Search Results

Search found 537 results on 22 pages for 'ffmpeg'.

facebook twitter digg google delicious technorati stumbleupon myspace wordpress linkedin gmail igoogle windows live tumblr viadeo yahoo buzz yahoo mail yahoo bookmarks favorites email print

Page 15/22 | < Previous Page | 11 12 13 14 15 16 17 18 19 20 21 22  | Next Page >

  • avconv - not working with movflags

    - by MarKa
    With the Parameter "-movflags frag_custom" my avconv says [mp4 muxer @ 0x8d05f60] [Eval @ 0x7fffcc763f00] Undefined constant or missing '(' in 'frag_custom' [mp4 muxer @ 0x8d05f60] Unable to parse option value "frag_custom" [mp4 muxer @ 0x8d05f60] Error setting option movflags to value frag_custom. But with avconv is compiled with a mp4 muxer, so whats the problem ? Version is avconv 0.8.1-4:0.8.1-0ubuntu1

    Read the article

  • avconv gets killed if mkv has subtitles

    - by Lukas Knuth
    What I'm trying to do is to take a movie (in an Matroska container), convert all audio tracks to AC3 and don't touch anything else. I'm using this line: avconv -i infile.mkv -map 0 -vcodec copy -scodec copy -acodec ac3 -ab 256k outfile.mkv This works fine, except when there are subtitles embedded. Then, after some time processing with no progress, avconv just "dies" (output shortened, these seem to be the interesting parts): [matroska,webm @ 0xf867a0] max_analyze_duration reached [matroska,webm @ 0xf867a0] Estimating duration from bitrate, this may be inaccurate ... Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' ... Stream #0.0(eng): Video: H264 / 0x34363248, yuv420p, 1280x536 [PAR 1:1 DAR 160:67], q=2-31, 1k tbn, 1k tbc (default) Stream #0.1(ger): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s (default) Stream #0.2(eng): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s Stream #0.3(ger): Subtitle: dvdsub (default) (forced) Metadata: title : forced Stream #0.4(ger): Subtitle: dvdsub Metadata: title : complete Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (dca -> ac3) Stream #0:2 -> #0:2 (dca -> ac3) Stream #0:3 -> #0:3 (copy) Stream #0:4 -> #0:4 (copy) Input stream #0:2 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 Input stream #0:1 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 frame= 2606 fps=1303 q=-1.0 size= 3kB time=107.36 bitrate= 0.2kbits/s ... frame=96141 fps=813 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s frame=96251 fps=810 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s ... frame=97015 fps=397 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s Getötet ["Killed", in English] I have no idea why this happens, as there is no error-output. I'd like to just copy the subtitles over, not touch them at all. If that won't work, they can be completely dropped.

    Read the article

  • Tool to bulk speed up/convert an audio file

    - by User1
    I want to listen to certain podcasts on my phone but I have two common problems: The audio is in some weird format (some don't play on my phone). The audio is slow. I want to use something like sox or avconv to bulk convert the files. Since this is just voice and going on a cell phone, small low-quality files would be best for me. I had some good success using avconv: avconv -i weird.wma normal.ogg Unforunately, this command creates an enormous ogg file and I can't get it play faster. Ideally, this particular file would play at 170% of the original speed.

    Read the article

  • Problem with cvCreateVideoWriter in OpenCv. Again )

    - by lyuba
    I know, that the issue had been widely discussed before, but after 5 hours of inefficient googling I guess I deserve to ask :) By the way, all such problems concerned earlier versions of OpenCV, so.. I've compiled fresh OpenCV 2.1. from source under Ubuntu 9.10. It works fine except of cvCreateVideoWriter, which returns null to the following request: CvVideoWriter *writer = cvCreateVideoWriter("video.avi", CV_FOURCC('M','J','P','G'), fps, size, 0); I've walked through the OpenCv folders - it even seems to have ffmpeg inside. I've also installed it on the system to make sure. I've changed CV_FOURCC('M','J','P','G') to -1 - all worthless. I would appreciate your help soo much! P.S. I've also explored some new way for writing videos here: http://opencv.willowgarage.com/documentation/cpp/reading_and_writing_images_and_video.html#videowriter But it fails to work as well, showing the mistake: error: no match for ‘operator>>’ in ‘writer >> frame’ My code is here: http://pastie.org/984734

    Read the article

  • x264 IDR access unit with a SPS and a PPS

    - by Gcoop
    Hi All, I am trying to encode video in h.264 that when split with Apples HTTP Live Streaming tools media file segmenter will pass the media file validator I am getting two errors on the split MPEG-TS file WARNING: Media segment contains a video track but does not contain any IDR access unit with a SPS and a PPS. WARNING: 7 samples (17.073 %) do not have timestamps in track 257 (avc1). After hours of research I think the "IDR" warning relates to not having keyframes in the right place on the segmented MPEG-TS file so in my ffmpeg command I set -keyint_min 1 to ensure keyframes where at every frame, but this didn't work. Although it would be great to get an answer, if anyone can shed any light on what a "IDR access unit with a SPS and a PPS" is or what the timestamps warning means I would be very grateful, thanks.

    Read the article

  • Rails paperclip problem

    - by palani
    I have uploaded the video into my rails application by using thoughtbot-paperclip then the video is converted into "flv" format by using ffmpeg. For your reference here I specified some of my model sample code: model.rb: has_attached_file :source,:styles => {:thumb => "137x85>" } If i specified :url or :path option it doesn't worked correctly. In my view I played my video by using the following line: <%= @model.source.url.gsub(/\?.*/,'')%> If i use <%= @model.source.url%>, the video is not played. When do the puts for video url it shows me the video URL as /source/original/sample/sample.fly?22000009. I knew that the last portion is a timestamp, but i want to use <%= @model.source.url%>. What's my mistake here can any one correct me please?

    Read the article

  • Best cheap GUI for converting to f4v

    - by Ben
    Hi, I need to give one client the ability to convert some videos to f4v h264 before loading it up to an AIR app. I normally doing with the Adobe Media Encoder CS4 but that only ships with adobe products (you can't download it as a free standalone app - which, start rant is odd, you'd think they would push the format - Microsoft's competing expresssion encoder is free end rant) Anyway, I need to get a (hopefully not too expensive, but willing to pay for it) good 3rd part app that can take any video and convert it to an f4v. Can you suggest any? Everything i've found is horrible or jammed with ads and crap. What would you use? Any suggestion? (please don't just say ffmpeg - I know it can do it but we need to good simple GUI) Thanks!

    Read the article

  • Linux application that bundles multiple incoming audio and video streams into one container file?

    - by StackedCrooked
    I've been assigned to implement a video on-demand service for a local university. Different aspects of the lectures (video, audio, screen cast, white board) will be recorded. During a lecture all these data streams arrive at one Linux server. This server should transcode and bundle all these streams into one container (Matroska) file. My options seem to be: Write a GStreamer application do something with FFMPEG do something with VLC ...? Has anyone done something similar in the past? Can you recommend something? Edit For those interested, here are a few of my findings: Matroska is not a good format for streaming (it's possible, but it's not its primary intent) For Flash streaming you can use MPEG4 If you want to combine different videos into one video where each subvideo occupies a rectangular portion of the total screen, then this GStreamer script is useful (I found it on this blog post). Desktop capture works fine with VLC

    Read the article

  • How can I concatenate two mp3 files with different bit rates

    - by Scott
    I have FFmpeg installed on my linux web server. When I execute the following code, I have intermittent results. I think I have figured out that the MP3s do not compile when they have different bitrates. exec ('cat '. $pair['source_file'] . ' ' . $pair['translated_word_file'] . '>' . $temp_mp3); I might have found some articles online that reference taking them apart and then bundling them back together at a consistent bitrates. I have confirmed that this won't really work with basic "cat" function and that "sox" can be used IF they have the same sample rate. The issue now becomes "What is the best way to get them to the same sample rate?"

    Read the article

  • Conversion from Iphone Core Surface RGB Frame into ffmepg AVFarme

    - by Sridhar
    Hello, I am trying to convert Core Surface RGB frame buffer(Iphone) to ffmpeg Avfarme to encode into a movie file. But I am not getting the correct video output (video showing colors dazzling not the correct picture) I guess there is something wrong with converting from core surface frame buffer into AVFrame. Here is my code : Surface *surface = [[Surface alloc]initWithCoreSurfaceBuffer:coreSurfaceBuffer]; [surface lock]; unsigned int height = surface.height; unsigned int width = surface.width; unsigned int alignmentedBytesPerRow = (width * 4); if (!readblePixels) { readblePixels = CGBitmapAllocateData(alignmentedBytesPerRow * height); NSLog(@"alloced readablepixels"); } unsigned int bytesPerRow = surface.bytesPerRow; void *pixels = surface.baseAddress; for (unsigned int j = 0; j < height; j++) { memcpy(readblePixels + alignmentedBytesPerRow * j, pixels + bytesPerRow * j, bytesPerRow); } pFrameRGB->data[0] = readblePixels; // I guess here is what I am doing wrong. pFrameRGB->data[1] = NULL; pFrameRGB->data[2] = NULL; pFrameRGB->data[3] = NULL; pFrameRGB->linesize[0] = pCodecCtx->width; pFrameRGB->linesize[1] = 0; pFrameRGB->linesize[2] = 0; pFrameRGB->linesize[3] = 0; sws_scale (img_convert_ctx, pFrameRGB->data, pFrameRGB->linesize, 0, pCodecCtx->height, pFrameYUV->data, pFrameYUV->linesize); Please help me out. Thanks, Raghu

    Read the article

  • split video (avi/h264) on keyframe

    - by m.sr
    Hallo. I have a big video file. ffmpeg, tcprobe and other tool say, it is an h264-stream in an AVI-container. Now i'd like to cut out small chunks form the video. Problem: The index of the video seam corrupted/destroyed. I kind of fixed this via mplayer -forceidx -saveidx <IndexFile> <BigVideoFile>. The Problem here is, that I'm now stuck with mplayer/mencoder which can use this index file via -loadidx <IndexFile>. I have tried correcting the index like described in man aviindex (mplayer -frames 0 -saveidx mpidx broken.avi ; aviindex -i mpidx -o tcindex ; avimerge -x tcindex -i broken.avi -o fixed.avi), but this didn't fix my video - meaning that most tools i've tested couldn't search in the video file. Problem: I sut out parts of the video via following command: mencoder -loadidx in.idx -ss 8578 -endpos 20 -oac faac -ovc x264 -sws 9 -lavfopts format=mp4 -x264encopts <LotsOfOpts> -of lavf -vf scale=800:-10,harddup in.avi -o out.mp4. Now here the problem is, that some videos are corrupted at the beginning. I think this is because the fact, that i do not necessarily cut at keyframe. Questions: What is the best way to fix the index of an avi "inline" so that every tool can again work as expected with it? How can i split at the keyframes? Is there an mencoder-option for this? Are Keyframes coming in a frequency? How to find out this frequency? (So with a bit of math it should be possible to calculate the next keyframe and cut there) Is ther perhaps some completely other way to split this movie? Doing it by hand is no option, i've to cut out 1000+ chunks ... Thanks a lot!

    Read the article

  • Sudden issues reading uncompressed video using opencv

    - by JohnSavage
    I have been using a particular pipeline to process video using opencv to encode uncompressed video (fourcc = 0), and opencv python bindings to then open and work on these files. This has been working fine for me on OpenCV 2.3.1a on Ubuntu 11.10 until just a few days ago. For some reason it currently is only allowing me to read the first frame of a given file the first time I open that file. Further frames are not read, and once I touch the file once with my program, it then cannot even read the first frame. More detail: I created the uncompressed video files as follows: out_video.open(out_vid_name, 0, // FOURCC = 0 means record raw fps, Size(640, 480)) Again, these videos worked fine for me until about a week ago. Now, when I try to open one of these I get the following message (from what I think is ffmpeg): Processing video.avi Using network protocols without global network initialization. Please use avformat_network_init(), this will become mandatory later. [avi @ 0x29251e0] parser not found for codec rawvideo, packets or times may be invalid. It reads and displays the first frame fine, but then fails to read the next frame. Then, when I try to run my code on the same video, the capture still opens with the same message as above. However, it cannot even read the very first frame. Here is the code to open the capture: self.capture = cv2.VideoCapture(filename) if not self.capture.isOpened() print "Error: could not open capture" sys.exit() Again, this part is passed without any issue, but then the break happens at: success, rgb = self.capture.read() if not success: print "error: could not read frame" return False This part breaks at the second frame on the first run of the video file, and then on the first frame on subsequent runs. I really don't know where to even begin debugging this. Please help!

    Read the article

  • MPlayer does not work

    - by Soham Pal
    Using the xubuntu desktop, on Ubuntu Raring updated from Quantal. MPlayer never really worked. No video, no audio, nothing. I really can't be any more helpful, so here's the log: petey@home-pc:~$ mplayer "/home/petey/Downloads/Polar Bear Cafe (480p)HorribleSubs]/[HorribleSubs] Polar Bear Cafe - 01 [480p].mkv" MPlayer SVN-r35984-4.7 (C) 2000-2013 MPlayer Team Playing /home/petey/Downloads/Polar Bear Cafe (480p)[HorribleSubs]/[HorribleSubs] Polar Bear Cafe - 01 [480p].mkv. libavformat version 55.0.100 (internal) libavformat file format detected. [lavf] stream 0: video (h264), -vid 0 [lavf] stream 1: audio (aac), -aid 0 [lavf] stream 2: subtitle (ass), -sid 0 VIDEO: [H264] 848x480 0bpp 23.810 fps 0.0 kbps ( 0.0 kbyte/s) Clip info: creation_time: 2012-04-05 21:36:10 Load subtitles in /home/petey/Downloads/Polar Bear Cafe (480p)[HorribleSubs]/ Can't open /dev/fb0: Permission denied [fbdev2] Can't open /dev/fb0: Permission denied VO: [v4l2] No such file or directory vo_cvidix: No vidix driver name provided, probing available ones (-v option for details)! [cyberblade] Error occurred during pci scan: Operation not permitted [mach64] Error occurred during pci scan: Operation not permitted [mga] Error occurred during pci scan: Operation not permitted [mga] Error occurred during pci scan: Operation not permitted [nvidia_vid] Error occurred during pci scan: Operation not permitted [pm3] Error occurred during pci scan: Operation not permitted [radeon] Error occurred during pci scan: Operation not permitted [rage128] Error occurred during pci scan: Operation not permitted [s3_vid] Error occurred during pci scan: Operation not permitted [SiS] Error occurred during pci scan: Operation not permitted [unichrome] Error occurred during pci scan: Operation not permitted [VO_SUB_VIDIX] Couldn't find working VIDIX driver. ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family libavcodec version 55.0.100 (internal) Selected video codec: [ffh264] vfm: ffmpeg (FFmpeg H.264) ========================================================================== ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 44100 Hz, 2 ch, floatle, 0.0 kbit/0.00% (ratio: 0->352800) Selected audio codec: [ffaac] afm: ffmpeg (FFmpeg AAC (MPEG-2/MPEG-4 Audio)) ========================================================================== [AO OSS] audio_setup: Can't open audio device /dev/dsp: No such file or directory DVB card number must be between 1 and 4 AO: [null] 44100Hz 2ch floatle (4 bytes per sample) Starting playback... Movie-Aspect is 1.78:1 - prescaling to correct movie aspect. VO: [null] 848x480 = 854x480 Planar YV12 A: 4.7 V: 4.7 A-V: 0.002 ct: 0.083 0/ 0 22% 0% 0.5% 0 0 MPlayer interrupted by signal 2 in module: sleep_timer A: 4.7 V: 4.7 A-V: 0.001 ct: 0.083 0/ 0 21% 0% 0.5% 0 0 Exiting... (Quit)

    Read the article

  • How do I set libavcodec to use 4:2:2 chroma when encoding MPEG-2 4:2:2 profile?

    - by Mike Pollitt
    I have a project using libavcodec (ffmpeg). I'm using it to encode MPEG-2 video at 4:2:2 Profile, Main Level. I have the pixel format PIX_FMT_YUV422P selected in the AVCodecContext, however the video output I'm getting has all the colours wrong, and looks to me like the encoder is incorrectly reading the buffers as though it thinks it is 4:2:0 chroma rather than 4:2:2. Here's my codec setup: // // AVFormatContext* _avFormatContext previously defined as mpeg2video // // // Set up the video stream for output // AVVideoStream* _avVideoStream = av_new_stream(_avFormatContext, 0); if (!_avVideoStream) { err = ccErrWFFFmpegUnableToAllocateStream; goto bail; } _avCodecContext = _avVideoStream->codec; _avCodecContext->codec_id = CODEC_ID_MPEG2VIDEO; _avCodecContext->codec_type = CODEC_TYPE_VIDEO; // // Set up required parameters // _avCodecContext->rc_max_rate = _avCodecContext->rc_min_rate = _avCodecContext->bit_rate = src->_avCodecContext->bit_rate; _avCodecContext->flags = CODEC_FLAG_INTERLACED_DCT; _avCodecContext->flags2 = CODEC_FLAG2_INTRA_VLC | CODEC_FLAG2_NON_LINEAR_QUANT; _avCodecContext->qmin = 1; _avCodecContext->qmax = 1; _avCodecContext->rc_buffer_size = _avCodecContext->rc_initial_buffer_occupancy = 2000000; _avCodecContext->rc_buffer_aggressivity = 0.25; _avCodecContext->profile = 0; _avCodecContext->level = 5; _avCodecContext->width = f->GetWidth(); // f is a private Frame class with width, height properties etc. _avCodecContext->height = f->GetHeight(); _avCodecContext->time_base.den = 25; _avCodecContext->time_base.num = 1; _avCodecContext->gop_size = 12; _avCodecContext->max_b_frames = 2; _avCodecContext->pix_fmt = PIX_FMT_YUV422P; if (_avFormatContext->oformat->flags & AVFMT_GLOBALHEADER) { _avCodecContext->flags |= CODEC_FLAG_GLOBAL_HEADER; } if (av_set_parameters(_avFormatContext, NULL) < 0) { err = ccErrWFFFmpegUnableToSetParameters; goto bail; } // // Set up video codec for encoding // AVCodec* _avCodec = avcodec_find_encoder(_avCodecContext->codec_id); if (!_avCodec) { err = ccErrWFFFmpegUnableToFindCodecForOutput; goto bail; } if (avcodec_open(_avCodecContext, _avCodec) < 0) { err = ccErrWFFFmpegUnableToOpenCodecForOutput; goto bail; } A screengrab of the resulting video frame can be seen at http://ftp.limeboy.com/images/screen_grab.png (the input was standard colour bars). I've checked by outputting debug frames to TGA format at various points in the process, and I can confirm that it is all fine and dandy up until the point that libavcodec encodes the frame. Any assistance most appreciated! Cheers, Mike.

    Read the article

  • Video encoding Help

    - by Pedro
    Hi guys, I'm doing one research on video encoding tools for flv. I tested flvtool2 and Yamddi, but I'm losing lots of quality of video. Does anyone recommend any other tool or algorithm to keep the maximum quality of the movie in flv? Regards, Pedro

    Read the article

  • Mobile Video Detection

    - by aaroninfidel
    Hi, I'm using DeviceAtlas to detect mobile phones, I was wondering if anyone had some good resources in terms of standard codecs, video dimensions that are used and how you go about serving video to mobile devices. Thanks! -Aaron

    Read the article

  • Understanding PTS and DTS in video frames

    - by theateist
    I had fps issues when transcoding from avi to mp4(x264). Eventually the problem was in PTS and DTS values, so lines 12-15 where added before av_interleaved_write_frame function: 1. AVFormatContext* outContainer = NULL; 2. avformat_alloc_output_context2(&outContainer, NULL, "mp4", "c:\\test.mp4"; 3. AVCodec *encoder = avcodec_find_encoder(AV_CODEC_ID_H264); 4. AVStream *outStream = avformat_new_stream(outContainer, encoder); 5. // outStream->codec initiation 6. // ... 7. avformat_write_header(outContainer, NULL); 8. // reading and decoding packet 9. // ... 10. avcodec_encode_video2(outStream->codec, &encodedPacket, decodedFrame, &got_frame) 11. 12. if (encodedPacket.pts != AV_NOPTS_VALUE) 13. encodedPacket.pts = av_rescale_q(encodedPacket.pts, outStream->codec->time_base, outStream->time_base); 14. if (encodedPacket.dts != AV_NOPTS_VALUE) 15. encodedPacket.dts = av_rescale_q(encodedPacket.dts, outStream->codec->time_base, outStream->time_base); 16. 17. av_interleaved_write_frame(outContainer, &encodedPacket) After reading many posts I still do not understand: outStream->codec->time_base = 1/25 and outStream->time_base = 1/12800. The 1st one was set by me but I cannot figure out why and who set 12800? I noticed that before line (7) outStream->time_base = 1/90000 and right after it it changes to 1/12800, why? When I transcode from avi to avi, meaning changing the line (2) to avformat_alloc_output_context2(&outContainer, NULL, "avi", "c:\\test.avi"; , so before and after line (7) outStream->time_base remains always 1/25 and not like in mp4 case, why? What is the difference between time_base of outStream->codec and outStream? To calc the pts av_rescale_q does: takes 2 time_base, multiplies their fractions in cross and then compute the pts. Why it does this in this way? As I debugged, the encodedPacket.pts has value incremental by 1, so why changing it if it does has value? At the beginning the dts value is -2 and after each rescaling it still has negative number, but despite this the video played correctly! Shouldn't it be positive?

    Read the article

  • Qt SDk 4.6.2 on mac os x: invoke ffmpeg ??

    - by varunmagical
    Hello, I am writing an FFmpeg frontend in Qt & testing it on linux, windows & Mac. (FFmpeg is a popular command line tool for video operations) My project is working well on Linux & windows but I cannot invoke FFmpeg on Mac! I have compiled it from svn source on Mac & I have ensured that it is working properly by running it in Mac terminal. In my project, I have created a widget that shows FFmpeg output during conversion, but on mac, It always stays blank. Need help!

    Read the article

  • Is there a best practice for concatenating MP3 Files, adjusting sample rates to match, while preserving original files?

    - by Scott
    Hello overflow community! Does anyone know if there is a "best practice" to concatenate mp3 files to create new files, while preserving the original files? I am working on a CentOS Linux machine, in command line. I will eventually call the command line from a PHP script. I have been doing research and I have come up with a process that I think could work. It combines general advice from different forums, blogs, and sources like this one. So here I go: Create a temporary folder Loop through files to create a new, converted copy, of file into a "raw" format (which one, I don't know. I didn't know "raw" files existed before too long ago. I could use some suggestions on this) Store the path to the temporary files, in the temporary folder, and then loop through the files to concatenate them and then put the new merged file the final "processed directory" Delete the contents of the temporary file with the temporary raw files inside. Convert the final file from "raw" to mp3 and enjoy the finished result I'm thinking that this course of action might be best because I can't necessarily control the quality of the original "source" mp3s. The only other option I could think of would be to create a script that would perform a similar process upon files being added to the system leaving only the files with the "proper" format and removing the original "erroneous" file. Hopefully you can see that I have put some thought into this and that I'm trying to leverage the collective knowledge of this community to choose the best direction. Perhaps there is a better path that I could take? By concatenate, I mean to join together in sequence to create a new audio file from the "concatenated files."

    Read the article

  • Echo cancellation

    - by Jorg B Jorge
    Can any of you suggest a good and stable echo cancelation package (gnu or not) to be linked with my videoconference application (C/C++) (Windows / Linux / MacOSX) ? My application should be freeware, so i do not want to pay for each user who download the app.

    Read the article

facebook twitter digg google delicious technorati stumbleupon myspace wordpress linkedin gmail igoogle windows live tumblr viadeo yahoo buzz yahoo mail yahoo bookmarks favorites email print

< Previous Page | 11 12 13 14 15 16 17 18 19 20 21 22  | Next Page >