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January 2015

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ffmpeg-devel.log.20150120
by burek 21 Jan '15

21 Jan '15

[02:05] <cone-663> ffmpeg.git 03Michael Niedermayer 07master:e069a4bf2082: avcodec/snowdec: set AVFrame.pict_type correctly [02:05] <cone-663> ffmpeg.git 03Michael Niedermayer 07master:7873f9aa727b: avcodec/snowdec: AVMotionVector support [02:08] <kierank> michaelni: can you revert fe439c20698fc4992b9373cb068efb438fe1a098 [02:46] <michaelni> kierank, why? have you implemented decoding of that sample so this wont be needed anymore ? [02:46] <michaelni> or is there some other issue with that commit ? [02:46] <kierank> it breaks normal samples [02:46] <kierank> https://trac.ffmpeg.org/ticket/4274 [02:57] <k_sze[work]> How does frame-dropping work when dealing with DirecShow H.264 video sources? [02:57] <k_sze[work]> *DirectShow [03:11] <cone-663> ffmpeg.git 03Michael Niedermayer 07master:ff5b9a1cbe96: Revert "avcodec/h264: also show frames with missing fields when CODEC_FLAG2_SHOW_ALL is set" [03:36] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:e9ddf726aa72: avformat/segment: Use av_freep() avoid leaving stale pointers in memory [03:36] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:a6d59978a00e: avformat/cdxl: Fix integer overflow of image_size [03:36] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:1c983ee2c10e: avformat/flvdec: do not inject dts=0 metadata packets which failed to be parsed into a new data stream [03:36] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:ef33242c2ac5: avformat/flvdec: Increase string array size [03:36] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:92a36a4e78e6: avfilter/vf_sab: fix filtering tiny images [03:36] <cone-663> ffmpeg.git 03wm4 07release/2.2:0d481efb7b81: avcodec/dvdsubdec: fix out of bounds accesses [03:36] <cone-663> ffmpeg.git 03Dale Curtis 07release/2.2:57710c3646d4: mov: Avoid overflow with mov_metadata_raw() [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:20a03d5c9323: avformat/mov: fix integer overflow in mov_read_udta_string() [03:37] <cone-663> ffmpeg.git 03Dale Curtis 07release/2.2:7c270a5e3b60: mov: Fix negative size calculation in mov_read_default(). [03:37] <cone-663> ffmpeg.git 03Paul B Mahol 07release/2.2:bebe3d35f362: lavfi: check av_strdup() return value [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:8abe459ac6ce: avformat/matroskadec: Use av_freep() to avoid leaving stale pointers in memory [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:3b17c1e13e45: avformat/utils: Clear pointer in ff_alloc_extradata() to avoid leaving a stale pointer in memory [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:3e3193f03c7d: cmdutils: Use 64bit for file size/offset related variable in cmdutils_read_file() [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:81bdaacb6556: avformat/mov: Fix mixed declaration and statement warning [03:37] <cone-663> ffmpeg.git 03James Almer 07release/2.2:6e2204b15292: configure: bump year [03:37] <cone-663> ffmpeg.git 03wm4 07release/2.2:b76dc8b5b8f7: avcodec/dvdsubdec: error on bitmaps with size 0 [03:37] <cone-663> ffmpeg.git 03wm4 07release/2.2:a4a87a7efd8b: avcodec/dvdsubdec: fix accessing dangling pointers [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:0684cd5d8c92: ffmpeg: Clear error message array at init. [03:37] <cone-663> ffmpeg.git 03wm4 07release/2.2:fc1fed62d9da: vp9: fix parser return values in error case [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:33a67961c8c6: avformat/movenc: workaround bug in "PathScale EKOPath(tm) Compiler Suite Version 4.0.12.1" [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:ecae610207e3: avcodec/hevc: Fix handling of skipped_bytes() reallocation failures [03:37] <cone-663> ffmpeg.git 03wm4 07release/2.2:b1959b1719c8: qpeg: avoid pointless invalid memcpy() [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:855ae45c5a64: avdevice: Use av_format_get_control_message_cb() [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:e91df69cf204: avfilter/vf_framepack: Check and update frame_rate [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:8babbdc9b15f: avcodec/flac_parser: fix handling EOF if no headers are found [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:3e8b73e65bb0: avformat/rmdec: Check for overflow in ff_rm_read_mdpr_codecdata() [03:37] <cone-663> ffmpeg.git 03Rob Sykes 07release/2.2:a40e6a214ec6: swresample/soxr_resample: fix error handling [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:f2fde86dae21: avformat/aviobuf: Check that avio_seek() target is non negative [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:30099bf0f014: avformat/utils: Do not update programs streams from program-less streams in update_wrap_reference() [03:37] <cone-663> ffmpeg.git 03wm4 07release/2.2:b9510b3274cf: lavu/frame: fix malloc error path in av_frame_copy_props() [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:3b9cb8d7d819: configure: create the tests directory like the doc directory [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:de434423910e: avformat/hdsenc: Use av_freep() avoid leaving stale pointers in memory [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:f5fd937fc5df: avformat/flvdec: Use av_freep() avoid leaving stale pointers in memory [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:7279be7c75c3: avcodec/vmdvideo: Check len before using it in method 3 [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:c351cd720a0c: avcodec/utvideodec: Fix handling of slice_height=0 [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:a0f4f12b6273: avformat/mov: check atom nesting depth [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:b8021620e21a: avformat/mov: fix integer overflow of size [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:226727f08fb0: swscale: increase yuv2rgb table headroom [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:50e04b3f3ca9: avcodec/h264: make the first field of H264Context an AVClass [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:9ce4686bfef8: avcodec/indeo3: use signed variables to avoid underflow [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:133dc77da96f: avcodec/dcadec: Check that the added xch channel isnt already there [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:13838647ca63: avcodec/hevc: clear filter_slice_edges() on allocation [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:25dc978bb1b5: avcodec/h264: Clear delayed_pic on deallocation [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:85b2396265f0: avcodec/hevc_ps: Check diff_cu_qp_delta_depth [03:37] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:f1d59a207fef: avcodec/h264: Check *log2_weight_denom [03:38] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:2528468e204c: avcodec/indeo3: ensure offsets are non negative [03:38] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:3769601fb6d8: Add FFMPEG_VERSION into the binary libs [03:38] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:9789612689d7: Makefile: add dependencies which require ffversion.h [03:45] <cone-663> ffmpeg.git 03Michael Niedermayer 07release/2.2:86a01362c0e4: Update for FFmpeg 2.2.12 [04:00] <cone-663> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n2.2.12': unknown revision or path not in the working tree. [04:00] <cone-663> Use '--' to separate paths from revisions [04:00] <cone-663> refs/tags/n2.2.12:HEAD: Revert "avcodec/h264: also show frames with missing fields when CODEC_FLAG2_SHOW_ALL is set" [08:24] <ubitux> damn there are a lot of good things to take in these subtitles files [08:24] <ubitux> maybe i should ask akira4 to work on these instead... [08:34] <ubitux> thx [08:34] <ubitux> oups [11:51] <ubitux> anshul_: ping [11:53] <ubitux> i might make akira4 (the subtitles opw student) work on the patchset from the Shan guy to extract the different features (i saw many sub text codecs improvements) [11:54] <ubitux> and i might make her work on the cc thing [11:54] <ubitux> please tell me if you have pending work or plans on this [12:12] <anshul_> ubitux: I am very busy these day, thanks to take load from me [12:13] <anshul_> I have completed those roullup as you said in mail [12:14] <anshul_> but I could not make rollup and reap_output apart, they were tightly coupled [12:15] <anshul_> just not I am working ISDB closed caption, I will put that later in our closed caption [12:17] <anshul_> I have also done, webVtt in hls if someone want to take it over, I would be happy to transfer it [13:56] <kierank> anshul_: your patch doesn't solve 1778 [14:41] <anshul_> kierank: reopened it, thanks for reminding [14:45] <anshul_> kierank: I looked at the specs of other codec, none of the codec data sheet specify anything about nut muxer, most of them do not specify anything other then ts [14:45] <anshul_> its the container people decide where they will put the codec data [14:45] <kierank> sigh [14:45] <kierank> you fail to understand [14:45] <anshul_> In Mp4 same closed caption is there in different stream [14:45] <kierank> i am vlc and i see captions in nut and in the elementary stream [14:46] <kierank> not standardised afaik [14:46] <kierank> apple proprietary [14:47] <anshul_> ok I got your point, do you know of any opensource closed caption encoder [14:47] <kierank> no [15:07] <cone-305> ffmpeg.git 03Anshul Maheshwari 07master:012559f2bb59: Cosmetic changes in cc_caption [15:07] <cone-305> ffmpeg.git 03Anshul Maheshwari 07master:88ba88e8877f: CHAR_DEBUG to av_dlog [15:07] <cone-305> ffmpeg.git 03Anshul Maheshwari 07master:d6d354254f68: Initilaize array to known size [15:07] <cone-305> ffmpeg.git 03Anshul Maheshwari 07master:1013c15f72f9: changed name of erase_display_memory to screen_changed [16:29] <LConstantin> Hi. Does anyone know if these vulnerabilities were patched in FFmpeg? http://www.binarysniper.net/2015/01/vlc-media-player-215-memory-corruption.… [16:30] <LConstantin> VLC devs said the vulnerabilities are in libavcodec [16:30] <LConstantin> https://trac.videolan.org/vlc/ticket/13390 [16:30] <LConstantin> https://trac.videolan.org/vlc/ticket/13389 [16:32] <wm4> LConstantin: they're ancient, so yes [16:33] <nevcairiel> who the fuck does security research on a OS for which official security patches have ceased [16:34] <LConstantin> do you happen to know the corresponding CVEs in FFmpeg advisories? [16:34] <nevcairiel> the issues could never be reproduced in ffmpeg [16:34] <LConstantin> just trying to determine exactly when they were patched and whether other projects that use FFmpeg are vulnerable because they haven't updated FFmpeg [16:34] <nevcairiel> if you can make it crash, please report a bug [16:35] <nevcairiel> apparently vlc also claims that updated versions of avcodec are not even affected [16:37] <wm4> LConstantin: well, take the samples (or ask for them), and try to reproduce? [16:37] <wm4> and no reason to distrust the word of vlc here [16:37] <thardin> don't this depends on how various programs are packaged? [16:39] <LConstantin> I'm a security reporter with the IDG News Service (PCWorld, Computerworld, etc.). Since FFmpeg has no official media contact I thought I'd ask here. I understand that the latest version of FFmpeg is not vulnerable. I was trying to determine when exactly it was patched (in which version) so that I can determine if other players that use older versions of FFmpeg might also be vulnerable, not just VLC. [16:41] <kierank> j-b: ^ [16:41] <thardin> you could use git-bisect to figure that out automagically [16:41] <nevcairiel> I stick to my opinion that a security researcher working on Windows XP nearly a year after official security patches for this OS have ceased should be discredited and his work tripple checked before making any claims. [16:41] <wm4> thardin: he doesn't/can't do any real work [16:41] <thardin> oh [16:41] <wm4> thardin: he's just here to annoy everyone and to make questionable statements about open source security [16:41] <kierank> wm4: ? [16:42] <wm4> that's what it sounds like [16:42] <kierank> wm4: right, because reporters generally go on irc to check their facts [16:45] <j-b> kierank: sorry, no, it's just bullshit. [16:45] <j-b> it doesnot even crash vLC [16:45] <kierank> I mean you should respond to the guy [16:45] <LConstantin> I didn't make any statements. I asked questions. [16:45] <thardin> hanlon's razor [16:46] <cone-305> ffmpeg.git 03Paul B Mahol 07master:fd33658f866d: lavfi: remove mp=ilpack [16:48] <kierank> durandal_1707: !! [16:49] <cone-305> ffmpeg.git 03Paul B Mahol 07master:be1b8b4da5e4: fix Makefile: remove ilpack [16:50] <ubitux> what about soft pulldown? [16:50] <wm4> doesn't it do nothing? [16:50] <durandal_1707> it should be ported [16:51] <iive> it turns soft telecine into hard telecine [16:51] <kepstin-laptop> soft pulldown is kind of useful, to fix dvds that were incorrectly soft-telecined :/ [16:51] <LConstantin> j-b I understand that you're probably upset with the media coverage of this issue reflecting badly on VLC. If it doesn't even crash VLC and the report is not valid why not take the opportunity to formulate a response and set the record straight. [16:52] <kepstin-laptop> (or rather, were incorrectly detelecined before being soft-telecined) [16:52] <wm4> durandal_1707: didn't you say the mp filter was broken? or was that in mpv only? [16:53] <durandal_1707> wm4: it was broken in mp but i belive i fixed that [16:53] <LConstantin> If it is a valid issue, just say what users need to do and that's it. Software has flaws, people accept that. It's how you respond to them that matters. [16:53] <wm4> durandal_1707: oh I see, field flags were not passed [16:54] <wm4> LConstantin: if it's really a libavcodec issue, I either expect it was reported to ffmpeg (and fixed), or vlc used an old version of libavcodec, and it was already fixed in a newer version [16:57] <j-b> LConstantin: because I have better things to do. [16:58] <LConstantin> wm4 Thanks. [17:06] <cone-305> ffmpeg.git 03Michael Niedermayer 07master:48895a23fc03: avdevice/alsa-audio-common: Use *alloc_array() [17:06] <cone-305> ffmpeg.git 03Michael Niedermayer 07master:081e117bfee0: avfilter/af_ashowinfo: Use av_realloc_array() [17:46] <cone-305> ffmpeg.git 03Paul B Mahol 07master:357c6a92bc94: avfilter/af_join: Use av_mallocz_array() [17:48] <ubitux> durandal_1707: please swap the args [17:49] <durandal_1707> why? [17:49] <ubitux> for consistency with the prototype [17:49] <ubitux> libavutil/mem.h:av_alloc_size(1, 2) static inline void *av_mallocz_array(size_t nmemb, size_t size) [17:55] <cone-305> ffmpeg.git 03Paul B Mahol 07master:f191c98e4d55: avfilter/af_join: swap av_mallocz_array() args for consistency with the prototype [18:00] <wm4> not like it matters...? [18:11] <akira4> ubitux, could you explain the WebVTT and CEA-608 work in a little more detail? [18:12] <ubitux> akira4: yep sure [18:12] <ubitux> akira4: did you see the 3 .c files? do you have them opened? [18:12] <akira4> just a sec. [18:13] <akira4> yep [18:14] <ubitux> look into srtenc.c [18:14] <ubitux> and webvtt thing [18:15] <ubitux> iirc there are some color tag support and stuff like [18:15] <ubitux> you need to extract that code and put it into libavcodec/webvttenc.c [18:15] <ubitux> which is currently a very basic encoder not supporting much tags [18:15] <ubitux> basically, try extracting all the webvtt stuff for now [18:16] <ubitux> the goal will be to extract all the features from these files [18:16] <ubitux> it seems there are quite a bunch of things, like ttml, cc, etc [18:16] <ubitux> for the cc (CEA-608 thing), you need to edit the cc decoder recently added from Anshul [18:17] <ubitux> so same thing, you need to find how they constructed the ASS markup into these .c [18:17] <ubitux> and add them in the appropriate place (in our cc decoder) [18:17] <ubitux> but start with the webvtt, it's easier [18:17] <akira4> hmm. alright. [18:18] <akira4> oh and about the merge that I did :( do you think I should get a new copy of the ffmpeg repo or work with the same thing? [18:19] <ubitux> akira4: branch out your work (git checkout -b dvdwhatever) [18:20] <ubitux> then reset your master on origin/master [18:20] <ubitux> your master must stay clean all the time [18:20] <ubitux> then branch out that clean master as a new base [18:20] <ubitux> it will look like this: [18:20] <ubitux> git checkout -b dvd [18:20] <ubitux> git checkout master [18:21] <ubitux> git reset --hard origin/master [18:21] <ubitux> git pull [18:21] <ubitux> git checkout -b webvtt [18:21] <akira4> got it. Thanks. [18:21] <ubitux> origin must point on the correct remote (the git @ source.ffmpeg.org) [18:21] <akira4> okay. [18:49] <akira4> ubitux, how would I test the encoder after making changes? Use a srt file and encode it into webvtt something like that? [18:49] <ubitux> yes, in fate you have a .srt file with colors and stuff like that [18:49] <ubitux> convert it to webvtt [18:50] <akira4> alright. [18:50] <ubitux> this will probably change the webvtt test btw [18:50] <akira4> yeah. [18:50] <ubitux> run a make fate-sub-webvttenc GEN=1 [18:50] <ubitux> and look at the diff [18:50] <ubitux> (after your modification) [18:51] <akira4> okay. [20:53] <wm4> how does ffmpeg handle locales? what does av_asprintf("%f", 1.5) return? [21:01] <rcombs> av_asprintf uses vsnprintf [21:01] <wm4> yeah [21:01] <wm4> but vsnprintf uses the current locale [21:01] <wm4> so it might result in "1,5" [21:02] <wm4> I guess libav* doesn't convert floats to strings that often as that it could be a problem... [21:03] <rcombs> something something C locales considered harmful [21:04] <wm4> certainly [21:04] <wm4> some frameworks (Qt) even set it on program start, and the recommended way to fix it is setting it back to C [21:05] <rcombs> libmediainfo fucks with it too [21:07] <rcombs> hey maybe C1Y will have locale-independent versions of everything [21:07] <wm4> yeah, let's add more shit, instead of fixing it [21:08] <wm4> standards committees are all about adding shit to existing shit [21:09] <rcombs> well making everything locale-independent would be nice but probably even less likely than new standardized locale-independent variants [21:09] <wm4> I want to see who the fuck even "needs" the locale shit (especially LC_NUMERIC, which breaks snprintf/strtod) [21:14] <cone-305> ffmpeg.git 03Michael Niedermayer 07master:ea3fc9fe6875: smoothstreamingenc: Add a missing "goto fail" [21:14] <cone-305> ffmpeg.git 03Michael Niedermayer 07master:9cbaf5849b21: Merge commit 'ea3fc9fe68752eb1b9078b28bea18a1e781d1c2c' [21:29] <cone-305> ffmpeg.git 03Martin Storsjö 07master:bba02479260d: libopenh264enc: Remove a workaround for silencing warnings about unused variables in the OpenH264 header [21:29] <cone-305> ffmpeg.git 03Michael Niedermayer 07master:87c4a80e563a: Merge commit 'bba02479260d0e7dec8c530a7e75a1c7aa53c06e' [21:55] <cone-305> ffmpeg.git 03Martin Storsjö 07master:440119b18836: libopenh264enc: Move a declaration of a variable into an ifdef [21:55] <cone-305> ffmpeg.git 03Michael Niedermayer 07master:6c9dfa5a1b45: Merge commit '440119b18836887d98c9e337c5911563bb43588c' [00:00] --- Wed Jan 21 2015

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ffmpeg.log.20150120
by burek 21 Jan '15

21 Jan '15

[03:58] <greves_> can anyone help me out here? I'm trying to add cover art with ffmpeg via pydub (a python wrapper for ffmpeg) [03:59] <greves_> i'm having trouble with the -map command(s?) and if i need some other metadata argument or something? [04:02] <c_14> https://ffmpeg.org/ffmpeg-formats.html#mp3 [04:02] <c_14> That's for mp3 [04:04] <greves__> yeah i was looking at that already, let me paste the command pydub is giving to ffmpeg [04:07] <greves__> ah i think i' mmissing metadata:s:v, what is that? [04:08] <c_14> It writes id3 tags. Probably so that the image is detected as cover art [04:12] <greves__> so what is the difference between -metadata and -metadata:s:v ? [04:12] <c_14> -metadata is global metadata, -metadata:s:v is stream metedata for the video stream [04:15] <greves__> http://pastebin.com/tqW8fwbf [04:16] <c_14> wait [04:16] <c_14> what [04:16] <greves__> thats what pydub is sending to the command line [04:16] <greves__> it all gets joined together in one string [04:16] <c_14> You only have 2 inputs, why are you mapping the 3rd and 4th input? [04:17] <greves__> oh i thought it was argument position [04:17] <greves__> so i should do map 0 1 ? [04:17] <c_14> yes [04:18] <greves__> no, still nothing [04:18] <c_14> what [04:19] <c_14> Why do you have id3v2_version 3 and 4 ? [04:19] <c_14> Is that even valid? [04:19] <greves__> oh right that's a pydub thing, let me fix that [04:20] <c_14> and the (Front) is being lost [04:21] <greves__> http://pastebin.com/zqYzXSDE [04:21] <greves__> oh do i actually need (Front)? [04:21] <c_14> You might? [04:22] <greves__> ok, i just tried that it doesn't work either [04:23] <greves__> actually since i started doing the -i cover.jpg -c copy -map x -map y, my mp3 output is blank, 0 bytes [04:24] <c_14> Is there any ffmpeg output? [04:24] <greves__> hmm i'm not sure how to access that since it's wrapped in pydub... [04:25] <greves__> maybe let me try running the command direct from terminal [04:30] <greves__> wow this is so hard lol i can't even type the command [04:33] <greves__> i think ffmpeg thinks my image is an audio stream [04:35] <greves__> http://pastebin.com/sJh0F5Sv [04:35] <greves__> does anything there help? [04:36] <c_14> >[mp3 @ 0000000002c20900] Invalid audio stream. Exactly one MP3 audio stream is required. [04:37] <c_14> Something's b0rked [04:37] <c_14> oh [04:37] <c_14> You can't put pcm into mp3 [04:37] <c_14> You'll need to encode it [04:37] <c_14> derp [04:37] <c_14> change -c copy to -c:a libmp3lame [04:38] <greves__> wow [04:38] <greves__> how did you do that [04:38] <greves__> it's like magic [04:39] <greves__> thanks a lot :) [04:40] <c_14> npnp [04:40] <greves__> how would i have learned that from the docs? [04:40] <greves__> like, which section? [04:41] <greves__> wow my other computer is going haywire [04:42] <c_14> That error message [04:43] <c_14> Then looking at the bottom near output0 stream 0 and noticing that it's pcm and not mp3 [04:43] <c_14> The Could not write header is usually also indicative. [07:08] <k_sze> Is there a particular (video) container file format that supports simultaneous reading and writing (as long as all reading and writing are sequential) [07:14] <BtbN> should work for mpegts and flv [07:14] <BtbN> depends a lot on the player though [07:14] <BtbN> if it just scans the length one at startup, it will stop playing once i reaches that point [07:14] <BtbN> +c [07:36] <tmh1999> Hi guys, I hope someone could help me :D. I want to rescale/resize a video. So as far as I understand, the procedure is : 1. Open video file, read video stream. 2. Read packet from that stream. 3. Decode the packet into frames. Resize those frames. 4. Encode those frames into packets. 5. Write those packets to new file. [07:36] <tmh1999> Is that correct? Thanks. [07:37] <BtbN> tmh1999, you want to do that with the ffmpeg cli tool, or by using libav*? [07:38] <tmh1999> BtbN : I am using ffmpeg API, none of them. [07:38] <tmh1999> oh [07:38] <tmh1999> sorry [07:38] <tmh1999> libav* [07:38] <tmh1999> misunderstanding your question [07:41] <tmh1999> as far as I know I use libavcodec, libavformat, libavswscale [07:41] <BtbN> basicaly you use libavformat to access the container, use libavcodec to decode it, scale it with libavfilter, encode it with libavcodec, and then mux it again with libavformat. [07:41] <BtbN> Ah, yes. libswscale for scaling. Not avfilter. But yes, you got the idea. [07:42] <BtbN> might still need libavfilter for stuff like deinterlacing. At least that's where i think that stuff is. [07:42] <tmh1999> So, my procedure above is right? [07:42] <BtbN> I'd personaly just use the ffmpeg cli tool. [07:43] <tmh1999> wow, that's new, deinterlacing? [07:43] <BtbN> If the source is interlaced, that would be a good idea, yes [07:43] <tmh1999> TBH I am new to both ffmpeg and video processing in general. [07:43] <BtbN> Or you need to use interlacing-aware scaling [07:44] <BtbN> With the ffmpeg cli tool scaling a video is one of the more simple tasks. [07:44] <BtbN> Doing that manualy via libav* requires quite a lot of work. [07:45] <tmh1999> Actually, I want to call ffmpeg/libav* API in my threads. If I use ffmpeg CLI like, system("ffmpeg ..."); that would be a process [07:48] <tmh1999> OK I will consider your opinion. [07:48] <tmh1999> Thanks :) [08:43] <Elirips> Hello everyone. I have an app that creates one frame per second. On Startup it creates an ffmpeg process and then pipes those frames to ffmpeg, instructing ffmpeg to create a rtmp stream with 5 fps from the single frame. [08:44] <Elirips> I use ffmpeg like this: 'ffmpeg.exe -framerate 1 -i - -an -c:v libx264 -preset fast -profile baseline -pix_fmt yuv420p -x264opts keyint=5 -r 5 -s 260x208 -f flv rtmp://<some-ip>/streams/name' [08:44] <Elirips> I'm wondering: What would happen if my input-source to ffmpeg does not provide 1 fps reliable? For example, if the input source has a delay, and does not provide a frame for 3 seconds. will ffmpeg then just write out 15 frames of the last image passed to ffmpeg? [08:45] <Elirips> or will ffmpeg 'do nothing' and wait for the next frame, as I told it that it will receive 1fps? [08:53] <sfan5> Elirips: iirc ffmpeg waits for input [08:53] <Elirips> sfan5: mmhh. this is bad for me :P [08:54] <Elirips> Is it possible to instruct ffmpeg something like 'if you do not get the framerate reported on the command-line, "pad" with the last frame? [08:54] <sfan5> Elirips: you could have a different thread that passes frames to ffmpeg [08:55] <sfan5> and that thread would then write the same frame if there isn't a new one [08:55] <Elirips> sfan5: you're right, that would maybe be the way to go [08:55] <Elirips> But I would need to implement quite some timing stuff [08:55] <sfan5> actually ffmpeg does not get any "frame rate", ffmpeg just (tries) to read as many frames as possible and encodes/copies/whatever-you-instructed-it-to-do them [08:56] <Elirips> but wat is then the effect of the parameter 'framerate 1' for the input? [08:57] <sfan5> that is how many frames the input has [08:57] <sfan5> that is needed to playback [08:58] <Elirips> I see. I will add some tests to see whats going on [09:00] <sfan5> Elirips: maybe my explanation was confusing; ffmpeg doesn't try to read n frames per second, it reads as many frames per second as it manages to process [09:01] <Elirips> sfan5: and if I specify -r 5 on the output, it will try to make an output of 5fps. So, in my case, where I send the frames using a pipe, I could ommit the -framerate 1 parameter, not? [09:01] <sfan5> depends [09:01] <sfan5> is the input in a container? [09:01] <Elirips> the input are single bitmaps [09:02] <sfan5> i don't know what ffmpeg defaults to in that case [09:02] <Elirips> anyway thx for the hints. I'll try to add a text displaying the current frame / timestamp on the video-created, then I can start testing more [09:02] <sfan5> Stream #0:0: Video: bmp, bgr24, 454x454, 25 tbr, 25 tbn, 25 tbc [09:02] <sfan5> seems like bitmaps have no framerate at all [09:03] <Elirips> Where did you find that? [09:03] <sfan5> which makes sense considering that they are static images [09:03] <sfan5> ffmpeg -i somebitmap.bmp [09:05] <Elirips> Hm, what would happen if I transcode one single bitmap into an infinite rtmp-stream? Well I can test that on my own [09:05] <Elirips> I regret that I was too lazy to implement the pipe to read from ffmpegs stderr to my app :P [09:07] <Elirips> If only ffmpeg could tell me what I'm doing wrong with my -vf [09:14] <Zombie> Does anyon know how to use de-interlacing filters to de-interlace video the way VLC does? [09:14] <sfan5> Zombie: what is "the way vlc does"? [09:15] <BtbN> vlc has its own deinterlacing code. It doesn't use ffmpeg. [09:16] <BtbN> It has a lot of diffrent deinterlacing algorithms. [09:16] <sfan5> last time i looked at man ffmpeg-filter i found a lot of ways to deinterlace video [09:16] <BtbN> yes, ffmpeg itself also has deinterlacing support [09:16] <BtbN> but vlc doesn't use that. [09:20] <Zombie> I'm trying to de-interlace a batch of VHS Tapes. [09:51] <Zombie> Hello? [10:04] <relaxed> Zombie: are you using the yadif filter? [10:04] <Zombie> I have done one video that way. [10:04] <Zombie> Is that optimal? [10:04] <relaxed> yes, how did the output look? [10:05] <Zombie> I tried it once, but there was an improvement. [10:38] <galba> hi, is there a simple command to unshake (stabilize) a video? [10:42] <BtbN> ffmpeg -i input.mp4 -filter:v deshake -c:a copy -c:v libx264 ..... output.mp4 [10:49] <galba> thanks, I just found vidstab, is it different? [10:52] <ubitux> it's way better [10:52] <galba> ok :) [10:54] <galba> do I have to use "-threads n" to use n processors or is it automatic? [10:55] <ubitux> it's automatic [11:56] <TonyIenciu> hello all, I'm trying to decode audio stream provided by internet radio stations, but I seem to get memory leak when reading the stream with av_read_frame. I'm also calling av_free_packet after each call... can someone help me, please? [11:58] <Mavrik> valgrind can help you alot more than we can. [12:30] <Ludowicus> Hello! Could someone tell me where I can find FFmpeg with x264 10bit included? [12:50] <relaxed> Ludowicus: http://johnvansickle.com/ffmpeg/ [12:50] <Ludowicus> relaxed, Thank you! [12:51] <relaxed> you're welcome [13:02] <Wader8> helllo [13:03] <Wader8> MP4 metadata doesn't work well from this wiki http://wiki.multimedia.cx/index.php?title=FFmpeg_Metadata [13:03] <Wader8> most of the documentation is oboslete [13:03] <Wader8> only title and genre worked [13:04] <Wader8> -metadata title="my title" -metadata author="my author" -metadata TIT3="my subtitle" -metadata comment="my comment" -metadata genre="my genre" -metadata comment="my comments" -metadata year="2014" -metadata year="2015" -metadata description="description" -metadata episode_id="episode id" [13:05] <Wader8> pretty much any ffmpeg documentation found on the web older than 3 years is useless [13:06] <Wader8> okay date="2014" works too [13:06] <Wader8> not year= [13:07] <Wader8> this is on Windows [13:07] <Wader8> comments also work [13:24] <theholyduck> Wader8, i found it sort of depressive how far the -sameq falacy [13:24] <theholyduck> spread [13:25] <theholyduck> i mean, the fact they had to resort to removing it due to all the people being wrong on the internet [13:34] <Wader8> well most programmers aren't known for good namings :p [13:34] <Wader8> I find it often very confusing, not pointing at this case but in general, usually programmers out of colleges just weird me out [13:35] <Wader8> you don't suppose to use advanced/misc stuff to mix with basic abbreviations [13:36] <Wader8> I don't who the heck came up with sameq, but glad it's removed :p [13:38] <Wader8> and I already had a big rant on ffmpeg documentation on a forum and what I think about that, a pity [13:39] <dv_> hi, I have a question about vc-1 . I keep reading about repeated frames in it. is this a vc-1 specific feature? can in vc-1 one input frame produce N output frames (that is, the same output frame repeated N times)? [13:40] <Wader8> it's really a semi-cult, where you have the industry folks who study ffmpeg for a job ofcourse they're going to understand all the commands, and other paid software that uses ffmpeg as the backend, seems like they want to intentionally make poor documentation [13:41] <Wader8> for example, I find linux namings extremely weird [13:42] <Wader8> folders with names "abt" ebt afv dda wsa wsdd sda sa q sys dsa odo sos rwd wa BIN BIN BIN BIN BIN [13:45] <Wader8> probably wouldn't be a problem if I devoted 5 years of my life trying to adapt to linux GUI and commands, anything's possible [13:46] <Wader8> im ranting again! oh well [14:20] <visiot> Solver: how ffmpeg get input frames in filtergraph [16:53] <Elirips> Anonyone a hint how to use -vf on windows? I'm stuck on the problem described here: http://stackoverflow.com/questions/28035800/ffmpeg-could-not-load-font-c-ca… [16:56] <Elirips> Also if I add c:\windows\fonts to the path, it fails to open [16:58] <Elirips> ah, just copy arial.ttf to the same dir as ffmpeg [20:01] <baran> hey, can somebody tell me values of CodecContext->level for MPEG-2 format? [20:01] <baran> are they determined by ffmpeg, or are they purely set by a user? [23:15] <Fraifrai> hello [23:16] <Fraifrai> I use ffmpeg to modify videos , for example to merge separated audio and videos files [23:16] <Fraifrai> or to extract parts [23:16] <Fraifrai> but my problem is new and I found no help onforums [23:16] <Fraifrai> I have a 1920x1152 videos which I would like to reduce to 1080 by STRIPPING 36 pixels top and bottom [23:17] <Fraifrai> without reencoding nor rescaling [23:17] <Fraifrai> is that possible with ffmeg [23:17] <Fraifrai> sorry for mypoor english I'm french [23:17] <Fraifrai> any help appreciated [23:18] <JEEB> <Fraifrai> without reencoding nor rescaling <- Impossible [23:19] <Fraifrai> you mean that i'm forced to loose quality ? [23:19] <Fraifrai> or may be there is an adapted filter i do not know [23:19] <JEEB> no, but you are forced to re-encode. you can encode with a lossless encoder and there will be no loss (and the file will be huge), or you encode with enough rate to make sure it looks good enough [23:20] <JEEB> but yes, re-encoding is inevitable [23:22] <Fraifrai> ok [23:22] <Fraifrai> which options would you use to convert from 1920x1152 to 1920x1080 deleting the 36 top and bottom lins? [23:22] <Fraifrai> croping? [23:23] <Fraifrai> with keep_aspect? [23:24] <Fraifrai> http://ffmpeg.org/ffmpeg-filters.html#crop [23:24] <Fraifrai> using Crop the input video central square ? [23:24] <Fraifrai> no [23:24] <Fraifrai> Crop the central input area with size 100x100: [23:25] <Fraifrai> crop=1920:1080 [23:25] <Fraifrai> ? [23:26] <foonix> crop=1920:1080:0:36 [23:28] <Fraifrai> it's running [23:28] <Fraifrai> thx a lot [23:28] <Fraifrai> it seems to work [23:28] <Fraifrai> Thansk a lot [23:39] <Fraifrai> hum [23:39] <Fraifrai> it seems that it changes encoders http://pastebin.com/DyxTx617 [00:00] --- Wed Jan 21 2015

1 0

ffmpeg-devel.log.20150119
by burek 20 Jan '15

20 Jan '15

[00:44] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:bf8bcd3b2b51: avdevice/openal-dec: Check the return code of av_new_packet() [03:21] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:a582705d1540: ffmpeg: Favor filter graph based duration over ost->frame_rate in fps code [03:47] <ac_slater> hey all, this might be a question for the other channel. Anyways, without looking to much into the 'plugin' system ffmpeg has in place, would it be possible to register plugins at runtime? ie - I want to create some demuxers/decoders but I'm not sure if I *HAVE* to compile them in [03:47] <ac_slater> thanks [04:00] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:90c98994f68b: ffmpeg: Reduce noisiness of frame clipping in rate convertion [04:02] <compn> ac_slater : theres a rfc on the devel list asking about creating such a feature [04:02] <compn> with a intro patch posted there [04:02] <compn> you may want to look at it [04:02] <ac_slater> compn: interesting, thanks mate [04:03] <compn> i think the thread title is dynamically loading something or other [04:03] <compn> should find it easily [04:14] <ac_slater> compn: thanks again mate. [08:11] <jamrial> nevcairiel: https://github.com/jamrial/FFmpeg/commits/hevc_intr [11:31] <k_sze> How does the DirectShow support in ffmpeg work? Does it just use a SampleGrabber to grab samples from a source filter and feed it to the encoder? [11:34] <ramiro> k_sze: did you take a look at the source code? [11:36] <ramiro> k_sze: it implements a custom grabber filter and uses that to output frames through libavdevice [11:57] <k_sze> ah, you wrote it. [11:57] <k_sze> so it doesn't implement ISampleGrabber at all. [12:02] <ramiro> k_sze: exactly [12:03] <k_sze> libAVFilter, right? [12:04] <k_sze> It's ... a struct. o.O funky [12:07] <ramiro> it's in C but it has to work like a C++ class [12:07] <k_sze> oh, right [12:07] <k_sze> I forgot it's C. [12:17] <k_sze> thanks for the tips. I'll be back later. [12:34] <wm4> I get a huge memory leak when seeking (probably when resetting the h264 decoder), does anyone observe the same? [12:34] <wm4> I get it with my own player, but also with ffplay [12:35] <wm4> (in ffplay, keep the mouse button down, and move the mouse between left and right window a few times) [12:38] <wm4> it happens with the ffmpeg from now, and also from a week ago [12:43] <nevcairiel> wm4: seems fine here [12:44] <wm4> time to bisect [12:44] <nevcairiel> define "huge" [12:45] <nevcairiel> memory usage seems stable here anyway seeking a stream [12:45] <wm4> quickly goes into the gigabytes [12:47] <nevcairiel> seems fine here in any case, so maybe something OS specific? but that would be odd [12:48] <nevcairiel> of course i tried with my own code [12:48] Action: nevcairiel grabs some ffplay [12:49] <wm4> only tried on Linux [12:50] <nevcairiel> can reproduce with ffplay, so i must be doing something "right" in my code. ;) [12:51] <wm4> do you reset the decoder with avcodec_flush (or whatever it was)? [12:56] <nevcairiel> of course [13:22] <wm4> and of course this whole merge bullshit makes bisecting very hard [13:23] <nevcairiel> using the ffbisect thing actually worked ok for me [13:23] <wm4> what was ffbisect again? [13:24] <nevcairiel> it automatically skips points on the libav branch, so you dont end up having to switch to avplay for testing [13:24] <nevcairiel> instead you'll end up on the merge that merged said commit from libav (if the libav change was faulty) [13:56] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:723c37d3b7b5: avfilter/vf_idet: reduce noisyness if the filter has been auto inserted [13:56] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:56a33b232cbb: avfilter/vf_idet: flush internal buffers on parameter changes [13:56] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:6c702c3c6314: avfilter/avfilter: allow idet like scale to have parameter changes [13:59] <ubitux> michaelni: is it really wise to insert an idet filter everytime we have a yuv420p output? [14:00] <ubitux> it doesn't sound like a good idea to me, unless it's really a no-op for progressive content? [14:00] <ubitux> i'm afraid of various slowdowns with complex filtergraph graph [14:01] <ubitux> it's probably a better idea to let users insert it themselves, but well.. [14:03] <michaelni> it should be a no-op for progressive content but if someone finds a slowdown with progressive i would be very interrested [14:04] <michaelni> the problem with letting the user do it is that the user #1 doesnt know she shoule do that and #2 doesnt know that some scale was auto inserted and needed an idet [14:05] <michaelni> also note idet is not auto inserted yet, the patch is still on the ML [14:43] <arwa> I am getting errors while using rsync : rsync: failed to connect to fate-suite.ffmpeg.org (192.190.173.45): Connection refused (111). [14:43] <arwa> I think this maybe because of the proxy settings. [14:44] <arwa> I tried out a lot of things to bypass the proxy, but I am unable to do so. [15:02] <wm4> kierank is going to get a stroke from all those nicolas posts [15:22] <kierank> wm4: i saw a wall of text and ignored [17:15] <wm4> man this thing sure took a while to find [17:15] <nevcairiel> i still wonder why mine didnt leak [17:16] <wm4> I'm wondering how the other frames are unreffed, but I'm too scared to ask [17:16] <wm4> nevcairiel: a single seek is still minor [17:16] <nevcairiel> i seeked a lot [17:16] <wm4> hm [17:17] <wm4> maybe it has to do with whether there are still frames in flight? [17:17] <iive> threading? [17:17] <wm4> part of why it took me so long was: memcpy(h, h1, offsetof(H264Context, intra_pcm_ptr)); [17:17] <wm4> truly evil [17:18] <wm4> this copies a big load of stuff from one context to another, including all refs [17:18] <wm4> maybe I didn't fix it properly [17:28] <nevcairiel> i shall remember to check for applied h264 patches soon then and make sure they are applied [17:30] <nevcairiel> (the second applied to my own version, i mean) [18:43] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:438c0a665af3: avcodec/xan: Use av_realloc_array() [18:43] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:f9a5a89bbfde: avcodec/xsubdec: Check av_mallocz() return codes [19:39] <kurosu> I have recently tried msys2, seeing how some people here seem to use it [19:40] <kurosu> the OS is identified as msys_nt61, causing a configure failure: incorrect installation? [19:41] <Daemon404> did you use the installer from msys2.github.io? [19:41] <kurosu> I think so [19:41] <Daemon404> (thats what i used) [19:41] <Daemon404> actually i was just going to send a patch to change msys links to msys2 links [19:43] <kurosu> it's on a computer I can't access atm, though, so what I'm writing isn't 100% sure [19:44] <Daemon404> ok [19:45] <kurosu> that was the 20141113 version though [19:45] <jamrial> kurosu: try --target-os=mingw32 [19:46] <kurosu> jamrial, will do that, thanks [19:47] <jamrial> also, assuming you're using their mingw toolchains, you should use mingw64_shell.bat or mingw32_shell.bat to start the environment [19:47] <jamrial> that way uname -s will output MINGW64_NT-6.1 or MINGW32_NT-6.1 respectively [19:48] <kurosu> oh right, I did run the msys shell [19:48] <jamrial> yeah [19:48] <kurosu> what's the toolchain to install then? the mingw64 one and not the msys one ? [19:48] <Daemon404> you want the mingw-w64 one [19:48] <jamrial> they also change the PATH enviroment var, to add either /mingw64 or /mingw32 [19:49] <kurosu> ok, I didn't make sure if the mingw-w64 was prefixed yet [19:49] <jamrial> Yeah, "pacman -S -l | grep mingw64" will show you all the mingw64 packages, for example [19:49] <kurosu> (hence the failure to detect initially) [19:50] <kurosu> well, actually, the fewer packages installed and picked up by ffmpeg, the shorter the compilation and link times, so I don't expect to install a lot of libs :D [19:50] <kurosu> thanks [19:51] <Daemon404> although i dislike pacman, it is roughly 1000x better than mingw-get ever was [19:52] <jamrial> check "pacman -S mingw-w64-x86_64-ffmpeg". it will try to install like 40 dependencies :p [19:52] <kurosu> I only work on ffmpeg as a hobby, so I tend to avoid such distractions as updates [19:53] <Daemon404> nobody sane uses a packaged ffmpeg [19:54] <kurosu> (and thus I only ran mingw-get on install) [19:57] <wm4> I use mxe-packaged ffmpeg sometimes [20:00] <jamrial> configure needs to be fixed anyway. if you use the mingw64 shell from msys2, uname -s will output mingw64 and configure only checks for "mingw32" [20:01] <kurosu> btw, does the package mingw-w64 toolchain accepts -m32, so as to be able to build both 64b and 32b versions of ffmpeg? [20:02] <kurosu> *packaged [20:03] <jamrial> i don't think so, but i haven't tried [20:03] <jamrial> they offer the x86 toolchain separately after all [20:06] <jamrial> you have both mingw-w64-x86_64-gcc and mingw-w64-i686-gcc [20:06] <kurosu> :/ [20:10] <kurosu> jamrial, do you know the package name for bc ? [20:10] <kurosu> might also be useful for the doc patch from Daemon404 [20:12] <jamrial> no, i couldn't find it in any package, so i had to download it from http://gnuwin32.sourceforge.net/packages/bc.htm [20:13] <jamrial> i assume they offer it. i just didn't bother much searching for it :p [20:15] <Daemon404> kurosu, i thouht about that too [20:20] <cone-871> ffmpeg.git 03wm4 07master:e0e33ebe6bd7: h264: unref leaking error recovery picture [20:22] <wm4> so apparently the fix was correct enough [20:24] <nevcairiel> apparently [20:39] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:5c1a8d3b7d33: avfilter/vf_idet: factorize av_frame_free(&idet->prev) [21:16] <aetasx> any of you guys know why when encoding to libfdk_aac, it widens mono to stereo output so that there's 2 channels duplicating the original? [22:17] <kurosu> Daemon404/jamrial, I don't think bc is packaged: pkgfile.exe -r ".*bc\.exe" returns nothing valid [22:18] <nevcairiel> bc is something thats missing often :d [22:18] <ubitux> bc is in the... bc package on archlinux [22:19] <ubitux> (where that pacman comes from) [22:19] <Daemon404> bc is one of those gnu projects with no real upstream iirc [22:19] <Daemon404> and no vcs at all [22:19] <ubitux> we should probably try to exec a zsh or something if bc is not found [22:19] <kurosu> I'll be waiting for ffbc then [22:19] <ubitux> (zsh supports the float arithmetic which we use bc for) [22:20] <ubitux> kurosu: actually... we have ffeval [22:20] <ubitux> that might work.. [22:20] <wm4> what's bc used for? [22:20] <Daemon404> float arith [22:20] <Daemon404> some sort of metric [22:20] <ubitux> tests/fate-run.sh: echo "scale=2; v = $1 - $2; if (v < 0) v = -v; if (v > $3) r = 1; r" | bc [22:20] <ubitux> just here. [22:20] <Daemon404> i thought t was elsewhere [22:28] <ubitux> actually i don't get that bc like [22:29] <ubitux> oh r is the exit value maybe [22:29] <ubitux> seems not [22:29] <nevcairiel> bc is just weird, deal with it! :D [22:29] <ubitux> so it's simply a abs($1-$2)>$3?1:0 [22:31] <ubitux> so something like echo "gt(abs($1-$2),$3)" | run tools/ffeval might actually do the trick? [22:34] <ubitux> i should have read the comment above [22:38] <ubitux> - echo "scale=2; v = $1 - $2; if (v < 0) v = -v; if (v > $3) r = 1; r" | bc [22:38] <ubitux> + [ "$(echo "gt(abs($1-$2),$3)" | ./tools/ffeval)" == "=> 1.000000" ] && echo 1 || echo 0 [22:38] <ubitux> seems to do the trick :p [22:38] <ubitux> (yes that's a bit ugly but well) [22:41] <ubitux> ah, got a better one :) [22:44] <ubitux> dammit bash [22:44] <ubitux> - echo "scale=2; v = $1 - $2; if (v < 0) v = -v; if (v > $3) r = 1; r" | bc [22:44] <ubitux> + printf "%d\n" $(echo "gt(abs($1-$2),$3)" | ./tools/ffeval -p '') [22:44] <ubitux> this doesn't work with the printf implementation of printf [22:45] <ubitux> $ printf "%d\n" "1.0" [22:45] <ubitux> bash: printf: 1.0: invalid number [22:45] <ubitux> :( [22:45] <Daemon404> because its a float [22:45] <ubitux> i know, but i'm not trying to do bash arithmetic here [22:46] <nevcairiel> what kind of printf do you have, seems to work here [22:46] <ubitux> bash builtin [22:46] <ubitux> it works fine with zsh [22:47] <ubitux> it warns but it works though.. [22:47] <ubitux> [ux@en ~]$ printf "%d\n" "0.0000" 2>/dev/null [22:47] <ubitux> 0 [22:47] <ubitux> [ux@en ~]$ printf "%d\n" "1.0000" 2>/dev/null [22:47] <ubitux> 1 [22:47] <ubitux> now how portable is that? [22:47] <nevcairiel> not very? [22:47] <nevcairiel> :D [22:47] <ubitux> :( [22:47] <ubitux> dammit [22:47] <nevcairiel> i dont know [22:48] <nevcairiel> why are you fighting bc anyway [22:48] <nevcairiel> its been a dep for ever [22:48] <ubitux> because it's not in the base system :p [22:48] <ubitux> yeah but it's a stupid one [22:48] <Daemon404> neither is rsync [22:48] <Daemon404> who cares [22:48] <nevcairiel> nor is git, or yasm, or gcc (on most base systems) [22:49] <nevcairiel> probably not even make [22:49] <nevcairiel> i should go sleep, 11pm is no time to be fighting 20 year old automation interfaces [22:49] <ubitux> maybe i can just use awk [22:50] <ubitux> since we use it in other places [22:50] <aetasx> you guys have any clue about the libfdk question I asked a bit ago? [22:53] <ubitux> - echo "scale=2; v = $1 - $2; if (v < 0) v = -v; if (v > $3) r = 1; r" | bc [22:53] <ubitux> + awk "BEGIN { v=$1-$2; printf \"%d\\n\", ((v<0?v:-v) > $3) ? 1 : 0 }" [22:53] <ubitux> this works \o/ [22:54] <aetasx> what the hell is that? [22:54] <nevcairiel> shell magic [22:55] <aetasx> yeah I got that, for what [22:55] <ubitux> doing a check of abs(a-b)>c [22:55] <ubitux> with a, b and c floats [23:03] <aetasx> isnt there a bc for that? [23:04] <nevcairiel> he wants to get rid of bc [23:04] <nevcairiel> thats the entire point of this endavour :D [23:04] <aetasx> ah, missed that part [23:20] <aetasx> do people still use acovea? looks like their website is long since gone [00:00] --- Tue Jan 20 2015

1 0

ffmpeg.log.20150119
by burek 20 Jan '15

20 Jan '15

[01:32] <ac_slater> hey all. I have an AVPacket that is a demuxed h.264 stream. What's the best way to separate the packet.data into NAL units ? [01:34] <fmiz> Why the startpts of a webcam input is not 0? [03:38] <pentanol> fmiz not /dev/video0 ? [06:30] <ac_slater> hey all avformat_find_stream_info() is timing out for me with RTSP streams sometimes. What do you think the chances are that I could fill a demuxer/decoder structure pair myself? [06:32] <ac_slater> ie - I know the properties before hand [08:10] <jeanre> guys, im extracting a frame from a video [08:11] <jeanre> 00:05:00, frame 2 @ 29.95 FPS should be frame 8987 [08:11] <jeanre> but when I open the screenshot [08:11] <jeanre> the time code is 00:04:59;25 [08:12] <jeanre> so its using the 25th frame on the previous second [08:14] <jeanre> its not really accurate [08:17] <jeanre> the command I am using is ffmpeg -i ./fixtures/timecode.mp4 -vf "select=gte(n\, 8987)" -vframes 1 8987.png [08:21] <jeanre> its accurate until the minutes start inceasing [08:25] <jeanre> any ideas guys? [08:32] <jeanre> how can it be this inaccurate? [08:34] <BtbN> No idea, have you tried frame 9000? [08:35] <BtbN> which would be 30 fps [08:36] <jeanre> hmm [08:37] <jeanre> that gives me 00:05:00;10 [08:37] <jeanre> wtf [08:38] <jeanre> BtbN thanks though :P [08:54] <jeanre> im not sure why it is 7 frames out [08:54] <jeanre> but it seems to shift as the time increases [08:55] <jeanre> BtbN any other ideas? [09:12] <jeanre> this has to be possible [09:28] <jeanre> no one has any ideas? [09:34] Action: jeanre is confused [10:06] <jeanre> ok so I got it slightly more accurate but the first frame is still screwed [10:58] <jeanre> im doing an extraction of frames [10:59] <jeanre> if I want the first frame is returns 00:04:59;29 the send one is 00:05:00;2 [10:59] <jeanre> why is the first frame off [11:05] <relaxed> you could cheat and dump them all, then pick what you need [11:10] <BtbN> maybe there simply is no frame which is at exactly 00:05:00? [11:22] <k_sze> Who uses the directshow input of ffmpeg on Windows? [11:24] <BtbN> If you want help with that, asking your actual question is much more likely to get you any. [11:26] <jeanre> BtbNit should be 00:05:00;01 [11:27] <jeanre> because its the first frame on 5 minutes [11:27] <jeanre> if there are 29.97 frames per second [11:27] <jeanre> I think the calculation I am doing [11:28] <jeanre> BtbN https://github.com/ikhasi/video-frame-extractor [11:28] <jeanre> thats the code I am running now [11:28] <jeanre> the first frame is always off [11:30] <jeanre> it jumps from 00:04:59;29 -> 00:05:00;02 [11:30] <jeanre> which makes no sense [11:30] <jeanre> its skipping the first frame (maybe dropframes) [11:31] <jeanre> but not too sure [11:34] <jeanre> how can it be [11:40] <jeanre> BtbN any ideas? [11:41] <BtbN> no [11:41] <jeanre> BtbN thanks dude [11:42] <BtbN> I'm not even sure i understand what you are trying to do, except you want to extract one or multiple frames from a video, and some timestamp isn't what you think it should be. [11:43] <jeanre> one frame [11:43] <jeanre> so each second has 29.97 frames [11:44] <jeanre> I want frame 2 on the fifth second [11:44] <jeanre> or frame 1 [11:44] <jeanre> or frame 3 [11:44] <BtbN> And what makes you think that you don't get that frame? [11:44] <BtbN> Is it some test video with timestamps in each frame, or some ffmpeg output? [11:44] <jeanre> because I am using a video with timecode burn [11:45] <BtbN> And you are sure that the entire test video isn't just off by one? [11:45] <jeanre> when I open the images created the first extract on 00:05:00 is actually 00:04:59 [11:45] <jeanre> the video is fine, because all other frames are correct [11:45] <jeanre> you can however check my code on github that does the frame etracting [11:45] <jeanre> *extracting [11:46] <jeanre> the second frame and all the others are correct [11:46] <jeanre> just the first one [11:46] <BtbN> did you check that the number it calculates are actualy consecutive frames? [11:47] <jeanre> and the funny thing is [11:47] <jeanre> if I start extracting from 00:00:00 [11:47] <jeanre> it works fine [11:47] <jeanre> yes I have [11:48] <BtbN> I'd blame that on floating point rounding inaccuracies [11:48] <jeanre> yes [11:48] <jeanre> but I have no idea how to compensate for that [11:48] <BtbN> well, don't use floats [11:48] <jeanre> im not parsing floats [11:49] <jeanre> well even without doing it [11:50] <jeanre> same effect without or with floats [11:50] <jeanre> I think its drop frames [11:51] <jeanre> this is my calculation to calc the frame [11:51] <jeanre> seconds = (+hours) * 60 * 60 + (+minutes) * 60 + (+seconds); [11:51] <jeanre> return (seconds * frameRate) + frameNumber; [11:51] <jeanre> where frame rate will be 29.97 [11:51] <jeanre> and the frame number 1 or 2 or 3 [11:53] <jeanre> its confusing as hell [11:54] <jeanre> for instance passing time of 00:01:00 in and extracting the first 10 frames [11:54] <BtbN> what ffmpeg version are you using? [11:55] <jeanre> the first extract is 00:01:00;02 [11:55] <jeanre> wtf [11:55] <ramiro> jeanre: did you try changing -vsync? [11:56] <jeanre> hmm [11:56] <jeanre> no idea [11:57] <BtbN> Also, did you try the -ss option to seek via ffmpeg itself? [11:57] <BtbN> Instead of calculating frame numbers [11:57] <BtbN> since ffmpeg 2.1 -ss should be accurate [11:57] <nichego> hi. is there a way to tell ffmpeg what source address to use for hls? (i have a network interface with multiple addresses) [11:58] <jeanre> 2.4.4 [11:58] <jeanre> is what I am using [11:58] <BtbN> So, then just try with -ss [11:58] <jeanre> so hh:mm:ss.frames? [11:58] <BtbN> no idea if it expects frames after the seconds [11:59] <jeanre> its miliseconds [11:59] <Elirips> Hello. I have an external source which creates one bitmap per second. I need to transform those bitmamps into a "live Stream". Can I take an ffmpeg, and serve one bitmap every second to that ffmpeg and let ffmpeg create a 5 fps stream out of that? [11:59] <jeanre> so I need to calculate the milisecond per frame [11:59] <jeanre> if there are 29.97 frames per second [12:01] <BtbN> why calculate frames at all? [12:01] <BtbN> -ss takes the exact timestamp you want [12:02] <jeanre> I need to use the frame [12:02] <BtbN> But take a timestamp as input? [12:02] <jeanre> I have a frame no timestamp [12:02] <Elirips> Could I just use 'ffmpeg -framerate 1/5 -i img%03d.png -c:v libx264 -r 30 -pix_fmt yuv420p out.mp4' but replace 'img%03d.png' with '-' to make ffmpeg read from stdin and then pipe the frames to ffmpeg? Something as simple as 'cat myimg.bmp | ffmpeg-process' ? [12:03] <BtbN> your code on githut uses a timestamp though [12:03] <jeanre> timestamp for hh:mm:ss + frame [12:03] <jeanre> it takes a frame [12:03] <relaxed> Elirips: cat myimg.bmp | ffmpeg -f image2pipe -i - [12:03] <jeanre> so it will be 29.97 * frame Iwant [12:03] <BtbN> yes, i don't see the problem there? [12:04] <BtbN> Just keep getting the frame like you are right now. [12:04] <BtbN> And seek to that timestamp via -ss [12:04] <BtbN> no calculations needed at all [12:05] <Elirips> relaxed: thank you. My real problem is a little bit more: The images are created constantly from another programm, and I need one ffmpeg that "runs forever" and listens to stdin, and then every second a cron-job will feed one image to ffmpeg. But I guess that could be done somehow. Or is this already built-in? [12:05] <Elirips> ie, is there a function if ffmpeg like 'Create a video from the file in directory X and refresh that directory every y seconds' ? [12:06] <jeanre> BtbN [12:06] <jeanre> I am not understanding what you are saying [12:06] <jeanre> I need to seek to frame 1 on 00:05:00 [12:06] <BtbN> yes, then just seek to 5 minutes with -ss, and then extract frame 1? [12:06] <jeanre> 300 seconds in [12:07] <jeanre> so it should in theory seak to 329.97 [12:07] <jeanre> wait no [12:07] <relaxed> Elirips: No, you would have to script it [12:08] <jeanre> it should seek to 300 + frameNumber * frameRate / 60 [12:08] <Elirips> relaxed: okay, I will try. Bad luck that I'm on windows for that part :( [12:08] <BtbN> jeanre, no, why? Just seek to your timestamp, and extract your frameNumbr. [12:08] <BtbN> +e [12:08] <jeanre> BtbN how would I do that [12:08] <BtbN> <BtbN> yes, then just seek to 5 minutes with -ss, and then extract frame 1? [12:08] <jeanre> I think my command is wrong [12:09] <Elirips> relaxed: Need a mfc app that creates the bitmaps every second (needs activeX to access the source, so it must be MFC), on start creates a ffmpeg-instance and the stdouts to that ffmpeg [12:09] <Elirips> I already now that it will be a pain to pipe to stdin of an external process from an mfc app - hell yea [12:09] <jeanre> ffmpeg -r 1 -i ' + filePath +' -ss ' + time +' -vframes 1 ' + frameNumber + '.png -y [12:10] <BtbN> -ss is an input option [12:10] <BtbN> so it has to be in front of your input file. [12:10] <BtbN> and you propably want to get rid of that 1 [12:10] <BtbN> also, the -frames:v parameter is the number of frames to extract [12:10] <relaxed> Elirips: I know zilch about windows scripting [12:10] <BtbN> why not just keep using the method you currently use? [12:10] <jeanre> ok [12:10] <BtbN> and only add -ss [12:11] <jeanre> command = 'ffmpeg -r 1 -ss ' + time + ' -i ' + filePath +' -vframes 1 ' + frameNumber + '.png -y'; [12:12] <BtbN> That's now how your code on github extracts a specific frame number. [12:12] <jeanre> ok [12:12] <jeanre> thats works sorta [12:12] <jeanre> command = 'ffmpeg -r 1 -ss ' + time + ' -i ' + filePath +' -vf "select=gte(n\\, ' + frameNumber +')" -vframes 1 ' + frameNumber + '.png -y' [12:12] <jeanre> executing with 00:01:00 [12:12] <jeanre> as the time [12:13] <jeanre> the video burn is 00:00:59;01 for the first frame [12:13] <jeanre> second is 00:00:59;02 [12:13] <jeanre> why is it loosing a second? [12:14] <jeanre> if I remove the -r 1 [12:14] <jeanre> the first image is 00:00:01;02 [12:14] <jeanre> urg [12:16] <jeanre> if I change the time to 00:05:00 [12:16] <jeanre> the first extracted image is 00:04:59;09 [12:17] <jeanre> that makes no sense [12:17] <jeanre> even if frame number is 1 [12:17] <BtbN> well, propably something doesn't agree with that framerate. [12:18] <jeanre> im not even touching the frame rate now [12:18] <BtbN> ffmpeg still needs to take it into account when seeking to the timestamp [12:19] <jeanre> so -r is the rate IIRC.. [12:19] <jeanre> so if I set that to 29.97 it still freaks out [12:19] <jeanre> if I remove -r [12:20] <jeanre> the first frame is 00:05:00;03 [12:20] <relaxed> -ss is more accurate after the -i [12:20] <relaxed> (in case you didn't know) [12:20] <BtbN> I'm not even sure if -r does anything for non-raw input format. At least that's what the manpage says [12:21] <jeanre> relaxed if I move it to after [12:21] <jeanre> everything is 00:05:00;02 [12:22] Action: jeanre bashes head [12:23] <jeanre> relaxed any other ideas? [12:23] <_vaibhav_> Hi, am trying to make hls playlist, using following command : ./ffmpeg -i ~/source.mp4 -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list playlist.m3u8 -segment_list_flags +live -segment_time 10 out%03d.ts [12:24] <_vaibhav_> but it generates playlist which has floating point segments. I want to make those in pure integer so that older players also support. [12:25] <_vaibhav_> is there any way I can get -no-floating-time , something like that/ [12:25] <_vaibhav_> ? [12:26] <relaxed> jeanre: maybe, ffmpeg -vsync passthrough -i input -ss 00:05:00.02 ... [12:27] <BtbN> i think he means frame no. 2 with his timestamp, not milliseconds or something [12:27] <relaxed> oh [12:27] <jeanre> relaxed .02 in that timecode is milliseconds not frames [12:30] <relaxed> do you know the exact timestamp? [12:30] <jeanre> only hour minutes and seconds [12:30] <jeanre> and the frame number [12:32] <jeanre> I'm thinking I could calculate the exact millisecond by taking frameRate and calculating the correct milisecond [12:32] <jeanre> but alas [12:32] <jeanre> no idera [12:32] <jeanre> *idea [12:34] <relaxed> It's hard to troubleshoot when I can't run it locally [12:34] <jeanre> you can [12:34] <jeanre> I have code on github [13:20] <Elirips> Is there somewhere a windows build from the version 2.6.54 available? Or are there only the 'latest builds' available? [13:21] <Elirips> err, I mean the version 2.5. 64bit [13:57] <HebusLeTroll> Hello, is it possible to add a stream and set its language without knowing its output's steam number ? [13:58] <HebusLeTroll> (or to reference the last steam, no matter how many stream there is ?) [13:58] <HebusLeTroll> stream* [15:57] <tmh1999> Hi guys, I hope someone could help me. Can you explain briefly all/ or most of it the options in enum AVPixelFormat in libavutil/pixfmt.h ? [15:59] <tmh1999> I kind of now have to rescale a video from 1920x1080 to smaller size like 360p 480p 720p, so which source pixel format and destination pixel format should I use? [15:59] <tmh1999> Thanks. [18:38] <grepper> what is wrong with this command? ffmpeg -i INFILE.mp4 -c:v copy -i audio1.wav -c:a aac -b:a 240k -y OUTFILE.mp4 . I get "Unknown decoder 'copy'" [18:39] <Mavrik> grepper, parameter order [18:39] <Mavrik> grepper, everything before "-i" means "apply to input" [18:40] <grepper> yes, I know. But I'm not sure how to arrange that then, I thought I was putting it AFTER [18:41] <grepper> Mavrik: ? [18:41] <Mavrik> grepper, your "-c:v" parameter is certanly before "-i" [18:41] <Mavrik> ;) [18:42] <grepper> ah, so all the input files before the other options [18:43] <dunpeal> Hi. I have a bunch of classical music CDs. What's a good lossy conversion for them? [18:44] <Mavrik> dunpeal, high-bitrate mp3 is still the most compatible one [18:44] <Mavrik> dunpeal, good AAC encoder should be better, but you should probably use iTunes for ripping then [18:44] <dunpeal> Mavrik: I'm using Rockbox, so compatibility isn't an issue. My priorities are 1) quality, 2) file size [18:44] <dunpeal> and I'm on Ubuntu, so no iTunes :) [18:45] <Mavrik> well, filesize is set by the bitrate [18:45] <Mavrik> and higher bitrate means more quality [18:45] <Mavrik> also, better encoder means more quality for same bitrate :) [18:45] <Mavrik> dunpeal, but honestly, 320kbps/48000 mp3 is going to sound awesome [18:45] <Mavrik> or if you want to listen on the way, 192kbps/44100 AAC should be more than enough [18:45] <Googol30> I'm partial to AAC as a lossy format. There's no shortage of options, though. [18:46] <dunpeal> so far in my inquiries, it seemed like the very best option is libfdk_aac with a -vbr setting of 4 or 5 [18:46] <grepper> Thanks Mavrik , that worked. I had wrongly thought since -c:v applied to the video, that it was "after" . Anyway ... [18:47] <Mavrik> grepper, yeah, ffmpeg is a bit funny about ordering [18:47] <Mavrik> dunpeal, yes, FDK AAC is the best AAC encoder ffmpeg has [18:47] <Mavrik> I have no idea what bitrate vbr4/5 will be, but sounds reasonable :) [18:48] <dunpeal> Mavrik: I just encoded with -vbr 4, and it was about 128 kbps [18:49] <Mavrik> dunpeal, if it sounds good to you, keep it :) [18:49] <dunpeal> bitrate= 136.9kbits/s [18:49] <dunpeal> it does, I can't distinguish between that and the original [18:49] <dunpeal> for casual listening, I bet even 3 may be good enough [18:50] <Mavrik> hmm, I wouldn't go lower than that with classical, since it tends to cut off some stuff [18:51] <dunpeal> Mavrik: one more question: is there a way to preserve the tracks? I have my CDs ripped to FLAC with a .cue file. [18:52] <Mavrik> dunpeal, you'll have to split the one-piece FLAC file into separate tracks [18:52] <Mavrik> something like http://danilodellaquila.com/blog/how-to-split-an-audio-.flac-file-using-ubu… [18:53] <dunpeal> Mavrik: I thought M4A supported chaptering? [18:53] <Mavrik> Hmm, I'm not aware of players / tools that would support that well in terms of audio :/ [18:54] <Mavrik> but I might be wrong there [20:32] <kapetr> Hello. I use avconv to encode MP4 directly from DVB card (m2ts stream). Problem is, that the captured video do not begin with key-frame and output video has damaged beginning. Is there a way to tell avconv to begin encoding at first receive key-frame (to skip no-key-frames on begin of input) ? Thanks. [20:34] <c_14> kapetr: support for avconv is at #libav, if you want support here try again with ffmpeg from FFmpeg [20:35] <kapetr> Sorry - is libav and ffmpeg not the same ? Even if not - it is possible in ffmpeg ? THX. [20:37] <c_14> I don't know if there's an option for it, but I'd try again with a static build and check if the same error occurs. [20:37] <Mavrik> um [20:37] <Mavrik> there's no way to encode a video without starting keyframe [20:37] <Mavrik> so you have an issue somewhere else. [20:37] <c_14> Mavrik: you can't encode a stream without a starting keyframe, but you can receive a stream without having the starting keyframe [20:37] <kapetr> See: http://linux.goeszen.com/simple-video-editing-with-ffmpeg.html [20:38] <Mavrik> c_14, ah, I misread his post -_- [20:38] <Mavrik> *grumbl* [20:38] <Mavrik> gonna go sleep rather [20:38] <Mavrik> "-sameq"? [20:38] <c_14> kapetr: -sameq doesn't do what you think it does [20:38] <Mavrik> No. [20:38] <c_14> And shouldn't be used. [20:38] <kapetr> CITE: "so the outputted video file might start with garbled data or simply black video (might depend on how your media player renders garbled data/ data without a starting keyframe&)." [20:40] <Mavrik> it also seems that ffmpeg should discard broken frames before keyframe [20:40] <c_14> kapetr: if you are using -codec copy, there is an option to copy non-key frames at the beginning (in ffmpeg at least) [20:41] <Mavrik> when running actual encode [20:41] <c_14> So as Mavrik just said, the default should be to discard non-keyframes before the first keyframe [20:47] <kapetr> should ... BUT seems not t be so. E.g. Avidemux (uses libav) set start point at first key-frame, but shows, that there are some frames before it. That's why i mean, there is a way to say ffmpeg/libav to skip them. [20:48] <c_14> Did you try using ffmpeg? You can get a static build here: [20:48] <c_14> http://johnvansickle.com/ffmpeg/ [20:49] <c_14> And if you're using the library (which avidemux probably is) finding the first keyframe is as easy as asking for all the frames until one of them is a keyframe [21:08] <Mavrik> kapetr, libav ISN'T ffmpeg. [21:08] <Mavrik> so we can't really help you with issues from that software :/ [21:13] <kapetr> I have test now static ffmeg to do the same work on same m2ts file. [21:15] <kapetr> There is difference - ffmpeg version has on begin till picture and video starts to play in the same "timestamp" as in avconf version the video begins to play withou damage. [21:15] <kapetr> ... not till -> stil picture [21:17] <kapetr> ffmeg output is more nice, but there are frames (as audio shows) before first key-frame too. [21:18] <kapetr> ?? maybe - file output begins from first key-frame - but this frame is from audio stream ? [21:23] <lyscer> anyone have any good tutorials they recommend for adding audio to an existing mp4? I have tried a couple different ones and can't seem to get the audio to merge into the video [21:23] <c_14> ffmpeg won't drop the audio just because there isn't a video keyframe, so yeah that makes sense [21:23] <c_14> lyscer: ffmpeg -i video -i audio -map 0:v -map 1:a -c copy outfile [21:26] <lyscer> c_14, thanks; Any ideas how to troubleshoot it if the sound still isn't in my video? [21:27] <lyscer> I tested the mp3 and it plays fine [21:27] <Mavrik> lyscer, by reading the ffmpegs output [21:27] <c_14> Throw your full commandline and complete console output on a pastebin service and throw the link here [21:27] <Mavrik> ^^ [21:27] <lyscer> please hold... [21:31] <lyscer> http://pastebin.com/N6Hh9959 [21:32] <c_14> So the mp3 plays fine, the video plays fine, but if you play the merged video you don't get audio? [21:32] <lyscer> correct [21:32] <c_14> What player? [21:33] <lyscer> oh... I just had another idea and tried a new player and it works under the new player [21:33] <lyscer> gah, I wasted soo much time with a non broken file on that player! Thank you for the help! [21:33] <lyscer> c_14, thanks for the help!! [21:37] <Mavrik> lyscer, yeah, do use aac in mp4s for maximum compatiblity :) [21:46] <kapetr> c_14: ant is it possible to say ffmbeg to do so ? I mean to begin encode all streams from first key-frame in selected (video) stream ? [21:47] <c_14> Can't think of one off the top of my head. [21:49] <kapetr> thanks anyway. Good night. [21:49] <Nopik> hi all [21:51] <Nopik> i'm streaming some music, the source is combining songs from various sources, and therefore producing songs with different volumes, so when you listen to resulting stream, volume is often jumping up or down at song boundaries. Is there any simple way to alter the stream to normalize the volume with ffmpeg? i mean to do that in 1 pass, without knowing the [21:51] <Nopik> volume adjustment beforehand, i.e. the filter would need to be adaptive [21:52] <c_14> Try using the compand filter? [21:52] <Nopik> c_14: let me see its docs first :) [21:55] <Nopik> c_14: looks very interesting, I'll try it out. Any other options? [21:57] <c_14> If the tracks have replaygain side-data, I think the volume filter can act on that. [21:59] <Nopik> yeah, I've just found it as well. But I'm afraid that most of the tracks wont have the replaygain, I cannot reliably base on it [21:59] <Nopik> anyway, thanks, it seems that compand will do the job, although I'll apparently have to experiment with it a bit [00:00] --- Tue Jan 20 2015

1 0

ffmpeg-devel.log.20150118
by burek 19 Jan '15

19 Jan '15

[00:04] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:23fe589e1948: prores: Evaluate all the quantizers [00:04] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:20d6ae4626a4: hevc: always clip luma_log2_weight_denom [00:04] <cone-46> ffmpeg.git 03Thiago Santos 07release/2.4:20f9cf744a9a: vaapi: wrap codec specific functions in appropiate #ifs [00:05] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:73046fa19593: Merge commit '23fe589e19487bc9a40b77dae9509068da5b14b8' into release/2.4 [00:05] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:9d43e51eec49: Merge commit '20d6ae4626a42fbc6e7b9c0040bdd64397b23e11' into release/2.4 [00:05] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:0d801c0bfdec: Merge commit '20f9cf744a9a82ac4b269cb4317a5d59a8553baf' into release/2.4 [00:23] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:1fd55ec507f6: svq1dec: Unbreak the scratch buffer allocation [00:23] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:bbb86717b303: vf_frei0r: do not increment string if it reached the end [00:23] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:4baee1124b90: mp3: Tweak the probe scores [00:23] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:12d6bbf7cbcc: Merge commit '1fd55ec507f6f47b4d9fddf8e79a0df4540ef6e4' into release/2.4 [00:23] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:c8a2b1ac3c62: Merge commit 'bbb86717b303a3e4c0809d3cc6fb55580766a17e' into release/2.4 [00:23] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:85d05fdc8000: Merge commit '4baee1124b905cbd75240530e081c8ffa68fddbe' into release/2.4 [00:23] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:705e0e05131f: avformat/mpeg: do not count PES packets inside PES packets during probing [00:37] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:da35008c314e: latm: Do not give a score for a single instance [00:37] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:036f5c5420e4: rm: Use the correct codec_data_size signedness [00:37] <cone-46> ffmpeg.git 03Michael Lynch 07release/2.4:b31bb39bdd7b: rtsp: Check a memory allocation [00:37] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:1dc53463d9c9: Merge commit 'da35008c314eae5a10a8c070f5185b7694c0a37e' into release/2.4 [00:37] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:365e0652f94d: Merge commit '036f5c5420e4529f05fa5180f5fa28ca2c5c4065' into release/2.4 [00:37] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:aded1110a32e: avformat/rmdec: rm_read_extradata: add error message for oversized extradata [00:37] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:564d943b2755: avformat/rmdec: Check for overflow in ff_rm_read_mdpr_codecdata() [00:37] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:1da83d52bc0f: Merge commit 'b31bb39bdd7b5a53e0d282acc0f0f62b32b17acc' into release/2.4 [00:39] <cone-46> ffmpeg.git 03Clément BSsch 07master:ca3c7be7c68a: avcodec/tiffenc: remove double ;; [00:39] <cone-46> ffmpeg.git 03Clément BSsch 07master:4d028bf202bd: avcodec/vp8: remove double ;; [00:39] <cone-46> ffmpeg.git 03Clément BSsch 07master:fa6c9e54df59: avdevice/pulse_audio_dec: remove double ;; [00:39] <cone-46> ffmpeg.git 03Clément BSsch 07master:14f3e8b821d6: avformat/sbgdec: remove double ;; [00:47] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:bb823e26b1cb: avformat: Make avformat_free_context handle NULL [00:47] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:4d4869162214: cmdutils: Use the correct guard [00:47] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:608e8d8dd754: vf_drawtext: Do not leak the mmapped textfile [00:47] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:2496dbd68b29: vf_showinfo: Forward the av_image_get_linesize error [00:47] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:679c3d210f5d: Merge commit 'bb823e26b1cbb91b475c10772cdfd867d4809a65' into release/2.4 [00:47] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:ee025a0c6137: Merge commit '4d48691622149ba5998de08a0acec85d1f4ed46a' into release/2.4 [00:47] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:6fd7456f8572: Merge commit '608e8d8dd754199b657b439f9e722e0b45f84461' into release/2.4 [00:47] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:158cf1d52df6: Merge commit '2496dbd68b29bd36fbf4753a46163d33f6dae70e' into release/2.4 [00:49] <kierank> why is it so hard to understand [00:49] <kierank> bt601 SARs are not arithmetic [00:49] <kierank> nor do they apply to the entire digital picture [00:50] <Daemon404> nicholas does not live in reality [00:50] <Daemon404> just ask wm4. [01:01] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:daef7feb09a0: af_resample: check av_opt_set_dict return value [01:01] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:c246b0b4c3a3: avresample: Make sure the even check does not overflow [01:01] <cone-46> ffmpeg.git 03Timothy Gu 07release/2.4:51f76e4e932e: oss_audio: use a macro to simplify ioctl() error checking [01:01] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:27487944eff7: swscale: fix sign extensions in yuv planar conversion [01:01] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:b4004124432e: Merge commit 'daef7feb09a0dde2265d56f77de8ae03f9612d47' into release/2.4 [01:01] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:adfb4661d9ab: Merge commit 'c246b0b4c3a3b02a714e99423cf23d59f8f81409' into release/2.4 [01:01] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:39f2e53a2457: Merge commit '51f76e4e932ebdce8ccf6cf0797651d632cfc3e2' into release/2.4 [01:01] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:a299d93928ae: Merge commit '27487944eff721ef8e310db1a2a52329d9377f71' into release/2.4 [01:09] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:5aceced0a0d2: avio: fix sizeof argument [01:09] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:6cf27b550d66: aviobuf: check context before using it [01:09] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:d1ad85fc0206: vf_format: check input validity [01:09] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:4cd0041d3866: rmdec: check av_new_packet return value [01:09] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:d6f4e475a107: Merge commit '5aceced0a0d235d485e88fda502019b221dbf3d2' into release/2.4 [01:10] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:153a012384b3: Merge commit '6cf27b550d66963b89b5917568b85c4b49fb18ba' into release/2.4 [01:10] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:34b28d3823af: Merge commit 'd1ad85fc020653be5ad25ed1d3d5091e89fee782' into release/2.4 [01:10] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:8120061be269: Merge commit '4cd0041d38664adcb6f4b3038e277631b85d5dc8' into release/2.4 [01:18] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:1551602b4237: nutdec: check av_new_packet return value [01:18] <cone-46> ffmpeg.git 03Anton Khirnov 07release/2.4:556a5090f2b0: mp3dec: fix reading the Xing tag [01:18] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:a7e79f6a43bd: avfilter: check filter link validity [01:18] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:749fadaa651e: Merge commit '1551602b423755c4ed98c5b7b2c2d6504416726e' into release/2.4 [01:18] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:275cd55a0f4b: Merge commit '556a5090f2b0a20fd9998e1a327875f5b0c8d1d5' into release/2.4 [01:18] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:fefb152cd94c: Merge commit 'a7e79f6a43bdc7e3b164dd78b060f0d5575a0cf0' into release/2.4 [01:28] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:7c710c38f648: audiointerleave: check av_new_packet return value [01:29] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:b9b689550e75: img2dec: check av_new_packet return value [01:29] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:19fc283dbb53: lavf: replace rename() with ff_rename() [01:29] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:298f7f1fac0b: Merge commit '7c710c38f6481b892e01d3c1e4781ad160b2935e' into release/2.4 [01:29] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:324797eb83df: Merge commit 'b9b689550e7531b1a2cc893d2af623e37f266936' into release/2.4 [01:29] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:d3316c3f85c2: Merge commit '19fc283dbb53a5f7d6658fd4edcfa59b99369b58' into release/2.4 [01:39] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:8a982092cc34: mxfdec: add missing break [01:39] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:26ba78adacd8: mov: fix assigment check [01:39] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:bae05e532670: matroskaenc: write correct Display{Width, Height} in stereo encoding [01:39] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:b6cd6da17329: Merge commit '8a982092cc3436c25d68ec15b27277c176ce0061' into release/2.4 [01:39] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:97a55f00d35c: Merge commit '26ba78adacd8469fca97c8c833e2e6364b13b7c8' into release/2.4 [01:39] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:6f9bb83dbb93: Merge commit 'bae05e5326703dad3bfe0ed5b31ba73ee9254515' into release/2.4 [01:59] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:242fc6394fec: mtv: improve header check and avoid division by zero [01:59] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:34e7f70f9f49: assdec: check av_new_packet return value [01:59] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:2a75c0b1ca16: aacsbr: change order of operation to prevent out of array read [01:59] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:4edbb0955e04: png_parser: fix size of chunk_lenght [01:59] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:c7831e55b517: Merge commit '242fc6394fecb403bcbd0f652920f2647d0b08ae' into release/2.4 [01:59] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:6aece62b50cc: Merge commit '34e7f70f9f493f340daab80eba4f12d005ec3e63' into release/2.4 [01:59] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:f038623d5342: Merge commit '2a75c0b1ca16b5480497de0d4c79ef122406a0b5' into release/2.4 [01:59] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:b5b36e45b328: Merge commit '4edbb0955e043d698dcc1b5073b6e714f9cc7960' into release/2.4 [02:08] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:e7ee74485b43: hnm4: Use av_image_check_size [02:08] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:12e1a7013a53: roqaudio: Always use the frame buffer on flush [02:08] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:b82170336f90: tiffenc: initialize return value [02:09] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:484e015dc8b9: cook: Make sure there is enough extradata [02:09] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:6cb8e40f4372: Merge commit 'e7ee74485b436c34591177c18c8643764a55d516' into release/2.4 [02:09] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:5b965a508ca3: Merge commit '12e1a7013a53ad957c4ff11a3aebc0763024d24b' into release/2.4 [02:09] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:49f477071dcd: Merge commit 'b82170336f90d06c645d8252ddeccfc92c2f9ccb' into release/2.4 [02:09] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:6f1443421887: Merge commit '484e015dc8b9983297e9269b406c65084daf4528' into release/2.4 [02:23] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:fbc20c3b85be: aacdec: avoid an out-of-bounds write [02:23] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:cce99f72d1b4: mpegenc: prevent a NULL pointer dereference [02:23] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:e9aeaa6441f6: qdm2: avoid integer overflow [02:23] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:5891fd017aa7: dvdsubdec: Do not leak on failure path [02:23] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:4f72aef342c8: Merge commit 'fbc20c3b85be169389b6c9f8806a311d3dea91ea' into release/2.4 [02:23] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:e3ce2a87731d: Merge commit 'cce99f72d1b49d3dfee859136eeff3db32553750' into release/2.4 [02:23] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:f251409adc04: Merge commit 'e9aeaa6441f6fd18fc951d9737887dcf8a9584c0' into release/2.4 [02:23] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:c2b658c0e7b7: Merge commit '5891fd017aa7bed4c423b8511090cf8641a0afa4' into release/2.4 [02:33] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:4d11e4b47db6: libtwolame: prevent a NULL pointer dereference [02:33] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:32701252af65: xwma: Do not leak on failure path [02:33] <cone-46> ffmpeg.git 03Martin Storsjö 07release/2.4:527617485914: lavc: Move the libtwolame encoder registration to the list for external libraries [02:33] <cone-46> ffmpeg.git 03Martin Storsjö 07release/2.4:93bf4a74de99: configure: Fix enabling memalign_hack automatically [02:33] <cone-46> ffmpeg.git 03Paul B Mahol 07release/2.4:375c1050bf74: libavcodec/libtwolame: fix null pointer dereference [02:33] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:206ca5c7c820: Merge commit '4d11e4b47db6387077682595d53e9a17b50511cb' into release/2.4 [02:33] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:dc84cf241ccc: Merge commit '32701252af65014bb68194bb61d67ec1882ae75d' into release/2.4 [02:33] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:099e0178bf56: Merge commit '527617485914004dc8b772056322ea5ae74c800d' into release/2.4 [02:33] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:9b669f9b1413: Merge commit '93bf4a74de99300fdc0deb83f672bf12e6f1c262' into release/2.4 [02:38] <nevcairiel> is it just me, or does nicolas just show up to troll around on certain topics on the ML recently, and that being the entire extend of what he is doing =p [02:39] <nevcairiel> without having checked, whatever the outcome, nvenc and libx264 should produce identical behaviour in any case [02:39] <nevcairiel> if the patch in question does that, it should be applied, and we can fix bt601 things later [02:47] <compn> nevcairiel : i think most devs just show up to troll actually :P [02:50] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:61fdbf7ff64c: lavc: fix bitshifts amount bigger than the type [02:50] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:29e720da76ca: librtmp: append the correct field to the string [02:50] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:dd195c2c587f: on2avc: Fix out of array access [02:50] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:21aeae9c6796: svq1enc: check ff_get_buffer return value [02:50] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:4e251b23c31a: Merge commit '61fdbf7ff64c0ae1bdd6a8d573092dc6924c1dba' into release/2.4 [02:50] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:e82140b09b0f: avformat/librtmp: fix swfurl [02:50] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:d48d2f166d65: Merge commit '29e720da76ca353dbda9f881562902b41c355b77' into release/2.4 [02:50] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:a0bad45cd8e7: Merge commit 'dd195c2c587f44dbc4be7f059ed182f7d83e6cb4' into release/2.4 [02:50] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:65b2973a5433: Merge commit '21aeae9c679657a1537d0d9127eff280bafc901a' into release/2.4 [03:20] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:39e07ac9fcaf: ansi: check ff_set_dimensions return value [03:20] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:9e0a38d32b36: avs: check ff_set_dimensions return value [03:20] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:796bd81835eb: g2meet: check ff_set_dimensions return value [03:20] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:3aba00b1b331: indeo3: check ff_set_dimensions return value [03:20] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:baef2736b07d: Merge commit '39e07ac9fcaf3d412f9a33f427072e8ded032d24' into release/2.4 [03:20] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:98ba2887370e: Merge commit '9e0a38d32b36fac7fd73bdb93e820ae0b9e03616' into release/2.4 [03:20] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:faa67e9777eb: Merge commit '796bd81835ebcb69d40bc87b33f13924a5937a1b' into release/2.4 [03:20] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:55c6a482d4fc: Merge commit '3aba00b1b331461611d32cfd56ebe8334f6d1b93' into release/2.4 [04:06] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:3f09d4f6d434: ffv1: fix out-of-bounds read [04:06] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:88411b87b4bb: display: fix order of operands [04:06] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:431f57f04672: libopusenc: prevent an out-of-bounds read by returning early [04:06] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:ff77fa754bf2: vp7: fix checking vp7_feature_value_size() [04:06] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:5f6f24122116: doc: Update the Changelog for release 11.2 [04:06] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:4039f11ed82d: Prepare for 11.2 Release [04:06] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:fb62580fb6cb: Merge commit '3f09d4f6d43468dbc9307bb937516a32287008dc' into release/2.4 [04:06] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:02d979d0074a: Merge commit '88411b87b4bb3c5820ec232f26ba4a284c11a7f9' into release/2.4 [04:06] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:6feb72a094de: Merge commit '431f57f0467244686ae63a3d06a8cf51f60090ed' into release/2.4 [04:06] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:f5dd1270d9dd: Merge commit 'ff77fa754bf2d6e7231d2e050babefa9a13d1fec' into release/2.4 [04:06] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:2b95572b10ec: Merge commit '5f6f2412211609e46d3f3573f4ac4e136534f1d4' into release/2.4 [04:06] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:fa1ca88cb6fc: Merge commit '4039f11ed82d84009b5146f0a563f33ced33e4db' into release/2.4 [04:41] <j-b> 'morning [04:46] <michaelni> 'night [05:15] <cone-46> ffmpeg.git 03Michael Niedermayer 07master:741fa0493b94: avcodec/mpegvideo_parser: more robust dimension setup [05:15] <cone-46> ffmpeg.git 03Michael Niedermayer 07master:dff6c9250ba8: avcodec/mpegvideo_parser: Check the ff_set_dimensions() return code [10:47] <wm4> ubitux: not sure if you can use the DTS field for readorder [10:47] <wm4> ubitux: DTS must be a valid timestamp, and I'd argue it should at least be "close enough" to the PTS [10:48] <nevcairiel> does the readorder have to be a strictly continous stream? ie. 2 4 6 is not valid? [10:48] <wm4> yes [10:49] <nevcairiel> why does this field even exist anyway [10:49] <wm4> well it's probably ok, but it will at least confuse the ffmpeg ASS muxer [10:49] <wm4> to serialize .ass files to a muxed packet stream [10:49] <nevcairiel> but it has timestamps [10:49] <wm4> events in ass files don't have to be sorted by timestamps [10:49] <wm4> and the order matters [10:49] <wm4> for subtle reasons [10:49] <nevcairiel> borrow my time machine, shoot whoever though random order was a good idea [10:49] <nevcairiel> +t [10:50] <wm4> it was created with the "total compatibility" mindset [10:50] <wm4> before ass-in-mkv, it was an accidental implication at best [10:51] <wm4> ubitux: also I'm noticing you use the timestamp 9:59:59.99 for subtitles that have no end time? [12:08] <wm4> so the nvenc behavior makes "sense" after all... [12:16] <kierank> wm4: yes [12:17] <nevcairiel> it jsut doesnt match other encoders behaviour in ffmpeg, which is bad imho [12:19] <kierank> that's because other encoders don't work in terms of DAR [12:19] <kierank> working in terms of DAR is actually the correct way to do it [12:21] <kierank> x262 has this problem as well [12:21] <kierank> and so I had to hack the sar field in x262 to be a DAR [12:23] <kierank> unfortunately the "SARs must be mathematical" crowd won't understand that [12:24] <JEEB> uhh, I'm a person like that yet I'm alright with the simplified representations f.ex. [12:25] <JEEB> like 40:33 for 16:9 480p instead of the longer mathematical one :D [12:25] <Mavrik> I've actually met some decoders that choke on more "complex" SARs [12:26] <Mavrik> some Windows Phones for example O.o [12:27] <kierank> JEEB: yeah [12:27] <kierank> https://github.com/kierank/broadcastencoder/blob/staging/filters/video/vide… [12:28] <JEEB> yup [12:30] <wm4> makes me glad that many file format specify aspect in the container [12:31] <JEEB> I made a bug report on one of MS's parsers of AVC streams once, and it seems like they just failed to understand some of the predefined SAR values, and only parsed if the "custom SAR" flag was set [13:03] <kierank> heh [14:30] <kierank> I give up [14:42] <compn> as well you should [14:43] <compn> spend hours fighting over some random aspect ratio [14:43] <compn> :P [14:43] <compn> ehe [14:43] <JEEB> what... [14:43] <JEEB> I should read the ML [14:43] <compn> should have applied nvidia's patch instead [14:43] <compn> let them deal with it :D [14:43] Action: compn trooooolls [14:43] <compn> this is what happen when NIH [14:44] <compn> arguing about things and citing wikipedia [14:49] <JEEB> I do understand the point that the encoder should encode with the aspect ratio that you set, and it should be a straightforward thing. The proper way would be to have the proper SAR/DAR set when the source is read so people wouldn't expect the f.ex. 16:9 part of 480p to be the full picture [14:51] <JEEB> and for the nonexact aspect ratios one just has to look at the DVD and BD specs [14:51] <JEEB> oh well [14:51] <compn> just let nvidia change aspect, its their crap, let them fix it [14:51] <compn> kierank : you need to reply in patches, not english [14:51] <compn> thats the only way to solve this issue of not understanding each other [14:52] <JEEB> well my idea is more like ffmpeg should read the aspect ratio correctly to begin with, which is not always possible of course [14:52] <kierank> you'll all go beserk if I submitted a patch that makes it right [14:52] <kierank> I can tell you that now [14:52] <compn> kierank : try it [14:52] <compn> do it!!!! [14:53] <kierank> I would change the API to set DAR [14:53] <kierank> and if DAR wasn't set you would read SAR [14:53] <kierank> because SAR and DAR *don't* necessarily have a mathematical relationship [14:54] <JEEB> but wouldn't the DAR just say "16:9" [14:54] <JEEB> or "4:3" [14:54] <kierank> yes [14:54] <kierank> and you then use a LUT for the bt601 case [14:54] <kierank> for other cases I don't really care what ffmpeg does [14:54] <kierank> nvidia uses DAR [14:54] <kierank> and ffmpeg assumes you can just do a multiplication to get DAR [14:55] <kierank> that's why the thread doesn't like the "correction" [14:55] <JEEB> I'm just thinking of how you could possibly see which of the two is set [14:55] <JEEB> as in, is it the BT.601 case [14:55] <kierank> if( dar.width && dar.height ) [14:55] <compn> just have it report 'par: not set' [14:55] <JEEB> or is it the "standard" case [14:56] <JEEB> (where the full picture is meant" [14:56] <wm4> making api changes is hard [14:56] <JEEB> or is it a if ( dar.width && dar.height && BT.601 ) kind of thing [14:56] <kierank> wm4: henc [14:56] <kierank> e [14:56] <kierank> 1:52 PM <"kierank> you'll all go beserk if I submitted a patch that makes it right [14:56] <compn> kierank : submit rfc about dar api change [14:56] Action: JEEB scratches his head [14:56] <kierank> anyway it's good - I can add a slide about how open source devs don't understand aspect ratios in my fosdem presentation [14:57] <compn> JEEB : solution is to troll people to send patches, not complain endlessly on irc [14:57] <JEEB> I understand the two types of aspect ratios but I'm not wholly sure how you'd automatize the correct behavior [14:57] <kierank> compn: the nvidia behaviour is correct [14:57] <kierank> so no patches are necessary [14:57] <JEEB> you can see a lot of OSS shit doing it the wrong way, but things that do it correctly are usually limited to a single field [14:57] <compn> isnt that what i said ? [14:57] <JEEB> (like DVD players) [14:58] <JEEB> where all of their input is of that one specific type [14:58] <compn> let nvidia deal with it, roughly [14:58] <kierank> but the people in that thread want things to work based on their beliefs [14:58] <JEEB> but yeah, I would rather bring this up as the wider issue of handling the BT.601 ARs correctly [15:01] <JEEB> like, if you read a 16:9 DVD with ffmpeg it should set the BT.601 AR, not the one that sets the full picture to be 16:9 [15:01] <JEEB> that said [15:01] <JEEB> how do you do it properly? [15:02] <JEEB> limit to specific resolutions, specific colorimetry and DAR being set? [15:02] <kierank> mpeg-2 is easy [15:02] <kierank> it just has DAR [15:02] <kierank> done [15:02] <JEEB> true [15:02] <kierank> h264 is tricker [15:02] <kierank> because you have to map the aspect_ratio_idc to a DAR [15:03] <JEEB> didn't AVC have SAR? [15:03] <JEEB> and at least the blu-ray specs mandate the correct ones [15:03] <JEEB> same for PSP video [15:04] <JEEB> s/PSP/UMD/ [15:04] <kierank> sar is used if you don't have a valid aspect_ratio_idc [15:04] <kierank> in x264 when you use SAR it gets mapped to an aspect_ratio_idc [15:04] <JEEB> yes [15:04] <kierank> otherwise, an escape value is used and the SAR is coded directly [15:04] <JEEB> and those IIRC in the specs are set to specific ones [15:04] <JEEB> like you have a value for 40:33 [15:04] <JEEB> etc [15:04] <kierank> yes they are set to the bt601 values [15:04] <kierank> and the simple HD ones [15:04] <JEEB> yup [15:05] <JEEB> so unless someone did some funky set-up there it should be pretty simple to get the correct one, since it should in correct cases be already set [15:06] <Daemon404> i am surprised nobody has linked The Link yet [15:07] <JEEB> yeah, just rechecked the spec and it seems simple enough, unless someone set the value incorrectly (each of the aspect_ratio_idc values has a SAR except for zero) [15:07] <JEEB> although that is "not present" [15:07] <kierank> Daemon404: which one [15:07] <Daemon404> 1 sec [15:07] <JEEB> but yeah, with MPEG-2 it should be simple enough [15:08] <JEEB> might post a patch at some point and get all the hatred [15:08] <Daemon404> hmm [15:08] <Daemon404> http://lipas.uwasa.fi/~f76998/video/conversion/ [15:08] <Daemon404> but its gone now [15:08] <Daemon404> :( [15:08] <kierank> i have "The Link" for interlacing [15:08] <JEEB> Daemon404, that's what nicholas is actually holding upon for "nv is doing the wrong thing" [15:08] <Daemon404> heh [15:09] <JEEB> unless it also explains the simplified versions" [15:09] <Daemon404> whats there to explain though? [15:09] <Daemon404> it's just "shits in a standard" [15:09] <kierank> that's what i'm saying [15:09] <JEEB> ^ [15:09] <kierank> but he wants things to mathematically work [15:10] <JEEB> should I take a look at mpegdec? [15:10] <JEEB> at how it sets the sar/dar? [15:10] <Daemon404> [23:50] <@Daemon404> nicholas does not live in reality [15:10] <Daemon404> [23:50] <@Daemon404> just ask wm4. [15:10] <wm4> once nicholas thinks he's right, there's nothing you can do [15:10] <Daemon404> pretty much [15:11] <JEEB> and I guess you'd also have to poke the raw video formats? [15:11] <JEEB> because when you capture shit it generally should be BT.601 if it's SD [15:12] <JEEB> (and that would be an even bigger can of worms tbqh) [15:14] <kierank> ok i posted the links [15:16] <JEEB> hmm, where did mpegvideo set the sar/dar? [15:16] <kierank> in the sequence header there is a DAR fiel [15:16] <kierank> field somewhere [15:16] <JEEB> mpegvideo_extract_headers doesn't seem to touch it at least [15:16] <JEEB> where it sets width/height etc [15:16] <kierank> loool [15:17] <kierank> it must do [15:17] <JEEB> https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/mpegvideo_parser.c#… [15:17] <JEEB> sequence extension [15:17] <JEEB> a lot of shit gets set [15:18] <JEEB> except for sar/dar :D [15:18] <nevcairiel> the actual decoder sets its anyway [15:18] <nevcairiel> thats the parser [15:19] <JEEB> so it's in mpegvideo.c? [15:19] <JEEB> I didn't notice it :/ [15:19] <nevcairiel> its in mpeg12dec.c [15:19] <JEEB> ugh [15:20] <JEEB> I don't know why I was looking only at mpegvideo [15:20] <wm4> I thought parser and decoder code was shared to some extend, and that's why parsers have an AVCodecContext... [15:22] <nevcairiel> mpeg2 aspect ratio info is kinda confusing in the code, there is aspect_ratio_info in the sequence header, but it also takes the pan_scan info into account when calculating the final aspect ratio .. probably because of dar/sar differences [15:22] <wm4> ah [15:23] <kierank> i have no idea why it cares about pan scan [15:24] <nevcairiel> AVRational dar = av_mul_q(av_div_q(ff_mpeg2_aspect[s->aspect_ratio_info], (AVRational) { s1->pan_scan.width, s1->pan_scan.height }), (AVRational) { s->width, s->height }); [15:24] <nevcairiel> thats what it does :D [15:24] <kierank> yeah i saw [15:24] <JEEB> yeah, I would simplify a lot of that shit [15:25] <JEEB> meanwhile I really wonder what should be done with MPEG-2 where the colorimetry is set to BT.709 specifically [15:25] <JEEB> Japan airs such stuff in 720x480 [15:27] <JEEB> I should check the broadcast specs for that [15:34] <JEEB> thanks for the ETSI doc btw kierank [15:35] <JEEB> reading the japanese side of things too [15:39] <JEEB> anyways, food and then maybe something :3 [15:45] <Daemon404> kierank, fyi i mirrored the other link here: https://chromashift.org/video/ [15:45] <Daemon404> (because it is easier for me to remember) [15:45] <kierank> I was going to do that [15:45] <kierank> but well done [15:46] <kierank> the notable part of that document is that it doesn't use SAR anywhere [15:46] <kierank> SAR is a construct of ffmpeg and pals [15:46] <kierank> i mean the way SAR is calculated is a construct [15:58] <JEEB> wow [15:58] <JEEB> the ARIB spec sure has nice tables [16:08] <JEEB> kierank, how do you read tables 6-5 to 6-7 http://www.arib.or.jp/english/html/overview/doc/6-STD-B21v5_4-E1.pdf [16:08] <JEEB> of chapter 6.1 [16:08] <kierank> s000 slow [16:10] <JEEB> because my reading of table 6.7 (output to a progressive 720p screne) [16:10] <JEEB> seems to hint that no cropping should be done [16:11] <JEEB> it does use the wording "in 750p format" [16:11] <kierank> 750 i guess because of vbi [16:12] <JEEB> 480 * (3/2) and 720 (* (16/9)) [16:12] <JEEB> is what it says to do for interlaced 720x480 [16:12] <JEEB> as far as I can tell [16:12] <JEEB> which hits 1280x720 [16:14] <JEEB> I'm just trying to understand if this means that they indeed use BT.709 stuff [16:14] <JEEB> instead of BT.601 aspect ratios [16:15] <JEEB> the colorimetry is specified to always be BT.709 at least [19:24] <rcombs> that doesn't look like a crash to me [20:44] <cone-798> ffmpeg.git 03Muhammad Faiz 07master:eba5cb2028f1: avfilter/showcqt: add gamma2 option [21:07] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:8d7ce5cdb707: avformat/nsvdec: Check the return value of av_get_packet() [21:07] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:65b83ce01b94: avformat/rmdec: Check the return value of av_get_packet() [21:56] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:2a3af7728441: avcodec/svq1enc: Check memory allocations [21:56] <cone-798> ffmpeg.git 03Michael Niedermayer 07master:79888388e7c4: avcodec/svq1enc: fix error handling/cleanup in case of ff_get_buffer() or scratchbuffer alloc failure [22:09] <cone-798> ffmpeg.git 03Mark Reid 07master:b396914c6813: libavformat/mxfdec.c: support demuxing opatom audio without index [22:20] <ubitux> wm4: well, it felt like it was the exact purpose of dts to deal with reordering... now does dts make any sense in a text subtitle context? [22:28] <wm4> well it would be nice if subtitles wouldn't just blatantly break assumptions about PTS which are otherwise always true [22:29] <wm4> also, it seems PGS uses PTS for a real purpose [22:29] <ubitux> you mean dts? [22:32] <wm4> err [22:32] <wm4> yes [22:33] <ubitux> i wanted to apply that just for text subtitles [22:33] <ubitux> but i guess i need to find another way? [23:44] <kierank> ubitux: for closed captions it does [23:44] <kierank> it keeps a buffer in memory and then presents them [23:46] <nevcairiel> pgs uses them the same way [00:00] --- Mon Jan 19 2015

1 0

ffmpeg.log.20150118
by burek 19 Jan '15

19 Jan '15

[01:53] <windu> im getting an error when trying to compile ffmpeg on centos6.6 [01:54] <windu> anyone on that is able to help? [01:55] <stefkos> im not an expert here, can you paste error? [01:55] <windu> configure: error: unrecognized option: --extra-cflags=-I/home/windu/ffmpeg_build/include [01:56] <stefkos> hmm did you check configure script? [01:56] <windu> here is my cmd [01:56] <windu> configure: error: unrecognized option: --extra-cflags=-I/home/windu/ffmpeg_build/include [01:56] <windu> sorry [01:57] <windu> PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig" ./configure --prefix="$HOME/ffmpeg_build" --extra-cflags="-I$HOME/ffmpeg_build/include" --extra-ldflags="-L$HOME/ffmpeg_build/lib" --bindir="$HOME/bin" --enable-gpl --enable-nonfree --enable-libfdk_aac --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 [01:58] <stefkos> hmm [01:58] <windu> this is per the wiki for centos [01:58] <stefkos> configure says it exist [01:59] <stefkos> windu, maybe check ' then " [01:59] <stefkos> just guessing [02:00] <windu> ok...im looking at the wiki now [02:01] <windu> could there be a typo on the wiki possible? [02:01] <stefkos> dont think so [02:01] <stefkos> but maybe console need that? [02:01] <stefkos> or maybe try without this line --extra-cflags [02:02] <stefkos> "--extra-cflags="-I$HOME/ffmpeg_build/include"" [02:02] <windu> ok let me try that [02:02] <stefkos> but thats weird [02:02] <stefkos> I was building ffmpeg for android last time [02:03] <stefkos> let me check [02:03] <windu> lol..took it out they i get erro for the --xtra-ldflags line [02:04] <stefkos> on ubuntu [02:05] <stefkos> ./configure $FFMPEG_FLAGS --extra-cflags="...." \ ... [02:06] <windu> hmmm ill try to do this on my debian box and see if i get similar error [02:06] <stefkos> well it works here on ubuntu... at least something similar [02:07] <stefkos> your $HOME is defined somewhere? [02:07] <stefkos> HOME = /home/stefkos fe. [02:08] <stefkos> sorry for guessing:) Im trying to help...and give some hints [02:08] <windu> yes I just ran echo $HOME and it rported back /home/windu [02:08] <stefkos> ok [02:08] <stefkos> but you put echo inside your script? [02:08] <windu> no i just ran it alone [02:09] <stefkos> put HOME = /home/windu/ before ./configure [02:10] <stefkos> maybe somehow he cannot find this global variable and "put some shit" [02:11] <windu> naw same error [02:11] <stefkos> ok [02:11] <stefkos> dont have new ideas now...and need some sleep [02:11] <stefkos> good luck [02:12] <windu> cool thx for the help [02:12] <klaxa> windu: pastebin your config.log [02:12] <windu> ok stand by [02:14] <windu> wait. i think i may have it [02:14] <windu> lol..im such an ass. i was in teh worng directoy [02:14] <klaxa> haha [02:49] <misterno> should i switch completely to h.265? [03:06] <c_14> In short, probably not. [03:07] <misterno> so if i should still stick with h.264 when i make my videos [03:08] <c_14> Probably, yes. Unless you have unlimited amounts of cpu-time and like playing around with new things. [03:11] <misterno> no i was just thinking making my desktop screencast videos would take less space encoding in h.265 [03:11] <misterno> and who knows maybe even better quality [03:12] <klaxa> sounds like you are looking for h265 real-time encodes. buy asics or come back in 2020 [03:13] <misterno> lol wut? [03:13] <misterno> brb [03:14] <klaxa> if he comes back in 2020 for the first time again i'll be impressed [03:14] <c_14> He's just spinning up ye ole time machine [03:16] <klaxa> that's funny because time machines should be somewhat immune to age [03:17] <klaxa> hmm... or are they? [03:18] <c_14> Depends, time machine as in accelerating an object as close to the speed of light as possible: that one will age, but only very slowly when in transit. It'll age normally out-of-transit though. [03:19] <klaxa> looks like he doesn't have a time machine after all... [03:19] <klaxa> well there goes Plan T for world domination [03:22] <BtbN> nvidia has a h265 hardware encoder which should be able to do it in real time. Intel also has one, but without library support for now. [03:22] <c_14> With decent compression support? [03:25] <rcombs> with comparable or better performance than x264? [03:25] <BtbN> No idea, i wasn't yet able to test it, because it only works on the most expensive cards. [03:25] <BtbN> So far the gtx980 is the only consumer card that supports it [03:35] <fSCatxycle> hi guys, i have a question. I'm new to this, and it's hard for me to understand the answer to this question: http://stackoverflow.com/questions/5784661/ffmpeg-convert-entire-directory [03:36] <fSCatxycle> Shouild it be http://pastebin.com/99auZmhK [03:37] <klaxa> >E\:physio [03:37] <klaxa> that makes me think you are using windows [03:37] <fSCatxycle> I am [03:37] <klaxa> can powershell run for loops? [03:37] <fSCatxycle> hm I have no idea [03:38] <BtbN> even basic windows cmd can, but no idea about the syntax for it [03:39] <fSCatxycle> hnm. But is my placement right? Of the E\physio I mean, and quotes [03:40] <c_14> if you were using bash, maybe [03:40] <fSCatxycle> Because I can just try, and see if it doesn't work. I jsut want to make sure it's first not going to like reencode everyhing in E for some reason cuz of my placement [03:40] <klaxa> that's pretty windows specific i don't know the first thing about windows shell [03:41] <fSCatxycle> I got "i was unexpected at this time" [03:41] <BtbN> Just use the answer starting with "And on Windows:" [03:41] <BtbN> Or install bash [03:42] <fSCatxycle> wow I'm dumb, I missed that [03:44] <fSCatxycle> It uses flac and such though, I have no idea how to insert my directory in there and make it from mov to avi [03:47] <c_14> FOR /F "tokens=*" %G IN ('dir /b E:physio\*.mov') DO ffmpeg -i "%G" -c copy "%~nG.avi" [03:48] <c_14> probably anyway [03:48] <fSCatxycle> ok thanks I'll try [07:07] <tmh1999> Hi guys, I hope someone could help me. I build ffmpeg from source code in Centos 6.6 as the following the guide : https://trac.ffmpeg.org/wiki/CompilationGuide/Centos. I just checked and it seems libavresample didn't get built. Is that ok? [07:07] <tmh1999> Thanks [07:40] <tmh1999> And I get the undefined references to "av_get_cpu_flags" error. I did try this https://github.com/xplodwild/bbqscreen_client/issues/2 but it doesn't work. [07:40] <tmh1999> Hope someone could enlighten me. Thanks a lot. [07:47] <tmh1999> Oh this linker order works for me : -lavformat -lavcodec -lswscale -lavutil -lSDL -lm -lz -lswresample -lavdevice -lavfilter -lpostproc -lavfilter -lrt -lx264 -lmp3lame -lvpx -lyasm -lfdk-aac [11:36] <elii> How do I make sure that ffmpeg uses the latest x264 that I just installed? [11:36] <elii> I specify libx264 in the command, but I dont think that it uses the binaries that I installed [11:37] <elii> I followed this guide: https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [11:38] <elii> I want to use the latest x264 version (r2525) [12:02] <Mavrik> elii, is your ffmpeg statically built or not? [12:03] <Mavrik> elii, since that guide will make a static build - that is, it's going to bake in x264 into ffmpeg binary [12:03] <Mavrik> which means you have to recompile x264 and ffmpeg to get the new one [12:03] <elii> aha [12:03] <elii> how should I compile ffmpeg then? [12:06] <Mavrik> um [12:06] <Mavrik> Don't understand the question. [12:06] <elii> I guess my ffmpeg build is static [12:06] <Mavrik> Static builds are ok, but updating dependant libs requires you to recompile ffmpeg. [12:07] <Mavrik> Dynamic builds require you to have all the dependant libraries available as separate files, but usually they can be updated without recompiling ffmpeg as long as major version doesn't change. [12:07] <elii> yea, it is fine. I can recompile it. But how do I recompile it so it "takes" the new x264 [12:07] <Mavrik> elii, follow the libx264 part again (which will compile and install new x264 to your /usr/local) [12:08] <Mavrik> that will replace old libx264, so when you follow the ffmpeg step again to recompile it, it'll grab the new x264 [12:08] <elii> hmm, I followed those steps before [12:09] <elii> hmm [12:09] <elii> I guess I can redo all steps [12:11] <Mavrik> you don't have to redo all steps, just x264 and ffmpeg if you have rest of the libs ready :) [12:12] <elii> But, I should be able to confirm the correct x264 version after the x264-step? [12:12] <elii> with like x264 --version or something? [12:15] <elii> Strange that http://download.videolan.org/pub/x264/snapshots/last_x264.tar.bz2 is the r2245 version I think [12:15] <elii> not the newer r2525 [12:15] <elii> eli@workstation:~/ffmpeg_sources/x264-snapshot-20141218-2245$ pwd [12:15] <elii> /home/eli/ffmpeg_sources/x264-snapshot-20141218-2245 [12:15] <elii> after extraction [12:22] <elii> Mavrik: I got no x264 in my /usr/local after compiling [12:28] <Dimtree> Someone have an idea how to get libfaac.so to be seen by ffmpeg? I can't seem to figure out pkg-config. Relevant script at http://pastebin.com/ptUqrD6W [12:29] <JEEB> at this point I would recommend not using faac [12:30] <JEEB> it is nonfree (as in, not GPL unlike its license says) and in nonfree encoders fdk-aac is now better than that [12:30] <JEEB> (nonfree in the --enable-nonfree meaning, you can't distribute the binaries) [12:31] <Dimtree> JEEB: It's a prerequisite of a package I don't control, is fdk-aac a drop-in equivalent? I don't plan to redistribute or violate GPL [12:32] <JEEB> so it's something else than ffmpeg? [12:32] <Dimtree> It's another program calling ffmpeg and expecting that specific encoder by name [12:32] <JEEB> and you can most probably switch that around [12:33] <JEEB> so as long as the encoder name can be switched it's a drop-in replacement [12:33] <JEEB> Dimtree, also it's not a case of violating GPL, it's just not compatible with (L)GPL, which makes the end result not distributable [12:33] <Dimtree> I have absolutely no way to switch the encoder name in the application [12:34] <JEEB> ok [12:34] <JEEB> then you either patch the fdk-aac wrapper in libavcodec or try to use faac [12:35] <Dimtree> So how would I go about getting ffmpeg to see libfaac.so? [12:35] <JEEB> have you done ldconfig? [12:35] <JEEB> with root [12:36] <JEEB> oh, a nonstandard path [12:36] <Dimtree> This is a freeBSD box [12:36] <JEEB> uhh [12:36] <Dimtree> I'm also compiling faac [12:36] <JEEB> I'm looking at that pc file [12:37] <Dimtree> Yeah, it's probably completely wrong [12:37] <JEEB> 1) why in cflags you have the file name in -I (it's supposed to set a directory, like -L with libs) [12:37] <JEEB> 2) usually it's -l<name of the file without "lib"> [12:38] <JEEB> also is that way of having \ there correct in the prefix etc. ? [12:38] <JEEB> and then we get to building ffmpeg, PKG_CONFIG_LIBDIR is once again a directory [12:38] <Dimtree> The \$ is just escaping, it comes out intact [12:39] <JEEB> and it's supposed to be prefix/lib/pkgconfig usually :P [12:39] <JEEB> I'm surprised why you don't have a pc file come with faac though, if ffmpeg expects one [12:39] <JEEB> sounds like an old version or whatever [12:39] <Dimtree> In PKG_CONFIG_DIR, I should have libfaac.so and libfaac.pc ? [12:39] <JEEB> no [12:40] <JEEB> as I said, things usually install their pc files to prefix/lib/pkg(-)config [12:40] <JEEB> but as long as the pc file can be found there with the correct values that should work [12:41] <JEEB> also I must say that the prefix being set to tmp thing doesn't really reek of "I want to do this correctly" [12:41] <JEEB> even though the needed files might be installed to correct places [12:42] <JEEB> also I must say that at least with lunix machines trying to build shit with static libraries is much easier than with shared ones if I'm using a custom prefix [12:42] <Dimtree> Yeah, this is probably the worst way of doing this with --prefix-/tmp , but it's the only way I can get it to not puke files everywhere [12:43] <JEEB> that is why you usually reserve a directory for the prefix in your homedir [12:43] <JEEB> instead of using your homedir [12:43] <JEEB> :P [12:45] <Dimtree> so, leaving the clusterf**k aside, I need to properly format libfaac.pc, but how do I get pkg-config to see it? [12:45] <JEEB> PKG_CONFIG_LIBDIR being set to the directory, not the file :P [12:46] <JEEB> pkg-config will then scan the dir for what the configure script requests [12:46] <Dimtree> And it searches for *.pc in there? [12:46] <JEEB> yes [12:46] <Dimtree> Ah, ok. This is making sense now. Thanks [12:46] <JEEB> which is why in general the pc files are installed to prefix/lib/pkgconfig [12:46] <JEEB> instead of prefix/lib [12:47] <Dimtree> pkgconfig being a directory, or a file with everything? [12:47] <JEEB> dir [12:47] <Dimtree> Ah, ok [12:47] <Dimtree> In my defence, I was hungover when I wrote this [12:48] <JEEB> also there are two pkg-config variables usually used. one of them adds a dir to the list of default ones for this pass, and the other overrides the whole list [12:48] <JEEB> PKG_CONFIG_LIBDIR is the one that overrides the whole thing [12:48] <Dimtree> I only have the one, so it should be fine? [12:49] <JEEB> yeah, you're probably fine. I'm just saying that so you know what you've set there [12:49] <Dimtree> gotcha [12:49] <JEEB> also I'm really surprised if your faac thing didn't create a pc file [12:51] <Dimtree> Nothing in ./lib or ./include [12:51] <JEEB> try setting just prefix and making it a directory within home instead of having all the stuff right inside home [12:51] <Dimtree> will do [12:53] <JEEB> yeah, checked the debian patches and none of them adds pkg-config stuff as far as I can see... [12:53] <JEEB> so I would be really surprised if faac didn't create one :) [12:55] <JEEB> also I guess I'm helping you with using some closed source proprietary crap that you're getting paid for :P [12:55] <Dimtree> I'm trying to get a networked cable tuner to stream to a TV for a church [12:57] <Dimtree> And I donated to the FFmpeg project, don't worry :P [12:58] <JEEB> uh-huh. I don't care about that at all tbqh :P [12:58] <JEEB> also I don't think it's impossible to switch the encoder from that proprietary blob. you can use a tool to modify the strings in the binary [12:58] <JEEB> but it's probably simpler to patch the libfdk-aac encoder to be called faac [13:00] <Dimtree> Here's a find . of the prefix dir: http://pastebin.com/jxynBQ7c [13:00] <Dimtree> No .pc file in sight [13:00] <JEEB> intesting [13:03] <JEEB> then just use --extra-cflags="-I/your/prefix/include" and --extra-ldflags="-L/your/prefix/lib -lfaac" or so [13:59] <Dimtree> Ok, now I'm having an issue with the makefile, 'PATH="$HOME/ffmpeg_build/bin:$PATH" make' throws http://pastebin.com/0SuYawPx [14:01] <JEEB> use gmake [14:03] <Dimtree> Derp, thanks [14:04] <Dimtree> Would that also be 'gmake install', and etc.? [14:04] <JEEB> yes [14:09] <Dimtree> So, I compiled with fdk-aac and ran 'sed -i.bak s/faac/fdk-aac/g' on the binary, and surprisingly it seems to work [14:09] <JEEB> lol [14:09] <JEEB> I'm not sure how the string structures in binaries work [14:10] <JEEB> depending on where they are it might have fucked up some lengths and offsets [14:10] <Dimtree> Also, I now hate BSD [14:11] <Dimtree> It segfaulted I think [14:11] <JEEB> I would recheck the structure of BSD binaries and make sure that when you modify the binary it doesn't fuck anything up [14:11] <JEEB> because just using sed prolly fucks it up [14:12] <Dimtree> apparantly sed has a --binary flag [14:13] <Dimtree> But yeah, it shits all over the binary structure [14:14] <JEEB> personally I would just patch the fdk-aac wrapper to be called faac [14:14] <JEEB> that is simpler [14:17] <Dimtree> I'll investigate that, seems straightforward enough [14:21] <JEEB> Dimtree, also I recommend making sure that you're not on the default gcc 4.2 on that bsd [14:22] <Dimtree> Err, how bout 4.2.1? [14:23] <JEEB> I recommend you install either 4.8.x or 4.9.x [14:23] <JEEB> latest version of either branch [14:29] <JEEB> bsd things generally tend to have an ancient gcc installed because they keep away from GPLv3. thankfully they usually package a newer one as well [15:19] <relaxed> Dimtree: using clang instead of gcc will save you some time [15:20] <relaxed> pass --cc=/path/to/clang to ffmpeg's configure [15:21] <foonix> btw, is there any noticeable speed difference between ffmpeg (+libx264) on bsd/linux? [15:23] <relaxed> never tested but I wouldn't think so [15:23] <foonix> best way is to test it myself, but just wondering.. [15:23] <JEEB> no [15:23] <foonix> ok, good to know [15:57] <elii> Might ass well ask again: If I want x264 r2525 (got the binary from videolan.org) to work with ffmpeg [15:57] <elii> Where should I put the x264 binary? [15:57] <elii> when compiling ffmpeg [16:04] <c_14> IIRC ffmpeg wants the libraries, not the binary. [16:06] <elii> I followed this guide: https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [16:07] <elii> Yea I am a bit lost [16:07] <elii> I want ffmpeg to use this x264: http://download.videolan.org/x264/binaries/linux-x86_64/x264-10b-r2525-40bb… [16:09] <elii> Anyone know how I should do to accomplish that? [16:09] <c_14> That's just the binary. You need the libraries [16:09] <elii> aha, where do I get those? [16:10] <elii> by compiling my self? [16:10] <c_14> ye [16:10] <elii> c_14: Or can I download them some how? [16:10] <c_14> Not that I know of [16:10] <__jack__> elii: just clone the git, compile the lib, compile ffmpeg [16:11] <elii> I am a bit new to all this compiling it my self [16:13] <elii> Do I have to remove the currently wrong libraries? or will they get overrided by me compiling the new x264? [16:15] <c_14> You compile the libraries, install them somewhere and then add that path to your CFLAGS and LDFLAGS and then ffmpeg will compile against those [16:15] <elii> __jack__: Got new new x264 compiled. Do i need to do something in order for ffmpeg to know which libraries to include when compiling it? [16:15] <elii> c_14: ok [16:17] <elii> c_14: Uhm, how do I know where x264s libraries got installed? [16:17] <elii> i just did: make; make install [16:17] <c_14> Did you set a prefix ? [16:17] <elii> nno [16:18] <c_14> Then it's in /usr/local/{lib,include} [16:18] <elii> I feel a bit lost, sorry about these basic questions [16:18] <elii> okey, yea [16:18] <elii> and now how do i configure ffmpeg? [16:19] <elii> --extra-cflags="-I/usr/local/include" ? [16:19] <elii> and the other one to ../lib? [16:19] <c_14> Like it says on the CompilationGuide/Ubuntu page except that you need to add /usr/local/lib/pkgconfig to PKG_CONFIG_PATH and with the include dir added to the cflags and the libdir to ldflags [16:19] <__jack__> elii: make sure you compiled x264 with the --enable-shared flag [16:20] <elii> __jack__: I did [16:20] <__jack__> ok, now if you have installed libx264 from your distribution, they might conflict; else, just compile ffmpeg (./configure --enable-libx264 && make) [16:21] <elii> __jack__: But what about the flags c_14 is talking about? [16:21] <elii> I did not have a previous x264 [16:22] <__jack__> not required, I guess, ffmpeg's configure will find the lib if they are in common places (they are by default) [16:22] <elii> okey [16:22] <__jack__> if the configure say something like "cannot find libx264 headers", have a look at them [16:23] <elii> looks alright [16:24] <elii> so every time I compile it goes to /usr/local/ ? [16:25] <elii> or, make just creates the files and make installs move them to /usr/local/ [16:29] <__jack__> make install moves files in places, commonly in /usr/local (but not always) [16:34] <elii> ok thanks [16:34] <elii> Think I got it sorted now [16:34] <elii> hopfully [17:37] <netw1z> how can i get a good high contrast look using ffmpeg on videos? [18:13] <opus> question [18:25] <martindelille> Hi, [18:25] <martindelille> where is the file_size member mentioned here: http://ffmpeg.org/doxygen/trunk/structAVFormatContext.html#a4f7a7b9b1313907… [18:26] <martindelille> I'm having a problem with an AVStream with negative value [18:35] <opus> so i was gonna ask: when i use ffmpeg in command mode, is it possible to make to convert something 1:1 to another format (say, m2v/m1v to mpg/mp4), loosing little to no quality whatsoever? [18:36] <opus> like, can i just type "ffmpeg -i input.m2v output.mp4" or something or does it not work like that [18:36] <JEEB> default mode is whatever encoder ffmpeg happens to pick by default, stream copy is not done by default [18:36] <JEEB> you need to set -c copy for stream copy [18:37] <opus> do i put the -c after the -i or does it not matter where it's positioned [18:39] <JEEB> after -i yes [18:43] <opus> okay, weird [18:44] <opus> i tried doing the command [18:44] <opus> ffmpeg -i [input file] -c [output file] [18:44] <JEEB> uhh [18:44] <JEEB> -c copy [18:44] <opus> and ffmpeg said "at least one output file should be specified" [18:45] <opus> i tried "ffmpeg -i -c [input file] [output file]" but it said that "-c: no such file or directory" [18:45] <JEEB> yes [18:45] <JEEB> not -c [18:45] <JEEB> but -c copy [18:45] <opus> OH [18:45] <JEEB> it sets the "codec" option to "copy" [18:45] <opus> i thought you meant that "c- = copy" [18:45] <opus> sorry [18:54] <opus> ok i got what i needed, thanks [23:08] <fmiz> Hi all. I'm trying to use ffmpeg to live stream to youtube. This is the command I use: http://pastebin.com/U6WbTTgW . After about 1 hour ffmpeg still runs fine, youtube says "stream status good", but actually the video isn't playing online, it's stuck [23:10] <fmiz> Before locking completely, there are huge artifacts in the video frames, for example half the frame sometimes is grey [23:13] <sfan5> your audio bitrate seems to be low, but that is probably not a problem [23:13] <fmiz> I don't know where to look for errors anymore. The last time I had this issue I was running ffmpeg for fedora 21 from rpmfusion, the rpm version is ffmpeg-2.4.6-1.fc21 [23:14] <fmiz> Yes, I keep it really low because I don't need it, but I found that youtube does not accept the stream if it doesn't have also an audio stream [23:15] <fmiz> Also bandwidth shouldn't be an issue, upload should be at least something around 20 mb/s [23:25] <fmiz> ah, the source of video is a logitech c310 webcam [23:47] <netwiz> is there a easy way [23:47] <netwiz> to increase contrast on a black and white video? [00:00] --- Mon Jan 19 2015

1 0

ffmpeg-devel.log.20150117
by burek 18 Jan '15

18 Jan '15

[01:54] <heffer2k02> Does anyone understand AAC + MP4 internals a bit? FFmpeg is reported that my audio file has a "start offset" of 0.02333 (1024 samples / 44100), i.e. a single audio frame. Looking at ffprobe, it doesn't reveal and delay (first pts is 0). Digging further, the MP4 files first sample actually points to an AAC FIL frame with some string indentifying the encoder software, however it has a duration of 1024... Is this a bug in ffmpeg? [01:54] <heffer2k02> Or is FIL frame interpreted as silence? Confused... [02:02] <thardin> ask on the mailing list maybe? [02:10] <aetasx> where does it report the start offset? [02:11] <cone-576> ffmpeg.git 03Michael Niedermayer 07master:040e5703e6b8: avcodec/flac_parser: print error messages in case of errors [02:11] <cone-576> ffmpeg.git 03Michael Niedermayer 07master:c4d85fc23c10: avcodec/flac_parser: fix handling EOF if no headers are found [02:18] <heffer2k02> aetasx: ffmpeg -i gives a line like: Duration: 00:00:16.19, start: 0.023220, bitrate: 78 kb/s [02:20] <heffer2k02> aetasx: I've just been doing some ffprobes and managed to get this line: "start_time": "-0.023220". Which is negative, so now I'm more confused. The negative may make sense though as it offset the apparent "delay" inserted by the MP4 giving the FIL frame a duration of 1024. Although I might be reaching. [02:21] <heffer2k02> Not sure how ffmpeg is computing these values. [02:27] <aetasx> you sure thats not just the information you see when pulling up an mp4 file in mediainfo? [02:28] <heffer2k02> mediainfo doesn't actually say anything about start offsets... [02:29] <aetasx> in ffprobe, whats the actual file start_time listed as? [02:29] <aetasx> vs the stream start_time [02:30] <heffer2k02> FORMAT section and STREAM section both report: start_time=-0.023220 [02:31] <aetasx> and any other streams in there? [02:31] <heffer2k02> Nope [02:32] <heffer2k02> What does a negative start time usually acheive? Is it just applied as an offset to all pts? [02:32] <aetasx> usually its when having multiple streams, not sure what players do with it, probably drop it [02:33] <aetasx> try doing "ffmpeg -ss 0 -i <file> -c:a copy <outfile>" [02:43] <kierank> heffer2k02: aac start time is not a trivial problem [02:43] <kierank> https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFAp… [02:43] <heffer2k02> aetasx: Ah, so that produced a file that now reports 0 as start time. The first frame pts now reports as 0.02322. I should point out that this was originally extracted from a video, and it will be remuxed with it at a later time. Has this effectively changed the presentation time of the audio? [02:43] <kierank> and https://developer.apple.com/library/mac/technotes/tn2258/_index.html [02:46] <heffer2k02> kierank: thanks. these look useful. [02:46] <aetasx> thanks kierank [03:21] <heffer2k02> kierank: Thanks again. Was able to track this offset down to the elst box. [03:21] <kierank> the offset is correct iirc [13:30] <cone-46> ffmpeg.git 03Michael Niedermayer 07master:7532f0e14c5e: avcodec/libmp3lame: use av_malloc_array() [13:30] <cone-46> ffmpeg.git 03Michael Niedermayer 07master:1f4ef3ccaee6: avcodec/libopusenc: Use av_mallocz_array() [13:30] <cone-46> ffmpeg.git 03Michael Niedermayer 07master:f8ff76199c38: avcodec/mss12: Use av_malloc_array() [14:38] <arwa> does eq/eq2 filter need to be ported in one single filter? [14:40] <compn> i dont think so [14:40] <compn> but they are very similar [14:40] <compn> iirc [14:50] <wm4> having two of them makes no sense at all [14:51] <wm4> and it was probably only done because of shitty mplayer dev culture [14:51] <arwa> okay thanks :) [15:20] <ubitux> compn: yes they need to, obviously. [15:20] <ubitux> arwa: they're probably going to be merged into hue filter [15:20] <ubitux> one is a superset of the other [15:21] <arwa> Okay, so should I directly go for the merged filter? [15:21] <ubitux> i thought there was mail exchange about this [15:23] <arwa> Yeah, there was. But there was nothing mentioned about the merging part. [15:26] <ubitux> we just need the feature from eq & eq2 (which are the same basically) into hue [15:26] <ubitux> i think that was the plan, and you can look at J. Darnley code [15:26] <arwa> Yes, I was going through it when I realized that it is merged [15:28] <arwa> Does the arm code need to be ported? [15:29] <wm4> arm code? or x86 asm code? [15:29] <arwa> sorry the asm code [15:30] <wm4> I think porting the C first, and adding asm as yasm later would be a reasonable proposition [15:30] <wm4> although it could well be that this old mmx code is slower than the C code [15:30] <arwa> okay. [15:33] <ubitux> hue works in a LUT approach iirc [15:44] <tim> Hi! [15:51] <tim> nobody response? [15:52] <tim> *àJ [15:53] <compn> hey tim [15:53] <compn> what country are you from ? [15:53] <tim> china [15:53] <tim> and you ? [15:53] <compn> awesome. i'm from usa [16:33] <cone-46> ffmpeg.git 03Michael Niedermayer 07master:353cf95f948e: avfilter/vf_frei0r: also set AVFilterLink.frame_rate [16:33] <cone-46> ffmpeg.git 03Michael Niedermayer 07master:e0623953e1e6: avfilter/vf_frei0r: fix build with DEBUG enabled [16:33] <cone-46> ffmpeg.git 03Michael Niedermayer 07master:edf1cb7aeea0: avformat/nsvdec: Check av_get_packet() return value [16:33] <cone-46> ffmpeg.git 03Michael Niedermayer 07master:8de0990e9f22: avcodec/dvbsubdec: Check get_region() return value [18:31] <arwa> I added the attachments to a wrong page. How can I delete those attachments? [18:34] <cone-46> ffmpeg.git 03Timo Rothenpieler 07master:bc3f76779b6c: avcodec/nvenc: Add support for nvenc api version 5 [18:38] <michaelni> arwa, deleted them, only admins can delete stuff [18:39] <arwa> Okay, thanks and sorry. [20:28] <kierank> so what exactly is wrong with the nvidia SAR/DAR? [20:29] <kierank> anyhow the important thing is that the aspect ratio applies to the 702 pixels [20:30] <nevcairiel> but that stops being relevant for other resolutions [20:30] <nevcairiel> and apparently its applied everywhere [20:30] <kierank> ok [20:53] <cone-46> ffmpeg.git 03Michael Niedermayer 07master:4adf75cade69: avformat/rtpdec_hevc: Fix 2 "may be used uninitialized in this function" warnings [20:53] <cone-46> ffmpeg.git 03Michael Niedermayer 07master:ab161bfa4b3e: avformat/asfdec: Check the av_get_packet() return value [21:04] <kierank> sigh [21:04] <kierank> just because you can make a 720x576 testsrc in ffmpeg doesn't change the fact that the rest of the world does it right [21:04] <kierank> and has a 702 active picture [21:05] <nevcairiel> arent dvds typically 720x576 [21:06] <JEEB> according to the specs they should be with the active picture based AR though (often simplified) [21:06] <kierank> yes with 702 active pixels [21:06] <kierank> look at the downscaled parkjoy samples [21:06] <kierank> they do it correctly [21:06] <kierank> 9 pixels of grey on either side [21:06] <JEEB> that's also why the PSP crops the picture [21:06] <JEEB> when you encode 480p or 576p [22:41] <wm4> talking about ffmpeg advancements on #libav-devel is mean [22:47] <ubitux> yeah, sorry [22:47] <ubitux> i was highlighted there [22:49] <ubitux> j45: https://github.com/ubitux/FFmpeg/commit/a73a19cfc5c5e27d06b6e0cdb4ec06b1c45… just an experiment, but you get the idea [22:49] <ubitux> typically all the timing go out of the decoders [22:49] <ubitux> and you can see the readorder heuristic [22:50] <ubitux> the dts being in a bunch of demuxers (the one for standalone subs) handled @ https://github.com/ubitux/FFmpeg/commit/a7d1c5917c36b66f3bcdcaf369f12edbc34… (but i need to do that properly in mkv demuxer at least, and probably several others) [22:50] <ubitux> i could probably extract this https://github.com/ubitux/FFmpeg/commit/a73a19cfc5c5e27d06b6e0cdb4ec06b1c45… in a commit [22:50] <ubitux> to at least have the code properly refactored [22:51] <ubitux> but the AVTextSubtitles thing wasn't a good idea, i realized that too late [22:51] <ubitux> anyway, experiments. [22:51] <ubitux> will try to get done with this... but i'm on something more fun :) [22:56] <cone-46> ffmpeg.git 03Martin Storsjö 07release/2.4:932caa50a5de: libavformat: Check for malloc failures in avformat_new_stream [22:56] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:61335ab33e59: opt: check memory allocation [22:56] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:3280b86c85f5: swscale: check memory allocations [22:56] <cone-46> ffmpeg.git 03wm4 07release/2.4:b96163f055ee: avformat/utils: check for malloc failure [22:56] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:cdf37f25489f: Merge commit '932caa50a5dec7f2a69da8a417ccae84d19ae5e5' into release/2.4 [22:56] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:ee910aef38cb: Merge commit '61335ab33e594d47b84fde7367ee10dc66fc3465' into release/2.4 [22:56] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:dd4adba8112b: Merge commit '3280b86c85f5a88397fce0230d1878fee79fbb93' into release/2.4 [23:07] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:f6c82b34a320: segment: Fix the failure paths [23:07] <cone-46> ffmpeg.git 03Anton Khirnov 07release/2.4:bfe18be88a66: mpeg4audio: check the init_get_bits() return value [23:07] <cone-46> ffmpeg.git 03Alexandre Colucci 07release/2.4:3d0752d82f8e: xsub: Support DXSA subtitles [23:07] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:cc598bf7dc8a: Merge commit 'f6c82b34a320f105af266997f5951cbe7dfc8a05' into release/2.4 [23:07] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:05b1ac24709b: Merge commit 'bfe18be88a66da25b60a091de6011197dcb231fd' into release/2.4 [23:07] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:3cc8822c93e9: Merge commit '3d0752d82f8eaa326cff306ae50b0186a5b4d304' into release/2.4 [23:20] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:76e9a17f3392: lavfi: always check av_expr_parse_and_eval() return value [23:20] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:1411f073fdce: aacenc: correctly check returned value [23:20] <cone-46> ffmpeg.git 03Vittorio Giovara 07release/2.4:cbfdbba58e14: cmdutils: check file access functions return values [23:21] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:98d4a07ccf97: Merge commit '76e9a17f3392e752193015765e9216c2f0716b96' into release/2.4 [23:21] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:0afeb76e75f0: Merge commit '1411f073fdceeff1f39dbaa035c3c0275f69095f' into release/2.4 [23:21] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:e7b8fa2c005b: Merge commit 'cbfdbba58e1460bd0791911ad84a6c76b5500a0e' into release/2.4 [23:55] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:21683549edf4: h261dec: Fix context initialization sequence [23:55] <cone-46> ffmpeg.git 03Martin Storsjö 07release/2.4:cbf31d5f1577: rtpdec_h263_rfc2190: Clear the stored bits if discarding buffered data [23:55] <cone-46> ffmpeg.git 03Luca Barbato 07release/2.4:9bd4561d74a4: tiff: Check the check_size() return value and forward it [23:55] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:3c33983b7435: Merge commit '21683549edf436024cc3efbc4f8d1d55221c5336' into release/2.4 [23:55] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:c69fff3790ae: Merge commit 'cbf31d5f15774b3ffd1e2009159dc7154a767b09' into release/2.4 [23:55] <cone-46> ffmpeg.git 03Michael Niedermayer 07release/2.4:0295615f0a01: Merge commit '9bd4561d74a46dd6557140e286228e09e380674c' into release/2.4 [00:00] --- Sun Jan 18 2015

1 0

ffmpeg.log.20150117
by burek 18 Jan '15

18 Jan '15

[00:04] <voip__> Can you guys please check: http://pastebin.com/B6LXb7ik [00:04] <voip__> I taking stream from source and sending to 2 servers. Works well with transcoding, doesn't work when I copying video [00:05] <c_14> >[flv @ 0x2f100c0] Tag [27][0][0][0]/0x0000001b incompatible with output codec id '28' ([7][0][0][0]) [00:05] <c_14> The source video stream is incompatible with the output format. [00:22] <voip__> c_14, what kind of outuput shuld i setup for cmpaility ? How to fix ? [00:29] <c_14> voip__: Either get a different source, stop using rtmp, or reencode, not sure if there's another way to get around it [00:29] <voip__> c_14, same command works when i sending to only 1 server ffmpeg -i http://Input -acodec aac -strict experimental -ar 44100 -ac 2 -b:a 96k -vcodec copy -f flv rtmp://Output/live/ars [00:31] <c_14> it does [00:31] <c_14> hmm [00:32] <c_14> the tee muxer can be a bit iffy sometimes [00:32] <c_14> try adding the annexb bitstream filter [00:32] <c_14> or removing the global headers [00:33] <voip__> if i add annexb i tink ffmpeg vi;ll start transcode video, i dont wana use cpu power [00:34] <c_14> The bitstream filter shouldn't make ffmpeg start transcoding [00:36] <halfie> oh, I have a ADTS + AAC writer working now and it works good. however, mediainfo reports that my generated files are VBR and not CBR. any tips to fix this part? http://wiki.multimedia.cx/index.php?title=ADTS [00:37] <c_14> halfie: what's your current cmdline ? [00:37] <ac_slater_> hey all... with libavformat, what's a good way to avoid `avformat_find_stream_info` and set up a demuxer myself? [00:38] <voip__> i added -bsf:v h264_mp4toannexb , didnt work again [00:38] <halfie> c_14, umm, I wrote my own code for ADTS. http://pastebin.com/TuTWmt9k [00:39] <c_14> voip__: it's probably some sort of format flag or bitstream filter, not sure which. Try the working to single server one with -loglevel debug [00:39] <c_14> Maybe it outputs what filters are inserted [00:40] <halfie> c_14, I make sure that 0x7FF (VBR indicator) does not appear in my ADTS header but still mediainfo reports my files as using VBR. weird stuff. [00:40] <voip__> c_14, thanks i will debug [00:40] <c_14> halfie: maybe it's just mediainfo being weird, does it matter? [00:41] <halfie> c_14, yes, I think you are right. I was looking at mediainfo code and it isn't the greatest. seems to be doing nothing much. [00:44] <halfie> on to finding how the AES key is derived, so that I can modify "aacdec.c" to work with DRM files :) [01:14] <voip__> c_14, plaese check debugs http://pastebin.com/ZKsVut1c [01:21] <c_14> hmm, can't see it stating that a filter is added anywhere. It might be a bug. Could you report it on trac with the content of that pastebin and mention that it looks like the tag isn't being set correctly when you use the tee muxer? Oh, and just to make sure, can you try with only one output in the tee muxer? [01:25] <voip__> in http://pastebin.com/ZKsVut1c you can see 2 examples. with 1 output and 2 outputs [01:28] <c_14> I meant with one output in the tee muxer [01:32] <voip__> can you please type command ? [01:32] <c_14> ffmpeg/ffmpeg -i http://InputIP:8042/InputStream -loglevel debug -preset fast -acodec aac -strict experimental -ar 44100 -ac 2 -b:a 96k -vcodec copy -f tee -map 0:v -map 0:a -flags +global_header "[f=flv]rtmp://OutputIP/live/ars" [01:34] <voip__> tnx [01:46] <voip__> c_14, same problem cant start http://pastebin.com/EmNWfk38 [01:46] <c_14> Right, bug report time. [01:48] <voip__> c_14 OMG :) [01:49] <voip__> where is yor bug trcler or ticketing system ? [01:49] <c_14> https://trac.ffmpeg.org/ [01:53] <heffer2k02> Does anyone understand AAC + MP4 internals a bit? FFmpeg is reported that my audio file has a "start offset" of 0.02333 (1024 samples / 44100), i.e. a single audio frame. Looking at ffprobe, it doesn't reveal and delay (first pts is 0). Digging further, the MP4 files first sample actually points to an AAC FIL frame with some string indentifying the encoder software, however it has a duration of 1024... Is this a bug in ffmpeg? Is the [01:53] <heffer2k02> FIL frame interpreted as silence? Confused... [02:04] <voip__> c_14, thank you [06:39] <notnola> Howdy world! [06:40] <notnola> I have a queston about publishing streams to an RTMP server [06:41] <notnola> Actually - more on viewing streams for now. I should learn to handle one problem at a time :P [06:42] <notnola> Basicly, I am able to make the connection to the RTMP server via an external python lib and am able to view the other clients connected to it [06:42] <notnola> so I am aware of each playpath that each published stream would have [06:43] <notnola> problem is, I want to recieve upto 12 videos per single connection of the same RTMP server [07:33] <Matador> hmmm [07:33] <Matador> What does "Past duration %f too large" mean the problem relates to ? [07:42] <Matador> Past duration 0.98& too large [07:46] <Matador> I went down to 2 threads from 4, and no more error... [07:46] <Matador> actually n/m.. same error [07:59] <notnola> Does anyone kno how to save multiple live streams from the same rtmp connection? [10:14] <vladimir> hi to all [10:18] <eddiefingerz> greetings having a terrible time compiling ffmpeg with lame for arm64 http://pastebin.com/bLZ2185T thanks! [10:31] <x41> I have ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -c:v libx264 -b:v 500k -preset ultrafast test.avi ... the bitrate is being ignored. [10:34] <x41> k nvm [10:35] <Mavrik> why are you trying to put H.264 into an AVI container? O.o [10:36] <x41> not sure what container is recommended? [10:37] <x41> -f flv *.flv what is what I am going to be using in the end. [10:37] <Mavrik> well, then use flv. [10:37] <Mavrik> or mkv or mp4 [10:37] <Mavrik> the ones that actually support all metadata from H.264 video [10:37] <x41> what does that change? [10:37] <x41> ah [10:38] <x41> well [10:38] <x41> metadata is not that important [11:01] <vladimir> I get this error what is problem [11:01] <vladimir> Trailing options were found on the commandline. [ingenient @ 0x2c1cbe0] Could not find codec parameters for stream 0 (Video: mjpeg, none(bt470bg/unknown/unknown)): unspecified size [11:02] <vladimir> ffmpeg -i http://root:pass@192.168.1.52/axis-cgi/mjpg/video.cgi out.mjpeg [11:02] <vladimir> camera give as mjpeg stream on this url and it's work on vlc [11:03] <vladimir> I what to mix 2 stream in one... [11:04] <vladimir> I tray whit this [11:04] <vladimir> ffmpeg -i http://root:pass@192.168.1.51/axis-cgi/mjpg/video.cgi -i http://root:pass@192.168.1.52/axis-cgi/mjpg/video.cgi -filter_complex '[0:v]pad=2560x720;[0:v][1:v]overlay=1280:0[v]' -map '[v]' test.mjpg [13:52] <danell> Hi! I keep getting the "packet too small" message in the ffmpeg comand view while is working... the files seem to play afterwards but it is very unsettling. I have searched around but haven't found any solution. What does it mean that the "packet too small"? [15:05] <vladimir> hi [18:48] <Guest93839> If I have FFmpeg installed on my system, do I need to install it separately on my localhost in order to execute it from PHP? [18:49] <c_14> If you have ffmpeg installed locally, you do not need to install it again for use with PHP [18:49] <Guest93839> Inspite of that, I am not able to execute it using the exec() or shell_exec() command. [18:50] <c_14> Is it in your path? [18:50] <Guest93839> I am using ffmpeg -i file.mp4 file1.mpg [18:50] <Guest93839> The input file is in the same path as my php file [18:51] <Guest93839> Folder, same folder as my php file [18:51] <c_14> `which ffmpeg' [18:51] <c_14> `echo $PATH' [18:53] <Guest93839> Could you please give me a paste for this on pastebin? [18:56] <c_14> Not entirely sure what you're asking. [18:56] <Guest93839> Okay. [18:56] <Guest93839> I;ll start again. [18:56] <Guest93839> My PHP file looks like this. [18:57] <Guest93839> <?php exec('ffmpeg -i filename.mp4 fileout.mpg); ?> [18:57] <Guest93839> This should ideally generate an mpg file, right? [18:57] <Guest93839> However, when I run it through my localhost, it returns nothing. [18:58] <Guest93839> The file filename.mp4 and my php file are in the same directory. [18:58] <c_14> That single ' looks wrong. [18:59] <c_14> Other than that, make sure that ffmpeg is in the PATH of whatever is executing php [18:59] <Guest93839> okay, I'll give it another shot. [19:00] <Guest93839> You mean, the ffmpeg.exe file should be in the same directory as the PHP file? [19:01] <c_14> The ffmpeg binary should be in a directory referenced by the PATH environment variable from the environment of the process executing php [19:03] <Guest93839> Hmm.. I''ll give it a try. [20:14] <Nitori> .exe [20:54] <geraldn> trying to transcode an mpeg2 stream to h264. And for some reason, the output stream has a lot more frames than the input stream. Any ideas? [20:54] <geraldn> like 140,000 frames for mpeg and 180,000 frames for the h264 stream [20:55] <geraldn> right now i am attempting to use x264 directly to see if it does the same thing [21:08] <geraldn> Might have figured it out. Looks like "-vsync 0" keeps a 1:1 ratio of input and output framerate [21:14] <IsoDesu> klaxa JEEB and iive [21:15] <IsoDesu> the problem was that my system didn't have the fonts installed, I had to make fontconfig cache them with "fc-config dir" [21:16] <IsoDesu> now ffmpeg does the job wonderfully [21:16] <JEEB> yes, that's why you use something like mpv [21:16] <JEEB> no need to install the fonts from the file [21:16] <JEEB> :P [21:16] <IsoDesu> doesn't mpv use fontconfig's cache? [21:17] <JEEB> it uses fontconfig but it can load the fonts from the matroska file [21:17] <JEEB> for the player only [21:17] <JEEB> and not affecting anything else [21:18] <IsoDesu> for the player? but the others players also do that [21:19] <IsoDesu> you mean the same way mpv load it from the file itself, it does it too when encoding? [21:19] <JEEB> yes [21:19] <JEEB> I mean it uses the same stuff for playback as for encoding :P [21:19] <IsoDesu> hm [21:19] <JEEB> it just pushes the decoded pictures through possibly some filters and then to the encoders [21:19] <JEEB> and who here told me that ffmpeg can now handle the fonts from files :P [21:20] <JEEB> that person needs a spanking [21:20] <c_14> it works for me [21:20] <IsoDesu> why ffmpeg doesn't do that yet I have no idea... [21:21] <c_14> The output of ffmpeg corresponds exactly to what mpv produces [21:21] <JEEB> do you have the fonts installed system-wide? [21:21] <c_14> nope [21:21] <JEEB> well then I have no idea :P [21:21] <IsoDesu> maybe the fonts you use are common [21:22] <IsoDesu> anime fonts are stylish [21:23] <c_14> Most of the stuff I tested on were anime ops/eds [21:23] <IsoDesu> whats ur OS? [21:24] <c_14> linux [21:24] <IsoDesu> mine is Fedora 21 [21:24] <IsoDesu> I had to cache the fonts manually with fontconfig before hardsubbind the with ffmpeg [21:24] <IsoDesu> them* [21:25] <JEEB> anyways, fedora doesn't package libav* and friends at all, so that's some random packaged binary then [21:27] <IsoDesu> ok guys i'm going now [21:27] <IsoDesu> thanks for the support [21:52] <c_14> For anybody who cares, here are two screenshots. The first one taken with mpv from the source file with ass subs. The second taken with mpv after I hardsubbed the source file with ffmpeg and the subtitles filter. The fonts are Monotype Corsiva and DFKai-SB neither of which I have installed on my system and are only contained within the source mkv. http://dl.c-14.de/ffmpeg/mpv_ass_fonts.png [21:52] <c_14> http://dl.c-14.de/ffmpeg/mpv_ass_fonts.png [21:53] <c_14> eh, wait [21:53] <c_14> https://dl.c-14.de/ffmpeg/ffmpeg_hardsubbed.png [21:53] <c_14> I pasted the same picture twice accidentally. [21:55] <relaxed> looks good [21:56] <relaxed> there was a bug before where ffmpeg wouldn't use the embedded font [21:56] <relaxed> is that what was causing that user issues? [21:57] <relaxed> user's* [22:00] <c_14> Not sure, I never got to see his full console output with the subtitles filter, but from what I did saw he was using the 2.5.3 static build so that issue should have been solved. [22:08] <relaxed> hmm, my build? [22:09] <c_14> yep [22:23] <c_14> I just tested with your static build, looks fine with that as well. [00:00] --- Sun Jan 18 2015

1 0

ffmpeg-devel.log.20150116
by burek 17 Jan '15

17 Jan '15

[01:14] <cone-975> ffmpeg.git 03raxon1s 07master:4184d344af4a: ffmpeg: Add bitrate value under -progress action [02:59] <cone-975> ffmpeg.git 03Michael Niedermayer 07master:ba97cf2c4562: avdevice: Use av_format_get_control_message_cb() [02:59] <cone-975> ffmpeg.git 03Anshul Maheshwari 07master:a47c9331ef13: avformat: Improve data streams/codec support [02:59] <cone-975> ffmpeg.git 03Anshul Maheshwari 07master:bb0344dc5582: ffmpeg: improve data stream support [03:02] <aetasx> hey guys, when Im doing a static build with avx disabled, there are still the runtime tables that get build and make their way into the static libs. Is there something different about these that keeps them from being skipped like the others? [03:09] <Timothy_Gu> aetasx: what kind of runtime tables are you talking about? [03:11] <aetasx> Timothy_Gu: they're used in the h264 codecs. these ones ff_hevc_v_loop_filter_luma_8_avx.weakfilter [03:11] <aetasx> ff_vp9_ipred_h_16x16_avx [03:37] <kierank> those look like functions [03:39] <aetasx> they're the table generated functions related to the hardcoded-tables configure flag [06:47] <fffan> why av_frame_get_best_effort_timestamp does not work for wma audio for wmv? [13:09] <cone-530> ffmpeg.git 03Vittorio Giovara 07master:89df3fd49e99: swscale: fix warning about incompatible function pointer type [13:09] <cone-530> ffmpeg.git 03Michael Niedermayer 07master:a0aa3eaafc79: Merge commit '89df3fd49e9992441f680326902b4912d79f514f' [14:20] <cone-530> ffmpeg.git 03Vittorio Giovara 07master:2df7277711cb: swscale: fix gbrap to gbrap alpha scaling [14:20] <cone-530> ffmpeg.git 03Michael Niedermayer 07master:aee51ee8b9b8: Merge commit '2df7277711cbe3ca5415ff8e31228dcffa309211' [14:47] <cone-530> ffmpeg.git 03Nidhi Makhijani 07master:3941df546276: aea: Return proper error code on invalid header [14:47] <cone-530> ffmpeg.git 03Michael Niedermayer 07master:2d0440f93f93: Merge commit '3941df546276b190cc9362fd093e6721e8e52f50' [20:06] <cone-168> ffmpeg.git 03Michael Niedermayer 07master:261e8ccbd1e3: ffmpeg: drop redundant frame_pts value [20:06] <cone-168> ffmpeg.git 03Michael Niedermayer 07master:7b32856011d1: ffmpeg: Maintain higher precision pts between video filter and fps code [20:41] <cone-168> ffmpeg.git 03Michael Niedermayer 07master:d3d9a00bcdfd: ffmpeg: Use filter graph output frame rate also for frame duration estimation [20:45] <akira4> ubitux,do you have a minute? [21:14] <michaelni> argh, next time akira is here someone please tell her to ask her question as someone else than ubitux might know the awnser ... [21:14] <Daemon404> also also stay for more than 20 mins [21:14] <Daemon404> or use email [21:14] <michaelni> yes [21:41] <wm4> wow why is the flac parser so complex [21:43] <kierank> lovely http://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=7b32856011d1859a974b88… [21:43] <kierank> ffmpeg.c is such a masterpiece [21:44] <wm4> kierank: at least strict reviews ensure future contributions will be of high quality? [21:45] <kierank> wm4: yep so strict that the review was done in secret [21:45] <kierank> and by one person on his own code [21:49] <Daemon404> [20:41] < wm4> wow why is the flac parser so complex <-- because the flac format is shite? [21:49] <Daemon404> bitpack all the things [21:50] <Daemon404> byte alignment is for chumps [21:50] <Daemon404> etc [22:00] <BtbN> This sample_aspect_ratio thing confuses me. [22:00] <BtbN> The -aspect cli option does not seem to mean the sample_aspect_ratio option. [22:01] <nevcairiel> there is two of those, one in AVStream for the container and one in AVCodecContext for the codec [22:03] <BtbN> And how would i get the display aspect ratio, so 16:9, 4:3 or whatever it can be, from within an encoder? [22:11] <cone-168> ffmpeg.git 03Michael Niedermayer 07master:a46a23d30fea: avfilter/vf_framepack: Check and update frame_rate [22:13] <aetasx> you guys familiar with the runtime generated prediction tables code? [22:27] <michaelni> kierank, I hadnt posted it to devel because i thought noone wants to maintain it, does understand and is interrested in that code except me. If thats wrong, please do review it now, iam happy to make any corrections needed if reviewes find issues [22:34] <aetasx> do you guys know why the runtime predicted table functions like ff_hevc_h_loop_filter_chroma_10_avx still get generated and included in the static libs even if you disable avx? wasn't sure if there was a technical reason for it [22:35] <ubitux> michaelni: precision* ;) [22:36] <ubitux> (akira4 sent me a mail) [22:46] <michaelni> also if someone wants to review patches to the fps code theres still one on ffmpeg-devel "[FFmpeg-devel] [PATCH 1/2] ffmpeg: last frame duplication heuristic" [22:49] <halfie> hi! I have a raw bitstream which i suspect is encoded AAC audio samples. is there a way to prove that samples are really AAC encoded? [22:50] <cone-168> ffmpeg.git 03Michael Niedermayer 07master:f96148913b31: Fix precission typos [22:56] <ubitux> thx :) [00:00] --- Sat Jan 17 2015

1 0

ffmpeg.log.20150116
by burek 17 Jan '15

17 Jan '15

[00:19] <knob> Hello everyone. Got a question... looking for some direction. [00:19] <knob> I pull a 30 second video from a rPi. I save it locally as a .h264. I then use MP4Box to convert it into .mp4 [00:19] <knob> All perfect in that part. [00:19] <knob> I want to take that same .h264 file, and convert it to webm. [00:20] <knob> I tried doing it with ffmpeg, yet I seem to be missing a codec (??). [00:20] <knob> I ran: ffmpeg -codecs | fgrep vpx and got back this: http://pastebin.com/Sx4CnArm [00:20] <knob> I am lost as to... what do I need? Do I need to re-compile ffmpeg to have this new codec for webm? [00:21] <iive> knob: http://ffmpeg.org/general.html#libvpx ? [00:21] <knob> iive, on my way [00:22] <knob> Thank you iive ... I hope it's simple. I am a n00b to ffmpeg... and, kinda lost. [00:22] <knob> Question: When that page says Then pass --enable-libvpx to configure to enable it. How do I "pass it to configure"? [00:22] <knob> Where is the ./configure? [00:22] <iive> in the ffmpeg source [00:23] <knob> Ahh ok ok. Ok [00:23] <iive> it's the first step to compile mplayer... after extracting the source. [00:23] <iive> i mean, ffmpeg [00:24] <knob> Yes... now I see what I ran when I compiled ffmpeg... and ... I see where it is ./configure -----options---- [00:24] <knob> Reading up on your link... [00:26] <knob> iive, this is what I run to "install" ffmpeg. I am not sure how to "add" the WebM repo to this, so ffmpeg gets compiled with the VP8/VP9 codecs [00:26] <knob> http://pastebin.com/Y934Fqr6 [00:28] <iive> why disable-mmx? [00:28] <iive> btw, what distribution are you running? [00:29] <knob> iive, Raspbian (from Debian). It's on a Raspberry Pi [00:29] <knob> btw, thank you for the help. I have like 18 tabs open... and still not sure which direction I have to go [00:30] <iive> knob: first, you have to install libvpx, if you are lucky, that debian flavor may already have it. [00:30] <knob> Ok. [00:31] <iive> just don't forget that you need the -dev packages to compile stuff. [00:31] <knob> For WebM? http://www.webmproject.org/code/ ? [00:33] <iive> knob: https://sites.google.com/a/webmproject.org/wiki/ffmpeg/building-with-libvpx [00:34] <iive> i wonder if libvpx is gpl... [00:35] <iive> have in mind, that page tells you how to install every library manually. it might be better to use pre-built system version if available. [00:38] <knob> iive, checking that link now... thank you [00:39] <iive> i'll be off soon [00:40] <knob> Yes... thank you. I am going to try and install libvpx, and then re-compile ffmpeg [00:41] <dirty_d> what am i missing here? ffmpeg -i /mnt/misc/VIDEO/MOVI0007.avi -c:v libx264 -crf 18 -c:a aac -profile:v baseline output.mp4 [00:41] <dirty_d> i get Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height [00:41] <dirty_d> im using the examples from https://trac.ffmpeg.org/wiki/Encode/H.264 [00:48] <iive> dirty_d: is #0:0 the video? it looks like it does have rate (factor). try without "-profile:v baseline" [00:49] <dirty_d> no dice [00:49] <dirty_d> let me get you the full output [00:49] <iive> try without "-c:a aac" and instead use -an [00:49] <iive> that disables audio [00:50] <dirty_d> http://codepad.org/FF1Q37pJ [00:50] <dirty_d> that did the trick [00:52] <dirty_d> hmm, what am i missing for audio though? [00:52] <iive> aac might be the internal aac encoder, and there is only one encoder that is worse that it. [00:52] <dirty_d> no dice with -c:a aac -ac 1 -ab 128k -ar 44100 [00:52] <iive> than... [00:53] <dirty_d> what do you think i should use? [00:53] <dirty_d> i dont realy care about audio quality as much as it working on every device possible [00:53] <iive> faac or fdk-aac [00:54] <dirty_d> -c:a faac? [00:54] <dirty_d> if so its not compiled in [00:57] <dirty_d> oh, i guess its supposed to be libfdk_aac, but i dont have that either [00:57] <dirty_d> ill just recompile [00:58] <iive> all externals start with lib* [00:58] <iive> `ffmpeg -codecs` [00:59] <iive> dirty_d: also, you might want to add -pix_fmt 420p , as the log hints, because the mjpeg is 422p. 422 is mostly used by production, it handles interlace better. [01:00] <iive> gtg [01:00] <iive> n8 ppl have fun. [01:00] <dirty_d> later, thanks for the help [01:11] <dirty_d> hmm, if anyone else has any ideas, -c:a libfaac did not work [02:24] <fffan> is there a simple way to get pts after avcodec_decode_audio4 [02:53] <voip__> Hello guys, I transcoding and sending stream to 2 different servers, it works well. [02:53] <voip__> But I have 1 stream which I cant start [02:53] <voip__> There is difference between inputs. The input stream which cant start has 4 audio streams. [02:53] <voip__> How to fix this issue? [02:53] <voip__> http://pastebin.com/XisBgYwv [06:21] <jjohn> Hello, guys. [06:22] <jjohn> I am having a problem with ffmpeg capturing audio, but it's too long to explain here, so I used pastebin: http://pastebin.com/BEnUUn9E [06:28] <klaxa> jjohn: see https://trac.ffmpeg.org/wiki/Capture/ALSA [06:28] <klaxa> under recording an application [06:34] <jjohn> klaxa: like type plug slave.pcm "hw:Loopback,3,0"? [06:34] <klaxa> i don't know the first thing about alsa [06:34] <klaxa> the page was linked by someone else when someone else wanted to record from alsa [06:38] <jjohn> The thing is: when I set the device to hw:0,0, I have no audio anymore. [06:39] <jjohn> And recording from 3,0 (when issuing ffmpeg) does again just produce silence. [06:42] <jjohn> And if I set the loopback device in the /etc/asound.conf to 0,0 and then record something (which I do not hear), then the file contains silence again (after I commented the line so that I'd get back output). [06:46] <fffan> why av_frame_get_best_effort_timestamp does not work for wma audio for wmv? [06:48] <jjohn> But the module is already built-in in my kernel, so maybe the default value for pcm_substreams is 0? I will add this to my boot line and reboot the kernel, and see if that did any difference. [06:50] <jjohn> Will take a little while until the kernel is compiled and I restarted it. I will be back in a few minutes. [07:05] <jjohn> Ahoy, guys. [07:06] <jjohn> Recompiled the kernel to apply the new boot command line, but it didn't work out. [07:06] <jjohn> Still no audio. [07:07] <jjohn> http://pastebin.com/BEnUUn9E [07:22] <jjohn> Of course I made sure that the kernel had the line: "snd_aloop.pcm_substreams=1" [07:24] <pzich> are there two things trying to read from the pipe? do you need to tee it in some way? [07:25] <jjohn> Did you read the pastebin? I guess I explained what I want. [07:54] <john___> http://pastebin.com/BEnUUn9E [08:24] <jjohn> http://pastebin.com/BEnUUn9E [08:29] <klaxa> jjohn: what happens if you omit -ac 2? [08:35] <jjohn> klaxa: in which command? /opt/local/ffmpeg/bin/ffmpeg -f alsa -i hw:3,0 -ac 2 -c pcm_s16le output.wav [08:35] <klaxa> yes [08:36] <jjohn> $ /opt/local/ffmpeg/bin/ffmpeg -f alsa -i hw:3,0 -c pcm_s16le output.wav [08:36] <jjohn> >cannot set channel count to 2 (Invalid argument) [08:36] <jjohn> >hw:3,0: Input/output error [08:37] <jjohn> Stays the same. [08:38] <jjohn> Right now there is VLC running with some Chrono Trigger tracks, which I'd like to capture from the device. But somehow it does not work. [08:39] <jjohn> VLC outputs it to hw:3,0 (my USB headphones), and I can hear them. But I cannot capture them. [08:50] <klaxa> did you do the loopback stuff? [08:50] <klaxa> from the wiki entry about capturing alsa [08:54] <jjohn> You mean cm.!default { type plug slave.pcm "hw:Loopback,X,0" }? Yeah, I did. [08:54] <jjohn> When I set it, I got no sound to my headphones anymore. [08:54] <jjohn> Doesn't matter if X is 0 or 3. [08:56] <jjohn> I wouldn't know which device I'd have to put in exactly, because the paragraph isn't explained at all. [08:56] <jjohn> So I tried out several values, but without success. [09:00] <jjohn> But there is still a difference between X being 0 and 3: if it's 3, VLC gives me an error message that the default device is not ready. I guess this is because the option for asound.conf sets the target loopback device. If X is 0, I still have no sound, but VLC does not give me an error message. [09:09] <jjohn> Sorry, hardware hickup, had to restart. [09:23] <jjohn> klaxa you are out of ideas? [09:24] <klaxa> have you done the "Record audio from an application while also routing the audio to an output device" part? [09:24] <klaxa> also notice how the command line says: [09:24] <klaxa> >ffmpeg -f alsa -ac 2 -ar 44100 -i hw:Loopback,1,0 out.wav [09:24] <klaxa> notice the hw:Loopback,1,0 [10:03] <jjohn> klaxa: OK, I somehow got it by using the informations in the "Record audio from an application while also routing the audio to an output device" paragraph. The command now is this: /opt/local/ffmpeg/bin/ffmpeg -f alsa -ac 2 -ar 48000 -i loopout -f alsa -ac 1 -ar 44100 -i mic -c pcm_s16le output.wav [10:03] <klaxa> so it works properly now? [10:03] <jjohn> But it seems that my mic is ignored - I do not hear my voice in output.wav, only the music. [10:04] <jjohn> Not yet, but at least I hear music from VLC now in output.wav [10:04] <klaxa> oh right that is probably because you have two input streams and only one output file. you might want to look at amix or even better amerge since you have streams with different channel layouts [10:05] <klaxa> http://www.ffmpeg.org/ffmpeg-filters.html#amerge [10:12] <jjohn> OK, now I hear both my voice and the music: /opt/local/ffmpeg/bin/ffmpeg -f alsa -ac 2 -ar 48000 -i loopout -f alsa -ac 1 -ar 44100 -i mic -filter_complex "[0:0][1:0] amerge=inputs=2" -c pcm_s16le output.wav [10:12] <jjohn> But the background noise (first problem) is still there. [10:13] <klaxa> background noise from the mic? [10:13] <jjohn> Seemingly. A high-pitched, constant noise. [10:14] <klaxa> you can either figure out the frequency and try to do some equalizer filtering or try to use sox with some piping stuff to apply a noise removal filter [10:14] <klaxa> that will, however, add latency [10:16] <jjohn> Hm. I got a program called spek which shows the spectrum of an audio file. [10:17] <jjohn> And the noise seems to be in different heights ... from 0 kHz to 24 kHz. [10:17] <klaxa> noise removal filter it is? [10:19] <jjohn> Single lines only, but constant ... not sure if that would help. But see for yourself: https://imgur.com/EaKj40F [10:19] <jjohn> This is noise only. I deactivated the mic, so this comes from a basically empty source. [10:20] <klaxa> wot [10:20] <klaxa> that looks broken [10:20] <klaxa> well uh... i guess one could fix that one way or another [10:21] <jjohn> Any ideas? :) [10:21] <klaxa> write down the frequencies and block them with the equalizer filter? [10:22] <klaxa> or rather, reduce them [10:23] <baran> does anybody know how to properly calculate bitrate of a stream/video? [10:23] <klaxa> filesize / fileduration ? [10:23] <baran> formatcontext->bit_rate = 0 [10:23] <baran> well, the problem is i get only byte stream from java :) [10:24] <klaxa> can you elaborate the problem in a more precise way? [10:24] <klaxa> pastebin code or something [10:25] <baran> i cant paste code [10:25] <baran> bt i will try to elaborate [10:30] <jjohn> klaxa: OK, I tried this out: /opt/local/ffmpeg/bin/ffmpeg -f alsa -ac 2 -ar 48000 -i loopout -f alsa -ac 1 -ar 44100 -i mic -filter_complex "[0:0][1:0] amerge=inputs=2" -af "equalizer=f=22000:width_type=h:width=200:g=-10" output.wav [10:30] <jjohn> And it gave me this error: "-vf/-af/-filter and -filter_complex cannot be used together for the same stream." [10:31] <klaxa> you can use equalizer in filter_complex [10:32] <klaxa> like so: -filter_complex "[0:0]equalizer=f=22000:width_type=h:width=200:g=-10[loopback];[loopback][1:0] amerge=inputs=2" [10:34] <baran> i read the file through jni (java native interface), so i dont know it's size nor duration. I read AVpackets, so i know their size (in bytes) and duration (in ffmpeg units) + the average framerate. so to my knowledge bit rate should be calculated something like this: bitrate = (packet_size * 8) * avg_framerate, since 1 packet should contain a single frame [10:35] <klaxa> i doubt packets will be of similar size [10:35] <klaxa> you might want to average over a number of packets [10:35] <baran> but i didnt came close to actual results calculating file_size/ file_duration [10:35] <jjohn> Now I get another error: The following filters could not choose their formats: Parsed_amerge_1 - Consider inserting the (a)format filter near their input or output. [10:36] <jjohn> /opt/local/ffmpeg/bin/ffmpeg -f alsa -ac 2 -ar 48000 -i loopout -f alsa -ac 1 -ar 44100 -i mic -filter_complex "[0:0]equalizer=f=22000:width_type=h:width=200:g=-10[loopback];[loopback][1:0] amerge=inputs=2" -c pcm_s16le output.wav [10:36] <klaxa> huh [10:36] <klaxa> that's odd [10:36] <klaxa> is there any additional info regarding that error? [10:37] <jjohn> > [Parsed_amerge_1 @ 0x1a3cbe0] No channel layout for input 1 - Last message repeated 1 times [10:37] <jjohn> And the previous error message was printed by [AVFilterGraph @ 0x1a3a300]. [10:38] <pzich> are you passing in a separate audio file/pipe/stream? or did you want [0:1] for separate stream, same input? [10:38] <pzich> oh, derp, -i mic, missed that the command was there [10:39] <klaxa> maybe add an aformat filter and tell it to to use 2 channels? [10:39] <pzich> what does ffmpeg interpret the input as, can you pastebin the console output? [10:39] <klaxa> -filter_complex "[0:0]equalizer=f=22000:width_type=h:width=200:g=-10[loopback1];[loopback1]aformat=channel_layouts=stereo[loopback2];[loopback2][1:0] amerge=inputs=2" [10:40] <jjohn> The loopback output as stereo (0:0), the mic as mono (1:0). [10:40] <pzich> is that considered input 1 though? I think it's [1:0] (mic) [10:41] <klaxa> is it the loopback causing troubles or the microphone? i would think the loopback since we have been messing with that with the equalizer and it worked earlier [10:41] <jjohn> pzich input 0 is the loopback device, input 1 is the mic. The mic is causing the problems. [10:42] <pzich> might be worth converting the mic to stereo? [10:42] <jjohn> klaxa the filter that you provided works ... but I guess I'll have to tamper with it a bit. [10:42] <pzich> I see [1:0][1:0]amerge=inputs=2[mic] being one of the ways to do that [10:43] <jjohn> pzich even if I don't merge the mic into the loopback output the noise is still there. If I just write the mono channel into the file there is still the noise. [10:43] <pzich> ah, don't know what to do about the noise, sorry [10:44] <jjohn> Let me stress this: only record mic => still noise. [10:44] <jjohn> Me neither. :( [10:44] <pzich> is it just ffmpeg? [10:44] <jjohn> I am as far as ditching the mic and getting a new one. I have no idea about it. [10:45] <jjohn> pzich no. If I record with audacity the noise is still there. [10:45] <pzich> are you looking for a noise reduction filter? [10:46] <klaxa> pzich: this is the spectrum graph: https://imgur.com/EaKj40F [10:46] <jjohn> I dunno if audacity relies on ffmpeg ... I am currently tampering with noise reduction, yes. [10:46] <pzich> ah, hmm, that's interesting [10:46] <jjohn> But ... the specturm is ... well ... klaxa showed it to you. [10:46] <klaxa> if you can do post production i would advice to record to different audio streams and clean them separately [10:47] <klaxa> if you can't, maybe sox is the best solution for noise removal [10:47] <jjohn> Can't do. The output is going to be streamed in the end, and I want to avoid latency at all cose. [10:47] <jjohn> These are just tests. [10:48] <jjohn> costs* [10:48] <klaxa> voice latency is actually not as much of a deal as in-game audio latency is [10:48] <klaxa> unless you are trying to be in sync with the in-game audio (if it's a game even) [10:48] <jjohn> Who said anything about in-game? So far I just wanna have VLC running in the background. [10:48] <klaxa> then it's pretty bad yeah [10:48] <klaxa> like if you try to play along to something [10:49] <jjohn> I mean, for now. [10:49] <klaxa> well most people who come with these kinds of requests want to stream video games on twitch :P [10:49] <klaxa> or some other streaming service [10:49] <jjohn> If that would be the case I'd just use OBS or something like that. [10:50] <jjohn> Or directly use Windows. [10:51] <klaxa> well what is your exact use case then? maybe we are doing this too complicated [10:51] <jjohn> Which makes me wonder ... maybe there is no problem with the recording on Windows - maybe it's just the driver being sucky? [10:51] <pzich> seems like it could be the mic, cable, card or driver [10:52] <jjohn> OK. I will do a test on Windows to check if it's the kernel driver. If the record is OK, then the kernel hacker will have something to do ... :) [10:52] <jjohn> If not ... well, guess I will need to invest into new hardware then. [10:53] <jjohn> Or do you have any more ideas? [10:53] <klaxa> nope [10:55] <jjohn> Thought so. Well, I am going to go now and will do the test later this day. I will come back to chat regardless whether the driver is the problem or not ... [10:55] <jjohn> ... hm, waitasec. Are there developers present as well? [10:55] <jjohn> Because I have another issues, which has nothing to do with audio recording, but X11 grabbing. [10:55] <klaxa> ffmpeg-developers? some read this channel i guess? [10:56] <klaxa> well what's the issue? [10:56] <jjohn> I noticed that if I record my desktop at 60 FPS and save the record to the HDD immediately, I quickly drop frames. [10:57] <klaxa> probably slow I/O [10:57] <jjohn> But if the video gets stored on a ramdisk, the FPSs are constant. [10:57] <klaxa> that's what i resorted to [10:57] <pzich> mmm, ramdisks [10:58] <jjohn> That's what I thought as well. And I took a look into the sources. Seems like X11grab uses a AV interface to save single frames ... or at least small units of data. [10:59] <jjohn> I believe that ffmpeg does not cache some data in RAM, but writes it to the HDD immediately. I know that this is appropiate, because if ffmpeg gets interrupted, the data is still there. [10:59] <klaxa> maybe tweaking disk caches might help? [11:00] <jjohn> But maybe ffmpeg could offer an option to create a small/bigger buffer in RAM, maybe for a second, and flush the data not immediately, but just every second or two ... or every 30 seconds. As the user wants it. [11:01] <klaxa> open a feature-request/bug-report [11:01] <jjohn> Hm, OK. Will do. [11:03] <jjohn> Should I use ffmpeg-devel for this? [11:03] <jjohn> The mailing list, I mean. [11:05] <jjohn> or ffmpeg-trac? [11:11] <ubitux> try -flush_packets 0 maybe [11:16] <klaxa> i didn't even know about that [11:23] <jjohn> ubitux that looks rather good. I am recording for some time now - usually I would have dropped to 50 frames by now, but I am still at my full 60 FPS. [11:26] <jjohn> But the commit message is the joke of the century: 0 disables it and may slightly increase performance in some cases. [11:27] <jjohn> I am impressed by this imcrease of performance. Thank you very very much. [11:27] <jjohn> increase* [11:29] <ubitux> what format do you output? [11:30] <jjohn> libx264 in mp4. [11:31] <jjohn> I am already at 40,000 frames, and I didn't drop a single frame. [11:39] <ubitux> no difference here [12:05] <jeanre> hi all [12:12] <jeanre> im trying to figure out the best way to get a frame from a video [12:12] <jeanre> say 00:00:14 frame 2 [12:16] <jeanre> ffmpeg -i timecode\ 3h.mp4 -r 1 -ss 00:00:14 -frames:v 1 image-%d.jpeg [12:17] <jeanre> this will extract 1 frame so 00:00:14 frame 1 [12:17] <jeanre> but I need frame 5 [12:23] <jeanre> the only way I can think of is to create the amount of images for each frame [12:23] <jeanre> take the last one [12:23] <jeanre> trash the other [12:26] <ribasushi> jeanre: that won't be the exact frame though - -ss on input is not precise [12:26] <ribasushi> (documentation explains why) [12:57] <OxFEEDBACC> high, i used "ffmpeg -i stem-%3d.png ...." to create a video from rendered images... now i want it to put the same sequence into the video 3 times ... "-i stem-%3d.png -i stem-%3d.png -i stem-%3d.png" doesn't do that... what am i missing? [12:58] <jeanre> ribasushi any other ideas? [12:59] <jeanre> how I can get it more accurate? [13:01] <knob> Hello everyone. I am converting a 55secon .h264 file to webM. The line I am using is this one: ffmpeg -i /var/tmp/stream.h264 -strict experimental $TMP/stream.webm [13:01] <jeanre> ribasushi so extracting all the frames will not work? [13:01] <knob> Yet it is taking a LOT of time to "create" the webm file. Is there a way I can... lower the quality? Or bit rate? What do you suggest? [13:01] <OxFEEDBACC> it doesn't even work when i copy all stem*png to sten*png and steo.png, then use "-i stem-%3d.png -i sten-%3d.png -i steo-%3d.png" [13:02] <knob> Or a new command line arguments for ffmpeg? [13:02] <knob> ^^ "a new different set" of command line arguments... sorry [13:03] <OxFEEDBACC> -qscale? [13:04] <knob> OxFEEDBACC, that's for me? [13:04] <OxFEEDBACC> takes a parameter 1 (best quality) to 31 (least quality) [13:04] <OxFEEDBACC> yup, knob [13:04] <knob> OxFEEDBACC, ahh... ok. Will try that now. Thansk... I am totally new to ffmpeg, and a little bit overwhelmed. On my way to google those parameters. [13:04] <knob> Thank you [13:04] <jeanre> I also see hh:mm:ss[.xxx] [13:04] <knob> Will report back in a while on how it goes1 [13:04] <OxFEEDBACC> i use it to somehow influence the size of the output file [13:04] <jeanre> whats the use of .xxx? [13:05] <OxFEEDBACC> milliseconds? [13:06] <OxFEEDBACC> so, answered two question within my first 5 minutes here... could now someone look at making me happy, please? :-D [13:06] <jeanre> im trying to extract a specific frame from the video [13:06] <jeanre> like 00:00:14 frame 2 [13:07] <jeanre> the docs don't say anything about this [13:07] <OxFEEDBACC> i guess it's not a good idea to mix hh:mm:ss times with frame numbers... [13:07] <OxFEEDBACC> if you know the frame rate of your video, you should compute the absolute frame number and use that... [13:07] <jeanre> well each second has 25 frames [13:07] <jeanre> so frame 100 [13:08] <OxFEEDBACC> so, just convert your hh:mm:ss to seconds, multiply by 25, and there you go... [13:08] <jeanre> OxFEEDBACChow would I do that [13:08] <OxFEEDBACC> seconds=hh*3600+mm*60+ss [13:08] <jeanre> no not that [13:08] <jeanre> lol [13:08] <OxFEEDBACC> wtf... [13:08] <jeanre> extract frame 100 [13:08] <jeanre> or frame 125 [13:08] <OxFEEDBACC> how would i know? i'm new to ffmpeg... and never extracted anything... [13:08] <jeanre> just that frame the calculation is the easy part [13:09] <OxFEEDBACC> you could try mplayer for extracting stuff, however... i did it with sound clips... [13:21] <knob> ox [13:21] <knob> OxFEEDBACC, I used -qscale and it returned Please use -q:a or -q:v, -qscale is ambiguous [13:21] <knob> is the the same thing [13:24] <OxFEEDBACC> how did you use it? [13:25] <knob> I did::: ffmpeg -i /var/tmp/stream.h264 -qscale 30 -strict experimental $TMP/stream.webm [13:25] <knob> OxFEEDBACC, I am reading up... yet, ... not sure if the "position" of the argument is correct [13:26] <hans_s> jeanre: if you know the exact frame number to extract, you could try to adjust input frame rate to 1 fps and therefore specify a more accurate -ss parameter [13:26] <OxFEEDBACC> well, i specify it between the codec and out file name... [13:26] <jeanre> hans_s I can calculate the frame that I need to extract [13:27] <jeanre> so on 29 frames 00:00:1 frame 2 will be 31 frames [13:27] <jeanre> it should work [13:27] <knob> OxFEEDBACC, going to try now like this:: ffmpeg -i /var/tmp/stream.h264 -strict experimental -qscale 30 $TMP/stream.webm [13:27] <hans_s> jeanre: like: ffmpeg -r 1 -i whatever -ss 104 ... to get the 105th frame [13:27] <knob> will report back [13:28] <jeanre> ye [13:28] <OxFEEDBACC> gl, knob [13:28] <jeanre> i'm doing ffmpeg -i timecode\ 3h.mp4 -vf "select=gte(n\,32)" -vframes 1 out_img.png [13:32] <hans_s> jeanre: specifying "-r 1" before "-i" will rewrite the timecodes of the frames, so frame 1 will be at 00:00:00, frame 2 at 00:00:01 and so on [13:35] <OxFEEDBACC> please, how can i repeatedly put a sequence of images into a video? [13:36] <jeanre> hans_s [13:36] <jeanre> ffmpeg -r 1 -i timecode\ 3h.mp4 -ss 3 -vframes 1 out_img.png [13:36] <jeanre> always reutrns 00:00:00,2 [13:36] <jeanre> wtf [13:38] <hans_s> so? [13:40] <OxFEEDBACC> a funny thing is that when i extend it to even "-i stem-%3d.png -i sten-%3d.png -i steo-%3d.png -i step-%3d.png", when step* does not exist, it complains... [13:41] <hans_s> OxFEEDBACC: I think specifying multiple -i means to add multiple parallel streams (of which probably only one is selected for output) [13:42] <hans_s> maybe the helps: https://trac.ffmpeg.org/wiki/Concatenate [13:42] <OxFEEDBACC> oh, okay, thanks a lot! [13:42] <Carlo67> Hello, I am a new user to ffmpeg and I have some videos that I want to convert to .webm. I looked at the wiki page for VP8 http://trac.ffmpeg.org/wiki/Encode/VP8, so far so good. The article further links to a page about encoding parameters http://www.webmproject.org/docs/encoder-parameters/ .. can I pass these VP8 encoding parameters to ffmpeg? [13:46] <OxFEEDBACC> ya, "-f concat" can't be used with -i stem-%03d form of input spec... :-/ [13:46] <jeanre> hans_s -ss 4 also returns the same [13:47] <hans_s> what does the 2 in 00:00:00,2 mean? 2nd frame? 200ms? [13:47] <jeanre> second frame [13:47] <OxFEEDBACC> where in the man pages can i read more about the -i option that in man ffmpeg? [13:55] <edoloughlin> I've got a file of RGBA frames @25fps that I'm encoding to MP4. Is there anything I could do on the input side to improve encoding performance? E.g., any extra metadata I could supply or use JPEG instead of RGBA? [14:01] <OxFEEDBACC> man, it can't be THAT complicated to put some fucking simple pictures into a video? [15:29] <herbySk> Hi! Is there any option to make ffmpeg end with nonzero exit code if input file is partial? [15:30] <herbySk> (I just realized if ends with exit code 0 for partial file :-( which I would really not expect) [15:30] <herbySk> s/if/it/ [15:32] <c_14> Define "partial file" [15:33] <herbySk> c_14 an mp4 file which is not full. After all, ffmpeg output "partial file" with red letters. Just, exits with code 0. [15:33] <herbySk> I use it to get frames of the video. Though, I'd presume it does not end with exit code 0 if the video was not complete. [15:52] <c_14> hmm, utils.c appears to be ignoring AVERROR_INVALIDDATA [15:52] <c_14> Not entirely sure why. [15:53] <c_14> Make a bug report on trac mentioning it [15:54] <herbySk> c_14 Hm. Maybe it's a feature :-( to allow recovering broken videos, or so. Where's the trac? [15:55] <c_14> https://trac.ffmpeg.org/ [16:26] <netwiz> do you guys know a good way to crank up contract on a black and white video [18:29] <ike> can anyone help me, im having an issue trying to get ffmpeg to work on my amazon server [18:31] <ike> anyone here [18:50] Last message repeated 1 time(s). [18:52] <c_14> Maybe if you asked a question, someone could help you. [18:54] <ike> fair enough [18:59] <ike> ive installed ffmpeg on my amazon EC2 server to convert files to mp3. I have android and iphone app that is sending native audio recording to the server to be converted into an mp3. it is converting the audio file from android as .m4a or .amr to mp3, and for iphone .wav to .mp3. Right now we run -i /var/www/filepath..../audiofile.m4a /var/www/filepathe...../out_audiofile.mp3... the result is [18:59] <ike> a file with 0 size and the error i get is "unsupported codec for output stream #0.0 . On my computer i try to convert the files using ffmpeg and it works fine. So the server is having the issue. Ive looked through stack overflow and have tried various solutions with no luck. Please let me know if anyone can help me. [18:59] <c_14> pastebin the output of `ffmpeg -version' please [18:59] <c_14> On the server [18:59] <c_14> It presumably was built without mp3 encoding support [19:00] <ike> SVN-r0.5.10-4:0.5.10-1 [19:00] <c_14> yeah [19:00] <c_14> I don't even know what version that's supposed to be. [19:00] <c_14> Either use a static build or build from source. [19:01] <c_14> Or find a way to get a newer version of ffmpeg. [19:02] <ike> do i get the statc build or the source from GIT? or somewhere specific [19:05] <c_14> http://johnvansickle.com/ffmpeg/ [19:05] <c_14> for the static builds [19:05] <c_14> source is at https://git.videolan.org/?p=ffmpeg.git [19:06] <ike> thank you i will work with that for now [19:35] <jjohn> klaxa: are you still around? [19:36] <klaxa> yes [19:36] <jjohn> OK, just wanted to give you an update. [19:36] <jjohn> So, I tried out to record on Windows, and it had the same static noise in the background. Thus I consider the hardware broken. [19:37] <klaxa> ok [19:37] <klaxa> time for new hardware then probably [19:37] <jjohn> But I managed to record from another internal device which seems to be way more reliable, and together with the changes that I made this morning I am now able to record other applications. [19:37] <klaxa> ah right [19:37] <jjohn> So, I wanted to thank you for your help and your patience with me. [19:37] <klaxa> you had another soundcard [19:38] <jjohn> I have three, to be precise. [19:38] <klaxa> you're welcome, you explained your problem in a very precise way [19:38] <jjohn> Just wanted to make sure that you see the exact things that I see. That's just how I work. :) [19:39] <jjohn> Again, thanks and a very nice weekend. [19:40] <am11> hey guys can anyone here help me with a gcc-c++/libstdc++ problem? [19:44] <shudouken> when using 2 pass encode, is there any command that makes ffmpeg automatically remove the ffmpeg2pass-0.log file when done with the second pass? [19:45] <c_14> shudouken: not that I know of [19:45] <torrente> hi [19:45] <c_14> am11: what is the problem? [19:46] <klaxa> what about "rm ffmpeg2pass-0.log" ? [19:46] <c_14> shudouken: use && rm ffmpeg2pass-0.log [19:46] <torrente> who use mediacoder? [19:47] <shudouken> yes I will but that won't work for every OS, which is why I asked [19:47] <torrente> because it has many features [19:51] <torrente> who convert video files? [20:06] <voip__> Hello guys, I transcoding and sending stream to 2 different servers, it works well. [20:06] <voip__> But I have 1 stream which I cant start [20:06] <voip__> There is difference between inputs. The input stream which cant start has 4 audio streams. [20:06] <voip__> How to fix this issue? [20:06] <voip__> http://pastebin.com/XisBgYwv [21:22] <voip__> any help [21:22] <voip__> ? [21:24] <c_14> [flv @ 0x4bc6940] at most one audio stream is supported in flv [21:24] <sfan5> voip__: [flv @ 0x4bc6940] at most one audio stream is supported in flv [21:24] <c_14> that timing [21:24] <sfan5> D: [21:24] <voip__> what it means how to fix problem ? [21:24] <yarilo_> Hi guys [21:25] <c_14> voip__: drop audio channels [21:25] <yarilo_> I have a question which I guess is really popular [21:26] <yarilo_> Is there a way that I can create a video that would play on all the major browsers? [21:26] <voip__> c_14, i need audio chanels :) what te correct ddmpeg commands [21:26] <yarilo_> now I have to create .mp4 and .webm videos [21:26] <c_14> voip__: if you need more than one audio channel per stream, don't use flv [21:27] <c_14> yarilo_: I am relatively certain you're sol [21:28] <yarilo_> c_14 I don't understand you :) [21:28] <voip__> c_14, i need take live stream, keep video (no transcoding), resend to 2 servers with 1 audio. flv [21:29] <c_14> yarilo_: in short, I know of no codec/format that is supported in every browser [21:29] <c_14> voip__: then map the audio stream you want [21:29] <yarilo_> <c_14> ok [21:29] <yarilo_> uhmm [21:29] <yarilo_> than I guess I'll have to go with creating both videos [21:30] <voip__> where is my mistake ? tee -map 0:v -map 0:a -flags +global_header [21:30] <c_14> -map 0:a:0 if you want the first audio stream only [21:31] <yarilo_> before going that way, do you have any experience if it will be faster to create both videos at the same time or create the one first and then convert to the other format. I'm doing the following(in order): create several videos from images, merge those several videos with oteh [21:31] <yarilo_> before going that way, do you have any experience if it will be faster to create both videos at the same time or create the one first and then convert to the other format. I'm doing the following(in order): [21:31] <yarilo_> 1. reate several videos from images [21:32] <voip__> c_14, thanks ! :) [21:32] <yarilo_> 2. Merge those videos with other videos [21:32] <yarilo_> at that point I get the final video [21:32] <yarilo_> 3. Merge the final video with audio and get the complete final video [21:33] <yarilo_> so [21:33] <c_14> I'd split after 2. If I'm understanding you correctly. [21:33] <yarilo_> should I add step 4. Convert the final video(which is .mp4) to webm [21:33] <yarilo_> you'd split? [21:34] <yarilo_> I need to merge all the videos together to get one final video which I then merge with audio file [21:34] <c_14> as in ffmpeg -i final_video -i audio -map 0 -map 1 -c:v copy -c:a copy out.mp4 -map 0 -map 1 -c:v libvpx -c:a libvorbis out.webm [21:34] <yarilo_> aha [21:35] <yarilo_> so this will create both at the same time [21:35] <c_14> yep [21:35] <yarilo_> ok [21:35] <yarilo_> I'll test :) [21:35] <yarilo_> thank you! [21:36] <yarilo_> Btw I know this is not exactly ffmpeg question [21:36] <yarilo_> but if someone has experience executing ffmpeg through PHP could maybe help me [21:37] <yarilo_> I have a problem when I use -filter-complex [21:37] <yarilo_> -filter_complex "[1:0] setsar=sar=1,format=rgba [1sared]; [0:0]format=rgba [0rgbd]; [0rgbd][1sared]blend=all_mode='overlay':all_opacity=0.8,format=yuva422p10le" [21:38] <c_14> what's the problem? [21:38] <yarilo_> it adds backslashes before \[1:0\] etc.. [21:38] <yarilo_> so when it's executed it looks sth like [21:38] <yarilo_> ok hold on let me paste the exact output [21:39] <c_14> use -filter_complex_script and the path to a file containing the actual complex [21:39] <c_14> Then, no more escaping things. [21:40] <yarilo_> wow [21:40] <yarilo_> let me test that [23:06] <halfie> hi, how do I generate ADTS headers? I already have encoded AAC data. [23:07] <vladimir> hi to all [23:08] <vladimir> can some one help me [23:08] <vladimir> I what 2 stream from IP camera (axis mjpeg stream) to make like one [23:08] <vladimir> 2 mjpeg > 1mjpeg [23:09] <vladimir> 1280x720 (x2) ====> 2560x720 [23:10] <c_14> pad and overlay [23:11] <vladimir> url from camera is http://user:pass@ipofcamera/axis-cgi/mjpg/video.cgi?camera=1 [23:13] <vladimir> c_14 how is going full comand ? [23:16] <c_14> ffmpeg -i url -i url2 -filter_complex '[0:v]pad=2560x720;[0:v][1:v]overlay=1280:0[v]' -map '[v]' outfile [23:16] <c_14> or something [23:17] <c_14> https://ffmpeg.org/ffmpeg-filters.html read the manpage [23:18] <yarilo_> I'm back again <c_14> [23:18] <yarilo_> so [23:19] <yarilo_> as filter_complex parameters I send [1:0] setsar=sar=1,format=rgba [1sared]; [0:0]format=rgba [0rgbd]; [0rgbd][1sared]blend=all_mode='overlay':all_opacity=0.8,format=yuva422p10le [23:19] <yarilo_> if I put this in a file named filtercomplex [23:19] <yarilo_> I try to run the command with ffmpeg .. .. .. filter_complex_script filtercomplex ... [23:20] <yarilo_> right? [23:20] <c_14> probably [23:20] <c_14> if filtercomplex is in your cwd [23:20] <yarilo_> I give absolute path [23:27] <vladimir> c_14... bat how to make outgoung http mjpeg stream to be... not file.mjpg... [23:28] <c_14> Get a http server that can serve mjpeg and send the stream there? [23:33] <vladimir> hm nooo [23:34] <vladimir> like I said I have 2 AXIS ip camera... [23:34] <vladimir> and I need from to camera to make one mjpeg strem.... [23:34] <vladimir> after I make one 1stream..... I willl record in another solutions 1 stream... [23:35] <c_14> yees, I heard that. I also told you what you need to do. Take the two input streams, make one out of them like I told you, send the result wherever you want. [23:35] <vladimir> aha [23:35] <vladimir> I make and it's work [23:35] <vladimir> like this... [23:35] <vladimir> -filter_complex '[0:v]pad=2560x720;[0:v][1:v]overlay=1280:0[v]' -map '[v]' test.mjpg [23:36] <vladimir> bat I don;t know how to make "stream" http mjpeg stream [23:36] <c_14> You can try streaming to ffserver and producing an mjpeg stream that way, or find another server that accepts input in whatever form and makes an mjpeg stream from it. [23:38] <vladimir> what u thnik is best solution to==> ffserver ...and after to http stream [23:39] <c_14> ffmpeg can stream to ffserver and ffserver can serve the http stream [23:44] <yarilo_> uhmmm [23:45] <yarilo_> with the current filter_complex params I receive First input link top parameters (size 836x360, SAR 1:1) do not match the corresponding second input link bottom parameters (846x360, SAR 1:1) [23:45] <yarilo_> here they are again [23:46] <yarilo_> [1:0] setsar=sar=1,format=rgba [1sared]; [0:0]format=rgba [0rgbd]; [0rgbd][1sared]blend=all_mode='overlay':all_opacity=0.8,format=yuva422p10le [23:47] <vladimir> axis-cgi/mjpg/video.cgi?camera=1&fps=5 -filter_complex '[0:v]pad=2560x720;[0:v][1:v]overlay=1280:0[v]' -map '[v]' test.mjpg [9] 29062 [10] 29063 -bash: -filter_complex: command not found root@ns-test:~# -bash: -i: command not found [23:47] <vladimir> there is some error on command [23:48] <c_14> -i isn't a bash command... [23:48] <c_14> you want ffmpeg -i [23:49] <c_14> yarilo_: resize/pad the top video to the bottom width [23:49] <yarilo_> hmmm [23:50] <yarilo_> the first video is a dynamically created video [23:50] <yarilo_> that's the video created from images [23:51] <yarilo_> and that should remain as it is [23:52] <yarilo_> but the second one is an overlay video that actually is static [23:52] <yarilo_> meaning [23:52] <yarilo_> I can change that one [23:52] <yarilo_> oh FUCK [23:52] <yarilo_> sorry [23:52] <yarilo_> I just saw this [23:55] <yarilo_> the videos were supposed to have the same dimensions [23:56] <yarilo_> that's how it was initially [23:56] <yarilo_> ughhhhh [23:56] <yarilo_> anyways [23:56] <yarilo_> thank you <c_14> [23:56] <yarilo_> I owe you a beer [00:00] --- Sat Jan 17 2015

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