
Recherche avancée
Médias (1)
-
Revolution of Open-source and film making towards open film making
6 octobre 2011, par
Mis à jour : Juillet 2013
Langue : English
Type : Texte
Autres articles (56)
-
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 (...) -
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 : (...) -
Automated installation script of MediaSPIP
25 avril 2011, parTo overcome the difficulties mainly due to the installation of server side software dependencies, an "all-in-one" installation script written in bash was created to facilitate this step on a server with a compatible Linux distribution.
You must have access to your server via SSH and a root account to use it, which will install the dependencies. Contact your provider if you do not have that.
The documentation of the use of this installation script is available here.
The code of this (...)
Sur d’autres sites (6495)
-
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 :)
-
Find a great Google Tag Manager alternative in Matomo Tag Manager
-
Google Speech API returns empty result for some FLAC files, and not for the others although they have same codec and sample rate
15 mars 2021, par ChadBelow code is what I used to make request for transcription.


import io
from google.cloud import speech_v1p1beta1 as speech
def transcribe_file(speech_file):
 """Transcribe the given audio file."""

 client = speech.SpeechClient()

 encoding = speech.RecognitionConfig.AudioEncoding.FLAC
 if os.path.splitext(speech_file)[1] == ".wav":
 encoding = speech.RecognitionConfig.AudioEncoding.LINEAR16
 with io.open(speech_file, "rb") as audio_file:
 content = audio_file.read()

 audio = speech.RecognitionAudio(content=content)
 config = speech.RecognitionConfig(
 encoding=speech.RecognitionConfig.AudioEncoding.FLAC,
 sample_rate_hertz=32000,
 language_code="ja-JP",
 max_alternatives=3,
 enable_word_time_offsets=True,
 enable_automatic_punctuation=True,
 enable_word_confidence=True,
 )

 response = client.recognize(config=config, audio=audio)
 #print(speech_file, "Recognition Done")
 return response



As I wrote in title, the results of response has empty list for some files, and not for some files.
They have same sample rate and codec(32000, FLAC)


Below is the result of
ffprobe -i "AUDIOFILE" -show_streams
for one of each cases.

Left one is empty one. The only difference is duration of file.


How can I get non empty results ?




Edit :


Result of ffprobe show stream show format


Something not captured in one screen


Sadly, re-mux didn't work.


I used ffmpeg-git-20210225


ffbrobe result of broken one


./ffprobe -show_streams -show_format broken.flac 
ffprobe version N-56320-ge937457b7b-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2007-2021 the FFmpeg developers
 built with gcc 8 (Debian 8.3.0-6)
 configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
 libavutil 56. 66.100 / 56. 66.100
 libavcodec 58.125.101 / 58.125.101
 libavformat 58. 68.100 / 58. 68.100
 libavdevice 58. 12.100 / 58. 12.100
 libavfilter 7.107.100 / 7.107.100
 libswscale 5. 8.100 / 5. 8.100
 libswresample 3. 8.100 / 3. 8.100
 libpostproc 55. 8.100 / 55. 8.100
Input #0, flac, from 'broken.flac':
 Metadata:
 encoder : Lavf58.45.100
 Duration: 00:00:00.90, start: 0.000000, bitrate: 342 kb/s
 Stream #0:0: Audio: flac, 32000 Hz, mono, s16
[STREAM]
index=0
codec_name=flac
codec_long_name=FLAC (Free Lossless Audio Codec)
profile=unknown
codec_type=audio
codec_tag_string=[0][0][0][0]
codec_tag=0x0000
sample_fmt=s16
sample_rate=32000
channels=1
channel_layout=mono
bits_per_sample=0
id=N/A
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/32000
start_pts=0
start_time=0.000000
duration_ts=28672
duration=0.896000
bit_rate=N/A
max_bit_rate=N/A
bits_per_raw_sample=16
nb_frames=N/A
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=0
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
[/STREAM]
[FORMAT]
filename=broken.flac
nb_streams=1
nb_programs=0
format_name=flac
format_long_name=raw FLAC
start_time=0.000000
duration=0.896000
size=38362
bit_rate=342517
probe_score=100
TAG:encoder=Lavf58.45.100
[/FORMAT]



ffprobe result of non_broken one


./ffprobe -show_streams -show_format non_broken.flac 
ffprobe version N-56320-ge937457b7b-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2007-2021 the FFmpeg developers
 built with gcc 8 (Debian 8.3.0-6)
 configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
 libavutil 56. 66.100 / 56. 66.100
 libavcodec 58.125.101 / 58.125.101
 libavformat 58. 68.100 / 58. 68.100
 libavdevice 58. 12.100 / 58. 12.100
 libavfilter 7.107.100 / 7.107.100
 libswscale 5. 8.100 / 5. 8.100
 libswresample 3. 8.100 / 3. 8.100
 libpostproc 55. 8.100 / 55. 8.100
Input #0, flac, from 'non_broken.flac':
 Metadata:
 encoder : Lavf58.45.100
 Duration: 00:00:00.86, start: 0.000000, bitrate: 358 kb/s
 Stream #0:0: Audio: flac, 32000 Hz, mono, s16
[STREAM]
index=0
codec_name=flac
codec_long_name=FLAC (Free Lossless Audio Codec)
profile=unknown
codec_type=audio
codec_tag_string=[0][0][0][0]
codec_tag=0x0000
sample_fmt=s16
sample_rate=32000
channels=1
channel_layout=mono
bits_per_sample=0
id=N/A
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/32000
start_pts=0
start_time=0.000000
duration_ts=27648
duration=0.864000
bit_rate=N/A
max_bit_rate=N/A
bits_per_raw_sample=16
nb_frames=N/A
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=0
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
[/STREAM]
[FORMAT]
filename=non_broken.flac
nb_streams=1
nb_programs=0
format_name=flac
format_long_name=raw FLAC
start_time=0.000000
duration=0.864000
size=38701
bit_rate=358342
probe_score=100
TAG:encoder=Lavf58.45.100
[/FORMAT]



And the result of
ffmpeg -f lavfi -i sine=d=0.864:r=32000 output.flac


ffmpeg version 3.4.8-0ubuntu0.2 Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
 configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
 WARNING: library configuration mismatch
 avcodec configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared --enable-version3 --disable-doc --disable-programs --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libtesseract --enable-libvo_amrwbenc
 libavutil 55. 78.100 / 55. 78.100
 libavcodec 57.107.100 / 57.107.100
 libavformat 57. 83.100 / 57. 83.100
 libavdevice 57. 10.100 / 57. 10.100
 libavfilter 6.107.100 / 6.107.100
 libavresample 3. 7. 0 / 3. 7. 0
 libswscale 4. 8.100 / 4. 8.100
 libswresample 2. 9.100 / 2. 9.100
 libpostproc 54. 7.100 / 54. 7.100
Input #0, lavfi, from 'sine=d=0.864:r=32000':
 Duration: N/A, start: 0.000000, bitrate: 512 kb/s
 Stream #0:0: Audio: pcm_s16le, 32000 Hz, mono, s16, 512 kb/s
File 'output.flac' already exists. Overwrite ? [y/N] y
Stream mapping:
 Stream #0:0 -> #0:0 (pcm_s16le (native) -> flac (native))
Press [q] to stop, [?] for help
Output #0, flac, to 'output.flac':
 Metadata:
 encoder : Lavf57.83.100
 Stream #0:0: Audio: flac, 32000 Hz, mono, s16, 128 kb/s
 Metadata:
 encoder : Lavc57.107.100 flac
[Parsed_sine_0 @ 0x55c317ddda00] EOF timestamp not reliable
size= 16kB time=00:00:00.86 bitrate= 154.0kbits/s speed= 205x 
video:0kB audio:8kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 99.364586%