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MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
The zip file provided here only contains the sources of MediaSPIP in its standalone version.
To get a working installation, you must manually install all-software dependencies on the server.
If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...) -
MediaSPIP version 0.1 Beta
16 avril 2011, parMediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
Amélioration de la version de base
13 septembre 2013Jolie sélection multiple
Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...)
Sur d’autres sites (15790)
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Google Speech API returns empty result for some FLAC files, and not for the others although they have same codec and sample rate
15 mars 2021, par ChadBelow code is what I used to make request for transcription.


import io
from google.cloud import speech_v1p1beta1 as speech
def transcribe_file(speech_file):
 """Transcribe the given audio file."""

 client = speech.SpeechClient()

 encoding = speech.RecognitionConfig.AudioEncoding.FLAC
 if os.path.splitext(speech_file)[1] == ".wav":
 encoding = speech.RecognitionConfig.AudioEncoding.LINEAR16
 with io.open(speech_file, "rb") as audio_file:
 content = audio_file.read()

 audio = speech.RecognitionAudio(content=content)
 config = speech.RecognitionConfig(
 encoding=speech.RecognitionConfig.AudioEncoding.FLAC,
 sample_rate_hertz=32000,
 language_code="ja-JP",
 max_alternatives=3,
 enable_word_time_offsets=True,
 enable_automatic_punctuation=True,
 enable_word_confidence=True,
 )

 response = client.recognize(config=config, audio=audio)
 #print(speech_file, "Recognition Done")
 return response



As I wrote in title, the results of response has empty list for some files, and not for some files.
They have same sample rate and codec(32000, FLAC)


Below is the result of
ffprobe -i "AUDIOFILE" -show_streams
for one of each cases.

Left one is empty one. The only difference is duration of file.


How can I get non empty results ?




Edit :


Result of ffprobe show stream show format


Something not captured in one screen


Sadly, re-mux didn't work.


I used ffmpeg-git-20210225


ffbrobe result of broken one


./ffprobe -show_streams -show_format broken.flac 
ffprobe version N-56320-ge937457b7b-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2007-2021 the FFmpeg developers
 built with gcc 8 (Debian 8.3.0-6)
 configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
 libavutil 56. 66.100 / 56. 66.100
 libavcodec 58.125.101 / 58.125.101
 libavformat 58. 68.100 / 58. 68.100
 libavdevice 58. 12.100 / 58. 12.100
 libavfilter 7.107.100 / 7.107.100
 libswscale 5. 8.100 / 5. 8.100
 libswresample 3. 8.100 / 3. 8.100
 libpostproc 55. 8.100 / 55. 8.100
Input #0, flac, from 'broken.flac':
 Metadata:
 encoder : Lavf58.45.100
 Duration: 00:00:00.90, start: 0.000000, bitrate: 342 kb/s
 Stream #0:0: Audio: flac, 32000 Hz, mono, s16
[STREAM]
index=0
codec_name=flac
codec_long_name=FLAC (Free Lossless Audio Codec)
profile=unknown
codec_type=audio
codec_tag_string=[0][0][0][0]
codec_tag=0x0000
sample_fmt=s16
sample_rate=32000
channels=1
channel_layout=mono
bits_per_sample=0
id=N/A
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/32000
start_pts=0
start_time=0.000000
duration_ts=28672
duration=0.896000
bit_rate=N/A
max_bit_rate=N/A
bits_per_raw_sample=16
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=broken.flac
nb_streams=1
nb_programs=0
format_name=flac
format_long_name=raw FLAC
start_time=0.000000
duration=0.896000
size=38362
bit_rate=342517
probe_score=100
TAG:encoder=Lavf58.45.100
[/FORMAT]



ffprobe result of non_broken one


./ffprobe -show_streams -show_format non_broken.flac 
ffprobe version N-56320-ge937457b7b-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2007-2021 the FFmpeg developers
 built with gcc 8 (Debian 8.3.0-6)
 configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
 libavutil 56. 66.100 / 56. 66.100
 libavcodec 58.125.101 / 58.125.101
 libavformat 58. 68.100 / 58. 68.100
 libavdevice 58. 12.100 / 58. 12.100
 libavfilter 7.107.100 / 7.107.100
 libswscale 5. 8.100 / 5. 8.100
 libswresample 3. 8.100 / 3. 8.100
 libpostproc 55. 8.100 / 55. 8.100
Input #0, flac, from 'non_broken.flac':
 Metadata:
 encoder : Lavf58.45.100
 Duration: 00:00:00.86, start: 0.000000, bitrate: 358 kb/s
 Stream #0:0: Audio: flac, 32000 Hz, mono, s16
[STREAM]
index=0
codec_name=flac
codec_long_name=FLAC (Free Lossless Audio Codec)
profile=unknown
codec_type=audio
codec_tag_string=[0][0][0][0]
codec_tag=0x0000
sample_fmt=s16
sample_rate=32000
channels=1
channel_layout=mono
bits_per_sample=0
id=N/A
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/32000
start_pts=0
start_time=0.000000
duration_ts=27648
duration=0.864000
bit_rate=N/A
max_bit_rate=N/A
bits_per_raw_sample=16
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=non_broken.flac
nb_streams=1
nb_programs=0
format_name=flac
format_long_name=raw FLAC
start_time=0.000000
duration=0.864000
size=38701
bit_rate=358342
probe_score=100
TAG:encoder=Lavf58.45.100
[/FORMAT]



And the result of
ffmpeg -f lavfi -i sine=d=0.864:r=32000 output.flac


ffmpeg version 3.4.8-0ubuntu0.2 Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
 configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
 WARNING: library configuration mismatch
 avcodec configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared --enable-version3 --disable-doc --disable-programs --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libtesseract --enable-libvo_amrwbenc
 libavutil 55. 78.100 / 55. 78.100
 libavcodec 57.107.100 / 57.107.100
 libavformat 57. 83.100 / 57. 83.100
 libavdevice 57. 10.100 / 57. 10.100
 libavfilter 6.107.100 / 6.107.100
 libavresample 3. 7. 0 / 3. 7. 0
 libswscale 4. 8.100 / 4. 8.100
 libswresample 2. 9.100 / 2. 9.100
 libpostproc 54. 7.100 / 54. 7.100
Input #0, lavfi, from 'sine=d=0.864:r=32000':
 Duration: N/A, start: 0.000000, bitrate: 512 kb/s
 Stream #0:0: Audio: pcm_s16le, 32000 Hz, mono, s16, 512 kb/s
File 'output.flac' already exists. Overwrite ? [y/N] y
Stream mapping:
 Stream #0:0 -> #0:0 (pcm_s16le (native) -> flac (native))
Press [q] to stop, [?] for help
Output #0, flac, to 'output.flac':
 Metadata:
 encoder : Lavf57.83.100
 Stream #0:0: Audio: flac, 32000 Hz, mono, s16, 128 kb/s
 Metadata:
 encoder : Lavc57.107.100 flac
[Parsed_sine_0 @ 0x55c317ddda00] EOF timestamp not reliable
size= 16kB time=00:00:00.86 bitrate= 154.0kbits/s speed= 205x 
video:0kB audio:8kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 99.364586%



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Start RTSP server on Linux with USB Web-cam source [on hold]
4 juin 2019, par NecroFillinI would like to write a script that runs the RTSP server on a PC with Linux, using ffmpeg, gstreamer, or other media framework.
- Video source : Web camera connected via USB to PC.
- Provide the ability to view the video stream using VLC or another RTSP client.
- Dump traffic using tcpdump
What have I done for now :
streamed the USB camera to a rtsp server by using ffmpeg :ffmpeg -f v4l2 -input_format h264 -timestamps abs -video_size hd720 -i /dev/video0 -c:v copy -c:a none -f rtsp rtsp://10.52.9.104:45002/cameraTx1
What to do next ? Is it possible to make some bush-script for it ?
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Can you force ffmpeg hls demuxing to start from the beginning in a live hls stream ?
13 octobre 2016, par jdramerWhen remuxing an hls stream into an mp4 file I use the following command.
ffmpeg -i "http://example.com/master.m3u8" -c copy -bsf:a aac_adtstoasc output.mp4
This works fine for VOD content, but if the stream is live it starts from the live position, rather than the very first segment in the m3u8 file. Does the applehttp demuxer have any parameters that would force starting from the first segment ?