I have a simple implementation of an audio mixer using the web audio API. I'm basically creating a buffer for each source, adding some filters and then connecting to the destination:
this.track.connect(this.highPassFilter);
this.highPassFilter.connect(this.lowShelfFilter)
this.lowShelfFilter.connect(this.highShelfFilter)
this.highShelfFilter.connect(this.midFilter);
this.midFilter.connect(this.panner)
this.panner.connect(this.gain)
this.gain.connect(this.ctx.destination)
What I want to do is be able to create a master gain and master compressor. Seeing as all my 'tracks' are connected directly to the context.destination and then all essentially played in unison when the play button is pressed, how can I connect a master gain in to the chain?
I think you already know how to do this. What you need to do is to create your master gain and compressor, connect those two together and then connect all your tracks to them, like so:
[track] -> masterGain -> masterCompressor -> ctx.destination
Or the reverse, however you want it (masterCompressor -> masterGain -> ctx.destination).
Related
I am working on a Chromecast custom receiver app, built on top of the sample app provided by Google (sampleplayer.CastPlayer)
The app manages a playlist, I would like the player to move on to the next item in the list after a video fails to play for whatever reason.
I am running into a situation where, after a video fails to load because of a network error, the player becomes unresponsive: in the 'onError_()' handler, my custom code will do this
var queueLoadRequest = ...
var mediaManager = ...
setTimeout (function(){mediaManager.queueLoad(queueLoadRequest)}), 5000
...the player does receive the LOAD event according to the receiver logs, but nothing happens on the screen, the player's status remains IDLEand the mediaManager.getMediaQueue().getItems() remains undefined. Same result trying to use the client controller to try to load a different video.
I have tried to recover with mediaManager.resetMediaElement() and player.reset() in the onError_ handler, but no luck.
For reference, here is a screenshot of the logs (filtered for errors only) leading up to the player becoming unresponsive. Note that I am not interested in fixing the original error, what I need to figure out is how to recover from it:
My custom code is most likely responsible for the issue, however after spending many hours + stripping the custom code to a bare minimum in an effort to isolate the responsible bit of code, I have not made any progress. I am not looking for a fix but rather for some guidance in troubleshooting the root cause: what could possibly cause the Player to become unresponsive? or alternatively how can one recover from an unresponsive Player?
I'm rather new to the Lync 2013 SDK (been using it for a couple weeks now) and have been able to figure out mostly everything I need except for this...
When I'm joining a conference (using ConversationManager.JoinConference()) it joins fine. However, in certain cases (not all), I want to then connect the AVModality on the conference. Sometimes it works, sometimes it just sits in "Connecting" and never connects (even though I've called EndConnect).
What I've found is the setting in Skype's Options -> Skype Meetings -> Joining conference calls section, seems to override my code. Maybe a race condition?
When the setting is "Do not join audio" and "Before I join meetings, ask me which audio device I want to use" is NOT CHECKED (meaning I get no prompt when joining): the conference joins, the AVModality goes Disconnected -> Connecting -> Disconnected. Then my code triggers a BeginConnect and the AVModality goes Disconnected -> Connecting - and never resolves (sometimes I get a fast busy tone audio sound).
When the "Before I join meetings, ask me which audio device I want to use" IS CHECKED (meaning I get the prompt): the conference joins, the prompt asks how to connect, if I select Skype for business - it connects audio fine (expected). Interestingly, if I hang up the call using the Lync UI (AVModality goes to Disconnected), it then immediately connects back again (assuming my BeginConnect doing this).
Here's where it gets really convoluted:
If I call BeginConnect when the state is Connecting on the AVmodality within the ModalityStateChanged event handler... the following happens:
Conference joins, prompt asks me how to connect (AVmodality state is "Connecting" at this point until a decision is made on the prompt) - this means my BeginConnect fires. Then if I choose "Do not join audio" in the prompt... the AVModality status goes Connecting -> Disconnected -> Connecting -> Joining -> Connected. So - my BeginConnect is already in progress and still works in this case so long as it fires BEFORE the selection of "Do not join audio".
So I'm wondering if the "Do not join audio" selection (whether with or without the prompt) actually sets some other properties on something which prevents the AVModality from being connected after that point without doing some additional hocus pocus? If so - I'd like to know the additional hocus pocus I need to perform :)
Thanks for any and all help!
It's come down to this... whether the conference joining does join the audio or not - I've handled every scenario except one, which I still can't figure out:
1. I need the conference audio to be joined, but the user selects to NOT join the audio (either on the prompt, or from the Skype options settings).
In this case - I have added an event handler to the modality state change event, and when the NewState == Disconnected, I trigger a BeginConnect on the modality itself. This works fine. Within the callback, I have the EndConnect call. However - the AVModality state continues to stay in "Connecting" and never resolves to being connected. On the UI - it shows the audio buttons, but all grayed out (like normal when it's connecting). I'm not sure how to make it finish connecting?
Here's a snippet of code:
if (merge)
{
myHandler = delegate (object sender1, ModalityStateChangedEventArgs e1)
{
AVModality avModality = (AVModality)sender1;
Globals.ThisAddIn.confConvo = avModality.Conversation;
if (e1.NewState == ModalityState.Connected)
{
DialNumberInSkype(meetingInfo);
avModality.ModalityStateChanged -= myHandler;
}
if (e1.NewState == ModalityState.Disconnected)
{
object[] asyncState = { avModality, "CONNECT" };
avModality.BeginConnect((ar) =>
{
avModality.EndConnect(ar);
DialNumberInSkype(meetingInfo);
}, asyncState);
avModality.ModalityStateChanged -= myHandler;
}
};
}
EDIT:
For some reason, I'm not able to add a comment right now...
I tried setting the endpoint as you suggested. However, I get an ArgumentException error "Value does not fall within the expected range." So I tried hardcoding the uri value in the CreateContactEndpoint to "sip:my_login#domain.com" (except real value of course) - and got the same ArgumentException error. I added a breakpoint before this and was able to see the value for the avModality.Endpoint - and it is actually set to me the entire time... it's not null or unset when I'm trying to call BeginConnect.
When JoinConference() is invoked audio modality will be connected even without explicitly invoking BeginConnect().
When prompt asking for audio device selection is shown(when ask before join option is set in skype) conversation property ConferenceEscalationProgress will be having value AwaitingJoinDialogResponse.
Setting conversation property ConferenceJoinDialogCompleted as true will initiate Modality connection even though the prompt is not closed.
Edited
If do not join audio is selected, modality will be disconnected, at this point you are trying to invoke BeginConnect(). Try setting modality endpoint before invoking BeginConnect().
conversation.Modalities[ModalityTypes.AudioVideo].Endpoint = lyncClient.Self.Contact.CreateContactEndpoint(lyncClient.Self.Contact.Uri);
I have a EmbeddedMediaPlayerComponent and I want to check before playing if the video has audio track.
The getMediaPlayer().getAudioTrackCount() method works fine but only when I play the video and I am inside the public void playing(MediaPlayer mp) event.
I also tryed
getMediaPlayer().prepareMedia("/path/to/media", null);
getMediaPlayer().play();
System.out.println("TRACKS: "+getMediaPlayer().getAudioTrackCount());
But it does not work. it says 0.
I also tryed:
MediaPlayerFactory factory = new MediaPlayerFactory();
HeadlessMediaPlayer p = factory.newHeadlessMediaPlayer();
p.prepareMedia("/path/to/video", null);
p.parseMedia();
System.out.println("TRACKS: "+p.getAudioTrackCount());
But it also says -1. Is there a way I can do that ? or using another technique?
The track count is not metadata, so using parseMedia() here is not going to help.
parseMedia() will work to get e.g. ID3 tag data, title, artist, album, and so on.
The track data is usually not available until after the media has started playing, since it is the particular decoder plugin that knows how many tracks there are. Even then, it is not always available immediately after the media has started playing, sometimes there's an indeterminate delay (and no LibVLC event).
In applications where I need the track information before playing the media, I usually would use something like the native MediaInfo application and parse the output - this has a plain-text out format, or an XML output format and IIRC the newer versions have a JSON output format. The downside is you have to launch a native process to do this, I use CommonsExec for things like this. It's pretty simple and does work even though it's not a pure Java solution, but neither is vlcj!
A slight aside if you did actually want the meta data there is an easier way, just use
this method on the MediaPlayerFactory:
public MediaMeta getMediaMeta(String mediaPath, boolean parse);
This gives you the meta data without having to prepare, play or parse media.
Within our organisation we use quite a few different models of telephone sets. The only thing they have in common, apart from the dialpad, is the ability to "send" hook flash. I prefer using this type of signaling for attended transfers above combinations of the usual dialpad keys to prevent the other end from receiving DTMF-tones (to prevent unwanted interactions with IVRs or bothering people on the other end).
2 questions:
How is a flash hook represented in features.conf? According to RFC2833 section 3.10 (DTMF Events) and this article (which is about a ZAP- instead of SIP-configuration, thus my doubt... see next question, also), it should be just "flash".
From my Google-quest I've learned that hook flash gets ignored by the PBX when using the SIP-protocol in Asterisk... I do get an error message when sending it: "WARNING[26159]: chan_sip.c:6487 sip_indicate: Don't know how to indicate condition 9". Is there a way to fix it/work around it?
Asterisk version: 1.8.3.2
Using "info" for dtmfmode
Tnx in advance!
In most cases you have in you adapter settings what to do with hook. IF you have, you can change that to transfer code.
Update: after code review i can say that DTMF 16 received ok and sended in 1.8.x. BUT features.c have no any action on flash(event 16)
So posible create audiohook application for asterisk to change that DTMF 16 to 2 DTMF values or invoke transfer. Will work for DTMF method SIPInfo, and such patch complexity is below-average(5-6 hours for expert)
Netty-Gurus,
I've been wondering if there is a shortcut/Netty-Utility/smart-trick
for connecting the input of one Channel to the output of
an other channel. In more details consider the following:
Set-Up a Netty (http) server
For an incoming MessageEvent get its ChannelBuffer
and pipe its input to a NettyClient-ChannelBuffer
(which is to be set up along the lines of the NettyServer).
I'm interested in how to achieve bullet-point 3. since my first
thoughts along the lines
// mock messageReceived(ChannelHandlerContext ctx, MessageEvent e):
ChannelBuffer bufIn = (ChannelBuffer) e.getMessage();
ChannelBuffer bufOut = getClientChannelBuffer();// Set-up somewhere else
bufOut.write(bufIn);
seem to me awkward because
A. I have to determine for each and every messageReceived-Event
the target ChannelBuffer
B. To much Low-Level tinkering
My wish/vision would be to connect
--> the input of one Channel
--> to the output of an other channel
and let them do their I/O without any additional coding.
Many thanks in advance!,
Traude
P.S: Issue has arisen as I'm trying to dispatch the various HTTP-requests to the
server (one entry point) to several other servers, depending on
the input content (mapping based on the first HTTP request line).
Obviously, I also need to do the inverse trick -- pipeing back client
to server -- but I guess it'll be similar to the solution of
the question before.
Looks like you need to use a multiplexer in you business handler. The business handler could have a map. With key as "first http request line" and value as the output channel for the server. Once you do a lookup you just do a channel.write(channelBuffer);
Also take a look at bruno de carvalho's tcp tunnel, which may give you more ideas on how to deal with these kind of requirements.