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  • FFmpeg audio dont work in converted videos

    - by Juddy Swaft
    NOTICE: when i convert videos via terminal and download them from ftp into pc the audio works fine. I use: if($ext == "avi" && $convert_avi == true) { $convert_source = _VIDEOS_DIR_PATH.$new_name; $conv_name = substr(md5($file['name'].rand(1,888)), 2, 10).".mp4"; $converted_file = _VIDEOS_DIR_PATH.$conv_name; $ffmpeg_command = 'ffmpeg -i '.$convert_source.' -acodec libmp3lame -vcodec libx264 -s 1280x720 -ar 44100 -async 44100 -r 29.970 -ac 2 -qscale 5 '.$converted_file; echo exec($ffmpeg_command); $sql = "UPDATE pm_temp SET url = '".$conv_name."' WHERE url = '".$new_name."' LIMIT 1"; $result = @mysql_query($sql); unlink($convert_source); } This code to convert avi to mp4 ffmpeg concole output: root@1tb:~# ffmpeg -i sample.avi -acodec libmp3lame -vcodec libx264 -s 1280x720 -ar 44100 -async 44100 -r 29.970 -ac 2 -qscale 5 goodsample.mp4 ffmpeg version 0.7.15, Copyright (c) 2000-2013 the FFmpeg developers built on Feb 22 2013 07:18:58 with gcc 4.4.5 configuration: --enable-libdc1394 --prefix=/usr --extra-cflags='-Wall -g ' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-avfilter --enable-libdirac --disable-decoder=libdirac --enable-libfreetype --enable-libschroedinger --disable-encoder=libschroedinger - s libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.123. 0 / 52.123. 0 libavformat 52.111. 0 / 52.111. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 [mp3 @ 0x191d4100] Header missing [mpeg4 @ 0x191d1dc0] Invalid and inefficient vfw-avi packed B frames detected Input #0, avi, from 'sample.avi': Metadata: encoder : VirtualDubMod 1.5.10.2 (build 2540/release) Duration: 00:01:01.81, start: 0.000000, bitrate: 1194 kb/s Stream #0.0: Video: mpeg4, yuv420p, 640x352 [PAR 1:1 DAR 20:11], 23.98 tbr, Stream #0.1: Audio: mp3, 48000 Hz, stereo, s16, 128 kb/s [buffer @ 0x191d1c80] w:640 h:352 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param: [scale @ 0x191d6880] w:640 h:352 fmt:yuv420p -> w:1280 h:720 fmt:yuv420p flags:0 [libx264 @ 0x191ce5a0] Default settings detected, using medium profile [libx264 @ 0x191ce5a0] using SAR=45/44 [libx264 @ 0x191ce5a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle S [libx264 @ 0x191ce5a0] profile High, level 3.1 [libx264 @ 0x191ce5a0] 264 - core 118 - H.264/MPEG-4 AVC codec - Copyleft 2003-2 6 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_off 1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_l Output #0, mp4, to 'goodsample.mp4': Metadata: encoder : Lavf52.111.0 Stream #0.0: Video: libx264, yuv420p, 1280x720 [PAR 45:44 DAR 20:11], q=2-31 Stream #0.1: Audio: libmp3lame, 44100 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop, [?] for help [mp3 @ 0x191d4100] Header missing Error while decoding stream #0.1 [mpeg4 @ 0x191d1dc0] Invalid and inefficient vfw-avi packed B frames detected [mp3 @ 0x191d4100] incomplete frame 9467kB time=00:01:00.32 bitrate=1285.5kbits/ Error while decoding stream #0.1 frame= 1852 fps= 20 q=29.0 Lsize= 9652kB time=00:01:01.72 bitrate=1280.9kbits video:9121kB audio:483kB global headers:0kB muxing overhead 0.499688% frame I:11 Avg QP:16.78 size: 51456 [libx264 @ 0x191ce5a0] frame P:784 Avg QP:20.81 size: 8954 [libx264 @ 0x191ce5a0] frame B:1057 Avg QP:26.06 size: 1659 [libx264 @ 0x191ce5a0] consecutive B-frames: 22.0% 3.1% 7.5% 67.4% [libx264 @ 0x191ce5a0] mb I I16..4: 31.1% 59.8% 9.1% [libx264 @ 0x191ce5a0] mb P I16..4: 1.8% 2.6% 0.2% P16..4: 24.3% 7.0% 4.0 [libx264 @ 0x191ce5a0] mb B I16..4: 0.1% 0.1% 0.0% B16..8: 22.7% 0.8% 0.2 [libx264 @ 0x191ce5a0] 8x8 transform intra:57.0% inter:72.6% [libx264 @ 0x191ce5a0] coded y,uvDC,uvAC intra: 44.4% 33.3% 10.3% inter: 7.6% 5. [libx264 @ 0x191ce5a0] i16 v,h,dc,p: 68% 14% 8% 10% [libx264 @ 0x191ce5a0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 14% 27% 5% 7% 7% 6 [libx264 @ 0x191ce5a0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 28% 14% 14% 6% 10% 9% 7 [libx264 @ 0x191ce5a0] i8c dc,h,v,p: 67% 13% 17% 3% [libx264 @ 0x191ce5a0] Weighted P-Frames: Y:1.9% UV:0.4% [libx264 @ 0x191ce5a0] ref P L0: 62.2% 12.8% 10.3% 14.5% 0.2% [libx264 @ 0x191ce5a0] ref B L0: 88.1% 5.5% 6.4% [libx264 @ 0x191ce5a0] ref B L1: 95.7% 4.3% [libx264 @ 0x191ce5a0] kb/s:1209.03 I know there is couple errors tough, but i dont know hot to fix it. Also i would be very thankfull if someone can help reduce video size but is not main problem video weights as original avi but sill.

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  • Laptop wakes from sleep, once, due to audio controller (Windows 7)

    - by stijn
    The laptop is a recent Dell XPS 15z and the problem is as follows (reproducible about 90% of tries): put laptop to sleep using either Start-Sleep or closing the lid laptop goes to sleep after about 5 seconds, but instantly wakes again showing a black screen (touching the keyboard or moving the mouse shows the login screen one normally gets after wake) login again, put laptop to sleep latop stays in sleep mode output of powercfg -lastwake after the first instant wake shows the audio controller is responsible. Why would that be, why only the first try, and how to fix this? Wake History Count - 1 Wake History [0] Wake Source Count - 1 Wake Source [0] Type: Device Instance Path: PCI\VEN_8086&DEV_1C20&SUBSYS_04461028&REV_05\3&11583659&0&D8 Friendly Name: Description: High Definition Audio Controller Manufacturer: Microsoft

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  • Use all 5.1 speakers with a 2.1 audio source

    - by thegreyspot
    Hi! I just bought a 5.1 surround sound speaker set for my computer in my bedroom. The rear speakers are next to me in bed while the front speakers are at the other end of the bed at my feet. While I enjoy the surround sound during movies that support 5.1 sound, I would like to have my rear speakers working when listening to podcasts, or other 2.1 channel sound. How can I do this? When I enable "Speaker Fill" in the Realtek Hd Audio manager the sound only comes out of the front and center speakers with a few background noises that come out the rear ones. But since my ears are closer to the rear speakers, I'd rather have the sound come out of them. Let me know of any ideas! Hmm seems like the only option is to set the rear speakers to "Front Speakers" and change it to stereo in the Realtek HD audio. But still that take alot of steps and it doesnt not use the center speaker Thanks

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  • Realtek HD Audio playing weird with certain video formats

    - by dyasny
    Hi, I have a Gigabyte motherboard with an onboard Realtek HD sound card. The card is working perfectly everywhere, except for a single video format, where the voice is distorted, sounds as if it's been passed through a metal tube. Been googling for this, but couldn't find an answer anywhere. The movie plays fine on other systems (got Linux everywhere else), but on this one (winXP-x64-sp2) it just doesn't. Here are some details: MPC: Type: KLCP WMV File Audio: 0x000a 22050Hz mono 20Kbps [Raw Audio 0] Video: Windows Media Video 9 400x300 29.97fps 227Kbps [Raw Video 1] VLC: Codec: wmas Sample rate: 22050 Bits per sample: 16 Bitrate: 20kb/s

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  • In search of a good audio player for Ubuntu 9.10

    - by Joe Casadonte
    If this should be marked Community Wiki, please let me know. I'm switching from XP to Ubuntu, and I have been very disappointed with the selection of media players available. I'm primarily interested in an audio player, but integrated video and library management is OK, too. My criteria: Must be able to play audio CDs (I'm shocked how many apps this does away with, right away) Must be able to play MP3 & WAV; OGG, SHN, FLAC are all bonuses Repeat and Shuffle modes are a must FreeDB / GraceNote through a proxy is a must (if it can read a PAC file, that would be awesome) It needs to be really small, e.g. skinnable or an applet Ability to execute a playlist is a plus Gapless MP3 playback a plus I'm running Gnome, but I'm not totally adverse to a KDE app. Command-line only is also a viable option. Some that I've tried: RhythmBox - probably the best of the lot that I've tried; I don't like its mini mode (doesn't show the song being played) and I can't figure out how to get it to hit FreeDB/GraceNote through a proxy Songbird - can't play CDs, playlist management is atrocious Banshee Jajuk Maybe a couple of more. Thanks! UPDATE I tried out VLC, Amarok and Songbord (again). VLC I eventually got to work (I had some kind of bad configuration). It seemed way more involved than I was looking for out of a music player, and in general more geared to video than audio. I couldn't fathom its library management, which I think it has; maybe it doesn't, and that's why I couldn't figure it out. Amaork looked very promising but the library management was not to my liking, and the way it handled a playlist with both MP3 and WAV is inexplicable at best. I did like some aspects of the UI, but not enough to keep it. Songbird is very finicky, but I like the library management. Sort of. It kept telling me my Watch folder was invalid, even thought it clearly was accessible. Playlist management is bizarre, and the message that it was deleting source files whenever I deleted a playlist had me too worried to keep using it. Had it been able to play CDs, maybe I would have persevered. Audacious, while a bit odd at times, does seem to do what I want. If it had a library manager, I wouldn't have bothered trying any of the others. Thanks for the help, everyone!

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  • Creating video with audio and still image for YouTube

    - by scottlabs
    I'm running the following command: ffmpeg -i audio.mp3 -ar 44100 -f image2 -i logo.jpg -r 15 -b 1800 -s 640x480 foo.mov Which successfully outputs a video with my recorded audio and an image on it. When I try and upload this to YouTube it fails to process, regardless of the formats I try: .mov, .avi, .flv, .mp4 Is there some setting I'm missing in the above that would generate a format Youtube will accept? I've tried looking through the ffmpeg documentation but I'm in over my head. I did an experiment by putting a 2 second video with a 30 second mp3. When I uploaded to youtube, the resulting video was only 2 seconds long. So it may be that YouTube looks only to the video track for the length, and since a picture is only a frame long or whatever, maybe that borks it.

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  • Another sound not working post

    - by Thomas Smart
    Tried all the other "sound not working" posts i think, lost count. purge/reinstall alsa and pulse, reboot, add user to audio group, various lines in the alsa config file such as "options snd-hda-intel model=" then tried different options like generic, auto, basic, default, etc. tried pulseaudio -k && sudo alsa force-reload a few times, with and without rebooting. Hardware: 16gb ram, core I7-4790, Intel Haswell mboard with onboard sound and graphics Multimedia: Audio Adapter: HDA-Intel-HDA Intel HDMI OS: Ubuntu server 14.04 with ubuntu-desktop installed. GUI sound settings lists only the dummy sound card alsamixer -c 0 ¦ Card: HDA Intel HDMI F1: Help ¦ ¦ Chip: Intel Haswell HDMI F2: System information ¦ ¦ View: F3:[Playback] F4: Capture F5: All F6: Select sound card ¦ ¦ Item: S/PDIF ¦ ¦ +--+ ¦ ¦ ¦OO¦ ¦ ¦ +--+ ¦ ¦ < S/PDIF > ¦ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 aplay -L default Playback/recording through the PulseAudio sound server null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server hdmi:CARD=HDMI,DEV=0 HDA Intel HDMI, HDMI 0 HDMI Audio Output dmix:CARD=HDMI,DEV=3 HDA Intel HDMI, HDMI 0 Direct sample mixing device dsnoop:CARD=HDMI,DEV=3 HDA Intel HDMI, HDMI 0 Direct sample snooping device hw:CARD=HDMI,DEV=3 HDA Intel HDMI, HDMI 0 Direct hardware device without any conversions plughw:CARD=HDMI,DEV=3 HDA Intel HDMI, HDMI 0 Hardware device with all software conversions cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xf7d14000 irq 46 cat /proc/asound/devices 1: : sequencer 2: [ 0- 3]: digital audio playback 3: [ 0- 0]: hardware dependent 4: [ 0] : control 33: : timer mplayer -ao alsa:device=hdmi /usr/share/sounds/ubuntu/stereo/system-ready.ogg MPlayer 1.1-4.8 (C) 2000-2012 MPlayer Team mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing /usr/share/sounds/ubuntu/stereo/system-ready.ogg. libavformat version 54.20.4 (external) Mismatching header version 54.20.3 libavformat file format detected. [lavf] stream 0: audio (vorbis), -aid 0 Load subtitles in /usr/share/sounds/ubuntu/stereo/ ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders libavcodec version 54.35.0 (external) AUDIO: 44100 Hz, 1 ch, floatle, 80.0 kbit/5.67% (ratio: 10000->176400) Selected audio codec: [ffvorbis] afm: ffmpeg (FFmpeg Vorbis) ========================================================================== [AO_ALSA] alsa-lib: confmisc.c:768:(parse_card) cannot find card '1' [AO_ALSA] alsa-lib: conf.c:4248:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory [AO_ALSA] alsa-lib: confmisc.c:392:(snd_func_concat) error evaluating strings [AO_ALSA] alsa-lib: conf.c:4248:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory [AO_ALSA] alsa-lib: confmisc.c:1251:(snd_func_refer) error evaluating name [AO_ALSA] alsa-lib: conf.c:4248:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory [AO_ALSA] alsa-lib: conf.c:4727:(snd_config_expand) Evaluate error: No such file or directory [AO_ALSA] alsa-lib: pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM hdmi [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hdmi' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video Exiting... (End of file) mplayer -ao alsa:device=hw=0.3 /usr/share/sounds/ubuntu/stereo/system-ready.ogg MPlayer 1.1-4.8 (C) 2000-2012 MPlayer Team mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing /usr/share/sounds/ubuntu/stereo/system-ready.ogg. libavformat version 54.20.4 (external) Mismatching header version 54.20.3 libavformat file format detected. [lavf] stream 0: audio (vorbis), -aid 0 Load subtitles in /usr/share/sounds/ubuntu/stereo/ ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders libavcodec version 54.35.0 (external) AUDIO: 44100 Hz, 1 ch, floatle, 80.0 kbit/5.67% (ratio: 10000->176400) Selected audio codec: [ffvorbis] afm: ffmpeg (FFmpeg Vorbis) ========================================================================== [AO_ALSA] Format floatle is not supported by hardware, trying default. AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... A: 0.4 (00.4) of 0.8 (00.7) 0.1% Exiting... (End of file) Thank you for your time and help :)

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  • MP3 fingerprint tagger

    - by droberts
    Does anyone know of a tool which will read mp3 audio information directly (not the tag information), generate a fingerprint of that audio information, recommend tags based on the fingerprint and retag your MP3 collection? Last.FM released a console application which did all but retag your collection.

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  • Using Sigmatel STAC 92XX on Windows 7 RTM x64, cannot get 5.1 surround sound to work

    - by Roy Rico
    I have a Dell XPS 420. I've installed Windows 7 RTM (x64) I have this audio device: SIGMATEL STAC 92XX C-Major HD Audio I have tried using the windows 7 basic driver, and also the Vista 64-bit driver from Dell's website (details): Date: 10/29/2007 Version: 6.10.0.5511, A04 File Size: 7 MB With both drivers, I get normal stereo sound from the driver, but my 5.1 surround sound doesn't work. Has anyone experienced this? Is there a fix?

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  • Burn 24/96 flac files to play on standalone player

    - by takeshin
    I have vinyl record rip in 24/96 flac format. Each track is almost 200 MB big, so the album won't fit on CD. How to burn these files on a DVD to play with the same quality on standalone DVD player? My player supports SACD, DVD Audio and DVD video as well. My OS is Ubuntu Lucid (preferred), but I have also WinXp with Nero installed. BTW, is there any difference between DVD+ and DVD- for audio?

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  • Replace sound in another YouTube video

    - by Tom
    I have received permission from someone to translate the audio in their movies. The problem I am facing is that the video quality is quite poor and the author does not have the original videos any more. How can I replace the audio in the YouTube videos without further degrading the quality of the videos? Thanks, Tom

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  • Any plugins for Skype that support "Soundboard" usage?

    - by Axxmasterr
    I would like to find a program or plugin for Skype that allows you to pipe sound samples in to the outgoing audio stream when you are on a call. Ideally it would have some sort of soundboard functionality so that I could have a group of audio samples at the touch of a button. I'd also prefer something that supports mp3 but wav support will also do.

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  • Debian sound on hdmi instead of jack

    - by Hans de Jong
    I installed debian (gnome) and i can't get my sound working. When i use inxi -A i get the following result: Audio: Card-1: Advanced Micro Devices [AMD] nee ATI Cayman/Antilles HDMI Audio [Radeon HD 6900 Series] driver: snd_hda_intel Card-2: Advanced Micro Devices [AMD] nee ATI SBx00 Azalia (Intel HDA) driver: snd_hda_intel Sound: Advanced Linux Sound Architecture ver: 1.0.24 My feeling tells me my sound output is on the HDMI instead of my jackplug on my motherboard. How can i change this to my motherboard sound output?

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  • Which connector do I need for a "line level" subwoofer?

    - by Ben Brocka
    I've got a separate pair of speakers and I'm looking at adding a subwoofer (this, specifically). I noticed on the detail page it's inputs are listed as such: Inputs: Speaker level, line level If I'm not mistaken "line level" are the standard 3.5 audio jacks on your motherboard/sound card, right? My motherboard has the standard 6 ports for sound, if I get a subwoofer like this can I simply plug the input into the orange 3.5 jack? My audio software supports up to 7.1 so software-wise, 2.1 wouldn't be a problem.

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  • Using Sigmatel STAC 92XX on Windows 7 RTM x64, cannot get 5.1 surround sound to work

    - by Roy Rico
    I have a Dell XPS 420. I've installed Windows 7 RTM (x64) I have this audio device: SIGMATEL STAC 92XX C-Major HD Audio I have tried using the windows 7 basic driver, and also the Vista 64-bit driver from Dell's website (details): Date: 10/29/2007 Version: 6.10.0.5511, A04 File Size: 7 MB With both drivers, I get normal stereo sound from the driver, but my 5.1 surround sound doesn't work. Has anyone experienced this? Is there a fix?

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  • Changing default playback device on Windows 8

    - by emartel
    Previously, on Vista and Windows 7, changing the Default Playback device would occur instantly. For example, audio is coming out of my speakers, I right click the Volume Control, click Playback Devices then I select another device and click Set Default. Audio would be transferred immediately. Unfortunately, now, with Windows 8, I need to kill whatever process what outputting sound, and restart it for the change to take effect. Is there something that can be done about it so that changes are taken into account immediately?

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  • Synchronizing audio and video using MP4Box / ffmpeg to concatenate files

    - by jdl2003
    I have two H.264 encoded MPEG-4 files that I need to concatenate. I have been using MP4Box for this task by first ensuring the files are encoded identically (even went so far as to compare output from h264_parse on their video tracks) and then concatenating with this command: MP4Box -cat file1.mp4 -cat file2.mp4 output_file.mp4 This works and the output file is playable, but on playback in Quicktime or VLC the second video's audio starts too soon, making the entire second part of the concatenated file out of sync. I have tried reencoding the output through ffmpeg with -vcodec copy and -acodec copy but the sync issue persists. I have also tried converting to MPEG-2 format first, concatenating with a simple cat file1.mpg file2.mpg > output.mpg and reencoding the result with ffmpeg. This was even worse. I know that I can pass commands to MP4Box to adjust the start time of the audio track, but I am trying to do this programmatically for any input video (in the same encoding of course). I am looking for possible solutions that would not require human intervention / manual adjustments. Or, at least, an understanding of what is happening to make the second part of the concatenated video go out of sync.

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  • Sending microphone input over Remote Desktop 7.0

    - by Taylor Price
    I am using Remote Desktop 7 (the new version that came out with Windows 7) to control a Windows XP Pro machine. I have selected "Record from this computer" in the Remote Audio settings. When I connect to the machine, go to the control panel, open the sound panel, and go to the audio tab, I find that the default sound playback device is "Microsoft RDP Audio Driver". However, there is no default sound recording device. As expected, my IP phone thinks there is no recording device. If I am sitting in front of the computer with a mic plugged in, it works just fine. Has anybody else been able to get this work appropriately? Is there anything that I have to setup on the XP machine to get this working? Thanks in advance. Edit: As John T pointed out below, you have to be connecting to a Windows 7 Enterprise or Ultimate machine for this to work. I've also found out that Multi-monitor support has the same requirement.

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  • QuickTime Player sounds much better than iTunes

    - by Gene Goykhman
    I am playing a 320 kpbs encoded music MP3 in iTunes and the sound is substantially worse than the exact same file played back in QuickTime Player (Max OS X 10.8.5). I have maxed out system volume and iTunes playback volume. I have disabled all the audio processing features in iTunes (equalization, sound enhancer, etc.) The audio coming from iTunes still sounds resampled and/or processed, whereas QuickTime Player appears to be playing it "as is". Even when I Get Info on the MP3 file in Finder and play it back directly from the Get Info window it sounds good. It's just iTunes that seems to be mangling the song. I can notice a difference on virtually all my music, so it's not just one particular MP3. I suspect the issue is that iTunes is doing some kind of audio processing but I can't find a way to turn it off. This is the newest iTunes (11.1), but the problem has probably been going on for a while... I just switched to decent earbuds and started noticing the difference. What's the best way to force iTunes to play back the file as-is, or as close as possible to how QuickTime Player/Finder would play it?

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  • Speakers silent, headphones work in Ubuntu 9.04

    - by CarlF
    I'm running Ubuntu 9.04. Worked fine for months, then I rebooted yesterday after weeks of continuous operation. Now audio won't play through the speakers. The USB headset works fine, but the Conexant audio (CX20549) does not. Weirdly, it thinks it's playing. pavumeter shows appropriate levels, volume looks OK in alsamixer, but no sound. I did find this page: http://www.eugeneteplitsky.com/fixing-silent-pulseaudio-in-ubuntu-9-04/ Unfortunately the advice there doesn't help me. For one thing, the syntax for the alsa-base.conf file is apparently not actually documented anywhere. For another, my chipset isn't listed in the kernel.org docs he links to! EDIT: would upgrading to 9.10 help? Is there a major change in the audio subsystem between 9.04 and 9.10? Any suggestions? EDIT 2: This is stranger than I thought. Sound works normally in Xine, but is silent in Audacity, VLC and mplayer. What the?

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  • avconv gets killed if mkv has subtitles

    - by Lukas Knuth
    What I'm trying to do is to take a movie (in an Matroska container), convert all audio tracks to AC3 and don't touch anything else. I'm using this line: avconv -i infile.mkv -map 0 -vcodec copy -scodec copy -acodec ac3 -ab 256k outfile.mkv This works fine, except when there are subtitles embedded. Then, after some time processing with no progress, avconv just "dies" (output shortened, these seem to be the interesting parts): [matroska,webm @ 0xf867a0] max_analyze_duration reached [matroska,webm @ 0xf867a0] Estimating duration from bitrate, this may be inaccurate ... Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' ... Stream #0.0(eng): Video: H264 / 0x34363248, yuv420p, 1280x536 [PAR 1:1 DAR 160:67], q=2-31, 1k tbn, 1k tbc (default) Stream #0.1(ger): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s (default) Stream #0.2(eng): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s Stream #0.3(ger): Subtitle: dvdsub (default) (forced) Metadata: title : forced Stream #0.4(ger): Subtitle: dvdsub Metadata: title : complete Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (dca -> ac3) Stream #0:2 -> #0:2 (dca -> ac3) Stream #0:3 -> #0:3 (copy) Stream #0:4 -> #0:4 (copy) Input stream #0:2 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 Input stream #0:1 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 frame= 2606 fps=1303 q=-1.0 size= 3kB time=107.36 bitrate= 0.2kbits/s ... frame=96141 fps=813 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s frame=96251 fps=810 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s ... frame=97015 fps=397 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s Getötet ["Killed", in English] I have no idea why this happens, as there is no error-output. I'd like to just copy the subtitles over, not touch them at all. If that won't work, they can be completely dropped.

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  • Is ffmpeg incorrectly interpreting .aif files?

    - by marue
    Being on an Ubuntu 10.04 server i installed the ffmpeg packages with apt. ffmpeg is working afterwards, and doing as it should. Almost. For testing purposes i uploaded a few audiofiles. One of them, an aif file, is not being correctly interpreted. While on my workhorse (Mac SnowLeopard) ffmpeg tells the format as Stream #0.0: Audio: pcm_s24be, 44100 Hz, 2 channels, s32, 2116 kb/s my Ubuntu server says it is: Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s which is the wrong bitdepth. Ubuntu then fails to convert the file with the error message [pcm_s24be @ 0xcd4b580]invalid PCM packet Error while decoding stream #0.0 which certainly is not true. The file is perfectly valid. Are there any know issues for ffmpeg interpreting the aif format? How can i find out which version of the aif-codec ffmpeg is using? Any ideas how to approach this issue? ffprobe output: FFprobe version SVN-r20090707, Copyright (c) 2007-2009 Stefano Sabatini libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 built on Jan 20 2010 00:13:01, gcc: 4.4.3 20100116 (prerelease) Input #0, aiff, from 'testfile.aif': Duration: 00:00:04.00, start: 0.000000, bitrate: 2117 kb/s Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s update 2: Forcing the conversion with -sample_fmt s32 doesn't change anything. Strange thing is: Even without using -sample_fmt s32 i just realized that the conversion is working and creates valid audiofiles. There just is the error message from above.

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  • Problems with 5.1 digital out on Ubuntu 12.04

    - by user895319
    I've recently bought a new PC, installed Ubuntu and am now unable to get 5.1 digital sound working. Simple analogue stereo works fine on both the front and rear connectors. On my old box I connected the coax connection from my soundcard to my surround sound amplifier, set Settings-Sound to "Digital Stereo Duplex" and it worked. My old soundcard doesn't fit in my new machine so I'm using the built-in sound hardware. I'm connecting the combination output socket on the back of the PC via the same cable to my surround amp as before. The MB is an MSI Global H61M-P31 with an RealTek ALC887 sound chip. When I go to Settings-Sound I only see "Headphone Built-in Audio" and "Analogue Output Built-in Audio" - no digitial options. The output from aplay -l is: default Playback/recording through the PulseAudio sound server sysdefault:CARD=PCH HDA Intel PCH, ALC887-VD Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers dmix:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Hardware device with all software conversions While googling for ALC887 I've seen some references to "ALC887 -VD Analog" and some to "ALC887 -VD Digital". Does anyone know if I need to force it to chance mode somehow? It's worth mentioning that when I set the output to 5.1 digital surround in Windows 7 on the same machine I still don't get any sound so it's not a unique Linux problem. Thanks for any help.

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  • 5.1 Surround Channels are Jumbled

    - by stickynips
    I had this exact setup working previously, but after a reformat it went screwy on me. I have an Onkyo A/V Receiver hooked up to my PC, via optical S/PDIF. Attached to the receiver is a 5.1 speaker setup (tested and working fine with my Xbox via the receiver). It seems to me that the audio channels are getting mixed up somehow between the PC and the receiver. I have a 5.1 test file which plays a sounds through each speaker individually. The channels are mixed as such: "Left Front" plays through my Right Front speaker "Center" plays through my Left Front speaker "Right Front" plays through my Center speaker "Left Rear" plays through my Subwoofer "Right Rear" plays through my Left Rear speaker I've tried downloading the latest Realtek HD Audio Drivers and the Realtek HD Audio Manager, but neither makes any difference. If there's a way I can manually rearrange the channels I believe it would fix the problem, but as far as I know this is impossible. edit: Sorry, I've forgotten some basic info. I'm running Windows 7 x64. The sound card is Realtek ALC892 embedded in a GIGABYTE GA-890GPA-UD3H AM3 motherboard.

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