pure sdk for main
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divadiow
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/*
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* Copyright (C) 2004 Nathan Lutchansky <lutchann@litech.org>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program 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
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software Foundation,
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* Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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#include <stdio.h>
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#include "lwip\sockets.h"
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#include "lwip\netif.h"
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#include "lwip\dns.h"
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#include "lwip\api.h"
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#include "lwip\tcp.h"
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#include <event.h>
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#include <log.h>
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#include <frame.h>
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#include <stream.h>
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#include <rtp.h>
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#include <rtp_media.h>
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#include <spook_config.h>
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#include "list.h"
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#include "jpgdef.h"
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#include "rtp.h"
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#include "session.h"
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#include "osal/sleep.h"
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#include "newaudio/newaudio.h"
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#define AUDIO_SAMPLE 8000
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struct rtp_audio {
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//为了获取到frame,因为使用了链表形式
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struct frame *f;
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unsigned int timestamp;
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int max_send_size;
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};
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static int audio_get_sdp( char *dest, int len, int payload, int port, void *d )
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{
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uint32_t audio_sample_rate = (uint32)d;
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_os_printf("audio_sample_rate:%d\n",audio_sample_rate);
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if(audio_sample_rate == 0)
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{
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audio_sample_rate = AUDIO_SAMPLE;//默认采样率
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}
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return snprintf( dest, len, "m=audio %d RTP/AVP 97\r\na=rtpmap:97 L16/%d/1\r\n", port,audio_sample_rate);
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}
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static int audio_process_frame( struct frame *f, void *d )
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{
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_os_printf("+");
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struct rtp_media *rtp = (struct rtp_media *)d;
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struct rtp_audio *out = (struct rtp_audio *)rtp->private;
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out->f = f;
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out->timestamp = f->timestamp*rtp->sample_rate/(1000);
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return 1;
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}
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static int audio_get_payload( int payload, void *d )
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{
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_os_printf("%s:%d\n",__FUNCTION__,__LINE__);
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return 97;
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}
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static uint8_t *audio_send_buf = (uint8_t*)live_buf_cache;
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#if 0
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static int audio_send( struct rtp_endpoint *ep, void *d )
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{
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//_os_printf("%s:%d\n",__FUNCTION__,__LINE__);
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struct rtp_audio *out = (struct rtp_audio*)d;
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uint8_t *inter,*rtphdr,*realdata;
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int size = -1;
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int timeouts = 0;
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int fd = ep->trans.udp.rtp_fd;
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uint32_t i;
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struct sockaddr_in rtpaddr;
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unsigned int namelen = sizeof( rtpaddr );
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if( getsockname( fd, (struct sockaddr *)&rtpaddr, &namelen ) < 0 ) {
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spook_log( SL_ERR, "sendmsg getsockname error");
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}
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inter = audio_send_buf;
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rtphdr = inter;
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realdata = rtphdr + 12;
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struct frame *f = out->f;
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uint8_t *real_buf = GET_AUDIO_BUF(f->get_f);
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//_os_printf("out:%X\n",out);
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//_os_printf("real_buf:%X\t%X\t%X\t%d\n",real_buf,real_buf[0],real_buf[1],f->length);
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//memcpy(realdata,real_buf,f->length);
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for(i=0;i<f->length;i+=2)
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{
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realdata[i] = real_buf[i+1];
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realdata[i+1] = real_buf[i];
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}
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//_os_printf("length111:%d\n",f->length);
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//_os_printf("real:%X\t%X\t%d\n",realdata[0],realdata[1],out->timestamp);
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ep->last_timestamp = ( ep->start_timestamp + out->timestamp )& 0xFFFFFFFF;
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inter[0] = 2 << 6; /* version */
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//帧结束
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inter[1] = ep->payload | 0x80;
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PUT_16(rtphdr + 2, ep->seqnum );
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PUT_32(rtphdr+4, ep->last_timestamp);
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PUT_32( rtphdr + 8, ep->ssrc );
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ep->seqnum = ( ep->seqnum + 1 ) & 0xFFFF;
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while(size < 0 )
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{
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//size = sendto(fd, sendtobuf, total_len, MSG_DONTWAIT, (struct sockaddr *)&rtpaddr, namelen);
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//-8是音频过来前8字节是空的
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size = sendto(fd, audio_send_buf, f->length+12, MSG_DONTWAIT, (struct sockaddr *)&rtpaddr, namelen);
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//_os_printf("seq:%d\t%X\t%X\n",ep->seqnum,audio_send_buf[12],audio_send_buf[13] );
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timeouts++;
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if(timeouts>20)
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{
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break;
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}
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if(timeouts%2==0)
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{
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os_sleep_ms(1);
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}
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}
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//累计发送的数据量大小
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ep->octet_count += (f->length+12);
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//累计发送的数据包数量,由于现在音频只有1K左右,所以这里+1,实际要看发送多少包,然后增加多少包
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++ep->packet_count;
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//_os_printf("%s:%d\ttime:%d\n",__FUNCTION__,__LINE__,krhino_sys_tick_get());
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return 0;
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}
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#else
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static int audio_send( struct rtp_endpoint *ep, void *d )
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{
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return 0;
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}
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#endif
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static unsigned char *get_send_real_data_audio()
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{
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return audio_send_buf;
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}
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//设置数据头部,并且将音频数据拷贝到需要发送的buf中,返回需要发送的buf长度
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static int set_send_rtp_packet_head_audio(struct rtp_endpoint *ep, unsigned int timestamp, int marker,unsigned char *real_buf,unsigned char *send_buf,int plen )
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{
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uint8_t *inter,*rtphdr;
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unsigned char *data_buf = send_buf+12;
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inter = send_buf;
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rtphdr = inter;
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int i;
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ep->last_timestamp = ( ep->start_timestamp + timestamp )& 0xFFFFFFFF;
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inter[0] = 2 << 6; /* version */
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if(marker)
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{
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inter[1] = ep->payload | 0x80;
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}
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else
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{
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inter[1] = ep->payload;
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}
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PUT_32(rtphdr+4, ep->last_timestamp);
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PUT_32( rtphdr + 8, ep->ssrc );
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PUT_16(rtphdr + 2, ep->seqnum );
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ep->seqnum = ( ep->seqnum + 1 ) & 0xFFFF;
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++ep->packet_count;
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ep->octet_count += (plen);
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//音频数据进行大小端互换
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for(i=0;i<plen;i+=2)
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{
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data_buf[i] = real_buf[i+1];
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data_buf[i+1] = real_buf[i];
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}
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return plen+12;
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}
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//端口发送数据,
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int fd_send_data_audio(int fd,unsigned char *sendbuf,int sendLen,int times)
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{
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int size = -1;
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int timeouts = 0;
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struct sockaddr_in rtpaddr;
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unsigned int namelen = sizeof( rtpaddr );
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if( getsockname( fd, (struct sockaddr *)&rtpaddr, &namelen ) < 0 ) {
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spook_log( SL_ERR, "sendmsg getsockname error");
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}
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while(size < 0 )
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{
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//size = sendto(fd, sendtobuf, total_len, MSG_DONTWAIT, (struct sockaddr *)&rtpaddr, namelen);
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size = sendto(fd, sendbuf, sendLen, MSG_DONTWAIT, (struct sockaddr *)&rtpaddr, namelen);
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//_os_printf("P:%d ",size);
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timeouts++;
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if(timeouts>times)
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{
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break;
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}
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if(size < 0)
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{
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os_sleep_ms(3);
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}
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}
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if(size<0)
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{
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_os_printf("%s err size:%d\n",__FUNCTION__,size);
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return -1;
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}
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else
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{
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return 0;
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}
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}
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//数据发送
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int send_rtp_packet_more_audio( struct rtp_endpoint *ep, unsigned char *sendbuf, int sendLen,int times )
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{
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fd_send_data_audio(ep->trans.udp.rtp_fd,sendbuf,sendLen,times);
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return 0;
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}
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//支持大于1460的音频数量
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static int audio_send_more( rtp_loop_search_ep search,void *ls,void *track, void *d )
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{
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int max_data_size;
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unsigned char *send_buf;
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int plen;
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struct rtp_endpoint *ep;
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void *head;
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int send_total_len;
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int i;
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struct rtp_audio *out = (struct rtp_audio *)d;
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struct frame *f = out->f;
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uint8_t *real_buf = (uint8_t *)GET_DATA_BUF(f->get_f);
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int audio_flen = GET_DATA_LEN(f->get_f);
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//12是音频中rtp的头
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max_data_size = out->max_send_size-12;
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for( i = 0; i < audio_flen; i += plen )
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{
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if((audio_flen - i) > max_data_size)
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{
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plen = max_data_size;
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}
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else
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{
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plen = audio_flen - i;
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}
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send_buf = get_send_real_data_audio();
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//重新赋值ls的头
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head = ls;
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while(head)
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{
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//获取ep
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head = search(head,track,(void*)&ep);
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if(ep)
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{
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//数据内容是一样的,修改头部就可以了
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send_total_len = set_send_rtp_packet_head_audio(ep,out->timestamp, plen + i == audio_flen,&real_buf[i],send_buf,plen );
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send_rtp_packet_more(ep, send_buf, send_total_len,30 );
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}
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}
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}
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return 0;
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}
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struct rtp_media *new_rtp_media_audio_stream( struct stream *stream )
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{
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struct rtp_audio *out;
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int fincr, fbase;
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struct rtp_media *m;
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stream->get_framerate( stream, &fincr, &fbase );
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out = (struct rtp_audio *)malloc( sizeof( struct rtp_audio ) );
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out->f = NULL;
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out->timestamp = 0;
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out->max_send_size = MAX_DATA_PACKET_SIZE;
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//return new_rtp_media( audio_get_sdp, audio_get_payload,audio_process_frame, audio_send, out );
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m = new_rtp_rtcp_media( audio_get_sdp, audio_get_payload,audio_process_frame, audio_send,new_rtcp_send, out );
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if(m)
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{
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m->sample_rate = AUDIO_SAMPLE;//默认
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m->type = 1; //音频
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m->send_more = audio_send_more;
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m->per_ms_incr = AUDIO_SAMPLE/(1000);
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}
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return m;
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}
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