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• How to make pulseaudio and ubuntu detect the same audio device as alsa driver

  - by Kiwy

  I use Ubuntu 14.04 x64 and I use gnome-shell on my laptop. I have a Bose companion 5 (which is basically a USB sound system) and a HDMI port, both does work perfectly when I just boot with the cable plugin. However, when my laptop go to sleep or get unplugged from those two outputs, if I plug back the device, I end up without any hardware detection (only the built-in speakers) from pulse and gnome-shell sound output selector while if I use alsamixer, the device look up and ready. gstreamer-properties allow me to select and test effectively any device but while alsa recognize any device on the run, pulse is not capable of handling things correctly, my question is then: How can I make pulse detect and use the same hardware as alsa, or how to remove completely and gracefully pulseaudio (meaning volume applet running in gnome shell) I don't mind if the project implies to recompile half gnome shell if it implies those audio outputs work all the time. Pulse does not list my soundcard when I use command pactl list cards while the module plug&play for sound card is loaded in pactl list modules. I really don't know what to do, the behavior seems pretty random.

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• 12.10 no audio via hdmi and video speeds up

  - by jackson

  I have a laptop with an ati radeon 4200, on 12.04 everything worked fine, since upgrading to 12.10 I cannot get sound over the hdmi. When I switch to hdmi audio the video speeds up to about 2x. I can use the speakers in my laptop and watch video via hdmi with no problems. Things I have tried: Various tutorials to install the AMD/ATI drivers, all of which resulted in low graphics mode. Checked that everything is properly set in alsamixer, the sound utility and - installed pavucontrol and checked everything in there. Verified the output from cat /proc/asound/cards looks normal When I initially upgraded there was a plethora of problems which I believe were due to the old proprietary driver still being used but not compatible, after a few hours trying to fix that I decided just to back up and do a fresh install which works great except for the above stated problem. Any help would be greatly appreciated!! Finally hopefully this hasn't already been answered, I have tried a few different searches on the boards and haven't come up with anything. $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC269VB Analog [ALC269VB Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 0/1 Subdevice #0: subdevice #0

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• No audio in Google Chrome

  - by Z9iT

  I started with Ubuntu 12.04 Minimal. Then installed only 3 utils sudo apt-get install xorg xinit google-chrome-stable alsa-base alsa-utils alsa-oss I have added google-chrome to .xinitrc file. Used sudo alsamixer to unmute everything using M. Also I am able to hear sound when I run this independently in a terminal sudo aplay /usr/share/sounds/alsa/Front_Center.wav However Google Chrome is not giving any sound output be it on youtube or the same file (/usr/share/sounds/alsa/Front_Center.wav) opened by browsing in chrome. UPDATE : the moment i install some Desktop (display) Manager like gnome or lxde and launch chrome then, the audio is perfect success. However if i kill the xsession and the desktop manager (lxde) AND then start with loading only the chrome (without DM) then again i loose the sound. This makes me wonder that there is something which is not allowing the sound to be loaded into chrome directly, but once the session like lxde loads, then it works flawless. I am thinking that i should rather ask, how to authorize google-chrome to use sound software? Miscellaneous : I am surprised to know that I cannot start google-chrome by sudo command (it asks to be a normal user) && that i cannot start alsamixer as a normal user (i must use sudo alsamixer ) May someone please help what i need to do so that google chrome speaks????

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• My sound stopped working today, how can I fix it?

  - by Oli

  This seems to be a problem with pulseaudio. I was logged in over VNC on my phone and started playing a video this caused X to crash (as sometimes happens). I restarted and suddenly the sound doesn't work. I have a Intel HDA/Realtek ALC889 00:1b.0 Audio device: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller alsamixer is detecting this just fine. PulseAudio doesn't detect this alsa device so is using auto_null as the default sink (logs below). When I properly kill PulseAudio (tell it not to auto-start) direct ALSA communication with the sound card works just fine. speaker-test, for example, works. So the hardware and ALSA layers are fine IMO. In the logs, it seems that the card might be "busy" but I really don't know how or why it would be now (and never before). Is there an ALSA lock file somewhere that it still there because of my crash? I just ran sudo fuser /dev/snd/* and saw this: oli@bert:~$ sudo fuser /dev/snd/* /dev/snd/controlC0: 1884 /dev/snd/pcmC0D0c: 1884m /dev/snd/timer: 1884 A look at the process list (ps aux | grep 1884) tells me process 1884 is arecord -c 1 -f S16_LE -r 8000 -t raw. No idea what this is or why it's running. When I try and kill arecord (as root), it just respawns and rebinds on the hardware. I'm in a very annoying situation where I don't know what is going on and don't know how to find out. I'm open to all suggestions to get this working again. Fire away. And here's what I get when I stop PA auto-loading, kill it and then start it with -vvvv. oli@bert:~$ pulseaudio -vvvvv I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted D: core-rtclock.c: Timer slack is set to 50 us. D: core-util.c: RealtimeKit worked. I: core-util.c: Successfully gained nice level -11. I: main.c: This is PulseAudio 0.9.21-63-gd3efa-dirty D: main.c: Compilation host: x86_64-pc-linux-gnu D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option D: main.c: Running on host: Linux x86_64 2.6.38-rc3 #1 SMP Tue Feb 1 10:53:04 GMT 2011 D: main.c: Found 8 CPUs. I: main.c: Page size is 4096 bytes D: main.c: Compiled with Valgrind support: no D: main.c: Running in valgrind mode: no D: main.c: Running in VM: no D: main.c: Optimised build: yes D: main.c: All asserts enabled. I: main.c: Machine ID is 8310740c4729ef474fe5ecec4bbf5a6b. I: main.c: Session ID is 8310740c4729ef474fe5ecec4bbf5a6b-1297338553.571075-1050119523. I: main.c: Using runtime directory /home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-runtime. I: main.c: Using state directory /home/oli/.pulse. I: main.c: Using modules directory /usr/lib/pulse-0.9.21/modules. I: main.c: Running in system mode: no I: main.c: Fresh high-resolution timers available! Enjoy ol' chap! I: cpu-x86.c: CPU flags: CMOV MMX SSE SSE2 SSE3 SSSE3 SSE4_1 SSE4_2 I: svolume_mmx.c: Initialising MMX optimized functions. I: remap_mmx.c: Initialising MMX optimized remappers. I: svolume_sse.c: Initialising SSE2 optimized functions. I: remap_sse.c: Initialising SSE2 optimized remappers. I: sconv_sse.c: Initialising SSE2 optimized conversions. D: memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472 D: database-tdb.c: Opened TDB database '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-device-volumes.tdb' I: module-device-restore.c: Sucessfully opened database file '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-device-volumes'. I: module.c: Loaded "module-device-restore" (index: #0; argument: ""). D: database-tdb.c: Opened TDB database '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-stream-volumes.tdb' I: module-stream-restore.c: Sucessfully opened database file '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-stream-volumes'. I: module.c: Loaded "module-stream-restore" (index: #1; argument: ""). D: database-tdb.c: Opened TDB database '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-card-database.tdb' I: module-card-restore.c: Sucessfully opened database file '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-card-database'. I: module.c: Loaded "module-card-restore" (index: #2; argument: ""). I: module.c: Loaded "module-augment-properties" (index: #3; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-udev-detect.so': success D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes D: module-udev-detect.c: /devices/pci0000:00/0000:00:1b.0/sound/card0 is busy: yes I: module-udev-detect.c: Found 1 cards. I: module.c: Loaded "module-udev-detect" (index: #4; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-bluetooth-discover.so': success D: dbus-util.c: Successfully connected to D-Bus system bus ba7c9a1f90b3d49d930bca2100000015 as :1.62 D: bluetooth-util.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired D: bluetooth-util.c: Bluetooth daemon is apparently not available. I: module.c: Loaded "module-bluetooth-discover" (index: #5; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-esound-protocol-unix.so': success I: module.c: Loaded "module-esound-protocol-unix" (index: #6; argument: ""). I: module.c: Loaded "module-native-protocol-unix" (index: #7; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-gconf.so': success I: module.c: Loaded "module-gconf" (index: #8; argument: ""). I: module-default-device-restore.c: Saved default sink 'auto_null' not existant, not restoring default sink setting. I: module-default-device-restore.c: Saved default source 'auto_null.monitor' not existant, not restoring default source setting. I: module.c: Loaded "module-default-device-restore" (index: #9; argument: ""). I: module.c: Loaded "module-rescue-streams" (index: #10; argument: ""). D: module-always-sink.c: Autoloading null-sink as no other sinks detected. I: sink.c: Created sink 0 "auto_null" with sample spec s16le 6ch 44100Hz and channel map front-left,front-left-of-center,front-center,front-right,front-right-of-center,rear-center I: sink.c: device.description = "Dummy Output" I: sink.c: device.class = "abstract" I: sink.c: device.icon_name = "audio-card" D: core-subscribe.c: Dropped redundant event due to change event. I: source.c: Created source 0 "auto_null.monitor" with sample spec s16le 6ch 44100Hz and channel map front-left,front-left-of-center,front-center,front-right,front-right-of-center,rear-center I: source.c: device.description = "Monitor of Dummy Output" I: source.c: device.class = "monitor" I: source.c: device.icon_name = "audio-input-microphone" D: module-null-sink.c: Thread starting up I: module.c: Loaded "module-null-sink" (index: #11; argument: "sink_name=auto_null sink_properties='device.description="Dummy Output"'"). I: module.c: Loaded "module-always-sink" (index: #12; argument: ""). I: module.c: Loaded "module-intended-roles" (index: #13; argument: ""). D: module-suspend-on-idle.c: Sink auto_null becomes idle, timeout in 5 seconds. I: module.c: Loaded "module-suspend-on-idle" (index: #14; argument: ""). I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session1" D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session1 I: module.c: Loaded "module-console-kit" (index: #15; argument: ""). I: module.c: Loaded "module-position-event-sounds" (index: #16; argument: ""). D: dbus-util.c: Successfully connected to D-Bus session bus efbffc6788fad56cfd64d40c00000018 as :1.182 D: main.c: Got org.pulseaudio.Server! I: main.c: Daemon startup complete. I: client.c: Created 1 "Native client (UNIX socket client)" I: client.c: Created 2 "Native client (UNIX socket client)" D: protocol-native.c: Protocol version: remote 16, local 16 I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: yes D: protocol-native.c: Protocol version: remote 16, local 16 I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: yes D: module-augment-properties.c: Looking for .desktop file for gnome-volume-control-applet D: module-augment-properties.c: Looking for .desktop file for gnome-settings-daemon D: core-subscribe.c: Dropped redundant event due to change event. I: module-suspend-on-idle.c: Sink auto_null idle for too long, suspending ... D: sink.c: Suspend cause of sink auto_null is 0x0004, suspending Note the one section that seems to find the hardware but says it's busy (no idea if this is relevant). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-udev-detect.so': success D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes D: module-udev-detect.c: /devices/pci0000:00/0000:00:1b.0/sound/card0 is busy: yes I: module-udev-detect.c: Found 1 cards.

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• Dealing with numerous, simultaneous sounds in unity

  - by luxchar

  I've written a custom class that creates a fixed number of audio sources. When a new sound is played, it goes through the class, which creates a queue of sounds that will be played during that frame. The sounds that are closer to the camera are given preference. If new sounds arrive in the next frame, I have a complex set of rules that determines how to replace the old ones. Ideally, "big" or "important" sounds should not be replaced by small ones. Sound replacement is necessary since the game can be fast-paced at times, and should try to play new sounds by replacing old ones. Otherwise, there can be "silent" moments when an old sound is about to stop playing and isn't replaced right away by a new sound. The drawback of replacing old sounds right away is that there is a harsh transition from the old sound clip to the new one. But I wonder if I could just remove that management logic altogether, and create audio sources on the fly for new sounds. I could give "important" sounds more priority (closer to 0 in the corresponding property) as opposed to less important ones, and let Unity take care of culling out sound effects that exceed the channel limit. The only drawback is that it requires many heap allocations. I wonder what strategy people use here?

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• How safe is chatting in iGoogle?

  - by Abhijeet

  How safe is chatting in iGoogle - Chat in office? I am chatting over HTTPS.

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• Chat with Audio/Video and DesktopSharing

  - by RavIncredible

  Hi All, I am working on an application where I like to bundle 3 things: 1. Chat 2. Audio / Video Conversation 3. Desktop Sharing I would like to know the approach and where to look in for this, few of the things that I am aware of are: Chat and Audio – I can go with Jabber server and configure any SIP server like asterix for audio calls. Desktop Sharing – I have read about silverlite coming up with Desktop Sharing modules, but what would be only targeted to Windows. I would like to have sharing for windows, mac and linux OS. I don’t mind building separate clients for each. But I like to know which common protocol has to be used for Desktop sharing. In other words something similar to team viewer. Please suggest. Video Conference – I totally don’t have any idea about this. The application that I am supposed to build has to target the below platforms: 1. Window, Mac, Linux Desktop 2. iPhone, iPad and Android Devices. Would appreciate any help or reference or links to any of the topics (Chat, A/V and desktop sharing). Thanks Ravi

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• how to start atop audio in j2me

  - by haaris_86

  i want to add audio to the application. i habe added audio but the problem is that the audio does not stop till the audio clip is over. How do i make it stop. Please give axample.

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• How to detect generation loss of a transcoded audio.

  - by The Rook

  Lets say you have a 96 kbit mp3 and you Transcode the file into a 320 kbit mp3. How could you programmatically detect the original bit rate or quality? Generation loss is created because each time a lossy algorithm is applied new information will be deemed "unnecessary" and is discarded. How could an algorithm use this property to detect the transcoding of audio. 128 kbps LAME mp3 transcoded to 320 kbps LAME mp3 (I Feel You, Depeche Mode) 10.8 MB. This image was taken from the bottom of this site. The 2 tracks above look nearly identical, but the difference is enough to support this argument.

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• What language/API to use for a standalone live-input audio visualizer app?

  - by knuckfubuck

  I develop with Actionscript and was glad to see that AIR 2.0 was going to give access to mic input data. I planned to use this to create a visualizer set to the tempo of the incoming live audio. After doing a few days of google research it seems unlikely that it will be possible to analyze the data of the mic input in Flash/AIR. If anyone has ideas on how I can achieve this in AIR please let me know. (I'm open to workarounds.) That being said, I don't want to give up on the idea so I'm interested in suggestions for other language/API to use. My requirements for the app are: Run on OSX Two windows - one that can go fullscreen while the other(controller GUI) stays put Able to access live mic input data I've done reading on FFT and understand what needs to be done on the sound side so no need to help with that.

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• Streaming audio to mobile phones, what technology to use ?

  - by Alx

  I'm planning on building an application where audio media is going to be streamed to the mobile phone for the user to listen. The targets are smartphones: iPhone/Blackberry/Android/(J2ME ?). I see that streaming on iPhone has to be done with HTTP Live streaming, but I don't see it supported by other platforms. Should I broadcast the streams via rstp ? http ? Is there any way to use a unified solution for all the different mobile platform ? If anyone already had to go through this, help would be gratly appreciated.

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• Android: Using MediaRecorder to crop an existing audio file?

  - by user141146

  Hi, I'd like to take an existing mp3 file located on an SD card and arbitrarily crop it (e.g. crop from 0:12 to 1:14 in a 3 minute song). The only class that I've seen that seems remotely relevant to do this is the MediaRecorder class. My 'hope' would be to "record" an existing file like this: MediaRecorder recorder = new MediaRecorder(); recorder.setAudioSource(###some magical way of specifying an existing file??###); But this obviously doesn't work (setAudioSource() takes an int and seems to default to the phone's microphone). Is there a class or an approach that can be used to crop audio on the phone itself? TKS!!

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• How does one record audio from a Javascript based webapp?

  - by username

  I'm trying to write a web-app that records WAV files (eg: from the user's microphone). I know Javascript alone can not do this, but I'm interested in the least proprietary method to augment my Javascript with. My targeted browsers are Firefox for PC and Mac (so no ActiveX). Please share your experiences with this. I gather it can be done with Flash (but not as a WAV formated file). I gather it can be done with Java (but not without code-signing). Are these the only options? @dominic-mazzoni I'd like to record the file as a WAV because because the purpose of the webapp will be to assemble a library of good quality short soundbites. I estimate upload will be 50 MB, which is well worth it for the quality. The app will only be used on our intranet. UPDATE: There's now an alternate solution thanks to JetPack's upcoming Audio API: See https://wiki.mozilla.org/Labs/Jetpack/JEP/18

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• is there any Simple opensource live audio streaming Server using WCF? (see specification below)

  - by Ole Jak

  is there any Simple opensource live audio streaming Server using WCF? I need it to have simple structure: it should listen to some url format like http://example.com/service/stream?write&id=ANY_STRING and if any data comes to such address format it'll start making it avaliable by something like this http://example.com/service/stream?read&id=ANY_STRING Main thing here to be able to stream live data thru WCF service not buffering it just sharing stream. So can please any one help me with such idea? I think not only I have seen such problem with WCF alot on different sites so answer will help the WCF comunyty alot. I hope. BTW: I know some people say WCF is not prepared for live streaming over bacikHTTPbinding but hey! We all need it to, and we ask MS alot so some day they'll make it beter and we all want to be prepared for it.

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• How to initiate chatting between two clients and two clients only, using applets and servlets?

  - by mithun1538

  Hello everyone, I first need to apologize for my earlier questions. (You can check my profile for them)They seemed to ask more questions than give answers. Hence, I am laying down the actual question that started all them absurd questions. I am trying to design a chat applet. Till now, I have coded the applet, servlet and communication between the applet and the servlet. The code in the servlet side is such that I was able to establish chatting between clients using the applets, but the code was more like a broadcast all feature, i.e. all clients would be chatting with each other. That was my first objective when I started designing the chat applet. The second step is chatting between only two specific users, much like any other chat application we have. So this was my idea for it: I create an instance of the servlet that has the 'broadcast-all' code. I then pass the address of this instance to the respective clients. 2 client applets use the address to then chat. Technically the code is 'broadcast-all', but since only 2 clients are connected to it, it gives the chatting between two clients feature. Thus, groups of 2 clients have different instances of the same servlet, and each instance handles chatting between two clients at a max. However, as predicted, the idea didn't materialize! I tried to create an instance of the servlet but the only solution for that was using sessions on the servlet side, and I don't know how to use this session for later communications. I then tried to modify my broadcast-all code. In that code, I was using classes that implemented Observer and Observable interfaces. So the next idea that I got was: Create a new object of the Observable class(say class_1). This object be common to 2 clients. 2 clients that wish to chat will use same object of the class_1. 2 other clients will use a different object of class_1. But the problem here lies with the class that implements the Observer interface(say class_2). Since this has observers monitoring the same type of class, namely class_1, how do I establish an observer monitoring one object of class_1 and another observer monitoring another object of the same class class_1 (Because notifyObservers() would notify all the observers and I can't assign a particular observer to a particular object)? I first decided to ask individual problems, like how to create instances of servlets, using objects of observable and observer and so on in stackoverflow... but I got confused even more. Can anyone give me an idea how to establish chatting between two clients only?(I am using Http and not sockets or RMI). Regards, Mithun. P.S. Thanks to all who replied to my previous (absurd) queries. I should have stated the purpose earlier so that you guys could help me better.

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• why does this knockout method receive a form element instead of the object its nested in?

  - by ladookie

  I have this HTML: <ul class="chat_list" data-bind="foreach: chats"> <li> <div class="chat_response" data-bind="visible: CommentList().length == 0"> <form data-bind="submit: $root.addComment"> <input class="comment_field" placeholder="Comment…" data-bind="value: NewCommentText" /> </form> </div> </li> </ul> and this JavaScript: function ChatListViewModel(chats) { // var self = this; self.chats = ko.observableArray(ko.utils.arrayMap(chats, function (chat) { return { CourseItemDescription: chat.CourseItemDescription, CommentList: ko.observableArray(chat.CommentList), CourseItemID: chat.CourseItemID, UserName: chat.UserName, ChatGroupNumber: chat.ChatGroupNumber, ChatCount: chat.ChatCount, NewCommentText: ko.observable("") }; })); self.newChatText = ko.observable(); self.addComment = function (chat) { var newComment = { CourseItemDescription: chat.NewCommentText(), ParentCourseItemID: chat.CourseItemID, CourseID: $.CourseLogic.dataitem.CourseID, AccountID: $.CourseLogic.dataitem.AccountID, SystemObjectID: $.CourseLogic.dataitem.CommentSystemObjectID, SystemObjectName: "Comments", UserName: chat.UserName }; chat.CommentList.push(newComment); chat.NewCommentText(""); }; } ko.applyBindings(new ChatListViewModel(initialData)); When I go into the debugger it shows that the chat parameter of the addComment() function is a form element instead of a chat object. Why is this happening?

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• Would like to change audio codec, but keep video settings with ffmpeg

  - by Craig Tataryn

  I have a video for which I'd like to convert the audio codec to AAC 320 kbps / 44.100 kHz. What would I use for ffmpeg switches such that all the video settings and codec remain the same, but only the audio codec and settings change? Here's my video: $ ffmpeg -i Winnipeg.rb\ Scala-Talk.mov FFmpeg version SVN-r25375, Copyright (c) 2000-2010 the FFmpeg developers built on Oct 6 2010 13:02:41 with gcc 4.2.1 (Apple Inc. build 5664) configuration: --enable-libmp3lame --enable-shared --disable-mmx --arch=x86_64 libavutil 50.32. 2 / 50.32. 2 libavcore 0. 9. 1 / 0. 9. 1 libavcodec 52.92. 0 / 52.92. 0 libavformat 52.80. 0 / 52.80. 0 libavdevice 52. 2. 2 / 52. 2. 2 libavfilter 1.48. 0 / 1.48. 0 libswscale 0.12. 0 / 0.12. 0 Seems stream 0 codec frame rate differs from container frame rate: 2000.00 (2000/1) -> 10.00 (10/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Winnipeg.rb Scala-Talk.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt Duration: 01:10:53.00, start: 0.000000, bitrate: 283 kb/s Stream #0.0(eng): Video: h264, yuv420p, 800x598, 94 kb/s, 10 fps, 10 tbr, 1k tbn, 2k tbc Stream #0.1(eng): Audio: adpcm_ima_qt, 22050 Hz, 1 channels, s16 Stream #0.2(eng): Audio: adpcm_ima_qt, 22050 Hz, 1 channels, s16 At least one output file must be specified Many thanks in advance! One with with ffmpeg I've never been able to grok is how to just "tweak" files without having to regurgitate every little setting for things you don't want changes.

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• Convert MPG w/ AC3 audio to something else - on a Mac

  - by anonymous coward

  I'm helping with a small volunteer media team, and they have several .mpg videos that don't appear to have sound when played in QuickTime, iTunes, Real Player, etc, on the local Mac machine. I was able to hear audio after transferring one of the movies to a Windows machine that had VLC media player on it. Through VLC I was able to discover that the audio stream is a52 / AC3 format. We use Autodesk Cleaner in our normal workflow of converting the format of our videos to FLV, but for some reason it's unable to convert this particular batch of videos (well, the video converts fine, but with no audio). Obviously, it seems that there's a codec issue here, but I'm not sure how to correct it. (I'm not extremely familiar with Macs, and/or Autodesk Cleaner). I've seen the Perian codec pack, but I'm not sure that having the codecs on the system will enable Cleaner to convert these videos (particularly the audio stream, since the video converts fine). Is there something obvious that I'm overlooking, or will we have to use something else for this particular batch of videos? If so, what?

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• Audio playback, creating nested loop for fade in/out.

  - by Dave Slevin

  Hi Folks, First time poster here. A quick question about setting up a loop here. I want to set up a for loop for the first 1/3 of the main loop that will increase a value from .00001 or similar to 1. So I can use it to multiply a sample variable so as to create a fade-in in this simple audio file playback routine. So far it's turning out to be a bit of a head scratcher, any help greatfully recieved. for(i=0; i < end && !feof(fpin); i+=blockframes) { samples = fread(audioblock, sizeof(short), blocksamples, fpin); frames = samples; for(j=0; j < frames; j++) { for (f = 0; f< frames/3 ;f++) { fade = fade--; } output[j] = audioblock[j]/fade; } fwrite(output,sizeof(short), frames, fpoutput); } Apologies, So far I've read and re-write the file successfully. My problem is I'm trying to figure out a way to loop the variable 'fade' so it either increases or decreases to 1, so as I can modify the output variable. I wanted to do this in say 3 stages: 1. From 0 to frames/3 to increace a multiplication factor from .0001 to 1 2. from frames 1/3 to frames 2/3 to do nothing (multiply by 1) and 3. For the factor to decrease again below 1 so as for the output variable to decrease back to the original point. How can I create a loop that will increase and decrease these values over the outside loop?

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• Simple way to embed an MP3 audio file in a (static) HTML file, starting to play at a specifc time?

  - by Marcel

  Hi all, I want to produce a simple, static HTML file, that has one or more embedded MP3 files in it. The idea is to have a simple mean of listening to specific parts of an mp3 file. On a single click on a visual element, the sound should start to play; However, not from the beginning of the file, but from a specified starting point in that file (and play to the end of the file). This should work all locally from the client's local filesystem, the HTML file and the MP3 files do not reside on a webserver. So, how to play the MP3 audio from a specific starting point? The solution I am looking for should be as simple as possible, while working on most browsers, including IE, Firefox and Safari. Note: I know the <embed> tag as described here, but this seems not to work with my target browsers. Also I have read about jPlayer and other Java-Script-based players, but since I have never coded some JavaScript, I would prefer a HTML-only solution, if possible.

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• yahoo media player not working.

  - by luca590

  I have a yahoo media player embedded in my webpage. I am currently using Ruby on Rails to create/edit my web page. When i click the play button next to a track the YMP waits a while and then goes to the next track without playing the first one. I then get a warning on my second (last) track that its file could not be found. Does anyone has a better recommendation for an audio player or a way to fix this one?

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• How to convert from wav or mp3 to raw PCM [on hold]

  - by Komyg

  I am developing a game using Cocos2d-X and Marmalade SDK, and I am looking for any recommendations of programs that can convert audio files in mp3 or wav format to raw PCM 16 format. The problem is that I am using the SimpleAudioEngine class to play sounds in my game and in Marmalade it only supports files that are encoded as raw PCM 16. Unfortunately I've been having a very hard time finding a program that can do this type of conversion, so I am looking for a recommendation.

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• yahoo media player not working

  - by luca590

  I have a yahoo media player embedded in my webpage. I am currently using Ruby on Rails to create/edit my web page. When i click the play button next to a track the YMP waits a while and then goes to the next track without playing the first one. I then get a warning on my second (last) track that its file could not be found. Does anyone has a better recommendation for an audio player or a way to fix this one?

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• Yahoo media player not working with Ruby on rails

  - by luca590

  I have a yahoo media player embedded in my webpage. I am currently using Ruby on Rails to create/edit my web page. When i click the play button next to a track the YMP waits a while and then goes to the next track without playing the first one. I then get a warning on my second (last) track that its file could not be found. Does anyone has a better recommendation for an audio player or a way to fix this one?

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• Virtual Microphone and skype.

  - by Dario

  Hello, I need to have at least one microphone on Windows to make Skype calls, but i have a VPS with Windows 2003 server with no audio device. I googled a lot and finally i found something called "Virtual Audio Cable", a tool to install virtual audio drivers ( http://software.muzychenko.net/eng/vac.html ). I tried many times but i couldn't get this driver work, so i'm asking if someone know a similar solution, i mean a virtual microphone or a way to make skype working without any microphone. Thanks all!

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