I would like to know the algorithm for converting .amr to .wav . What I have done till now is remove the .amr header and replace it with .wav header.The format gets changed but the audio plays some vague sound. I think this is because the raw data of .amr is different. How do I convert this to .wav?
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So I've done some research and suppose, correct me if I'm wrong, a .wav file is basically I/Q(file) samples merged with extra information(header part).
What I'm trying to do is get that data from the .wav file(I already did that, also can convert bytes to samples).
My question here is how can I convert that .wav file to an I/Q file with the data/samples I have? is it possible? if it is, how?
I am pretty new with processing audio file. '
I want to build a web app that can take audio file and turn the into visualization for user like this https://github.com/CrowdCurio/audio-annotator
Right now I want to research on visualize audio datas. Original data that was stored in S3 come in two form .ts and .flac. That's why I want to ask if there's any visualization tool which can directly use .ts or .flac audio file.
Because right now the solution I think of will be first convert them into .wav or .mp3, so most visualization tool can process them, but .wav file is really storage-wasting as far as I know.
So if you know any approach or tool to do this. Please let me know!
Audio visualization requires audio data. Your compressed audio isn't audible until decoded. Therefore, you must decode them to PCM before visualizing.
This doesn't require that you store the files as WAV, but you'll at least have to decode them on-the-fly.
im trying to write a little client for rtmp(audio only). so far i got the communication working (red5 server) but now im stuck with the audio data.
the server is sending in MP3 44KHz 16bit stereo.
i get my Audiomessage which consists of the byte identifying the codec (0x2f) and the audio data which looks for example like this
ff:fb:92:64:eb:80:03:98:58:d2:e9:26:1b:7e:5d:e7:4a:1a:19:26:5c:8b:89:07:47:44:98:6b:91:2d:9c:28:b4:33:15:70:82:c9:29:87:8d:e4:8f:31:83:84:7b:e5:82:b5:57:62:00:02:e5:bb:f1:86:15:7a:8f:da:9e:ca:4f:83:9d:0a:c4:56:7b:b3:3d:56:43:ba:2b:28:b8:9d:0c:e1:82:0c:08:36:24:f3:39:67:54:b7:41:d9:8e:ef:36:96:56:22:d2:b9:9f:ae:40:43:8e:ea:39:52:0c:a4:48:25:02:54:91:c7:35:37:2d:be:f2:37:23:61:65:35:d9:0f:aa:18:b4:37:d9:d4:c8:68:21:3c:bd:ea:c1:d0:98:df:eb:96:59:99:88:09:37:36:c3:8b:47:80:64:84:41:ba:35:ea:a6:0a:d6:74:9e:09:f6:a5:d7:3f:1f:53:d8:fb:8d:d9:d3:f8:ee:c7:c1:68:25:25:8e:ae:6a:1c:08:52:9d:58:cf:cf:87:c1:ba:a4:f0:63:76:b0:b4:65:79:1b:3b:21:5f:2f:b5:7a:18:43:af:f7:fd:15:0c:87:c9:73:54:95:22:94:cc:cb:e3:da:4d:e0:f3:8a:95:69:69:eb:32:71:57:08:49:76:e0:f3:84:8c:4b:4c:84:6b:5d:7a:c8:c9:d7:df:d5:e2:68:bb:5f:6c:9f:ba:f4:0a:6c:6e:51:8a:b3:59:9a:07:0c:e4:2a:9d:ec:d1:99:53:48:f2:8b:22:b2:d3:bf:e1:5b:9f:ee:49:9f:2c:ee:63:1f:6f:da:90:e7:65:00:55:99:97:77:b9:e8:97:43:81:fd:32:e4:81:20:d0:78:f5:4f:59:47:39:f2:57:5d:f4:d5:91:48:c9:45:10:52:49:4d:04:87:6b:0e:a5:72:ed:34:74:08:93:5b:8a:54:3a:d9:7e:53:8f:c7:5e:b1:99:f3:55:63:72:49:99:55:3a:b8:0d:73:3b:2a:ea:9a:b5:32:d2:3b:61:c2:4e:e9:56:78:99:14:4a:a7:46:f4:ee:ae:6f:ff:c8:85:2d:07:68:ad:e2:84:dd:0a:bd:2e:93:12:43
i dont find a little thing about the data format. as the first byte is always 0xff i assume every chunk of audio data has a little header describing its contents.
the rtmp spec from adobe doesnt loose a single word about the format of the audio message package (just two lines saying its an audio message... wow).
does anyone know the format for the audio messages or at least a source where i find something?
The Adobe spec doesn't document the elementary stream formats because they are covered in their own documents, and usually quite large. MP3 is covered by ISO/IEC 11172-3.
There is a good rundown available here:
http://www.mpgedit.org/mpgedit/mpeg_format/mpeghdr.htm
Recently i have been trying to convert an audio file from one format to another through ffmpeg. i was trying to do some google but results made me a little confused about the difference between encoding and decoding an audio file and converting from one format to another.
Let me describe it this way: There are several different file formats for video files (sometimes also called "wrappers"). There are also several different codecs which can be used to encode (or compress) the audio and video. Audio and video use different codecs - and the encoded formats can be sorted in different file types/formats.
So when you talk about "encoding" vs. "converting" a couple of things come into play.
"Encoding" would be the act of taking audio/video and encoding them into a given codec(s). "Converting" implies having stuff in one format, but wanting it in another. There are two ways of looking at this:
Often called "repackaging" - this is when the video (for example) has been encoded correctly (let's say h264, with a bunch of parameters), but you want it in a different file-type - maybe it's an .AVI and you wanted it in an .MP4. This doesn't involve changing the actual video - just re-wraping the h264 stream in a new "wrapper", and is thus a fast operation.
Re-encoding. Let's say your audio was in a MP3 format, and you wanted it in an AAC format. This would require decoding the entire MP3 stream, and re-encoding it into AAC.
Obviously you can also do "1" and "2" together.
Refer Formats and Codecs for detailed information.
Hope it helps!
What is the difference between compressed and uncompressed .wav files?
The WAV format is a container format for audio files in Windows.
The WAV file consists of a header and the contents. The header contains information about the size, duration, sampling frequency, resolution, and other information about the audio contained in the WAV file. Generally, after the header is the actual audio data.
Since WAV is a container format, the data it contains can be stored in various formats. One of which is uncompressed PCM, but it can also store ADPCM, MP3 and other formats, and can be read and written if an audio codec for the format is available.
The difference between compressed and uncompressed WAV files is that the data contained within the WAV file is either uncompressed raw audio samples, or it is compressed using an audio codec, in which case, it must be decompressed before it can be played back.
Further reading:
Wikipedia: Audio compression (data)
Wikipedia: WAV
Wikipedia: Codec
There's a great explanation here. The basic difference is that an uncompressed wave file has just the raw bits in it as they "appear". There is nothing done to compress or shrink them. A compressed wave file uses some sort of codec to shrink down the data before putting it in the file.
The difference between these two things is basically in the size of object, the compressed one might have low size compared to uncompressed basically the content are the same.
You have to be very careful when using the word "uncompressed" when talking about media.
Basically ALL digital media is compressed in some way. Audio, or video. No matter what it is, it is compressed in some way. Its intrinsic to converting from analog to digital.
The problem isn't really technical, its lingual.
People think that uncompressed means "nothing done to it" when in reality there really isnt any way you can do this. There is always some kind of compression done when you convert the analog signal coming out of the mic and going into a file...Its essential.
What uncompressed means is very high quality. And different "Uncompressed" codecs do things differently.
I know more about video codecs, so i will base my example in those.
Black Magic (A company that makes video Out Cards) has an Uncompressed Codec. Its very good. Makes Beautiful images.. But its not really "uncompressed". Sure its big. But compare it to a DPX of TIFF image sequence...and it aint that big, and is quite compressed. Its only 10 bit, but something like an OpenEXR image sequence is like 32 bit...and coming from film, that is still technically compressed. It has to be.
Its just the nature of the beast.