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Search found 167 results on 7 pages for 'pcm'.

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  • Qt Audio Recording Question

    - by Cenoc
    This is sort of a follow-up/branch off a previous question, which still stands unresolved. Are there other codecs besides pcm for qt QAudio class? I cant seem to find any... I want to have a way of playing stuff recorded by qt on vlc. Thanks in advance.

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  • What is a lightweight cross platform WAV playing library?

    - by Lokkju
    I'm looking for a lightweight way to make my program (written in C) be able to play audio files on either windows or linux. I am currently using windows native calls, which is essentially just a single call that is passed a filename. I would like something similar that works on linux. The audio files are Microsoft PCM, Single channel, 22Khz Any Suggestions?

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  • what is the best mid/high-end class audio/music creation audio sound card?

    - by Chris
    Hello, I have a computershop myself, and I repair computers. But one of the things I really don't know (yet) is the performace od audio cards for music creation with midi. I have searched and searched and came up with some good reviews, but after browsing for a couple of hours I could't see the trees trough the forrest :-D (it's a dutch expression) At one moment I thought the M-Audio - Delta 1010LT would be a good PCIe card, later on I read that this card was released years ago. (but that could be false information) Also any personal expierence would be great, but not necessairy. I have searched a few cards, and I hope someone can help me make a choice for a friend of mine. He's buget is between $100 and $350 I know there are audio cards from $ 500 - $1850,- this is just too expensive. The following specs are crucial: ASIO Midi Mic in minimal 5.1, 7.1 recommended it's not for airplay, but just to compose music at home. using Ableton and midi keyboard. 1. M-Audio - Delta 1010LT: 8 x 8 analog I/O 2 mic preamps or line inputs S/PDIF digital I/O (coaxial) with 2-channel PCM SCMS copy protection control digital I/O supports surround-encoded AC-3 and DTS pass-through 1 x 1 MIDI I/O directly drive up to 7.1 surround (bass management software included) software controlled 36-bit internal DSP digital mixing/routing +4dbu/-10dBV operation individually switched in software word clock I/O for sample accurate device synchronization 2. RME HDSP 9632: * Stereo Analog Ein- und Ausgang, symmetrisch*, 24-Bit/192kHz, > 110 dB SNR * Optionale Erweiterungsboards mit je 4 symmetrischen Ein- und Ausgängen * Alle analogen I/Os voll 192 kHz-fähig, also keine Reduzierung der Kanalzahl * 1 x ADAT Digital In/Out, 96 kHz-fähig (S/MUX) * 1 x SPDIF Digital In/Out, 192 kHz-fähig * 1 x Breakout Kabel für koaxialen SPDIF-Betrieb* * Also bis zu 16 Ein-und Ausgänge gleichzeitig nutzbar! * 1 x Stereo Kopfhörerausgang, parallel zum analogen Ausgang, aber eigene Pegelanpassung * 1 x MIDI I/O für 16 Kanäle Hi-Speed MIDI über Breakout Kabel * DIGICheck, RMEs einzigartiges Meter- und Analysetool mit Spectral Analyser, Professionelle Level Meter 2/8/16-Kanalig, Vector Audio Scope und diversen weiteren Analysefunktionen * HDSP Meter Bridge: Frei skalierbare Levelmeter mit Peak- und RMS Berechnung in Hardware * TotalMix: 512-Kanal Mischer mit 40 Bit interner Auflösung 3. EMU 1212M (1212 M) PCIe: * Top kwaliteit convertors 24-bit/192kHz convertors. * Hardware gestuurde effecten. * DSP zero-latency hardware mixen en monitoring. * Analoge en digitale I/O plus MIDI. * EMU Production Tools Software Bundle - Cakewalk SONAR , Steinberg Cubase LE, Ableton Live E-MU Edition **EMU 1212M PCI-e inputs/outputs:** * 2 balanced jack inputs. * 2 balanced jack outputs. * 24-bit/192kHz ADAT I/O. * 24-bit/192kHz Coaxiale S/PDif I/O switchable to AES/EBU. * MIDI I/O. 4. M-Audio Audiophile 192: - Up to 24-bit/192kHz audio - 2 balanced analog inputs (1/4” TRS) - 2 balanced analog outputs (1/4” TRS) - S/PDIF digital I/O (coaxial RCA connectors) with 2-channel PCM - SCMS copy protection control - Digital I/O supports surround-encoded AC-3 and DTS pass-through - Direct hardware input monitoring via separate balanced 1/4” TRS monitor outputs - Software routing of inputs and outputs - Digital I/O can be routed to/from external effects - 16-channel MIDI I/O - ASIO, WDM, GSIF 2 and Core Audio driver support for compatibility with most applications - 64-bit driver support for Windows - PCI 2.2 compatibility - Apple G5 compatible - Incompatible exceptions - Includes Ableton Live Lite music production software, so you can make music right away - Works with other Delta cards Technical Specifcations: - Compatibility - ASIO - WDM - GSIF 2 - Core Audio

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  • Out of sync audio video using mencoder

    - by 1ch1g0
    hi i converted a mkv (matroska) file to avi using ffmpeg: ffmpeg -i input.mkv -f mp4 -vcodec mpeg4 -sameq -r 29.97 -b 512kb -acodec ac3 -ab 128kb -vol 512 output.avi the output file plays fine using mplayer. after that, i using mencoder to insert subtitles: mencoder output.avi -o new.avi -oac pcm -ovc lavc -subfont-text-scale 3 -sub subtitle.srt however, after i play back the video "new.avi", the video and audio is out of sync. What options can i put into mencoder to sync the A/V. ? I have also tried ffmpeg -newsubtitle option but can't get it work. Any examples of usage of -newsubtitle would be greatly appreciated. thanks

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  • Creating a DVD-player compatible movie file

    - by Robert Munteanu
    I've created encoded a MPEG-4 file with mplayer and placed it on a DVD. The file is identified as RIFF (little-endian) data, AVI, 320 x 240, 25.00 fps, video: FFMpeg MPEG-4, audio: uncompressed PCM (stereo, 96000 Hz) I've tried playing it on a Samsung 1080p DVD player and the codecs were not recognised. There are no firmware upgrades available for my region (Romania). How should I pick the codecs to make sure that the files are readable by this DVD player? Update: The command line I used is similar to mencoder -dvd 2 -ovc lavc -lavcopts vcodec=mpeg4:vpass=1 -oac copy -o movie.avi mencoder -dvd 2 -ovc lavc -lavcopts vcodec=mpeg4:vpass=2 -oac copy -o movie.avi

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  • Make SoX not show output

    - by Ram Rachum
    I'm recording with SoX, and I want to make it not show output at all. When recording, it shows this output: Input File : 'default' (waveaudio) Channels : 2 Sample Rate : 48000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM In:0.00% 00:00:00.68 [00:00:00.00] Out:28.7k [ | ] Clip:0 I tried setting verbosity to 0, but it has no effect. (I'm guessing it's meant for messages other than this.) I don't just want to hide the output, which I could do easily; I want SoX to not generate it in the first place, for performance on a weak computer.

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  • Why is my Output distorted after encoding with Expression Encoder?

    - by WernerCD
    I'm a "n00b" when it comes to re-encoding files. I'm trying to re-encode an AVI into a silverlight container via Encoding Video using Expression Encoder 4.0. As you can see in the video, the left is the input and it looks/sounds fine. The right is the output and it... doesn't. I'm unsure of where to go from here. I'm not sure why the output is jacked up, since the input looks fine. Input Video properties: AVI 2.49GB 22:34 809x605 Video: TSCC 809x605 15fps [Stream 00] Audio: PCM 22050Hz mono 352kbps [Stream 01] Choice of output doesn't seem to matter, they all end up distorted like the picture shows.

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  • How to keep source frame rate with mencoder/ffmpeg?

    - by Sandra
    I would like to crop and rotate a video, and then encode it to mp4 or mkv. mencoder video.mp4 -vf rotate=1,crop=720:1280:0:0 -oac pcm -ovc x264 -x264encopts preset=veryslow:tune=film:crf=15:frameref=15:fast_pskip=0:threads=auto -lavfopts format=matroska -o test.mkv But when I do the above encoding, the frame rate is way too fast. The encoding options were something I found, so I don't know if that is the problem. Question All I want is to crop and rotate the video, and keep the audio/video quality as good as possible. Have anyone tried this?

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  • xvidcap: Error accessing sound input from /dev/dsp

    - by stivlo
    I'm running Ubuntu 11.10 and I'm trying xvidcap to record a screencast with audio from the microphone, however it can't record any sound: $ xvidcap --file appo.avi --cap_geometry 700x500-0+0 Error accessing sound input from /dev/dsp Sound disabled! Sure enough /dev/dsp doesn't even exist: $ sudo ls -lh /dev/dsp ls: cannot access /dev/dsp: No such file or directory I found a blog post about fixing xvidcap sound input, however if I try the suggestion I get: $ sudo modprobe snd-pcm-oss FATAL: Module snd_pcm_oss not found. So the question is, how can I create /dev/dsp? The problem behind the problem is: how can I record sound from the microphone with xvidcap? So workarounds are welcome too. UPDATE: I've followed the suggestion of James, and something has improved. The error accessing /dev/dsp is gone, however now I get: [oss @ 0x8e0c120] Estimating duration from bitrate, this may be inaccurate xtoffmpeg.c add_audio_stream(): Can't initialize fifo for audio recording Now when I record xvidcap appears in the recording tab of pavucontrol and I can choose Audio stream from Internal Audio Analog Stereo or Monitor of Internal Audio Analog Stereo, I tried both just in case, but the video is still mute. UPDATE 2: I found that "Monitor of" is the one to record application sounds, while for microphone, I should choose "Internal Audio Analog Stereo". To rule out other problems, such as with the microphone, I tried with gnome-sound-recorder and it works. Actually I jumped on my chair, since the volume was too high! :-)

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  • recordMyDesktop stopped working after upgrade

    - by anfeo
    Hi, I've done the upgrade to Ubuntu 10.10 from Ubuntu 10.04, and recordMydesktop don't work now. If I start it from command line it seam to work, but the interface don't start and I have this error: Initial recording window is set to: X:0 Y:0 Width:1680 Height:945 Adjusted recording window is set to: X:0 Y:0 Width:1680 Height:944 Your window manager appears to be Metacity Initializing... Buffer size adjusted to 4096 from 4096 frames. Opened PCM device default Recording on device default is set to: 1 channels at 22050Hz X Error: BadAccess (attempt to access private resource denied) Bad Access on XGrabKey. Shortcut already assigned. X Error: BadAccess (attempt to access private resource denied) Bad Access on XGrabKey. Shortcut already assigned. X Error: BadAccess (attempt to access private resource denied) Bad Access on XGrabKey. Shortcut already assigned. X Error: BadAccess (attempt to access private resource denied) Bad Access on XGrabKey. Shortcut already assigned. Capturing! X Error: BadAccess (attempt to access private resource denied) Bad Access on XGrabKey. Shortcut already assigned. X Error: BadAccess (attempt to access private resource denied) Bad Access on XGrabKey. Shortcut already assigned. X Error: BadAccess (attempt to access private resource denied) Bad Access on XGrabKey. Shortcut already assigned. X Error: BadAccess (attempt to access private resource denied) Bad Access on XGrabKey. Shortcut already assigned.

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  • Ubuntu 12.04 taking too much time to boot

    - by adarshdinesh
    Ubuntu 12.04 is taking much time for booting, Here is the system kernel message while booting .It is showing that some anacron was killed ,why ? and how to fix the problem ? [ 2.241047] scsi6 : usb-storage 2-1.6:1.0 [ 2.241501] usbcore: registered new interface driver usb-storage [ 2.241895] USB Mass Storage support registered. [ 3.240670] scsi 6:0:0:0: Direct-Access Multiple Card Reader 1.00 PQ: 0 ANSI: 0 [ 3.241791] sd 6:0:0:0: Attached scsi generic sg2 type 0 [ 3.243083] sd 6:0:0:0: [sdb] Attached SCSI removable disk [ 12.568641] Adding 4037904k swap on /dev/sda3. Priority:-1 extents:1 across:4037904k [ 12.615014] udevd[462]: starting version 175 [ 12.651334] mei: module is from the staging directory, the quality is unknown, you have been warned. [ 12.655283] [drm] Initialized drm 1.1.0 20060810 ................... [ 14.118369] init: alsa-restore main process (982) terminated with status 19 [ 14.252595] init: anacron main process (1033) killed by TERM signal [ 14.285763] HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 [ 14.285841] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 [ 14.285925] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 [ 14.285991] input: HDA Intel PCH Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 [ 14.615073] init: plymouth-stop pre-start process (1222) terminated with status 1 [ 16.447287] wlan0: authenticate with c0:8a:de:7c:60:e8 (try 1) [ 16.448858] wlan0: authenticated [ 16.453405] wlan0: associate with c0:8a:de:7c:60:e8 (try 1) [ 16.456392] wlan0: RX AssocResp from c0:8a:de:7c:60:e8 (capab=0x431 status=0 aid=2) [ 16.456398] wlan0: associated [ 16.457014] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 16.457017] ieee80211 phy0: brcmsmac: brcms_ops_bss_info_changed: associated [ 16.457019] ieee80211 phy0: changing basic rates failed: -22 [ 16.457021] ieee80211 phy0: brcms_ops_bss_info_changed: arp filtering: enabled true, count 0 (implement) [ 16.457226] ADDRCONF(NETDEV_CHANGE): wlan0: link becomes ready [ 16.654196] ieee80211 phy0: brcms_ops_bss_info_changed: arp filtering: enabled true, count 1 (implement) [ 17.823565] ieee80211 phy0: wl0: brcms_c_d11hdrs_mac80211: txop exceeded phylen 180/256 dur 1946/1504 [ 18.220865] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 26.881422] wlan0: no IPv6 routers present [ 68.228293] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 73.240133] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 76.574490] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 102.180006] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 103.100984] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 124.171624] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement)

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  • What packages are neccessary to have sound output from java applets?

    - by MvG
    I've got a very minimalistic setup of ubuntu precise, created using debootstrap. So please don't assume that any packages are installed just because they usually are. On that system, I'd like to play some sounds from a java applet. However, this always fails with the following error message: javax.sound.midi.MidiUnavailableException: Can not open line at com.sun.media.sound.SoftSynthesizer.open(SoftSynthesizer.java:1132) at com.sun.media.sound.SoftSynthesizer.open(SoftSynthesizer.java:1036) ... Caused by: java.lang.IllegalArgumentException: No line matching interface SourceDataLine supporting format PCM_SIGNED 44100.0 Hz, 16 bit, stereo, 4 bytes/frame, little-endian is supported. at javax.sound.sampled.AudioSystem.getLine(AudioSystem.java:476) at javax.sound.sampled.AudioSystem.getSourceDataLine(AudioSystem.java:604) at com.sun.media.sound.SoftSynthesizer.open(SoftSynthesizer.java:1066) ... 35 more As the messages mention a soft synthesizer, and pcm lines, I expect that the lack of some midi daemon is not the issue here. As far as I can tell, the alsa kernel modules are loaded, including snd_hda_intel, snd_pcm, snd_seq_midi among others. I've also included the alsa-base and alsa-utils packages in my installation. alsa-mixer looks good, using “HDA Intel PCH” as its default device. What other packages, configuration settings or daemon startups does java require to make its sound output work?

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  • aplay -l says no soundcards found; alsaconf says no supported cords; yet /proc/asound contains cards

    - by nimasmi
    I am trying to get HDMI output using a Gainward Nvidia 210 512 MB on Ubuntu 10.04 Lucid Lynx. I have upgraded alsa-driver, alsa-lib and alsa-utils to 1.0.24 by building from source, thanks to this blog post. Some relevant output... user@box:~$ lspci | grep Audio 00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2) 01:09.0 Multimedia video controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder (rev 05) 01:09.2 Multimedia controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder [MPEG Port] (rev 05) 01:09.4 Multimedia controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder [IR Port] (rev 05) 02:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) user@box:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.24. Compiled on Sep 15 2012 for kernel 2.6.32-42-generic (SMP). user@box:~$ ls /proc/asound` card0 cards hwdep NVidia oss seq version card1 devices modules NVidia_1 pcm timers user@box:~$ aplay -l aplay: device_list:240: no soundcards found... user@box:~$ sudo /sbin/alsa-utils start * Setting up ALSA... * warning: 'alsactl restore' failed with error message 'alsactl: set_control:1403: Cannot write control '2:0:0:IEC958 Playback Default:0' : Operation not permitted'... amixer: Invalid command! ...done. Any help appreciated. PS my video card is connected only through the PCI-E slot. I assume there is no extra audio connection required.

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  • Best way to learn iphone audio queue services, step by step tutorial

    - by optician
    Hi Everyone, I'm trying to learn how to handle audio at a fairly low level with audio queue services. I have been progrmaing in memory managed languages for quite a while, and have just completed the c programing tutorial by vtc (2007). This has left me comfortable with the understanding of pointers and memory allocation, but the apple documention still leaves me wanting for a simpler implenation and explaination. Maybe I need to learn objective c and cocoa better. I have heard that this book is good. Cocoa(R) Programming for Mac(R) OS X (3rd Edition) Could someone suggest a learning path that is going to help me get an better understanding of working with audio and an iphone. I want to be able to play mp3 files back and also alter the pitch of them as they are playing. I am prepared that I may have to temporarily convert the mp3 files into pcm files to do things like that to them. Thanks everyone.

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  • Adding audio channel using ffmpeg

    - by Raj
    Hi all, I am working on ffmpeg and trying to add a audio stream on the fly. I am using AudioQueues and I get raw audio buffer. I am encoding audio with linear PCM and hence the audio I get will be of raw format, which I know ffmpeg does accept it. But I cannot figure out how. I have looked into AVStream, where in we have to create a new stream for this audio channel but how do I encode it to a video which is already initialized in another AVStream structure. Overall, I would like to have an idea of the architecture of ffmpeg. I found it difficult to work since it is least documented. Any pointers or details are appreciated. Thanks and Regards, Raj Pawan G

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  • ffmpeg 0.5 flv to wav conversion creates wav files that other programs won't open.

    - by superrebel
    Hi, I am using the following command to convert FLV files to audio files to feed into julian, a speech to text program. cat ./jon2.flv | ffmpeg -i - -vn -acodec pcm_s16le -ar 16000 -ac 1 -f wav - | cat - > jon2.wav The cat's are there for debugging purposes as the final use will be a running program that will pipe FLV into ffmpeg's stdin and the stdout going to julian. The resulting wave files are identified by "file" as: jon3.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz VLC (based on ffmpeg) plays the file, but no other tools will open/see the data. They show empty wav files or won't open/play. For example Sound Booth from CS4. Has anyone else had similar problems? Julian requires wav files 16bit mono at 16000 Hz. Julian does seem to read the file, but doesn't seem to go through the entire file (may be unrelated). Thanks, -rr

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  • How can I play compressed sound files in C# in a portable way?

    - by skolima
    Is there a portable, not patent-restricted way to play compressed sound files in C# / .Net? I want to play short "jingle" sounds on various events occuring in the program. System.Media.SoundPlayer can handle only WAV, but those are typically to big to embed in a downloadable apllication. MP3 is protected with patents, so even if there was a fully managed decoder/player it wouldn't be free to redistribute. The best format available would seem to be OGG Vorbis, but I had no luck getting any C# Vorbis libraries to work (I managed to extract a raw PCM with csvorbis but I don't know how to play it afterwards). I neither want to distribute any binaries with my application nor depend on P/Invoke, as the project should run at least on Windows and Linux. I'm fine with bundling .Net assemblies as long as they are license-compatible with GPL. [this question is a follow up to a mailing list discussion on mono-dev mailing list a year ago]

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  • Write wave files to memory in Java

    - by Cliff
    I'm trying to figure out why my servlet code creates wave files with improper headers. I use: AudioSystem.write( new AudioInputStream( new ByteArrayInputStream(memoryBytes), new AudioFormat(22000, 16, 1, true,false), memoryBytes.length ), AudioFileFormat.Type.WAVE, servletOutputStream ); taking a byte array from memory containing raw PCM samples and a servlet output stream that gets returned to the client. In the result I get a normal wave file but with zeros in the chunk size fields. Is the API broken? I would think that the size could be filled in using the size passed in the audio input stream. But now, after typing this out I'm thinking its not making this info available to the outer write() method on AudioSystem. It seems like the AudioSystem.write call needs a size parameter unless it is able to pull the size from the stream... which wouldn't work with an arbitrary sized stream. Does anyone know how to make this example work?

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  • Why does Silverlight provides webcam and microphone support without any encoding API?

    - by Shurup
    In the list of new features in Silverlight 4 you will find following: Webcam and microphone to allow sharing of video and audio for instance for chat or customer service applications. Silverlight captures an audio stream as raw pcm. So how would you realize for example audio/video chat or client/server audio recording application without any encoding on the client side, where there is no APIs in Silverlight available? Much less in a Silverlight you cannot use an unmanaged dll. You can use a com automation (a new feature of the Silverlight 4, I think only for Windows) but only if it was already installed on the client side (do you know any encoding COM servers that are installed with the windows). Otherwise, how would you deploy a custom COM server within you Silverlight application? The only way I found is either to deploy a command-line encoding and use it with COM AutomationFactory.CreateObject("WScript.Shell") or to implement an encoding to use it in your own AudioSink.

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  • C++ Microsoft SAPI: How to set Windows text-to-speech output to a memory buffer?

    - by Vladimir
    Hi all, I have been trying to figure out how to "speak" a text into a memory buffer using Windows SAPI 5.1 but so far no success, even though it seems it should be quite simple. There is an example of streaming the synthesized speech into a .wav file, but no examples of how to stream it to a memory buffer. In the end I need to have the synthesized speech in a char* array in 16 kHz 16-bit little-endian PCM format. Currently I create a temp .wav file, redirect speech output there, then read it, but it seems to be a rather stupid solution. Anyone knows how to do that? Thanks!

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  • Is it possible to access raw iphone audio output?

    - by Peter Hall
    Is it possible access raw PCM data from the iphone audio output? I know I can embed an MP3 and use AudioUnit. But if the user is playing music in the background from their itunes library, is it possible to access that audio data? This is for an app that shows visual effects, which react to the music. From what I can tell, it isn't possible, but that's just from lack of finding any information at all, rather than actual confirmation that it can't be done. If it isn't possible to access the audio stream from the ipod, is it possible to access raw audio output from the Media Player inside an app, or is pretty much not permitted to access raw audio data from the itunes library at all? EDIT: I found this question: iOS - Access output audio from background program, which say I can't access the audio from a background app. But is it possible to get the audio data from the itunes library if I play it inside the app?

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  • audio stream sampling rate in linux

    - by farhan
    Im trying read and store samples from an audio microphone in linux using C/C++. Using PCM ioctls i setup the device to have a certain sampling rate say 10Khz using the SOUND_PCM_WRITE_RATE ioctl etc. The device gets setup correctly and im able to read back from the device after setup using the "read". int got = read(itsFd, b.getDataPtr(), b.sizeBytes()); The problem i have is that after setting the appropriate sampling rate i have a thread that continuously reads from /dev/dsp1 and stores these samples, but the number of samples that i get for 1 second of recording are way off the sampling rate and always orders of magnitude more than the set sampling rate. Any ideas where to begin on figuring out what might be the problem?

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  • how to play an encrypted file in Android.

    - by user306517
    I need to be able to play an encrypted file in Android. The file is AAC. The only way I can see to do this is either: decrypt the file to internal private storage and point the player at that file to play, or decrypt & decode the file to pcm and feed it to an AudioTrack. 1 isn't great because it takes a long time to do that. 2 isn't great either because I don't know how I can take advantage of the HW decoder to do this. Any ideas? tia.

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  • C# - .WAV Playback Randomly High Pitch

    - by Nate Shoffner
    For some reason, when a WAV file is played back using the snippet below, it randomly plays back screwy, like a high pitch noise. It doesn't happen all the time, just randomly. It seems to happen more often when it is played back more frequently. The WAV properties are below along with the code snippet I am using. WAV Properties: Bit Rate - 750kbps Audio Sample Size - 16 bit Channels - 1 (mono) Audio Sample Rate - 44kHz Audio Format - PCM Snippet: System.Media.SoundPlayer myPlayer = new System.Media.SoundPlayer(Captcha.Properties.Resources.sound1); myPlayer.Play(); Is this because of the way I am playing the file or the file itself? Thank you.

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