/* * Copyright (C) 2008 Ole André Vadla Ravnås * Copyright (C) 2013 Collabora Ltd. * Author: Sebastian Dröge * Copyright (C) 2018 Centricular Ltd. * Author: Nirbheek Chauhan * Copyright (C) 2020 Seungha Yang * * 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 * |[ * gst-launch-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapi2sink * ]| Generate 20 ms buffers and render to the default audio device. * * |[ * gst-launch-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapi2sink low-latency=true * ]| Same as above, but with the minimum possible latency * */ #ifdef HAVE_CONFIG_H #include #endif #include "gstwasapi2sink.h" #include "gstwasapi2util.h" #include "gstwasapi2client.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 #define GST_WASAPI2_SINK_LOCK(s) g_mutex_lock(&(s)->lock) #define GST_WASAPI2_SINK_UNLOCK(s) g_mutex_unlock(&(s)->lock) enum { PROP_0, PROP_DEVICE, PROP_LOW_LATENCY, PROP_MUTE, PROP_VOLUME, PROP_DISPATCHER, }; struct _GstWasapi2Sink { GstAudioSink parent; GstWasapi2Client *client; GstCaps *cached_caps; gboolean started; /* properties */ gchar *device_id; gboolean low_latency; gboolean mute; gdouble volume; gpointer dispatcher; gboolean mute_changed; gboolean volume_changed; /* to protect audioclient from set/get property */ GMutex lock; }; static void gst_wasapi2_sink_dispose (GObject * object); 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); static GstCaps *gst_wasapi2_sink_get_caps (GstBaseSink * bsink, GstCaps * filter); static gboolean gst_wasapi2_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec); static gboolean gst_wasapi2_sink_unprepare (GstAudioSink * asink); static gboolean gst_wasapi2_sink_open (GstAudioSink * asink); static gboolean gst_wasapi2_sink_close (GstAudioSink * asink); static gint gst_wasapi2_sink_write (GstAudioSink * asink, gpointer data, guint length); static guint gst_wasapi2_sink_delay (GstAudioSink * asink); static void gst_wasapi2_sink_reset (GstAudioSink * asink); 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 G_DEFINE_TYPE_WITH_CODE (GstWasapi2Sink, gst_wasapi2_sink, GST_TYPE_AUDIO_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); GstAudioSinkClass *audiosink_class = GST_AUDIO_SINK_CLASS (klass); gobject_class->dispose = gst_wasapi2_sink_dispose; 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 , " "Ole André Vadla Ravnås , " "Seungha Yang "); basesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_get_caps); audiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_prepare); audiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_unprepare); audiosink_class->open = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_open); audiosink_class->close = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_close); audiosink_class->write = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_write); audiosink_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_delay); audiosink_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_reset); 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; g_mutex_init (&self->lock); } static void gst_wasapi2_sink_dispose (GObject * object) { GstWasapi2Sink *self = GST_WASAPI2_SINK (object); GST_WASAPI2_SINK_LOCK (self); gst_clear_object (&self->client); gst_clear_caps (&self->cached_caps); GST_WASAPI2_SINK_UNLOCK (self); G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_wasapi2_sink_finalize (GObject * object) { GstWasapi2Sink *self = GST_WASAPI2_SINK (object); g_free (self->device_id); g_mutex_clear (&self->lock); 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; } } static GstCaps * gst_wasapi2_sink_get_caps (GstBaseSink * bsink, GstCaps * filter) { GstWasapi2Sink *self = GST_WASAPI2_SINK (bsink); GstCaps *caps = NULL; /* In case of UWP, device activation might not be finished yet */ if (self->client && !gst_wasapi2_client_ensure_activation (self->client)) { GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (NULL), ("Failed to activate device")); return NULL; } if (self->client) caps = gst_wasapi2_client_get_caps (self->client); /* store one caps here so that we can return device caps even if * audioclient was closed due to unprepare() */ if (!self->cached_caps && caps) self->cached_caps = gst_caps_ref (caps); if (!caps && self->cached_caps) caps = gst_caps_ref (self->cached_caps); 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; } GST_DEBUG_OBJECT (self, "returning caps %" GST_PTR_FORMAT, caps); return caps; } static gboolean gst_wasapi2_sink_open_unlocked (GstAudioSink * asink) { GstWasapi2Sink *self = GST_WASAPI2_SINK (asink); gst_clear_object (&self->client); self->client = gst_wasapi2_client_new (GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER, self->low_latency, -1, self->device_id, self->dispatcher); if (!self->client) return FALSE; return TRUE; } static gboolean gst_wasapi2_sink_open (GstAudioSink * asink) { GstWasapi2Sink *self = GST_WASAPI2_SINK (asink); gboolean ret; GST_DEBUG_OBJECT (self, "Opening device"); GST_WASAPI2_SINK_LOCK (self); ret = gst_wasapi2_sink_open_unlocked (asink); GST_WASAPI2_SINK_UNLOCK (self); if (!ret) { GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (NULL), ("Failed to open device")); return FALSE; } return TRUE; } static gboolean gst_wasapi2_sink_close (GstAudioSink * asink) { GstWasapi2Sink *self = GST_WASAPI2_SINK (asink); GST_WASAPI2_SINK_LOCK (self); gst_clear_object (&self->client); gst_clear_caps (&self->cached_caps); self->started = FALSE; GST_WASAPI2_SINK_UNLOCK (self); return TRUE; } static gboolean gst_wasapi2_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec) { GstWasapi2Sink *self = GST_WASAPI2_SINK (asink); GstAudioBaseSink *bsink = GST_AUDIO_BASE_SINK (asink); gboolean ret = FALSE; HRESULT hr; GST_WASAPI2_SINK_LOCK (self); if (!self->client && !gst_wasapi2_sink_open_unlocked (asink)) { GST_ERROR_OBJECT (self, "No audio client was configured"); goto done; } if (!gst_wasapi2_client_ensure_activation (self->client)) { GST_ERROR_OBJECT (self, "Couldn't activate audio device"); goto done; } hr = gst_wasapi2_client_open (self->client, spec, bsink->ringbuffer); if (!gst_wasapi2_result (hr)) { GST_ERROR_OBJECT (self, "Couldn't open audio client"); goto done; } /* Set mute and volume here again, maybe when "mute" property was set, audioclient * might not be configured at that moment */ if (self->mute_changed) { gst_wasapi2_client_set_mute (self->client, self->mute); self->mute_changed = FALSE; } if (self->volume_changed) { gst_wasapi2_client_set_volume (self->client, self->volume); self->volume_changed = FALSE; } /* Will start IAudioClient on the first write request */ self->started = FALSE; ret = TRUE; done: GST_WASAPI2_SINK_UNLOCK (self); return ret; } static gboolean gst_wasapi2_sink_unprepare (GstAudioSink * asink) { GstWasapi2Sink *self = GST_WASAPI2_SINK (asink); self->started = FALSE; /* Will reopen device later prepare() */ GST_WASAPI2_SINK_LOCK (self); if (self->client) { gst_wasapi2_client_stop (self->client); gst_clear_object (&self->client); } GST_WASAPI2_SINK_UNLOCK (self); return TRUE; } static gint gst_wasapi2_sink_write (GstAudioSink * asink, gpointer data, guint length) { GstWasapi2Sink *self = GST_WASAPI2_SINK (asink); HRESULT hr; guint write_len = 0; if (!self->client) { GST_ERROR_OBJECT (self, "No audio client was configured"); return -1; } if (!self->started) { HRESULT hr = gst_wasapi2_client_start (self->client); if (!gst_wasapi2_result (hr)) { GST_ERROR_OBJECT (self, "Failed to re-start client"); return -1; } self->started = TRUE; } hr = gst_wasapi2_client_write (self->client, data, length, &write_len); if (!gst_wasapi2_result (hr)) { GST_ERROR_OBJECT (self, "Failed to write"); return -1; } return (gint) write_len; } static guint gst_wasapi2_sink_delay (GstAudioSink * asink) { GstWasapi2Sink *self = GST_WASAPI2_SINK (asink); guint32 delay; HRESULT hr; if (!self->client) return 0; hr = gst_wasapi2_client_delay (self->client, &delay); if (!gst_wasapi2_result (hr)) { GST_WARNING_OBJECT (self, "Failed to get delay"); return 0; } return delay; } static void gst_wasapi2_sink_reset (GstAudioSink * asink) { GstWasapi2Sink *self = GST_WASAPI2_SINK (asink); GST_INFO_OBJECT (self, "reset called"); self->started = FALSE; if (!self->client) return; gst_wasapi2_client_stop (self->client); } static void gst_wasapi2_sink_set_mute (GstWasapi2Sink * self, gboolean mute) { GST_WASAPI2_SINK_LOCK (self); self->mute = mute; self->mute_changed = TRUE; if (self->client) { HRESULT hr = gst_wasapi2_client_set_mute (self->client, mute); if (FAILED (hr)) { 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_SINK_UNLOCK (self); } static gboolean gst_wasapi2_sink_get_mute (GstWasapi2Sink * self) { gboolean mute; GST_WASAPI2_SINK_LOCK (self); mute = self->mute; if (self->client) { HRESULT hr = gst_wasapi2_client_get_mute (self->client, &mute); if (FAILED (hr)) { 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_SINK_UNLOCK (self); return mute; } static void gst_wasapi2_sink_set_volume (GstWasapi2Sink * self, gdouble volume) { GST_WASAPI2_SINK_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) { HRESULT hr = gst_wasapi2_client_set_volume (self->client, (gfloat) self->volume); if (FAILED (hr)) { 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_SINK_UNLOCK (self); } static gdouble gst_wasapi2_sink_get_volume (GstWasapi2Sink * self) { gfloat volume; GST_WASAPI2_SINK_LOCK (self); volume = (gfloat) self->volume; if (self->client) { HRESULT hr = gst_wasapi2_client_get_volume (self->client, &volume); if (FAILED (hr)) { 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_SINK_UNLOCK (self); volume = MAX (0.0, volume); volume = MIN (1.0, volume); return volume; }