mp3

Calculate Mp3 duration based on bitrate and file size

拈花ヽ惹草 提交于 2019-12-04 19:25:42
问题 I try to calculate mp3 duration by using bitrate and file size , after some search i found this formula: (mp3sizeInByte*0.008)/bitrate i am using mp3sizeInByte*0.008 to convert byte to Kbits. but its not so accurate , in result there is couple second different compare to actual mp3 duration. i want know this right formula ? 回答1: You can calculate the size using the following formula: x = length of song in seconds y = bitrate in kilobits per second (x * y) / 8 We divide by 8 to get the result

libmad playback too fast if read in chunks

一世执手 提交于 2019-12-04 17:27:17
I took libmad example C file, and played an mp3, which played just fine. However, when I try to read the file in chunks, as opposed to the example, which reads the file in one go, I hear "breaks" and the playback is way too fast. Here's my input callback, and my output callback static enum mad_flow input(void *data, struct mad_stream *stream) { struct buffer *buffer = data; // char* raw_data[buffer->size]; // if(fgets(*raw_data, buffer->size, buffer->file) == NULL) { // file is finished! // in our case we would want to move to next file here! // when we get there, we will get data from node-

eyed3 package for Python not properly setting ID3 metadata

半世苍凉 提交于 2019-12-04 17:09:08
For this I am using Python 2.7.13, Windows 10, and the eyed3 package as documented here . Goal: I am trying to create a script that can input any desired ID3 metadata for MP3 files that are missing information. Problem: The script appears to update the metadata correctly but fails to add the information to the "Details" screen of the MP3 properties ( MP3 Details screen ). However, if I first manually input data in those fields before running the script, it correctly both adds the metadata and shows it on the Details screen! Another thing I've noticed is that I only need to enter data in at

Converting a 32 bit wave form to a 16 bit wave form

心已入冬 提交于 2019-12-04 17:08:00
I've been capturing audio using the loopback capture mode. The captured waveform is a 32 bit waveform. I'm struggling with converting this to a 16 bit waveform so encoders like lame can deal with it (it says Unsupported data format: 0x0003). I've tried shifting the bits (not my strong point) in the wave stream itself from 32 bit to 16 bit but the result still sounds distorted. The Wave32To16Stream class seems to blow up on this case: if (sourceStream.WaveFormat.Encoding != WaveFormatEncoding.IeeeFloat) throw new ApplicationException("Only 32 bit Floating point supported"); Ideally I would want

Convert PCM to MP3/OGG

此生再无相见时 提交于 2019-12-04 16:30:59
I need to convert a continuous stream of PCM, or encoded audio (ADPCM, uLaw, Opus), into MP3/OGG format so that it can be streamed to a browser (using html's audio tag). I have the "stream-mp3/ogg-using-audio-tag" part working, now I need to develop the conversion layer. I have two questions: How can I convert PCM into MP3/OGG using NAudio and/or some other C# library/framework? I assume there is a code snippet or two in the NAudio demo app that may be doing this, but I haven't been able to find it. Do I have to convert the encoded data (ADPCM, uLaw, OPUS) into PCM (which I can) before I

concatenating mp3 files or joining mp3 files using java [closed]

a 夏天 提交于 2019-12-04 15:47:54
Closed. This question is off-topic . It is not currently accepting answers. Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 5 years ago . We would like to concatenate/merge/join mp3 files seamlessly using "java" in any environment. We are trying the following options at the moment ( please let us know any other options): Using JMF -- ruled out as it supported only in windows http://java.sun.com/javase/technologies/desktop/media/jmf/reference/faqs/index.html Using tritinous , jlayer and Lame combination. Please let us know thoughts and the links

Looking for a simple MP3 library for C# [closed]

[亡魂溺海] 提交于 2019-12-04 15:31:42
I'm writing a larger project and a part of the project is playing MP3 files. Nothing fancy, just playing files from a playlist, start/stop, next/previous (can be done outside the library), fast forward/rewind. I'm looking for a (free) library, or any other method of doing that. Relying on an external player is not an option. Bass Audio Library is one option. Check out SDL.NET http://cs-sdl.sourceforge.net/ It's a library and has ways to play music. NAudio is an open source .NET audio library that can play back WAV files, using ACM codecs installed on your computer for decompression purposes.

Is it possible, using TagLibSharp, to remove a Lyrics3v2 tag from a MP3 file?

天涯浪子 提交于 2019-12-04 14:35:10
I wonder if it's possible to remove a Lyrics3v2 tag type from a MP3 file using TagLibSharp library. This documentation says that the block entry starts with word " LYRICSBEGIN " and ends with " LYRICS200 ", also it says that the ID3 tag should be present to let exists the Lyrics3v2 tag ...but it doesn't specifies if reffers to ID3v1 or ID3v2 tag, or any of them, anyways I don't understand that part, because Lyrics3v2 tag is a single tag type, is not part of an ID3v1/ID3v2 tag type, it has its own entry on the mp3 header so... I don't understand what it means about the ID3v1/ID3v2 "dependancy".

Mix Audio tracks with offset in SOX

旧时模样 提交于 2019-12-04 13:43:17
问题 From ASP.Net, I am using FFMPEG to convert flv files on a Flash Media Server to wavs that I need to mix into a single MP3 file. I originally attempted this entirely with FFMPEG but eventually gave up on the mixing step because I don't believe it it possible to combine audio only tracks into a single result file. I would love to be wrong. I am now using FFMPEG to access the FLV files and extract the audio track to wav so that SOX can mix them. The problem is that I must offset one of the audio

mp3剪切器如何剪切mp3音频文件

僤鯓⒐⒋嵵緔 提交于 2019-12-04 13:34:05
相信很多人都会有这种想法吧,有些音乐太长,但是只觉得音乐的高潮部分比较好听,其他的地方则是一般。那么如果想只听一小段的话,可能需要把一部分音乐剪下来,也就是音乐剪辑。剪辑音乐需要专门的软件才可以做到。那么,下面就是MP3剪切器怎么剪辑音乐的教程了。一起来看看如何操作吧。 1、首先就是下载安装剪辑音乐的软件了。也就是迅捷音频转换器了。点击下面的按钮,然后把软件下载到桌面上。在软件中选择D盘为安装软件的位置,如果直接点击立即安装,然后会默认安装在C:\Users\AppData\Roaming\AudioEditor中。 2、然后就是打开软件进行使用音频剪辑功能了。打开之后软件会默认使用音频转换功能的。所以需要更换一个音频剪辑的功能,在软件上方点击选择即可。当然软件当中其他的功能也是可以直接选择使用的。 3、然后就把需要剪辑的MP3音频文件添加到迅捷音频转换器当中。点击软件当中的添加文件按钮或者点击添加文件夹,把需要剪辑的音乐导入到软件中。或者可以选中文件然后拖拽到软件中也能添加。 4、然后就是设置转换输出文件的保存位置了。一般软件都是默认输出在桌面上C:/Users/Desktop,不过也可以保存在其他的地方,那就是点击更改路径按钮,然后选择其他的位置保存就可以了。 5、然后就是设置剪辑输出的文件参数了。设置参数的位置在软件右侧。软件分为手动分割、平均分割和按时间分割