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  • Concatenation of a 2 second silence audio with a normal audio not working

    - by user1665130
    I have a code for concatenation of files using ffmpeg.Here silence.wav is a mute audio file with 2 seconds length. I need to prepend this mut audio file to REC00096_Jun-06-2014 16.47.28.wav. I tried the folowing code. ffmpeg -i D:\vishnu\silence.wav -i D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav \-filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' \-map '[out]' output.wav Following is the error i am getting. D:\vishnu>ffmpeg -i silence.wav -i "D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav" -filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' -map '[out]' outp ut.wav ffmpeg version N-59036-g5d8e4f6 Copyright (c) 2000-2013 the FFmpeg developers built on Dec 12 2013 22:01:01 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 58.100 / 52. 58.100 libavcodec 55. 45.101 / 55. 45.101 libavformat 55. 22.100 / 55. 22.100 libavdevice 55. 5.102 / 55. 5.102 libavfilter 3. 92.100 / 3. 92.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Input #0, wav, from 'silence.wav': Metadata: encoder : Lavf55.22.100 Duration: 00:00:02.02, bitrate: 4234 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 5.1, s16, 4 233 kb/s Guessed Channel Layout for Input Stream #1.0 : mono Input #1, wav, from 'D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav': Duration: 00:00:08.04, bitrate: 384 kb/s Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 24000 Hz, mono, s16, 384 kb/s [wav @ 036f5e40] Invalid stream specifier: '[out]'. Last message repeated 1 times Stream map ''[out]'' matches no streams. D:\vishnu>

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  • Sound/Silence in a wav file.

    - by Vivek
    Hi, I am searching for a utility/code that could detect and let me know if my 1 minute wav file contains sound or not ? Other way, if it could detect the duration of the silence(if exists) at any position in the wav file, that would also server the purpose. Does SOX support any command for that ? I tried with Java, but didnt found anything in JMF. Thanks Vivek

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  • How do I silence the following RightAWS messages when running tests

    - by Laurie Young
    I'm using the RighAWS gem, and mocking at the http level so that the RightAWS code is being executed as part of my tests. When this happens I get the following output ....New RightAws::S3Interface using per_request-connection mode Opening new HTTP connection to s3.amazonaws.com:80 .New RightAws::S3Interface using per_request-connection mode . Even though all the tests pass, when I do have errors its harder to scan them because of this output. is there a nice way to silence it?

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  • Silence output from SimpleXMLRPCServer

    - by Corey Goldberg
    I am running an xml-rpc server using SimpleXMLRPCServer from the stdlib. My code looks something like this: import SimpleXMLRPCServer import socket class RemoteStarter: def start(self): return 'foo' rs = RemoteStarter() host = socket.gethostbyaddr(socket.gethostname())[0] port = 9000 server = SimpleXMLRPCServer.SimpleXMLRPCServer((host, port)) server.register_instance(rs) server.serve_forever() every time the 'start' method gets called remotely, the server prints an access line like this: <server_name> - - [10/Mar/2010 13:06:20] "POST /RPC2 HTTP/1.0" 200 - I can't figure out a way to silence the output so it doesn't print these access lines to stdout. anyone?

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  • How to produce precisely-timed tone and silence in C#

    - by Bob Denny
    I have a C# project that plays Morse code for RSS feeds. I write it using Managed DirectX, only to discover that Managed DirectX is old and deprecated. The task I have is to play pure sine wave bursts interspersed with silence periods (the code) which are precisely timed as to their duration. I need to be able to call a function which plays a pure tone for so many milliseconds, then Thread.Sleep() then play another, etc. At its fastest, the tones and spaces can be as short as 40ms. It's working quite well in Managed DirectX. To get the precisely timed tone I create 1 sec. of sine wave into a secondary buffer, then to play a tone of a certain duration I seek forward to within x milliseconds of the end of the buffer then play. I've tried System.Media.SoundPlayer. It's a loser because you have to Play(), Sleep(), then Stop() for arbitrary tone lengths. The result is a tone that is too long, variable by CPU load. It takes an indeterminate amount of time to actually stop the tone. I then embarked on a lengthy attempt to use NAudio 1.3. I ended up with a memory resident stream providing the tone data, and again seeking forward leaving the desired length of tone remaining in the stream, then playing. This worked OK on the DirectSoundOut class for a while (see below) but the WaveOut class quickly dies with an internal assert saying that buffers are still on the queue despite PlayerStopped = true. This is odd since I play to the end then put a wait of the same duration between the end of the tone and the start of the next. You'd think that 80ms after starting Play of a 40 ms tone that it wouldn't have buffers on the queue. DirectSoundOut works well for a while, but its problem is that for every tone burst Play() it spins off a separate thread. Eventually (5 min or so) it just stops working. You can see thread after thread after thread exiting in the Output window while running the project in VS2008 IDE. I don't create new objects during playing, I just Seek() the tone stream then call Play() over and over, so I don't think it's a problem with orphaned buffers/whatever piling up till it's choked. I'm out of patience on this one, so I'm asking in the hopes that someone here has faced a similar requirement and can steer me in a direction with a likely solution. Thanks in advance...

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  • How to produce precisely-timed tone and silence?

    - by Bob Denny
    I have a C# project that plays Morse code for RSS feeds. I write it using Managed DirectX, only to discover that Managed DirectX is old and deprecated. The task I have is to play pure sine wave bursts interspersed with silence periods (the code) which are precisely timed as to their duration. I need to be able to call a function which plays a pure tone for so many milliseconds, then Thread.Sleep() then play another, etc. At its fastest, the tones and spaces can be as short as 40ms. It's working quite well in Managed DirectX. To get the precisely timed tone I create 1 sec. of sine wave into a secondary buffer, then to play a tone of a certain duration I seek forward to within x milliseconds of the end of the buffer then play. I've tried System.Media.SoundPlayer. It's a loser because you have to Play(), Sleep(), then Stop() for arbitrary tone lengths. The result is a tone that is too long, variable by CPU load. It takes an indeterminate amount of time to actually stop the tone. I then embarked on a lengthy attempt to use NAudio 1.3. I ended up with a memory resident stream providing the tone data, and again seeking forward leaving the desired length of tone remaining in the stream, then playing. This worked OK on the DirectSoundOut class for a while (see below) but the WaveOut class quickly dies with an internal assert saying that buffers are still on the queue despite PlayerStopped = true. This is odd since I play to the end then put a wait of the same duration between the end of the tone and the start of the next. You'd think that 80ms after starting Play of a 40 ms tone that it wouldn't have buffers on the queue. DirectSoundOut works well for a while, but its problem is that for every tone burst Play() it spins off a separate thread. Eventually (5 min or so) it just stops working. You can see thread after thread after thread exiting in the Output window while running the project in VS2008 IDE. I don't create new objects during playing, I just Seek() the tone stream then call Play() over and over, so I don't think it's a problem with orphaned buffers/whatever piling up till it's choked. I'm out of patience on this one, so I'm asking in the hopes that someone here has faced a similar requirement and can steer me in a direction with a likely solution.

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  • How to enable 'playback follows cursor' / 'skip silence' in itunes?

    - by xor_eq
    foobar has got two pretty simple, yet very useful features (at least for me) that iTunes seems to be lacking: Playback follows cursor - Playback continues at the selected track Skip silence - Those 25 minute songs, containing 20 minutes of silence and then some final brainfart that's called "hidden track"... this feature just skips the silence, plays the final bit and continues with the next song Is it possible somehow to enable those features in iTunes?

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  • How do I overcome my silence, Patience and be talkative? [closed]

    - by Pallavi
    Hi Everyone, I am a professional working in a industry for 6 years worked in 3 differnt companies. In previous current company and previous company I had bad bosses. As a result I could not talk with them properly and being a honest person I didnt have rapo with them. I am nt able to open up and communicate my views clearly to anyone. I guess I live in my shell if I dont like the comments or surrounding. I want to be talkative and expressive. Can anybody help me to do so?

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  • How do I silence strace's message "[ Process PID=15733 runs in 64 bit mode. ]" ?

    - by Ross Rogers
    I'm using memoize.py, but strace keeps injecting the following into the program output each time a process is executed: [ Process PID=15733 runs in 64 bit mode. ] or [ Process PID=16503 runs in 32 bit mode. ] How can I silence strace such that it doesn't inject these statements into the log file? At the very least, I'd like these statements to only go into the output file that memoize.py is instructing strace to use. It's already telling strace to put its output into a specific file ithrough arguments -o /tmp/OUTFILE. Note that strace is being called with the -f parameter to follow child processes.

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  • Fix DB duplicate entries (MySQL bug)

    - by Silence
    I'm using MySQL 4.1. Some tables have duplicates entries that go against the constraints. When I try to group rows, MySQL doesn't recognise the rows as being similar. Example: Table A has a column "Name" with the Unique proprety. The table contains one row with the name 'Hach?' and one row with the same name but a square at the end instead of the '?' (which I can't reproduce in this textfield) A "Group by" on these 2 rows return 2 separate rows This cause several problems including the fact that I can't export and reimport the database. On reimporting an error mentions that a Insert has failed because it violates a constraint. In theory I could try to import, wait for the first error, fix the import script and the original DB, and repeat. In pratice, that would take forever. Is there a way to list all the anomalies or force the database to recheck constraints (and list all the values/rows that go against them) ? I can supply the .MYD file if it can be helpful.

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  • SQL swap field values

    - by Silence
    I have a SQL table (MYSQL 4.0 to be precise) composed of the following : int no (primary) int field1 int field2 I would like to swap the values of field1 with the values of field2 and vice-versa. Ex.: 1;2.5;3.76 becomes 1;3.76;2.5 I need a swapping temporary variable of some sort. But I don't think I can use something like Set @var1 = field1 inside my update statement. And I don't think altering the table to add a temporary column is my best best either.

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  • Goldwave Autotrim over Command Prompt

    - by Vivek
    Hi, I have 1 minute wav file recordings. To check whether there is no voice present in the file, I am using Goldwave to trim the silence which lefts the file size very small in few Kbs. This tells me that there is no sound in most part of the file. But I am not able to do this via Command prompt using the Command line utility, Does Goldwave aupport Autotrim Silence through Command Prompt ? If so, what is the command for it ? Thanks Vivek

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  • AudioQueue ate my buffer (first 15 milliseconds of it)

    - by iter
    I am generating audio programmatically. I hear gaps of silence between my buffers. When I hook my phone to a scope, I see that the first few samples of each buffer are missing, and in their place is silence. The length of this silence varies from almost nothing to as much as 20 ms. My first thought is that my original callback function takes too much time. I replace it with the shortest one possible--it re-renqueues the same buffer over and over. I observe the same behavior. AudioQueueRef aq; AudioQueueBufferRef aq_buffer; AudioStreamBasicDescription asbd; void aq_callback (void *aqData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer) { OSStatus s = AudioQueueEnqueueBuffer(aq, aq_buffer, 0, NULL); } void aq_init(void) { OSStatus s; asbd.mSampleRate = AUDIO_SAMPLES_PER_S; asbd.mFormatID = kAudioFormatLinearPCM; asbd.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; asbd.mBytesPerPacket = 1; asbd.mFramesPerPacket = 1; asbd.mBytesPerFrame = 1; asbd.mChannelsPerFrame = 1; asbd.mBitsPerChannel = 8; asbd.mReserved = 0; int PPM_PACKETS_PER_SECOND = 50; // one buffer is as long as one PPM frame int BUFFER_SIZE_BYTES = asbd.mSampleRate/PPM_PACKETS_PER_SECOND*asbd.mBytesPerFrame; s = AudioQueueNewOutput(&asbd, aq_callback, NULL, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &aq); s = AudioQueueAllocateBuffer(aq, BUFFER_SIZE_BYTES, &aq_buffer); // put samples in the buffer buffer_data(my_data, aq_buffer); s = AudioQueueStart(aq, NULL); s = AudioQueueEnqueueBuffer(aq, aq_buffer, 0, NULL); }

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  • Debug NAudio MP3 reading difference?

    - by Conrad Albrecht
    My code using NAudio to read one particular MP3 gets different results than several other commercial apps. Specifically: My NAudio-based code finds ~1.4 sec of silence at the beginning of this MP3 before "audible audio" (a drum pickup) starts, whereas other apps (Windows Media Player, RealPlayer, WavePad) show ~2.5 sec of silence before that same drum pickup. The particular MP3 is "Like A Rolling Stone" downloaded from Amazon.com. Tested several other MP3s and none show any similar difference between my code and other apps. Most MP3s don't start with such a long silence so I suspect that's the source of the difference. Debugging problems: I can't actually find a way to even prove that the other apps are right and NAudio/me is wrong, i.e. to compare block-by-block my code's results to a "known good reference implementation"; therefore I can't even precisely define the "error" I need to debug. Since my code reads thousands of samples during those 1.4 sec with no obvious errors, I can't think how to narrow down where/when in the input stream to look for a bug. The heart of the NAudio code is a P/Invoke call to acmStreamConvert(), which is a Windows "black box" call which I can't think how to error-check. Can anyone think of any tricks/techniques to debug this?

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  • Seeking automated toolchain for WAV cleanup and conversion to MP3 (currently using Audacity manually

    - by Andrew M. Greene
    I use Finale with the Garritan sounds to convert my music into WAV files; after that I manually load each one into Audacity and select all use the "normalize" effect (with default settings) manually select the silence at the beginning of the file and delete it manually select the silence at the end of the file and delete it finally export the whole thing as an MP3 (with specific tag data) This gets tedious, especially as I work on a ten-movement piece with six part-prominent rehearsal "tapes" to accompany the full-score rendition. And there are too many variations in what's going on for me to trust AutoHotKey for this one. Is there a way to script Audacity, or is there another toolchain that would do what I want? I'm running on a WindowsXP laptop with cygwin installed. Thanks!

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  • Etiquette when asking questions in an IRC channel

    - by Zarkonnen
    Many larger OSS projects maintain IRC channels to discuss their usage or development. When I get stuck on using a project, having tried and failed to find information on the web, one of the ways I try to figure out what to do is to go into the IRC channel and ask. But my questions are invariably completely ignored by the people in the channel. If there was silence when I entered, there will still be silence. If there is an ongoing conversation, it carries on unperturbed. I leave the channel open for a few hours, hoping that maybe someone will eventually engage me, but nothing happens. So I worry that I'm being rude in some way I don't understand, or breaking some unspoken rule and being ignored for it. I try to make my questions polite, to the point, and grammatical, and try to indicate that I've tried the obvious solutions and why they didn't work. I understand that I'm obviously a complete stranger to the people on the channel, but I'm not sure how to fix this. Should I just lurk in the channel, saying nothing, for a week? That seems absurd too. A typical message I send might be "Hello all - I've been trying to get Foo to work, but I keep on getting a BarException. I tried resetting the Quux, but this doesn't seem to do anything. Does anyone have a suggestion on what I could try?"

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  • performing a javascript event without triggering that event handler

    - by bento
    In my latest code, I have an event handler for a focus on a textarea. When the user clicks on the textarea, that event-handler is triggered which sets some other DOM states based on the selected textarea. However, elsewhere in my program I want to programmatically set the focus of the textarea without triggering that event handler. I know Backbone, for instance, has a way to silently perform an action. My only pseudo-solution is to temporarily set a variable: var silence = true; And then, in my event handler, only perform the logic if silence is false. The handler is still triggered, but the logic doesn't run. Does anyone else know of better strategies for this?

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  • Which adjustable ergonomic keyboard do you recommend to a fellow coder and why?

    - by thefonso
    My beloved Goldtouch has died (a moment of silence)....the hinge at the middle which allows you to adjust the board finally broke and the manufacturer does not do repairs for free. I'd have to buy a new board. So...since this I'm in the market for a new keyboard and I'm looking for an adjustable one comparable or better to the Goldentouch. I post this question to all you coders out there. Which adjustable ergo keyboard do you recommend to a fellow coder and why?

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  • Which adjustable ergo keyboard do you recommend to a fellow coder and why?

    - by thefonso
    My beloved Goldtouch has died (a moment of silence)....the hinge at the middle which allows you to adjust the board finally broke and the manufacturer does not do repairs for free. I'd have to buy a new board. So...since this I'm in the market for a new keyboard and I'm looking for an adjustable one comparable or better to the Goldentouch. I post this question to all you coders out there. Which adjustable ergo keyboard do you recommend to a fellow coder and why?

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