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27f2c9b255
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)
324 lines
8.9 KiB
C
324 lines
8.9 KiB
C
/* GStreamer
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* Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
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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
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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 <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpamrpay.h"
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GST_DEBUG_CATEGORY (rtpamrpay_debug);
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#define GST_CAT_DEFAULT (rtpamrpay_debug)
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/* references:
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*
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* RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
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* Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive
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* Multi-Rate Wideband (AMR-WB) Audio Codecs.
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*/
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/* elementfactory information */
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static const GstElementDetails gst_rtp_amrpay_details =
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GST_ELEMENT_DETAILS ("RTP packet parser",
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"Codec/Payloader/Network",
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"Payload-encode AMR audio into RTP packets (RFC 3267)",
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"Wim Taymans <wim@fluendo.com>");
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static GstStaticPadTemplate gst_rtp_amr_pay_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/AMR, channels=(int)1, rate=(int)8000")
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);
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static GstStaticPadTemplate gst_rtp_amr_pay_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) [ 96, 127 ], "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) \"AMR\", "
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"encoding-params = (string) \"1\", "
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"octet-align = (string) \"1\", "
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"crc = (string) \"0\", "
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"robust-sorting = (string) \"0\", "
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"interleaving = (string) \"0\", "
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"mode-set = (int) [ 0, 7 ], "
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"mode-change-period = (int) [ 1, MAX ], "
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"mode-change-neighbor = (string) { \"0\", \"1\" }, "
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"maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ]")
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);
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static void gst_rtp_amr_pay_class_init (GstRtpAMRPayClass * klass);
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static void gst_rtp_amr_pay_base_init (GstRtpAMRPayClass * klass);
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static void gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay);
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static gboolean gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * pad,
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GstBuffer * buffer);
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static GstBaseRTPPayloadClass *parent_class = NULL;
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static GType
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gst_rtp_amr_pay_get_type (void)
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{
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static GType rtpamrpay_type = 0;
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if (!rtpamrpay_type) {
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static const GTypeInfo rtpamrpay_info = {
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sizeof (GstRtpAMRPayClass),
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(GBaseInitFunc) gst_rtp_amr_pay_base_init,
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NULL,
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(GClassInitFunc) gst_rtp_amr_pay_class_init,
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NULL,
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NULL,
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sizeof (GstRtpAMRPay),
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0,
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(GInstanceInitFunc) gst_rtp_amr_pay_init,
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};
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rtpamrpay_type =
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g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpAMRPay",
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&rtpamrpay_info, 0);
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}
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return rtpamrpay_type;
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}
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static void
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gst_rtp_amr_pay_base_init (GstRtpAMRPayClass * 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_amr_pay_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_amr_pay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_amrpay_details);
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}
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static void
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gst_rtp_amr_pay_class_init (GstRtpAMRPayClass * 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_peek_parent (klass);
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gstbasertppayload_class->set_caps = gst_rtp_amr_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_amr_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtpamrpay_debug, "rtpamrpay", 0,
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"AMR RTP Payloader");
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}
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static void
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gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay)
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{
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}
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static gboolean
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gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
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{
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GstRtpAMRPay *rtpamrpay;
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rtpamrpay = GST_RTP_AMR_PAY (basepayload);
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gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
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gst_basertppayload_set_outcaps (basepayload,
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"encoding-params", G_TYPE_STRING, "1", "octet-align", G_TYPE_STRING, "1",
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/* don't set the defaults
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*
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* "crc", G_TYPE_STRING, "0",
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* "robust-sorting", G_TYPE_STRING, "0",
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* "interleaving", G_TYPE_STRING, "0",
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*/
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NULL);
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return TRUE;
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}
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/* -1 is invalid */
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static gint frame_size[16] = {
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12, 13, 15, 17, 19, 20, 26, 31,
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5, -1, -1, -1, -1, -1, -1, 0
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};
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static GstFlowReturn
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gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpAMRPay *rtpamrpay;
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GstFlowReturn ret;
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guint size, payload_len;
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GstBuffer *outbuf;
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guint8 *payload, *data, *payload_amr;
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GstClockTime timestamp;
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guint packet_len, mtu;
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gint i, num_packets, num_nonempty_packets;
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gint amr_len;
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rtpamrpay = GST_RTP_AMR_PAY (basepayload);
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mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpamrpay);
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size = GST_BUFFER_SIZE (buffer);
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data = GST_BUFFER_DATA (buffer);
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timestamp = GST_BUFFER_TIMESTAMP (buffer);
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/* FIXME, only
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* octet aligned, no interleaving, single channel, no CRC,
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* no robust-sorting. */
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GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
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/* first count number of packets and total amr frame size */
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amr_len = num_packets = num_nonempty_packets = 0;
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for (i = 0; i < size; i++) {
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guint8 FT;
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gint fr_size;
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FT = (data[i] & 0x78) >> 3;
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fr_size = frame_size[FT];
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GST_DEBUG_OBJECT (basepayload, "frame size %d", fr_size);
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/* FIXME, we don't handle this yet.. */
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if (fr_size <= 0)
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goto wrong_size;
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amr_len += fr_size;
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num_nonempty_packets++;
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num_packets++;
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i += fr_size;
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}
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if (amr_len > size)
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goto incomplete_frame;
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/* we need one extra byte for the CMR, the ToC is in the input
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* data */
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payload_len = size + 1;
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/* get packet len to check against MTU */
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packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
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if (packet_len > mtu)
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goto too_big;
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/* now alloc output buffer */
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outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
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/* copy timestamp, or fabricate one */
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if (timestamp != GST_CLOCK_TIME_NONE)
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GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
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else {
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/* AMR (nb) and AMR-WB both have 20 ms per frame */
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/* FIXME: when we do more than one AMR frame per packet, fix this */
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gint count = basepayload->seqnum - basepayload->seqnum_base;
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GST_BUFFER_TIMESTAMP (outbuf) = count * 20 * GST_MSECOND;
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}
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/* get payload, this is now writable */
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payload = gst_rtp_buffer_get_payload (outbuf);
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/* 0 1 2 3 4 5 6 7
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* +-+-+-+-+-+-+-+-+
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* | CMR |R|R|R|R|
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* +-+-+-+-+-+-+-+-+
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*/
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payload[0] = 0xF0; /* CMR, no specific mode requested */
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/* this is where we copy the AMR data, after num_packets FTs and the
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* CMR. */
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payload_amr = payload + num_packets + 1;
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/* copy data in payload, first we copy all the FTs then all
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* the AMR data. The last FT has to have the F flag cleared. */
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for (i = 1; i <= num_packets; i++) {
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guint8 FT;
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gint fr_size;
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/* 0 1 2 3 4 5 6 7
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* +-+-+-+-+-+-+-+-+
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* |F| FT |Q|P|P| more FT...
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* +-+-+-+-+-+-+-+-+
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*/
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FT = (*data & 0x78) >> 3;
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fr_size = frame_size[FT];
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if (i == num_packets)
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/* last packet, clear F flag */
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payload[i] = *data & 0x7f;
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else
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/* set F flag */
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payload[i] = *data | 0x80;
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memcpy (payload_amr, &data[1], fr_size);
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/* all sizes are > 0 since we checked for that above */
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data += fr_size + 1;
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payload_amr += fr_size;
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}
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gst_buffer_unref (buffer);
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ret = gst_basertppayload_push (basepayload, outbuf);
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return ret;
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/* ERRORS */
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wrong_size:
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{
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GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
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(NULL), ("received AMR frame with size <= 0"));
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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incomplete_frame:
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{
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GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
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(NULL), ("received incomplete AMR frames"));
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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too_big:
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{
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GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
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(NULL), ("received too many AMR frames for MTU"));
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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}
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gboolean
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gst_rtp_amr_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpamrpay",
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GST_RANK_NONE, GST_TYPE_RTP_AMR_PAY);
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}
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