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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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cd3aa029d6
There was a race when going to PAUSED while pushing a buffer to the pipeline process (where we weren't even cancelling anything). This rework base all the cancellation around the GCancellable "cancelled" signal trying to ensure that the streaming thread will not block once a cancel operation happens. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2504>
311 lines
8.5 KiB
C
311 lines
8.5 KiB
C
/* Copyright (C) <2020> Philippe Normand <philn@igalia.com>
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* Copyright (C) <2021> Thibault Saunier <tsaunier@igalia.com>
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*
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* This library is free software; you can redistribute it and/or modify it under the terms of the
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* GNU Library General Public License as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without
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* even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public License along with this
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* library; if not, write to the Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#define _GNU_SOURCE
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#include <stdio.h>
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#include <unistd.h>
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#include <sys/mman.h>
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#include <sys/types.h>
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#include "gstwpeextension.h"
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#define gst_wpe_audio_sink_parent_class parent_class
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GST_DEBUG_CATEGORY (wpe_audio_sink_debug);
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#define GST_CAT_DEFAULT wpe_audio_sink_debug
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struct _GstWpeAudioSink
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{
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GstBaseSink parent;
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guint32 id;
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GCancellable *cancellable;;
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gchar *caps;
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GMutex buf_lock;
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GCond buf_cond;
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GUnixFDList *fdlist;
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};
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static guint id = -1; /* atomic */
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G_DEFINE_TYPE_WITH_CODE (GstWpeAudioSink, gst_wpe_audio_sink,
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GST_TYPE_BASE_SINK, GST_DEBUG_CATEGORY_INIT (wpe_audio_sink_debug,
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"wpeaudio_sink", 0, "WPE Sink"););
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static GstStaticPadTemplate audio_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw"));
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static void
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message_consumed_cb (GObject * source_object, GAsyncResult * res,
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GstWpeAudioSink * self)
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{
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g_mutex_lock (&self->buf_lock);
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g_cond_broadcast (&self->buf_cond);
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g_mutex_unlock (&self->buf_lock);
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}
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static GstFlowReturn
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render (GstBaseSink * sink, GstBuffer * buf)
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{
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gsize written_bytes;
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static int init = 0;
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char filename[1024];
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const gint *fds;
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WebKitUserMessage *msg;
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GstMapInfo info;
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GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (sink);
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if (!self->caps) {
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GST_ELEMENT_ERROR (self, CORE, NEGOTIATION,
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("Trying to render buffer before caps were set"), (NULL));
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return GST_FLOW_ERROR;
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}
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if (!gst_buffer_map (buf, &info, GST_MAP_READ)) {
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GST_ELEMENT_ERROR (self, RESOURCE, READ, ("Failed to map input buffer"),
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(NULL));
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return GST_FLOW_ERROR;
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}
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if (!self->fdlist) {
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gint fds[1] = { -1 };
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#ifdef HAVE_MEMFD_CREATE
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fds[0] = memfd_create ("gstwpe-shm", MFD_CLOEXEC);
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#endif
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if (fds[0] < 0) {
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/* allocate shm pool */
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snprintf (filename, 1024, "%s/%s-%d-%s", g_get_user_runtime_dir (),
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"gstwpe-shm", init++, "XXXXXX");
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fds[0] = g_mkstemp (filename);
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if (fds[0] < 0) {
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gst_buffer_unmap (buf, &info);
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GST_ELEMENT_ERROR (self, RESOURCE, READ,
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("opening temp file %s failed: %s", filename, strerror (errno)),
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(NULL));
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return GST_FLOW_ERROR;
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}
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unlink (filename);
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}
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if (fds[0] <= 0)
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goto write_error;
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self->fdlist = g_unix_fd_list_new_from_array (fds, 1);
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msg = webkit_user_message_new_with_fd_list ("gstwpe.set_shm",
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g_variant_new ("(u)", self->id), self->fdlist);
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gst_wpe_extension_send_message (msg, self->cancellable, NULL, NULL);
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}
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fds = g_unix_fd_list_peek_fds (self->fdlist, NULL);
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if (ftruncate (fds[0], info.size) == -1)
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goto write_error;
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written_bytes = write (fds[0], info.data, info.size);
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if (written_bytes < 0)
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goto write_error;
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if (written_bytes != info.size)
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goto write_error;
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if (lseek (fds[0], 0, SEEK_SET) == -1)
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goto write_error;
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msg = webkit_user_message_new ("gstwpe.new_buffer",
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g_variant_new ("(ut)", self->id, info.size));
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g_mutex_lock (&self->buf_lock);
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gst_wpe_extension_send_message (msg, self->cancellable,
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(GAsyncReadyCallback) message_consumed_cb, self);
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g_cond_wait (&self->buf_cond, &self->buf_lock);
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g_mutex_unlock (&self->buf_lock);
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gst_buffer_unmap (buf, &info);
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return GST_FLOW_OK;
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write_error:
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gst_buffer_unmap (buf, &info);
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GST_ELEMENT_ERROR (self, RESOURCE, WRITE, ("Couldn't write memfd: %s",
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strerror (errno)), (NULL));
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return GST_FLOW_ERROR;
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}
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static gboolean
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set_caps (GstBaseSink * sink, GstCaps * caps)
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{
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GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (sink);
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gchar *stream_id;
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if (self->caps) {
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GST_ERROR_OBJECT (sink, "Renegotiation is not supported yet");
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return FALSE;
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}
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self->caps = gst_caps_to_string (caps);
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self->id = g_atomic_int_add (&id, 1);
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stream_id = gst_pad_get_stream_id (GST_BASE_SINK_PAD (sink));
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gst_wpe_extension_send_message (webkit_user_message_new ("gstwpe.new_stream",
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g_variant_new ("(uss)", self->id, self->caps, stream_id)),
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self->cancellable, NULL, NULL);
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g_free (stream_id);
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return TRUE;
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}
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static gboolean
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unlock (GstBaseSink * sink)
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{
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GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (sink);
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g_cancellable_cancel (self->cancellable);
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return TRUE;
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}
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static gboolean
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unlock_stop (GstBaseSink * sink)
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{
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GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (sink);
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GCancellable *cancellable = self->cancellable;
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self->cancellable = g_cancellable_new ();
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g_object_unref (cancellable);
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return TRUE;
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}
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static void
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_cancelled_cb (GCancellable * _cancellable, GstWpeAudioSink * self)
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{
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g_mutex_lock (&self->buf_lock);
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g_cond_broadcast (&self->buf_cond);
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g_mutex_unlock (&self->buf_lock);
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}
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static gboolean
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stop (GstBaseSink * sink)
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{
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GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (sink);
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if (!self->caps) {
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GST_DEBUG_OBJECT (sink, "Stopped before started");
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return TRUE;
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}
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/* Stop processing and claim buffers back */
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g_cancellable_cancel (self->cancellable);
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GST_DEBUG_OBJECT (sink, "Stopping %d", self->id);
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gst_wpe_extension_send_message (webkit_user_message_new ("gstwpe.stop",
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g_variant_new_uint32 (self->id)), self->cancellable, NULL, NULL);
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return TRUE;
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}
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static GstStateChangeReturn
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change_state (GstElement * element, GstStateChange transition)
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{
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GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (element);
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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{
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if (g_cancellable_is_cancelled (self->cancellable)) {
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GCancellable *cancellable = self->cancellable;
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self->cancellable = g_cancellable_new ();
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g_object_unref (cancellable);
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}
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break;
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}
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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g_cancellable_cancel (self->cancellable);
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gst_wpe_extension_send_message (webkit_user_message_new ("gstwpe.pause",
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g_variant_new_uint32 (self->id)), NULL, NULL, NULL);
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break;
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default:
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break;
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}
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return GST_CALL_PARENT_WITH_DEFAULT (GST_ELEMENT_CLASS,
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change_state, (element, transition), GST_STATE_CHANGE_SUCCESS);
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}
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static void
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dispose (GObject * object)
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{
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GstWpeAudioSink *self = GST_WPE_AUDIO_SINK (object);
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g_clear_object (&self->cancellable);
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g_clear_pointer (&self->caps, g_free);
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}
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static void
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gst_wpe_audio_sink_init (GstWpeAudioSink * self)
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{
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GstElementClass *klass = GST_ELEMENT_GET_CLASS (self);
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GstPadTemplate *pad_template =
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gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
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g_return_if_fail (pad_template != NULL);
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self->cancellable = g_cancellable_new ();
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g_cancellable_connect (self->cancellable,
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G_CALLBACK (_cancelled_cb), self, NULL);
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}
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static void
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gst_wpe_audio_sink_class_init (GstWpeAudioSinkClass * klass)
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{
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GstPadTemplate *tmpl;
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstBaseSinkClass *gstbasesink_class = (GstBaseSinkClass *) klass;
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object_class->dispose = dispose;
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gst_element_class_set_static_metadata (gstelement_class,
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"WPE internal audio sink", "Sink/Audio",
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"Internal sink to be used in wpe when running inside gstwpe",
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"Thibault Saunier <tsaunier@igalia.com>");
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tmpl = gst_static_pad_template_get (&audio_sink_factory);
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gst_element_class_add_pad_template (gstelement_class, tmpl);
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (change_state);
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gstbasesink_class->stop = GST_DEBUG_FUNCPTR (stop);
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gstbasesink_class->unlock = GST_DEBUG_FUNCPTR (unlock);
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gstbasesink_class->unlock_stop = GST_DEBUG_FUNCPTR (unlock_stop);
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gstbasesink_class->render = GST_DEBUG_FUNCPTR (render);
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gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (set_caps);
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}
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