mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-06 10:42:22 +00:00
Original commit message from CVS: * ext/libvisual/visual.c: (gst_visual_getcaps), (gst_visual_src_setcaps), (gst_visual_sink_setcaps): * ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect): * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src), (gst_vorbisenc_convert_sink): * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size), (gst_audio_duration_from_pad_buffer): * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link), (gst_audio_filter_chain): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_setcaps): * gst-libs/gst/video/video.c: (gst_video_frame_rate), (gst_video_get_size): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps): Don't leak references returned by gst_pad_get_parent() (#333663, based on patch by: Christophe Fergeau).
441 lines
13 KiB
C
441 lines
13 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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GST_DEBUG_CATEGORY (audio_rate_debug);
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#define GST_CAT_DEFAULT audio_rate_debug
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#define GST_TYPE_AUDIO_RATE \
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(gst_audio_rate_get_type())
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#define GST_AUDIO_RATE(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RATE,GstAudioRate))
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#define GST_AUDIO_RATE_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RATE,GstAudioRate))
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#define GST_IS_AUDIO_RATE(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RATE))
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#define GST_IS_AUDIO_RATE_CLASS(obj) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RATE))
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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 offset;
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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 audio_rate_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_audio_rate_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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GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)
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);
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static GstStaticPadTemplate gst_audio_rate_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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GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)
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);
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static void gst_audio_rate_base_init (gpointer g_class);
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static void gst_audio_rate_class_init (GstAudioRateClass * klass);
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static void gst_audio_rate_init (GstAudioRate * audiorate);
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static GstFlowReturn gst_audio_rate_chain (GstPad * pad, GstBuffer * buf);
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static void gst_audio_rate_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_audio_rate_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn gst_audio_rate_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_audio_rate_signals[LAST_SIGNAL] = { 0 }; */
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static GType
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gst_audio_rate_get_type (void)
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{
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static GType audio_rate_type = 0;
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if (!audio_rate_type) {
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static const GTypeInfo audio_rate_info = {
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sizeof (GstAudioRateClass),
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gst_audio_rate_base_init,
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NULL,
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(GClassInitFunc) gst_audio_rate_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_audio_rate_init,
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};
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audio_rate_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstAudioRate", &audio_rate_info, 0);
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}
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return audio_rate_type;
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}
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static void
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gst_audio_rate_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, &audio_rate_details);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_audio_rate_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_audio_rate_src_template));
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}
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static void
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gst_audio_rate_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_audio_rate_set_property;
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object_class->get_property = gst_audio_rate_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_audio_rate_change_state;
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}
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static gboolean
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gst_audio_rate_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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gboolean ret = FALSE;
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gint channels, width;
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audiorate = GST_AUDIO_RATE (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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goto beach;
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "channels", &channels) ||
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!gst_structure_get_int (structure, "width", &width)) {
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goto beach;
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}
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audiorate->bytes_per_sample = channels * (width / 8);
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if (audiorate->bytes_per_sample == 0)
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audiorate->bytes_per_sample = 1;
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ret = TRUE;
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beach:
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gst_object_unref (audiorate);
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return ret;
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}
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static void
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gst_audio_rate_init (GstAudioRate * audiorate)
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{
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audiorate->sinkpad =
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gst_pad_new_from_static_template (&gst_audio_rate_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_audio_rate_chain);
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gst_pad_set_setcaps_function (audiorate->sinkpad, gst_audio_rate_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_static_template (&gst_audio_rate_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_audio_rate_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_audio_rate_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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guint in_size;
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GstFlowReturn ret = GST_FLOW_OK;
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audiorate = GST_AUDIO_RATE (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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GST_LOG_OBJECT (audiorate,
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"in_time:%" GST_TIME_FORMAT ", in_duration:%" GST_TIME_FORMAT
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", in_size:%u, in_offset:%lld, in_offset_end:%lld" ", ->next_offset:%lld",
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GST_TIME_ARGS (in_time), GST_TIME_ARGS (in_duration), in_size, in_offset,
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in_offset_end, audiorate->next_offset);
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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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in_offset = audiorate->offset;
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in_offset_end = audiorate->offset + in_size / audiorate->bytes_per_sample;
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GST_WARNING_OBJECT (audiorate, "in_offset:%lld, in_offset_end:%lld",
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in_offset, in_offset_end);
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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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if ((ret = gst_pad_push (audiorate->srcpad, fill) != GST_FLOW_OK))
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goto beach;
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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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goto beach;
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} else {
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guint64 truncsamples;
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guint 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, leftsize);
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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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ret = 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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beach:
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audiorate->offset += in_size / audiorate->bytes_per_sample;
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return ret;
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}
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static void
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gst_audio_rate_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_AUDIO_RATE (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_audio_rate_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_AUDIO_RATE (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_audio_rate_change_state (GstElement * element, GstStateChange transition)
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{
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GstAudioRate *audiorate = GST_AUDIO_RATE (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->offset = 0;
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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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GST_DEBUG_CATEGORY_INIT (audio_rate_debug, "audiorate", 0,
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"AudioRate stream fixer");
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return gst_element_register (plugin, "audiorate", GST_RANK_NONE,
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GST_TYPE_AUDIO_RATE);
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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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