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| diff --git a/plugins/obs-webrtc/whip-output.cpp b/plugins/obs-webrtc/whip-output.cpp | |
| index e83cc03b8..0e3693992 100644 | |
| --- a/plugins/obs-webrtc/whip-output.cpp | |
| +++ b/plugins/obs-webrtc/whip-output.cpp | |
| @@ -60,6 +60,10 @@ bool WHIPOutput::Start() | |
| return false; | |
| } | |
| + obs_data_t *video_settings = obs_encoder_get_settings(encoder); | |
| + video_bitrate = (int)obs_data_get_int(video_settings, "bitrate"); |
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| <html> | |
| <head> | |
| <title>whep-static</title> | |
| </head> | |
| <body> | |
| <h3> Video </h3> | |
| <video id="videoPlayer" autoplay muted controls style="width: 500"> </video> |
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| 2024/02/16 02:25:23 Loading `.env.production` | |
| [0;33mNETWORK_TEST_ON_START is enabled. If the test fails Broadcast Box will exit. | |
| See the README for how to debug or disable NETWORK_TEST_ON_START[0m | |
| 2024/02/16 02:25:23 Running HTTP Server at `:8080` | |
| [0;32mNetwork Test passed. | |
| Have fun using Broadcast Box.[0m | |
| 2024/02/16 05:24:25 Authorization was not set | |
| 2024/02/16 05:25:38 Authorization was not set | |
| 2024/02/16 14:22:00 Authorization was not set |
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| class Ffmpeg < Formula | |
| desc "Play, record, convert, and stream audio and video" | |
| homepage "https://ffmpeg.org/" | |
| url "https://ffmpeg.org/releases/ffmpeg-6.0.tar.xz" | |
| sha256 "57be87c22d9b49c112b6d24bc67d42508660e6b718b3db89c44e47e289137082" | |
| # None of these parts are used by default, you have to explicitly pass `--enable-gpl` | |
| # to configure to activate them. In this case, FFmpeg's license changes to GPL v2+. | |
| license "GPL-2.0-or-later" | |
| head "https://github.com/FFmpeg/FFmpeg.git", branch: "master" |
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| Pion has lots of opportunities to get involved. We have lots of self contained projects that would be a lot of fun. Even if you don’t finish the project that is fine. The great thing about Open Source is someone can pick up where you left off. | |
| It’s a great way to learn deep WebRTC knowledge. It is also a great pathway into a career in WebRTC. Multiple Pion contributors now work at companies that use the code they wrote. | |
| ————————————— | |
| Network Simulator - Pion needs an easy to use network simulator. Something that developers can use in a few minutes and doesn’t require a Docker container or a long setup process. It can run in memory, but also provides virtual interfaces to test non-Go programs. | |
| The simulator will model real world networks. Nog just setting a static packet loss or bandwidth cap. It can dynamically adjust all the attributes of the link. |
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| -DCMAKE_CXX_FLAGS="-static-libgcc -static-libstdc++ -w -pipe -fno-semantic-interposition -static" | |
| -DCMAKE_C_FLAGS="-static-libgcc -w -pipe -fno-semantic-interposition -static" | |
| -DCMAKE_SHARED_LINKER_FLAGS="-static-libgcc -static-libstdc++ -L${target_config[output_dir]}/lib -Wl,--exclude-libs,ALL -static" |
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| a=rtpmap:102 H264/90000 | |
| a=rtcp-fb:102 goog-remb | |
| a=rtcp-fb:102 transport-cc | |
| a=rtcp-fb:102 ccm fir | |
| a=rtcp-fb:102 nack | |
| a=rtcp-fb:102 nack pli | |
| a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f | |
| a=rtpmap:122 rtx/90000 | |
| a=fmtp:122 apt=102 | |
| a=rtpmap:127 H264/90000 |
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| peerConnection, err := webrtc.NewPeerConnection(webrtc.Configuration{}) | |
| if err != nil { | |
| panic(err) | |
| } | |
| // Allow us to receive 1 audio track, and 1 video track | |
| if _, err = peerConnection.AddTransceiverFromKind(webrtc.RTPCodecTypeAudio, webrtc.RTPTransceiverInit{Direction: webrtc.RTPTransceiverDirectionRecvonly}); err != nil { | |
| panic(err) | |
| } else if _, err = peerConnection.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo, webrtc.RTPTransceiverInit{Direction: webrtc.RTPTransceiverDirectionRecvonly}); err != nil { | |
| panic(err) |
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| const H264Codecs = RTCRtpReceiver.getCapabilities('video').codecs.sort((a, b) => { | |
| if (a.mimeType.includes('H264') && b.mimeType.includes('H264')) { | |
| return 0 | |
| } else if (a.mimeType.includes('H264')) { | |
| return -1 | |
| } else { | |
| return 1 | |
| } | |
| }) |
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| module set-receiver-codec | |
| go 1.18 | |
| require github.com/pion/webrtc/v3 v3.1.41 | |
| require ( | |
| github.com/google/uuid v1.3.0 // indirect | |
| github.com/pion/datachannel v1.5.2 // indirect | |
| github.com/pion/dtls/v2 v2.1.5 // indirect |
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