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• Sound problems in Windows 7

  - by Plastkort

  hi! I hope I reached the correct forum for this question. I have a media computer with a third party audio card (Soundblaster audigy SE) I use a coaxial digital audio cable connected to a Onkyo TX SR508. if I use normal audio, the sound seems very low, I have to set volume atleast to 62 in my amplifier to hear anything, however if I set digital SPDIF I have no means of controlling the audio volume from the PC (only from the amplifier) and this can be nasty if I toggle between movies that uses Digital AC3/THX and movies wihtouth, if I look at movies capable of AC3 the volume gets VERY loud if amp is set to 62, 32 is more than enough volume when using passthrough. so this bothers me is how can I get the same amount of volume with or without digital output? I tried also other soundcards, internal red light Digital audio cable... if I connect to my television I get ok sound on any sound source via HDMI... help :)

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• How to trim AVAudioRecorder recording visually?

  - by Dyldo42

  Hi, I have an app that records and plays audio with AVFoundation. I'm trying to work out how I would implement an audio trimmer (kind of like the one in Quicktime for Mac) and am totally stumped. Can anyone offer any help on how I would do this? Thanks!

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• Front audio jack not working or missconfigured?

  - by Nicholas

  I have win xp and asus p5ql-e motherboard. The problem is that when i plug in my headphones on the front audio jack, the computer doesn't recognizes it. They work on the back audio jack but not on the front - how can i be sure that it;s not a software problem (something miss configured or not configured at all), before i conclude that it's a hardware problem (broken front audio jack or miss connected to the motherboard)?

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• Is there a way to "burn" audio to an ISO? (as an audio CD)

  - by Sootah

  I have an audiobook that I've downloaded via their download manager, and it's loaded into their cutesy little audio program that they force you to use. I can play the book just fine using their proprietary software, and while it's annoying when using my PC, it's utterly UNBEARABLE when I try to listen to it on my Blackberry. The program is INSANELY slow, it literally takes around 30 seconds to switch between tracks, so if I've forgotten where I am in the book it takes me around 15 minutes to finally get to where I was at. I've looked everywhere on how to transcode the book to .MP3, but evidently with their current format it's either extremely convoluted (and I have no desire to dick around with installing some older version of the codec, getting a different transcoding app, and then wrestling with getting it to actually work). Since I'm able to burn a copy of the book to an audio CD, I figure the best way to go about this is to just make the CDs and then rip them off of those to .MP3. In order to avoid wasting two hours, not to mention 14 CD-R's, I was wondering if there's a way to "burn" to an .ISO instead of an actual CD-R. I currently have SlySoft's Virtual CloneDrive installed, so I can mount .ISO's easily enough, but now I want to actually create an ISO via the CD burning process. Just in case I've not explained myself very well, here is an overview of what I intend to do: "Burn" a set of Audio CD .ISOs from the audiobook (hopefully I can do this using Windows Media Player, otherwise I'll be forced to use the audiobook app) Mount an .ISO in Virtual CloneDrive Rip the audio tracks on the mounted .ISO to .MP3s Repeat steps 2-3 until the entire book is in .MP3 format Copy .MP3s to my Blackberry so that I'm not driven insane every time I want to listen to the book in the car, and be able to use Winamp when listening on my computer EDIT: I'd suppose a rather concise way to put it is that I need something that will emulate a CD-R drive, so that you can select it as the output drive in whatever app your burning the audio CD from. (I'd suppose that when you "insert a blank CD-R" the app would then ask you what file to save to)

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• No audio from USB device on iMac

  - by brandon

  My computer froze the other day and started a buzzing noise. After I restarted in the middle of it, the buzzing persisted. A few restarts later, the buzzing stopped, but I can't get audio to come from my audio interface setup anymore. I'm using an M-Audiophile USB to connect to studio monitors, and there's absolutely no audio. I've tried restarting many times and unplugging - still nothing. Any help would be appreciated.

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• Flash game size and distribution between asset types

  - by EyeSeeEm

  I am currently developing a Flash game and am planning on a large amount of graphics and audio assets, which has led to some questions about Flash game size. By looking at some of the popular games on NG, there seem to be many in the 5-10Mb and a few in the 10-25Mb range. I was wondering if anyone knew of other notable large-scale games and what their sizes were, and if there have been any cases of games being disadvantaged because of their size. What is a common distribution of game size between code, graphics and audio? I know this can vary strongly, but I would like to hear your thoughts on an average case for a high-quality game.

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• Lubuntu upgrade to 13.04 killed sound with ALSA. How to troubleshoot?

  - by Sven

  After upgrading to 13.04 from 12.10 Lubuntu lost audio playback after unplugging usb soundcard (Polycom) and plugging it back in. Volume control was gray and leading to pulseaudio mixer (not installed) so I uninstalled the pulseaudio package. I also removed and reinstalled the alsa-base package. After restart I have the alsamixer back everything seemingly as usual(volume 100%, unmute) but every sound program gets me errors no matter what device I select. aplay -L: null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default:CARD=NVidia HDA NVidia, ALC662 rev1 Analog Default Audio Device sysdefault:CARD=NVidia HDA NVidia, ALC662 rev1 Analog Default Audio Device front:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output dmix:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Direct sample mixing device dmix:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Direct sample mixing device dmix:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct sample mixing device dsnoop:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Direct sample snooping device dsnoop:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Direct sample snooping device dsnoop:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct sample snooping device hw:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Direct hardware device without any conversions hw:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Direct hardware device without any conversions hw:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Direct hardware device without any conversions plughw:CARD=NVidia,DEV=0 HDA NVidia, ALC662 rev1 Analog Hardware device with all software conversions plughw:CARD=NVidia,DEV=1 HDA NVidia, ALC662 rev1 Digital Hardware device with all software conversions plughw:CARD=NVidia,DEV=3 HDA NVidia, HDMI 0 Hardware device with all software conversions default:CARD=Communicator Default Audio Device sysdefault:CARD=Communicator Default Audio Device front:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio Front speakers surround40:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio IEC958 (S/PDIF) Digital Audio Output dmix:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio Direct sample mixing device dsnoop:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio Direct sample snooping device hw:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio Direct hardware device without any conversions plughw:CARD=Communicator,DEV=0 Polycom Communicator, USB Audio Hardware device with all software conversions etc/asound.conf: defaults.ctl.card 1 defaults.pcm.card 1 defaults.pcm.device 1 Following gets same result with both devices. aplay -vv -D front:CARD=NVidia,DEV=0 "Release the Pressure.wav": Playing WAVE 'Release the Pressure.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono aplay: set_params:1087: Channels count non available Guayadeque mp3 playback: AL lib: alsa_open_playback: Could not open playback device 'default': No such file or directory 21:32:14: Error: Gstreamer error 'Configured audiosink playbackbin is not working.' Audacious: ALSA error: snd_mixer_attach failed: No such file or directory. ALSA error: snd_pcm_open failed: No such device. So How do I fix my audio? UPDATE: I removed the usb soundcard and got rid of all alsa config. Everything is working as before the install but it sure feels fragile.

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• How to automatically change volume level when un-/plugging headphones?

  - by htorque

  What I want is the following: When I plug in my headphones, I want the sound to be un-muted and set to a specific volume level. When I unplug my headphones, I want the sound to be muted (or set to a specific volume level). Setting the volume levels isn't the problem, but I somehow need to do this when un-/plugging the headphones, so I'm looking for a way to get notified of those events. I quickly found /proc/asound/card0/codec#0 to indicate whether headphones are plugged in or not, so I tried to monitor it using inotifywait and change the volume level based on modified notifications. Unfortunately inotifywait failed because proc isn't an ordinary filesystem. Are there other ways to do this (maybe via PulseAudio)? Audio device: Intel HDA, audio codec: Conexant CX20585.

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• Remove Audio stream from XVID files

  - by Kyle Brandt

  I have a bunch of Xvid files that each have an audio stream that I do not want. How can I strip the audio track I don't want using the Linux command line? I don't need the whole script (loop), just what command I would use to process each avi file individually (unless the cmd itself has batch modification built into it). I don't believe the file is in an mkv container, as mkvinfo doesn't find anything. Here is part of mplayer's output (thanks ~quack): [aviheader] Video stream found, -vid 0 ID_AUDIO_ID=1 [aviheader] Audio stream found, -aid 1 ID_AUDIO_ID=2 [aviheader] Audio stream found, -aid 2 VIDEO: [XVID] 512x384 12bpp 25.000 fps 1013.4 kbps (123.7 kbyte/s)

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• Optimal Compression for Speech

  - by ashes999

  I'm designing a game that depends heavily on audio; I will have some 300+ speech files (most of them just a word or two long). This can very quickly escalate the size of my final game. What's the optimal way to encode/compress speech files to keep the size minimal without getting audio artifacts? Please address both per-file compression/encoding, and also zipping/compressing the set of all speech files together in your answer. Because I'm not sure which (or combination of both) factors will give me the best results. Edit: I need this to run in Silverlight and Android, so I'm presumably stuck with only MP3 as my option (other than uncompressed wave files).

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• Checking rtp stream audio quality.

  - by chills42

  We are working in a test environment and need to monitor the audio quality of an rtp stream that is being captured using tshark. Right now we are able to capture the audio and access the file through wireshark, but we would like to find a way to save the audio to a .wav file (or similar) via the command line. Does anyone know of a tool that can do this?

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• A seekable one-frame FLV video (with audio)?

  - by George Stephanos

  Is it possible to generate an FLV out of an MP3 and a JPG, without uselessly looping the image and still be able to seek the audio ? This command generates a non-seekable video: ffmpeg -y -i audio.mp3 -i image.jpg -r 1 -acodec copy video.flv and this one generates a seekable one, but with uselessly looping the image occupying both space and time: ffmpeg -y -loop_input -i audio.mp3 -i image.jpg -r 1 -acodec copy video.flv -shortest

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• Legal concerns with orchestrating a music submission contest

  - by Amplify91

  My team and I are getting pretty far along in the development of our latest game and have been thinking about audio. We decided to host an audio submission contest where we will offer a little cash and some equity stake in the game as prizes. We are also giving away copies of the game to participants. We hope not only to find audio for our game, but to meet some cool sound artists and promote the game a bit through the process. First of all, is this even a good idea? What are some potential dangers in doing this? Will it even be well received among artists? Secondly, I wrote up some Terms and Conditions in my best legal-speak to try to protect us and clarify how the contest will be run. Are these sufficient to make sure everyone involved is treated fairly and is legally protected? They are as follows: All submissions (The Submission) must be licensed under a Creative Commons Attribution 3.0 Unported License (CC-BY-3.0) By applying a CC-BY-3.0 license, you (The Submitter) expressly give Detour Games (and all members wherein) permission to copy, distribute, transmit, modify, adapt, and make commercial use of The Submission. The Submitter must own all rights to The Submission and be within their rights to license it as specified and submit it. The Submitter claims responsibility for the legality of The Submission. If The Submission is found to infringe on the rights of a person or entity other than those of The Submitter, Detour Games will not be held liable as all responsibility and liability for the legality of The Submission is that of The Submitter's. No more than two free copies of The Game per submitter. All flat cash prizes will only be disbursed pending the success of our first $5,000 Kickstarter campaign. These prizes will be disbursed 30 days after Detour Games receives the Kickstarter funds. All equity prizes (percentage of profits) are defined as the given percent of total profits after costs for a period of one year (12 months) after the release of RAW. These prizes will be disbursed semi-annually. All prize money will be disbursed through either an electronic fund transfer through a service such as PayPal or by a mailed money order. It is The Submitter's responsibility to cooperate with Detour Games in the disbursement of the funds. Detour Games reserves the right to change these Terms and Conditions at any time without notice. By participating in the contest, The Submitter agrees to and accepts all terms and conditions listed. What else could I do (legally) to protect everyone involved?

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• "HDA audio bus driver is required and not found" on Dell Optiplex

  - by user1666698

  I have Dell Optiplex 745 with Windows 7 installed on it. I'm trying to use the Windows XP audio driver as Windows 7 drivers aren't available for Optiplex 745 and Windows Vista driver is displaying that it's not compatible with my hardware. When I try to install the Windows XP audio driver, it's displaying an error HDA audio bus driver is required and not found The installation fails then. I have researched thourghly and used many drivers but my audio is not working at all. I was also told that it might be a problem with my hardware – that is, a problem with the board.

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• Playing/extracting audio file from PDF

  - by ravl1084

  I use Ubuntu and I have a PDF file that contains an audio annotation. It won't play on Okular, it treats it as a text annotation. Following an old blog post where the poster created a small C script to extract the audio didn't work either, I suspect the format of these audio annotations has changed. Using the information on it I managed to uncompress the PDF and with vim, I found the audio data in the file. I tried copying this into its own file and changed the extension from mp3, wav, mid, but none of them would play. Is there a way of achieving this?

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• ffdshow h.264 audio desync

  - by Core Xii

  When I encode video with ffdshow with h.264, the audio is out of sync. At the very beginning of the video, the picture freezes for about 1 second, while the audio plays fine, resulting in the audio being that 1 second ahead of the picture throughout the entire video. Any ideas on possible causes or, obviously, solutions?

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• Why won't AVI2DVD load the audio stream?

  - by Xavierjazz

  XP SP3 I have an .avi file. It is in a folder on my "C:" drive. There are no disallowed characters in either the folder or file name. It has audio as I have watched it on my computer. I want to burn it to a DVD. When I load the file into AVI2DVD, no audio stream shows, and the program will not work without an audio stream. I have used the net extensively to try and solve this, with no success. AFICT I have followed all instructions exactly, but no audio stream. Very frustrating. Does anyone have a clue? Can you help me? Thank you. Regards,

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• Microphone audio streaming from Cocoa mac app to iPhone

  - by Benzamin

  Hi devs, I'm trying to build a microphone audio streamer to iPhone. The server software will be a mac desktop app and the client will be iPhone, and they are connected via tcp port. I've successfully connected the mac app and iPhone, and tried to send a fixed test.m4a audio file first. But at the iPhone i grabbed the data well, when tried to play it i used AVAudioPlayer and its returning OSStatus error. I played around with the audio queue service but its very tricky and i only got some example for fixed length audio playing like http://cocoawithlove.com/2009/06/revisiting-old-post-streaming-and.html Now i need help on two things, how can i continuously grab audio data from Mac desktop microphone? And then after grabbing the data how i can play this unfixed length audio data in the iPhone. What exactly i need to do? Please please help me on this......

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• AudioQueue recording as float

  - by niklassaers

  Hi guys, I would like to have the result from my recording as a float in the range [0.0, 1.0], alternatively [-1.0, 1.0] because of a bit of math I want to do on it. When I set my recordingformat to be in float, like this: mRecordFormat.mFormatFlags = kLinearPCMFormatFlagIsFloat; I get: Error: AudioQueueNewInput failed ('fmt?') Does this mean the hardware doesn't support recording to floats? If not, how do I set it to record in floats? If so, are there any processor-friendly ways I can convert a signed integer array to a float array? Cheers Nik

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• WPF MediaElement pauses when trying to play Windows Media Center recording

  - by eborlas

  I have an WPF application that starts a live tv recording using Windows Media Center, using the Windows Media Center SDK, waits 2 minutes and then plays the currently recording file (.wtv) in a WPF MediaElement. The problem is that the video visual will freeze and it is not consistent on when it happens. I have looked into it and discovered that the timeline is still moving so the MediaElement still thinks that the video is playing. This happens only when trying to play a video file that is currently being recorded, previously recorded shows play just fine. Both Media Center and Media Player can play those files without any hiccups; it is only the WPF MediaElement that has this problem. Any help would be appreciated.

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• How to adjust video recording brightness?

  - by Paul

  When I record videos in low-light conditions on my Motorola Droid results are poor... very dark images. On my old phone, an LG Dare, recording in low light works better thanks to a brightness adjustment in the video recorder which can be set manually... Android does not have a similar brightness adjustment when recording video. Does anyone have suggestions for how to adjust the brightness using Android's APIs? I have searched the official APIs, but I cannot find a way to control brightness.

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• Get binary data from audio impulses

  - by Timo

  I have IR sensor which have TRS plug and I can record my remotes signals into audio. Now I want to control my computer with TV remote, but I don't have any clue how to compare audio input with pre-recorded audio. But after I realized that these audio waves contains only some kind data (binary) I can turn these into binary or hex, so it is much easier to compare. Waves look just like this: http://i.imgur.com/lCIyl.png And this: ttp://i.imgur.com/goJ6d.png These are records of "OK" button, sometimes there are some impulses on right channel too and I don't know why, it seems like connections in sensor are damaged maybe. Ok thats not matter, anyway I need help with python program which read these impulses and turn these into binary, in realtime from audio input(mic). I know it's sounds like "Do it for me, while I enjoy my life", but I don't have experiences with sound transforming/reading... I've looking for python examples for recording and reading audio, but unsuccessfully.

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• ffmpeg: video file played OK on Ubuntu, but no sound on XP

  - by Andy Le

  I created a video clip using ffmpeg (vcodec: mpeg2video, acodec: AC3 5.1). The file can be played normally on Ubuntu, but when I play it on an XP machine, there is no sound. I can play AC3 files and other movies with AC3 sound. I already tried many codec packs and many players. When I compare the MediaInfo tab of the Properties window of the file with another playable movie, I see that the Audio Identifier of the audio stream in my file is 0x80 while it is 0x02 in the other movie. So I guess that's why players on XP can't recognize the audio codec. When I use an MKV container instead of MPEG (still mpeg2video codec), then the result is OK on both Ubuntu and XP (with the correct Audio ID). I really need MPEG though. Any idea? This is the command I used: ~/ffmpeg/ffmpeg/ffmpeg -loop_input \ -t 97 -r 30000/1001 -i v%4d.tga -i final.ac3 \ -vcodec mpeg2video -qscale 1 -s 400x400 -r 30000/1001 \ -acodec copy -y out6.mpeg 2 This is the output of mediainfo (on Ubuntu): General Complete name : out6.mpeg Format : MPEG-PS File size : 6.86 MiB Duration : 1mn 37s Overall bit rate : 593 Kbps Video ID : 224 (0xE0) Format : MPEG Video Format version : Version 2 Format profile : Main@Main Format settings, BVOP : No Format settings, Matrix : Default Format_Settings_GOP : M=1, N=12 Duration : 1mn 37s Bit rate mode : Variable Bit rate : 122 Kbps Width : 400 pixels Height : 400 pixels Display aspect ratio : 1.000 Frame rate : 29.970 fps Resolution : 8 bits Colorimetry : 4:2:0 Scan type : Progressive Bits/(Pixel*Frame) : 0.025 Stream size : 1.41 MiB (21%) Audio ID : 128 (0x80) Format : AC-3 Format/Info : Audio Coding 3 Duration : 1mn 36s Bit rate mode : Constant Bit rate : 448 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 44.1 KHz Stream size : 5.18 MiB (75%)

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• Downmix surround to Dolby Pro-Logic at the OS/driver level in Windows 7?

  - by davr

  First off, I'm talking about Dolby Pro-Logic, a really old tech for encoding 4 audio channels (L/R/C/SR) into two analog outputs, and then extracting them again. It was used in surround sound systems in the last century. I have a modern PC that can output 5.1 analog audio (Three outputs on the back carry six channels of audio). But I have a really old surround sound reciever that only has a two-channel, L/R input, which it extracts 4 channels of audio from, and outputs to 5.1 speakers. What I want is some way for the OS, Windows 7, to act as if I really had 5.1 audio channels available, so applications produce surround audio, but before outputting it out of the back of my PC, apply Dolby Pro-Logic matrix encoding so that it outputs over only two channels. These two channels would then get sent to my receiver via a RCA cable, which would decode it again and drive the surround speakers. Is anything like this possible? I'm pretty sure I could do it at an application / codec level, but I'm looking for something that I just have to set once.

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• XPP-32 over W7-64 on music production laptop

  - by quarlo

  I need to upgrade my laptop and need high performance for music production (recording and mixing). My audio interface manufacturer seems to be unable to successfully convert their drivers to 64-bit. I do not trust a virtual machine to handle real-time audio recording at low enough latency so ... I would like to install XP Pro 32-bit on a separate partition and dual boot since most of the machines that can handle this application now ship with Windows 7 64-bit flavors. I'd like to transit to 64-bit over time assuming M-Audio does eventually get a handle on 64-bit drivers, but really need to ensure that I can stay at 32-bit for now. Does anyone have any experience with this or something similar?

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