gstreamer/subprojects/gst-plugins-bad/ext/wpe/wpe-extension/gstwpeaudiosink.c

311 lines
8.5 KiB
C

/* 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;
}
if (!self->fdlist) {
gint fds[1] = { -1 };
#ifdef HAVE_MEMFD_CREATE
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);
self->id = g_atomic_int_add (&id, 1);
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);
return TRUE;
}
static gboolean
unlock_stop (GstBaseSink * sink)
{
GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (sink);
GCancellable *cancellable = self->cancellable;
self->cancellable = g_cancellable_new ();
g_object_unref (cancellable);
return TRUE;
}
static void
_cancelled_cb (GCancellable * _cancellable, GstWpeAudioSink * self)
{
g_mutex_lock (&self->buf_lock);
g_cond_broadcast (&self->buf_cond);
g_mutex_unlock (&self->buf_lock);
}
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 */
g_cancellable_cancel (self->cancellable);
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_PAUSED_TO_PLAYING:
{
if (g_cancellable_is_cancelled (self->cancellable)) {
GCancellable *cancellable = self->cancellable;
self->cancellable = g_cancellable_new ();
g_object_unref (cancellable);
}
break;
}
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
g_cancellable_cancel (self->cancellable);
gst_wpe_extension_send_message (webkit_user_message_new ("gstwpe.pause",
g_variant_new_uint32 (self->id)), NULL, 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 ();
g_cancellable_connect (self->cancellable,
G_CALLBACK (_cancelled_cb), self, NULL);
}
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->unlock_stop = GST_DEBUG_FUNCPTR (unlock_stop);
gstbasesink_class->render = GST_DEBUG_FUNCPTR (render);
gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (set_caps);
}