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Video d’abeille en portrait
14 mai 2011, par
Mis à jour : Février 2012
Langue : français
Type : Video
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Sur d’autres sites (11224)
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ffmpeg 4 : Using the stream_loop parameter to loop the audio during a video ends up with an infinite loop
17 juin 2020, par JarsOfJam-SchedulerSummary



- 

- Context
- The software I use
- The problem
-
Results

4.1. Actual Results


4.2. Expected Results
-
What did I try to fix the bug ?
-
How to reproduce this bug : minimal and testable example with the provided required data
-
The question
-
Sources





















Context



I would want to set an audio WAV as the background sound of a video WEBM. The video can be shorter or longer than the audio. At the moment I add the audio over the video, I don't know the length of both streams. The audio must repeat until the video ends (the audio can be truncated if the video ends before the end of the last repetition of the audio).



The software I use



I use ffmpeg version 4.2.2-1ubuntu1 18.04.sav0.



The problem



ffmpeg seems to enter in an infinite loop when it proccesses in order to mix the audio and the video. Also, the length of the currently-generating-output-file (which contains both video and audio) is equal to the length of the audio, instead of the length of the video.



The problem seems to be triggered by this command line :



ffmpeg -i directory_1/video.webm -stream_loop -1 -fflags +shortest -max_interleave_delta 50000 -i directory_2/audio.wav directory_3/video_and_audio.webm



Results



Actual Results



Three things :



- 

-
The infinite loop of the ffmpeg process : I must manually stop the ffmpeg process
-
The output video file with music (which is currently generating but output anyway) : it contains both audio and video. But the length of the output file is equal to the length of the audio, instead of the length of the video.
-
The following output logs :











ffmpeg version 4.2.2-1ubuntu1 18.04.sav0 Copyright (c) 2000-2019 the
 FFmpeg developers built with gcc 7 (Ubuntu 7.5.0-3ubuntu1 18.04)

 configuration : —prefix=/usr —extra-version='1ubuntu1 18.04.sav0'
 —toolchain=hardened —libdir=/usr/lib/x86_64-linux-gnu —incdir=/usr/include/x86_64-linux-gnu —arch=amd64 —enable-gpl —disable-stripping —enable-avresample —disable-filter=resample —enable-avisynth —enable-gnutls —enable-ladspa —enable-libaom —enable-libass —enable-libbluray —enable-libbs2b —enable-libcaca —enable-libcdio —enable-libcodec2 —enable-libflite —enable-libfontconfig —enable-libfreetype —enable-libfribidi —enable-libgme —enable-libgsm —enable-libjack —enable-libmp3lame —enable-libmysofa —enable-libopenjpeg —enable-libopenmpt —enable-libopus —enable-libpulse —enable-librsvg —enable-librubberband —enable-libshine —enable-libsnappy —enable-libsoxr —enable-libspeex —enable-libssh —enable-libtheora —enable-libtwolame —enable-libvidstab —enable-libvorbis —enable-libvpx —enable-libwavpack —enable-libwebp —enable-libx265 —enable-libxml2 —enable-libxvid —enable-libzmq —enable-libzvbi —enable-lv2 —enable-omx —enable-openal —enable-opencl —enable-opengl —enable-sdl2 —enable-libdc1394 —enable-libdrm —enable-libiec61883 —enable-nvenc —enable-chromaprint —enable-frei0r —enable-libx264 —enable-shared libavutil 56. 31.100 / 56. 31.100 libavcodec 58. 54.100 / 58. 54.100 libavformat 58. 29.100 / 58. 29.100 libavdevice 58. 8.100 /
 58. 8.100 libavfilter 7. 57.100 / 7. 57.100 libavresample 4. 0. 0 / 4. 0. 0 libswscale 5. 5.100 / 5. 5.100 libswresample 3. 5.100 / 3. 5.100 libpostproc 55. 5.100 /
 55. 5.100 Input #0, matroska,webm, from 'youtubed/my_youtube_video.webm' : Metadata :
 encoder : Chrome Duration : N/A, start : 0.000000, bitrate : N/A
 Stream #0:0(eng) : Video : vp8, yuv420p(progressive), 3200x1608, SAR 1:1 DAR 400:201, 1k tbr, 1k tbn, 1k tbc (default)
 Metadata :
 alpha_mode : 1 Guessed Channel Layout for Input Stream #1.0 : stereo Input #1, wav, from 'tmp_music/original_music.wav' :

 Duration : 00:00:11.78, bitrate : 1411 kb/s
 Stream #1:0 : Audio : pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s Stream mapping : Stream #0:0 -> #0:0 (vp8
 (native) -> vp9 (libvpx-vp9)) Stream #1:0 -> #0:1 (pcm_s16le
 (native) -> opus (libopus)) Press [q] to stop, [?] for help
 [libvpx-vp9 @ 0x5645268aed80] v1.8.2 [libopus @ 0x5645268b09c0] No bit
 rate set. Defaulting to 96000 bps. Output #0, webm, to
 'youtubed/my_youtube_video_with_music.webm' : Metadata :
 encoder : Lavf58.29.100
 Stream #0:0(eng) : Video : vp9 (libvpx-vp9), yuv420p(progressive), 3200x1608 [SAR 1:1 DAR 400:201], q=-1—1, 200 kb/s, 1k fps, 1k tbn, 1k
 tbc (default)
 Metadata :
 alpha_mode : 1
 encoder : Lavc58.54.100 libvpx-vp9
 Side data :
 cpb : bitrate max/min/avg : 0/0/0 buffer size : 0 vbv_delay : -1
 Stream #0:1 : Audio : opus (libopus), 48000 Hz, stereo, s16, 96 kb/s
 Metadata :
 encoder : Lavc58.54.100 libopus




Expected Results



- 

-
No infinite loop during the ffmpeg process
-
Concerning the output logs, I don't know what it should look.
-
The output file with the audio and the video should :



3.1. If the video is longer than the audio, then the audio is repeated until it exactly fits the video. The audio can be truncated.



3.2. If the video is shorter than the audio, then the audio is truncated and exactly fits the video.



3.3. If both video and audio are of the same length, then the audio exactly fits the video.









How to reproduce this bug ? (+ required data)



- 

-
Download the following files (resp. audio and video) (I must refresh these download links every 24 hours) :



1.1. https://a.uguu.se/dmgsmItjJMDq_audio.wav



-
Move them into the directory/directories of your choice.
-
Open your CLI, move to the adequat directory and copy/paste/execute the instruction given in Part. The Problem (don't forget to eventually modify this instruction by indicating the adequat directories, according to step 2.).
-
You'll face my problem.











What did I try to fix the bug ?



Nothing, since I don't even understand why the bug occures.



The question



How to correct my command in order to mix these audio and video streams without any infinite loop during the ffmpeg process, keeping in mind that I don't know their length, and that audio must be repeated in order to fit the video, even if audio must be truncated (in the case of the last repetition of the audio file must be truncated because the video stream has just ended) ?



Sources



The source is the command line you can find in Part. The problem.


-
FFmpeg with Nvidia GPU - full HW transcode with 50i to 50p deinterlacing
5 janvier 2018, par Jernej StopinšekI’m trying to do a full hardware transcode of an udp stream to hls
with 50i to 50p deinterlacing.I’m using ffmpeg and Nvidia GPU.
Since HLS requires deinterlacing
I would like to deinterlace an interlaced source stream and preserve
as much smooth motion and picture quality as possible.My hardware, software and driver info :
GPU : Tesla P100-PCIE-12GB
Nvidia Driver Version : 387.26
Cuda compilation tools, release 9.1, V9.1.85
FFmpeg from git on 20171218
ffmpeg version N-89520-g3f88744067 Copyright (c) 2000-2017 the FFmpeg
developers built with gcc 6.3.0 (Debian 6.3.0-18) 20170516
configuration : —enable-gpl
—enable-cuda-sdk —enable-libx264 —enable-libx265 —enable-nonfree —enable-libnpp —enable-opengl —enable-opencl —enable-libfreetype —enable-openssl —enable-libzvbi —enable-libfontconfig —enable-libfreetype —enable-libfribidi —extra-cflags=-I/usr/local/cuda/include —extra-ldflags=-L/usr/local/cuda/lib64 —arch=x86_64libavutil 56. 6.100 / 56. 6.100
libavcodec 58. 8.100 / 58.
8.100
libavformat 58. 3.100 / 58. 3.100
libavdevice 58. 0.100 / 58. 0.100
libavfilter 7. 7.100 / 7. 7.100
libswscale 5.
0.101 / 5. 0.101
libswresample 3. 0.101 / 3. 0.101
libpostproc 55. 0.100 / 55. 0.100Input stream info :
ffmpeg -t 00:05:00 -i udp://xxx.xxx.xxx.xxx:xxxx -map 0:0 -vf idet -c rawvideo -y -f rawvideo /dev/nullInput #0, mpegts, from ’udp ://xxx.xxx.xxx.xxx:xxxx’ :
Duration :
N/A, start : 49634.159411, bitrate : N/A
Program xxxxx
Metadata : service_name :
service_provider : Stream
#0:0[0x44d] : Video : h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709, top first), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 50 tbr, 90k
tbn, 50 tbc
Stream #0:10x19de : Audio : mp2 ([3][0][0][0] /
0x0003), 48000 Hz, stereo, s16p, 192 kb/s
Stream
#0:20x19e1 : Subtitle : dvb_subtitle ([6][0][0][0] / 0x0006)Output #0, rawvideo, to ’/dev/null’ :
Metadata :
encoder :
Lavf58.3.100
Stream #0:0 : Video : rawvideo (I420 / 0x30323449),
yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 622080 kb/s, 25 fps, 25
tbn, 25 tbc
Metadata :
encoder : Lavc58.8.100 rawvideo
frame= 7538 fps= 25 q=-0.0 Lsize=22896675kB time=00:05:01.52
bitrate=622080.0kbits/s dup=38 drop=0 speed=1.02x
video:22896675kB audio:0kB subtitle:0kB other streams:0kB global
headers:0kB muxing overhead : 0.000000%
[Parsed_idet_0 @
0x56370b3c5080] Repeated Fields : Neither : 7458 Top : 24 Bottom : 18
[Parsed_idet_0 @ 0x56370b3c5080] Single frame detection : TFF : 281 BFF :
13 Progressive : 5639 Undetermined : 1567
[Parsed_idet_0 @
0x56370b3c5080] Multi frame detection : TFF : 380 BFF : 0 Progressive :
7120 Undetermined : 0
This is my command for adaptive hardware deinterlacing. It gives great results with picture, but sound is out of sync.
ffmpeg -y -err_detect ignore_err -loglevel debug -vsync -1 -hwaccel cuvid -hwaccel_device 1 -c:v h264_cuvid -deint adaptive -r:v 50 -gpu:v 1 -i "udp://xxx.xxx.xxx.xxx:xxxx=?overrun_nonfatal=1&fifo_size=84450&buffer_size=33554432" -map 0:0 -map 0:1 -c:a aac -b:a 196k -c:v h264_nvenc -flags -global_header+cgop -gpu:v 1 -g:v 50 -bf:v 4 -coder:v cabac -b_adapt:v false -b:v 5184000 -minrate:v 5184000 -maxrate:v 5184000 -bufsize:v 2488320 -rc:v cbr_hq -2pass:v true -rc-lookahead:v 25 -no-scenecut:v 1 -profile:v high -preset:v slow -color_range:v 1 -color_trc:v 1 -color_primaries:v 1 -colorspace:v 1 -f hls -hls_time 5 -hls_list_size 3 -start_number 0 -hls_flags delete_segments /srv/hls/program_01/1080p/index.m3u8If I add option "-drop_second_field 1" to h264_cuvid and remove -r:v 50 from input and put it to h264_nvenc - then transcoded stream has synced audio, but I think I’m losing quality due to drop_second_field option.
ffmpeg -y -err_detect ignore_err -loglevel debug -vsync -1 -hwaccel cuvid -hwaccel_device 1 -c:v h264_cuvid -deint adaptive -drop_second_field 1 -gpu:v 1 -i "udp://xxx.xxx.xxx.xxx:xxxx=?overrun_nonfatal=1&fifo_size=84450&buffer_size=33554432" -map 0:0 -map 0:1 -c:a aac -b:a 196k -c:v h264_nvenc -flags -global_header+cgop -gpu:v 1 -g:v 50 -r:v 50 -bf:v 4 -coder:v cabac -b_adapt:v false -b:v 5184000 -minrate:v 5184000 -maxrate:v 5184000 -bufsize:v 2488320 -rc:v cbr_hq -2pass:v true -rc-lookahead:v 25 -no-scenecut:v 1 -profile:v high -preset:v slow -color_range:v 1 -color_trc:v 1 -color_primaries:v 1 -colorspace:v 1 -f hls -hls_time 5 -hls_list_size 3 -start_number 0 -hls_flags delete_segments /srv/hls/program_01/1080p/index.m3u8Could someone please point me in the right direction how to properly deinterlace with cuvid and minimal possible loss of quality ?
-
ffmpeg capture from ip camera video in h264 stream [closed]
23 mars 2023, par Иванов ИванI can't read the frames from the camera and then write them to a video file (any). The fact is that I even get crooked frames, they seem to have violated the coordinates of the position of each point, the video is crooked, distorted


c++ code.


https://drive.google.com/file/d/1W2sZMR5D5pvVmnhiQyhiaQhC9frhdeII/view?usp=sharing


#define INBUF_SIZE 4096


 //writing the minimal required header for a pgm file format
 //portable graymap format-> https://en.wikipedia.org/wiki/Netpbm_format#PGM_example
 fprintf (f, "P5\n%d %d\n%d\n", xsize, ysize, 255);

 //writing line by line
 for (i = 0; i /contains data on a configuration of media content, such as bitrate, 
 //frame rate, sampling frequency, channels, height and many other things.
 AVCodecContext * AVCodecContext_ = NULL;
 AVCodecParameters * AVCodecParametr_ = NULL;
 FILE * f;
 //This structure describes decoded (raw) audio- or video this.
 AVFrame * frame;
 uint8_t inbuf [INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
 uint8_t * data;
 size_t data_size;
 int ret;
 int eof;
 AVFormatContext * AVfc = NULL;
 int ERRORS;
 //AVCodec * codec;
 char buf [1024];
 const char * FileName;
 
 //https://habr.com/ru/post/137793/
 //Stores the compressed one shot.
 AVPacket * pkt;
 
 //**********************************************************************
 //Beginning of reading video from the camera. 
 //**********************************************************************
 
 avdevice_register_all ();
 
 filename = "rtsp://admin: 754HG@192.168.1.75:554/11";
 //filename = "c:\\1.avi";
 outfilename = "C:\\2.MP4";
 
 //We open a flow of video (it is the file or the camera). 
 ERRORS = avformat_open_input (& AVfc, filename, NULL, NULL);
 if (ERRORS <0) {
 fprintf (stderr, "ffmpeg: could not open file \n");
 return-1;
 }
 
 //After opening, we can print out information on the video file (iformat = the name of a format; 
 //duration = duration). But as I connected the camera to me wrote: Duration: N/A, 
 //start: 0.000000, bitrate: N/A
 printf ("Format %s, duration %lld us", AVfc-> iformat-> long_name, AVfc-> duration);
 
 
 ERRORS = avformat_find_stream_info (AVfc, NULL);
 if (ERRORS <0) {
 fprintf (stderr, "ffmpeg: Unable to find stream info\n");
 return-1;
 }
 
 
 int CountStream;
 
 //We learn quantity of streams. 
 CountStream = AVfc-> nb_streams;
 
 //Let's look for the codec. 
 int video_stream;
 for (video_stream = 0; video_stream nb_streams; ++ video_stream) {
 if (AVfc-> streams[video_stream]-> codecpar-> codec_type == AVMEDIA_TYPE_VIDEO) {
 break;
 }
 
 }
 
 if (video_stream == AVfc-> nb_streams) {
 fprintf (stderr, "ffmpeg: Unable to find video stream\n");
 return-1;
 }
 
 //Here we define a type of the codec, for my camera it is equal as AV_CODEC_ID_HEVC (This that in what is broadcast by my camera)
 codec = avcodec_find_decoder(AVfc-> streams [video_stream]-> codecpar-> codec_id);
 //--------------------------------------------------------------------------------------
 
 //Functions for inquiry of opportunities of libavcodec,
 AVCodecContext_ = avcodec_alloc_context3(codec);
 if (! AVCodecContext _) {
 fprintf (stderr, "Was not succeeded to allocate a video codec context, since it not poddrerzhivayetsya\n");
 exit(1);
 }
 
 //This function is used for initialization 
 //AVCodecContext of video and audio of the codec. The announcement of avcodec_open2 () is in libavcodecavcodec.h
 //We open the codec. 
 
 ERRORS = avcodec_open2 (AVCodecContext _, codec, NULL);
 if (ERRORS <0) {
 fprintf (stderr, "ffmpeg: It is not possible to open codec \n");
 return-1;
 }
 
 //It for processing of a sound - a reserve.
 //swr_alloc_set_opts ()
 //swr_init (); 
 
 //To output all information on the video file. 
 av_dump_format (AVfc, 0, argv[1], 0);
 
 //=========================================================================================
 //Further, we receive frames. before we only received all infomration about the entering video.
 //=========================================================================================
 
 //Now we are going to read packages from a stream and to decode them in shots, but at first 
 //we need to mark out memory for both components (AVPacket and AVFrame).
 frame = av_frame_alloc ();
 
 if (! frame) {
 fprintf (stderr, "Is not possible to mark out memory for video footage \n");
 exit(1);
 }
 //We mark out memory for a package 
 pkt = av_packet_alloc ();
 //We define a file name for saving the picture.
 const char * FileName1 = "C:\\Users\\Павел\\Desktop\\NyFile.PGM";
 //Data reading if they is. 
 while (av_read_frame (AVfc, pkt)> = 0) {
 //It is a package from a video stream? Because there is still a soundtrack.
 if (pkt-> stream_index == video_stream) {
 int ret;
 
 //Transfer of the raw package data as input data in the decoder
 ret = avcodec_send_packet (AVCodecContext _, pkt);
 if (ret <0 | | ret == AVERROR(EAGAIN) | | ret == AVERROR_EOF) {
 std:: cout <<"avcodec_send_packet:" <<ret while="while"> = 0) {
 
 //Returns the decoded output data from the decoder or the encoder
 ret = avcodec_receive_frame (AVCodecContext _, frame);
 if (ret == AVERROR(EAGAIN) | | ret == AVERROR_EOF) {
 //std:: cout <<"avcodec_receive_frame:" <<ret cout="cout"> of frame_number </============================================================================================
 
 //Experimentally - we will keep a shot in the picture. 
 
 save_gray_frame(frame-> data [0], frame-> linesize [0], frame-> width, frame-> height, (char *) FileName1);
 }
 }
 }
 
 //av_parser_close(parser);
 avcodec_free_context (& AVCodecContext _);
 av_frame_free (& frame);
 av_packet_free (& pkt);
 
 return 0;
</ret></ret>

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