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DJ Dolores - Oslodum 2004 (includes (cc) sample of “Oslodum” by Gilberto Gil)
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
Autres articles (43)
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Les formats acceptés
28 janvier 2010, parLes commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
ffmpeg -codecs ffmpeg -formats
Les format videos acceptés en entrée
Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
Les formats vidéos de sortie possibles
Dans un premier temps on (...) -
Ajouter notes et légendes aux images
7 février 2011, parPour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
Modification lors de l’ajout d’un média
Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...) -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...)
Sur d’autres sites (7023)
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unable to steam rtsp from mp4(h264) file using ffmpeg on os x : Connection refused Could not write header for output file
28 janvier 2023, par TalGim ussing the following command on my macbook os high sierra to stream rtsp from mp4 file using ffmpeg :


sudo ffmpeg -re -i ./Big_Buck_Bunny_1080_10s_1MB.mp4 -c:v libx264 -preset superfast -tune zerolatency -c:a aac -ar 44100 -f rtsp -rtsp_transport udp rtsp://127.0.0.1:8888/live


but get the following error :


[tcp @ 0x7fb979e22ec0] Connection to tcp://127.0.0.1:1935?timeout=0 failed: Connection refused
Could not write header for output file #0 (incorrect codec parameters ?): Connection refused
Error initializing output stream 0:0 -- 
Conversion failed!


here is the whole output of the command :


ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
 built with Apple LLVM version 10.0.0 (clang-1000.11.45.5)
 configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.1 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
 libavutil 56. 51.100 / 56. 51.100
 libavcodec 58. 91.100 / 58. 91.100
 libavformat 58. 45.100 / 58. 45.100
 libavdevice 58. 10.100 / 58. 10.100
 libavfilter 7. 85.100 / 7. 85.100
 libavresample 4. 0. 0 / 4. 0. 0
 libswscale 5. 7.100 / 5. 7.100
 libswresample 3. 7.100 / 3. 7.100
 libpostproc 55. 7.100 / 55. 7.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from './Big_Buck_Bunny_1080_10s_1MB.mp4':
 Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 title : Big Buck Bunny, Sunflower version
 artist : Blender Foundation 2008, Janus Bager Kristensen 2013
 composer : Sacha Goedegebure
 encoder : Lavf57.63.100
 comment : Creative Commons Attribution 3.0 - http://bbb3d.renderfarming.net
 genre : Animation
 Duration: 00:00:10.00, start: 0.000000, bitrate: 815 kb/s
 Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 812 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
 Metadata:
 handler_name : VideoHandler
Stream mapping:
 Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
Press [q] to stop, [?] for help
[libx264 @ 0x7fb97a00de00] using SAR=1/1
[libx264 @ 0x7fb97a00de00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0x7fb97a00de00] profile High, level 4.0, 4:2:0, 8-bit
[libx264 @ 0x7fb97a00de00] 264 - core 160 r3011 cde9a93 - H.264/MPEG-4 AVC codec - Copyleft 2003-2020 - http://www.videolan.org/x264.html - options: cabac=1 ref=1 deblock=1:0:0 analyse=0x3:0x3 me=dia subme=1 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=12 lookahead_threads=12 sliced_threads=1 slices=12 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
[tcp @ 0x7fb979e22ec0] Connection to tcp://127.0.0.1:1935?timeout=0 failed: Connection refused
Could not write header for output file #0 (incorrect codec parameters ?): Connection refused
Error initializing output stream 0:0 -- 
Conversion failed!


tried with and without sudo, tried changing rtsp ://... to http://
also tried udp but get same output..


chacked that the port is not in use(8888) and different ports (1935...) but still the same.


i installed ffmpeg via brew install...


when i run some test server on my localhost i never have issues ussing an unused port


really stuck here and any help would be amazing...thank you


EDIT :
Problem was in the command i used : "rtsp ://127.0.0.1:8888/live" - but i did not have a running server capable of accepting the data from ffmpeg and redestributing it - so i had to first run such server and only after that to run ffmpeg :


Servers which can receive from FFmpeg (to restream to multiple clients) include ffserver (linux only, though with cygwin it might work on windows), or Wowza Media Server, or Flash Media Server, Red5, or various others. Even VLC can pick up the stream from ffmpeg, then redistribute it, acting as a server.


i used the VLC option. You can read about it here : http://trac.ffmpeg.org/wiki/StreamingGuide


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How to simultaneously capture mic, stream it to RTSP server and play it on iPhone's speaker ?
24 août 2021, par Norbert TowiańskiI want to capture sound from mic, stream it to RTSP server and play it simultaneously on iPhone's speaker after getting samples from RTSP server. I mean such kind of loop. I use FFMPEGKit and I want to use MobileVLCKit, but unfortunately microphone is off when I start play stream.
I think I've done first step (capturing from microphone and send OutputStream to RTSP server) :


@IBAction func transmitBtnPressed(_ sender: Any) {
 ffmpeg_transmit()
}

@IBAction func recordBtnPressed(_ sender: Any) {
 switch recordingState {
 case .idle:
 recordingState = .start
 startRecording()
 recordBtn.setTitle("Started", for: .normal)
 let urlToFile = URL(fileURLWithPath: outPipePath!)
 outputStream = OutputStream(url: urlToFile, append: false)
 outputStream!.open()
 case .capturing:
 recordingState = .end
 stopRecording()
 recordBtn.setTitle("End", for: .normal)
 default:
 break
 }
}

override func viewDidLoad() {
 super.viewDidLoad()
 outPipePath = FFmpegKitConfig.registerNewFFmpegPipe()
 self.setup()
}

override func viewDidAppear(_ animated: Bool) {
 super.viewDidAppear(animated)
 setUpAuthStatus()
}

func setUpAuthStatus() {
 if AVCaptureDevice.authorizationStatus(for: AVMediaType.audio) != .authorized {
 AVCaptureDevice.requestAccess(for: AVMediaType.audio, completionHandler: { (authorized) in
 DispatchQueue.main.async {
 if authorized {
 self.setup()
 }
 }
 })
 }
}

func setup() {
 self.session.sessionPreset = AVCaptureSession.Preset.high
 
 self.recordingURL = URL(fileURLWithPath: "\(NSTemporaryDirectory() as String)/file.m4a")
 if self.fileManager.isDeletableFile(atPath: self.recordingURL!.path) {
 _ = try? self.fileManager.removeItem(atPath: self.recordingURL!.path)
 }
 
 self.assetWriter = try? AVAssetWriter(outputURL: self.recordingURL!,
 fileType: AVFileType.m4a)
 self.assetWriter!.movieFragmentInterval = CMTime.invalid
 self.assetWriter!.shouldOptimizeForNetworkUse = true
 
 let audioSettings = [
 AVFormatIDKey: kAudioFormatLinearPCM,
 AVSampleRateKey: 48000.0,
 AVNumberOfChannelsKey: 1,
 AVLinearPCMIsFloatKey: false,
 AVLinearPCMBitDepthKey: 16,
 AVLinearPCMIsBigEndianKey: false,
 AVLinearPCMIsNonInterleaved: false,
 
 ] as [String : Any]
 
 
 self.audioInput = AVAssetWriterInput(mediaType: AVMediaType.audio,
 outputSettings: audioSettings)
 
 self.audioInput?.expectsMediaDataInRealTime = true
 
 if self.assetWriter!.canAdd(self.audioInput!) {
 self.assetWriter?.add(self.audioInput!)
 }
 
 self.session.startRunning()
 
 DispatchQueue.main.async {
 self.session.beginConfiguration()
 
 self.session.commitConfiguration()
 
 let audioDevice = AVCaptureDevice.default(for: AVMediaType.audio)
 let audioIn = try? AVCaptureDeviceInput(device: audioDevice!)
 
 if self.session.canAddInput(audioIn!) {
 self.session.addInput(audioIn!)
 }
 
 if self.session.canAddOutput(self.audioOutput) {
 self.session.addOutput(self.audioOutput)
 }
 
 self.audioConnection = self.audioOutput.connection(with: AVMediaType.audio)
 }
}

func startRecording() {
 if self.assetWriter?.startWriting() != true {
 print("error: \(self.assetWriter?.error.debugDescription ?? "")")
 }
 
 self.audioOutput.setSampleBufferDelegate(self, queue: self.recordingQueue)
}

func stopRecording() {
 self.audioOutput.setSampleBufferDelegate(nil, queue: nil)
 
 self.assetWriter?.finishWriting {
 print("Saved in folder \(self.recordingURL!)")
 }
}
func captureOutput(_ captureOutput: AVCaptureOutput, didOutput
 sampleBuffer: CMSampleBuffer, from connection: AVCaptureConnection) {
 
 if !self.isRecordingSessionStarted {
 let presentationTime = CMSampleBufferGetPresentationTimeStamp(sampleBuffer)
 self.assetWriter?.startSession(atSourceTime: presentationTime)
 self.isRecordingSessionStarted = true
 recordingState = .capturing
 }
 
 var blockBuffer: CMBlockBuffer?
 var audioBufferList: AudioBufferList = AudioBufferList.init()
 
 CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampleBuffer, bufferListSizeNeededOut: nil, bufferListOut: &audioBufferList, bufferListSize: MemoryLayout<audiobufferlist>.size, blockBufferAllocator: nil, blockBufferMemoryAllocator: nil, flags: kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment, blockBufferOut: &blockBuffer)
 let buffers = UnsafeMutableAudioBufferListPointer(&audioBufferList)
 
 for buffer in buffers {
 let u8ptr = buffer.mData!.assumingMemoryBound(to: UInt8.self)
 let output = outputStream!.write(u8ptr, maxLength: Int(buffer.mDataByteSize))
 
 if (output == -1) {
 let error = outputStream?.streamError
 print("\(#file) > \(#function) > Error on outputStream: \(error!.localizedDescription)")
 }
 else {
 print("\(#file) > \(#function) > Data sent")
 }
 }
}

func ffmpeg_transmit() {
 
 let cmd1: String = "-f s16le -ar 48000 -ac 1 -i "
 let cmd2: String = " -probesize 32 -analyzeduration 0 -c:a libopus -application lowdelay -ac 1 -ar 48000 -f rtsp -rtsp_transport udp rtsp://localhost:18556/mystream"
 let cmd = cmd1 + outPipePath! + cmd2
 
 print(cmd)
 
 ffmpegSession = FFmpegKit.executeAsync(cmd, withExecuteCallback: { ffmpegSession in
 
 let state = ffmpegSession?.getState()
 let returnCode = ffmpegSession?.getReturnCode()
 if let returnCode = returnCode, let get = ffmpegSession?.getFailStackTrace() {
 print("FFmpeg process exited with state \(String(describing: FFmpegKitConfig.sessionState(toString: state!))) and rc \(returnCode).\(get)")
 }
 }, withLogCallback: { log in
 
 }, withStatisticsCallback: { statistics in
 
 })
}
</audiobufferlist>

I want to use MobileVLCKit in that way :


func startStream(){
 guard let url = URL(string: "rtsp://localhost:18556/mystream") else {return}
 audioPlayer!.media = VLCMedia(url: url)

 audioPlayer!.media.addOption( "-vv")
 audioPlayer!.media.addOption( "--network-caching=10000")

 audioPlayer!.delegate = self
 audioPlayer!.audio.volume = 100

 audioPlayer!.play()

}


Could you give me some hints how to implement that ?


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Why does File upload for moving image and Audio to tmp PHP folder work on Windows but only image upload portion works on Mac using MAMP ?
31 mai 2021, par YazdanSo according to my colleague who tested this on Windows says it works perfectly fine , but in my case when I use it on a Mac with MAMP for Moodle , the image files get uploaded to the correct destination folder without an issue whereas the audio files don't move from the tmp folder to the actual destination folder and to check if this was the case ... I just changed and gave a fixed path instead of
$fileTmpLocand the file made it to the correct destination. Sorry I know the first half of the code isn't the main issue but I still wanted to post the whole code so one could understand it easily, moreover I am just beginning to code so please "have a bit of patience with me" . Thanks in advance


// this file contains upload function 
// checks if the file exists in server
include("../db/database.php");
require_once(dirname(__FILE__) . '/../../../config.php');
global $IP;

$ajaxdata = $_POST['mediaUpload'];

$FILENAME = $ajaxdata[1];
$IMAGE=$ajaxdata[0];
// an array to check which category the media belongs too
$animal= array("bird","cat","dog","horse","sheep","cow","elephant","bear","giraffe","zebra");
$allowedExts = array("mp3","wav");
$temp = explode(".", $_FILES["audio"]["name"]);
$extension = end($temp);



$test = $_FILES["audio"]["type"]; 


if (
 $_FILES["audio"]["type"] == "audio/wav"||
 $_FILES["audio"]["type"] == "audio/mp3"||
 $_FILES["audio"]["type"] == "audio/mpeg"
 &&
 in_array($extension, $allowedExts)
 )
 {

 // if the name detected by object detection is present in the animal array
 // then initialize target path to animal database or to others
 if (in_array($FILENAME, $animal)) 
 { 
 $image_target_dir = "image_dir/";
 $audio_target_dir = "audio_dir/";
 } 
 else
 { 
 $image_target_dir = "other_image_dir/";
 $audio_target_dir = "other_audio_dir/";
 } 
 // Get file path
 
 $img = $IMAGE;
 // decode base64 image
 $img = str_replace('data:image/png;base64,', '', $img);
 $img = str_replace(' ', '+', $img);
 $image_data = base64_decode($img);

 //$extension = pathinfo( $_FILES["fileUpload"]["name"], PATHINFO_EXTENSION ); // jpg
 $image_extension = "png";
 $image_target_file =$image_target_dir . basename($FILENAME . "." . $image_extension);
 $image_file_upload = "http://localhost:8888/moodle310/blocks/testblock/classes/".$image_target_file;
 
 
 $audio_extension ="mp3";
 $audio_target_file= $audio_target_dir . basename($FILENAME. "." . $audio_extension) ;
 $audio_file_upload = "http://localhost:8888/moodle310/blocks/testblock/classes/".$audio_target_file;

 // file size limit
 if(($_FILES["audio"]["size"])<=51242880)
 {

 $fileName = $_FILES["audio"]["name"]; // The file name
 $fileTmpLoc = $_FILES["audio"]["tmp_name"]; // File in the PHP tmp folder
 $fileType = $_FILES["audio"]["type"]; // The type of file it is
 $fileSize = $_FILES["audio"]["size"]; // File size in bytes
 $fileErrorMsg = $_FILES["audio"]["error"]; // 0 for false... and 1 for true
 
 if (in_array($FILENAME, $animal)) 
 { 
 $sql = "INSERT INTO mdl_media_animal (animal_image_path,animal_name,animal_audio_path) VALUES ('$image_file_upload','$FILENAME','$audio_file_upload')";
 } else {
 $sql = "INSERT INTO mdl_media_others (others_image_path,others_name,others_audio_path) VALUES ('$image_file_upload','$FILENAME','$audio_file_upload')";
 }

 // if file exists
 if (file_exists($audio_target_file) || file_exists($image_target_file)) {
 echo "alert";
 } else {
 // write image file
 if (file_put_contents($image_target_file, $image_data) ) {
 // ffmpeg to write audio file
 $output = shell_exec("ffmpeg -i $fileTmpLoc -ab 160k -ac 2 -ar 44100 -vn $audio_target_file");
 echo $output;
 
 // $stmt = $conn->prepare($sql);
 $db = mysqli_connect("localhost", "root", "root", "moodle310"); 
 // echo $sql;
 if (!$db) {
 echo "nodb";
 die("Connection failed: " . mysqli_connect_error());
 }
 // echo"sucess";
 if(mysqli_query($db, $sql)){
 // if($stmt->execute()){
 echo $fileTmpLoc;
 echo "sucess"; 
 echo $output;
 }
 else {
 // echo "Error: " . $sql . "<br />" . mysqli_error($conn);
 echo "failed";
 }

 }else {
 echo "failed";
 }

 
 
 
 }
 
 // $test = "ffmpeg -i $outputfile -ab 160k -ac 2 -ar 44100 -vn bub.wav";
 } else
 {
 echo "File size exceeds 5 MB! Please try again!";
 }
}
else
{
 echo "PHP! Not a video! ";//.$extension." ".$_FILES["uploadimage"]["type"];
 }

?>


I am a student learning frontend but a project of mine requires a fair bit of backend. So forgive me if my question sounds silly.


What I meant by manually overriding it was creating another folder and a index.php file with
echo "hello"; $output = shell_exec("ffmpeg -i Elephant.mp3 -ab 160k -ac 2 -ar 44100 -vn bub.mp3"); echo $output;so only yes in this caseElephant.mp3was changed as the initial tmp path so in this case as suggested by Mr.CBroe the permissons shouldn't be an issue.

Okay I checked my
Apache_error.logonly to find out ffmpeg is indeed the culprit ... I had installedffmpegglobally so I am not sure if it is an access problem but here is a snippet of the log

I checked my php logs and found out that
FFmpegis the culprit.
Attached is a short log file

[Mon May 31 18:11:33 2021] [notice] caught SIGTERM, shutting down
[Mon May 31 18:11:40 2021] [notice] Digest: generating secret for digest authentication ...
[Mon May 31 18:11:40 2021] [notice] Digest: done
[Mon May 31 18:11:40 2021] [notice] Apache/2.2.34 (Unix) mod_ssl/2.2.34 OpenSSL/1.0.2o PHP/7.2.10 configured -- resuming normal operations
sh: ffmpeg: command not found
sh: ffmpeg: command not found
sh: ffmpeg: command not found


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