Read input from stdin (input is provided by FFmpeg via pipe) and write output to Wave file:įfmpeg.exe -i "movie.mkv" -loglevel quiet -vn -f s16le -c:a pcm_s16le - | DynamicAudioNormalizerCLI.exe -i -input-bits 16 -input-chan 2 -input-rate 48000 -o "output.wav" Read input from Wave file and write output to a Wave file again:ĭynamicAudioNormalizerCLI.exe -i "in_original.wav" -o "out_normalized.wav" Also see to the configuration chapter for more details! When reading from the stdin, you have to explicitly specify the input sample format, channel count and sampling rate.įor a list of all available options, please run DynamicAudioNormalizerCLI.exe -help from the command prompt. Just specify the file name "-" in order to read from or write to the stdin or stdout stream, respectively. Passing "raw" PCM data via pipe is supported too. But take care, an existing output file will be overwritten!Īlso note that the Dynamic Audio Normalizer program uses libsndfile for input/output, so a wide range of file formats (WAV, W64, FLAC, Ogg/Vorbis, AIFF, AU/SND, etc) as well as various sample types (ranging from 8-Bit Integer to 64-Bit floating point) are supported. Note that the input file and the output file always have to be specified, while all other parameters are optional. The basic Dynamic Audio Normalizer command-line syntax is as follows:ĭynamicAudioNormalizerCLI.exe -i -o Fixed bugs.Dynamic Audio Normalizer program can be invoked via command-line interface (CLI), either manually from the command prompt or automatically by a batch file. What's New: Added support of ID3 Tags ("User Text", "Encoder by", "Modifed by", "Orig Artist", "Orig Album", "Orig Filename", "Orig Lyricist", "Orig Year", "Lyricist", "SubTitle", "Content Group", "Key", "BPM", "Publisher", "Copyright", "User URL", "ISRC", "Play Count", "Album Artist", "Disc", "Conductor"). Here are some key features of "Sound Normalizer":īatch processor for Mp3, Mp4, FLAC, Ogg, APE, AAC, Wav (PCM 8, 16, 24, 32 bits, DSP, GSM, IMA ADPCM, MS ADPCM, AC3, MP3, MP2) files īatch normalizing for Mp3, Mp4, FLAC, Ogg, APE, AAC, Wav (PCM 8, 16, 24, 32 bits, DSP, GSM, IMA ADPCM, MS ADPCM, AC3, MP3, MP2) files īatch converting for Mp3, Mp4, FLAC, Ogg, APE, AAC, Wav (PCM 8, 16, 24, 32 bits, DSP, GSM, IMA ADPCM, MS ADPCM, AC3, MP3, MP2) files īatch test for Mp3, Mp4, FLAC, Ogg, APE, AAC, Wav (PCM 8, 16, 24, 32 bits, DSP, GSM, IMA ADPCM, MS ADPCM, AC3, MP3, MP2) files. The Sound Normalizer also allows editing ID3, Mp4, FLAC, Ogg Tags, converting Mp3, Mp4, Wav, FLAC, Ogg, APE, AAC files using Lame MP3 Encoder 3.99.2, FLAC Encoder 1.2.1, Monkey's Audio Encoder 4.11, Ogg Vorbis Encoder 1.3.2 (aoTuV 6.03), FAAC Encoder 1.28, listening Mp3, Mp4, FLAC, Ogg, APE, AAC, Wav files using the build-in audio player. The Mp3 Normalizer allows to modify the volume of a scanned file directly without usage tags. The Mp3 normalization and test is fulfilled on an average level (RMS normalization). The Mp4, Wav, Ogg, APE, AAC and FLAC normalization and test is fulfilled on a peak level (Peak Normalization) and on an average level (RMS normalization). The volume level is represented graphically in percentage or decibels (dB). It contains batch processor, which allows to fulfill the batch test, batch normalization and batch converting of Mp3, Mp4, Wav, FLAC, Ogg, APE, AAC files. It is reached by the test and normalization of the volume level of Mp3, Mp4, Wav, FLAC, Ogg, APE, AAC files. The Sound Normalizer increases, reduce, improves, regains a volume and file size without losing ID3, Mp4, FLAC, Ogg tags of Mp3, Mp4, FLAC, Ogg, APE, AAC and Wav (PCM 8, 16, 24, 32 bits, DSP, GSM, IMA ADPCM, MS ADPCM, AC3, MP3, MP2, OGG, A-LAW, u-LAW) files. Sound Normalizer improves volume of Mp3, Mp4, FLAC, Ogg, APE, AAC, Wav files.
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