Hello! I'm trying to nomalize an audio file with FFmpeg. I'm using the loudnorm filter. The source loudness is -23 LUFS and I want to make it -17 LUFS. As far as I know, loudnorm has 2 modes of normalizing audio: linear and dynamic (analysing small parts vs. analysing the whole file). The problem is that when I have an audio file where someone is speaking, the pauses in the speech get louder and louder and thus a hissing noise is clearly audible. Thats why I need linear normalization. But for some reason, that I can't explain, FFmpeg always switches to dynamic mode. I've considered all the requirements for liner scaling in the loudnorm documentation, but I can't figure out whats wrong. I've specified all 4 values, target LRA isn't lower than input LRA, and when I normalize the file in Adobe Audition to -17 LUFS, I can't see any peeking. What would be the best way to get linear normalization with FFmpeg? Here is what I'm doing: 1. Analyze the source audio file: ffmpeg -i input.wav -filter:a loudnorm=I=-17:TP=-1:LRA=9:print_format=json -f null - ffmpeg version N-100942-gc596e82155-gb7251aed46+3 Copyright (c) 2000-2021 the FFmpeg developers built with gcc 10.2.0 (Rev6, Built by MSYS2 project) configuration: --pkg-config=pkgconf --cc='ccache gcc' --cxx='ccache g++' --ld='ccache g++' --disable-autodetect --enable-amf --enable-bzlib --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-iconv --enable-lzma --enable-nvenc --enable-zlib --enable-sdl2 --enable-ffnvcodec --enable-nvdec --enable-cuda-llvm --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libdav1d --enable-libaom --disable-debug --enable-fontconfig --enable-libass --enable-libbluray --enable-libfreetype --enable-libmfx --enable-libmysofa --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libwebp --enable-libxml2 --enable-libzimg --enable-libshine --enable-gpl --enable-avisynth --enable-libxvid --enable-libopenmpt --enable-version3 --enable-chromaprint --enable-decklink --enable-frei0r --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libfdk-aac --enable-libflite --enable-libfribidi --enable-libgme --enable-libgsm --enable-libilbc --enable-libsvthevc --enable-libsvtav1 --enable-libkvazaar --enable-libmodplug --enable-librtmp --enable-librubberband --enable-libtesseract --enable-libxavs --enable-libzmq --enable-libzvbi --enable-openal --enable-libvmaf --enable-libcodec2 --enable-libsrt --enable-ladspa --enable-librav1e --enable-libglslang --enable-vulkan --enable-openssl --extra-cflags=-fopenmp --extra-libs=-lgomp --extra-cflags=-DLIBTWOLAME_STATIC --extra-libs=-lstdc++ --extra-cflags=-DCACA_STATIC --extra-cflags=-DMODPLUG_STATIC --extra-cflags=-DCHROMAPRINT_NODLL --extra-libs=-lstdc++ --extra-cflags=-DZMQ_STATIC --extra-libs=-lpsapi --extra-cflags=-DLIBXML_STATIC --extra-libs=-liconv --disable-w32threads --extra-cflags=-DKVZ_STATIC_LIB --enable-nonfree --extra-cflags=-DAL_LIBTYPE_STATIC --extra-cflags='-ID:/ab-suite/local64/include/AL' libavutil 56. 64.100 / 56. 64.100 libavcodec 58.120.100 / 58.120.100 libavformat 58. 65.101 / 58. 65.101 libavdevice 58. 11.103 / 58. 11.103 libavfilter 7.101.100 / 7.101.100 libswscale 5. 8.100 / 5. 8.100 libswresample 3. 8.100 / 3. 8.100 libpostproc 55. 8.100 / 55. 8.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, wav, from 'input.wav': Metadata: encoded_by : Adobe Adobe Media Encoder 2020.0 encoder : Adobe Adobe Media Encoder 2020.0 (Windows) date : 2021-02-15 creation_time : 15:31:34 time_reference : 0 Duration: 00:37:57.52, bitrate: 1539 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native)) Press [q] to stop, [?] for help Output #0, null, to 'pipe:': Metadata: encoded_by : Adobe Adobe Media Encoder 2020.0 time_reference : 0 date : 2021-02-15 encoder : Lavf58.65.101 Stream #0:0: Audio: pcm_s16le, 192000 Hz, stereo, s16, 6144 kb/s Metadata: encoder : Lavc58.120.100 pcm_s16le size=N/A time=00:37:54.62 bitrate=N/A speed=33.9x video:0kB audio:1708140kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown [Parsed_loudnorm_0 @ 00000121fcf41e00] { "input_i" : "-22.72", "input_tp" : "-2.67", "input_lra" : "6.10", "input_thresh" : "-33.31", "output_i" : "-16.95", "output_tp" : "-1.00", "output_lra" : "6.00", "output_thresh" : "-27.53", "normalization_type" : "dynamic", "target_offset" : "-0.05" } 2. Encode the audio with: ffmpeg -i input.wav -filter:a loudnorm=I=-17:TP=-1:LRA=9:measured_I=-22.72:measured_TP=-2.67:measured_LRA=6.10:measured_thresh=-33.31:offset=-0.05:linear=true:print_format=summary output.wav ffmpeg version N-100942-gc596e82155-gb7251aed46+3 Copyright (c) 2000-2021 the FFmpeg developers built with gcc 10.2.0 (Rev6, Built by MSYS2 project) configuration: --pkg-config=pkgconf --cc='ccache gcc' --cxx='ccache g++' --ld='ccache g++' --disable-autodetect --enable-amf --enable-bzlib --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-iconv --enable-lzma --enable-nvenc --enable-zlib --enable-sdl2 --enable-ffnvcodec --enable-nvdec --enable-cuda-llvm --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libdav1d --enable-libaom --disable-debug --enable-fontconfig --enable-libass --enable-libbluray --enable-libfreetype --enable-libmfx --enable-libmysofa --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libwebp --enable-libxml2 --enable-libzimg --enable-libshine --enable-gpl --enable-avisynth --enable-libxvid --enable-libopenmpt --enable-version3 --enable-chromaprint --enable-decklink --enable-frei0r --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libfdk-aac --enable-libflite --enable-libfribidi --enable-libgme --enable-libgsm --enable-libilbc --enable-libsvthevc --enable-libsvtav1 --enable-libkvazaar --enable-libmodplug --enable-librtmp --enable-librubberband --enable-libtesseract --enable-libxavs --enable-libzmq --enable-libzvbi --enable-openal --enable-libvmaf --enable-libcodec2 --enable-libsrt --enable-ladspa --enable-librav1e --enable-libglslang --enable-vulkan --enable-openssl --extra-cflags=-fopenmp --extra-libs=-lgomp --extra-cflags=-DLIBTWOLAME_STATIC --extra-libs=-lstdc++ --extra-cflags=-DCACA_STATIC --extra-cflags=-DMODPLUG_STATIC --extra-cflags=-DCHROMAPRINT_NODLL --extra-libs=-lstdc++ --extra-cflags=-DZMQ_STATIC --extra-libs=-lpsapi --extra-cflags=-DLIBXML_STATIC --extra-libs=-liconv --disable-w32threads --extra-cflags=-DKVZ_STATIC_LIB --enable-nonfree --extra-cflags=-DAL_LIBTYPE_STATIC --extra-cflags='-ID:/ab-suite/local64/include/AL' libavutil 56. 64.100 / 56. 64.100 libavcodec 58.120.100 / 58.120.100 libavformat 58. 65.101 / 58. 65.101 libavdevice 58. 11.103 / 58. 11.103 libavfilter 7.101.100 / 7.101.100 libswscale 5. 8.100 / 5. 8.100 libswresample 3. 8.100 / 3. 8.100 libpostproc 55. 8.100 / 55. 8.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, wav, from 'input.wav': Metadata: encoded_by : Adobe Adobe Media Encoder 2020.0 encoder : Adobe Adobe Media Encoder 2020.0 (Windows) date : 2021-02-15 creation_time : 15:31:34 time_reference : 0 Duration: 00:37:57.52, bitrate: 1539 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s File 'output.wav' already exists. Overwrite? [y/N] y Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native)) Press [q] to stop, [?] for help Output #0, wav, to 'output.wav': Metadata: ITCH : Adobe Adobe Media Encoder 2020.0 time_reference : 0 ICRD : 2021-02-15 ISFT : Lavf58.65.101 Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 192000 Hz, stereo, s16, 6144 kb/s Metadata: encoder : Lavc58.120.100 pcm_s16le size= 1708140kB time=00:37:54.62 bitrate=6151.8kbits/s speed=33.6x video:0kB audio:1708140kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000009% [Parsed_loudnorm_0 @ 000001a3673c0780] Input Integrated: -22.7 LUFS Input True Peak: -2.7 dBTP Input LRA: 6.1 LU Input Threshold: -33.3 LUFS Output Integrated: -17.0 LUFS Output True Peak: -1.0 dBTP Output LRA: 6.0 LU Output Threshold: -27.6 LUFS Normalization Type: Dynamic Target Offset: -0.0 LU
On Mon, Feb 22, 2021 at 6:52 PM Jonathan Baecker <jonbae77@gmail.com> wrote:
Hello!
I'm trying to nomalize an audio file with FFmpeg. I'm using the loudnorm filter. The source loudness is -23 LUFS and I want to make it -17 LUFS. As far as I know, loudnorm has 2 modes of normalizing audio: linear and dynamic (analysing small parts vs. analysing the whole file).
The problem is that when I have an audio file where someone is speaking, the pauses in the speech get louder and louder and thus a hissing noise is clearly audible. Thats why I need linear normalization. But for some reason, that I can't explain, FFmpeg always switches to dynamic mode.
I've considered all the requirements for liner scaling in the loudnorm documentation, but I can't figure out whats wrong. I've specified all 4 values, target LRA isn't lower than input LRA, and when I normalize the file in Adobe Audition to -17 LUFS, I can't see any peeking. What would be the best way to get linear normalization with FFmpeg?
Could provide input.wav somehow?
Here is what I'm doing:
1. Analyze the source audio file:
ffmpeg -i input.wav -filter:a loudnorm=I=-17:TP=-1:LRA=9:print_format=json -f null - ffmpeg version N-100942-gc596e82155-gb7251aed46+3 Copyright (c) 2000-2021 the FFmpeg developers built with gcc 10.2.0 (Rev6, Built by MSYS2 project) configuration: --pkg-config=pkgconf --cc='ccache gcc' --cxx='ccache g++' --ld='ccache g++' --disable-autodetect --enable-amf --enable-bzlib --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-iconv --enable-lzma --enable-nvenc --enable-zlib --enable-sdl2 --enable-ffnvcodec --enable-nvdec --enable-cuda-llvm --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libdav1d --enable-libaom --disable-debug --enable-fontconfig --enable-libass --enable-libbluray --enable-libfreetype --enable-libmfx --enable-libmysofa --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libwebp --enable-libxml2 --enable-libzimg --enable-libshine --enable-gpl --enable-avisynth --enable-libxvid --enable-libopenmpt --enable-version3 --enable-chromaprint --enable-decklink --enable-frei0r --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libfdk-aac --enable-libflite --enable-libfribidi --enable-libgme --enable-libgsm --enable-libilbc --enable-libsvthevc --enable-libsvtav1 --enable-libkvazaar --enable-libmodplug --enable-librtmp --enable-librubberband --enable-libtesseract --enable-libxavs --enable-libzmq --enable-libzvbi --enable-openal --enable-libvmaf --enable-libcodec2 --enable-libsrt --enable-ladspa --enable-librav1e --enable-libglslang --enable-vulkan --enable-openssl --extra-cflags=-fopenmp --extra-libs=-lgomp --extra-cflags=-DLIBTWOLAME_STATIC --extra-libs=-lstdc++ --extra-cflags=-DCACA_STATIC --extra-cflags=-DMODPLUG_STATIC --extra-cflags=-DCHROMAPRINT_NODLL --extra-libs=-lstdc++ --extra-cflags=-DZMQ_STATIC --extra-libs=-lpsapi --extra-cflags=-DLIBXML_STATIC --extra-libs=-liconv --disable-w32threads --extra-cflags=-DKVZ_STATIC_LIB --enable-nonfree --extra-cflags=-DAL_LIBTYPE_STATIC --extra-cflags='-ID:/ab-suite/local64/include/AL' libavutil 56. 64.100 / 56. 64.100 libavcodec 58.120.100 / 58.120.100 libavformat 58. 65.101 / 58. 65.101 libavdevice 58. 11.103 / 58. 11.103 libavfilter 7.101.100 / 7.101.100 libswscale 5. 8.100 / 5. 8.100 libswresample 3. 8.100 / 3. 8.100 libpostproc 55. 8.100 / 55. 8.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, wav, from 'input.wav': Metadata: encoded_by : Adobe Adobe Media Encoder 2020.0 encoder : Adobe Adobe Media Encoder 2020.0 (Windows) date : 2021-02-15 creation_time : 15:31:34 time_reference : 0 Duration: 00:37:57.52, bitrate: 1539 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native)) Press [q] to stop, [?] for help Output #0, null, to 'pipe:': Metadata: encoded_by : Adobe Adobe Media Encoder 2020.0 time_reference : 0 date : 2021-02-15 encoder : Lavf58.65.101 Stream #0:0: Audio: pcm_s16le, 192000 Hz, stereo, s16, 6144 kb/s Metadata: encoder : Lavc58.120.100 pcm_s16le size=N/A time=00:37:54.62 bitrate=N/A speed=33.9x video:0kB audio:1708140kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown [Parsed_loudnorm_0 @ 00000121fcf41e00] { "input_i" : "-22.72", "input_tp" : "-2.67", "input_lra" : "6.10", "input_thresh" : "-33.31", "output_i" : "-16.95", "output_tp" : "-1.00", "output_lra" : "6.00", "output_thresh" : "-27.53", "normalization_type" : "dynamic", "target_offset" : "-0.05" }
2. Encode the audio with:
ffmpeg -i input.wav -filter:a loudnorm=I=-17:TP=-1:LRA=9:measured_I=-22.72:measured_TP=-2.67:measured_LRA=6.10:measured_thresh=-33.31:offset=-0.05:linear=true:print_format=summary
output.wav ffmpeg version N-100942-gc596e82155-gb7251aed46+3 Copyright (c) 2000-2021 the FFmpeg developers built with gcc 10.2.0 (Rev6, Built by MSYS2 project) configuration: --pkg-config=pkgconf --cc='ccache gcc' --cxx='ccache g++' --ld='ccache g++' --disable-autodetect --enable-amf --enable-bzlib --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-iconv --enable-lzma --enable-nvenc --enable-zlib --enable-sdl2 --enable-ffnvcodec --enable-nvdec --enable-cuda-llvm --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libdav1d --enable-libaom --disable-debug --enable-fontconfig --enable-libass --enable-libbluray --enable-libfreetype --enable-libmfx --enable-libmysofa --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libwebp --enable-libxml2 --enable-libzimg --enable-libshine --enable-gpl --enable-avisynth --enable-libxvid --enable-libopenmpt --enable-version3 --enable-chromaprint --enable-decklink --enable-frei0r --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libfdk-aac --enable-libflite --enable-libfribidi --enable-libgme --enable-libgsm --enable-libilbc --enable-libsvthevc --enable-libsvtav1 --enable-libkvazaar --enable-libmodplug --enable-librtmp --enable-librubberband --enable-libtesseract --enable-libxavs --enable-libzmq --enable-libzvbi --enable-openal --enable-libvmaf --enable-libcodec2 --enable-libsrt --enable-ladspa --enable-librav1e --enable-libglslang --enable-vulkan --enable-openssl --extra-cflags=-fopenmp --extra-libs=-lgomp --extra-cflags=-DLIBTWOLAME_STATIC --extra-libs=-lstdc++ --extra-cflags=-DCACA_STATIC --extra-cflags=-DMODPLUG_STATIC --extra-cflags=-DCHROMAPRINT_NODLL --extra-libs=-lstdc++ --extra-cflags=-DZMQ_STATIC --extra-libs=-lpsapi --extra-cflags=-DLIBXML_STATIC --extra-libs=-liconv --disable-w32threads --extra-cflags=-DKVZ_STATIC_LIB --enable-nonfree --extra-cflags=-DAL_LIBTYPE_STATIC --extra-cflags='-ID:/ab-suite/local64/include/AL' libavutil 56. 64.100 / 56. 64.100 libavcodec 58.120.100 / 58.120.100 libavformat 58. 65.101 / 58. 65.101 libavdevice 58. 11.103 / 58. 11.103 libavfilter 7.101.100 / 7.101.100 libswscale 5. 8.100 / 5. 8.100 libswresample 3. 8.100 / 3. 8.100 libpostproc 55. 8.100 / 55. 8.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, wav, from 'input.wav': Metadata: encoded_by : Adobe Adobe Media Encoder 2020.0 encoder : Adobe Adobe Media Encoder 2020.0 (Windows) date : 2021-02-15 creation_time : 15:31:34 time_reference : 0 Duration: 00:37:57.52, bitrate: 1539 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s File 'output.wav' already exists. Overwrite? [y/N] y Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native)) Press [q] to stop, [?] for help Output #0, wav, to 'output.wav': Metadata: ITCH : Adobe Adobe Media Encoder 2020.0 time_reference : 0 ICRD : 2021-02-15 ISFT : Lavf58.65.101 Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 192000 Hz, stereo, s16, 6144 kb/s Metadata: encoder : Lavc58.120.100 pcm_s16le size= 1708140kB time=00:37:54.62 bitrate=6151.8kbits/s speed=33.6x video:0kB audio:1708140kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000009% [Parsed_loudnorm_0 @ 000001a3673c0780] Input Integrated: -22.7 LUFS Input True Peak: -2.7 dBTP Input LRA: 6.1 LU Input Threshold: -33.3 LUFS
Output Integrated: -17.0 LUFS Output True Peak: -1.0 dBTP Output LRA: 6.0 LU Output Threshold: -27.6 LUFS
Normalization Type: Dynamic Target Offset: -0.0 LU
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I'm trying to nomalize an audio file with FFmpeg. I'm using the loudnorm filter. The source loudness is -23 LUFS and I want to make it -17 LUFS. As far as I know, loudnorm has 2 modes of normalizing audio: linear and
2. Encode the audio with:
ffmpeg -i input.wav -filter:a loudnorm=I=-17:TP=-1:LRA=9:measured_I=-22.72:measured_TP=-2.67:measured_LRA=6.10:measured_thresh=-33.31:offset=-0.05:linear=true:print_format=summary
Why not just add 6 dB to the input? That ’should’ do the trick just fine imho. (As one LU should correspond with 1 dB) Bouke
participants (3)
-
Bouke / edit 'B -
Jonathan Baecker -
Paul B Mahol