#endif
#include "gstaudioencoder.h"
+#include "gstaudioutilsprivate.h"
#include <gst/base/gstadapter.h>
#include <gst/audio/audio.h>
#include <gst/pbutils/descriptions.h>
gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc, GstCaps * caps,
GstCaps * filter)
{
- GstCaps *templ_caps = NULL;
- GstCaps *allowed = NULL;
- GstCaps *fcaps, *filter_caps;
- gint i, j;
-
- /* we want to be able to communicate to upstream elements like audioconvert
- * and audioresample any rate/channel restrictions downstream (e.g. muxer
- * only accepting certain sample rates) */
- templ_caps =
- caps ? gst_caps_ref (caps) : gst_pad_get_pad_template_caps (enc->sinkpad);
- allowed = gst_pad_get_allowed_caps (enc->srcpad);
- if (!allowed || gst_caps_is_empty (allowed) || gst_caps_is_any (allowed)) {
- fcaps = templ_caps;
- goto done;
- }
-
- GST_LOG_OBJECT (enc, "template caps %" GST_PTR_FORMAT, templ_caps);
- GST_LOG_OBJECT (enc, "allowed caps %" GST_PTR_FORMAT, allowed);
-
- filter_caps = gst_caps_new_empty ();
-
- for (i = 0; i < gst_caps_get_size (templ_caps); i++) {
- GQuark q_name;
-
- q_name = gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
-
- /* pick rate + channel fields from allowed caps */
- for (j = 0; j < gst_caps_get_size (allowed); j++) {
- const GstStructure *allowed_s = gst_caps_get_structure (allowed, j);
- const GValue *val;
- GstStructure *s;
-
- s = gst_structure_new_id_empty (q_name);
- if ((val = gst_structure_get_value (allowed_s, "rate")))
- gst_structure_set_value (s, "rate", val);
- if ((val = gst_structure_get_value (allowed_s, "channels")))
- gst_structure_set_value (s, "channels", val);
- /* following might also make sense for some encoded formats,
- * e.g. wavpack */
- if ((val = gst_structure_get_value (allowed_s, "channel-mask")))
- gst_structure_set_value (s, "channel-mask", val);
-
- filter_caps = gst_caps_merge_structure (filter_caps, s);
- }
- }
-
- fcaps = gst_caps_intersect (filter_caps, templ_caps);
- gst_caps_unref (filter_caps);
- gst_caps_unref (templ_caps);
-
- if (filter) {
- GST_LOG_OBJECT (enc, "intersecting with %" GST_PTR_FORMAT, filter);
- filter_caps = gst_caps_intersect_full (filter, fcaps,
- GST_CAPS_INTERSECT_FIRST);
- gst_caps_unref (fcaps);
- fcaps = filter_caps;
- }
-
-done:
- gst_caps_replace (&allowed, NULL);
-
- GST_LOG_OBJECT (enc, "proxy caps %" GST_PTR_FORMAT, fcaps);
-
- return fcaps;
+ return __gst_audio_element_proxy_getcaps (GST_ELEMENT_CAST (enc),
+ GST_AUDIO_ENCODER_SINK_PAD (enc), GST_AUDIO_ENCODER_SRC_PAD (enc),
+ caps, filter);
}
static GstCaps *
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
+ * Copyright (C) 2011 Nokia Corporation. All rights reserved.
+ * Contact: Stefan Kost <stefan.kost@nokia.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 <gst/audio/audio.h>
+#include "gstaudioutilsprivate.h"
+
+/*
+ * Takes caps and copies its audio fields to tmpl_caps
+ */
+static GstCaps *
+__gst_audio_element_proxy_caps (GstElement * element, GstCaps * templ_caps,
+ GstCaps * caps)
+{
+ GstCaps *result = gst_caps_new_empty ();
+ gint i, j;
+ gint templ_caps_size = gst_caps_get_size (templ_caps);
+ gint caps_size = gst_caps_get_size (caps);
+
+ for (i = 0; i < templ_caps_size; i++) {
+ GQuark q_name =
+ gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
+
+ for (j = 0; j < caps_size; j++) {
+ const GstStructure *caps_s = gst_caps_get_structure (caps, j);
+ const GValue *val;
+ GstStructure *s;
+
+ s = gst_structure_new_id_empty (q_name);
+ if ((val = gst_structure_get_value (caps_s, "rate")))
+ gst_structure_set_value (s, "rate", val);
+ if ((val = gst_structure_get_value (caps_s, "channels")))
+ gst_structure_set_value (s, "channels", val);
+ if ((val = gst_structure_get_value (caps_s, "channels-mask")))
+ gst_structure_set_value (s, "channels-mask", val);
+
+ result = gst_caps_merge_structure (result, s);
+ }
+ }
+
+ return result;
+}
+
+/**
+ * __gst_audio_element_proxy_getcaps:
+ * @element: a #GstElement
+ * @sinkpad: the element's sink #GstPad
+ * @srcpad: the element's source #GstPad
+ * @initial_caps: initial caps
+ * @filter: filter caps
+ *
+ * Returns caps that express @initial_caps (or sink template caps if
+ * @initial_caps == NULL) restricted to rate/channels/...
+ * combinations supported by downstream elements (e.g. muxers).
+ *
+ * Returns: a #GstCaps owned by caller
+ */
+GstCaps *
+__gst_audio_element_proxy_getcaps (GstElement * element, GstPad * sinkpad,
+ GstPad * srcpad, GstCaps * initial_caps, GstCaps * filter)
+{
+ GstCaps *templ_caps, *src_templ_caps;
+ GstCaps *peer_caps;
+ GstCaps *allowed;
+ GstCaps *fcaps, *filter_caps;
+
+ /* Allow downstream to specify rate/channels constraints
+ * and forward them upstream for audio converters to handle
+ */
+ templ_caps = initial_caps ? gst_caps_ref (initial_caps) :
+ gst_pad_get_pad_template_caps (sinkpad);
+ src_templ_caps = gst_pad_get_pad_template_caps (srcpad);
+ if (filter && !gst_caps_is_any (filter)) {
+ GstCaps *proxy_filter =
+ __gst_audio_element_proxy_caps (element, src_templ_caps, filter);
+
+ peer_caps = gst_pad_peer_query_caps (srcpad, proxy_filter);
+ gst_caps_unref (proxy_filter);
+ } else {
+ peer_caps = gst_pad_peer_query_caps (srcpad, NULL);
+ }
+
+ allowed = gst_caps_intersect_full (peer_caps, src_templ_caps,
+ GST_CAPS_INTERSECT_FIRST);
+
+ gst_caps_unref (src_templ_caps);
+ gst_caps_unref (peer_caps);
+
+ if (!allowed || gst_caps_is_any (allowed)) {
+ fcaps = templ_caps;
+ goto done;
+ } else if (gst_caps_is_empty (allowed)) {
+ fcaps = gst_caps_ref (allowed);
+ goto done;
+ }
+
+ GST_LOG_OBJECT (element, "template caps %" GST_PTR_FORMAT, templ_caps);
+ GST_LOG_OBJECT (element, "allowed caps %" GST_PTR_FORMAT, allowed);
+
+ filter_caps = __gst_audio_element_proxy_caps (element, templ_caps, allowed);
+
+ fcaps = gst_caps_intersect (filter_caps, templ_caps);
+ gst_caps_unref (filter_caps);
+ gst_caps_unref (templ_caps);
+
+ if (filter) {
+ GST_LOG_OBJECT (element, "intersecting with %" GST_PTR_FORMAT, filter);
+ filter_caps = gst_caps_intersect (fcaps, filter);
+ gst_caps_unref (fcaps);
+ fcaps = filter_caps;
+ }
+
+done:
+ gst_caps_replace (&allowed, NULL);
+
+ GST_LOG_OBJECT (element, "proxy caps %" GST_PTR_FORMAT, fcaps);
+
+ return fcaps;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
+ * Copyright (C) 2011 Nokia Corporation. All rights reserved.
+ * Contact: Stefan Kost <stefan.kost@nokia.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.
+ */
+
+#ifndef __GST_AUDIO_H__
+#include <gst/audio/audio.h>
+#endif
+
+#ifndef _GST_AUDIO_UTILS_PRIVATE_H_
+#define _GST_AUDIO_UTILS_PRIVATE_H_
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+/* Element utility functions */
+GstCaps *__gst_audio_element_proxy_getcaps (GstElement * element, GstPad * sinkpad,
+ GstPad * srcpad, GstCaps * initial_caps,
+ GstCaps * filter);
+
+G_END_DECLS
+
+#endif
projects / platform / upstream / gstreamer.git / commitdiff