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• No external microphone Acer AO722

  - by Leeghwater

  The ACER AO722 comes with an external mic input, and this input is not recognised by Alsa mixer or Sound (in System Settings). There are various comments on this problem, but no real solutions. For example External Mic not working but Internal Mic works on an Acer Aspiron AO722. Using the internal mic is not an option, as I need to use skype professionally. I have tried everything in alsamixer (accessible through the Terminal Ctrl+Alt+t, command: alsamixer), and in Sound (under System Settings). I have also installed Pulseaudio. But to no avail. The headset is working normally under Skype in Windows. My AO722 came with Windows 7 on it, so I have installed Skype there too. My headset has separate connectors for ears and mic, and these go into the respective output and input on the right side of the laptop. This location: http://bernaerts.dyndns.org/linux/202-ubuntu-acer-ao722 sounds like an effective solution but it is for Ubuntu Natty 11.04. The solution suggested sounds drastic to me: replace the kernel 2.6.38-13 with version 2.6.38-12. I use Ubuntu 12.04, and my kernel is 3.2.0-30-generic-pae. Question: could I try this solution with Ubuntu 12.04? Is this a risky thing to do? I have found harware work around this problem. The audio output seems to be a combi output with also a microphone connection. I have made an adapter for this output. I used a 4 contacts 3,5 mm audio jack plug. To this plug I have soldered 2 female (common stereo) connectors, one for ears and one for the mic of my headset. The 4 contacts jack, which goes into the laptop (in audio OUTput) is wired as follows: tip = hot audio right; first sleeve after tip = hot audio left; second sleeve = common earth (for both ears and microphone); the 3rd sleeve = microphone signal input. In the connector which I could buy, the 3rd sleeve is not so much a sleeve, but part of the metal base of the connector; normally you would expect this one to be connect to earth. But connecting the mic signal to it works. Maybe ready made adapters of this kind and even headsets with a combi jack can simply be purchased; I didn't check. When I plug in the 4 contacts jack, Sound and Alsamixer immediately recognise an external microphone (even if no mic is connected to the adapter). In Sound, under the Input tab, 'Settings for internal microphone' changes into 'Setting for microphone'. The microphone comes through loud and clear, however there is a constant noise in the background. Others have reported this too. If I disconnect the external mic from the adapter, or shortcircuit the external microphone, the noise gets less but does not disappear. Therefore, it is not background noise from the room, but it comes from the computer itself. However, if you talk directly in the microphone of the headset, the noise level is acceptable for VOIP. The headset of my mobile phone Nokia C1 mobile comes wwith a 4 contacts combi 3,5mm jack plug. However, this one works (ear and mic) with the AO722 only if not inserted fully. Possibly the wiring of this headset jack is different. I cannot find detailed specs of the AO722, and don't know whether the audio 'output' was actually designed as a combi input/output. I have seen that at least one other AO model has a combi connector only. In any case, I do not believe that connecting your headset in this way will harm your computer. I would still appreciate a software solution. This must be possible, because the proper microphone input connector works under MS Windows.

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• How to create a screencast?

  - by Riccardo Murri

  How can I create a screencast on Ubuntu? What applications are available? The app I'm looking for has ideally all of these features: Can record in a format that can be played back easily on any platform and/or accepted by youtube or another popular video site Can record just a window (instead of the whole screen), possibly selecting it with a mouse click Can start recording after a configurable delay (e.g., I launch the app and have time to do arrangements to my desktop/window before actual recording starts)

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• How to record streaming camera video and auto-erase old data before drive fills up?

  - by nLinked

  I'm interested in making my own home CCTV system using Ubuntu. I want to get network cameras similar to this: http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&item=260745150596 I want a way to dump or record the live stream to a large hard drive, but have Ubuntu automatically delete the oldest parts of the video while the drive fills, so it can continue recording new data continuously. How can this auto-delete while recording be accomplished? I've searched and searched.

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• How to create a screencast?

  - by Riccardo Murri

  How can I create a screencast on Ubuntu? What applications are available? The app I'm looking for has ideally all of these features: Can record in a format that can be played back easily on any platform and/or accepted by youtube or another popular video site Can record just a window (instead of the whole screen), possibly selecting it with a mouse click Can start recording after a configurable delay (e.g., I launch the app and have time to do arrangements to my desktop/window before actual recording starts)

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• How do I search the MediaStore for a specific directory instead of entire external storage?

  - by Nick Lopez

  In my app I have an option that allows users to browse for audio files on their phone to add to the app. I am having trouble however with creating a faster way of processing the query code. Currently it searches the entire external storage and causes the phone to prompt a force close/wait warning. I would like to take the code I have posted below and make it more efficient by either searching in a specific folder on the phone or by streamlining the process to make the file search quicker. I am not sure how to do this however. Thanks! public class BrowseActivity extends DashboardActivity implements OnClickListener, OnItemClickListener { private List<Sound> soundsInDevice = new ArrayList<Sound>(); private List<Sound> checkedList; private ListView browsedList; private BrowserSoundAdapter adapter; private long categoryId; private Category category; private String currentCategoryName; private String description; // private Category newCategory ; private Button doneButton; @Override protected void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); requestWindowFeature(Window.FEATURE_NO_TITLE); setContentView(R.layout.activity_browse); checkedList = new ArrayList<Sound>(); browsedList = (ListView) findViewById(android.R.id.list); doneButton = (Button) findViewById(R.id.doneButton); soundsInDevice = getMediaSounds(); if (soundsInDevice.size() > 0) { adapter = new BrowserSoundAdapter(this, R.id.browseSoundName, soundsInDevice); } else { Toast.makeText(getApplicationContext(), getString(R.string.no_sounds_available), Toast.LENGTH_SHORT) .show(); } browsedList.setAdapter(adapter); browsedList.setOnItemClickListener(this); doneButton.setOnClickListener(this); } private List<Sound> getMediaSounds() { List<Sound> mediaSoundList = new ArrayList<Sound>(); ContentResolver cr = getContentResolver(); String[] projection = {MediaStore.Audio.Media._ID, MediaStore.Audio.Media.DISPLAY_NAME, MediaStore.Audio.Media.TITLE, MediaStore.Audio.Media.DATA, MediaStore.Audio.Media.DURATION}; final Uri uri = MediaStore.Audio.Media.EXTERNAL_CONTENT_URI; Log.v("MediaStore.Audio.Media.EXTERNAL_CONTENT_URI", "" + uri); final Cursor cursor = cr.query(uri, projection, null, null, null); int n = cursor.getCount(); Log.v("count", "" + n); if (cursor.moveToFirst()) { do { String soundName = cursor .getString(cursor .getColumnIndexOrThrow(MediaStore.Audio.Media.DISPLAY_NAME)); Log.v("soundName", "" + soundName); String title = cursor .getString(cursor .getColumnIndexOrThrow(MediaStore.Audio.Media.TITLE)); Log.v("title", "" + title); String path = cursor.getString(cursor .getColumnIndexOrThrow(MediaStore.Audio.Media.DATA)); Log.v("path", "" + path); Sound browsedSound = new Sound(title, path, false, false, false, false, 0); Log.v("browsedSound", "" + browsedSound); mediaSoundList.add(browsedSound); Log.v("mediaSoundList", "" + mediaSoundList.toString()); } while (cursor.moveToNext()); } return mediaSoundList; } public class BrowserSoundAdapter extends ArrayAdapter<Sound> { public BrowserSoundAdapter(Context context, int textViewResourceId, List<Sound> objects) { super(context, textViewResourceId, objects); } @Override public View getView(final int position, View convertView, ViewGroup parent) { ViewHolder viewHolder; View view = convertView; LayoutInflater inflater = getLayoutInflater(); if (view == null) { view = inflater.inflate(R.layout.list_item_browse, null); viewHolder = new ViewHolder(); viewHolder.soundNameTextView = (TextView) view .findViewById(R.id.browseSoundName); viewHolder.pathTextView = (TextView) view .findViewById(R.id.browseSoundPath); viewHolder.checkToAddSound = (CheckBox) view .findViewById(R.id.browse_checkbox); view.setTag(viewHolder); } else { viewHolder = (ViewHolder) view.getTag(); } final Sound sound = soundsInDevice.get(position); if (sound.isCheckedState()) { viewHolder.checkToAddSound.setChecked(true); } else { viewHolder.checkToAddSound.setChecked(false); } viewHolder.soundNameTextView.setText(sound.getName()); viewHolder.pathTextView.setText(sound.getUri()); viewHolder.checkToAddSound .setOnClickListener(new OnClickListener() { @Override public void onClick(View v) { CheckBox cb = (CheckBox) v .findViewById(R.id.browse_checkbox); boolean checked = cb.isChecked(); boolean newValue = checked; updateView(position, newValue); doneButtonStatus(checkedList.size()); } }); return view; } } // Adapter view holder class private class ViewHolder { private TextView soundNameTextView; private TextView pathTextView; private CheckBox checkToAddSound; } // done button On Click @Override public void onClick(View view) { boolean status = getIntent().getBooleanExtra("FromAddCat", false); Log.v("for add category","enters in if"); if(status){ Log.v("for add category","enters in if1"); currentCategoryName = getIntent().getStringExtra("categoryName"); description = getIntent().getStringExtra("description"); boolean existCategory = SQLiteHelper.getCategoryStatus(currentCategoryName); if (!existCategory) { category = new Category(currentCategoryName, description, false); category.insert(); category.update(); Log.v("for add category","enters in if2"); } }else{ categoryId = getIntent().getLongExtra("categoryId",-1); category = SQLiteHelper.getCategory(categoryId); } for (Sound checkedsound : checkedList) { checkedsound.setCheckedState(false); checkedsound.insert(); category.getSounds().add(checkedsound); final Intent intent = new Intent(this, CategoriesActivity.class); finish(); startActivity(intent); } } @Override public void onItemClick(AdapterView<?> arg0, View view, int position, long arg3) { boolean checked = true; boolean newValue = false; CheckBox cb = (CheckBox) view.findViewById(R.id.browse_checkbox); if (cb.isChecked()) { cb.setChecked(!checked); newValue = !checked; } else { cb.setChecked(checked); newValue = checked; } updateView(position, newValue); doneButtonStatus(checkedList.size()); } private void doneButtonStatus(int size) { if (size > 0) { doneButton.setEnabled(true); doneButton.setBackgroundResource(R.drawable.done_button_drawable); } else { doneButton.setEnabled(false); doneButton.setBackgroundResource(R.drawable.done_btn_disabled); } } private void updateView(int index, boolean newValue) { System.out.println(newValue); Sound sound = soundsInDevice.get(index); if (newValue == true) { checkedList.add(sound); sound.setCheckedState(newValue); } else { checkedList.remove(sound); sound.setCheckedState(newValue); } } }

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• How would the conversion of a custom CMS using a text-file-based database to Drupal be tackled?

  - by James Morris

  Just today I've started using Drupal for a site I'm designing/developing. For my own site http://jwm-art.net I wrote a user-unfriendly CMS in PHP. My brief experience with Drupal is making me want to convert from the CMS I wrote. A CMS whose sole method (other than comments) of automatically publishing content is by logging in via SSH and using NANO to create a plain text file in a format like so*: head<<END_HEAD title = Audio keywords= open,source,audio,sequencing,sampling,synthesis descr = Music, noise, and audio, created by James W. Morris. parent = home END_HEAD main<<END_MAIN text<<END_TEXT Digital music, noise, and audio made exclusively with @=xlink=http://www.linux-sound.org@:Linux Audio Software@_=@. END_TEXT image=gfb@--@;Accompanying image for penonpaper-c@right ilink=audio_2008 br= ilink=audio_2007 br= ilink=audio_2006 END_MAIN info=text<<END_TEXT I've been making PC based music since the early nineties - fortunately most of it only exists as tape recordings. END_TEXT ( http://jwm-art.net/dark.php?p=audio - There's just over 400 pages on there. ) *The jounal-entry form which takes some of the work out of it, has mysteriously broken. And it still required SSH access to copy the file to the main dat dir and to check I had actually remembered the format correctly and the code hadn't mis-formatted anything (which it always does). I don't want to drop all the old content (just some), but how much work would be involved in converting it, factoring into account I've been using Drupal for a day, have not written any PHP for a couple of years, and have zero knowledge of SQL? How might a team of developers tackle this? How do-able is it for one guy in his spare time?

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• Handling selection and deselection of objects in a user interface

  - by Dennis

  I'm writing a small audio application (in Silverlight, but that's not really relevant I suppose), and I'm a struggling with a problem I've faced before and never solved properly. This time I want to get it right. In the application there is one Arrangement control, which contains several Track controls, and every Track can contain AudioObject controls (these are all custom user controls). The user needs to be able to select audio objects, and when these objects are selected they are rendered differently. I can make this happen by hooking into the MouseDown event of the AudioObject control and setting state accordingly. So far so good, but when an audio object is selected, all other audio objects need to be deselected (well, unless the user holds the shift key). Audio objects don't know about other audio objects though, so they have no way to tell the rest to deselect themselves. Now if I would approach this problem like I did the last time I would pass a reference to the Arrangement control in the constructor of the AudioObject control and give the Arrangement control a DeselectAll() method or something like that, which would tell all Track controls to deselect all their AudioObject controls. This feels wrong, and if I apply this strategy to similar issues I'm afraid I will soon end up with every object having a reference to every other object, creating one big tightly coupled mess. It feels like opening the floodgates for poorly designed code. Is there a better way to handle this?

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• Advanced Localization with Omission of Arguments in Xcode

  - by coneybeare

  I have this formatted string that I am having a translator work on. ENGLISH "Check out the %1$@ %2$@ in %3$@: %4$@" = "Check out the %1$@ %2$@ in %3$@: %4$@" GERMAN TRANSLATION "Check out the %1$@ %2$@ in %3$@: %4$@" = "Hör Dir mal %2$@ in %3$@ an: %4$@"; These are passed to a [NSString stringWithFormat:] call: ////////////////////////////////////// // Share Over Twitter NSString *frmt = NSLocalizedString(@"Check out the %1$@ %2$@ in %3$@: %4$@", @"The default tweet for sharing sounds. Use %1$@ for where the sound type (Sound, mix, playlist) will be, %2$@ for where the audio name will be, %3$@ for the app name, and %3$@ for where the sound link will be."); NSString *urlString = [NSString stringWithFormat:@"sounds/%@", SoundSoundID(audio)]; NSString *url = ([audio audioType] == UAAudioTypeSound ? UrlFor(urlString) : APP_SHORTLINK); NSString *msg = [NSString stringWithFormat: frmt, [[Audio titleForAudioType:[audio audioType]] lowercaseString], [NSString stringWithFormat:@"\"%@\"", AudioName(audio)], APP_NAME, url]; NSString *applink = [NSString stringWithFormat:@" %@", APP_SHORTLINK]; if (msg.length <= (140 - applink.length)) { msg = [msg stringByAppendingString:applink]; } returnString = msg; With the desired and actual outcome of: ENGLISH desired: "Check out the sound "This Sound Name" in My App Name: link_to_sound link_to_app" actual: "Check out the sound "This Sound Name" in My App Name: link_to_sound link_to_app" GERMAN desired: "Hör Dir mal "This Sound Name" in My App Name an: link_to_sound link_to_app" actual: "Hör Dir mal sound in "This Sound Name" an: My App Name link_to_app" THE PROBLEM The problem is that I was under the assumption that by using numbered variable in the NSLocalizedString, I could do things like this, where the %1$@ variable is completely omitted. If you notice, the German translation of the format string does not use the first argument (%1$@) at all but it ("sound") still appears in the output string. What am I doing wrong?

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• Android - Looping Activity to Repeat MediaPlayer

  - by Austin Anderson

  I'm trying to create a soundboard for longer audio files and can't figure out how to stop an audio file and start it again without closing the activity. Let's say each audio file is one minute long. If I play the first audio file for 20 seconds and start the next audio file, the first stops playing and the second starts playing. However, if I click the first audio file again, the second stops playing and the first does not. I need help. This is driving me insane. bAudio1 = (ImageButton) findViewById(R.id.bAudio1); bAudio2 = (ImageButton) findViewById(R.id.bAudio2); mpAudio1 = MediaPlayer.create(this, R.raw.audio1); mpAudio2 = MediaPlayer.create(this, R.raw.audio2); bAudio1.setOnClickListener(new View.OnClickListener() { public void onClick(View v) { if(mpAudio1.isPlaying()) { mpAudio1.stop(); } else { if(mpAudio2.isPlaying()) { mpAudio2.stop(); } mpAudio1.start(); } } }); bAudio2.setOnClickListener(new View.OnClickListener() { public void onClick(View v) { if(mpAudio2.isPlaying()) { mpAudio2.stop(); } else { if(mpAudio1.isPlaying()) { mpAudio1.stop(); } mpAudio2.start(); } } }); Thanks.

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• gstreamer pulseaudio echo cancellation

  - by user3618055

  I'm implementing a voip application using gstreamer, i use the example of the rtp in the plugin-good! i want to implement echo cancellation, i couldn't use the speex echo canceller with gstreamer because the input and the output are not in the same process. So, i want to use pulse audio to make echo cancellation? can any one help me how to deal with? the sender voice is pipeline = gst_pipeline_new (NULL); g_assert (pipeline); /* the audio capture and format conversion */ audiosrc = gst_element_factory_make (pulsesrc, "audiosrc"); g_assert (audiosrc); audioconv = gst_element_factory_make ("audioconvert", "audioconv"); g_assert (audioconv); audiores = gst_element_factory_make ("audioresample", "audiores"); g_assert (audiores); /* the encoding and payloading */ audioenc = gst_element_factory_make (AUDIO_ENC, "audioenc"); g_assert (audioenc); audiopay = gst_element_factory_make (AUDIO_PAY, "audiopay"); g_assert (audiopay); /* add capture and payloading to the pipeline and link */ gst_bin_add_many (GST_BIN (pipeline), audiosrc, audioconv, audiores, audioenc, audiopay, NULL); if (!gst_element_link_many (audiosrc, audioconv, audiores, audioenc, audiopay, NULL)) { g_error ("Failed to link audiosrc, audioconv, audioresample, " "audio encoder and audio payloader"); } and the receiver is : gst_bin_add_many (GST_BIN (pipeline), rtpsrc, rtcpsrc, rtcpsink, NULL); /* the depayloading and decoding */ audiodepay = gst_element_factory_make (AUDIO_DEPAY, "audiodepay"); g_assert (audiodepay); audiodec = gst_element_factory_make (AUDIO_DEC, "audiodec"); g_assert (audiodec); /* the audio playback and format conversion */ audioconv = gst_element_factory_make ("audioconvert", "audioconv"); g_assert (audioconv); audiores = gst_element_factory_make ("audioresample", "audiores"); g_assert (audiores); audiosink = gst_element_factory_make (pulsesink, "audiosink"); g_assert (audiosink); /* add depayloading and playback to the pipeline and link */ gst_bin_add_many (GST_BIN (pipeline), audiodepay, audiodec, audioconv, audiores, audiosink, NULL); res = gst_element_link_many (audiodepay, audiodec, audioconv, audiores, audiosink, NULL); g_assert (res == TRUE); i tried to change gstreamer proprietes to pulseaudio server in input and output and i used "pactl load-module module-echo-cancel aec_method=adrian" but i still listen to echo!! any one could help please thanks!!

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• No Sound in 12.04 on Acer TimelineX 3830TG

  - by Fawkes5

  When i first installed ubuntu 12.04 everything worked fine. All of a sudden my sound has stopped working!! This has happened before [like 2 days ago], however the next day it started working again. That may happen again...[I've already rebooted 5 times] My Sound Card is a HDA intel PCH My Sound Chip is Intel CougarPoint HDMI My machine is an Acer TimelineX 3830TG, heres my lcpci output 00:00.0 Host bridge: Intel Corporation 2nd Generation Core Processor Family DRAM Controller (rev 09) 00:01.0 PCI bridge: Intel Corporation Xeon E3-1200/2nd Generation Core Processor Family PCI Express Root Port (rev 09) 00:02.0 VGA compatible controller: Intel Corporation 2nd Generation Core Processor Family Integrated Graphics Controller (rev 09) 00:16.0 Communication controller: Intel Corporation 6 Series/C200 Series Chipset Family MEI Controller #1 (rev 04) 00:1a.0 USB controller: Intel Corporation 6 Series/C200 Series Chipset Family USB Enhanced Host Controller #2 (rev 04) 00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 04) 00:1c.0 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 1 (rev b4) 00:1c.1 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 2 (rev b4) 00:1c.2 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 3 (rev b4) 00:1c.3 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 4 (rev b4) 00:1d.0 USB controller: Intel Corporation 6 Series/C200 Series Chipset Family USB Enhanced Host Controller #1 (rev 04) 00:1f.0 ISA bridge: Intel Corporation HM65 Express Chipset Family LPC Controller (rev 04) 00:1f.2 SATA controller: Intel Corporation 6 Series/C200 Series Chipset Family 6 port SATA AHCI Controller (rev 04) 00:1f.3 SMBus: Intel Corporation 6 Series/C200 Series Chipset Family SMBus Controller (rev 04) 01:00.0 VGA compatible controller: NVIDIA Corporation GF108 [GeForce GT 540M] (rev ff) 02:00.0 Ethernet controller: Atheros Communications Inc. AR8151 v2.0 Gigabit Ethernet (rev c0) 03:00.0 Network controller: Atheros Communications Inc. AR9287 Wireless Network Adapter (PCI-Express) (rev 01) 04:00.0 Unassigned class [ff00]: Realtek Semiconductor Co., Ltd. RTS5116 PCI Express Card Reader (rev 01) 05:00.0 USB controller: NEC Corporation uPD720200 USB 3.0 Host Controller (rev 04) I'm running a dual boot with Windows 7, sound always works there. I've tried loading earlier kernels [3.2.0 ] but that hasn't helped. I've tried playing with alsamixer, unmuting and raising volume, doesn't work I've tried playing with pauvcontrol, doesnt work. Althought, strangely, when i play a song, under output i can see a bar moving implying that something is working, just the sound isn't getting through my speakers. Thus it might be a hardware problem. I've tried unistalling and reinstalling pulsaudio annd alsamixer, doesn't work [maybe made it worse...]. I've heard reverting back to kernel 3.0... works, but i'm reluctant to try that[i dont know much about kernels]. Thank you. Please tell me what other information you need. EDIT: My audio has started working again after another reboot. However this time im not connected to my external audio on startup. I will see if this is the issue next time i'm at home Edit: Connecting to external audio is not the problem.

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• Record Screen Activity with CamStudio

  - by Asian Angel

  Sometimes a visual demonstration works much better than a list of instructions. If you need to make a demo video for family and/or friends then you might want to have a look at CamStudio. Using CamStudio To get properly set up you will need to install two different files (the main program followed by the codec). Once that is done you are ready to get started. When you start the program you will see a surprisingly small window. Notice the highlighted Record to text…it serves as a visual indicator for the video type selected for recording. Before you start creating a video it would be a good idea to look through some of the settings. The first one to look at is the region or area that you want to record. Next you will want to look through the video options since these will affect the quality and final size of your video files. The default setting for quality is 70…adjust that to the level that best suits your needs. Note: For our example we maxed out the various video settings for best quality. On our system Microsoft Video 1 was listed as the default compressor but as you can see there were other options available. You can configure the settings for the compressor you want to use if desired. Keep in mind that each compressor will have unique settings of their own, so if you change it, be certain to go back and check. We decided to use the CamStudio Lossless Codec for our example (it gave the best results while trying the software). Going back to the main window you can toggle back and forth between .avi and .swf output using the last button. Once you are satisfied with the settings click on the red record button to start. If you need to pause while recording or stop recording click on the system tray icon and select the appropriate command. When you are finished recording, you will be presented with the save file window. Browse for the desired save location and name your new file. Once you have saved the file the movie player window will automatically open so that you view your new video. Our sample video shown here is at 50% of original size so may look slightly “gritty”. The detail was much better at 100%. If you decide to record and save as .swf the process will be identical to recording in .avi format until the movie player window opens. At that time the conversion process from .avi to .swf will begin. When complete you will have a new flash video and html file that goes with it. Depending on which browser you have set as default, you may run into a small problem when the preview for your new .swf file tries to open. There is a small bug in the generated html file. You can use this work-around or… Just open the .swf file directly in your favorite browser. Conclusion CamStudio may not produce the highest quality videos, but it’s free and does a very nice job nonetheless. If you are working on a tight budget or only need to make an occasional video then CamStudio is a very sensible choice. Links Download CamStudio Stable Version & CamStudio Codec *Download links are approximately half-way down the page. Download CamStudio Stable Version & CamStudio Codec at SourceForge *Beta version also available here. Similar Articles Productive Geek Tips Get the Classic Style Network Activity Indicator Back in Windows 7How To Copy a DVD with VLC 1.0ALLCapture 3.0 [Review]Listen and Record Over 12,000 Online Radio Stations with RadioSureGeek Reviews: Play And Record Internet Radio With Screamer Radio TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips Xobni Plus for Outlook All My Movies 5.9 CloudBerry Online Backup 1.5 for Windows Home Server Snagit 10 TimeToMeet is a Simple Online Meeting Planning Tool Easily Create More Bookmark Toolbars in Firefox Filevo is a Cool File Hosting & Sharing Site Get a free copy of WinUtilities Pro 2010 World Cup Schedule Boot Snooze – Reboot and then Standby or Hibernate

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• Converting .wav (CCITT A-Law format) to .mp3 using LAME

  - by iar

  I would like to convert wav files to mp3 using the lame encoder (lame.exe). The wav files are recorded along the following specifications: Bit Rate: 64kbps Audio sample size: 8 bit Channels: 1 (mono) Audio sample rate: 8 kHz Audio format: CCITT A-Law If I try to convert such a wav file using lame, I get the following error message: Unsupported data format: 0x0006 Could anyone provide me with a command line string using lame.exe that will enable me to convert these kind of wav files?

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• How to get headphones and speakers working at the same time?

  - by Borek

  I have a Gigabyte P55-UD3 motherboard which comes with a Realtek sound card and audio header for the front panel. My headset is permanently connected to the front panel (mic in, headphones in) and I also have speakers connected at the rear of the computer. What I want to achieve is that both the headphones and the speakers work at the same time. On my old PC with Realtek, I was able to do this by clicking "Device advanced settings" in the Realtek HD Audio Manager and then clicking "Make front and rear output devices playback two different audio streams simultaneously" and then doing some reassigning of front/rear settings in the Realtek Audio Manager (can't remember the details). But that doesn't seem to be working now - either the speakers play sounds or the headphones, not both at the same time. To pose the question differently, Windows plays the sounds on the "default device". I'd like to make both headphones and the speakers kind of like the default devices.

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• Portable USB drives hidden pertition - New request

  - by ZXC

  This question was made by Francesco on Jul 29 '11 at 17:14. and the replies were not satisfactory due they not point to an important problem that´s: Why could anyone want to make certain data only accesible for a program but not to the users?. For example: If I want to do a safe distribution of original music for demostration purposes I will need several requisites: 1) The music should be heard using a simple procedure like selecting the name of each song on a playlist of a mediaplayer. 2) The portable media, ussually a portable USB drive, must hide for complete and should make unaccesible the files that contain the audio data to anything but the mediaplayer, that must be in the first partition, the one that is visible. 3) Considering that´s impossible to really hide files in a non-hidden partition, a second hidden partition should be created in the USB drive and the audio data will be stored there. 4) The trick is to read the audio data files stored in the hidden partition with a mediaplayer stored in the visible partition, the media player also should be a complete standalone program and independent from any library of the operating system except of the OS audio system. 5) The hidden partition should have a copy protection scheme that could impede to do copies of the data or create working ISO images of it. I know that this description could not be technically accurate but it has a complete logic from the needs of a music producer against the problem of piracy. The philosophy that surrounds the concept is to transform a virtual object like a digital string of audio in a solid object like the analog vinyl discs are.

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• How to extract a Vorbis stream from a WAVE file?

  - by H.B.

  I would like to move the Vorbis stream into an ogg container but ffmpeg does not seem to recognize the stream. Even though MPlayer gives this output upon playback: Opening audio decoder: [acm] Win32/ACM decoders Loading codec DLL: 'vorbis.acm' Loaded DLL driver vorbis.acm at 10000000 Warning! ACM codec reports srcsize=0 AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 16000-176400) Selected audio codec: [vorbisacm] afm: acm (OggVorbis ACM) ffmpeg: ffmpeg -i Source.wav -acodec copy Target.ogg Input #0, wav, from 'Source.wav': Duration: 00:02:15.17, bitrate: 128 kb/s Stream #0.0: Audio: qg[0][0] / 0x6771, 44100 Hz, 2 channels, 128 kb/s [ogg @ 00000000003096C0] Unsupported codec id in stream 0 Output #0, ogg, to 'Target.ogg': Metadata: encoder : Lavf53.6.0 Stream #0.0: Audio: qg[0][0] / 0x6771, 44100 Hz, 2 channels, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 Could not write header for output file #0 (incorrect codec parameters ?) Of course this does not necessarily need to be done via ffmpeg, any method that is workable would be fine... I have cut down one of the files to 512KB: sample.wav (Changed two chunk size fields in the wave header to account for this, the embedded stream is cut "without notice")

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• What is the fastest and best way to convert an rmvb video to mp4/mkv without losing any quality?

  - by Eric Leung

  the file will be played in a popbox3d. my old method was to convert the video using vidcoder (an offshoot of handbrake) using normal settings, but i've recently confirmed that this significantly reduces video and audio quality. i bumped up the conversion quality to 'high profile' and this produced a higher quality video but raised the conversion time to about twice the video length (95 minutes to convert a 45 minute video) on a core2duo laptop. this is less than ideal when a large number of videos need to be converted. i have tried a direct remuxing using mkv toolnix but this produced a video that refused to display video on the popbox3d, which is consistent with the reported: [quote=other old thread] it is possible to put RealMedia A/V in MKV container (used MKVtoolnix) - however, it is awkward to play later. RV40 is only suspected to be based on H.264 - simplify, is not consistent with MPEG-4 AVC specification. [/quote] i have read that ... [quote=from old thread] Under normal circumstances, [ffmpeg] should convert the video to .video.mp4 and the audio to (.wav then to) .audio.mp4, then mux the video and audio into a new .mp4 file and delete the temporary video-only and audio-only files.[/quote] and i am currently attempting to discover how this is done. help? PS: i download a lot of series from asia and for some strange reason, rmvb is a really popular format over there. sometimes, it's the only format that's available. unfortunately, it's a format that is incompatible with the popbox3d, so i have to convert the files before i can watch them on my tv.

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• Merging cuesheet chapter halves into single track for an audiobook

  - by TheSavo

  I have an audiobook that I have ripped and I need some help constructing chapters. I have already made some cue sheets TITLE "Bookname" PERFORMER "the Author" FILE "File1.FLAC" wave ; 23971906.667 milliseconds TRACK 01 AUDIO TITLE "_Intro" INDEX 01 00:00:00 TRACK 02 AUDIO TITLE "CH 01" INDEX 01 24:15:50 TRACK 03 AUDIO TITLE "CH 02" INDEX 01 66:21:00 TRACK 04 AUDIO TITLE "CH 03" INDEX 01 87:05:00 The audio book is in two files. The chapter at the end of the first file is continued in the second file. However, the second file restates: The publisher Book Title List item Blah blah blah I would like to merge the two 'halves' of the chapter in one seamless track. The only way I can think to do this would be be: Bulk cut down the tracks. Drop the junk info into junk track Continue the track listings as normal Take the two "halves" of the target chapter and build a separate cue sheet for it. I know there has to be an easier way. I am ok with making the 'junk' info a 'gap' or something. These are are FLAC files that will be converted to MP3 for my phone and other potable devices. I have read the primers on cue sheets, but I am just not getting it.

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• how to use time out in mplayer?

  - by manoj

  I am trying to save audio using mplayer from a live http stream. saving audio is successful. If there is no live stream playing it does not exit automatically. Is there any way to set timeout if there is no live stream? code : mplayer -i -t 00:00:10 -acodec libmp3lame -ab 24 -ar 8000 audio.mp3 Thanks in advance.

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• How can I fix Problems with interlaced video jerking/flicking when playedback on DVD players? (Mixin

  - by Simon P Stevens

  I'm trying to make a DVD and the final DVD jerks when played on standalone DVD players. It seems to play fine on PCs. I think the problem may be to do with interlacing settings when rendering the final output, but I'll outline the whole editing process I have followed in case I've made a mistake somewhere else. Most of the footage comes from a sony handy cam (one of those mini DVD ones) so isn't great quality. It was set to "high quality" (haha) and 16:9 aspect ratio when it was recorded. I copy the files directly from the mini DVDs onto the hard drive and import them into Cinelerra. In Cinelerra I set the format to 25fps, 720x576, RGBA-8bit, 16:9, interlaced bottom fields first. When I've finished the editing, I add a Fields to frames effect (set to bottom first) to each video track. I render to audio and video separately: Audio: AC3, 128kbps Video: YUV4MPEG steam, video pipe settings: ffmpeg -f yuv4mpegpipe -i - -y -target dvd -flags +ilme+ildct mpeg2video % Cinelerra often crashes during the rendering, so I set it to generate a new video file at each label, and combine them using cat when I've got a sucesful render of each one. Once I've combined them, I use mencoder to re-index them: mencoder -forceidx -oac copy -ovc copy merged.m2v -o mergedReIndexed.m2v I combine the audio and video files using ffmpeg: ffmpeg -i AudioFile.ac3 -i VideoFile.m2v -target dvd -flags +ilme+ildct FinalMovie.mpg Then I build the menus with spumux and I create the DVD file system with dvdauthor, and finally I write it do a dvd-r like this: nice -n -20 growisofs -dvd-compat -speed=2 -Z /dev/dvd -dvd-video -V VIDEO ./ && eject /dev/dvd Originally, when I did it the DVD flickered badly, so as suggested in a guide I added the fields to frames effect in cinelerra. Now it doesn't "flicker", but has become "jerky" when there is lots of motion, particularly when the camera is moving, so the whole background moves. This is what I've tried so far: Removed "mpeg2video" from cinelerra video render pipe. Removed +ilme from render pipe. Removed +ildct from render pipe. Removed +ilme from render audio/video rejoin command. Removed +ildct from render audio/video rejoin command. Added -alt to render pipe. Added -alt to render audio/video rejoin command. Tried with and without the frames to fields effect in Cinelerra. and various combinations of the above. I've also tried this: change the Cinelerra fps to 50, use fields to frames (instead of frames to fields), render to an intermediate QTforlinux jpeg video stream, re-importing that back into Cinelerra, adding a frames to fields effect and then rendering that output as normal (@25fps), and I still have the same problem. Has anyone experienced this "jerking" playback before? Can anyone give any suggestions on how to fix it? (Like I say, it plays back fine on a PC, but not on any of the standalone players I've tried)

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• Why does my computer run slowly and freeze sometimes?

  - by Brae

  EDIT I disabled sound in the BIOS, rebooted and it hung. I removed the (previously) faulty HDD, rebooted and it hung. I have managed to get my Realtek audio manager open again after its mysterious disappearance. Subsequently my microphone is now working again, to fix it I had to uninstall audio drivers, disable audio in BIOS, install audio drivers, enable audio in BIOS. Access via network (with faulty HDD in) seems to not be triggering hangs at the moment. I think with the sound problem fixed it might play a little nicer, but I think its still going to hang. If it does, then I'm fairly sure its been narrowed down to the mobo. EDIT Pretty convinced my motherboard is the culprit, because nothing else seems to have any obvious problems (bar the hard drive, which the PC still hangs without it being plugged in) So thank you all for helping, once I get more rep ill up a few of the answers. My PC is doing some weird things... Sometimes when I open up a program, lets say Adobe Photoshop, it will load everything normally, nice and quick no problems and I can use it fine. Other times its a little odd, and it loads the program as if it's only using half of the CPU. It's pretty obvious when it does it, normally the loading screen skims past, but when it does this weird load you see it slowly tick though each thing, and the program itself becomes incredibly slow. Even Google Chrome does it sometimes. Yet when I exit and reopen the program without doing anything else, it typically opens fine without lagging. I think this problem is probably a symptom of something bigger, because of other problems I'm having. Random hangs; no matter what I have open or what I'm doing. My PC will sort of freeze up for a few seconds. If music is playing it will either loop the last second or two, or it will buzz. This only lasts for a few seconds then it returns to normal without having to restart. During this time all programs lock up and freeze, and the mouse and keyboard are useless. I am also having a weird issue with my audio jacks, without touching my PC at all, sometimes I will see a popup saying that I have unplugged something or plugged something in, neither of which has actually happened. Pretty sure this is cause by the motherboard. I recently had a 'Pink Screen of Death' (yes pink) which pointed to my audio drivers. The lockups seem to happen with some consistency when someone is accessing my shared files via my home LAN. Which leads me to believe either one of my hard drives is acting up or more likely the controller. One of my drives had a bit of a crash before, I used Spinrite and managed to recover my stuff and now the drive appears to be working okay. This is possibly adding to the problems. My best guess is something has gone wrong with my motherboard, possibly a power issue or a chip has died, I really don't know. So what I would like to know is: Have I have missed some obvious diagnostic to help figure out what it is, or should I just bite the bullet and assume its my motherboard acting up and buy a new system? dxdiag[64-bit] - http://pastebin.com/G30kb2TL PSOD (minidump) - http://pastebin.com/aZsv0H56 HWiNFO64 (system info / specs) - http://pastebin.com/X6h3K8g6

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• Multiple Graphics Cards on Fedora 15 (Gnome 3)

  - by Michael

  I have the (delightful) misfortune of having 3 graphics cards. They are XFX Radeon 5750s. Each drives 2 monitors via dvi. I am having a really hard time getting these running on fedora 15 (gnome 3). So my setup is 3 columns of 2 monitors (the upper monitor is mounted upside down in each column to reduce the bezel between monitors). When the (graphical) login screen comes up all 6 have the blue stripey background that must be the default, but then when I login, things get interesting. In the xorg.conf below, you will see only 2 of the screens in the serverlayout while the other 4 are commented out. Logging in with only 2 of the screens active works well (and it even remembers that the top one is upside down, and should be considered above the lower, i am not sure where it stores this info, but i set it using the graphical "Displays" settings) However, as soon as I uncomment a third screen, or more, it gives me an error message when I login. It's one of those friendly, less helpful messages (Oh no! Something has gone wrong. A problem has occurred and the system can't recover. Please log out and try again). If i do not use an xorg.conf, then the "Displays" prefs pane shows only the two monitors on one of my graphics cards Thanks to anyone who can help me get going! (xorg.conf and then lspci below, and xorg log) xorg.conf Section "ServerLayout" Identifier "X.org Configured" Screen "Screen0" 0 0 Screen "Screen1" Below "Screen0" # Screen "Screen2" RightOf "Screen0" # Screen "Screen3" RightOf "Screen1" # Screen "Screen4" RightOf "Screen3" # Screen "Screen5" RightOf "Screen4" InputDevice "Mouse0" "CorePointer" InputDevice "Keyboard0" "CoreKeyboard" EndSection Section "Files" ModulePath "/usr/lib/xorg/modules" FontPath "catalogue:/etc/X11/fontpath.d" FontPath "built-ins" EndSection Section "Module" Load "record" Load "dri" Load "dbe" Load "extmod" Load "dri2" Load "glx" EndSection Section "InputDevice" Identifier "Keyboard0" Driver "kbd" EndSection Section "InputDevice" Identifier "Mouse0" Driver "mouse" Option "Protocol" "auto" Option "Device" "/dev/input/mice" Option "ZAxisMapping" "4 5 6 7" EndSection Section "Monitor" Identifier "Monitor0" VendorName "Monitor Vendor" ModelName "Monitor Model" EndSection Section "Monitor" Identifier "Monitor1" VendorName "Monitor Vendor" ModelName "Monitor Model" EndSection Section "Monitor" Identifier "Monitor2" VendorName "Monitor Vendor" ModelName "Monitor Model" EndSection Section "Monitor" Identifier "Monitor3" VendorName "Monitor Vendor" ModelName "Monitor Model" EndSection Section "Monitor" Identifier "Monitor4" VendorName "Monitor Vendor" ModelName "Monitor Model" EndSection Section "Monitor" Identifier "Monitor5" VendorName "Monitor Vendor" ModelName "Monitor Model" EndSection Section "Device" Identifier "Card0" Driver "radeon" BusID "PCI:4:0:0" EndSection Section "Device" Identifier "Card1" Driver "radeon" BusID "PCI:5:0:0" EndSection Section "Device" Identifier "Card2" Driver "radeon" BusID "PCI:6:0:0" EndSection Section "Screen" Identifier "Screen0" Device "Card0" Monitor "Monitor0" SubSection "Display" Viewport 0 0 Depth 24 EndSubSection EndSection Section "Screen" Identifier "Screen1" Device "Card0" Monitor "Monitor1" SubSection "Display" Viewport 0 0 Depth 24 EndSubSection EndSection Section "Screen" Identifier "Screen2" Device "Card1" Monitor "Monitor2" SubSection "Display" Viewport 0 0 Depth 24 EndSubSection EndSection Section "Screen" Identifier "Screen3" Device "Card1" Monitor "Monitor3" SubSection "Display" Viewport 0 0 Depth 24 EndSubSection EndSection Section "Screen" Identifier "Screen4" Device "Card2" Monitor "Monitor4" SubSection "Display" Viewport 0 0 Depth 24 EndSubSection EndSection Section "Screen" Identifier "Screen5" Device "Card2" Monitor "Monitor5" SubSection "Display" Viewport 0 0 Depth 24 EndSubSection EndSection lspci output follows [tgm@tgm ~]$ lspci 00:00.0 Host bridge: Intel Corporation 5520/5500/X58 I/O Hub to ESI Port (rev 13) 00:01.0 PCI bridge: Intel Corporation 5520/5500/X58 I/O Hub PCI Express Root Port 1 (rev 13) 00:03.0 PCI bridge: Intel Corporation 5520/5500/X58 I/O Hub PCI Express Root Port 3 (rev 13) 00:07.0 PCI bridge: Intel Corporation 5520/5500/X58 I/O Hub PCI Express Root Port 7 (rev 13) 00:14.0 PIC: Intel Corporation 5520/5500/X58 I/O Hub System Management Registers (rev 13) 00:14.1 PIC: Intel Corporation 5520/5500/X58 I/O Hub GPIO and Scratch Pad Registers (rev 13) 00:14.2 PIC: Intel Corporation 5520/5500/X58 I/O Hub Control Status and RAS Registers (rev 13) 00:14.3 PIC: Intel Corporation 5520/5500/X58 I/O Hub Throttle Registers (rev 13) 00:1a.0 USB Controller: Intel Corporation 82801JI (ICH10 Family) USB UHCI Controller #4 00:1a.1 USB Controller: Intel Corporation 82801JI (ICH10 Family) USB UHCI Controller #5 00:1a.2 USB Controller: Intel Corporation 82801JI (ICH10 Family) USB UHCI Controller #6 00:1a.7 USB Controller: Intel Corporation 82801JI (ICH10 Family) USB2 EHCI Controller #2 00:1b.0 Audio device: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller 00:1c.0 PCI bridge: Intel Corporation 82801JI (ICH10 Family) PCI Express Root Port 1 00:1c.1 PCI bridge: Intel Corporation 82801JI (ICH10 Family) PCI Express Port 2 00:1c.2 PCI bridge: Intel Corporation 82801JI (ICH10 Family) PCI Express Root Port 3 00:1d.0 USB Controller: Intel Corporation 82801JI (ICH10 Family) USB UHCI Controller #1 00:1d.1 USB Controller: Intel Corporation 82801JI (ICH10 Family) USB UHCI Controller #2 00:1d.2 USB Controller: Intel Corporation 82801JI (ICH10 Family) USB UHCI Controller #3 00:1d.7 USB Controller: Intel Corporation 82801JI (ICH10 Family) USB2 EHCI Controller #1 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev 90) 00:1f.0 ISA bridge: Intel Corporation 82801JIR (ICH10R) LPC Interface Controller 00:1f.2 IDE interface: Intel Corporation 82801JI (ICH10 Family) 4 port SATA IDE Controller #1 00:1f.3 SMBus: Intel Corporation 82801JI (ICH10 Family) SMBus Controller 00:1f.5 IDE interface: Intel Corporation 82801JI (ICH10 Family) 2 port SATA IDE Controller #2 02:00.0 PCI bridge: nVidia Corporation NF200 PCIe 2.0 switch for mainboards (rev a3) 03:00.0 PCI bridge: nVidia Corporation NF200 PCIe 2.0 switch for mainboards (rev a3) 03:02.0 PCI bridge: nVidia Corporation NF200 PCIe 2.0 switch for mainboards (rev a3) 04:00.0 VGA compatible controller: ATI Technologies Inc Juniper [Radeon HD 5750 Series] 04:00.1 Audio device: ATI Technologies Inc Juniper HDMI Audio [Radeon HD 5700 Series] 05:00.0 VGA compatible controller: ATI Technologies Inc Juniper [Radeon HD 5750 Series] 05:00.1 Audio device: ATI Technologies Inc Juniper HDMI Audio [Radeon HD 5700 Series] 06:00.0 VGA compatible controller: ATI Technologies Inc Juniper [Radeon HD 5750 Series] 06:00.1 Audio device: ATI Technologies Inc Juniper HDMI Audio [Radeon HD 5700 Series] 07:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 02) 08:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 02) xorg log (posted to fpaste due to how long it is. thanks to marcusw for the request) http://www.fpaste.org/r5ww/ xorg log with all 6 monitors enabled in xorg.conf (they all turn on and have blue, but then one gets the aforementioned user-friendly error message). http://www.fpaste.org/X63H/

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• How to tell if my sound card is listed in Device Manager?

  - by Bruhan

  The sound on my computer suddenly stopped working. When I check Sounds and Audio Devices in the Control Panel, I get "No Audio Device" with everything grayed out. When I check the Device Manager under "Sound, video and game controllers" I see the following list: Audio Codecs Legacy Audio Drivers Legacy Video Capture Devices Media Control Devices MPU-401 Compatible MIDI Device Standard Game Port Video Codecs None of these looks like my sound card. Of course, my sound "card" is not really a sound card, it's integrated with the nVidia-nForce motherboard. I'm running Windows XP. So is one of the above my sound device, or is the OS not detecting it? If the latter, how do I get it to detect it?

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• Windows 7 Desktop as Bluetooth Mixer

  - by J.R. Lillard

  I would like to stream all the audio from my Droid X2 to my desktop running Windows 7 via bluetooth. When I pair the two devices I can make this work but the audio plays through the internal speakers. I am actually wanting to listen to everything through a pair of bluetooth headphones. The end result would be that I could listen to music from my phone but easily pause it to listen to audio from the desktop instead. So far I have not had any luck. I thought about connecting the line out to the line in so I could monitor the line in from my headphones. My current setup is wiring the phone to the line in and then monitoring the line in through the headphones. I'm just wanting to eliminate the wire and go wireless the whole way. Is there something in Windows that prevents it from taking an inbound audio stream and sending it out another device?

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• How to get headphones and speakers working at the same time?

  - by Borek

  I have a Gigabyte P55-UD3 motherboard which comes with a Realtek sound card and audio header for the front panel. My headset is permanently connected to the front panel (mic in, headphones in) and I also have speakers connected at the rear of the computer. What I want to achieve is that both the headphones and the speakers work at the same time. On my old PC with Realtek, I was able to do this by clicking "Device advanced settings" in the Realtek HD Audio Manager and then clicking "Make front and rear output devices playback two different audio streams simultaneously" and then doing some reassigning of front/rear settings in the Realtek Audio Manager (can't remember the details). But that doesn't seem to be working now - either the speakers play sounds or the headphones, not both at the same time. To pose the question differently, Windows plays the sounds on the "default device". I'd like to make both headphones and the speakers kind of like the default devices.

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