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  • waveInProc / Windows audio question...

    - by BTR
    I'm using the Windows API to get audio input. I've followed all the steps on MSDN and managed to record audio to a WAV file. No problem. I'm using multiple buffers and all that. I'd like to do more with the buffers than simply write to a file, so now I've got a callback set up. It works great and I'm getting the data, but I'm not sure what to do with it once I have it. Here's my callback... everything here works: // Media API callback void CALLBACK AudioRecorder::waveInProc(HWAVEIN hWaveIn, UINT uMsg, DWORD dwInstance, DWORD dwParam1, DWORD dwParam2) { // Data received if (uMsg == WIM_DATA) { // Get wav header LPWAVEHDR mBuffer = (WAVEHDR *)dwParam1; // Now what? for (unsigned i = 0; i != mBuffer->dwBytesRecorded; ++i) { // I can see the char, how do get them into my file and audio buffers? cout << mBuffer->lpData[i] << "\n"; } // Re-use buffer mResultHnd = waveInAddBuffer(hWaveIn, mBuffer, sizeof(mInputBuffer[0])); // mInputBuffer is a const WAVEHDR * } } // waveInOpen cannot use an instance method as its callback, // so we create a static method which calls the instance version void CALLBACK AudioRecorder::staticWaveInProc(HWAVEIN hWaveIn, UINT uMsg, DWORD_PTR dwInstance, DWORD_PTR dwParam1, DWORD_PTR dwParam2) { // Call instance version of method reinterpret_cast<AudioRecorder *>(dwParam1)->waveInProc(hWaveIn, uMsg, dwInstance, dwParam1, dwParam2); } Like I said, it works great, but I'm trying to do the following: Convert the data to short and copy into an array Convert the data to float and copy into an array Copy the data to a larger char array which I'll write into a WAV Relay the data to an arbitrary output device I've worked with FMOD a lot and I'm familiar with interleaving and all that. But FMOD dishes everything out as floats. In this case, I'm going the other way. I guess I'm basically just looking for resources on how to go from LPSTR to short, float, and unsigned char. Thanks much in advance!

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  • PhP/HTML play button [migrated]

    - by Marian
    I'm wanting to make my own small webpage, I've got a domain Saoo.eu As you see there is a small play button in the corner witch plays a playlist. Is there anyway to have that playbutton on each page I'd add in the future without resetting every time the page changes? Am I forced to use iFrames for that? This is my player code <button id="audioControl" style="width:30px;height:25px;"></button> <audio id="aud" src="" autoplay autobuffer /> Script: $(document).ready(function() { $('#audioControl').html('II'); if(Modernizr.audio && Modernizr.audio.mp3) { audio.setAttribute("src",'http://daokun.webs.com/play0.mp3'); } else if(Modernizr.audio && Modernizr.audio.wav) { audio.setAttribute("src", 'http://daokun.webs.com/play0.ogg'); } }); var audio = document.getElementById('aud'), count = 0; $('#audioControl').toggle( function () { audio.pause(); $('#audioControl').html('>'); }, function () { audio.play(); $('#audioControl').html('II'); } ); audio.addEventListener("ended", function() { count++; if(count == 4){count = 0;} if(Modernizr.audio && Modernizr.audio.mp3) { audio.setAttribute("src",'http://daokun.webs.com/play'+count+'.mp3'); } else if(Modernizr.audio && Modernizr.audio.wav) { audio.setAttribute("src", 'http://daokun.webs.com/play'+count+'.ogg'); } audio.load(); });

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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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  • The Best Tools for Enhancing and Expanding the Features of the Windows Clipboard

    - by Lori Kaufman
    The Windows clipboard is like a scratch pad used by the operating system and all running applications. When you copy or cut some text or a graphic, it is temporarily stored in the clipboard and then retrieved later when you paste the data. We’ve previously showed you how to store multiple items to the clipboard (using Clipboard Manager) in Windows, how to copy a file path to the clipboard, how to create a shortcut to clear the clipboard, and how to copy a list of files to the clipboard. There are some limitations of the Windows clipboard. Only one item can be stored at a time. Each time you copy something, the current item in the clipboard is replaced. The data on the clipboard also cannot be viewed without pasting it into an application. In addition, the data on the clipboard is cleared when you log out of your Windows session. NOTE: The above image shows the clipboard viewer from Windows XP (clipbrd.exe), which is not available in Windows 7 or Vista. However, you can download the file from deviantART and run it to view the current entry in the clipboard in Windows 7. Here are some additional useful tools that help enhance or expand the features of the Windows clipboard and make it more useful. Can Dust Actually Damage My Computer? What To Do If You Get a Virus on Your Computer Why Enabling “Do Not Track” Doesn’t Stop You From Being Tracked

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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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  • 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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  • 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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  • Speech to text software (audio transcribing) for mac

    - by GiH
    What is the best speech to text software for mac? I have an hours worth of audio that I need to transcribe, and I'd really like to not have to do it manually :-). I prefer free options and I like open source so if there is a project I'd like to know. All answers are welcome though.

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  • Ruby/Rails Audio Conversion Plugins?

    - by coneybeare
    I am looking for a good gem/plugin to convert user-uploaded audio files to different formats. One format in particular that I am interested in is converting to Apple .caf with ima4 compression for inclusion in an iPhone app. I have been using afconvert on my mac for this so far, but I need to do it on my linux box, server-side. Ideally, I would be able to work into paperclip. As an additional solution, ffmpeg could work, but I have not seen any .caf options for it. Anybody know of one?

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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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  • How to convert .avi video to .mp4(for Motorola Milestone, Android 2.3.4) with Avidemux

    - by kv1dr
    I open .avi video with Avidemux and I set Video format to MPEG-4 AVC(under Configure, Bitrate tab I choose "Single Pass - Bitrate (Average)" and Target bitrate to 256 kb/s, under Filters I choose MPlayer resize to 480x360 and I also add a subtitles) audio format to AAC (Faac)(Under Configure, I choose Bitrate 96) and format to MP4(like a image below). When Avidemux convert video to .mp4 format I can play the file on my copmuter, but on my phone I can't. When I want to play it on my phone with native video player, it just show the error something like "Can't play this video". So the question is how to convert .avi video to .mp4 with Avidemux(because I want to have subtitles inside movie) to be playable with android phone(Android version 2.3.4) with native player. Any help will be highly appreciated. :)

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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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  • Convert mkv to mp4 with ffmpeg

    - by JohnS
    When I try converting mkv to mp4 using ffmpeg, the following error occurs: [ipod @ 0x16fa0a0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: -2 = -2 av_interleaved_write_frame(): Invalid argument I used this command to convert the file: ffmpeg -i input.mkv -vcodec copy -acodec copy -absf aac_adtstoasc output.m4v The input file has the following characteristics: mediainfo input.mkv General Unique ID : 200459305952356554213392832683163418790 (0x96CF0ED8DB5914CBB9E18163689280A6) Complete name : input.mkv Format : Matroska Format version : Version 2 File size : 1.46 GiB Duration : 1h 5mn Overall bit rate : 3 168 Kbps Encoded date : UTC 2010-09-26 21:44:02 Writing application : mkvmerge v2.9.5 ('Tu es le seul') built on Jun 17 2009 16:28:30 Writing library : libebml v0.7.8 + libmatroska v0.8.1 Video ID : 1 Format : AVC Format/Info : Advanced Video Codec Format profile : [email protected] Format settings, CABAC : Yes Format settings, ReFrames : 4 frames Codec ID : V_MPEG4/ISO/AVC Duration : 1h 5mn Bit rate : 2 910 Kbps Width : 1 280 pixels Height : 720 pixels Display aspect ratio : 16:9 Frame rate : 25.000 fps Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Bits/(Pixel*Frame) : 0.126 Stream size : 1.31 GiB (90%) Writing library : x264 core 105 r1724 b02df7b Encoding settings : cabac=1 / ref=3 / deblock=1:0:0 / analyse=0x3:0x113 / me=hex / subme=6 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=0 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=1 / cqm=0 / deadzone=21,11 / fast_pskip=0 / chroma_qp_offset=-2 / threads=18 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / constrained_intra=0 / bframes=3 / b_pyramid=2 / b_adapt=1 / b_bias=0 / direct=3 / weightb=1 / open_gop=0 / weightp=0 / keyint=250 / keyint_min=25 / scenecut=40 / intra_refresh=0 / rc=2pass / mbtree=0 / bitrate=2910 / ratetol=1.0 / qcomp=0.60 / qpmin=10 / qpmax=51 / qpstep=4 / cplxblur=20.0 / qblur=0.5 / ip_ratio=1.40 / pb_ratio=1.30 / aq=1:1.00 Default : Yes Forced : No Audio ID : 2 Format : AC-3 Format/Info : Audio Coding 3 Mode extension : CM (complete main) Codec ID : A_AC3 Duration : 1h 5mn Bit rate mode : Constant Bit rate : 256 Kbps Channel(s) : 2 channels Channel positions : Front: L R Sampling rate : 48.0 KHz Bit depth : 16 bits Compression mode : Lossy Stream size : 121 MiB (8%) Language : English Default : Yes Forced : No Being new to ffmpeg, I'm not sure what the error means or how to correct it. Thanks!

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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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  • How to get musicbrainz track information from audio file

    - by Baki
    Can anyone tell me how to get track information from the MusicBrainz database from an audio file (mp3, wav, wma, ogg, etc...) using audio fingerprinting. I'm using MusicBrainz Sharp library, but any other library is ok. I've seen that you must use the libofa library, that you can't use MusicBrainz Sharp to get puid from the audio file, but I can't figure out how to use libofa with C#. Please show some examples and code snippets to help me, because I can't find them anywhere. Thanks in advance!

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  • Converting rich text that contains backslashes to plain text or html

    - by Allison
    I am trying to convert a rich text string to plain text or html. I am currently using the RichTextBox.Text feature which works correctly for almost all cases except when the text contains backslashes then some of the text is stripped out as the converter believes that it is part of the rtf formatting. Does anyone have any ideas of how to get the backslashes to stay in that instance. Here is an example of a string I would have {\rtf1\ansi\ansicpg1252\deff0\deflang1033{\fonttbl{\f0\fnil\fcharset0 Arial;}}\viewkind4\uc1\pard\fs17 Testing Export \with comments\par} The text I would need would be "Testing Export \with comments" and the text I am getting back from the rtf converter is "Testing Export comments". Any help would be greatly appreciated. Please respond if you have further questions.

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