mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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wpe: Base wpe audio implementation on a web extension
This makes the implementation simpler and enable us to map webviews and audio stream much more easily Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2252>
This commit is contained in:
Thibault Saunier
GStreamer Marge Bot
parent
81ced7932f
commit
a92d4373ad
12 changed files with 578 additions and 101 deletions
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@ -22,6 +22,8 @@
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#endif
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#include "WPEThreadedView.h"
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#include "gstwpe.h"
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#include "gstwpesrcbin.h"
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#include <gst/gl/gl.h>
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#include <gst/gl/egl/gsteglimage.h>
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@ -165,6 +167,56 @@ gpointer WPEContextThread::s_viewThread(gpointer data)
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return nullptr;
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}
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#ifdef G_OS_UNIX
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static void
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initialize_web_extensions (WebKitWebContext *context)
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{
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webkit_web_context_set_web_extensions_directory (context, gst_wpe_get_extension_path ());
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}
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static gboolean
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webkit_extension_msg_received (WebKitWebContext *context,
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WebKitUserMessage *message,
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GstWpeSrc *src)
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{
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const gchar *name = webkit_user_message_get_name (message);
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GVariant *params = webkit_user_message_get_parameters (message);
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gboolean res = TRUE;
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if (!g_strcmp0(name, "gstwpe.new_stream")) {
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guint32 id = g_variant_get_uint32 (g_variant_get_child_value (params, 0));
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const gchar *capsstr = g_variant_get_string (g_variant_get_child_value (params, 1), NULL);
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GstCaps *caps = gst_caps_from_string (capsstr);
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const gchar *stream_id = g_variant_get_string (g_variant_get_child_value (params, 2), NULL);
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gst_wpe_src_new_audio_stream(src, id, caps, stream_id);
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gst_caps_unref (caps);
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} else if (!g_strcmp0(name, "gstwpe.set_shm")) {
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auto fdlist = webkit_user_message_get_fd_list (message);
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gint id = g_variant_get_uint32 (g_variant_get_child_value (params, 0));
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gst_wpe_src_set_audio_shm (src, fdlist, id);
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} else if (!g_strcmp0(name, "gstwpe.new_buffer")) {
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guint32 id = g_variant_get_uint32 (g_variant_get_child_value (params, 0));
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guint64 size = g_variant_get_uint64 (g_variant_get_child_value (params, 1));
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gst_wpe_src_push_audio_buffer (src, id, size);
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webkit_user_message_send_reply(message, webkit_user_message_new ("gstwpe.buffer_processed", NULL));
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} else if (!g_strcmp0(name, "gstwpe.pause")) {
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guint32 id = g_variant_get_uint32 (params);
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gst_wpe_src_pause_audio_stream (src, id);
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} else if (!g_strcmp0(name, "gstwpe.stop")) {
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guint32 id = g_variant_get_uint32 (params);
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gst_wpe_src_stop_audio_stream (src, id);
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} else {
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res = FALSE;
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g_error("Unknown event: %s", name);
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}
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return res;
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}
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#endif
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WPEView* WPEContextThread::createWPEView(GstWpeVideoSrc* src, GstGLContext* context, GstGLDisplay* display, int width, int height)
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{
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GST_DEBUG("context %p display %p, size (%d,%d)", context, display, width, height);
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@ -179,13 +231,11 @@ WPEView* WPEContextThread::createWPEView(GstWpeVideoSrc* src, GstGLContext* cont
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WPEView* view = nullptr;
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dispatch([&]() mutable {
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if (!glib.web_context) {
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auto* manager = webkit_website_data_manager_new_ephemeral();
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glib.web_context = webkit_web_context_new_with_website_data_manager(manager);
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g_object_unref(manager);
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}
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auto* manager = webkit_website_data_manager_new_ephemeral();
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auto web_context = webkit_web_context_new_with_website_data_manager(manager);
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g_object_unref(manager);
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view = new WPEView(glib.web_context, src, context, display, width, height);
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view = new WPEView(web_context, src, context, display, width, height);
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});
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if (view && view->hasUri()) {
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@ -233,6 +283,26 @@ static void s_loadProgressChaned(GObject* object, GParamSpec*, gpointer data)
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WPEView::WPEView(WebKitWebContext* web_context, GstWpeVideoSrc* src, GstGLContext* context, GstGLDisplay* display, int width, int height)
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{
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#ifdef G_OS_UNIX
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{
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GstObject *parent = gst_object_get_parent (GST_OBJECT (src));
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if (parent && GST_IS_WPE_SRC (parent)) {
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audio.init_ext_sigid = g_signal_connect (web_context,
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"initialize-web-extensions",
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G_CALLBACK (initialize_web_extensions),
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NULL);
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audio.extension_msg_sigid = g_signal_connect (web_context,
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"user-message-received",
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G_CALLBACK (webkit_extension_msg_received),
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parent);
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GST_INFO_OBJECT (parent, "Enabled audio");
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}
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gst_clear_object (&parent);
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}
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#endif // G_OS_UNIX
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g_mutex_init(&threading.ready_mutex);
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g_cond_init(&threading.ready_cond);
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threading.ready = FALSE;
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@ -353,6 +423,15 @@ WPEView::~WPEView()
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if (shm_committed)
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gst_buffer_unref (shm_committed);
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if (audio.init_ext_sigid) {
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WebKitWebContext* web_context = webkit_web_view_get_context (webkit.view);
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g_signal_handler_disconnect(web_context, audio.init_ext_sigid);
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g_signal_handler_disconnect(web_context, audio.extension_msg_sigid);
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audio.init_ext_sigid = 0;
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audio.extension_msg_sigid = 0;
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}
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WPEContextThread::singleton().dispatch([&]() {
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if (webkit.view) {
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g_object_unref(webkit.view);
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@ -523,11 +602,6 @@ void WPEView::setDrawBackground(gboolean drawsBackground)
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webkit_web_view_set_background_color(webkit.view, &color);
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}
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void WPEView::registerAudioReceiver(const struct wpe_audio_receiver* audioReceiver, gpointer userData)
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{
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wpe_audio_register_receiver(audioReceiver, userData);
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}
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void WPEView::releaseImage(gpointer imagePointer)
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{
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s_view->dispatch([&]() {
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@ -23,7 +23,6 @@
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#include <glib.h>
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#include <gst/gl/gstglfuncs.h>
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#include <gst/gl/egl/gstgldisplay_egl.h>
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#include <wpe/extensions/audio.h>
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#include <wpe/fdo.h>
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#include <wpe/fdo-egl.h>
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#include <wpe/webkit.h>
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@ -52,8 +51,6 @@ public:
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void loadData(GBytes*);
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void setDrawBackground(gboolean);
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void registerAudioReceiver(const struct wpe_audio_receiver*, gpointer);
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GstEGLImage* image();
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GstBuffer* buffer();
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@ -129,6 +126,11 @@ private:
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GstBuffer* committed;
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} shm { nullptr, nullptr };
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struct {
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gulong init_ext_sigid;
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gulong extension_msg_sigid;
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} audio {0, 0};
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};
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class WPEContextThread {
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@ -65,19 +65,32 @@
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#include "gstwpevideosrc.h"
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#include "gstwpesrcbin.h"
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#include "gstwpe.h"
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static gchar *extension_path = NULL;
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GST_DEBUG_CATEGORY (wpe_video_src_debug);
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GST_DEBUG_CATEGORY (wpe_view_debug);
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GST_DEBUG_CATEGORY (wpe_src_debug);
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const gchar *gst_wpe_get_extension_path (void)
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{
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return extension_path;
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}
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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gboolean result;
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gchar *dirname = g_path_get_dirname (gst_plugin_get_filename (plugin));
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GST_DEBUG_CATEGORY_INIT (wpe_video_src_debug, "wpevideosrc", 0, "WPE Video Source");
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GST_DEBUG_CATEGORY_INIT (wpe_view_debug, "wpeview", 0, "WPE Threaded View");
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GST_DEBUG_CATEGORY_INIT (wpe_src_debug, "wpesrc", 0, "WPE Source");
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gboolean result = gst_element_register (plugin, "wpevideosrc", GST_RANK_NONE,
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extension_path = g_build_filename (dirname, "wpe-extension", NULL);
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g_free (dirname);
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result = gst_element_register (plugin, "wpevideosrc", GST_RANK_NONE,
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GST_TYPE_WPE_VIDEO_SRC);
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result &= gst_element_register(plugin, "wpesrc", GST_RANK_NONE, GST_TYPE_WPE_SRC);
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return result;
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@ -21,4 +21,4 @@
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#include <gst/gst.h>
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void gst_wpe_video_src_register_audio_receiver(GstElement*, const struct wpe_audio_receiver*, gpointer);
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const gchar *gst_wpe_get_extension_path (void);
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@ -36,13 +36,16 @@
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#include "gstwpesrcbin.h"
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#include "gstwpevideosrc.h"
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#include "gstwpe-private.h"
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#include "gstwpe.h"
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#include "WPEThreadedView.h"
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#include <gst/allocators/allocators.h>
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#include <gst/base/gstflowcombiner.h>
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#include <wpe/extensions/audio.h>
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#include <sys/mman.h>
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#include <unistd.h>
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G_DEFINE_TYPE (GstWpeAudioPad, gst_wpe_audio_pad, GST_TYPE_GHOST_PAD);
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static void
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@ -106,6 +109,11 @@ GST_DEBUG_CATEGORY_EXTERN (wpe_src_debug);
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G_DEFINE_TYPE_WITH_CODE (GstWpeSrc, gst_wpe_src, GST_TYPE_BIN,
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G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_wpe_src_uri_handler_init));
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/**
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* GstWpeSrc!video
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*
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* Since: 1.20
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*/
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static GstStaticPadTemplate video_src_factory =
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GST_STATIC_PAD_TEMPLATE ("video", GST_PAD_SRC, GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("video/x-raw(memory:GLMemory), "
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@ -125,6 +133,14 @@ GST_STATIC_PAD_TEMPLATE ("video", GST_PAD_SRC, GST_PAD_SOMETIMES,
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#endif
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));
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/**
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* GstWpeSrc!audio_%u
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*
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* Each audio stream in the renderer web page will expose the and `audio_%u`
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* #GstPad.
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*
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* Since: 1.20
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*/
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static GstStaticPadTemplate audio_src_factory =
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GST_STATIC_PAD_TEMPLATE ("audio_%u", GST_PAD_SRC, GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ( \
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@ -149,18 +165,15 @@ gst_wpe_src_chain_buffer (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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return result;
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}
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static void
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on_audio_receiver_handle_start(void* data, uint32_t id, int32_t channels, const char* format, int32_t sampleRate)
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void
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gst_wpe_src_new_audio_stream(GstWpeSrc *src, guint32 id, GstCaps *caps, const gchar *stream_id)
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{
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GstWpeSrc* src = GST_WPE_SRC (data);
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GstWpeAudioPad *audio_pad;
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GstPad *pad;
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gchar *name;
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GstEvent *stream_start;
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GstSegment segment;
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GstCaps *caps;
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GST_DEBUG_OBJECT (src, "Exposing audio pad for stream %u", id);
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name = g_strdup_printf ("audio_%u", id);
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audio_pad = gst_wpe_audio_pad_new (name);
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pad = GST_PAD_CAST (audio_pad);
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@ -170,19 +183,13 @@ on_audio_receiver_handle_start(void* data, uint32_t id, int32_t channels, const
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gst_element_add_pad (GST_ELEMENT_CAST (src), pad);
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gst_flow_combiner_add_pad (src->flow_combiner, pad);
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name = gst_pad_create_stream_id_printf(pad, GST_ELEMENT_CAST (src), "%03u", id);
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stream_start = gst_event_new_stream_start (name);
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gst_pad_push_event (pad, stream_start);
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g_free (name);
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GST_DEBUG_OBJECT (src, "Adding pad: %" GST_PTR_FORMAT, pad);
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stream_start = gst_event_new_stream_start (stream_id);
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gst_pad_push_event (pad, stream_start);
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caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, format,
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"rate", G_TYPE_INT, sampleRate,
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"channels", G_TYPE_INT, channels,
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"channel-mask", GST_TYPE_BITMASK, gst_audio_channel_get_fallback_mask (channels),
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"layout", G_TYPE_STRING, "interleaved", NULL);
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gst_audio_info_from_caps (&audio_pad->info, caps);
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gst_pad_push_event (pad, gst_event_new_caps (caps));
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gst_caps_unref (caps);
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gst_segment_init (&segment, GST_FORMAT_TIME);
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gst_pad_push_event (pad, gst_event_new_segment (&segment));
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@ -190,23 +197,40 @@ on_audio_receiver_handle_start(void* data, uint32_t id, int32_t channels, const
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g_hash_table_insert (src->audio_src_pads, GUINT_TO_POINTER (id), audio_pad);
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}
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static void
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on_audio_receiver_handle_packet(void* data, struct wpe_audio_packet_export* packet_export, uint32_t id, int32_t fd, uint32_t size)
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void
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gst_wpe_src_set_audio_shm (GstWpeSrc* src, GUnixFDList *fds, guint32 id)
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{
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gint fd;
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GstWpeAudioPad *audio_pad = GST_WPE_AUDIO_PAD (g_hash_table_lookup (src->audio_src_pads, GUINT_TO_POINTER (id)));
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g_return_if_fail (GST_IS_WPE_SRC (src));
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g_return_if_fail (fds);
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g_return_if_fail (g_unix_fd_list_get_length (fds) == 1);
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g_return_if_fail (audio_pad->fd <= 0);
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fd = g_unix_fd_list_get (fds, 0, NULL);
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g_return_if_fail (fd >= 0);
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if (audio_pad->fd > 0)
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close(audio_pad->fd);
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audio_pad->fd = dup(fd);
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}
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void
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gst_wpe_src_push_audio_buffer (GstWpeSrc* src, guint32 id, guint64 size)
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{
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GstWpeSrc* src = GST_WPE_SRC (data);
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GstWpeAudioPad *audio_pad = GST_WPE_AUDIO_PAD (g_hash_table_lookup (src->audio_src_pads, GUINT_TO_POINTER (id)));
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GstPad *pad = GST_PAD_CAST (audio_pad);
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GstBuffer *buffer;
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GstClock *clock;
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g_return_if_fail (GST_IS_PAD (pad));
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g_return_if_fail (fd >= 0);
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g_return_if_fail (audio_pad->fd > 0);
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GST_TRACE_OBJECT (pad, "Handling incoming audio packet");
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buffer = gst_buffer_new ();
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GST_TRACE_OBJECT (audio_pad, "Handling incoming audio packet");
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GstMemory *mem = gst_fd_allocator_alloc (src->fd_allocator, dup (fd), size, GST_FD_MEMORY_FLAG_KEEP_MAPPED);
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gst_buffer_append_memory (buffer, mem);
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gpointer data = mmap (0, size, PROT_READ, MAP_PRIVATE, audio_pad->fd, 0);
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buffer = gst_buffer_new_wrapped (g_memdup(data, size), size);
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munmap (data, size);
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gst_buffer_add_audio_meta (buffer, &audio_pad->info, size, NULL);
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clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
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@ -231,30 +255,33 @@ on_audio_receiver_handle_packet(void* data, struct wpe_audio_packet_export* pack
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audio_pad->discont_pending = FALSE;
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}
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gst_flow_combiner_update_pad_flow (src->flow_combiner, pad, gst_pad_push (pad, buffer));
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wpe_audio_packet_export_release (packet_export);
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close (fd);
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gst_flow_combiner_update_pad_flow (src->flow_combiner, GST_PAD (audio_pad),
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gst_pad_push (GST_PAD_CAST (audio_pad), buffer));
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}
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static void
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on_audio_receiver_handle_stop(void* data, uint32_t id)
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gst_wpe_src_remove_audio_pad (GstWpeSrc *src, GstPad *pad)
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{
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GstWpeSrc* src = GST_WPE_SRC (data);
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GstWpeAudioPad *audio_pad = GST_WPE_AUDIO_PAD (g_hash_table_lookup (src->audio_src_pads, GUINT_TO_POINTER (id)));
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GstPad *pad = GST_PAD_CAST (audio_pad);
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GST_DEBUG_OBJECT (src, "Removing pad: %" GST_PTR_FORMAT, pad);
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gst_element_remove_pad (GST_ELEMENT_CAST (src), pad);
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gst_flow_combiner_remove_pad (src->flow_combiner, pad);
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}
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void
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gst_wpe_src_stop_audio_stream(GstWpeSrc* src, guint32 id)
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{
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GstPad *pad = GST_PAD (g_hash_table_lookup (src->audio_src_pads, GUINT_TO_POINTER (id)));
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g_return_if_fail (GST_IS_PAD (pad));
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|
||||
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));
|
||||
|
|
@ -265,26 +292,6 @@ on_audio_receiver_handle_pause(void* data, uint32_t id)
|
|||
audio_pad->discont_pending = TRUE;
|
||||
}
|
||||
|
||||
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)
|
||||
{
|
||||
|
|
@ -366,9 +373,14 @@ static gchar *
|
|||
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
|
||||
|
|
@ -403,8 +415,14 @@ gst_wpe_src_init (GstWpeSrc * src)
|
|||
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);
|
||||
}
|
||||
|
||||
gst_wpe_video_src_register_audio_receiver (src->video_src, &audio_receiver, src);
|
||||
static gboolean
|
||||
gst_wpe_audio_remove_audio_pad (gint32 *id, GstPad *pad, GstWpeSrc *self)
|
||||
{
|
||||
gst_wpe_src_remove_audio_pad (self, pad);
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static GstStateChangeReturn
|
||||
|
|
@ -418,6 +436,7 @@ gst_wpe_src_change_state (GstElement * element, GstStateChange transition)
|
|||
|
||||
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;
|
||||
}
|
||||
|
|
@ -459,7 +478,8 @@ gst_wpe_src_class_init (GstWpeSrcClass * klass)
|
|||
(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>");
|
||||
|
||||
|
|
|
|||
|
|
@ -23,6 +23,10 @@
|
|||
#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);
|
||||
|
|
@ -43,6 +47,7 @@ struct _GstWpeAudioPad
|
|||
GstAudioInfo info;
|
||||
GstClockTime buffer_time;
|
||||
gboolean discont_pending;
|
||||
gint fd;
|
||||
};
|
||||
|
||||
struct _GstWpeAudioPadClass
|
||||
|
|
@ -67,4 +72,10 @@ struct _GstWpeSrcClass
|
|||
|
||||
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
|
||||
|
|
|
|||
|
|
@ -75,7 +75,6 @@
|
|||
|
||||
/*
|
||||
* 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)
|
||||
|
|
@ -86,7 +85,6 @@
|
|||
#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>
|
||||
|
|
@ -129,8 +127,6 @@ struct _GstWpeVideoSrc
|
|||
gint64 n_frames; /* total frames sent */
|
||||
|
||||
WPEView *view;
|
||||
const struct wpe_audio_receiver *audio_receiver;
|
||||
gpointer audio_receiver_data;
|
||||
|
||||
GMutex lock;
|
||||
};
|
||||
|
|
@ -299,11 +295,6 @@ gst_wpe_video_src_start (GstWpeVideoSrc * src)
|
|||
|
||||
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;
|
||||
|
|
@ -743,17 +734,4 @@ gst_wpe_video_src_class_init (GstWpeVideoSrcClass * klass)
|
|||
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);
|
||||
}
|
||||
|
|
@ -16,12 +16,17 @@ if not wpe_dep.found() or not wpe_fdo_dep.found() or not egl_dep.found() or not
|
|||
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]
|
||||
|
|
|
|||
278
ext/wpe/wpe-extension/gstwpeaudiosink.c
Normal file
278
ext/wpe/wpe-extension/gstwpeaudiosink.c
Normal file
|
|
@ -0,0 +1,278 @@
|
|||
/* 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);
|
||||
}
|
||||
63
ext/wpe/wpe-extension/gstwpeextension.c
Normal file
63
ext/wpe/wpe-extension/gstwpeextension.c
Normal file
|
|
@ -0,0 +1,63 @@
|
|||
/* 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);
|
||||
}
|
||||
25
ext/wpe/wpe-extension/gstwpeextension.h
Normal file
25
ext/wpe/wpe-extension/gstwpeextension.h
Normal file
|
|
@ -0,0 +1,25 @@
|
|||
/* 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);
|
||||
8
ext/wpe/wpe-extension/meson.build
Normal file
8
ext/wpe/wpe-extension/meson.build
Normal file
|
|
@ -0,0 +1,8 @@
|
|||
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')
|
||||
|
||||
Loading…
Reference in a new issue