I am looking for a library (.jar) which help me to convert sound file from .wav format to .mp3 format. Infact i want to write an application which reads .wav file from a specified location in linux, convert it to mp3 and save it in DB. Kindly suggest any opensource java library which serve this purpose. waiting for your feedback. Regards, Amir Usage: > wav2mp3 wav_folder_uri All wav files in the folder wav_folder_uri will be encoded to mp3. Supports WAV files containing uncompressed audio (PCM) with no more than 2 channels for now. It will be difficult to use multiple encoding threads on a single wav file, because it is not clear whether LAME library structures are thread-safe. src_simple is for converting a whole file in one go. It does not give good results when called multiple times on discrete chunks of the file like you are doing. If you want to convert the file in chunks like that, you should be using the src_process API. Also be aware that the libsamplerate source code tarball includes sources to a program FLAC files can then be played with your favorite player, just like ordinary MP3's. If you already have WAV files, then you simply need to convert them to FLAC (and not worry about losing quality). You can either use CLI via SoX (although other solutions exist, like flac itself): sox track_01.wav track_01.flac --extract-audio as the name implies, it extracts the audio from the video--audio-format mp3 specifies the audio format - mp3 in this case--audio-quality 160K specifies the audio quality that will be used by ffmpeg/avconv when converting to mp3 in this case. You can specify an exact bitrate, like 128K, 160K, etc., or a VBR quality value between This depends on the phone of course. Many modern smartphones like AAC audio: ffmpeg -i weird.wma -filter:a "atempo=1.7" -c:a libfaac -q:a 100 final.m4a. Here, you can change the quality with the -q:a option, where the value is in percent and higher means better. Or, MP3 audio with an (average) quality of 4, where less means better (0 resulting Depending on the initial format of the mp3, you may need two separate commands. First convert your existing audio file to the mandatory input format: ffmpeg -i file.mp3 -ar 16000 -ac 1 file.wav. The run pocketsphinx. pocketsphinx_continuous -infile file.wav 2> pocketsphinx.log > myspeech.txt. To do so, run the following command: 1. ./AAXtoMP3 . The token has to be replaced with the authcode (“activation_bytes”), which we have just read via audible-activator. The token must be replaced by the path to the aax file. The rest is done by the tool. We piped the output of cdparanoia to the lame application which is used to create mp3 audio files. Since mp3 is a lossy format, we used the -b option of the latter to specify a fixed bitrate of 320kbps and, again, the - symbol to instruct the application to read from its standard input; finally, we provided the name of the output file, just 4UbAmo.

convert wav to mp3 linux