mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 07:47:17 +00:00
audioencoder: refactor getcaps proxy function to be reusable
Makes the audioencoder's getcaps function that proxies downstream restriction available to other elements in the audio module to use it
This commit is contained in:
parent
8085352fb3
commit
5e3405bd08
4 changed files with 187 additions and 65 deletions
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@ -38,6 +38,7 @@ libgstaudio_@GST_API_VERSION@_la_SOURCES = \
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gstaudiometa.c \
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gstaudiometa.c \
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gstaudiosink.c \
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gstaudiosink.c \
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gstaudiosrc.c \
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gstaudiosrc.c \
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gstaudioutilsprivate.c \
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streamvolume.c \
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streamvolume.c \
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gstaudioiec61937.c
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gstaudioiec61937.c
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@ -64,7 +65,8 @@ libgstaudio_@GST_API_VERSION@include_HEADERS = \
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gstaudioiec61937.h
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gstaudioiec61937.h
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nodist_libgstaudio_@GST_API_VERSION@include_HEADERS = \
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nodist_libgstaudio_@GST_API_VERSION@include_HEADERS = \
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audio-enumtypes.h
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audio-enumtypes.h \
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gstaudioutilsprivate.h
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libgstaudio_@GST_API_VERSION@_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) \
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libgstaudio_@GST_API_VERSION@_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) \
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$(ORC_CFLAGS)
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$(ORC_CFLAGS)
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@ -152,6 +152,7 @@
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#endif
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#endif
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#include "gstaudioencoder.h"
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#include "gstaudioencoder.h"
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#include "gstaudioutilsprivate.h"
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#include <gst/base/gstadapter.h>
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#include <gst/base/gstadapter.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/audio.h>
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#include <gst/pbutils/descriptions.h>
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#include <gst/pbutils/descriptions.h>
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@ -1359,70 +1360,9 @@ GstCaps *
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gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc, GstCaps * caps,
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gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc, GstCaps * caps,
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GstCaps * filter)
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GstCaps * filter)
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{
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{
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GstCaps *templ_caps = NULL;
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return __gst_audio_element_proxy_getcaps (GST_ELEMENT_CAST (enc),
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GstCaps *allowed = NULL;
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GST_AUDIO_ENCODER_SINK_PAD (enc), GST_AUDIO_ENCODER_SRC_PAD (enc),
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GstCaps *fcaps, *filter_caps;
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caps, filter);
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gint i, j;
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/* we want to be able to communicate to upstream elements like audioconvert
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* and audioresample any rate/channel restrictions downstream (e.g. muxer
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* only accepting certain sample rates) */
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templ_caps =
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caps ? gst_caps_ref (caps) : gst_pad_get_pad_template_caps (enc->sinkpad);
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allowed = gst_pad_get_allowed_caps (enc->srcpad);
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if (!allowed || gst_caps_is_empty (allowed) || gst_caps_is_any (allowed)) {
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fcaps = templ_caps;
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goto done;
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}
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GST_LOG_OBJECT (enc, "template caps %" GST_PTR_FORMAT, templ_caps);
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GST_LOG_OBJECT (enc, "allowed caps %" GST_PTR_FORMAT, allowed);
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filter_caps = gst_caps_new_empty ();
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for (i = 0; i < gst_caps_get_size (templ_caps); i++) {
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GQuark q_name;
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q_name = gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
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/* pick rate + channel fields from allowed caps */
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for (j = 0; j < gst_caps_get_size (allowed); j++) {
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const GstStructure *allowed_s = gst_caps_get_structure (allowed, j);
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const GValue *val;
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GstStructure *s;
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s = gst_structure_new_id_empty (q_name);
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if ((val = gst_structure_get_value (allowed_s, "rate")))
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gst_structure_set_value (s, "rate", val);
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if ((val = gst_structure_get_value (allowed_s, "channels")))
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gst_structure_set_value (s, "channels", val);
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/* following might also make sense for some encoded formats,
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* e.g. wavpack */
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if ((val = gst_structure_get_value (allowed_s, "channel-mask")))
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gst_structure_set_value (s, "channel-mask", val);
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filter_caps = gst_caps_merge_structure (filter_caps, s);
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}
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}
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fcaps = gst_caps_intersect (filter_caps, templ_caps);
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gst_caps_unref (filter_caps);
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gst_caps_unref (templ_caps);
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if (filter) {
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GST_LOG_OBJECT (enc, "intersecting with %" GST_PTR_FORMAT, filter);
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filter_caps = gst_caps_intersect_full (filter, fcaps,
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GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (fcaps);
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fcaps = filter_caps;
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}
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done:
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gst_caps_replace (&allowed, NULL);
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GST_LOG_OBJECT (enc, "proxy caps %" GST_PTR_FORMAT, fcaps);
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return fcaps;
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}
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}
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static GstCaps *
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static GstCaps *
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140
gst-libs/gst/audio/gstaudioutilsprivate.c
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140
gst-libs/gst/audio/gstaudioutilsprivate.c
Normal file
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@ -0,0 +1,140 @@
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/* GStreamer
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* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
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* Copyright (C) 2011 Nokia Corporation. All rights reserved.
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* Contact: Stefan Kost <stefan.kost@nokia.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 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,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* 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
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* License along with this library; if not, write to the
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* 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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/audio/audio.h>
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#include "gstaudioutilsprivate.h"
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/*
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* Takes caps and copies its audio fields to tmpl_caps
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*/
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static GstCaps *
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__gst_audio_element_proxy_caps (GstElement * element, GstCaps * templ_caps,
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GstCaps * caps)
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{
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GstCaps *result = gst_caps_new_empty ();
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gint i, j;
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gint templ_caps_size = gst_caps_get_size (templ_caps);
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gint caps_size = gst_caps_get_size (caps);
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for (i = 0; i < templ_caps_size; i++) {
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GQuark q_name =
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gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
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for (j = 0; j < caps_size; j++) {
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const GstStructure *caps_s = gst_caps_get_structure (caps, j);
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const GValue *val;
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GstStructure *s;
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s = gst_structure_new_id_empty (q_name);
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if ((val = gst_structure_get_value (caps_s, "rate")))
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gst_structure_set_value (s, "rate", val);
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if ((val = gst_structure_get_value (caps_s, "channels")))
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gst_structure_set_value (s, "channels", val);
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if ((val = gst_structure_get_value (caps_s, "channels-mask")))
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gst_structure_set_value (s, "channels-mask", val);
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result = gst_caps_merge_structure (result, s);
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}
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}
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return result;
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}
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/**
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* __gst_audio_element_proxy_getcaps:
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* @element: a #GstElement
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* @sinkpad: the element's sink #GstPad
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* @srcpad: the element's source #GstPad
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* @initial_caps: initial caps
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* @filter: filter caps
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*
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* Returns caps that express @initial_caps (or sink template caps if
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* @initial_caps == NULL) restricted to rate/channels/...
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* combinations supported by downstream elements (e.g. muxers).
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*
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* Returns: a #GstCaps owned by caller
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*/
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GstCaps *
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__gst_audio_element_proxy_getcaps (GstElement * element, GstPad * sinkpad,
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GstPad * srcpad, GstCaps * initial_caps, GstCaps * filter)
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{
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GstCaps *templ_caps, *src_templ_caps;
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GstCaps *peer_caps;
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GstCaps *allowed;
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GstCaps *fcaps, *filter_caps;
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/* Allow downstream to specify rate/channels constraints
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* and forward them upstream for audio converters to handle
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*/
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templ_caps = initial_caps ? gst_caps_ref (initial_caps) :
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gst_pad_get_pad_template_caps (sinkpad);
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src_templ_caps = gst_pad_get_pad_template_caps (srcpad);
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if (filter && !gst_caps_is_any (filter)) {
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GstCaps *proxy_filter =
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__gst_audio_element_proxy_caps (element, src_templ_caps, filter);
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peer_caps = gst_pad_peer_query_caps (srcpad, proxy_filter);
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gst_caps_unref (proxy_filter);
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} else {
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peer_caps = gst_pad_peer_query_caps (srcpad, NULL);
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}
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allowed = gst_caps_intersect_full (peer_caps, src_templ_caps,
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GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (src_templ_caps);
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gst_caps_unref (peer_caps);
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if (!allowed || gst_caps_is_any (allowed)) {
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fcaps = templ_caps;
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goto done;
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} else if (gst_caps_is_empty (allowed)) {
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fcaps = gst_caps_ref (allowed);
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goto done;
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}
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GST_LOG_OBJECT (element, "template caps %" GST_PTR_FORMAT, templ_caps);
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GST_LOG_OBJECT (element, "allowed caps %" GST_PTR_FORMAT, allowed);
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filter_caps = __gst_audio_element_proxy_caps (element, templ_caps, allowed);
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fcaps = gst_caps_intersect (filter_caps, templ_caps);
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gst_caps_unref (filter_caps);
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gst_caps_unref (templ_caps);
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if (filter) {
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GST_LOG_OBJECT (element, "intersecting with %" GST_PTR_FORMAT, filter);
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filter_caps = gst_caps_intersect (fcaps, filter);
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gst_caps_unref (fcaps);
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fcaps = filter_caps;
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}
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done:
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gst_caps_replace (&allowed, NULL);
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GST_LOG_OBJECT (element, "proxy caps %" GST_PTR_FORMAT, fcaps);
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return fcaps;
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}
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40
gst-libs/gst/audio/gstaudioutilsprivate.h
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40
gst-libs/gst/audio/gstaudioutilsprivate.h
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@ -0,0 +1,40 @@
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/* GStreamer
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* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
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* Copyright (C) 2011 Nokia Corporation. All rights reserved.
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* Contact: Stefan Kost <stefan.kost@nokia.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 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,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* 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
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* License along with this library; if not, write to the
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* 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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#ifndef __GST_AUDIO_H__
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#include <gst/audio/audio.h>
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#endif
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#ifndef _GST_AUDIO_UTILS_PRIVATE_H_
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#define _GST_AUDIO_UTILS_PRIVATE_H_
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#include <gst/gst.h>
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G_BEGIN_DECLS
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/* Element utility functions */
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GstCaps *__gst_audio_element_proxy_getcaps (GstElement * element, GstPad * sinkpad,
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GstPad * srcpad, GstCaps * initial_caps,
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GstCaps * filter);
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G_END_DECLS
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#endif
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