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191 lines
5.6 KiB
C
191 lines
5.6 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtpbvdepay
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* @see_also: rtpbvpay
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*
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* Extract BroadcomVoice audio from RTP packets according to RFC 4298.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt
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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 <gst/audio/audio.h>
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#include "gstrtpbvdepay.h"
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#include "gstrtputils.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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"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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"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 (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp);
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static gboolean gst_rtp_bv_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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#define gst_rtp_bv_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRTPBVDepay, gst_rtp_bv_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
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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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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_bv_depay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_bv_depay_sink_template);
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gst_element_class_set_static_metadata (gstelement_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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gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_bv_depay_process;
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gstrtpbasedepayload_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)
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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 (GstRTPBaseDepayload * 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_RTP_BASE_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 (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
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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 (rtp);
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GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
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gst_buffer_get_size (rtp->buffer), marker,
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gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
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outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
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if (marker && outbuf) {
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/* mark start of talkspurt with RESYNC */
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
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}
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if (outbuf) {
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gst_rtp_drop_meta (GST_ELEMENT_CAST (depayload), outbuf,
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g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
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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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