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  • ffmpeg hangs when creating a video

    - by FearUs
    I am trying to insert an audio channel with a video: first of all I extract the audio from the original video for processing: ffmpeg -i lotr.mp4 lotr.wav I then extract all frames for later processing too: ffmpeg -i lotr.mp4 -f image2 %d.jpg When done processing audio and video streams, I try to create the video ffmpeg -f image2 -r 15 -i %d.jpg new.mp4 then merge with the audio: ffmpeg -i new.mp4 -i lotr.wav -map 0:0 -map 1:0 new_w_audio.mp4 Result: CPU activity = 100%, the process hangs and never returns. PS: I even tried it without modifying the images or the audio (so just trying to unpack then repack the video) but still the same output FFmpeg version SVN-r26400, Copyright (c) 2000-2011 the FFmpeg developers built on Jan 18 2011 04:07:05 with gcc 4.4.2 configuration: --enable-gpl --enable-version3 --enable-libgsm --enable-libvorb is --enable-libtheora --enable-libspeex --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-libvpx --disable-decoder=libvpx --arch=x86 --enable-runtime-cpudetect - -enable-libxvid --enable-libx264 --enable-librtmp --extra-libs='-lrtmp -lpolarss l -lws2_32 -lwinmm' --target-os=mingw32 --enable-avisynth --enable-w32threads -- cross-prefix=i686-mingw32- --cc='ccache i686-mingw32-gcc' --enable-memalign-hack libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'new.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Duration: 00:00:29.66, start: 0.000000, bitrate: 193 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], 192 k b/s, 15 fps, 15 tbr, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 [wav @ 01fed010] max_analyze_duration reached Input #1, wav, from 'lotr.wav': Duration: 00:00:29.90, bitrate: 176 kb/s Stream #1.0: Audio: pcm_s16le, 11025 Hz, 1 channels, s16, 176 kb/s File 'new_w_audio.mp4' already exists. Overwrite ? [y/N] y [buffer @ 01b03820] w:200 h:134 pixfmt:yuv420p Output #0, mp4, to 'new_w_audio.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], q=2-3 1, 200 kb/s, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1: Audio: aac, 11025 Hz, 1 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding

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  • Is it possible to transcode audio in C# using DirectSound?

    - by Robert Davis
    I want to transcode a lot of audio from its source format to PCM without resampling or messing with the sample size. I figure if Windows Media Player can play the file and it doesn't use a legacy ACM codecs it must be using DirectSound to do so (this is on Windows XP and Windows Server 2k3). So is it possible to access DirectSound from C# and do so? I've tried searching the web but all the examples have been about playback which I have no interest in doing.

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  • What's the best way of playing media files (esp. audio) with Mono/C#?

    - by supercheetah
    I'm trying to create something that will be playing some sound and music for some things in Mono+C#, but I'm not sure what the best thing will be for that. I'm trying to make it usable with things like Ogg Vorbis, MP3s, and wave files. My primary platform will be Linux, although a cross platform solution would be nice. Anyone have any suggestions for libraries for playing audio files?

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  • Where to start learning about audio or video codecs ?

    - by Vamsi
    Hi, I am very much confused to know what happens inside the codecs. I want to learn about the elements inside audio encoders and decoders. Would be very happy if you can provide me some links where i can find some good study material. Thanks precisely i would like to know how the codec parses the a media file.

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  • Compiling a list of audio + video players (flash / javascript / otehr) that I can embed into a websi

    - by FiveTools
    I'm compiling a list of audio + video players (flash / javascript / other) that I can embed into a website. flowplayer: http://flowplayer.org/ jw player: http://www.longtailvideo.com/players/ premium beat: http://www.premiumbeat.com/flash_resources/free_flash_music_player/ xspf web player: http://musicplayer.sourceforge.net/ yahoo media player: http://mediaplayer.yahoo.com/ any popular ones I'm missing? (anyone know if I can skin / customize any of them to operate similar to the Windows vista volume control?)

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  • Where can I get a splitter to connect a device with a single 3.5 mm plug into the audio input/output jacks on my laptop?

    - by XinJeisan
    I recently bought the :Hype Retro Handset for Mobile Phone" -- its just a device that looks like a handset to use when chatting on a computer or mobile phone that plugs into the phone/computer with a single 3.5 mm plug. I was hoping to use it on my windows 7 Toshiba laptop. I can hear audio fine through the handset but what I'm saying is not being picked up on the handset. On the box it says "some phones and computers may need additional adapters," so I'm hoping it is possible to get a splitter or something for this to work properly. I did email the parent company (http://dglusa.com/) but I haven't heard from them, and, looking over their website, I doubt I will. I also went to the local radio shack, and the guy said I needed a splitter, but he didn't know where to get one. I can find the kind of splitter I think I need online, but I'm unsure whether they are just for output or can also do input/output.

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  • Flash in browsers does not play sound accurately using Pulse network audio

    - by Dave M G
    I use PulseAudio to send sound over the LAN to an audio server. When playing any Flash media in Firefox or Chrome, the sound flutters, as if the volume were going up and down every second. The problem does not exhibit with any other software, and I think it's specific to how Flash interacts with my sound set up. How do I get Flash to play nice with the PulseAudio network sound server? Update I have discovered that I can stop the sound fluttering if I follow these steps: Start a Flash video Run pulseaudio --kill on the server Wait about 7 seconds After this, the PulseAudio server automatically respawns, and the sound in the Flash video is perfect. The problem now, though, is that I have to do this every time I start a Flash video. This is obviously not desireable. So, the question is, how do I make whatever it is that makes the sound work when I go through these steps stick so that I don't have to do them? Also, I've uploaded some PulseAudio log output to Pastebin, taken while attempting to play a Flash video, if that helps. I've tried to get logging details from Flash, but despite installing and enabling Flash for debugging, it has not generated any ouput at all. Details I have uploaded an example video of the problem onto Youtube. In the video you can see the opening of a Ted Talk video, and the sound flutters as it plays. The video also stutters while playing back. Here are my sound device output settings:

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  • Jack Audio ubuntu 12.10

    - by Shaneo1
    I used to have Jack Server working with 10.10, 11.04, 11.10 but not 12.04 and now 12.10. I have installed jackd jackd2 qjackctl surfed many forums and even given advice of how to get jack working, but now I am stuck. Tue Nov 27 22:30:46 2012: Saving settings to "/home/shane/.config/jack/conf.xml" ... 22:31:19.960 D-BUS: JACK server could not be started. Sorry Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started Tue Nov 27 22:31:19 2012: Starting jack server... Tue Nov 27 22:31:19 2012: JACK server starting in realtime mode with priority 10 Tue Nov 27 22:31:19 2012: [1m[31mERROR: cannot register object path "/org/freedesktop/ReserveDevice1/Audio0": A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Failed to acquire device name : Audio0 error : A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Audio device hw:0,0 cannot be acquired...[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Cannot initialize driver[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: JackServer::Open failed with -1[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Failed to open server[0m Tue Nov 27 22:31:21 2012: Saving settings to "/home/shane/.config/jack/conf.xml" ... 22:31:22.047 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started Can anyone assist?

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  • Double audio cd ripping weirdness

    - by jqno
    Since I installed Ubuntu 12.04, Rhythmbox, Banshee and Sound Juicer have started acting weird around double cd's, and specifically, cd #2 of said double cd. Sometimes, they will show the information of cd #1. Track names, durations, and even count are incorrect. Sometimes, they will first show the tracks for cd #1, then continue onto cd #2 if cd #2 has more tracks than #1. Sound Juicer seems to be unable to find any track durations at all, even for single cd's. Obviously, this is a pain when I'm trying to rip double cd's. And I have a fair number of them, which I want to rip. This happens on both my machines (a slightly aging iMac, and a 1-year-old Sony Vaio). However, on previous versions of Ubuntu, this never happened. All on the same machines. So I suspect 12.04 is using a different lib for extracting audio cd data. Just for kicks, I tried with Linux Mint 13, and there it works correctly, even though it claims to be based on Ubuntu 12.04 and therefore should be using (partially) the same software. So if the Mint guys can fix it, I should be able to do it too, right? So, my question: what changed in 12.04 that could cause this? And more importantly: what can I do to fix it?

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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 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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