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The Great Big Beautiful Tomorrow
28 octobre 2011, par
Mis à jour : Octobre 2011
Langue : English
Type : Texte
Autres articles (13)
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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 (...) -
Encoding and processing into web-friendly formats
13 avril 2011, parMediaSPIP automatically converts uploaded files to internet-compatible formats.
Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
All uploaded files are stored online in their original format, so you can (...) -
Taille des images et des logos définissables
9 février 2011, parDans beaucoup d’endroits du site, logos et images sont redimensionnées pour correspondre aux emplacements définis par les thèmes. L’ensemble des ces tailles pouvant changer d’un thème à un autre peuvent être définies directement dans le thème et éviter ainsi à l’utilisateur de devoir les configurer manuellement après avoir changé l’apparence de son site.
Ces tailles d’images sont également disponibles dans la configuration spécifique de MediaSPIP Core. La taille maximale du logo du site en pixels, on permet (...)
Sur d’autres sites (6297)
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MPEG DASH - do I need to have audio and video tracks as seperate source file for creating DASH package using mp4box
10 juillet 2016, par TarunI have one source mp4, I tried to create MPEG DASH package using mp4box by GPAC.
I am able to play output MPD files in OSMO4 player by GPAC.However I am not able to play the same in DASH JS player @ http://dashif.org/reference/players/javascript/0.2.3/index.html
When I try to play the mpd in it I get error "Error creating source buffer"
I tried reading their MPD files, and I found that those guys are using audio and video as separate source track.
Ques1) Does DASH specs states that audio and video tracks should be seprate source tracks ?
Ques2) Please find below the MPD file created by me, Let me know if anybody thinks that there is a problem in it
<mpd type="static" xmlns="urn:mpeg:DASH:schema:MPD:2011" profiles="urn:mpeg:dash:profile:full:2011" minbuffertime="PT1.5S" mediapresentationduration="PT0H2M31.63S">
<programinformation moreinformationurl="http://gpac.sourceforge.net">
</programinformation>
<period start="PT0S" duration="PT0H2M31.63S">
<adaptationset>
<contentcomponent contenttype="video"></contentcomponent>
<contentcomponent contenttype="audio" lang="und"></contentcomponent>
<segmenttemplate initialization="flight_init.mp4"></segmenttemplate>
<representation mimetype="video/mp4" codecs="avc1.64001f,mp4a.40.02" width="1280" height="720" samplerate="44100" numchannels="2" lang="und" startwithsap="1" bandwidth="3096320">
<segmenttemplate timescale="1000" duration="20164" media="flight_test_flight_3000$Number$.mp4" startnumber="1"></segmenttemplate>
</representation>
<representation mimetype="video/mp4" codecs="avc1.64001e,mp4a.40.02" width="640" height="360" samplerate="44100" numchannels="2" lang="und" startwithsap="1" bandwidth="1119428">
<segmenttemplate timescale="1000" duration="20099" media="flight_test_flight_1000$Number$.mp4" startnumber="1"></segmenttemplate>
</representation>
<representation mimetype="video/mp4" codecs="avc1.640014,mp4a.40.02" width="320" height="180" samplerate="44100" numchannels="2" lang="und" startwithsap="1" bandwidth="722208">
<segmenttemplate timescale="1000" duration="20164" media="flight_test_flight_600$Number$.mp4" startnumber="1"></segmenttemplate>
</representation>
</adaptationset>
</period>
</mpd> -
Libav (ffmpeg) copying decoded video timestamps to encoder
31 octobre 2016, par Jason CI am writing an application that decodes a single video stream from an input file (any codec, any container), does a bunch of image processing, and encodes the results to an output file (single video stream, Quicktime RLE, MOV). I am using ffmpeg’s libav 3.1.5 (Windows build for now, but the application will be cross-platform).
There is a 1:1 correspondence between input and output frames and I want the frame timing in the output to be identical to the input. I am having a really, really hard time accomplishing this. So my general question is : How do I reliably (as in, in all cases of inputs) set the output frame timing identical to the input ?
It took me a very long time to slog through the API and get to the point I am at now. I put together a minimal test program to work with :
#include <cstdio>
extern "C" {
#include <libavcodec></libavcodec>avcodec.h>
#include <libavformat></libavformat>avformat.h>
#include <libavutil></libavutil>avutil.h>
#include <libavutil></libavutil>imgutils.h>
#include <libswscale></libswscale>swscale.h>
}
using namespace std;
struct DecoderStuff {
AVFormatContext *formatx;
int nstream;
AVCodec *codec;
AVStream *stream;
AVCodecContext *codecx;
AVFrame *rawframe;
AVFrame *rgbframe;
SwsContext *swsx;
};
struct EncoderStuff {
AVFormatContext *formatx;
AVCodec *codec;
AVStream *stream;
AVCodecContext *codecx;
};
template <typename t="t">
static void dump_timebase (const char *what, const T *o) {
if (o)
printf("%s timebase: %d/%d\n", what, o->time_base.num, o->time_base.den);
else
printf("%s timebase: null object\n", what);
}
// reads next frame into d.rawframe and d.rgbframe. returns false on error/eof.
static bool read_frame (DecoderStuff &d) {
AVPacket packet;
int err = 0, haveframe = 0;
// read
while (!haveframe && err >= 0 && ((err = av_read_frame(d.formatx, &packet)) >= 0)) {
if (packet.stream_index == d.nstream) {
err = avcodec_decode_video2(d.codecx, d.rawframe, &haveframe, &packet);
}
av_packet_unref(&packet);
}
// error output
if (!haveframe && err != AVERROR_EOF) {
char buf[500];
av_strerror(err, buf, sizeof(buf) - 1);
buf[499] = 0;
printf("read_frame: %s\n", buf);
}
// convert to rgb
if (haveframe) {
sws_scale(d.swsx, d.rawframe->data, d.rawframe->linesize, 0, d.rawframe->height,
d.rgbframe->data, d.rgbframe->linesize);
}
return haveframe;
}
// writes an output frame, returns false on error.
static bool write_frame (EncoderStuff &e, AVFrame *inframe) {
// see note in so post about outframe here
AVFrame *outframe = av_frame_alloc();
outframe->format = inframe->format;
outframe->width = inframe->width;
outframe->height = inframe->height;
av_image_alloc(outframe->data, outframe->linesize, outframe->width, outframe->height,
AV_PIX_FMT_RGB24, 1);
//av_frame_copy(outframe, inframe);
static int count = 0;
for (int n = 0; n < outframe->width * outframe->height; ++ n) {
outframe->data[0][n*3+0] = ((n+count) % 100) ? 0 : 255;
outframe->data[0][n*3+1] = ((n+count) % 100) ? 0 : 255;
outframe->data[0][n*3+2] = ((n+count) % 100) ? 0 : 255;
}
++ count;
AVPacket packet;
av_init_packet(&packet);
packet.size = 0;
packet.data = NULL;
int err, havepacket = 0;
if ((err = avcodec_encode_video2(e.codecx, &packet, outframe, &havepacket)) >= 0 && havepacket) {
packet.stream_index = e.stream->index;
err = av_interleaved_write_frame(e.formatx, &packet);
}
if (err < 0) {
char buf[500];
av_strerror(err, buf, sizeof(buf) - 1);
buf[499] = 0;
printf("write_frame: %s\n", buf);
}
av_packet_unref(&packet);
av_freep(&outframe->data[0]);
av_frame_free(&outframe);
return err >= 0;
}
int main (int argc, char *argv[]) {
const char *infile = "wildlife.wmv";
const char *outfile = "test.mov";
DecoderStuff d = {};
EncoderStuff e = {};
av_register_all();
// decoder
avformat_open_input(&d.formatx, infile, NULL, NULL);
avformat_find_stream_info(d.formatx, NULL);
d.nstream = av_find_best_stream(d.formatx, AVMEDIA_TYPE_VIDEO, -1, -1, &d.codec, 0);
d.stream = d.formatx->streams[d.nstream];
d.codecx = avcodec_alloc_context3(d.codec);
avcodec_parameters_to_context(d.codecx, d.stream->codecpar);
avcodec_open2(d.codecx, NULL, NULL);
d.rawframe = av_frame_alloc();
d.rgbframe = av_frame_alloc();
d.rgbframe->format = AV_PIX_FMT_RGB24;
d.rgbframe->width = d.codecx->width;
d.rgbframe->height = d.codecx->height;
av_frame_get_buffer(d.rgbframe, 1);
d.swsx = sws_getContext(d.codecx->width, d.codecx->height, d.codecx->pix_fmt,
d.codecx->width, d.codecx->height, AV_PIX_FMT_RGB24,
SWS_POINT, NULL, NULL, NULL);
//av_dump_format(d.formatx, 0, infile, 0);
dump_timebase("in stream", d.stream);
dump_timebase("in stream:codec", d.stream->codec); // note: deprecated
dump_timebase("in codec", d.codecx);
// encoder
avformat_alloc_output_context2(&e.formatx, NULL, NULL, outfile);
e.codec = avcodec_find_encoder(AV_CODEC_ID_QTRLE);
e.stream = avformat_new_stream(e.formatx, e.codec);
e.codecx = avcodec_alloc_context3(e.codec);
e.codecx->bit_rate = 4000000; // arbitrary for qtrle
e.codecx->width = d.codecx->width;
e.codecx->height = d.codecx->height;
e.codecx->gop_size = 30; // 99% sure this is arbitrary for qtrle
e.codecx->pix_fmt = AV_PIX_FMT_RGB24;
e.codecx->time_base = d.stream->time_base; // ???
e.codecx->flags |= (e.formatx->flags & AVFMT_GLOBALHEADER) ? AV_CODEC_FLAG_GLOBAL_HEADER : 0;
avcodec_open2(e.codecx, NULL, NULL);
avcodec_parameters_from_context(e.stream->codecpar, e.codecx);
//av_dump_format(e.formatx, 0, outfile, 1);
dump_timebase("out stream", e.stream);
dump_timebase("out stream:codec", e.stream->codec); // note: deprecated
dump_timebase("out codec", e.codecx);
// open file and write header
avio_open(&e.formatx->pb, outfile, AVIO_FLAG_WRITE);
avformat_write_header(e.formatx, NULL);
// frames
while (read_frame(d) && write_frame(e, d.rgbframe))
;
// write trailer and close file
av_write_trailer(e.formatx);
avio_closep(&e.formatx->pb);
}
</typename></cstdio>A few notes about that :
- Since all of my attempts at frame timing so far have failed, I’ve removed almost all timing-related stuff from this code to start with a clean slate.
- Almost all error checking and cleanup omitted for brevity.
- The reason I allocate a new output frame with a new buffer in
write_frame
, rather than usinginframe
directly, is because this is more representative of what my real application is doing. My real app also uses RGB24 internally, hence the conversions here. - The reason I generate a weird pattern in
outframe
, rather than using e.g.av_copy_frame
, is because I just wanted a test pattern that compressed well with Quicktime RLE (my test input ends up generating a 1.7GB output file otherwise). - The input video I am using, "wildlife.wmv", can be found here. I’ve hard-coded the filenames.
- I am aware that
avcodec_decode_video2
andavcodec_encode_video2
are deprecated, but don’t care. They work fine, I’ve already struggled too much getting my head around the latest version of the API, ffmpeg changes their API with nearly every release, and I really don’t feel like dealing withavcodec_send_*
andavcodec_receive_*
right now. - I think I’m supposed to be finishing off by passing a NULL frame to
avcodec_encode_video2
to flush some buffers or something but I’m a bit confused about that. Unless somebody feels like explaining that let’s ignore it for now, it’s a separate question. The docs are as vague about this point as they are about everything else. - My test input file’s frame rate is 29.97.
Now, as for my current attempts. The following timing related fields are present in the above code, with details/confusion in bold. There’s a lot of them, because the API is mind-bogglingly convoluted :
main: d.stream->time_base
: Input video stream time base. For my test input file this is 1/1000.main: d.stream->codec->time_base
: Not sure what this is (I never could make sense of whyAVStream
has anAVCodecContext
field when you always use your own new context anyways) and also thecodec
field is deprecated. For my test input file this is 1/1000.main: d.codecx->time_base
: Input codec context time-base. For my test input file this is 0/1. Am I supposed to set it ?main: e.stream->time_base
: Time base of the output stream I create. What do I set this to ?main: e.stream->codec->time_base
: Time base of the deprecated and mysterious codec field of the output stream I create. Do I set this to anything ?main: e.codecx->time_base
: Time base of the encoder context I create. What do I set this to ?read_frame: packet.dts
: Decoding timestamp of packet read.read_frame: packet.pts
: Presentation timestamp of packet read.read_frame: packet.duration
: Duration of packet read.read_frame: d.rawframe->pts
: Presentation timestamp of raw frame decoded. This is always 0. Why isn’t it read by the decoder...?read_frame: d.rgbframe->pts
/write_frame: inframe->pts
: Presentation timestamp of decoded frame converted to RGB. Not set to anything currently.read_frame: d.rawframe->pkt_*
: Fields copied from packet, discovered after reading this post. They are set correctly but I don’t know if they are useful.write_frame: outframe->pts
: Presentation timestamp of frame being encoded. Should I set this to something ?write_frame: outframe->pkt_*
: Timing fields from a packet. Should I set these ? They seem to be ignored by the encoder.write_frame: packet.dts
: Decoding timestamp of packet being encoded. What do I set it to ?write_frame: packet.pts
: Presentation timestamp of packet being encoded. What do I set it to ?write_frame: packet.duration
: Duration of packet being encoded. What do I set it to ?
I have tried the following, with the described results. Note that
inframe
isd.rgbframe
:-
- Init
e.stream->time_base = d.stream->time_base
- Init
e.codecx->time_base = d.codecx->time_base
- Set
d.rgbframe->pts = packet.dts
inread_frame
- Set
outframe->pts = inframe->pts
inwrite_frame
- Result : Warning that encoder time base is not set (since
d.codecx->time_base was 0/1
), seg fault.
- Init
-
- Init
e.stream->time_base = d.stream->time_base
- Init
e.codecx->time_base = d.stream->time_base
- Set
d.rgbframe->pts = packet.dts
inread_frame
- Set
outframe->pts = inframe->pts
inwrite_frame
- Result : No warnings, but VLC reports frame rate as 480.048 (no idea where this number came from) and file plays too fast.
Also the encoder sets all the timing fields in(Edit : Turns out this is becausepacket
to 0, which was not what I expected.av_interleaved_write_frame
, unlikeav_write_frame
, takes ownership of the packet and swaps it with a blank one, and I was printing the values after that call. So they are not ignored.)
- Init
-
- Init
e.stream->time_base = d.stream->time_base
- Init
e.codecx->time_base = d.stream->time_base
- Set
d.rgbframe->pts = packet.dts
inread_frame
- Set any of pts/dts/duration in
packet
inwrite_frame
to anything. - Result : Warnings about packet timestamps not set. Encoder seems to reset all packet timing fields to 0, so none of this has any effect.
- Init
-
- Init
e.stream->time_base = d.stream->time_base
- Init
e.codecx->time_base = d.stream->time_base
- I found these fields,
pkt_pts
,pkt_dts
, andpkt_duration
inAVFrame
after reading this post, so I tried copying those all the way through tooutframe
. - Result : Really had my hopes up, but ended up with same results as attempt 3 (packet timestamp not set warning, incorrect results).
- Init
I tried various other hand-wavey permutations of the above and nothing worked. What I want to do is create an output file that plays back with the same timing and frame rate as the input (29.97 constant frame rate in this case).
So how do I do this ? Of the zillions of timing related fields here, what do I do to make the output be the same as the input ? And how do I do it in such a way that handles arbitrary video input formats that may store their time stamps and time bases in different places ? I need this to always work.
For reference, here is a table of all the packet and frame timestamps read from the video stream of my test input file, to give a sense of what my test file looks like. None of the input packet pts’ are set, same with frame pts, and for some reason the duration of the first 108 frames is 0. VLC plays the file fine and reports the frame rate as 29.9700089 :
- Table is here since it was too large for this post.
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ffmpeg - convert image sequence to video with reversed order
8 avril 2017, par 0__Looking at the docs, it is not apparent to me whether ffmpeg would allow me to convert an image sequence to a video in reverse order, for example using this sequence :
frame-1000.jpg
frame-999.jpg
frame-998.jpg
...
frame-1.jpgIs it possible to give a "step direction" for the frame indices ?