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GetID3 - Boutons supplémentaires
9 avril 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Image
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Core Media Video
4 avril 2013, par
Mis à jour : Juin 2013
Langue : français
Type : Video
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The pirate bay depuis la Belgique
1er avril 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Image
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Bug de détection d’ogg
22 mars 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Video
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Exemple de boutons d’action pour une collection collaborative
27 février 2013, par
Mis à jour : Mars 2013
Langue : français
Type : Image
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Exemple de boutons d’action pour une collection personnelle
27 février 2013, par
Mis à jour : Février 2013
Langue : English
Type : Image
Autres articles (101)
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Use, discuss, criticize
13 avril 2011, parTalk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
A discussion list is available for all exchanges between users. -
Le plugin : Podcasts.
14 juillet 2010, parLe problème du podcasting est à nouveau un problème révélateur de la normalisation des transports de données sur Internet.
Deux formats intéressants existent : Celui développé par Apple, très axé sur l’utilisation d’iTunes dont la SPEC est ici ; Le format "Media RSS Module" qui est plus "libre" notamment soutenu par Yahoo et le logiciel Miro ;
Types de fichiers supportés dans les flux
Le format d’Apple n’autorise que les formats suivants dans ses flux : .mp3 audio/mpeg .m4a audio/x-m4a .mp4 (...) -
Des sites réalisés avec MediaSPIP
2 mai 2011, parCette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page.
Sur d’autres sites (5337)
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MP3 files created using FFmpeg are not starting playback in browser immediately. Is there any major difference between FFmpeg and AVCONV ?
23 janvier 2019, par AR5I am working on a website that streams music. We recently changed server from Debian (with avconv) to a Centos7 (with FFmpeg) server.
The mp3 files created on Debian server start playback on browser (I have tested Chrome and Firefox) start almost at the same time they start loading into the browser (I used Network tab on Developer Tools to monitor this)Now after the switch to Centos/FFmpeg server, the files being created on this new server are displaying a strange behavior. They only start playback after about 1MB is loaded into the browser.
I have used identical settings for converting original file into MP3 in both AVCONV and FFmpeg but the files created using FFmpeg are showing this issue. Is there something that might be causing such an issue ? Are there differences in terms of audio conversion between AVCONV and FFmpeg ?
I have already tried
I first found that the files created on old server (Debian/Avconv) were VBR (variable bitrate) and the ones created on new server were CBR (constant bitrate), so I tried switching to VBR but the issue still persisted.
I checked the mp3 files using MediaInfo app and there seems to be no difference between the files.
I also checked if both files were being served as 206 Partial Content and they both are indeed.
I am trying to create mp3 files using FFmpeg that work exactly like the ones created before using avconv
I am trying to make the streaming site work on the new server but the mp3 files created using FFmpeg are not playing back correctly as compared to the ones created on the old server. I am trying to figure out what I might be doing wrong ? or if there is a difference between avconv and FFmpeg that is causing this issue.
I am really stuck on this issue, any help will be really appreciated.
Edit
I don’t have access to old server anymore so I couldn’t retrieve the log output of avconv. The command that I was using was as follows :
avconv -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"Here is the command and log output from new server :
ffmpeg -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"
ffmpeg version 2.8.15 Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-28)
configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
libavutil 54. 31.100 / 54. 31.100
libavcodec 56. 60.100 / 56. 60.100
libavformat 56. 40.101 / 56. 40.101
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 40.101 / 5. 40.101
libavresample 2. 1. 0 / 2. 1. 0
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 2.101 / 1. 2.101
libpostproc 53. 3.100 / 53. 3.100
[mp3 @ 0xd60be0] Skipping 0 bytes of junk at 240044.
Input #0, mp3, from '/test/Track 01.mp3':
Metadata:
album : Future Hndrxx Presents: The WIZRD
artist : Future
genre : Hip-Hop
title : Never Stop
track : 1
lyrics-eng : rgf.is
WEB SITE : rgf.is
TAGGINGTIME : rgf.is
WEB : rgf.is
date : 2019
encoder : Lavf56.40.101
Duration: 00:04:51.40, start: 0.025056, bitrate: 121 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 114 kb/s
Metadata:
encoder : Lavc56.60
Stream #0:1: Video: png, rgb24(pc), 333x333 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
Metadata:
comment : Cover (front)
[mp3 @ 0xd66ec0] Frame rate very high for a muxer not efficiently supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2
Output #0, mp3, to '/test/Track 01 (converted).mp3':
Metadata:
TALB : Future Hndrxx Presents: The WIZRD
TPE1 : Future
TCON : Hip-Hop
TIT2 : Never Stop
TRCK : 1
lyrics-eng : rgf.is
WEB SITE : rgf.is
TAGGINGTIME : rgf.is
WEB : rgf.is
TDRC : 2019
TSSE : Lavf56.40.101
Stream #0:0: Video: png, rgb24, 333x333 [SAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc
Metadata:
comment : Cover (front)
encoder : Lavc56.60.100 png
Stream #0:1: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 128 kb/s
Metadata:
encoder : Lavc56.60.100 libmp3lame
Stream mapping:
Stream #0:1 -> #0:0 (png (native) -> png (native))
Stream #0:0 -> #0:1 (mp3 (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
[libmp3lame @ 0xd9b0c0] Trying to remove 1152 samples, but the queue is emptys/s
frame= 1 fps=0.1 q=-0.0 Lsize= 4788kB time=00:04:51.39 bitrate= 134.6kbits/s
video:234kB audio:4553kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.014809%Samples of MP3 files
I have uploaded samples of mp3 files created using both avconv and FFmpeg. Please find these here : https://drive.google.com/drive/folders/1gRTmMM2iSK0VWQ4Zaf_iBNQe5laFJl08?usp=sharing
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ffmpeg avcodec_send_packet/avcodec_receive_frame memory leak
23 janvier 2019, par G HamlinI’m attempting to decode frames, but memory usage grows with every frame (more specifically, with every call to avcodec_send_packet) until finally the code crashes with a bad_alloc. Here’s the basic decode loop :
int rfret = 0;
while((rfret = av_read_frame(inctx.get(), &packet)) >= 0){
if (packet.stream_index == vstrm_idx) {
//std::cout << "Sending Packet" << std::endl;
int ret = avcodec_send_packet(ctx.get(), &packet);
if (ret < 0 || ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
std::cout << "avcodec_send_packet: " << ret << std::endl;
break;
}
while (ret >= 0) {
//std::cout << "Receiving Frame" << std::endl;
ret = avcodec_receive_frame(ctx.get(), fr);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
//std::cout << "avcodec_receive_frame: " << ret << std::endl;
av_frame_unref(fr);
// av_frame_free(&fr);
break;
}
std::cout << "frame: " << ctx->frame_number << std::endl;
// eventually do something with the frame here...
av_frame_unref(fr);
// av_frame_free(&fr);
}
}
else {
//std::cout << "Not Video" << std::endl;
}
av_packet_unref(&packet);
}Memory usage/leakage seems to scale with the resolution of the video I’m decoding. For example, for a 3840x2160 resolution video, the memory usage in windows task manager consistently jumps up by about 8mb (1 byte per pixel ??) for each received frame. Do I need to do something besides call av_frame_unref to release the memory ?
(more) complete code below
void AVFormatContextDeleter(AVFormatContext* ptr)
{
if (ptr) {
avformat_close_input(&ptr);
}
}
void AVCodecContextDeleter(AVCodecContext* ptr)
{
if (ptr) {
avcodec_free_context(&ptr);
}
}
typedef std::unique_ptr AVFormatContextPtr;
typedef std::unique_ptr AVCodecContextPtr;
AVCodecContextPtr createAvCodecContext(AVCodec *vcodec)
{
AVCodecContextPtr ctx(avcodec_alloc_context3(vcodec), AVCodecContextDeleter);
return ctx;
}
AVFormatContextPtr createFormatContext(const std::string& filename)
{
AVFormatContext* inctxPtr = nullptr;
int ret = avformat_open_input(&inctxPtr, filename.c_str(), nullptr, nullptr);
// int ret = avformat_open_input(&inctx, "D:/Videos/test.mp4", nullptr, nullptr);
if (ret != 0) {
inctxPtr = nullptr;
}
return AVFormatContextPtr(inctxPtr, AVFormatContextDeleter);
}
int testDecode()
{
// open input file context
AVFormatContextPtr inctx = createFormatContext("D:/Videos/Matt Chapman Hi Greg.MOV");
if (!inctx) {
// std::cerr << "fail to avforamt_open_input(\"" << infile << "\"): ret=" << ret;
return 1;
}
// retrieve input stream information
int ret = avformat_find_stream_info(inctx.get(), nullptr);
if (ret < 0) {
//std::cerr << "fail to avformat_find_stream_info: ret=" << ret;
return 2;
}
// find primary video stream
AVCodec* vcodec = nullptr;
const int vstrm_idx = av_find_best_stream(inctx.get(), AVMEDIA_TYPE_VIDEO, -1, -1, &vcodec, 0);
if (vstrm_idx < 0) {
//std::cerr << "fail to av_find_best_stream: vstrm_idx=" << vstrm_idx;
return 3;
}
AVCodecParameters* origin_par = inctx->streams[vstrm_idx]->codecpar;
if (vcodec == nullptr) { // is this even necessary?
vcodec = avcodec_find_decoder(origin_par->codec_id);
if (!vcodec) {
// Can't find decoder
return 4;
}
}
AVCodecContextPtr ctx = createAvCodecContext(vcodec);
if (!ctx) {
return 5;
}
ret = avcodec_parameters_to_context(ctx.get(), origin_par);
if (ret) {
return 6;
}
ret = avcodec_open2(ctx.get(), vcodec, nullptr);
if (ret < 0) {
return 7;
}
//print input video stream informataion
std::cout
//<< "infile: " << infile << "\n"
<< "format: " << inctx->iformat->name << "\n"
<< "vcodec: " << vcodec->name << "\n"
<< "size: " << origin_par->width << 'x' << origin_par->height << "\n"
<< "fps: " << av_q2d(ctx->framerate) << " [fps]\n"
<< "length: " << av_rescale_q(inctx->duration, ctx->time_base, {1,1000}) / 1000. << " [sec]\n"
<< "pixfmt: " << av_get_pix_fmt_name(ctx->pix_fmt) << "\n"
<< "frame: " << inctx->streams[vstrm_idx]->nb_frames << "\n"
<< std::flush;
AVPacket packet;
av_init_packet(&packet);
packet.data = nullptr;
packet.size = 0;
AVFrame *fr = av_frame_alloc();
if (!fr) {
return 8;
}
int rfret = 0;
while((rfret = av_read_frame(inctx.get(), &packet)) >= 0){
if (packet.stream_index == vstrm_idx) {
//std::cout << "Sending Packet" << std::endl;
int ret = avcodec_send_packet(ctx.get(), &packet);
if (ret < 0 || ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
std::cout << "avcodec_send_packet: " << ret << std::endl;
break;
}
while (ret >= 0) {
//std::cout << "Receiving Frame" << std::endl;
ret = avcodec_receive_frame(ctx.get(), fr);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
//std::cout << "avcodec_receive_frame: " << ret << std::endl;
av_frame_unref(fr);
// av_frame_free(&fr);
break;
}
std::cout << "frame: " << ctx->frame_number << std::endl;
// do something with the frame here...
av_frame_unref(fr);
// av_frame_free(&fr);
}
}
else {
//std::cout << "Not Video" << std::endl;
}
av_packet_unref(&packet);
}
std::cout << "RFRET = " << rfret << std::endl;
return 0;
}Update 1 : (1/21/2019) Compiling on a different machine and running with different video files I am not seeing the memory usage growing without bound. I’ll try to narrow down where the difference lies (compiler ?, ffmpeg version ?, or video encoding ?)
Update 2 : (1/21/2019) Ok, it looks like there is some interaction occurring between ffmpeg and Qt’s QCamera. In my application, I’m using Qt to manage the webcam, but decided to use ffmpeg libraries to handle decoding/encoding since Qt doesn’t have as comprehensive support for different codecs. If I have the camera turned on (through Qt), ffmpeg decoding memory consumption grows without bound. If the camera is off, ffmpeg behaves fine. I’ve tried this both with a physical camera (Logitech C920) and with a virtual camera using OBS-Virtualcam, with the same result. So far I’m baffled as to how the two systems are interacting...
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Concatenating multiple mp4 files [duplicate]
18 octobre 2019, par Mr MedusaThis question already has an answer here :
I’m currently trying to join multiple mp4 video files into one file, and I found the ffmpeg library and looked at a lot of ways on how I should use it.
Currently I’m usingffmpeg -f concat -safe 0 -i mylist.txt -c copy output.mp4This should work, but for some reason it doesn’t. My terminal is filled with this
(Can’t give anymore output than this)
[mp4 @ 000002ce260bc180] Non-monotonous DTS in output stream 0:0; previous: 24541175, current: 5482875; changing to 24541176. This may result in incorrect timestamps in the output file.
[mp4 @ 000002ce260bc180] Non-monotonous DTS in output stream 0:0; previous: 24541176, current: 5483397; changing to 24541177. This may result in incorrect timestamps in the output file.
[mp4 @ 000002ce260bc180] Non-monotonous DTS in output stream 0:0; previous: 24541177, current: 5483904; changing to 24541178. This may result in incorrect timestamps in the output file.
[mp4 @ 000002ce260bc180] Non-monotonous DTS in output stream 0:0; previous: 24541178, current: 5484426; changing to 24541179. This may result in incorrect timestamps in the output file.
frame=19167 fps=4682 q=-1.0 Lsize= 249689kB time=00:26:37.73 bitrate=1280.2kbits/s speed= 390x
video:243606kB audio:5584kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.200045%Mylist.txt
file 'ab.mp4'
file 'bb.mp4'
file 'cc.mp4'
file 'dd.mp4'
file 'ee.mp4'
file 'ff.mp4'
file 'gg.mp4'
file 'hh.mp4'
file 'jj.mp4'
file 'kk.mp4'Info about each input :
ffmpeg version git-2019-10-13-4f4334b Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 9.2.1 (GCC) 20191010
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
libavutil 56. 35.100 / 56. 35.100
libavcodec 58. 59.102 / 58. 59.102
libavformat 58. 33.100 / 58. 33.100
libavdevice 58. 9.100 / 58. 9.100
libavfilter 7. 62.100 / 7. 62.100
libswscale 5. 6.100 / 5. 6.100
libswresample 3. 6.100 / 3. 6.100
libpostproc 55. 6.100 / 55. 6.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'ab.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.83.100
Duration: 00:00:26.59, start: 0.000000, bitrate: 4610 kb/s
Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, unknown/bt470bg/unknown), 1280x720, 4466 kb/s, 60.04 fps, 60 tbr, 15360 tbn, 2k tbc (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : SoundHandler
Input #1, mov,mp4,m4a,3gp,3g2,mj2, from 'bb.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.83.100
Duration: 00:00:40.77, start: 0.000000, bitrate: 3384 kb/s
Stream #1:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1920x1080, 3251 kb/s, 29.99 fps, 29.97 tbr, 15360 tbn, 2k tbc (default)
Metadata:
handler_name : VideoHandler
Stream #1:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : SoundHandler
Input #2, mov,mp4,m4a,3gp,3g2,mj2, from 'cc.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.83.100
Duration: 00:00:06.29, start: 0.000000, bitrate: 8029 kb/s
Stream #2:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, unknown/bt470bg/unknown), 1600x900, 7896 kb/s, 60.16 fps, 60 tbr, 15360 tbn, 2k tbc (default)
Metadata:
handler_name : VideoHandler
Stream #2:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : SoundHandler
Input #3, mov,mp4,m4a,3gp,3g2,mj2, from 'dd.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.83.100
Duration: 00:00:50.90, start: 0.000000, bitrate: 7851 kb/s
Stream #3:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuvj420p(pc, unknown/bt709/unknown), 1920x1080, 7708 kb/s, 60.02 fps, 60 tbr, 15360 tbn, 2k tbc (default)
Metadata:
handler_name : VideoHandler
Stream #3:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : SoundHandler
Input #4, mov,mp4,m4a,3gp,3g2,mj2, from 'ee.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.83.100
Duration: 00:00:27.10, start: 0.000000, bitrate: 6056 kb/s
Stream #4:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1920x1080, 5912 kb/s, 60.03 fps, 60 tbr, 15360 tbn, 2k tbc (default)
Metadata:
handler_name : VideoHandler
Stream #4:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : SoundHandler
Input #5, mov,mp4,m4a,3gp,3g2,mj2, from 'ff.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.83.100
Duration: 00:00:31.00, start: 0.000000, bitrate: 6290 kb/s
Stream #5:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1920x1080, 6150 kb/s, 60.03 fps, 60 tbr, 15360 tbn, 2k tbc (default)
Metadata:
handler_name : VideoHandler
Stream #5:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : SoundHandler
Input #6, mov,mp4,m4a,3gp,3g2,mj2, from 'gg.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.83.100
Duration: 00:01:00.00, start: 0.000000, bitrate: 4531 kb/s
Stream #6:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, unknown/bt470bg/unknown), 1280x720, 4387 kb/s, 60.02 fps, 60 tbr, 15360 tbn, 2k tbc (default)
Metadata:
handler_name : VideoHandler
Stream #6:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : SoundHandler
Input #7, mov,mp4,m4a,3gp,3g2,mj2, from 'hh.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf58.20.100
Duration: 00:00:30.00, start: 0.000000, bitrate: 8186 kb/s
Stream #7:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 1920x1080, 8046 kb/s, 60 fps, 60 tbr, 90k tbn, 120 tbc (default)
Metadata:
handler_name : VideoHandler
Stream #7:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : SoundHandler
Input #8, mov,mp4,m4a,3gp,3g2,mj2, from 'jj.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.83.100
Duration: 00:00:49.81, start: 0.000000, bitrate: 5579 kb/s
Stream #8:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, unknown/bt470bg/unknown), 1600x900, 5437 kb/s, 60.02 fps, 60 tbr, 15360 tbn, 2k tbc (default)
Metadata:
handler_name : VideoHandler
Stream #8:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : SoundHandler
Input #9, mov,mp4,m4a,3gp,3g2,mj2, from 'kk.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.83.100
Duration: 00:00:34.69, start: 0.000000, bitrate: 5223 kb/s
Stream #9:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1920x1080, 5088 kb/s, 30.03 fps, 30 tbr, 15360 tbn, 2k tbc (default)
Metadata:
handler_name : VideoHandler
Stream #9:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : SoundHandler
At least one output file must be specifiedI’ve read most of the posts on here about ffmpeg causing this issue for other people, but none of the fixes that are explained have worked for me.
The video output I get seems fine for the first video, but then once it’s supposed to transition into the other videos it just "freezes" or sets the speed to something like 100x. Output video also ends up being 28 minutes long instead of 2.8 minutes.
Anyone able to help me ?
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