Table of Contents |
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Example of streamer with access to media devices
This streamer can be used to publish or playback WebRTC streams on Web Call Server and allows to select media devices and parameters for the published video
- camera
- microphone
- FPS (Frames Per Second)
- resolution (width, height)
On the screenshot below a stream is being published from the client.
Two video elements are displayed on the page
- 'Local' - video from the camera
- 'Player' - the video received from the server
Code of the example
The path to the source code of the example on WCS server is:
/usr/local/FlashphonerWebCallServer/client/examples/demo/streaming/media_devices_manager
manager.css - file with styles
media_device_manager.html - page of the streamer
manager.js - script providing functionality for the streamer
This example can be tested using the following address:
https://host:8888/client/examples/demo/streaming/media_devices_manager/media_device_manager.html
Here host is the address of the WCS server.
Analyzing the code
To analyze the code, let's take the version of file manager.js whith hash ecbadc3, which is available here and can be downloaded with corresponding build 2.0.212.
1. Initialization of the API.
Flashphoner.init() code
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Flashphoner.init({
screenSharingExtensionId: extensionId,
mediaProvidersReadyCallback: function (mediaProviders) {
if (Flashphoner.isUsingTemasys()) {
$("#audioInputForm").hide();
$("#videoInputForm").hide();
}
}
}) |
2. List available input media devices.
Flashphoner.getMediaDevices() code
When input media devices are listed, drop-down lists of microphones and cameras on client page are filled.
Code Block | ||||
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Flashphoner.getMediaDevices(null, true).then(function (list) {
list.audio.forEach(function (device) {
...
});
list.video.forEach(function (device) {
...
});
...
}).catch(function (error) {
$("#notifyFlash").text("Failed to get media devices");
}); |
3. List available output media devices
Flashphoner.getMediaDevices() code
When output media devices are listed, drop-down lists of spakers and headphones on client page are filled.
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Flashphoner.getMediaDevices(null, true, MEDIA_DEVICE_KIND.OUTPUT).then(function (list) {
list.audio.forEach(function (device) {
...
});
...
}).catch(function (error) {
$("#notifyFlash").text("Failed to get media devices");
}); |
4. Get audio and video publishing constraints from client page
getConstraints() code
Publishing sources:
- camera (sendVideo)
- microphone (sendAudio)
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constraints = {
audio: $("#sendAudio").is(':checked'),
video: $("#sendVideo").is(':checked'),
};
|
Audio constraints:
- microphone choise (deviceId)
- error correction for Opus codec (fec)
- stereo mode (stereo)
- audio bitrate (bitrate)
Code Block | ||||
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if (constraints.audio) {
constraints.audio = {
deviceId: $('#audioInput').val()
};
if ($("#fec").is(':checked'))
constraints.audio.fec = $("#fec").is(':checked');
if ($("#sendStereoAudio").is(':checked'))
constraints.audio.stereo = $("#sendStereoAudio").is(':checked');
if (parseInt($('#sendAudioBitrate').val()) > 0)
constraints.audio.bitrate = parseInt($('#sendAudioBitrate').val());
}
|
Video constraints:
- camera choise (deviceId)
- publishing video size (width, height)
- minimal and maximal video bitrate (minBitrate, maxBitrate)
- FPS (frameRate)
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constraints.video = {
deviceId: {exact: $('#videoInput').val()},
width: parseInt($('#sendWidth').val()),
height: parseInt($('#sendHeight').val())
};
if (Browser.isSafariWebRTC() && Browser.isiOS() && Flashphoner.getMediaProviders()[0] === "WebRTC") {
constraints.video.deviceId = {exact: $('#videoInput').val()};
}
if (parseInt($('#sendVideoMinBitrate').val()) > 0)
constraints.video.minBitrate = parseInt($('#sendVideoMinBitrate').val());
if (parseInt($('#sendVideoMaxBitrate').val()) > 0)
constraints.video.maxBitrate = parseInt($('#sendVideoMaxBitrate').val());
if (parseInt($('#fps').val()) > 0)
constraints.video.frameRate = parseInt($('#fps').val());
|
5. Get access to media devices for local test
Flashphoner.getMediaAccess() code
Audio and video constraints and <div>-element to display captured video are passed to the method.
Code Block | ||||
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Flashphoner.getMediaAccess(getConstraints(), localVideo).then(function (disp) {
$("#testBtn").text("Release").off('click').click(function () {
$(this).prop('disabled', true);
stopTest();
}).prop('disabled', false);
...
testStarted = true;
}).catch(function (error) {
$("#testBtn").prop('disabled', false);
testStarted = false;
}); |
6. Connecting to the server
Flashphoner.createSession() code
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Flashphoner.createSession({urlServer: url, timeout: tm}).on(SESSION_STATUS.ESTABLISHED, function (session) {
...
}).on(SESSION_STATUS.DISCONNECTED, function () {
...
}).on(SESSION_STATUS.FAILED, function () {
...
}); |
7. Receiving the event confirming successful connection
ConnectionStatusEvent ESTABLISHED code
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Flashphoner.createSession({urlServer: url, timeout: tm}).on(SESSION_STATUS.ESTABLISHED, function (session) {
setStatus("#connectStatus", session.status());
onConnected(session);
...
}); |
8. Stream publishing
session.createStream(), publishStream.publish() code
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publishStream = session.createStream({
name: streamName,
display: localVideo,
cacheLocalResources: true,
constraints: constraints,
mediaConnectionConstraints: mediaConnectionConstraints,
sdpHook: rewriteSdp,
transport: transportInput,
cvoExtension: cvo,
stripCodecs: strippedCodecs,
videoContentHint: contentHint
...
});
publishStream.publish(); |
9. Receiving the event confirming successful streaming
StreamStatusEvent PUBLISHING code
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publishStream = session.createStream({
...
}).on(STREAM_STATUS.PUBLISHING, function (stream) {
$("#testBtn").prop('disabled', true);
var video = document.getElementById(stream.id());
//resize local if resolution is available
if (video.videoWidth > 0 && video.videoHeight > 0) {
resizeLocalVideo({target: video});
}
enablePublishToggles(true);
if ($("#muteVideoToggle").is(":checked")) {
muteVideo();
}
if ($("#muteAudioToggle").is(":checked")) {
muteAudio();
}
//remove resize listener in case this video was cached earlier
video.removeEventListener('resize', resizeLocalVideo);
video.addEventListener('resize', resizeLocalVideo);
publishStream.setMicrophoneGain(currentGainValue);
setStatus("#publishStatus", STREAM_STATUS.PUBLISHING);
onPublishing(stream);
}).on(STREAM_STATUS.UNPUBLISHED, function () {
...
}).on(STREAM_STATUS.FAILED, function () {
...
});
publishStream.publish(); |
10. Stream playback
session.createStream(), previewStream.play() code
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previewStream = session.createStream({
name: streamName,
display: remoteVideo,
constraints: constraints,
transport: transportOutput,
stripCodecs: strippedCodecs
...
});
previewStream.play(); |
11. Receiving the event confirming successful playback
StreamStatusEvent PLAYING code
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previewStream = session.createStream({
...
}).on(STREAM_STATUS.PLAYING, function (stream) {
playConnectionQualityStat.connectionQualityUpdateTimestamp = new Date().valueOf();
setStatus("#playStatus", stream.status());
onPlaying(stream);
document.getElementById(stream.id()).addEventListener('resize', function (event) {
$("#playResolution").text(event.target.videoWidth + "x" + event.target.videoHeight);
resizeVideo(event.target);
});
//wait for incoming stream
if (Flashphoner.getMediaProviders()[0] == "WebRTC") {
setTimeout(function () {
if(Browser.isChrome()) {
detectSpeechChrome(stream);
} else {
detectSpeech(stream);
}
}, 3000);
}
...
});
previewStream.play(); |
12. Stop stream playback
stream.stop() code
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$("#playBtn").text("Stop").off('click').click(function () {
$(this).prop('disabled', true);
stream.stop();
}).prop('disabled', false); |
13. Receiving the event confirming successful playback stop
StreamStatusEvent STOPPED code
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previewStream = session.createStream({
...
}).on(STREAM_STATUS.STOPPED, function () {
setStatus("#playStatus", STREAM_STATUS.STOPPED);
onStopped();
...
});
previewStream.play(); |
14. Stop stream publishing
stream.stop() code
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$("#publishBtn").text("Stop").off('click').click(function () {
$(this).prop('disabled', true);
stream.stop();
}).prop('disabled', false); |
15. Receiving the event confirming successful publishsing stop
StreamStatusEvent UNPUBLISHED code
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publishStream = session.createStream({
...
}).on(STREAM_STATUS.UNPUBLISHED, function () {
setStatus("#publishStatus", STREAM_STATUS.UNPUBLISHED);
onUnpublished();
...
});
publishStream.publish(); |
16. Mute publisher audio
stream.muteAudio() code:
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function muteAudio() {
if (publishStream) {
publishStream.muteAudio();
}
} |
17. Mute publisher video
stream.muteVideo() code:
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function muteVideo() {
if (publishStream) {
publishStream.muteVideo();
}
} |
18. Show WebRTC stream publishing statistics
stream.getStats() code:
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publishStream.getStats(function (stats) {
if (stats && stats.outboundStream) {
if (stats.outboundStream.video) {
showStat(stats.outboundStream.video, "outVideoStat");
let vBitrate = (stats.outboundStream.video.bytesSent - videoBytesSent) * 8;
if ($('#outVideoStatBitrate').length == 0) {
let html = "<div>Bitrate: " + "<span id='outVideoStatBitrate' style='font-weight: normal'>" + vBitrate + "</span>" + "</div>";
$("#outVideoStat").append(html);
} else {
$('#outVideoStatBitrate').text(vBitrate);
}
videoBytesSent = stats.outboundStream.video.bytesSent;
...
}
if (stats.outboundStream.audio) {
showStat(stats.outboundStream.audio, "outAudioStat");
let aBitrate = (stats.outboundStream.audio.bytesSent - audioBytesSent) * 8;
if ($('#outAudioStatBitrate').length == 0) {
let html = "<div>Bitrate: " + "<span id='outAudioStatBitrate' style='font-weight: normal'>" + aBitrate + "</span>" + "</div>";
$("#outAudioStat").append(html);
} else {
$('#outAudioStatBitrate').text(aBitrate);
}
audioBytesSent = stats.outboundStream.audio.bytesSent;
}
}
...
}); |
19. Show WebRTC stream playback statistics
stream.getStats() code:
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previewStream.getStats(function (stats) {
if (stats && stats.inboundStream) {
if (stats.inboundStream.video) {
showStat(stats.inboundStream.video, "inVideoStat");
let vBitrate = (stats.inboundStream.video.bytesReceived - videoBytesReceived) * 8;
if ($('#inVideoStatBitrate').length == 0) {
let html = "<div>Bitrate: " + "<span id='inVideoStatBitrate' style='font-weight: normal'>" + vBitrate + "</span>" + "</div>";
$("#inVideoStat").append(html);
} else {
$('#inVideoStatBitrate').text(vBitrate);
}
videoBytesReceived = stats.inboundStream.video.bytesReceived;
...
}
if (stats.inboundStream.audio) {
showStat(stats.inboundStream.audio, "inAudioStat");
let aBitrate = (stats.inboundStream.audio.bytesReceived - audioBytesReceived) * 8;
if ($('#inAudioStatBitrate').length == 0) {
let html = "<div style='font-weight: bold'>Bitrate: " + "<span id='inAudioStatBitrate' style='font-weight: normal'>" + aBitrate + "</span>" + "</div>";
$("#inAudioStat").append(html);
} else {
$('#inAudioStatBitrate').text(aBitrate);
}
audioBytesReceived = stats.inboundStream.audio.bytesReceived;
}
...
}
}); |
20. Speech detection using ScriptProcessor interface (any browser except Chrome)
audioContext.createMediaStreamSource(), audioContext.createScriptProcessor() code
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function detectSpeech(stream, level, latency) {
var mediaStream = document.getElementById(stream.id()).srcObject;
var source = audioContext.createMediaStreamSource(mediaStream);
var processor = audioContext.createScriptProcessor(512);
processor.onaudioprocess = handleAudio;
processor.connect(audioContext.destination);
processor.clipping = false;
processor.lastClip = 0;
// threshold
processor.threshold = level || 0.10;
processor.latency = latency || 750;
processor.isSpeech =
function () {
if (!this.clipping) return false;
if ((this.lastClip + this.latency) < window.performance.now()) this.clipping = false;
return this.clipping;
};
source.connect(processor);
// Check speech every 500 ms
speechIntervalID = setInterval(function () {
if (processor.isSpeech()) {
$("#talking").css('background-color', 'green');
} else {
$("#talking").css('background-color', 'red');
}
}, 500);
} |
Audio data handler code
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function handleAudio(event) {
var buf = event.inputBuffer.getChannelData(0);
var bufLength = buf.length;
var x;
for (var i = 0; i < bufLength; i++) {
x = buf[i];
if (Math.abs(x) >= this.threshold) {
this.clipping = true;
this.lastClip = window.performance.now();
}
}
} |
21. Speech detection using incoming audio WebRTC statistics in Chrome browser
stream.getStats() code
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function detectSpeechChrome(stream, level, latency) {
statSpeechDetector.threshold = level || 0.010;
statSpeechDetector.latency = latency || 750;
statSpeechDetector.clipping = false;
statSpeechDetector.lastClip = 0;
speechIntervalID = setInterval(function() {
stream.getStats(function(stat) {
let audioStats = stat.inboundStream.audio;
if(!audioStats) {
return;
}
// Using audioLevel WebRTC stats parameter
if (audioStats.audioLevel >= statSpeechDetector.threshold) {
statSpeechDetector.clipping = true;
statSpeechDetector.lastClip = window.performance.now();
}
if ((statSpeechDetector.lastClip + statSpeechDetector.latency) < window.performance.now()) {
statSpeechDetector.clipping = false;
}
if (statSpeechDetector.clipping) {
$("#talking").css('background-color', 'green');
} else {
$("#talking").css('background-color', 'red');
}
});
},500);
} |