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Sur d’autres sites (5525)
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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 :)
-
On-premise analytics demand grows as Google Analytics GDPR uncertainties continue
7 janvier 2020, par Jake Thornton — Privacy -
How to grab ffmpeg's output as binary and write it to a file on the fly such that video players can play it in real time ?
29 décembre 2022, par Mister MystèreI want to stream a RTSP-streaming device to a video player such as VLC but the catch is that, in between, the binary data needs to go through a custom high-speed serial link. I control what goes in this link from a C++ program.


I was happily surprised to see that the following line allowed me to watch the RTSP stream by just opening "out.bin" from VLC which was a good lead for fast and efficient binary transmission of the stream :


ffmpeg -i "rtsp://admin:password@X.X.X.X:554/h264Preview_01_main" -c:v copy -c:a copy -f mpegts out.bin



I already wondered how ffmpeg manages to allow VLC to read that file, while itself writing to it at the same time. Turns out I was right to wonder, see below.


I told myself I could make this command pipe its output to the standard output, and then in turn pipe the standard output to a file that I can read, (later, slice it, transmit the chunks and reconstruct it) and then write to an output file. However, this does not work :


#include 
#include 
#include 

#define BUFSIZE 188 //MPEG-TS packet size

int main()
{
 char *cmd = (char*)"ffmpeg -i \"rtsp://admin:password@X.X.X.X:554/h264Preview_01_main\" -c:v copy -c:a copy -f mpegts pipe:1 -loglevel quiet";
 char buf[BUFSIZE];
 FILE *ptr, *file;

 file = fopen("./out.bin", "w");

 if (!file)
 {
 printf("Failed to open output file for writing, aborting");
 abort();
 }

 if ((ptr = popen(cmd, "r")) != NULL) {
 printf("Writing RTSP stream to file...");

 while (!kbhit())
 {
 if(fread(&buf, sizeof(char), BUFSIZE, ptr) != 0)
 {
 fwrite(buf, sizeof(char), BUFSIZE, file);
 }
 else
 {
 printf("No data\n");
 }
 }
 pclose(ptr);
 }
 else
 {
 printf("Failed to open pipe from ffmpeg command, aborting");
 }

 printf("End of program");

 fclose(file);
 return 0;
}



Since VLC says "your input can't be opened" - whereas this works just fine :


ffmpeg -i "rtsp://admin:password@X.X.X.X:554/h264Preview_01_main" -c:v copy -c:a copy -f mpegts pipe:1 -loglevel quiet > out.bin



This is what ends up in the file after I close the program, versus the result of the command immediately above :



The file is always 2kB regardless of how long I run the program : "No data" is shown repeatedly in the console output.


Why doesn't it work ? If it is not just a bug, how can I grab the stream as binary at some point, and write it at the end to a file that VLC can read ?


Update


New code after applying Craig Estey's fix to my stupid mistake. The end result is that the MPEG-TS frames don't seem to shift anymore but the file writing stops partway into one of the first few frames (the console only shows a few ">" symbols and then stays silent, c.f. code).


#include 
#include 
#include 

#define BUFSIZE 188 // MPEG-TS packet size

int
main()
{
 char *cmd = (char *) "ffmpeg -i \"rtsp://127.0.0.1:8554/test.sdp\" -c:v copy -c:a copy -f mpegts pipe:1 -loglevel quiet";
 char buf[BUFSIZE];
 FILE *ptr,
 *file;

 file = fopen("./out.ts", "w");

 if (!file) {
 printf("Failed to open output file for writing, aborting");
 abort();
 }

 if ((ptr = popen(cmd, "r")) != NULL) {
 printf("Writing RTSP stream to file...");

 while(!kbhit()) {
 ssize_t rlen = fread(&buf, sizeof(char), BUFSIZE, ptr);
 if(rlen != 0)
 {
 printf(">");
 fwrite(buf, sizeof(char), rlen, file);
 fflush(file);
 }
 }
 pclose(ptr);
 }
 else {
 printf("Failed to open pipe from ffmpeg command, aborting");
 }

 printf("End of program");

 fclose(file);
 return 0;
}



This can be tested on any computer with VLC and a webcam : open VLC, open capture device, capture mode directshow, (switch "play" for "stream"), next, display locally, select RTSP, Add, path=/test.sdp, next, transcoding=H264+MP3 (TS), replace rtsp ://:8554/ with rtsp ://127.0.0.1:8554/ in the generated command line, stream.


To test that streaming is ok, you can just open a command terminal and enter "ffmpeg -i "rtsp ://127.0.0.1:8554/test.sdp" -c:v copy -c:a copy -f mpegts pipe:1 -loglevel quiet", the terminal should fill up with binary data.


To test the program, just compile, run, and open out.ts after the program has run.