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Médias (16)
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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
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FFMPEG sound not playing in video converted from audio in browsers (chrome/firefox) except safari
7 février 2019, par Bhavik RathodSound not playing in Video (mp4) converted from Audio (mp3), Below is my code
I am using FFMPEG version 4.1
ffmpeg -y -loop 1 -i background.jpg -i dia.mp3 -c:v libx264 -tune stillimage -pix_fmt yuv420p -c:a ac3_fixed -b:a 128k -shortest -vf "[in]drawtext=text=Hi:x=(w-text_w)/2:y=(h-text_h):fontfile=arial.ttf:fontsize=100:fontcolor=ffffff:alpha='if(lt(t,9.2763066666667),0,if(lt(t,9.2763066666667+1),(t-9.2763066666667)/1,if(lt(t,12.83428),1,if(lt(t,12.83428+1),(1-(t-12.83428))/1,0))))'" output.mp4 2>&1I have tried putting -movflags +faststart as an output option
I am using
ac3_fixedbecause i found that fix for handle memory management, My server is getting hang on usingaacno video created nothing i have to restart instance to get back to work. by usingac3_fixedeverything is perfect but issue is only sound playing in chrome and firefox.Below is full log while using
aac`ffmpeg version 4.1-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc-6 --enable-fontconfig --enable-frei0r --enable-gnutls --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-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg
libavutil 56. 22.100 / 56. 22.100
libavcodec 58. 35.100 / 58. 35.100
libavformat 58. 20.100 / 58. 20.100
libavdevice 58. 5.100 / 58. 5.100
libavfilter 7. 40.101 / 7. 40.101
libswscale 5. 3.100 / 5. 3.100
libswresample 3. 3.100 / 3. 3.100
libpostproc 55. 3.100 / 55. 3.100
Input #0, image2, from '/var/www/html/xxxxx/assets/media/guest/e0ce3bfdab4b305d00459328afb94ca6.jpg':
Duration: 00:00:00.04, start: 0.000000, bitrate: 196382 kb/s
Stream #0:0: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 25 tbc
[mp3 @ 0x5970440] Estimating duration from bitrate, this may be inaccurate
Input #1, mp3, from '/var/www/html/xxxxx/assets/media/guest/e0ce3bfdab4b305d00459328afb94ca6.mp3':
Metadata:
title : In the Cafe
album : British Council
genre : Blues
Duration: 00:01:02.77, start: 0.000000, bitrate: 128 kb/s
Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 128 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (mjpeg (native) -> h264 (libx264))
Stream #1:0 -> #0:1 (mp3 (mp3float) -> aac (native))
Press [q] to stop, [?] for help
[swscaler @ 0x5b91ec0] deprecated pixel format used, make sure you did set range correctly
[libx264 @ 0x598e640] using SAR=1/1
[libx264 @ 0x598e640] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0x598e640] profile High, level 4.0, 4:2:0, 8-bit
[libx264 @ 0x598e640] 264 - core 157 r2935 545de2f - H.264/MPEG-4 AVC codec - Copyleft 2003-2018 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:-3:-3 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=2.00:0.70 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-4 threads=1 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.20
Output #0, mp4, to '/var/www/html/xxxxx/assets/media/guest/created_video/e0ce3bfdab4b305d00459328afb94ca6.mp4':
Metadata:
encoder : Lavf58.20.100
Stream #0:0: Video: h264 (libx264) (avc1 / 0x31637661), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 25 fps, 12800 tbn, 25 tbc
Metadata:
encoder : Lavc58.35.100 libx264
Side data:
cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
Stream #0:1: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s
Metadata:
encoder : Lavc58.35.100 aac
frame= 14 fps=0.0 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A speed= frame= 28 fps= 27 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A speed= frame= 42 fps= 27 q=28.0 size= 0kB time=00:00:00.00 bitrate=N/A speed= frame= 44 fps= 21 q=28.0 size= 256kB time=00:00:00.02 bitrate=72266.5kbitsframe= 49 fps= 19 q=28.0 size= 256kB time=00:00:00.20 bitrate=10323.8kbitsframe= 55 fps= 18 q=28.0 size= 256kB time=00:00:00.48 bitrate=4353.4kbits/frame= 62 fps= 17 q=28.0 size= 256kB time=00:00:00.72 bitrate=2890.7kbits/frame= 68 fps= 16 q=28.0 size= 256kB time=00:00:01.00 bitrate=2088.6kbits/
` -
"ffmpeg has text relocations" error in Android
5 janvier 2015, par djzmoI’m building an Android video-sending application. I need the video to be compressed before being uploaded. In this case, I chose to use this Android wrapper of FFmpeg : https://github.com/hiteshsondhi88/ffmpeg-android-java
FFmpeg error: WARNING: linker: /data/data/com.mycompany.myapplication/files/ffmpeg has text relocations. This is wasting memory and prevents security hardening. Please fix.
ffmpeg version n2.4.2 Copyright (c) 2000-2014 the FFmpeg developers
built on Oct 7 2014 15:08:46 with gcc 4.8 (GCC)
configuration: --target-os=linux --cross-prefix=/home/sb/Source-Code/ffmpeg-android/toolchain-android/bin/arm-linux-androideabi- --arch=arm --cpu=cortex-a8 --enable-runtime-cpudetect --sysroot=/home/sb/Source-Code/ffmpeg-android/toolchain-android/sysroot --enable-pic --enable-libx264 --enable-libass --enable-libfreetype --enable-libfribidi --enable-fontconfig --enable-pthreads --disable-debug --disable-ffserver --enable-version3 --enable-hardcoded-tables --disable-ffplay --disable-ffprobe --enable-gpl --enable-yasm --disable-doc --disable-shared --enable-static --pkg-config=/home/sb/Source-Code/ffmpeg-android/ffmpeg-pkg-config --prefix=/home/sb/Source-Code/ffmpeg-android/build/armeabi-v7a-neon --extra-cflags='-I/home/sb/Source-Code/ffmpeg-android/toolchain-android/include -U_FORTIFY_SOURCE -D_FORTIFY_SOURCE=2 -fno-strict-overflow -fstack-protector-all -mfpu=neon' --extra-ldflags='-L/home/sb/Source-Code/ffmpeg-android/toolchain-android/lib -Wl,-z,relro -Wl,-z,now -pie' --extra-libs='-lpng -lexpat -lm' --extra-cxxflags=
libavutil 54. 7.100 / 54. 7.100
libavcodec 56. 1.100 / 56. 1.100
libavformat 56. 4.101 / 56. 4.101
libavdevice 56. 0.100 / 56. 0.100
libavfilter 5. 1.100 / 5. 1.100
libswscale 3. 0.100 / 3. 0.100
libswresample 1. 1.100 / 1. 1.100
libpostproc 53. 0.100 / 53. 0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/storage/emulated/0/DCIM/Camera/VID_20150104_235926.mp4':
Metadata:
major_brand : mp42
minor_version : 0
compatible_brands: isommp42
creation_time : 2015-01-04 16:00:40
Duration: 00:01:12.60, start: 0.000000, bitrate: 19590 kb/s
Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080, 19364 kb/s, SAR 65536:65536 DAR 16:9, 26.73 fps, 26.92 tbr, 90k tbn, 180k tbc (default)
Metadata:
creation_time : 2015-01-04 16:00:40
handler_name : VideoHandle
Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 196 kb/s (default)
Metadata:
creation_time : 2015-01-04 16:00:40
handler_name : SoundHandle
[libx264 @ 0x40208800] using SAR=1/1
[libx264 @ 0x40208800] frame MB size (120x68) > level limit (1620)
[libx264 @ 0x40208800] DPB size (1 frames, 8160 mbs) > level limit (0 frames, 8100 mbs)
[libx264 @ 0x40208800] MB rate (219640) > level limit (40500)
[libx264 @ 0x40208800] using cpu capabilities: none!
[libx264 @ 0x40208800] profile Constrained Baseline, level 3.0
[libx264 @ 0x40208800] 264 - core 142 - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc_lookahead=0 rc=crf mbtree=0 crf=28.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=2500 vbv_bufsize=1835 crf_max=0.0 nal_hrd=none filler=0 ip_ratio=1.40 aq=0
Output #0, mp4, to '/storage/emulated/0/Android/data/com.mycompany.myapplication/cache/VID_20150104_235926.mp4':
Metadata:
major_brand : mp42
minor_version : 0
compatible_brands: isommp42
encoder : Lavf56.4.101
Stream #0:0(eng): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 1920x1080 [SAR 65536:65536 DAR 16:9], q=-1--1, 2500 kb/s, 26.92 fps, 10336 tbn, 26.92 tbc (default)
Metadata:
creation_time : 2015-01-04 16:00:40
handler_name : VideoHandle
encoder : Lavc56.1.100 libx264
Stream #0:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 4Everything seemed fine when I compress small files (below 20 seconds), but when I tried to compress larger files, I seldom get the above error in the middle of the compression process, and the process was also terminated immediately. I need the application to be able to compress one minute video to about 20-35 MB.
Here is the FFmpeg command I used :
ffmpeg -y -i -codec:v libx264 -profile:v baseline -level 3.0 -b:v 2500k -minrate 2500k -maxrate 2500k -bufsize 1835k -pix_fmt yuv420p -preset ultrafast -crf 28 -codec:a copy
Any idea on what went wrong ?
Any kind of help would be very appreciated. -
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)


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