Recherche avancée
Médias (2)
-
Granite de l’Aber Ildut
9 septembre 2011, par
Mis à jour : Septembre 2011
Langue : français
Type : Texte
-
Géodiversité
9 septembre 2011, par ,
Mis à jour : Août 2018
Langue : français
Type : Texte
Autres articles (79)
-
Personnaliser en ajoutant son logo, sa bannière ou son image de fond
5 septembre 2013, parCertains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;
-
Ecrire une actualité
21 juin 2013, parPrésentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
Dans le thème par défaut spipeo de MédiaSPIP, les actualités sont affichées en bas de la page principale sous les éditoriaux.
Vous pouvez personnaliser le formulaire de création d’une actualité.
Formulaire de création d’une actualité Dans le cas d’un document de type actualité, les champs proposés par défaut sont : Date de publication ( personnaliser la date de publication ) (...) -
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
Sur d’autres sites (4504)
-
ffmpeg scale 4k video and add subtitles
29 novembre 2020, par goodkid38I am having trouble scaling video from 4k to 1080 while still retaining the video sub titles. These subtitles are PGS based and therefore need to be baked into the output video. In order to scale the video I am using ffmpeg with a filter_complex like so :


'-filter_complex', `[0:v][0:s:randomSubtitleHere]scale=1920:1080[scaled];[scaled][0:s:0]overlay[v]`, 
'-map', '[v]', '-map', 
`0:a:${movieTitle['audioSelect']}`,


This will give me the video at 1080, but it i still am not seeing the subtitles I have selected.


-
FFMPEG Presentation Time Stamps (PTS) calculation in RTSP stream
8 décembre 2020, par BadaBudaBuduBelow please find en raw example of my code for your better understanding of what it does. Please note that this is an updated (removed deprecated methods, etc.) example code by myself from the official FFMPEG documentation complemented by my encoder.


/// STD
#include <iostream>
#include <string>

/// FFMPEG
extern "C"
{
 #include <libavformat></libavformat>avformat.h>
 #include <libswscale></libswscale>swscale.h>
 #include <libavutil></libavutil>imgutils.h>
}

/// VideoLib
#include <tools></tools>multimediaprocessing.h>
#include 
#include 
#include <enums></enums>codec.h>
#include <enums></enums>pixelformat.h>

/// OpenCV
#include <opencv2></opencv2>opencv.hpp>

inline static const char *inputRtspAddress = "rtsp://192.168.0.186:8080/video/h264";

int main()
{
 AVFormatContext* formatContext = nullptr;

 AVStream* audioStream = nullptr;
 AVStream* videoStream = nullptr;
 AVCodec* audioCodec = nullptr;
 AVCodec* videoCodec = nullptr;
 AVCodecContext* audioCodecContext = nullptr;
 AVCodecContext* videoCodecContext = nullptr;
 vl::AudioSettings audioSettings;
 vl::VideoSettings videoSettings;

 int audioIndex = -1;
 int videoIndex = -1;

 SwsContext* swsContext = nullptr;
 std::vector frameBuffer;
 AVFrame* frame = av_frame_alloc();
 AVFrame* decoderFrame = av_frame_alloc();

 AVPacket packet;
 cv::Mat mat;

 vl::tools::MultimediaProcessing multimediaProcessing("rtsp://127.0.0.1:8080/stream", vl::configs::rtspStream, 0, vl::enums::EPixelFormat::ABGR);

 // *** OPEN STREAM *** //
 if(avformat_open_input(&formatContext, inputRtspAddress, nullptr, nullptr) < 0)
 {
 std::cout << "Failed to open input." << std::endl;
 return EXIT_FAILURE;
 }

 if(avformat_find_stream_info(formatContext, nullptr) < 0)
 {
 std::cout << "Failed to find stream info." << std::endl;
 return EXIT_FAILURE;
 }

 // *** FIND DECODER FOR BOTH AUDIO AND VIDEO STREAM *** //
 audioCodec = avcodec_find_decoder(AVCodecID::AV_CODEC_ID_AAC);
 videoCodec = avcodec_find_decoder(AVCodecID::AV_CODEC_ID_H264);

 if(audioCodec == nullptr || videoCodec == nullptr)
 {
 std::cout << "No AUDIO or VIDEO in stream." << std::endl;
 return EXIT_FAILURE;
 }

 // *** FIND STREAM FOR BOTH AUDIO AND VIDEO STREAM *** //

 audioIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, &audioCodec, 0);
 videoIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_VIDEO, -1, -1, &videoCodec, 0);

 if(audioIndex < 0 || videoIndex < 0)
 {
 std::cout << "Failed to find AUDIO or VIDEO stream." << std::endl;
 return EXIT_FAILURE;
 }

 audioStream = formatContext->streams[audioIndex];
 videoStream = formatContext->streams[videoIndex];

 // *** ALLOC CODEC CONTEXT FOR BOTH AUDIO AND VIDEO STREAM *** //
 audioCodecContext = avcodec_alloc_context3(audioCodec);
 videoCodecContext = avcodec_alloc_context3(videoCodec);

 if(audioCodecContext == nullptr || videoCodecContext == nullptr)
 {
 std::cout << "Can not allocate AUDIO or VIDEO context." << std::endl;
 return EXIT_FAILURE;
 }

 if(avcodec_parameters_to_context(audioCodecContext, formatContext->streams[audioIndex]->codecpar) < 0 || avcodec_parameters_to_context(videoCodecContext, formatContext->streams[videoIndex]->codecpar) < 0)
 {
 std::cout << "Can not fill AUDIO or VIDEO codec context." << std::endl;
 return EXIT_FAILURE;
 }

 if(avcodec_open2(audioCodecContext, audioCodec, nullptr) < 0 || avcodec_open2(videoCodecContext, videoCodec, nullptr) < 0)
 {
 std::cout << "Failed to open AUDIO codec" << std::endl;
 return EXIT_FAILURE;
 }

 // *** INITIALIZE MULTIMEDIA PROCESSING *** //
 std::vector<unsigned char="char"> extraData(audioStream->codecpar->extradata_size);
 std::copy_n(audioStream->codecpar->extradata, extraData.size(), extraData.begin());

 audioSettings.sampleRate = audioStream->codecpar->sample_rate,
 audioSettings.bitrate = audioStream->codecpar->bit_rate,
 audioSettings.codec = vl::enums::EAudioCodec::AAC,
 audioSettings.channels = audioStream->codecpar->channels,
 audioSettings.bitsPerCodedSample = audioStream->codecpar->bits_per_coded_sample,
 audioSettings.bitsPerRawSample = audioStream->codecpar->bits_per_raw_sample,
 audioSettings.blockAlign = audioStream->codecpar->block_align,
 audioSettings.channelLayout = audioStream->codecpar->channel_layout,
 audioSettings.format = audioStream->codecpar->format,
 audioSettings.frameSize = audioStream->codecpar->frame_size,
 audioSettings.codecExtraData = std::move(extraData);

 videoSettings.width = 1920;
 videoSettings.height = 1080;
 videoSettings.framerate = 25;
 videoSettings.pixelFormat = vl::enums::EPixelFormat::ARGB;
 videoSettings.bitrate = 8000 * 1000;
 videoSettings.codec = vl::enums::EVideoCodec::H264;

 multimediaProcessing.initEncoder(videoSettings, audioSettings);

 // *** INITIALIZE SWS CONTEXT *** //
 swsContext = sws_getCachedContext(nullptr, videoCodecContext->width, videoCodecContext->height, videoCodecContext->pix_fmt, videoCodecContext->width, videoCodecContext->height, AV_PIX_FMT_RGBA, SWS_FAST_BILINEAR, nullptr, nullptr, nullptr);

 if (const auto inReturn = av_image_get_buffer_size(AV_PIX_FMT_RGBA, videoCodecContext->width, videoCodecContext->height, 1); inReturn > 0)
 {
 frameBuffer.reserve(inReturn);
 }
 else
 {
 std::cout << "Can not get buffer size." << std::endl;
 return EXIT_FAILURE;
 }

 if (const auto inReturn = av_image_fill_arrays(frame->data, frame->linesize, frameBuffer.data(), AV_PIX_FMT_RGBA, videoCodecContext->width, videoCodecContext->height, 1); inReturn < 0)
 {
 std::cout << "Can not fill buffer arrays." << std::endl;
 return EXIT_FAILURE;
 }

 // *** MAIN LOOP *** //
 while(true)
 {
 // Return the next frame of a stream.
 if(av_read_frame(formatContext, &packet) == 0)
 {
 if(packet.stream_index == videoIndex) // Check if it is video packet.
 {
 // Send packet to decoder.
 if(avcodec_send_packet(videoCodecContext, &packet) == 0)
 {
 int returnCode = avcodec_receive_frame(videoCodecContext, decoderFrame); // Get Frame from decoder.

 if (returnCode == 0) // Transform frame and send it to encoder. And re-stream that.
 {
 sws_scale(swsContext, decoderFrame->data, decoderFrame->linesize, 0, decoderFrame->height, frame->data, frame->linesize);

 mat = cv::Mat(videoCodecContext->height, videoCodecContext->width, CV_8UC4, frameBuffer.data(), frame->linesize[0]);

 cv::resize(mat, mat, cv::Size(1920, 1080), cv::INTER_NEAREST);

 multimediaProcessing.encode(mat.data, packet.dts, packet.dts, packet.flags == AV_PKT_FLAG_KEY); // Thise line sends cv::Mat to encoder and re-streams it.

 av_packet_unref(&packet);
 }
 else if(returnCode == AVERROR(EAGAIN))
 {
 av_frame_unref(decoderFrame);
 av_freep(decoderFrame);
 }
 else
 {
 av_frame_unref(decoderFrame);
 av_freep(decoderFrame);

 std::cout << "Error during decoding." << std::endl;
 return EXIT_FAILURE;
 }
 }
 }
 else if(packet.stream_index == audioIndex) // Check if it is audio packet.
 {
 std::vector vectorPacket(packet.data, packet.data + packet.size);

 multimediaProcessing.addAudioPacket(vectorPacket, packet.dts, packet.dts);
 }
 else
 {
 av_packet_unref(&packet);
 }
 }
 else
 {
 std::cout << "Can not send video packet to decoder." << std::endl;
 std::this_thread::sleep_for(std::chrono::seconds(1));
 }
 }

 return EXIT_SUCCESS;
 }
</unsigned></string></iostream>

What does It do ?


It takes a single RTSP stream to decode its data so I can, for example, draw something to its frames or whatever, and then stream it under a different address.


Basically, I am opening the RTSP stream, check if it does contain both audio and video streams, and find a decoder for them. Then I create an encoder to which I will tell how the output stream should look like and that's it.


And this point I will create an endless loop Where I will read all packets coming from the input stream, then decode it does something to it and again encode it and re=stream it.


What is the issue ?


If you take a closer look I am sending both video and audio frame together with lastly received PTS and DTS contained in AVPacket, to the encoder.


The PTS and DTS from the point when I receive the first AVPacket looks for example like this.


IN AUDIO STREAM :




-22783, -21759, -20735, -19711, -18687, -17663, -16639, -15615, -14591, -13567, -12543, -11519, -10495, -9471, -8447, -7423, -6399, -5375, -4351, -3327, -2303, -1279, -255, 769, 1793, 2817, 3841, 4865, 5889, 6913, 7937, 8961, 9985, 11009, 12033, 13057, 14081, 15105, 16129, 17153




As you can see it is every time incremented by 1024 and that is a sample rate of the audio stream. Quite clear here.


IN VIDEO STREAM :




86400, 90000, 93600, 97200, 100800, 104400, 108000, 111600, 115200, 118800, 122400, 126000, 129600, 133200, 136800, 140400, 144000, 147600, 151200, 154800, 158400, 162000, 165600




As you can see it is every time incremented by 3600 but WHY ?. What this number actually mean ?


From what I can understand, those received PTS and DTS are for the following :


DTS should tell the encoder when it should start encoding the frame so the frame in time are in the correct order and not mishmashed.


PTS should say the correct time when the frame should be played/displayed in the output stream so the frame in time are in the correct order and not mishmashed.


What I am trying to achieve ?


As I said I need to restream a RTSP stream. I can not use PTS and DTS which comes from received AVPackets, because at some point it can happen that the input stream can randomly close and I need to open it again. The problem is that when I actually do it, then the PTS and DTS start to generate again from the minus values same as you could see in the samples. I CAN NOT send those "new" PTS and DTS to the encoder because they are now lower than the encoder/muxer expects.


I need to continually stream something (both audio and video), even it is a blank black screen or silent audio. And each frame the PTS and DTS should rise by a specific number. I need to figure out how the increment is calculated.


----------------------------------


The final result should look like a mosaic of multiple input streams in a single output stream. A single input stream (main) has both audio and video and the rest (side) has just video. Some of those streams can randomly close in time and I need to ensure that it will be back again once it is possible.


-
Invalid data stream in media could not be discarded by FFMPEG. Why is it staying and how to discard it ?
5 décembre 2020, par Link-akroI have downloaded a [short media][1] i intend to convert then i will use the result as basis to practice and test any and all FFMPEG commands i learn or use later.


That video seems to have an unknown invalid stream which never disappears no matter everything i tried to discard it. When i try to work with it later i get various problems like missing codec or no stream which is not the focus here but the reason why i got stubborn to remove the things i cannot deal with.


The following probing prints a warning in yellow color on last row.


> ffprobe -hide_banner -show_streams Movie_Countdown-youtube_I1vMKZ1kvg0.mov

Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Movie_Countdown-youtube_I1vMKZ1kvg0.mov':
 Metadata:
 major_brand : qt
 minor_version : 537199360
 compatible_brands: qt
 creation_time : 2015-05-20T13:45:55.000000Z
 Duration: 00:00:10.00, start: 0.000000, bitrate: 11474 kb/s
 Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080, 9930 kb/s, SAR 1:1 DAR 16:9, 25 fps, 25 tbr, 25 tbn, 50 tbc (default)
 Metadata:
 creation_time : 2015-05-20T13:45:55.000000Z
 handler_name : Apple Video Media Handler
 encoder : H.264
 timecode : 00:00:00:00
 Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, s16, 1536 kb/s (default)
 Metadata:
 creation_time : 2015-05-20T13:45:56.000000Z
 handler_name : Apple Sound Media Handler
 timecode : 00:00:00:00
 Stream #0:2(eng): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
 Metadata:
 creation_time : 2015-05-20T13:46:11.000000Z
 handler_name : Time Code Media Handler
 timecode : 00:00:00:00
Unsupported codec with id 0 for input stream 2


Below the streams output. I split the text so you see the warning without searching in the middle.


[STREAM]
index=0
codec_name=h264
codec_long_name=H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10
profile=Main
codec_type=video
codec_time_base=1/50
codec_tag_string=avc1
codec_tag=0x31637661
width=1920
height=1080
coded_width=1920
coded_height=1088
closed_captions=0
has_b_frames=0
sample_aspect_ratio=1:1
display_aspect_ratio=16:9
pix_fmt=yuv420p
level=40
color_range=tv
color_space=bt709
color_transfer=bt709
color_primaries=bt709
chroma_location=left
field_order=unknown
timecode=N/A
refs=1
is_avc=true
nal_length_size=4
id=N/A
r_frame_rate=25/1
avg_frame_rate=25/1
time_base=1/25
start_pts=0
start_time=0.000000
duration_ts=250
duration=10.000000
bit_rate=9930739
max_bit_rate=N/A
bits_per_raw_sample=8
nb_frames=250
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=1
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
TAG:creation_time=2015-05-20T13:45:55.000000Z
TAG:language=eng
TAG:handler_name=Apple Video Media Handler
TAG:encoder=H.264
TAG:timecode=00:00:00:00
[/STREAM]
[STREAM]
index=1
codec_name=pcm_s16le
codec_long_name=PCM signed 16-bit little-endian
profile=unknown
codec_type=audio
codec_time_base=1/48000
codec_tag_string=sowt
codec_tag=0x74776f73
sample_fmt=s16
sample_rate=48000
channels=2
channel_layout=stereo
bits_per_sample=16
id=N/A
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/48000
start_pts=0
start_time=0.000000
duration_ts=480000
duration=10.000000
bit_rate=1536000
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=480000
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=1
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
TAG:creation_time=2015-05-20T13:45:56.000000Z
TAG:language=eng
TAG:handler_name=Apple Sound Media Handler
TAG:timecode=00:00:00:00
[/STREAM]
[STREAM]
index=2
codec_name=unknown
codec_long_name=unknown
profile=unknown
codec_type=data
codec_tag_string=tmcd
codec_tag=0x64636d74
id=N/A
r_frame_rate=0/0
avg_frame_rate=25/1
time_base=1/25
start_pts=0
start_time=0.000000
duration_ts=250
duration=10.000000
bit_rate=3
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=1
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=1
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
TAG:creation_time=2015-05-20T13:46:11.000000Z
TAG:language=eng
TAG:handler_name=Time Code Media Handler
TAG:timecode=00:00:00:00
[/STREAM]


I scaled it down and recoded it for starters and i tried to discard the data stream with
-dnand failed other methods i will mention at the end.
The output so far suggests that discarding should have worked since alternative datas are 0kB and no stream 2 is listed.

> ffmpeg -hide_banner -dn -i C:\Users\admin-dix\Downloads\Movie_Countdown-youtube_I1vMKZ1kvg0.mov -vf "scale=h=450:w=800" -f mp4 -c:a aac -c:v libx264 mov_countdown.mp4

Output #0, mp4, to 'mov_countdown.mp4':
 Metadata:
 major_brand : qt
 minor_version : 537199360
 compatible_brands: qt
 encoder : Lavf58.45.100
 Stream #0:0(eng): Video: h264 (libx264) (avc1 / 0x31637661), yuv420p(progressive), 800x450 [SAR 1:1 DAR 16:9], q=-1--1, 0.04 fps, 12800 tbn, 25 tbc (default)
 Metadata:
 creation_time : 2015-05-20T13:45:55.000000Z
 handler_name : Apple Video Media Handler
 timecode : 00:00:00:00
 encoder : Lavc58.91.100 libx264
 Side data:
 cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
 Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)
 Metadata:
 creation_time : 2015-05-20T13:45:56.000000Z
 handler_name : Apple Sound Media Handler
 timecode : 00:00:00:00
 encoder : Lavc58.91.100 aac
frame= 250 fps= 43 q=-1.0 Lsize= 342kB time=00:00:10.00 bitrate= 280.1kbits/s speed= 1.7x
video:175kB audio:159kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.710857%


Then ffprobe disagrees, there is still the stream and the warning.


ffprobe mov_countdown.mp4

Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'mov_countdown.mp4':
 Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 encoder : Lavf58.45.100
 Duration: 00:00:10.02, start: 0.000000, bitrate: 279 kb/s
 Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 800x450 [SAR 1:1 DAR 16:9], 142 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
 Metadata:
 handler_name : Apple Video Media Handler
 timecode : 00:00:00:00
 Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 129 kb/s (default)
 Metadata:
 handler_name : Apple Sound Media Handler
 Stream #0:2(eng): Data: none (tmcd / 0x64636D74), 0 kb/s
 Metadata:
 handler_name : Apple Video Media Handler
 timecode : 00:00:00:00
Unsupported codec with id 0 for input stream 2


I tried negative mapping as per this answer
-map 0:dwhich failed. I did not understand-discardoption as ffmpeg documentation refers to from the-vn,-an,-dnentries as it does not specify a stream.

Why does it do that and how can i remove that stream i do not know and do not want to care about in future tests ?


[1] : Clean Retro Movie Countdown - YouTube from Philippe Moesch https://www.youtube.com/watch?v=I1vMKZ1kvg0


Syndiquer tout le site
