gstreamer/gst/rtp/gstrtpmpapay.c
Stefan Kost 27f2c9b255 Define GstElementDetails as const and also static (when defined as global)
Original commit message from CVS:
* ext/aalib/gstaasink.c:
* ext/annodex/gstcmmldec.c:
* ext/annodex/gstcmmlenc.c:
* ext/cairo/gsttextoverlay.c:
* ext/cairo/gsttimeoverlay.c:
* ext/cdio/gstcdiocddasrc.c:
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdemux.c:
* ext/esd/esdmon.c:
* ext/esd/esdsink.c:
* ext/flac/gstflacenc.c:
* ext/flac/gstflactag.c:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init):
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init):
* ext/gdk_pixbuf/pixbufscale.c:
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
* ext/jpeg/gstjpegdec.c:
* ext/jpeg/gstjpegenc.c:
* ext/jpeg/gstsmokedec.c:
* ext/jpeg/gstsmokeenc.c:
* ext/libcaca/gstcacasink.c:
* ext/libmng/gstmngdec.c:
* ext/libmng/gstmngenc.c:
* ext/libpng/gstpngdec.c:
* ext/libpng/gstpngenc.c:
* ext/mikmod/gstmikmod.c:
* ext/raw1394/gstdv1394src.c:
* ext/shout2/gstshout2.c: (gst_shout2send_init):
* ext/shout2/gstshout2.h:
* ext/speex/gstspeexdec.c:
* ext/speex/gstspeexenc.c:
* gst/alpha/gstalpha.c:
* gst/alpha/gstalphacolor.c:
* gst/apetag/gstapedemux.c:
* gst/auparse/gstauparse.c:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_base_init):
* gst/avi/gstavidemux.c: (gst_avi_demux_base_init):
* gst/avi/gstavimux.c: (gst_avimux_base_init):
* gst/cutter/gstcutter.c:
* gst/debug/breakmydata.c:
* gst/debug/efence.c:
* gst/debug/gstnavigationtest.c:
* gst/debug/gstnavseek.c:
* gst/debug/negotiation.c:
* gst/debug/progressreport.c:
* gst/debug/testplugin.c:
* gst/effectv/gstaging.c:
* gst/effectv/gstdice.c:
* gst/effectv/gstedge.c:
* gst/effectv/gstquark.c:
* gst/effectv/gstrev.c:
* gst/effectv/gstshagadelic.c:
* gst/effectv/gstvertigo.c:
* gst/effectv/gstwarp.c:
* gst/flx/gstflxdec.c:
* gst/goom/gstgoom.c:
* gst/icydemux/gsticydemux.c:
* gst/id3demux/gstid3demux.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/law/alaw-decode.c: (gst_alawdec_base_init):
* gst/law/alaw-encode.c: (gst_alawenc_base_init):
* gst/law/mulaw-decode.c: (gst_mulawdec_base_init):
* gst/law/mulaw-encode.c: (gst_mulawenc_base_init):
* gst/level/gstlevel.c:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init):
* gst/median/gstmedian.c:
* gst/monoscope/gstmonoscope.c:
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartmux.c:
* gst/oldcore/gstaggregator.c:
* gst/oldcore/gstfdsink.c:
* gst/oldcore/gstmd5sink.c:
* gst/oldcore/gstmultifilesrc.c:
* gst/oldcore/gstpipefilter.c:
* gst/oldcore/gstshaper.c:
* gst/oldcore/gststatistics.c:
* gst/rtp/gstasteriskh263.c:
* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpL16pay.c:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpgsmpay.c:
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vdepay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmpadepay.c:
* gst/rtp/gstrtpmpapay.c:
* gst/rtp/gstrtppcmadepay.c:
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtspsrc.c:
* gst/smpte/gstsmpte.c:
* gst/udp/gstdynudpsink.c:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstudpsink.c:
* gst/udp/gstudpsrc.c:
* gst/videobox/gstvideobox.c:
* gst/videofilter/gstgamma.c: (gst_gamma_base_init):
* gst/videofilter/gstvideobalance.c:
* gst/videofilter/gstvideoflip.c:
* gst/videofilter/gstvideotemplate.c:
(gst_videotemplate_base_init):
* gst/videomixer/videomixer.c:
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_class_init), (gst_wavparse_dispose),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_peek_chunk), (gst_wavparse_stream_headers),
(gst_wavparse_parse_stream_init), (gst_wavparse_send_event),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_chain), (gst_wavparse_srcpad_event),
(gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull),
(gst_wavparse_change_state):
* gst/wavparse/gstwavparse.h:
* sys/oss/gstossmixerelement.c:
* sys/oss/gstosssink.c:
* sys/oss/gstosssrc.c:
* sys/osxaudio/gstosxaudioelement.c:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosrc.c:
* sys/sunaudio/gstsunaudiomixer.c:
* sys/sunaudio/gstsunaudiosink.c:
Define GstElementDetails as const and also static (when defined as
global)
2006-04-25 21:39:46 +00:00

267 lines
7.7 KiB
C

/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
*
* 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 <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpmpapay.h"
/* elementfactory information */
static const GstElementDetails gst_rtp_mpapay_details =
GST_ELEMENT_DETAILS ("RTP packet parser",
"Codec/Payloader/Network",
"Payode MPEG audio as RTP packets (RFC 2038)",
"Wim Taymans <wim@fluendo.com>");
static GstStaticPadTemplate gst_rtp_mpa_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg")
);
static GstStaticPadTemplate gst_rtp_mpa_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) 90000, " "encoding-name = (string) \"MPA\"")
);
static void gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass);
static void gst_rtp_mpa_pay_base_init (GstRtpMPAPayClass * klass);
static void gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay);
static void gst_rtp_mpa_pay_finalize (GObject * object);
static gboolean gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_mpa_pay_handle_buffer (GstBaseRTPPayload * payload,
GstBuffer * buffer);
static GstBaseRTPPayloadClass *parent_class = NULL;
static GType
gst_rtp_mpa_pay_get_type (void)
{
static GType rtpmpapay_type = 0;
if (!rtpmpapay_type) {
static const GTypeInfo rtpmpapay_info = {
sizeof (GstRtpMPAPayClass),
(GBaseInitFunc) gst_rtp_mpa_pay_base_init,
NULL,
(GClassInitFunc) gst_rtp_mpa_pay_class_init,
NULL,
NULL,
sizeof (GstRtpMPAPay),
0,
(GInstanceInitFunc) gst_rtp_mpa_pay_init,
};
rtpmpapay_type =
g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMPAPay",
&rtpmpapay_info, 0);
}
return rtpmpapay_type;
}
static void
gst_rtp_mpa_pay_base_init (GstRtpMPAPayClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mpa_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mpa_pay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_mpapay_details);
}
static void
gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * 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);
gobject_class->finalize = gst_rtp_mpa_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_mpa_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_mpa_pay_handle_buffer;
}
static void
gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay)
{
rtpmpapay->adapter = gst_adapter_new ();
}
static void
gst_rtp_mpa_pay_finalize (GObject * object)
{
GstRtpMPAPay *rtpmpapay;
rtpmpapay = GST_RTP_MPA_PAY (object);
g_object_unref (rtpmpapay->adapter);
rtpmpapay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gst_basertppayload_set_options (payload, "audio", TRUE, "MPA", 90000);
gst_basertppayload_set_outcaps (payload, NULL);
return TRUE;
}
static GstFlowReturn
gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
guint16 frag_offset;
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. In the case the
* adapter has more than one MTU, we need to split the MPA data
* over multiple packets. The frag_offset in each packet header
* needs to be updated with the position in the MPA frame. */
avail = gst_adapter_available (rtpmpapay->adapter);
ret = GST_FLOW_OK;
frag_offset = 0;
while (avail > 0) {
guint towrite;
guint8 *payload;
guint8 *data;
guint payload_len;
guint packet_len;
/* this will be the total lenght of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpmpapay));
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
payload_len -= 4;
gst_rtp_buffer_set_payload_type (outbuf, GST_RTP_PAYLOAD_MPA);
/*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | MBZ | Frag_offset |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
payload = gst_rtp_buffer_get_payload (outbuf);
payload[0] = 0;
payload[1] = 0;
payload[2] = frag_offset >> 8;
payload[3] = frag_offset & 0xff;
data = (guint8 *) gst_adapter_peek (rtpmpapay->adapter, payload_len);
memcpy (&payload[4], data, payload_len);
gst_adapter_flush (rtpmpapay->adapter, payload_len);
avail -= payload_len;
frag_offset += payload_len;
if (avail == 0)
gst_rtp_buffer_set_marker (outbuf, TRUE);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmpapay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmpapay->duration;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpapay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_mpa_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpMPAPay *rtpmpapay;
GstFlowReturn ret;
guint size, avail;
guint packet_len;
GstClockTime duration;
rtpmpapay = GST_RTP_MPA_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
duration = GST_BUFFER_DURATION (buffer);
avail = gst_adapter_available (rtpmpapay->adapter);
if (avail == 0) {
rtpmpapay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpmpapay->duration = 0;
}
/* get packet length of previous data and this new data,
* payload length includes a 4 byte header */
packet_len = gst_rtp_buffer_calc_packet_len (4 + avail + size, 0, 0);
/* if this buffer is going to overflow the packet, flush what we
* have. */
if (gst_basertppayload_is_filled (basepayload,
packet_len, rtpmpapay->duration + duration)) {
ret = gst_rtp_mpa_pay_flush (rtpmpapay);
rtpmpapay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpmpapay->duration = 0;
} else {
ret = GST_FLOW_OK;
}
gst_adapter_push (rtpmpapay->adapter, buffer);
rtpmpapay->duration += duration;
return ret;
}
gboolean
gst_rtp_mpa_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmpapay",
GST_RANK_NONE, GST_TYPE_RTP_MPA_PAY);
}