I made an audio recording with Audacity but before I had a chance to save the file, the power cord to my computer was accidently knocked out. So I downloaded and tryed the recovery program to get the sound file back and I saved the file. That worked. But when I played the file back it the audio sounded speeded up as if Alvin and the Chipmunks were talking. Any advice on how to fix the file / recording?
Let me pose the question a different way.
I have a audio recording that, when I play it back, it is too fast. How do I change the play back speed so that it sounds more normal?
I made an audio recording with Audacity but before I had a chance to save the file, the power cord to my computer was accidently knocked out. So I downloaded and tryed the recovery program to get the sound file back and I saved the file. That worked. But when I played the file back it the audio sounded speeded up as if Alvin and the Chipmunks were talking. Any advice on how to fix the file / recording?
Let me pose the question a different way.
I have a audio recording that, when I play it back, it is too fast. How do I change the play back speed so that it sounds more normal?
In my experience, "chipmonking" has generally been the result of a mixup in sample rate. Typically (for me) it was 1 channel encodings and players that could only play stereo, so they essentially sent every other sample to the left and right channels and the file played at double speed.
So, load into Audacity and see if the sample rate and number of channels matches what it ought to be. If there's no utility for directly changing that while leaving the samples intact, Audacity has a time/pitch stretching function.
Like donP has said, the most probable cause of this is that you are working now with a different samplerate. Check which one you have in the bottom left toolbar of audacity, and which samplerate is defined in the toolbar of the audio track.
Just in case it isn’t clear, having a file that thinks it is a different sample rate does not involve any changed audio data. The audio is the same, only the file header information is wrong. This is very easily corrected with CoolEdit, it just writes the correct sample rate (supplied by you) to the header. This can most likely also be easily done by various other editors.
If the header is missing or fouled-up...
In Audacity, File -> Import -> Raw Data
Start with the paramaters matching whatever you used for recording. WAV files should be Little-endian. If you get noise try an offset or 1, 2, or 3.
I made an audio recording with Audacity but before I had a chance to save the file, the power cord to my computer was accidently knocked out. So I downloaded and tryed the recovery program to get the sound file back and I saved the file. That worked. But when I played the file back it the audio sounded speeded up as if Alvin and the Chipmunks were talking. Any advice on how to fix the file / recording?
Let me pose the question a different way.
I have a audio recording that, when I play it back, it is too fast. How do I change the play back speed so that it sounds more normal?
In my experience, "chipmonking" has generally been the result of a mixup in sample rate. Typically (for me) it was 1 channel encodings and players that could only play stereo, so they essentially sent every other sample to the left and right channels and the file played at double speed.
So, load into Audacity and see if the sample rate and number of channels matches what it ought to be. If there's no utility for directly changing that while leaving the samples intact, Audacity has a time/pitch stretching function.
I am sorry it has been seveal days since I have addressed this issue.
I think we are on the right proverbial track. The recovery option created two seperate files:
Recovery_block1_channel1
Recovery_block1_channel2
And the input was from a single microphone input jack that was connected to a device that records audio off of a telephone line. I know. I know you have to tell the other speaker on the line that the conversation is being recorded. So let's not derail the technical nature of this discussion.
How to I merge these two files into one. Clearly it seems what has happened is, like you have said, the files were generated by taking every other sound piece and alternating the output thus generating two files that play back at double speed.
If the header is missing or fouled-up...
In Audacity, File -> Import -> Raw Data
Start with the paramaters matching whatever you used for recording. WAV files should be Little-endian. If you get noise try an offset or 1, 2, or 3.
Nope. It is
Project -> Import -> Raw DataAnd I do not understand the rest of your post here. I have two audio channels. How do I merge them into one?
Just in case it isn’t clear, having a file that thinks it is a different sample rate does not involve any changed audio data. The audio is the same, only the file header information is wrong. This is very easily corrected with CoolEdit, it just writes the correct sample rate (supplied by you) to the header. This can most likely also be easily done by various other editors.
Do you use Cool Edit?
Is it this?
http://www.softpedia.com/get/Multimedia/Au...-Edit-Pro.shtml (http://www.softpedia.com/get/Multimedia/Audio/Audio-Editors-Recorders/Cool-Edit-Pro.shtml)
That is the program but it is no longer legally available; the current program is Audition, from Adobe. The trial version of CoolEdit, when it was legitimate, would demo most functions but added random tones to the output file, making it pretty useless for anything except a program demo.
I only mention CoolEdit because that is what I use and I have not become familiar with other editors.
A recovery from a file without a header (raw data) should have an option for # of channels, along with the other information you need to feed it. If the file were raw PCM data, the program would not open it at all until you supplied a number of pieces of information.
However, if you can play the original file and its only problem is as your first post described, it isn’t without a header. In CoolEdit, the proper function is labeled Adjust Sample Rate. The only value you supply is the new sample rate, the one you were recording at. No data is changed, only the header value for sample rate is rewritten.
This is different than Convert Sample Rate, where you can change a number of parameters, none of which will effect playback speed.
Most audio editors also have some form of Stretch transform which can do pretty much the same thing as Adjust Sample Rate: make changes in the playback speed and change pitch.
I made an audio recording with Audacity but before I had a chance to save the file, the power cord to my computer was accidently knocked out. So I downloaded and tryed the recovery program to get the sound file back and I saved the file. That worked. But when I played the file back it the audio sounded speeded up as if Alvin and the Chipmunks were talking. Any advice on how to fix the file / recording?
Let me pose the question a different way.
I have a audio recording that, when I play it back, it is too fast. How do I change the play back speed so that it sounds more normal?
In my experience, "chipmonking" has generally been the result of a mixup in sample rate. Typically (for me) it was 1 channel encodings and players that could only play stereo, so they essentially sent every other sample to the left and right channels and the file played at double speed.
So, load into Audacity and see if the sample rate and number of channels matches what it ought to be. If there's no utility for directly changing that while leaving the samples intact, Audacity has a time/pitch stretching function.
Where is the tme/pitch stretching function?
I have managed, it seems, to have fixed the problem by changing this setting. I set the rate from 44100 to 22050:
(http://i67.photobucket.com/albums/h292/Athono/audio_fix.png)