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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 (...) -
Multilang : améliorer l’interface pour les blocs multilingues
18 février 2011, parMultilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela. -
Le profil des utilisateurs
12 avril 2011, parChaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...)
Sur d’autres sites (10310)
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google speech to text errors out (grpc invalid deadline NaN)
15 décembre 2019, par jamescharlesworthI have a ffmpeg script that cuts an audio file into a short 5 second clip, however after I cut the file, calling the google speech
recognize
command errors out.Creating a clip - full code link :
const uri = 'http://traffic.libsyn.com/joeroganexp/p1400.mp3?dest-id=19997';
const command = ffmpeg(got.stream(uri));
command
.seek(0)
.duration(5)
.audioBitrate(128)
.format('mp3')
...which works fine and creates
./clip2.mp3
.I then take that file and upload it to speech to text api and it times out (script here. When I put
timeout
andmaxRetries
argument I can get the actual error :Error: 2 UNKNOWN: Getting metadata from plugin failed with error: Deadline is too far in the future
at Object.callErrorFromStatus (/Users/jamescharlesworth/Downloads/demo/node_modules/@grpc/grpc-js/build/src/call.js:30:26)
at Http2CallStream.<anonymous> (/Users/jamescharlesworth/Downloads/demo/node_modules/@grpc/grpc-js/build/src/client.js:96:33)
at Http2CallStream.emit (events.js:215:7)
at /Users/jamescharlesworth/Downloads/demo/node_modules/@grpc/grpc-js/build/src/call-stream.js:98:22
at processTicksAndRejections (internal/process/task_queues.js:75:11) {
code: 2,
details: 'Getting metadata from plugin failed with error: Deadline is too far in the future',
metadata: Metadata { internalRepr: Map {}, options: {} },
note: 'Exception occurred in retry method that was not classified as transient'
}
</anonymous>Stepping through the grpc code i see that the deadline is an invalid date.
This seems to be causing the issue but i assume it may be from incorrect params passed into the speechclient.recognize()
method.A few other things to note :
- The script works for some audio files, not all
- I can upload the broken my clip mp3 clip2.mp3 to the demo app here and it works fine.
- If I change the seek command of my ffmpeg script to start at
0.01
speech recognize command will work (however it breaks other audio clips as its not the correct starting point). I notice that when i do this the png of the mp3 gets stripped out and is a much smaller file size
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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.
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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