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Des sites réalisés avec MediaSPIP
2 mai 2011, parCette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page. -
Librairies et binaires spécifiques au traitement vidéo et sonore
31 janvier 2010, parLes logiciels et librairies suivantes sont utilisées par SPIPmotion d’une manière ou d’une autre.
Binaires obligatoires FFMpeg : encodeur principal, permet de transcoder presque tous les types de fichiers vidéo et sonores dans les formats lisibles sur Internet. CF ce tutoriel pour son installation ; Oggz-tools : outils d’inspection de fichiers ogg ; Mediainfo : récupération d’informations depuis la plupart des formats vidéos et sonores ;
Binaires complémentaires et facultatifs flvtool2 : (...) -
Changer son thème graphique
22 février 2011, parLe thème graphique ne touche pas à la disposition à proprement dite des éléments dans la page. Il ne fait que modifier l’apparence des éléments.
Le placement peut être modifié effectivement, mais cette modification n’est que visuelle et non pas au niveau de la représentation sémantique de la page.
Modifier le thème graphique utilisé
Pour modifier le thème graphique utilisé, il est nécessaire que le plugin zen-garden soit activé sur le site.
Il suffit ensuite de se rendre dans l’espace de configuration du (...)
Sur d’autres sites (5805)
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Jitsi and ffplay
15 juin 2014, par KotkotI’m playing with jitsi. Got examples form source code. I modified it a bit.
Here is what I’ve got.
I am trying to play the transmitted stream in VLC of ffplay or any other player,
but I cannot.I use these application parameters to run the code :
--local-port-base=5000 --remote-host=localhost --remote-port-base=10000
What am I doing wrong ?
package com.company;
/*
* Jitsi, the OpenSource Java VoIP and Instant Messaging client.
*
* Distributable under LGPL license.
* See terms of license at gnu.org.
*/
import org.jitsi.service.libjitsi.LibJitsi;
import org.jitsi.service.neomedia.*;
import org.jitsi.service.neomedia.device.MediaDevice;
import org.jitsi.service.neomedia.format.MediaFormat;
import org.jitsi.service.neomedia.format.MediaFormatFactory;
import java.io.PrintStream;
import java.net.DatagramSocket;
import java.net.InetAddress;
import java.net.InetSocketAddress;
import java.util.HashMap;
import java.util.Map;
/**
* Implements an example application in the fashion of JMF's AVTransmit2 example
* which demonstrates the use of the <tt>libjitsi</tt> library for the purposes
* of transmitting audio and video via RTP means.
*
* @author Lyubomir Marinov
*/
public class VideoTransmitter {
/**
* The port which is the source of the transmission i.e. from which the
* media is to be transmitted.
*
* @see #LOCAL_PORT_BASE_ARG_NAME
*/
private int localPortBase;
/**
* The <tt>MediaStream</tt> instances initialized by this instance indexed
* by their respective <tt>MediaType</tt> ordinal.
*/
private MediaStream[] mediaStreams;
/**
* The <tt>InetAddress</tt> of the host which is the target of the
* transmission i.e. to which the media is to be transmitted.
*
* @see #REMOTE_HOST_ARG_NAME
*/
private InetAddress remoteAddr;
/**
* The port which is the target of the transmission i.e. to which the media
* is to be transmitted.
*
* @see #REMOTE_PORT_BASE_ARG_NAME
*/
private int remotePortBase;
/**
* Initializes a new <tt>AVTransmit2</tt> instance which is to transmit
* audio and video to a specific host and a specific port.
*
* @param localPortBase the port which is the source of the transmission
* i.e. from which the media is to be transmitted
* @param remoteHost the name of the host which is the target of the
* transmission i.e. to which the media is to be transmitted
* @param remotePortBase the port which is the target of the transmission
* i.e. to which the media is to be transmitted
* @throws Exception if any error arises during the parsing of the specified
* <tt>localPortBase</tt>, <tt>remoteHost</tt> and <tt>remotePortBase</tt>
*/
private VideoTransmitter(
String localPortBase,
String remoteHost, String remotePortBase)
throws Exception {
this.localPortBase
= (localPortBase == null)
? -1
: Integer.valueOf(localPortBase).intValue();
this.remoteAddr = InetAddress.getByName(remoteHost);
this.remotePortBase = Integer.valueOf(remotePortBase).intValue();
}
/**
* Starts the transmission. Returns null if transmission started ok.
* Otherwise it returns a string with the reason why the setup failed.
*/
private String start()
throws Exception {
/*
* Prepare for the start of the transmission i.e. initialize the
* MediaStream instances.
*/
MediaType[] mediaTypes = MediaType.values();
MediaService mediaService = LibJitsi.getMediaService();
int localPort = localPortBase;
int remotePort = remotePortBase;
mediaStreams = new MediaStream[mediaTypes.length];
for (MediaType mediaType : mediaTypes) {
if(mediaType != MediaType.VIDEO) continue;
/*
* The default MediaDevice (for a specific MediaType) is configured
* (by the user of the application via some sort of UI) into the
* ConfigurationService. If there is no ConfigurationService
* instance known to LibJitsi, the first available MediaDevice of
* the specified MediaType will be chosen by MediaService.
*/
MediaDevice device
= mediaService.getMediaDeviceForPartialDesktopStreaming(100,100,100,100);
if (device == null) {
continue;
}
MediaStream mediaStream = mediaService.createMediaStream(device);
// direction
/*
* The AVTransmit2 example sends only and the AVReceive2 receives
* only. In a call, the MediaStream's direction will most commonly
* be set to SENDRECV.
*/
mediaStream.setDirection(MediaDirection.SENDONLY);
// format
String encoding;
double clockRate;
/*
* The AVTransmit2 and AVReceive2 examples use the H.264 video
* codec. Its RTP transmission has no static RTP payload type number
* assigned.
*/
byte dynamicRTPPayloadType;
switch (device.getMediaType()) {
case AUDIO:
encoding = "PCMU";
clockRate = 8000;
/* PCMU has a static RTP payload type number assigned. */
dynamicRTPPayloadType = -1;
break;
case VIDEO:
encoding = "H264";
clockRate = MediaFormatFactory.CLOCK_RATE_NOT_SPECIFIED;
/*
* The dymanic RTP payload type numbers are usually negotiated
* in the signaling functionality.
*/
dynamicRTPPayloadType = 99;
break;
default:
encoding = null;
clockRate = MediaFormatFactory.CLOCK_RATE_NOT_SPECIFIED;
dynamicRTPPayloadType = -1;
}
if (encoding != null) {
MediaFormat format
= mediaService.getFormatFactory().createMediaFormat(
encoding,
clockRate);
/*
* The MediaFormat instances which do not have a static RTP
* payload type number association must be explicitly assigned
* a dynamic RTP payload type number.
*/
if (dynamicRTPPayloadType != -1) {
mediaStream.addDynamicRTPPayloadType(
dynamicRTPPayloadType,
format);
}
mediaStream.setFormat(format);
}
// connector
StreamConnector connector;
if (localPortBase == -1) {
connector = new DefaultStreamConnector();
} else {
int localRTPPort = localPort++;
int localRTCPPort = localPort++;
connector
= new DefaultStreamConnector(
new DatagramSocket(localRTPPort),
new DatagramSocket(localRTCPPort));
}
mediaStream.setConnector(connector);
// target
/*
* The AVTransmit2 and AVReceive2 examples follow the common
* practice that the RTCP port is right after the RTP port.
*/
int remoteRTPPort = remotePort++;
int remoteRTCPPort = remotePort++;
mediaStream.setTarget(
new MediaStreamTarget(
new InetSocketAddress(remoteAddr, remoteRTPPort),
new InetSocketAddress(remoteAddr, remoteRTCPPort)));
// name
/*
* The name is completely optional and it is not being used by the
* MediaStream implementation at this time, it is just remembered so
* that it can be retrieved via MediaStream#getName(). It may be
* integrated with the signaling functionality if necessary.
*/
mediaStream.setName(mediaType.toString());
mediaStreams[mediaType.ordinal()] = mediaStream;
}
/*
* Do start the transmission i.e. start the initialized MediaStream
* instances.
*/
for (MediaStream mediaStream : mediaStreams) {
if (mediaStream != null) {
mediaStream.start();
}
}
return null;
}
/**
* Stops the transmission if already started
*/
private void stop() {
if (mediaStreams != null) {
for (int i = 0; i < mediaStreams.length; i++) {
MediaStream mediaStream = mediaStreams[i];
if (mediaStream != null) {
try {
mediaStream.stop();
} finally {
mediaStream.close();
mediaStreams[i] = null;
}
}
}
mediaStreams = null;
}
}
/**
* The name of the command-line argument which specifies the port from which
* the media is to be transmitted. The command-line argument value will be
* used as the port to transmit the audio RTP from, the next port after it
* will be to transmit the audio RTCP from. Respectively, the subsequent
* ports will be used to transmit the video RTP and RTCP from."
*/
private static final String LOCAL_PORT_BASE_ARG_NAME
= "--local-port-base=";
/**
* The name of the command-line argument which specifies the name of the
* host to which the media is to be transmitted.
*/
private static final String REMOTE_HOST_ARG_NAME = "--remote-host=";
/**
* The name of the command-line argument which specifies the port to which
* the media is to be transmitted. The command-line argument value will be
* used as the port to transmit the audio RTP to, the next port after it
* will be to transmit the audio RTCP to. Respectively, the subsequent ports
* will be used to transmit the video RTP and RTCP to."
*/
private static final String REMOTE_PORT_BASE_ARG_NAME
= "--remote-port-base=";
/**
* The list of command-line arguments accepted as valid by the
* <tt>AVTransmit2</tt> application along with their human-readable usage
* descriptions.
*/
private static final String[][] ARGS
= {
{
LOCAL_PORT_BASE_ARG_NAME,
"The port which is the source of the transmission i.e. from"
+ " which the media is to be transmitted. The specified"
+ " value will be used as the port to transmit the audio"
+ " RTP from, the next port after it will be used to"
+ " transmit the audio RTCP from. Respectively, the"
+ " subsequent ports will be used to transmit the video RTP"
+ " and RTCP from."
},
{
REMOTE_HOST_ARG_NAME,
"The name of the host which is the target of the transmission"
+ " i.e. to which the media is to be transmitted"
},
{
REMOTE_PORT_BASE_ARG_NAME,
"The port which is the target of the transmission i.e. to which"
+ " the media is to be transmitted. The specified value"
+ " will be used as the port to transmit the audio RTP to"
+ " the next port after it will be used to transmit the"
+ " audio RTCP to. Respectively, the subsequent ports will"
+ " be used to transmit the video RTP and RTCP to."
}
};
public static void main(String[] args)
throws Exception {
// We need two parameters to do the transmission. For example,
// ant run-example -Drun.example.name=AVTransmit2 -Drun.example.arg.line="--remote-host=127.0.0.1 --remote-port-base=10000"
if (args.length < 2) {
prUsage();
} else {
Map argMap = parseCommandLineArgs(args);
LibJitsi.start();
try {
// Create a audio transmit object with the specified params.
VideoTransmitter at
= new VideoTransmitter(
argMap.get(LOCAL_PORT_BASE_ARG_NAME),
argMap.get(REMOTE_HOST_ARG_NAME),
argMap.get(REMOTE_PORT_BASE_ARG_NAME));
// Start the transmission
String result = at.start();
// result will be non-null if there was an error. The return
// value is a String describing the possible error. Print it.
if (result == null) {
System.err.println("Start transmission for 600 seconds...");
// Transmit for 60 seconds and then close the processor
// This is a safeguard when using a capture data source
// so that the capture device will be properly released
// before quitting.
// The right thing to do would be to have a GUI with a
// "Stop" button that would call stop on AVTransmit2
try {
Thread.sleep(600_000);
} catch (InterruptedException ie) {
}
// Stop the transmission
at.stop();
System.err.println("...transmission ended.");
} else {
System.err.println("Error : " + result);
}
} finally {
LibJitsi.stop();
}
}
}
/**
* Parses the arguments specified to the <tt>AVTransmit2</tt> application on
* the command line.
*
* @param args the arguments specified to the <tt>AVTransmit2</tt>
* application on the command line
* @return a <tt>Map</tt> containing the arguments specified to the
* <tt>AVTransmit2</tt> application on the command line in the form of
* name-value associations
*/
static Map parseCommandLineArgs(String[] args) {
Map argMap = new HashMap();
for (String arg : args) {
int keyEndIndex = arg.indexOf('=');
String key;
String value;
if (keyEndIndex == -1) {
key = arg;
value = null;
} else {
key = arg.substring(0, keyEndIndex + 1);
value = arg.substring(keyEndIndex + 1);
}
argMap.put(key, value);
}
return argMap;
}
/**
* Outputs human-readable description about the usage of the
* <tt>AVTransmit2</tt> application and the command-line arguments it
* accepts as valid.
*/
private static void prUsage() {
PrintStream err = System.err;
err.println("Usage: " + VideoTransmitter.class.getName() + " <args>");
err.println("Valid args:");
for (String[] arg : ARGS)
err.println(" " + arg[0] + " " + arg[1]);
}
}
</args> -
Reading in pydub AudioSegment from url. BytesIO returning "OSError [Errno 2] No such file or directory" on heroku only ; fine on localhost
24 octobre 2014, par MarkEDIT 1 for anyone with the same error : installing ffmpeg did indeed solve that BytesIO error
EDIT 1 for anyone still willing to help : my problem is now that when I AudioSegment.export("filename.mp3", format="mp3"), the file is made, but has size 0 bytes — details below (as "EDIT 1")
EDIT 2 : All problems now solved.
- Files can be read in as AudioSegment using BytesIO
- I found buildpacks to ensure ffmpeg was installed correctly on my app, with lame support for exporting proper mp3 files
Answer below
Original question
I have pydub working nicely locally to crop a particular mp3 file based on parameters in the url.
(?start_time=3.8&end_time=5.1)When I run
foreman start
it all looks good on localhost. The html renders nicely.
The key lines from the views.py include reading in a file from a url usingurl = "https://s3.amazonaws.com/shareducate02/The_giving_tree__by_Alex_Blumberg__sponsored_by_mailchimp-short.mp3"
mp3 = urllib.urlopen(url).read() # inspired by http://nbviewer.ipython.org/github/ipython-books/cookbook-code/blob/master/notebooks/chapter11_image/06_speech.ipynb
original=AudioSegment.from_mp3(BytesIO(mp3)) # AudioSegment.from_mp3 is a pydub command, see http://pydub.com
section = original[start_time_ms:end_time_ms]That all works great... until I push to heroku (django app) and run it online.
then when I load the same page now on the herokuapp.com, I get this errorOSError at /path/to/page
[Errno 2] No such file or directory
Request Method: GET
Request URL: http://my.website.com/path/to/page?start_time=3.8&end_time=5
Django Version: 1.6.5
Exception Type: OSError
Exception Value:
[Errno 2] No such file or directory
Exception Location: /app/.heroku/python/lib/python2.7/subprocess.py in _execute_child, line 1327
Python Executable: /app/.heroku/python/bin/python
Python Version: 2.7.8
Python Path:
['/app',
'/app/.heroku/python/bin',
'/app/.heroku/python/lib/python2.7/site-packages/setuptools-5.4.1-py2.7.egg',
'/app/.heroku/python/lib/python2.7/site-packages/distribute-0.6.36-py2.7.egg',
'/app/.heroku/python/lib/python2.7/site-packages/pip-1.3.1-py2.7.egg',
'/app',
'/app/.heroku/python/lib/python27.zip',
'/app/.heroku/python/lib/python2.7',
'/app/.heroku/python/lib/python2.7/plat-linux2',
'/app/.heroku/python/lib/python2.7/lib-tk',
'/app/.heroku/python/lib/python2.7/lib-old',
'/app/.heroku/python/lib/python2.7/lib-dynload',
'/app/.heroku/python/lib/python2.7/site-packages',
'/app/.heroku/python/lib/python2.7/site-packages/setuptools-0.6c11-py2.7.egg-info']
Traceback:
File "/app/.heroku/python/lib/python2.7/site-packages/django/core/handlers/base.py" in get_response
112. response = wrapped_callback(request, *callback_args, **callback_kwargs)
File "/app/evernote/views.py" in finalize
105. original=AudioSegment.from_mp3(BytesIO(mp3))
File "/app/.heroku/python/lib/python2.7/site-packages/pydub/audio_segment.py" in from_mp3
318. return cls.from_file(file, 'mp3')
File "/app/.heroku/python/lib/python2.7/site-packages/pydub/audio_segment.py" in from_file
302. retcode = subprocess.call(convertion_command, stderr=open(os.devnull))
File "/app/.heroku/python/lib/python2.7/subprocess.py" in call
522. return Popen(*popenargs, **kwargs).wait()
File "/app/.heroku/python/lib/python2.7/subprocess.py" in __init__
710. errread, errwrite)
File "/app/.heroku/python/lib/python2.7/subprocess.py" in _execute_child
1327. raise child_exceptionI have commented out some of the original to convince myself that sure enough the single line
original=AudioSegment.from_mp3(BytesIO(mp3))
is where the problem kicks in... but this is not a problem locallyThe full function in views.py starts like this :
from django.shortcuts import render, get_object_or_404
from django.http import HttpResponseRedirect #, Http404, HttpResponse
from django.core.urlresolvers import reverse
from django.views import generic
import pydub
# Maybe only need:
from pydub import AudioSegment # == see below
from time import gmtime, strftime
import boto
from boto.s3.connection import S3Connection
from boto.s3.key import Key
# http://nbviewer.ipython.org/github/ipython-books/cookbook-code/blob/master/notebooks/chapter11_image/06_speech.ipynb
import urllib
from io import BytesIO
# import numpy as np
# import scipy.signal as sg
# import pydub # mentioned above already
# import matplotlib.pyplot as plt
# from IPython.display import Audio, display
# import matplotlib as mpl
# %matplotlib inline
import os
# from settings import AWS_ACCESS_KEY, AWS_SECRET_KEY, AWS_BUCKET_NAME
AWS_ACCESS_KEY = os.environ.get('AWS_ACCESS_KEY') # there must be a better way?
AWS_SECRET_KEY = os.environ.get('AWS_SECRET_KEY')
AWS_BUCKET_NAME = os.environ.get('S3_BUCKET_NAME')
# http://stackoverflow.com/questions/415511/how-to-get-current-time-in-python
boto_conn = S3Connection(AWS_ACCESS_KEY, AWS_SECRET_KEY)
bucket = boto_conn.get_bucket(AWS_BUCKET_NAME)
s3_url_format = 'https://s3.amazonaws.com/shareducate02/{end_path}'and specifically the view in views.py that’s called when I visit the page :
def finalize(request):
start_time = request.GET.get('start_time')
end_time = request.GET.get('end_time')
original_file = "https://s3.amazonaws.com/shareducate02/The_giving_tree__by_Alex_Blumberg__sponsored_by_mailchimp-short.mp3"
if start_time:
# original=AudioSegment.from_mp3(original_file) #...that didn't work
# but this works below:
# next three uncommented lines from http://nbviewer.ipython.org/github/ipython-books/cookbook-code/blob/master/notebooks/chapter11_image/06_speech.ipynb
# python 2.x
url = original_file
# req = urllib.Request(url, headers={'User-Agent': ''}) # Note: I commented out this because I got error that "Request" did not exist
mp3 = urllib.urlopen(url).read()
# That's for my 2.7
# If I ever upgrade to python 3.x, would need to change it to:
# req = urllib.request.Request(url, headers={'User-Agent': ''})
# mp3 = urllib.request.urlopen(req).read()
# as per instructions on http://nbviewer.ipython.org/github/ipython-books/cookbook-code/blob/master/notebooks/chapter11_image/06_speech.ipynb
original=AudioSegment.from_mp3(BytesIO(mp3))
# original=AudioSegment.from_mp3("static/givingtree.mp3") # alternative that works locally (on laptop) but no use for heroku
start_time_ms = int(float(start_time) * 1000)
if end_time:
end_time_ms = int(float(end_time) * 1000)
else:
end_time_ms = int(float(original.duration_seconds) * 1000)
duration_ms = end_time_ms - start_time_ms
# duration = end_time - start_time
duration = duration_ms/1000
# section = original[start_time_ms:end_time_ms]
# section_with_fading = section.fade_in(100).fade_out(100)
clip = "demo-"
number = strftime("%Y-%m-%d_%H-%M-%S", gmtime())
clip += number
clip += ".mp3"
# DON'T BOTHER writing locally:
# clip_with_path = "evernote/static/"+clip
# section_with_fading.export(clip_with_path, format = "mp3")
# tempclip = section_with_fading.export(format = "mp3")
# commented out while de-bugging, but was working earlier if run on localhost
# c = boto.connect_s3()
# b = c.get_bucket(S3_BUCKET_NAME) # as defined above
# k = Key(b)
# k.key=clip
# # k.set_contents_from_filename(clip_with_path)
# k.set_contents_from_file(tempclip)
# k.set_acl('public-read')
clip_made = True
else:
duration = 0.0
clip_made = False
clip = ""
context = {'original_file':original_file, 'new_file':clip, 'start_time': start_time, 'end_time':end_time, 'duration':duration, 'clip_made':clip_made}
return render(request, 'finalize.html' , context)Any suggestions ?
Potentially related :
I have ffmpeg installed locallyBut have been unable to install it onto heroku, due to not understanding buildpacks. I tried just a moment ago (
http://stackoverflow.com/questions/14407388/how-to-install-ffmpeg-for-a-django-app-on-heroku
andhttps://github.com/shunjikonishi/heroku-buildpack-ffmpeg
) but so far ffmpeg is not working on heroku (ffmpeg is not recognised when I do "heroku run ffmpeg —version")
...do you think this is the reason ?An answer like any of these would be much appreciated as I’m going round in circles here :
- "I think ffmpeg is indeed your problem. Try harder to sort that out, to get it installed on heroku"
- "Actually, I think this is why BytesIO is not working for you : ..."
- "Your approach is terrible anyway... if you want to read in an audio file to process using pydub, you should just do this instead : ..." (since I’m just hacking my way through pydub for my first time... my approach may be poor)
EDIT 1
ffmpeg is now installed (e.g., I can output wav files)
However, I can’t create mp3 files, still... or more correctly, I can, but the filesize is zero
(venv-app)moriartymacbookair13:getstartapp macuser$ heroku config:add BUILDPACK_URL=https://github.com/ddollar/heroku-buildpack-multi.git
Setting config vars and restarting awe01... done, v93
BUILDPACK_URL: https://github.com/ddollar/heroku-buildpack-multi.git
(venv-app)moriartymacbookair13:getstartapp macuser$ vim .buildpacks
(venv-app)moriartymacbookair13:getstartapp macuser$ cat .buildpacks
https://github.com/shunjikonishi/heroku-buildpack-ffmpeg.git
https://github.com/heroku/heroku-buildpack-python.git
(venv-app)moriartymacbookair13:getstartapp macuser$ git add --all
(venv-app)moriartymacbookair13:getstartapp macuser$ git commit -m "need multi, not just ffmpeg, so adding back in multi + shun + heroku, with trailing .git in .buildpacks file"
[master cd99fef] need multi, not just ffmpeg, so adding back in multi + shun + heroku, with trailing .git in .buildpacks file
1 file changed, 2 insertions(+), 2 deletions(-)
(venv-app)moriartymacbookair13:getstartapp macuser$ git push heroku master
Fetching repository, done.
Counting objects: 5, done.
Delta compression using up to 4 threads.
Compressing objects: 100% (3/3), done.
Writing objects: 100% (3/3), 372 bytes | 0 bytes/s, done.
Total 3 (delta 2), reused 0 (delta 0)
-----> Fetching custom git buildpack... done
-----> Multipack app detected
=====> Downloading Buildpack: https://github.com/shunjikonishi/heroku-buildpack-ffmpeg.git
=====> Detected Framework: ffmpeg
-----> Install ffmpeg
DOWNLOAD_URL = http://flect.github.io/heroku-binaries/libs/ffmpeg.tar.gz
exporting PATH and LIBRARY_PATH
=====> Downloading Buildpack: https://github.com/heroku/heroku-buildpack-python.git
=====> Detected Framework: Python
-----> Installing dependencies with pip
Cleaning up...
-----> Preparing static assets
Collectstatic configuration error. To debug, run:
$ heroku run python ./example/manage.py collectstatic --noinput
Using release configuration from last framework (Python).
-----> Discovering process types
Procfile declares types -> web
-----> Compressing... done, 198.1MB
-----> Launching... done, v94
http://[redacted].herokuapp.com/ deployed to Heroku
To git@heroku.com:awe01.git
78d6b68..cd99fef master -> master
(venv-app)moriartymacbookair13:getstartapp macuser$ heroku run ffmpeg
Running `ffmpeg` attached to terminal... up, run.6408
ffmpeg version git-2013-06-02-5711e4f Copyright (c) 2000-2013 the FFmpeg developers
built on Jun 2 2013 07:38:40 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
configuration: --enable-shared --disable-asm --prefix=/app/vendor/ffmpeg
libavutil 52. 34.100 / 52. 34.100
libavcodec 55. 13.100 / 55. 13.100
libavformat 55. 8.102 / 55. 8.102
libavdevice 55. 2.100 / 55. 2.100
libavfilter 3. 74.101 / 3. 74.101
libswscale 2. 3.100 / 2. 3.100
libswresample 0. 17.102 / 0. 17.102
Hyper fast Audio and Video encoder
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}...
Use -h to get full help or, even better, run 'man ffmpeg'
(venv-app)moriartymacbookair13:getstartapp macuser$ heroku run bash
Running `bash` attached to terminal... up, run.9660
~ $ python
Python 2.7.8 (default, Jul 9 2014, 20:47:08)
[GCC 4.4.3] on linux2
Type "help", "copyright", "credits" or "license" for more information.
>>> import pydub
>>> from pydub import AudioSegment
>>> exit()
~ $ which ffmpeg
/app/vendor/ffmpeg/bin/ffmpeg
~ $ python
Python 2.7.8 (default, Jul 9 2014, 20:47:08)
[GCC 4.4.3] on linux2
Type "help", "copyright", "credits" or "license" for more information.
>>> import pydub
>>> from pydub import AudioSegment
>>> AudioSegment.silent(5000).export("/tmp/asdf.mp3", "mp3")
<open file="file"></open>tmp/asdf.mp3', mode 'wb+' at 0x7f9a37d44780>
>>> exit ()
~ $ cd /tmp/
/tmp $ ls
asdf.mp3
/tmp $ open asdf.mp3
bash: open: command not found
/tmp $ ls -lah
total 8.0K
drwx------ 2 u36483 36483 4.0K 2014-10-22 04:14 .
drwxr-xr-x 14 root root 4.0K 2014-09-26 07:08 ..
-rw------- 1 u36483 36483 0 2014-10-22 04:14 asdf.mp3Note the file size of 0 above for the mp3 file... when I do the same thing on my macbook, the file size is never zero
Back to the heroku shell :
/tmp $ python
Python 2.7.8 (default, Jul 9 2014, 20:47:08)
[GCC 4.4.3] on linux2
Type "help", "copyright", "credits" or "license" for more information.
>>> import pydub
>>> from pydub import AudioSegment
>>> pydub.AudioSegment.ffmpeg = "/app/vendor/ffmpeg/bin/ffmpeg"
>>> AudioSegment.silence(1200).export("/tmp/herokuSilence.mp3", format="mp3")
Traceback (most recent call last):
File "<stdin>", line 1, in <module>
AttributeError: type object 'AudioSegment' has no attribute 'silence'
>>> AudioSegment.silent(1200).export("/tmp/herokuSilence.mp3", format="mp3")
<open file="file"></open>tmp/herokuSilence.mp3', mode 'wb+' at 0x7fcc2017c780>
>>> exit()
/tmp $ ls
asdf.mp3 herokuSilence.mp3
/tmp $ ls -lah
total 8.0K
drwx------ 2 u36483 36483 4.0K 2014-10-22 04:29 .
drwxr-xr-x 14 root root 4.0K 2014-09-26 07:08 ..
-rw------- 1 u36483 36483 0 2014-10-22 04:14 asdf.mp3
-rw------- 1 u36483 36483 0 2014-10-22 04:29 herokuSilence.mp3
</module></stdin>I realised the first time that I had forgotten the
pydub.AudioSegment.ffmpeg = "/app/vendor/ffmpeg/bin/ffmpeg"
command, but as you can see above, the file is still zero sizeOut of desperation, I even tried adding the ".heroku" into the path to be as verbatim as your example, but that didn’t fix it :
/tmp $ python
Python 2.7.8 (default, Jul 9 2014, 20:47:08)
[GCC 4.4.3] on linux2
Type "help", "copyright", "credits" or "license" for more information.
>>> import pydub
>>> from pydub import AudioSegment
>>> pydub.AudioSegment.ffmpeg = "/app/.heroku/vendor/ffmpeg/bin/ffmpeg"
>>> AudioSegment.silent(1200).export("/tmp/herokuSilence03.mp3", format="mp3")
<open file="file"></open>tmp/herokuSilence03.mp3', mode 'wb+' at 0x7fc92aca7780>
>>> exit()
/tmp $ ls -lah
total 8.0K
drwx------ 2 u36483 36483 4.0K 2014-10-22 04:31 .
drwxr-xr-x 14 root root 4.0K 2014-09-26 07:08 ..
-rw------- 1 u36483 36483 0 2014-10-22 04:14 asdf.mp3
-rw------- 1 u36483 36483 0 2014-10-22 04:31 herokuSilence03.mp3
-rw------- 1 u36483 36483 0 2014-10-22 04:29 herokuSilence.mp3Finally, I tried exporting a .wav file to check pydub was at least working correctly
/tmp $ python
Python 2.7.8 (default, Jul 9 2014, 20:47:08)
[GCC 4.4.3] on linux2
Type "help", "copyright", "credits" or "license" for more information.
>>> import pydub
>>> from pydub import AudioSegment
>>> pydub.AudioSegment.ffmpeg = "/app/vendor/ffmpeg/bin/ffmpeg"
>>> AudioSegment.silent(1300).export("/tmp/heroku_wav_silence01.wav", format="wav")
<open file="file"></open>tmp/heroku_wav_silence01.wav', mode 'wb+' at 0x7fa33cbf3780>
>>> exit()
/tmp $ ls
asdf.mp3 herokuSilence03.mp3 herokuSilence.mp3 heroku_wav_silence01.wav
/tmp $ ls -lah
total 40K
drwx------ 2 u36483 36483 4.0K 2014-10-22 04:42 .
drwxr-xr-x 14 root root 4.0K 2014-09-26 07:08 ..
-rw------- 1 u36483 36483 0 2014-10-22 04:14 asdf.mp3
-rw------- 1 u36483 36483 0 2014-10-22 04:31 herokuSilence03.mp3
-rw------- 1 u36483 36483 0 2014-10-22 04:29 herokuSilence.mp3
-rw------- 1 u36483 36483 29K 2014-10-22 04:42 heroku_wav_silence01.wav
/tmp $At least that filesize for .wav is non-zero, so pydub is working
My current theory is that either I’m still not using ffmpeg correctly, or it’s insufficient... maybe I need an mp3 additional install on top of basic ffmpeg.
Several sites mention "libavcodec-extra-53" but I’m not sure how to install that on heroku, or to check if I have it ?
https://github.com/jiaaro/pydub/issues/36
Similarly tutorials on libmp3lame seem to be geared towards laptop installation rather than installation on heroku, so I’m at a losshttp://superuser.com/questions/196857/how-to-install-libmp3lame-for-ffmpeg
In case relevant, I also have youtube-dl in my requirements.txt... this also works locally on my macbook, but fails when I run it in the heroku shell :
~/ytdl $ youtube-dl --restrict-filenames -x --audio-format mp3 n2anDgdUHic
[youtube] Setting language
[youtube] Confirming age
[youtube] n2anDgdUHic: Downloading webpage
[youtube] n2anDgdUHic: Downloading video info webpage
[youtube] n2anDgdUHic: Extracting video information
[download] Destination: Boyce_Avenue_feat._Megan_Nicole_-_Skyscraper_Patrick_Ebert_Edit-n2anDgdUHic.m4a
[download] 100% of 5.92MiB in 00:00
[ffmpeg] Destination: Boyce_Avenue_feat._Megan_Nicole_-_Skyscraper_Patrick_Ebert_Edit-n2anDgdUHic.mp3
ERROR: audio conversion failed: Unknown encoder 'libmp3lame'
~/ytdl $The informative link is that it too specificies an mp3 failure, so perhaps they two issues are related.
EDIT 2
See answer, all problems solved
-
FFMPEG : Offseting & merging audios [migrated]
5 novembre 2014, par user1064504I am trying to offset multiple audios into one, each with different offset.
<code>ffmpeg -i a.ogg -i 1.ogg -filter_complex "amix=inputs=2[op];[op]adelay=5000|15000" out.ogg
Can someone help with understand how to correctly use adelay with amix for multiple files, I am trying to achieve something like this.
<code>
<-ist audio-> <---2nd-audio---><---------------------------------------------->