2002-07-04 12:38:01 +02:00
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/*
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* RTP input/output format
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* Copyright (c) 2002 Fabrice Bellard.
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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 Lesser 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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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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2002-11-05 01:56:08 +01:00
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#include <sys/socket.h>
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2002-07-04 12:38:01 +02:00
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#include "avformat.h"
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#include <unistd.h>
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#include <sys/types.h>
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#include <netinet/in.h>
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2002-11-02 11:35:07 +01:00
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#ifndef __BEOS__
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# include <arpa/inet.h>
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#else
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# include "barpainet.h"
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#endif
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2002-07-04 12:38:01 +02:00
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#include <netdb.h>
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//#define DEBUG
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/* TODO: - add RTCP statistics reporting (should be optional).
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- add support for h263/mpeg4 packetized output : IDEA: send a
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buffer to 'rtp_write_packet' contains all the packets for ONE
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frame. Each packet should have a four byte header containing
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the length in big endian format (same trick as
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'url_open_dyn_packet_buf')
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*/
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#define RTP_VERSION 2
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#define RTP_MAX_SDES 256 /* maximum text length for SDES */
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/* RTCP paquets use 0.5 % of the bandwidth */
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#define RTCP_TX_RATIO_NUM 5
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#define RTCP_TX_RATIO_DEN 1000
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typedef enum {
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RTCP_SR = 200,
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RTCP_RR = 201,
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RTCP_SDES = 202,
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RTCP_BYE = 203,
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RTCP_APP = 204
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} rtcp_type_t;
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typedef enum {
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RTCP_SDES_END = 0,
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RTCP_SDES_CNAME = 1,
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RTCP_SDES_NAME = 2,
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RTCP_SDES_EMAIL = 3,
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RTCP_SDES_PHONE = 4,
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RTCP_SDES_LOC = 5,
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RTCP_SDES_TOOL = 6,
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RTCP_SDES_NOTE = 7,
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RTCP_SDES_PRIV = 8,
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RTCP_SDES_IMG = 9,
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RTCP_SDES_DOOR = 10,
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RTCP_SDES_SOURCE = 11
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} rtcp_sdes_type_t;
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enum RTPPayloadType {
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RTP_PT_ULAW = 0,
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RTP_PT_GSM = 3,
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RTP_PT_G723 = 4,
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RTP_PT_ALAW = 8,
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RTP_PT_S16BE_STEREO = 10,
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RTP_PT_S16BE_MONO = 11,
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RTP_PT_MPEGAUDIO = 14,
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RTP_PT_JPEG = 26,
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RTP_PT_H261 = 31,
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RTP_PT_MPEGVIDEO = 32,
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RTP_PT_MPEG2TS = 33,
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RTP_PT_H263 = 34, /* old H263 encapsulation */
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};
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typedef struct RTPContext {
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int payload_type;
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UINT32 ssrc;
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UINT16 seq;
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UINT32 timestamp;
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UINT32 base_timestamp;
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UINT32 cur_timestamp;
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int max_payload_size;
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/* rtcp sender statistics receive */
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INT64 last_rtcp_ntp_time;
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UINT32 last_rtcp_timestamp;
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/* rtcp sender statistics */
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unsigned int packet_count;
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unsigned int octet_count;
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unsigned int last_octet_count;
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int first_packet;
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/* buffer for output */
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UINT8 buf[RTP_MAX_PACKET_LENGTH];
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UINT8 *buf_ptr;
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} RTPContext;
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int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
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{
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switch(payload_type) {
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case RTP_PT_ULAW:
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codec->codec_id = CODEC_ID_PCM_MULAW;
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codec->channels = 1;
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codec->sample_rate = 8000;
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break;
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case RTP_PT_ALAW:
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codec->codec_id = CODEC_ID_PCM_ALAW;
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codec->channels = 1;
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codec->sample_rate = 8000;
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break;
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case RTP_PT_S16BE_STEREO:
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codec->codec_id = CODEC_ID_PCM_S16BE;
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codec->channels = 2;
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codec->sample_rate = 44100;
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break;
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case RTP_PT_S16BE_MONO:
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codec->codec_id = CODEC_ID_PCM_S16BE;
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codec->channels = 1;
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codec->sample_rate = 44100;
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break;
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case RTP_PT_MPEGAUDIO:
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codec->codec_id = CODEC_ID_MP2;
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break;
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case RTP_PT_JPEG:
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codec->codec_id = CODEC_ID_MJPEG;
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break;
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case RTP_PT_MPEGVIDEO:
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codec->codec_id = CODEC_ID_MPEG1VIDEO;
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break;
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default:
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return -1;
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}
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return 0;
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}
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/* return < 0 if unknown payload type */
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int rtp_get_payload_type(AVCodecContext *codec)
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{
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int payload_type;
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/* compute the payload type */
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payload_type = -1;
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switch(codec->codec_id) {
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case CODEC_ID_PCM_MULAW:
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payload_type = RTP_PT_ULAW;
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break;
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case CODEC_ID_PCM_ALAW:
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payload_type = RTP_PT_ALAW;
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break;
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case CODEC_ID_PCM_S16BE:
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if (codec->channels == 1) {
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payload_type = RTP_PT_S16BE_MONO;
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} else if (codec->channels == 2) {
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payload_type = RTP_PT_S16BE_STEREO;
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}
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break;
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case CODEC_ID_MP2:
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case CODEC_ID_MP3LAME:
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payload_type = RTP_PT_MPEGAUDIO;
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break;
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case CODEC_ID_MJPEG:
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payload_type = RTP_PT_JPEG;
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break;
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case CODEC_ID_MPEG1VIDEO:
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payload_type = RTP_PT_MPEGVIDEO;
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break;
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default:
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break;
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}
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return payload_type;
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}
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static inline UINT32 decode_be32(const UINT8 *p)
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{
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return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
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}
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static inline UINT32 decode_be64(const UINT8 *p)
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{
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return ((UINT64)decode_be32(p) << 32) | decode_be32(p + 4);
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}
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static int rtcp_parse_packet(AVFormatContext *s1, const unsigned char *buf, int len)
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{
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RTPContext *s = s1->priv_data;
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if (buf[1] != 200)
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return -1;
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s->last_rtcp_ntp_time = decode_be64(buf + 8);
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s->last_rtcp_timestamp = decode_be32(buf + 16);
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return 0;
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}
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/**
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* Parse an RTP packet directly sent as raw data. Can only be used if
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* 'raw' is given as input file
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* @param s1 media file context
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* @param pkt returned packet
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* @param buf input buffer
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* @param len buffer len
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* @return zero if no error.
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*/
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int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt,
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const unsigned char *buf, int len)
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{
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RTPContext *s = s1->priv_data;
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unsigned int ssrc, h;
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int payload_type, seq, delta_timestamp;
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AVStream *st;
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UINT32 timestamp;
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if (len < 12)
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return -1;
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if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
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return -1;
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if (buf[1] >= 200 && buf[1] <= 204) {
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rtcp_parse_packet(s1, buf, len);
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return -1;
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}
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payload_type = buf[1] & 0x7f;
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seq = (buf[2] << 8) | buf[3];
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timestamp = decode_be32(buf + 4);
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ssrc = decode_be32(buf + 8);
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if (s->payload_type < 0) {
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s->payload_type = payload_type;
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if (payload_type == RTP_PT_MPEG2TS) {
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/* XXX: special case : not a single codec but a whole stream */
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return -1;
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} else {
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st = av_new_stream(s1, 0);
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if (!st)
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return -1;
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rtp_get_codec_info(&st->codec, payload_type);
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}
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}
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/* NOTE: we can handle only one payload type */
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if (s->payload_type != payload_type)
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return -1;
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#if defined(DEBUG) || 1
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if (seq != ((s->seq + 1) & 0xffff)) {
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printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n",
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payload_type, seq, ((s->seq + 1) & 0xffff));
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}
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s->seq = seq;
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#endif
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len -= 12;
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buf += 12;
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st = s1->streams[0];
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switch(st->codec.codec_id) {
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case CODEC_ID_MP2:
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/* better than nothing: skip mpeg audio RTP header */
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if (len <= 4)
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return -1;
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h = decode_be32(buf);
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len -= 4;
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buf += 4;
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av_new_packet(pkt, len);
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memcpy(pkt->data, buf, len);
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break;
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case CODEC_ID_MPEG1VIDEO:
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/* better than nothing: skip mpeg audio RTP header */
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if (len <= 4)
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return -1;
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h = decode_be32(buf);
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buf += 4;
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len -= 4;
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if (h & (1 << 26)) {
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/* mpeg2 */
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if (len <= 4)
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return -1;
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buf += 4;
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len -= 4;
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}
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av_new_packet(pkt, len);
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memcpy(pkt->data, buf, len);
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break;
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default:
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av_new_packet(pkt, len);
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memcpy(pkt->data, buf, len);
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break;
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}
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if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
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/* compute pts from timestamp with received ntp_time */
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delta_timestamp = timestamp - s->last_rtcp_timestamp;
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/* XXX: do conversion, but not needed for mpeg at 90 KhZ */
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pkt->pts = s->last_rtcp_ntp_time + delta_timestamp;
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}
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return 0;
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}
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static int rtp_read_header(AVFormatContext *s1,
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AVFormatParameters *ap)
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{
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RTPContext *s = s1->priv_data;
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s->payload_type = -1;
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s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
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return 0;
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}
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static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt)
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{
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char buf[RTP_MAX_PACKET_LENGTH];
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int ret;
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/* XXX: needs a better API for packet handling ? */
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for(;;) {
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ret = url_read(url_fileno(&s1->pb), buf, sizeof(buf));
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if (ret < 0)
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return AVERROR_IO;
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if (rtp_parse_packet(s1, pkt, buf, ret) == 0)
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break;
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}
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return 0;
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}
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static int rtp_read_close(AVFormatContext *s1)
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{
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// RTPContext *s = s1->priv_data;
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return 0;
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}
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static int rtp_probe(AVProbeData *p)
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{
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if (strstart(p->filename, "rtp://", NULL))
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return AVPROBE_SCORE_MAX;
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return 0;
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}
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/* rtp output */
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static int rtp_write_header(AVFormatContext *s1)
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{
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RTPContext *s = s1->priv_data;
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int payload_type, max_packet_size;
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AVStream *st;
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if (s1->nb_streams != 1)
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return -1;
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st = s1->streams[0];
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payload_type = rtp_get_payload_type(&st->codec);
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if (payload_type < 0)
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return -1;
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s->payload_type = payload_type;
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s->base_timestamp = random();
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s->timestamp = s->base_timestamp;
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s->ssrc = random();
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s->first_packet = 1;
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max_packet_size = url_fget_max_packet_size(&s1->pb);
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if (max_packet_size <= 12)
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return AVERROR_IO;
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s->max_payload_size = max_packet_size - 12;
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switch(st->codec.codec_id) {
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case CODEC_ID_MP2:
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|
|
case CODEC_ID_MP3LAME:
|
|
|
|
s->buf_ptr = s->buf + 4;
|
|
|
|
s->cur_timestamp = 0;
|
|
|
|
break;
|
|
|
|
case CODEC_ID_MPEG1VIDEO:
|
|
|
|
s->cur_timestamp = 0;
|
|
|
|
break;
|
|
|
|
default:
|
|
|
|
s->buf_ptr = s->buf;
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* send an rtcp sender report packet */
|
|
|
|
static void rtcp_send_sr(AVFormatContext *s1, INT64 ntp_time)
|
|
|
|
{
|
|
|
|
RTPContext *s = s1->priv_data;
|
|
|
|
#if defined(DEBUG)
|
|
|
|
printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
|
|
|
|
#endif
|
|
|
|
put_byte(&s1->pb, (RTP_VERSION << 6));
|
|
|
|
put_byte(&s1->pb, 200);
|
|
|
|
put_be16(&s1->pb, 6); /* length in words - 1 */
|
|
|
|
put_be32(&s1->pb, s->ssrc);
|
|
|
|
put_be64(&s1->pb, ntp_time);
|
|
|
|
put_be32(&s1->pb, s->timestamp);
|
|
|
|
put_be32(&s1->pb, s->packet_count);
|
|
|
|
put_be32(&s1->pb, s->octet_count);
|
|
|
|
put_flush_packet(&s1->pb);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* send an rtp packet. sequence number is incremented, but the caller
|
|
|
|
must update the timestamp itself */
|
|
|
|
static void rtp_send_data(AVFormatContext *s1, UINT8 *buf1, int len)
|
|
|
|
{
|
|
|
|
RTPContext *s = s1->priv_data;
|
|
|
|
|
|
|
|
#ifdef DEBUG
|
|
|
|
printf("rtp_send_data size=%d\n", len);
|
|
|
|
#endif
|
|
|
|
|
|
|
|
/* build the RTP header */
|
|
|
|
put_byte(&s1->pb, (RTP_VERSION << 6));
|
|
|
|
put_byte(&s1->pb, s->payload_type & 0x7f);
|
|
|
|
put_be16(&s1->pb, s->seq);
|
|
|
|
put_be32(&s1->pb, s->timestamp);
|
|
|
|
put_be32(&s1->pb, s->ssrc);
|
|
|
|
|
|
|
|
put_buffer(&s1->pb, buf1, len);
|
|
|
|
put_flush_packet(&s1->pb);
|
|
|
|
|
|
|
|
s->seq++;
|
|
|
|
s->octet_count += len;
|
|
|
|
s->packet_count++;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* send an integer number of samples and compute time stamp and fill
|
|
|
|
the rtp send buffer before sending. */
|
|
|
|
static void rtp_send_samples(AVFormatContext *s1,
|
|
|
|
UINT8 *buf1, int size, int sample_size)
|
|
|
|
{
|
|
|
|
RTPContext *s = s1->priv_data;
|
|
|
|
int len, max_packet_size, n;
|
|
|
|
|
|
|
|
max_packet_size = (s->max_payload_size / sample_size) * sample_size;
|
|
|
|
/* not needed, but who nows */
|
|
|
|
if ((size % sample_size) != 0)
|
|
|
|
av_abort();
|
|
|
|
while (size > 0) {
|
|
|
|
len = (max_packet_size - (s->buf_ptr - s->buf));
|
|
|
|
if (len > size)
|
|
|
|
len = size;
|
|
|
|
|
|
|
|
/* copy data */
|
|
|
|
memcpy(s->buf_ptr, buf1, len);
|
|
|
|
s->buf_ptr += len;
|
|
|
|
buf1 += len;
|
|
|
|
size -= len;
|
|
|
|
n = (s->buf_ptr - s->buf);
|
|
|
|
/* if buffer full, then send it */
|
|
|
|
if (n >= max_packet_size) {
|
|
|
|
rtp_send_data(s1, s->buf, n);
|
|
|
|
s->buf_ptr = s->buf;
|
|
|
|
/* update timestamp */
|
|
|
|
s->timestamp += n / sample_size;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* NOTE: we suppose that exactly one frame is given as argument here */
|
|
|
|
/* XXX: test it */
|
|
|
|
static void rtp_send_mpegaudio(AVFormatContext *s1,
|
|
|
|
UINT8 *buf1, int size)
|
|
|
|
{
|
|
|
|
RTPContext *s = s1->priv_data;
|
|
|
|
AVStream *st = s1->streams[0];
|
|
|
|
int len, count, max_packet_size;
|
|
|
|
|
|
|
|
max_packet_size = s->max_payload_size;
|
|
|
|
|
|
|
|
/* test if we must flush because not enough space */
|
|
|
|
len = (s->buf_ptr - s->buf);
|
|
|
|
if ((len + size) > max_packet_size) {
|
|
|
|
if (len > 4) {
|
|
|
|
rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
|
|
|
|
s->buf_ptr = s->buf + 4;
|
|
|
|
/* 90 KHz time stamp */
|
|
|
|
s->timestamp = s->base_timestamp +
|
|
|
|
(s->cur_timestamp * 90000LL) / st->codec.sample_rate;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* add the packet */
|
|
|
|
if (size > max_packet_size) {
|
|
|
|
/* big packet: fragment */
|
|
|
|
count = 0;
|
|
|
|
while (size > 0) {
|
|
|
|
len = max_packet_size - 4;
|
|
|
|
if (len > size)
|
|
|
|
len = size;
|
|
|
|
/* build fragmented packet */
|
|
|
|
s->buf[0] = 0;
|
|
|
|
s->buf[1] = 0;
|
|
|
|
s->buf[2] = count >> 8;
|
|
|
|
s->buf[3] = count;
|
|
|
|
memcpy(s->buf + 4, buf1, len);
|
|
|
|
rtp_send_data(s1, s->buf, len + 4);
|
|
|
|
size -= len;
|
|
|
|
buf1 += len;
|
|
|
|
count += len;
|
|
|
|
}
|
|
|
|
} else {
|
|
|
|
if (s->buf_ptr == s->buf + 4) {
|
|
|
|
/* no fragmentation possible */
|
|
|
|
s->buf[0] = 0;
|
|
|
|
s->buf[1] = 0;
|
|
|
|
s->buf[2] = 0;
|
|
|
|
s->buf[3] = 0;
|
|
|
|
}
|
|
|
|
memcpy(s->buf_ptr, buf1, size);
|
|
|
|
s->buf_ptr += size;
|
|
|
|
}
|
|
|
|
s->cur_timestamp += st->codec.frame_size;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* NOTE: a single frame must be passed with sequence header if
|
|
|
|
needed. XXX: use slices. */
|
|
|
|
static void rtp_send_mpegvideo(AVFormatContext *s1,
|
|
|
|
UINT8 *buf1, int size)
|
|
|
|
{
|
|
|
|
RTPContext *s = s1->priv_data;
|
|
|
|
AVStream *st = s1->streams[0];
|
|
|
|
int len, h, max_packet_size;
|
|
|
|
UINT8 *q;
|
|
|
|
|
|
|
|
max_packet_size = s->max_payload_size;
|
|
|
|
|
|
|
|
while (size > 0) {
|
|
|
|
/* XXX: more correct headers */
|
|
|
|
h = 0;
|
|
|
|
if (st->codec.sub_id == 2)
|
|
|
|
h |= 1 << 26; /* mpeg 2 indicator */
|
|
|
|
q = s->buf;
|
|
|
|
*q++ = h >> 24;
|
|
|
|
*q++ = h >> 16;
|
|
|
|
*q++ = h >> 8;
|
|
|
|
*q++ = h;
|
|
|
|
|
|
|
|
if (st->codec.sub_id == 2) {
|
|
|
|
h = 0;
|
|
|
|
*q++ = h >> 24;
|
|
|
|
*q++ = h >> 16;
|
|
|
|
*q++ = h >> 8;
|
|
|
|
*q++ = h;
|
|
|
|
}
|
|
|
|
|
|
|
|
len = max_packet_size - (q - s->buf);
|
|
|
|
if (len > size)
|
|
|
|
len = size;
|
|
|
|
|
|
|
|
memcpy(q, buf1, len);
|
|
|
|
q += len;
|
|
|
|
|
|
|
|
/* 90 KHz time stamp */
|
|
|
|
/* XXX: overflow */
|
|
|
|
s->timestamp = s->base_timestamp +
|
|
|
|
(s->cur_timestamp * 90000LL * FRAME_RATE_BASE) / st->codec.frame_rate;
|
|
|
|
rtp_send_data(s1, s->buf, q - s->buf);
|
|
|
|
|
|
|
|
buf1 += len;
|
|
|
|
size -= len;
|
|
|
|
}
|
|
|
|
s->cur_timestamp++;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* write an RTP packet. 'buf1' must contain a single specific frame. */
|
|
|
|
static int rtp_write_packet(AVFormatContext *s1, int stream_index,
|
|
|
|
UINT8 *buf1, int size, int force_pts)
|
|
|
|
{
|
|
|
|
RTPContext *s = s1->priv_data;
|
|
|
|
AVStream *st = s1->streams[0];
|
|
|
|
int rtcp_bytes;
|
|
|
|
INT64 ntp_time;
|
|
|
|
|
|
|
|
#ifdef DEBUG
|
|
|
|
printf("%d: write len=%d\n", stream_index, size);
|
|
|
|
#endif
|
|
|
|
|
|
|
|
/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
|
|
|
|
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
|
|
|
|
RTCP_TX_RATIO_DEN;
|
|
|
|
if (s->first_packet || rtcp_bytes >= 28) {
|
|
|
|
/* compute NTP time */
|
|
|
|
ntp_time = force_pts; // ((INT64)force_pts << 28) / 5625
|
|
|
|
rtcp_send_sr(s1, ntp_time);
|
|
|
|
s->last_octet_count = s->octet_count;
|
|
|
|
s->first_packet = 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
switch(st->codec.codec_id) {
|
|
|
|
case CODEC_ID_PCM_MULAW:
|
|
|
|
case CODEC_ID_PCM_ALAW:
|
|
|
|
case CODEC_ID_PCM_U8:
|
|
|
|
case CODEC_ID_PCM_S8:
|
|
|
|
rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
|
|
|
|
break;
|
|
|
|
case CODEC_ID_PCM_U16BE:
|
|
|
|
case CODEC_ID_PCM_U16LE:
|
|
|
|
case CODEC_ID_PCM_S16BE:
|
|
|
|
case CODEC_ID_PCM_S16LE:
|
|
|
|
rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
|
|
|
|
break;
|
|
|
|
case CODEC_ID_MP2:
|
|
|
|
case CODEC_ID_MP3LAME:
|
|
|
|
rtp_send_mpegaudio(s1, buf1, size);
|
|
|
|
break;
|
|
|
|
case CODEC_ID_MPEG1VIDEO:
|
|
|
|
rtp_send_mpegvideo(s1, buf1, size);
|
|
|
|
break;
|
|
|
|
default:
|
|
|
|
return AVERROR_IO;
|
|
|
|
}
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int rtp_write_trailer(AVFormatContext *s1)
|
|
|
|
{
|
|
|
|
// RTPContext *s = s1->priv_data;
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
AVInputFormat rtp_demux = {
|
|
|
|
"rtp",
|
|
|
|
"RTP input format",
|
|
|
|
sizeof(RTPContext),
|
|
|
|
rtp_probe,
|
|
|
|
rtp_read_header,
|
|
|
|
rtp_read_packet,
|
|
|
|
rtp_read_close,
|
2002-10-07 01:06:06 +02:00
|
|
|
.flags = AVFMT_NOHEADER,
|
2002-07-04 12:38:01 +02:00
|
|
|
};
|
|
|
|
|
|
|
|
AVOutputFormat rtp_mux = {
|
|
|
|
"rtp",
|
|
|
|
"RTP output format",
|
|
|
|
NULL,
|
|
|
|
NULL,
|
|
|
|
sizeof(RTPContext),
|
|
|
|
CODEC_ID_PCM_MULAW,
|
|
|
|
CODEC_ID_NONE,
|
|
|
|
rtp_write_header,
|
|
|
|
rtp_write_packet,
|
|
|
|
rtp_write_trailer,
|
|
|
|
};
|
|
|
|
|
|
|
|
int rtp_init(void)
|
|
|
|
{
|
|
|
|
av_register_output_format(&rtp_mux);
|
|
|
|
av_register_input_format(&rtp_demux);
|
|
|
|
return 0;
|
|
|
|
}
|