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  • Web Audio API and mobile browsers

    - by Michael
    I've run into a problem while implementing sound and music into an HTML game that I'm building. I'm using the Web Audio API, loading all the sound files with XMLHttpRequests and decoding them into an AudioBufferSourceNode with AudioContext.prototype.decodeAudioData(). It looks something like this: var request = new XMLHttpRequest(); request.open("GET", "soundfile.ogg", true); request.responseType = "arraybuffer"; request.onload = function() { context.decodeAudioData(request.response) } request.send(); Everything plays fine, but on mobile the decodeAudioData takes an absurdly long time for the background music. I then tried using AudioContext.prototype.createMediaElementSource() to load the music from an HTML Audio object, since they support streaming and don't have to load the whole file into memory at once. It looked something like this: var audio = new Audio('soundfile.ogg'); var source = context.createMediaElementSource(audio); var mainVolume = context.createGain(); source.connect(mainVolume); mainVolume.connect(context.destination); This loads much faster, but the audio volume isn't affected by the gain node. Works fine on desktop, so I'm assuming this is a bug/limitation of mobile Chrome (testing on Android). Is there actually no good, well-performing way to handle sound on mobile browsers or am just I doing something stupid?

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  • Audio programming resources

    - by rashleighp
    I've been very interested in the last few months about getting in to audio programming (I'm from a musical background). I've been a .NET developer for two years and have also done some objective c for an iPhone app recently. I realise I would probably need to work on my C++ chops and have been having a play around with FMOD EX and doing a lot of research into the industry. I was just wondering if anyone could suggest some good resources for audio programming (be they websites, podcasts, books, videos, online courses etc). Anything from Fourier analysis, low level coding, audio engine creation to audio APIs. I just want to learn as much as possible! Thanks in advance.

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  • Setting Up Audio on a Server Install

    - by tdcrenshaw
    I'm running on a clean install of 10.10 Server edition and have alsa-base, alsa-tools, alsa-utils, alsaplayer, and alsa-firmware-loader installed. At one point I installed pulseaudio, but I have since removed it. I've tried the following lspci | grep audio 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 01) 01:06.0 Multimedia audio controller: Creative Labs [SB Live! Value] EMU10k1X aplay -l aplay: device_list:235: no soundcards found... alsamixer can not open mixer: No such file or directory When I search for modules with find /lib/modules/`uname -r` | grep snd I do get a list of modules I'm not very experienced with alsa setup, so I'm not sure where to go from here

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  • How to stream semi-live audio over internet

    - by Thomas Tempelmann
    I want to write something like Skype, i.e. I have a constant audio stream on one computer and then recompress it in a format that's suitable for a latent internet connection, receive it on the other end and play it. Let's also assume that the internet connection is fairly modern and fast, i.e. DSL or alike, no slow connections over phone and such. The involved computers will also be rather modern (Dual Core Intel CPUs at 2GHz or more). I know how to handle the audio on the machines. What I don't know is how to transmit the audio in an efficient way. The challenges are: I'd like get good audio quality across the line. The stream should be received without drops. The stream may, however, be received with a little delay (a second delay is acceptable). I imagine that the transport software could first determine the average (and max) latency, then start the stream and tell the receiver to wait for that max latency before starting to play the audio. With that, if the latency doesn't get any higher, the entire stream will be playable on the other side without stutter or drops. If, due to unexpected IP latencies or blockages, the stream does get cut off, I want to be able to notice this so that I can take actions (e.g. abort the stream) and eventually start a new transmission. What are my options if I want do use ready-made software for the compression and tranmission? I have no intention to write my own audio compression engine, really. OTOH, I plan to sell the solution in a vertical market, meaning I can afford a few dollars of license fees per copy, but not $100s. I guess the simplest solution would be to just open a TCP stream, send a few packets back and forth to determine their running time (or even use UDP for that), then use the results as the guide for my max latency value, then simply fire the audio data in its raw form (uncompressed 16 bit stereo), along with a timing code over the TCP connection. The receiver reads the data and plays it with the pre-determined delay. That might just work with the type of fast connection I expect. I just wonder if there are better solutions to reach this goal, with better performance (lower latency) and less data (compressed). BTW, I first try to implement this on OS X, but might want to do it on Windows, too, if it proves successful.

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  • HTTP Live Streaming, FFMPEG & FFSERVER, and iPhone OS 3

    - by jcnnghm
    In iPhone OS 3, Apple has introduced HTTP Live Streaming which should allow live streaming of video from the internet. I am currently operating a webcam, which feeds into my server, and is then converted into a flv stream by ffmpeg, and streamed back out using ffserver. Does anyone know how to setup a video stream the iPhone can use using ffmpeg and ffserver? I should be able to re-encode into just about any format on the fly.

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  • Audio Streaming API's: Wifi vs what?

    - by Moshe
    I've noticed certain radio apps, that some stations required wifi and others did not. What were those other stations possibly using? Are there other methods of streaming audio on iOS? Apparently, I was not clear in my question before. I'm asking in terms of API's. Is there an API to interact directly with say, FM radio, on iOS? Is wifi the only way of streaming audio?

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  • Record audio via MediaRecorder

    - by Isuru Madusanka
    I am trying to record audio by MediaRecorder, and I get an error, I tried to change everything and nothing works. Last two hours I try to find the error, I used Log class too and I found out that error occurred when it call recorder.start() method. What could be the problem? public class AudioRecorderActivity extends Activity { MediaRecorder recorder; File audioFile = null; private static final String TAG = "AudioRecorderActivity"; private View startButton; private View stopButton; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); startButton = findViewById(R.id.start); stopButton = findViewById(R.id.stop); setContentView(R.layout.main); } public void startRecording(View view) throws IOException{ startButton.setEnabled(false); stopButton.setEnabled(true); File sampleDir = Environment.getExternalStorageDirectory(); try{ audioFile = File.createTempFile("sound", ".3gp", sampleDir); }catch(IOException e){ Toast.makeText(getApplicationContext(), "SD Card Access Error", Toast.LENGTH_LONG).show(); Log.e(TAG, "Sdcard access error"); return; } recorder = new MediaRecorder(); recorder.setAudioSource(MediaRecorder.AudioSource.MIC); recorder.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP); recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB); recorder.setAudioEncodingBitRate(16); recorder.setAudioSamplingRate(44100); recorder.setOutputFile(audioFile.getAbsolutePath()); recorder.prepare(); recorder.start(); } public void stopRecording(View view){ startButton.setEnabled(true); stopButton.setEnabled(false); recorder.stop(); recorder.release(); addRecordingToMediaLibrary(); } protected void addRecordingToMediaLibrary(){ ContentValues values = new ContentValues(4); long current = System.currentTimeMillis(); values.put(MediaStore.Audio.Media.TITLE, "audio" + audioFile.getName()); values.put(MediaStore.Audio.Media.DATE_ADDED, (int)(current/1000)); values.put(MediaStore.Audio.Media.MIME_TYPE, "audio/3gpp"); values.put(MediaStore.Audio.Media.DATA, audioFile.getAbsolutePath()); ContentResolver contentResolver = getContentResolver(); Uri base = MediaStore.Audio.Media.EXTERNAL_CONTENT_URI; Uri newUri = contentResolver.insert(base, values); sendBroadcast(new Intent(Intent.ACTION_MEDIA_SCANNER_SCAN_FILE, newUri)); Toast.makeText(this, "Added File" + newUri, Toast.LENGTH_LONG).show(); } } And here is the xml layout. <?xml version="1.0" encoding="utf-8"?> <RelativeLayout xmlns:android="http://schemas.android.com/apk/res/android" android:id="@+id/RelativeLayout1" android:layout_width="fill_parent" android:layout_height="fill_parent" android:orientation="vertical" > <Button android:id="@+id/start" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_alignParentTop="true" android:layout_centerHorizontal="true" android:layout_marginTop="146dp" android:onClick="startRecording" android:text="Start Recording" /> <Button android:id="@+id/stop" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_alignLeft="@+id/start" android:layout_below="@+id/start" android:layout_marginTop="41dp" android:enabled="false" android:onClick="stopRecording" android:text="Stop Recording" /> </RelativeLayout> And I added permission to AndroidManifest file. <?xml version="1.0" encoding="utf-8"?> <manifest xmlns:android="http://schemas.android.com/apk/res/android" package="in.isuru.audiorecorder" android:versionCode="1" android:versionName="1.0" > <uses-sdk android:minSdkVersion="8" /> <application android:icon="@drawable/ic_launcher" android:label="@string/app_name" > <activity android:name=".AudioRecorderActivity" android:label="@string/app_name" > <intent-filter> <action android:name="android.intent.action.MAIN" /> <category android:name="android.intent.category.LAUNCHER" /> </intent-filter> </activity> </application> <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/> <uses-permission android:name="android.permission.RECORD_AUDIO" /> </manifest> I need to record high quality audio. Thanks!

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  • VLC streaming testbench

    - by Vineet Menon
    Has anyone tried streaming media with VLC as server? I want to deploy VLC as streaming server, but my department didn't had a nice experience with VLC streaming. My question is has anyone tried VLC streaming over LAN with as many as 200 clients? What were the precautions to be taken before going for the actual showdown? What kind of transport stream is better for a smoother live streaming? Are any test bench I can use to convince my superiors?

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  • Streaming in justin.tv using Ubuntu 11.10

    - by Orlando Gonzales
    I used to streaming videos in justin.tv..using Windows 7 and I had no problems. But a month ago, I installed Ubuntu 11.10 (32 bits) in my notebook and now I can't streaming videos. The problem is that when I click on the button "Go Live!", it´s supposed that the screen where I have to clic on "Allow" it should appear, but it doesn't. The web only displays the message: "You can't start until you click allow below.", but the "allow button" doesn't appear, instead of that I only see a black screen. I guessed that it was a problem with the Adobe Flash Player, but I have the lastest version for Linux installed and I have no problems with Youtube. Also, my webcam works without problem in others Ubuntu's applications. I hope you can help me. This are my notebook's features: Toshiba Satellite C645-SP4135L / 3GB RAM / Intel Core i3-2310M 2.10GHz x 4 / HHD:320GB Thanks!! Orlando

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  • How to fix these compiler errors?

    - by Sandra Schlichting
    I have this source code from 2001 that I would like to compile. It gives this: $ make g++ -O99 -Wall -DLINUX -pedantic -c -o audio.o audio.cpp In file included from audio.cpp:7: audio.h:14: error: use of enum ‘mad_flow’ without previous declaration audio.h:15: error: use of enum ‘mad_flow’ without previous declaration audio.h:17: error: use of enum ‘mad_flow’ without previous declaration audio.cpp: In function ‘mad_flow audio::input(void*, mad_stream*)’: audio.cpp:19: error: new declaration ‘mad_flow audio::input(void*, mad_stream*)’ audio.h:14: error: ambiguates old declaration ‘int audio::input(void*, mad_stream*)’ audio.h:11: error: ‘size_t audio::stream::BufferPos’ is private audio.cpp:23: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:23: error: within this context audio.h:10: error: ‘char* audio::stream::Buffer’ is private audio.cpp:26: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:26: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferPos’ is private audio.cpp:27: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:27: error: within this context audio.cpp: In function ‘mad_flow audio::output(void*, const mad_header*, mad_pcm*)’: audio.cpp:49: error: new declaration ‘mad_flow audio::output(void*, const mad_header*, mad_pcm*)’ audio.h:15: error: ambiguates old declaration ‘int audio::output(void*, const mad_header*, mad_pcm*)’ audio.cpp: In function ‘mad_flow audio::error(void*, mad_stream*, mad_frame*)’: audio.cpp:83: error: new declaration ‘mad_flow audio::error(void*, mad_stream*, mad_frame*)’ audio.h:17: error: ambiguates old declaration ‘int audio::error(void*, mad_stream*, mad_frame*)’ audio.cpp: In constructor ‘audio::stream::stream(const char*)’: audio.cpp:119: error: ‘input’ was not declared in this scope audio.cpp:122: error: ‘output’ was not declared in this scope audio.cpp:123: error: ‘error’ was not declared in this scope make: *** [audio.o] Error 1 audio.h contains #include <stdlib.h> #include "mad.h" namespace audio { class stream { private: char* Buffer; size_t BufferSize, BufferPos; struct mad_decoder Decoder; friend enum mad_flow input(void* Data, struct mad_stream* MadStream); friend enum mad_flow output(void* Data, const struct mad_header* Header, struct mad_pcm* PCM); friend enum mad_flow error(void* Data, struct mad_stream* MadStream, struct mad_frame* Frame); public: stream(const char* FileName); ~stream(); void play(); }; } I have tried to just insert enum mad_flow {}; but that just gave a new problem. Can anyone see how to fix this?

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  • Recording Audio from WMP Stream

    - by Jonathan Sampson
    I'm sitting here listening to a radio show that is being broadcast live over an internet stream, but I would like to keep bits and pieces for later-enjoyment. Is there a way I can easily record streams in real-time? I should note also (not sure if it's necessary or not) that this stream requires me to first login before listening.

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  • App / protocol to tune into live audio and video based on schedule or subscription

    - by Richard
    Many of us have embraced the podcasting revolution enabled by rss feeds and podcatchers. Alot of sites now broadcast live streams of what is eventually edited into a podcast. In most cases listening to the live stream gets you the info several days sooner then the podcast. So I was wondering if anybody knows of a notification protocol / app that allows me to auto tune into certain streams when they go live, or based on a schedule. I imagine twitter could be used for the notification but It'd be better not to be tied to a proprietary service. Example podcasts / live streams noagenda.squarespace.com jupiterbroadcasting.com twit.tv

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  • Streaming server required with JW Player?

    - by Aaron
    Currently, a site I developed plays mp3 files directly in JW Player using the file attribute and public URLs to the mp3 file. This is now an issue with the client for legal reasons, and they now need to stream the audio files so that the users can't open up their cache and nab the files directly after downloading. The JW player site has a bunch of examples for streaming video, but nothing for audio. Is it possible to stream audio files with JW player, and do we have to pay a lot of money for a streaming provider? Is it possible to do on the local php server?

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  • help me pick the right iPhone audio class - MPMoviePlayer vs AVAudioPlayer vs MPMusicPlayer

    - by huevos de oro
    Does anyone know of a good tutorial on the distinction between the MPMoviePlayer vs AVAudioPlayer vs MPMusicPlayer? I want to play audio from an mp3 file available at an external URL. Ideally it is played in an iPod-like audio view. I toyed with MPMoviePlayer but it appears to be more suitable for video, as when audio starts a "movie playing" message displays, the controls disappear and a white quicktime splash page displays. I would like the standard ipod audio controls to display all the time, and to customize the image behind them.

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  • Is there an audio recording application/tool that has Tivo-like functionality?

    - by Bob
    I do a lot of live speech recording that requires me to quickly jump back and then transcribe a particular piece of the audio, then go back to recording again, while still maintaining the full audio file. So Far I've done this by splitting the audio and running one line to a recorder (for the whole audio), and one to my computer. Then I use something like Audacity to record, and then stop/go back whenever I hear something worth transcribing. This requires me to stop the recording, then start it up again and I end up missing chunks of the speech I'm listening to. Is there a tool that would let me rewind, then listen again and continue listening at a buffered distance from the audio recording, the way Tivo does with television shows?

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  • HTML Audio performance

    - by user1888309
    I'm working on HTML drum machine, and I`ve met some performance issues, rhythm start to break if BPM is higher than 110 but I'm expecting to make it work on BPM over 180. I guess that it can be related with format or codec of audio files, however it also maybe that my code is not very optimised (as I can see from JS CPU profiling it's not). So I'm expecting you guys give me some code review or some hints on optimisation. Although all similar projects I've found on internet didn't work good and maybe it's just restrictions of Audio API. By the way, it's very raw and sounds works only on Chrome under Mac OS, so any advise on audio encoding for web also would be great Project on Github pages Screenshot of Groove which breaks UPDATE Ok, I've found that I was encoding audio files incorrectly, after fixing that rhythm stopped breaking, and also it started working in Mozilla. But still there are issues on windows OS.

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  • Streaming Media Server and Hosting

    - by Ryan Max
    My partner and I have a webcam site that basically runs the old-school method....Every 0.5 seconds the javascript reloads the image in the browser from the webcam. However we are wanting to upgrade to a streaming media server to get higher quality video, and possibly audio. We aren't tied to any one specific file format or server type, as of right now we are leaning towards slicehost (as scalability is important), and installing darwin streaming server or wowza. This is meant to be a live stream. Does anyone have any suggestions for hosts/server software?

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  • Multiple audio sources on a single gameObject in unity

    - by angryInsomniac
    So, I have an audio system set up wherein I have loaded all my audio clips centrally and play them on demand by passing the requesting audioSource into the sound manager. However, there is a complication wherein if I want to overlay multiple looping sounds, I need to have multiple audio sources on an object, which is fine , so I created two in my script instantiated them and played my clips on them and then the world went crazy. For some reason, when I create two audio Sources in an object only the latest one is ever used, even if I explicitly keep objects separated, playing a clip on one or the other plays the clip on the last one that was created, furthermore, either this last one is not created in the right place or somehow messes with the rolloff rules because I can hear it all across my level, havign just one source works fine, but putting a second one on it causes shit to go batshit insane. Does anyone know the reason / solution for this ? Some pseudocode : guardSoundsSource = (AudioSource)gameObject.AddComponent("AudioSource"); guardSoundsSource.name = "Guard_Sounds_source"; // Setup this source guardThrusterSource = (AudioSource)gameObject.AddComponent("AudioSource"); guardThrusterSource.name = "Guard_Thruster_Source"; // setup this source // play using custom Sound manager soundMan.soundMgr.playOnSource(guardSoundsSource,"Guard_Idle_loop" ,true,GameManager.Manager.PlayerType); // this method prints out the name of the source the sound was to be played on and it always shows "Guard_Thruster_Source" even on the "Guard_Idle_loop" even though I clearly told it to use "Guard_Sounds_source"

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  • Play an audio file using RemoteIO and Audio Unit

    - by NeilMonday
    I am looking at Apple's 'aurioTouch' example for the iPhone and I would like to play an mp3 or wav instead of using the built in mic. I am very new to the audio portion of iPhone programming, but I think I need to modify the SetupRemoteIO(...) function and replace the AudioComponent named 'comp' with a custom AudioComponent that plays a file. Basically I want the app to function exactly the same as the original, but with an audio file as the input instead of the mic.

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  • Audio decoding delay when changing the audio language

    - by mahendiran.b
    My gstreamer Pipeline is like this Approach1 --------------input-selector->Queue->AduioParser->AudioSink | Souphttpsrc->tsdemux-->| | --------------- Queue->videoParser->videoSink In this approach 1, there is a delay in audio decoding when I toggle between various audio language. Approach2 ------ input-selector-> Queue->AduioParser->AudioSink | Souphttpsrc->tsdemux---multiqueue>| | ------- Queue->videoParser->VideoSink But there is no delay is observed in approach2. Can anyone please explain the reason behind this ? what is the specialty of multiqueue here?

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  • Embed audio broadcasting on web page

    - by giargo
    Hi, I'd like to embed simple audio player on my webpage and I want it to get the audio from a stream broadcasted from my server. I read I can use IceCast on my web-server, getting an audio stream from a client using IceS (or this is what i got from other questions and articles) but once I have my stream, IceCast is supposed to broadcast it on an URL, that can be opened from pkayers like winamp or similar. I've found out this is quite a rare topic, usually people just want to broadcast "radio" where files are taken from a static playlist. In this case I have to get a stream from an IceCast URL and embed it with a player on a web page. Thank.

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  • Audio libraries for PC indie games [closed]

    - by bluescrn
    Possible Duplicate: Cross-Platform Audio API Suggestions What options are out there these days for audio playback/mixing in C++? Primarily for Windows, but portability (particularly to Mac and iOS) would be desirable. For a small indie game, potentially commercial, though - so I'm looking for something free/low-cost. My requirements are fairly basic - I don't need 3D sound, or many-channels - simple stereo is fine. Just need to be able to mix sound effects and a music stream, maybe decoding one or more compressed audio formats (.ogg/.mp3 etc), with all the basic controls over looping, pitch, volume, etc. Is OpenAL more-or-less the standard choice, or are there other good options out there?

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  • Verizon SongID - How is it programmed?

    - by CheeseConQueso
    For anyone not familiar with Verizon's SongID program, it is a free application downloadable through Verizon's VCast network. It listens to a song for 10 seconds at any point during the song and then sends this data to some all-knowing algorithmic beast that chews it up and sends you back all the ID3 tags (artist, album, song, etc...) The first two parts and last part are straightforward, but what goes on during the processing after the recorded sound is sent? I figure it must take the sound file (what format?), parse it (how? with what?) for some key identifiers (what are these? regular attributes of wave functions? phase/shift/amplitude/etc), and check it against a database. Everything I find online about how this works is something generic like what I typed above. From audiotag.info This service is based on a sophisticated audio recognition algorithm combining advanced audio fingerprinting technology and a large songs' database. When you upload an audio file, it is being analyzed by an audio engine. During the analysis its audio “fingerprint” is extracted and identified by comparing it to the music database. At the completion of this recognition process, information about songs with their matching probabilities are displayed on screen.

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