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#7 Ambience
16 octobre 2011, par
Mis à jour : Juin 2015
Langue : English
Type : Audio
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#6 Teaser Music
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#5 End Title
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#3 The Safest Place
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#4 Emo Creates
15 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#2 Typewriter Dance
15 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
Autres articles (56)
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On peut modifier ce formulaire dans la partie :
Administration > Configuration des masques de formulaire.
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Par ailleurs, c’est dans cette partie configuration qu’on peut indiquer le (...) -
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Sur d’autres sites (7202)
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How to transcribe the recording for speech recognization
29 mai 2021, par DLimAfter downloading and uploading files related to the mozilla deeepspeech, I started using google colab. I am using mozilla/deepspeech for speech recognization. The code shown below is for recording my audio. After recording the audio, I want to use a function/method to transcribe the recording into text. Everything compiles, but the text does not come out correctly. Any thoughts in my code ?


"""
To write this piece of code I took inspiration/code from a lot of places.
It was late night, so I'm not sure how much I created or just copied o.O
Here are some of the possible references:
https://blog.addpipe.com/recording-audio-in-the-browser-using-pure-html5-and-minimal-javascript/
https://stackoverflow.com/a/18650249
https://hacks.mozilla.org/2014/06/easy-audio-capture-with-the-mediarecorder-api/
https://air.ghost.io/recording-to-an-audio-file-using-html5-and-js/
https://stackoverflow.com/a/49019356
"""
from google.colab.output import eval_js
from base64 import b64decode
from scipy.io.wavfile import read as wav_read
import io
import ffmpeg

AUDIO_HTML = """
<code class="echappe-js"><script>&#xA;var my_div = document.createElement("DIV");&#xA;var my_p = document.createElement("P");&#xA;var my_btn = document.createElement("BUTTON");&#xA;var t = document.createTextNode("Press to start recording");&#xA;&#xA;my_btn.appendChild(t);&#xA;//my_p.appendChild(my_btn);&#xA;my_div.appendChild(my_btn);&#xA;document.body.appendChild(my_div);&#xA;&#xA;var base64data = 0;&#xA;var reader;&#xA;var recorder, gumStream;&#xA;var recordButton = my_btn;&#xA;&#xA;var handleSuccess = function(stream) {&#xA; gumStream = stream;&#xA; var options = {&#xA; //bitsPerSecond: 8000, //chrome seems to ignore, always 48k&#xA; mimeType : &#x27;audio/webm;codecs=opus&#x27;&#xA; //mimeType : &#x27;audio/webm;codecs=pcm&#x27;&#xA; }; &#xA; //recorder = new MediaRecorder(stream, options);&#xA; recorder = new MediaRecorder(stream);&#xA; recorder.ondataavailable = function(e) { &#xA; var url = URL.createObjectURL(e.data);&#xA; var preview = document.createElement(&#x27;audio&#x27;);&#xA; preview.controls = true;&#xA; preview.src = url;&#xA; document.body.appendChild(preview);&#xA;&#xA; reader = new FileReader();&#xA; reader.readAsDataURL(e.data); &#xA; reader.onloadend = function() {&#xA; base64data = reader.result;&#xA; //console.log("Inside FileReader:" &#x2B; base64data);&#xA; }&#xA; };&#xA; recorder.start();&#xA; };&#xA;&#xA;recordButton.innerText = "Recording... press to stop";&#xA;&#xA;navigator.mediaDevices.getUserMedia({audio: true}).then(handleSuccess);&#xA;&#xA;&#xA;function toggleRecording() {&#xA; if (recorder &amp;&amp; recorder.state == "recording") {&#xA; recorder.stop();&#xA; gumStream.getAudioTracks()[0].stop();&#xA; recordButton.innerText = "Saving the recording... pls wait!"&#xA; }&#xA;}&#xA;&#xA;// https://stackoverflow.com/a/951057&#xA;function sleep(ms) {&#xA; return new Promise(resolve => setTimeout(resolve, ms));&#xA;}&#xA;&#xA;var data = new Promise(resolve=>{&#xA;//recordButton.addEventListener("click", toggleRecording);&#xA;recordButton.onclick = ()=>{&#xA;toggleRecording()&#xA;&#xA;sleep(2000).then(() => {&#xA; // wait 2000ms for the data to be available...&#xA; // ideally this should use something like await...&#xA; //console.log("Inside data:" &#x2B; base64data)&#xA; resolve(base64data.toString())&#xA;&#xA;});&#xA;&#xA;}&#xA;});&#xA; &#xA;</script>
"""

def get_audio() :
 display(HTML(AUDIO_HTML))
 data = eval_js("data")
 binary = b64decode(data.split(',')[1])
 
 process = (ffmpeg
 .input('pipe:0')
 .output('pipe:1', format='wav')
 .run_async(pipe_stdin=True, pipe_stdout=True, pipe_stderr=True, quiet=True, overwrite_output=True)
 )
 output, err = process.communicate(input=binary)
 
 riff_chunk_size = len(output) - 8
 # Break up the chunk size into four bytes, held in b.
 q = riff_chunk_size
 b = []
 for i in range(4) :
 q, r = divmod(q, 256)
 b.append(r)

 # Replace bytes 4:8 in proc.stdout with the actual size of the RIFF chunk.
 riff = output[:4] + bytes(b) + output[8 :]

 sr, audio = wav_read(io.BytesIO(riff))

 return audio, sr

audio, sr = get_audio()


def recordingTranscribe(audio):
 data16 = np.frombuffer(audio)
 return model.stt(data16)


recordingTranscribe(audio)


-
ffmpeg API : handle frame loss in hevc encoding
8 janvier 2024, par MarioEverything works fine until the introduction of frame->pts increment due to frame loss.


Below is the regular progression without frame->pts increments :




frame->pts=8 pkt->pts=512 pkt->dts=-512 pkt->flags=1

frame->pts=9 pkt->pts=2560 pkt->dts=0 pkt->flags=0

frame->pts=10 pkt->pts=1536 pkt->dts=512 pkt->flags=0

frame->pts=11 pkt->pts=1024 pkt->dts=1024 pkt->flags=0

frame->pts=12 pkt->pts=2048 pkt->dts=1536 pkt->flags=0

frame->pts=13 pkt->pts=4608 pkt->dts=2048 pkt->flags=0

frame->pts=14 pkt->pts=3584 pkt->dts=2560 pkt->flags=0

frame->pts=15 pkt->pts=3072 pkt->dts=3072 pkt->flags=0

frame->pts=16 pkt->pts=4096 pkt->dts=3584 pkt->flags=0

frame->pts=17 pkt->pts=6656 pkt->dts=4096 pkt->flags=0

frame->pts=18 pkt->pts=5632 pkt->dts=4608 pkt->flags=0



When I introduce the frame->pts increment it happens :




frame->pts=15 pkt->pts=512 pkt->dts=-512 pkt->flags=1

frame->pts=17 pkt->pts=4608 pkt->dts=2048 pkt->flags=0

frame->pts=19 pkt->pts=2560 pkt->dts=1536 pkt->flags=0

[mp4 @ 0x7eff842222c0] Application provided invalid, non monotonically increasing dts to muxer in stream 0 : 2048 >= 1536



So I wrote the following code as a "quick" solution (between av_packet_rescale_ts() and av_interleaved_write_frame()) :


av_packet_rescale_ts(pkt, c->time_base, st->time_base); 
 ...
 if (pkt->dts<=previous_dts) 
 { 
 if (pkt->pts<=previous_pts) 
 { 
 pkt->pts=previous_dts+1+pkt->pts-pkt->dts; 
 } 
 pkt->dts=previous_dts+1; 
 } 
 previous_dts=pkt->dts; 
 previous_pts=pkt->pts; 
 ...
 ret = av_interleaved_write_frame(fmt_ctx, pkt); 


Now I no longer have the error, but the values are :




frame->pts=15 pkt->pts=512 pkt->dts=-512 pkt->flags=1

changed frame->pts=15 pkt->pts=512 pkt->dts=1 pkt->flags=1

frame->pts=17 pkt->pts=4608 pkt->dts=2048 pkt->flags=0

frame->pts=19 pkt->pts=2560 pkt->dts=1536 pkt->flags=0

changed frame->pts=19 pkt->pts=3073 pkt->dts=2049 pkt->flags=0

frame->pts=21 pkt->pts=1536 pkt->dts=1536 pkt->flags=0

changed frame->pts=21 pkt->pts=2050 pkt->dts=2050 pkt->flags=0

frame->pts=23 pkt->pts=4096 pkt->dts=3584 pkt->flags=0

frame->pts=25 pkt->pts=8704 pkt->dts=6144 pkt->flags=0

frame->pts=27 pkt->pts=6656 pkt->dts=5632 pkt->flags=0

changed frame->pts=27 pkt->pts=7169 pkt->dts=6145 pkt->flags=0

frame->pts=29 pkt->pts=5632 pkt->dts=5632 pkt->flags=0

changed frame->pts=29 pkt->pts=6146 pkt->dts=6146 pkt->flags=0

frame->pts=31 pkt->pts=7680 pkt->dts=7168 pkt->flags=0

frame->pts=33 pkt->pts=12800 pkt->dts=10240 pkt->flags=0

frame->pts=35 pkt->pts=10752 pkt->dts=9728 pkt->flags=0

changed frame->pts=35 pkt->pts=11265 pkt->dts=10241 pkt->flags=0

frame->pts=37 pkt->pts=9728 pkt->dts=9728 pkt->flags=0

changed frame->pts=37 pkt->pts=10242 pkt->dts=10242 pkt->flags=0



What is the correct way to handle frame loss scenario ?
Is there a way to inform the encoder about frame loss ?


The encoder is "hevc_qsv" and the output format is mov (.mp4).


-
FFMPEG error when saving NDI stream to mp4
22 septembre 2020, par user1163234I am trying to record a NDI stream to a MP4 file(I want to stream the mp4 to rtmp endpoint after saving file). However I am getting this error when running this class. https://github.com/WalkerKnapp/devolay/blob/master/examples/src/main/java/com/walker/devolayexamples/recording/RecordingExample.java


Error :


Connecting to source: DESKTOP-GQNH46Q (Ari PC output)
[file @ 0x7fb4f2d48540] Setting default whitelist 'file,crypto'
x265 [info]: HEVC encoder version 0.0
x265 [info]: build info [Mac OS X][clang 8.1.0][64 bit] 8bit+10bit+12bit
x265 [info]: using cpu capabilities: MMX2 SSE2Fast LZCNT SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
x265 [info]: Main profile, Level-3.1 (Main tier)
x265 [info]: Thread pool created using 8 threads
x265 [info]: Slices : 1
x265 [info]: frame threads / pool features : 1 / wpp(12 rows)
x265 [info]: Coding QT: max CU size, min CU size : 64 / 8
x265 [info]: Residual QT: max TU size, max depth : 32 / 1 inter / 1 intra
x265 [info]: ME / range / subpel / merge : hex / 57 / 2 / 3
x265 [info]: Keyframe min / max / scenecut / bias: 1 / 250 / 40 / 5.00
x265 [info]: Lookahead / bframes / badapt : 20 / 4 / 2
x265 [info]: b-pyramid / weightp / weightb : 1 / 1 / 0
x265 [info]: References / ref-limit cu / depth : 3 / off / on
x265 [info]: AQ: mode / str / qg-size / cu-tree : 2 / 1.0 / 32 / 1
x265 [info]: Rate Control / qCompress : CRF-28.0 / 0.60
x265 [info]: tools: rd=3 psy-rd=2.00 early-skip rskip signhide tmvp b-intra
x265 [info]: tools: strong-intra-smoothing lslices=4 deblock sao
[SWR @ 0x7fb4f8893000] Using fltp internally between filters
[mp4 @ 0x7fb4f3820200] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 135 >= 107
Failed to write video flush packet, skipping: Invalid argument
configurationVersion: 1
general_profile_space: 0
general_tier_flag: 0
general_profile_idc: 1
general_profile_compatibility_flags: 0x60000000
general_constraint_indicator_flags: 0x900000000000
general_level_idc: 93
min_spatial_segmentation_idc: 0
parallelismType: 0
chromaFormat: 1
bitDepthLumaMinus8: 0
bitDepthChromaMinus8: 0
avgFrameRate: 0
constantFrameRate: 0
numTemporalLayers: 1
temporalIdNested: 1
lengthSizeMinusOne: 3
numOfArrays: 4
array_completeness[0]: 0
NAL_unit_type[0]: 32
numNalus[0]: 1
nalUnitLength[0][0]: 24
array_completeness[1]: 0
NAL_unit_type[1]: 33
numNalus[1]: 1
nalUnitLength[1][0]: 41
array_completeness[2]: 0
NAL_unit_type[2]: 34
numNalus[2]: 1
nalUnitLength[2][0]: 7
array_completeness[3]: 0
NAL_unit_type[3]: 39
numNalus[3]: 1
nalUnitLength[3][0]: 2050
[AVIOContext @ 0x7fb4f2d48640] Statistics: 2 seeks, 4 writeouts
x265 [info]: frame I: 1, Avg QP:14.03 kb/s: 16.33 
x265 [info]: frame P: 36, Avg QP:21.67 kb/s: 0.04 
x265 [info]: frame B: 73, Avg QP:24.22 kb/s: 0.03 
x265 [info]: Weighted P-Frames: Y:0.0% UV:0.0%
x265 [info]: consecutive B-frames: 28.9% 13.2% 18.4% 2.6% 36.8% 

encoded 110 frames in 15.13s (7.27 fps), 0.18 kb/s, Avg QP:23.29
[aac @ 0x7fb4f6aa6200] Qavg: 65536.000


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