gstreamer/gst/rtp/gstrtpspeexdepay.c

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/* 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 details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpspeexdepay.h"
/* elementfactory information */
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
static const GstElementDetails gst_rtp_speexdepay_details =
GST_ELEMENT_DETAILS ("RTP packet depayloader",
"Codec/Depayloader/Network",
better/unified long descriptions 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/esd/esdmon.c: * ext/esd/esdsink.c: * ext/flac/gstflacdec.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/gstgdkpixbuf.c: * 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: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.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/gstavimux.c: (gst_avimux_base_init): * gst/cutter/gstcutter.c: * gst/debug/breakmydata.c: * gst/debug/efence.c: * gst/debug/gstnavigationtest.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/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.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/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/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/smpte/gstsmpte.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/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init): better/unified long descriptions Fixed #336602 Some cleanups to auparse, don't send multiple newsegments.
2006-03-30 15:37:05 +00:00
"Extracts Speex audio from RTP packets",
"Edgard Lima <edgard.lima@indt.org.br>");
/* RtpSPEEXDepay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0
};
static GstStaticPadTemplate gst_rtp_speex_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) [6000, 48000], "
"encoding-name = (string) \"SPEEX\", "
"encoding-params = (string) \"1\"")
);
static GstStaticPadTemplate gst_rtp_speex_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-speex")
);
static GstBuffer *gst_rtp_speex_depay_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
static gboolean gst_rtp_speex_depay_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
GST_BOILERPLATE (GstRtpSPEEXDepay, gst_rtp_speex_depay, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static void
gst_rtp_speex_depay_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_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_speex_depay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_speexdepay_details);
}
static void
gst_rtp_speex_depay_class_init (GstRtpSPEEXDepayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
gstbasertpdepayload_class->process = gst_rtp_speex_depay_process;
gstbasertpdepayload_class->set_caps = gst_rtp_speex_depay_setcaps;
}
static void
gst_rtp_speex_depay_init (GstRtpSPEEXDepay * rtpspeexdepay,
GstRtpSPEEXDepayClass * klass)
{
GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay)->clock_rate = 8000;
rtpspeexdepay->start_ts = 0;
}
static gint
gst_rtp_speex_depay_get_mode (gint rate)
{
if (rate > 25000)
return 2;
else if (rate > 12500)
return 1;
else
return 0;
}
/* len 4 bytes LE,
* vendor string (len bytes),
* user_len 4 (0) bytes LE
*/
static const gchar gst_rtp_speex_comment[] =
"\045\0\0\0Depayloaded with GStreamer speexdepay\0\0\0\0";
static gboolean
gst_rtp_speex_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpSPEEXDepay *rtpspeexdepay;
gint clock_rate, nb_channels;
GstBuffer *buf;
guint8 *data;
const gchar *params;
rtpspeexdepay = GST_RTP_SPEEX_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "clock-rate", &clock_rate);
depayload->clock_rate = clock_rate;
if (!(params = gst_structure_get_string (structure, "encoding-params")))
nb_channels = 1;
else {
nb_channels = atoi (params);
}
/* construct minimal header and comment packet for the decoder */
buf = gst_buffer_new_and_alloc (80);
data = GST_BUFFER_DATA (buf);
memcpy (data, "Speex ", 8);
data += 8;
memcpy (data, "1.1.12", 7);
data += 20;
GST_WRITE_UINT32_LE (data, 1); /* version */
data += 4;
GST_WRITE_UINT32_LE (data, 80); /* header_size */
data += 4;
GST_WRITE_UINT32_LE (data, clock_rate); /* rate */
data += 4;
GST_WRITE_UINT32_LE (data, gst_rtp_speex_depay_get_mode (clock_rate)); /* mode */
data += 4;
GST_WRITE_UINT32_LE (data, 4); /* mode_bitstream_version */
data += 4;
GST_WRITE_UINT32_LE (data, nb_channels); /* nb_channels */
data += 4;
GST_WRITE_UINT32_LE (data, -1); /* bitrate */
data += 4;
GST_WRITE_UINT32_LE (data, 0xa0); /* frame_size */
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* VBR */
data += 4;
GST_WRITE_UINT32_LE (data, 1); /* frames_per_packet */
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* extra_headers */
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* reserved1 */
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* reserved2 */
gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay), buf);
buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_speex_comment));
memcpy (GST_BUFFER_DATA (buf), gst_rtp_speex_comment,
sizeof (gst_rtp_speex_comment));
gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay), buf);
return TRUE;
}
static GstBuffer *
gst_rtp_speex_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf = NULL;
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
GST_BUFFER_SIZE (buf),
gst_rtp_buffer_get_marker (buf),
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
GstRtpSPEEXDepay *speexdepayload = GST_RTP_SPEEX_DEPAY (depayload);
/* apply correct timestamp from rtp packet */
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
if (speexdepayload->start_ts == 0)
speexdepayload->start_ts = gst_rtp_buffer_get_timestamp (buf);
GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale (GST_SECOND,
gst_rtp_buffer_get_timestamp (buf) - speexdepayload->start_ts,
depayload->clock_rate);
return outbuf;
}
gboolean
gst_rtp_speex_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpspeexdepay",
GST_RANK_MARGINAL, GST_TYPE_RTP_SPEEX_DEPAY);
}