I have a app where I play different code-generated sounds. I place these sounds in a AudioBufferSourceNode.
I allow the the user to choose what output device to play the sound through, so I use a MediaStreamAudioDestinationNode with its stream used as the source for an Audio Element. This way when the user chooses an audio output to play the sound to, I set the Sink Id of the Audio element to the requested audio output.
So I have AudioBufferSourceNode -> some Audio Graph (gain nodes, etc) -> MediaStreamAudioDestinationNode -> Audio element.
When I Play the first sound, it sound fine. But when I create a new source and connect it to the same MediaStreamAudioDestinationNode, the sound is played with the wrong pitch.
I created a Fiddle that shows the problem.
Is this a bug, or am I doing something wrong?
The problem was identified based on the OP Chrome Ticket.
It seems to come from the lack of sync between AudioElement and its source AudioNode (AudioBufferSourceNode, OscillatorNode, etc.) when you pause the source and play it back again.
The solution is to always call AudioElement.pause() and AudioElement.start() alongside your source stop and start.
https://jsfiddle.net/k1r7o0xj/3/
It's possible to dynamically change your graph layout by using .connect() and .disconnect(), even when audio is playing or sent through a stream (which could even be streamed over WebRTC).
I couldn't find a reference in the spec, so I'm pretty sure this is taken for granted.
For example, if you have two AudioBufferSourceNodes bufferSource1 and bufferSource2, and a MediaStreamAudioDestinationNode streamDestination:
bufferSource1.connect(streamDestination);
//do some other things here, and after some time, switch to bufferSource2:
//(streamDestination doesn't need to be explicitly specified here)
bufferSource1.disconnect(streamDestination);
bufferSource2.connect(streamDestination);
Example in action.
Edit 1:
Proper implementation:
According to the Editors Draft on the Audio Output API, it is planned/will be possible to choose a custom audio output device for the AudioContext as well (by means of new AudioContext({ sinkId: requestedSinkId });). I couldn't find any info on the progress, and even found a related discussion which the asker apparently read already. According to this and (many) other references, it doesn't seem te be an easy task, but it's planned for WA V1.
Edit:
That section has been removed from the API Draft, but you can still find it in an older version.
Current workaround:
I played around with your workaround (using a MediaStreamAudioDestinationNode and Audio object), and it seems to be related to nothing being connected. I modified my example to toggle a single buffer (similar to your example but with an AudioBufferSourceNode), and observed a similar frequency drop. However, when using a GainNode inbetween and setting it's gain.value to either 0 or 1, the frequency drops disappeared (this isn't gonna be the solution if you want to create and connect new AudioBuffers dynamically).
Related
I'm currently working on an HTML5/WebGL based project that requires the use of stereo audio for locating objects.
In order to handle this I need to vary the Left/Right Stereo Pan based on the location and distance the audio source is from the user.
However, when trying to use the following code as suggested by the SoundManager2 Documentation (http://www.schillmania.com/projects/soundmanager2/doc/#sm-config) nothing happens, the audio remains balance between left/right
soundManager.setPan('soundObjectId', -100);
I've tried other forms, such as:
soundManager.createSound({
url: '/path/to/some.mp3',
autoPlay: false,
pan: -75
});
I don't know how common SoundManager2 is as Googling around doesn't seem to show that many answers, I'm happy to try other libraries if they can be recommended for playing browser audio (must support IE11)
I ended up doing a very nasty workaround for this - I created two copies of my audio file in Audacity, one with the audio panned all the way left, and one panned all the way right. And then I 'mixed' these in SM2 by reducing/increasing the volume of the appropriate 'side' audio file. It works, but isn't a particularly elegant solution
I understand how to record microphone input in AS3 from this doc.
Is it possible to record sound exactly as they are being output/played?
The reason is I applied some sound transform (via the global SoundMixer) to sounds that are currently playing; and I also want to record this sound data while it is being played.
I just saw this question, to clarify, I am not trying to record just all sounds on the user's computer (which is not possible). My flash app has a Youtube player in it (via their AS3 API), and it's playing some sounds. I applied transforms using SoundMixer.soundTransform, and I want to record what's being played when the user is playing it.
Thanks in advance.
Just a passing suggestion.. on my desktop it seems ABLE to record sound into Flash from a different tab playing Youtube (HTML5).. I don't know how it's doing that!!
I allow microphone here.. (none actually plugged in, and speaker out has in-ear headphones)
http://code.tutsplus.com/tutorials/create-a-useful-audio-recorder-app-in-actionscript-3--active-5836
PS: Anyone trying this must reduce Windows volume since anything above 10-20% is distorted audio into the Flash app.
And this HTML5 youtube trailer was recorded fine into the Wav file produced by Flash app above
http://www.youtube.com/watch?v=MVt32qoyhi0
So after a quick search it seems my Realtek Audio is classed as a Full-Duplex soundcard and also within its own control panel I have an option called "Multi-streaming" which is enabled/ticked. I think Full-Duplex is enough to do this though. Try options within your soundcard's own settings software. Don't know about your end-users. Some hardware will do it, some wont, there is no all-round solution outside of AIR (which makes desktop apps out of your AS3 code).
I am using Vimeo's Flash API so that I can embed and read the timecode of a video using the playProgressHandler, pause it at certain times, pop a menu, and use buttons that trigger seekTo calls. Although everything works, the timecode is inaccurate to varying degrees. Anywhere from 1-2 seconds. I can tell this because:
1) If I play my video on Vimeo and pause it at 6:03 and do the same with it embedded in Flash the visuals do not match up. Flash is lagging behind a tad.
2) I did a test using the JavaScript API. My seekTo calls were consistently accurate. To seek to the same spot using the AS3 API I had to add 1.5 seconds. But even this isn't foolproof. Sometimes it works, but sometimes it's still off.
Any ideas what would account for this inaccuracy and how I might fix this problem? Yes, I can ditch the AS3 and use the JS version, but I'd prefer to just fix what I've already built.
(I also posted this on Vimeo's forum, but I'm following their "Limited support in API Forum" post which suggests to post here)
Unfortunately, there's not much we can do to fix this other than to recommend that you use our iframe embed.
It has to do with the way that we retrieve files from our CDN. Because Flash doesn't support byterange requests, we pass a parameter that returns part of the file starting at that position. The nature of how that works means it's always going to be imprecise.
I was watching a youtube video and I decided to investigate some parts of its video player. I noticed that unlike most HTML5 video I have seen, Youtube's video player does not do a normal video source and instead utilizes a blob url as the source.
Previously I have tested HTML5 videos and I found that the server starts streaming the whole video from the start and buffers in the background the complete rest of the video. This means that if your video is 300 megs, all 300 megs will be downloaded. If you seek to the middle, it will start downloading from the seek position all the way to the end.
Youtube does not work this way (at least in chrome). Instead it manages to control buffering so it only buffers a certain amount while paused. It also seems to only buffer the relevant pieces, so if you skip around it will make sure not to buffer pieces that are unlikely to be watched.
In my attempts to investigate how this worked, I noticed the video src tag has a value of blob:http%3A//www.youtube.com/ee625eee-2802-49b2-a13f-eb374d551d54, which pointed me to blobs, which then led me to typed arrays. Using those two resources I am able to load a mp4 video into a blob and display it in a HTML5 video tag.
However, what I am now stuck on is how Youtube deals with the pieces. Looking at the network traffic it appears to sends requests to http://r6---sn-p5q7ynee.c.youtube.com/videoplayback which returns binary video data back in chunks of 1.1mb. It also seems worth noting that most normal requests due to HTML5 video requests seem to receive a 206 response code back while it streams, yet youtube's playvideo calls get a 200 back.
I tried to attempt to only load a range of bytes (via setting the Range http header) which unfortunately failed (I'm assuming because there was no meta-data for the video coming with the video).
At this point I'm stuck on figuring out how Youtube accomplishes this. I came up with several ideas though none of which I am completely sold on:
1) Youtube is sending down self contained video and audio chunks with each /videoplayback call. This seems like a pretty heavy burden on the upload side and it seems like it would be difficult to stitch these together to make it appear like it's one seemless video. Also, the video tag seems to think it's one full video, judging from calling $('video').duration and $('video').currentTime, which leads me to believe that the video tag thinks it's a single video file. Finally, the vidoe src tag never changes which makes me believe it is working with a singular blob and not switching out blobs.
2) Youtube constructs an empty blob pre-sized to the full video array and updates the blob with pieces as it downloads it. It would then make sure the user has not gotten too close to the last downloaded piece (to prevent the user from entering an undownloaded section of the blob). The problem that I see with this that I don't see any way to dynamically update a blob through javascript (although maybe I'm just having trouble googling for it)
3) Youtube downloads the meta data and then starts constructing the blob in order by appending the video pieces as it downloads them. The problem I see with this method is I don't understand how it would handle seeks in post-buffered territory.
Maybe I"m just missing an obvious answer that's right in front of me. Anyone have any ideas?
edit: I just thought of a fourth option. Another idea is they might use the file API to write the binary chunks to a file and use that file to stream off of. The file API seems to have the ability to seek to specific positions, therefore allowing you to fill a video with empty bytes and fill them in as they are received. This would definitely accommodate video seeking as well.
Okay, so few things you need to know is that YouTube is based on this great open source Project. It behaves different for every browser and if your browser supports more intensive decoding like WEBM it will use that to save Google's bandwidth. Also if you look at this Demo
Then you will find a section which downloads the entire video into a thing called "offline storage". I know chrome has it and some other browsers not every in some cases they do have to use the entire video source instead of a blob. So that blob is streaming depending on the user interaction with the video. Yes the video is just 1 file and they have metadata for that video like a little database that tells the time of the video and the points at which chunks can be divided in.
You can find out more by reading the Project's documentation. I really recommend you have a look at the demo.
When you look at the AppData of GoogleChrome, while playing a youtube video, you will see that it buffers in segmented files. The videos uploaded to youtube are segmented, which is why you can't perfectly pinpoint a timeframe in the first click on the bar if that timeframe is outside of the current segment.
The amount of segments depends on the length of the video, and the time from which you start and stop playing back the video.
When you are linked to a timeframe of a video, it will simply skip the buffering of the segments that come before that timeframe.
Unfortunately I don't know much about the coding for video playback, but I hope this points you in the right direction.
there is a canvas element in the page ,Maybe This Will Help
http://html5doctor.com/video-canvas-magic/
we knew the video is been segmented,the question is how to stitch them together.i think the real video element doesn't do the play work,it support the datasource,and draw the seagments each frame to the canvas element。
var v = document.getElementById('v');
var canvas = document.getElementById('c');
v.addEventListener('play', function(){
if(v.paused || v.ended) return false;
c.drawImage(v,0,0,w,h);
setTimeout(draw,20,v,c,w,h);
},false);
Youtube is using this feature only in browsers that support Media Source Extensions so it is up to the browser decide about all the rest because of this feature.
Is there some straightforward technique to play only a certain part of a HTML5 video? For example in a 30 second clip I would like to play only the part 5-20 sec. Additionally the rest of the video should not be accessible from the UI at all (meaning the video timeline should only show the 5-20 sec part).
I've been going through some HTML5 video players but none of them seem to be supporting this kind of functionality. If anyone knows a (good) way to implement this feature please give me a hint.
Thanks in advance!
Even though this question has already been marked as answered, here's something that may interest you and anyone else who stumbles across here: Specifying playback range.
It's part of the Media Fragment API and currently works in the latest versions of Firefox, Chrome and Safari 6+.
You can implement it there is a Player.Play() event in a players, whenever Play() called start a timer and call the Player.stop() on the specific time you want.
I got the same problem and I didn't found something that can solves this problem, what you can do is to implement your own controls and display the video with a canvas...
and if you are trying to implement this in IOS you will not be able to do it.
My thought would be to use custom controls, as I don't believe that functionality is available natively. All the functionality of html5 controls (play, pause, start at timestamp, etc.) can be called via javascript. In this case you are going to want to edit the currentTime variable.
So you may want to consider setting up your own slider, where the start of the slider represents your starting point, and the end your ending point. Set the video to not play on page load. Then on page load have a javascript function change the currentTime to your starting point. For stopping you could occasionally query the currentTime. I wouldn't use a timer as delays like slow loading could throw it off.
I am trying to implement a similar video in "Preview Mode". I am utilizing the methods stated above by adding an event listener and then pausing the video at a currentTime()=='x' postition. To prevent user from simply hitting play again, the currentTime listener wont allow playback past the 'x' time in the timeline so each time the user hits play, it is automatically instantly paused again. Furthermore, at time 'x' the video container will become hidden through CSS thus preventing interaction from the user with the video.