mirror of
https://github.com/outbackdingo/UltraGrid.git
synced 2026-03-22 05:40:27 +00:00
c3bb31928e
When destroying the state, ServerMediaSubsession::deleteStream callbacks are called but we are already deconfiguring so do not send messages to change destination (eg. audio is already destroyed so the message wont be delivered causing warnings like: ``` Receiver audio.sender does not exist. Receiver audio.sender does not exist. Warning: Message queue not empty! ``` Currently this is just a dirty fix, rtsp_serv::watch would be better to pass.
366 lines
12 KiB
C++
366 lines
12 KiB
C++
/*
|
|
* FILE: rtsp/BasicRTSPOnlySubsession.cpp
|
|
* AUTHORS: David Cassany <david.cassany@i2cat.net>
|
|
* Gerard Castillo <gerard.castillo@i2cat.net>
|
|
*
|
|
* Copyright (c) 2005-2010 Fundació i2CAT, Internet I Innovació Digital a Catalunya
|
|
* Copyright (c) 2014-2023 CESNET, z. s. p. o.
|
|
*
|
|
* Redistribution and use in source and binary forms, with or without
|
|
* modification, is permitted provided that the following conditions
|
|
* are met:
|
|
*
|
|
* 1. Redistributions of source code must retain the above copyright
|
|
* notice, this list of conditions and the following disclaimer.
|
|
*
|
|
* 2. Redistributions in binary form must reproduce the above copyright
|
|
* notice, this list of conditions and the following disclaimer in the
|
|
* documentation and/or other materials provided with the distribution.
|
|
*
|
|
* 3. All advertising materials mentioning features or use of this software
|
|
* must display the following acknowledgement:
|
|
*
|
|
* This product includes software developed by the Fundació i2CAT,
|
|
* Internet I Innovació Digital a Catalunya. This product also includes
|
|
* software developed by CESNET z.s.p.o.
|
|
*
|
|
* 4. Neither the name of the University nor of the Institute may be used
|
|
* to endorse or promote products derived from this software without
|
|
* specific prior written permission.
|
|
*
|
|
* THIS SOFTWARE IS PROVIDED BY THE AUTHORS AND CONTRIBUTORS
|
|
* "AS IS" AND ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING,
|
|
* BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY
|
|
* AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
|
|
* EVENT SHALL THE AUTHORS OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
|
|
* INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
|
|
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
|
|
* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
|
|
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
|
|
* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
|
|
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE,
|
|
* EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
|
*
|
|
*/
|
|
|
|
#include "rtsp/BasicRTSPOnlySubsession.hh"
|
|
#include <BasicUsageEnvironment.hh>
|
|
#include <RTSPServer.hh>
|
|
#include <GroupsockHelper.hh>
|
|
|
|
#include "messaging.h"
|
|
|
|
BasicRTSPOnlySubsession*
|
|
BasicRTSPOnlySubsession::createNew(UsageEnvironment& env,
|
|
Boolean reuseFirstSource, struct module *mod, rtps_types_t avType,
|
|
audio_codec_t audio_codec, int audio_sample_rate, int audio_channels,
|
|
int audio_bps, int rtp_port, int rtp_port_audio) {
|
|
return new BasicRTSPOnlySubsession(env, reuseFirstSource, mod, avType,
|
|
audio_codec, audio_sample_rate, audio_channels, audio_bps, rtp_port, rtp_port_audio);
|
|
}
|
|
|
|
BasicRTSPOnlySubsession::BasicRTSPOnlySubsession(UsageEnvironment& env,
|
|
Boolean reuseFirstSource, struct module *mod, rtps_types_t avType,
|
|
audio_codec_t audio_codec, int audio_sample_rate, int audio_channels,
|
|
int audio_bps, int rtp_port, int rtp_port_audio) :
|
|
ServerMediaSubsession(env), fSDPLines(NULL), fReuseFirstSource(
|
|
reuseFirstSource), fLastStreamToken(NULL) {
|
|
Vdestination = NULL;
|
|
Adestination = NULL;
|
|
gethostname(fCNAME, sizeof fCNAME);
|
|
this->fmod = mod;
|
|
this->avType = avType;
|
|
this->audio_codec = audio_codec;
|
|
this->audio_sample_rate = audio_sample_rate;
|
|
this->audio_channels = audio_channels;
|
|
this->audio_bps = audio_bps;
|
|
this->rtp_port = rtp_port;
|
|
this->rtp_port_audio = rtp_port_audio;
|
|
fCNAME[sizeof fCNAME - 1] = '\0';
|
|
}
|
|
|
|
BasicRTSPOnlySubsession::~BasicRTSPOnlySubsession() {
|
|
delete[] fSDPLines;
|
|
delete Adestination;
|
|
delete Vdestination;
|
|
}
|
|
|
|
char const* BasicRTSPOnlySubsession::sdpLines() {
|
|
if (fSDPLines == NULL) {
|
|
setSDPLines();
|
|
}
|
|
return fSDPLines;
|
|
}
|
|
|
|
void BasicRTSPOnlySubsession::setSDPLines() {
|
|
//TODO: should be more dynamic
|
|
//VStream
|
|
if (avType == video || avType == av) {
|
|
unsigned estBitrate = 5000;
|
|
char const* mediaType = "video";
|
|
uint8_t rtpPayloadType = 96;
|
|
AddressString ipAddressStr(fServerAddressForSDP);
|
|
char* rtpmapLine = strdup("a=rtpmap:96 H264/90000\n");
|
|
//char const* auxSDPLine = "";
|
|
|
|
char const* const sdpFmt = "m=%s %u RTP/AVP %u\r\n"
|
|
"c=IN IP4 %s\r\n"
|
|
"b=AS:%u\r\n"
|
|
"a=rtcp:%d\r\n"
|
|
"%s"
|
|
"a=control:%s\r\n";
|
|
unsigned sdpFmtSize = strlen(sdpFmt) + strlen(mediaType) + 5 /* max short len */
|
|
+ 3 /* max char len */
|
|
+ strlen(ipAddressStr.val()) + 20 /* max int len */
|
|
+ strlen(rtpmapLine) + strlen(trackId());
|
|
char* sdpLines = new char[sdpFmtSize];
|
|
|
|
snprintf(sdpLines, sdpFmtSize, sdpFmt, mediaType, // m= <media>
|
|
rtp_port,//fPortNumForSDP, // m= <port>
|
|
rtpPayloadType, // m= <fmt list>
|
|
ipAddressStr.val(), // c= address
|
|
estBitrate, // b=AS:<bandwidth>
|
|
rtp_port + 1,
|
|
rtpmapLine, // a=rtpmap:... (if present)
|
|
trackId()); // a=control:<track-id>
|
|
|
|
fSDPLines = sdpLines;
|
|
free(rtpmapLine);
|
|
}
|
|
//AStream
|
|
if (avType == audio || avType == av) {
|
|
unsigned estBitrate = 384;
|
|
char const* mediaType = "audio";
|
|
AddressString ipAddressStr(fServerAddressForSDP);
|
|
uint8_t rtpPayloadType;
|
|
|
|
if (audio_sample_rate == 8000 && audio_channels == 1) { //NOW NOT COMPUTING 1 BPS BECAUSE RESAMPLER FORCES TO 2 BPS...
|
|
if (audio_codec == AC_MULAW)
|
|
rtpPayloadType = 0;
|
|
else if (audio_codec == AC_ALAW)
|
|
rtpPayloadType = 8;
|
|
else rtpPayloadType = 97;
|
|
} else {
|
|
rtpPayloadType = 97;
|
|
}
|
|
|
|
char* rtpmapLine = strdup("a=rtpmap:97 PCMU/48000/2\n"); //only to alloc max possible size
|
|
//char const* auxSDPLine = "";
|
|
|
|
const int sdp_ch_count = audio_codec == AC_OPUS ? 2 :
|
|
audio_channels; // RFC 7587 enforces 2 for Opus
|
|
char const* const sdpFmt = "m=%s %u RTP/AVP %u\r\n"
|
|
"c=IN IP4 %s\r\n"
|
|
"b=AS:%u\r\n"
|
|
"a=rtcp:%d\r\n"
|
|
"a=rtpmap:%u %s/%d/%d\r\n"
|
|
"a=control:%s\r\n";
|
|
unsigned sdpFmtSize = strlen(sdpFmt) + strlen(mediaType) + 5 /* max short len */
|
|
+ 3 /* max char len */
|
|
+ strlen(ipAddressStr.val()) + 20 /* max int len */
|
|
+ strlen(rtpmapLine) + strlen(trackId());
|
|
char* sdpLines = new char[sdpFmtSize];
|
|
|
|
snprintf(sdpLines, sdpFmtSize, sdpFmt,
|
|
mediaType, // m= <media>
|
|
rtp_port_audio,//fPortNumForSDP, // m= <port>
|
|
rtpPayloadType, // m= <fmt list>
|
|
ipAddressStr.val(), // c= address
|
|
estBitrate, // b=AS:<bandwidth>
|
|
//rtpmapLine, // a=rtpmap:... (if present)
|
|
rtp_port_audio + 1,
|
|
rtpPayloadType,
|
|
audio_codec == AC_MULAW ? "PCMU" : audio_codec == AC_ALAW ? "PCMA" : "opus",
|
|
audio_sample_rate,
|
|
sdp_ch_count,
|
|
trackId()); // a=control:<track-id>
|
|
|
|
fSDPLines = sdpLines;
|
|
free(rtpmapLine);
|
|
}
|
|
}
|
|
|
|
void BasicRTSPOnlySubsession::getStreamParameters(unsigned /* clientSessionId */,
|
|
netAddressBits clientAddress, Port const& clientRTPPort,
|
|
Port const& clientRTCPPort, int /* tcpSocketNum */,
|
|
unsigned char /* rtpChannelId */, unsigned char /* rtcpChannelId */,
|
|
netAddressBits& destinationAddress, uint8_t& /*destinationTTL*/,
|
|
Boolean& /* isMulticast */, Port& serverRTPPort, Port& serverRTCPPort,
|
|
void*& /* streamToken */) {
|
|
if (avType == video || avType == av) {
|
|
Port rtp(rtp_port);
|
|
serverRTPPort = rtp;
|
|
Port rtcp(rtp_port + 1);
|
|
serverRTCPPort = rtcp;
|
|
|
|
if (fSDPLines == NULL) {
|
|
setSDPLines();
|
|
}
|
|
if (destinationAddress == 0) {
|
|
destinationAddress = clientAddress;
|
|
}
|
|
struct in_addr destinationAddr;
|
|
destinationAddr.s_addr = destinationAddress;
|
|
delete Vdestination;
|
|
Vdestination = new Destinations(destinationAddr, clientRTPPort,
|
|
clientRTCPPort);
|
|
}
|
|
if (avType == audio || avType == av) {
|
|
Port rtp(rtp_port_audio);
|
|
serverRTPPort = rtp;
|
|
Port rtcp(rtp_port_audio + 1);
|
|
serverRTCPPort = rtcp;
|
|
|
|
if (fSDPLines == NULL) {
|
|
setSDPLines();
|
|
}
|
|
if (destinationAddress == 0) {
|
|
destinationAddress = clientAddress;
|
|
}
|
|
struct in_addr destinationAddr;
|
|
destinationAddr.s_addr = destinationAddress;
|
|
delete Adestination;
|
|
Adestination = new Destinations(destinationAddr, clientRTPPort,
|
|
clientRTCPPort);
|
|
}
|
|
}
|
|
|
|
void BasicRTSPOnlySubsession::startStream(unsigned /* clientSessionId */,
|
|
void* /* streamToken */, TaskFunc* /* rtcpRRHandler */,
|
|
void* /* rtcpRRHandlerClientData */, unsigned short& /* rtpSeqNum */,
|
|
unsigned& /* rtpTimestamp */,
|
|
ServerRequestAlternativeByteHandler* /* serverRequestAlternativeByteHandler */,
|
|
void* /* serverRequestAlternativeByteHandlerClientData */) {
|
|
struct response *resp = NULL;
|
|
|
|
if (Vdestination != NULL) {
|
|
if (avType == video || avType == av) {
|
|
char pathV[1024];
|
|
|
|
memset(pathV, 0, sizeof(pathV));
|
|
enum module_class path_sender[] = { MODULE_CLASS_SENDER,
|
|
MODULE_CLASS_NONE };
|
|
append_message_path(pathV, sizeof(pathV), path_sender);
|
|
|
|
//CHANGE DST PORT
|
|
struct msg_sender *msgV1 = (struct msg_sender *) new_message(
|
|
sizeof(struct msg_sender));
|
|
msgV1->tx_port = ntohs(Vdestination->rtpPort.num());
|
|
msgV1->type = SENDER_MSG_CHANGE_PORT;
|
|
resp = send_message(fmod, pathV, (struct message *) msgV1);
|
|
free_response(resp);
|
|
|
|
//CHANGE DST ADDRESS
|
|
struct msg_sender *msgV2 = (struct msg_sender *) new_message(
|
|
sizeof(struct msg_sender));
|
|
strncpy(msgV2->receiver, inet_ntoa(Vdestination->addr),
|
|
sizeof(msgV2->receiver) - 1);
|
|
msgV2->type = SENDER_MSG_CHANGE_RECEIVER;
|
|
|
|
resp = send_message(fmod, pathV, (struct message *) msgV2);
|
|
free_response(resp);
|
|
}
|
|
}
|
|
|
|
if (Adestination != NULL) {
|
|
if (avType == audio || avType == av) {
|
|
char pathA[1024];
|
|
|
|
memset(pathA, 0, sizeof(pathA));
|
|
enum module_class path_sender[] = { MODULE_CLASS_AUDIO,
|
|
MODULE_CLASS_SENDER, MODULE_CLASS_NONE };
|
|
append_message_path(pathA, sizeof(pathA), path_sender);
|
|
|
|
//CHANGE DST PORT
|
|
struct msg_sender *msgA1 = (struct msg_sender *) new_message(
|
|
sizeof(struct msg_sender));
|
|
msgA1->tx_port = ntohs(Adestination->rtpPort.num());
|
|
msgA1->type = SENDER_MSG_CHANGE_PORT;
|
|
resp = send_message(fmod, pathA, (struct message *) msgA1);
|
|
free_response(resp);
|
|
resp = NULL;
|
|
|
|
//CHANGE DST ADDRESS
|
|
struct msg_sender *msgA2 = (struct msg_sender *) new_message(
|
|
sizeof(struct msg_sender));
|
|
strncpy(msgA2->receiver, inet_ntoa(Adestination->addr),
|
|
sizeof(msgA2->receiver) - 1);
|
|
msgA2->type = SENDER_MSG_CHANGE_RECEIVER;
|
|
|
|
resp = send_message(fmod, pathA, (struct message *) msgA2);
|
|
free_response(resp);
|
|
resp = NULL;
|
|
}
|
|
}
|
|
}
|
|
|
|
void BasicRTSPOnlySubsession::deleteStream(unsigned /* clientSessionId */,
|
|
void*& /* streamToken */) {
|
|
if (exitting) {
|
|
return;
|
|
}
|
|
|
|
if (Vdestination != NULL) {
|
|
if (avType == video || avType == av) {
|
|
char pathV[1024];
|
|
delete Vdestination;
|
|
Vdestination = NULL;
|
|
memset(pathV, 0, sizeof(pathV));
|
|
enum module_class path_sender[] = { MODULE_CLASS_SENDER,
|
|
MODULE_CLASS_NONE };
|
|
append_message_path(pathV, sizeof(pathV), path_sender);
|
|
|
|
//CHANGE DST PORT
|
|
struct msg_sender *msgV1 = (struct msg_sender *) new_message(
|
|
sizeof(struct msg_sender));
|
|
msgV1->tx_port = rtp_port;
|
|
msgV1->type = SENDER_MSG_CHANGE_PORT;
|
|
struct response *resp;
|
|
resp = send_message(fmod, pathV, (struct message *) msgV1);
|
|
free_response(resp);
|
|
|
|
//CHANGE DST ADDRESS
|
|
struct msg_sender *msgV2 = (struct msg_sender *) new_message(
|
|
sizeof(struct msg_sender));
|
|
strncpy(msgV2->receiver, "127.0.0.1", sizeof(msgV2->receiver) - 1);
|
|
msgV2->type = SENDER_MSG_CHANGE_RECEIVER;
|
|
resp = send_message(fmod, pathV, (struct message *) msgV2);
|
|
free_response(resp);
|
|
}
|
|
}
|
|
|
|
if (Adestination != NULL) {
|
|
if (avType == audio || avType == av) {
|
|
char pathA[1024];
|
|
delete Vdestination;
|
|
Adestination = NULL;
|
|
memset(pathA, 0, sizeof(pathA));
|
|
enum module_class path_sender[] = { MODULE_CLASS_AUDIO,
|
|
MODULE_CLASS_SENDER, MODULE_CLASS_NONE };
|
|
append_message_path(pathA, sizeof(pathA), path_sender);
|
|
|
|
//CHANGE DST PORT
|
|
struct msg_sender *msgA1 = (struct msg_sender *) new_message(
|
|
sizeof(struct msg_sender));
|
|
|
|
//TODO: GET AUDIO PORT SET (NOT A COMMON CASE WHEN RTSP IS ENABLED: DEFAULT -> vport + 2)
|
|
msgA1->tx_port = rtp_port_audio;
|
|
msgA1->type = SENDER_MSG_CHANGE_PORT;
|
|
struct response *resp;
|
|
resp = send_message(fmod, pathA, (struct message *) msgA1);
|
|
free_response(resp);
|
|
|
|
//CHANGE DST ADDRESS
|
|
struct msg_sender *msgA2 = (struct msg_sender *) new_message(
|
|
sizeof(struct msg_sender));
|
|
strncpy(msgA2->receiver, "127.0.0.1", sizeof(msgA2->receiver) - 1);
|
|
msgA2->type = SENDER_MSG_CHANGE_RECEIVER;
|
|
resp = send_message(fmod, pathA, (struct message *) msgA2);
|
|
free_response(resp);
|
|
}
|
|
}
|
|
}
|
|
|
|
volatile bool BasicRTSPOnlySubsession::exitting = false;
|
|
/* vi: set noexpandtab: */
|