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Autres articles (20)
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Qualité du média après traitement
21 juin 2013, parLe bon réglage du logiciel qui traite les média est important pour un équilibre entre les partis ( bande passante de l’hébergeur, qualité du média pour le rédacteur et le visiteur, accessibilité pour le visiteur ). Comment régler la qualité de son média ?
Plus la qualité du média est importante, plus la bande passante sera utilisée. Le visiteur avec une connexion internet à petit débit devra attendre plus longtemps. Inversement plus, la qualité du média est pauvre et donc le média devient dégradé voire (...) -
Websites made with MediaSPIP
2 mai 2011, parThis page lists some websites based on MediaSPIP.
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Creating farms of unique websites
13 avril 2011, parMediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...)
Sur d’autres sites (6542)
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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;
}



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ffmpeg file conversion AWS Lambda
10 avril 2021, par eartoolboxI 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')
 }
 



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ffmpeg file conversion AWS Lamda
10 avril 2021, par eartoolboxI 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')
 }