#endif
#include "WPEThreadedView.h"
+#include "gstwpe.h"
+#include "gstwpesrcbin.h"
#include <gst/gl/gl.h>
#include <gst/gl/egl/gsteglimage.h>
return nullptr;
}
+#ifdef G_OS_UNIX
+static void
+initialize_web_extensions (WebKitWebContext *context)
+{
+ webkit_web_context_set_web_extensions_directory (context, gst_wpe_get_extension_path ());
+}
+
+static gboolean
+webkit_extension_msg_received (WebKitWebContext *context,
+ WebKitUserMessage *message,
+ GstWpeSrc *src)
+{
+ const gchar *name = webkit_user_message_get_name (message);
+ GVariant *params = webkit_user_message_get_parameters (message);
+ gboolean res = TRUE;
+
+ if (!g_strcmp0(name, "gstwpe.new_stream")) {
+ guint32 id = g_variant_get_uint32 (g_variant_get_child_value (params, 0));
+ const gchar *capsstr = g_variant_get_string (g_variant_get_child_value (params, 1), NULL);
+ GstCaps *caps = gst_caps_from_string (capsstr);
+ const gchar *stream_id = g_variant_get_string (g_variant_get_child_value (params, 2), NULL);
+ gst_wpe_src_new_audio_stream(src, id, caps, stream_id);
+ gst_caps_unref (caps);
+ } else if (!g_strcmp0(name, "gstwpe.set_shm")) {
+ auto fdlist = webkit_user_message_get_fd_list (message);
+ gint id = g_variant_get_uint32 (g_variant_get_child_value (params, 0));
+ gst_wpe_src_set_audio_shm (src, fdlist, id);
+ } else if (!g_strcmp0(name, "gstwpe.new_buffer")) {
+ guint32 id = g_variant_get_uint32 (g_variant_get_child_value (params, 0));
+ guint64 size = g_variant_get_uint64 (g_variant_get_child_value (params, 1));
+ gst_wpe_src_push_audio_buffer (src, id, size);
+
+ webkit_user_message_send_reply(message, webkit_user_message_new ("gstwpe.buffer_processed", NULL));
+ } else if (!g_strcmp0(name, "gstwpe.pause")) {
+ guint32 id = g_variant_get_uint32 (params);
+
+ gst_wpe_src_pause_audio_stream (src, id);
+ } else if (!g_strcmp0(name, "gstwpe.stop")) {
+ guint32 id = g_variant_get_uint32 (params);
+
+ gst_wpe_src_stop_audio_stream (src, id);
+ } else {
+ res = FALSE;
+ g_error("Unknown event: %s", name);
+ }
+
+ return res;
+}
+#endif
+
WPEView* WPEContextThread::createWPEView(GstWpeVideoSrc* src, GstGLContext* context, GstGLDisplay* display, int width, int height)
{
GST_DEBUG("context %p display %p, size (%d,%d)", context, display, width, height);
WPEView* view = nullptr;
dispatch([&]() mutable {
- if (!glib.web_context) {
- auto* manager = webkit_website_data_manager_new_ephemeral();
- glib.web_context = webkit_web_context_new_with_website_data_manager(manager);
- g_object_unref(manager);
- }
+ auto* manager = webkit_website_data_manager_new_ephemeral();
+ auto web_context = webkit_web_context_new_with_website_data_manager(manager);
+ g_object_unref(manager);
- view = new WPEView(glib.web_context, src, context, display, width, height);
+ view = new WPEView(web_context, src, context, display, width, height);
});
if (view && view->hasUri()) {
WPEView::WPEView(WebKitWebContext* web_context, GstWpeVideoSrc* src, GstGLContext* context, GstGLDisplay* display, int width, int height)
{
+#ifdef G_OS_UNIX
+{
+ GstObject *parent = gst_object_get_parent (GST_OBJECT (src));
+
+ if (parent && GST_IS_WPE_SRC (parent)) {
+ audio.init_ext_sigid = g_signal_connect (web_context,
+ "initialize-web-extensions",
+ G_CALLBACK (initialize_web_extensions),
+ NULL);
+ audio.extension_msg_sigid = g_signal_connect (web_context,
+ "user-message-received",
+ G_CALLBACK (webkit_extension_msg_received),
+ parent);
+ GST_INFO_OBJECT (parent, "Enabled audio");
+ }
+
+ gst_clear_object (&parent);
+}
+#endif // G_OS_UNIX
+
g_mutex_init(&threading.ready_mutex);
g_cond_init(&threading.ready_cond);
threading.ready = FALSE;
if (shm_committed)
gst_buffer_unref (shm_committed);
+ if (audio.init_ext_sigid) {
+ WebKitWebContext* web_context = webkit_web_view_get_context (webkit.view);
+
+ g_signal_handler_disconnect(web_context, audio.init_ext_sigid);
+ g_signal_handler_disconnect(web_context, audio.extension_msg_sigid);
+ audio.init_ext_sigid = 0;
+ audio.extension_msg_sigid = 0;
+ }
+
WPEContextThread::singleton().dispatch([&]() {
if (webkit.view) {
g_object_unref(webkit.view);
webkit_web_view_set_background_color(webkit.view, &color);
}
-void WPEView::registerAudioReceiver(const struct wpe_audio_receiver* audioReceiver, gpointer userData)
-{
- wpe_audio_register_receiver(audioReceiver, userData);
-}
-
void WPEView::releaseImage(gpointer imagePointer)
{
s_view->dispatch([&]() {
#include <glib.h>
#include <gst/gl/gstglfuncs.h>
#include <gst/gl/egl/gstgldisplay_egl.h>
-#include <wpe/extensions/audio.h>
#include <wpe/fdo.h>
#include <wpe/fdo-egl.h>
#include <wpe/webkit.h>
void loadData(GBytes*);
void setDrawBackground(gboolean);
- void registerAudioReceiver(const struct wpe_audio_receiver*, gpointer);
-
GstEGLImage* image();
GstBuffer* buffer();
GstBuffer* committed;
} shm { nullptr, nullptr };
+ struct {
+ gulong init_ext_sigid;
+ gulong extension_msg_sigid;
+ } audio {0, 0};
+
};
class WPEContextThread {
#include "gstwpevideosrc.h"
#include "gstwpesrcbin.h"
+#include "gstwpe.h"
+
+static gchar *extension_path = NULL;
GST_DEBUG_CATEGORY (wpe_video_src_debug);
GST_DEBUG_CATEGORY (wpe_view_debug);
GST_DEBUG_CATEGORY (wpe_src_debug);
+const gchar *gst_wpe_get_extension_path (void)
+{
+ return extension_path;
+}
+
static gboolean
plugin_init (GstPlugin * plugin)
{
+ gboolean result;
+ gchar *dirname = g_path_get_dirname (gst_plugin_get_filename (plugin));
+
GST_DEBUG_CATEGORY_INIT (wpe_video_src_debug, "wpevideosrc", 0, "WPE Video Source");
GST_DEBUG_CATEGORY_INIT (wpe_view_debug, "wpeview", 0, "WPE Threaded View");
GST_DEBUG_CATEGORY_INIT (wpe_src_debug, "wpesrc", 0, "WPE Source");
- gboolean result = gst_element_register (plugin, "wpevideosrc", GST_RANK_NONE,
+ extension_path = g_build_filename (dirname, "wpe-extension", NULL);
+ g_free (dirname);
+ result = gst_element_register (plugin, "wpevideosrc", GST_RANK_NONE,
GST_TYPE_WPE_VIDEO_SRC);
result &= gst_element_register(plugin, "wpesrc", GST_RANK_NONE, GST_TYPE_WPE_SRC);
return result;
#include <gst/gst.h>
-void gst_wpe_video_src_register_audio_receiver(GstElement*, const struct wpe_audio_receiver*, gpointer);
+const gchar *gst_wpe_get_extension_path (void);
#include "gstwpesrcbin.h"
#include "gstwpevideosrc.h"
-#include "gstwpe-private.h"
+#include "gstwpe.h"
#include "WPEThreadedView.h"
#include <gst/allocators/allocators.h>
#include <gst/base/gstflowcombiner.h>
#include <wpe/extensions/audio.h>
+#include <sys/mman.h>
+#include <unistd.h>
+
G_DEFINE_TYPE (GstWpeAudioPad, gst_wpe_audio_pad, GST_TYPE_GHOST_PAD);
static void
G_DEFINE_TYPE_WITH_CODE (GstWpeSrc, gst_wpe_src, GST_TYPE_BIN,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_wpe_src_uri_handler_init));
+/**
+ * GstWpeSrc!video
+ *
+ * Since: 1.20
+ */
static GstStaticPadTemplate video_src_factory =
GST_STATIC_PAD_TEMPLATE ("video", GST_PAD_SRC, GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("video/x-raw(memory:GLMemory), "
#endif
));
+/**
+ * GstWpeSrc!audio_%u
+ *
+ * Each audio stream in the renderer web page will expose the and `audio_%u`
+ * #GstPad.
+ *
+ * Since: 1.20
+ */
static GstStaticPadTemplate audio_src_factory =
GST_STATIC_PAD_TEMPLATE ("audio_%u", GST_PAD_SRC, GST_PAD_SOMETIMES,
GST_STATIC_CAPS ( \
return result;
}
-static void
-on_audio_receiver_handle_start(void* data, uint32_t id, int32_t channels, const char* format, int32_t sampleRate)
+void
+gst_wpe_src_new_audio_stream(GstWpeSrc *src, guint32 id, GstCaps *caps, const gchar *stream_id)
{
- GstWpeSrc* src = GST_WPE_SRC (data);
GstWpeAudioPad *audio_pad;
GstPad *pad;
gchar *name;
GstEvent *stream_start;
GstSegment segment;
- GstCaps *caps;
- GST_DEBUG_OBJECT (src, "Exposing audio pad for stream %u", id);
name = g_strdup_printf ("audio_%u", id);
audio_pad = gst_wpe_audio_pad_new (name);
pad = GST_PAD_CAST (audio_pad);
gst_element_add_pad (GST_ELEMENT_CAST (src), pad);
gst_flow_combiner_add_pad (src->flow_combiner, pad);
- name = gst_pad_create_stream_id_printf(pad, GST_ELEMENT_CAST (src), "%03u", id);
- stream_start = gst_event_new_stream_start (name);
+ GST_DEBUG_OBJECT (src, "Adding pad: %" GST_PTR_FORMAT, pad);
+
+ stream_start = gst_event_new_stream_start (stream_id);
gst_pad_push_event (pad, stream_start);
- g_free (name);
- caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, format,
- "rate", G_TYPE_INT, sampleRate,
- "channels", G_TYPE_INT, channels,
- "channel-mask", GST_TYPE_BITMASK, gst_audio_channel_get_fallback_mask (channels),
- "layout", G_TYPE_STRING, "interleaved", NULL);
gst_audio_info_from_caps (&audio_pad->info, caps);
gst_pad_push_event (pad, gst_event_new_caps (caps));
- gst_caps_unref (caps);
gst_segment_init (&segment, GST_FORMAT_TIME);
gst_pad_push_event (pad, gst_event_new_segment (&segment));
g_hash_table_insert (src->audio_src_pads, GUINT_TO_POINTER (id), audio_pad);
}
-static void
-on_audio_receiver_handle_packet(void* data, struct wpe_audio_packet_export* packet_export, uint32_t id, int32_t fd, uint32_t size)
+void
+gst_wpe_src_set_audio_shm (GstWpeSrc* src, GUnixFDList *fds, guint32 id)
+{
+ gint fd;
+ GstWpeAudioPad *audio_pad = GST_WPE_AUDIO_PAD (g_hash_table_lookup (src->audio_src_pads, GUINT_TO_POINTER (id)));
+
+ g_return_if_fail (GST_IS_WPE_SRC (src));
+ g_return_if_fail (fds);
+ g_return_if_fail (g_unix_fd_list_get_length (fds) == 1);
+ g_return_if_fail (audio_pad->fd <= 0);
+
+ fd = g_unix_fd_list_get (fds, 0, NULL);
+ g_return_if_fail (fd >= 0);
+
+ if (audio_pad->fd > 0)
+ close(audio_pad->fd);
+
+ audio_pad->fd = dup(fd);
+}
+
+void
+gst_wpe_src_push_audio_buffer (GstWpeSrc* src, guint32 id, guint64 size)
{
- GstWpeSrc* src = GST_WPE_SRC (data);
GstWpeAudioPad *audio_pad = GST_WPE_AUDIO_PAD (g_hash_table_lookup (src->audio_src_pads, GUINT_TO_POINTER (id)));
- GstPad *pad = GST_PAD_CAST (audio_pad);
GstBuffer *buffer;
GstClock *clock;
- g_return_if_fail (GST_IS_PAD (pad));
- g_return_if_fail (fd >= 0);
+ g_return_if_fail (audio_pad->fd > 0);
- GST_TRACE_OBJECT (pad, "Handling incoming audio packet");
- buffer = gst_buffer_new ();
+ GST_TRACE_OBJECT (audio_pad, "Handling incoming audio packet");
- GstMemory *mem = gst_fd_allocator_alloc (src->fd_allocator, dup (fd), size, GST_FD_MEMORY_FLAG_KEEP_MAPPED);
- gst_buffer_append_memory (buffer, mem);
+ gpointer data = mmap (0, size, PROT_READ, MAP_PRIVATE, audio_pad->fd, 0);
+ buffer = gst_buffer_new_wrapped (g_memdup(data, size), size);
+ munmap (data, size);
gst_buffer_add_audio_meta (buffer, &audio_pad->info, size, NULL);
clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
audio_pad->discont_pending = FALSE;
}
- gst_flow_combiner_update_pad_flow (src->flow_combiner, pad, gst_pad_push (pad, buffer));
- wpe_audio_packet_export_release (packet_export);
- close (fd);
+ gst_flow_combiner_update_pad_flow (src->flow_combiner, GST_PAD (audio_pad),
+ gst_pad_push (GST_PAD_CAST (audio_pad), buffer));
}
static void
-on_audio_receiver_handle_stop(void* data, uint32_t id)
+gst_wpe_src_remove_audio_pad (GstWpeSrc *src, GstPad *pad)
{
- GstWpeSrc* src = GST_WPE_SRC (data);
- GstWpeAudioPad *audio_pad = GST_WPE_AUDIO_PAD (g_hash_table_lookup (src->audio_src_pads, GUINT_TO_POINTER (id)));
- GstPad *pad = GST_PAD_CAST (audio_pad);
+ GST_DEBUG_OBJECT (src, "Removing pad: %" GST_PTR_FORMAT, pad);
+ gst_element_remove_pad (GST_ELEMENT_CAST (src), pad);
+ gst_flow_combiner_remove_pad (src->flow_combiner, pad);
+}
+
+void
+gst_wpe_src_stop_audio_stream(GstWpeSrc* src, guint32 id)
+{
+ GstPad *pad = GST_PAD (g_hash_table_lookup (src->audio_src_pads, GUINT_TO_POINTER (id)));
g_return_if_fail (GST_IS_PAD (pad));
GST_INFO_OBJECT(pad, "Stopping");
gst_pad_push_event (pad, gst_event_new_eos ());
- gst_element_remove_pad (GST_ELEMENT_CAST (src), pad);
- gst_flow_combiner_remove_pad (src->flow_combiner, pad);
+ gst_wpe_src_remove_audio_pad (src, pad);
g_hash_table_remove (src->audio_src_pads, GUINT_TO_POINTER (id));
}
-static void
-on_audio_receiver_handle_pause(void* data, uint32_t id)
+void
+gst_wpe_src_pause_audio_stream(GstWpeSrc* src, guint32 id)
{
- GstWpeSrc* src = GST_WPE_SRC (data);
GstWpeAudioPad *audio_pad = GST_WPE_AUDIO_PAD (g_hash_table_lookup (src->audio_src_pads, GUINT_TO_POINTER (id)));
GstPad *pad = GST_PAD_CAST (audio_pad);
g_return_if_fail (GST_IS_PAD (pad));
}
static void
-on_audio_receiver_handle_resume(void* data, uint32_t id)
-{
- GstWpeSrc* src = GST_WPE_SRC (data);
- GstWpeAudioPad *audio_pad = GST_WPE_AUDIO_PAD (g_hash_table_lookup (src->audio_src_pads, GUINT_TO_POINTER (id)));
- GstPad *pad = GST_PAD_CAST (audio_pad);
- g_return_if_fail (GST_IS_PAD (pad));
-
- GST_INFO_OBJECT(pad, "Resuming");
-}
-
-
-static const struct wpe_audio_receiver audio_receiver = {
- .handle_start = on_audio_receiver_handle_start,
- .handle_packet = on_audio_receiver_handle_packet,
- .handle_stop = on_audio_receiver_handle_stop,
- .handle_pause = on_audio_receiver_handle_pause,
- .handle_resume = on_audio_receiver_handle_resume
-};
-
-static void
gst_wpe_src_load_bytes (GstWpeVideoSrc * src, GBytes * bytes)
{
GstWpeSrc *self = GST_WPE_SRC (src);
gst_wpe_src_get_uri (GstURIHandler * handler)
{
GstWpeSrc *src = GST_WPE_SRC (handler);
- const gchar *location;
+ gchar *location;
+ gchar *res;
+
g_object_get (src->video_src, "location", &location, NULL);
- return g_strdup_printf ("wpe://%s", location);
+ res = g_strdup_printf ("wpe://%s", location);
+ g_free (location);
+
+ return res;
}
static gboolean
src->audio_src_pads = g_hash_table_new (g_direct_hash, g_direct_equal);
src->flow_combiner = gst_flow_combiner_new ();
src->video_src = gst_element_factory_make ("wpevideosrc", NULL);
+}
+
+static gboolean
+gst_wpe_audio_remove_audio_pad (gint32 *id, GstPad *pad, GstWpeSrc *self)
+{
+ gst_wpe_src_remove_audio_pad (self, pad);
- gst_wpe_video_src_register_audio_receiver (src->video_src, &audio_receiver, src);
+ return TRUE;
}
static GstStateChangeReturn
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:{
+ g_hash_table_foreach_remove (src->audio_src_pads, (GHRFunc) gst_wpe_audio_remove_audio_pad, src);
gst_flow_combiner_reset (src->flow_combiner);
break;
}
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)));
gst_element_class_set_static_metadata (element_class, "WPE source",
- "Source/Video/Audio", "Creates a video stream from a WPE browser",
+ "Source/Video/Audio", "Creates Audio/Video streams from a web"
+ " page using WPE web engine",
"Philippe Normand <philn@igalia.com>, Žan Doberšek "
"<zdobersek@igalia.com>");
#include <gst/gst.h>
#include <gst/audio/audio.h>
+#ifdef G_OS_UNIX
+#include <gio/gunixfdlist.h>
+#endif
+
G_BEGIN_DECLS
GType gst_wpe_audio_pad_get_type(void);
GstAudioInfo info;
GstClockTime buffer_time;
gboolean discont_pending;
+ gint fd;
};
struct _GstWpeAudioPadClass
GType gst_wpe_src_get_type (void);
+void gst_wpe_src_new_audio_stream(GstWpeSrc *src, guint32 id, GstCaps *caps, const gchar *stream_id);
+void gst_wpe_src_set_audio_shm (GstWpeSrc* src, GUnixFDList *fds, guint32 id);
+void gst_wpe_src_push_audio_buffer (GstWpeSrc* src, guint32 id, guint64 size);
+void gst_wpe_src_pause_audio_stream (GstWpeSrc* src, guint32 id);
+void gst_wpe_src_stop_audio_stream (GstWpeSrc* src, guint32 id);
+
G_END_DECLS
/*
* TODO:
- * - Audio support (requires an AudioSession implementation in WebKit and a WPEBackend-fdo API for it)
* - DMABuf support (requires changes in WPEBackend-fdo to expose DMABuf planes and fds)
* - Custom EGLMemory allocator
* - Better navigation events handling (would require a new GstNavigation API)
#endif
#include "gstwpevideosrc.h"
-#include "gstwpe-private.h"
#include <gst/gl/gl.h>
#include <gst/gl/egl/gstglmemoryegl.h>
#include <gst/gl/wayland/gstgldisplay_wayland.h>
gint64 n_frames; /* total frames sent */
WPEView *view;
- const struct wpe_audio_receiver *audio_receiver;
- gpointer audio_receiver_data;
GMutex lock;
};
if (created_view) {
src->n_frames = 0;
- if (src->audio_receiver) {
- src->view->registerAudioReceiver(src->audio_receiver, src->audio_receiver_data);
- src->audio_receiver = NULL,
- src->audio_receiver_data = NULL;
- }
}
WPE_UNLOCK (src);
return TRUE;
static_cast < GSignalFlags > (G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION),
G_CALLBACK (gst_wpe_video_src_load_bytes), NULL, NULL, NULL,
G_TYPE_NONE, 1, G_TYPE_BYTES);
-}
-
-void
-gst_wpe_video_src_register_audio_receiver(GstElement* video_src, const struct wpe_audio_receiver* receiver, gpointer user_data)
-{
- GstWpeVideoSrc* src = GST_WPE_VIDEO_SOURCE(video_src);
-
- if (!src->view) {
- src->audio_receiver = receiver;
- src->audio_receiver_data = user_data;
- return;
- }
- src->view->registerAudioReceiver(receiver, user_data);
}
\ No newline at end of file
subdir_done()
endif
+giounix_dep = dependency('gio-unix-2.0', required: false)
gstwpe = library('gstwpe',
['WPEThreadedView.cpp', 'gstwpe.cpp', 'gstwpevideosrc.cpp', 'gstwpesrcbin.cpp'],
- dependencies : [egl_dep, wpe_dep, wpe_fdo_dep, gstallocators_dep, gstaudio_dep, gstvideo_dep, gstbase_dep, gstgl_dep, xkbcommon_dep, wl_server_dep],
+ dependencies : [egl_dep, wpe_dep, wpe_fdo_dep, gstallocators_dep, gstaudio_dep, gstvideo_dep, gstbase_dep, gstgl_dep, xkbcommon_dep, wl_server_dep, giounix_dep],
cpp_args : gst_plugins_bad_args + ['-DHAVE_CONFIG_H=1'],
include_directories : [configinc],
install : true,
install_dir : plugins_install_dir)
+
+if giounix_dep.found()
+ subdir('wpe-extension')
+endif
pkgconfig.generate(gstwpe, install_dir : plugins_pkgconfig_install_dir)
plugins += [gstwpe]
--- /dev/null
+/* Copyright (C) <2020> Philippe Normand <philn@igalia.com>
+ * Copyright (C) <2021> Thibault Saunier <tsaunier@igalia.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.
+ */
+
+#define _GNU_SOURCE
+#include <stdio.h>
+#include <unistd.h>
+#include <sys/mman.h>
+#include <sys/types.h>
+#include "gstwpeextension.h"
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#define gst_wpe_audio_sink_parent_class parent_class
+GST_DEBUG_CATEGORY (wpe_audio_sink_debug);
+#define GST_CAT_DEFAULT wpe_audio_sink_debug
+
+struct _GstWpeAudioSink
+{
+ GstBaseSink parent;
+
+ guint32 id;
+ GCancellable *cancellable;;
+
+ gchar *caps;
+
+ GMutex buf_lock;
+ GCond buf_cond;
+ GUnixFDList *fdlist;
+};
+
+static guint id = -1; /* atomic */
+
+G_DEFINE_TYPE_WITH_CODE (GstWpeAudioSink, gst_wpe_audio_sink,
+ GST_TYPE_BASE_SINK, GST_DEBUG_CATEGORY_INIT (wpe_audio_sink_debug,
+ "wpeaudio_sink", 0, "WPE Sink"););
+
+static GstStaticPadTemplate audio_sink_factory =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw"));
+
+static void
+message_consumed_cb (GObject * source_object, GAsyncResult * res,
+ GstWpeAudioSink * self)
+{
+ g_mutex_lock (&self->buf_lock);
+ g_cond_broadcast (&self->buf_cond);
+ g_mutex_unlock (&self->buf_lock);
+}
+
+static GstFlowReturn
+render (GstBaseSink * sink, GstBuffer * buf)
+{
+ gsize written_bytes;
+ static int init = 0;
+ char filename[1024];
+ const gint *fds;
+ WebKitUserMessage *msg;
+ GstMapInfo info;
+ GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (sink);
+
+ if (!self->caps) {
+ GST_ELEMENT_ERROR (self, CORE, NEGOTIATION,
+ ("Trying to render buffer before caps were set"), (NULL));
+
+ return GST_FLOW_ERROR;
+ }
+
+ if (!gst_buffer_map (buf, &info, GST_MAP_READ)) {
+ GST_ELEMENT_ERROR (self, RESOURCE, READ, ("Failed to map input buffer"),
+ (NULL));
+
+ return GST_FLOW_ERROR;
+ }
+#ifdef HAVE_MEMFD_CREATE
+ if (!self->fdlist) {
+ gint fds[1] = { -1 };
+
+ fds[0] = memfd_create ("gstwpe-shm", MFD_CLOEXEC);
+#endif
+
+ if (fds[0] < 0) {
+ /* allocate shm pool */
+ snprintf (filename, 1024, "%s/%s-%d-%s", g_get_user_runtime_dir (),
+ "gstwpe-shm", init++, "XXXXXX");
+
+ fds[0] = g_mkstemp (filename);
+ if (fds[0] < 0) {
+ gst_buffer_unmap (buf, &info);
+ GST_ELEMENT_ERROR (self, RESOURCE, READ,
+ ("opening temp file %s failed: %s", filename, strerror (errno)),
+ (NULL));
+ return GST_FLOW_ERROR;
+ }
+
+ unlink (filename);
+ }
+
+ if (fds[0] <= 0)
+ goto write_error;
+
+ self->fdlist = g_unix_fd_list_new_from_array (fds, 1);
+ msg = webkit_user_message_new_with_fd_list ("gstwpe.set_shm",
+ g_variant_new ("(u)", self->id), self->fdlist);
+ gst_wpe_extension_send_message (msg, self->cancellable, NULL, NULL);
+ }
+
+ fds = g_unix_fd_list_peek_fds (self->fdlist, NULL);
+ if (ftruncate (fds[0], info.size) == -1)
+ goto write_error;
+
+ written_bytes = write (fds[0], info.data, info.size);
+ if (written_bytes < 0)
+ goto write_error;
+
+ if (written_bytes != info.size)
+ goto write_error;
+
+ if (lseek (fds[0], 0, SEEK_SET) == -1)
+ goto write_error;
+
+ msg = webkit_user_message_new ("gstwpe.new_buffer",
+ g_variant_new ("(ut)", self->id, info.size));
+
+ g_mutex_lock (&self->buf_lock);
+ gst_wpe_extension_send_message (msg, self->cancellable,
+ (GAsyncReadyCallback) message_consumed_cb, self);
+ g_cond_wait (&self->buf_cond, &self->buf_lock);
+ g_mutex_unlock (&self->buf_lock);
+
+ gst_buffer_unmap (buf, &info);
+
+ return GST_FLOW_OK;
+
+write_error:
+ gst_buffer_unmap (buf, &info);
+ GST_ELEMENT_ERROR (self, RESOURCE, WRITE, ("Couldn't write memfd: %s",
+ strerror (errno)), (NULL));
+
+ return GST_FLOW_ERROR;
+}
+
+static gboolean
+set_caps (GstBaseSink * sink, GstCaps * caps)
+{
+ GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (sink);
+ gchar *stream_id;
+
+ if (self->caps) {
+ GST_ERROR_OBJECT (sink, "Renegotiation is not supported yet");
+
+ return FALSE;
+ }
+
+ self->caps = gst_caps_to_string (caps);
+ g_atomic_int_inc (&id);
+ self->id = g_atomic_int_get (&id);
+ stream_id = gst_pad_get_stream_id (GST_BASE_SINK_PAD (sink));
+ gst_wpe_extension_send_message (webkit_user_message_new ("gstwpe.new_stream",
+ g_variant_new ("(uss)", self->id, self->caps, stream_id)),
+ self->cancellable, NULL, NULL);
+ g_free (stream_id);
+
+ return TRUE;
+}
+
+static gboolean
+unlock (GstBaseSink * sink)
+{
+ GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (sink);
+
+ g_cancellable_cancel (self->cancellable);
+ g_mutex_lock (&self->buf_lock);
+ g_cond_broadcast (&self->buf_cond);
+ g_mutex_unlock (&self->buf_lock);
+
+ return TRUE;
+}
+
+static gboolean
+stop (GstBaseSink * sink)
+{
+ GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (sink);
+
+ if (!self->caps) {
+ GST_DEBUG_OBJECT (sink, "Stopped before started");
+
+ return TRUE;
+ }
+
+ /* Stop processing and claim buffers back */
+ unlock (sink);
+
+ GST_DEBUG_OBJECT (sink, "Stopping %d", self->id);
+ gst_wpe_extension_send_message (webkit_user_message_new ("gstwpe.stop",
+ g_variant_new_uint32 (self->id)), self->cancellable, NULL, NULL);
+
+ return TRUE;
+}
+
+static GstStateChangeReturn
+change_state (GstElement * element, GstStateChange transition)
+{
+ GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+ gst_wpe_extension_send_message (webkit_user_message_new ("gstwpe.pause",
+ g_variant_new_uint32 (self->id)), self->cancellable, NULL, NULL);
+ break;
+ default:
+ break;
+ }
+
+ return GST_CALL_PARENT_WITH_DEFAULT (GST_ELEMENT_CLASS,
+ change_state, (element, transition), GST_STATE_CHANGE_SUCCESS);
+}
+
+static void
+dispose (GObject * object)
+{
+ GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (object);
+
+ g_clear_object (&self->cancellable);
+ g_clear_pointer (&self->caps, g_free);
+}
+
+static void
+gst_wpe_audio_sink_init (GstWpeAudioSink * self)
+{
+ GstElementClass *klass = GST_ELEMENT_GET_CLASS (self);
+ GstPadTemplate *pad_template =
+ gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
+ g_return_if_fail (pad_template != NULL);
+
+ self->cancellable = g_cancellable_new ();
+}
+
+static void
+gst_wpe_audio_sink_class_init (GstWpeAudioSinkClass * klass)
+{
+ GstPadTemplate *tmpl;
+ GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+ GObjectClass *object_class = G_OBJECT_CLASS (klass);
+ GstBaseSinkClass *gstbasesink_class = (GstBaseSinkClass *) klass;
+
+ object_class->dispose = dispose;
+
+ gst_element_class_set_static_metadata (gstelement_class,
+ "WPE internal audio sink", "Sink/Audio",
+ "Internal sink to be used in wpe when running inside gstwpe",
+ "Thibault Saunier <tsaunier@igalia.com>");
+
+ tmpl = gst_static_pad_template_get (&audio_sink_factory);
+ gst_element_class_add_pad_template (gstelement_class, tmpl);
+
+ gstelement_class->change_state = GST_DEBUG_FUNCPTR (change_state);
+ gstbasesink_class->stop = GST_DEBUG_FUNCPTR (stop);
+ gstbasesink_class->unlock = GST_DEBUG_FUNCPTR (unlock);
+ gstbasesink_class->render = GST_DEBUG_FUNCPTR (render);
+ gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (set_caps);
+}
--- /dev/null
+/* Copyright (C) <2021> Thibault Saunier <tsaunier@igalia.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.
+ */
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "gstwpeextension.h"
+
+#include <errno.h>
+#include <fcntl.h>
+#include <sys/socket.h>
+
+#include <gst/gst.h>
+#include <gmodule.h>
+#include <gio/gunixfdlist.h>
+#include <wpe/webkit-web-extension.h>
+
+G_MODULE_EXPORT void webkit_web_extension_initialize (WebKitWebExtension *
+ extension);
+
+static WebKitWebExtension *global_extension = NULL;
+
+void
+webkit_web_extension_initialize (WebKitWebExtension * extension)
+{
+ g_return_if_fail (!global_extension);
+
+ gst_init (NULL, NULL);
+
+ /* Register our own audio sink to */
+ gst_element_register (NULL, "gstwpeaudiosink", GST_RANK_PRIMARY + 500,
+ gst_wpe_audio_sink_get_type ());
+
+ GST_INFO ("Mark processus as WebProcess");
+ if (!g_setenv ("GST_WPE_ID", "1", TRUE))
+ g_error ("Could not set GST_WPE_ID envvar\n");
+
+ global_extension = extension;
+ GST_INFO_OBJECT (global_extension, "Setting as global extension.");
+}
+
+void
+gst_wpe_extension_send_message (WebKitUserMessage * msg,
+ GCancellable * cancellable, GAsyncReadyCallback cb, gpointer udata)
+{
+ webkit_web_extension_send_message_to_context (global_extension, msg,
+ cancellable, cb, udata);
+}
--- /dev/null
+/* Copyright (C) <2021> Thibault Saunier <tsaunier@igalia.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.
+ */
+
+#pragma once
+
+#include <wpe/webkit-web-extension.h>
+#include <gio/gunixfdlist.h>
+#include <gst/gst.h>
+#include <gst/base/gstbasesink.h>
+
+void gst_wpe_extension_send_message (WebKitUserMessage *msg, GCancellable *cancellable, GAsyncReadyCallback cb, gpointer udata);
+
+G_DECLARE_FINAL_TYPE (GstWpeAudioSink, gst_wpe_audio_sink, GST, WPE_AUDIO_SINK, GstBaseSink);
\ No newline at end of file
--- /dev/null
+library('gstwpeextension',
+ ['gstwpeextension.c', 'gstwpeaudiosink.c'],
+ dependencies : [wpe_dep, gst_dep, gstbase_dep, giounix_dep],
+ c_args : ['-DHAVE_CONFIG_H=1'],
+ include_directories : [configinc],
+ install : true,
+ install_dir : plugins_install_dir / 'wpe-extension')
+
projects / platform / upstream / gstreamer.git / commitdiff