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SPIP - plugins - embed code - Exemple
2 septembre 2013, par
Mis à jour : Septembre 2013
Langue : français
Type : Image
Autres articles (69)
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Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir -
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" (...) -
Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs
Sur d’autres sites (11481)
-
Decoding and resampling audio with FFmpeg for output with libao
13 mai 2020, par DoctorSelarI'm trying to write a program to read and play an audio file using FFmpeg and libao. I've been following the procedure outlined in the FFmpeg documentation for decoding audio using the new
avcodec_send_packetandavcodec_receive_framefunctions, but the examples I've been able to find are few and far between (the ones in the FFmpeg documentation either don't use libavformat or use the deprecatedavcodec_decode_audio4). I've based a lot of my program off of the transcode_aac.c example (up toinit_resampler) in the FFmpeg documentation, but that also uses the deprecated decoding function.


I believe I have the decoding part of the program working, but I need to resample the audio in order to convert it into an interleaved format to send to libao, for which I'm attempting to use libswresample. Whenever the program is run in its current state, it outputs (many times) "Error resampling : Output changed". The test file I've been using is just a YouTube rip that I had on hand. ffprobe reports the only stream as :



Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 125 kb/s (default)



This is my first program with FFmpeg (and I'm still relatively new to C), so any advice on how to improve/fix other parts of the program would be welcome.



#include
#include<libavcodec></libavcodec>avcodec.h>
#include<libavformat></libavformat>avformat.h>
#include<libavutil></libavutil>avutil.h>
#include<libswresample></libswresample>swresample.h>
#include<ao></ao>ao.h>

#define OUTPUT_CHANNELS 2
#define OUTPUT_RATE 44100
#define BUFFER_SIZE 192000
#define OUTPUT_BITS 16
#define OUTPUT_FMT AV_SAMPLE_FMT_S16

static char *errtext (int err) {
 static char errbuff[256];
 av_strerror(err,errbuff,sizeof(errbuff));
 return errbuff;
}

static int open_audio_file (const char *filename, AVFormatContext **context, AVCodecContext **codec_context) {
 AVCodecContext *avctx;
 AVCodec *codec;
 int ret;
 int stream_id;
 int i;

 // Open input file
 if ((ret = avformat_open_input(context,filename,NULL,NULL)) < 0) {
 fprintf(stderr,"Error opening input file '%s': %s\n",filename,errtext(ret));
 *context = NULL;
 return ret;
 }

 // Get stream info
 if ((ret = avformat_find_stream_info(*context,NULL)) < 0) {
 fprintf(stderr,"Unable to find stream info: %s\n",errtext(ret));
 avformat_close_input(context);
 return ret;
 }

 // Find the best stream
 if ((stream_id = av_find_best_stream(*context,AVMEDIA_TYPE_AUDIO,-1,-1,&codec,0)) < 0) {
 fprintf(stderr,"Unable to find valid audio stream: %s\n",errtext(stream_id));
 avformat_close_input(context);
 return stream_id;
 }

 // Allocate a decoding context
 if (!(avctx = avcodec_alloc_context3(codec))) {
 fprintf(stderr,"Unable to allocate decoder context\n");
 avformat_close_input(context);
 return AVERROR(ENOMEM);
 }

 // Initialize stream parameters
 if ((ret = avcodec_parameters_to_context(avctx,(*context)->streams[stream_id]->codecpar)) < 0) {
 fprintf(stderr,"Unable to get stream parameters: %s\n",errtext(ret));
 avformat_close_input(context);
 avcodec_free_context(&avctx);
 return ret;
 }

 // Open the decoder
 if ((ret = avcodec_open2(avctx,codec,NULL)) < 0) {
 fprintf(stderr,"Could not open codec: %s\n",errtext(ret));
 avformat_close_input(context);
 avcodec_free_context(&avctx);
 return ret;
 }

 *codec_context = avctx;
 return 0;
}

static void init_packet (AVPacket *packet) {
 av_init_packet(packet);
 packet->data = NULL;
 packet->size = 0;
}

static int init_resampler (AVCodecContext *codec_context, SwrContext **resample_context) {
 int ret;

 // Set resampler options
 *resample_context = swr_alloc_set_opts(NULL,
 av_get_default_channel_layout(OUTPUT_CHANNELS),
 OUTPUT_FMT,
 codec_context->sample_rate,
 av_get_default_channel_layout(codec_context->channels),
 codec_context->sample_fmt,
 codec_context->sample_rate,
 0,NULL);
 if (!(*resample_context)) {
 fprintf(stderr,"Unable to allocate resampler context\n");
 return AVERROR(ENOMEM);
 }

 // Open the resampler
 if ((ret = swr_init(*resample_context)) < 0) {
 fprintf(stderr,"Unable to open resampler context: %s\n",errtext(ret));
 swr_free(resample_context);
 return ret;
 }

 return 0;
}

static int init_frame (AVFrame **frame) {
 if (!(*frame = av_frame_alloc())) {
 fprintf(stderr,"Could not allocate frame\n");
 return AVERROR(ENOMEM);
 }
 return 0;
}

int main (int argc, char *argv[]) {
 AVFormatContext *context = 0;
 AVCodecContext *codec_context;
 SwrContext *resample_context = NULL;
 AVPacket packet;
 AVFrame *frame = 0;
 AVFrame *resampled = 0;
 int16_t *buffer;
 int ret, packet_ret, finished;

 ao_device *device;
 ao_sample_format format;
 int default_driver;

 if (argc != 2) {
 fprintf(stderr,"Usage: %s <filename>\n",argv[0]);
 return 1;
 }

 av_register_all();
 printf("Opening file...\n");
 if (open_audio_file(argv[1],&context,&codec_context) < 0)
 return 1;

 printf("Initializing resampler...\n");
 if (init_resampler(codec_context,&resample_context) < 0) {
 avformat_close_input(&context);
 avcodec_free_context(&codec_context);
 return 1;
 }

 // Setup libao
 printf("Starting audio device...\n");
 ao_initialize();
 default_driver = ao_default_driver_id();
 format.bits = OUTPUT_BITS;
 format.channels = OUTPUT_CHANNELS;
 format.rate = codec_context->sample_rate;
 format.byte_format = AO_FMT_NATIVE;
 format.matrix = 0;
 if ((device = ao_open_live(default_driver,&format,NULL)) == NULL) {
 fprintf(stderr,"Error opening audio device\n");
 avformat_close_input(&context);
 avcodec_free_context(&codec_context);
 swr_free(&resample_context);
 return 1;
 }

 // Mainloop
 printf("Beginning mainloop...\n");
 init_packet(&packet);
 // Read packets until done
 while (1) {
 packet_ret = av_read_frame(context,&packet);
 // Send a packet
 if ((ret = avcodec_send_packet(codec_context,&packet)) < 0)
 fprintf(stderr,"Error sending packet to decoder: %s\n",errtext(ret));

 av_packet_unref(&packet);

 while (1) {
 if (!frame)
 frame = av_frame_alloc();

 ret = avcodec_receive_frame(codec_context,frame);
 if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) // Need more input
 break;
 else if (ret < 0) {
 fprintf(stderr,"Error receiving frame: %s\n",errtext(ret));
 break;
 }
 // We have a valid frame, need to resample it
 if (!resampled)
 resampled = av_frame_alloc();

 resampled->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
 resampled->sample_rate = codec_context->sample_rate;
 resampled->format = OUTPUT_FMT;

 if ((ret = swr_convert_frame(resample_context,resampled,frame)) < 0) {
 fprintf(stderr,"Error resampling: %s\n",errtext(ret));
 } else {
 ao_play(device,(char*)resampled->extended_data[0],resampled->linesize[0]);
 }
 av_frame_unref(resampled);
 av_frame_unref(frame);
 }

 if (packet_ret == AVERROR_EOF)
 break;
 }

 printf("Closing file and freeing contexts...\n");
 avformat_close_input(&context);
 avcodec_free_context(&codec_context);
 swr_free(&resample_context);

 printf("Closing audio device...\n");
 ao_close(device);
 ao_shutdown();

 return 0;
}
</filename>


UPDATE : I've got it playing sound now, but it sounds like samples are missing (and MP3 files warn that "Could not update timestamps for skipped samples"). The issue was that the
resampledframe needed to have certain attributes set before being passed toswr_convert_frame. I've also addedav_packet_unrefandav_frame_unref, but I'm still unsure as to where to best locate them.

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What's the most desireable way to capture system display and audio in the form of individual encoded audio and video packets in go (language) ? [closed]
11 janvier 2023, par Tiger YangQuestion (read the context below first) :


For those of you familiar with the capabilities of go, Is there a better way to go about all this ? Since ffmpeg is so ubiquitous, I'm sure it's been optomized to perfection, but what's the best way to capture system display and audio in the form of individual encoded audio and video packets in go (language), so that they can be then sent via webtransport-go ? I wish for it to prioritize efficiency and low latency, and ideally capture and encode the framebuffer directly like ffmpeg does.


Thanks ! I have many other questions about this, but I think it's best to ask as I go.


Context and what I've done so far :


I'm writing a remote desktop software for my personal use because of grievances with current solutions out there. At the moment, it consists of a web app that uses the webtransport API to send input datagrams and receive AV packets on two dedicated unidirectional streams, and the webcodecs API to decode these packets. On the serverside, I originally planned to use python with the aioquic library as a webtransport server. Upon connection and authentication, the server would start ffmpeg as a subprocess with this command :


ffmpeg -init_hw_device d3d11va -filter_complex ddagrab=video_size=1920x1080:framerate=60 -vcodec hevc_nvenc -tune ll -preset p7 -spatial_aq 1 -temporal_aq 1 -forced-idr 1 -rc cbr -b:v 400K -no-scenecut 1 -g 216000 -f hevc -


What I really appreciate about this is that it uses windows' desktop duplication API to copy the framebuffer of my GPU and hand that directly to the on-die hardware encoder with zero round trips to the CPU. I think it's about as efficient and elegant a solution as I can manage. It then outputs the encoded stream to the stdout, which python can read and send to the client.


As for the audio, there is another ffmpeg instance :


ffmpeg -f dshow -channels 2 -sample_rate 48000 -sample_size 16 -audio_buffer_size 15 -i audio="RD Audio (High Definition Audio Device)" -acodec libopus -vbr on -application audio -mapping_family 0 -apply_phase_inv true -b:a 25K -fec false -packet_loss 0 -map 0 -f data -


which listens to a physical loopback interface, which is literally just a short wire bridging the front panel headphone and microphone jacks (I'm aware of the quality loss of converting to analog and back, but the audio is then crushed down to 25kbps so it's fine) ()


Unfortunately, aioquic was not easy to work with IMO, and I found webtransport-go https://github.com/adriancable/webtransport-go, which was a hell of a lot better in both simplicity and documentation. However, now I'm dealing with a whole new language, and I wanna ask : (above)


EDIT : Here's the code for my server so far :




package main

import (
 "bytes"
 "context"
 "fmt"
 "log"
 "net/http"
 "os/exec"
 "time"

 "github.com/adriancable/webtransport-go"
)

func warn(str string) {
 fmt.Printf("\n===== WARNING ===================================================================================================\n %s\n=================================================================================================================\n", str)
}

func main() {

 password := []byte("abc")

 videoString := []string{
 "ffmpeg",
 "-init_hw_device", "d3d11va",
 "-filter_complex", "ddagrab=video_size=1920x1080:framerate=60",
 "-vcodec", "hevc_nvenc",
 "-tune", "ll",
 "-preset", "p7",
 "-spatial_aq", "1",
 "-temporal_aq", "1",
 "-forced-idr", "1",
 "-rc", "cbr",
 "-b:v", "500K",
 "-no-scenecut", "1",
 "-g", "216000",
 "-f", "hevc", "-",
 }

 audioString := []string{
 "ffmpeg",
 "-f", "dshow",
 "-channels", "2",
 "-sample_rate", "48000",
 "-sample_size", "16",
 "-audio_buffer_size", "15",
 "-i", "audio=RD Audio (High Definition Audio Device)",
 "-acodec", "libopus",
 "-mapping_family", "0",
 "-b:a", "25K",
 "-map", "0",
 "-f", "data", "-",
 }

 connected := false

 http.HandleFunc("/", func(writer http.ResponseWriter, request *http.Request) {
 session := request.Body.(*webtransport.Session)

 session.AcceptSession()
 fmt.Println("\nAccepted incoming WebTransport connection.")
 fmt.Println("Awaiting authentication...")

 authData, err := session.ReceiveMessage(session.Context()) // Waits here till first datagram
 if err != nil { // if client closes connection before sending anything
 fmt.Println("\nConnection closed:", err)
 return
 }

 if len(authData) >= 2 && bytes.Equal(authData[2:], password) {
 if connected {
 session.CloseSession()
 warn("Client has authenticated, but a session is already taking place! Connection closed.")
 return
 } else {
 connected = true
 fmt.Println("Client has authenticated!\n")
 }
 } else {
 session.CloseSession()
 warn("Client has failed authentication! Connection closed. (" + string(authData[2:]) + ")")
 return
 }

 videoStream, _ := session.OpenUniStreamSync(session.Context())

 videoCmd := exec.Command(videoString[0], videoString[1:]...)
 go func() {
 videoOut, _ := videoCmd.StdoutPipe()
 videoCmd.Start()

 buffer := make([]byte, 15000)
 for {
 len, err := videoOut.Read(buffer)
 if err != nil {
 break
 }
 if len > 0 {
 videoStream.Write(buffer[:len])
 }
 }
 }()

 time.Sleep(50 * time.Millisecond)

 audioStream, err := session.OpenUniStreamSync(session.Context())

 audioCmd := exec.Command(audioString[0], audioString[1:]...)
 go func() {
 audioOut, _ := audioCmd.StdoutPipe()
 audioCmd.Start()

 buffer := make([]byte, 15000)
 for {
 len, err := audioOut.Read(buffer)
 if err != nil {
 break
 }
 if len > 0 {
 audioStream.Write(buffer[:len])
 }
 }
 }()

 for {
 data, err := session.ReceiveMessage(session.Context())
 if err != nil {
 videoCmd.Process.Kill()
 audioCmd.Process.Kill()

 connected = false

 fmt.Println("\nConnection closed:", err)
 break
 }

 if len(data) == 0 {

 } else if data[0] == byte(0) {
 fmt.Printf("Received mouse datagram: %s\n", data)
 }
 }

 })

 server := &webtransport.Server{
 ListenAddr: ":1024",
 TLSCert: webtransport.CertFile{Path: "SSL/fullchain.pem"},
 TLSKey: webtransport.CertFile{Path: "SSL/privkey.pem"},
 QuicConfig: &webtransport.QuicConfig{
 KeepAlive: false,
 MaxIdleTimeout: 3 * time.Second,
 },
 }

 fmt.Println("Launching WebTransport server at", server.ListenAddr)
 ctx, cancel := context.WithCancel(context.Background())
 if err := server.Run(ctx); err != nil {
 log.Fatal(err)
 cancel()
 }

}






-
Motion : Raspian webcam Unable to find a compatible palette format
25 juin 2018, par mrtumnusI am trying to get a Creative Live ! Motion webcam working on a Raspberry Pi 3 B+.
fswebcamandmotionboth give an error, "Unable to find a compatible palette format".ffmpeg, on the other hand, is able to easily capture video from the webcam.Device info :
$ lsusb
Bus 001 Device 004: ID 041e:4041 Creative Technology, Ltd Webcam Live! Motion`
$ v4l2-ctl --all
Driver Info (not using libv4l2):
Driver name : sq930x
Card type : Creative WebCam Live! Motion
Bus info : usb-3f980000.usb-1.2
Driver version: 4.14.34
Capabilities : 0x85200001
Video Capture
Read/Write
Streaming
Extended Pix Format
Device Capabilities
Device Caps : 0x05200001
Video Capture
Read/Write
Streaming
Extended Pix Format
Priority: 2
Video input : 0 (sq930x: ok)
Format Video Capture:
Width/Height : 640/480
Pixel Format : 'RGGB'
Field : None
Bytes per Line : 640
Size Image : 307200
Colorspace : sRGB
Transfer Function : Default
YCbCr/HSV Encoding: Default
Quantization : Default
Flags :
Streaming Parameters Video Capture:
Frames per second: invalid (0/0)
Read buffers : 2
User Controls
exposure (int) : min=1 max=4095 step=1 default=854 value=854
gain (int) : min=1 max=255 step=1 default=141 value=141Log of
motionfailure to open device :$ cat /var/log/motion/motion.log
[0:motion] [NTC] [ALL] [Jun 24 16:16:42] motion_startup: Using log type (ALL) log level (NTC)
[0:motion] [NTC] [ENC] [Jun 24 16:16:42] ffmpeg_init: ffmpeg libavcodec version 57.48.101 libavformat version 57.41.100
[0:motion] [NTC] [ALL] [Jun 24 16:16:42] main: Motion running in setup mode.
[0:motion] [NTC] [ALL] [Jun 24 16:16:42] main: Camera 0 is from /etc/motion/motion.conf
[0:motion] [NTC] [ALL] [Jun 24 16:16:42] main: Camera 0 is device: /dev/video0 input -1
[0:motion] [NTC] [ALL] [Jun 24 16:16:42] main: Stream port 8081
[0:motion] [NTC] [ALL] [Jun 24 16:16:42] main: Waiting for threads to finish, pid: 12067
[0:web_control] [NTC] [STR] [Jun 24 16:16:42] http_bindsock: listening on 127.0.0.1 port 8080
[1:ml1] [NTC] [ALL] [Jun 24 16:16:42] motion_init: Camera 0 started: motion detection Enabled
[0:web_control] [NTC] [STR] [Jun 24 16:16:42] httpd_run: Started motion-httpd server on port 8080 (auth Disabled)
[1:ml1] [NTC] [VID] [Jun 24 16:16:42] vid_v4lx_start: Using videodevice /dev/video0 and input -1
[1:ml1] [NTC] [VID] [Jun 24 16:16:42] v4l2_get_capability:
------------------------
cap.driver: "sq930x"
cap.card: "Creative WebCam Live! Motion"
cap.bus_info: "usb-3f980000.usb-1.2"
cap.capabilities=0x85200001
------------------------
[1:ml1] [NTC] [VID] [Jun 24 16:16:42] v4l2_get_capability: - VIDEO_CAPTURE
[1:ml1] [NTC] [VID] [Jun 24 16:16:42] v4l2_get_capability: - READWRITE
[1:ml1] [NTC] [VID] [Jun 24 16:16:42] v4l2_get_capability: - STREAMING
[1:ml1] [NTC] [VID] [Jun 24 16:16:42] v4l2_select_input: name = "sq930x", type 0x00000002, status 00000000
[1:ml1] [NTC] [VID] [Jun 24 16:16:42] v4l2_select_input: - CAMERA
[1:ml1] [WRN] [VID] [Jun 24 16:16:42] v4l2_select_input: Device doesn't support VIDIOC_G_STD
[1:ml1] [NTC] [VID] [Jun 24 16:16:42] v4l2_set_pix_format: Config palette index 17 (YU12) doesn't work.
[1:ml1] [NTC] [VID] [Jun 24 16:16:42] v4l2_set_pix_format: Supported palettes:
[1:ml1] [NTC] [VID] [Jun 24 16:16:42] v4l2_set_pix_format: (0) RGGB (8-bit Bayer RGRG/GBGB)
[1:ml1] [NTC] [VID] [Jun 24 16:16:42] v4l2_set_pix_format: 0 - 8-bit Bayer RGRG/GBGB (compressed : 0) (0x42474752)
[1:ml1] [ERR] [VID] [Jun 24 16:16:42] v4l2_set_pix_format: Unable to find a compatible palette format.
[1:ml1] [NTC] [VID] [Jun 24 16:16:42] vid_v4lx_start: Using V4L1
[1:ml1] [NTC] [ALL] [Jun 24 16:16:42] image_ring_resize: Resizing pre_capture buffer to 1 items
[1:ml1] [ERR] [ALL] [Jun 24 16:16:52] motion_init: Error capturing first image
[1:ml1] [NTC] [STR] [Jun 24 16:16:52] http_bindsock: listening on 127.0.0.1 port 8081
[1:ml1] [NTC] [ALL] [Jun 24 16:16:52] motion_init: Started motion-stream server on port 8081 (auth Disabled)
[1:ml1] [ERR] [ALL] [Jun 24 16:16:52] motion_loop: Video device fatal error - Closing video device
[1:ml1] [NTC] [VID] [Jun 24 16:16:52] vid_close: Closing video device /dev/video0Successful
ffmpegrecording :$ ffmpeg -i /dev/video0 test.mp4
ffmpeg version 3.2.10-1~deb9u1+rpt1 Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 6.3.0 (Raspbian 6.3.0-18+rpi1) 20170516
configuration: --prefix=/usr --extra-version='1~deb9u1+rpt1' --toolchain=hardened --libdir=/usr/lib/arm-linux-gnueabihf --incdir=/usr/include/arm-linux-gnueabihf --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx-rpi --enable-mmal --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 34.101 / 55. 34.101
libavcodec 57. 64.101 / 57. 64.101
libavformat 57. 56.101 / 57. 56.101
libavdevice 57. 1.100 / 57. 1.100
libavfilter 6. 65.100 / 6. 65.100
libavresample 3. 1. 0 / 3. 1. 0
libswscale 4. 2.100 / 4. 2.100
libswresample 2. 3.100 / 2. 3.100
libpostproc 54. 1.100 / 54. 1.100
[video4linux2,v4l2 @ 0x16035c0] Time per frame unknown
Input #0, video4linux2,v4l2, from '/dev/video0':
Duration: N/A, start: 2523.746120, bitrate: N/A
Stream #0:0: Video: rawvideo ([186]RG[8] / 0x84752BA), bayer_rggb8, 640x480, 25.42 tbr, 1000k tbn, 1000k tbcRelated questions :
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