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UltraGrid/src/rtsp/BasicRTSPOnlySubsession.cpp
Martin Pulec 41b0388624 rtsp server: fixed audio props 
Do not deduce audio properties from command-line parameters but from
the actual format that is being to sent.

Until now, the properties like ch_count and sample rate was deduced from
command-line parameters (ch_count from `-a channels=` and sample rate from
`-A xxx:sample_rate`). However, this is a bit fragile - the source may not
respect the `-a` parameters and also the `-A sample_rate` doesn't need
to be set, like in the following case:
```
uv -x rtsp -A A-law -a sample_r=8000 -s testcard
```

Here the capture sample_rate is set to 8000 but not parsed from parameters
(it must have been in `-A`). So SDP parameeters for RTSP was set
incorrectly (48000).
2024-08-09 09:22:59 +02:00

350 lines
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/*
* 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 <cassert>
#include <BasicUsageEnvironment.hh>
#include <RTSPServer.hh>
#include <GroupsockHelper.hh>
#include "audio/codec.h" // get_name_to_audio_codec
#include "debug.h" // for MSG
#include "messaging.h"
#include "module.h" // for module_class, append_message...
#include "utils/macros.h"
#include "utils/net.h"
#include "utils/sdp.h"
#include "video_codec.h" // for get_codec_name
#define MOD_NAME "[RTSP] "
BasicRTSPOnlySubsession*
BasicRTSPOnlySubsession::createNew(UsageEnvironment& env,
Boolean reuseFirstSource, rtsp_types_t avType, int rtpPort,
struct rtsp_server_parameters params) {
return new BasicRTSPOnlySubsession(env, reuseFirstSource, avType,
rtpPort, params);
}
BasicRTSPOnlySubsession::BasicRTSPOnlySubsession(UsageEnvironment& env,
Boolean reuseFirstSource, rtsp_types_t avType, int rtpPort,
struct rtsp_server_parameters params) :
ServerMediaSubsession(env), fReuseFirstSource(reuseFirstSource),
fLastStreamToken(nullptr), rtsp_params(params)
{
assert(avType == rtsp_type_audio || avType == rtsp_type_video);
Vdestination = NULL;
Adestination = NULL;
gethostname(fCNAME, sizeof fCNAME);
this->avType = avType;
this->rtpPort = rtpPort;
fCNAME[sizeof fCNAME - 1] = '\0';
// print (preliminary) SDP
setSDPLines(AF_UNSPEC);
delete[] fSDPLines;
fSDPLines = nullptr;
}
BasicRTSPOnlySubsession::~BasicRTSPOnlySubsession() {
delete[] fSDPLines;
delete Adestination;
delete Vdestination;
}
const static struct media_spec {
unsigned estBitrate;
const char *mname;
} media_params[] = {
{ 0, nullptr }, // none
{ 384, "audio" },
{ 5000, "video" },
};
static_assert(rtsp_type_audio == 1); // ensure the above mapping is correct
static_assert(rtsp_type_video == 2);
char const* BasicRTSPOnlySubsession::sdpLines(int addressFamily) {
if (fSDPLines == NULL) {
setSDPLines(addressFamily);
}
return fSDPLines;
}
void BasicRTSPOnlySubsession::setSDPLines(int addressFamily) {
//TODO: should be more dynamic
const char *ip_ver_list_addr = nullptr;
const char *control = trackId();
switch (addressFamily) {
case AF_INET:
ip_ver_list_addr = "4 0.0.0.0";
break;
case AF_INET6:
ip_ver_list_addr = "6 ::";
break;
case AF_UNSPEC:
ip_ver_list_addr = "<VER> <TO_BE_FILLED>";
control = "<CONTROL>"; // is null here
break;
default:
abort();
}
const struct media_spec *mspec = &media_params[avType];
char rtpmapLine[STR_LEN];
int rtpPayloadType = avType == rtsp_type_audio
? get_audio_rtp_pt_rtpmap(
rtsp_params.adesc.codec, rtsp_params.adesc.sample_rate,
rtsp_params.adesc.ch_count, rtpmapLine)
: get_video_rtp_pt_rtpmap(rtsp_params.video_codec, rtpmapLine);
if (rtpPayloadType < 0) {
MSG(ERROR, "Unsupported %s codec %s!\n", mspec[avType].mname,
avType == rtsp_type_audio
? get_name_to_audio_codec(rtsp_params.adesc.codec)
: get_codec_name(rtsp_params.video_codec));
}
//char const* auxSDPLine = "";
char const *const sdpFmt = "m=%s %u RTP/AVP %d\r\n"
"c=IN IP%s\r\n"
"b=AS:%u\r\n"
"a=rtcp:%d\r\n"
"%s"
"a=control:%s\r\n";
unsigned sdpFmtSize = strlen(sdpFmt) + strlen(mspec->mname) +
5 /* max short len */
+ 3 /* max char len */
+ strlen(ip_ver_list_addr) + 20 /* max int len */
+ strlen(rtpmapLine) + strlen(control);
char *sdpLines = new char[sdpFmtSize];
snprintf(sdpLines, sdpFmtSize, sdpFmt, mspec->mname, // m= <media>
rtpPort, // fPortNumForSDP, // m= <port>
rtpPayloadType, // m= <fmt list>
ip_ver_list_addr, // c= address
mspec->estBitrate, // b=AS:<bandwidth>
rtpPort + 1,
rtpmapLine, // a=rtpmap:... (if present)
control); // a=control:<track-id>
fSDPLines = sdpLines;
MSG(VERBOSE, "SDP%s:\n%s\n",
addressFamily == AF_UNSPEC ? " (preliminary)" : "", fSDPLines);
}
void BasicRTSPOnlySubsession::getStreamParameters(unsigned /* clientSessionId */,
struct sockaddr_storage const &clientAddress, Port const& clientRTPPort,
Port const& clientRTCPPort, int /* tcpSocketNum */,
unsigned char /* rtpChannelId */, unsigned char /* rtcpChannelId */,
TLSState * /* tlsState */,
struct sockaddr_storage& /*destinationAddress*/, uint8_t& /*destinationTTL*/,
Boolean& /* isMulticast */, Port& serverRTPPort, Port& serverRTCPPort,
void*& /* streamToken */) {
if (avType == rtsp_type_video) {
Port rtp(rtsp_params.rtp_port_video);
serverRTPPort = rtp;
Port rtcp(rtsp_params.rtp_port_video + 1);
serverRTCPPort = rtcp;
delete Vdestination;
Vdestination = new Destinations(clientAddress, clientRTPPort,
clientRTCPPort);
}
if (avType == rtsp_type_audio) {
Port rtp(rtsp_params.rtp_port_audio);
serverRTPPort = rtp;
Port rtcp(rtsp_params.rtp_port_audio + 1);
serverRTCPPort = rtcp;
delete Adestination;
Adestination = new Destinations(clientAddress, 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 == rtsp_type_video) {
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(rtsp_params.parent, pathV, (struct message *) msgV1);
free_response(resp);
//CHANGE DST ADDRESS
struct msg_sender *msgV2 = (struct msg_sender *) new_message(
sizeof(struct msg_sender));
char host[IN6_MAX_ASCII_LEN + 1];
const int ret =
getnameinfo((struct sockaddr *) &Vdestination->addr,
sizeof Vdestination->addr, host,
sizeof host, nullptr, 0, NI_NUMERICHOST);
assert(ret == 0);
strncpy(msgV2->receiver, host,
sizeof(msgV2->receiver) - 1);
msgV2->type = SENDER_MSG_CHANGE_RECEIVER;
resp = send_message(rtsp_params.parent, pathV, (struct message *) msgV2);
free_response(resp);
}
}
if (Adestination != NULL) {
if (avType == rtsp_type_audio) {
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(rtsp_params.parent, 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));
char host[IN6_MAX_ASCII_LEN + 1];
const int ret =
getnameinfo((struct sockaddr *) &Adestination->addr,
sizeof Adestination->addr, host,
sizeof host, nullptr, 0, NI_NUMERICHOST);
assert(ret == 0);
strncpy(msgA2->receiver, host,
sizeof(msgA2->receiver) - 1);
msgA2->type = SENDER_MSG_CHANGE_RECEIVER;
resp = send_message(rtsp_params.parent, pathA, (struct message *) msgA2);
free_response(resp);
resp = NULL;
}
}
}
void BasicRTSPOnlySubsession::deleteStream(unsigned /* clientSessionId */,
void*& /* streamToken */) {
if (Vdestination != NULL) {
if (avType == rtsp_type_video) {
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 = rtsp_params.rtp_port_video;
msgV1->type = SENDER_MSG_CHANGE_PORT;
struct response *resp;
resp = send_message(rtsp_params.parent, 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(rtsp_params.parent, pathV, (struct message *) msgV2);
free_response(resp);
}
}
if (Adestination != NULL) {
if (avType == rtsp_type_audio) {
char pathA[1024];
delete Adestination;
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 = rtsp_params.rtp_port_audio;
msgA1->type = SENDER_MSG_CHANGE_PORT;
struct response *resp;
resp = send_message(rtsp_params.parent, 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(rtsp_params.parent, pathA, (struct message *) msgA2);
free_response(resp);
}
}
}
/* vi: set noexpandtab: */
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