Initial commit: TXW82x FPV v2.7.0.7-42229 SDK + project sources
This commit is contained in:
199
sdk/app/spook/rtp-audio.c
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199
sdk/app/spook/rtp-audio.c
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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 "lib/audio/audio_code/audio_code.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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};
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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 mpeg4-generic/%d/1\r\na=fmtp:97 profile-level-id=1; mode=AAC-hbr; config=1588; sizeLength=13; indexlength=3; indexdeltalength=3\r\n", port,audio_sample_rate);
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// return snprintf( dest, len, "m=audio %d RTP/AVP 97\r\na=rtpmap:97 PCMA/%d/1\r\n", port,audio_sample_rate);
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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 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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static void audio_build_au_header( unsigned char *au_hdr, int au_size )
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{
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au_hdr[0] = 0x00;
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au_hdr[1] = 0x10;
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au_hdr[2] = ( au_size & 0x1FE0 ) >> 5;
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au_hdr[3] = ( au_size & 0x1F ) << 3;
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}
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static int audio_send_au_to_endpoint( struct rtp_endpoint *ep, uint32_t timestamp, uint8_t *au_data, uint32_t au_size, int times )
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{
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struct iovec v[3];
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uint8_t au_hdr[4];
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int max_payload;
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uint32_t max_fragment_size;
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uint32_t offset = 0;
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uint32_t fragment_len;
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if( !ep || !ep->sendEnable || !au_data || au_size <= 0 ) return -1;
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if( au_size > 0x1FFF )
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{
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spook_log( SL_ERR, "AAC AU too large for 13-bit AU-size field: %d", au_size );
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ep->sendEnable = 0;
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return -1;
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}
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max_payload = rtp_get_payload_size_limit( ep, RTP_HEADER_SIZE );
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if( max_payload <= (int)sizeof( au_hdr ) )
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{
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ep->sendEnable = 0;
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return -1;
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}
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max_fragment_size = max_payload - sizeof( au_hdr );
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audio_build_au_header( au_hdr, au_size );
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v[1].iov_base = au_hdr;
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v[1].iov_len = sizeof( au_hdr );
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while( offset < au_size )
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{
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fragment_len = au_size - offset;
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if( fragment_len > max_fragment_size )
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fragment_len = max_fragment_size;
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v[2].iov_base = (void *)( au_data + offset );
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v[2].iov_len = fragment_len;
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if( rtp_sendmsg( ep, v, 3, timestamp, offset + fragment_len >= au_size, times ) < 0 )
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{
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ep->sendEnable = 0;
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return -1;
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}
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offset += fragment_len;
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}
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return 0;
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}
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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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struct rtp_endpoint *ep;
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void *head;
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uint32_t au_size;
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struct rtp_audio *out = (struct rtp_audio *)d;
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struct framebuff *fb = (struct framebuff *)out->f->get_f;
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if( !fb || fb->len <= 7 )
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{
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return 0;
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}
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au_size = fb->len - 7;
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head = ls;
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while(head)
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{
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head = search(head,track,(void*)&ep);
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if(!ep)
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{
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continue;
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}
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if(ep->sendEnable)
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{
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if( audio_send_au_to_endpoint( ep, out->timestamp, fb->data + 7, au_size, 10 ) < 0 )
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{
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ep->sendEnable = 0;
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}
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}
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ep->sendEnable = 1;
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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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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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