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Autres articles (18)
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Supporting all media types
13 avril 2011, parUnlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)
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HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...) -
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.
Sur d’autres sites (3329)
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FFMPEG - Struggling to find correct input audio codec parameters on macOS
20 septembre 2024, par XaviI am trying to read my external stereo microphone with ffmpeg within my Qt Windows+macOs application, but I am struggling to obtain consistent correct input codec parameters on macOs. My findings and suspicions so far :


The code I'm using in macOs is the following, where everything returns a successful return code :


avdevice_register_all();
 
 //macOs only, the same code in windows looks for "dshow" 
 const AVInputFormat *inputFormat = av_find_input_format("avfoundation");
 
 AVFormatContext* inputFormatContext;
 avformat_open_input(&inputFormatContext, inputDevice, inputFormat, NULL);

 avformat_find_stream_info(inputFormatContext, NULL);
 
 //... allocate the codec context for the single input stream and
 // copy the parameters from the stream to the context




In my standalone minimal reproducer this always results on the codec ID of the single stream being AV_CODEC_ID_PCM_F32LE, in both macOS and Windows. When I integrate this code in my Qt application on Windows, I get the same result. However, on macOS, most of the times results in the codec id of the stream being AV_CODEC_ID_PCM_S16LE (via AV_CODEC_ID_FIRST_AUDIO) and sometimes AV_CODEC_ID_PCM_F32LE. Both sample formats are supported by my microphone.


AV_CODEC_ID_PCM_F32LE always results in a correct output. AV_CODEC_ID_PCM_S16LE results on buzzy noisy audio slowed down to 0.5x, and If in this case I decode with AV_CODEC_ID_PCM_F32LE instead of copying the codec parameters from the stream, the output sounds correct again.


I am trying to write generic code, so while enforcing the AV_CODEC_ID_PCM_F32LE codec works, I'd rather understand what is happening.


What am I missing ? Is Qt interacting in some way that I can't think of ? I am compiling and linking my own ffmpeg libraries (6.1.1) and not using Qt's ones.


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matroskaenc : implement CueRelativePosition
23 juillet 2013, par Bernie Habermeiermatroskaenc : implement CueRelativePosition
This is a minimal change to matroskaenc that implements CueRelativePosition in the output.
Most players will probably ignore this additional information, but it is in the
matroska spec, and it’d be nice to be able to make use of it.Signed-off-by : Bernt Habermeier <bernt@wulfram.com>
Tested-by : wm4 <nfxjfg@googlemail.com>
Signed-off-by : Michael Niedermayer <michaelni@gmx.at> -
Ffmpeg Android - Minimum binary size to convert WAV to MP3
22 novembre 2020, par timsonThe only thing I want to do is convert wav files to mp3 inside my Android application.


I am currently using https://github.com/tanersener/mobile-ffmpeg and with audio-release everything is working fine. As the lib size is about 40 MB and I only need a single command, I'd like to build my own .aar file as described in the Wiki to reduce the application size.


I edited the
android-ffmpeh.sh ./configure:


--disable-everything \ 
--enable-pthreads \
--enable-avcodec \
--enable-avformat \
--enable-swresample \
--enable-avfilter \
--enable-libmp3lame \
--enable-parser=mpegaudio \
--enable-demuxer=mp3,wav,pcm_s16le \
--enable-muxer=mp3,wav,pcm_s16le \
--enable-decoder=pcm*,mp3*,wav,pcm_s16le \
--enable-encoder=pcm*,pcm_s16le,wav,mp3,libmp3lame \
--enable-filter=aresample \
--enable-protocol=file \



and then ran
./android.sh -l --enable-lame --enable-libiconv


In my Android app FFmpeg loads but the conversion doesn't succed with following error :


E/mobile-ffmpeg: [AVFilterGraph @ 0x7209dfec40] No such filter: 'anull'
E/mobile-ffmpeg: Error reinitializing filters!
E/mobile-ffmpeg: Failed to inject frame into filter network: Invalid argument
E/mobile-ffmpeg: Error while processing the decoded data for stream #0:0
I/mobile-ffmpeg: Conversion failed!



Does anyone know what I'm missing or another config to build a minimal size binary for this.
Any help is highly appreciated !