mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-11 18:05:37 +00:00
Move siren rtp pay/depay to good
This commit is contained in:
Olivier Crête
Edward Hervey
parent
158e8692c5
commit
ce558ab988
6 changed files with 1 additions and 426 deletions
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@ -3,8 +3,6 @@ plugin_LTLIBRARIES = libgstsiren.la
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libgstsiren_la_SOURCES = gstsiren.c \
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gstsirendec.c gstsirendec.h \
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gstsirenenc.c gstsirenenc.h \
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gstrtpsirenpay.c gstrtpsirenpay.h \
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gstrtpsirendepay.c gstrtpsirendepay.h \
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common.c dct4.c decoder.c encoder.c huffman.c \
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huffman.h rmlt.h common.h dct4.h decoder.h \
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encoder.h huffman_consts.h rmlt.c siren7.h
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@ -1,134 +0,0 @@
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/*
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* Siren Depayloader Gst Element
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*
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* @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 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 <stdlib.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpsirendepay.h"
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/* elementfactory information */
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static const GstElementDetails gst_rtp_siren_depay_details =
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GST_ELEMENT_DETAILS ("RTP Siren packet depayloader",
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"Codec/Depayloader/Network",
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"Extracts Siren audio from RTP packets",
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"Philippe Kalaf <philippe.kalaf@collabora.co.uk>");
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static GstStaticPadTemplate gst_rtp_siren_depay_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 ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 16000, "
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"encoding-name = (string) \"SIREN\", "
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"dct-length = (int) 320")
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);
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static GstStaticPadTemplate gst_rtp_siren_depay_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 ("audio/x-siren, " "dct-length = (int) 320")
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);
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static GstBuffer *gst_rtp_siren_depay_process (GstBaseRTPDepayload * depayload,
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GstBuffer * buf);
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static gboolean gst_rtp_siren_depay_setcaps (GstBaseRTPDepayload * depayload,
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GstCaps * caps);
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GST_BOILERPLATE (GstRTPSirenDepay, gst_rtp_siren_depay, GstBaseRTPDepayload,
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GST_TYPE_BASE_RTP_DEPAYLOAD);
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static void
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gst_rtp_siren_depay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_siren_depay_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_siren_depay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_siren_depay_details);
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}
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static void
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gst_rtp_siren_depay_class_init (GstRTPSirenDepayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
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gstbasertpdepayload_class->process = gst_rtp_siren_depay_process;
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gstbasertpdepayload_class->set_caps = gst_rtp_siren_depay_setcaps;
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}
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static void
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gst_rtp_siren_depay_init (GstRTPSirenDepay * rtpsirendepay,
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GstRTPSirenDepayClass * klass)
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{
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}
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static gboolean
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gst_rtp_siren_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
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{
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GstCaps *srccaps;
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gboolean ret;
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srccaps = gst_caps_new_simple ("audio/x-siren",
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"dct-length", G_TYPE_INT, 320, NULL);
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ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
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GST_DEBUG ("set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
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gst_caps_unref (srccaps);
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/* always fixed clock rate of 16000 */
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depayload->clock_rate = 16000;
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return ret;
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}
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static GstBuffer *
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gst_rtp_siren_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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{
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GstBuffer *outbuf;
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outbuf = gst_rtp_buffer_get_payload_buffer (buf);
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return outbuf;
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}
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gboolean
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gst_rtp_siren_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpsirendepay",
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GST_RANK_MARGINAL, GST_TYPE_RTP_SIREN_DEPAY);
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}
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@ -1,60 +0,0 @@
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/*
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* Siren Depayloader Gst Element
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*
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* @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef __GST_RTP_SIREN_DEPAY_H__
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#define __GST_RTP_SIREN_DEPAY_H__
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#include <gst/gst.h>
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#include <gst/rtp/gstbasertpdepayload.h>
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G_BEGIN_DECLS
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typedef struct _GstRTPSirenDepay GstRTPSirenDepay;
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typedef struct _GstRTPSirenDepayClass GstRTPSirenDepayClass;
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#define GST_TYPE_RTP_SIREN_DEPAY \
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(gst_rtp_siren_depay_get_type())
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#define GST_RTP_SIREN_DEPAY(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SIREN_DEPAY,GstRTPSirenDepay))
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#define GST_RTP_SIREN_DEPAY_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SIREN_DEPAY,GstRTPSirenDepayClass))
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#define GST_IS_RTP_SIREN_DEPAY(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SIREN_DEPAY))
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#define GST_IS_RTP_SIREN_DEPAY_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SIREN_DEPAY))
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struct _GstRTPSirenDepay
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{
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GstBaseRTPDepayload depayload;
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};
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struct _GstRTPSirenDepayClass
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{
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GstBaseRTPDepayloadClass parent_class;
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};
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gboolean gst_rtp_siren_depay_plugin_init (GstPlugin * plugin);
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G_END_DECLS
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#endif /* __GST_RTP_SIREN_DEPAY_H__ */
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@ -1,163 +0,0 @@
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/*
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* Siren Payloader Gst Element
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*
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* @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 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 "gstrtpsirenpay.h"
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#include <gst/rtp/gstrtpbuffer.h>
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/* elementfactory information */
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static GstElementDetails gst_rtpsirenpay_details = {
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"RTP Payloader for Siren Audio",
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"Codec/Payloader/Network",
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"Packetize Siren audio streams into RTP packets",
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"Youness Alaoui <kakaroto@kakaroto.homelinux.net>"
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};
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GST_DEBUG_CATEGORY_STATIC (rtpsirenpay_debug);
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#define GST_CAT_DEFAULT (rtpsirenpay_debug)
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static GstStaticPadTemplate gst_rtpsirenpay_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 ("audio/x-siren, " "dct-length = (int) 320")
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);
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static GstStaticPadTemplate gst_rtpsirenpay_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 ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 16000, "
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"encoding-name = (string) \"SIREN\", "
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"dct-length = (int) 320")
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);
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static gboolean gst_rtpsirenpay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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GST_BOILERPLATE (GstRTPSirenPay, gst_rtpsirenpay, GstBaseRTPAudioPayload,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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static void
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gst_rtpsirenpay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtpsirenpay_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtpsirenpay_src_template));
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gst_element_class_set_details (element_class, &gst_rtpsirenpay_details);
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}
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static void
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gst_rtpsirenpay_class_init (GstRTPSirenPayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
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gstbasertppayload_class->set_caps = gst_rtpsirenpay_setcaps;
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GST_DEBUG_CATEGORY_INIT (rtpsirenpay_debug, "rtpsirenpay", 0,
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"siren audio RTP payloader");
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}
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static void
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gst_rtpsirenpay_init (GstRTPSirenPay * rtpsirenpay, GstRTPSirenPayClass * klass)
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{
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GstBaseRTPPayload *basertppayload;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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basertppayload = GST_BASE_RTP_PAYLOAD (rtpsirenpay);
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpsirenpay);
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/* we don't set the payload type, it should be set by the application using
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* the pt property or the default 96 will be used */
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basertppayload->clock_rate = 16000;
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/* tell basertpaudiopayload that this is a frame based codec */
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gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
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}
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static gboolean
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gst_rtpsirenpay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
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{
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GstRTPSirenPay *rtpsirenpay;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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gboolean ret;
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gint dct_length;
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GstStructure *structure;
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const char *payload_name;
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rtpsirenpay = GST_RTP_SIREN_PAY (basertppayload);
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (structure, "dct-length", &dct_length);
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if (dct_length != 320)
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goto wrong_dct;
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payload_name = gst_structure_get_name (structure);
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if (g_strcasecmp ("audio/x-siren", payload_name))
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goto wrong_caps;
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gst_basertppayload_set_options (basertppayload, "audio", TRUE, "SIREN", 16000);
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/* set options for this frame based audio codec */
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gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 20, 40);
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ret = gst_basertppayload_set_outcaps (basertppayload, NULL);
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return TRUE;
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/* ERRORS */
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wrong_dct:
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{
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GST_ERROR_OBJECT (rtpsirenpay, "dct-length must be 320, received %d", dct_length);
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return FALSE;
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}
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wrong_caps:
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{
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GST_ERROR_OBJECT (rtpsirenpay, "expected audio/x-siren, received %s",
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payload_name);
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return FALSE;
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}
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}
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gboolean
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gst_rtp_siren_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpsirenpay",
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GST_RANK_NONE, GST_TYPE_RTP_SIREN_PAY);
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}
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@ -1,58 +0,0 @@
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/*
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* Siren Payloader Gst Element
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*
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* @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
|
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*
|
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* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2 of the License, or (at your option) any later version.
|
||||
*
|
||||
* This library is distributed in the hope that it will be useful,
|
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
* Library General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Library General Public
|
||||
* License along with this library; if not, write to the
|
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
||||
* Boston, MA 02111-1307, USA.
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*/
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#ifndef __GST_RTP_SIREN_PAY_H__
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#define __GST_RTP_SIREN_PAY_H__
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#include <gst/gst.h>
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#include <gst/rtp/gstbasertpaudiopayload.h>
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G_BEGIN_DECLS
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#define GST_TYPE_RTP_SIREN_PAY \
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(gst_rtpsirenpay_get_type())
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#define GST_RTP_SIREN_PAY(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SIREN_PAY,GstRTPSirenPay))
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#define GST_RTP_SIREN_PAY_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SIREN_PAY,GstRTPSirenPayClass))
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#define GST_IS_RTP_SIREN_PAY(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SIREN_PAY))
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#define GST_IS_RTP_SIREN_PAY_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SIREN_PAY))
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|
||||
typedef struct _GstRTPSirenPay GstRTPSirenPay;
|
||||
typedef struct _GstRTPSirenPayClass GstRTPSirenPayClass;
|
||||
|
||||
struct _GstRTPSirenPay
|
||||
{
|
||||
GstBaseRTPAudioPayload audiopayload;
|
||||
};
|
||||
|
||||
struct _GstRTPSirenPayClass
|
||||
{
|
||||
GstBaseRTPAudioPayloadClass parent_class;
|
||||
};
|
||||
|
||||
gboolean gst_rtp_siren_pay_plugin_init (GstPlugin * plugin);
|
||||
|
||||
G_END_DECLS
|
||||
|
||||
#endif /* __GST_RTP_SIREN_PAY_H__ */
|
||||
|
|
@ -26,8 +26,6 @@
|
|||
|
||||
#include "gstsirendec.h"
|
||||
#include "gstsirenenc.h"
|
||||
#include "gstrtpsirenpay.h"
|
||||
#include "gstrtpsirendepay.h"
|
||||
|
||||
|
||||
static gboolean
|
||||
|
|
@ -39,12 +37,6 @@ plugin_init (GstPlugin * plugin)
|
|||
if (!gst_siren_enc_plugin_init (plugin))
|
||||
return FALSE;
|
||||
|
||||
if (!gst_rtp_siren_pay_plugin_init (plugin))
|
||||
return FALSE;
|
||||
|
||||
if (!gst_rtp_siren_depay_plugin_init (plugin))
|
||||
return FALSE;
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
|
|
@ -52,4 +44,4 @@ GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|||
GST_VERSION_MINOR,
|
||||
"gstsiren",
|
||||
"Siren encoder/decoder/payloader/depayloader plugins",
|
||||
plugin_init, "0.1" , "LGPL", "Siren", "");
|
||||
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
|
||||
|
|
|
|||
Loading…
Reference in a new issue