08:52
Why is it that when I create a livestream in Python using ffmpeg, and then I open the browser and visit the page, the page keeps loading continuously, and in PyCharm logs, I see binary data? There are no errors displayed, and the code seems correct to me. I even tried saving to a file for testing purposes, and when I play the video, everything works fine. Does anyone know what might be wrong here?
Code:
def generate_frames():
cap = cv2.VideoCapture(os.path.normpath(app_root_dir().joinpath("data/temp", "video-979257305707693982.mp4")))
while cap.isOpened():
ret, (...)
07:05
I'm trying to set the minimum, maximum, and average bitrate or vbv delay for a WebM livestream in Python using FFMPEG, but it still shows "0" or "N/A". I've checked the documentation, and everything seems correct. I've also searched for solutions online, but none of them have solved my problem. Does anyone know what I might be doing wrong and how it should be done correctly?
FFMPEG Command:
ffmpeg_command = [
'ffmpeg', '-f', 'rawvideo', '-pix_fmt', 'bgr24',
'-s:v', '1920x1080', (...)
20:05
I am facing a weird error in FFMPG while doing any operation of a particular video. I did a lot of R&D but did not found any solution regarding this. Can anyone of you please look into it, it will be so kind.
here is a command i am using to mute the video, but end up with the following error.
fun videoMuteCmd(
filePath: Uri,
outputPath: String,
): Array
return arrayOf(
"-y", "-i", filePath.toString(),
"-c", "copy", "-an", outputPath
)
ERROR
: LogMessageexecutionId=3010, (...)
16:34
I am facing a weird error in FFMPG while doing any operation of a particular video. I did a lot of R&D but did not found any solution regarding this. Can anyone of you please look into it, it will be so kind.
here is a command i am using to mute the video, but end up with the following error.
fun videoMuteCmd(
filePath: Uri,
outputPath: String,
): Array
return arrayOf(
"-y", "-i", filePath.toString(),
"-c", "copy", "-an", outputPath
)
ERROR
: LogMessageexecutionId=3010, (...)
09:35
maybe someone can help me. I just installed ffmpeg on my Windows 10 system. I downloaded the latest release from the gyan.dev website, and i already tried different builds too, but every time I try to download an m3u8 live stream, it gives me these errors:
[tls ⓐ 00000237a151f580] Error in the pull function.
[tls ⓐ 00000237a151f580] IO error: End of file
[in#0 ⓐ 00000237a151ed00] Error opening input: End of file
Error opening input file (...)
13:25
I am trying to convert a .wav audio file generated from a flutter's text to speech package - "flutter_tts" to mp3 file but it is failing everytime.
I have written the below code for file conversion. I have imported the package ffmpeg_kit_flutter. It doesnt even show why the conversion is failing.
I have looked up in stackoverflow and other sites but could not find any relevant solutions. I am using vscode as editor. I have attached flutter doctor output below as well. Could anyone please guide me? Let me know if you need more information.
List command = [
(...)
06:48
I have a buffer containing audio data representing a voice recording, and I need to determine its loudness in Node.js. I tried using the fluent-ffmpeg library, as it seemed to offer functionality for audio analysis. However, my attempts to use it to analyze the loudness of the audio buffers were unsuccessful.
Could someone please suggest a reliable approach or provide a code example to help me analyze the loudness of the audio buffers accurately in Node.js? Any guidance or assistance would be greatly (...)
04:15
Trying to use hardware encoding with ffmpeg (version 7.0-essentials_build-www.gyan.dev) on Windows 10
Command
ffmpeg -i -v verbose -c:v h264_amf -acodec copy -y
Error
[h264_amf ⓐ 0000014dc3388580] AMF initialisation succeeded via D3D11.
[h264_amf ⓐ 0000014dc3388900] CreateComponent(AMFVideoEncoderVCE_AVC) failed with error 36
[vost#0:0/h264_amf ⓐ 0000014dc3bf7c80] Error while opening encoder - maybe incorrect parameters such as bit_rate, rate, width or height.
How to resolve this (...)
03:01
I am trying to make an audio file be exactly x second.
So far i tried using the atempo filter by doing the following calculation
Audio length / desired length = atempo.
But this is not accurate, and I am having to tweak the tempo manually to get it to an exact fit.
Are there any other solutions to get this work ? Or am I doing this incorrectly?
My original file is a wav file, and my output in an mp3
Here is a sample command
ffmpeg -i input.wav -codec:a libmp3lame -filter:a "atempo=0.9992323" -b:a 320K output.mp3
UPDATE:
I was able to (...)
00:04
I have ffmpeg setup which produces rtmp stream from remote rtsp stream. The rtsp stream comes from ip camera which support multiple profiles. Each profile has 1080p 30 fps, 1080p 15 fps.
The weird thing is that when ffmpeg analyze its input stream, one profile is recognized as 1 fps as below
Input #0, rtsp, from 'rtsp://...':
Metadata:
title : Media Presentation
comment : samsung
Duration: N/A, start: 0.064144, bitrate: N/A
Stream #0:0: Video: h264 (High), yuvj420p(pc, bt709, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 1 fps, 1 tbr, 90k tbn, 2 (...)