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Search found 167 results on 7 pages for 'pcm'.

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  • How to get espeak working?

    - by wisemonkey
    I'm trying read it loud feature of acrobat, so need a text synthesizer, I've installed espeak and libgnome-speech libraries (it didn't work for acrobat right out of the box) so when I started espeak-gui through command line it gave me segmentation fault next I tried only espeak and here is output: ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib audio/pcm_bluetooth.c:1613:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1613:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1613:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1613:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream Cannot connect to server socket err = No such file or directory Cannot connect to server socket jack server is not running or cannot be started Any ideas? or any alternative solutions for read it loud? Thanks

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  • Flash swf to play PCM WAV files?

    - by Geuis
    I am in need of a Flash swf that is capable of loading PCM WAV files via a url passed to it. An example of the file can be found here: http://msdn.microsoft.com/en-us/library/ff512405.aspx The swf does not need a visible interface, as it is meant for audio playing only and not user interaction. The swf should have a simple javascript interface for page-level interactions. These include: load(url): loads a PCM WAV file over the internet using the url that is passed through it. play(): Should play the PCM WAV file that was loaded stop(): Should stop playing the current file. I can provide a sample audio file that matches the specifications if the developer is unable to obtain a url from the link posted above.

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  • Android PCM Bytes

    - by Pintac
    Hi I am using the AudioRecord class to analize raw pcm bytes as it comes in the mic. So thats working nicely. Now i need convert the pcm bytes into decibel. I have a formula that takes sound presure in Pa into db. db = 20 * log10(Pa/ref Pa) So the question is the bytes i am getting from audiorecorder from the buffer what is it is it amplitude pascal sound pressure or what. I tried to putting the value into te formula but it comes back with very hight db so i do not think its right thanks

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  • Audio output from Silverlight

    - by leecarter
    I'm looking to develop a Silverlight application which will take a stream of data (not an audio stream as such) from a web server. The data stream would then be manipulated to give audio of a certain format (G.711 a-Law for example) which would then be converted into PCM so that additional effects can be applied (such as boosting the volume). I'm OK up to this point. I've got my data, converted the G.711 into PCM but my problem is being able to output this PCM audio to the sound card. I basing a solution on some C# code intended for a .Net application but in Silverlight there is a problem with trying to take a copy of a delegate (function pointer) which will be the topic of a separate question once I've produced a simple code sample. So, the question is... How can I output the PCM audio that I have held in a data structure (currently an array) in my Silverlight to the user? (Please don't say write the byte values to a text box) If it were a MP3 or WMA file I would play it using a MediaElement but I don't want to have to make it into a file as this would put a crimp on applying dynamic effects to the audio. I've seen a few posts from people saying low level audio support is poor/non-existant in Silverlight so I'm open to any suggestions/ideas people may have.

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  • check_mk IPMI PCM sensor reading randomly fails

    - by Julian Kessel
    I use check_mk_agent for monitoring a server with IPMI and the freeipmi-tools installed. As far as I can see, the monitoring randomly detects no value returned by the IPMI Sensor "Temperature_PCH_Temp". That's a problem since it results in a CRITICAL state triggering a notification. The interruption lasts only over one check, the following is always OK. The temperature is in no edge area and neither the readings before the fail nor after show a Temp that is tending to overrun a treshold. Has someone an idea on what could be the reason for this behaviour and how prevent it?

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  • android playback suddenly stopping...

    - by user306517
    I have an app that is streaming audio content and sometimes it just stops all of the suddent. the logcat windows shows -- AudioHardware pcm playback is going to standby and that's it. I saw on another thread (pun intended) that someone was saying it was because he was using too many threads. Could that really be causing this? Could i give the audio thread higher priority? Anyway to prevent the audio hardware pcm from going to standby?

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  • Getting level values from PCM raw data using Core Audio

    - by John
    I am trying to extract level data from a PCM audio file using core audio. I have gotten as far as (I believe) getting the raw data into a byte array (UInt8) but it is 16 bit PCM data and I am having trouble reading the data out. The input is from the iPhone microphone, which I have set as: [recordSetting setValue:[NSNumber numberWithInt:kAudioFormatLinearPCM] forKey:AVFormatIDKey]; [recordSetting setValue:[NSNumber numberWithFloat:44100.0] forKey:AVSampleRateKey]; [recordSetting setValue:[NSNumber numberWithInt:1] forKey:AVNumberOfChannelsKey]; [recordSetting setValue:[NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey]; [recordSetting setValue:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsBigEndianKey]; [recordSetting setValue:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsFloatKey]; which is obviously 16 bits. I am then trying to just print out a few values to see if they look reasonable for debug purposes below, and they do not look reasonable (many 0's). ExtAudioFileRef inputFile = NULL; ExtAudioFileOpenURL(track.location, &inputFile); AudioStreamBasicDescription inputFileFormat; UInt32 dataSize = (UInt32)sizeof(inputFileFormat); ExtAudioFileGetProperty(inputFile, kExtAudioFileProperty_FileDataFormat, &dataSize, &inputFileFormat); UInt8 *buffer = malloc(BUFFER_SIZE); AudioBufferList bufferList; bufferList.mNumberBuffers = 1; bufferList.mBuffers[0].mNumberChannels = 1; bufferList.mBuffers[0].mData = buffer; //pointer to buffer of audio data bufferList.mBuffers[0].mDataByteSize = BUFFER_SIZE; //number of bytes in the buffer while(true) { UInt32 frameCount = (bufferList.mBuffers[0].mDataByteSize / inputFileFormat.mBytesPerFrame); // Read a chunk of input OSStatus status = ExtAudioFileRead(inputFile, &frameCount, &bufferList); // If no frames were returned, conversion is finished if(0 == frameCount) break; NSLog(@"---"); int16_t *bufferl = &buffer; for(int i=0;i<100;i++){ //const int16_t *bufferl = bufferl[i]; NSLog(@"%d",bufferl[i]); } } Not sure what I am doing wrong, I think it has to do with reading the byte array. Sorry for the long code post...

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  • Steaming a non-PCM WAV file to a SilverLight application

    - by Satumba
    Hi, I would like to allow users to play recorded WAV files that stored on a server back to a Silverlight application as a client to play them. I saw that there is a way to play a WAV file on Silverlight (here), but when i tried to impliment it, i got an error playing the file because it is not in PCM format but encoded. The files that i'm trying to play are encoded with a special encoder, so i thought that the only way is to decode the WAV file on the server and stream it back to the client. The limitation is that the decode process should occur in real time because it is not reasonable to convert all the WAV files that exists. Is it possible to do it? Which streamer can i use? (Windows Media Service can help here?) Does somebody has any experience with such a scenario? Appreciate your help.

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  • Making a DVD video with a still image and PCM 16bit audio with ffmpeg

    - by João
    I'm trying to make a small video with a still image and a sound file playing in the background to pass it to dvdauthor and create a DVD. The command I'm using is this: ffmpeg -loop_input -i image.jpg -qscale 2 -i song.flac -aspect 4:3 -target pal-dvd -acodec pcm_s16le -shortest output.mpg However, the resulting video file doesn't have sound at all (testing it on VLC Player). I don't know if I can't combine "-acodec pcm_s16le" with "-target pal-dvd" to override the later, or if there is something else wrong with the command. If I try without the "-acodec pcm_s16le" parameter the video and audio works, I can even create a DVD ISO with it. However, the audio stays as AC3. I wanted to include with the video the lossless audio, not a compressed one. I suppose the DVD standart allows to have PCM audio in it, am I right?

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  • Streaming a non-PCM WAV file to a SilverLight application

    - by Satumba
    Hi, I would like to allow users to play recorded WAV files that stored on a server back to a Silverlight application as a client to play them. I saw that there is a way to play a WAV file on Silverlight (here), but when i tried to impliment it, i got an error playing the file because it is not in PCM format but encoded. The files that i'm trying to play are encoded with a special encoder, so i thought that the only way is to decode the WAV file on the server and stream it back to the client. The limitation is that the decode process should occur in real time because it is not reasonable to convert all the WAV files that exists. Is it possible to do it? Which streamer can i use? (Windows Media Service can help here?) Does somebody has any experience with such a scenario? Appreciate your help.

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  • Resampling audio output for A2DP (from PCM WAV)

    - by user1669982
    The question is how to bring stereo PCM WAV 32,000 Hz with a stream of 1024 kbps (125 KB) to the headset with Bluetooth 2.1 on a CM7 smartphone with DSPManager. Is it possible? SBC is really bad idea. To TJD: Because it compresses the compressed stream. My Epic 4G don`t have Apt-X support. My headset Gemix BH-04A yellow. May be its possible with the Headset Profile (HSP)? I dont know about supported codecs in this profile.

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  • Another guvcview issue? Core dump!

    - by user290498
    guvcview is core dumping with my Sony Handycam plugged in. It works fine with my standard Logitech Webcam. I deleted the config file so it could re-create it. here is the output: $ guvcview guvcview 1.5.3 Could not open /home/rayj/.guvcviewrc for read, will try to create it write /home/rayj/.guvcviewrc OK ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream Cannot connect to server socket err = No such file or directory Cannot connect to server socket jack server is not running or cannot be started video device: /dev/video0 ERROR opening V4L2 interface for /dev/video1 Init. stk1160 (location: usb-0000:00:1d.0-1.5) { pixelformat = 'UYVY', description = '16 bpp YUY2, 4:2:2, packed' } VIDIOC_ENUM_FRAMESIZES - Error enumerating frame sizes: Inappropriate ioctl for device Unable to enumerate frame sizes. : Inappropriate ioctl for device { pixelformat = 'RGB3', description = 'RGB3' } { ?GSPCA? : width = 720, height = 480 } fmtind:2 fsizeind: 1 { pixelformat = 'BGR3', description = 'BGR3' } { ?GSPCA? : width = 720, height = 480 } fmtind:3 fsizeind: 1 { pixelformat = 'YU12', description = 'YU12' } { ?GSPCA? : width = 720, height = 480 } fmtind:4 fsizeind: 1 { pixelformat = 'YV12', description = 'YV12' } { ?GSPCA? : width = 720, height = 480 } fmtind:5 fsizeind: 1 vid:05e1 pid:0408 driver:stk1160 checking format: 1196444237 Format unavailable: 1196444237. Init v4L2 failed !! Init video returned -2 trying minimum setup ... Segmentation fault (core dumped)

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  • Static background noise while using new headset Ubuntu 13.04

    - by ThundLayr
    Today I bought a new gaming headset (Gx-Gaming Lychas), and when I tried to record some gameplay-comentary I noticed that there always is a static background noise, I just recorded an example so you guys can listen it (no downloaded needed): http://www47.zippyshare.com/v/65167832/file.html I'm using Kubuntu 13.04 and Kernel version is 3.8.0-19, my laptop is an Acer Travelmate 5760Z, I tried tons of configurations on Alsamixer and none of them made result, I really need to get this working so any kind of help will be very aprecciated. cat /proc/asound/cards: 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xc6400000 irq 44 cat /proc/asound/card0/codec#0 Codec: Conexant CX20588 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x14f1506c Subsystem Id: 0x10250574 Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 D3cold CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20588 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x4a 0x4a] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x80 0x80] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0f, mute=0 Amp-Out vals: [0x00] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="CX20588 Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x50 0x50] [0x80 0x80] [0x80 0x80] [0x80 0x80] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x16 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x17 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x04 0x04] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a 0x1b* 0x1d 0x1e Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a* 0x1b 0x1d 0x1e Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Control: name="Headphone Jack", index=0, device=0 Pincap 0x0000001c: OUT HP Detect Pin Default 0x04214040: [Jack] HP Out at Ext Right Conn = 1/8, Color = Green DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1a [Pin Complex] wcaps 0x400481: Stereo Control: name="Internal Mic Phantom Jack", index=0, device=0 Pincap 0x00001324: IN Detect Vref caps: HIZ 50 80 Pin Default 0x90a70130: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Control: name="Mic Jack", index=0, device=0 Pincap 0x00011334: IN OUT EAPD Detect Vref caps: HIZ 50 80 EAPD 0x0: Pin Default 0x04a19020: [Jack] Mic at Ext Right Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1c [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1d [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x0: Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1e [Pin Complex] wcaps 0x400481: Stereo Pincap 0x00000024: IN Detect Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1f [Pin Complex] wcaps 0x400501: Stereo Control: name="Speaker Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11* Node 0x20 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x22 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x21 Node 0x23 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x24 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11 Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono

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  • Using Audio Queue Services to play PCM data over a socket connection

    - by Rohan
    I'm writing a remote desktop client for the iPhone and I'm trying to implement audio redirection. The client is connected to the server over a socket connection, and the server sends 32K chunks of PCM data at a time. I'm trying to use AQS to play the data and it plays the first two seconds (1 buffer worth). However, since the next chunk of data hasn't come in over the socket yet, the next AudioQueueBuffer is empty. When the data comes in, I fill the next available buffer with the data and enqueue it with AudioQueueEnqueueBuffer. However, it never plays these buffers. Does the queue stop playing if there are no buffers in the queue, even if you later add a buffer? Here's the relevant part of the code: void wave_out_write(STREAM s, uint16 tick, uint8 index) { if(items_in_queue == NUM_BUFFERS){ return; } if(!playState.busy){ OSStatus status; status = AudioQueueNewOutput(&playState.dataFormat, AudioOutputCallback, &playState, CFRunLoopGetCurrent(), NULL, 0, &playState.queue); if(status == 0){ for(int i=0; i<NUM_BUFFERS; i++){ AudioQueueAllocateBuffer(playState.queue, 40000, &playState.buffers[i]); } AudioQueueAddPropertyListener(playState.queue, kAudioQueueProperty_IsRunning, MyAudioQueuePropertyListenerProc, &playState); status = AudioQueueStart(playState.queue, NULL); if(status ==0){ playState.busy = True; } else{ return; } } else{ return; } } playState.buffers[queue_hi]->mAudioDataByteSize = s->size; memcpy(playState.buffers[queue_hi]->mAudioData, s->data, s->size); AudioQueueEnqueueBuffer(playState.queue, playState.buffers[queue_hi], 0, 0); queue_hi++; queue_hi = queue_hi % NUM_BUFFERS; items_in_queue++; } void AudioOutputCallback(void* inUserData, AudioQueueRef outAQ, AudioQueueBufferRef outBuffer) { PlayState *playState = (PlayState *)inUserData; items_in_queue--; } Thanks!

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  • AS3 microphone recording/saving works, in-flash PCM playback double speed

    - by Lowgain
    I have a working mic recording script in AS3 which I have been able to successfully use to save .wav files to a server through AMF. These files playback fine in any audio player with no weird effects. For reference, here is what I am doing to capture the mic's ByteArray: (within a class called AudioRecorder) public function startRecording():void { _rawData = new ByteArray(); _microphone.addEventListener(SampleDataEvent.SAMPLE_DATA, _samplesCaptured, false, 0, true); } private function _samplesCaptured(e:SampleDataEvent):void { _rawData.writeBytes(e.data); } This works with no problems. After the recording is complete I can take the _rawData variable and run it through a WavWriter class, etc. However, if I run this same ByteArray as a sound using the following code which I adapted from the adobe cookbook: (within a class called WavPlayer) public function playSound(data:ByteArray):void { _wavData = data; _wavData.position = 0; _sound.addEventListener(SampleDataEvent.SAMPLE_DATA, _playSoundHandler); _channel = _sound.play(); _channel.addEventListener(Event.SOUND_COMPLETE, _onPlaybackComplete, false, 0, true); } private function _playSoundHandler(e:SampleDataEvent):void { if(_wavData.bytesAvailable <= 0) return; for(var i:int = 0; i < 8192; i++) { var sample:Number = 0; if(_wavData.bytesAvailable > 0) sample = _wavData.readFloat(); e.data.writeFloat(sample); } } The audio file plays at double speed! I checked recording bitrates and such and am pretty sure those are all correct, and I tried changing the buffer size and whatever other numbers I could think of. Could it be a mono vs stereo thing? Hope I was clear enough here, thanks!

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  • Downsampling and applying a lowpass filter to digital audio

    - by twk
    I've got a 44Khz audio stream from a CD, represented as an array of 16 bit PCM samples. I'd like to cut it down to an 11KHz stream. How do I do that? From my days of engineering class many years ago, I know that the stream won't be able to describe anything over 5500Hz accurately anymore, so I assume I want to cut everything above that out too. Any ideas? Thanks. Update: There is some code on this page that converts from 48KHz to 8KHz using a simple algorithm and a coefficient array that looks like { 1, 4, 12, 12, 4, 1 }. I think that is what I need, but I need it for a factor of 4x rather than 6x. Any idea how those constants are calculated? Also, I end up converting the 16 byte samples to floats anyway, so I can do the downsampling with floats rather than shorts, if that helps the quality at all.

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  • AudioConverterConvertBuffer problem with insz error

    - by Samuel
    Hi Codegurus, I have a problem with the this function AudioConverterConvertBuffer. Basically I want to convert from this format _ streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked |0 ; _streamFormat.mBitsPerChannel = 16; _streamFormat.mChannelsPerFrame = 2; _streamFormat.mBytesPerPacket = 4; _streamFormat.mBytesPerFrame = 4; _streamFormat.mFramesPerPacket = 1; _streamFormat.mSampleRate = 44100; _streamFormat.mReserved = 0; to this format _streamFormatOutput.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked|0 ;//| kAudioFormatFlagIsNonInterleaved |0; _streamFormatOutput.mBitsPerChannel = 16; _streamFormatOutput.mChannelsPerFrame = 1; _streamFormatOutput.mBytesPerPacket = 2; _streamFormatOutput.mBytesPerFrame = 2; _streamFormatOutput.mFramesPerPacket = 1; _streamFormatOutput.mSampleRate = 44100; _streamFormatOutput.mReserved = 0; and what i want to do is to extract an audio channel(Left channel or right channel) from an LPCM buffer based on the input format to make it mono in the output format. Some logic code to convert is as follows This is to set the channel map for PCM output file SInt32 channelMap[1] = {0}; status = AudioConverterSetProperty(converter, kAudioConverterChannelMap, sizeof(channelMap), channelMap); and this is to convert the buffer in a while loop AudioBufferList audioBufferList; CMBlockBufferRef blockBuffer; CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampBuffer, NULL, &audioBufferList, sizeof(audioBufferList), NULL, NULL, 0, &blockBuffer); for (int y=0; y<audioBufferList.mNumberBuffers; y++) { AudioBuffer audioBuffer = audioBufferList.mBuffers[y]; //frames = audioBuffer.mData; NSLog(@"the number of channel for buffer number %d is %d",y,audioBuffer.mNumberChannels); NSLog(@"The buffer size is %d",audioBuffer.mDataByteSize); numBytesIO = audioBuffer.mDataByteSize; convertedBuf = malloc(sizeof(char)*numBytesIO); status = AudioConverterConvertBuffer(converter, audioBuffer.mDataByteSize, audioBuffer.mData, &numBytesIO, convertedBuf); char errchar[10]; NSLog(@"status audio converter convert %d",status); if (status != 0) { NSLog(@"Fail conversion"); assert(0); } NSLog(@"Bytes converted %d",numBytesIO); status = AudioFileWriteBytes(mRecordFile, YES, countByteBuf, &numBytesIO, convertedBuf); NSLog(@"status for writebyte %d, bytes written %d",status,numBytesIO); free(convertedBuf); if (numBytesIO != audioBuffer.mDataByteSize) { NSLog(@"Something wrong in writing"); assert(0); } countByteBuf = countByteBuf + numBytesIO; But the insz problem is there... so it cant convert. I would appreciate any input Thanks in advance

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  • Where to get pure C++ Lame MP3 encoder - PCM to MP3 example?

    - by Ole Jak
    So all I need is a simple function that sets it up (eating incoming PCM RATE (for example: rate near to 44100) It's channels (for example: 2) and -bits (for example: 16) and desirable 128 kb\s rate) and another one that takes PCM data and encodes it into pure MP3 frames. I know it looks like a silly homework task but I assure you - it is not. I hope it will be of help to all C++ developers starting with MP3s. So can anybody please help me with that?

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  • Integrated webcam in lenovo t410 not working with 12.04

    - by kristianp
    I have a Lenovo T410 with an inbuilt webcam and I haven't been able to get the webcam working. I tried skype, cheese, both just give me a black window. The microphone works fine with skype, by the way. Can anyone provide any clues please? The webcam is enabled in the bios, but there is no light indicating the webcam is on (not sure if there should be, though). I tried this on Kubuntu 11.10 and have upgraded to 12.04 with the same results. The Fn-F6 keyboard combination doens't seem to do anything either. EDIT: I got the webcam replaced, it looks like it was a hardware problem, because it works fine now. Thanks guys. $ ls /dev/v4l/* /dev/v4l/by-id: usb-Chicony_Electronics_Co.__Ltd._Integrated_Camera-video-index0 /dev/v4l/by-path: pci-0000:00:1a.0-usb-0:1.6:1.0-video-index0 And lsusb: $ lsusb Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 001 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 003: ID 147e:2016 Upek Biometric Touchchip/Touchstrip Fingerprint Sensor Bus 001 Device 004: ID 0a5c:217f Broadcom Corp. Bluetooth Controller Bus 001 Device 005: ID 17ef:480f Lenovo Integrated Webcam [R5U877] Bus 002 Device 003: ID 05c6:9204 Qualcomm, Inc. Bus 002 Device 004: ID 17ef:1003 Lenovo Integrated Smart Card Reader Here is the output from guvcview, minus lots of lines describing the available capture formats. It says "unable to start with minimum setup. Please reconnect your camera.". guvcview 1.5.3 ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave Cannot connect to server socket err = No such file or directory Cannot connect to server socket jack server is not running or cannot be started video device: /dev/video0 Init. Integrated Camera (location: usb-0000:00:1a.0-1.6) { pixelformat = 'YUYV', description = 'YUV 4:2:2 (YUYV)' } { discrete: width = 640, height = 480 } Time interval between frame: 1/30, .... { discrete: width = 1600, height = 1200 } Time interval between frame: 1/15, vid:17ef pid:480f driver:uvcvideo checking format: 1196444237 libv4l2: error setting pixformat: Device or resource busy VIDIOC_S_FORMAT - Unable to set format: Device or resource busy Init v4L2 failed !! Init video returned -2 trying minimum setup ... video device: /dev/video0 Init. Integrated Camera (location: usb-0000:00:1a.0-1.6) { pixelformat = 'YUYV', description = 'YUV 4:2:2 (YUYV)' } { discrete: width = 640, height = 480 } .... vid:17ef pid:480f driver:uvcvideo checking format: 1448695129 libv4l2: error setting pixformat: Device or resource busy VIDIOC_S_FORMAT - Unable to set format: Device or resource busy Init v4L2 failed !! ERROR: Minimum Setup Failed. Exiting... VIDIOC_REQBUFS - Failed to delete buffers: Invalid argument (errno 22) cleaned allocations - 100% Closing portaudio ...OK Terminated.

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  • JNI - GetObjectField returns NULL

    - by Daniel
    I'm currently working on Mangler's Android implementation. I have a java class that looks like so: public class VentriloEventData { public short type; public class _pcm { public int length; public short send_type; public int rate; public byte channels; }; _pcm pcm; } The signature for my pcm object: $ javap -s -p VentriloEventData ... org.mangler.VentriloEventData$_pcm pcm; Signature: Lorg/mangler/VentriloEventData$_pcm; I am implementing a native JNI function called getevent, which will write to the fields in an instance of the VentriloEventData class. For what it's worth, it's defined and called in Java like so: public static native int getevent(VentriloEventData data); VentriloEventData data = new VentriloEventData(); getevent(data); And my JNI implementation of getevent: JNIEXPORT jint JNICALL Java_org_mangler_VentriloInterface_getevent(JNIEnv* env, jobject obj, jobject eventdata) { v3_event *ev = v3_get_event(V3_BLOCK); if(ev != NULL) { jclass event_class = (*env)->GetObjectClass(env, eventdata); // Event type. jfieldID type_field = (*env)->GetFieldID(env, event_class, "type", "S"); (*env)->SetShortField( env, eventdata, type_field, 1234 ); // Get PCM class. jfieldID pcm_field = (*env)->GetFieldID(env, event_class, "pcm", "Lorg/mangler/VentriloEventData$_pcm;"); jobject pcm = (*env)->GetObjectField( env, eventdata, pcm_field ); jclass pcm_class = (*env)->GetObjectClass(env, pcm); // Set PCM fields. jfieldID pcm_length_field = (*env)->GetFieldID(env, pcm_class, "length", "I"); (*env)->SetIntField( env, pcm, pcm_length_field, 1337 ); free(ev); } return 0; } The code above works fine for writing into the type field (that is not wrapped by the _pcm class). Once getevent is called, data.type is verified to be 1234 at the java side :) My problem is that the assertion "pcm != NULL" will fail. Note that pcm_field != NULL, which probably indicates that the signature to that field is correct... so there must be something wrong with my call to GetObjectField. It looks fine though if I compare it to the official JNI docs. Been bashing my head on this problem for the past 2 hours and I'm getting a little desperate.. hoping a different perspective will help me out on this one.

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  • Limit PulseAudio volume to prevent overdrive

    - by el.pescado
    Is it possible to limit maximum volume in PulseAudio? Currently, PulseAudio sets PCM channel too loud which results in distorted sound. I use aumix to turn volume down, but whenever any other sound is played (IM notification etc), PA plays with knobs, turning master volume down and PCM up. aumix ++++++++++++++++++++O+++++<Vol ++++++++++++++++++++O+++++ Pcm ++++++++++++O+++++++++++++ becomes: aumix ++++++++++++++++O+++++++++<Vol ++++++++++++++++++++++++O+ Pcm ++++++++++++O+++++++++++++ I use OpenSUSE 11.2, pulseaudio 0.9.21, ALSA 1.0.21 and ALC889A sound codec.

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  • hdmi AC-3 audio broke after upgrading from 11.10 to 12.04.3

    - by Jim LastName
    I just updated my MythBuntu 11.10 to 12.04.3. Now, when I try to play 5.1 content (ripped DVD), my TV (and receiver) plays a "chattering" sound. I check my receiver and the digital dolby light isn't on--it's in PCM mode. So, either the audio is getting sent as AC-3, but the TV and receiver think it's PCM or the AC-3 audio got converted to multichannel PCM and they can't handle it. My setup: hdmi cable from htpc to TV. TV has an s/pdif output to my receiver. I know TV sends AC-3 audio out correctly because I see digital dolby light come on when I view a digital TV channel and PCM come on when I view an old analog channel. I can connect s/pdif from my htpc to my receiver and the digital dolby light comes on and it can decode the audio just fine. It's just not sending it right over hdmi. Now for some hints to the issue: I noticed in MythTV audio setup when I select alsa:hdmi.... the description only lists 2 channel PCM audio capability. speaker-test -Dhdmi:PCH -c6 errors about a bad channel count (only -c2 works). Finally, I tried vlc and it does the same chattering sound. These all make me think this isn't a MythTV issue, it's something lower than that. I think the best way to troubleshoot this is to start at the drivers and check each layer, one at a time all the way to alsa. I just don't know what the layers are and how to do it. So, I need to find some audio troubleshooting guide to assist me. Or, if one doesn't exist, I'd appreciate some steps. Thanks much, Jim

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  • Opening Skype, Opera, OpenOffice logs me off

    - by anjanesh
    Whats common among Skype, Opera, OpenOffice in Ubuntu ? Whenever I open these applications I get logged off and shows back me the login screen. This started happening since the 10.10 upgrade. Forgot to mention : Yes, its x64.Each time I open these applications, the UI shows and then crashes. I started each app & logged the last few lines of /var/log/syslog after each crash. Looks like something to do with sound drivers ? Opera :Jan 8 09:33:20 al-ubuntu pulseaudio[11532]: pid.c: Daemon already running. Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: snd_pcm_avail_delay() returned strange values: delay 0 is less than avail 8. Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: snd_pcm_dump(): Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: Soft volume PCM Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: Control: PCM Playback Volume Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: min_dB: -51 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: max_dB: 0 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: resolution: 256 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: Its setup is: Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: stream : CAPTURE Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: access : MMAP_INTERLEAVED Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: format : S16_LE Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: subformat : STD Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: channels : 2 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: rate : 44100 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: exact rate : 44100 (44100/1) Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: msbits : 16 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: buffer_size : 88192 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: period_size : 44096 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: period_time : 999909 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: tstamp_mode : ENABLE Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: period_step : 1 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: avail_min : 87310 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: period_event : 0 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: start_threshold : -1 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: stop_threshold : 6205960286516543488 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: silence_threshold: 0 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: silence_size : 0 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: boundary : 6205960286516543488 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: Its setup is: Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: stream : CAPTURE Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: access : MMAP_INTERLEAVED Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: format : S16_LE Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: subformat : STD Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: channels : 2 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: rate : 44100 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: exact rate : 44100 (44100/1) Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: msbits : 16 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: buffer_size : 88192 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: period_size : 44096 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: period_time : 999909 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: tstamp_mode : ENABLE Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: period_step : 1 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: avail_min : 87310 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: period_event : 0 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: start_threshold : -1 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: stop_threshold : 6205960286516543488 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: silence_threshold: 0 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: silence_size : 0 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: boundary : 6205960286516543488 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: appl_ptr : 87320 Jan 8 09:33:21 al-ubuntu pulseaudio[11429]: alsa-util.c: hw_ptr : 87320 Jan 8 09:33:22 al-ubuntu kernel: [ 4962.078306] opera[11036]: segfault at 261 ip 0000000000000261 sp 00007fffed7cd9a8 error 14 in opera[400000+122b000] anjanesh@al-ubuntu:~$ SkypeJan 8 09:40:21 al-ubuntu pulseaudio[12602]: pid.c: Daemon already running. Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: snd_pcm_avail_delay() returned strange values: delay 0 is less than avail 8. Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: snd_pcm_dump(): Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: Soft volume PCM Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: Control: PCM Playback Volume Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: min_dB: -51 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: max_dB: 0 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: resolution: 256 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: Its setup is: Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: stream : CAPTURE Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: access : MMAP_INTERLEAVED Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: format : S16_LE Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: subformat : STD Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: channels : 2 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: rate : 44100 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: exact rate : 44100 (44100/1) Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: msbits : 16 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: buffer_size : 88192 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: period_size : 44096 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: period_time : 999909 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: tstamp_mode : ENABLE Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: period_step : 1 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: avail_min : 87310 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: period_event : 0 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: start_threshold : -1 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: stop_threshold : 6205960286516543488 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: silence_threshold: 0 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: silence_size : 0 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: boundary : 6205960286516543488 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: Its setup is: Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: stream : CAPTURE Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: access : MMAP_INTERLEAVED Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: format : S16_LE Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: subformat : STD Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: channels : 2 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: rate : 44100 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: exact rate : 44100 (44100/1) Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: msbits : 16 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: buffer_size : 88192 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: period_size : 44096 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: period_time : 999909 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: tstamp_mode : ENABLE Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: period_step : 1 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: avail_min : 87310 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: period_event : 0 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: start_threshold : -1 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: stop_threshold : 6205960286516543488 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: silence_threshold: 0 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: silence_size : 0 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: boundary : 6205960286516543488 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: appl_ptr : 87312 Jan 8 09:40:23 al-ubuntu pulseaudio[12485]: alsa-util.c: hw_ptr : 87312 anjanesh@al-ubuntu:~$ Open OfficeJan 8 09:43:46 al-ubuntu pulseaudio[13157]: pid.c: Daemon already running. Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: snd_pcm_avail_delay() returned strange values: delay 0 is less than avail 16. Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: snd_pcm_dump(): Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: Soft volume PCM Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: Control: PCM Playback Volume Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: min_dB: -51 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: max_dB: 0 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: resolution: 256 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: Its setup is: Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: stream : CAPTURE Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: access : MMAP_INTERLEAVED Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: format : S16_LE Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: subformat : STD Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: channels : 2 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: rate : 44100 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: exact rate : 44100 (44100/1) Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: msbits : 16 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: buffer_size : 88192 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: period_size : 44096 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: period_time : 999909 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: tstamp_mode : ENABLE Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: period_step : 1 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: avail_min : 87310 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: period_event : 0 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: start_threshold : -1 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: stop_threshold : 6205960286516543488 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: silence_threshold: 0 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: silence_size : 0 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: boundary : 6205960286516543488 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: Its setup is: Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: stream : CAPTURE Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: access : MMAP_INTERLEAVED Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: format : S16_LE Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: subformat : STD Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: channels : 2 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: rate : 44100 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: exact rate : 44100 (44100/1) Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: msbits : 16 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: buffer_size : 88192 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: period_size : 44096 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: period_time : 999909 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: tstamp_mode : ENABLE Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: period_step : 1 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: avail_min : 87310 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: period_event : 0 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: start_threshold : -1 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: stop_threshold : 6205960286516543488 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: silence_threshold: 0 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: silence_size : 0 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: boundary : 6205960286516543488 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: appl_ptr : 87320 Jan 8 09:43:48 al-ubuntu pulseaudio[13064]: alsa-util.c: hw_ptr : 87320 anjanesh@al-ubuntu:~$

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