mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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32ba17cd0f
where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there.
433 lines
13 KiB
C
433 lines
13 KiB
C
/* GStreamer
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* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
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*
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* gstaudioparse.c:
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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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/**
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* SECTION:element-audioparse
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*
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* Converts a byte stream into audio frames.
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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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/* FIXME 0.11: suppress warnings for deprecated API such as g_value_array stuff
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* for now with newer GLib versions (>= 2.31.0) */
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#include "gstaudioparse.h"
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#include <string.h>
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typedef enum
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{
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GST_AUDIO_PARSE_FORMAT_RAW,
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GST_AUDIO_PARSE_FORMAT_MULAW,
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GST_AUDIO_PARSE_FORMAT_ALAW
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} GstAudioParseFormat;
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typedef enum
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{
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GST_AUDIO_PARSE_ENDIANNESS_LITTLE = 1234,
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GST_AUDIO_PARSE_ENDIANNESS_BIG = 4321
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} GstAudioParseEndianness;
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static void gst_audio_parse_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audio_parse_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_audio_parse_finalize (GObject * object);
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static GstCaps *gst_audio_parse_get_caps (GstRawParse * rp);
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static void gst_audio_parse_update_frame_size (GstAudioParse * ap);
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GST_DEBUG_CATEGORY_STATIC (gst_audio_parse_debug);
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#define GST_CAT_DEFAULT gst_audio_parse_debug
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enum
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{
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PROP_0,
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PROP_FORMAT,
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PROP_RAW_FORMAT,
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PROP_RATE,
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PROP_CHANNELS,
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PROP_INTERLEAVED,
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PROP_CHANNEL_POSITIONS,
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PROP_USE_SINK_CAPS
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};
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#define GST_AUDIO_PARSE_FORMAT (gst_audio_parse_format_get_type ())
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static GType
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gst_audio_parse_format_get_type (void)
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{
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static GType audio_parse_format_type = 0;
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static const GEnumValue format_types[] = {
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{GST_AUDIO_PARSE_FORMAT_RAW, "Raw", "raw"},
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{GST_AUDIO_PARSE_FORMAT_ALAW, "A-Law", "alaw"},
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{GST_AUDIO_PARSE_FORMAT_MULAW, "\302\265-Law", "mulaw"},
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{0, NULL, NULL}
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};
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if (!audio_parse_format_type) {
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audio_parse_format_type =
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g_enum_register_static ("GstAudioParseFormat", format_types);
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}
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return audio_parse_format_type;
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}
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#define gst_audio_parse_parent_class parent_class
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G_DEFINE_TYPE (GstAudioParse, gst_audio_parse, GST_TYPE_RAW_PARSE);
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static void
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gst_audio_parse_class_init (GstAudioParseClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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GstRawParseClass *rp_class = GST_RAW_PARSE_CLASS (klass);
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GstCaps *caps;
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gobject_class->set_property = gst_audio_parse_set_property;
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gobject_class->get_property = gst_audio_parse_get_property;
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gobject_class->finalize = gst_audio_parse_finalize;
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rp_class->get_caps = gst_audio_parse_get_caps;
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g_object_class_install_property (gobject_class, PROP_FORMAT,
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g_param_spec_enum ("format", "Format",
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"Format of audio samples in raw stream", GST_AUDIO_PARSE_FORMAT,
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GST_AUDIO_PARSE_FORMAT_RAW,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_RAW_FORMAT,
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g_param_spec_enum ("raw-format", "Raw Format",
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"Format of audio samples in raw stream", GST_TYPE_AUDIO_FORMAT,
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GST_AUDIO_FORMAT_S16, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_RATE,
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g_param_spec_int ("rate", "Rate", "Rate of audio samples in raw stream",
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1, INT_MAX, 44100, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_CHANNELS,
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g_param_spec_int ("channels", "Channels",
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"Number of channels in raw stream", 1, 64, 2,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_INTERLEAVED,
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g_param_spec_boolean ("interleaved", "Interleaved Layout",
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"True if audio has interleaved layout", TRUE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_CHANNEL_POSITIONS,
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g_param_spec_value_array ("channel-positions", "Channel positions",
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"Channel positions used on the output",
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g_param_spec_enum ("channel-position", "Channel position",
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"Channel position of the n-th input",
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GST_TYPE_AUDIO_CHANNEL_POSITION,
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GST_AUDIO_CHANNEL_POSITION_NONE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_USE_SINK_CAPS,
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g_param_spec_boolean ("use-sink-caps", "Use sink caps",
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"Use the sink caps for the format, only performing timestamping",
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FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_set_static_metadata (gstelement_class, "Audio Parse",
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"Filter/Audio",
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"Converts stream into audio frames",
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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caps = gst_caps_from_string (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)
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", layout = (string) { interleaved, non-interleaved }; "
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"audio/x-alaw, rate=(int)[1,MAX], channels=(int)[1,MAX]; "
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"audio/x-mulaw, rate=(int)[1,MAX], channels=(int)[1,MAX]");
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gst_raw_parse_class_set_src_pad_template (rp_class, caps);
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gst_raw_parse_class_set_multiple_frames_per_buffer (rp_class, TRUE);
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gst_caps_unref (caps);
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GST_DEBUG_CATEGORY_INIT (gst_audio_parse_debug, "audioparse", 0,
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"audioparse element");
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}
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static void
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gst_audio_parse_init (GstAudioParse * ap)
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{
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ap->format = GST_AUDIO_PARSE_FORMAT_RAW;
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ap->raw_format = GST_AUDIO_FORMAT_S16;
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ap->channels = 2;
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ap->interleaved = TRUE;
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gst_audio_parse_update_frame_size (ap);
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gst_raw_parse_set_fps (GST_RAW_PARSE (ap), 44100, 1);
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}
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static void
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gst_audio_parse_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAudioParse *ap = GST_AUDIO_PARSE (object);
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g_return_if_fail (!gst_raw_parse_is_negotiated (GST_RAW_PARSE (ap)));
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switch (prop_id) {
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case PROP_FORMAT:
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ap->format = g_value_get_enum (value);
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break;
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case PROP_RAW_FORMAT:
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ap->raw_format = g_value_get_enum (value);
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break;
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case PROP_RATE:
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gst_raw_parse_set_fps (GST_RAW_PARSE (ap), g_value_get_int (value), 1);
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break;
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case PROP_CHANNELS:
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ap->channels = g_value_get_int (value);
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break;
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case PROP_INTERLEAVED:
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ap->interleaved = g_value_get_boolean (value);
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break;
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case PROP_CHANNEL_POSITIONS:
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if (ap->channel_positions)
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g_value_array_free (ap->channel_positions);
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ap->channel_positions = g_value_dup_boxed (value);
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break;
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case PROP_USE_SINK_CAPS:
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ap->use_sink_caps = 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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gst_audio_parse_update_frame_size (ap);
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}
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static void
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gst_audio_parse_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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GstAudioParse *ap = GST_AUDIO_PARSE (object);
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switch (prop_id) {
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case PROP_FORMAT:
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g_value_set_enum (value, ap->format);
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break;
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case PROP_RAW_FORMAT:
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g_value_set_enum (value, ap->raw_format);
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break;
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case PROP_RATE:{
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gint fps_n, fps_d;
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gst_raw_parse_get_fps (GST_RAW_PARSE (ap), &fps_n, &fps_d);
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g_value_set_int (value, fps_n);
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break;
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}
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case PROP_CHANNELS:
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g_value_set_int (value, ap->channels);
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break;
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case PROP_INTERLEAVED:
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g_value_set_boolean (value, ap->interleaved);
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break;
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case PROP_CHANNEL_POSITIONS:
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g_value_set_boxed (value, ap->channel_positions);
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break;
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case PROP_USE_SINK_CAPS:
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g_value_set_boolean (value, ap->use_sink_caps);
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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_parse_finalize (GObject * object)
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{
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GstAudioParse *ap = GST_AUDIO_PARSE (object);
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if (ap->channel_positions) {
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g_value_array_free (ap->channel_positions);
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ap->channel_positions = NULL;
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}
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g_free (ap->channel_pos);
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g_free (ap->channel_order);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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void
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gst_audio_parse_update_frame_size (GstAudioParse * ap)
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{
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gint framesize, width;
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switch (ap->format) {
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case GST_AUDIO_PARSE_FORMAT_ALAW:
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case GST_AUDIO_PARSE_FORMAT_MULAW:
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width = 8;
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break;
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case GST_AUDIO_PARSE_FORMAT_RAW:
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default:
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{
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GstAudioInfo info;
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gst_audio_info_init (&info);
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/* rate, etc do not really matter here */
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gst_audio_info_set_format (&info, ap->raw_format, 44100, ap->channels,
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NULL);
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width = GST_AUDIO_INFO_WIDTH (&info);
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break;
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}
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}
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framesize = (width / 8) * ap->channels;
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gst_raw_parse_set_framesize (GST_RAW_PARSE (ap), framesize);
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}
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static GstAudioChannelPosition *
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gst_audio_parse_get_channel_positions (GValueArray * positions)
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{
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gint i;
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guint channels;
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GstAudioChannelPosition *pos;
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channels = positions->n_values;
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pos = g_new (GstAudioChannelPosition, positions->n_values);
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for (i = 0; i < channels; i++) {
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GValue *v = g_value_array_get_nth (positions, i);
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pos[i] = g_value_get_enum (v);
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}
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return pos;
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}
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static void
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gst_audio_parse_setup_channel_positions (GstAudioParse * ap)
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{
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GstAudioChannelPosition *pos, *to;
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g_free (ap->channel_pos);
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g_free (ap->channel_order);
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ap->channel_pos = NULL;
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ap->channel_order = NULL;
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if (!ap->channel_positions) {
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GST_DEBUG_OBJECT (ap, "no channel positions");
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/* implicit mapping for 1- and 2-channel audio is okay */
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/* will come up with one in other cases also */
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return;
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}
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pos = gst_audio_parse_get_channel_positions (ap->channel_positions);
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if (ap->channels != ap->channel_positions->n_values ||
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!gst_audio_check_valid_channel_positions (pos, ap->channels, FALSE)) {
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GST_DEBUG_OBJECT (ap, "invalid channel position");
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g_free (pos);
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return;
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}
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/* ok, got something we can work with now */
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to = g_new (GstAudioChannelPosition, ap->channels);
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memcpy (to, pos, ap->channels * sizeof (to[0]));
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gst_audio_channel_positions_to_valid_order (to, ap->channels);
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ap->channel_pos = pos;
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ap->channel_order = to;
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}
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static GstCaps *
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gst_audio_parse_get_caps (GstRawParse * rp)
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{
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GstAudioParse *ap = GST_AUDIO_PARSE (rp);
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GstCaps *caps, *ncaps;
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GstAudioInfo info;
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gint fps_n, fps_d;
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const GValue *val;
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if (ap->use_sink_caps) {
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gint rate;
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GstCaps *caps = gst_pad_get_current_caps (rp->sinkpad);
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gst_audio_info_from_caps (&info, caps);
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ap->format = GST_AUDIO_PARSE_FORMAT_RAW;
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ap->raw_format = GST_AUDIO_INFO_FORMAT (&info);
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ap->channels = GST_AUDIO_INFO_CHANNELS (&info);
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ap->interleaved = info.layout == GST_AUDIO_LAYOUT_INTERLEAVED;
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rate = GST_AUDIO_INFO_RATE (&info);
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gst_raw_parse_set_fps (GST_RAW_PARSE (ap), rate, 1);
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gst_audio_parse_update_frame_size (ap);
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return caps;
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}
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gst_raw_parse_get_fps (rp, &fps_n, &fps_d);
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gst_audio_parse_setup_channel_positions (ap);
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/* yes, even when format not raw */
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gst_audio_info_init (&info);
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gst_audio_info_set_format (&info, ap->raw_format, fps_n, ap->channels,
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ap->channel_order);
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info.layout = ap->interleaved ? GST_AUDIO_LAYOUT_INTERLEAVED :
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GST_AUDIO_LAYOUT_NON_INTERLEAVED;
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caps = gst_audio_info_to_caps (&info);
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switch (ap->format) {
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case GST_AUDIO_PARSE_FORMAT_RAW:
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break;
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case GST_AUDIO_PARSE_FORMAT_ALAW:
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ncaps = gst_caps_new_simple ("audio/x-alaw",
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"rate", G_TYPE_INT, fps_n,
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"channels", G_TYPE_INT, ap->channels, NULL);
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/* pick mask stuff from faked raw format */
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val = gst_structure_get_value (gst_caps_get_structure (caps, 0),
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"channel-mask");
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if (val)
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gst_caps_set_value (ncaps, "channel-mask", val);
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gst_caps_unref (caps);
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caps = ncaps;
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break;
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case GST_AUDIO_PARSE_FORMAT_MULAW:
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ncaps = gst_caps_new_simple ("audio/x-mulaw",
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"rate", G_TYPE_INT, fps_n,
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"channels", G_TYPE_INT, ap->channels, NULL);
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/* pick mask stuff from faked raw format */
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val = gst_structure_get_value (gst_caps_get_structure (caps, 0),
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"channel-mask");
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if (val)
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gst_caps_set_value (ncaps, "channel-mask", val);
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gst_caps_unref (caps);
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caps = ncaps;
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break;
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default:
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caps = gst_caps_new_empty ();
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GST_ERROR_OBJECT (rp, "unexpected format %d", ap->format);
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break;
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}
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return caps;
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}
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