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SPIP - plugins - embed code - Exemple
2 septembre 2013, par
Mis à jour : Septembre 2013
Langue : français
Type : Image
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Unable to stream file onto localhost - ffmpeg
18 octobre 2013, par trueblueI am new to ffmpeg/ffserver. I am trying to stream a local file named Trial onto a localhost using ffserver. I want to run the file in browser as
http://localhost:8090/feed1.ffm
I am executing the below command in Ubuntu(Trial is a Mpeg TS file) :ffmpeg -i Trial http://localhost:8090/feed1.ffmUpon execution of above command I am getting below error :
FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.3, Copyright (c) 2000-2009 Fabrice Bellard, et al.
configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.3 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static
libavutil 49.15. 0 / 49.15. 0
libavcodec 52.20. 1 / 52.20. 1
libavformat 52.31. 0 / 52.31. 0
libavdevice 52. 1. 0 / 52. 1. 0
libavfilter 0. 4. 0 / 0. 4. 0
libswscale 0. 7. 1 / 0. 7. 1
libpostproc 51. 2. 0 / 51. 2. 0
built on Jan 24 2013 19:42:59, gcc: 4.4.3
Seems stream 0 codec frame rate differs from container frame rate: 119.88 (120000/1001) -> 59.94 (60000/1001)
Input #0, mpegts, from 'Trial':
Duration: 00:00:04.22, start: 0.177633, bitrate: 40368 kb/s
Program 2
Stream #0.0[0x21]: Video: mpeg2video, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 45000 kb/s, 59.94 tbr, 90k tbn, 119.88 tbc
Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
Stream #0.0: Video: flv, yuv420p, 352x288, q=1-5, 100 kb/s, 1000k tbn, 15 tbc
Stream #0.1: Audio: mp2, 44100 Hz, mono, s16, 32 kb/s
Stream #0.2: Video: mpeg1video, yuv420p, 160x128, q=3-31, 64 kb/s, 1000k tbn, 3 tbc
Stream #0.3: Audio: mp2, 22050 Hz, mono, s16, 64 kb/s
Stream #0.4: Video: msmpeg4, yuv420p, 352x240, q=3-31, 256 kb/s, 1000k tbn, 15 tbc
Could not find input stream matching output stream #0.1My ffserver.conf file goes like this :
# Port on which the server is listening. You must select a different
# port from your standard HTTP web server if it is running on the same
# computer.
Port 8090
# Address on which the server is bound. Only useful if you have
# several network interfaces.
BindAddress 0.0.0.0
# Number of simultaneous HTTP connections that can be handled. It has
# to be defined *before* the MaxClients parameter, since it defines the
# MaxClients maximum limit.
MaxHTTPConnections 2000
# Number of simultaneous requests that can be handled. Since FFServer
# is very fast, it is more likely that you will want to leave this high
# and use MaxBandwidth, below.
MaxClients 1000
# This the maximum amount of kbit/sec that you are prepared to
# consume when streaming to clients.
MaxBandwidth 1000
# Access log file (uses standard Apache log file format)
# '-' is the standard output.
CustomLog -
# Suppress that if you want to launch ffserver as a daemon.
NoDaemon
##################################################################
# Definition of the live feeds. Each live feed contains one video
# and/or audio sequence coming from an ffmpeg encoder or another
# ffserver. This sequence may be encoded simultaneously with several
# codecs at several resolutions.
<feed>
# You must use 'ffmpeg' to send a live feed to ffserver. In this
# example, you can type:
#
# ffmpeg http://localhost:8090/feed1.ffm
# ffserver can also do time shifting. It means that it can stream any
# previously recorded live stream. The request should contain:
# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
# a path where the feed is stored on disk. You also specify the
# maximum size of the feed, where zero means unlimited. Default:
# File=/tmp/feed_name.ffm FileMaxSize=5M
File /tmp/feed1.ffm
FileMaxSize 5M
# You could specify
# ReadOnlyFile /saved/specialvideo.ffm
# This marks the file as readonly and it will not be deleted or updated.
# Specify launch in order to start ffmpeg automatically.
# First ffmpeg must be defined with an appropriate path if needed,
# after that options can follow, but avoid adding the http:// field
#Launch ffmpeg
# Only allow connections from localhost to the feed.
ACL allow 127.0.0.1
</feed>
<stream>
Feed feed1.ffm
Format swf
VideoCodec flv
VideoFrameRate 15
VideoBufferSize 80000
VideoBitRate 100
VideoQMin 1
VideoQMax 5
VideoSize 352x288
PreRoll 0
Noaudio
</stream>
##################################################################
# Now you can define each stream which will be generated from the
# original audio and video stream. Each format has a filename (here
# 'test1.mpg'). FFServer will send this stream when answering a
# request containing this filename.
<stream>
# coming from live feed 'feed1'
Feed feed1.ffm
# Format of the stream : you can choose among:
# mpeg : MPEG-1 multiplexed video and audio
# mpegvideo : only MPEG-1 video
# mp2 : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
# ogg : Ogg format (Vorbis audio codec)
# rm : RealNetworks-compatible stream. Multiplexed audio and video.
# ra : RealNetworks-compatible stream. Audio only.
# mpjpeg : Multipart JPEG (works with Netscape without any plugin)
# jpeg : Generate a single JPEG image.
# asf : ASF compatible streaming (Windows Media Player format).
# swf : Macromedia Flash compatible stream
# avi : AVI format (MPEG-4 video, MPEG audio sound)
Format mpeg
# Bitrate for the audio stream. Codecs usually support only a few
# different bitrates.
AudioBitRate 32
# Number of audio channels: 1 = mono, 2 = stereo
AudioChannels 1
# Sampling frequency for audio. When using low bitrates, you should
# lower this frequency to 22050 or 11025. The supported frequencies
# depend on the selected audio codec.
AudioSampleRate 44100
# Bitrate for the video stream
VideoBitRate 64
# Ratecontrol buffer size
VideoBufferSize 40
# Number of frames per second
VideoFrameRate 3
# Size of the video frame: WxH (default: 160x128)
# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
# hd1080
VideoSize 160x128
# Transmit only intra frames (useful for low bitrates, but kills frame rate).
#VideoIntraOnly
# If non-intra only, an intra frame is transmitted every VideoGopSize
# frames. Video synchronization can only begin at an intra frame.
VideoGopSize 12
# More MPEG-4 parameters
# VideoHighQuality
# Video4MotionVector
# Choose your codecs:
#AudioCodec mp2
#VideoCodec mpeg1video
# Suppress audio
#NoAudio
# Suppress video
#NoVideo
#VideoQMin 3
#VideoQMax 31
# Set this to the number of seconds backwards in time to start. Note that
# most players will buffer 5-10 seconds of video, and also you need to allow
# for a keyframe to appear in the data stream.
#Preroll 15
# ACL:
# You can allow ranges of addresses (or single addresses)
#ACL ALLOW <first address="address"> <last address="address">
# You can deny ranges of addresses (or single addresses)
#ACL DENY <first address="address"> <last address="address">
# You can repeat the ACL allow/deny as often as you like. It is on a per
# stream basis. The first match defines the action. If there are no matches,
# then the default is the inverse of the last ACL statement.
#
# Thus 'ACL allow localhost' only allows access from localhost.
# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
# allow everybody else.
</last></first></last></first></stream>
##################################################################
# Example streams
# Multipart JPEG
#<stream>
#Feed feed1.ffm
#Format mpjpeg
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#Strict -1
#</stream>
# Single JPEG
#<stream>
#Feed feed1.ffm
#Format jpeg
#VideoFrameRate 2
#VideoIntraOnly
##VideoSize 352x240
#NoAudio
#Strict -1
#</stream>
# Flash
#<stream>
#Feed feed1.ffm
#Format swf
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#</stream>
# ASF compatible
<stream>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</stream>
# MP3 audio
#<stream>
#Feed feed1.ffm
#Format mp2
#AudioCodec mp3
#AudioBitRate 64
#AudioChannels 1
#AudioSampleRate 44100
#NoVideo
#</stream>
# Ogg Vorbis audio
#<stream>
#Feed feed1.ffm
#Title "Stream title"
#AudioBitRate 64
#AudioChannels 2
#AudioSampleRate 44100
#NoVideo
#</stream>
# Real with audio only at 32 kbits
#<stream>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#NoVideo
#NoAudio
#</stream>
# Real with audio and video at 64 kbits
#<stream>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#VideoBitRate 128
#VideoFrameRate 25
#VideoGopSize 25
#NoAudio
#</stream>
##################################################################
# A stream coming from a file: you only need to set the input
# filename and optionally a new format. Supported conversions:
# AVI -> ASF
#<stream>
#File "/usr/local/httpd/htdocs/tlive.rm"
#NoAudio
#</stream>
#<stream>
#File "/usr/local/httpd/htdocs/test.asf"
#NoAudio
#Author "Me"
#Copyright "Super MegaCorp"
#Title "Test stream from disk"
#Comment "Test comment"
#</stream>
##################################################################
# RTSP examples
#
# You can access this stream with the RTSP URL:
# rtsp://localhost:5454/test1-rtsp.mpg
#
# A non-standard RTSP redirector is also created. Its URL is:
# http://localhost:8090/test1-rtsp.rtsp
#<stream>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#</stream>
##################################################################
# SDP/multicast examples
#
# If you want to send your stream in multicast, you must set the
# multicast address with MulticastAddress. The port and the TTL can
# also be set.
#
# An SDP file is automatically generated by ffserver by adding the
# 'sdp' extension to the stream name (here
# http://localhost:8090/test1-sdp.sdp). You should usually give this
# file to your player to play the stream.
#
# The 'NoLoop' option can be used to avoid looping when the stream is
# terminated.
#<stream>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#MulticastAddress 224.124.0.1
#MulticastPort 5000
#MulticastTTL 16
#NoLoop
#</stream>
##################################################################
# Special streams
# Server status
<stream>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
</stream>
# Redirect index.html to the appropriate site
<redirect>
URL http://www.ffmpeg.org/
</redirect>Kindly anyone please assist me whether I am missing something or do i need to change my server.conf file ? I have referred many websites. But still I am unable to fix it. Thanks in advance.
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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=10000What 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> -
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.oggCan 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---><---------------------------------------------->
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