mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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acddbd83ff
Original commit message from CVS: * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process): Fix depayloader clock_rate and some cleanups. * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_finalize), (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process): * gst/rtp/gstrtph264depay.h: Don't push codec_data in the adapter because it might get flushed when we get a discont. * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process): Handle multiple AU per packet. * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process), (gst_rtp_sv3v_depay_plugin_init): Disable rank, this one does not work. Remove timestamping, base class does that.
423 lines
11 KiB
C
423 lines
11 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 details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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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 <gst/rtp/gstrtpbuffer.h>
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#include <stdlib.h>
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#include <string.h>
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#include "gstrtpamrdepay.h"
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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 Multi-Rate
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* Wideband (AMR-WB) Audio Codecs.
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*/
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/* elementfactory information */
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static const GstElementDetails gst_rtp_amrdepay_details =
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GST_ELEMENT_DETAILS ("RTP packet depayloader",
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"Codec/Depayloader/Network",
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"Extracts AMR audio from RTP packets (RFC 3267)",
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"Wim Taymans <wim@fluendo.com>");
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/* RtpAMRDepay signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0
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};
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/* input is an RTP packet
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*
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* params see RFC 3267, section 8.1
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*/
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static GstStaticPadTemplate gst_rtp_amr_depay_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 ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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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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/* NOTE that all values must be strings in orde to be able to do SDP <->
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* GstCaps mapping. */
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"octet-align = (string) \"1\", "
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"crc = (string) { \"0\", \"1\" }, "
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"robust-sorting = (string) \"0\", " "interleaving = (string) \"0\""
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/* following options are not needed for a decoder
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*
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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 = (boolean) { TRUE, FALSE }, "
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"maxptime = (int) [ 20, MAX ], "
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"ptime = (int) [ 20, MAX ]"
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*/
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)
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);
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static GstStaticPadTemplate gst_rtp_amr_depay_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 ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000")
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);
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static gboolean gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload,
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GstCaps * caps);
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static GstBuffer *gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload,
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GstBuffer * buf);
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GST_BOILERPLATE (GstRtpAMRDepay, gst_rtp_amr_depay, GstBaseRTPDepayload,
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GST_TYPE_BASE_RTP_DEPAYLOAD);
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static void
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gst_rtp_amr_depay_base_init (gpointer 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_depay_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_depay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_amrdepay_details);
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}
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static void
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gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gstbasertpdepayload_class->process = gst_rtp_amr_depay_process;
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gstbasertpdepayload_class->set_caps = gst_rtp_amr_depay_setcaps;
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}
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static void
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gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay,
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GstRtpAMRDepayClass * klass)
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{
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GstBaseRTPDepayload *depayload;
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depayload = GST_BASE_RTP_DEPAYLOAD (rtpamrdepay);
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depayload->clock_rate = 8000;
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gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
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}
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static gboolean
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gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
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{
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GstStructure *structure;
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GstCaps *srccaps;
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GstRtpAMRDepay *rtpamrdepay;
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const gchar *params;
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const gchar *str;
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gint clock_rate;
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rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
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structure = gst_caps_get_structure (caps, 0);
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if (!(str = gst_structure_get_string (structure, "octet-align")))
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rtpamrdepay->octet_align = FALSE;
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else
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rtpamrdepay->octet_align = (atoi (str) == 1);
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if (!(str = gst_structure_get_string (structure, "crc")))
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rtpamrdepay->crc = FALSE;
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else
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rtpamrdepay->crc = (atoi (str) == 1);
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if (rtpamrdepay->crc) {
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/* crc mode implies octet aligned mode */
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rtpamrdepay->octet_align = TRUE;
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}
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if (!(str = gst_structure_get_string (structure, "robust-sorting")))
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rtpamrdepay->robust_sorting = FALSE;
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else
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rtpamrdepay->robust_sorting = (atoi (str) == 1);
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if (rtpamrdepay->robust_sorting) {
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/* robust_sorting mode implies octet aligned mode */
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rtpamrdepay->octet_align = TRUE;
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}
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if (!(str = gst_structure_get_string (structure, "interleaving")))
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rtpamrdepay->interleaving = FALSE;
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else
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rtpamrdepay->interleaving = (atoi (str) == 1);
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if (rtpamrdepay->interleaving) {
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/* interleaving mode implies octet aligned mode */
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rtpamrdepay->octet_align = TRUE;
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}
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if (!(params = gst_structure_get_string (structure, "encoding-params")))
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rtpamrdepay->channels = 1;
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else {
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rtpamrdepay->channels = atoi (params);
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}
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if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
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clock_rate = 8000;
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depayload->clock_rate = clock_rate;
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/* we require 1 channel, 8000 Hz, octet aligned, no CRC,
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* no robust sorting, no interleaving for now */
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if (rtpamrdepay->channels != 1)
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return FALSE;
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if (clock_rate != 8000)
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return FALSE;
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if (rtpamrdepay->octet_align != TRUE)
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return FALSE;
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if (rtpamrdepay->robust_sorting != FALSE)
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return FALSE;
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if (rtpamrdepay->interleaving != FALSE)
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return FALSE;
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srccaps = gst_caps_new_simple ("audio/AMR",
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"channels", G_TYPE_INT, rtpamrdepay->channels,
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"rate", G_TYPE_INT, clock_rate, NULL);
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gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
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gst_caps_unref (srccaps);
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rtpamrdepay->negotiated = TRUE;
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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 GstBuffer *
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gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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{
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GstRtpAMRDepay *rtpamrdepay;
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GstBuffer *outbuf = NULL;
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gint payload_len;
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rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
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if (!rtpamrdepay->negotiated)
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goto not_negotiated;
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if (!gst_rtp_buffer_validate (buf))
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goto invalid_packet;
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/* when we get here, 1 channel, 8000 Hz, octet aligned, no CRC,
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* no robust sorting, no interleaving data is to be depayloaded */
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{
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guint8 *payload, *p, *dp;
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guint8 CMR;
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gint i, num_packets, num_nonempty_packets;
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gint amr_len;
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gint ILL, ILP;
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payload_len = gst_rtp_buffer_get_payload_len (buf);
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/* need at least 2 bytes for the header */
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if (payload_len < 2)
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goto too_small;
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payload = gst_rtp_buffer_get_payload (buf);
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/* depay CMR. The CMR is used by the sender to request
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* a new encoding mode.
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*
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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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CMR = (payload[0] & 0xf0) >> 4;
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/* strip CMR header now, pack FT and the data for the decoder */
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payload_len -= 1;
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payload += 1;
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GST_DEBUG_OBJECT (rtpamrdepay, "payload len %d", payload_len);
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if (rtpamrdepay->interleaving) {
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ILL = (payload[0] & 0xf0) >> 4;
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ILP = (payload[0] & 0x0f);
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payload_len -= 1;
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payload += 1;
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if (ILP > ILL)
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goto wrong_interleaving;
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}
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/*
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
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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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/* count number of packets by counting the FTs. Also
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* count number of amr data bytes and number of non-empty
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* packets (this is also the number of CRCs if present). */
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amr_len = 0;
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num_nonempty_packets = 0;
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num_packets = 0;
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for (i = 0; i < payload_len; i++) {
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gint fr_size;
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guint8 FT;
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FT = (payload[i] & 0x78) >> 3;
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fr_size = frame_size[FT];
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GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size);
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if (fr_size == -1)
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goto wrong_framesize;
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if (fr_size > 0) {
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amr_len += fr_size;
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num_nonempty_packets++;
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}
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num_packets++;
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if ((payload[i] & 0x80) == 0)
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break;
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}
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if (rtpamrdepay->crc) {
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/* data len + CRC len + header bytes should be smaller than payload_len */
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if (num_packets + num_nonempty_packets + amr_len > payload_len)
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goto wrong_length_1;
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} else {
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/* data len + header bytes should be smaller than payload_len */
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if (num_packets + amr_len > payload_len)
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goto wrong_length_2;
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}
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outbuf = gst_buffer_new_and_alloc (payload_len);
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/* point to destination */
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p = GST_BUFFER_DATA (outbuf);
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/* point to first data packet */
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dp = payload + num_packets;
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if (rtpamrdepay->crc) {
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/* skip CRC if present */
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dp += num_nonempty_packets;
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}
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for (i = 0; i < num_packets; i++) {
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gint fr_size;
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/* copy FT, clear F bit */
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*p++ = payload[i] & 0x7f;
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fr_size = frame_size[(payload[i] & 0x78) >> 3];
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if (fr_size > 0) {
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/* copy data packet, FIXME, calc CRC here. */
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memcpy (p, dp, fr_size);
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p += fr_size;
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dp += fr_size;
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}
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}
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gst_buffer_set_caps (outbuf,
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GST_PAD_CAPS (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload)));
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GST_DEBUG ("gst_rtp_amr_depay_chain: pushing buffer of size %d",
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GST_BUFFER_SIZE (outbuf));
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}
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return outbuf;
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/* ERRORS */
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invalid_packet:
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{
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GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
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(NULL), ("AMR RTP packet did not validate"));
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goto bad_packet;
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}
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not_negotiated:
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{
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GST_ELEMENT_ERROR (rtpamrdepay, STREAM, NOT_IMPLEMENTED,
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(NULL), ("not negotiated"));
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return NULL;
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}
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too_small:
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{
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GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
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(NULL), ("AMR RTP payload too small (%d)", payload_len));
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goto bad_packet;
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}
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wrong_interleaving:
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{
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GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
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(NULL), ("AMR RTP wrong interleaving"));
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goto bad_packet;
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}
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wrong_framesize:
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{
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GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
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(NULL), ("AMR RTP frame size == -1"));
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goto bad_packet;
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}
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wrong_length_1:
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{
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GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
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(NULL), ("AMR RTP wrong length 1"));
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goto bad_packet;
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}
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wrong_length_2:
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{
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GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
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(NULL), ("AMR RTP wrong length 2"));
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goto bad_packet;
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}
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bad_packet:
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{
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/* no fatal error */
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return NULL;
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}
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}
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gboolean
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gst_rtp_amr_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpamrdepay",
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GST_RANK_MARGINAL, GST_TYPE_RTP_AMR_DEPAY);
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}
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