With VAST complaint video ads, you may get a "nested" VAST (VAST Wrappers), which will eventually yield a MediaFile that is a media that you could preload (e.g. MP4/WebM/flv, etc.).
However in the case of VPAID (flash, yuck!), which contains swf as the media files, at the end of the wrappers chain there lies yet another swf - for example, an "ad manager" swf.
So the effect of preloading a VPAID ad is essentially not very useful, because you could not extract the actual media file for preloading merely by inspecting the VAST XMLs. At best - you may preload the final ads manager swf.
The final swf, when played, may then begin to fetch ads from different providers (e.g. brightroll, liverail, etc.), and this process takes a while. At the end of this, an ad might pop up.
Is it allowed to try to preload the actual eventual ad media, or to find out if there is a "fill" (an ad media will be yielded by the ads manager)? For example, by "playing" the ads manager swf in the background, hidden and muted, this might be possible, but it sounds a bit fishy (what if the user eventually does not reach the point where the ad should actually be shown? it would count as fraud).
The answer is No,
When you load the VPAID swf --> the ad server count impression and start the bid between the ads providers.
You can only load the swf when the user is watching the ad/movie.
Related
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).
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 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.
I have a combined Flash Builder/Flash Pro project. Because of the hassles involving in maintaining sound assets on the timeline, my sounds are all embedded into Class files, like:
[Embed (source="/mp3/Welcome_01_V.mp3", mimeType="audio/mpeg")]
private static const WELCOME_1:Class;
These files are then referenced by the base Classes for the symbols that need them, embedded for Actionscript on Frame 10 (because the second frame label is on Frame 10 to give space for you to read the first one).
What I'm finding is that a few of these Sounds don't play all the way through, but the SoundChannel dispatches the "soundComplete" event, and its final position matches the Sound's length.
All sounds are converted from wav to mp3 at 44Hz / 16 kbps. I faked out the compiler to avoid a reference to Flex by including a dummy SoundAsset that extends Sound.
I don't know what other steps to take to debug this. Is there a way to figure out whether the problem is on the compile side or on the run side?
Updated
More things I have tried:
Looked at the Size report: The nonworking sounds were smaller in
their embedded form than the source mp3
Got rid of my own BitmapAsset and let Flash link in the Flex Framework and do whatever that does (definitely worse)
Dropped the encoding from 44 kHz to 22 kHz (no improvement or worse)
Dropped the bit rate to 8kbps (the lowest dbPowerAmp, the tool I use, supports). This usually helps somewhat, but I still usually use a word or two from the end of the file
Dropped both parameters in the encoding. This helped a few that just dropping the bit rate didn't, but not all files. Plus it sounds tinny.
Thanks!
For Flash audio, I recommend importing the sound assets into a FLA using wav files if you have the high quality source wavs. Otherwise, you can consider converting your mp3 into a wav as well. Then set the FLA export settings to the quality you want and Flash will convert your wavs into its own format at the quality you set with hopefully less issues.
Once you do that, you can export the sound symbol for actionscript in your library and set a class name just like how you would embed it.
One other trick I use is I have one FLA just for sound assets which can be used to store as big waves as I want. And when I export that, it becomes a small SWF file which I can then embed in my main application. That way, I never have Flash reconvert the wavs to the swf every single time I export the swf. Instead it just copies the swf data which is much faster as well.
[Embed(source="Audio/Sfx.swf", symbol="WELCOME_1_WAV")]
private static const WELCOME_1:Class;
If you are having audio cut off issues in Flash Pro, you may want to check your frame rate.
I had an issue with sounds cutting off (in Flash pro CC 2014). My issue turned out to be related to the frame rate being set to 25 rather than the default 24. I have been using 25 to resolve an issue unrelated to anything in this project, so my solution was to change the FPS to 24, which invoked the necessity to move all of the synchronized animations to re-align with the corresponding audio.
Why long(ish) audio tracks get cut off at the end when the frame rate is at 25 regardless of using proper keyframing is a mystery. This solved the symptoms however, so if you are having audio cut off issues in Flash Pro, you may want to check your frame rate.
My symptoms were specifically when an audio clip was particularly long, and deep into the time line.
What worked for me: I opened the audio files in an audio editor and added a few seconds of silence to the end.
Good luck! - J.Hall
So, hypothetical situation here:
I have an SWF that's 30MB. Sound files (music) make up 25MB, art and other things make up the remaining 5MB.
Would it be possible for me to load the 5MB of necessary art and other things first to allow the user to operate the app, then after that's all loaded and they are operating the app, load the remaining 25MB of sound files in the background?
UPDATE:
Loading SWF (or other entities) externally is not an option.
You can do this by modifying the compilation settings and strategically placing the sound.
By default, anything placed on the timeline will be loaded sequentially; the Flash Player will play the SWF as soon as the first frame is loaded. If you use the default compilation settings, all library content will be placed in the first frame, so the movie doesn't start until everything is available.
You can modify these settings, however, to allow for a more sequential load order: For each of your library elements, you can uncheck "export into frame 1" in the properties window. Now these elements won't be loaded, until they appear in the main timeline. This way, if you place your content carefully, you can allow for all important elements to be loaded in the first frame, or if you have a progress bar, until the main movie starts, while all streaming elements load with the animation, which has to be placed accordingly. Make sure though, that you don't leave anything out (by not placing it on the timeline), or call elements from ActionScript before they are loaded completely. It is very important to test this thoroughly, because if anything goes wrong in the load order, your entire SWF might stop working.
Also, remember that the SWF loads sequentially: If you have a sound in, say, frame 300, and another in frame 1000, the one in 300 will be loaded first. If you jump to frame 1000 from a menu in frame 10, you have to take into account that the frame might not be loaded yet. So there has to be some sort of checking mechanism (framesLoaded) and/or dialog to inform the user about additional loading time, and prevent the application from crashing.