mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-28 12:41:05 +00:00
4f8f42b0b6
Original commit message from CVS: restructure configure.ac, use correct libtool LDFLAGS, fix up defines
415 lines
12 KiB
C
415 lines
12 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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 <string.h>
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#define GST_TYPE_AUDIORATE \
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(gst_audiorate_get_type())
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#define GST_AUDIORATE(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORATE,GstAudiorate))
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#define GST_AUDIORATE_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORATE,GstAudiorate))
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#define GST_IS_AUDIORATE(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORATE))
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#define GST_IS_AUDIORATE_CLASS(obj) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORATE))
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typedef struct _GstAudiorate GstAudiorate;
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typedef struct _GstAudiorateClass GstAudiorateClass;
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struct _GstAudiorate
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{
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GstElement element;
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GstPad *sinkpad, *srcpad;
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gint bytes_per_sample;
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/* audio state */
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guint64 next_offset;
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guint64 in, out, add, drop;
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gboolean silent;
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};
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struct _GstAudiorateClass
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{
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GstElementClass parent_class;
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};
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/* elementfactory information */
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static GstElementDetails audiorate_details =
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GST_ELEMENT_DETAILS ("Audio rate adjuster",
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"Filter/Effect/Audio",
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"Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream",
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"Wim Taymans <wim@fluendo.com>");
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/* GstAudiorate signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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#define DEFAULT_SILENT TRUE
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enum
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{
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ARG_0,
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ARG_IN,
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ARG_OUT,
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ARG_ADD,
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ARG_DROP,
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ARG_SILENT,
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/* FILL ME */
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};
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static GstStaticPadTemplate gst_audiorate_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
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);
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static GstStaticPadTemplate gst_audiorate_sink_template =
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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 (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
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);
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static void gst_audiorate_base_init (gpointer g_class);
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static void gst_audiorate_class_init (GstAudiorateClass * klass);
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static void gst_audiorate_init (GstAudiorate * audiorate);
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static GstFlowReturn gst_audiorate_chain (GstPad * pad, GstBuffer * buf);
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static void gst_audiorate_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_audiorate_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn gst_audiorate_change_state (GstElement * element,
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GstStateChange transition);
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static GstElementClass *parent_class = NULL;
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/*static guint gst_audiorate_signals[LAST_SIGNAL] = { 0 }; */
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static GType
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gst_audiorate_get_type (void)
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{
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static GType audiorate_type = 0;
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if (!audiorate_type) {
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static const GTypeInfo audiorate_info = {
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sizeof (GstAudiorateClass),
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gst_audiorate_base_init,
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NULL,
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(GClassInitFunc) gst_audiorate_class_init,
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NULL,
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NULL,
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sizeof (GstAudiorate),
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0,
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(GInstanceInitFunc) gst_audiorate_init,
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};
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audiorate_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstAudiorate", &audiorate_info, 0);
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}
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return audiorate_type;
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}
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static void
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gst_audiorate_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_set_details (element_class, &audiorate_details);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_audiorate_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_audiorate_src_template));
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}
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static void
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gst_audiorate_class_init (GstAudiorateClass * klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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parent_class = g_type_class_peek_parent (klass);
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object_class->set_property = gst_audiorate_set_property;
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object_class->get_property = gst_audiorate_get_property;
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g_object_class_install_property (object_class, ARG_IN,
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g_param_spec_uint64 ("in", "In",
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"Number of input samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
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g_object_class_install_property (object_class, ARG_OUT,
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g_param_spec_uint64 ("out", "Out",
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"Number of output samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
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g_object_class_install_property (object_class, ARG_ADD,
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g_param_spec_uint64 ("add", "Add",
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"Number of added samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
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g_object_class_install_property (object_class, ARG_DROP,
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g_param_spec_uint64 ("drop", "Drop",
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"Number of dropped samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
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g_object_class_install_property (object_class, ARG_SILENT,
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g_param_spec_boolean ("silent", "silent",
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"Don't emit notify for dropped and duplicated frames",
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DEFAULT_SILENT, G_PARAM_READWRITE));
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element_class->change_state = gst_audiorate_change_state;
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}
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static gboolean
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gst_audiorate_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstAudiorate *audiorate;
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GstStructure *structure;
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GstPad *otherpad;
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gint ret, channels, depth;
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audiorate = GST_AUDIORATE (gst_pad_get_parent (pad));
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otherpad = (pad == audiorate->srcpad) ? audiorate->sinkpad :
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audiorate->srcpad;
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if (!gst_pad_set_caps (otherpad, caps))
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return FALSE;
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structure = gst_caps_get_structure (caps, 0);
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ret = gst_structure_get_int (structure, "channels", &channels);
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ret &= gst_structure_get_int (structure, "depth", &depth);
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if (!ret)
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return FALSE;
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audiorate->bytes_per_sample = channels * (depth / 8);
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if (audiorate->bytes_per_sample == 0)
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audiorate->bytes_per_sample = 1;
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return TRUE;
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}
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static void
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gst_audiorate_init (GstAudiorate * audiorate)
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{
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audiorate->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_audiorate_sink_template), "sink");
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gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad);
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gst_pad_set_chain_function (audiorate->sinkpad, gst_audiorate_chain);
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gst_pad_set_setcaps_function (audiorate->sinkpad, gst_audiorate_setcaps);
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gst_pad_set_getcaps_function (audiorate->sinkpad, gst_pad_proxy_getcaps);
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audiorate->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_audiorate_src_template), "src");
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gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad);
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gst_pad_set_setcaps_function (audiorate->srcpad, gst_audiorate_setcaps);
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gst_pad_set_getcaps_function (audiorate->srcpad, gst_pad_proxy_getcaps);
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audiorate->bytes_per_sample = 1;
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audiorate->in = 0;
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audiorate->out = 0;
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audiorate->drop = 0;
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audiorate->add = 0;
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audiorate->silent = DEFAULT_SILENT;
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}
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static GstFlowReturn
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gst_audiorate_chain (GstPad * pad, GstBuffer * buf)
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{
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GstAudiorate *audiorate;
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GstClockTime in_time, in_duration;
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guint64 in_offset, in_offset_end;
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gint in_size;
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audiorate = GST_AUDIORATE (gst_pad_get_parent (pad));
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audiorate->in++;
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in_time = GST_BUFFER_TIMESTAMP (buf);
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in_duration = GST_BUFFER_DURATION (buf);
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in_size = GST_BUFFER_SIZE (buf);
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in_offset = GST_BUFFER_OFFSET (buf);
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in_offset_end = GST_BUFFER_OFFSET_END (buf);
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if (in_offset == GST_CLOCK_TIME_NONE || in_offset_end == GST_CLOCK_TIME_NONE) {
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GST_WARNING_OBJECT (audiorate, "audiorate got buffer without offsets");
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}
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/* do we need to insert samples */
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if (in_offset > audiorate->next_offset) {
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GstBuffer *fill;
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gint fillsize;
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guint64 fillsamples;
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fillsamples = in_offset - audiorate->next_offset;
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fillsize = fillsamples * audiorate->bytes_per_sample;
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fill = gst_buffer_new_and_alloc (fillsize);
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memset (GST_BUFFER_DATA (fill), 0, fillsize);
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GST_LOG_OBJECT (audiorate, "inserting %lld samples", fillsamples);
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GST_BUFFER_DURATION (fill) = in_duration * fillsize / in_size;
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GST_BUFFER_TIMESTAMP (fill) = in_time - GST_BUFFER_DURATION (fill);
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GST_BUFFER_OFFSET (fill) = audiorate->next_offset;
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GST_BUFFER_OFFSET_END (fill) = in_offset;
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gst_pad_push (audiorate->srcpad, fill);
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audiorate->out++;
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audiorate->add += fillsamples;
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if (!audiorate->silent)
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g_object_notify (G_OBJECT (audiorate), "add");
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} else if (in_offset < audiorate->next_offset) {
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/* need to remove samples */
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if (in_offset_end <= audiorate->next_offset) {
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guint64 drop = in_size / audiorate->bytes_per_sample;
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audiorate->drop += drop;
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GST_LOG_OBJECT (audiorate, "dropping %lld samples", drop);
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/* we can drop the buffer completely */
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gst_buffer_unref (buf);
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if (!audiorate->silent)
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g_object_notify (G_OBJECT (audiorate), "drop");
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return GST_FLOW_OK;
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} else {
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guint64 truncsamples, truncsize, leftsize;
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GstBuffer *trunc;
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/* truncate buffer */
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truncsamples = audiorate->next_offset - in_offset;
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truncsize = truncsamples * audiorate->bytes_per_sample;
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leftsize = in_size - truncsize;
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trunc = gst_buffer_create_sub (buf, truncsize, in_size);
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GST_BUFFER_DURATION (trunc) = in_duration * leftsize / in_size;
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GST_BUFFER_TIMESTAMP (trunc) =
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in_time + in_duration - GST_BUFFER_DURATION (trunc);
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GST_BUFFER_OFFSET (trunc) = audiorate->next_offset;
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GST_BUFFER_OFFSET_END (trunc) = in_offset_end;
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GST_LOG_OBJECT (audiorate, "truncating %lld samples", truncsamples);
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gst_buffer_unref (buf);
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buf = trunc;
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audiorate->drop += truncsamples;
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}
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}
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gst_pad_push (audiorate->srcpad, buf);
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audiorate->out++;
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audiorate->next_offset = in_offset_end;
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return GST_FLOW_OK;
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}
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static void
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gst_audiorate_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec)
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{
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GstAudiorate *audiorate = GST_AUDIORATE (object);
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switch (prop_id) {
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case ARG_SILENT:
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audiorate->silent = g_value_get_boolean (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_audiorate_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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GstAudiorate *audiorate = GST_AUDIORATE (object);
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switch (prop_id) {
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case ARG_IN:
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g_value_set_uint64 (value, audiorate->in);
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break;
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case ARG_OUT:
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g_value_set_uint64 (value, audiorate->out);
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break;
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case ARG_ADD:
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g_value_set_uint64 (value, audiorate->add);
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break;
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case ARG_DROP:
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g_value_set_uint64 (value, audiorate->drop);
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break;
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case ARG_SILENT:
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g_value_set_boolean (value, audiorate->silent);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstStateChangeReturn
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gst_audiorate_change_state (GstElement * element, GstStateChange transition)
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{
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GstAudiorate *audiorate = GST_AUDIORATE (element);
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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audiorate->next_offset = 0;
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break;
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default:
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break;
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}
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if (parent_class->change_state)
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return parent_class->change_state (element, transition);
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return GST_STATE_CHANGE_SUCCESS;
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}
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "audiorate", GST_RANK_NONE,
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GST_TYPE_AUDIORATE);
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}
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GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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"audiorate",
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"Adjusts audio frames",
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plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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