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Transcode webcam blob to RTMP using ffmpeg.wasm
29 novembre 2023, par hassan moradnezhadI'm trying transcode webcam blob data to a rtmp server from browser by using ffmpeg.wasm .

first, i create a MediaStream.

const stream = await navigator.mediaDevices.getUserMedia({
 video: true,
 });



then, i create a MediaRecorder.


const recorder = new MediaRecorder(stream, {mimeType: "video/webm; codecs:vp9"});
 recorder.ondataavailable = handleDataAvailable;
 recorder.start(0)



when data is available, i call a function called
handleDataAvailable
.

here is the function.

const handleDataAvailable = (event: BlobEvent) => {
 console.log("data-available");
 if (event.data.size > 0) {
 recordedChunksRef.current.push(event.data);
 transcode(event.data)
 }
 };



in above code, i use another function which called
transcode
it's goal is going to send data to rtmp server using useffmpeg.wasm
.

here it is.

const transcode = async (inputVideo: Blob | undefined) => {
 const ffmpeg = ffmpegRef.current;
 const fetchFileOutput = await fetchFile(inputVideo)
 ffmpeg?.writeFile('input.webm', fetchFileOutput)

 const data = await ffmpeg?.readFile('input.webm');
 if (videoRef.current) {
 videoRef.current.src =
 URL.createObjectURL(new Blob([(data as any)?.buffer], {type: 'video/webm'}));
 }

 // execute by node-media-server config 1
 await ffmpeg?.exec(['-re', '-i', 'input.webm', '-c', 'copy', '-f', 'flv', "rtmp://localhost:1935/live/ttt"])

 // execute by node-media-server config 2
 // await ffmpeg?.exec(['-re', '-i', 'input.webm', '-c:v', 'libx264', '-preset', 'veryfast', '-tune', 'zerolatency', '-c:a', 'aac', '-ar', '44100', '-f', 'flv', 'rtmp://localhost:1935/live/ttt']);

 // execute by stack-over-flow config 1
 // await ffmpeg?.exec(['-re', '-i', 'input.webm', '-c:v', 'h264', '-c:a', 'aac', '-f', 'flv', "rtmp://localhost:1935/live/ttt"]);

 // execute by stack-over-flow config 2
 // await ffmpeg?.exec(['-i', 'input.webm', '-c:v', 'libx264', '-flags:v', '+global_header', '-c:a', 'aac', '-ac', '2', '-f', 'flv', "rtmp://localhost:1935/live/ttt"]);

 // execute by stack-over-flow config 3
 // await ffmpeg?.exec(['-i', 'input.webm', '-acodec', 'aac', '-ac', '2', '-strict', 'experimental', '-ab', '160k', '-vcodec', 'libx264', '-preset', 'slow', '-profile:v', 'baseline', '-level', '30', '-maxrate', '10000000', '-bufsize', '10000000', '-b', '1000k', '-f', 'flv', 'rtmp://localhost:1935/live/ttt']);

 }



after running app and start streaming, console logs are as below.


ffmpeg >>> ffmpeg version 5.1.3 Copyright (c) 2000-2022 the FFmpeg developers index.tsx:81:20
ffmpeg >>> built with emcc (Emscripten gcc/clang-like replacement + linker emulating GNU ld) 3.1.40 (5c27e79dd0a9c4e27ef2326841698cdd4f6b5784) index.tsx:81:20
ffmpeg >>> configuration: --target-os=none --arch=x86_32 --enable-cross-compile --disable-asm --disable-stripping --disable-programs --disable-doc --disable-debug --disable-runtime-cpudetect --disable-autodetect --nm=emnm --ar=emar --ranlib=emranlib --cc=emcc --cxx=em++ --objcc=emcc --dep-cc=emcc --extra-cflags='-I/opt/include -O3 -msimd128' --extra-cxxflags='-I/opt/include -O3 -msimd128' --disable-pthreads --disable-w32threads --disable-os2threads --enable-gpl --enable-libx264 --enable-libx265 --enable-libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libopus --enable-zlib --enable-libwebp --enable-libfreetype --enable-libfribidi --enable-libass --enable-libzimg index.tsx:81:20
ffmpeg >>> libavutil 57. 28.100 / 57. 28.100 index.tsx:81:20
ffmpeg >>> libavcodec 59. 37.100 / 59. 37.100 index.tsx:81:20
ffmpeg >>> libavformat 59. 27.100 / 59. 27.100 index.tsx:81:20
ffmpeg >>> libavdevice 59. 7.100 / 59. 7.100 index.tsx:81:20
ffmpeg >>> libavfilter 8. 44.100 / 8. 44.100 index.tsx:81:20
ffmpeg >>> libswscale 6. 7.100 / 6. 7.100 index.tsx:81:20
ffmpeg >>> libswresample 4. 7.100 / 4. 7.100 index.tsx:81:20
ffmpeg >>> libpostproc 56. 6.100 / 56. 6.100 index.tsx:81:20
ffmpeg >>> Input #0, matroska,webm, from 'input.webm': index.tsx:81:20
ffmpeg >>> Metadata: index.tsx:81:20
ffmpeg >>> encoder : QTmuxingAppLibWebM-0.0.1 index.tsx:81:20
ffmpeg >>> Duration: N/A, start: 0.000000, bitrate: N/A index.tsx:81:20
ffmpeg >>> Stream #0:0(eng): Video: vp8, yuv420p(progressive), 640x480, SAR 1:1 DAR 4:3, 15.50 tbr, 1k tbn (default)



the problem is when
ffmpeg.wasm
try to execute the last command.

await ffmpeg?.exec(['-re', '-i', 'input.webm', '-c', 'copy', '-f', 'flv', "rtmp://localhost:1935/live/ttt"])
.

it just calls aGET Request
, I will send further details about this request.

as u can see, i try to use lots of arg sample withffmpeg?.exec
, but non of them works.

the network tab in browser, after
ffmpeg.wasm
execute the command is as below.



it send a
GET request
tows://localhost:1935/

and nothing happened after that.

for backend, i use node-media-server and here is my output logs when
ffmpeg.wasm
trying to execute the args

11/28/2023 19:33:18 55301 [INFO] [rtmp disconnect] id=JL569YOF
[NodeEvent on doneConnect] id=JL569YOF args=undefined



at last here are my ques




- 

- how can i achive this option ?
- is it possible to share webcam to rtmp server ?








-
ffmpeg : Strange behaviour with fade-in and fade-out in conjunction vs successively
29 novembre 2023, par CErI have a video file I want to fade in and out over a duration of three seconds each.


My code is :


ffmpeg -i video.mp4 -vf "fade=t=in:st=0:d=3,fade=t=out:st=169.288000:d=3" videofaded.mp4


Expected behaviour : Fade from and to black over 3s.
What I get : No fade in at all but a fade out as specified.


If I run
ffmpeg -i video.mp4 -vf "fade=t=in:st=0:d=3" -c:a copy fadein.mp4
to juste fade in everything works as expected.

Can anybody help my identify the problem ? I want to fade in and out in the same turn as there are a couple of videos to edit.


Thanks,
Chris


Here is my ffmpeg output :


ffmpeg version 4.2.7-0ubuntu0.1 Copyright (c) 2000-2022 the FFmpeg developers
 built with gcc 9 (Ubuntu 9.4.0-1ubuntu1~20.04.1)
 configuration: --prefix=/usr --extra-version=0ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
 libavutil 56. 31.100 / 56. 31.100
 libavcodec 58. 54.100 / 58. 54.100
 libavformat 58. 29.100 / 58. 29.100
 libavdevice 58. 8.100 / 58. 8.100
 libavfilter 7. 57.100 / 7. 57.100
 libavresample 4. 0. 0 / 4. 0. 0
 libswscale 5. 5.100 / 5. 5.100
 libswresample 3. 5.100 / 3. 5.100
 libpostproc 55. 5.100 / 55. 5.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fffda8a7300] Could not find codec parameters for stream 0 (Video: h264 (avc1 / 0x31637661), none, 1920x1080, 1087 kb/s): unspecified pixel format
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '20231106_Einfuehrung_Word_Mark_Fragen.mp4':
 Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 encoder : Lavf58.29.100
 Duration: 00:02:52.29, start: 0.016000, bitrate: 1126 kb/s
 Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), none, 1920x1080, 1087 kb/s, 16 fps, 16 tbr, 10k tbn, 20k tbc (default)
 Metadata:
 handler_name : VideoHandler
 Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 16000 Hz, mono, fltp, 72 kb/s (default)
 Metadata:
 handler_name : SoundHandler
File 'videofaded.mp4' already exists. Overwrite ? [y/N] y
Stream mapping:
 Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
 Stream #0:1 -> #0:1 (aac (native) -> aac (native))
Press [q] to stop, [?] for help
[libx264 @ 0x7fffda8c2c40] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2 AVX512
[libx264 @ 0x7fffda8c2c40] profile High, level 4.0
[libx264 @ 0x7fffda8c2c40] 264 - core 155 r2917 0a84d98 - H.264/MPEG-4 AVC codec - Copyleft 2003-2018 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=24 lookahead_threads=4 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=16 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=1:1.00
Output #0, mp4, to 'videofaded.mp4':
 Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 encoder : Lavf58.29.100
 Stream #0:0(eng): Video: h264 (libx264) (avc1 / 0x31637661), yuv420p, 1920x1080, q=-1--1, 16 fps, 16384 tbn, 16 tbc (default)
 Metadata:
 handler_name : VideoHandler
 encoder : Lavc58.54.100 libx264
 Side data:
 cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
 Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 16000 Hz, mono, fltp, 69 kb/s (default)
 Metadata:
 handler_name : SoundHandler
 encoder : Lavc58.54.100 aac
frame= 2757 fps=273 q=-1.0 Lsize= 21394kB time=00:02:52.28 bitrate=1017.2kbits/s dup=92 drop=0 speed=17.1x
video:19842kB audio:1492kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.280957%
[libx264 @ 0x7fffda8c2c40] frame I:12 Avg QP:15.18 size:232156
[libx264 @ 0x7fffda8c2c40] frame P:1751 Avg QP:20.38 size: 9540
[libx264 @ 0x7fffda8c2c40] frame B:994 Avg QP:23.59 size: 832
[libx264 @ 0x7fffda8c2c40] consecutive B-frames: 40.7% 29.5% 13.1% 16.8%
[libx264 @ 0x7fffda8c2c40] mb I I16..4: 44.5% 15.6% 39.9%
[libx264 @ 0x7fffda8c2c40] mb P I16..4: 0.5% 0.2% 0.2% P16..4: 9.3% 1.1% 1.0% 0.0% 0.0% skip:87.8%
[libx264 @ 0x7fffda8c2c40] mb B I16..4: 0.1% 0.1% 0.0% B16..8: 6.2% 0.2% 0.1% direct: 0.1% skip:93.1% L0:31.0% L1:66.2% BI: 2.9%
[libx264 @ 0x7fffda8c2c40] 8x8 transform intra:20.6% inter:17.2%
[libx264 @ 0x7fffda8c2c40] coded y,uvDC,uvAC intra: 40.5% 40.6% 18.4% inter: 3.2% 2.1% 0.3%
[libx264 @ 0x7fffda8c2c40] i16 v,h,dc,p: 40% 33% 8% 19%
[libx264 @ 0x7fffda8c2c40] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 36% 14% 43% 2% 1% 1% 1% 1% 2%
[libx264 @ 0x7fffda8c2c40] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 37% 32% 15% 2% 3% 3% 3% 3% 3%
[libx264 @ 0x7fffda8c2c40] i8c dc,h,v,p: 56% 20% 19% 5%
[libx264 @ 0x7fffda8c2c40] Weighted P-Frames: Y:0.7% UV:0.7%
[libx264 @ 0x7fffda8c2c40] ref P L0: 85.4% 5.7% 6.2% 2.7% 0.0%
[libx264 @ 0x7fffda8c2c40] ref B L0: 79.1% 18.3% 2.6%
[libx264 @ 0x7fffda8c2c40] ref B L1: 96.6% 3.4%
[libx264 @ 0x7fffda8c2c40] kb/s:943.30
[aac @ 0x7fffda8f1e00] Qavg: 18927.953



-
Extract all audio from a mov file into a aac file [closed]
5 février 2024, par Sushrut KaulI have a video which contains several streams as shown below :


Stream #0:0[0x1](eng): Video: prores (XQ) (ap4x / 0x78347061), yuv444p12le(bt709/smpte432/smpte2084, progressive), 3840x2160, 1222881 kb/s, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 24k tbn (default)
 Metadata:
 creation_time : 2022-08-26T22:20:26.000000Z
 handler_name : Apple Video Media Handler
 vendor_id : appl
 encoder : Apple ProRes 4444 XQ
 timecode : 00:59:59:00
 Side data:
 Content Light Level Metadata, MaxCLL=726, MaxFALL=93
 Stream #0:1[0x2](eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (FL), s32 (24 bit), 1152 kb/s (default)
 Metadata:
 creation_time : 2022-08-26T22:20:26.000000Z
 handler_name : Apple Sound Media Handler
 vendor_id : [0][0][0][0]
 Stream #0:2[0x3](eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (FR), s32 (24 bit), 1152 kb/s (default)
 Metadata:
 creation_time : 2022-08-26T22:20:26.000000Z
 handler_name : Apple Sound Media Handler
 vendor_id : [0][0][0][0]
 Stream #0:3[0x4](eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, s32 (24 bit), 1152 kb/s (default)
 Metadata:
 creation_time : 2022-08-26T22:20:26.000000Z
 handler_name : Apple Sound Media Handler
 vendor_id : [0][0][0][0]
 Stream #0:4[0x5](eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (LFE), s32 (24 bit), 1152 kb/s (default)
 Metadata:
 creation_time : 2022-08-26T22:20:26.000000Z
 handler_name : Apple Sound Media Handler
 vendor_id : [0][0][0][0]
 Stream #0:5[0x6](eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (BL), s32 (24 bit), 1152 kb/s (default)
 Metadata:
 creation_time : 2022-08-26T22:20:26.000000Z
 handler_name : Apple Sound Media Handler
 vendor_id : [0][0][0][0]
 Stream #0:6[0x7](eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (BR), s32 (24 bit), 1152 kb/s (default)
 Metadata:
 creation_time : 2022-08-26T22:20:26.000000Z
 handler_name : Apple Sound Media Handler
 vendor_id : [0][0][0][0]
 Stream #0:7[0x8](eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (DL), s32 (24 bit), 1152 kb/s (default)
 Metadata:
 creation_time : 2022-08-26T22:20:26.000000Z
 handler_name : Apple Sound Media Handler
 vendor_id : [0][0][0][0]
 Stream #0:8[0x9](eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels (DR), s32 (24 bit), 1152 kb/s (default)
 Metadata:
 creation_time : 2022-08-26T22:20:26.000000Z
 handler_name : Apple Sound Media Handler
 vendor_id : [0][0][0][0]



As you can see, these audio streams are single channel as each stream is mapping to a unique channel.


I want to combine these audio streams into a aac file such that the output has the channel information retained.


I use the following command to do this :


ffmpeg -i -filter _complex "[0:a:0][0:a:1][0:a:2][0:a:3][0:a:4][0:a:5][0:a:6][0:a:7]join=inputs=8:channel_layout=7.1[aout]" -map "[aout]" -c:a aac output_audio_join.aac


The individual audio streams were having a duration of duration=103.728625 seconds.


The output however is 862.207660 seconds. Why ?