It lets you record the audio of a video you are watching like from a YouTube music video. You only have to press the Record button to start recording and save it to a high quality MP3 or M4A file. One good thing with the program is having an 'Add to iTunes' button which lets you add your recorded audio to your iTunes Library.
It is capable of identifying recorded music files and can automatically get music information such as Title, Artist, Genre and Album. If you notice any mistakes with the information, you can edit the tag. It records exactly what it hears from your Mac. You will enjoy exactly the same music with the same quality, no more and no less.
What's more, the application has its own media player. You can play the audio you have recorded and check the quality of the said audio. You can also organize your audio files and delete unwanted music files.
LAME is a free software project that was first released in , and has incorporated many improvements since then, including an improved psychoacoustic model. This avoided including LAME itself, which use patented techniques, and so required patent licenses in some countries. For instance, it is now bundled with Audacity , [4].
After some quality concerns raised by others, he decided to start again from scratch based on the 'dist10' MPEG reference software sources. His goal was only to speed up the dist10 sources, and leave its quality untouched. That branch a patch against the reference sources became Lame 2. Still useful and relevant. You need a Dropbox or GUI for this hence my low review on ease of use. By the way the latest Rarewares Lame Dropbox has a resampling bug so that option has been removed until it's fixed.
I reverted to the last Dropbox box which is bug free. Review by Tim on Jun 6, Version: 3. Thankyou for this excellent tool. Review by Kevin on May 11, Version: 3. It may be disabled when installing or after installation.
Free Trial version available for download and testing with usually a time limit or limited functions. No installation is required. It works on bit and bit Windows. It works only on bit Windows. It works on bit and bit Mac OS. It works only on bit Mac OS. Be careful when you install the software and disable addons that you don't want!
It may not contain the latest versions. Our hosted tools are virus and malware scanned with several antivirus programs using www. Rating Rating from Encoders requiring an external library must be enabled manually via the corresponding --enable-lib option. You can list all available encoders using the configure option --list-encoders. Setting this automatically activates constant bit rate CBR mode.
If this option is unspecified it is set to kbps. Set quality for variable bit rate VBR mode. This option is valid only using the ffmpeg command-line tool. Set cutoff frequency. If unspecified will allow the encoder to dynamically adjust the cutoff to improve clarity on low bitrates. This method first sets quantizers depending on band thresholds and then tries to find an optimal combination by adding or subtracting a specific value from all quantizers and adjusting some individual quantizer a little.
This is an experimental coder which currently produces a lower quality, is more unstable and is slower than the default twoloop coder but has potential. Not currently recommended. Worse with low bitrates less than 64kbps , but is better and much faster at higher bitrates. Can be forced for all bands using the value "enable", which is mainly useful for debugging or disabled using "disable".
Sets intensity stereo coding tool usage. Can be disabled for debugging by setting the value to "disable". Uses perceptual noise substitution to replace low entropy high frequency bands with imperceptible white noise during the decoding process. Enables the use of a multitap FIR filter which spans through the high frequency bands to hide quantization noise during the encoding process and is reverted by the decoder.
As well as decreasing unpleasant artifacts in the high range this also reduces the entropy in the high bands and allows for more bits to be used by the mid-low bands. Enables the use of the long term prediction extension which increases coding efficiency in very low bandwidth situations such as encoding of voice or solo piano music by extending constant harmonic peaks in bands throughout frames. Use in conjunction with -ar to decrease the samplerate.
Enables the use of a more traditional style of prediction where the spectral coefficients transmitted are replaced by the difference of the current coefficients minus the previous "predicted" coefficients. In theory and sometimes in practice this can improve quality for low to mid bitrate audio. The default, AAC "Low-complexity" profile. Is the most compatible and produces decent quality. Introduced in MPEG4. Introduced in MPEG2. This does not mean that one is always faster, just that one or the other may be better suited to a particular system.
The AC-3 metadata options are used to set parameters that describe the audio, but in most cases do not affect the audio encoding itself.
Some of the options do directly affect or influence the decoding and playback of the resulting bitstream, while others are just for informational purposes. A few of the options will add bits to the output stream that could otherwise be used for audio data, and will thus affect the quality of the output.
Those will be indicated accordingly with a note in the option list below. Allow Per-Frame Metadata. Specifies if the encoder should check for changing metadata for each frame.
Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo. This field will only be written to the bitstream if a center channel is present. The value is specified as a scale factor. There are 3 valid values:. Surround Mix Level. The amount of gain the decoder should apply to the surround channel s when downmixing to stereo.
This field will only be written to the bitstream if one or more surround channels are present. Audio Production Information is optional information describing the mixing environment. Either none or both of the fields are written to the bitstream.
Mixing Level. Specifies peak sound pressure level SPL in the production environment when the mix was mastered. Valid values are 80 to , or -1 for unknown or not indicated. The default value is -1, but that value cannot be used if the Audio Production Information is written to the bitstream.
Room Type. Describes the equalization used during the final mixing session at the studio or on the dubbing stage. A large room is a dubbing stage with the industry standard X-curve equalization; a small room has flat equalization. Dialogue Normalization. This parameter determines a level shift during audio reproduction that sets the average volume of the dialogue to a preset level. The goal is to match volume level between program sources. A value of dB will result in no volume level change, relative to the source volume, during audio reproduction.
Valid values are whole numbers in the range to -1, with being the default. Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround Pro Logic. This field will only be written to the bitstream if the audio stream is stereo.
Original Bit Stream Indicator. Specifies whether this audio is from the original source and not a copy. It is grouped into 2 parts. If any one parameter in a group is specified, all values in that group will be written to the bitstream. Default values are used for those that are written but have not been specified. Preferred Stereo Downmix Mode. Dolby Surround EX Mode.
Indicates whether the stream uses Dolby Surround EX 7. Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone encoding multi-channel matrixed to 2.
Stereo Rematrixing. This option is enabled by default, and it is highly recommended that it be left as enabled except for testing purposes.
Set lowpass cutoff frequency. If unspecified, the encoder selects a default determined by various other encoding parameters. These options are only valid for the floating-point encoder and do not exist for the fixed-point encoder due to the corresponding features not being implemented in fixed-point. The per-channel high frequency information is sent with less accuracy in both the frequency and time domains. This allows more bits to be used for lower frequencies while preserving enough information to reconstruct the high frequencies.
This option is enabled by default for the floating-point encoder and should generally be left as enabled except for testing purposes or to increase encoding speed. Coupling Start Band. Sets the channel coupling start band, from 1 to If a value higher than the bandwidth is used, it will be reduced to 1 less than the coupling end band. If auto is used, the start band will be determined by the encoder based on the bit rate, sample rate, and channel layout.
This option has no effect if channel coupling is disabled. Sets the compression level, which chooses defaults for many other options if they are not set explicitly. Valid values are from 0 to 12, 5 is the default. Chooses if rice parameters are calculated exactly or approximately.
Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is applied after the first stage to finetune the coefficients.
This is quite slow and slightly improves compression. This is a native FFmpeg encoder for the Opus format. Currently its in development and only implements the CELT part of the codec. Its quality is usually worse and at best is equal to the libopus encoder.
If unspecified it uses the number of channels and the layout to make a good guess. Sets the maximum delay in milliseconds. Lower delays than 20ms will very quickly decrease quality. Requires the presence of the libfdk-aac headers and library during configuration. You need to explicitly configure the build with --enable-libfdk-aac.
The library is also incompatible with GPL, so if you allow the use of GPL, you should configure with --enable-gpl --enable-nonfree --enable-libfdk-aac. If the bitrate is not explicitly specified, it is automatically set to a suitable value depending on the selected profile.
Note that VBR is implicitly enabled when the vbr value is positive. If not specified or explicitly set to 0 it will use a value automatically computed by the library. Enable afterburner feature if set to 1, disabled if set to 0. This improves the quality but also the required processing power. Set VBR mode, from 1 to 5. Requires the presence of the libmp3lame headers and library during configuration. You need to explicitly configure the build with --enable-libmp3lame.
See libshine for a fixed-point MP3 encoder, although with a lower quality. The following options are supported by the libmp3lame wrapper. The lame -equivalent of the options are listed in parentheses. Set constant quality setting for VBR. Set algorithm quality.
Valid arguments are integers in the range, with 0 meaning highest quality but slowest, and 9 meaning fastest while producing the worst quality. Enable use of bit reservoir when set to 1.
LAME has this enabled by default, but can be overridden by use --nores option. Enable the encoder to use ABR when set to 1. The lame --abr sets the target bitrate, while this options only tells FFmpeg to use ABR still relies on b to set bitrate. Requires the presence of the libopencore-amrnb headers and library during configuration. You need to explicitly configure the build with --enable-libopencore-amrnb --enable-version3.
This is a mono-only encoder. Set bitrate in bits per second. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate. Allow discontinuous transmission generate comfort noise when set to 1. The default value is 0 disabled. Most libopus options are modelled after the opusenc utility from opus-tools.
The following is an option mapping chart describing options supported by the libopus wrapper, and their opusenc -equivalent in parentheses. Set VBR mode. The FFmpeg vbr option has the following valid arguments, with the opusenc equivalent options in parentheses:. Set encoding algorithm complexity. Valid options are integers in the range.
The default is Set maximum frame size, or duration of a frame in milliseconds. The argument must be exactly the following: 2. Smaller frame sizes achieve lower latency but less quality at a given bitrate. Sizes greater than 20ms are only interesting at fairly low bitrates. The default is 20ms. We will be grateful to know where we can improve our posts. LAME is free,but in some countries you may need to pay a license fee in orderto legally encode MP3 files.
Audacity is a free and open source Audio Editor which allows you totransform ogg to mp3, transform mp3 to ogg, transform vinyls to mp3 or ogg, do anykind of home recording, remove noise, etc. I have used it torecord and mix some of my bands songs.
Feel free to check out this page to download some songs. To use LAME or FFmpeg with Audacity, you can put it anywhereyou want, but the first time you want to export an MP3 file,Audacity will ask you for the location of this file, so you willwant to remember where you put it. EXE or. DMG for Mac versions. If you use the installers, and Audacity does not detect LAME, download the ZIP option, extract the files inside to a well known folder, thenopen Audacity, go to Library Preferences and configure it to search on the well known folder you extracted the files to.
If you need or want a newer version of Lame, because of the performance improvements with newer AMD and Intel processors , here is v3.
You can also find 3. You can also download the standalone lame v3. Audacity 2. It is recommended to use the. FFmpeg Binary for Audacity 1. You can use Audacity to,Record live audio,convert tapes and records into digital recordings or CDs,cut, copy, splice or mix sounds together. Download the file to your computer and put C: Program Files Audacity in the Audacity program folder or another one if you installed it not here.
Run Audacity, go to 'Edit' - 'Options' - 'Libraries'.
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