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  • What's the best way of playing media files (esp. audio) with Mono/C#?

    - by supercheetah
    I'm trying to create something that will be playing some sound and music for some things in Mono+C#, but I'm not sure what the best thing will be for that. I'm trying to make it usable with things like Ogg Vorbis, MP3s, and wave files. My primary platform will be Linux, although a cross platform solution would be nice. Anyone have any suggestions for libraries for playing audio files?

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  • Where to start learning about audio or video codecs ?

    - by Vamsi
    Hi, I am very much confused to know what happens inside the codecs. I want to learn about the elements inside audio encoders and decoders. Would be very happy if you can provide me some links where i can find some good study material. Thanks precisely i would like to know how the codec parses the a media file.

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  • Compiling a list of audio + video players (flash / javascript / otehr) that I can embed into a websi

    - by FiveTools
    I'm compiling a list of audio + video players (flash / javascript / other) that I can embed into a website. flowplayer: http://flowplayer.org/ jw player: http://www.longtailvideo.com/players/ premium beat: http://www.premiumbeat.com/flash_resources/free_flash_music_player/ xspf web player: http://musicplayer.sourceforge.net/ yahoo media player: http://mediaplayer.yahoo.com/ any popular ones I'm missing? (anyone know if I can skin / customize any of them to operate similar to the Windows vista volume control?)

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  • Where can I get a splitter to connect a device with a single 3.5 mm plug into the audio input/output jacks on my laptop?

    - by XinJeisan
    I recently bought the :Hype Retro Handset for Mobile Phone" -- its just a device that looks like a handset to use when chatting on a computer or mobile phone that plugs into the phone/computer with a single 3.5 mm plug. I was hoping to use it on my windows 7 Toshiba laptop. I can hear audio fine through the handset but what I'm saying is not being picked up on the handset. On the box it says "some phones and computers may need additional adapters," so I'm hoping it is possible to get a splitter or something for this to work properly. I did email the parent company (http://dglusa.com/) but I haven't heard from them, and, looking over their website, I doubt I will. I also went to the local radio shack, and the guy said I needed a splitter, but he didn't know where to get one. I can find the kind of splitter I think I need online, but I'm unsure whether they are just for output or can also do input/output.

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  • Jack Audio ubuntu 12.10

    - by Shaneo1
    I used to have Jack Server working with 10.10, 11.04, 11.10 but not 12.04 and now 12.10. I have installed jackd jackd2 qjackctl surfed many forums and even given advice of how to get jack working, but now I am stuck. Tue Nov 27 22:30:46 2012: Saving settings to "/home/shane/.config/jack/conf.xml" ... 22:31:19.960 D-BUS: JACK server could not be started. Sorry Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started Tue Nov 27 22:31:19 2012: Starting jack server... Tue Nov 27 22:31:19 2012: JACK server starting in realtime mode with priority 10 Tue Nov 27 22:31:19 2012: [1m[31mERROR: cannot register object path "/org/freedesktop/ReserveDevice1/Audio0": A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Failed to acquire device name : Audio0 error : A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Audio device hw:0,0 cannot be acquired...[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Cannot initialize driver[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: JackServer::Open failed with -1[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Failed to open server[0m Tue Nov 27 22:31:21 2012: Saving settings to "/home/shane/.config/jack/conf.xml" ... 22:31:22.047 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started Can anyone assist?

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  • Double audio cd ripping weirdness

    - by jqno
    Since I installed Ubuntu 12.04, Rhythmbox, Banshee and Sound Juicer have started acting weird around double cd's, and specifically, cd #2 of said double cd. Sometimes, they will show the information of cd #1. Track names, durations, and even count are incorrect. Sometimes, they will first show the tracks for cd #1, then continue onto cd #2 if cd #2 has more tracks than #1. Sound Juicer seems to be unable to find any track durations at all, even for single cd's. Obviously, this is a pain when I'm trying to rip double cd's. And I have a fair number of them, which I want to rip. This happens on both my machines (a slightly aging iMac, and a 1-year-old Sony Vaio). However, on previous versions of Ubuntu, this never happened. All on the same machines. So I suspect 12.04 is using a different lib for extracting audio cd data. Just for kicks, I tried with Linux Mint 13, and there it works correctly, even though it claims to be based on Ubuntu 12.04 and therefore should be using (partially) the same software. So if the Mint guys can fix it, I should be able to do it too, right? So, my question: what changed in 12.04 that could cause this? And more importantly: what can I do to fix it?

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  • 12.10 no audio via hdmi and video speeds up

    - by jackson
    I have a laptop with an ati radeon 4200, on 12.04 everything worked fine, since upgrading to 12.10 I cannot get sound over the hdmi. When I switch to hdmi audio the video speeds up to about 2x. I can use the speakers in my laptop and watch video via hdmi with no problems. Things I have tried: Various tutorials to install the AMD/ATI drivers, all of which resulted in low graphics mode. Checked that everything is properly set in alsamixer, the sound utility and - installed pavucontrol and checked everything in there. Verified the output from cat /proc/asound/cards looks normal When I initially upgraded there was a plethora of problems which I believe were due to the old proprietary driver still being used but not compatible, after a few hours trying to fix that I decided just to back up and do a fresh install which works great except for the above stated problem. Any help would be greatly appreciated!! Finally hopefully this hasn't already been answered, I have tried a few different searches on the boards and haven't come up with anything. $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC269VB Analog [ALC269VB Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 0/1 Subdevice #0: subdevice #0

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  • No audio in Google Chrome

    - by Z9iT
    I started with Ubuntu 12.04 Minimal. Then installed only 3 utils sudo apt-get install xorg xinit google-chrome-stable alsa-base alsa-utils alsa-oss I have added google-chrome to .xinitrc file. Used sudo alsamixer to unmute everything using M. Also I am able to hear sound when I run this independently in a terminal sudo aplay /usr/share/sounds/alsa/Front_Center.wav However Google Chrome is not giving any sound output be it on youtube or the same file (/usr/share/sounds/alsa/Front_Center.wav) opened by browsing in chrome. UPDATE : the moment i install some Desktop (display) Manager like gnome or lxde and launch chrome then, the audio is perfect success. However if i kill the xsession and the desktop manager (lxde) AND then start with loading only the chrome (without DM) then again i loose the sound. This makes me wonder that there is something which is not allowing the sound to be loaded into chrome directly, but once the session like lxde loads, then it works flawless. I am thinking that i should rather ask, how to authorize google-chrome to use sound software? Miscellaneous : I am surprised to know that I cannot start google-chrome by sudo command (it asks to be a normal user) && that i cannot start alsamixer as a normal user (i must use sudo alsamixer ) May someone please help what i need to do so that google chrome speaks????

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  • How do I control Front and Back volume independently with cloned stereo signal

    - by someguy1984
    hello I have my 7.1 soundcard set to stereo and plugged in two stereo thingies (a. my desktop speakers b. my stereo) to the front respectively rear plug. Now I would like to an easy way to mute/volume control them independently - gui if possible. If I set my soundcard to 4.0, the pulse audio mixer gives me sliders for front and back but the problem is that's the input signal. so if i mute the front there and I'm listening to stereo everything remains silent. But what I'd have liked in that scenario would be the front to be silent and the back playing the full stereo signal. Is there a way to do that or something similar?

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  • Dealing with numerous, simultaneous sounds in unity

    - by luxchar
    I've written a custom class that creates a fixed number of audio sources. When a new sound is played, it goes through the class, which creates a queue of sounds that will be played during that frame. The sounds that are closer to the camera are given preference. If new sounds arrive in the next frame, I have a complex set of rules that determines how to replace the old ones. Ideally, "big" or "important" sounds should not be replaced by small ones. Sound replacement is necessary since the game can be fast-paced at times, and should try to play new sounds by replacing old ones. Otherwise, there can be "silent" moments when an old sound is about to stop playing and isn't replaced right away by a new sound. The drawback of replacing old sounds right away is that there is a harsh transition from the old sound clip to the new one. But I wonder if I could just remove that management logic altogether, and create audio sources on the fly for new sounds. I could give "important" sounds more priority (closer to 0 in the corresponding property) as opposed to less important ones, and let Unity take care of culling out sound effects that exceed the channel limit. The only drawback is that it requires many heap allocations. I wonder what strategy people use here?

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  • How to detect generation loss of a transcoded audio.

    - by The Rook
    Lets say you have a 96 kbit mp3 and you Transcode the file into a 320 kbit mp3. How could you programmatically detect the original bit rate or quality? Generation loss is created because each time a lossy algorithm is applied new information will be deemed "unnecessary" and is discarded. How could an algorithm use this property to detect the transcoding of audio. 128 kbps LAME mp3 transcoded to 320 kbps LAME mp3 (I Feel You, Depeche Mode) 10.8 MB. This image was taken from the bottom of this site. The 2 tracks above look nearly identical, but the difference is enough to support this argument.

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  • What language/API to use for a standalone live-input audio visualizer app?

    - by knuckfubuck
    I develop with Actionscript and was glad to see that AIR 2.0 was going to give access to mic input data. I planned to use this to create a visualizer set to the tempo of the incoming live audio. After doing a few days of google research it seems unlikely that it will be possible to analyze the data of the mic input in Flash/AIR. If anyone has ideas on how I can achieve this in AIR please let me know. (I'm open to workarounds.) That being said, I don't want to give up on the idea so I'm interested in suggestions for other language/API to use. My requirements for the app are: Run on OSX Two windows - one that can go fullscreen while the other(controller GUI) stays put Able to access live mic input data I've done reading on FFT and understand what needs to be done on the sound side so no need to help with that.

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  • Streaming audio to mobile phones, what technology to use ?

    - by Alx
    I'm planning on building an application where audio media is going to be streamed to the mobile phone for the user to listen. The targets are smartphones: iPhone/Blackberry/Android/(J2ME ?). I see that streaming on iPhone has to be done with HTTP Live streaming, but I don't see it supported by other platforms. Should I broadcast the streams via rstp ? http ? Is there any way to use a unified solution for all the different mobile platform ? If anyone already had to go through this, help would be gratly appreciated.

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  • Android: Using MediaRecorder to crop an existing audio file?

    - by user141146
    Hi, I'd like to take an existing mp3 file located on an SD card and arbitrarily crop it (e.g. crop from 0:12 to 1:14 in a 3 minute song). The only class that I've seen that seems remotely relevant to do this is the MediaRecorder class. My 'hope' would be to "record" an existing file like this: MediaRecorder recorder = new MediaRecorder(); recorder.setAudioSource(###some magical way of specifying an existing file??###); But this obviously doesn't work (setAudioSource() takes an int and seems to default to the phone's microphone). Is there a class or an approach that can be used to crop audio on the phone itself? TKS!!

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  • How does one record audio from a Javascript based webapp?

    - by username
    I'm trying to write a web-app that records WAV files (eg: from the user's microphone). I know Javascript alone can not do this, but I'm interested in the least proprietary method to augment my Javascript with. My targeted browsers are Firefox for PC and Mac (so no ActiveX). Please share your experiences with this. I gather it can be done with Flash (but not as a WAV formated file). I gather it can be done with Java (but not without code-signing). Are these the only options? @dominic-mazzoni I'd like to record the file as a WAV because because the purpose of the webapp will be to assemble a library of good quality short soundbites. I estimate upload will be 50 MB, which is well worth it for the quality. The app will only be used on our intranet. UPDATE: There's now an alternate solution thanks to JetPack's upcoming Audio API: See https://wiki.mozilla.org/Labs/Jetpack/JEP/18

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  • is there any Simple opensource live audio streaming Server using WCF? (see specification below)

    - by Ole Jak
    is there any Simple opensource live audio streaming Server using WCF? I need it to have simple structure: it should listen to some url format like http://example.com/service/stream?write&id=ANY_STRING and if any data comes to such address format it'll start making it avaliable by something like this http://example.com/service/stream?read&id=ANY_STRING Main thing here to be able to stream live data thru WCF service not buffering it just sharing stream. So can please any one help me with such idea? I think not only I have seen such problem with WCF alot on different sites so answer will help the WCF comunyty alot. I hope. BTW: I know some people say WCF is not prepared for live streaming over bacikHTTPbinding but hey! We all need it to, and we ask MS alot so some day they'll make it beter and we all want to be prepared for it.

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  • Debian no lib32asound2-plugins=1.0.25

    - by erai
    I compiled git wine 1.5.6 and the sound is not working correctly. winecfg gives a message: ALSA lib ../../pulse/pcm_pulse.c:995:(_snd_pcm_pulse_open) Unknown field handle_underrun err:winediag:AUDDRV_GetAudioEndpoint PulseAudio "default" -22 without handle_underrun. Audio may hang. Please upgrade to alsa_plugins >= 1.0.24 err:ole:CoInitializeEx Attempt to change threading model of this apartment from multi-threaded to apartment threaded But on wheezy there is no package lib32asound2-plugins 1.0.25 So, how do I build and package alsa-plugins to install to lib32? Or how do I use automake's configure to compile a 32bit lib on a 64 bit system. All I Need now is the libavcodec-dev for 32bit, but I don't understand. What is the 32bit libavcodec-dev package for amd64 debian?

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  • Would like to change audio codec, but keep video settings with ffmpeg

    - by Craig Tataryn
    I have a video for which I'd like to convert the audio codec to AAC 320 kbps / 44.100 kHz. What would I use for ffmpeg switches such that all the video settings and codec remain the same, but only the audio codec and settings change? Here's my video: $ ffmpeg -i Winnipeg.rb\ Scala-Talk.mov FFmpeg version SVN-r25375, Copyright (c) 2000-2010 the FFmpeg developers built on Oct 6 2010 13:02:41 with gcc 4.2.1 (Apple Inc. build 5664) configuration: --enable-libmp3lame --enable-shared --disable-mmx --arch=x86_64 libavutil 50.32. 2 / 50.32. 2 libavcore 0. 9. 1 / 0. 9. 1 libavcodec 52.92. 0 / 52.92. 0 libavformat 52.80. 0 / 52.80. 0 libavdevice 52. 2. 2 / 52. 2. 2 libavfilter 1.48. 0 / 1.48. 0 libswscale 0.12. 0 / 0.12. 0 Seems stream 0 codec frame rate differs from container frame rate: 2000.00 (2000/1) -> 10.00 (10/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Winnipeg.rb Scala-Talk.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt Duration: 01:10:53.00, start: 0.000000, bitrate: 283 kb/s Stream #0.0(eng): Video: h264, yuv420p, 800x598, 94 kb/s, 10 fps, 10 tbr, 1k tbn, 2k tbc Stream #0.1(eng): Audio: adpcm_ima_qt, 22050 Hz, 1 channels, s16 Stream #0.2(eng): Audio: adpcm_ima_qt, 22050 Hz, 1 channels, s16 At least one output file must be specified Many thanks in advance! One with with ffmpeg I've never been able to grok is how to just "tweak" files without having to regurgitate every little setting for things you don't want changes.

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  • Convert MPG w/ AC3 audio to something else - on a Mac

    - by anonymous coward
    I'm helping with a small volunteer media team, and they have several .mpg videos that don't appear to have sound when played in QuickTime, iTunes, Real Player, etc, on the local Mac machine. I was able to hear audio after transferring one of the movies to a Windows machine that had VLC media player on it. Through VLC I was able to discover that the audio stream is a52 / AC3 format. We use Autodesk Cleaner in our normal workflow of converting the format of our videos to FLV, but for some reason it's unable to convert this particular batch of videos (well, the video converts fine, but with no audio). Obviously, it seems that there's a codec issue here, but I'm not sure how to correct it. (I'm not extremely familiar with Macs, and/or Autodesk Cleaner). I've seen the Perian codec pack, but I'm not sure that having the codecs on the system will enable Cleaner to convert these videos (particularly the audio stream, since the video converts fine). Is there something obvious that I'm overlooking, or will we have to use something else for this particular batch of videos? If so, what?

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  • Audio playback, creating nested loop for fade in/out.

    - by Dave Slevin
    Hi Folks, First time poster here. A quick question about setting up a loop here. I want to set up a for loop for the first 1/3 of the main loop that will increase a value from .00001 or similar to 1. So I can use it to multiply a sample variable so as to create a fade-in in this simple audio file playback routine. So far it's turning out to be a bit of a head scratcher, any help greatfully recieved. for(i=0; i < end && !feof(fpin); i+=blockframes) { samples = fread(audioblock, sizeof(short), blocksamples, fpin); frames = samples; for(j=0; j < frames; j++) { for (f = 0; f< frames/3 ;f++) { fade = fade--; } output[j] = audioblock[j]/fade; } fwrite(output,sizeof(short), frames, fpoutput); } Apologies, So far I've read and re-write the file successfully. My problem is I'm trying to figure out a way to loop the variable 'fade' so it either increases or decreases to 1, so as I can modify the output variable. I wanted to do this in say 3 stages: 1. From 0 to frames/3 to increace a multiplication factor from .0001 to 1 2. from frames 1/3 to frames 2/3 to do nothing (multiply by 1) and 3. For the factor to decrease again below 1 so as for the output variable to decrease back to the original point. How can I create a loop that will increase and decrease these values over the outside loop?

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