT (self, "Opening device");
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GST_WASAPI2_SRC_LOCK (self);
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ret = gst_wasapi2_src_open_unlocked (asrc);
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GST_WASAPI2_SRC_UNLOCK (self);
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if (!ret) {
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GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
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("Failed to open device"));
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return FALSE;
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}
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return TRUE;
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}
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static gboolean
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gst_wasapi2_src_close (GstAudioSrc * asrc)
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{
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GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
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GST_WASAPI2_SRC_LOCK (self);
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gst_clear_object (&self->client);
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gst_clear_caps (&self->cached_caps);
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self->started = FALSE;
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GST_WASAPI2_SRC_UNLOCK (self);
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return TRUE;
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}
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static gboolean
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gst_wasapi2_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
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{
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GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
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GstAudioBaseSrc *bsrc = GST_AUDIO_BASE_SRC (asrc);
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gboolean ret = FALSE;
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GST_WASAPI2_SRC_LOCK (self);
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if (!self->client && !gst_wasapi2_src_open_unlocked (asrc)) {
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GST_ERROR_OBJECT (self, "No audio client was configured");
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goto done;
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}
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if (!gst_wasapi2_client_ensure_activation (self->client)) {
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GST_ERROR_OBJECT (self, "Couldn't activate audio device");
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goto done;
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}
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if (!gst_wasapi2_client_open (self->client, spec, bsrc->ringbuffer)) {
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GST_ERROR_OBJECT (self, "Couldn't open audio client");
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goto done;
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}
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/* Set mute and volume here again, maybe when "mute" property was set, audioclient
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* might not be configured at that moment */
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if (self->mute_changed) {
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gst_wasapi2_client_set_mute (self->client, self->mute);
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self->mute_changed = FALSE;
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}
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if (self->volume_changed) {
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gst_wasapi2_client_set_volume (self->client, self->volume);
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self->volume_changed = FALSE;
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}
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/* Will start IAudioClient on the first read request */
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self->started = FALSE;
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ret = TRUE;
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done:
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GST_WASAPI2_SRC_UNLOCK (self);
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return ret;
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}
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static gboolean
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gst_wasapi2_src_unprepare (GstAudioSrc * asrc)
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{
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GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
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self->started = FALSE;
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/* Will reopen device later prepare() */
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GST_WASAPI2_SRC_LOCK (self);
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if (self->client) {
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gst_wasapi2_client_stop (self->client);
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gst_clear_object (&self->client);
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}
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GST_WASAPI2_SRC_UNLOCK (self);
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return TRUE;
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}
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static guint
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gst_wasapi2_src_read (GstAudioSrc * asrc, gpointer data, guint length,
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GstClockTime * timestamp)
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{
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GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
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if (!self->client) {
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GST_ERROR_OBJECT (self, "No audio client was configured");
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return -1;
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}
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if (!self->started) {
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if (!gst_wasapi2_client_start (self->client)) {
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GST_ERROR_OBJECT (self, "Failed to re-start client");
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return -1;
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}
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self->started = TRUE;
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}
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return gst_wasapi2_client_read (self->client, data, length);
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}
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static guint
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gst_wasapi2_src_delay (GstAudioSrc * asrc)
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{
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GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
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if (!self->client)
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return 0;
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return gst_wasapi2_client_delay (self->client);
|
|
}
|
|
|
|
static void
|
|
gst_wasapi2_src_reset (GstAudioSrc * asrc)
|
|
{
|
|
GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
|
|
|
|
GST_DEBUG_OBJECT (self, "reset called");
|
|
|
|
self->started = FALSE;
|
|
|
|
if (!self->client)
|
|
return;
|
|
|
|
gst_wasapi2_client_stop (self->client);
|
|
}
|
|
|
|
static void
|
|
gst_wasapi2_src_set_mute (GstWasapi2Src * self, gboolean mute)
|
|
{
|
|
GST_WASAPI2_SRC_LOCK (self);
|
|
|
|
self->mute = mute;
|
|
self->mute_changed = TRUE;
|
|
|
|
if (self->client) {
|
|
if (!gst_wasapi2_client_set_mute (self->client, mute)) {
|
|
GST_INFO_OBJECT (self, "Couldn't set mute");
|
|
} else {
|
|
self->mute_changed = FALSE;
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (self, "audio client is not configured yet");
|
|
}
|
|
|
|
GST_WASAPI2_SRC_UNLOCK (self);
|
|
}
|
|
|
|
static gboolean
|
|
gst_wasapi2_src_get_mute (GstWasapi2Src * self)
|
|
{
|
|
gboolean mute;
|
|
|
|
GST_WASAPI2_SRC_LOCK (self);
|
|
|
|
mute = self->mute;
|
|
|
|
if (self->client) {
|
|
if (!gst_wasapi2_client_get_mute (self->client, &mute)) {
|
|
GST_INFO_OBJECT (self, "Couldn't get mute state");
|
|
} else {
|
|
self->mute = mute;
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (self, "audio client is not configured yet");
|
|
}
|
|
|
|
GST_WASAPI2_SRC_UNLOCK (self);
|
|
|
|
return mute;
|
|
}
|
|
|
|
static void
|
|
gst_wasapi2_src_set_volume (GstWasapi2Src * self, gdouble volume)
|
|
{
|
|
GST_WASAPI2_SRC_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 (self->client) {
|
|
if (!gst_wasapi2_client_set_volume (self->client, (gfloat) self->volume)) {
|
|
GST_INFO_OBJECT (self, "Couldn't set volume");
|
|
} else {
|
|
self->volume_changed = FALSE;
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (self, "audio client is not configured yet");
|
|
}
|
|
|
|
GST_WASAPI2_SRC_UNLOCK (self);
|
|
}
|
|
|
|
static gdouble
|
|
gst_wasapi2_src_get_volume (GstWasapi2Src * self)
|
|
{
|
|
gfloat volume;
|
|
|
|
GST_WASAPI2_SRC_LOCK (self);
|
|
|
|
volume = (gfloat) self->volume;
|
|
|
|
if (self->client) {
|
|
if (!gst_wasapi2_client_get_volume (self->client, &volume)) {
|
|
GST_INFO_OBJECT (self, "Couldn't get volume");
|
|
} else {
|
|
self->volume = volume;
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (self, "audio client is not configured yet");
|
|
}
|
|
|
|
GST_WASAPI2_SRC_UNLOCK (self);
|
|
|
|
volume = MAX (0.0, volume);
|
|
volume = MIN (1.0, volume);
|
|
|
|
return volume;
|
|
}
|