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Autres articles (101)
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MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
The zip file provided here only contains the sources of MediaSPIP in its standalone version.
To get a working installation, you must manually install all-software dependencies on the server.
If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...) -
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 ) (...)
Sur d’autres sites (13694)
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What is the best way to create video transcoding server for google cloud storage [on hold]
20 mai 2017, par ramana vvI would like to create best video transcoding server for my project . I’m already using ffmpeg , python for transcoding videos. My existing application is like this "after uploading the video to the GCE it will transcode that video and copy to cloud storage, after copy to cloud storage the file in the instance will be deleted. But when the video count is in 100’s then I need to increase the capacity of GCE then the cost will be very expensive so please help me to solve this issue.
Thanks in advance -
Can google cloud dataflow (apache beam) use ffmpeg to process video or image data
12 février 2016, par MichaelCan a dataflow process use ffmpeg to process video or images and if so what would a sample workflow look like
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Google Speech - Streaming Request Returns EOF
9 octobre 2017, par JoshUsing Go, I’m taking a RTMP stream, transcoding it to FLAC (using ffmpeg) and attempting to stream to Google’s Speech API to transcribe the audio. However, I keep getting
EOF
errors when sending the data. I can’t find any information on this error in the docs so I’m not exactly sure what’s causing it.I’m chunking the received data into 3s clips (length isn’t relevant as long as it’s less than the maximum length of a streaming recognition request).
Here is the core of my code :
func main() {
done := make(chan os.Signal)
received := make(chan []byte)
go receive(received)
go transcribe(received)
signal.Notify(done, os.Interrupt, syscall.SIGTERM)
select {
case <-done:
os.Exit(0)
}
}
func receive(received chan<- []byte) {
var b bytes.Buffer
stdout := bufio.NewWriter(&b)
cmd := exec.Command("ffmpeg", "-i", "rtmp://127.0.0.1:1935/live/key", "-f", "flac", "-ar", "16000", "-")
cmd.Stdout = stdout
if err := cmd.Start(); err != nil {
log.Fatal(err)
}
duration, _ := time.ParseDuration("3s")
ticker := time.NewTicker(duration)
for {
select {
case <-ticker.C:
stdout.Flush()
log.Printf("Received %d bytes", b.Len())
received <- b.Bytes()
b.Reset()
}
}
}
func transcribe(received <-chan []byte) {
ctx := context.TODO()
client, err := speech.NewClient(ctx)
if err != nil {
log.Fatal(err)
}
stream, err := client.StreamingRecognize(ctx)
if err != nil {
log.Fatal(err)
}
// Send the initial configuration message.
if err = stream.Send(&speechpb.StreamingRecognizeRequest{
StreamingRequest: &speechpb.StreamingRecognizeRequest_StreamingConfig{
StreamingConfig: &speechpb.StreamingRecognitionConfig{
Config: &speechpb.RecognitionConfig{
Encoding: speechpb.RecognitionConfig_FLAC,
LanguageCode: "en-GB",
SampleRateHertz: 16000,
},
},
},
}); err != nil {
log.Fatal(err)
}
for {
select {
case data := <-received:
if len(data) > 0 {
log.Printf("Sending %d bytes", len(data))
if err := stream.Send(&speechpb.StreamingRecognizeRequest{
StreamingRequest: &speechpb.StreamingRecognizeRequest_AudioContent{
AudioContent: data,
},
}); err != nil {
log.Printf("Could not send audio: %v", err)
}
}
}
}
}Running this code gives this output :
2017/10/09 16:05:00 Received 191704 bytes
2017/10/09 16:05:00 Saving 191704 bytes
2017/10/09 16:05:00 Sending 191704 bytes
2017/10/09 16:05:00 Could not send audio: EOF
2017/10/09 16:05:03 Received 193192 bytes
2017/10/09 16:05:03 Saving 193192 bytes
2017/10/09 16:05:03 Sending 193192 bytes
2017/10/09 16:05:03 Could not send audio: EOF
2017/10/09 16:05:06 Received 193188 bytes
2017/10/09 16:05:06 Saving 193188 bytes
2017/10/09 16:05:06 Sending 193188 bytes // Notice that this doesn't error
2017/10/09 16:05:09 Received 191704 bytes
2017/10/09 16:05:09 Saving 191704 bytes
2017/10/09 16:05:09 Sending 191704 bytes
2017/10/09 16:05:09 Could not send audio: EOFNotice that not all of the
Send
s fail.Could anyone point me in the right direction here ? Is it something to do with the FLAC headers or something ? I also wonder if maybe resetting the buffer causes some of the data to be dropped (i.e. it’s a non-trivial operation that actually takes some time to complete) and it doesn’t like this missing information ?
Any help would be really appreciated.