gstreamer/gst/rtp/gstrtpamrdepay.c

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/* 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 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 <gst/rtp/gstrtpbuffer.h>
#include <stdlib.h>
#include <string.h>
#include "gstrtpamrdepay.h"
/* references:
*
* RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
* Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
* Wideband (AMR-WB) Audio Codecs.
*/
/* 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)
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static const GstElementDetails gst_rtp_amrdepay_details =
GST_ELEMENT_DETAILS ("RTP packet depayloader",
"Codec/Depayloader/Network",
"Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)",
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
"Wim Taymans <wim@fluendo.com>");
/* RtpAMRDepay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0
};
/* input is an RTP packet
*
* params see RFC 3267, section 8.1
*/
static GstStaticPadTemplate gst_rtp_amr_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) 8000, "
"encoding-name = (string) \"AMR\", "
"encoding-params = (string) \"1\", "
/* NOTE that all values must be strings in orde to be able to do SDP <->
* GstCaps mapping. */
"octet-align = (string) \"1\", "
"crc = (string) { \"0\", \"1\" }, "
"robust-sorting = (string) \"0\", " "interleaving = (string) \"0\";"
/* following options are not needed for a decoder
*
"mode-set = (int) [ 0, 7 ], "
"mode-change-period = (int) [ 1, MAX ], "
"mode-change-neighbor = (boolean) { TRUE, FALSE }, "
"maxptime = (int) [ 20, MAX ], "
"ptime = (int) [ 20, MAX ]"
*/
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 16000, "
"encoding-name = (string) \"AMR-WB\", "
"encoding-params = (string) \"1\", "
/* NOTE that all values must be strings in orde to be able to do SDP <->
* GstCaps mapping. */
"octet-align = (string) \"1\", "
"crc = (string) { \"0\", \"1\" }, "
"robust-sorting = (string) \"0\", " "interleaving = (string) \"0\""
/* following options are not needed for a decoder
*
"mode-set = (int) [ 0, 7 ], "
"mode-change-period = (int) [ 1, MAX ], "
"mode-change-neighbor = (boolean) { TRUE, FALSE }, "
"maxptime = (int) [ 20, MAX ], "
"ptime = (int) [ 20, MAX ]"
*/
)
);
static GstStaticPadTemplate gst_rtp_amr_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000;"
"audio/AMR-WB, " "channels = (int) 1," "rate = (int) 16000")
);
static gboolean gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
GST_BOILERPLATE (GstRtpAMRDepay, gst_rtp_amr_depay, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static void
gst_rtp_amr_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_amr_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_amr_depay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_amrdepay_details);
}
static void
gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
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
parent_class = g_type_class_peek_parent (klass);
gstbasertpdepayload_class->process = gst_rtp_amr_depay_process;
gstbasertpdepayload_class->set_caps = gst_rtp_amr_depay_setcaps;
}
static void
gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay,
GstRtpAMRDepayClass * klass)
{
GstBaseRTPDepayload *depayload;
depayload = GST_BASE_RTP_DEPAYLOAD (rtpamrdepay);
gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
}
static gboolean
gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstCaps *srccaps;
GstRtpAMRDepay *rtpamrdepay;
const gchar *params;
const gchar *str, *type;
gint clock_rate, need_clock_rate;
rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
/* figure out the mode first and set the clock rates */
if ((str = gst_structure_get_string (structure, "encoding-name"))) {
if (strcmp (str, "AMR") == 0) {
rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_NB;
clock_rate = need_clock_rate = 8000;
type = "audio/AMR";
} else if (strcmp (str, "AMR-WB") == 0) {
rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_WB;
clock_rate = need_clock_rate = 16000;
type = "audio/AMR-WB";
} else
goto invalid_mode;
} else
goto invalid_mode;
if (!(str = gst_structure_get_string (structure, "octet-align")))
rtpamrdepay->octet_align = FALSE;
else
rtpamrdepay->octet_align = (atoi (str) == 1);
if (!(str = gst_structure_get_string (structure, "crc")))
rtpamrdepay->crc = FALSE;
else
rtpamrdepay->crc = (atoi (str) == 1);
if (rtpamrdepay->crc) {
/* crc mode implies octet aligned mode */
rtpamrdepay->octet_align = TRUE;
}
if (!(str = gst_structure_get_string (structure, "robust-sorting")))
rtpamrdepay->robust_sorting = FALSE;
else
rtpamrdepay->robust_sorting = (atoi (str) == 1);
if (rtpamrdepay->robust_sorting) {
/* robust_sorting mode implies octet aligned mode */
rtpamrdepay->octet_align = TRUE;
}
if (!(str = gst_structure_get_string (structure, "interleaving")))
rtpamrdepay->interleaving = FALSE;
else
rtpamrdepay->interleaving = (atoi (str) == 1);
if (rtpamrdepay->interleaving) {
/* interleaving mode implies octet aligned mode */
rtpamrdepay->octet_align = TRUE;
}
if (!(params = gst_structure_get_string (structure, "encoding-params")))
rtpamrdepay->channels = 1;
else {
rtpamrdepay->channels = atoi (params);
}
gst_structure_get_int (structure, "clock-rate", &clock_rate);
depayload->clock_rate = clock_rate;
/* we require 1 channel, 8000 Hz, octet aligned, no CRC,
* no robust sorting, no interleaving for now */
if (rtpamrdepay->channels != 1)
return FALSE;
if (clock_rate != need_clock_rate)
return FALSE;
if (rtpamrdepay->octet_align != TRUE)
return FALSE;
if (rtpamrdepay->robust_sorting != FALSE)
return FALSE;
if (rtpamrdepay->interleaving != FALSE)
return FALSE;
srccaps = gst_caps_new_simple (type,
"channels", G_TYPE_INT, rtpamrdepay->channels,
"rate", G_TYPE_INT, clock_rate, NULL);
gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
rtpamrdepay->negotiated = TRUE;
return TRUE;
/* ERRORS */
invalid_mode:
{
GST_ERROR_OBJECT (rtpamrdepay, "invalid encoding-name");
return FALSE;
}
}
/* -1 is invalid */
static gint nb_frame_size[16] = {
12, 13, 15, 17, 19, 20, 26, 31,
5, -1, -1, -1, -1, -1, -1, 0
};
static gint wb_frame_size[16] = {
17, 23, 32, 36, 40, 46, 50, 58,
60, -1, -1, -1, -1, -1, -1, 0
};
static GstBuffer *
gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpAMRDepay *rtpamrdepay;
GstBuffer *outbuf = NULL;
gint payload_len;
gint *frame_size;
rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
if (!rtpamrdepay->negotiated)
goto not_negotiated;
if (!gst_rtp_buffer_validate (buf))
goto invalid_packet;
/* setup frame size pointer */
if (rtpamrdepay->mode == GST_RTP_AMR_DP_MODE_NB)
frame_size = nb_frame_size;
else
frame_size = wb_frame_size;
/* when we get here, 1 channel, 8000/16000 Hz, octet aligned, no CRC,
* no robust sorting, no interleaving data is to be depayloaded */
{
guint8 *payload, *p, *dp;
guint8 CMR;
gint i, num_packets, num_nonempty_packets;
gint amr_len;
gint ILL, ILP;
payload_len = gst_rtp_buffer_get_payload_len (buf);
/* need at least 2 bytes for the header */
if (payload_len < 2)
goto too_small;
payload = gst_rtp_buffer_get_payload (buf);
/* depay CMR. The CMR is used by the sender to request
* a new encoding mode.
*
* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* | CMR |R|R|R|R|
* +-+-+-+-+-+-+-+-+
*/
CMR = (payload[0] & 0xf0) >> 4;
/* strip CMR header now, pack FT and the data for the decoder */
payload_len -= 1;
payload += 1;
GST_DEBUG_OBJECT (rtpamrdepay, "payload len %d", payload_len);
if (rtpamrdepay->interleaving) {
ILL = (payload[0] & 0xf0) >> 4;
ILP = (payload[0] & 0x0f);
payload_len -= 1;
payload += 1;
if (ILP > ILL)
goto wrong_interleaving;
}
/*
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
* +-+-+-+-+-+-+-+-+..
* |F| FT |Q|P|P| more FT..
* +-+-+-+-+-+-+-+-+..
*/
/* count number of packets by counting the FTs. Also
* count number of amr data bytes and number of non-empty
* packets (this is also the number of CRCs if present). */
amr_len = 0;
num_nonempty_packets = 0;
num_packets = 0;
for (i = 0; i < payload_len; i++) {
gint fr_size;
guint8 FT;
FT = (payload[i] & 0x78) >> 3;
fr_size = frame_size[FT];
GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size);
if (fr_size == -1)
goto wrong_framesize;
if (fr_size > 0) {
amr_len += fr_size;
num_nonempty_packets++;
}
num_packets++;
if ((payload[i] & 0x80) == 0)
break;
}
if (rtpamrdepay->crc) {
/* data len + CRC len + header bytes should be smaller than payload_len */
if (num_packets + num_nonempty_packets + amr_len > payload_len)
goto wrong_length_1;
} else {
/* data len + header bytes should be smaller than payload_len */
if (num_packets + amr_len > payload_len)
goto wrong_length_2;
}
outbuf = gst_buffer_new_and_alloc (payload_len);
/* point to destination */
p = GST_BUFFER_DATA (outbuf);
/* point to first data packet */
dp = payload + num_packets;
if (rtpamrdepay->crc) {
/* skip CRC if present */
dp += num_nonempty_packets;
}
for (i = 0; i < num_packets; i++) {
gint fr_size;
/* copy FT, clear F bit */
*p++ = payload[i] & 0x7f;
fr_size = frame_size[(payload[i] & 0x78) >> 3];
if (fr_size > 0) {
/* copy data packet, FIXME, calc CRC here. */
memcpy (p, dp, fr_size);
p += fr_size;
dp += fr_size;
}
}
gst_buffer_set_caps (outbuf,
GST_PAD_CAPS (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload)));
GST_DEBUG ("gst_rtp_amr_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
}
return outbuf;
/* ERRORS */
invalid_packet:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP packet did not validate"));
goto bad_packet;
}
not_negotiated:
{
GST_ELEMENT_ERROR (rtpamrdepay, STREAM, NOT_IMPLEMENTED,
(NULL), ("not negotiated"));
return NULL;
}
too_small:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP payload too small (%d)", payload_len));
goto bad_packet;
}
wrong_interleaving:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP wrong interleaving"));
goto bad_packet;
}
wrong_framesize:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP frame size == -1"));
goto bad_packet;
}
wrong_length_1:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP wrong length 1"));
goto bad_packet;
}
wrong_length_2:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
(NULL), ("AMR RTP wrong length 2"));
goto bad_packet;
}
bad_packet:
{
/* no fatal error */
return NULL;
}
}
gboolean
gst_rtp_amr_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpamrdepay",
GST_RANK_MARGINAL, GST_TYPE_RTP_AMR_DEPAY);
}