gstreamer/gst/rtp/gstrtpspeexpay.c
Stefan Kost b5af832d7b Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
Original commit message from CVS:
* ext/aalib/gstaasink.c: (gst_aasink_class_init):
* ext/esd/esdsink.c: (gst_esdsink_class_init):
* ext/flac/gstflactag.c: (gst_flac_tag_class_init):
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_class_init):
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init):
* ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init):
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init):
* ext/libcaca/gstcacasink.c: (gst_cacasink_class_init):
* ext/libmng/gstmngdec.c: (gst_mngdec_class_init):
* ext/libmng/gstmngenc.c: (gst_mngenc_class_init):
* ext/libpng/gstpngdec.c: (gst_pngdec_class_init):
* ext/libpng/gstpngenc.c: (gst_pngenc_class_init):
* ext/mikmod/gstmikmod.c: (gst_mikmod_class_init):
* ext/shout2/gstshout2.c: (gst_shout2send_class_init):
* ext/speex/gstspeexenc.c: (gst_speexenc_class_init):
* gst/alpha/gstalpha.c: (gst_alpha_class_init):
* gst/avi/gstavimux.c: (gst_avimux_class_init):
* gst/debug/efence.c: (gst_efence_class_init):
* gst/debug/negotiation.c: (gst_negotiation_class_init):
* gst/flx/gstflxdec.c: (gst_flxdec_class_init):
* gst/goom/gstgoom.c: (gst_goom_class_init):
* gst/id3demux/gstid3demux.c: (gst_id3demux_class_init):
* gst/interleave/deinterleave.c: (deinterleave_class_init):
* gst/interleave/interleave.c: (interleave_class_init):
* gst/law/alaw-decode.c: (gst_alawdec_class_init):
* gst/law/alaw-encode.c: (gst_alawenc_class_init):
* gst/law/mulaw-encode.c: (gst_mulawenc_class_init):
* gst/median/gstmedian.c: (gst_median_class_init):
* gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init):
* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init):
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_class_init):
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_class_init):
* gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_class_init):
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init):
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init):
* gst/rtp/gstrtpdepay.c: (gst_rtp_depay_class_init):
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_class_init):
* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_class_init):
* gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_class_init):
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_class_init):
* gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init):
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init):
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init):
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init):
* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_class_init):
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_class_init):
* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_class_init):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_class_init):
* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_class_init):
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init):
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init):
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init):
* gst/smpte/gstsmpte.c: (gst_smpte_class_init):
* gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init):
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init):
* gst/udp/gstudpsink.c: (gst_udpsink_class_init):
* gst/videomixer/videomixer.c: (gst_videomixer_class_init):
* gst/wavenc/gstwavenc.c: (gst_wavenc_class_init):
* sys/oss/gstossdmabuffer.c: (gst_ossdmabuffer_class_init):
* sys/oss/gstosssink.c: (gst_oss_sink_class_init):
* sys/osxaudio/gstosxaudioelement.c:
(gst_osxaudioelement_class_init):
* sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_class_init):
* sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_class_init):
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init):
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-04-08 21:21:45 +00:00

150 lines
4.4 KiB
C

/* GStreamer
* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
*
* 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
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpspeexpay.h"
/* elementfactory information */
static GstElementDetails gst_rtp_speex_pay_details =
GST_ELEMENT_DETAILS ("RTP packet parser",
"Codec/Payloader/Network",
"Payload-encodes Speex audio into a RTP packet",
"Edgard Lima <edgard.lima@indt.org.br>");
static GstStaticPadTemplate gst_rtp_speex_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-speex")
);
static GstStaticPadTemplate gst_rtp_speex_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) [ 96, 127 ], "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"speex\", "
"encoding-params = (string) \"1\"")
);
static gboolean gst_rtp_speex_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
GST_BOILERPLATE (GstRtpSPEEXPay, gst_rtp_speex_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtp_speex_pay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_speex_pay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_speex_pay_src_template));
gst_element_class_set_details (element_class, &gst_rtp_speex_pay_details);
}
static void
gst_rtp_speex_pay_class_init (GstRtpSPEEXPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gstbasertppayload_class->set_caps = gst_rtp_speex_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_speex_pay_handle_buffer;
}
static void
gst_rtp_speex_pay_init (GstRtpSPEEXPay * rtpspeexpay,
GstRtpSPEEXPayClass * klass)
{
GST_BASE_RTP_PAYLOAD (rtpspeexpay)->clock_rate = 8000;
GST_BASE_RTP_PAYLOAD_PT (rtpspeexpay) = 110; /* Create String */
}
static gboolean
gst_rtp_speex_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gst_basertppayload_set_options (payload, "audio", FALSE, "speex", 8000);
gst_basertppayload_set_outcaps (payload, NULL);
return TRUE;
}
static GstFlowReturn
gst_rtp_speex_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpSPEEXPay *rtpspeexpay;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data;
GstClockTime timestamp;
GstFlowReturn ret;
rtpspeexpay = GST_RTP_SPEEX_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* FIXME, only one SPEEX frame per RTP packet for now */
payload_len = size;
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* FIXME, assert for now */
g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpspeexpay));
/* copy timestamp */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
/* get payload */
payload = gst_rtp_buffer_get_payload (outbuf);
data = GST_BUFFER_DATA (buffer);
/* copy data in payload */
memcpy (&payload[0], data, size);
gst_buffer_unref (buffer);
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}
gboolean
gst_rtp_speex_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpspeexpay",
GST_RANK_NONE, GST_TYPE_RTP_SPEEX_PAY);
}