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  • MediaSPIP Core : La Configuration

    9 novembre 2010, par

    MediaSPIP Core fournit par défaut trois pages différentes de configuration (ces pages utilisent le plugin de configuration CFG pour fonctionner) : une page spécifique à la configuration générale du squelettes ; une page spécifique à la configuration de la page d’accueil du site ; une page spécifique à la configuration des secteurs ;
    Il fournit également une page supplémentaire qui n’apparait que lorsque certains plugins sont activés permettant de contrôler l’affichage et les fonctionnalités spécifiques (...)

  • MediaSPIP Player : problèmes potentiels

    22 février 2011, par

    Le lecteur ne fonctionne pas sur Internet Explorer
    Sur Internet Explorer (8 et 7 au moins), le plugin utilise le lecteur Flash flowplayer pour lire vidéos et son. Si le lecteur ne semble pas fonctionner, cela peut venir de la configuration du mod_deflate d’Apache.
    Si dans la configuration de ce module Apache vous avez une ligne qui ressemble à la suivante, essayez de la supprimer ou de la commenter pour voir si le lecteur fonctionne correctement : /** * GeSHi (C) 2004 - 2007 Nigel McNie, (...)

  • Encoding and processing into web-friendly formats

    13 avril 2011, par

    MediaSPIP automatically converts uploaded files to internet-compatible formats.
    Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
    Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
    Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
    All uploaded files are stored online in their original format, so you can (...)

Sur d’autres sites (9122)

  • Encoding audio_common messages to OPUS

    14 juin 2023, par djangbahevans

    


    I am trying to stream microphone and camera data to Amazon KVS WebRTC. I'm able to make video work using this package (adapted for noetic) however I am struggling to make audio work. I'm using the audio_capture package to get mp3 frames. I'm trying to convert this to OPUS frames before streaming to KVS, but I'm unsure how to do this. I wrote this bit of code based on the small resources I can find on using ffmpeg, but it's not working. avcodec_fill_audio_frame is returning -22.

    


    #include "opus_encoder.h"

OPUSEncoder::OPUSEncoder() {
  av_register_all();
  codecContext == nullptr;
}

OPUSEncoder::~OPUSEncoder() {
  if (codecContext != nullptr) {
    avcodec_free_context(&codecContext);
  }
}

int OPUSEncoder::Initialize(int Fs, int channels) {
  AVCodec *codec = avcodec_find_encoder(AV_CODEC_ID_OPUS);
  if (!codec) {
    printf("Codec not found\n");
    return -1;
  }

  codecContext = avcodec_alloc_context3(codec);
  if (!codecContext) {
    printf("Could not allocate audio codec context\n");
    return -1;
  }

  codecContext->sample_fmt = AV_SAMPLE_FMT_S16;
  codecContext->bit_rate = 128000;
  codecContext->sample_rate = Fs;
  codecContext->channel_layout = av_get_default_channel_layout(channels);
  codecContext->channels = channels;

  if (avcodec_open2(codecContext, codec, nullptr) < 0) {
    printf("Could not open codec\n");
    return -1;
  }

  return 0;
}

int OPUSEncoder::Encode(const uint8_t *audio_data, int frameSize,
                        uint8_t *out) {
  AVPacket pkt;
  av_init_packet(&pkt);
  pkt.data = nullptr;
  pkt.size = 0;

  AVFrame *frame = av_frame_alloc();
  frame->nb_samples = frameSize;
  frame->format = codecContext->sample_fmt;
  frame->channel_layout = codecContext->channel_layout;

  int ret = avcodec_fill_audio_frame(frame, codecContext->channels,
                                     codecContext->sample_fmt, audio_data,
                                     frameSize * 2, 0);
  if (ret < 0) {
    printf("Error filling audio frame: %d\n", ret);
    return -1;
  }

  ret = avcodec_send_frame(codecContext, frame);
  if (ret < 0) {
    printf("Error sending the frame to the encoder\n");
    return -1;
  }

  while (ret >= 0) {
    ret = avcodec_receive_packet(codecContext, &pkt);
    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
      return 0;
    } else if (ret < 0) {
      printf("Error encoding audio frame\n");
      return -1;
    }

    memcpy(out, pkt.data, pkt.size);
    out += pkt.size;
    av_packet_unref(&pkt);
  }

  av_frame_free(&frame);

  return 0;
}


    


  • ffmpeg file conversion AWS Lambda

    10 avril 2021, par eartoolbox

    I want a .webm file to be converted to a .wav file after it hits my S3 bucket. I followed this tutorial and tried to adapt it from my use case using the .webm -> .wav ffmpeg command described here.

    


    My AWS Lambda function generally works, in that when my .webm file hits the source bucket, it is converted to .wav and ends up in the destination bucket. However, the resulting file .wav is always 0 bytes (though the .webm not, including the appropriate audio). Did I adapt the code wrong ? I only changed the ffmpeg_cmd line from the first link.

    


    import json
import os
import subprocess
import shlex
import boto3

S3_DESTINATION_BUCKET = "hmtm-out"
SIGNED_URL_TIMEOUT = 60

def lambda_handler(event, context):

    s3_source_bucket = event['Records'][0]['s3']['bucket']['name']
    s3_source_key = event['Records'][0]['s3']['object']['key']

    s3_source_basename = os.path.splitext(os.path.basename(s3_source_key))[0]
    s3_destination_filename = s3_source_basename + ".wav"

    s3_client = boto3.client('s3')
    s3_source_signed_url = s3_client.generate_presigned_url('get_object',
        Params={'Bucket': s3_source_bucket, 'Key': s3_source_key},
        ExpiresIn=SIGNED_URL_TIMEOUT)
    
    ffmpeg_cmd = "/opt/bin/ffmpeg -i \"" + s3_source_signed_url + "\" -c:a pcm_f32le " + s3_destination_filename + " -"
    
    
    command1 = shlex.split(ffmpeg_cmd)
    p1 = subprocess.run(command1, stdout=subprocess.PIPE, stderr=subprocess.PIPE)

    resp = s3_client.put_object(Body=p1.stdout, Bucket=S3_DESTINATION_BUCKET, Key=s3_destination_filename)

    return {
        'statusCode': 200,
        'body': json.dumps('Processing complete successfully')
    }
 


    


  • ffmpeg file conversion AWS Lamda

    10 avril 2021, par eartoolbox

    I want a .webm file to be converted to a .wav file after it hits my S3 bucket. I followed this tutorial and tried to adapt it from my use case using the .webm -> .wav ffmpeg command described here.

    


    My AWS Lambda function generally works, in that when my .webm file hits the source bucket, it is converted to .wav and ends up in the destination bucket. However, the resulting file .wav is always 0 bytes (though the .webm not, including the appropriate audio). Did I adapt the code wrong ? I only changed the ffmpeg_cmd line from the first link.

    


    import json
import os
import subprocess
import shlex
import boto3

S3_DESTINATION_BUCKET = "hmtm-out"
SIGNED_URL_TIMEOUT = 60

def lambda_handler(event, context):

    s3_source_bucket = event['Records'][0]['s3']['bucket']['name']
    s3_source_key = event['Records'][0]['s3']['object']['key']

    s3_source_basename = os.path.splitext(os.path.basename(s3_source_key))[0]
    s3_destination_filename = s3_source_basename + ".wav"

    s3_client = boto3.client('s3')
    s3_source_signed_url = s3_client.generate_presigned_url('get_object',
        Params={'Bucket': s3_source_bucket, 'Key': s3_source_key},
        ExpiresIn=SIGNED_URL_TIMEOUT)
    
    ffmpeg_cmd = "/opt/bin/ffmpeg -i \"" + s3_source_signed_url + "\" -c:a pcm_f32le " + s3_destination_filename + " -"
    
    
    command1 = shlex.split(ffmpeg_cmd)
    p1 = subprocess.run(command1, stdout=subprocess.PIPE, stderr=subprocess.PIPE)

    resp = s3_client.put_object(Body=p1.stdout, Bucket=S3_DESTINATION_BUCKET, Key=s3_destination_filename)

    return {
        'statusCode': 200,
        'body': json.dumps('Processing complete successfully')
    }