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Sur d’autres sites (11855)
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I tried to play the audio on Alexa skill from my S3 Bucket, from the test tab, **it show but in fact, I can't hear any sound
19 avril 2022, par Siti MaynaSo I tried to play the audio on Alexa skill from my S3 Bucket, from the test tab, it show but in fact, I can't hear any sound. Another fact is, that I tried to use the sample audio from https://developer.amazon.com/en-US/docs/alexa/custom-skills/ask-soundlibrary.html and it is worked, but why it won't work when it comes from my own S3 Bucket ?


Notes :


I've tried to test the skill using my mobile phone also.


I've tried to encode the audio using FFmpeg.


I've tried to use Jovo to convert the audio. https://v3.jovo.tech/audio-converter


I don't know how to fix this error.


There is no error message on cloud watch.


Assumptions :
There is some problem related to the audio resources or there is more set to play audio from S3 Bucket since the sample audio is working.


Steps to reproduce :




Build the interaction model






Encode the audio to make it Alexa skill friendly (fulfill the requirements, like sample rate, etc), I used and tried all of these :




A :


ffmpeg -i -ac 2 -codec:a libmp3lame -b:a 48k -ar 16000 -write_xing 0 


B :


ffmpeg -i -ac 2 -codec:a libmp3lame -b:a 48k -ar 24000 -write_xing 0 


C :


ffmpeg -y -i input.mp3 -ar 16000 -ab 48k -codec:a libmp3lame -ac 1 output.mp3




Upload the audio resources on S3Bucket
Audio sample on s3 storage but none of them are produce any sounds






Use the link and insert it to APLA.json





 {
 "type": "APLA",
 "version": "0.91",
 "description": "Simple document that generates speech",
 "mainTemplate": {
 "parameters": [
 "payload"
 ],
 "type": "Sequencer",
 "items": [
 {
 "type": "Audio",
 "source": "https://72578561-d9d8-47b4-811c-cafbcbc5ddb9-us-east-1.s3.amazonaws.com/Media/one-small-step-alexa-24.mp3"
 }
 ]
 }
 }



notes : I change the link sources based on audio that I tried.




the intent on lambda_function.py :




def _load_apl_document(file_path):
 # type: (str) -> Dict[str, Any]
 """Load the apl json document at the path into a dict object."""
 with open(file_path) as f:
 return json.load(f)

class LaunchRequestHandler(AbstractRequestHandler):
 """Handler for Skill Launch."""
 def can_handle(self, handler_input):
 # type: (HandlerInput) -> bool

 return ask_utils.is_request_type("LaunchRequest")(handler_input)

 def handle(self, handler_input):
 # type: (HandlerInput) -> Response
 logger.info("In LaunchRequestHandler")

 # type: (HandlerInput) -> Response
 speak_output = "Hello World!"
 # .ask("add a reprompt if you want to keep the session open for the user to respond")

 return (
 handler_input.response_builder
 #.speak(speak_output)
 .add_directive(
 RenderDocumentDirective(
 token="pagerToken",
 document=_load_apl_document("APLA.json"),
 datasources={}
 )
 )
 .response
 )




Deploy






Test it






The result of the test on my end :

The response for testing




the JSON response :


{
 "body": {
 "version": "1.0",
 "response": {
 "directives": [
 {
 "type": "Alexa.Presentation.APLA.RenderDocument",
 "token": "pagerToken",
 "document": {
 "type": "APLA",
 "version": "0.91",
 "description": "Simple document that generates speech",
 "mainTemplate": {
 "parameters": [
 "payload"
 ],
 "type": "Sequencer",
 "items": [
 {
 "type": "Audio",
 "source": "https://72578561-d9d8-47b4-811c-cafbcbc5ddb9-us-east-1.s3.amazonaws.com/Media/one-small-step-alexa-24.mp3"
 }
 ]
 }
 },
 "datasources": {}
 }
 ],
 "type": "_DEFAULT_RESPONSE"
 },
 "sessionAttributes": {},
 "userAgent": "ask-python/1.16.1 Python/3.7.12"
 }
}




On my cloud Watch :
Cloud Watch




-
JSmpeg is not playing audio from websocket stream
5 juin 2023, par NikI am trying to stream RTSP to web browser using ffmpeg through web socket relay written in node js taken from https://github.com/phoboslab/jsmpeg , and on the browser i am using JSMpeg to display the RTSP stream, the video is playing fine, but audio is not playing,


The ffmpeg command :


ffmpeg -rtsp_transport tcp -i rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mp4 
 -f mpegts -c:v mpeg1video -c:a mp2 http://127.0.0.1:8081/stream_from_ffmpeg/


The node js web socket relay :


// Use the websocket-relay to serve a raw MPEG-TS over WebSockets. You can use
// ffmpeg to feed the relay. ffmpeg -> websocket-relay -> browser
// Example:
// node websocket-relay yoursecret 8081 8082
// ffmpeg -i <some input="input"> -f mpegts http://localhost:8081/yoursecret

var fs = require('fs'),
 http = require('http'),
 WebSocket = require('ws');

if (process.argv.length < 3) {
 console.log(
 'Usage: \n' +
 'node websocket-relay.js <secret> [ ]'
 );
 process.exit();
}

var STREAM_SECRET = process.argv[2],
 STREAM_PORT = process.argv[3] || 8081,
 WEBSOCKET_PORT = process.argv[4] || 8082,
 RECORD_STREAM = false;

// Websocket Server
var socketServer = new WebSocket.Server({port: WEBSOCKET_PORT, perMessageDeflate: false});
socketServer.connectionCount = 0;
socketServer.on('connection', function(socket, upgradeReq) {
 socketServer.connectionCount++;
 console.log(
 'New WebSocket Connection: ',
 (upgradeReq || socket.upgradeReq).socket.remoteAddress,
 (upgradeReq || socket.upgradeReq).headers['user-agent'],
 '('+socketServer.connectionCount+' total)'
 );
 socket.on('close', function(code, message){
 socketServer.connectionCount--;
 console.log(
 'Disconnected WebSocket ('+socketServer.connectionCount+' total)'
 );
 });
});
socketServer.broadcast = function(data) {
 socketServer.clients.forEach(function each(client) {
 if (client.readyState === WebSocket.OPEN) {
 client.send(data);
 }
 });
};

// HTTP Server to accept incoming MPEG-TS Stream from ffmpeg
var streamServer = http.createServer( function(request, response) {
 var params = request.url.substr(1).split('/');

 if (params[0] !== STREAM_SECRET) {
 console.log(
 'Failed Stream Connection: '+ request.socket.remoteAddress + ':' +
 request.socket.remotePort + ' - wrong secret.'
 );
 response.end();
 }

 response.connection.setTimeout(0);
 console.log(
 'Stream Connected: ' +
 request.socket.remoteAddress + ':' +
 request.socket.remotePort
 );
 request.on('data', function(data){
 socketServer.broadcast(data);
 if (request.socket.recording) {
 request.socket.recording.write(data);
 }
 });
 request.on('end',function(){
 console.log('close');
 if (request.socket.recording) {
 request.socket.recording.close();
 }
 });

 // Record the stream to a local file?
 if (RECORD_STREAM) {
 var path = 'recordings/' + Date.now() + '.ts';
 request.socket.recording = fs.createWriteStream(path);
 }
})
// Keep the socket open for streaming
streamServer.headersTimeout = 0;
streamServer.listen(STREAM_PORT);

console.log('Listening for incoming MPEG-TS Stream on http://127.0.0.1:'+STREAM_PORT+'/<secret>');
console.log('Awaiting WebSocket connections on ws://127.0.0.1:'+WEBSOCKET_PORT+'/');
</secret></secret></some>

The front end code




 
 
 
 
 <code class="echappe-js"><script src='http://stackoverflow.com/feeds/tag/jsmpeg.min.js'></script>
 
 
 
 
 
<script>&#xA; let url;&#xA; let player;&#xA; let canvas = document.getElementById("video-canvas");&#xA; let ipAddr = "127.0.0.1:8082";&#xA; window.onload = async() => {&#xA; url = `ws://${ipAddr}`;&#xA; player = new JSMpeg.Player(url, { canvas: canvas, });&#xA; };&#xA;&#xA; </script>




The above code works fine and plays the video, but no audio is playing
Things I tried :


Changed the audio context state inside the player object from suspended to running


player.audioOut.context.onstatechange = async () => {
 console.log("Event triggered by audio");

 if (player.audioOut.context === "suspended") {
 await player.audioOut.context.resume();
 }
}


-
libavcodec : how to encode with h264 codec ,with mp4 container using controllable frame rate and bitrate(through c code)
26 mai 2016, par musimbateI am trying to record the screen of a pc and encode the recorded frames using h264 encoder
and wrap them into a mp4 container.I want to do this because this super user link http://superuser.com/questions/300897/what-is-a-codec-e-g-divx-and-how-does-it-differ-from-a-file-format-e-g-mp/300997#300997 suggests it allows good trade-off between size and quality of the output file.The application I am working on should allow users to record a few hours of video and have the minimum output file size with decent quality.
The code I have cooked up so far allows me to record and save .mpg(container) files with the mpeg1video encoder
Running :
ffmpeg -i test.mpg
on the output file gives the following output :
[mpegvideo @ 028c7400] Estimating duration from bitrate, this may be inaccurate
Input #0, mpegvideo, from 'test.mpg':
Duration: 00:00:00.29, bitrate: 104857 kb/s
Stream #0:0: Video: mpeg1video, yuv420p(tv), 1366x768 [SAR 1:1 DAR 683:384], 104857 kb/s, 25 fps, 25 tbr, 1200k tbn, 25 tbcI have these settings for my output :
const char * filename="test.mpg";
int codec_id= AV_CODEC_ID_MPEG1VIDEO;
AVCodec *codec11;
AVCodecContext *outContext= NULL;
int got_output;
FILE *f;
AVPacket pkt;
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
/* put sample parameters */
outContext->bit_rate = 400000;
/* resolution must be a multiple of two */
outContext->width=pCodecCtx->width;
outContext->height=pCodecCtx->height;
/* frames per second */
outContext->time_base.num=1;
outContext->time_base.den=25;
/* emit one intra frame every ten frames
* check frame pict_type before passing frame
* to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
* then gop_size is ignored and the output of encoder
* will always be I frame irrespective to gop_size
*/
outContext->gop_size = 10;
outContext->max_b_frames = 1;
outContext->pix_fmt = AV_PIX_FMT_YUV420P;When I change int codec_id= AV_CODEC_ID_MPEG1VIDEO to int codec_id= AV_CODEC_ID_H264 i get a file that does not play with vlc.
I have read that writing the
uint8_t endcode[] = { 0, 0, 1, 0xb7 };array at the end of your file when finished encoding makes your file a legitimate mpeg file.It is written like this :
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);in my code. Should I do the same thing when I change my encoder to AV_CODEC_ID_H264 ?
I am capturing using gdi input like this :
AVDictionary* options = NULL;
//Set some options
//grabbing frame rate
av_dict_set(&options,"framerate","30",0);
AVInputFormat *ifmt=av_find_input_format("gdigrab");
if(avformat_open_input(&pFormatCtx,"desktop",ifmt,&options)!=0){
printf("Couldn't open input stream.\n");
return -1;
}I want to be able to modify my grabbing rate to optimize for the outptut file size
but When I change it to 20 for example I get a video that plays so fast.How do
I get a video that plays with normal speed with frames captured at 20 fps or any
lower frame rate value ?While recording I get the following output on the standard error output :
[gdigrab @ 00cdb8e0] Capturing whole desktop as 1366x768x32 at (0,0)
Input #0, gdigrab, from '(null)':
Duration: N/A, start: 1420718663.655713, bitrate: 1006131 kb/s
Stream #0:0: Video: bmp, bgra, 1366x768, 1006131 kb/s, 29.97 tbr, 1000k tbn, 29.97 tbc
[swscaler @ 00d24120] Warning: data is not aligned! This can lead to a speedloss
[mpeg1video @ 00cdd160] AVFrame.format is not set
[mpeg1video @ 00cdd160] AVFrame.width or height is not set
[mpeg1video @ 00cdd160] AVFrame.format is not set
[mpeg1video @ 00cdd160] AVFrame.width or height is not set
[mpeg1video @ 00cdd160] AVFrame.format is not setHow do I get rid of this error in my code ?
In summary :
1) How do I encode h264 video wrapped into mp4 container ?2) How do I capture at lower frame rates and still play
the encoded video at normal speed ?3) How do I set the format(and which format—depends on the codec ?)
and width and height info on the frames I write ?The code I am using in its entirety is shown below
extern "C"
{
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libswscale/swscale.h"
#include "libavdevice/avdevice.h"
#include <libavutil></libavutil>opt.h>
#include <libavutil></libavutil>channel_layout.h>
#include <libavutil></libavutil>common.h>
#include <libavutil></libavutil>imgutils.h>
#include <libavutil></libavutil>mathematics.h>
#include <libavutil></libavutil>samplefmt.h>
//SDL
#include "SDL.h"
#include "SDL_thread.h"
}
//Output YUV420P
#define OUTPUT_YUV420P 0
//'1' Use Dshow
//'0' Use GDIgrab
#define USE_DSHOW 0
int main(int argc, char* argv[])
{
//1.WE HAVE THE FORMAT CONTEXT
//THIS IS FROM THE DESKTOP GRAB STREAM.
AVFormatContext *pFormatCtx;
int i, videoindex;
AVCodecContext *pCodecCtx;
AVCodec *pCodec;
av_register_all();
avformat_network_init();
//ASSIGN STH TO THE FORMAT CONTEXT.
pFormatCtx = avformat_alloc_context();
//Register Device
avdevice_register_all();
//Windows
#ifdef _WIN32
#if USE_DSHOW
//Use dshow
//
//Need to Install screen-capture-recorder
//screen-capture-recorder
//Website: http://sourceforge.net/projects/screencapturer/
//
AVInputFormat *ifmt=av_find_input_format("dshow");
//if(avformat_open_input(&pFormatCtx,"video=screen-capture-recorder",ifmt,NULL)!=0){
if(avformat_open_input(&pFormatCtx,"video=UScreenCapture",ifmt,NULL)!=0){
printf("Couldn't open input stream.\n");
return -1;
}
#else
//Use gdigrab
AVDictionary* options = NULL;
//Set some options
//grabbing frame rate
av_dict_set(&options,"framerate","30",0);
//The distance from the left edge of the screen or desktop
//av_dict_set(&options,"offset_x","20",0);
//The distance from the top edge of the screen or desktop
//av_dict_set(&options,"offset_y","40",0);
//Video frame size. The default is to capture the full screen
//av_dict_set(&options,"video_size","640x480",0);
AVInputFormat *ifmt=av_find_input_format("gdigrab");
if(avformat_open_input(&pFormatCtx,"desktop",ifmt,&options)!=0){
printf("Couldn't open input stream.\n");
return -1;
}
#endif
#endif//FOR THE WIN32 THING.
if(avformat_find_stream_info(pFormatCtx,NULL)<0)
{
printf("Couldn't find stream information.\n");
return -1;
}
videoindex=-1;
for(i=0; inb_streams; i++)
if(pFormatCtx->streams[i]->codec->codec_type
==AVMEDIA_TYPE_VIDEO)
{
videoindex=i;
break;
}
if(videoindex==-1)
{
printf("Didn't find a video stream.\n");
return -1;
}
pCodecCtx=pFormatCtx->streams[videoindex]->codec;
pCodec=avcodec_find_decoder(pCodecCtx->codec_id);
if(pCodec==NULL)
{
printf("Codec not found.\n");
return -1;
}
if(avcodec_open2(pCodecCtx, pCodec,NULL)<0)
{
printf("Could not open codec.\n");
return -1;
}
//THIS IS WHERE YOU CONTROL THE FORMAT(THROUGH FRAMES).
AVFrame *pFrame;
pFrame=av_frame_alloc();
int ret, got_picture;
AVPacket *packet=(AVPacket *)av_malloc(sizeof(AVPacket));
//TRY TO INIT THE PACKET HERE
av_init_packet(packet);
//Output Information-----------------------------
printf("File Information---------------------\n");
av_dump_format(pFormatCtx,0,NULL,0);
printf("-------------------------------------------------\n");
//<<--FOR WRITING MPG FILES
//<<--START:PREPARE TO WRITE YOUR MPG FILE.
const char * filename="test.mpg";
int codec_id= AV_CODEC_ID_MPEG1VIDEO;
AVCodec *codec11;
AVCodecContext *outContext= NULL;
int got_output;
FILE *f;
AVPacket pkt;
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
printf("Encode video file %s\n", filename);
/* find the mpeg1 video encoder */
codec11 = avcodec_find_encoder((AVCodecID)codec_id);
if (!codec11) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
outContext = avcodec_alloc_context3(codec11);
if (!outContext) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
/* put sample parameters */
outContext->bit_rate = 400000;
/* resolution must be a multiple of two */
outContext->width=pCodecCtx->width;
outContext->height=pCodecCtx->height;
/* frames per second */
outContext->time_base.num=1;
outContext->time_base.den=25;
/* emit one intra frame every ten frames
* check frame pict_type before passing frame
* to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
* then gop_size is ignored and the output of encoder
* will always be I frame irrespective to gop_size
*/
outContext->gop_size = 10;
outContext->max_b_frames = 1;
outContext->pix_fmt = AV_PIX_FMT_YUV420P;
if (codec_id == AV_CODEC_ID_H264)
av_opt_set(outContext->priv_data, "preset", "slow", 0);
/* open it */
if (avcodec_open2(outContext, codec11, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
AVFrame *outframe = av_frame_alloc();
int nbytes = avpicture_get_size(outContext->pix_fmt,
outContext->width,
outContext->height);
uint8_t* outbuffer = (uint8_t*)av_malloc(nbytes);
//ASSOCIATE THE FRAME TO THE ALLOCATED BUFFER.
avpicture_fill((AVPicture*)outframe, outbuffer,
AV_PIX_FMT_YUV420P,
outContext->width, outContext->height);
SwsContext* swsCtx_ ;
swsCtx_= sws_getContext(pCodecCtx->width,
pCodecCtx->height,
pCodecCtx->pix_fmt,
outContext->width, outContext->height,
outContext->pix_fmt,
SWS_BICUBIC, NULL, NULL, NULL);
//HERE WE START PULLING PACKETS FROM THE SPECIFIED FORMAT CONTEXT.
while(av_read_frame(pFormatCtx, packet)>=0)
{
if(packet->stream_index==videoindex)
{
ret= avcodec_decode_video2(pCodecCtx,
pFrame,
&got_picture,packet );
if(ret < 0)
{
printf("Decode Error.\n");
return -1;
}
if(got_picture)
{
sws_scale(swsCtx_, pFrame->data, pFrame->linesize,
0, pCodecCtx->height, outframe->data,
outframe->linesize);
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
ret = avcodec_encode_video2(outContext, &pkt, outframe, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
}
av_free_packet(packet);
}//THE LOOP TO PULL PACKETS FROM THE FORMAT CONTEXT ENDS HERE.
//
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
//fflush(stdout);
ret = avcodec_encode_video2(outContext, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
/* add sequence end code to have a real mpeg file */
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);
avcodec_close(outContext);
av_free(outContext);
//av_freep(&frame->data[0]);
//av_frame_free(&frame);
//THIS WAS ADDED LATER
av_free(outbuffer);
avcodec_close(pCodecCtx);
avformat_close_input(&pFormatCtx);
return 0;
}Thank you for your time.
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