Search Results

Search found 5304 results on 213 pages for 'audio streaming'.

Page 61/213 | < Previous Page | 57 58 59 60 61 62 63 64 65 66 67 68  | Next Page >

  • Streaming webcam video in Flash using MP4 encoding

    - by Herms
    One of the features of the Flash app I'm working on is to be able to stream a webcam to others. We're just using the built-in webcam support in Flash and sending it through FMS. We've had some people ask for higher quality video, but we're already using the highest quality setting we can in Flash (setting quality to 100%). My understanding is that in the newer flash players they added support for MPEG-4 encoding for the videos. I created a simple test Flex app to try and compare the video quality of the MP4 vs FLV encodings. However, I can't seem to get MP4 to work at all. According to the Flex documentation the only thing I need to do to use MP4 instead of FLV is prepend "mp4:" to the name of the stream when calling publish: Specify the stream name as a string with the prefix mp4: with or without the filename extension. The prefix indicates to the server that the file contains H.264-encoded video and AAC-encoded audio within the MPEG-4 Part 14 container format. When I try this nothing happens. I don't get any events raised on the client side, no exceptions thrown, and my logging on the server side doesn't show any streams starting. Here's the relevant code: // These are all defined and created within the class. private var nc:NetConnection; private var sharing:Boolean; private var pubStream:NetStream; private var format:String; private var streamName:String; private var camera:Camera; // called when the user clicks the start button private function startSharing():void { if (!nc.connected) { return; } if (sharing) { return; } if(pubStream == null) { pubStream = new NetStream(nc); pubStream.attachCamera(camera); } startPublish(); sharing = true; } private function startPublish():void { var name:String; if (this.format == "mp4") { name = "mp4:" + streamName; } else { name = streamName; } //pubStream.publish(name, "live"); pubStream.publish(name, "record"); }

    Read the article

  • iPhone SDK: How to record voices with ambient noise supression?

    - by Harkonian
    Can anyone point me in the right direction on how I would minimize ambient noise while recording someone speaking using the iPhone SDK Core Audio? I'm guessing a band-pass filter that eliminates any frequencies above and below the human vocal range might work. I have no idea how I would implement band filters on audio in the SDK though. The optimum solution would be one that eliminates the noise from the stream before it is written to memory/disk. Some sample code would be appreciated.

    Read the article

  • iPhone: CPU power to do DSP/Fourier transform/frequency domain?

    - by mahboudz
    I want to analyze MIC audio on an ongoing basis (not just a snipper or prerecorded sample), and display frequency graph and filter out certain aspects of the audio. Is the iPhone powerful enough for that? I suspect the answer is a yes, given the Google and iPhone voice recognition, Shazaam and other music recognition apps, and guitar tuner apps out there. However, I don't know what limitations I'll have to deal with. Anyone play around with this area?

    Read the article

  • iPhone SDK audioSession question.

    - by Morion
    Hi to all. In my app i record and play audio at the same time. The app is almost finished. But there is one thing, that annoying me. When audio session is set to PlayAndRecord, sounds become quiet in comparison with the same sounds with the SoloAmbient category. Is there any way to make sound louder using PlayAndRecord?

    Read the article

  • 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?

    Read the article

  • Identifying voice as male or female

    - by duder
    I'm not much into audio engineering, so please be easy on me. I'm receiving an audio file as input, and need to detect whether the speaker is male or female. Any ideas how to go about doing this? I'm using php, but am open to using other languages, and don't mind learning a little bit of sound theory as long as the time is proportionate to the task.

    Read the article

  • How does the Ableton warp algorithm work exactly?

    - by pepperdreamteam
    I'm looking for any documentation or definitive information on Ableton's warp feature. I understand that it has something to do with finding transients, aligning them with an even rhythm and shifting audio samples accordingly. I'm hoping to find ways to approximate warping with more basic audio editing tools. I understand that this is ableton's unique device, really any information about how it works would be helpful. So...does anyone have any 411?

    Read the article

  • Any tips for how to build a LED sytem thet will light up to music?

    - by daniels
    So basically I would like somehow that given an audio file as input (most likely mp3 or I can use some audio engine that will handle other types too) from my computer to control some LED lights so they will be something like an oscilloscope, like the one in winamp. What would I need to be able to do this? I'm interested in building thing up all by myself, coding, hardware, etc.. I'm going with C++ on Windows.

    Read the article

  • AudioTrack lag: obtainBuffer timed out

    - by BTR
    I'm playing WAVs on my Android phone by loading the file and feeding the bytes into AudioTrack.write() via the FileInputStream BufferedInputStream DataInputStream method. The audio plays fine and when it is, I can easily adjust sample rate, volume, etc on the fly with nice performance. However, it's taking about two full seconds for a track to start playing. I know AudioTrack has an inescapable delay, but this is ridiculous. Every time I play a track, I get this: 03-13 14:55:57.100: WARN/AudioTrack(3454): obtainBuffer timed out (is the CPU pegged?) 0x2e9348 user=00000960, server=00000000 03-13 14:55:57.340: WARN/AudioFlinger(72): write blocked for 233 msecs, 9 delayed writes, thread 0xba28 I've noticed that the delayed write count increases by one every time I play a track -- even across multiple sessions -- from the time the phone has been turned on. The block time is always 230 - 240ms, which makes sense considering a minimum buffer size of 9600 on this device (9600 / 44100). I've seen this message in countless searches on the Internet, but it usually seems to be related to not playing audio at all or skipping audio. In my case, it's just a delayed start. I'm running all my code in a high priority thread. Here's a truncated-yet-functional version of what I'm doing. This is the thread callback in my playback class. Again, this works (only playing 16-bit, 44.1kHz, stereo files right now), it just takes forever to start and has that obtainBuffer/delayed write message every time. public void run() { // Load file FileInputStream mFileInputStream; try { // mFile is instance of custom file class -- this is correct, // so don't sweat this line mFileInputStream = new FileInputStream(mFile.path()); } catch (FileNotFoundException e) {} BufferedInputStream mBufferedInputStream = new BufferedInputStream(mFileInputStream, mBufferLength); DataInputStream mDataInputStream = new DataInputStream(mBufferedInputStream); // Skip header try { if (mDataInputStream.available() > 44) mDataInputStream.skipBytes(44); } catch (IOException e) {} // Initialize device mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, ConfigManager.SAMPLE_RATE, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT, ConfigManager.AUDIO_BUFFER_LENGTH, AudioTrack.MODE_STREAM); mAudioTrack.play(); // Initialize buffer byte[] mByteArray = new byte[mBufferLength]; int mBytesToWrite = 0; int mBytesWritten = 0; // Loop to keep thread running while (mRun) { // This flag is turned on when the user presses "play" while (mPlaying) { try { // Check if data is available if (mDataInputStream.available() > 0) { // Read data from file and write to audio device mBytesToWrite = mDataInputStream.read(mByteArray, 0, mBufferLength); mBytesWritten += mAudioTrack.write(mByteArray, 0, mBytesToWrite); } } catch (IOException e) { } } } } If I can get past the artificially long lag, I can easily deal with the inherit latency by starting my write at a later, predictable position (ie, skip past the minimum buffer length when I start playing a file).

    Read the article

  • How to set a native object property

    - by theunilife
    ok so im creating a jquery plugin that will allow me to use the new html5 Audio interface and im trying to create an option that is an object that you will be able to set the various listeners but i dont seem to be able to set those options to the listener property of the Audio object.

    Read the article

  • Any tips for how to build a LED system thet will light up to music?

    - by daniels
    So basically I would like somehow that given an audio file as input (most likely mp3 or I can use some audio engine that will handle other types too) from my computer to control some LED lights so they will be something like an oscilloscope, like the one in winamp. What would I need to be able to do this? I'm interested in building thing up all by myself, coding, hardware, etc.. I'm going with C++ on Windows.

    Read the article

  • Streaming a webcam from Silverlight 4 (Beta)

    - by Ken Smith
    The new webcam stuff in Silverlight 4 is darned cool. By exposing it as a brush, it allows scenarios that are way beyond anything that Flash has. At the same time, accessing the webcam locally seems like it's only half the story. Nobody buys a webcam so they can take pictures of themselves and make funny faces out of them. They buy a webcam because they want other people to see the resulting video stream, i.e., they want to stream that video out to the Internet, a lay Skype or any of the dozens of other video chat sites/applications. And so far, I haven't figured out how to do that with It turns out that it's pretty simple to get a hold of the raw (Format32bppArgb formatted) bytestream, as demonstrated here. But unless we want to transmit that raw bytestream to a server (which would chew up way too much bandwidth), we need to encode that in some fashion. And that's more complicated. MS has implemented several codecs in Silverlight, but so far as I can tell, they're all focused on decoding a video stream, not encoding it in the first place. And that's apart from the fact that I can't figure out how to get direct access to, say, the H.264 codec in the first place. There are a ton of open-source codecs (for instance, in the ffmpeg project here), but they're all written in C, and they don't look easy to port to C#. Unless translating 10000+ lines of code that look like this is your idea of fun :-) const int b_xy= h->mb2b_xy[left_xy[i]] + 3; const int b8_xy= h->mb2b8_xy[left_xy[i]] + 1; *(uint32_t*)h->mv_cache[list][cache_idx ]= *(uint32_t*)s->current_picture.motion_val[list][b_xy + h->b_stride*left_block[0+i*2]]; *(uint32_t*)h->mv_cache[list][cache_idx+8]= *(uint32_t*)s->current_picture.motion_val[list][b_xy + h->b_stride*left_block[1+i*2]]; h->ref_cache[list][cache_idx ]= s->current_picture.ref_index[list][b8_xy + h->b8_stride*(left_block[0+i*2]>>1)]; h->ref_cache[list][cache_idx+8]= s->current_picture.ref_index[list][b8_xy + h->b8_stride*(left_block[1+i*2]>>1)]; The mooncodecs folder within the Mono project (here) has several audio codecs in C# (ADPCM and Ogg Vorbis), and one video codec (Dirac), but they all seem to implement just the decode portion of their respective formats, as do the java implementations from which they were ported. I found a C# codec for Ogg Theora (csTheora, http://www.wreckedgames.com/forum/index.php?topic=1053.0), but again, it's decode only, as is the jheora codec on which it's based. Of course, it would presumably be easier to port a codec from Java than from C or C++, but the only java video codecs that I found were decode-only (such as jheora, or jirac). So I'm kinda back at square one. It looks like our options for hooking up a webcam (or microphone) through Silverlight to the Internet are: (1) Wait for Microsoft to provide some guidance on this; (2) Spend the brain cycles porting one of the C or C++ codecs over to Silverlight-compatible C#; (3) Send the raw, uncompressed bytestream up to a server (or perhaps compressed slightly with something like zlib), and then encode it server-side; or (4) Wait for someone smarter than me to figure this out and provide a solution. Does anybody else have any better guidance? Have I missed something that's just blindingly obvious to everyone else? (For instance, does Silverlight 4 somewhere have some classes I've missed that take care of this?)

    Read the article

  • How to show a video stored on server on iphone

    - by Amitkumar
    Hi, I have a query regarding showing a video (which is stored on server) on iPhone. I want show a video in an iPhone Application. This is not live streaming. So how the video can be shown? I have read the Apple's documentation for HTTP streaming of video. Do I need to call a Web Service? Is there any tutorial for this? Thanks in advance..

    Read the article

  • Encode real-time dvb-s stream using mencoder

    - by karatchov
    My satellite receiver can stream the mpeg-2 video/audio output through lan. Using mencoder, I'm trying to build a script to encode and save the stream in real time with my Core2Duo 1.8 Ghz. Right now, I'm using a single pass, it produces good quality for a video rate of 800Kb/s, but takes more then 95% of CPU power, thus making a lot of frameskips is the computer is used while encoding. mencoder -o -vf lavcdeint -oac mp3lame -lameopts abr:q=2:aq=2 -ovc x264 -ffourcc avc1 -x264encopts crf=25:me=hex:subq=9:frameref=2:nocabac:threads=auto -mc 3 So, I'm considering using a 2-pass encoding to alleviate the processor and record 100% of the stream. But I have no idea how to start. For the info: Standard Stream: mpeg-2 720*576 25fps HD Stream: 1920*1080 50fps (this is not my goal to record it, but it will be super cool if I could)

    Read the article

  • Stream video file in debian?

    - by Rob
    I've tried ffserver with ffmpeg, I've tried VLC, and I'm not sure what else to try or what I've done wrong. I've gone through, with VLC +-[ robert@s10 ]--[ ~ ] +[#!]¬ vlc --version VLC media player 2.0.0 Twoflower (revision 2.0.0-0-g421a4fc) VLC version 2.0.0 Twoflower (2.0.0-0-g421a4fc) Compiled by buildd on biber.debian.org (Mar 1 2012 22:21:37) Compiler: gcc version 4.6.2 (Debian 4.6.2-14) This program comes with NO WARRANTY, to the extent permitted by law. You may redistribute it under the terms of the GNU General Public License; see the file named COPYING for details. Written by the VideoLAN team; see the AUTHORS file. and tried everything I could in the streaming section, but I can't get the stream to actually work. Looking around, apparently debian strips the encoders from the package? I want to do share some videos I've made with friends on IRC, and it would be easiest if I could just stream it so we can all watch at the same time and critique parts of it in real time. Has anyone done something similar? Linux s10 3.2.0-2-686-pae #1 SMP Tue Mar 20 19:48:26 UTC 2012 i686 GNU/Linux Basic home network, I am behind a NAT (192.168.1.*) and have dynamic DNS set up. That doesn't really matter too much, I can figure that out, but it's not even working locally. I have a file server set up and could just share the files that way, but I'd rather have everyone watching at the same time (or just about). Not worried about installing new packages or building something from source, that's not a big issue, just want to get it working. Big plus if I can do it from command line.

    Read the article

  • Why do my speakers get distorted randomly on Windows 7?

    - by Daniel Fischer
    I have a studio monitor setup. I have 2 KRK 6's and a Focusrite Firewire Pro 24. Every few hours my speakers sound distorted and my solution has been go to sound levels Properties of Saffire Audio Device Advanced Default Format Toggle to 16 bit then back to 24bit. Why does it screw up every few hours? Sometimes one speaker doesn't output too and this same process resets it but that's more rare. Is this a OS issue or Focusrite Driver Issue?

    Read the article

  • How to open an iPhone compatible M3U file on Windows?

    - by user1158667
    This is how the M3U file looks like: #EXTM3U #EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=1400000 http://maskedip/http_livestr.str?r=true&id=mbit-test&k=testkey #EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=900000 http://maskedip/http_livestr.str?r=true&id=test&k=testkey #EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=450000 http://maskedip/http_livestr.str?r=true&id=mobile-test&k=testkey #EXT-X-STREAM-INF:PROGRAM-ID=1,CODECS="mp4a.40.2",BANDWIDTH=64000 http://maskedup/http_livestr.str?r=true&id=test-audio&k=testkey Clicking on http://maskedip/http_livestr.str?r=true&id=mbit-test&k=testkey then returns another M3U file in this format: #EXTM3U #EXT-X-TARGETDURATION:10 #EXT-X-MEDIA-SEQUENCE:1361 #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1361.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1362.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1363.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1364.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1365.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1366.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1367.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1368.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1369.ts Anyways, VLC won't recognize it. How can I play this on the PC?

    Read the article

  • OS X Headphone jack issue [closed]

    - by Alex Coady
    Possible Duplicate: Optical Audio out stuck on on a MacBook When I plug my headphones into my iMac (27-inch, Mid 2011; OSX 10.8.1) and try to adjust the volume, the volume popup shows a greyed out speaker and there's a circle with a line through it signalling that it isn't working. I've tried the headphones with my iPhone, other iMacs etc and they're fine. This is incredibly frustrating. Other headphones don't generally work either. In Sound preferences the headphones are being listed as "Optical digital-out port" which is incorrect and would explain the problem, but doesn't help me fix it. Any ideas?

    Read the article

  • Songs/Movies are not being played

    - by romilnagrani
    Hi the problem i am facing is very odd. None of my songs (.mp3/.avi etc) are not able to run in any of the media players: Windows media player, Windows media center, VLC, DIV etc. What i have done before posting the question here. a. Check Device and Drivers and they are working OK and Updated even. b. Run Sample Audio (Downloaded from Internet) in Adobe Sound-booth and it worked. c. Plus i run a video in YouTube and i was able to hear it! d. I am able to hear windows Error Beeps too..!!! e. While running Movies i could see playing but not hear it What could be the problem? please help

    Read the article

< Previous Page | 57 58 59 60 61 62 63 64 65 66 67 68  | Next Page >