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  • Finding out if a FLAC or WAVPACK audio file is NOT originally encoded from a lossy source

    - by cornel
    Is there a way of checking that the so-called FLAC or WAVPACK audio file was originally encoded from a lossless source (WAV, CDA, APE, etc.) instead of a lossy source (MP3, AAC, ATRAC, etc.)? Say I have a lossy MP3 audio file (5.17Mb, 87% compressed from its original, source unknown). I then encode it to another lossless format, say FLAC or WAVPACK. The size increases (23.14Mb, 39% compressed from its original, source MP3)! ID tags, etc, remain the same and there's no way of checking the integrity of its origin. How do I go about doing that?

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  • How do I make my Geforce GTS 250's power save mode stop causing audio stuttering?

    - by Matt
    Whenever my GTS 250 enters its power save mode, downscaling its frequencies, my audio stutters. This affects both my onboard audio and my Audigy Soundblaster 2 ZS. Changing Windows power save mode options such as PCI-E link state power management or Power Management Mode in the nVidia control panel have no effect on this issue. Replacing the power supply had no effect on this issue. The BIOS is the latest version, and I have the latest motherboard chipset and graphics drivers installed. I do not overclock. I started to see this issue after I upgraded my rig from its Socket 939 board to a Socket 1156 board with a Core i5-750 while simultaneously upgrading from Vista to 7.

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  • How do I swap audio output of the left and right speakers?

    - by Manga Lee
    I have two speakers stereo speakers but when I use the sound control panel applet to test my audio configuration I get sound in the right speaker when the user interface indicates the right speaker and vice versa. Is there a way to swap the audio output from left to right and right to left? UPDATE: The reason for this question is that I've recently rearranged my workspace and because of physical constraints the left speaker has to go on the right side and vice versa. I could of course solve this problem with a hardware solution but I'd rather use a software solution if one is available.

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  • How can you convert audio (3.5mm) to S/PDIF?

    - by SSumner
    I have a monitor that I want to use as my 'TV' for my gaming system. I connect it via HDMI, so sound and video go through the monitor, but I want sound to my headphones, which travels via optical (S/PDIF) cable. The monitor (Dell U2713HM) has a 3.5mm audio jack on it for line out, but I couldn't find anything that simply plugs in an converts the analog audio signal to a digital one so I can plug in a S/PDIF cable. What sort of device do I need to do this? (I am not asking for shopping recommendations, merely what options allow this conversion. I would prefer the smallest option, as space is limited).

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  • Anyone experiencing audio issues with VirtualBox on Linux and has a solution?

    - by DoxaLogos
    I've been using Virtualbox (now at 3.0.2) on Kubuntu (now at 9.04) for a while now, and I seem to have a problem when running Windows. Sometime after a while the audio will cut out in Kubuntu. The only way I can get it to recover is to make sure VirtualBox is completely shutdown and either going into multimedia under "system settings" and test the audio or restart. I'm wondering if anyone else here has experienced similar issues and has come up with a more elegant solution. I can't seem to find a reasonable one at virtualbox.org.

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  • I just don't get AudioFileReadPackets

    - by Eric Christensen
    I've tried to write the smallest chunk of code to narrow down a problem. It's now just a few lines and it doesn't work, which makes it pretty clear that I have a fundamental misunderstanding of how to use AudioFileReadPackets. I've read the docs and other examples online, and apparently I'm just not getting. Could you explain it to me? Here's what this block should do: I've previously opened a file. I want to read just one packet - the first one of the file - and then print it. But it crashes on the AudioFileReadPackets line: AudioFileID mAudioFile2; AudioFileOpenURL (audioFileURL, 0x01, 0, &mAudioFile2); UInt32 *audioData2 = (UInt32 *)malloc(sizeof(UInt32) * 1); AudioFileReadPackets(mAudioFile2, false, NULL, NULL, 0, (UInt32*)1, audioData2); NSLog(@"first packet:%i",audioData2[0]); (For clarity, I've stripped out all error handling.) It's the AFRP line that crashes out. (I understand that the third and fourth argument are useful, and in my "real" code, I use them, but they're not required, right? So NULL in this case should work, right?) So then what's going on? Any guidance would be much appreciated. Thanks.

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  • record output sound in python

    - by aaronstacy
    i want to programatically record sound coming out of my laptop in python. i found PyAudio and came up with the following program that accomplishes the task: import pyaudio, wave, sys chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 RECORD_SECONDS = 5 WAVE_OUTPUT_FILENAME = sys.argv[1] p = pyaudio.PyAudio() channel_map = (0, 1) stream_info = pyaudio.PaMacCoreStreamInfo( flags = pyaudio.PaMacCoreStreamInfo.paMacCorePlayNice, channel_map = channel_map) stream = p.open(format = FORMAT, rate = RATE, input = True, input_host_api_specific_stream_info = stream_info, channels = CHANNELS) all = [] for i in range(0, RATE / chunk * RECORD_SECONDS): data = stream.read(chunk) all.append(data) stream.close() p.terminate() data = ''.join(all) wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb') wf.setnchannels(CHANNELS) wf.setsampwidth(p.get_sample_size(FORMAT)) wf.setframerate(RATE) wf.writeframes(data) wf.close() the problem is i have to connect the headphone jack to the microphone jack. i tried replacing these lines: input = True, input_host_api_specific_stream_info = stream_info, with these: output = True, output_host_api_specific_stream_info = stream_info, but then i get this error: Traceback (most recent call last): File "./test.py", line 25, in data = stream.read(chunk) File "/Library/Python/2.5/site-packages/pyaudio.py", line 562, in read paCanNotReadFromAnOutputOnlyStream) IOError: [Errno Not input stream] -9975 is there a way to instantiate the PyAudio stream so that it inputs from the computer's output and i don't have to connect the headphone jack to the microphone? is there a better way to go about this? i'd prefer to stick w/ a python app and avoid cocoa.

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  • getAudioInputStream can not convert [stereo, 4 bytes/frame] stream to [mono, 2 bytes/frame]

    - by brian_d
    Hello. I am using javasound and have an AudioInputStream of format PCM_SIGNED 8000.0 Hz, 16 bit, stereo, 4 bytes/frame, little-endian Using AudioSystem.getAudioInputStream(target_format, original_stream) produces an 'IllegalArgumentException: Unsupported Conversion' when the target_format is PCM_SIGNED 8000.0 Hz, 16 bit, mono, 2 bytes/frame, little-endian Is it possible to convert this stream manually after every read() call? And if yes, how? In general, how can you compare two formats and tell if a conversion is possible?

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  • Pay per view video solution

    - by Bassem Hefny
    Hello, We are planning on building a pay per view (PPV) video solution but we have no idea from where to start. Here are the current givens: it will be hosted on Linux using PHP Database: MySQL And by PPV I mean: - going to website, selecting a movie to watch/download - going to payment portal and paying - being now able to watch/download So here is my question, from where to start? is there an existing (recommended) solution that we can download/buy? Any information would be really appreciated

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  • How to stream video content in asp.net?

    - by Kon
    Hi, I have the following code which downloads video content: WebRequest wreq = (HttpWebRequest)WebRequest.Create(url); using (HttpWebResponse wresp = (HttpWebResponse)wreq.GetResponse()) using (Stream mystream = wresp.GetResponseStream()) { using (BinaryReader reader = new BinaryReader(mystream)) { int length = Convert.ToInt32(wresp.ContentLength); byte[] buffer = new byte[length]; buffer = reader.ReadBytes(length); Response.Clear(); Response.Buffer = false; Response.ContentType = "video/mp4"; //Response.BinaryWrite(buffer); Response.OutputStream.Write(buffer, 0, buffer.Length); Response.End(); } } But the problem is that the whole file downloads before being played. How can I make it stream and play as it's still downloading? Or is this up to the client/receiver application to manage?

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  • How do I record streams in chunks on Flash Media Server.

    - by Vasil
    I want to record a stream which is published with Flash Live Encoder to FMS 3.5, but split the recording in files with predefined length. For example if a stream 'webcam' is published I want to record it in chunks of 10 minutes: 'webcam1.flv', 'webcam2.flv' ... From what I can tell there's no facility to work with timers. The only solution I could think of was using stream.record() with a time limit parameter but that seems like a hack because it triggers NetStream.Record.DiskQuotaExceeded on the stream when the recordin should stop and start recording another chunk. Has anyone done something similar?

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  • VLC Functionality: How to stream an .iso?

    - by MattUebel
    I gave a DVD ISO image and would like to use VLC's http function to create URL a user could access, after which the DVD would start up in full screen. I can make this happen by accepting the defaults vlc.exe -I http navigating to localhost:8080 and then browsing and opening the file. How would I modify this activity so that navigating to localhost:8080 instead opens a predetermined file and starts playing it?

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  • GStreamer record iradio-mode artifacts

    - by Kanzeon
    I'm trying to record internet radio while listen it. I use the following line, but comes to my attention that when I set the iradio-mode true some noises comes in the recorded file, not in the playback. Without iradio-mode, all is ok. But in my app I need this mode to get the title message. gst-launch souphttpsrc location="<radio channel>" iradio-mode=true ! tee name=t ! queue ! decodebin2 ! audioconvert ! audioresample ! osxaudiosink t. ! queue ! filesink location=rectest.mp3

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  • AudioTrack skipping after pause and resume

    - by Markus Drösser
    Hi, here is the problem. I play a wav file that i recorded earlier without problems. but when i call audiotrack.pause() and audiotrack.start() again after some waiting, it skips some frames of the file. why is that? here is my play listener // Start playback audioTrack.setPlaybackPositionUpdateListener(new OnPlaybackPositionUpdateListener() { @Override public void onPeriodicNotification(AudioTrack track) { try { if(ramfile!=null && ramfile.read(buffer)==-1) { audioTrack.release(); audioTrack = null; ramfile.close(); playing=false; } else { audioTrack.write(buffer, 0, buffer.length); } } catch (IOException e) { try { ramfile.close(); playing=false; } catch (IOException e1) { } } } @Override public void onMarkerReached(AudioTrack track) { playing=false; track.release(); } });

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  • What is meant by "streaming data access" in HDFS?

    - by Van Gale
    According to the HDFS Architecture page HDFS was designed for "streaming data access". I'm not sure what that means exactly, but would guess it means an operation like seek is either disabled or has sub-optimal performance. Would this be correct? I'm interested in using HDFS for storing audio/video files that need to be streamed to browser clients. Most of the streams will be start to finish, but some could have a high number of seeks. Maybe there is another file system that could do this better?

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  • Un groupe de développeurs sort Flac.js, un décodeur JavaScript pour la lecture du contenu audio dans le navigateur sans recours aux codecs

    Un groupe de développeurs sort Flac.js un décodeur audio en JavaScript pour la lecture du contenu audio dans le navigateur sans nécessiter de codecs HTML5, le futur standard du Web introduit la balise audio permettant de créer des applications fournissant le traitement et la synthèse audio dans le navigateur. Les navigateurs récents comme Chrome ou Firefox, intègrent déjà des bibliothèques Javascript qui fournissent des méthodes et propriétés permettant de manipuler l'élément audio. Cependant, les applications HTML 5 manipulant du contenu audio qui fonctionnent normalement dans un navigateur sur un système d'exploitation donné pourraient ne pas marcher correctement lors de...

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  • Rhythmbox - access Windows 7 Media Streaming

    - by rifferte
    I wish to be able to see and stream music to my Ubuntu 10.04 installation through Rhythmbox. I have enabled media streaming in Windows 7 and I can see Rhythmbox as an allowed device. I have installed the Coherence plugin for Rhythmbox. I can see my Windows 7 PC under the Shared folder in Rhythmbox, but I do not see any of my music. Is there a step along the way that I missed or something else that I have to enable?

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  • What music streaming app fits my needs on Ubuntu?

    - by Jim
    I'm looking for an application to stream Internet radio on Ubuntu. I like listening to Radio Paradise while I work. Right now, I'm using Amarok. "Movie Player" sometimes refuses to open the stream, and VLC doesn't keep its window title updated with the currently playing track. Amarok has nice translucent notifications when tracks change, but track changes in streams don't trigger the notifications. Mostly, I want something that reliably opens streams and makes it easy to see the name of the track that's playing. If it has a built-in directory of streaming radio stations, that would be a big benefit.

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