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replace.me › /04/17 › ffmpeg-for-windows-with-faac. Run the MinGW-get installer and select Download latest repository catalogs. In the options add the C++ compiler (needed for FAAC) and select.
 
 

FFmpeg Codecs Documentation

Avoid transcoding from a lossy format to the same or another lossy format when possible. Transcode to lossy from the lossless source if you have it , or just copy the lossy source audio track instead of transcoding. This quality degradation might not be audible to you but it might be audible to others. This post on hydrogenaudio. If the target container format supports the audio codec of the source file then consider just muxing it into the output file without re-encoding.

MKV supports virtually any audio codec. This can be achieved by specifying ‘copy’ as the audio codec. Another reason to transcode might be that the source audio track is too big it has a higher bitrate than what you want to use in the output file. I had about 2 or 3 versions. All worked for other encoders under Windows, but the libfaac. BTW, it must have been for older versions. The latest version of FFmpeg does not need it. I tried it without it and it still works.

I am using the following command. Thanks in advance. Originally Posted by velagac. Hi bat99 if I want to use libfaac, How can I do that? Last edited by bat; 19th Oct at Originally Posted by bat It shows that libfaac isn’t on the list. So probably you can’t use libfaac with “zeranoe” compiles. If you want to use libfaac, use a different compile of FFmpeg with the libfaac. See post 4 etc. Actually, I’ve found that “libfaac” isn’t that good after all.

I started experimenting with MEncoder first and found the sound to be very poor using libfaac, even with bitrates of 96 to kbps. FFmpeg ‘s was slightly better as was the stand alone faac. Strange, libfaac and faac. Version 1. From audiocoding. So much so, that I’ll be using this encoder from now on for aac audio. This will require more work because I’ll have to re-mux the sound back in for videos. Has anyone used pipes with faac. Last edited by wsc4; 28th Oct at Originally Posted by wsc4.

Last edited by bat; 28th Oct at 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. Enable inband forward error correction. Default is disabled. Set cutoff bandwidth in Hz. The argument must be exactly one of the following: , , , , or , corresponding to narrowband, mediumband, wideband, super wideband, and fullband respectively.

The default is 0 cutoff disabled. Set channel mapping family to be used by the encoder. The default value of -1 uses mapping family 0 for mono and stereo inputs, and mapping family 1 otherwise.

The default also disables the surround masking and LFE bandwidth optimzations in libopus, and requires that the input contains 8 channels or fewer. Other values include 0 for mono and stereo, 1 for surround sound with masking and LFE bandwidth optimizations, and for independent streams with an unspecified channel layout. If set to 0, disables the use of phase inversion for intensity stereo, improving the quality of mono downmixes, but slightly reducing normal stereo quality.

The default is 1 phase inversion enabled. Shine is a fixed-point MP3 encoder. It has a far better performance on platforms without an FPU, e. However, as it is more targeted on performance than quality, it is not on par with LAME and other production-grade encoders quality-wise. Requires the presence of the libshine headers and library during configuration. You need to explicitly configure the build with –enable-libshine.

The following options are supported by the libshine wrapper. The shineenc -equivalent of the options are listed in parentheses. Requires the presence of the libtwolame headers and library during configuration. You need to explicitly configure the build with –enable-libtwolame. The following options are supported by the libtwolame wrapper. The twolame -equivalent options follow the FFmpeg ones and are in parentheses. Default value is k. Set quality for experimental VBR support.

Maximum value range is from to 50, useful range is from to The higher the value, the better the quality. Set psychoacoustic model to use in encoding. The argument must be an integer between -1 and 4, inclusive.

The default value is 3. Requires the presence of the libvo-amrwbenc headers and library during configuration. You need to explicitly configure the build with –enable-libvo-amrwbenc –enable-version3. Requires the presence of the libvorbisenc headers and library during configuration. You need to explicitly configure the build with –enable-libvorbis. The following options are supported by the libvorbis wrapper. The oggenc -equivalent of the options are listed in parentheses.

The value should be a float number in the range of Set cutoff bandwidth in Hz, a value of 0 disables cutoff. This only has effect on ABR mode. Set noise floor bias for impulse blocks. The value is a float number from A negative bias instructs the encoder to pay special attention to the crispness of transients in the encoded audio. The tradeoff for better transient response is a higher bitrate.

The following shared options are effective for this encoder. Only special notes about this particular encoder will be documented here. This is the default AAC encoder. Effective range for -q:a is around 0. By default the MP4 muxer writes the ‘moov’ atom after the audio stream ‘mdat’ atom at the end of the file. This results in the user requiring to download the file completely before playback can occur.

Relocating this moov atom to the beginning of the file can facilitate playback before the file is completely downloaded by the client. Since the audio is simply being stream copied there is no re-encoding occurring, just re-muxing, so therefore there is no quality loss:. Powered by Trac 1.