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Normalize audio via ffmpeg

Project description

ffmpeg-normalize

Build Status PyPI version

A utility for batch-normalizing audio using ffmpeg.

This program normalizes media files to a certain loudness level using the EBU R128 loudness normalization procedure. It can also perform RMS-based normalization (where the mean is lifted or attenuated), or peak normalization to a certain target level.

Batch processing of several input files is possible, including video files.

Contents:


Requirements

Installation

pip3 install ffmpeg-normalize

Or download this repository, then run pip install ..

Usage

ffmpeg-normalize [-h] [-o OUTPUT [OUTPUT ...]] [-of OUTPUT_FOLDER] [-f]
                    [-d] [-v] [-n] [-pr] [--version] [-nt {ebu,rms,peak}]
                    [-t TARGET_LEVEL] [-p] [-lrt LOUDNESS_RANGE_TARGET]
                    [-tp TRUE_PEAK] [--offset OFFSET] [--dual-mono]
                    [-c:a AUDIO_CODEC] [-b:a AUDIO_BITRATE]
                    [-ar SAMPLE_RATE] [-koa] [-prf PRE_FILTER]
                    [-pof POST_FILTER] [-vn] [-c:v VIDEO_CODEC] [-sn]
                    [-mn] [-cn] [-e EXTRA_OUTPUT_OPTIONS]
                    [-ofmt OUTPUT_FORMAT] [-ext EXTENSION]
                    input [input ...]

For more information, run ffmpeg-normalize -h, or read on.

Description

Please read this section for a high level introduction.

What does the program do?

The program takes one or more input files and, by default, writes them to a folder called normalized, using an .mkv container. All audio streams will be normalized so that they have the same (perceived) volume.

How do I specify the input?

Just give the program one or more input files as arguments. It works with most media files.

How do I specify the output?

You can specify one output file name for each input file with the -o option. In this case, the container format (e.g. .wav) will be inferred from the file name extension that you've given.

Example:

ffmpeg-normalize 1.wav 2.wav -o 1n.wav 2n.wav

If you don't specify the output file name for an input file, the container format will be MKV, and the output will be written to normalized/<input>.mkv.

Using the -ext option, you can supply a different output extension common to all output files, e.g. -ext m4a.

What will get normalized?

By default, all streams from the input file will be written to the output file. For example, if your input is a video with two language tracks and a subtitle track, both audio tracks will be normalized independently. The video and subtitle tracks will be copied over to the output file.

How will the normalization be done?

The normalization will be performed with the loudnorm filter from FFmpeg, which was originally written by Kyle Swanson. It will bring the audio to a specified target level. This ensures that multiple files normalized with this filter will have the same perceived loudness.

What codec is chosen?

The default audio encoding method is uncompressed PCM (pcm_s16le) to avoid introducing compression artifacts. This will result in a much higher bitrate than you might want, for example if your input files are MP3s.

Some containers (like MP4) also cannot handle PCM audio. If you want to use such containers and/or keep the file size down, use -c:a and specify an audio codec (e.g., -c:a aac for ffmpeg's built-in AAC encoder).

Examples

Normalize two WAV files and write them to the specified output files with uncompressed PCM WAV as audio codec:

ffmpeg-normalize file1.wav file2.wav -o file1-normalized.wav file2-normalized.wav

Normalize a number of videos in the current folder and write them to a folder called normalized, converting all audio streams to AAC with 192 kBit/s.

ffmpeg-normalize *.mkv -c:a aac -b:a 192k

For Windows, the above would be written as a loop:

for %%f in ("*.mkv") do ffmpeg-normalize "%%f" -c:a aac -b:a 192k

Normalize an MP3 file and write an MP3 file (you have to explicitly specify the encoder):

ffmpeg-normalize input.mp3 -c:a libmp3lame -b:a 320k -o output.mp3

Normalize many files, keeping PCM audio, but choosing a different container:

ffmpeg-normalize *.wav -c:a pcm_s16le -ext aif

Instead of EBU R128, one might just want to use simple peak normalization to 0 dB:

ffmpeg-normalize test.wav --normalization-type peak --target-level 0 --output normalized.wav
ffmpeg-normalize test.wav -nt peak -t 0 -o normalized.wav

You can (if you really need to!) also overwrite your input file. Warning, this will destroy data:

ffmpeg-normalize input.mp4 -o input.mp4 -f

If you need some fancy extra options, such as setting vbr for the libfdk_aac encoder, pass them to the -e/--extra-options argument:

ffmpeg-normalize input.m4a -c:a libfdk_aac -e='-vbr 3' -o output.m4a

Further examples? Please submit a PR so I can collect them.

Detailed Options

File Input/Output

  • input: Input media file(s)

  • -o OUTPUT [OUTPUT ...], --output OUTPUT [OUTPUT ...]: Output file names.

    Will be applied per input file.

    If no output file name is specified for an input file, the output files will be written to the default output folder with the name <input>.<ext>, where <ext> is the output extension (see -ext option).

    Example: ffmpeg-normalize 1.wav 2.wav -o 1n.wav 2n.wav

  • -of OUTPUT_FOLDER, --output-folder OUTPUT_FOLDER: Output folder (default: normalized)

    This folder will be used for input files that have no explicit output name specified.

General

  • -f, --force: Force overwrite existing files

  • -d, --debug: Print debugging output

  • -v, --verbose: Print verbose output

  • -q, --quiet: Only print errors

  • -n, --dry-run: Do not run normalization, only print what would be done

  • -pr, --progress: Show progress bar for files and streams

  • --version: Print version and exit

Normalization

  • -nt {ebu,rms,peak}, --normalization-type {ebu,rms,peak}: Normalization type (default: ebu).

    EBU normalization performs two passes and normalizes according to EBU R128.

    RMS-based normalization brings the input file to the specified RMS level.

    Peak normalization brings the signal to the specified peak level.

  • -t TARGET_LEVEL, --target-level TARGET_LEVEL: Normalization target level in dB/LUFS (default: -23).

    For EBU normalization, it corresponds to Integrated Loudness Target in LUFS. The range is -70.0 - -5.0.

    Otherwise, the range is -99 to 0.

  • -p, --print-stats: Print first pass loudness statistics formatted as JSON to stdout.

EBU R128 Normalization

  • -lrt LOUDNESS_RANGE_TARGET, --loudness-range-target LOUDNESS_RANGE_TARGET: EBU Loudness Range Target in LUFS (default: 7.0).

    Range is 1.0 - 20.0.

  • -tp TRUE_PEAK, --true-peak TRUE_PEAK: EBU Maximum True Peak in dBTP (default: -2.0).

    Range is -9.0 - +0.0.

  • --offset OFFSET: EBU Offset Gain (default: 0.0).

    The gain is applied before the true-peak limiter in the first pass only. The offset for the second pass will be automatically determined based on the first pass statistics.

    Range is -99.0 - +99.0.

  • --dual-mono: Treat mono input files as "dual-mono".

    If a mono file is intended for playback on a stereo system, its EBU R128 measurement will be perceptually incorrect. If set, this option will compensate for this effect. Multi-channel input files are not affected by this option.

Audio Encoding

  • -c:a AUDIO_CODEC, --audio-codec AUDIO_CODEC: Audio codec to use for output files.

    See ffmpeg -encoders for a list.

    Will use PCM audio with input stream bit depth by default.

  • -b:a AUDIO_BITRATE, --audio-bitrate AUDIO_BITRATE: Audio bitrate in bits/s, or with K suffix.

    If not specified, will use codec default.

  • -ar SAMPLE_RATE, --sample-rate SAMPLE_RATE: Audio sample rate to use for output files in Hz.

    Will use input sample rate by default, except for EBU normalization, which will change the input sample rate to 192 kHz.

  • -koa, --keep-original-audio: Copy original, non-normalized audio streams to output file

  • -prf PRE_FILTER, --pre-filter PRE_FILTER: Add an audio filter chain before applying normalization.

    Multiple filters can be specified by comma-separating them.

  • -pof POST_FILTER, --post-filter POST_FILTER: Add an audio filter chain after applying normalization.

    Multiple filters can be specified by comma-separating them.

    For EBU, the filter will be applied during the second pass.

Other Encoding Options

  • -vn, --video-disable: Do not write video streams to output

  • -c:v VIDEO_CODEC, --video-codec VIDEO_CODEC: Video codec to use for output files (default: 'copy').

    See ffmpeg -encoders for a list.

    Will attempt to copy video codec by default.

  • -sn, --subtitle-disable: Do not write subtitle streams to output

  • -mn, --metadata-disable: Do not write metadata to output

  • -cn, --chapters-disable: Do not write chapters to output

Input/Output Format

  • -ei EXTRA_INPUT_OPTIONS, --extra-input-options EXTRA_INPUT_OPTIONS: Extra input options list.

    A list of extra ffmpeg command line arguments valid for the input, applied before ffmpeg's -i.

    You can either use a JSON-formatted list (i.e., a list of comma-separated, quoted elements within square brackets), or a simple string of space-separated arguments.

    If JSON is used, you need to wrap the whole argument in quotes to prevent shell expansion and to preserve literal quotes inside the string. If a simple string is used, you need to specify the argument with -e=.

    Examples: -e '[ "-f", "mpegts" ]' or -e="-f mpegts"

  • -e EXTRA_OUTPUT_OPTIONS, --extra-output-options EXTRA_OUTPUT_OPTIONS: Extra output options list.

    A list of extra ffmpeg command line arguments valid for the output.

    You can either use a JSON-formatted list (i.e., a list of comma-separated, quoted elements within square brackets), or a simple string of space-separated arguments.

    If JSON is used, you need to wrap the whole argument in quotes to prevent shell expansion and to preserve literal quotes inside the string. If a simple string is used, you need to specify the argument with -e=.

    Examples: -e '[ "-vbr", "3" ]' or -e="-vbr 3"

  • -ofmt OUTPUT_FORMAT, --output-format OUTPUT_FORMAT: Media format to use for output file(s).

    See ffmpeg -formats for a list.

    If not specified, the format will be inferred by ffmpeg from the output file name. If the output file name is not explicitly specified, the extension will govern the format (see '--extension' option).

  • -ext EXTENSION, --extension EXTENSION: Output file extension to use for output files that were not explicitly specified. (Default: mkv)

Environment Variables

The program additionally respects environment variables:

  • TMP / TEMP / TMPDIR

    Sets the path to the temporary directory in which files are stored before being moved to the final output directory. Note: You need to use full paths.

  • FFMPEG_PATH

    Sets the full path to an ffmpeg executable other than the system default.

FAQ

The program doesn't work because the "loudnorm" filter can't be found

Make sure you run ffmpeg v3.1 or higher and that loudnorm is part of the output when you run ffmpeg -filters. Many distributions package outdated ffmpeg 2.x versions, or (even worse), Libav's ffmpeg disguising as a real ffmpeg from the FFmpeg project.

Some ffmpeg builds also do not have the loudnorm filter enabled.

You can always download a static build from their website and use that.

If you have to use an outdated ffmpeg version, you can only use rms or peak as normalization types, but I can't promise that the program will work correctly.

Should I use this to normalize my music collection?

When you run ffmpeg-normalize and re-encode files with MP3 or AAC, you will inevitably introduce generation loss. Therefore, I do not recommend running this on your precious music collection, unless you have a backup of the originals or accept potential quality reduction. If you just want to normalize the subjective volume of the files without changing the actual content, consider using MP3Gain and aacgain.

Why are my output files MKV?

I chose MKV as a default output container since it handles almost every possible combination of audio, video, and subtitle codecs. If you know which audio/video codec you want, and which container is supported, use the output options to specify the encoder and output file name manually.

The conversion does not work and I get a cryptic ffmpeg error!

One possible reason is that the input file contains some streams that cannot be mapped to the output file. Examples:

  • You are trying to normalize a movie file, writing to a .wav or .mp3 file. WAV/MP3 files only support audio, not video. Disable video and subtitles with -vn and -sn, or choose a container that supports video (e.g. .mkv).

  • You are trying to normalize a file, writing to an .mp4 container. This program defaults to PCM audio, but MP4 does not support PCM audio. Make sure that your audio codec is set to something MP4 containers support (e.g. `-c:a aac).

The default output container is .mkv as it will support most input stream types. If you want a different output container, make sure that it supports your input file's video, audio, and subtitle streams (if any).

Also, if there is some other broken metadata, you can try to disable copying over of metadata with -mn.

What are the different normalization algorithms?

  • EBU R128 is an EBU standard that is commonly used in the broadcasting world. The normalization is performed using a psychoacoustic model that targets a subjective loudness level measured in LUFS (Loudness Unit Full Scale). R128 is subjectively more accurate than any peak or RMS-based normalization. More info on R128 can be found in the official document and the loudnorm filter description by its original author.

  • Peak Normalization analyzes the peak signal level in dBFS and increases the volume of the input signal such that the maximum in the output is 0 dB (or any other chosen threshold). Since spikes in the signal can cause high volume peaks, peak normalization might still result in files that are subjectively quieter than other, non-peak-normalized files.

  • RMS-based Normalization analyzes the RMS power of the signal and changes the volume such that a new RMS target is reached. Otherwise it works similar to peak normalization.

Couldn't I just run loudnorm with ffmpeg?

You absolutely can. However, you can get better accuracy and linear normalization with two passes of the filter. Since ffmpeg does not allow you to automatically run these two passes, you have to do it yourself and parse the output values from the first run. If this program is too over-engineered for you, you could also use an approach such as featured in this Ruby script that performs the two loudnorm passes.

After updating, this program does not work as expected anymore!

You are probably using a 0.x version of this program. There are significant changes to the command line arguments and inner workings of this program, so please adapt your scripts to the new one. Those changes were necessary to address a few issues that kept piling up; leaving the program as-is would have made it hard to extend it. You can continue using the old version (find it under Releases on GitHub or request the specific version from PyPi), but it will not be supported anymore.

Can I buy you a beer / coffee / random drink?

If you found this program useful and feel like giving back, feel free to send a donation via PayPal.

License

The MIT License (MIT)

Copyright (c) 2015-2018 Werner Robitza

Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files (the "Software"), to deal in the Software without restriction, including without limitation the rights to use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, and to permit persons to whom the Software is furnished to do so, subject to the following conditions:

The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software.

THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.

Changelog

v1.20.1 (2020-07-22)

  • Manually specify usage string, fixes #132.

  • Fix local import for tests.

v1.20.0 (2020-07-04)

  • Add extra input options.

v1.19.1 (2020-06-25)

  • Add colorama to requirements, fixes #131.

  • Fix warning that is printed with default options.

v1.19.0 (2020-05-02)

  • Fix issue with output folder, fixes #126.

  • Fix typo in README's table of contents link to "File Input/Output". (#124)

  • Clarify readme, fixes #122.

v1.18.2 (2020-04-19)

  • Add warning for automatic sample rate conversion, addresses #122.

  • Ignore vscode folder.

  • Fix printing of errors in conversion.

v1.18.1 (2020-04-16)

  • Fix unit tests.

  • Improve handling of output file folder and errors.

  • Clarify usage of output options, add warning.

  • Improve documentation, fixes #120.

  • Do not include bump messages in changelog.

v1.18.0 (2020-04-13)

  • Use measured offset in second pass, fixes #119.

  • Update release instructions.

  • Remove author names from changelog.

v1.17.0 (2020-04-10)

  • Update release script and changelog template.

  • Apply pre-filters in all first passes, fixes #118.

    This allows properly reading the level for any kind of normalization, even if filters affect the loudness in the first pass.

v1.16.0 (2020-04-07)

  • Add all commits to changelog.

  • Remove python 2 support.

  • Add quiet option, fixes #116.

    • Add a new quiet option
    • Promote some warnings to actual errors that need to be shown
    • Add a very basic test case

v1.15.8 (2020-03-15)

  • Improve release script.

  • Python 3.8.

v1.15.7 (2020-03-14)

  • Only print length warning for non-EBU type normalization.

v1.15.6 (2019-12-04)

  • Remove build and dist folder on release.

  • Do not exit on error in batch processing.

    Simply process the next file if one has errors, addresses #110.

v1.15.5 (2019-11-19)

  • Use minimal dependency for tqdm.

  • Remove specific python version requirement.

v1.15.4 (2019-11-19)

  • Freeze tqdm version.

  • Update python to 3.7.

  • Improve release documentation.

v1.15.3 (2019-10-15)

  • Do not print stream warning when there is only one stream.

  • Remove previous dist versions before release.

v1.15.2 (2019-07-12)

  • Warn when duration cannot be read, fixes #105.

  • Update README.

    minor improvements in the description

v1.15.1 (2019-06-17)

  • Add output to unit test failures.

  • Fix input label for audio stream.

v1.15.0 (2019-06-17)

  • Add pre-and post-filter hooks, fixes #67.

    This allows users to specify filters to be run before or after the actual normalization call, using regular ffmpeg syntax. Only applies to audio.

  • Document audiostream class.

  • Warn when file is too short, fixes #87.

  • Update release method to twine.

v1.14.1 (2019-06-14)

  • Handle progress output from ffmpeg, fixes #10.

  • Merge pull request #99 from Nottt/patch-1.

    fix -cn description

  • Fix -cn description.

  • Add nicer headers for options in README.

v1.14.0 (2019-04-24)

  • Add version file in release script before committing.

  • Add option to keep original audio, fixes #83.

  • Add pypi badge.

  • Allow release script to add changelog for future version; upload to pypi.

  • Add release script.

  • Add small developer guide on releasing.

  • Move HISTORY.md to CHANGELOG.md.

  • Fix ffmpeg static build download location.

v1.3.10 (2019-02-22)

  • Bump version.

  • Cap measured input loudness, fixes #92.

v1.3.9 (2019-01-10)

  • Bump version.

  • Fix handling of errors with tqdm.

  • Improve readme.

  • Delete issue template.

  • Bump version.

  • Clarify extra argument options, move to main entry point.

  • Update issue templates.

v1.3.8 (2018-11-28)

  • Bump version.

  • Clarify extra argument options, move to main entry point.

v1.3.7 (2018-10-28)

  • Bump version.

  • Copy metadata from individual streams, fixes #86.

  • Add python version for pyenv.

v1.3.6 (2018-07-09)

  • Bump version.

  • Update README, fixes #79 and addresses #80.

v1.3.5 (2018-06-12)

  • Bump version.

  • Minor README updates.

  • Fix documentation of TMPDIR parameter.

v1.3.4 (2018-05-05)

  • Bump version.

  • New way to specify extra options.

v1.3.3 (2018-05-05)

  • Update README.

  • Decode strings in extra options.

v1.3.2 (2018-04-25)

  • Bump version.

  • Merge pull request #69 from UbiCastTeam/master.

    Stderror decoding ignoring utf8 encoding errors

  • Stderror decoding ignoring utf8 encoding errors.

v1.3.1 (2018-04-24)

  • Bump version.

  • Do not require main module in setup.py, fixes #68.

v1.3.0 (2018-04-15)

  • Bump version.

  • Remove dead code.

  • Fix for python2 division.

  • Update documentation.

  • Progress bar.

  • Remove imports from test file.

  • Fix travis file.

  • WIP: progress bar.

  • Minor typo in option group.

  • Add simple unit test for disabling chapters.

v1.2.3 (2018-04-11)

  • Fix unit test.

  • Bump version.

  • Add option to disable chapters, fixes #65, also fix issue with metadata.

v1.2.2 (2018-04-10)

  • Bump version.

  • Set default loudness target to -23, fixes #48.

v1.2.1 (2018-04-04)

  • Bump version.

  • Merge pull request #64 from UbiCastTeam/encoding-issue.

    Stdout and stderror decoding ignoring utf8 encoding errors

  • Stdout and stderror decoding ignoring utf8 encoding errors.

v1.2.0 (2018-03-22)

  • Bump version.

  • Add errors for impossible format combinations, fixes #60.

  • Fix ordering of output maps, fixes #63.

  • Improve documentation.

v1.1.0 (2018-03-06)

  • Add option to print first pass statistics.

v1.0.10 (2018-03-04)

  • Bump version.

  • Restrict parsing to valid JSON part only, fixes #61.

  • Add an example for MP3 encoding.

  • Update paypal link.

v1.0.9 (2018-02-08)

  • Bump version.

  • Add normalized folder to gitignore.

  • Do not print escape sequences on Windows.

  • Do not check for file existence, fixes #57.

  • Add github issue template.

v1.0.8 (2018-02-01)

  • Bump version.

  • Do not check for ffmpeg upon module import.

v1.0.7 (2018-02-01)

  • Bump version.

  • Rename function test.

  • Fix issue with wrong adjustment parameters, fixes #54.

v1.0.6 (2018-01-30)

  • Allow setting FFMPEG_PATH and document TMP.

v1.0.5 (2018-01-26)

  • Handle edge case for short input clips.

v1.0.4 (2018-01-26)

  • Bump version.

  • Do not try to remove nonexisting file in case of error in command.

v1.0.3 (2018-01-26)

  • Bump version.

  • Always streamcopy when detecting streams to avoid initializing encoder.

  • Fix handling of temporary file.

  • Add build status.

  • Travis tests.

v1.0.2 (2018-01-25)

  • Fix bug with target level for peak/RMS.

  • Update documentation formatting.

  • Update history.

v1.0.1 (2018-01-24)

  • Bump version.

  • Set default target to -23.

v1.0.0 (2018-01-23)

  • Add version info and test case for dry run.

  • New feature detection, add documentation, contributors guide etc.

  • WIP: v1.0 rewrite.

v0.7.3 (2017-10-09)

  • Use shutil.move instead of os.rename.

v0.7.2 (2017-09-17)

  • Allow setting threshold to 0.

v0.7.1 (2017-09-14)

  • Bump version.

  • Update HISTORY.md.

  • Merge pull request #37 from Mathijsz/fix-which-path-expansion.

    expand tilde and environment variables, fixes #36

  • Expand tilde and environment variables, fixes #36.

  • Update HISTORY.md.

  • Update README w.r.t. loudnorm filter.

  • Update README and indentation.

v0.7.0 (2017-08-02)

  • Bump version.

  • Fix handling of extra options with spaces.

  • Include test script.

  • Logging and other improvements.

  • Add test files.

  • Autopep8 that thing.

  • Logger improvements.

  • Add example for overwriting.

v0.6.0 (2017-07-31)

  • Allow overwriting input file, fixes #22.

  • Version bump.

  • Better handle cmd arguments.

  • Update README.md.

    add another example

v0.5.1 (2017-04-04)

  • Fix for problem introduced in 304e8df.

v0.5 (2017-04-02)

  • Fix pypi topics.

  • Bump version and README.

  • Fix issue where output was wrong format.

  • Add EBU R128 filter.

  • Use Markdown instead of RST for README/HISTORY.

  • Define file encode for python3, fixes #24.

  • Fix history.

  • Fix option -np.

  • Clarify merge option.

  • Minor documentation improvements.

    • change README from CRLF to LF
    • add "attenuated" in description
    • extend LICENSE year
    • add license to main README

v0.4.1 (2017-02-13)

  • Update for release.

  • Merge pull request #21 from mpuels/patch-1.

    Fix for #13

  • Fix for #13.

  • Mention Python 3.

    mention that Python 3 may work, just didn't have time to test

  • Fix README's code blocks.

v0.4 (2017-01-24)

  • Code cleanup, add option to set format and audio codec.

v0.3 (2017-01-19)

  • Add option for no prefix, fixes #20.

  • Handle multiple spaces in path; fixes issue #18.

  • Handle spaces in path, fixes #12.

  • Update README.rst.

  • Change default level back to -26.

  • Typo in README example.

  • Update documentation.

  • Bump to v0.2.0.

    • Support for multiple files and output directories.
    • Support merging of audio with input file
    • Set audio codec and additional options
    • User-definable threshold
    • Better error handling and logging
    • Deprecates avconv
  • Change default level back to -28.

  • Merge pull request #15 from auricgoldfinger/master.

    Add extended normalisation options

  • Add extended normalisation options.

    • add program option to write output in a separate directory in stead of prefixing it

    • add program option to merge the normalized audio in the original (video) file rather than creating a separate WAV file

    • change the maximum setting: will now normalize so that max volume is set to 0, adjusted with the given level. e.g. : -m -l -5 will increase the audio level to max = -5.0dB

    • improve verbose logging: number of files are written to the info log

    • improve performance: check first whether the output file exists before calculating the volume levels + not modifying the file if the adjustment < 0.5dB (level is never exactly 0)

  • Update README, fixes #11.

v0.1.3 (2015-12-15)

  • Check for Windows .exe, fixes #10.

  • Check path and fix #9.

  • Merge pull request #8 from benjaoming/master.

    Add MANIFEST.in

  • Bump version.

  • Add manifest to include missing files in sdist.

  • Merge pull request #6 from jetpks/master.

    Fixed ffmpeg v2.6.3 compatibility and docopt config

  • Updated to work with ffmpeg v2.6.3, and fixed broken docopt config.

    ffmpeg update:

    ffmpeg v2.6.3 puts mean_volume on stderr instead of stdout, causing output in ffmpeg_get_mean to be completely empty, and no match for mean_volume or max_volume to be found.

    Fixed by adding stderr=subprocess.PIPE in both Popen calls in run_command, and combining stdout and stderr on return. We already exit with non-zero return, so combining stderr/stdout shouldn't cause any poor side-effects.

    docopt config:

    • args['--level'] was not recognizing its default because there was an errant comma between -l and --level, and it needed <level> after the arguments.
    • Fixed spacing for --max
    • Removed quotes around 'normalized' so single quote characters don't end up in the output file names.
  • Removed Windows carraige returns from main.py.

  • Merge pull request #5 from mvbattista/master.

    Installation update to ffmpeg

  • Installation update to ffmpeg.

  • Update to ffmpeg.

  • Update HISTORY.rst.

  • Update to ffmpeg.

  • Merge pull request #4 from benjaoming/rename.

    Rename project

  • Make at least one file mandatory.

  • Rename project and remove pyc file.

  • Merge pull request #2 from benjaoming/docopt-setuptools-avconv.

    Docopt, Setuptools, avconv compatibility

  • Use docopt.

  • Use normalize-audio when using avconv because it doesn't have a way to measure volume.

  • Functional setup.py, communicate with avconv/ffmpeg about overwriting.

  • Also detect avconv.

  • Use a main function instead.

  • Add a history for the project.

  • Move to more unique module name.

  • Update README.rst.

  • Change the README to rst (PyPi)

  • Delete .gitignore.

  • Update README.md.

  • Various improvements, fixes #1.

  • License.

  • Livense.

  • Update README.md.

  • Merge branch 'master' of https://github.com/slhck/audio-normalize.

  • Initial commit.

  • Initial commit.

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