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Médias (91)
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Richard Stallman et le logiciel libre
19 octobre 2011, par
Mis à jour : Mai 2013
Langue : français
Type : Texte
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Stereo master soundtrack
17 octobre 2011, par
Mis à jour : Octobre 2011
Langue : English
Type : Audio
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Elephants Dream - Cover of the soundtrack
17 octobre 2011, par
Mis à jour : Octobre 2011
Langue : English
Type : Image
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#7 Ambience
16 octobre 2011, par
Mis à jour : Juin 2015
Langue : English
Type : Audio
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#6 Teaser Music
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#5 End Title
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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Support de tous types de médias
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Sur d’autres sites (7099)
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How to convert aac to ogg opus keeping bit rate and sample rate unchanged
25 juin 2019, par DooviI’m trying to convert a .aac file to .opus but after inspecting with ffprobe I get different bit and sample rates.
While input file’s audio stream bit rate is 245995, the output file’s audio stream has no bit rate specified - "format" shows bit rate of 118788.
While input file’s audio stream sample rate is 44100, the output’s is 48000.
ffprobe -v error -show_format -show_streams input.aac
[STREAM]
index=0
codec_name=aac
codec_long_name=AAC (Advanced Audio Coding)
profile=LC
codec_type=audio
codec_time_base=1/44100
codec_tag_string=[0][0][0][0]
codec_tag=0x0000
sample_fmt=fltp
sample_rate=44100
channels=2
channel_layout=stereo
bits_per_sample=0
id=N/A
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/28224000
start_pts=N/A
start_time=N/A
duration_ts=106533390807
duration=3774.567418
bit_rate=245995
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=N/A
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=0
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
[/STREAM]
[FORMAT]
filename=input.aac
nb_streams=1
nb_programs=0
format_name=aac
format_long_name=raw ADTS AAC (Advanced Audio Coding)
start_time=N/A
duration=3774.567418
size=116065589
bit_rate=245995
probe_score=51
[/FORMAT]ffmpeg -nostdin -i input.aac -c:a libopus output.opus
ffmpeg version N-93449-g013f714 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 7 (Ubuntu 7.3.0-27ubuntu1~18.04)
configuration: --prefix=/home/vagrant/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/vagrant/ffmpeg_build/include --extra-ldflags=-L/home/vagrant/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/vagrant/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree
libavutil 56. 26.100 / 56. 26.100
libavcodec 58. 47.105 / 58. 47.105
libavformat 58. 26.101 / 58. 26.101
libavdevice 58. 7.100 / 58. 7.100
libavfilter 7. 48.100 / 7. 48.100
libswscale 5. 4.100 / 5. 4.100
libswresample 3. 4.100 / 3. 4.100
libpostproc 55. 4.100 / 55. 4.100
[aac @ 0x55d4b7e21d80] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'input.aac':
Duration: 01:02:54.57, bitrate: 245 kb/s
Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 245 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (aac (native) -> opus (libopus))
[libopus @ 0x55d4b7e3f8c0] No bit rate set. Defaulting to 96000 bps.
Output #0, opus, to 'output.opus':
Metadata:
encoder : Lavf58.26.101
Stream #0:0: Audio: opus (libopus), 48000 Hz, stereo, flt, 96 kb/s
Metadata:
encoder : Lavc58.47.105 libopus
size= 52103kB time=00:59:53.21 bitrate= 118.8kbits/s speed=66.2x
video:0kB audio:51733kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.715930%ffprobe -v error -show_format -show_streams output.opus
[STREAM]
index=0
codec_name=opus
codec_long_name=Opus (Opus Interactive Audio Codec)
profile=unknown
codec_type=audio
codec_time_base=1/48000
codec_tag_string=[0][0][0][0]
codec_tag=0x0000
sample_fmt=fltp
sample_rate=48000
channels=2
channel_layout=stereo
bits_per_sample=0
id=N/A
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/48000
start_pts=0
start_time=0.000000
duration_ts=172473677
duration=3593.201604
bit_rate=N/A
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=N/A
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=0
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
TAG:ENCODER=Lavc58.47.105 libopus
[/STREAM]
[FORMAT]
filename=output.opus
nb_streams=1
nb_programs=0
format_name=ogg
format_long_name=Ogg
start_time=0.000000
duration=3593.201604
size=53353867
bit_rate=118788
probe_score=100
[/FORMAT]How can I preserve the quality of the input file ? Am I missing something in the ffmpeg cmd ?
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ffmpeg Output file #0 does not contain any stream when trying to access 1 of 2 audio streams
7 juillet 2019, par nulltorpedoffmpeg -i input.mkv -map 0:2 -c copy -strict -2 audio.mkv
Hi I have the above command. The output shows that there are 2 audio streams. I want to copy just the ac3 audio (actually I want to convert it but even this copy does not work). I have truncated the output print where there is metadata
NEW updated sample with full log which results in same message
ffmpeg -i input.mka -map 0:0 -c:a libfdk_aac aac_out.m4a
ffmpeg version 2.7.1 Copyright (c) 2000-2015 the FFmpeg developers
built with gcc 4.9.3 (crosstool-NG 1.20.0) 20150311 (prerelease)
configuration: --prefix=/usr --incdir='${prefix}/include/ffmpeg' --arch=i686 --target-os=linux --cross-prefix=/usr/local/x86_64-pc-linux-gnu/bin/x86_64-pc-linux-gnu- --enable-cross-compile --enable-optimizations --enable-pic --enable-gpl --enable-shared --disable-static --enable-version3 --enable-nonfree --enable-libfaac --enable-encoders --enable-pthreads --disable-bzlib --disable-protocol=rtp --disable-muxer=image2 --disable-muxer=image2pipe --disable-swscale-alpha --disable-ffserver --disable-ffplay --disable-devices --disable-bzlib --disable-altivec --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libmp3lame --disable-vaapi --disable-decoder=amrnb --disable-decoder=ac3 --disable-decoder=ac3_fixed --disable-encoder=zmbv --disable-encoder=dca --disable-encoder=ac3 --disable-encoder=ac3_fixed --disable-encoder=eac3 --disable-decoder=dca --disable-decoder=eac3 --disable-decoder=truehd --cc=/usr/local/x86_64-pc-linux-gnu/bin/x86_64-pc-linux-gnu-ccache-gcc --enable-yasm --enable-libx264 --enable-encoder=libx264
libavutil 54. 27.100 / 54. 27.100
libavcodec 56. 41.100 / 56. 41.100
libavformat 56. 36.100 / 56. 36.100
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 16.101 / 5. 16.101
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 2.100 / 1. 2.100
libpostproc 53. 3.100 / 53. 3.100
Input #0, matroska,webm, from '/volume1/..../input.mka':
Metadata:
encoder : libebml v1.3.9 + libmatroska v1.5.2
creation_time : 2019-07-07 06:19:20
Duration: 02:29:21.98, start: 0.000000, bitrate: 640 kb/s
Stream #0:0(eng): Audio: ac3, 48000 Hz, 5.1(side), 640 kb/s
Metadata:
BPS-eng : 640000
DURATION-eng : 02:29:21.984000480
NUMBER_OF_FRAMES-eng: 280062
NUMBER_OF_BYTES-eng: 716958720
_STATISTICS_WRITING_APP-eng: mkvmerge v35.0.0 ('All The Love In The World') 64-bit
_STATISTICS_WRITING_DATE_UTC-eng: 2019-07-07 06:19:20
_STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
Output #0, ipod, to 'aac_out.m4a':
Metadata:
encoder : libebml v1.3.9 + libmatroska v1.5.2
Output file #0 does not contain any stream -
Audio stuttering every couple of seconds
27 juin 2019, par GlutchI’m merging a video (recorded with ffmpeg, good quality, all solid), with a musicfile.mp3. However every couple of seconds the music stutters and skips slightly. Which seems very strange since simply adding music on top of a video sounds like the engine could relax and take its time, creating no artifacts. (In comparison to recording live desktop footage). Can anyone help me sort this out ?
System : MacOS MBP 2015, 16gb ram 2.7ghz i5
ffmpeg -i "temp/1561246948349.mkv" -i "music/happy.mp3" -vcodec copy -filter_complex amix -map 0:v -map 0:a -map 1:a -shortest -b:a 144k "finished/2019-06-22/1561246948349/output.mkv"
ffmpeg version 4.1.3 Copyright (c) 2000-2019 the FFmpeg developers
built with Apple LLVM version 10.0.1 (clang-1001.0.46.4)
configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1.3_1 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags='-I/Library/Java/JavaVirtualMachines/adoptopenjdk-11.0.2.jdk/Contents/Home/include -I/Library/Java/JavaVirtualMachines/adoptopenjdk-11.0.2.jdk/Contents/Home/include/darwin' --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-videotoolbox --disable-libjack --disable-indev=jack --enable-libaom --enable-libsoxr
libavutil 56. 22.100 / 56. 22.100
libavcodec 58. 35.100 / 58. 35.100
libavformat 58. 20.100 / 58. 20.100
libavdevice 58. 5.100 / 58. 5.100
libavfilter 7. 40.101 / 7. 40.101
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 3.100 / 5. 3.100
libswresample 3. 3.100 / 3. 3.100
libpostproc 55. 3.100 / 55. 3.100
Input #0, matroska,webm, from 'temp/1561246948349.mkv':
Metadata:
ENCODER : Lavf58.20.100
Duration: 00:00:21.50, start: 0.000000, bitrate: 5834 kb/s
Stream #0:0: Video: h264 (High 4:4:4 Predictive), yuv422p(progressive), 2880x1800, 30 fps, 30 tbr, 1k tbn, 2000k tbc (default)
Metadata:
ENCODER : Lavc58.35.100 libx264
DURATION : 00:00:21.467000000
Stream #0:1: Audio: vorbis, 44100 Hz, stereo, fltp (default)
Metadata:
ENCODER : Lavc58.35.100 libvorbis
DURATION : 00:00:21.496000000
Input #1, mp3, from 'music/happy.mp3':
Metadata:
album : Random
genre : Jazz & Blues
Duration: 00:15:59.84, start: 0.025057, bitrate: 186 kb/s
Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 186 kb/s
Metadata:
encoder : LAME3.100
Stream mapping:
Stream #0:1 (vorbis) -> amix:input0
Stream #1:0 (mp3float) -> amix:input1
amix -> Stream #0:0 (libvorbis)
Stream #0:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
Output #0, matroska, to 'finished/2019-06-22/1561246948349/output.mkv':
Metadata:
encoder : Lavf58.20.100
Stream #0:0: Audio: vorbis (libvorbis) (oV[0][0] / 0x566F), 44100 Hz, stereo, fltp, 144 kb/s (default)
Metadata:
encoder : Lavc58.35.100 libvorbis
Stream #0:1: Video: h264 (High 4:4:4 Predictive) (H264 / 0x34363248), yuv422p(progressive), 2880x1800, q=2-31, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
Metadata:
ENCODER : Lavc58.35.100 libx264
DURATION : 00:00:21.467000000
frame= 640 fps=0.0 q=-1.0 Lsize= 15227kB time=00:00:21.46 bitrate=5810.3kbits/s speed=33.8x
video:14888kB audio:318kB subtitle:0kB other streams:0kB global headers:4kB muxing overhead: 0.139864%