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Autres articles (43)
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Websites made with MediaSPIP
2 mai 2011, parThis page lists some websites based on MediaSPIP.
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List of compatible distributions
26 avril 2011, parThe table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...) -
Creating farms of unique websites
13 avril 2011, parMediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...)
Sur d’autres sites (7390)
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avconv / ffmpeg webcam capture while using minimum CPU processing
10 septembre 2015, par user3585723I have a question about avconv (or ffmpeg) usage.
My goal is to capture video from a webcam and saving it to a file.
Also, I don’t want to use too much CPU processing. (I don’t want avconv to scale or re-encode the stream)So, I was thinking to use the compressed mjpeg video stream from the webcam and directly saving it to a file.
My webcam is a Microsoft LifeCam HD 3000 and its capabilities are :
ffmpeg -f v4l2 -list_formats all -i /dev/video0
Raw: yuyv422 : YUV 4:2:2 (YUYV) : 640x480 1280x720 960x544 800x448 640x360 424x240 352x288 320x240 800x600 176x144 160x120 1280x800
Compressed: mjpeg : MJPEG : 640x480 1280x720 960x544 800x448 640x360 800x600 416x240 352x288 176x144 320x240 160x120What would be the avconv command to save the Compressed stream directly without having avconv doing scaling or re-encoding.
For now, I am using this command :
avconv -f video4linux2 -r 30 -s 320x240 -i /dev/video0 test.avi
I’m not sure that this command is CPU efficient since I don’t tell anywhere to use the mjpeg Compressed capability of the webcam.
Is avconv taking care of the configuration of the webcam setting before starting to record the file ? Is it always working of raw stream and doing scaling and enconding on the raw stream ?
Thanks for your answer
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Can I use avcodec_free_context() on an opened context ?
23 mars 2017, par Ashe the humanThe latest documentation says here that opening a context that’s closed again is not supported any more. I see why. Some codecs don’t work properly when they’re reopened. So after finding this bug, I decided to not use
avcodec_close()
and callavcodec_free_context()
on the contexts right away instead.But I’m not sure if it’s safe to do so with 2.8.4, the version that I linked to my program. The documentation from that time doesn’t clarify. Does anyone know ? At least empirically ?
ffmpeg version 2.8.4 Copyright (c) 2000-2015 the FFmpeg developers
built with Microsoft (R) C/C++ 최적화 컴파일러 버전 18.00.31101(x64)
configuration: --toolchain=msvc --enable-gpl --enable-nonfree --enable-nvenc --enable-libvorbis --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-libx265 --enable-libxvid --enable-libopus --enable-libvpx --enable-static --disable-shared --disable-debug --extra-cflags=-MT --extra-cxxflags=-MT --extra-ldflags='/nodefaultlib:msvcrt.lib' --extra-libs='zlib.lib libogg_static.lib libvorbis_static.lib libmpghip-static.lib libmp3lame-static.lib libtheora_static.lib libx264.lib x265-static.lib libxvidcore.lib silk_fixed.lib silk_common.lib silk_float.lib celt.lib opus.lib vpxmt.lib'
libavutil 54. 31.100 / 54. 31.100
libavcodec 56. 60.100 / 56. 60.100
libavformat 56. 40.101 / 56. 40.101
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 40.101 / 5. 40.101
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 2.101 / 1. 2.101
libpostproc 53. 3.100 / 53. 3.100I know there’s a bunch of forums I could post on but I’d felt like to ask it here first.
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FFMPEG command from Python 3.5 does not actually create audio file
20 décembre 2017, par Nathan BlaineI have a Django web application that accepts user uploaded videos/audio and saves them into a folder ’../WebAppDirectory/media/recordings’.
I am then using a speech to text API to get a rough transcription of the audio. This is working fine for .wav and .mp4 files, but the web app also accepts videos (.MOV) that I would like to first convert to .wav, then pass off to the API.
Using ffmpeg from my command line like this
ffmpeg -i C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.MOV -ab 160k -ac 2 -ar 44100 -vn upload_sample.wav
Correctly creates the .wav file from the original .MOV.
However, when I run this from python with
subprocess.check_call(command, shell=True)
ffmpeg responds with
File ’upload_sample.wav’ already exists. Overwrite ? [y/N]
While Python tells me
FileNotFoundError : [Errno 2] No such file or directory : ’C :\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.wav’
It is also worth noting that I do not see a ’upload_sample.wav’ file in the media/recordings/ directory.
This leads me to believe that maybe Python and ffmpeg are looking in different folders, but I am not sure where I am going wrong. When I print the command from the subprocess.check_call and copy/paste it into cmd, the file is created as expected.
Hoping someone with some experience with ffmpeg/Python subprocess can help shed some light ! Here are the files I am working with :
Folder Structure
DjangoWebApp
|---media
|---|---imgs
|---|---recordings
|---|---|---upload_sample.MOV
|---uploaded_audio_to_text.pyuploaded_audio_to_text.py
import speech_recognition as sr
from os import path
import os
import subprocess
def speech_to_text(file_name):
AUDIO_FILE = path.join(path.dirname(path.realpath(__file__)), 'media','recordings', file_name)
print("Looking at path: ",AUDIO_FILE)
# get extension
AUDIO_FILE_EXT = os.path.splitext(AUDIO_FILE)[1]
if(AUDIO_FILE_EXT == '.MOV'):
print("File is not .wav: ", AUDIO_FILE_EXT, "found. Converting...")
# We will use subprocess and ffmpeg to convert this .MOV file to .wav, so we can send to API
temp_wav = os.path.splitext(file_name)[0] + '.wav'
print("New audio file will be: ", temp_wav)
# build CMD ffmpeg command
command = "ffmpeg -i "
command += AUDIO_FILE
command += " -ab 160k -ac 2 -ar 44100 -vn "
command += temp_wav
print("Attempting to run this command: \n",command)
print(subprocess.check_call(command, shell=True))
print("Past Subprocess.call")
AUDIO_FILE = path.join(path.dirname(path.realpath(__file__)), 'media','recordings', temp_wav)
print("AUDIO_FILE now set to: ", AUDIO_FILE)
else:
# continue with what we are doing
pass
r = sr.Recognizer()
with sr.AudioFile(AUDIO_FILE) as source:
audio = r.record(source) # read the entire audio file
text_transcription = "Sentinel"
# recognize speech using Microsoft Bing Voice Recognition
BING_KEY = "MY_KEY_:)"
try:
text_transcription = r.recognize_bing(audio, key=BING_KEY)
except sr.UnknownValueError:
print("Microsoft Bing Voice Recognition could not understand audio")
except sr.RequestError as e:
print("Could not request results from Microsoft Bing Voice Recognition service; {0}".format(e))
return text_transcription
#my tests
my_relative_file_path = "upload_sample.MOV"
print(speech_to_text(my_relative_file_path))Console output (traceback and my print()’s)
Looking at path: C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.MOV
File is not .wav: .MOV found. Converting...
New audio file will be: upload_sample.wav Attempting to run this command:
ffmpeg -i C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.MOV -ab 160k -ac 2 -ar 44100 -vn upload_sample.wav
ffmpeg version git-2017-12-18-74f408c Copyright (c) 2000-2017 the FFmpeg developers built with gcc 7.2.0 (GCC)
----REMOVED SOME FFMPEG OUTPUT FOR BREVITY----
File 'upload_sample.wav' already exists. Overwrite ? [y/N] y
Stream mapping: Stream #0:1 -> #0:0 (aac (native) -> pcm_s16le (native)) Press [q] to stop, [?] for help Output #0, wav, to 'upload_sample.wav': Metadata:
major_brand : qt
minor_version : 0
compatible_brands: qt
com.apple.quicktime.creationdate: 2017-12-19T16:06:10-0500
com.apple.quicktime.make: Apple
com.apple.quicktime.model: iPhone 6
com.apple.quicktime.software: 10.3.3
ISFT : Lavf58.3.100
Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s (default)
Metadata:
creation_time : 2017-12-19T21:06:11.000000Z
handler_name : Core Media Data Handler
encoder : Lavc58.8.100 pcm_s16le size= 1036kB time=00:00:06.01 bitrate=1411.3kbits/s speed=N/A video:0kB audio:1036kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.007352%
0
Traceback (most recent call last): Past Subprocess.call
File "C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\uploaded_audio_to_text.py", line 53, in <module>
AUDIO_FILE now set to: C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.wav
print(speech_to_text(my_relative_file_path))
File "C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\uploaded_audio_to_text.py", line 36, in speech_to_text
with sr.AudioFile(AUDIO_FILE) as source:
File "C:\Users\Nathan\AppData\Local\Programs\Python\Python36-32\lib\site-packages\speech_recognition\__init__.py", line 203, in __enter__
self.audio_reader = wave.open(self.filename_or_fileobject, "rb")
File "C:\Users\Nathan\AppData\Local\Programs\Python\Python36-32\lib\wave.py", line 499, in open
return Wave_read(f)
File "C:\Users\Nathan\AppData\Local\Programs\Python\Python36-32\lib\wave.py", line 159, in __init__
f = builtins.open(f, 'rb')
FileNotFoundError: [Errno 2] No such file or directory: 'C:\\Users\\Nathan\\Desktop\\MeetingRecorderWebAPP\\media\\recordings\\upload_sample.wav'
Process finished with exit code 1
</module>