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Médias (91)
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Chuck D with Fine Arts Militia - No Meaning No
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Paul Westerberg - Looking Up in Heaven
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Le Tigre - Fake French
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Thievery Corporation - DC 3000
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Dan the Automator - Relaxation Spa Treatment
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Gilberto Gil - Oslodum
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
Autres articles (112)
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Personnaliser en ajoutant son logo, sa bannière ou son image de fond
5 septembre 2013, parCertains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;
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Ecrire une actualité
21 juin 2013, parPrésentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
Dans le thème par défaut spipeo de MédiaSPIP, les actualités sont affichées en bas de la page principale sous les éditoriaux.
Vous pouvez personnaliser le formulaire de création d’une actualité.
Formulaire de création d’une actualité Dans le cas d’un document de type actualité, les champs proposés par défaut sont : Date de publication ( personnaliser la date de publication ) (...) -
Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir
Sur d’autres sites (12387)
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libavformat/libavfilter transcoded audio is choppy
9 mai 2016, par JohnnyDI am attempting to transcode an mp4 file into a standard format. The video seems ok but the audio is choppy (and out of sync with the video).
My test input file has the following properties :
Stream #0:1(eng), 0, 1/48000: Audio: aac (mp4a / 0x6134706D), 48000 Hz, 2 channels, 129 kb/s (default)
and I’m outputing to :
Stream #0:1, 0, 1/44100: Audio: aac (libfaac) (Main), 44100 Hz, stereo, s16, 128 kb/s
When I construct the audio filter graph I get the following debug output :
[in @ 0x103954380] Setting 'time_base' to value '1/48000'
[in @ 0x103954380] Setting 'sample_rate' to value '48000'
[in @ 0x103954380] Setting 'sample_fmt' to value 'fltp'
[in @ 0x103954380] Setting 'channel_layout' to value '0x3'
[in @ 0x103954380] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x3
[format @ 0x10390b3e0] Setting 'sample_fmts' to value 's16'
[format @ 0x10390b3e0] Setting 'sample_rates' to value '44100'
[format @ 0x10390b3e0] Setting 'channel_layouts' to value '0x3'
[format @ 0x10390b3e0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'in' and the filter 'format'
[AVFilterGraph @ 0x101f21b80] query_formats: 3 queried, 3 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 0x103952bc0] ch:2 chl:stereo fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:s16 r:44100HzThis looks correct to me but I’m getting a lot of the following messages when I process the file...
[libfaac @ 0x102063a00] Trying to remove 80 more samples than there are in the queue
...and the audio is choppy. Also I’m seeing that the sample format is the same as the original file (from ffprobe) :
Stream #0:1(und): Audio: aac (Main) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 127 kb/s (default)
i.e. it hasn’t done the conversion from AV_SAMPLE_FMT_FLT to AV_SAMPLE_FMT_S16.
I’m wondering if the bitrate is the cause of the problem but I can’t see any way to transform the input bitrate to the output bitrate. Any thoughts ?
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will Adult script pro work with google app engine as server ? [on hold]
1er février 2016, par niko nørrei want to user Google app engine as a server for my website and app.
but a programmer needs to set the script up for me.
the script is called Adult Script Pro and it need FFMpeg and more to Work.do google app engine provide me with ssh root access ?
The requirements are the following :
Apache, Lighttpd or Nginx Server (with rewrite support)
MySQL 5.x
PHP >= 5.2.x (mod_php or CGI/FastCGI)
GD2 Support
MySQL Support
CURL Support
SimpleXML Support
FTP Support
PCRE with UTF8/Unicode Properties
PHP CLI >= 5.2.x (see above for support)
FFMpeg => 0.11.5 (with support for lame, x264, theora, vpx, xvid, faac, faad2, amr, webm, jpeg, png, gif and freetype)
Will the site Work whid Google app engine as server ??
Pleas help thanks
Nikolaj -
Play video using mse (media source extension) in google chrome
23 août 2019, par liyuqihxcI’m working on a project that convert rtsp stream (ffmpeg) and play it on the web page (signalr + mse).
So far it works pretty much as I expected on the latest version of edge and firefox, but not chrome.
here’s the code
public class WebmMediaStreamContext
{
private Process _ffProcess;
private readonly string _cmd;
private byte[] _initSegment;
private Task _readMediaStreamTask;
private CancellationTokenSource _cancellationTokenSource;
private const string _CmdTemplate = "-i {0} -c:v libvpx -tile-columns 4 -frame-parallel 1 -keyint_min 90 -g 90 -f webm -dash 1 pipe:";
public static readonly byte[] ClusterStart = { 0x1F, 0x43, 0xB6, 0x75, 0x01, 0x00, 0x00, 0x00 };
public event EventHandler<clusterreadyeventargs> ClusterReadyEvent;
public WebmMediaStreamContext(string rtspFeed)
{
_cmd = string.Format(_CmdTemplate, rtspFeed);
}
public async Task StartConverting()
{
if (_ffProcess != null)
throw new InvalidOperationException();
_ffProcess = new Process();
_ffProcess.StartInfo = new ProcessStartInfo
{
FileName = "ffmpeg/ffmpeg.exe",
Arguments = _cmd,
UseShellExecute = false,
CreateNoWindow = true,
RedirectStandardOutput = true
};
_ffProcess.Start();
_initSegment = await ParseInitSegmentAndStartReadMediaStream();
}
public byte[] GetInitSegment()
{
return _initSegment;
}
// Find the first cluster, and everything before it is the InitSegment
private async Task ParseInitSegmentAndStartReadMediaStream()
{
Memory<byte> buffer = new byte[10 * 1024];
int length = 0;
while (length != buffer.Length)
{
length += await _ffProcess.StandardOutput.BaseStream.ReadAsync(buffer.Slice(length));
int cluster = buffer.Span.IndexOf(ClusterStart);
if (cluster >= 0)
{
_cancellationTokenSource = new CancellationTokenSource();
_readMediaStreamTask = new Task(() => ReadMediaStreamProc(buffer.Slice(cluster, length - cluster).ToArray(), _cancellationTokenSource.Token), _cancellationTokenSource.Token, TaskCreationOptions.LongRunning);
_readMediaStreamTask.Start();
return buffer.Slice(0, cluster).ToArray();
}
}
throw new InvalidOperationException();
}
private void ReadMoreBytes(Span<byte> buffer)
{
int size = buffer.Length;
while (size > 0)
{
int len = _ffProcess.StandardOutput.BaseStream.Read(buffer.Slice(buffer.Length - size));
size -= len;
}
}
// Parse every single cluster and fire ClusterReadyEvent
private void ReadMediaStreamProc(byte[] bytesRead, CancellationToken cancel)
{
Span<byte> buffer = new byte[5 * 1024 * 1024];
bytesRead.CopyTo(buffer);
int bufferEmptyIndex = bytesRead.Length;
do
{
if (bufferEmptyIndex < ClusterStart.Length + 4)
{
ReadMoreBytes(buffer.Slice(bufferEmptyIndex, 1024));
bufferEmptyIndex += 1024;
}
int clusterDataSize = BitConverter.ToInt32(
buffer.Slice(ClusterStart.Length, 4)
.ToArray()
.Reverse()
.ToArray()
);
int clusterSize = ClusterStart.Length + 4 + clusterDataSize;
if (clusterSize > buffer.Length)
{
byte[] newBuffer = new byte[clusterSize];
buffer.Slice(0, bufferEmptyIndex).CopyTo(newBuffer);
buffer = newBuffer;
}
if (bufferEmptyIndex < clusterSize)
{
ReadMoreBytes(buffer.Slice(bufferEmptyIndex, clusterSize - bufferEmptyIndex));
bufferEmptyIndex = clusterSize;
}
ClusterReadyEvent?.Invoke(this, new ClusterReadyEventArgs(buffer.Slice(0, bufferEmptyIndex).ToArray()));
bufferEmptyIndex = 0;
} while (!cancel.IsCancellationRequested);
}
}
</byte></byte></byte></clusterreadyeventargs>I use ffmpeg to convert the rtsp stream to vp8 WEBM byte stream and parse it to "Init Segment" (ebml head、info、tracks...) and "Media Segment" (cluster), then send it to browser via signalR
$(function () {
var mediaSource = new MediaSource();
var mimeCodec = 'video/webm; codecs="vp8"';
var video = document.getElementById('video');
mediaSource.addEventListener('sourceopen', callback, false);
function callback(e) {
var sourceBuffer = mediaSource.addSourceBuffer(mimeCodec);
var queue = [];
sourceBuffer.addEventListener('updateend', function () {
if (queue.length === 0) {
return;
}
var base64 = queue[0];
if (base64.length === 0) {
mediaSource.endOfStream();
queue.shift();
return;
} else {
var buffer = new Uint8Array(atob(base64).split("").map(function (c) {
return c.charCodeAt(0);
}));
sourceBuffer.appendBuffer(buffer);
queue.shift();
}
}, false);
var connection = new signalR.HubConnectionBuilder()
.withUrl("/signalr-video")
.configureLogging(signalR.LogLevel.Information)
.build();
connection.start().then(function () {
connection.stream("InitVideoReceive")
.subscribe({
next: function(item) {
if (queue.length === 0 && !!!sourceBuffer.updating) {
var buffer = new Uint8Array(atob(item).split("").map(function (c) {
return c.charCodeAt(0);
}));
sourceBuffer.appendBuffer(buffer);
console.log(blockindex++ + " : " + buffer.byteLength);
} else {
queue.push(item);
}
},
complete: function () {
queue.push('');
},
error: function (err) {
console.error(err);
}
});
});
}
video.src = window.URL.createObjectURL(mediaSource);
})chrome just play the video for 3 5 seconds and then stop for buffering, even though there are plenty of cluster transfered and inserted into SourceBuffer.
here’s the information in chrome ://media-internals/
Player Properties :
render_id: 217
player_id: 1
origin_url: http://localhost:52531/
frame_url: http://localhost:52531/
frame_title: Home Page
url: blob:http://localhost:52531/dcb25d89-9830-40a5-ba88-33c13b5c03eb
info: Selected FFmpegVideoDecoder for video decoding, config: codec: vp8 format: 1 profile: vp8 coded size: [1280,720] visible rect: [0,0,1280,720] natural size: [1280,720] has extra data? false encryption scheme: Unencrypted rotation: 0°
pipeline_state: kSuspended
found_video_stream: true
video_codec_name: vp8
video_dds: false
video_decoder: FFmpegVideoDecoder
duration: unknown
height: 720
width: 1280
video_buffering_state: BUFFERING_HAVE_NOTHING
for_suspended_start: false
pipeline_buffering_state: BUFFERING_HAVE_NOTHING
event: PAUSELog
Timestamp Property Value
00:00:00 00 origin_url http://localhost:52531/
00:00:00 00 frame_url http://localhost:52531/
00:00:00 00 frame_title Home Page
00:00:00 00 url blob:http://localhost:52531/dcb25d89-9830-40a5-ba88-33c13b5c03eb
00:00:00 00 info ChunkDemuxer: buffering by DTS
00:00:00 35 pipeline_state kStarting
00:00:15 213 found_video_stream true
00:00:15 213 video_codec_name vp8
00:00:15 216 video_dds false
00:00:15 216 video_decoder FFmpegVideoDecoder
00:00:15 216 info Selected FFmpegVideoDecoder for video decoding, config: codec: vp8 format: 1 profile: vp8 coded size: [1280,720] visible rect: [0,0,1280,720] natural size: [1280,720] has extra data? false encryption scheme: Unencrypted rotation: 0°
00:00:15 216 pipeline_state kPlaying
00:00:15 213 duration unknown
00:00:16 661 height 720
00:00:16 661 width 1280
00:00:16 665 video_buffering_state BUFFERING_HAVE_ENOUGH
00:00:16 665 for_suspended_start false
00:00:16 665 pipeline_buffering_state BUFFERING_HAVE_ENOUGH
00:00:16 667 pipeline_state kSuspending
00:00:16 670 pipeline_state kSuspended
00:00:52 759 info Effective playback rate changed from 0 to 1
00:00:52 759 event PLAY
00:00:52 759 pipeline_state kResuming
00:00:52 760 video_dds false
00:00:52 760 video_decoder FFmpegVideoDecoder
00:00:52 760 info Selected FFmpegVideoDecoder for video decoding, config: codec: vp8 format: 1 profile: vp8 coded size: [1280,720] visible rect: [0,0,1280,720] natural size: [1280,720] has extra data? false encryption scheme: Unencrypted rotation: 0°
00:00:52 760 pipeline_state kPlaying
00:00:52 793 height 720
00:00:52 793 width 1280
00:00:52 798 video_buffering_state BUFFERING_HAVE_ENOUGH
00:00:52 798 for_suspended_start false
00:00:52 798 pipeline_buffering_state BUFFERING_HAVE_ENOUGH
00:00:56 278 video_buffering_state BUFFERING_HAVE_NOTHING
00:00:56 295 for_suspended_start false
00:00:56 295 pipeline_buffering_state BUFFERING_HAVE_NOTHING
00:01:20 717 event PAUSE
00:01:33 538 event PLAY
00:01:35 94 event PAUSE
00:01:55 561 pipeline_state kSuspending
00:01:55 563 pipeline_state kSuspendedCan someone tell me what’s wrong with my code, or dose chrome require some magic configuration to work ?
Thanks
Please excuse my english :)