openpilot is an open source driver assistance system. openpilot performs the functions of Automated Lane Centering and Adaptive Cruise Control for over 200 supported car makes and models.
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import { getXY } from "./controls.js";
import { pingPoints, batteryPoints, chartPing, chartBattery } from "./plots.js";
webrtcd: webrtc streaming server (audio/video/cereal) (#30186) * WebRTCClient and WebRTCServer abstractions * webrtc client implementation * Interactive test scripts * Send localDescriptions as offer/asnwer, as they are different * Tracks need to be added after setting remote description for multi-cam streaming to work * Remove WebRTCStreamingMetadata * Wait for tracks * Move stuff to separate files, rename some things * Refactor everything, create WebRTCStreamBuilder for both offer and answers * ta flight done time to grind * wait for incoming tracks and channels * Dummy track and frame reader track. Fix timing. * dt based on camera type * first trial of the new api * Fix audio track * methods for checking for incoming tracks * Web migration part 2 * Fixes for stream api * use rtc description for web.py * experimental cereal proxy * remove old code from bodyav * fix is_started * serialize session description * fix audio * messaging channel wrapper * fix audiotrack * h264 codec preference * Add codec preference to tracks * override sdp codecs * add logging * Move cli stuff to separate file * slight cleanup * Fix audio track * create codec_mime inside force_codec function * fix incoming media estimation * move builders to __init__ * stream updates following builders * Update example script * web.py support for new builder * web speaker fixes * StreamingMediaInfo API * Move things around * should_add_data_channel rename * is_connected_and_ready * fix linter errors * make cli executable * remove dumb comments * logging support * fix parse_info_from_offer * improve type annotations * satisfy linters * Support for waiting for disconnection * Split device tracks into video/audio files. Move audio speaker to audio.py * default dt for dummy video track * Fix cli * new speaker fixes * Remove almost all functionality from web.py * webrtcd * continue refactoring web.py * after handling joystick reset in controlsd with #30409, controls are not necessary anymore * ping endpoint * Update js files to at least support what worked previously * Fixes after some tests on the body * Streaming fixes * Remove the use of WebRTCStreamBuilder. Subclass use is now required * Add todo * delete all streams on shutdown * Replace lastPing with lastChannelMessageTime * Update ping text only if rtc is still on * That should affect the chart too * Fix paths in web * use protocol in SSLContext * remove warnings since aiortc is not used directly anymore * check if task is done in stop * remove channel handler wrapper, since theres only one channel * Move things around * Moved webrtc abstractions to separate repository * Moved webrtcd to tools/webrtc * Update imports * Add bodyrtc as dependency * Add webrtcd to process_config * Remove usage of DummyVideoStreamTrack * Add main to webrtcd * Move webrtcd to system * Fix imports * Move cereal proxy logic outside of runner * Incoming proxy abstractions * Add some tests * Make it executable * Fix process config * Fix imports * Additional tests. Add tests to pyproject.toml * Update poetry lock * New line * Bump aiortc to 1.6.0 * Added teleoprtc_repo as submodule, and linked its source dir * Add init file to webrtc module * Handle aiortc warnings * Ignore deprecation warnings * Ignore resource warning too * Ignore the warnings * find free port for test_webrtcd * Start process inside the test case * random sleep test * test 2 * Test endpoint function instead * Update comment * Add system/webrtc to release * default arguments for body fields * Add teleoprtc to release * Bump teleoprtc * Exclude teleoprtc from static analysis * Use separate event loop for stream session tests old-commit-hash: f058b5d64eed31878533a85f25eb53386ff9e330
1 year ago
export let controlCommandInterval = null;
export let latencyInterval = null;
export let lastChannelMessageTime = null;
export function offerRtcRequest(sdp, type) {
return fetch('/offer', {
body: JSON.stringify({sdp: sdp, type: type}),
headers: {'Content-Type': 'application/json'},
method: 'POST'
});
}
export function playSoundRequest(sound) {
return fetch('/sound', {
body: JSON.stringify({sound}),
headers: {'Content-Type': 'application/json'},
method: 'POST'
});
}
export function pingHeadRequest() {
return fetch('/', {
method: 'HEAD'
});
}
export function createPeerConnection(pc) {
var config = {
sdpSemantics: 'unified-plan'
};
pc = new RTCPeerConnection(config);
// connect audio / video
pc.addEventListener('track', function(evt) {
console.log("Adding Tracks!")
if (evt.track.kind == 'video')
document.getElementById('video').srcObject = evt.streams[0];
else
document.getElementById('audio').srcObject = evt.streams[0];
});
return pc;
}
export function negotiate(pc) {
return pc.createOffer({offerToReceiveAudio:true, offerToReceiveVideo:true}).then(function(offer) {
return pc.setLocalDescription(offer);
}).then(function() {
return new Promise(function(resolve) {
if (pc.iceGatheringState === 'complete') {
resolve();
}
else {
function checkState() {
if (pc.iceGatheringState === 'complete') {
pc.removeEventListener('icegatheringstatechange', checkState);
resolve();
}
}
pc.addEventListener('icegatheringstatechange', checkState);
}
});
}).then(function() {
var offer = pc.localDescription;
webrtcd: webrtc streaming server (audio/video/cereal) (#30186) * WebRTCClient and WebRTCServer abstractions * webrtc client implementation * Interactive test scripts * Send localDescriptions as offer/asnwer, as they are different * Tracks need to be added after setting remote description for multi-cam streaming to work * Remove WebRTCStreamingMetadata * Wait for tracks * Move stuff to separate files, rename some things * Refactor everything, create WebRTCStreamBuilder for both offer and answers * ta flight done time to grind * wait for incoming tracks and channels * Dummy track and frame reader track. Fix timing. * dt based on camera type * first trial of the new api * Fix audio track * methods for checking for incoming tracks * Web migration part 2 * Fixes for stream api * use rtc description for web.py * experimental cereal proxy * remove old code from bodyav * fix is_started * serialize session description * fix audio * messaging channel wrapper * fix audiotrack * h264 codec preference * Add codec preference to tracks * override sdp codecs * add logging * Move cli stuff to separate file * slight cleanup * Fix audio track * create codec_mime inside force_codec function * fix incoming media estimation * move builders to __init__ * stream updates following builders * Update example script * web.py support for new builder * web speaker fixes * StreamingMediaInfo API * Move things around * should_add_data_channel rename * is_connected_and_ready * fix linter errors * make cli executable * remove dumb comments * logging support * fix parse_info_from_offer * improve type annotations * satisfy linters * Support for waiting for disconnection * Split device tracks into video/audio files. Move audio speaker to audio.py * default dt for dummy video track * Fix cli * new speaker fixes * Remove almost all functionality from web.py * webrtcd * continue refactoring web.py * after handling joystick reset in controlsd with #30409, controls are not necessary anymore * ping endpoint * Update js files to at least support what worked previously * Fixes after some tests on the body * Streaming fixes * Remove the use of WebRTCStreamBuilder. Subclass use is now required * Add todo * delete all streams on shutdown * Replace lastPing with lastChannelMessageTime * Update ping text only if rtc is still on * That should affect the chart too * Fix paths in web * use protocol in SSLContext * remove warnings since aiortc is not used directly anymore * check if task is done in stop * remove channel handler wrapper, since theres only one channel * Move things around * Moved webrtc abstractions to separate repository * Moved webrtcd to tools/webrtc * Update imports * Add bodyrtc as dependency * Add webrtcd to process_config * Remove usage of DummyVideoStreamTrack * Add main to webrtcd * Move webrtcd to system * Fix imports * Move cereal proxy logic outside of runner * Incoming proxy abstractions * Add some tests * Make it executable * Fix process config * Fix imports * Additional tests. Add tests to pyproject.toml * Update poetry lock * New line * Bump aiortc to 1.6.0 * Added teleoprtc_repo as submodule, and linked its source dir * Add init file to webrtc module * Handle aiortc warnings * Ignore deprecation warnings * Ignore resource warning too * Ignore the warnings * find free port for test_webrtcd * Start process inside the test case * random sleep test * test 2 * Test endpoint function instead * Update comment * Add system/webrtc to release * default arguments for body fields * Add teleoprtc to release * Bump teleoprtc * Exclude teleoprtc from static analysis * Use separate event loop for stream session tests old-commit-hash: f058b5d64eed31878533a85f25eb53386ff9e330
1 year ago
return offerRtcRequest(offer.sdp, offer.type);
}).then(function(response) {
console.log(response);
return response.json();
}).then(function(answer) {
return pc.setRemoteDescription(answer);
}).catch(function(e) {
alert(e);
});
}
function isMobile() {
let check = false;
(function(a){if(/(android|bb\d+|meego).+mobile|avantgo|bada\/|blackberry|blazer|compal|elaine|fennec|hiptop|iemobile|ip(hone|od)|iris|kindle|lge |maemo|midp|mmp|mobile.+firefox|netfront|opera m(ob|in)i|palm( os)?|phone|p(ixi|re)\/|plucker|pocket|psp|series(4|6)0|symbian|treo|up\.(browser|link)|vodafone|wap|windows ce|xda|xiino/i.test(a)||/1207|6310|6590|3gso|4thp|50[1-6]i|770s|802s|a wa|abac|ac(er|oo|s\-)|ai(ko|rn)|al(av|ca|co)|amoi|an(ex|ny|yw)|aptu|ar(ch|go)|as(te|us)|attw|au(di|\-m|r |s )|avan|be(ck|ll|nq)|bi(lb|rd)|bl(ac|az)|br(e|v)w|bumb|bw\-(n|u)|c55\/|capi|ccwa|cdm\-|cell|chtm|cldc|cmd\-|co(mp|nd)|craw|da(it|ll|ng)|dbte|dc\-s|devi|dica|dmob|do(c|p)o|ds(12|\-d)|el(49|ai)|em(l2|ul)|er(ic|k0)|esl8|ez([4-7]0|os|wa|ze)|fetc|fly(\-|_)|g1 u|g560|gene|gf\-5|g\-mo|go(\.w|od)|gr(ad|un)|haie|hcit|hd\-(m|p|t)|hei\-|hi(pt|ta)|hp( i|ip)|hs\-c|ht(c(\-| |_|a|g|p|s|t)|tp)|hu(aw|tc)|i\-(20|go|ma)|i230|iac( |\-|\/)|ibro|idea|ig01|ikom|im1k|inno|ipaq|iris|ja(t|v)a|jbro|jemu|jigs|kddi|keji|kgt( |\/)|klon|kpt |kwc\-|kyo(c|k)|le(no|xi)|lg( g|\/(k|l|u)|50|54|\-[a-w])|libw|lynx|m1\-w|m3ga|m50\/|ma(te|ui|xo)|mc(01|21|ca)|m\-cr|me(rc|ri)|mi(o8|oa|ts)|mmef|mo(01|02|bi|de|do|t(\-| |o|v)|zz)|mt(50|p1|v )|mwbp|mywa|n10[0-2]|n20[2-3]|n30(0|2)|n50(0|2|5)|n7(0(0|1)|10)|ne((c|m)\-|on|tf|wf|wg|wt)|nok(6|i)|nzph|o2im|op(ti|wv)|oran|owg1|p800|pan(a|d|t)|pdxg|pg(13|\-([1-8]|c))|phil|pire|pl(ay|uc)|pn\-2|po(ck|rt|se)|prox|psio|pt\-g|qa\-a|qc(07|12|21|32|60|\-[2-7]|i\-)|qtek|r380|r600|raks|rim9|ro(ve|zo)|s55\/|sa(ge|ma|mm|ms|ny|va)|sc(01|h\-|oo|p\-)|sdk\/|se(c(\-|0|1)|47|mc|nd|ri)|sgh\-|shar|sie(\-|m)|sk\-0|sl(45|id)|sm(al|ar|b3|it|t5)|so(ft|ny)|sp(01|h\-|v\-|v )|sy(01|mb)|t2(18|50)|t6(00|10|18)|ta(gt|lk)|tcl\-|tdg\-|tel(i|m)|tim\-|t\-mo|to(pl|sh)|ts(70|m\-|m3|m5)|tx\-9|up(\.b|g1|si)|utst|v400|v750|veri|vi(rg|te)|vk(40|5[0-3]|\-v)|vm40|voda|vulc|vx(52|53|60|61|70|80|81|83|85|98)|w3c(\-| )|webc|whit|wi(g |nc|nw)|wmlb|wonu|x700|yas\-|your|zeto|zte\-/i.test(a.substr(0,4))) check = true;})(navigator.userAgent||navigator.vendor||window.opera);
return check;
};
export const constraints = {
audio: {
autoGainControl: false,
sampleRate: 48000,
sampleSize: 16,
echoCancellation: true,
noiseSuppression: true,
channelCount: 1
},
video: isMobile()
};
export function start(pc, dc) {
pc = createPeerConnection(pc);
// add audio track
navigator.mediaDevices.enumerateDevices()
.then(function(devices) {
const hasAudioInput = devices.find((device) => device.kind === "audioinput");
var modifiedConstraints = {};
modifiedConstraints.video = constraints.video;
modifiedConstraints.audio = hasAudioInput ? constraints.audio : false;
return Promise.resolve(modifiedConstraints);
})
.then(function(constraints) {
if (constraints.audio || constraints.video) {
return navigator.mediaDevices.getUserMedia(constraints);
} else{
return Promise.resolve(null);
}
})
.then(function(stream) {
if (stream) {
stream.getTracks().forEach(function(track) {
pc.addTrack(track, stream);
});
}
return negotiate(pc);
})
.catch(function(err) {
alert('Could not acquire media: ' + err);
});
var parameters = {"ordered": true};
dc = pc.createDataChannel('data', parameters);
dc.onclose = function() {
webrtcd: webrtc streaming server (audio/video/cereal) (#30186) * WebRTCClient and WebRTCServer abstractions * webrtc client implementation * Interactive test scripts * Send localDescriptions as offer/asnwer, as they are different * Tracks need to be added after setting remote description for multi-cam streaming to work * Remove WebRTCStreamingMetadata * Wait for tracks * Move stuff to separate files, rename some things * Refactor everything, create WebRTCStreamBuilder for both offer and answers * ta flight done time to grind * wait for incoming tracks and channels * Dummy track and frame reader track. Fix timing. * dt based on camera type * first trial of the new api * Fix audio track * methods for checking for incoming tracks * Web migration part 2 * Fixes for stream api * use rtc description for web.py * experimental cereal proxy * remove old code from bodyav * fix is_started * serialize session description * fix audio * messaging channel wrapper * fix audiotrack * h264 codec preference * Add codec preference to tracks * override sdp codecs * add logging * Move cli stuff to separate file * slight cleanup * Fix audio track * create codec_mime inside force_codec function * fix incoming media estimation * move builders to __init__ * stream updates following builders * Update example script * web.py support for new builder * web speaker fixes * StreamingMediaInfo API * Move things around * should_add_data_channel rename * is_connected_and_ready * fix linter errors * make cli executable * remove dumb comments * logging support * fix parse_info_from_offer * improve type annotations * satisfy linters * Support for waiting for disconnection * Split device tracks into video/audio files. Move audio speaker to audio.py * default dt for dummy video track * Fix cli * new speaker fixes * Remove almost all functionality from web.py * webrtcd * continue refactoring web.py * after handling joystick reset in controlsd with #30409, controls are not necessary anymore * ping endpoint * Update js files to at least support what worked previously * Fixes after some tests on the body * Streaming fixes * Remove the use of WebRTCStreamBuilder. Subclass use is now required * Add todo * delete all streams on shutdown * Replace lastPing with lastChannelMessageTime * Update ping text only if rtc is still on * That should affect the chart too * Fix paths in web * use protocol in SSLContext * remove warnings since aiortc is not used directly anymore * check if task is done in stop * remove channel handler wrapper, since theres only one channel * Move things around * Moved webrtc abstractions to separate repository * Moved webrtcd to tools/webrtc * Update imports * Add bodyrtc as dependency * Add webrtcd to process_config * Remove usage of DummyVideoStreamTrack * Add main to webrtcd * Move webrtcd to system * Fix imports * Move cereal proxy logic outside of runner * Incoming proxy abstractions * Add some tests * Make it executable * Fix process config * Fix imports * Additional tests. Add tests to pyproject.toml * Update poetry lock * New line * Bump aiortc to 1.6.0 * Added teleoprtc_repo as submodule, and linked its source dir * Add init file to webrtc module * Handle aiortc warnings * Ignore deprecation warnings * Ignore resource warning too * Ignore the warnings * find free port for test_webrtcd * Start process inside the test case * random sleep test * test 2 * Test endpoint function instead * Update comment * Add system/webrtc to release * default arguments for body fields * Add teleoprtc to release * Bump teleoprtc * Exclude teleoprtc from static analysis * Use separate event loop for stream session tests old-commit-hash: f058b5d64eed31878533a85f25eb53386ff9e330
1 year ago
clearInterval(controlCommandInterval);
clearInterval(latencyInterval);
};
webrtcd: webrtc streaming server (audio/video/cereal) (#30186) * WebRTCClient and WebRTCServer abstractions * webrtc client implementation * Interactive test scripts * Send localDescriptions as offer/asnwer, as they are different * Tracks need to be added after setting remote description for multi-cam streaming to work * Remove WebRTCStreamingMetadata * Wait for tracks * Move stuff to separate files, rename some things * Refactor everything, create WebRTCStreamBuilder for both offer and answers * ta flight done time to grind * wait for incoming tracks and channels * Dummy track and frame reader track. Fix timing. * dt based on camera type * first trial of the new api * Fix audio track * methods for checking for incoming tracks * Web migration part 2 * Fixes for stream api * use rtc description for web.py * experimental cereal proxy * remove old code from bodyav * fix is_started * serialize session description * fix audio * messaging channel wrapper * fix audiotrack * h264 codec preference * Add codec preference to tracks * override sdp codecs * add logging * Move cli stuff to separate file * slight cleanup * Fix audio track * create codec_mime inside force_codec function * fix incoming media estimation * move builders to __init__ * stream updates following builders * Update example script * web.py support for new builder * web speaker fixes * StreamingMediaInfo API * Move things around * should_add_data_channel rename * is_connected_and_ready * fix linter errors * make cli executable * remove dumb comments * logging support * fix parse_info_from_offer * improve type annotations * satisfy linters * Support for waiting for disconnection * Split device tracks into video/audio files. Move audio speaker to audio.py * default dt for dummy video track * Fix cli * new speaker fixes * Remove almost all functionality from web.py * webrtcd * continue refactoring web.py * after handling joystick reset in controlsd with #30409, controls are not necessary anymore * ping endpoint * Update js files to at least support what worked previously * Fixes after some tests on the body * Streaming fixes * Remove the use of WebRTCStreamBuilder. Subclass use is now required * Add todo * delete all streams on shutdown * Replace lastPing with lastChannelMessageTime * Update ping text only if rtc is still on * That should affect the chart too * Fix paths in web * use protocol in SSLContext * remove warnings since aiortc is not used directly anymore * check if task is done in stop * remove channel handler wrapper, since theres only one channel * Move things around * Moved webrtc abstractions to separate repository * Moved webrtcd to tools/webrtc * Update imports * Add bodyrtc as dependency * Add webrtcd to process_config * Remove usage of DummyVideoStreamTrack * Add main to webrtcd * Move webrtcd to system * Fix imports * Move cereal proxy logic outside of runner * Incoming proxy abstractions * Add some tests * Make it executable * Fix process config * Fix imports * Additional tests. Add tests to pyproject.toml * Update poetry lock * New line * Bump aiortc to 1.6.0 * Added teleoprtc_repo as submodule, and linked its source dir * Add init file to webrtc module * Handle aiortc warnings * Ignore deprecation warnings * Ignore resource warning too * Ignore the warnings * find free port for test_webrtcd * Start process inside the test case * random sleep test * test 2 * Test endpoint function instead * Update comment * Add system/webrtc to release * default arguments for body fields * Add teleoprtc to release * Bump teleoprtc * Exclude teleoprtc from static analysis * Use separate event loop for stream session tests old-commit-hash: f058b5d64eed31878533a85f25eb53386ff9e330
1 year ago
function sendJoystickOverDataChannel() {
const {x, y} = getXY();
webrtcd: webrtc streaming server (audio/video/cereal) (#30186) * WebRTCClient and WebRTCServer abstractions * webrtc client implementation * Interactive test scripts * Send localDescriptions as offer/asnwer, as they are different * Tracks need to be added after setting remote description for multi-cam streaming to work * Remove WebRTCStreamingMetadata * Wait for tracks * Move stuff to separate files, rename some things * Refactor everything, create WebRTCStreamBuilder for both offer and answers * ta flight done time to grind * wait for incoming tracks and channels * Dummy track and frame reader track. Fix timing. * dt based on camera type * first trial of the new api * Fix audio track * methods for checking for incoming tracks * Web migration part 2 * Fixes for stream api * use rtc description for web.py * experimental cereal proxy * remove old code from bodyav * fix is_started * serialize session description * fix audio * messaging channel wrapper * fix audiotrack * h264 codec preference * Add codec preference to tracks * override sdp codecs * add logging * Move cli stuff to separate file * slight cleanup * Fix audio track * create codec_mime inside force_codec function * fix incoming media estimation * move builders to __init__ * stream updates following builders * Update example script * web.py support for new builder * web speaker fixes * StreamingMediaInfo API * Move things around * should_add_data_channel rename * is_connected_and_ready * fix linter errors * make cli executable * remove dumb comments * logging support * fix parse_info_from_offer * improve type annotations * satisfy linters * Support for waiting for disconnection * Split device tracks into video/audio files. Move audio speaker to audio.py * default dt for dummy video track * Fix cli * new speaker fixes * Remove almost all functionality from web.py * webrtcd * continue refactoring web.py * after handling joystick reset in controlsd with #30409, controls are not necessary anymore * ping endpoint * Update js files to at least support what worked previously * Fixes after some tests on the body * Streaming fixes * Remove the use of WebRTCStreamBuilder. Subclass use is now required * Add todo * delete all streams on shutdown * Replace lastPing with lastChannelMessageTime * Update ping text only if rtc is still on * That should affect the chart too * Fix paths in web * use protocol in SSLContext * remove warnings since aiortc is not used directly anymore * check if task is done in stop * remove channel handler wrapper, since theres only one channel * Move things around * Moved webrtc abstractions to separate repository * Moved webrtcd to tools/webrtc * Update imports * Add bodyrtc as dependency * Add webrtcd to process_config * Remove usage of DummyVideoStreamTrack * Add main to webrtcd * Move webrtcd to system * Fix imports * Move cereal proxy logic outside of runner * Incoming proxy abstractions * Add some tests * Make it executable * Fix process config * Fix imports * Additional tests. Add tests to pyproject.toml * Update poetry lock * New line * Bump aiortc to 1.6.0 * Added teleoprtc_repo as submodule, and linked its source dir * Add init file to webrtc module * Handle aiortc warnings * Ignore deprecation warnings * Ignore resource warning too * Ignore the warnings * find free port for test_webrtcd * Start process inside the test case * random sleep test * test 2 * Test endpoint function instead * Update comment * Add system/webrtc to release * default arguments for body fields * Add teleoprtc to release * Bump teleoprtc * Exclude teleoprtc from static analysis * Use separate event loop for stream session tests old-commit-hash: f058b5d64eed31878533a85f25eb53386ff9e330
1 year ago
var message = JSON.stringify({type: "testJoystick", data: {axes: [x, y], buttons: [false]}})
dc.send(message);
}
webrtcd: webrtc streaming server (audio/video/cereal) (#30186) * WebRTCClient and WebRTCServer abstractions * webrtc client implementation * Interactive test scripts * Send localDescriptions as offer/asnwer, as they are different * Tracks need to be added after setting remote description for multi-cam streaming to work * Remove WebRTCStreamingMetadata * Wait for tracks * Move stuff to separate files, rename some things * Refactor everything, create WebRTCStreamBuilder for both offer and answers * ta flight done time to grind * wait for incoming tracks and channels * Dummy track and frame reader track. Fix timing. * dt based on camera type * first trial of the new api * Fix audio track * methods for checking for incoming tracks * Web migration part 2 * Fixes for stream api * use rtc description for web.py * experimental cereal proxy * remove old code from bodyav * fix is_started * serialize session description * fix audio * messaging channel wrapper * fix audiotrack * h264 codec preference * Add codec preference to tracks * override sdp codecs * add logging * Move cli stuff to separate file * slight cleanup * Fix audio track * create codec_mime inside force_codec function * fix incoming media estimation * move builders to __init__ * stream updates following builders * Update example script * web.py support for new builder * web speaker fixes * StreamingMediaInfo API * Move things around * should_add_data_channel rename * is_connected_and_ready * fix linter errors * make cli executable * remove dumb comments * logging support * fix parse_info_from_offer * improve type annotations * satisfy linters * Support for waiting for disconnection * Split device tracks into video/audio files. Move audio speaker to audio.py * default dt for dummy video track * Fix cli * new speaker fixes * Remove almost all functionality from web.py * webrtcd * continue refactoring web.py * after handling joystick reset in controlsd with #30409, controls are not necessary anymore * ping endpoint * Update js files to at least support what worked previously * Fixes after some tests on the body * Streaming fixes * Remove the use of WebRTCStreamBuilder. Subclass use is now required * Add todo * delete all streams on shutdown * Replace lastPing with lastChannelMessageTime * Update ping text only if rtc is still on * That should affect the chart too * Fix paths in web * use protocol in SSLContext * remove warnings since aiortc is not used directly anymore * check if task is done in stop * remove channel handler wrapper, since theres only one channel * Move things around * Moved webrtc abstractions to separate repository * Moved webrtcd to tools/webrtc * Update imports * Add bodyrtc as dependency * Add webrtcd to process_config * Remove usage of DummyVideoStreamTrack * Add main to webrtcd * Move webrtcd to system * Fix imports * Move cereal proxy logic outside of runner * Incoming proxy abstractions * Add some tests * Make it executable * Fix process config * Fix imports * Additional tests. Add tests to pyproject.toml * Update poetry lock * New line * Bump aiortc to 1.6.0 * Added teleoprtc_repo as submodule, and linked its source dir * Add init file to webrtc module * Handle aiortc warnings * Ignore deprecation warnings * Ignore resource warning too * Ignore the warnings * find free port for test_webrtcd * Start process inside the test case * random sleep test * test 2 * Test endpoint function instead * Update comment * Add system/webrtc to release * default arguments for body fields * Add teleoprtc to release * Bump teleoprtc * Exclude teleoprtc from static analysis * Use separate event loop for stream session tests old-commit-hash: f058b5d64eed31878533a85f25eb53386ff9e330
1 year ago
function checkLatency() {
const initialTime = new Date().getTime();
pingHeadRequest().then(function() {
const currentTime = new Date().getTime();
if (Math.abs(currentTime - lastChannelMessageTime) < 1000) {
const pingtime = currentTime - initialTime;
pingPoints.push({'x': currentTime, 'y': pingtime});
if (pingPoints.length > 1000) {
pingPoints.shift();
}
chartPing.update();
$("#ping-time").text((pingtime) + "ms");
}
})
}
dc.onopen = function() {
webrtcd: webrtc streaming server (audio/video/cereal) (#30186) * WebRTCClient and WebRTCServer abstractions * webrtc client implementation * Interactive test scripts * Send localDescriptions as offer/asnwer, as they are different * Tracks need to be added after setting remote description for multi-cam streaming to work * Remove WebRTCStreamingMetadata * Wait for tracks * Move stuff to separate files, rename some things * Refactor everything, create WebRTCStreamBuilder for both offer and answers * ta flight done time to grind * wait for incoming tracks and channels * Dummy track and frame reader track. Fix timing. * dt based on camera type * first trial of the new api * Fix audio track * methods for checking for incoming tracks * Web migration part 2 * Fixes for stream api * use rtc description for web.py * experimental cereal proxy * remove old code from bodyav * fix is_started * serialize session description * fix audio * messaging channel wrapper * fix audiotrack * h264 codec preference * Add codec preference to tracks * override sdp codecs * add logging * Move cli stuff to separate file * slight cleanup * Fix audio track * create codec_mime inside force_codec function * fix incoming media estimation * move builders to __init__ * stream updates following builders * Update example script * web.py support for new builder * web speaker fixes * StreamingMediaInfo API * Move things around * should_add_data_channel rename * is_connected_and_ready * fix linter errors * make cli executable * remove dumb comments * logging support * fix parse_info_from_offer * improve type annotations * satisfy linters * Support for waiting for disconnection * Split device tracks into video/audio files. Move audio speaker to audio.py * default dt for dummy video track * Fix cli * new speaker fixes * Remove almost all functionality from web.py * webrtcd * continue refactoring web.py * after handling joystick reset in controlsd with #30409, controls are not necessary anymore * ping endpoint * Update js files to at least support what worked previously * Fixes after some tests on the body * Streaming fixes * Remove the use of WebRTCStreamBuilder. Subclass use is now required * Add todo * delete all streams on shutdown * Replace lastPing with lastChannelMessageTime * Update ping text only if rtc is still on * That should affect the chart too * Fix paths in web * use protocol in SSLContext * remove warnings since aiortc is not used directly anymore * check if task is done in stop * remove channel handler wrapper, since theres only one channel * Move things around * Moved webrtc abstractions to separate repository * Moved webrtcd to tools/webrtc * Update imports * Add bodyrtc as dependency * Add webrtcd to process_config * Remove usage of DummyVideoStreamTrack * Add main to webrtcd * Move webrtcd to system * Fix imports * Move cereal proxy logic outside of runner * Incoming proxy abstractions * Add some tests * Make it executable * Fix process config * Fix imports * Additional tests. Add tests to pyproject.toml * Update poetry lock * New line * Bump aiortc to 1.6.0 * Added teleoprtc_repo as submodule, and linked its source dir * Add init file to webrtc module * Handle aiortc warnings * Ignore deprecation warnings * Ignore resource warning too * Ignore the warnings * find free port for test_webrtcd * Start process inside the test case * random sleep test * test 2 * Test endpoint function instead * Update comment * Add system/webrtc to release * default arguments for body fields * Add teleoprtc to release * Bump teleoprtc * Exclude teleoprtc from static analysis * Use separate event loop for stream session tests old-commit-hash: f058b5d64eed31878533a85f25eb53386ff9e330
1 year ago
controlCommandInterval = setInterval(sendJoystickOverDataChannel, 50);
latencyInterval = setInterval(checkLatency, 1000);
sendJoystickOverDataChannel();
};
webrtcd: webrtc streaming server (audio/video/cereal) (#30186) * WebRTCClient and WebRTCServer abstractions * webrtc client implementation * Interactive test scripts * Send localDescriptions as offer/asnwer, as they are different * Tracks need to be added after setting remote description for multi-cam streaming to work * Remove WebRTCStreamingMetadata * Wait for tracks * Move stuff to separate files, rename some things * Refactor everything, create WebRTCStreamBuilder for both offer and answers * ta flight done time to grind * wait for incoming tracks and channels * Dummy track and frame reader track. Fix timing. * dt based on camera type * first trial of the new api * Fix audio track * methods for checking for incoming tracks * Web migration part 2 * Fixes for stream api * use rtc description for web.py * experimental cereal proxy * remove old code from bodyav * fix is_started * serialize session description * fix audio * messaging channel wrapper * fix audiotrack * h264 codec preference * Add codec preference to tracks * override sdp codecs * add logging * Move cli stuff to separate file * slight cleanup * Fix audio track * create codec_mime inside force_codec function * fix incoming media estimation * move builders to __init__ * stream updates following builders * Update example script * web.py support for new builder * web speaker fixes * StreamingMediaInfo API * Move things around * should_add_data_channel rename * is_connected_and_ready * fix linter errors * make cli executable * remove dumb comments * logging support * fix parse_info_from_offer * improve type annotations * satisfy linters * Support for waiting for disconnection * Split device tracks into video/audio files. Move audio speaker to audio.py * default dt for dummy video track * Fix cli * new speaker fixes * Remove almost all functionality from web.py * webrtcd * continue refactoring web.py * after handling joystick reset in controlsd with #30409, controls are not necessary anymore * ping endpoint * Update js files to at least support what worked previously * Fixes after some tests on the body * Streaming fixes * Remove the use of WebRTCStreamBuilder. Subclass use is now required * Add todo * delete all streams on shutdown * Replace lastPing with lastChannelMessageTime * Update ping text only if rtc is still on * That should affect the chart too * Fix paths in web * use protocol in SSLContext * remove warnings since aiortc is not used directly anymore * check if task is done in stop * remove channel handler wrapper, since theres only one channel * Move things around * Moved webrtc abstractions to separate repository * Moved webrtcd to tools/webrtc * Update imports * Add bodyrtc as dependency * Add webrtcd to process_config * Remove usage of DummyVideoStreamTrack * Add main to webrtcd * Move webrtcd to system * Fix imports * Move cereal proxy logic outside of runner * Incoming proxy abstractions * Add some tests * Make it executable * Fix process config * Fix imports * Additional tests. Add tests to pyproject.toml * Update poetry lock * New line * Bump aiortc to 1.6.0 * Added teleoprtc_repo as submodule, and linked its source dir * Add init file to webrtc module * Handle aiortc warnings * Ignore deprecation warnings * Ignore resource warning too * Ignore the warnings * find free port for test_webrtcd * Start process inside the test case * random sleep test * test 2 * Test endpoint function instead * Update comment * Add system/webrtc to release * default arguments for body fields * Add teleoprtc to release * Bump teleoprtc * Exclude teleoprtc from static analysis * Use separate event loop for stream session tests old-commit-hash: f058b5d64eed31878533a85f25eb53386ff9e330
1 year ago
const textDecoder = new TextDecoder();
var carStaterIndex = 0;
dc.onmessage = function(evt) {
webrtcd: webrtc streaming server (audio/video/cereal) (#30186) * WebRTCClient and WebRTCServer abstractions * webrtc client implementation * Interactive test scripts * Send localDescriptions as offer/asnwer, as they are different * Tracks need to be added after setting remote description for multi-cam streaming to work * Remove WebRTCStreamingMetadata * Wait for tracks * Move stuff to separate files, rename some things * Refactor everything, create WebRTCStreamBuilder for both offer and answers * ta flight done time to grind * wait for incoming tracks and channels * Dummy track and frame reader track. Fix timing. * dt based on camera type * first trial of the new api * Fix audio track * methods for checking for incoming tracks * Web migration part 2 * Fixes for stream api * use rtc description for web.py * experimental cereal proxy * remove old code from bodyav * fix is_started * serialize session description * fix audio * messaging channel wrapper * fix audiotrack * h264 codec preference * Add codec preference to tracks * override sdp codecs * add logging * Move cli stuff to separate file * slight cleanup * Fix audio track * create codec_mime inside force_codec function * fix incoming media estimation * move builders to __init__ * stream updates following builders * Update example script * web.py support for new builder * web speaker fixes * StreamingMediaInfo API * Move things around * should_add_data_channel rename * is_connected_and_ready * fix linter errors * make cli executable * remove dumb comments * logging support * fix parse_info_from_offer * improve type annotations * satisfy linters * Support for waiting for disconnection * Split device tracks into video/audio files. Move audio speaker to audio.py * default dt for dummy video track * Fix cli * new speaker fixes * Remove almost all functionality from web.py * webrtcd * continue refactoring web.py * after handling joystick reset in controlsd with #30409, controls are not necessary anymore * ping endpoint * Update js files to at least support what worked previously * Fixes after some tests on the body * Streaming fixes * Remove the use of WebRTCStreamBuilder. Subclass use is now required * Add todo * delete all streams on shutdown * Replace lastPing with lastChannelMessageTime * Update ping text only if rtc is still on * That should affect the chart too * Fix paths in web * use protocol in SSLContext * remove warnings since aiortc is not used directly anymore * check if task is done in stop * remove channel handler wrapper, since theres only one channel * Move things around * Moved webrtc abstractions to separate repository * Moved webrtcd to tools/webrtc * Update imports * Add bodyrtc as dependency * Add webrtcd to process_config * Remove usage of DummyVideoStreamTrack * Add main to webrtcd * Move webrtcd to system * Fix imports * Move cereal proxy logic outside of runner * Incoming proxy abstractions * Add some tests * Make it executable * Fix process config * Fix imports * Additional tests. Add tests to pyproject.toml * Update poetry lock * New line * Bump aiortc to 1.6.0 * Added teleoprtc_repo as submodule, and linked its source dir * Add init file to webrtc module * Handle aiortc warnings * Ignore deprecation warnings * Ignore resource warning too * Ignore the warnings * find free port for test_webrtcd * Start process inside the test case * random sleep test * test 2 * Test endpoint function instead * Update comment * Add system/webrtc to release * default arguments for body fields * Add teleoprtc to release * Bump teleoprtc * Exclude teleoprtc from static analysis * Use separate event loop for stream session tests old-commit-hash: f058b5d64eed31878533a85f25eb53386ff9e330
1 year ago
const text = textDecoder.decode(evt.data);
const msg = JSON.parse(text);
if (carStaterIndex % 100 == 0 && msg.type === 'carState') {
const batteryLevel = Math.round(msg.data.fuelGauge * 100);
$("#battery").text(batteryLevel + "%");
batteryPoints.push({'x': new Date().getTime(), 'y': batteryLevel});
if (batteryPoints.length > 1000) {
batteryPoints.shift();
}
chartBattery.update();
}
webrtcd: webrtc streaming server (audio/video/cereal) (#30186) * WebRTCClient and WebRTCServer abstractions * webrtc client implementation * Interactive test scripts * Send localDescriptions as offer/asnwer, as they are different * Tracks need to be added after setting remote description for multi-cam streaming to work * Remove WebRTCStreamingMetadata * Wait for tracks * Move stuff to separate files, rename some things * Refactor everything, create WebRTCStreamBuilder for both offer and answers * ta flight done time to grind * wait for incoming tracks and channels * Dummy track and frame reader track. Fix timing. * dt based on camera type * first trial of the new api * Fix audio track * methods for checking for incoming tracks * Web migration part 2 * Fixes for stream api * use rtc description for web.py * experimental cereal proxy * remove old code from bodyav * fix is_started * serialize session description * fix audio * messaging channel wrapper * fix audiotrack * h264 codec preference * Add codec preference to tracks * override sdp codecs * add logging * Move cli stuff to separate file * slight cleanup * Fix audio track * create codec_mime inside force_codec function * fix incoming media estimation * move builders to __init__ * stream updates following builders * Update example script * web.py support for new builder * web speaker fixes * StreamingMediaInfo API * Move things around * should_add_data_channel rename * is_connected_and_ready * fix linter errors * make cli executable * remove dumb comments * logging support * fix parse_info_from_offer * improve type annotations * satisfy linters * Support for waiting for disconnection * Split device tracks into video/audio files. Move audio speaker to audio.py * default dt for dummy video track * Fix cli * new speaker fixes * Remove almost all functionality from web.py * webrtcd * continue refactoring web.py * after handling joystick reset in controlsd with #30409, controls are not necessary anymore * ping endpoint * Update js files to at least support what worked previously * Fixes after some tests on the body * Streaming fixes * Remove the use of WebRTCStreamBuilder. Subclass use is now required * Add todo * delete all streams on shutdown * Replace lastPing with lastChannelMessageTime * Update ping text only if rtc is still on * That should affect the chart too * Fix paths in web * use protocol in SSLContext * remove warnings since aiortc is not used directly anymore * check if task is done in stop * remove channel handler wrapper, since theres only one channel * Move things around * Moved webrtc abstractions to separate repository * Moved webrtcd to tools/webrtc * Update imports * Add bodyrtc as dependency * Add webrtcd to process_config * Remove usage of DummyVideoStreamTrack * Add main to webrtcd * Move webrtcd to system * Fix imports * Move cereal proxy logic outside of runner * Incoming proxy abstractions * Add some tests * Make it executable * Fix process config * Fix imports * Additional tests. Add tests to pyproject.toml * Update poetry lock * New line * Bump aiortc to 1.6.0 * Added teleoprtc_repo as submodule, and linked its source dir * Add init file to webrtc module * Handle aiortc warnings * Ignore deprecation warnings * Ignore resource warning too * Ignore the warnings * find free port for test_webrtcd * Start process inside the test case * random sleep test * test 2 * Test endpoint function instead * Update comment * Add system/webrtc to release * default arguments for body fields * Add teleoprtc to release * Bump teleoprtc * Exclude teleoprtc from static analysis * Use separate event loop for stream session tests old-commit-hash: f058b5d64eed31878533a85f25eb53386ff9e330
1 year ago
carStaterIndex += 1;
lastChannelMessageTime = new Date().getTime();
$(".pre-blob").addClass('blob');
};
}
export function stop(pc, dc) {
if (dc) {
dc.close();
}
if (pc.getTransceivers) {
pc.getTransceivers().forEach(function(transceiver) {
if (transceiver.stop) {
transceiver.stop();
}
});
}
pc.getSenders().forEach(function(sender) {
sender.track.stop();
});
setTimeout(function() {
pc.close();
}, 500);
}