Video and audio content - Learn web development
ferent video (and audio) formats. fortunately, there are things you can do to help prevent this from being a problem. contents of a media file first, let's go through the terminology quickly. formats like mp3, mp4 and webm are called container formats. they define a structure in which the audio and/or video tracks that make up the media are stored, along with metadata describing the media, what codecs are used to encode its channels, and so forth. a webm file containing a movie which has a main video track and one alternate angle track, plus audio for both english and spanish, in addition to audio for an english commentary track can be conceptualized as shown in the diagram below. also included are text tracks containing closed captions for the feature film, spanish subtitles for the film, a...
...rack, plus audio for both english and spanish, in addition to audio for an english commentary track can be conceptualized as shown in the diagram below. also included are text tracks containing closed captions for the feature film, spanish subtitles for the film, and english captions for the commentary. the audio and video tracks within the container hold data in the appropriate format for the codec used to encode that media. different formats are used for audio tracks versus video tracks. each audio track is encoded using an audio codec, while video tracks are encoded using (as you probably have guessed) a video codec. as we talked about before, different browsers support different video and audio formats, and different container formats (like mp3, mp4, and webm, which in turn can contain d...
...ork. an mp4 container often packages aac or mp3 audio with h.264 video. this is also supported in all modern browsers, as well as internet explorer. the ogg container tends to use vorbis audio and theora video. this is best supported in firefox and chrome, but has basically been superseded by the better quality webm format. there are some special cases. for example, for some types of audio, a codec's data is often stored without a container, or with a simplified container. one such instance is the flac codec, which is stored most commonly in flac files, which are just raw flac tracks. another such situation is the always-popular mp3 file. an "mp3 file" is actually an mpeg-1 audio layer iii (mp3) audio track stored within an mpeg or mpeg-2 container. this is especially interesting since whi...
...And 6 more matches

MediaSource.endOfStream() - Web APIs
o call endofstream('network'), display a descriptive message in the ui, and maybe retry the network request immediately or wait until the network is back up (via some kind of polling.) decode: terminates playback and signals that a decoding error has occured. this can be used to indicate that a parsing error has occured while fetching media data; maybe the data is corrupt, or is encoded using a codec that the browser doesn't know how to decode. return value undefined exceptions exception explanation invalidstateerror mediasource.readystate is not equal to open, or one or more of the sourcebuffer objects in mediasource.sourcebuffers are being updated (i.e. their sourcebuffer.updating property is true.) example the following snippet is from a simple ...
...al to open, or one or more of the sourcebuffer objects in mediasource.sourcebuffers are being updated (i.e. ...
...ffer objects in mediasource.sourcebuffers are being updated (i.e. ...
...And 4 more matches

MediaSource.isTypeSupported() - Web APIs
user agent. that is, if it can successfully create sourcebuffer objects for that mime type. if the returned value is false, then the user agent is certain that it cannot access media of the specified format. syntax var isitsupported = mediasource.istypesupported(mimetype); parameters mimetype the mime media type that you want to test support for in the current browser. this may include the codecs parameter to provide added details about the codecs used within the file. return value a boolean which is true if the browser feels that it can probably play media of the specified type. this is not a guarantee, however, and your code must be prepared for the possibility that the media will not play correctly if at all. a value of false is a guarantee that media of the given type will not pla...
...dia files are complex, intricate constructs with far too many subtle variations to be absolutely certain of anything until you actually use the contents of the media. ...
...s with far too many subtle variations to be absolutely certain of anything until you actually use the contents of the media. ...
...And 4 more matches

Using the Media Capabilities API - Web APIs
the media capabilities api provides several key features to help you better decide how to handle media, but also to determine how well media is being handled, in real time. these features include: the ability to query the browser to determine its ability to encode or decode media given a specified set of encoding parameters. these parameters may include the codecs, resolutions, bit rates, frame rates, and other such details. with the media capabilities api, you can determine not just if the browser can support a given format, but whether or not it can do so efficiently and smoothly. in short, this api replaces—and improves upon—the mediasource method istypesupported() or the htmlmediaelement method canplaytype(). more and more finely-detailed information about the display's properties,...
...me type. the width and height are the horizontal and vertical dimensions of the video; these are also used to determine the aspect ratio. the bitrate is the number of bits used to encode one second of video. the framerate is the number of frames which are played per second of time when playing the video. ...
...coding example, a typeerror would be raised if the media configuration passed to the decodinginfo() method was invalid. ...
...And 4 more matches

Reading textual data - Archive of obsolete content
// now you can do something with line.value } while (cont); } the above example reads an entire stream until eof. see nsiconverterinputstream for nsiconverterinputstream.init() arguments. earlier versions reading strings earlier versions of gecko do not provide easy ways to read unicode data from a stream. ...
...stream for nsiconverterinputstream.init() arguments. earlier versions reading strings earlier versions of gecko do not provide easy ways to read unicode data from a stream. you will have to manually read a block of data and convert it using nsiscriptableunicodeconverter. ...
... versions of gecko do not provide easy ways to read unicode data from a stream. you will have to manually read a block of data and convert it using nsiscriptableunicodeconverter. ...
...And 3 more matches

MediaSource.addSourceBuffer() - Web APIs
't handle any more sourcebuffer objects, or creating a new sourcebuffer using the given mimetype would result in an unsupported configuration of sourcebuffers. ...
...r creating a new sourcebuffer using the given mimetype would result in an unsupported configuration of sourcebuffers. ...
...ew sourcebuffer using the given mimetype would result in an unsupported configuration of sourcebuffers. ...
...And 3 more matches

MediaSource - Web APIs
ebuffer objects for that mime type. examples the following simple example loads a video with xmlhttprequest, playing it as soon as it can. ...
...ples the following simple example loads a video with xmlhttprequest, playing it as soon as it can. ...
...owing simple example loads a video with xmlhttprequest, playing it as soon as it can. ...
...And 3 more matches

SourceBuffer - Web APIs
s have been removed. examples the following simple example loads a video chunk by chunk as fast as possible, playing it as soon as it can. ...
...wing simple example loads a video chunk by chunk as fast as possible, playing it as soon as it can. ...
...ample loads a video chunk by chunk as fast as possible, playing it as soon as it can. ...
...And 3 more matches

Setting up adaptive streaming media sources - Developer guides
s. a media presentation description (mpd) file is used to hold the information on the various streams and the bandwidths they are associated with. in your video source (src) attribute you point to the mpd instead of to the media file as you would with non-adaptive media. the mpd file tells the browser where the various pieces of media are located, it also includes meta data such as mimetype and codecs and there are other details such as byte-ranges in there too - it is an xml document and in many cases will be generated for you. there are a few profiles we can use. we're going to take a look at ondemand profile for video on demand (vod) and the live profile. for live services streaming, the live profile is a requirement. the stream switching capabilities are identical between the profiles. ...
...w3.org/2001/xmlschema-instance" xmlns="urn:mpeg:dash:schema:mpd:2011" xsi:schemalocation="urn:mpeg:dash:schema:mpd:2011 dash-mpd.xsd" type="static" mediapresentationduration="pt654s" minbuffertime="pt2s" profiles="urn:mpeg:dash:profile:isoff-on-demand:2011"> <baseurl>http://example.com/ondemand/</baseurl> <period> <!-- english audio --> <adaptationset mimetype="audio/mp4" codecs="mp4a.40.5" lang="en" subsegmentalignment="true" subsegmentstartswithsap="1"> <representation id="1" bandwidth="64000"> <baseurl>elephantsdream_aac48k_064.mp4.dash</baseurl> </representation> </adaptationset> <!-- video --> <adaptationset mimetype="video/mp4" codecs="avc1.42401e" subsegmentalignment="true" subsegmentstartswithsap="1"> <representation id="2" ...
...www.w3.org/2001/xmlschema-instance" xmlns="urn:mpeg:dash:schema:mpd:2011" xsi:schemalocation="urn:mpeg:dash:schema:mpd:2011" profiles="urn:mpeg:dash:profile:isoff-main:2011" type="static" mediapresentationduration="pt0h9m56.46s"> <baseurl> http://example.com/segments </baseurl> <period start="pt0s"> <adaptationset bitstreamswitching="true"> <representation id="0" codecs="avc1" mimetype="video/mp4" width="320" height="240" startwithsap="1" bandwidth="46986"> <segmentbase> <initialization sourceurl="main/news100/1.m4s" range="0-862"/> </segmentbase> <segmentlist duration="1"> <segmenturl media="main/news100/2.m4s" mediarange="863-7113"/> <segmenturl media="main/news100/3.m4s" mediarange="7114-14104...
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Media type and format guide: image, audio, and video content - Web media technologies
still and video imagery on the web. the modern web has powerful features to support the presentation and manipulation of media, with several media-related apis supporting various types of content. generally, the media formats supported by a browser are entirely up to the browser's creators, which can complicate the work of a web developer. this guide provides an overview of the media file types, codecs, and algorithms that may comprise media used on the web. it also provides browser support information for various combinations of these, and suggestions for prioritization of formats, as well as which formats excel at specific types of content. references images image file type and format guide covers support of image file types and content formats across the major web browsers, as well...
...on about each type: benefits, limitations, and use cases of interest to web designers and developers. image file types for web designers fundamental information about the various image file types that may be useful for web designers, including best practices and use cases for each type, and guidelines for choosing the right image file format for specific types of content. ...
...f content. media file types and codecs media containers (file types) a guide to the file types that contain media data. some are audio-specific, while others may be used for either audio or combined audiovisual content such as movies. includes overviews of each of the file types supported by the major web browsers, along with browser support information and supported features. web audio codec guide a guide to the audio codecs allowed for by the common media containers, as well as by the major browsers. includes benefits, limitations, key specifications and capabilities, and use cases. it also covers each browser's support for using the codec in given containers. ...
...And 3 more matches

Transcoding assets for Media Source Extensions - Web APIs
when working with media source extensions, it is likely that you need to condition your assets before you can stream them. this article takes you through the requirements and shows you a toolchain you can use to encode your assets appropriately. getting started the first and most important step is to ensure that your files are comprised of a container and codec that users' browsers support. depending on the codec, you might need to fragment the file to comply with the iso bmff spec. (optional) if you decide to use dynamic adaptive streaming over http (dash) for adaptive bitrate streaming, you need to transcode your assets into multiple resolutions. most dash clients expect a corresponding media presentation description (mpd) manifest file, which is typically generated while genera...
...rectory in the same place as ffmpeg. python2 — bento4 uses it. get these installed successfully before moving to the next step. sample media should be placed in the bento4 utils directory and worked here. note: the prebuilt ffmpeg does not include libfdk_aac due to licensing reasons. bento4 uses this by default, so you need to compile ffmpeg if necessary. if you don't need it, add --audio-codec=aac to the mp4-dash-encode.py command line. container and codec support as specified in section 1.1 of the mse spec: goals, mse is designed not to require support for any particular media format or codec. while this is true on paper, browser support varies for specific container/codec combinations. to check if the browser supports a particular container, you can pass a string of the mime typ...
...edia format or codec. while this is true on paper, browser support varies for specific container/codec combinations. to check if the browser supports a particular container, you can pass a string of the mime type to the mediasource.istypesupported method: mediasource.istypesupported('audio/mp3'); // false mediasource.istypesupported('video/mp4'); // true mediasource.istypesupported('video/mp4; codecs="avc1.4d4028, mp4a.40.2"'); // true the string is the mime type of the container, optionally followed by a list of codecs. while the mime type is fairly simple to figure out, we can get the codec string using the mp4info utility. currently, mp4 containers with h.264 video and aac audio codecs have support across all modern browsers, while others don't. to convert our sample media from a quick...
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RTCRtpCapabilities - Web APIs
the rtcrtpcapabilities dictionary is a data type used to describe the capabilities of an rtcrtpsender or rtcrtpreceiver in response to a call to the rtcrtpsender.getcapabilities() or rtcrtpreceiver.getcapabilities() static functions, both of which return an array of rtcrtpcapabilities objects. an rtcrtpcapabilities object contains an array of objects conforming to rtcrtpcodeccapability (each describing the capabilities of one codec) and an array of the supported rtp header extensions for that codec. properties codecs an array of rtcrtpcodeccapability objects, each describing one of the codecs supported by the rtcrtpsender or rtcrtpreceiver. there are some special entries in this array, described below in the section the codecs array. headerextensions an array of objects conforming to t...
...its properties are: the browser will only report distinct capability combinations separately. if two sets of capabilities can be described as one, they will be. ...
...s supported by the rtcrtpsender or rtcrtpreceiver. there are some special entries in this array, described below in the section the codecs array. headerextensions an array of objects conforming to the rtcrtpheaderextensioncapability dictionary. each object contains a single domstring, uri, specifying the uri of the header extension, as described in rfc 5285. description the codecs array the codecs array is an array of objects conforming to the dictionary rtcrtpcodeccapability. each of these objects describes a single codec and its basic capabilities. its properties are: the browser will only report distinct capability combinations separately. ...
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Enc Dec MAC Using Key Wrap CertReq PKCS10 CSR
ta = null; /* read in the entire file specified with the -i argument */ rv = readderfromfile(&certder, infilename, ascii); if (rv != secsuccess) { pr_fprintf(pr_stderr, "unable to read input file %s : %s\n", infilename, port_errortostring(rv)); goto cleanup; } /* read in an ascii cert and return a certcertificate */ cert = cert_decodecertfrompackage((char *)certder.data, certder.len); if (!cert) { pr_fprintf(pr_stderr, "could not obtain certificate from file\n"); rv = secfailure; goto cleanup; } /* create a cert trust */ trust = (certcerttrust *)port_zalloc(sizeof(certcerttrust)); if (!trust) { pr_fprintf(pr_stderr, "unable to allocate cert trust\n"); rv = se...
...== pr_success) { pr_delete(encryptedfilename); } /* read certificate from header file */ rv = readfromheaderfile(headerfilename, certenc, &data, pr_true); if (rv != secsuccess) { pr_fprintf(pr_stderr, "could not read certificate from header file\n"); goto cleanup; } /* read in an ascii cert and return a certcertificate */ cert = cert_decodecertfrompackage((char *)data.data, data.len); if (!cert) { pr_fprintf(pr_stderr, "could not obtain certificate from file\n"); rv = secfailure; goto cleanup; } /* extract the public key from certificate */ pubkey = cert_extractpublickey(cert); if (!pubkey) { pr_fprintf(pr_stderr, "could not get key from certificate\n"); rv = secfa...
...); rv = secfailure; goto cleanup; } /* read certificate from header file */ rv = readfromheaderfile(headerfilename, certvfy, &certdata, pr_true); if (rv != secsuccess) { pr_fprintf(pr_stderr, "could not read certificate from header file\n"); goto cleanup; } /* read in an ascii cert and return a certcertificate */ cert = cert_decodecertfrompackage((char *)certdata.data, certdata.len); if (!cert) { pr_fprintf(pr_stderr, "could not obtain certificate from file\n"); rv = secfailure; goto cleanup; } /* extract the public key from certificate */ pubkey = cert_extractpublickey(cert); if (!pubkey) { pr_fprintf(pr_stderr, "could not get key from certificate\n"); rv...
...adfromheaderfile(headerfilename, pad, &paditem, pr_true); if (rv != secsuccess) { pr_fprintf(pr_stderr, "could not retrieve pad detail from header file\n"); goto cleanup; } paddinglength = (unsigned int)paditem.data[0]; infilelength = filesize(encryptedfilename); /* read in an ascii cert and return a certcertificate */ cert = cert_decodecertfrompackage((char *)data.data, data.len); if (!cert) { pr_fprintf(pr_stderr, "could not obtain certificate from file\n"); rv = secfailure; goto cleanup; } /* find private key from certificate */ pvtkey = pk11_findkeybyanycert(cert, null); if (pvtkey == null) { fprintf(stderr, "couldn't find private key for cert\n"); rv = sec...

sample2
ertcerttrust *trust = null; certcertificate *cert = null; certder.data = null; /* read in the entire file specified with the -i argument */ rv = readderfromfile(&certder, infilename, ascii); if (rv != secsuccess) { pr_fprintf(pr_stderr, "unable to read input file %s : %s\n", infilename, port_errortostring(rv)); goto cleanup; } /* read in an ascii cert and return a certcertificate */ cert = cert_decodecertfrompackage((char *)certder.data, certder.len); if (!cert) { pr_fprintf(pr_stderr, "could not obtain certificate from file\n"); rv = secfailure; goto cleanup; } /* create a cert trust */ trust = (certcerttrust *)port_zalloc(sizeof(certcerttrust)); if (!trust) { pr_fprintf(pr_stderr, "unable to allocate cert trust\n"); rv = secfailure; goto cleanup; } rv = cert_decodetruststring(trust, trusts); ...
...elete it*/ if (pr_access(encryptedfilename, pr_access_exists) == pr_success) { pr_delete(encryptedfilename); } /* read certificate from header file */ rv = readfromheaderfile(headerfilename, certenc, &data, pr_true); if (rv != secsuccess) { pr_fprintf(pr_stderr, "could not read certificate from header file\n"); goto cleanup; } /* read in an ascii cert and return a certcertificate */ cert = cert_decodecertfrompackage((char *)data.data, data.len); if (!cert) { pr_fprintf(pr_stderr, "could not obtain certificate from file\n"); rv = secfailure; goto cleanup; } /* extract the public key from certificate */ pubkey = cert_extractpublickey(cert); if (!pubkey) { pr_fprintf(pr_stderr, "could not get key from certificate\n"); rv = secfailure; goto cleanup; } /* open the encrypted file for writing */ encfi...
...rintf(pr_stderr, "unable to open \"%s\" for writing.\n", headerfilename); rv = secfailure; goto cleanup; } /* read certificate from header file */ rv = readfromheaderfile(headerfilename, certvfy, &certdata, pr_true); if (rv != secsuccess) { pr_fprintf(pr_stderr, "could not read certificate from header file\n"); goto cleanup; } /* read in an ascii cert and return a certcertificate */ cert = cert_decodecertfrompackage((char *)certdata.data, certdata.len); if (!cert) { pr_fprintf(pr_stderr, "could not obtain certificate from file\n"); rv = secfailure; goto cleanup; } /* extract the public key from certificate */ pubkey = cert_extractpublickey(cert); if (!pubkey) { pr_fprintf(pr_stderr, "could not get key from certificate\n"); rv = secfailure; goto cleanup; } /* read signature from header file */ r...
...\n"); goto cleanup; } /* read padding from header file */ rv = readfromheaderfile(headerfilename, pad, &paditem, pr_true); if (rv != secsuccess) { pr_fprintf(pr_stderr, "could not retrieve pad detail from header file\n"); goto cleanup; } paddinglength = (unsigned int)paditem.data[0]; infilelength = filesize(encryptedfilename); /* read in an ascii cert and return a certcertificate */ cert = cert_decodecertfrompackage((char *)data.data, data.len); if (!cert) { pr_fprintf(pr_stderr, "could not obtain certificate from file\n"); rv = secfailure; goto cleanup; } /* find private key from certificate */ pvtkey = pk11_findkeybyanycert(cert, null); if (pvtkey == null) { fprintf(stderr, "couldn't find private key for cert\n"); rv = secfailure; goto cleanup; } /* open the out file to write */ outfile = pr_...

MediaRecorder() - Web APIs
syntax var mediarecorder = new mediarecorder(stream[, options]); parameters stream the mediastream that will be recorded. ...
... syntax var mediarecorder = new mediarecorder(stream[, options]); parameters stream the mediastream that will be recorded. ...
... syntax var mediarecorder = new mediarecorder(stream[, options]); parameters stream the mediastream that will be recorded. ...
...be recorded. this source media can come from a stream created using navigator.mediadevices.getusermedia() or from an <audio>, <video> or <canvas> element. options optional a dictionary object that can contain the following properties: mimetype: a mime type specifying the format for the resulting media; you may simply specify the container format (the browser will select its preferred codecs for audio and/or video), or you may use the codecs parameter and/or the profiles parameter to provide detailed information about which codecs to use and how to configure them. applications can check in advance if a mimetype is supported by the user agent by calling mediarecorder.istypesupported(). audiobitspersecond: the chosen bitrate for the audio component of the media. videobitspersecond...

MediaRecorder.mimeType - Web APIs
the mimetype read-only property returns the mime media type that was specified when creating the mediarecorder object, or, if none was specified, which was chosen by the browser. this is the file format of the file that would result from writing all of the recorded data to disk. keep in mind that not all codecs are supported by a given container; if you write media using a codec that is not supported by a given media container, the resulting file may not work reliably if at all when you try to play it back. see our media type and format guide for information about container and codec support across browsers. note: the term "mime type" is officially considered to be historical; these strings are now officially known as media types. mdn web docs content uses the terms interchangeably. synta...
... the media type strings are standardized by the internet assigned numbers authority (iana). for their official list of defined media type strings, see the article media types on the iana site. ...
...wsers. note: the term "mime type" is officially considered to be historical; these strings are now officially known as media types. mdn web docs content uses the terms interchangeably. syntax var mimetype = mediarecorder.mimetype value the mime media type which describes the format of the recorded media, as a domstring. this string may include the codecs parameter, giving details about the codecs and the codec configurations used by the media recorder. the media type strings are standardized by the internet assigned numbers authority (iana). for their official list of defined media type strings, see the article media types on the iana site. see also media types to learn more about media types and how they're used in web content and by web browers. example ... if (navigator.mediadevic...
...er; m.mimetype; // would return 'video/mp4' ... }) .catch(function(error) { console.log(error.message); }); changing line 14 to the following causes mediarecorder to try to use avc constrained baseline profile level 4 for video and aac-lc (low complexity) for audio, which is good for mobile and other possible resource-constrained situations. mimetype: 'video/mp4; codecs="avc1.424028, mp4a.40.2"' assuming this configuration is acceptable to the user agent, the value returned later by m.mimetype would then be video/mp4; codecs="avc1.424028, mp4a.40.2". specifications specification status comment mediastream recordingthe definition of 'mediarecorder.mimetype' in that specification. working draft initial definition ...

Writing textual data - Archive of obsolete content
will fail with ns_error_loss_of_significant_data. versions before gecko 1.8 firefox 1.0.x, mozilla 1.7.x and older versions do not support nsiconverteroutputstream. alternative ways usable from javascript do not support character encodings that use embedded nulls (such as utf-16 and utf-32). ...
...ve ways usable from javascript do not support character encodings that use embedded nulls (such as utf-16 and utf-32). they work by manually converting the string you want to write to a byte sequence using nsiscriptableunicodeconverter, and writing that to the stream. here's an example: // first, get and initialize the converter var converter = components.classes["@mozilla.org/intl/scriptableunicodeconverter"] .createinstance(components.interfaces.nsiscriptableunicodeconverter); converter.charset = /* the character encoding you want, using utf-8 for this example */ "utf-8"; now you can convert and write to the stream: // this code assumes that os is your nsioutputstream // your_string here is the string you want to write. var chunk = converter.convertfromunicode(y...
...it, but for those that do calling this is important for proper output. var fin = converter.finish(); if (fin.length > 0) os.write(fin, fin.length); os.close(); converting a string into a stream sometimes, it is useful to convert a string into a stream, for example for uploading it using nsiuploadchannel. the example here requires gecko 1.8 (firefox 1.5, seamonkey 1.0). nsiscriptableunicodeconverter has a simple method to do that: // first, get and initialize the converter var converter = components.classes["@mozilla.org/intl/scriptableunicodeconverter"] .createinstance(components.interfaces.nsiscriptableunicodeconverter); converter.charset = /* the charset you want to use. using utf-8 in this example */ "utf-8"; // now, convert a string to an nsiinputstre...

Aprender y obtener ayuda - Learn web development
videos for quick explanations of concepts and new features, but it is really up to you what you prefer to learn from. there is no right and wrong answer here. interactive code playgrounds you might be the kind of person that prefers minimal instructions and would prefer to jump straight in and start playing with code. this is also a reasonable approach, and some learning sites tend to favor it. codecademy for example is a learning site where the tutorials mainly consist of interactive code editors where you have to directly write code and see if the desired result was achieved. many mdn web docs reference pages provide interactive examples too, where you can alter the code and see how the live result changes. and there is also nothing wrong with creating your own code examples on your comput...
...cater to beginners. ...
...eginners. ...

Certificate functions
gecerttrust mxr 3.2 and later cert_checkcertvalidtimes mxr 3.2 and later cert_checknamespace mxr 3.12 and later cert_checkcertusage mxr 3.3 and later cert_comparename mxr 3.2 and later cert_comparevaliditytimes mxr 3.11 and later cert_completecrldecodeentries mxr 3.6 and later cert_convertanddecodecertificate mxr 3.9.3 and later cert_copyname mxr 3.4 and later cert_copyrdn mxr 3.5 and later cert_createava mxr 3.2.1 and later cert_createcertificate mxr 3.5 and later cert_createcertificaterequest mxr 3.2 and later cert_createname mxr 3.2.1 and later cert_createocspcertid mxr 3.6 and l...
...xr 3.5 and later cert_crlcacherefreshissuer mxr 3.7 and later cert_decodealtnameextension mxr 3.10 and later cert_decodeauthinfoaccessextension mxr 3.10 and later cert_decodeauthkeyid mxr 3.10 and later cert_decodeavavalue mxr 3.4 and later cert_decodebasicconstraintvalue mxr 3.2 and later cert_decodecertfrompackage mxr 3.4 and later cert_decodecertificatepoliciesextension mxr 3.2 and later cert_decodecertpackage mxr 3.2 and later cert_decodecrldistributionpoints mxr 3.10 and later cert_decodedercrl mxr 3.2 and later cert_decodedercrlwithflags mxr 3.6 and later cert_decodegeneralname mxr 3.4 and late...
...ficate mxr 3.2 and later cert_dupcertlist mxr 3.2 and later cert_enableocspchecking mxr 3.2 and later cert_encodealtnameextension mxr 3.7 and later cert_encodeandaddbitstrextension mxr 3.5 and later cert_encodeauthkeyid mxr 3.5 and later cert_encodebasicconstraintvalue mxr 3.5 and later cert_encodecertpoliciesextension mxr 3.12 and later cert_encodecrldistributionpoints mxr 3.5 and later cert_encodegeneralname mxr 3.4 and later cert_encodeinfoaccessextension mxr 3.12 and later cert_encodeinhibitanyextension mxr 3.12 and later cert_encodenoticereference mxr 3.12 and later cert_encodeocsprequest mxr ...

NSS functions
gecerttrust mxr 3.2 and later cert_checkcertvalidtimes mxr 3.2 and later cert_checknamespace mxr 3.12 and later cert_checkcertusage mxr 3.3 and later cert_comparename mxr 3.2 and later cert_comparevaliditytimes mxr 3.11 and later cert_completecrldecodeentries mxr 3.6 and later cert_convertanddecodecertificate mxr 3.9.3 and later cert_copyname mxr 3.4 and later cert_copyrdn mxr 3.5 and later cert_createava mxr 3.2.1 and later cert_createcertificate mxr 3.5 and later cert_createcertificaterequest mxr 3.2 and later cert_createname mxr 3.2.1 and later cert_createocspcertid mxr 3.6 and l...
...xr 3.5 and later cert_crlcacherefreshissuer mxr 3.7 and later cert_decodealtnameextension mxr 3.10 and later cert_decodeauthinfoaccessextension mxr 3.10 and later cert_decodeauthkeyid mxr 3.10 and later cert_decodeavavalue mxr 3.4 and later cert_decodebasicconstraintvalue mxr 3.2 and later cert_decodecertfrompackage mxr 3.4 and later cert_decodecertificatepoliciesextension mxr 3.2 and later cert_decodecertpackage mxr 3.2 and later cert_decodecrldistributionpoints mxr 3.10 and later cert_decodedercrl mxr 3.2 and later cert_decodedercrlwithflags mxr 3.6 and later cert_decodegeneralname mxr 3.4 and late...
...ficate mxr 3.2 and later cert_dupcertlist mxr 3.2 and later cert_enableocspchecking mxr 3.2 and later cert_encodealtnameextension mxr 3.7 and later cert_encodeandaddbitstrextension mxr 3.5 and later cert_encodeauthkeyid mxr 3.5 and later cert_encodebasicconstraintvalue mxr 3.5 and later cert_encodecertpoliciesextension mxr 3.12 and later cert_encodecrldistributionpoints mxr 3.5 and later cert_encodegeneralname mxr 3.4 and later cert_encodeinfoaccessextension mxr 3.12 and later cert_encodeinhibitanyextension mxr 3.12 and later cert_encodenoticereference mxr 3.12 and later cert_encodeocsprequest mxr ...

MediaRecorder.isTypeSupported - Web APIs
n value true if the mediarecorder implementation is capable of recording blob objects for the specified mime type. recording may still fail if there are insufficient resources to support the recording and encoding process. if the value is false, the user agent is incapable of recording the specified format. ...
...ementation is capable of recording blob objects for the specified mime type. recording may still fail if there are insufficient resources to support the recording and encoding process. if the value is false, the user agent is incapable of recording the specified format. ...
...bjects for the specified mime type. recording may still fail if there are insufficient resources to support the recording and encoding process. if the value is false, the user agent is incapable of recording the specified format. example var types = ["video/webm", "audio/webm", "video/webm\;codecs=vp8", "video/webm\;codecs=daala", "video/webm\;codecs=h264", "audio/webm\;codecs=opus", "video/mpeg"]; for (var i in types) { console.log( "is " + types[i] + " supported? " + (mediarecorder.istypesupported(types[i]) ? "maybe!" : "nope :(")); } specifications specification status comment mediastream recordingthe definition of 'istypesupported()' in that specification. working draft initia...

MediaSource.MediaSource() - Web APIs
e object with no associated source buffers. syntax var mediasource = new mediasource(); parameters none. ...
.... syntax var mediasource = new mediasource(); parameters none. ...
... mediasource = new mediasource(); parameters none. example the following snippet is taken from a simple example written by nick desaulniers (view the full demo live, or download the source for further investigation.) var video = document.queryselector('video'); var asseturl = 'frag_bunny.mp4'; // need to be specific for blink regarding codecs // ./mp4info frag_bunny.mp4 | grep codec var mimecodec = 'video/mp4; codecs="avc1.42e01e, mp4a.40.2"'; if ('mediasource' in window && mediasource.istypesupported(mimecodec)) { var mediasource = new mediasource; //console.log(mediasource.readystate); // closed video.src = url.createobjecturl(mediasource); mediasource.addeventlistener('sourceopen', sourceopen); } else { console.error('unsupported mime type or codec: ', mimecodec); } ... ...

Media Capabilities API - Web APIs
different browsers support different media types and new media types are always being developed. with the media capabilities api, developers can ensure each user is getting the best bitrate and storage savings for their browser, device, and os capabilities. whether a device uses hardware or software decoding impacts how smooth and power efficient the video decoding is and how efficient the pla...
...pes are always being developed. with the media capabilities api, developers can ensure each user is getting the best bitrate and storage savings for their browser, device, and os capabilities. whether a device uses hardware or software decoding impacts how smooth and power efficient the video decoding is and how efficient the playback will be. the media capabilities api enables determining which codecs are supported and how performant a media file will be both in terms of smoothness and power efficiency. the media capabilities api provide more powerful features than say mediarecorder.istypesupported() or htmlmediaelement.canplaytype(), which only address general browser support, not performance. the api also provides abilities to access display property information such as supported color ga...
...rface's encodinginfo() and decodinginfo() methods. media capabilities information enables websites to enable adaptative streaming to alter the quality of content based on actual user-perceived quality, and react to a pick of cpu/gpu usage in real time. media capabilities interfaces mediacapabilities provides information about the decoding abilities of the device, system and browser based on codecs, profile, resolution, and bitrates. the information can be used to serve optimal media streams to the user and determine if playback should be smooth and power efficient . mediacapabilitiesinfo the interface of the promise returned by the mediacapabilities's encodinginfo() and decodinginfo() methods; returning whether the media configuration tested is supported, smooth, and powerefficient...

RTCRtpReceiver.getCapabilities() static function - Web APIs
you can, similarly, obtain the capabilities of rtcrtpsenders by calling the static function rtcrtpsender.getcapabilities(). syntax let rtpcapabilities = rtcrtpreceiver.getcapabilities(kind); parameters kind a domstring indicating the type of media for which you wish to get the device's capability to receive. all browsers support the primary media kinds: audio and video. ...
...ork when you actually try to use them. calling rtcrtpreceiver.getcapabilities() doesn't prime the browser in any way to handle media. nothing is loaded, fetched, or otherwise prepared. it's simply a means of determining what might be usable before starting to try to access media. because the set of capabilities available tend to be stable for a length of time (people don't install and uninstall codecs and the like very often), the media capabilities can in whole or in part provide a cross-origin method for identifying a user. for that reason, in privacy-sensitive contexts, the browser may choose to obscure the capabilities; this might be done, for example, by leaving out rarely-used codec configurations. example the function below returns a boolean indicating whether or not the device suppo...
...oolean indicating whether or not the device supports receiving h.264 video on a webrtc connection. since rtcrtpreceiver.getcapabilities() actually only indicates probable support, attempting to receive h.264 video might still fail even after getting a positive response from this function. function canreceiveh264() { let capabilities = rtcrtpreceiver.getcapabilities("video"); capabilities.codecs.foreach((codec) => { if (codec.mimetype === "video/h264") { return true; } }); return false; } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcrtpreceiver.getcapabilities()' in that specification. candidate recommendation ...

RTCRtpSender.getCapabilities() static function - Web APIs
you can, similarly, obtain the capabilities of rtcrtpreceivers by calling the static function rtcrtpreceiver.getcapabilities(). syntax let rtpcapabilities = rtcrtpsender.getcapabilities(kind); parameters kind a domstring indicating the type of media for which you wish to get the sender's capability to receive. all browsers support the primary media kinds: audio and video. return value an rtcrtpcapabilities object stating what capabilities the browser has for sending the specified media kind over an rtcpeerconnection. ...
... work when you actually try to use them. calling rtcrtpsender.getcapabilities() doesn't prime the browser in any way to handle media. nothing is loaded, fetched, or otherwise prepared. it's simply a means of determining what might be usable before starting to try to access media. because the set of capabilities available tend to be stable for a length of time (people don't install and uninstall codecs and the like very often), the media capabilities can in whole or in part provide a cross-origin method for identifying a user. for that reason, in privacy-sensitive contexts, the browser may choose to obscure the capabilities; this might be done, for example, by leaving out rarely-used codec configurations. example the function below returns a boolean indicating whether or not the device suppo...
...e the function below returns a boolean indicating whether or not the device supports sending h.264 video on an rtcrtpsender. since rtcrtpsender.getcapabilities() actually only indicates probable support, h.264 support might still fail even after getting a positive response from this function. function cansendh264() { let capabilities = rtcrtpsender.getcapabilities("video"); capabilities.codecs.foreach((codec) => { if (codec.mimetype === "video/h264") { return true; } }); return false; } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcrtpsender.getcapabilities()' in that specification. candidate recommendation ...

RTCRtpStreamStats - Web APIs
ed rtcrtpstreamstats until a specification update in the spring of 2017. check the browser compatibility table to know if and when the name change was implemented in specific browsers. properties the rtcrtpstreamstats dictionary is based on rtcstats, and inherits its properties. in addition, some or all of the following properties are available. standard fields included for all media types codecid a domstring which uniquely identifies the object which was inspected to produce the rtccodecstats object associated with this rtp stream. kind a domstring whose value is "audio" if the associated mediastreamtrack is audio-only or "video" if the track contains video. this value will match that of the media type indicated by rtccodecstats.codec, as well as the track's kind property. previously...
...quantization parameter (qp) values associated with every frame received to date on the video track described by this rtcrtpstreamstats object. in general, the higher this number is, the more heavily compressed the video track was. combined with rtcreceivedrtpstreamstats.framesdecoded or rtcsentrtpstreamstats.framesencoded, you can approximate the average qp over those frames, keeping in mind that codecs often vary the quantizer values even within frames. also keep in mind that the values of qp can vary from codec to codec, so this value is only potentially useful when compared against the same codec. slicount the number of times the receiver notified the sender that one or more consecutive (in scan order) encoded video macroblocks have been lost or corrupted; this notification is sent by the ...
... specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcrtpstreamstats' in that specification. ...

WebRTC API - Web APIs
party software. webrtc consists of several interrelated apis and protocols which work together to achieve this. the documentation you'll find here will help you understand the fundamentals of webrtc, how to set up and use both data and media connections, and more. interoperability because implementations of webrtc are still evolving, and because each browser has different levels of support for codecs and webrtc features, you should strongly consider making use of the adapter.js library provided by google before you begin to write your code. adapter.js uses shims and polyfills to smooth over the differences among the webrtc implementations across the environments supporting it. adapter.js also handles prefixes and other naming differences to make the entire webrtc development process easier,...
...iation while allowing both sides to be either the offerer or the answerer, without significant coding needed to differentiate the two. signaling and two-way video calling a tutorial and example which turns a websocket-based chat system created for a previous example and adds support for opening video calls among participants. the chat server's websocket connection is used for webrtc signaling. codecs used by webrtc a guide to the codecs which webrtc requires browsers to support as well as the optional ones supported by various popular browsers. included is a guide to help you choose the best codecs for your needs. using webrtc data channels this guide covers how you can use a peer connection and an associated rtcdatachannel to exchange arbitrary data between two peers. using dtmf with we...
... sample the rtcdatachannel interface is a feature which lets you open a channel between two peers over which you may send and receive arbitrary data. the api is intentionally similar to the websocket api, so that the same programming model can be used for each. ...

<audio>: The Embed Audio element - HTML: Hypertext Markup Language
data, but data is unexpectedly not forthcoming. suspend media data loading has been suspended. timeupdate the time indicated by the currenttime attribute has been updated. volumechange the volume has changed. waiting playback has stopped because of a temporary lack of data usage notes browsers don't all support the same file types and audio codecs; you can provide multiple sources inside nested <source> elements, and the browser will then use the first one it understands: <audio controls> <source src="myaudio.mp3" type="audio/mpeg"> <source src="myaudio.ogg" type="audio/ogg"> <p>your browser doesn't support html5 audio. here is a <a href="myaudio.mp4">link to the audio</a> instead.</p> </audio> we offer a substantive and thor...
...sted <source> elements, and the browser will then use the first one it understands: <audio controls> <source src="myaudio.mp3" type="audio/mpeg"> <source src="myaudio.ogg" type="audio/ogg"> <p>your browser doesn't support html5 audio. here is a <a href="myaudio.mp4">link to the audio</a> instead.</p> </audio> we offer a substantive and thorough guide to media file types and the audio codecs that can be used within them. also available is a guide to the codecs supported for video. other usage notes: if you don't specify the controls attribute, the audio player won't include the browser's default controls. you can, however, create your own custom controls using javascript and the htmlmediaelement api. to allow precise control over your audio content, htmlmediaelements fire many ...
...av" type="audio/wav"> your browser does not support the <code>audio</code> element. </audio> <audio> with multiple <source> elements this example includes multiple <source> elements. the browser tries to load the first source element (opus) if it is able to play it; if not it falls back to the second (vorbis) and finally back to mp3: <audio controls> <source src="foo.opus" type="audio/ogg; codecs=opus"/> <source src="foo.ogg" type="audio/ogg; codecs=vorbis"/> <source src="foo.mp3" type="audio/mpeg"/> </audio> accessibility concerns audio with spoken dialog should provide both captions and transcripts that accurately describe its content. captions, which are specified using webvtt, allow people who are experiencing hearing loss to understand an audio recording's content as the recordi...

<video>: The Video Embed element - HTML: Hypertext Markup Language
urce> elements, and the browser will then use the first one it understands. <video controls> <source src="myvideo.mp4" type="video/mp4"> <source src="myvideo.webm" type="video/webm"> <p>your browser doesn't support html5 video. here is a <a href="myvideo.mp4">link to the video</a> instead.</p> </video> we offer a substantive and thorough guide to media file types and the guide to the codecs supported for video. also available is a guide to audio codecs that can be used with them. other usage notes: if you don't specify the controls attribute, the video won't include the browser's default controls; you can create your own custom controls using javascript and the htmlmediaelement api. see creating a cross-browser video player for more details. to allow precise control over your ...
...ith your favorite video player! </video> until the video starts playing, the image provided in the poster attribute is displayed in its place. if the browser doesn't support video playback, the fallback text is displayed. multiple sources example this example builds on the last one, offering three different sources for the media; this allows the video to be watched regardless of which video codecs are supported by the browser. <!-- using multiple sources as fallbacks for a video tag --> <!-- 'elephants dream' by orange open movie project studio, licensed under cc-3.0, hosted by archive.org --> <!-- poster hosted by wikimedia --> <video width="620" controls poster="https://upload.wikimedia.org/wikipedia/commons/e/e8/elephants_dream_s5_both.jpg" > <source src="https://archive.org/d...
..."https://archive.org/download/elephantsdream/ed_hd.avi" type="video/avi"> your browser doesn't support html5 video tag. </video> first webm is tried. if that can't be played, then mp4 is tried. finally, ogg is tried. a fallback message is displayed if the video element isn't supported, but not if all sources fail. some media file types let you provide more specific information using the codecs parameter as part of the file's type string. a relatively simple example is video/webm; codecs="vp8, vorbis", which says that the file is a webm video using vp8 for its video and vorbis for audio. server support for video if the mime type for the video is not set correctly on the server, the video may not show or show a gray box containing an x (if javascript is enabled). if you use apache we...

Chapter 5: Let's build a Firefox extension - Archive of obsolete content
content for _writefile method _writefile: function(afile, adata) { // init stream var stream = components.classes["@mozilla.org/network/safe-file-output-stream;1"]. ...
... var stream = components.classes["@mozilla.org/network/safe-file-output-stream;1"]. createinstance(components.interfaces.nsifileoutputstream); stream.init(afile, 0x02 | 0x08 | 0x20, 0600, 0); // convert to utf-8 var converter = components.classes["@mozilla.org/intl/scriptableunicodeconverter"]. ...

Chapter 4: Using XPCOM—Implementing advanced processes - Archive of obsolete content
converterinputstream.default_replacement_character); converterstream.writestring(string); converterstream.close(); filestream.close(); character encoding conversion firefox's internal representation of all text is in unicode. however, in some cases, you may want to be able to process text using another encoding. ...
...y want to be able to process text using another encoding. in these cases, you can easily convert between encodings using the nsiscriptableunicodeconverter. converting from unicode to other encodings listing 17 shows how to convert from text saved as unicode to euc-jp encoding. listing 17: converting text from unicode to euc-jp var converter = components.classes['@mozilla.org/intl/scriptableunicodeconverter'] .getservice(components.interfaces.nsiscriptableunicodeconverter); converter.charset = 'euc-jp'; var unicode_str = '\u5909\u63db\u30c6\u30b9\u30c8'; var eucjp_str = converter.convertfromunicode(unicode_str); converting from other encodings to unicode listing 18 shows how to do the reverse, converting from text saved as iso-2022-jp to unicode. listing 18: converting tex...

Index - MDN Web Docs Glossary: Definitions of Web-related terms
80 closure codingscripting, glossary the binding which defines the scope of execution. in javascript, functions create a closure context. 81 code splitting glossary, reference, web performance, code splitting, latency code splitting is the splitting of code into various bundles or components which can then be loaded on demand or in parallel. 82 codec glossary, webmechanics a codec (a blend word derived from "coder-decoder") is a program, algorithm, or device that encodes or decodes a data stream. a given codec knows how to handle a specific encoding or compression technology. 83 compile codingscripting, glossary compiling is the process of transforming a computer program written in a given language into an equiv...
...rol, multi-homing, and other features to improve reliability and stability of the connection. it's used for sending traditional telephone calls over the internet, but is also used for webrtc data. 382 sdp advanced, collaborating, glossary, infrastructure, protocol, webrtc sdp (session description protocol) is the standard describing a peer-to-peer connection. sdp contains the codec, source address, and timing information of audio and video. 383 seo glossary, intro, seo, search, webmechanic seo (search engine optimization) is the process of making a website more visible in search results, also termed improving search rankings. 384 sgml codingscripting, composing, glossary, sgml the standard generalized markup language (sgml) is an is...

imgIEncoder
lease add a summary to this article. ...
... data list of bytes in the format specified by inputformat. length missing description width width in pixels. height height in pixels. stride the number of bytes per row in the image. normally (width*3) or (width*4), depending on your input format, but some data uses padding at the end of each row, which would be extra. frameformat missing description frameoptions missing description encodeclipboardimage() obsolete since gecko 1.9 (firefox 3) void encodeclipboardimage( in nsiclipboardimage aclipboardimage, out nsifile aimagefile ); parameters aclipboardimage missing description aimagefile missing description endimageencode() void endimageencode(); parameters none. initfromdata() void initfromdata( [array, size_is(length), const] in pruint8 data, in unsign...

nsIContentSniffer
to read content from adata. ...
...ry { // this pref has been removed, see bug 910192 charset = services.prefs.getcomplexvalue("intl.charset.default", ci.nsipreflocalizedstring).data; } catch (e) { } let conv = cc["@mozilla.org/intl/scriptableunicodeconverter"] .createinstance(ci.nsiscriptableunicodeconverter); conv.charset = charset; try { let str = conv.convertfrombytearray(adata, alength); if (str.substring(0, 5) == "%pdf-") return "application/pdf"; // we detected a pdf file } catch (e) { // try to get information from arequest } ...

nsICryptoHash
the hash as a base 64 encoded string. in this example we take the binary data and produce a hexadecimal string of the result, as is commonly output by hashing programs. computing the hash of a string another common operation is computing the hash value of a string. since hash functions are computed over bytes, you will first need to convert the string to a series of bytes using nsiscriptableunicodeconverter and a unicode encoding that you specify. note: different encodings will produce different hash values! you should always use the same encoding if you intend to compare results. the example below shows how to convert a string to bytes in the utf-8 encoding, and then compute the sha256 hash of it. the result is computed as a hex string as in the previous example. in this example, we use ...
...string to bytes in the utf-8 encoding, and then compute the sha256 hash of it. the result is computed as a hex string as in the previous example. in this example, we use the update() method to pass an array of bytes to be hashed. as in the previous example, we convert the binary result to a hex string. var str = "hello world"; var converter = components.classes["@mozilla.org/intl/scriptableunicodeconverter"]. createinstance(components.interfaces.nsiscriptableunicodeconverter); // we use utf-8 here, you can choose other encodings. converter.charset = "utf-8"; // result is an out parameter, // result.value will contain the array length var result = {}; // data is an array of bytes var data = converter.converttobytearray(str, result); var ch = components.classes["@mozilla.org/security/has...

nsIDOMHTMLSourceElement
ly manipulating this value after the page has loaded has no effect on the containing element; instead, change the src attribute of that element (audio or video) instead. type domstring the type attribute gives the type of the media resource, to help the user agent determine if it can play this media resource before fetching it. if specified, its value must be a valid mime type. the codecs parameter may be specified and might be necessary to specify exactly how the resource is encoded. media unimplemented domstring the media attribute gives the intended media type of the media resource, to help the user agent determine if this media resource is useful to the user before fetching it. its value must be a valid media query. example <video controls> <source sr...
...d might be necessary to specify exactly how the resource is encoded. media unimplemented domstring the media attribute gives the intended media type of the media resource, to help the user agent determine if this media resource is useful to the user before fetching it. its value must be a valid media query. example <video controls> <source src="foo.webm" type="video/webm; codecs=&quot;vp9, opus&quot;"> your browser does not support the <code>video</code> element. </video> this example specifies that a video object with playback controls should be drawn, playing the file "foo.ogg". the type is specified as "video/ogg", and the source element also specifies that the video file uses the dirac and speex codecs. ...

MediaCapabilities.encodingInfo() - Web APIs
ither an audioconfiguration or videoconfiguration dictionary. ...
...because the type is not video or audio, the contenttype is not a valid codec mime type, or any other error in the media configuration passed to the method, including omitting any of the media encoding configuration elements. example //create media configuration to be tested const mediaconfig = { type : 'record', // or 'transmission' video : { contenttype : "video/webm;codecs=vp8.0", // valid content type width : 1920, // width of the video height : 1080, // height of the video bitrate : 120000, // number of bits used to encode 1s of video framerate : 48 // number of frames making up that 1s. } }; // check support and performance navigator.mediacapabilities.encodinginfo(mediaconfig).then(result => { console.log('this...

MediaRecorder: dataavailable event - Web APIs
umentation for the event handler property: ondataavailable. examples using addeventlistener to listen for dataavailable events: ... var chunks = []; mediarecorder.addeventlistener('stop', (event) => { console.log("data available after mediarecorder.stop() called."); var audio = document.createelement('audio'); audio.controls = true; var blob = new blob(chunks, { 'type' : 'audio/ogg; codecs=opus' }); var audiourl = window.url.createobjecturl(blob); audio.src = audiourl; console.log("recorder stopped"); }); mediarecorder.addeventlistener('dataavailable', (event) => { chunks.push(event.data); }); ... the same, but using the ondataavailable event handler property: ... var chunks = []; mediarecorder.onstop = function(e) { console.log("data available after mediarecorder.
...eventlistener('dataavailable', (event) => { chunks.push(event.data); }); ... the same, but using the ondataavailable event handler property: ... var chunks = []; mediarecorder.onstop = function(e) { console.log("data available after mediarecorder.stop() called."); var audio = document.createelement('audio'); audio.controls = true; var blob = new blob(chunks, { 'type' : 'audio/ogg; codecs=opus' }); var audiourl = window.url.createobjecturl(blob); audio.src = audiourl; console.log("recorder stopped"); } mediarecorder.ondataavailable = function(e) { chunks.push(e.data); } ... specifications specification status mediastream recording working draft ...

MediaRecorder - Web APIs
top event, which occurs when media recording ends, either when the mediastream ends — or after the mediarecorder.stop() method is called. events listen to these events using addeventlistener() or by assigning an event listener to the oneventname property of this interface. error fired when an error occurs: for example because recording wasn't allowed or was attempted using an unsupported codec. also available via the onerror property. example if (navigator.mediadevices) { console.log('getusermedia supported.'); var constraints = { audio: true }; var chunks = []; navigator.mediadevices.getusermedia(constraints) .then(function(stream) { var mediarecorder = new mediarecorder(stream); visualize(stream); record.onclick = function() { mediarecorder.start(...
...t.add('clip'); audio.setattribute('controls', ''); deletebutton.innerhtml = "delete"; cliplabel.innerhtml = clipname; clipcontainer.appendchild(audio); clipcontainer.appendchild(cliplabel); clipcontainer.appendchild(deletebutton); soundclips.appendchild(clipcontainer); audio.controls = true; var blob = new blob(chunks, { 'type' : 'audio/ogg; codecs=opus' }); chunks = []; var audiourl = url.createobjecturl(blob); audio.src = audiourl; console.log("recorder stopped"); deletebutton.onclick = function(e) { evttgt = e.target; evttgt.parentnode.parentnode.removechild(evttgt.parentnode); } } mediarecorder.ondataavailable = function(e) { chunks.push(e.data); } }) .catch(fu...

MediaSource.readyState - Web APIs
attached to a media element but the stream has been ended via a call to mediasource.endofstream(). syntax var myreadystate = mediasource.readystate; value a domstring. ...
...ers (view the full demo live, or download the source for further investigation.) if ('mediasource' in window && mediasource.istypesupported(mimecodec)) { var mediasource = new mediasource; //console.log(mediasource.readystate); // closed video.src = url.createobjecturl(mediasource); mediasource.addeventlistener('sourceopen', sourceopen); } else { console.error('unsupported mime type or codec: ', mimecodec); } function sourceopen (_) { //console.log(this.readystate); // open var mediasource = this; var sourcebuffer = mediasource.addsourcebuffer(mimecodec); fetchab(asseturl, function (buf) { sourcebuffer.addeventlistener('updateend', function (_) { mediasource.endofstream(); video.play(); //console.log(mediasource.readystate); // ended }); sourceb...

RTCInboundRtpStreamStats.qpSum - Web APIs
est you can do is make an estimate. you can, for example, use the value of rtcreceivedrtpstreamstats.framesdecoded if receiving the media or rtcsentrtpstreamstats.framesencoded if sending it to get the number of frames handled so far, and compute an average from there. see calculating average quantization below for a function that does this. ...
..., for example, use the value of rtcreceivedrtpstreamstats.framesdecoded if receiving the media or rtcsentrtpstreamstats.framesencoded if sending it to get the number of frames handled so far, and compute an average from there. see calculating average quantization below for a function that does this. also, the exact meaning of the qp value depends on the codec being used. for example, for the vp8 codec, the qp value can be anywhere from 1 to 127 and is found in the frame header element "y_ac_qi", whose value is defined in rfc 6386: 19.2. h.264 uses a qp which ranges from 0 to 51; in this case, it's an index used to derive a scaling matrix used during the quantization process. additionally, qp is not likely to be the only parameter the codec uses to adjust the compression. see the individual cod...

RTCOutboundRtpStreamStats.qpSum - Web APIs
so it's difficult to know for certain how substantial the compression is. the best you can do is make an estimate. you can use the value of rtcsentrtpstreamstats.framesencoded to get the number of frames that have been encoded so far, and compute an average from there. see calculating average quantization below for a function that does this. ...
...ubstantial the compression is. the best you can do is make an estimate. you can use the value of rtcsentrtpstreamstats.framesencoded to get the number of frames that have been encoded so far, and compute an average from there. see calculating average quantization below for a function that does this. also, the exact meaning of the qp value depends on the codec being used. for example, for the vp8 codec, the qp value can be anywhere from 1 to 127 and is found in the frame header element "y_ac_qi", whose value is defined in rfc 6386: 19.2. h.264 uses a qp which ranges from 0 to 51; in this case, it's an index used to derive a scaling matrix used during the quantization process. additionally, qp is not likely to be the only parameter the codec uses to adjust the compression. see the individual cod...

RTCPeerConnection.createOffer() - Web APIs
the createoffer() method of the rtcpeerconnection interface initiates the creation of an sdp offer for the purpose of starting a new webrtc connection to a remote peer. the sdp offer includes information about any mediastreamtracks already attached to the webrtc session, codec, and options supported by the browser, and any candidates already gathered by the ice agent, for the purpose of being sent over the signaling channel to a potential peer to request a connection or to update the configuration of an existing connection. the return value is a promise which, when the offer has been created, is resolved with a rtcsessiondescription object containing the newly-created offer. syntax apromise = mypeerconnection.createoffer([options]); mypeerconnection.createoffer(successcallback, failurecall...
..."inactive"). in modern implementations, the presence of this member with a value true, will ensure there is at least one transceiver set to receive video that has not been stopped, and if there isn't one, one will be created. note: you shouldn't use this legacy property. instead, use rtcrtptransceiver to control whether or not to accept incoming video. voiceactivitydetection optional some codecs and hardware are able to detect when audio begins and ends by watching for "silence" (or relatively low sound levels) to occur. this reduces network bandwidth used for audio by only sending audio data when there's actually something to broadcast. however, in some cases, this is unwanted. for example, in the case of music or other non-voice transmission, this can cause loss of important low-volum...

RTCRtpEncodingParameters - Web APIs
an instance of the webrtc api's rtcrtpencodingparameters dictionary describes a single configuration of a codec for an rtcrtpsender. it's used in the rtcrtpsendparameters describing the configuration of an rtp sender's encodings; rtcrtpdecodingparameters is used to describe the configuration of an rtp receiver's encodings. properties active if true, the described encoding is currently actively being used. that is, for rtp senders, the encoding is currently being used to send data, while for receivers, the encoding is being used to decode received data. the default value is true. ...
...uration of an rtp sender's encodings; rtcrtpdecodingparameters is used to describe the configuration of an rtp receiver's encodings. properties active if true, the described encoding is currently actively being used. that is, for rtp senders, the encoding is currently being used to send data, while for receivers, the encoding is being used to decode received data. the default value is true. codecpayloadtype when describing a codec for an rtcrtpsender, codecpayloadtype is a single 8-bit byte (or octet) specifying the codec to use for sending the stream; the value matches one from the owning rtcrtpparameters object's codecs parameter. this value can only be set when creating the transceiver; after that, this value is read only. dtx only used for an rtcrtpsender whose kind is audio, this ...

RTCRtpSendParameters.encodings - Web APIs
object providing configuration settings for the encoder being used for the rtcrtpsender's track. ...
...provide settings and parameters that describe and configure a single codec that could be used to encode the track. each object's properties are: active if true, the described encoding is currently actively being used. that is, for rtp senders, the encoding is currently being used to send data, while for receivers, the encoding is being used to decode received data. the default value is true. codecpayloadtype when describing a codec for an rtcrtpsender, codecpayloadtype is a single 8-bit byte (or octet) specifying the codec to use for sending the stream; the value matches one from the owning rtcrtpparameters object's codecs parameter. this value can only be set when creating the transceiver; after that, this value is read only. dtx only used for an rtcrtpsender whose kind is audio, this ...

RTCRtpSender.replaceTrack() - Web APIs
with which to replace the rtcrtpsender's current source track. the new track's kind must be the same as the current track's, or the replace track request will fail. return value a promise which is fulfilled once the track has been successfully replaced. the promise is rejected if the track cannot be replaced for any reason; this is commonly because the change would require renegotiation of the codec, which is not allowed (see things that require negotiation). if newtrack was omitted or was null, replacetrack() simply stops the sender. no negotiation is required in this case. when the promise is fulfilled, the fulfillment handler receives a value of undefined. exceptions if the returned promise is rejected, one of the following exceptions is provided to the rejection handler: invalidmo...
...all track replacements require renegotiation. in fact, even changes that seem huge can be done without requiring negotation. here are the changes that can trigger negotiaton: the new track has a resolution which is outside the bounds of the bounds of the current track; that is, the new track is either wider or taller than the current one. the new track's frame rate is high enough to cause the codec's block rate to be exceeded. the new track is a video track and its raw or pre-encoded state differs from that of the original track. the new track is an audio track with a different number of channels fom the original. media sources that have built-in encoders — such as hardware encoders — may not be able to provide the negotiated codec. software sources may not implement the negotiated c...

RTCRtpSender.setParameters() - Web APIs
ible. in other words, setparameters() updates the configuration of the rtp transmission as well as the encoding configuration for a specific media track on the webrtc connection. syntax var promise = rtcrtpsender.setparameters(parameters) parameters parameters an object conforming with the rtcrtpsendparameters dictionary, specifying options for the rtcrtpsender; these include potential codecs that could be use for encoding the sender's track. the available options are: degradationpreference specifies the preferred way the webrtc layer should handle optimizing bandwidth against quality in constrained-bandwidth situations; the value comes from the rtcdegradationpreference enumerated string type, and the default is balanced. encodings an array of rtcrtpencodingparameters objects...
...track. the available options are: degradationpreference specifies the preferred way the webrtc layer should handle optimizing bandwidth against quality in constrained-bandwidth situations; the value comes from the rtcdegradationpreference enumerated string type, and the default is balanced. encodings an array of rtcrtpencodingparameters objects, each specifying the parameters for a single codec that could be used to encode the track's media. priority a string from the rtcprioritytype enumerated type which indicates the encoding's priority. the default value is low. transactionid a string containing a unique id for the last set of parameters applied; this value is used to ensure that setparameters() can only be called to alter changes made by a specific previous call to getparameters...

RTCRtpStreamStats.qpSum - Web APIs
est you can do is make an estimate. you can, for example, use the value of rtcreceivedrtpstreamstats.framesdecoded if receiving the media or rtcsentrtpstreamstats.framesencoded if sending it to get the number of frames handled so far, and compute an average from there. see calculating average quantization below for a function that does this. ...
..., for example, use the value of rtcreceivedrtpstreamstats.framesdecoded if receiving the media or rtcsentrtpstreamstats.framesencoded if sending it to get the number of frames handled so far, and compute an average from there. see calculating average quantization below for a function that does this. also, the exact meaning of the qp value depends on the codec being used. for example, for the vp8 codec, the qp value can be anywhere from 1 to 127 and is found in the frame header element "y_ac_qi", whose value is defined in rfc 6386: 19.2. h.264 uses a qp which ranges from 0 to 51; in this case, it's an index used to derive a scaling matrix used during the quantization process. additionally, qp is not likely to be the only parameter the codec uses to adjust the compression. see the individual cod...

SourceBuffer.changeType() - Web APIs
syntax sourcebuffer.changetype(type); parameters type a domstring specifying the mime type that future buffers will conform to. return value undefined. ...
... syntax sourcebuffer.changetype(type); parameters type a domstring specifying the mime type that future buffers will conform to. return value undefined. ...

VideoConfiguration - Web APIs
eo file being tested when calling the mediacapabilities methods encodinginfo() and decodinginfo() to determine whether or not the described video configuration is supported, and how smoothly and how smoooth and power-efficient it can be handled. properties the videoconfiguration dictionary is made up of five video properties, including: contenttype: a valid video mime type. see our web video codec guide for types which may be supported. width: the width of the video. height: the height of the video. bitrate: the number of bits used to encode one second of the video file. framerate: the number of frames making up one second of video playback. examples // create media configuration to be tested const mediaconfig = { type : 'file', // see mediadecodingconfiguration and mediaencodi...
...e video. height: the height of the video. bitrate: the number of bits used to encode one second of the video file. framerate: the number of frames making up one second of video playback. examples // create media configuration to be tested const mediaconfig = { type : 'file', // see mediadecodingconfiguration and mediaencodingconfiguration video : { contenttype : "video/webm;codecs=vp8", // valid content type width : 800, // width of the video height : 600, // height of the video bitrate : 10000, // number of bits used to encode 1s of video framerate : 30 // number of frames making up that 1s. } }; specifications specification status comment media capabilitiesthe definition of 'videoconfiguration'...

Lifetime of a WebRTC session - Web APIs
by labeling it “michelle” and dropping it in a mailbox when you don't know her address. you need to look up her address and include it on the package, or she'll wind up wondering why you forgot her birthday again. this is where signaling comes in. signaling signaling is the process of sending control information between two devices to determine the communication protocols, channels, media codecs and formats, and method of data transfer, as well as any required routing information. the most important thing to know about the signaling process for webrtc: it is not defined in the specification. why, you may wonder, is something fundamental to the process of establishing a webrtc connection left out of the specification? the answer is simple: since the two devices have no way to directly c...
...changed during signaling there are three basic types of information that need to be exchanged during signaling: control messages used to set up, open, and close the communication channel, and to handle errors. information needed in order to set up the connection: the ip addressing and port information needed for the peers to be able to talk to one another. media capability negotiation: what codecs and media data formats can the peers understand? these need to be agreed upon before the webrtc session can begin. only once signaling has been successfully completed can the true process of opening the webrtc peer connection begin. it's worth noting that the signaling server does not actually need to understand or do anything with the data being exchanged through it by the two peers during s...

Signaling and video calling - Web APIs
. target the username of the person to receive the description (if the caller is sending the message, this specifies the callee, and vice-versa). sdp the sdp (session description protocol) string describing the local end of the connection from the perspective of the sender (or the remote end of the connection from the receiver's point of view). at this point, the two participants know which codecs and codec parameters are to be used for this call. they still don't know how to transmit the media data itself though. this is where interactive connectivity establishment (ice) comes in. exchanging ice candidates two peers need to exchange ice candidates to negotiate the actual connection between them. every ice candidate describes a method that the sending peer is able to use to communicate.
... begins to flow. each peer continues to send candidates until it runs out of options, even after the media has already begun to flow. this is done in hopes of identifying even better options than the one initially selected. if conditions change—for example the network connection deteriorates—one or both peers might suggest switching to a lower-bandwidth media resolution, or to an alternative codec. this triggers a new exchange of candidates, after which a another media format and/or codec change may take place. you can learn more about the codecs which webrtc requires browsers to support, which additional codecs are supported by which browsers, and how to choose the best codecs to use in the guide codecs used by webrtc. optionally, see rfc 8445: interactive connectivity establishment, sec...

Cross-browser audio basics - Developer guides
uffering events there are also a couple of events related to buffering: seeking the seeking event is fired when media is being sought. seeked seeked occurs when the seeking attribute changes to false. note: you can read more on buffering, seeking and time ranges elsewhere. browser support the following tables list basic audio support across desktop and mobile browsers, and what audio codecs are supported. desktop desktop browser version chrome 4+ firefox 3.5+ internet explorer 9+ opera 10.5+ safari 4+ mobile mobile browser version chrome (android) 32+ firefox (android) 26+ ie mobile 10+ opera mobile 11+ safari (ios) 4+ ...
... chrome 4+ firefox 3.5+ internet explorer 9+ opera 10.5+ safari 4+ mobile mobile browser version chrome (android) 32+ firefox (android) 26+ ie mobile 10+ opera mobile 11+ safari (ios) 4+ android browser 2.3+ blackberry 7+ audio codec support browser ogg mp3 aac pcm opus firefox 3.5+ ✓ ✓ *26+ ✓ ✓ *14+ safari 5+ ✓ ✓ chrome 6+ ✓ ✓ ✓ ✓ *9+ opera 10.5+ ✓ internet explorer 9+ ✓ ✓ firefox mobile ✓ ✓ ✓ ✓ ✓ s...

Live streaming web audio and video - Developer guides
rse, we then have the issue that we need to encode in three different formats. audio streaming file formats there are also some audio formats beginning to see support across browsers. opus opus is a royalty-free and open format that manages to optimize quality at various bit-rates for different types of audio. music and speech can be optimized in different ways and opus uses the silk and celt codecs to achieve this. currently, opus is supported by firefox desktop and mobile as well as the latest versions of desktop chrome and opera. note: opus is a mandatory format for webrtc browser implementations. mp3, aac, ogg vorbis most common audio formats can be streamed using specific server-side technologies. note: it's potentially easier to stream audio using non-streaming formats because ...
...technologies in order to stream live audio and video, you will need to run specific streaming software on your server or use third-party services. gstreamer gstreamer is an open source cross-platform multimedia framework that allows you to create a variety of media-handling components, including streaming components. through its plugin system, gstreamer provides support for more than a hundred codecs (including mpeg-1, mpeg-2, mpeg-4, h.261, h.263, h.264, realvideo, mp3, wmv, and flv.) gstreamer plugins such as souphttpclientsink and shout2send exist to stream media over http. you can also integrate with python's twisted framework or use something like flumotion (open source streaming software). for rtmp transfer you can use the nginx rtmp module. shoutcast shoutcast is a cross-platform ...

Audio and Video Delivery - Developer guides
ised to match the latest version of the html5 specification. instead of the error event being dispatched to the media element itself, it now gets delivered to the child <source> elements corresponding to the sources resulting in the error. this lets you detect which sources failed to load, which may be useful. ...
...d to the media element itself, it now gets delivered to the child <source> elements corresponding to the sources resulting in the error. this lets you detect which sources failed to load, which may be useful. consider this html: <video> <source id="mp4_src" src="video.mp4" type='video/mp4; codecs="avc1.42e01e, mp4a.40.2"'> </source> <source id="3gp_src" src="video.3gp" type='video/3gpp; codecs="mp4v.20.8, samr"'> </source> <source id="ogg_src" src="video.ogv" type='video/ogv; codecs="theora, vorbis"'> </source> </video> since firefox doesn't support mp4 and 3gp on some platforms due to their patent-encumbered nature, the <source> elements with the ids "mp4_src" and "3gp_src" will receive error events before the ogg resource is loaded. the sources are tried in the order in which t...

Audio and video manipulation - Developer guides
tesian coordinates. example var panner = context.createpanner(); panner.coneoutergain = 0.2; panner.coneouterangle = 120; panner.coneinnerangle = 0; panner.connect(context.destination); source.connect(panner); source.start(0); // position the listener at the origin. context.listener.setposition(0, 0, 0); note: you can find an example on our github repository (see it live also). javascript codecs it's also possible to manipulate audio at a low level using javascript. this can be useful should you want to create audio codecs. libraries currently exist for the following formats : aac: aac.js alac: alac.js flac: flac.js mp3: mp3.js opus: opus.js vorbis: vorbis.js note: at audiocogs, you can try out a few demos; audiocogs also provides a framework, aurora.js, which is intended t...
...nipulate audio at a low level using javascript. this can be useful should you want to create audio codecs. libraries currently exist for the following formats : aac: aac.js alac: alac.js flac: flac.js mp3: mp3.js opus: opus.js vorbis: vorbis.js note: at audiocogs, you can try out a few demos; audiocogs also provides a framework, aurora.js, which is intended to help you author your own codecs in javascript. examples various web audio api (and other) examples three.js video cube example convolution effects in real-time ...

Digital video concepts - Web media technologies
in this article, we explore important concepts that are useful to understand in order to fully grasp how to work with video on the web. encoding color representing the colors in an image or video requires several values for each pixel. what those values are depends on how you "split up" the color when converting it to numeric form. there are several color models, and video codec makes use of one or more of these to represent their pixels during the encoding process as well as after decoding the video frames. rgb most computer graphics models use the rgb color system, wherein some number of bits of data are used to represent each of the red, green, and blue components of the color of an individual pixel, and an image is comprised of a two-dimensional array of these pixels. there a...
.... yuv data representation because the image is represented using more detail in greyscale than in color, the values of y', u, and v are not typically stored together, one sample per pixel, the way rgb images are stored in memory. each frame of video is represented by a matrix of luma values, y, with each pixel's color computed using the values in the u and v matrices. for example, in the av1 codec, a record stores the encoded luma for a tile, and a second record contains the chroma data in the form of the u and v values. in broadcast television, the luma signal is the primary signal, while the chrominance data is sent on a subcarrier. the jpeg file type's compression works by first converting the image to the y'uv color system, then compressing that data. each of y', cb, and cr are able ...

File I/O - Archive of obsolete content
e/modules/netutil.jsm"); components.utils.import("resource://gre/modules/fileutils.jsm"); // file is nsifile, data is a string // you can also optionally pass a flags parameter here. it defaults to // fileutils.mode_wronly | fileutils.mode_create | fileutils.mode_truncate; var ostream = fileutils.opensafefileoutputstream(file); var converter = components.classes["@mozilla.org/intl/scriptableunicodeconverter"]. createinstance(components.interfaces.nsiscriptableunicodeconverter); converter.charset = "utf-8"; var istream = converter.converttoinputstream(data); // the last argument (the callback) is optional. netutil.asynccopy(istream, ostream, function(status) { if (!components.issuccesscode(status)) { // handle error! return; } // data has been written to the fi...

HTML - MDN Web Docs Glossary: Definitions of Web-related terms
any text, for instance <img>. ...

JavaScript - MDN Web Docs Glossary: Definitions of Web-related terms
e the browser. ...

Latency - MDN Web Docs Glossary: Definitions of Web-related terms
ested resource to reach its destination. low latency is good, meaning there is little or no delay. high latency is bad, meaning it takes a long time for the requested resource to reach its destination. latency can be a factor in any kind of data flow, but is most commonly discussed in terms of network latency (the time it takes for a packet of data to travel from source to destination) and media codec latency (the time it takes for the source data to be encoded or decoded and reach the consumer of the resulting data). learn more understanding latency ...

SDP - MDN Web Docs Glossary: Definitions of Web-related terms
here is a typical sdp message: v=0 o=alice 2890844526 2890844526 in ip4 host.anywhere.com s= c=in ip4 host.anywhere.com t=0 0 m=audio 49170 rtp/avp 0 a=rtpmap:0 pcmu/8000 m=video 51372 rtp/avp 31 a=rtpmap:31 h261/90000 m=video 53000 rtp/avp 32 a=rtpmap:32 mpv/90000 sdp is never used alone, but by protocols like rtp and rtsp. sdp is also as component of webrtc, which uses sdp as a way of describing a session. ...

MDN Web Docs Glossary: Definitions of Web-related terms
ting carddav caret cdn certificate authority certified challenge-response authentication character character encoding character set chrome cia cipher cipher suite ciphertext class client hints closure cms code splitting codec compile compile time computer programming conditional constant constructor continuous media control flow cookie copyleft cors cors-safelisted request header cors-safelisted response header crawler crlf cross axis cross-site scripting cru...

How can we design for all types of users? - Learn web development
rovide alternatives to multimedia content. subtitling/close-captioning you should include captions in your video to cater to visitors who can't hear the audio. some users have hearing challenges, lack functioning speakers, or work in a noisy environment (like on the train). transcript subtitles only work if somebody watches the video. many users don't have time, or lack the proper plugin or codec. additionally, search engines rely mainly on text to index your contents. for all these reasons, please provide a text transcript of the video/audio file. image compression some users may choose to display images, but still have limited bandwidth available, especially in developing countries and on mobile devices. if you want a successful website, please compress your images. there are various...

Publishing your website - Learn web development
chrisdavidmills, go to chrisdavidmills.github.io. note: it may take a few minutes for your website to go live. if your website does not display immediately, wait a few minutes. try again. to learn more, see github pages help. further reading what is a web server understanding domain names how much does it cost to do something on the web? deploy a website: a nice tutorial from codecademy that goes a bit further and shows some additional techniques. cheap or free static website hosting by scott murray has some useful ideas on available services. previous overview: getting started with the web next in this module installing basic software what will your website look like? dealing with files html basics css basics javascript basics publishing your website ...

Multimedia: video - Learn web development
attribute is required for autoplay in mobile browsers. playsinline is required for mobile safari, allowing videos to play without forcing fullscreen mode. remove audio from muted hero videos for hero-video or other video without audio, removing audio is smart. <video autoplay="" loop="" muted="true" playsinline="" id="hero-video"> <source src="banner_video.webm" type='video/webm; codecs="vp8, vorbis"'> <source src="web_banner.mp4" type="video/mp4"> </video> this hero-video code (above) is common to conference websites and corporate home pages. it includes a video that is auto-playing, looping, and muted. there are no controls, so there is no way to hear audio. the audio is often empty, but still present, and still using bandwidth. there is no reason to serve audio with video...

Client-side tooling overview - Learn web development
dme document in a code repository. the date-fns docs for example are pretty good, complete, and easy to follow. some documentation however can be rather technical and academic and not a good fit for your learning needs. instead, you might want to find some dedicated tutorials on getting started with particular types of tools. great starting places is to search web sites like css tricks, dev, freecodecamp, and smashing magazine, as they’re tailored to the web development industry. again, you’ll probably go through several different tools as you search for the right ones for you, trying them out to see if they make sense, are well-supported, and do what you want them to do. this is fine — it is all good for learning, and the road will get smoother as you get more experience. summary tha...

NSS_3.12.1_release_notes.html
code from nss source tree bug 372241: need more versatile form of cert_nametoascii bug 390296: nss ignores subject cn even when san contains no dnsname bug 401928: support generalized pkcs#5 v2 pbes bug 403543: pkix: need a way to enable/disable aia cert fetching bug 408847: pkix_ocspchecker_check does not support specified responder (and given signercert) bug 414003: crash [[@ cert_decodecertpackage] sometimes with this testcase bug 415167: memory leak in certutil bug 417399: arena allocation results are not checked in pkix_pl_infoaccess_parselocation bug 420644: improve ssl tracing of key derivation bug 426886: use const char* in pk11_importcertforkey bug 428103: cert_encodesubjectkeyid is not defined in any public header file bug 429716: debug builds of libpkix uncond...

NSS_3.12.2_release_notes.html
flags needed when libnss3 was linked with libsoftokn3 bug 454961: fix the implementation and use of pr_fgets in signtool bug 455348: change hyphens to underscores in debug_$(shell whoami). bug 455424: nssilckt.h defines the enumeration constant 'lock' bug 456036: stubs for deprecated functions in lib/certdb/stanpcertdb.c should set the pr_not_implemented_error error. bug 456854: cert_decodecertpackage does not set nspr error code upon error bug 457980: hundreds of kilobytes of useless strings in libpkix bug 457984: enable pkcs11 module logging in optimized builds bug 458905: memory leaks in pkix bridge certificates. bug 459231: memory leak in cert fetching - aia extension. bug 459248: support intel aes extensions. bug 459359: forwardbuilderstate object is leaked when aia ...

NSS 3.12.4 release notes
la.org/pub/mozilla.org/security/nss/releases/nss_3_12_4_rtm/src/. major changes in nss 3.12.4 nss 3.12.4 is the version that we submitted to nist for fips 140-2 validation. currently nss 3.12.4 is in the "review pending" state in the fips 140-2 pre-validation list at http://csrc.nist.gov/groups/stm/cmvp/documents/140-1/140inprocess.pdf added crl distribution point support (see cert.h). cert_decodecrlissuingdistributionpoint cert_findcrlissuingdistpointexten the old documentation of the expression matching syntax rules was incorrect, and the new corrected documentation is as follows for public nssutil functions (see portreq.h): port_regexpvalid port_regexpsearch port_regexpcasesearch these functions will match a string with a shell expression. the expressions accepted are based loos...

NSS_3.12_release_notes.html
g/nspr/releases/v4.7.1/. ...

NSS 3.44 release notes
in lib/certdb/cert.h cert_getcertificateder - access the der-encoded form of a certcertificate. notable changes in nss 3.44 it is now possible to build nss as a static library (bug 1543545) initial support for building for ios. ...

Index
ail. 249 js_dumpnamedroots jsapi reference, obsolete, spidermonkey each call to js_addnamedroot creates a record in a table of named roots maintained by the garbage collector. js_dumpnamedroots provides a way for the application to access the contents of that table. it calls the dump function once for each named root in the given runtime rt. in pseudocode: 250 js_encodecharacters jsapi reference, obsolete, spidermonkey js_encodecharacters copies the characters of a jschar array into a char array, converting the 16-bit values to 8-bit values. if spidermonkey was built with js_c_strings_are_utf8 defined or js_setcstringsareutf8 was called, the string is converted to utf-8. otherwise each character is simply truncated to 8 bits. 251 js_encodest...

JS_CStringsAreUTF8
is no way to pass in a string containing characters beyond u+00ff. however, spidermonkey can optionally interpret these strings as utf-8. there are two ways to enable this: at compile time, by building spidermonkey with js_c_strings_are_utf8 defined; or at run time, by calling js_setcstringsareutf8 before the first call to js_newruntime. enabling this option also causes js_getstringbytes and js_encodecharacters to produce utf-8 strings instead of latin-1. js_cstringsareutf8 returns js_true if the javascript engine is configured (either at compile time or at run time) to interpret char strings as utf-8. see also bug 805080 ...

JSAPI reference
js_internstring js_internstringn js_internucstring js_internucstringn js_internjsstring added in spidermonkey 1.8.5 js_stringhasbeeninterned added in spidermonkey 17 ...

XPCOM Stream Guide
strings versus nsiinputstream nsiinputstream and nsioutputstream work with 8-bit characters. however, javascript strings contain 16-bit characters. this can mean if you have characters beyond ascii code 255, you risk losing data using nsstringstream, for example. to get an input stream for javascript strings safely, try this. var converter = components.classes["@mozilla.org/intl/scriptableunicodeconverter"] .createinstance(components.interfaces.nsiscriptableunicodeconverter); converter.charset = "utf-8"; var stream = converter.converttoinputstream(string); javascript modules are your friends there are several useful javascript modules at your disposal. for streams, the most obvious are netutil.jsm and fileutils.jsm. netutil.jsm provides apis for copying an inpu...

Index
stream. 883 available no summary! 884 close no summary! 885 init no summary! 886 read no summary! 887 nsiscriptableunescapehtml interfaces, interfaces:scriptable, xpcom, xpcom interface reference implemented by: @mozilla.org/feed-unescapehtml;1 as a service: 888 nsiscriptableunicodeconverter interfaces, interfaces:scriptable, unicode, xpcom, xpcom interface reference this legacy api represents binary data using the lower haft of each 16-bit code unit in a javascript string. if the other apis you are reading data from or writing data to don't require you to use this legacy representation, you should use textdecoder and textencoder (available to chrome javascript via ...

Language bindings
ative component manager service. the scriptable methods on the nsicomponentmanager interface can be called directly on this object.components.resultscomponents.results is a read-only object whose properties are the names listed as the first parameters of the macros in js/xpconnect/src/xpc.msg (also at table of errors), with the value of each corresponding to that constant's value.components.returncodecomponents.stackcomponents.stack is a read only property of type nsistackframe (idl definition) that represents a snapshot of the current javascript callstack. this can be used for various diagnostic purposes.components.utilscomponents.utils is a collection of various useful xpconnect features. its interface is defined at js/xpconnect/idl/xpccomponents.idl.components.utils.cloneintothis function pr...

XPCOM Interface Reference
lproxyservicensiproxyinfonsipushmessagensipushservicensipushsubscriptionnsiradiointerfacelayernsirandomgeneratornsirequestnsirequestobservernsiresumablechannelnsirunnablensishentrynsishistorynsishistorylistenernsisockssocketinfonsisslerrorlistenernsisslsocketcontrolnsiscreennsiscreenmanagernsiscripterrornsiscripterror2nsiscriptableionsiscriptableinputstreamnsiscriptableunescapehtmlnsiscriptableunicodeconverternsiscrollablensisearchenginensisearchsubmissionnsisecuritycheckedcomponentnsiseekablestreamnsiselectionnsiselection2nsiselection3nsiselectioncontrollernsiselectionimageservicensiselectionprivatensiserversocketnsiserversocketlistenernsiservicemanagernsisessionstartupnsisessionstorensisimpleenumeratornsismsdatabaseservicensismsrequestmanagernsismsservicensisocketprovidernsisocketproviderserv...

XPCOM Interface Reference by grouping
cessiblehypertext nsiaccessibleimage nsiaccessibleprovider nsiaccessibleretrieval nsiaccessiblerole nsiaccessiblescrolltype nsiaccessibleselectable nsiaccessiblestates nsiaccessibletable nsiaccessibletext nsiaccessibletreecache nsiaccessiblevalue nsiaccessnode nsisyncmessagesender script nsiscriptableunescapehtml nsiscriptableunicodeconverter nsiscripterror nsiscripterror2 stylesheet nsistylesheetservice url nsiuri nsiurl util nsidomserializer nsidomxpathevaluator nsidomxpathexception nsidomxpathexpression nsidomxpathresult xslt nsixsltexception nsixsltprocessor download nsidownload nsidownloadmanager nsidownloadprogresslistener...

AudioConfiguration - Web APIs
e being tested when calling mediacapabilities.encodinginfo() or mediacapabilities.decodinginfo() to query whether a specific audio configuration is supported, smooth, and/or power efficient. properties the audioconfiguration dictionary is made up of four audio properties, including: contenttype: a valid audio mime type, for information on possible values and what they mean, see the web audio codec guide. channels: the number of channels used by the audio track. bitrate: the number of bits used to encode one second of the audio file. samplerate: the number of audio samples making up one second of the audio file. examples //create media configuration to be tested const mediaconfig = { type : 'file', // 'record', 'transmission', or 'media-source' audio : { contenttype ...

AudioContext.createMediaStreamDestination() - Web APIs
mediarecorder.stop(); osc.stop(0); e.target.disabled = true; } }); mediarecorder.ondataavailable = function(evt) { // push each chunk (blobs) in an array chunks.push(evt.data); }; mediarecorder.onstop = function(evt) { // make blob out of our blobs, and open it. var blob = new blob(chunks, { 'type' : 'audio/ogg; codecs=opus' }); document.queryselector("audio").src = url.createobjecturl(blob); }; </script> </body> </html> note: you can view this example live, or study the source code, on github. specifications specification status comment web audio apithe definition of 'createmediastreamdestination()' in that specification. working draft ...

BasicCardResponse.cardSecurityCode - Web APIs
browser compatibility the compatibility table on this page is generated from structured data. if you'd like to contribute to the data, please check out https://github.com/mdn/browser-compat-data and send us a pull request. ...

BasicCardResponse - Web APIs
ightly builds.disabled from version 56: this feature is behind the dom.payments.request.enabled preference (needs to be set to true). ...

DOMException.code - Web APIs
dation browser compatibility the compatibility table on this page is generated from structured data. if you'd like to contribute to the data, please check out https://github.com/mdn/browser-compat-data and send us a pull request. ...

HTMLMediaElement - Web APIs
interface also inherits methods from its ancestors htmlelement, element, node, and eventtarget. htmlmediaelement.addtexttrack() adds a text track (such as a track for subtitles) to a media element. htmlmediaelement.capturestream() returns mediastream, captures a stream of the media content. htmlmediaelement.canplaytype() given a string specifying a mime media type (potentially with the codecs parameter included), canplaytype() returns the string probably if the media should be playable, maybe if there's not enough information to determine whether the media will play or not, or an empty string if the media cannot be played. htmlmediaelement.fastseek() directly seeks to the given time. htmlmediaelement.load() resets the media to the beginning and selects the best available source f...

MediaCapabilities.decodingInfo() - Web APIs
ration: either an audioconfiguration or a videoconfiguration. return value a promise fulfilling with a mediacapabilitiesinfo interface containing three boolean attributes: supported smooth powerefficient exceptions a typeerror is raised if the mediaconfiguration passed to the decodinginfo() method is invalid, either because the type is not video or audio, the contenttype is not a valid codec mime type, the media decoding configuration is not a valid value for the media decoding type, or any other error in the media configuration passed to the method, including omitting values required in the media decoding configuration. example //create media configuration to be tested const mediaconfig = { type : 'file', // or 'media-source' audio : { contenttype : "aud...

MediaCapabilities - Web APIs
the information can be used to serve optimal media streams to the user and determine if playback should be smooth and power efficient. the information is accessed through the mediacapabilities property of the navigator interface. ...

MediaCapabilitiesInfo - Web APIs
owerefficient: given the properties defined in the mediaconfiguration, will the playback of the specified piece of media be power efficient? if supported is true, and playback will be power efficient, powerefficient is true, otherwise, is it false. browsers will report a supported media configuration as smooth and powerefficient until stats on this device have been recorded. all supported audio codecs are reported to be power efficient. example // mediaconfiguration to be tested const mediaconfig = { type : 'file', audio : { contenttype : "audio/ogg", channels : 2, bitrate : 132700, samplerate : 5200 }, }; // check support and performance navigator.mediacapabilities.decodinginfo(mediaconfig).then(result => { // result contains the media capabili...

MediaConfiguration - Web APIs
ype. width: width of the video. height: height of the video. bitrate: number of bits used to encode one second of the video file. framerate: number of frames making up one second of video playback. example //create a video configuration to be tested const videodecoderconfig = { type : 'file', // 'record', 'transmission', or 'media-source' video : { contenttype : "video/webm;codecs=vp8", // valid content type width : 800, // width of the video height : 600, // height of the video bitrate : 10000, // number of bits used to encode 1s of video framerate : 30 // number of frames making up that 1s. } }; const audioencoderconfig = { type : 'file', // 'record', 'transmission', or 'media-source' audio : { contenttype : "audio/ogg", // valid cont...

MediaDecodingConfiguration - Web APIs
t is meant to be used for a plain file playback. media-source: represents a configuration that is meant to be used for playback of a mediasource. a media configuration — a videoconfiguration or audioconfiguration dictionary. examples //create media configuration to be tested const mediaconfig = { type : 'file', // or 'media-source' video : { contenttype : "video/webm;codecs=vp8", // valid content type width : 800, // width of the video height : 600, // height of the video bitrate : 10000, // number of bits used to encode 1s of video framerate : 30 // number of frames making up that 1s. } }; // check support and performance navigator.mediacapabilities.decodinginfo(mediaconfig).then(result => { console.log('this conf...

MediaEncodingConfiguration - Web APIs
.g. using mediarecorder. transmission — represents a configuration meant to be transmitted over electronic means (e.g. using rtcpeerconnection). a media configuration — a videoconfiguration audioconfiguration dictionary. examples //create media configuration to be tested const mediaconfig = { type : 'record', // or 'transmission' video : { contenttype : "video/webm;codecs=vp8", // valid content type width : 800, // width of the video height : 600, // height of the video bitrate : 10000, // number of bits used to encode 1s of video framerate : 30 // number of frames making up that 1s. } }; // check support and performance navigator.mediacapabilities.encodinginfo(mediaconfig).then(result => { console.log('this conf...

MediaRecorder: error event - Web APIs
bubbles no cancelable no interface mediarecordererrorevent event handler property onerror for details of the all the possible errors see the documentation for the event handler property: onerror. ...

MediaRecorder.ondataavailable - Web APIs
recorder.ondataavailable = function(event) { ... } mediarecorder.addeventlistener('dataavailable', function(event) { ... }) example ... var chunks = []; mediarecorder.onstop = function(e) { console.log("data available after mediarecorder.stop() called."); var audio = document.createelement('audio'); audio.controls = true; var blob = new blob(chunks, { 'type' : 'audio/ogg; codecs=opus' }); var audiourl = window.url.createobjecturl(blob); audio.src = audiourl; console.log("recorder stopped"); } mediarecorder.ondataavailable = function(e) { chunks.push(e.data); } ... specifications specification status comment mediastream recordingthe definition of 'mediarecorder.ondataavailable' in that specification. working draft ...

MediaRecorder.onerror - Web APIs
s obtained using getusermedia() when the user denies permission to use an input device. this also happens when a mediastreamtrack within the stream is marked as isolated due to the peeridentity constraint on the source stream. notsupportederror an attempt was made to instantiate a mediarecorder using a mime type that isn't supported on the user's device; one or more of the requested container, codecs, or profiles as well as other information may be invalid. unknownerror an non-security related error occurred that cannot otherwise be categorized. recording stops, the mediarecorder's state becomes inactive, one last dataavailable event is sent to the mediarecorder with the remaining received data, and finally a stop event is sent. these errors may occur either directly because of a call to...

MediaRecorder.onstop - Web APIs
se in your application. syntax mediarecorder.onstop = function(event) { ... } mediarecorder.addeventlistener('stop', function(event) { ... }) example ... mediarecorder.onstop = function(e) { console.log("data available after mediarecorder.stop() called."); var audio = document.createelement('audio'); audio.controls = true; var blob = new blob(chunks, { 'type' : 'audio/ogg; codecs=opus' }); var audiourl = window.url.createobjecturl(blob); audio.src = audiourl; console.log("recorder stopped"); } mediarecorder.ondataavailable = function(e) { chunks.push(e.data); } ... specifications specification status comment mediastream recordingthe definition of 'mediarecorder.onstop' in that specification. working draft initial de...

MediaSource.activeSourceBuffers - Web APIs
a sourcebufferlist containing the sourcebuffer objects for each of the active tracks. ...

MediaSource.duration - Web APIs
ebuffers are being updated (i.e. their sourcebuffer.updating property is true.) example the following snippet is based on a simple example written by nick desaulniers (view the full demo live, or download the source for further investigation.) function sourceopen (_) { //console.log(this.readystate); // open var mediasource = this; var sourcebuffer = mediasource.addsourcebuffer(mimecodec); fetchab(asseturl, function (buf) { sourcebuffer.addeventlistener('updateend', function (_) { mediasource.endofstream(); mediasource.duration = 120; video.play(); //console.log(mediasource.readystate); // ended }); sourcebuffer.appendbuffer(buf); }); }; ... specifications specification status comment media source extensionsthe de...

MediaSource.removeSourceBuffer() - Web APIs
bject. syntax mediasource.removesourcebuffer(sourcebuffer); parameters sourcebuffer the sourcebuffer object to be removed. return value undefined exceptions exception explanation notfounderror the supplied sourcebuffer doesn't exist in mediasource.sourcebuffers. examples for (i = 0; i < 10; i++) { var sourcebuffer = mediasource.addsourcebuffer(mimecodec); } mediasource.removesourcebuffer(mediasource.sourcebuffers[0]); specifications specification status comment media source extensionsthe definition of 'removesourcebuffer()' in that specification. recommendation initial definition. ...

MediaSource.sourceBuffers - Web APIs
syntax var mysourcebuffers = mediasource.sourcebuffers; value a sourcebufferlist. example the following snippet is based on a simple example written by nick desaulniers (view the full demo live, or download the source for further investigation.) function sourceopen (_) { //console.log(this.readystate); // open var mediasource = this; var sourcebuffer = mediasource.addsourcebuffer(mimecodec); fetchab(asseturl, function (buf) { sourcebuffer.addeventlistener('updateend', function (_) { mediasource.endofstream(); console.log(mediasource.sourcebuffers); // will contain the source buffer that was added above video.play(); //console.log(mediasource.readystate); // ended }); sourcebuffer.appendbuffer(buf); }); }; ... specifications specif...

Using the MediaStream Recording API - Web APIs
first, we display a prompt asking the user to name their clip. next, we create an html struct...

MediaStream Recording API - Web APIs
recorder. you can listen for error events by setting up a onerror event handler. example here, we use an html canvas as source of the mediastream, and stop recording after 9 seconds. var canvas = document.queryselector("canvas"); // optional frames per second argument. var stream = canvas.capturestream(25); var recordedchunks = []; console.log(stream); var options = { mimetype: "video/webm; codecs=vp9" }; mediarecorder = new mediarecorder(stream, options); mediarecorder.ondataavailable = handledataavailable; mediarecorder.start(); function handledataavailable(event) { console.log("data-available"); if (event.data.size > 0) { recordedchunks.push(event.data); console.log(recordedchunks); download(); } else { // ... } } function download() { var blob = new blob(re...

Media Source API - Web APIs
lients (such as those using dash or hls) to be built on its extensible api. creating assets that work with mse in modern browsers is a laborious process, taking significant time, computing power, and energy. the usage of external utilities to massage the content into a suitable format is required. while browser support for the various media containers with mse is spotty, usage of the h.264 video codec, aac audio codec, and mp4 container format is a common baseline. mse also provides an api for runtime detection of container and codec support. if you do not require explicit control of video quality over time, the rate at which content is fetched, or the rate at which memory is evicted, then the <video> and <source> tags may well be a simple and adequate solution. dash dynamic adaptive stream...

Navigator.requestMediaKeySystemAccess() - Web APIs
ribed by keysystem and supportedconfigurations exceptions in case of an error, the returned promise is rejected with a domexception whose name indicates what kind of error occurred. notsupportederror either the specified keysystem isn't supported by the platform or the browser, or none of the configurations specified by supportedconfigurations can be satisfied (if, for example, none of the codecs specified in contenttype are available). typeerror either keysystem is an empty string or the supportedconfigurations array is empty. specifications specification status comment encrypted media extensionsthe definition of 'requestmediakeysystemaccess()' in that specification. recommendation initial definition ...

RTCInboundRtpStreamStats.sliCount - Web APIs
g corrupted between the sender and the receiver, requiring resends or dropping frames. syntax var slicount = rtcinboundrtpstreamstats.slicount; value an unsigned integer indicating the number of sli packets this receiver sent to the remote sender due to lost runs of macroblocks. a high value of slicount may be an indication of an unreliable network. this is a very technical part of how video codecs work. for details, see rfc 4585: 6.3.2. note: this value is only present for video media. specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcinboundrtpstreamstats.slicount' in that specification. candidate recommendation initial definition. ...

RTCOfferAnswerOptions - Web APIs
as the base type for the options parameter when calling createoffer() or createanswer() on an rtcpeerconnection. each of createoffer() and createanswer() use rtcofferansweroptions as the base type for their options parameter's dictionary. createoffer() uses rtcofferoptions and createanswer() uses rtcansweroptions. properties voiceactivitydetection optional for configurations of systems and codecs that are able to detect when the user is speaking and toggle muting on and off automatically, this option enables and disables that behavior. the default value is true, enabling this functionality specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcofferansweroptions' in that specification. candidate ...

RTCOutboundRtpStreamStats.pliCount - Web APIs
nize with the media. however, the primary purpose of a pli packet is to allow the rtcrtpsender for which this rtcoutboundrtpstreamstats object provides statistics to consider techniques to mitigate network performance issues. this is often achieved by methods such as increasing the compression or lowering resolution, although the mechanisms available to reduce the bit rate of the stream vary from codec to codec. specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcoutboundrtpstreamstats.plicount' in that specification. candidate recommendation initial definition. ...

RTCOutboundRtpStreamStats.qualityLimitationReason - Web APIs
this quality reduction may include changes such as reduced frame rate or resolution, or an increase in compression factor. the amount of time the encoded media has had its quality reduced in each of the potential ways that can be done can be found in qualitylimitationdurations. ...

RTCOutboundRtpStreamStats.sliCount - Web APIs
d the receiver, causing the receiver to request retransmits or to drop frames entirely. syntax var slicount = rtcoutboundrtpstreamstats.slicount; value an unsigned integer indicating the number of sli packets the sender received from the receiver due to lost runs of macroblocks. a high value of slicount may be an indication of an unreliable network. this is a very technical part of how video codecs work. for details, see rfc 4585: 6.3.2. note: this value is only present for video media. specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcoutboundrtpstreamstats.slicount' in that specification. candidate recommendation initial definition. ...

RTCOutboundRtpStreamStats - Web APIs
videosenderstats object containing statistics about this stream's rtcrtpsender. slicount an integer indicating the number of times this sender received a slice loss indication (sli) frame from the remote peer, indicating that one or more consecutive video macroblocks have been lost or corrupted. available only for video streams. targetbitrate a value indicating the bit rate the rtcrtpsender's codec is configured to attempt to achieve in its output media. totalencodedbytestarget a cumulative sum of the target frame sizes (the targeted maximum size of the frame in bytes when the codec is asked to compress it) for all of the frames encoded so far. this will likely differ from the total of the actual frame sizes. totalencodetime a floating-point value indicating the total number of seconds ...

RTCPeerConnection.createAnswer() - Web APIs
the answer is delivered to the returned promise, and should then be sent to the source of the offer to continue the negotiation process. ...

RTCPeerConnection - Web APIs
will be transmitted to the other peer.close() the rtcpeerconnection.close() method closes the current peer connection.createanswer() the createanswer() method on the rtcpeerconnection interface creates an sdp answer to an offer received from a remote peer during the offer/answer negotiation of a webrtc connection. the answer contains information about any media already attached to the session, codecs and options supported by the browser, and any ice candidates already gathered. the answer is delivered to the returned promise, and should then be sent to the source of the offer to continue the negotiation process.createdatachannel() the createdatachannel() method on the rtcpeerconnection interface creates a new channel linked with the remote peer, over which any kind of data may be transmitte...

RTCRtpParameters - Web APIs
the rtcrtpparameters dictionary is the basic object describing the parameters of an rtp transport. it is extended separately for senders and receivers in the form of the rtcrtpsendparameters and rtcrtpreceiveparameters dictionaries. to obtain the parameters of a sender or receiver, call its getparameters() method: getparameters() getparameters() properties codecs an array of rtcrtpcodecparameters objects describing the set of codecs from which the sender or receiver will choose. this parameter cannot be changed once initially set. headerextensions an array of zero or more rtp header extensions, each identifying an extension supported by the sender or receiver. header extensions are described in rfc 3550: 5.3.1. this parameter cannot be changed once initially set. rtcp an rtcrtcp...

RTCRtpSendParameters - Web APIs
om the rtcrtpparameters interface. degradationpreference specifies the preferred way the webrtc layer should handle optimizing bandwidth against quality in constrained-bandwidth situations; the value comes from the rtcdegradationpreference enumerated string type, and the default is balanced. encodings an array of rtcrtpencodingparameters objects, each specifying the parameters for a single codec that could be used to encode the track's media. priority a string from the rtcprioritytype enumerated type which indicates the encoding's priority. the default value is low. transactionid a string containing a unique id for the last set of parameters applied; this value is used to ensure that setparameters() can only be called to alter changes made by a specific previous call to getparameters...

RTCRtpStreamStats.kind - Web APIs
u're targeting. syntax mediakind = rtcrtpstreamstats.kind; value a domstring whose value is "audio" if the track whose statistics are given by the rtcrtpstreamstats object contains audio, or "video" if the track contains video media. this string will always be the same as the one provided by the associated mediastreamtrack object's kind property. it will also match the statistics object's rtccodecstats.codec property's media type. specifications specification status comment identifiers for webrtc's statistics apithe definition of 'rtcrtpstreamstats.kind' in that specification. candidate recommendation initial definition. ...

RTCRtpTransceiver - Web APIs
the rtcrtpreceiver object that handles receiving and decoding incoming media. sender read only the rtcrtpsender object responsible for encoding and sending data to the remote peer. stopped indicates whether or not sending and receiving using the paired rtcrtpsender and rtcrtpreceiver has been permanently disabled, either due to sdp offer/answer, or due to a call to stop(). methods setcodecpreferences() a list of rtcrtpcodecparameters objects which override the default preferences used by the user agent for the transceiver's codecs. stop() permanently stops the rtcrtptransceiver. the associated sender stops sending data, and the associated receiver likewise stops receiving and decoding incoming data. specifications specification status comment webrtc 1.

RTCStatsReport - Web APIs
pair for the transport are deleted when the rtcpeerconnection changes its rtcpeerconnection.icegatheringstate to new during an ice restart. the active candidate pair is deleted after the transport switches to another candidate pair; this change cannot be detected otherwise. certificate an rtccertificatestats object providing statistics related to a certificate being used by an rtcicetransport. codec an rtccodecstats object containing statistics about a codec currently being used by rtp streams to send or receive data for the rtcpeerconnection. csrc an rtccontributingsourcestats object which contains statistics related to a contributing source (csrc) that contributed to an inbound rtp stream. data-channel an rtcdatachannelstats object which contains statistics about each rtcdatachannel o...

RTCStatsType - Web APIs
pair for the transport are deleted when the rtcpeerconnection changes its rtcpeerconnection.icegatheringstate to new during an ice restart. the active candidate pair is deleted after the transport switches to another candidate pair; this change cannot be detected otherwise. certificate an rtccertificatestats object providing statistics related to a certificate being used by an rtcicetransport. codec an rtccodecstats object containing statistics about a codec currently being used by rtp streams to send or receive data for the rtcpeerconnection. csrc an rtccontributingsourcestats object which contains statistics related to a contributing source (csrc) that contributed to an inbound rtp stream. data-channel an rtcdatachannelstats object which contains statistics about each rtcdatachannel o...

WebRTC connectivity - Web APIs
cription contains not just the offer or answer under consideration, but any local ice candidates which have already been gathered since the offer or answer was created. similarly, pendingremotedescription includes any remote ice candidates which have been provided by calls to rtcpeerconnection.addicecandidate(). see the individual articles on these properties and methods for more specifics, and codecs used by webrtc for information about codecs supported by webrtc and which are compatible with which browsers. the codecs guide also offers guidance to help you choose the best codecs for your needs. ice candidates as well as exchanging information about the media (discussed above in offer/answer and sdp), peers must exchange information about the network connection. this is known as an ice can...

Introduction to WebRTC protocols - Web APIs
mation through that server. you would create a connection with a turn server and tell all peers to send packets to the server which will then be forwarded to you. this obviously comes with some overhead so it is only used if there are no other alternatives. sdp session description protocol (sdp) is a standard for describing the multimedia content of the connection such as resolution, formats, codecs, encryption, etc. so that both peers can understand each other once the data is transferring. this is, in essence, the metadata describing the content and not the media content itself. technically, then, sdp is not truly a protocol, but a data format used to describe connection that shares media between devices. documenting sdp is well outside the scope of this documentation; however, there ar...

WebRTC Statistics API - Web APIs
nted candidate-pair statistics describing the change from one rtcicetransport to another, such as during an ice restart. rtcicecandidatepairstats rtcstats certificate statistics about a certificate being used by an rtcicetransport. rtccertificatestats rtcstats codec statistics about a specific codec being used by streams being sent or received by this connection. rtccodecstats rtcstats csrc statistics for a single contributing source (csrc) that contributed to one of the connection's inbound rtp streams. rtcrtpcontributingsourcestats rtcstats data-channel statistics related ...

Web APIs
le rtciceserver rtcicetcpcandidatetype rtcicetransport rtcicetransportstate rtcidentityassertion rtcidentityerrorevent rtcidentityevent rtcinboundrtpstreamstats rtcnetworktype rtcofferansweroptions rtcofferoptions rtcoutboundrtpstreamstats rtcpeerconnection rtcpeerconnectioniceerrorevent rtcpeerconnectioniceevent rtcremoteoutboundrtpstreamstats rtcrtcpparameters rtcrtpcapabilities rtcrtpcodeccapability rtcrtpcodecparameters rtcrtpcontributingsource rtcrtpencodingparameters rtcrtpparameters rtcrtpreceiveparameters rtcrtpreceiver rtcrtpsendparameters rtcrtpsender rtcrtpstreamstats rtcrtpsynchronizationsource rtcrtptransceiver rtcrtptransceiverdirection rtcrtptransceiverinit rtcsctptransport rtcsessiondescription rtcsessiondescriptioncallback rtcstats rtcstatsicecandidatepairstate ...

DASH Adaptive Streaming for HTML 5 Video - HTML: Hypertext Markup Language
rom ffmpeg.org, with libvpx and libvorbis support for webm video and audio, at least version 2.5 (probably; this was tested ith 3.2.5). 1. use your existing webm file to create one audio file and multiple video files. for example: the file in.video can be any container with at least one audio and one video stream that can be decoded by ffmpeg, create the audio using: ffmpeg -i in.video -vn -acodec libvorbis -ab 128k -dash 1 my_audio.webm create each video variant. ffmpeg -i in.video -c:v libvpx-vp9 -keyint_min 150 -g 150 -tile-columns 4 -frame-parallel 1 -f webm -dash 1 \ -an -vf scale=160:90 -b:v 250k -dash 1 video_160x90_250k.webm ffmpeg -i in.video -c:v libvpx-vp9 -keyint_min 150 -g 150 -tile-columns 4 -frame-parallel 1 -f webm -dash 1 \ -an -vf scale=320:180 -b:v 500k -dash 1 vi...

<source>: The Media or Image Source element - HTML: Hypertext Markup Language
criptor to be valid. among the list, there must be only one string containing the same tuple of width descriptor and pixel density descriptor. the browser chooses the most adequate image to display at a given point of time. the srcset attribute has an effect only when the <source> element is the direct child of a <picture> element. type the mime media type of the resource, optionally with a codecs parameter. if the type attribute isn't specified, the media's type is retrieved from the server and checked to see if the user agent can handle it; if it can't be rendered, the next <source> is checked. if the type attribute is specified, it's compared against the types the user agent can present, and if it's not recognized, the server doesn't even get queried; instead, the next <source> eleme...

Configuring servers for Ogg media - HTTP
e browser will treat it as a video file. most servers don't by default serve ogg media with the correct mime types, so you'll likely need to add the appropriate configuration for this. for apache, you can add the following to your configuration: addtype audio/ogg .oga addtype video/ogg .ogv addtype application/ogg .ogg you can find specific information about possible media file types and the codecs used within them in our comprehensive guide to media types and formats on the web. in particular, the article on media container formats will be especially helpful when configuring serers to host media properly. handle http 1.1 byte range requests correctly in order to support seeking and playing back regions of the media that aren't yet downloaded, gecko uses http 1.1 byte-range requests to r...

Image file type and format guide - Web media technologies
are. compression most tiff files are uncompressed, but lossless packbits and lzw compression are supported, as is lossy jpeg compression. licensing no license required (aside from any associated with libraries you might use); all known patents have expired. webp image webp supports lossy compression via predictive coding based on the vp8 video codec, and lossless compression that uses substitutions for repeating data. lossy webp images average 25–35% smaller than jpeg images of visually similar compression levels. lossless webp images are typically 26% smaller than the same images in png format. webp also supports animation: in a lossy webp file, the image data is represented by a vp8 bitstream, which may contain multiple frames. lossless...

Handling media support issues in web content - Web media technologies
opularity and with a variety of capabilities. the availability of choices is good for the user, in that they can choose the format that suits their needs best. there is a drawback, however: because there are so many to choose from, with so many different kinds of licenses and design principles involved, each web browser developer is left to its own devices when deciding which media file types and codecs to support. this places a small, but reasonably easily overcome, burden on the web developer: to properly handle the situation when the user's browser can't handle a particular type of media. this guide covers techniques you can use to develop web content that meets your media needs while providing the most broadly compatible experience possible. topics we will examine fallbacks, baseline media...

Web media technologies
udio> and <video> elements. accessibility guide for media in web design in this guide, we cover ways web designers and developers can create content that is accessible to people with different capabilities. this ranges from simply using the alt attribute on <img> elements to captions to tagging media for screen readers. guide to media types and formats on the web a guide to the file types and codecs available for images, audio, and video media on the web. this includes recommendations for what formats to use for what kinds of content, best practices including how to provide fallbacks and how to prioritize media types, and also includes general browser support information for each media container and codec. streaming audio and video a guide which covers how to stream audio and video, as we...

Tutorials
module, we continue our coverage of all javascript's key fundamental features, turning our attention to commonly-encountered types of code block such as conditional statements, loops, functions, and events. you've seen this stuff already in the course, but only in passing — here we'll discuss it all explicitly. getting started with javascript what is javascript and how can it help you? codecademy codecademy is an easy way to learn how to code javascript. it's interactive and you can do it with your friends. freecodecamp freecodecamp teaches a variety of languages and frameworks for web development. it also has a forum, an internet radio station, and a blog. intermediate level introducing javascript objects in javascript, most things are objects, from core javascript fea...

Compiling an Existing C Module to WebAssembly - WebAssembly
a core use-case for webassembly is to take the existing ecosystem of c libraries and allow developers to use them on the web. these libraries often rely on c's standard library, an operating system, a file system and other things. emscripten provides most of these features, although there are some limitations. as an example, let's compile an encoder for webp to wasm. the source for the webp codec is written in c and available on github as well as some extensive api documentation. that's a pretty good starting point. $ git clone https://github.com/webmproject/libwebp to start off simple, expose webpgetencoderversion() from encode.h to javascript by writing a c file called webp.c: #include "emscripten.h" #include "src/webp/encode.h" emscripten_keepalive int version() { return webpgetenco...