This format uses MPEG Audio Layer 3 encoding to compress audio data, which compresses the original file to a small file and still maintains good sound quality. The MP3 files can also be encoded at higher or lower bit rates, with higher or lower resulting quality. MP3 (MPEG Audio Layer 3) is an efficient and lossy compression format for digital audio, it offers a variety of different bit rates. If the source file is a very large file, it will take more time to upload, be sure to select a valid file.This converter cannot support encrypted or protected audio files.The converter supports all the popular audio formats like MP3, OGG, WAV, WMA and others. Convertio advanced online tool that solving any problems with any files. If the file upload process takes a long time or is unresponsive or very slow, please try to cancel and resubmit. Our web-based application helps you to convert audio files in seconds.Before uploading, please make sure you agree to the terms of this website.The maximum upload file size is 200 MB.Once the upload is complete, the converter will redirect a web page to display the conversion results.Click the "Convert" button to start uploading your files.100 free, secure and easy to use Convertio advanced online tool that solving any problems with any files. The tool will try to maintain the audio quality of the source Video file and create a high quality MP3 file as much as possible. The output length should match the input.This free Video to MP3 converter can help you convert various video files to MP3 (MPEG Audio Layer 3) audio, such as 3GP, ASF, AVCHD, AVI, DIVX, F4V, FLV, HEVC, M2TS, M4V, MKV, MOV, MP4, MPEG, MPG, MTS, RM, RMVB, TS, VOB, WEBM, WMV, WTV, XVID and more. Your second question.did you notice the LAME command-line app's "-decode" option? :) This should solve your problem if you're using LAME as the encoder. so I would look toward making sure you use a compatible encoder/decoder combo to get correct-length, junk-trimmed files as output. Most decoders/players recognize the frame as special and skip those samples, but some don't, so there's another potential point of failure. This header is actually a frame of silence (usually 1152 samples) with some specially formatted info embedded in between the frame's header and the start of its null audio data. Some encoders, like LAME, will add such "gapless playback" or "delay and padding" info (in an encoder-specific format, because there's no standard for it) into the file's VBR Info (VBRI) or Xing header, which is sort-of standard even on CBR files, and which contains other info that is sometimes helpful to the player. The decoder will know how much it adds and will skip those samples, but the encoder-added junk isn't entirely predictable (different encoders add different amounts), so the decoder can't skip those samples unless it is somehow informed of what to skip. An MP3 stream contains info for generating a set of samples where each sample corresponds to a sample in the original LPCM data (like from the WAV file), but a side-effect of encoding is that there is some extra "junk" added to each end, and a side-effect of decoding is that there is even more junk added to the beginning.
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