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Comments (18)

mpromonet avatar mpromonet commented on May 18, 2024

Did you tried to add .release() to the std::unique_ptr ?
I will probably do a better fix later.

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juvasquezg avatar juvasquezg commented on May 18, 2024

Hi,

Not really, I could not compile the code.

The last error was src

The last error was, rtc :: HttpListenServer is not recognized

Thank you.

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mpromonet avatar mpromonet commented on May 18, 2024

You need to relink the .a library because I changed the Makefile.
Using "make clean && make" should work if you built webrtc with httpserver support

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juvasquezg avatar juvasquezg commented on May 18, 2024

Hi,

I built with

https://chromium.googlesource.com/external/webrtc/+log/branch-heads/56

I will test it.

Tks

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juvasquezg avatar juvasquezg commented on May 18, 2024

Hello,

Nothing :(

g++  -o src/HttpServerRequestHandler.o -c src/HttpServerRequestHandler.cpp -Wall -pthread -g -std=c++11 -Iinc -DWEBRTC_POSIX -fno-rtti -D_GLIBCXX_USE_CXX11_ABI=0 -I ../webrtc/src -I ../webrtc/src/chromium/src/third_party/jsoncpp/source/include
g++  -o src/main.o -c src/main.cpp -Wall -pthread -g -std=c++11 -Iinc -DWEBRTC_POSIX -fno-rtti -D_GLIBCXX_USE_CXX11_ABI=0 -I ../webrtc/src -I ../webrtc/src/chromium/src/third_party/jsoncpp/source/include
g++  -o src/PeerConnectionManager.o -c src/PeerConnectionManager.cpp -Wall -pthread -g -std=c++11 -Iinc -DWEBRTC_POSIX -fno-rtti -D_GLIBCXX_USE_CXX11_ABI=0 -I ../webrtc/src -I ../webrtc/src/chromium/src/third_party/jsoncpp/source/include
src/PeerConnectionManager.cpp: In member function ‘const Json::Value PeerConnectionManager::getDeviceList()’:
src/PeerConnectionManager.cpp:60:109: error: no matching function for call to ‘webrtc::VideoCaptureFactory::CreateDeviceInfo()’
  std::unique_ptr<webrtc::VideoCaptureModule::DeviceInfo> info(webrtc::VideoCaptureFactory::CreateDeviceInfo());
                                                                                                             ^
src/PeerConnectionManager.cpp:60:109: note: candidate is:
In file included from src/PeerConnectionManager.cpp:13:0:
../webrtc/src/webrtc/modules/video_capture/video_capture_factory.h:38:42: note: static webrtc::VideoCaptureModule::DeviceInfo* webrtc::VideoCaptureFactory::CreateDeviceInfo(int32_t)
   static VideoCaptureModule::DeviceInfo* CreateDeviceInfo(
                                          ^
../webrtc/src/webrtc/modules/video_capture/video_capture_factory.h:38:42: note:   candidate expects 1 argument, 0 provided
src/PeerConnectionManager.cpp: In member function ‘cricket::VideoCapturer* PeerConnectionManager::OpenVideoCaptureDevice(const string&)’:
src/PeerConnectionManager.cpp:496:110: error: no matching function for call to ‘webrtc::VideoCaptureFactory::CreateDeviceInfo()’
   std::unique_ptr<webrtc::VideoCaptureModule::DeviceInfo> info(webrtc::VideoCaptureFactory::CreateDeviceInfo());
                                                                                                              ^
src/PeerConnectionManager.cpp:496:110: note: candidate is:
In file included from src/PeerConnectionManager.cpp:13:0:
../webrtc/src/webrtc/modules/video_capture/video_capture_factory.h:38:42: note: static webrtc::VideoCaptureModule::DeviceInfo* webrtc::VideoCaptureFactory::CreateDeviceInfo(int32_t)
   static VideoCaptureModule::DeviceInfo* CreateDeviceInfo(
                                          ^
../webrtc/src/webrtc/modules/video_capture/video_capture_factory.h:38:42: note:   candidate expects 1 argument, 0 provided
src/PeerConnectionManager.cpp:510:59: error: request for member ‘release’ in ‘factory.cricket::WebRtcVideoDeviceCapturerFactory::Create((*(const cricket::Device*)(& cricket::Device((* & std::basic_string<char>(((const char*)(& name)), (*(const std::allocator<char>*)(& std::allocator<char>())))), 0))))’, which is of pointer type ‘cricket::VideoCapturer*’ (maybe you meant to use ‘->’ ?)
       capturer = factory.Create(cricket::Device(name, 0)).release();
                                                           ^
make: *** [src/PeerConnectionManager.o] Error 1

with branch-heads/51

and

with branch-heads/56

Thank you!!

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juvasquezg avatar juvasquezg commented on May 18, 2024

If you run

cd webrtc/src
git branch
git log

What is your result?

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mpromonet avatar mpromonet commented on May 18, 2024

The last webrtc release I built is from mester branch commit 1e1c1ff7b917b6c6403d078cfcf7b7b04a7836df

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juvasquezg avatar juvasquezg commented on May 18, 2024

Hello,

Great, with master branch commit 1e1c1ff7b917b6c6403d078cfcf7b7b04a7836df

But I use Ubuntu Server 14.04 LTS:

  1. I use Vagrant
  2. Provider is VirtualBox

when I run, Out:

Logger level:3
[000:000] [26425] (webrtcvoiceengine.cc:1053): webrtc:   failed to connect context, error=-1
[000:001] [26425] (audio_device_pulse_linux.cc:173): failed to initialize PulseAudio
[000:002] [26425] (audio_device_impl.cc:279): Linux PulseAudio is *not* supported => ALSA APIs will be utilized instead
[000:003] [26425] (audio_device_alsa_linux.cc:176): failed to open X display, typing detection will not work
[000:010] [26425] (webrtcvoiceengine.cc:1053): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib control.c:953:(snd_ctl_open_noupdate) Invalid CTL
[000:011] [26425] (webrtcvoiceengine.cc:1053): webrtc:      snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory
[000:012] [26425] (webrtcvoiceengine.cc:1053): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib control.c:953:(snd_ctl_open_noupdate) Invalid CTL
[000:014] [26425] (webrtcvoiceengine.cc:1053): webrtc:      snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory
[000:015] [26425] (webrtcvoiceengine.cc:1053): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib control.c:953:(snd_ctl_open_noupdate) Invalid CTL
[000:017] [26425] (webrtcvoiceengine.cc:1053): webrtc:      snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory
[000:017] [26425] (webrtcvoiceengine.cc:1053): webrtc:   InitSpeaker() failed
[000:017] [26425] (webrtcvoiceengine.cc:1053): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM
[000:018] [26425] (webrtcvoiceengine.cc:1053): webrtc:      unable to open playback device: No such file or directory (-2)
[000:019] [26425] (webrtcvoiceengine.cc:1053): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib control.c:953:(snd_ctl_open_noupdate) Invalid CTL
[000:020] [26425] (webrtcvoiceengine.cc:1053): webrtc:      snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory
[000:020] [26425] (webrtcvoiceengine.cc:1053): webrtc:   InitMicrophone() failed
[000:020] [26425] (webrtcvoiceengine.cc:1053): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM
[000:021] [26425] (webrtcvoiceengine.cc:1053): webrtc:     unable to open record device: No such file or directory
[000:022] [26425] (audio_device_generic.cc:51): BuiltInAECIsAvailable: Not supported on this platform
[000:026] [26425] (audio_device_generic.cc:61): BuiltInAGCIsAvailable: Not supported on this platform
[000:029] [26425] (audio_device_generic.cc:71): BuiltInNSIsAvailable: Not supported on this platform
[000:029] [26425] (audio_device_buffer.cc:217): Not implemented
[000:029] [26425] (webrtcvoiceengine.cc:1053): webrtc: SetRecordingChannel() unable to set the recording channel (error=10028)
[000:030] [26425] (webrtcvoiceengine.cc:1053): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib control.c:953:(snd_ctl_open_noupdate) Invalid CTL
[000:030] [26425] (webrtcvoiceengine.cc:1053): webrtc:      snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory
[000:030] [26425] (webrtcvoiceengine.cc:1053): webrtc: SetRecordingDevice() cannot access microphone (error=9004)
[000:030] [26425] (webrtcvoiceengine.cc:1053): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib control.c:953:(snd_ctl_open_noupdate) Invalid CTL
[000:030] [26425] (webrtcvoiceengine.cc:1053): webrtc:      snd_mixer_attach(_inputMixerHandle, ) error: No such file or directory
[000:030] [26425] (webrtcvoiceengine.cc:1053): webrtc:   InitMicrophone() failed
[000:030] [26425] (webrtcvoiceengine.cc:1053): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM
[000:030] [26425] (webrtcvoiceengine.cc:1053): webrtc:     unable to open record device: No such file or directory
[000:031] [26425] (webrtcvoiceengine.cc:1053): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib control.c:953:(snd_ctl_open_noupdate) Invalid CTL
[000:031] [26425] (webrtcvoiceengine.cc:1053): webrtc:      snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory
[000:031] [26425] (webrtcvoiceengine.cc:1053): webrtc: SetPlayoutDevice() cannot access speaker (error=9005)
[000:031] [26425] (webrtcvoiceengine.cc:1053): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib control.c:953:(snd_ctl_open_noupdate) Invalid CTL
[000:031] [26425] (webrtcvoiceengine.cc:1053): webrtc:      snd_mixer_attach(_outputMixerHandle, ) error: No such file or directory
[000:031] [26425] (webrtcvoiceengine.cc:1053): webrtc:   InitSpeaker() failed
[000:031] [26425] (webrtcvoiceengine.cc:1053): webrtc: GetDevicesInfo - Could not find device name or numbers
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM
[000:031] [26425] (webrtcvoiceengine.cc:1053): webrtc:      unable to open playback device: No such file or directory (-2)
HTTP Listen at 0.0.0.0:8000
HTTP Listening at 0.0.0.0:8000
STUN Listening at 127.0.0.1:3478

Thank you!!

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juvasquezg avatar juvasquezg commented on May 18, 2024

Hi,

Excuse me,

I started a new project and built it.

When run example, I get:

[011:946] [1876] (PeerConnectionManager.cpp:549): Cannot create capturer rtsp://217.17.220.110/axis-media/media.amp
[011:946] [1876] (PeerConnectionManager.cpp:597): Cannot find stream

complete stdout:

===> HTTP request path:/
filename:./html/index.html
===> HTTP request path:/favicon.ico
filename:./html/favicon.ico
===> HTTP request path:/ajax.js
filename:./html/ajax.js
===> HTTP request path:/webrtcstreamer.js
filename:./html/webrtcstreamer.js
===> HTTP request path:/getIceServers
body:
answer:{
   "iceServers" : [
      {
         "url" : "stun:stun.l.google.com:19302"
      }
   ]
}

===> HTTP request path:/getDeviceList
body:
answer:[
   "rtsp://217.17.220.110/axis-media/media.amp",
   "rtsp://85.255.175.241/h264",
   "rtsp://85.255.175.244/h264",
   "rtsp://184.72.239.149/vod/mp4:BigBuckBunny_175k.mov"
]

===> HTTP request path:/favicon.ico
filename:./html/favicon.ico
===> HTTP request path:/call
body:{"type":"offer","sdp":"v=0\r\no=- 8420262131234635730 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video\r\na=msid-semantic: WMS\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:32Xj\r\na=ice-pwd:G6NA1uvmCYbTouKq/oLwallB\r\na=fingerprint:sha-256 D3:75:46:08:C1:5F:0E:8D:E0:33:12:4C:35:61:4B:EC:F8:4E:9A:DE:AD:43:E1:74:55:CF:77:74:D5:5C:27:F3\r\na=setup:actpass\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=recvonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 isac/16000\r\na=rtpmap:104 isac/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 cn/32000\r\na=rtpmap:105 cn/16000\r\na=rtpmap:13 cn/8000\r\na=rtpmap:126 telephone-event/8000\r\nm=video 9 UDP/TLS/RTP/SAVPF 100 101 107 116 117 96 97 99 98\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:32Xj\r\na=ice-pwd:G6NA1uvmCYbTouKq/oLwallB\r\na=fingerprint:sha-256 D3:75:46:08:C1:5F:0E:8D:E0:33:12:4C:35:61:4B:EC:F8:4E:9A:DE:AD:43:E1:74:55:CF:77:74:D5:5C:27:F3\r\na=setup:actpass\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:4 urn:3gpp:video-orientation\r\na=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=recvonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:100 VP8/90000\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtpmap:101 VP9/90000\r\na=rtcp-fb:101 ccm fir\r\na=rtcp-fb:101 nack\r\na=rtcp-fb:101 nack pli\r\na=rtcp-fb:101 goog-remb\r\na=rtcp-fb:101 transport-cc\r\na=rtpmap:107 H264/90000\r\na=rtcp-fb:107 ccm fir\r\na=rtcp-fb:107 nack\r\na=rtcp-fb:107 nack pli\r\na=rtcp-fb:107 goog-remb\r\na=rtcp-fb:107 transport-cc\r\na=fmtp:107 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:116 red/90000\r\na=rtpmap:117 ulpfec/90000\r\na=rtpmap:96 rtx/90000\r\na=fmtp:96 apt=100\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=101\r\na=rtpmap:99 rtx/90000\r\na=fmtp:99 apt=107\r\na=rtpmap:98 rtx/90000\r\na=fmtp:98 apt=116\r\n"}
[011:946] [1876] (PeerConnectionManager.cpp:549): **Cannot create capturer rtsp://217.17.220.110/axis-media/media.amp**
[011:946] [1876] (PeerConnectionManager.cpp:597): **Cannot find stream**
===> HTTP request path:/addIceCandidate
body:{"candidate":"candidate:339720134 1 udp 2113937151 192.168.1.53 52552 typ host generation 0 ufrag 32Xj network-cost 50","sdpMid":"audio","sdpMLineIndex":0}
answer:1

===> HTTP request path:/addIceCandidate
body:{"candidate":"candidate:339720134 2 udp 2113937150 192.168.1.53 52554 typ host generation 0 ufrag 32Xj network-cost 50","sdpMid":"audio","sdpMLineIndex":0}
answer:1

===> HTTP request path:/addIceCandidate
body:{"candidate":"candidate:339720134 1 udp 2113937151 192.168.1.53 52556 typ host generation 0 ufrag 32Xj network-cost 50","sdpMid":"video","sdpMLineIndex":1}
answer:1

===> HTTP request path:/addIceCandidate
body:{"candidate":"candidate:339720134 2 udp 2113937150 192.168.1.53 52558 typ host generation 0 ufrag 32Xj network-cost 50","sdpMid":"video","sdpMLineIndex":1}
answer:1

===> HTTP request path:/addIceCandidate
body:{"candidate":"candidate:842163049 1 udp 1677729535 190.251.0.222 52552 typ srflx raddr 192.168.1.53 rport 52552 generation 0 ufrag 32Xj network-cost 50","sdpMid":"audio","sdpMLineIndex":0}
answer:1

===> HTTP request path:/addIceCandidate
body:{"candidate":"candidate:842163049 2 udp 1677729534 190.251.0.222 52554 typ srflx raddr 192.168.1.53 rport 52554 generation 0 ufrag 32Xj network-cost 50","sdpMid":"audio","sdpMLineIndex":0}
answer:1

===> HTTP request path:/addIceCandidate
body:{"candidate":"candidate:842163049 1 udp 1677729535 190.251.0.222 52556 typ srflx raddr 192.168.1.53 rport 52556 generation 0 ufrag 32Xj network-cost 50","sdpMid":"video","sdpMLineIndex":1}
answer:1

===> HTTP request path:/addIceCandidate
body:{"candidate":"candidate:842163049 2 udp 1677729534 190.251.0.222 52558 typ srflx raddr 192.168.1.53 rport 52558 generation 0 ufrag 32Xj network-cost 50","sdpMid":"video","sdpMLineIndex":1}
answer:1

Thank you so much.

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juvasquezg avatar juvasquezg commented on May 18, 2024

I have live555

last line to build is

g++  -o webrtc-server__Release src/HttpServerRequestHandler.o src/main.o src/PeerConnectionManager.o src/rtspvideocapturer.o libWebRTC__Release.a -pthread  -lX11 -ldl -lr
t

Include rtspvideocapturer

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saket424 avatar saket424 commented on May 18, 2024

The last webrtc release I built is from master branch commit 1e1c1ff7b917b6c6403d078cfcf7b7b04a7836df

Hi Michel,
I get the following error when I use https://chromium.googlesource.com/external/webrtc/+log/branch-heads/58 and the newest webrtc-streamer HEAD

anand@ubu1404:/webrtc-streamer$ make
g++ -o src/PeerConnectionManager.o -c src/PeerConnectionManager.cpp -Wall -pthread -g -std=c++11 -Iinc -DHAVE_LIVE555 -I live555helper/inc -I /usr/include/liveMedia -I /usr/include/groupsock -I /usr/include/UsageEnvironment -I /usr/include/BasicUsageEnvironment/ -DWEBRTC_POSIX -fno-rtti -D_GLIBCXX_USE_CXX11_ABI=0 -I ../webrtc/src -I ../webrtc/src/third_party/jsoncpp/source/include
src/PeerConnectionManager.cpp: In member function ‘cricket::VideoCapturer* PeerConnectionManager::OpenVideoCaptureDevice(const string&)’:
src/PeerConnectionManager.cpp:491:39: error: cannot allocate an object of abstract type ‘RTSPVideoCapturer’
capturer = new RTSPVideoCapturer(url);
^
In file included from src/PeerConnectionManager.cpp:20:0:
inc/rtspvideocapturer.h:26:7: note: because the following virtual functions are pure within ‘RTSPVideoCapturer’:
class RTSPVideoCapturer : public cricket::VideoCapturer, public RTSPConnection::Callback, public rtc::Thread, public webrtc::DecodedImageCallback
^
In file included from inc/rtspvideocapturer.h:17:0,
from src/PeerConnectionManager.cpp:20:
live555helper/inc/rtspconnectionclient.h:48:21: note: virtual bool RTSPConnection::Callback::onData(const char*, unsigned char*, ssize_t, timeval)
virtual bool onData(const char* id, unsigned char* buffer, ssize_t size, struct timeval presentationTime) = 0;
^
make: *** [src/PeerConnectionManager.o] Error 1
anand@ubu1404:/webrtc-streamer$

No idea why my compile output is all crossed out in this github comment section

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saket424 avatar saket424 commented on May 18, 2024

I reverted to an earlier commit of webrtc-streamer b8606e1 that was known to work with webrtc/branch-heads/57 and made the single change to src/PeerConnectionManager.cpp (capturer=factory.Create(cricket::Device(name, 0)).release()) so that I can try it with webrtc/branch-heads/58 without success

Now I get a linker error

g++ -o webrtc-server__Release src/HttpServerRequestHandler.o src/main.o src/PeerConnectionManager.o live555helper/live555helper.a libWebRTC__Release.a -pthread live555helper/live555helper.a -l:libliveMedia.a -l:libgroupsock.a -l:libUsageEnvironment.a -l:libBasicUsageEnvironment.a -llog4cpp -lX11 -ldl -lrt
src/HttpServerRequestHandler.o: In function HttpServerRequestHandler::OnRequest(rtc::HttpServer*, rtc::HttpServerTransaction*)': /home/anand/webrtc-streamer/src/HttpServerRequestHandler.cpp:152: undefined reference to rtc::HttpServer::Respond(rtc::HttpServerTransaction*)'
src/main.o: In function main': /home/anand/webrtc-streamer/src/main.cpp:79: undefined reference to rtc::HttpListenServer::HttpListenServer()'
/home/anand/webrtc-streamer/src/main.cpp:82: undefined reference to rtc::HttpListenServer::Listen(rtc::SocketAddress const&)' /home/anand/webrtc-streamer/src/main.cpp:109: undefined reference to rtc::HttpListenServer::~HttpListenServer()'
/home/anand/webrtc-streamer/src/main.cpp:109: undefined reference to `rtc::HttpListenServer::~HttpListenServer()'
collect2: error: ld returned 1 exit status
make: *** [webrtc-server__Release] Error 1

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saket424 avatar saket424 commented on May 18, 2024

They moved the gn target for httpserver.cc (HttpListenServer) to rtc_base_tests_utils in

https://chromium.googlesource.com/external/webrtc.git/+/14584623035b29ce2a7830ea38956a10c584248d%5E!/#F0

So I needed the Makefile changes also. rtsp with webrtc/branch-heads/58 now works with webrtc-streamer b8606e1 . Need to see if I can make it work with the HEAD revision of webrtc-streamer

anand@ubu1404:~/webrtc-streamer$ git diff Makefile
diff --git a/Makefile b/Makefile
index 667a8b7..a83a213 100644
--- a/Makefile
+++ b/Makefile
@@ -26,7 +26,7 @@ endif
WEBRTCLIBPATH=$(WEBRTCROOT)/src/$(GYP_GENERATOR_OUTPUT)/out/$(WEBRTCBUILD)

CFLAGS += -DWEBRTC_POSIX -fno-rtti -D_GLIBCXX_USE_CXX11_ABI=0
-CFLAGS += -I $(WEBRTCROOT)/src -I $(WEBRTCROOT)/src/chromium/src/third_party/js
+CFLAGS += -I $(WEBRTCROOT)/src -I $(WEBRTCROOT)/src/third_party/jsoncpp/source/
#detect
TESTDEBUG=$(shell nm $(wildcard $(WEBRTCLIBPATH)/obj/webrtc/media/rtc_media/vid
ifeq ($(TESTDEBUG),debug)
@@ -35,7 +35,7 @@ endif
LDFLAGS += -lX11 -ldl -lrt

WEBRTC_LIB = $(shell find $(WEBRTCLIBPATH)/obj/base -name '.o')
-WEBRTC_LIB += $(shell find $(WEBRTCLIBPATH)/obj/webrtc -name '.o' ! -path 'te
+WEBRTC_LIB += $(shell find $(WEBRTCLIBPATH)/obj/webrtc -name '.o')
WEBRTC_LIB += $(shell find $(WEBRTCLIBPATH)/obj/third_party -name '*.o')
LIBS+=libWebRTC_$(GYP_GENERATOR_OUTPUT)$(WEBRTCBUILD).a
libWebRTC$(GYP_GENERATOR_OUTPUT)_$(WEBRTCBUILD).a: $(WEBRTC_LIB)

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saket424 avatar saket424 commented on May 18, 2024

Back to webrtc-streamer HEAD as of March 11instead of b8606e1 and I get the following error when I enable live555 for rtsp with webrtc/branch-heads/58

anand@ubu1404:~/webrtc-streamer$ git diff Makefile
diff --git a/Makefile b/Makefile
index 580b584..180b29d 100644
--- a/Makefile
+++ b/Makefile
@@ -10,7 +10,7 @@ all: $(TARGET)

live555helper

ifneq ($(wildcard $(SYSROOT)/usr/include/liveMedia/liveMedia.hh),)
-ifneq ($(wildcard $(SYSROOT)/usr/lib/libliveMedia.a),)
+# ifneq ($(wildcard $(SYSROOT)/usr/lib/libliveMedia.a),)
LIBS+=live555helper/live555helper.a
live555helper/live555helper.a:
make -C live555helper
@@ -21,14 +21,14 @@ CFLAGS += -I $(SYSROOT)/usr/include/liveMedia -I $(SYSROOT)/usr/include/groupso

LDFLAGS += live555helper/live555helper.a
LDFLAGS += -l:libliveMedia.a -l:libgroupsock.a -l:libUsageEnvironment.a -l:libBasicUsageEnvironment
-endif
+# endif
endif

webrtc

WEBRTCLIBPATH=$(WEBRTCROOT)/src/$(GYP_GENERATOR_OUTPUT)/out/$(WEBRTCBUILD)

CFLAGS += -DWEBRTC_POSIX -fno-rtti -D_GLIBCXX_USE_CXX11_ABI=0
-CFLAGS += -I $(WEBRTCROOT)/src -I $(WEBRTCROOT)/src/chromium/src/third_party/jsoncpp/source/include
+CFLAGS += -I $(WEBRTCROOT)/src -I $(WEBRTCROOT)/src/third_party/jsoncpp/source/include
#detect
TESTDEBUG=$(shell nm $(wildcard $(WEBRTCLIBPATH)/obj/webrtc/media/rtc_media/videocapturer.o $(WEBRT
ifeq ($(TESTDEBUG),debug)
anand@ubu1404:/webrtc-streamer$ make
g++ -o src/PeerConnectionManager.o -c src/PeerConnectionManager.cpp -Wall -pthread -g -std=c++11 -Iinc -DHAVE_LIVE555 -I live555helper/inc -I /usr/include/liveMedia -I /usr/include/groupsock -I /usr/include/UsageEnvironment -I /usr/include/BasicUsageEnvironment/ -DWEBRTC_POSIX -fno-rtti -D_GLIBCXX_USE_CXX11_ABI=0 -I ../webrtc/src -I ../webrtc/src/third_party/jsoncpp/source/include
src/PeerConnectionManager.cpp: In member function ‘cricket::VideoCapturer* PeerConnectionManager::OpenVideoCaptureDevice(const string&)’:
src/PeerConnectionManager.cpp:491:39: error: cannot allocate an object of abstract type ‘RTSPVideoCapturer’
capturer = new RTSPVideoCapturer(url);
^
In file included from src/PeerConnectionManager.cpp:20:0:
inc/rtspvideocapturer.h:26:7: note: because the following virtual functions are pure within ‘RTSPVideoCapturer’:
class RTSPVideoCapturer : public cricket::VideoCapturer, public RTSPConnection::Callback, public rtc::Thread, public webrtc::DecodedImageCallback
^
In file included from inc/rtspvideocapturer.h:17:0,
from src/PeerConnectionManager.cpp:20:
live555helper/inc/rtspconnectionclient.h:48:21: note: virtual bool RTSPConnection::Callback::onData(const char*, unsigned char*, ssize_t, timeval)
virtual bool onData(const char* id, unsigned char* buffer, ssize_t size, struct timeval presentationTime) = 0;
^
make: *** [src/PeerConnectionManager.o] Error 1
anand@ubu1404:/webrtc-streamer$

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juvasquezg avatar juvasquezg commented on May 18, 2024

Hi @saket424 How can I enable live555. I installed but I don't know. I work with master branch 1e1c1ff7b917b6c6403d078cfcf7b7b04a7836df

Thank you.

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juvasquezg avatar juvasquezg commented on May 18, 2024

Hi,

Excuse me,

I ran

sudo apt-get install liblivemedia-dev
sudo apt-get install liblog4cpp5-dev

And work with master branch 1e1c1ff7b917b6c6403d078cfcf7b7b04a7836df

Thank you so much.

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mpromonet avatar mpromonet commented on May 18, 2024

Hi,
Anyway as webrtc remove its httpserver, I will replace it with another one.
Best Regards,
Michel

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mpromonet avatar mpromonet commented on May 18, 2024

Hi,
This commit cf84eae replace webrtc removed httpserver with civetweb http server.
Now it should build with the commit you tried before.
Best Regards,
Michel.

from webrtc-streamer.

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