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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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aa89ae8beb
There are three instances where in- and outbuf have been swapped. This didn't affect the correctness of the libs *filter code, but the videoscale implementation swapped the arguments of meta->transform_func. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/853>
238 lines
7.4 KiB
C
238 lines
7.4 KiB
C
/* GStreamer audio filter base class
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2003> David Schleef <ds@schleef.org>
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* Copyright (C) <2007> Tim-Philipp Müller <tim centricular net>
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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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/**
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* SECTION:gstaudiofilter
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* @title: GstAudioFilter
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* @short_description: Base class for simple audio filters
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*
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* #GstAudioFilter is a #GstBaseTransform<!-- -->-derived base class for simple audio
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* filters, ie. those that output the same format that they get as input.
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*
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* #GstAudioFilter will parse the input format for you (with error checking)
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* before calling your setup function. Also, elements deriving from
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* #GstAudioFilter may use gst_audio_filter_class_add_pad_templates() from
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* their class_init function to easily configure the set of caps/formats that
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* the element is able to handle.
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*
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* Derived classes should override the #GstAudioFilterClass.setup() and
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* #GstBaseTransformClass.transform_ip() and/or
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* #GstBaseTransformClass.transform()
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* virtual functions in their class_init function.
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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 "gstaudiofilter.h"
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#include <string.h>
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GST_DEBUG_CATEGORY_STATIC (audiofilter_dbg);
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#define GST_CAT_DEFAULT audiofilter_dbg
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/* cached quark to avoid contention on the global quark table lock */
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#define META_TAG_AUDIO meta_tag_audio_quark
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static GQuark meta_tag_audio_quark;
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static GstStateChangeReturn gst_audio_filter_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_audio_filter_set_caps (GstBaseTransform * btrans,
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GstCaps * incaps, GstCaps * outcaps);
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static gboolean gst_audio_filter_get_unit_size (GstBaseTransform * btrans,
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GstCaps * caps, gsize * size);
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static GstFlowReturn gst_audio_filter_submit_input_buffer (GstBaseTransform *
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btrans, gboolean is_discont, GstBuffer * input);
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#define do_init G_STMT_START { \
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GST_DEBUG_CATEGORY_INIT (audiofilter_dbg, "audiofilter", 0, "audiofilter"); \
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} G_STMT_END
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G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstAudioFilter, gst_audio_filter,
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GST_TYPE_BASE_TRANSFORM, do_init);
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static gboolean
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gst_audio_filter_transform_meta (GstBaseTransform * trans, GstBuffer * outbuf,
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GstMeta * meta, GstBuffer * inbuf)
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{
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const GstMetaInfo *info = meta->info;
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const gchar *const *tags;
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tags = gst_meta_api_type_get_tags (info->api);
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if (!tags || (g_strv_length ((gchar **) tags) == 1
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&& gst_meta_api_type_has_tag (info->api, META_TAG_AUDIO)))
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return TRUE;
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return
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GST_BASE_TRANSFORM_CLASS (gst_audio_filter_parent_class)->transform_meta
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(trans, outbuf, meta, inbuf);
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}
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static void
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gst_audio_filter_class_init (GstAudioFilterClass * klass)
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{
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GstBaseTransformClass *basetrans_class = (GstBaseTransformClass *) klass;
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_audio_filter_change_state);
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basetrans_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_filter_set_caps);
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basetrans_class->get_unit_size =
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GST_DEBUG_FUNCPTR (gst_audio_filter_get_unit_size);
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basetrans_class->transform_meta = gst_audio_filter_transform_meta;
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basetrans_class->submit_input_buffer = gst_audio_filter_submit_input_buffer;
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meta_tag_audio_quark = g_quark_from_static_string (GST_META_TAG_AUDIO_STR);
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}
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static void
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gst_audio_filter_init (GstAudioFilter * self)
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{
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gst_audio_info_init (&self->info);
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}
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/* we override the state change vfunc here instead of GstBaseTransform's stop
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* vfunc, so GstAudioFilter-derived elements can override ::stop() for their
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* own purposes without having to worry about chaining up */
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static GstStateChangeReturn
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gst_audio_filter_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret;
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GstAudioFilter *filter = GST_AUDIO_FILTER (element);
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ret =
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GST_ELEMENT_CLASS (gst_audio_filter_parent_class)->change_state (element,
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transition);
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if (ret == GST_STATE_CHANGE_FAILURE)
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return ret;
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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case GST_STATE_CHANGE_READY_TO_NULL:
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gst_audio_info_init (&filter->info);
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break;
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default:
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break;
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}
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return ret;
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}
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static gboolean
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gst_audio_filter_set_caps (GstBaseTransform * btrans, GstCaps * incaps,
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GstCaps * outcaps)
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{
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GstAudioFilterClass *klass;
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GstAudioFilter *filter = GST_AUDIO_FILTER (btrans);
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GstAudioInfo info;
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gboolean ret = TRUE;
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GST_LOG_OBJECT (filter, "caps: %" GST_PTR_FORMAT, incaps);
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GST_LOG_OBJECT (filter, "info: %d", GST_AUDIO_FILTER_RATE (filter));
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if (!gst_audio_info_from_caps (&info, incaps))
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goto invalid_format;
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klass = GST_AUDIO_FILTER_GET_CLASS (filter);
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if (klass->setup)
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ret = klass->setup (filter, &info);
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if (ret) {
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filter->info = info;
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GST_LOG_OBJECT (filter, "configured caps: %" GST_PTR_FORMAT, incaps);
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}
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return ret;
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/* ERROR */
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invalid_format:
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{
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GST_WARNING_OBJECT (filter, "couldn't parse %" GST_PTR_FORMAT, incaps);
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return FALSE;
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}
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}
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static GstFlowReturn
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gst_audio_filter_submit_input_buffer (GstBaseTransform * btrans,
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gboolean is_discont, GstBuffer * input)
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{
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GstAudioFilter *filter = GST_AUDIO_FILTER (btrans);
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if (btrans->segment.format == GST_FORMAT_TIME) {
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input =
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gst_audio_buffer_clip (input, &btrans->segment, filter->info.rate,
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filter->info.bpf);
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if (!input)
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return GST_FLOW_OK;
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}
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return
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GST_BASE_TRANSFORM_CLASS
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(gst_audio_filter_parent_class)->submit_input_buffer (btrans, is_discont,
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input);
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}
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static gboolean
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gst_audio_filter_get_unit_size (GstBaseTransform * btrans, GstCaps * caps,
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gsize * size)
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{
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GstAudioInfo info;
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if (!gst_audio_info_from_caps (&info, caps))
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return FALSE;
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*size = GST_AUDIO_INFO_BPF (&info);
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return TRUE;
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}
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/**
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* gst_audio_filter_class_add_pad_templates:
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* @klass: an #GstAudioFilterClass
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* @allowed_caps: what formats the filter can handle, as #GstCaps
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*
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* Convenience function to add pad templates to this element class, with
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* @allowed_caps as the caps that can be handled.
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*
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* This function is usually used from within a GObject class_init function.
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*/
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void
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gst_audio_filter_class_add_pad_templates (GstAudioFilterClass * klass,
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GstCaps * allowed_caps)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstPadTemplate *pad_template;
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g_return_if_fail (GST_IS_AUDIO_FILTER_CLASS (klass));
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g_return_if_fail (GST_IS_CAPS (allowed_caps));
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pad_template = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
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allowed_caps);
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gst_element_class_add_pad_template (element_class, pad_template);
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pad_template = gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
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allowed_caps);
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gst_element_class_add_pad_template (element_class, pad_template);
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}
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