mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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5b18c652fb
GCC 4.6.x spits warnings about variables that are unused but set. Such
variables have been removed where trivial but with comments left behind
for informational purposes in some cases.
gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4
to always return GST_FLOW_OK instead of the return value of
rtp_session_process_rtcp (), so we'll keep it that way.
181 lines
5.3 KiB
C
181 lines
5.3 KiB
C
/* GStreamer
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* Copyright (C) <2009> Wim Taymans <wim.taymans@gmail.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 <string.h>
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#include <stdlib.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpbvdepay.h"
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static GstStaticPadTemplate gst_rtp_bv_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) \"BV16\"; "
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"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) 16000, " "encoding-name = (string) \"BV32\"")
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);
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static GstStaticPadTemplate gst_rtp_bv_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/x-bv, " "mode = (int) { 16, 32 }")
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);
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static GstBuffer *gst_rtp_bv_depay_process (GstBaseRTPDepayload * depayload,
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GstBuffer * buf);
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static gboolean gst_rtp_bv_depay_setcaps (GstBaseRTPDepayload * depayload,
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GstCaps * caps);
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GST_BOILERPLATE (GstRTPBVDepay, gst_rtp_bv_depay, GstBaseRTPDepayload,
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GST_TYPE_BASE_RTP_DEPAYLOAD);
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static void
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gst_rtp_bv_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_bv_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_bv_depay_sink_template));
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gst_element_class_set_details_simple (element_class,
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"RTP BroadcomVoice depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts BroadcomVoice audio from RTP packets (RFC 4298)",
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"Wim Taymans <wim.taymans@collabora.co.uk>");
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}
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static void
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gst_rtp_bv_depay_class_init (GstRTPBVDepayClass * klass)
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{
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GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
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gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
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gstbasertpdepayload_class->process = gst_rtp_bv_depay_process;
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gstbasertpdepayload_class->set_caps = gst_rtp_bv_depay_setcaps;
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}
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static void
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gst_rtp_bv_depay_init (GstRTPBVDepay * rtpbvdepay, GstRTPBVDepayClass * klass)
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{
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rtpbvdepay->mode = -1;
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}
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static gboolean
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gst_rtp_bv_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
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{
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GstRTPBVDepay *rtpbvdepay = GST_RTP_BV_DEPAY (depayload);
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GstCaps *srccaps;
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GstStructure *structure;
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const gchar *mode_str = NULL;
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gint mode, clock_rate, expected_rate;
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gboolean ret;
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structure = gst_caps_get_structure (caps, 0);
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mode_str = gst_structure_get_string (structure, "encoding-name");
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if (!mode_str)
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goto no_mode;
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if (!strcmp (mode_str, "BV16")) {
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mode = 16;
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expected_rate = 8000;
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} else if (!strcmp (mode_str, "BV32")) {
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mode = 32;
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expected_rate = 16000;
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} else
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goto invalid_mode;
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if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
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clock_rate = expected_rate;
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else if (clock_rate != expected_rate)
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goto wrong_rate;
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depayload->clock_rate = clock_rate;
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rtpbvdepay->mode = mode;
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srccaps = gst_caps_new_simple ("audio/x-bv",
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"mode", G_TYPE_INT, rtpbvdepay->mode, NULL);
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ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
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GST_DEBUG ("set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
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gst_caps_unref (srccaps);
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return ret;
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/* ERRORS */
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no_mode:
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{
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GST_ERROR_OBJECT (rtpbvdepay, "did not receive an encoding-name");
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return FALSE;
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}
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invalid_mode:
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{
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GST_ERROR_OBJECT (rtpbvdepay,
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"invalid encoding-name, expected BV16 or BV32, got %s", mode_str);
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return FALSE;
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}
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wrong_rate:
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{
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GST_ERROR_OBJECT (rtpbvdepay, "invalid clock-rate, expected %d, got %d",
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expected_rate, clock_rate);
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return FALSE;
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}
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}
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static GstBuffer *
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gst_rtp_bv_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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{
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GstBuffer *outbuf;
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gboolean marker;
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marker = gst_rtp_buffer_get_marker (buf);
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GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
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GST_BUFFER_SIZE (buf), marker,
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gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
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outbuf = gst_rtp_buffer_get_payload_buffer (buf);
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if (marker) {
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/* mark start of talkspurt with DISCONT */
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
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}
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return outbuf;
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}
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gboolean
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gst_rtp_bv_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpbvdepay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_BV_DEPAY);
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}
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