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February 2015

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ffmpeg-devel.log.20150217
by burek 18 Feb '15

18 Feb '15

[00:16] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:cb9d40451cb2: avcodec/ac3: add () to protect the macro arguments [00:16] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:e3f71423068b: avcodec/vc1dsp: add () to protect the arguments of the op* macros [00:16] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:61eb602d27af: avutil/log: add () to protect the argument of AV_LOG_C() [00:16] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:527f1d22a1a4: avformat/tedcaptionsdec: add () to protect the argument of ERR_CODE() [00:30] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:9cde41cdb511: avutil/version: Add () to protect the AV_VERSION_INT() arguments [00:30] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:cdee0c9d5fed: avutil/murmur3: Add () to protect the ROT() arguments [00:31] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:f8607cfb0a82: avutil/intmath: Add () to protect the ff_log2() argument [00:31] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:06309fc704dc: avutil/attributes: Add () to protect the AV_GCC_VERSION_AT_LEAST() arguments [00:31] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:52d1929000c3: avutil/aes: Add () to protect the ROT() arguments [00:58] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:0cb95f908298: swresample/resample_template: Add () to protect the arguments of the OUT() macro [00:58] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:f15c734be165: avfilter/vf_phase: Add () to protect DIFF()s arguments [00:58] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:2bae7b337030: avfilter/vf_fieldmatch: Add () to protect the arguments of the HAS_FF_AROUND() macro [04:08] <jamrial> porting OpenHEVC's idct x86 intrinsics to yasm (or generally optimize hevc with assembly, arm/aarch64 included), and adding AESNI support to our aes module could be two new GSoC tasks [04:08] <jamrial> kierank would surely approve of the latter :P [04:21] <kierank> :) [06:02] <Daemon404> aesni should be pretty straighrforward no? [07:26] <sekon> Hello i want a dummy ffmpeg opt of type OPT_STRING [07:27] <sekon> it will not be used anywhere in ffmpeg .. but it should not make ffmpeg crash [07:27] <sekon> *crash as in segfault in av_free [07:28] <sekon> also could i suggest adding some documetation regarding what HAS_ARG, OPT_EXIT etc actually mean in cmdutils_common_opts.h [07:28] <sekon> ? [07:29] <sekon> documentation/comments [08:07] <ubitux> string needs to be dupped [09:15] <j-b> hello [12:48] <cone-408> ffmpeg.git 03Christophe Gisquet 07master:d9293c776ed9: x86inc: Correctly warn on use of SSE2 instructions in SSE functions [13:19] <cone-408> ffmpeg.git 03Christophe Gisquet 07master:bc8e044d8c32: x86/doc/Makefile: DBG=1 to preprocess external asm [13:46] <wm4> wow claws-mail is such a piece of shit [14:10] <compn> wm4 : compared to what? whats better ? [14:10] <compn> sylpheed claws? :P [14:13] <arwa> How do I set min and max values allowed by using AVExpr? [14:20] <michaelni> good question, probably best by simply checking or cliping with av_clipd() the "double" value from evaluating the expression [15:17] <cone-408> ffmpeg.git 03Carl Eugen Hoyos 07master:a9dc94c39c7e: Force linking against user32 if dxva2 is enabled. [15:19] <arwa> Okay, thanks :) [15:20] <j-b> Why do you need user32? [15:21] <arwa> No, It was for michaelni. [15:21] <j-b> I'm asking Carl [15:24] <arwa> Ohhh!! [15:30] <compn> j-b : i'm guessing because vfwcap provided a header file , and when thats missing/disabled need to link against user32 [15:30] <compn> but thats just my guessssss [15:30] Action: compn afk [15:34] <j-b> vfwcap for dxva? [15:34] <j-b> seriously? [15:45] <j-b> -lole32 -lshlwapi -luuid should be enough [15:45] <j-b> and shlwapi, I'm not sure [15:47] <nevcairiel> one function needs user32 apparently [15:49] <j-b> sure, but which one? [16:08] <cone-408> ffmpeg.git 03Christophe Gisquet 07master:398f53191560: x86: hevc_mc: fewer xmm regs used in epel h/v [16:09] <nevcairiel> j-b: GetShellWindow, i guess [16:10] <j-b> ah, yes. But it's used in DxVA? [16:10] <nevcairiel> not the hwaccel, only in ffmpeg_dxva2.c to create the d3d device [16:11] <nevcairiel> (which needs a window ref) [16:15] <wm4> of course ffmpeg.c deps are included in the libs [16:15] <Daemon404> of course. [16:16] <Daemon404> because carl. [16:16] <nevcairiel> i think that part is only for ffmpeg itself [16:16] <nevcairiel> it does say ffmpeg_libs [16:17] <Daemon404> o ok [16:53] <gerion> I just found out, that x265 could not be controlled by -crf but with -x265-params crf etc.. Is there a technically/design reason for this or is it just not implemented? [16:53] <Daemon404> because -crf is a private option for x264 [16:53] <Daemon404> which shouldnt exist IMO [16:54] <Daemon404> i thought about adding a similar thing for x265 but it's still a hack [16:54] <gerion> with "shouldnt exist" you mean someway globally? [16:54] <Daemon404> i dont think the x264 wrapper should have exposed it in such a way [16:54] <Daemon404> but that's in the past, can't change it... [16:55] <Daemon404> you're not the first to ask... i guess i should add it [16:55] <Daemon404> begrudgingly. [16:56] <gerion> ;) ok, would be more consistent [16:56] <gerion> (aka intuitive) [16:56] <Daemon404> it's not intuitive [16:57] <Daemon404> consistent yes [16:57] <gerion> if you know how to work with x264 you kind of automatically know how to work with x265 [16:57] <gerion> maybe intuitive is an uncorrect word for this [16:58] <gerion> atm you have to read documention to control x265 even if you can control x264 through ffmpeg [16:59] <Daemon404> that's probably a good thing. [16:59] <gerion> ;) [16:59] <Daemon404> because everyone uses x264 wrong. [16:59] <Daemon404> also hevc is not the same as avc [16:59] <Daemon404> you have to read its options anyway [17:00] <Daemon404> crf should be fine, sicne ratecontrol is generally always available via high level options [17:01] <Daemon404> ill send a patch shortly [17:01] <gerion> thank you [17:02] <gerion> why everyone uses x264 wrong? [17:02] <Daemon404> because they dont understand the options. [17:02] <gerion> cause of twopass/crf/specific bitrate confusion? [17:02] <Daemon404> nah not ratecontrol [17:02] <Daemon404> but stuff like ipratio/pbratio, etc [17:05] <gerion> is this often set? [17:05] <Daemon404> cargo culted stuff tends to be [17:05] <Daemon404> from random ffmpeg "tutorials" [17:06] <Daemon404> (or x264) [17:06] <Daemon404> anyway, i argue one should be reading the docs for encoder specific options. [17:07] <gerion> yeah ok, but if x265 and x264 behave in similar way regarding a specific option they could be mapped together [17:08] <Daemon404> funny you menton that [17:08] <Daemon404> because x264 and x265 have different 'scales' for crf. [17:08] <Daemon404> so technically they don'y. [17:08] <Daemon404> dont* [17:08] <thardin> whyyy [17:08] <Daemon404> thardin, it's the nature of crf [17:08] <Daemon404> even 10bit x264 and 8bit x264 differ [17:09] <Daemon404> it's an arbitrary scale [17:09] <thardin> ah [17:09] <Daemon404> you'd all know this if you read the docs! [17:09] Action: Daemon404 runs [17:09] <gerion> yay i noticed that, but x265 says: crf = quality based vbr the same x264 says [17:10] <gerion> so the effect is quite similar [17:10] <Daemon404> yeah but the scale is arbitrary [17:10] <Daemon404> ratecontrol is more or less consistent between encoders. hence ill expose it as such [17:11] <BtbN> Does ffmpeg have some code to write h264 nal bitstream? [17:13] <gerion> what do you mean with ratecontrol? the normal bitrate settings? you have to be very experienced to know which bitrate of a specific codec is necessary to produce the wanted result [17:13] <thardin> gerion: same goes for crf [17:14] <Daemon404> i mean interface-wise gerion [17:14] <Daemon404> BtbN, what do you mean [17:14] <BtbN> libva doesn't generate the sps/pps for you. You have to do that yourself... [17:14] <Daemon404> eh [17:14] <Daemon404> oh [17:14] <Daemon404> libva. [17:14] <Daemon404> well. [17:15] <Daemon404> the way libx264 and libx265 write elementary stream (nals) is that the encoders both have options to produce sps and pps [17:15] <Daemon404> and lavf just passes it through [17:15] <Daemon404> in rawenc.c iirc [17:15] <Daemon404> so you'd have to do it in the libavcodec end [17:15] <Daemon404> also wtf kind of crap lib makes you generate sps/pps yourself? :/ [17:16] <gerion> thardin: more less, crf 20 or so with x264 generate more less similar results (not depending on the inpu), that the equivalent on x265 is not 20 but 18 or so (don't know the real value) is kind of strange [17:16] <BtbN> It's next level stupid that libva doesn't do that. Specialy as it already has bit stream generation code in place. It also does generate large parts of the bitstream. [17:16] <BtbN> But not the SPS/PPS [17:16] <BtbN> I have no idea what the point if that is. [17:16] <Daemon404> you would think the encoder would be best suited to generate it... [17:16] <BtbN> You also have to pass the sps/pps bitstream into libva then, it has a special buffer for that. [17:16] <Daemon404> wut [17:16] <BtbN> Yes [17:17] <Daemon404> thats retarded [17:17] <BtbN> No wonder intel wrote its own wrapper for that lib [17:17] <Daemon404> and intel's wrapper isn't good either from what i hear [17:17] <BtbN> It's of questionable quality and future [17:18] <Daemon404> just like all their foss [17:18] <BtbN> I realy wonder who designed that libva encode api. Either i'm missing something, or this stuff is next level broken. [17:19] <Daemon404> it sounds like it's literally a thin wrapper over direct hw calls [17:19] <BtbN> I looked into the driver, and it does have h264 nal generation [17:19] <Daemon404> oh. [17:19] <BtbN> But not for the sps/pps [17:19] <BtbN> like, wtf [17:20] <BtbN> B frames also look like they are a lot of pain. [17:20] <Daemon404> well x264 and 5 just pass me nals back [17:20] <Daemon404> and it's 0 effort. so, sucks to be you? [17:20] Action: Daemon404 runs [17:20] <BtbN> so does nvenc [17:21] <cone-408> ffmpeg.git 03Clément BSsch 07master:a00bab347518: avfilter/paletteuse: raise cache size from 64k to 512k [17:21] <BtbN> nvenc is actualy a realy nice api. Does precisely what it's expected to do. [17:25] <iive> BtbN: there are some bistream filters, e.g. annexb and stuff, would any of these help you? [17:26] <BtbN> Don't think so, i have to generate the entire thing from scratch [17:26] <BtbN> I already have code that does that, but it's like 2000 lines [17:27] <BtbN> https://github.com/BtbN/vlc-vaapi-enc/blob/master/vlc-h264-vaapi-enc.c#L622 well, not 2000, but still annoyingly long [17:36] <iive> :| [17:41] <wm4> haha [17:42] <wm4> obviously you should use gstreamer [17:43] <BtbN> btw., in gstreamer, they internaly write a wrapper around libva [17:43] <BtbN> so the actual code doesn't have to deal with the worst of it [17:44] <wm4> kind of like ffmpeg has hwaccel for decoding? [17:44] <BtbN> No, more like libyami-light [19:43] <cone-408> ffmpeg.git 03Xiaohan Wang 07release/2.2:a9602c6cfbe6: matroskadec: Fix read-after-free in matroska_read_seek() [19:43] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:e5b878b37f64: Merge commit 'a9602c6cfbe6fa06ff97ad01c0ffa9ad5ccff30f' into release/2.2 [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:56976346d989: avformat/utils: Fix number suffixes in tb_unreliable() [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:e61a7c9b4517: swresample/dither: Cleanup number suffixes [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:56e07e4caf48: avcodec/dxtory: Use LL instead of L number suffix [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:d0ed336d8a4a: avformat/matroskadec: Fix number suffixes [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:ab17d11310a9: avformat/smacker: Fix number suffix [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:cd5c78c804d1: avformat/omadec: fix number suffix [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:004cdd8b1538: avcodec/h264_cabac: use int instead of long for mbb_xy [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:4f2299b6b55f: avcodec/mpegvideo_enc: Fix number suffixes in rc_buffer_size calculation [19:50] <cone-408> ffmpeg.git 03wm4 07release/2.2:c7dc73a6c333: avformat/tta: fix crash with corrupted files [19:50] <cone-408> ffmpeg.git 03wm4 07release/2.2:3d8c51d69916: avformat/mpc8: fix hang with fuzzed file [19:50] <cone-408> ffmpeg.git 03wm4 07release/2.2:79d86b844f08: avformat/mpc8: fix broken pointer math [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:b902eab45a79: avformat/mpc8: Use uint64_t in *_get_v() to avoid undefined behavior [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:b0d387308525: avcodec/mjpegdec: Check escape sequence validity [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:4c246c65bfa2: avcodec/mjpegdec: Check number of components for JPEG-LS [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:bd9c755b22ff: swscale/utils: Limit filter shifting so as not to read from prior the array [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:7c3a3d47cf7f: avformat/thp: Check av_get_packet() for failure not only for partial output [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:eca1e3dcc847: avcodec/h264_ps: More completely check the bit depths [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:a3dca10470ec: avcodec/h264: Be more strict on rejecting pps_id changes [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:0c9d465e98c7: avcodec/h264: Be more strict on rejecting pps/sps changes [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:68d83bc3a692: avutil/opt: Fix types used to access AV_OPT_TYPE_PIXEL_FMT [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:8323f0944283: avutil/opt: Fix type used to access AV_OPT_TYPE_SAMPLE_FMT [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:e6093f5b85b3: avcodec/h264_slice: Do not change frame_num after the first slice [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:0afe061f2877: avcodec/h264_slice: Check picture structure before setting the related fields [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:f0526bc21eb4: avcodec/h264_slice: ignore SAR changes in slices after the first [19:50] <cone-408> ffmpeg.git 03Carl Eugen Hoyos 07release/2.2:b1b69baa0110: lavc/aarch64: Do not use the neon horizontal chroma loop filter for H.264 4:2:2. (cherry picked from commit 4faea46bd906b3897018736208123aa36c3f45d5) [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:0ee5f46fd1d1: avcodec/mjpegdec: Skip blocks which are outside the visible area [19:50] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:2dca276dcb9c: avcodec/arm/videodsp_armv5te: Fix linking failure with "g++ -shared -D__STDC_CONSTANT_MACROS -o test.so ... libavcodec.a" [19:59] <Daemon404> michaelni, fyi your "24 bit support" for wmapro is entirely broken [19:59] <Daemon404> it is massive distorted. [19:59] <Daemon404> massively* [20:00] <cone-408> ffmpeg.git 03Michael Niedermayer 07release/2.2:36cfee3adc70: Update for 2.2.13 [20:02] <michaelni> Daemon404, is there a ticket for this ? how can it be reproduced ? is it a regression? [20:03] <Daemon404> its not a regression because the support was never real [20:03] <Daemon404> it was an untested hack afaict [20:03] <Daemon404> every 24bit wmapro file i play is distorted [20:03] <Daemon404> let me see if i can get a file. [20:05] <michaelni> i only remember a recent partial 24bit fix for wmalossless not wmapro [20:06] <Daemon404> sorry [20:06] <Daemon404> i meant wma lossless [20:14] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:03dab49a1267: avcodec/vp56: Add () to protect TRANSPOSE()s arguments [20:25] <cone-408> ffmpeg.git 03Diego Biurrun 07master:f8c1719771dc: dca: Remove separate header for DCA ExSS [20:25] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:d2668ae47269: Merge commit 'f8c1719771dc4ac2e13e6bc8bf741854a30e3a86' [20:47] <cone-408> ffmpeg.git 03Vittorio Giovara 07master:5dc47a2bd52e: matroskaenc: Validate chapter start and end times [20:48] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:d302853bcabf: Merge commit '5dc47a2bd52e375ed742c45d08356b45098f458d' [20:48] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:a4cd057bc7dd: avformat/matroskaenc: Use the correct data type for the chapter times [20:58] <cone-408> ffmpeg.git 03Vittorio Giovara 07master:9d3b752fceb0: graphparser: Check av_get_token() memory error [20:58] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:9d5017a20c26: Merge commit '9d3b752fceb0f2a42cac7c2a1109b0629823c99f' [21:08] <cone-408> ffmpeg.git 03Vittorio Giovara 07master:81688e68f93f: avconv: Check rc_override memory allocation [21:08] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:15cd2a93e7f6: Merge commit '81688e68f93f3142e2093f1a3d226edaeb179992' [21:16] <cone-408> ffmpeg.git 03Vittorio Giovara 07master:266f241193b2: avplay: Check frame allocation inside video_thread() [21:16] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:32b2b7bd8c5a: Merge commit '266f241193b2fa8c99bb8b1f007c66bedd3b7d97' [21:23] <cone-408> ffmpeg.git 03Vittorio Giovara 07master:733f4b05f0e1: avplay: Check format allocation inside decode_thread() [21:23] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:79216a189b4e: Merge commit '733f4b05f0e120ddd0393b23f2b6d9106cf922e4' [21:35] <cone-408> ffmpeg.git 03Vittorio Giovara 07master:cb70a93ca1e7: lavc: Document interaction between avcodec_open2() and decoding routines [21:35] <cone-408> ffmpeg.git 03Vittorio Giovara 07master:bb7701684c71: lavf: Document the codec context initialization [21:35] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:2980c8e7d34e: Merge commit 'cb70a93ca1e7fae1b5dfb65ad149d311a65206a4' [21:35] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:563ed912eaac: Merge commit 'bb7701684c7170dfd5f33a3c7d827265844619f2' [21:42] <cone-408> ffmpeg.git 03Vittorio Giovara 07master:d89e58f53939: dct-test: Fix initialization syntax [21:42] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:beae6ac5db9d: Merge commit 'd89e58f53939d91a88aa7042c94d9d1cd364da52' [21:44] <iive> finally I see some libav merges. I was starting to wonder is it me that always miss them, or there aren't that many anymore. [21:48] <cone-408> ffmpeg.git 03Hugo Beauzée-Luyssen 07master:da7e31a240f2: rmdec: Check memory allocations from ff_rm_alloc_rmstream() [21:48] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:575ed4fe37fd: Merge commit 'da7e31a240f276836a0b90ca6c0714181b353cc4' [22:00] <cone-408> ffmpeg.git 03Hugo Beauzée-Luyssen 07master:a8765852158e: movenc: Check memory allocations [22:00] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:80c5dc56104d: Merge commit 'a8765852158ecb2ae34895fa35ff51dc95c186f9' [22:00] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:95165f7c1b53: avformat/movenc: Fix stts_entries allocation check [22:01] <akira4> ubitux, I used ./ffmpeg -i test.ass -flag2 +rawtextsub out.srt to transcode ass subtitles to srt. Is it the right way? [22:02] <ubitux> yes [22:02] <cbsrobot_> ubitux: what do you mean with "string needs to be dupped" ? [22:03] <cbsrobot_> is there an example ? [22:04] <ubitux> print_format in ffprobe [22:14] <joeflateau> Is there a magic way of using tee to output hls and rtmp from one encode? I can't seem to get the bsf option right [22:18] <BtbN> nginx-rtmp can do that [22:18] <BtbN> You stream to it via rtmp, and it distributes the stream via rtmp and hls [22:21] <cone-408> ffmpeg.git 03Hugo Beauzée-Luyssen 07master:3035d21b5a77: nutdec: Check memory allocations [22:21] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:880cee004418: Merge commit '3035d21b5a77df0e9531df397fc06d59488996c7' [22:29] <cone-408> ffmpeg.git 03Federico Tomassetti 07master:e51f22122d23: swscale: Check memory allocations [22:29] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:2a1b79d7e659: Merge commit 'e51f22122d23589e93ac4f0b3e570bb925755915' [22:38] <cone-408> ffmpeg.git 03Vittorio Giovara 07master:ec17782e17de: dvdsubdec: Check memory allocations [22:38] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:3d5c0ba816b9: Merge commit 'ec17782e17de1e8501ca213e276dfe5412ff1d11' [22:46] <cone-408> ffmpeg.git 03Vittorio Giovara 07master:31dc73e92a96: vorbisdec: Check memory allocations [22:46] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:7a7b77e6c97a: Merge commit '31dc73e92a96f08d07650c0e7d31c0b9a1465d46' [22:55] <cone-408> ffmpeg.git 03Vittorio Giovara 07master:fe0f4e56577a: avpacket: Check buffer reference [22:55] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:62aec0c2d5b3: Merge commit 'fe0f4e56577a2dbd373bb50b0ae2d49e69d822fc' [23:05] <cone-408> ffmpeg.git 03Himangi Saraogi 07master:18f4fa251b0e: aac_adtstoasc_bsf: Check extradata memory allocation [23:05] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:38e867f6c08e: Merge commit '18f4fa251b0eb36392839f5bf6180f280dc04d8d' [23:06] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:2e786bc49898: avcodec/aac_adtstoasc_bsf: Clear extradata_size on allocation failure [23:26] <cone-408> ffmpeg.git 03Himangi Saraogi 07master:a1e2c47cd4fd: libxvid: Return meaningful error messages [23:26] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:d128794fbc5e: Merge commit 'a1e2c47cd4fdaa64beda7e6dfa623b65c46f5012' [23:34] <cone-408> ffmpeg.git 03Himangi Saraogi 07master:f3e045263e44: qdm2: Return meaningful error codes [23:34] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:41572ac698b3: Merge commit 'f3e045263e445c4ab54d85ecae359494cd96a3e2' [23:42] <cone-408> ffmpeg.git 03Himangi Saraogi 07master:42c8f92e2fa3: wmv2: Return meaningful error codes [23:42] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:d06dce21a454: Merge commit '42c8f92e2fa390fa17b03d37b4323ec0d721d4cd' [23:51] <cone-408> ffmpeg.git 03Vittorio Giovara 07master:c78dc12118cb: riff: Support ProRes in avi (APCH fourcc) [23:51] <cone-408> ffmpeg.git 03Michael Niedermayer 07master:97bd0e2a93f1: Merge commit 'c78dc12118cb5b2a69a17f38c69f77a42e89a9c4' [00:00] --- Wed Feb 18 2015

1 0

ffmpeg.log.20150217
by burek 18 Feb '15

18 Feb '15

[00:27] <latenite> Hi folks, I try to do something like: http://d1u2s20mo6at4b.cloudfront.net/wp-content/uploads/01_subkeys.gif [00:28] <latenite> How do you think did he create the frames? [00:28] <latenite> I did split the gif appart with: [00:28] <latenite> convert Desktop/GIF-in-PDF/01_subkeys.gif -coalesce Desktop/1/animate_%d.jpg [00:29] <latenite> and he somehow managed to get one frame per change/letter/output [00:29] <latenite> but how? [00:47] <latenite> Hi folks, how can I used ffmpeg to record my screen? -> NOT as mp4 or mkv or any movie format BUT with many many png image frames? I just need the frames! [00:47] <c_14> Same way as normal just with out%03d.png as the output filename [00:48] <ramiro> it might make sense to use -vsync so that you don't get duplicate or dropped frames [00:53] <latenite> ramiro, somemow the resulting mp4 looks bad [00:53] <latenite> ffmpeg -f x11grab -s 1440x900 -i :0.0 sc.mp4 [00:53] <latenite> I would like to habe lossless images eventually [00:54] <c_14> Then use a lossless codec? [00:57] <latenite> ffmpeg -f x11grab -s 1440x900 -i :0.0 Desktop/frames/out%03d.png [00:57] <latenite> seems to work. But adding -vsync gives me this error: [00:57] <latenite> Expected number for vsync but found: -f [00:57] <latenite> Conversion failed! [00:57] <latenite> Is there a way to only get "new" frames? [00:58] <latenite> I want to make an anitmation out of the frames and would like to omit the duplicates [01:03] <grepper> use diff or md5sum ? [01:06] <latenite> grepper, you mean to afterards sort the images out? [01:06] <latenite> Is there a way to ONLY capture a frame when there is a change? [01:09] <grepper> you mean like a sensor that activates a camera? [01:31] <latenite> Sorry my DSL is bad [01:32] <latenite> I make screen capture frames like this [01:32] <latenite> http://i.imgur.com/IUs8ORs.png [01:32] <latenite> next frame is: [01:32] <latenite> http://i.imgur.com/HvRg8jD.png [01:33] <latenite> with ffmpeg -f x11grab -s 1440x900 -i :0.0 $HOME/screencast-frames/out%010d.png [01:33] <latenite> why is is missing so many frames? [01:33] <latenite> I expected it to record every letter I type "l" "s" " " "-" "l" "a" [01:34] <latenite> I am typing super fast. And it just got the whole string [01:34] <c_14> Increase the recording fps [01:34] <latenite> how? [01:34] <c_14> -framerate number [01:34] <c_14> As an input option [01:34] <latenite> what is a "good" number? 30? [01:35] <c_14> How many frames do you want per second? [01:37] <spaam> over 9000 [01:46] <latenite> c_14, see [01:46] <latenite> http://i.imgur.com/KttF1kn.png [01:46] <latenite> and [01:46] <latenite> http://i.imgur.com/q2i3kZm.png [01:46] <latenite> the pfs was set to 30 [01:46] <latenite> and these are two consecutive frames [01:47] <latenite> does that show/explain my problem?! [01:49] <latenite> There are actually 30 pfs but they all look the same [01:49] <latenite> fps [02:02] <c_14> Works fine for me [02:02] <c_14> Try updating your version of ffmpeg [02:03] <c_14> with -framerate 30, and `while true; do echo 'foo'; sleep 0.1; done' I get 1 foo every 3 frames [02:04] <latenite> ffmpeg version 2.2.11 Copyright (c) 2000-2014 the FFmpeg developers [02:04] <latenite> built on Jan 17 2015 17:03:30 with gcc 4.8.3 (Gentoo 4.8.3 p1.1, pie-0.5.9) [02:04] <latenite> is that to old? [02:05] <latenite> c_14, did you record to movie or to images? [02:05] <c_14> latenite: images [02:06] <c_14> `./ffmpeg/ffmpeg -video_size 1280x720 -framerate 30 -f x11grab -i :0.0 out%03d.png' [02:06] <c_14> Don't think the video_size, should matter [02:06] <latenite> just like I did it. [02:06] <c_14> I can try it again without if you want though. [02:06] <latenite> hmm??? [02:06] <latenite> no I want images [02:06] <c_14> -video_size just limits the amount of my desktop that's captured [02:06] <c_14> I was referring to that [02:07] <latenite> yes [02:07] <latenite> I did: [02:07] <latenite> ffmpeg -f x11grab -s "$(xdpyinfo | grep 'dimensions:' | awk '{print $2}')" -i $DISPLAY test.mp4 [02:07] <latenite> and in the mp4 I see all the minor changes [02:07] <latenite> that is sooo weird?! [02:09] <c_14> Try building from git master head with a minimal configure [02:10] <latenite> I think I found it....let me experiment for a minute :D [02:14] <c_14> Also, there's a 90% probability that you really don't want to disable threads in ffmpeg. Just wanted to point that one out. [02:27] <latenite> c_14, its a performance issue [02:27] <latenite> when I drop the fps to 10 i get decent results [02:27] <latenite> almost evrey change is captured. [02:28] <latenite> Now why is my performace so bad? I have a core i5 4th gen and 16GB ram intel grafics [02:28] <c_14> Turn on threads [02:28] <c_14> As a start [02:30] <latenite> ok, let s see how that effects it. [02:44] <latenite> c_14, thread is what made it work. I am all good now. [02:44] <latenite> c_14, thank you tons for you help. :D [02:47] <c_14> np [06:32] <Matador> damn [06:32] <Matador> having problems cross-compiling ffmpeg [06:33] <Matador> libquvi not found using pkg-config [06:33] <relaxed> config.log will show you the reason [06:34] <Matador> I have libquvi installed though [06:35] <relaxed> that doesn't matter, ffmpeg's configure runs tests to make sure it can be linked. [06:35] <relaxed> and right now it can not be [06:35] <Matador> I was doing this one on CentOS [06:35] <Matador> https://github.com/rdp/ffmpeg-windows-build-helpers [06:36] <Matador> http://pastebin.com/HbN2kRmv [06:38] <relaxed> config.log [06:38] <Matador> I have a build.log [06:38] <Matador> ~19k lines [06:40] <Matador> Package libquvi was not found in the pkg-config search path. [06:40] <Matador> Perhaps you should add the directory containing `libquvi.pc' [06:41] <Matador> its there @ /usr/lib64/pkgconfig/libquvi.pc though [06:44] <relaxed> Are you using the most recent version of that build script? I see mention of a quvi update on the github page. [06:44] <Matador> ya I just grabbed it [06:47] <relaxed> ok, then look at config.log in the ffmpeg source dir [06:47] <Matador> How can I add the path to libquvi.pc to build ? [06:47] <Matador> to the search path... [06:48] <relaxed> PKG_CONFIG_PATH=/path/to/dir [06:50] <Matador> http://smoothstreams.tv/tmp/config.txt [06:52] <relaxed> cd ffmpeg; PKG_CONFIG_PATH=/path/to/pkgconfig ./configure --enable.... [06:54] <relaxed> although I find it hard to believe this script isn't building to a specific location and adding the pkgconfig path on its own [06:54] <Matador> I dont know what to enable.. prefer script just do it [06:54] <Matador> It did work before.. like last year [06:55] <relaxed> http://ffmpeg.zeranoe.com/builds/ [06:55] <Matador> was just trying to upgrade my version [06:55] <Matador> ya I'm aware of those builds.. but I want the better AAC [06:55] <Matador> the libfdk-aac [06:56] <relaxed> Then you need seek support with the authors of that script. [06:57] <sekon> Matador: if you want .. people can be paid to look into it too ?? elance maybe ? [06:58] <Matador> Sure [06:58] <Matador> I appreciate your time relaxed [06:58] <Matador> Would like the build to work on CentOS 7 [06:59] <relaxed> Do you even need libquvi support? [06:59] <Matador> I dont know [06:59] <relaxed> Then you don't :) [06:59] <Matador> Just streaming flash [07:00] <relaxed> Ok, then really don't need it. It's used for downloading video from streaming sites such as youtube. [07:00] <Matador> ok much appreciated [07:00] <sekon> Matador: just disable it and see if your site works as expected [07:00] <sekon> if not .. you will need professional help most likely [07:00] <Matador> I cleared whole sandbox [07:00] <Matador> so rebuilding * [07:01] <Matador> but I see the build path in the script, and will remove the --enable-libquvi if it comes up with a problem again [07:02] <Matador> I set thread count to 8 so should build pretty fast :) [07:23] <sekon> Hello i want a dummy ffmpeg opt of type OPT_STRING [07:23] <sekon> it will bot be used anywhere in ffmpeg .. but it should not make ffmpeg crash [07:23] <sekon> is there a way i can make it happen ? [07:24] <sekon> *crash as in segfault in av_free [08:07] <Matador> build w/o quvi [08:16] <Matador> but error on encoding... Past duration 0.984367 too large [08:21] <Matador> http://pastebin.com/H8a37b2c [08:49] <k_sze[work]> Hello [08:50] <k_sze[work]> I don't quite understand how to use codec options, like on this page: https://www.ffmpeg.org/ffmpeg-codecs.html#Codec-Options [08:50] <k_sze[work]> In particular, I don't understand the time_base option [08:50] <k_sze[work]> How do I specify a rational number on the command line? [08:54] <k_sze[work]> Like, why can't I do this: `ffmpeg -i 'in.nut' -c:v copy -copyinkf -vsync drop -time_base '1 / 30' -r 30 -y ~/out.nut` [08:54] <k_sze[work]> it tells me "Unrecognized option 'time_base'." [09:01] <relaxed> I think it's an input option [09:03] <relaxed> And I doubt you can drop frames while stream copying [09:41] <k_sze[work]> I don't want to drop frames. [09:41] <k_sze[work]> I want to change the time base. [09:42] <k_sze[work]> relaxed: `-vsync drop` doesn't mean dropping frames. [09:42] <k_sze[work]> it means the discard the video sync information from the input stream and completely recompute it. [09:43] <relaxed> ah, did you try -time_base 1/30 as an input option? [09:44] <k_sze[work]> relaxed: doesn't recognise it. [09:44] <k_sze[work]> even as input option. [10:02] <relaxed> Could someone please test alsa input using my builds? I just fixed a bug a user reported where it hardcoded the prefix of my build and was looking for alsa.conf in a non-stand location. It works here but I just want to be sure there's no other issues. http://johnvansickle.com/ffmpeg/ [10:17] <termos> Hi. Right now I'm generating hls segments and writing them to disk. Is it possible to push them directly to a http endpoint? [10:18] <sekon> termos: write ffmpeg out to stdout capture in popen and push any way you want it anywhere ? [10:21] <termos> I'm using the library directly, and my program can have multiple ongoing encodings [10:22] <sekon> termos: this might be better suited for ffmpeg-devel [10:22] <ubitux> sekon: no. [10:22] <ubitux> ffmpeg-devel is not for helping library users [10:22] <sekon> ok .. sorry my bad [10:23] <sekon> i had it really wrong then [10:26] <termos> Hehe :) I don't find it in the docs so I guess it won't do this. I could just try and see what happens [15:04] <chris_g4> hello guys, i have something interesting here, i have a live stream which when i open it on vlc i can see on codec page this image http://postimg.org/image/hiwawurlt/ , i think the guy has put the epg info on the flow stream but when i try to do the same with ffmpeg i cant.. i tried to put for example only a title as metadata but nothing... is not working.. [15:04] <chris_g4> anyone have any idea [15:04] <chris_g4> ? [15:17] <chris_g4> also when i tried ffmpeg -i source -f ffmetadata metadata.txt to take the output i have this :error" [15:17] <chris_g4> Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used) [15:18] <robotbrain> can I spevify the starting frame for the input? [16:24] <Cheenis> Due to a change in broadcast from the BBC I need to put FFMPEG on my synology DS210J NAS to enable my Squeezebox radio to work with the new BBC streams. From what I have read I need to get hold of a FFMPEG binary compiled for version 1.0 or higher for the Marvell Kirkwood mv6281 ARM. Can somebody point me to one as I can't seem to find it. [16:24] <Cheenis> Please [16:31] <Cheenis> Or any advice as to how I can do this for myself; i.e., create my own binary - not sure how realistic that is as I'm only partially technical. :$ [17:23] <circ-user-TNwVO> Hi, could anyone help me with this: http://stackoverflow.com/questions/28519403/ffmpeg-command-issue issue ? [17:31] <t4nk370> Hi, could anyone help me with this: http://stackoverflow.com/questions/28519403/ffmpeg-command-issue issue ? [17:52] <dimi> Hello, can you help me with the info how float conversion is implemented in libswresample and libavresample? Are the float conversion procedures identical in both? If i convert a float to s16 and back will i always get identical results? [18:03] <circ-user-TNwVO> No point of asking from what I can see [18:08] <hans_s> circ-user-TNwVO: fps=... is a video filter, you have to add -vf before that [18:09] <circ-user-TNwVO> Hi, thanks for an answer! [18:13] <circ-user-TNwVO> I have tried it in the past but all this resulted in was only one/two thumbnails being generated ( which shouldn't happen - at least according to where I got that command from: http://superuser.com/a/608125 ) [18:13] <circ-user-TNwVO> Really sorry for being an FFMPEG N00b [18:19] <chris_g4> hello again , someone can help me about the metadata, i have a live stream as an example and i would like to make the same metadata output like this one, i can pay for this thanks [18:25] <hans_s> circ-user-TNwVO: I think the "fps=fps=1/600" results in a frame rate of 1/600 fps (1 frame every 10 minutes), so maybe you have to adjust that to the length of your video [18:26] <circ-user-TNwVO> hans_s Oooh, I see! Thank you very much! [18:27] <hans_s> if you want 6 thumbnails, you probably have to replace 600 with <length of video in s>/6 [18:39] <z16> Hey guys, disclaimer: I know next to nothing about video conversion/encoding... [18:40] <z16> I have ogg vorbis video files that I need to convert to FLV, and when I try to do that the resolution changes [18:41] <z16> Er.... wait [18:42] <z16> The input resolution *says* 720x480... but it's actually 640x480 [18:42] <z16> The converted file has the correct value oO [18:42] <z16> That's... strange, but I guess not relevant to the channel anymore! [18:45] <Mavrik> z16, oh you met your first anamorphic video :P [18:48] <z16> Mavrik: Not sure what that is, but apparently! [18:54] <Mavrik> It's when you have a video of resolution A that should be shown at resolution B [18:54] <Mavrik> Because legacy! :P [19:13] <Cheenis> Re: The question earlier requesting a FFMPEG binary V1.0 or higher compiled for the Marvell Kirkwood mv6281 ARM - is my question poor, is there nobody that can help me or am I in the wrong place? Sorry, I'm totally new to what I'm trying to do so I don't know what I don't know if that makes sense. [19:22] <chris_g4> stronzi testi di cazzo che non rispondete mai.. che state facendo tutto il giono rompete coglioni! [19:29] <Mavrik> Cheenis, if noone built ffmpeg for your NAS, you'd have to do it yourself - but setting up a proper toolchain (if it exists) and configuring ffmpeg with dependencies does require rather alot of technical knowledge [19:30] <Mavrik> Cheenis, did you try to see if ipkg (optware) has it already built? [19:32] <Matador> Anyone know what this -- Past duration 0.984367 too large -- error means ? [19:32] <Cheenis> Thanks for the reply Mavrik! I'm currently trying to load bootstrap onto my NAS which I believe will then allow me to run ipkg, but when I try to run the .xsh script it tried to make a directory and fails due to permissions. I am logged in as admin and I am currently researching how I change permissions, but haven't been able to work it out so far. [19:33] <Cheenis> If there is an easier way to get ipkg working then I'm all ears. [19:33] <Matador> http://pastebin.com/H8a37b2c [19:34] <Mavrik> Cheenis, sorry, I haven't really dealt with Synologies all that much :/ [19:36] <Mavrik> Matador, it looks conected to the video sync / framerate handling code [19:36] <Mavrik> it seems like you might not be defining input framerate properly? [19:36] <Cheenis> No probs. I'll have to do the hard yards in the forums. I really appreciate you replying in the first place Mavrik! The frustrating thing here really is the BBC changing the world on us without there being time to create a user friendly solution for those that haven't touched Vi in a good 15 years. It used to be my friend, but those days are clearly long gone :o [19:36] <Mavrik> :)) [19:36] <Matador> It works fine on older ffmpeg build :| [19:37] <Matador> input frame is correct [19:37] <Matador> always been like that [19:38] <Matador> bloody crazy :| [21:34] <joeflateau> does anyone possibly have an example of using tee pseudo-muxer to output hls and flv from one encode? [21:34] <joeflateau> keep banging my head against bitstream filters [23:03] <Matador> hmmm [23:04] <msmithng> joeflateau: did anyone come up with an example? [23:33] <joeflateau> msmithng: no, i didn't [23:33] <joeflateau> no, nobody did, i mean [23:34] <joeflateau> msmithng: i'm stuck on global headers now [23:35] <c_14> joeflateau: you could always scrap the pseudo-muxer, pipe the output of ffmpeg into a second ffmpeg process which uses -c copy and has 2 outputs [23:35] <c_14> Can help get around bitstream filter/flag issues [23:36] <joeflateau> i'm encoding different bitrates from one source (live stream) [23:37] <joeflateau> so, 3 bitrates each to 2 outputs [23:37] <msmithng> you mentioned you were working with hls as one of those outputs correct? [23:37] <joeflateau> yes [23:38] <joeflateau> 3 bitrates each to flv (rtmp) and hls [23:38] <grepper> I am trying to use ffmpeg to identify if a file is a video or not. I have found the 'file' command to be somewhat problematic, depending on how updated the magic file is, and the video format/codec. One gottcha is that it will call images a video stream, but I am filtering on 'image2| _pipe'. Anything I am missing ? [23:39] <grepper> current bash function is: is_video() { local result=$($FFmpeg -i "$1" 2>&1); grep -E -qi 'image2|_pipe' <<< "$result" && return 1; grep -qi Stream.*Video <<< "$result"; } [23:40] <joeflateau> http://pastebin.com/KtrJBjPs [23:40] <joeflateau> this is my stripped down example [23:42] <c_14> grepper: This might be easier to parse: `ffprobe -show_entries format=format_name -hide_banner' [23:42] <joeflateau> flv freaks out without +global_header and hls freaks out with +global_header [23:43] <c_14> joeflateau: can't you add ':flags=+global_header' into the [] for the rtmp stream? [23:44] <grepper> c_14: yes, I like ffprobe and use it optionally for other things, but don't want to (yet) make it a dependancy. (yes I know it is almost always installed with ffmpeg) [23:44] <joeflateau> then i get "Could not write header for output file #0 (incorrect codec parameters ?): Option not found [23:44] <joeflateau> " [23:50] <c_14> joeflateau: it seems to work for me (with the exception that I don't have an rtmp server so I can't answer the handshake) [23:51] <c_14> grepper: is there a concrete problem or are you just wondering if there's an edge case you might miss? [23:51] <grepper> c_14: yes, the latter. I know about image2 and {png,ppm,etc}_pipe , just wondering if there is anything I might miss with that [23:52] <grepper> otherwise it seems to work well for the files I have tested [23:52] <c_14> And that function is just for checking if it's a video or an image? [23:53] <joeflateau> [NULL @ 0x3a66220] Packet header is not contained in global extradata, corrupted stream or invalid MP4/AVCC bitstream [23:53] <joeflateau> [hls @ 0x3a656a0] Failed to filter bitstream with filter h264_mp4toannexb for stream 0 in file '/var/www/joef1/2100/playlist.m3u8' with codec h264 [23:53] <joeflateau> [mpegts @ 0x3a671c0] H.264 bitstream malformed, no startcode found, use the video bitstream filter 'h264_mp4toannexb' to fix it ('-bsf:v h264_mp4toannexb' option with ffmpeg) [23:53] <joeflateau> av_interleaved_write_frame(): Invalid data found when processing input [23:54] <joeflateau> that's with -flags +global_header [23:54] <grepper> well, it assists with both image vs video checking, and also elsewhere to check if an output file is uncorrupted ... or just data ... etc [23:55] <grepper> I also test if its empty of course :) [23:55] <joeflateau> when i move flags into [] I get: [23:55] <joeflateau> [flv @ 0x40ff860] Codec for stream 0 does not use global headers but container format requires global headers [23:55] <joeflateau> [flv @ 0x40ff860] Codec for stream 1 does not use global headers but container format requires global headers [23:55] <joeflateau> [flv @ 0x40ff860] Unknown option 'flags' [00:00] --- Wed Feb 18 2015

1 0

ffmpeg-devel.log.20150216
by burek 17 Feb '15

17 Feb '15

[01:46] <cone-859> ffmpeg.git 03Yusuke Nakamura 07master:4abfa387b823: mov: Change DTS-based seek into CTS-based seek. [01:46] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:f9e76d66e93a: Merge commit '4abfa387b8234736f6e0e541951e3d5eb60eb843' [01:46] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:38e641a060e0: avformat/mov: Bypass av_add_index_entry() [01:54] <cone-859> ffmpeg.git 03Peter Meerwald 07master:67690683130f: g722: Split out g722_qmf_apply() function into g722dsp.c [01:54] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:47df21e1727c: Merge commit '67690683130faf37dd9d969ced15eba2a1940ade' [02:01] <cone-859> ffmpeg.git 03Peter Meerwald 07master:10f160768b82: g722: Reduce number of pointers passed to g722_apply_qmf() function [02:01] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:9f1792982eb6: Merge commit '10f160768b824f00933f33bc69f1fae89a25dfc8' [02:07] <cone-859> ffmpeg.git 03Peter Meerwald 07master:b16057b5fe9f: g722: Unroll g722_apply_qmf() [02:07] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:9e0a1bd2b977: Merge commit 'b16057b5fe9f02442c52388037891c2e381205fc' [02:15] <cone-859> ffmpeg.git 03Peter Meerwald 07master:73ae0a9d1285: g722: Split out computation of band->s_zero and unroll code [02:15] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:39b40be2a130: Merge commit '73ae0a9d12857852222363f9a7c14d07058ebfd3' [02:29] <cone-859> ffmpeg.git 03Peter Meerwald 07master:702458538d4e: g722: Add ARM NEON implementation for g722_apply_qmf() [02:29] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:e86f754b1c5f: Merge commit '702458538d4e52809bcef460d39baabf061b16b5' [02:35] <cone-859> ffmpeg.git 03Diego Biurrun 07master:65df9904ceb5: dca: Add array sizes to array declarations [02:35] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:9d4f8366271e: Merge commit '65df9904ceb5477a63e99e1e8fd8ace9a7cb8bda' [02:55] <cone-859> ffmpeg.git 03Diego Biurrun 07master:3a651f599a18: dca: Move data tables from a header to an object file [02:55] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:815daf1607b0: Merge commit '3a651f599a18b023602370b67a77eb0efa309b20' [03:02] <cone-859> ffmpeg.git 03Himangi Saraogi 07master:81ef46020fa6: bktr: Use av_strerror and propagate error codes [03:02] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:e97a9666ef45: Merge commit '81ef46020fa654720e3beb4f5551ba634fae262a' [04:42] <cone-391> ffmpeg.git 03James Almer 07master:03adafb318e4: x86/g722dsp: add ff_g722_apply_qmf_sse2 [09:22] <j-b> 'morning [09:25] <Rathann> good morning, j-b :) [11:11] <yaronj> Hi, I was looking around to find the reason for the warning while running with the compiled FFmpeg: [11:11] <yaronj> "No accelerated colorspace conversion found from yuv420p to rgb24" [11:11] <yaronj> and found it's related to the reason of not usging the GPU. [11:11] <yaronj> did the following with no difference: [11:11] <yaronj> 1) compiled the FFmpeg with x264 [11:11] <yaronj> 2) added to the configure flags: --enable-neon [11:11] <yaronj> this warning become when calling: [11:11] <yaronj> sws_getContext with PIX_FMT_RGB24 where the source is YUV420P [11:11] <yaronj> compiled for iOS arm64 [11:11] <yaronj> any idea? [11:55] <cone-225> ffmpeg.git 03zhaoxiu.zeng 07master:0f834155690a: avcodec/vc1_mc: move median4() to mathops.h [12:21] <yaronj> cone-225, I did as you wrote but no change, any other idea? [12:34] <ramiro> what's a good format to store arbitrary data besides audio and video? [12:34] <ramiro> I want to record gps position with proper pts [12:59] <kierank> there is a us military gps format for ts [12:59] <kierank> and something similar for phones with mp4 [13:25] <compn> (if ffmpeg didnt discard gps metadata in mp4... eheh) [13:26] <kierank> should change all the uav metadata to compn's home [13:43] <cone-225> ffmpeg.git 03Seppo Tomperi 07master:e40e446efdd0: hevcdsp: HEVC deblocking ARM NEON register clobber fix [17:23] <ramiro> I have an mts file with a hdmv_pgs_subtitle track. if I play that file in VLC, there is no subtitle on the screen. if I remux it to mkv (with -{a,v,s}codec copy), VLC will show the subtitles on screen. [17:24] <ramiro> is it possible that the hdmv_pgs_subtitle track is somehow hidden in vlc? [17:52] <cone-225> ffmpeg.git 03Clément BSsch 07master:f9240ec01abb: avcodec/gif: fix off by one in column offsetting finding [17:57] <ubitux> so what are the maintained releases? [17:57] <ubitux> 2.5, 2.4, 2.2 and 1.2? [17:58] <jamrial> michaelni sent a patch to drop 1.2, since the two distros still using it apparently haven't updated to the latest point release in a while [17:59] <ubitux> ok, so i have to backport to 2.5, 2.4 and 2.2 [17:59] <ubitux> ok [18:03] <michaelni> jamrial, ubitux yes [18:04] <cone-225> ffmpeg.git 03Clément BSsch 07release/2.5:e5cc73e0a589: avcodec/gif: fix off by one in column offsetting finding [18:05] <ubitux> we should really limit ourselves to 2 releases [18:05] <ubitux> a lts and the "latest" [18:06] <ubitux> rolling releases distro will pick the latest release, and stable distro will pick the lts [18:07] <ubitux> currently, since we don't define the lts, distro pick a random one and we are stuck to maintain it, because they consider that's their stable [18:11] <ubitux> and we need to define how long lts is [18:12] <ubitux> latest stable being something like 3 months right? (assuming the new release replaces the old one) [18:13] <cone-225> ffmpeg.git 03Clément BSsch 07release/2.4:dbda57469456: avcodec/gif: fix off by one in column offsetting finding [18:14] <jamrial> ubitux: in a couple distros they don't go with a random stable but rather the ones that "match" one of libav's release [18:15] <jamrial> hence Gentoo being on 1.2, because they are also using libav 9 [18:16] <ubitux> ah yeah, i almost forgot that... [18:16] <ubitux> dammit [18:17] <jamrial> coincidentally 2.2 and 2.4 match libav 10 and 11, so we're good in that regard [18:18] <ubitux> i see [18:18] <ubitux> that's a bit annoying. [18:21] <Daemon404> the important point is here is that now we are truly free [18:21] <Daemon404> without lena [18:21] <Daemon404> should drop all lena-including releases [18:22] <ramiro> next release should be called "We'll miss you Lena" [18:26] <cone-225> ffmpeg.git 03Clément BSsch 07release/2.2:d5b20daeb0e3: avcodec/gif: fix off by one in column offsetting finding [18:27] <ubitux> ramiro: well yeah we could name the next release Lena :) [18:28] <Daemon404> nah [18:28] <ubitux> why? [18:28] <Daemon404> let's codename it "non-free" [18:28] <Daemon404> for extra confusion [18:28] <ubitux> next one will be hitler if you want [18:29] <Daemon404> lul [18:42] <cone-225> ffmpeg.git 03Kevin Wheatley 07master:88868d81a5d7: avformat/rtpdec_h264: fix compile failure with -DDEBUG [19:46] <Daemon404> https://www.dropbox.com/s/x129vklzoi774d7/sf.png?dl=0 [19:46] <Daemon404> for all you msys2 users.. beware that sf.net derped up [19:53] <jamrial> happened a couple days ago as well. forcing pacman to update the packages with -yy once sourceforge is back online will fix the databases if they got corrupted like this [19:53] <Daemon404> yeah [19:53] <Daemon404> apparently its been like this on and off for a few days [19:53] <Daemon404> i had to use gmane to check the msys2 ML [19:53] <Daemon404> (since it is... also hosted of sf.net, which is down) [19:58] <cone-225> ffmpeg.git 03zhaoxiu.zeng 07master:bf9906808992: avcodec/vc1_mc: factor out helper functions for scaling blocks [20:44] <Daemon404> now it has upgraded from returnign garbage to timing out [20:48] <wm4> why do they host things on sf.net [20:48] <wm4> other than "because they always did" [20:48] <Daemon404> i have no idea [21:01] <cone-225> ffmpeg.git 03wm4 07master:4bf3fab28115: hls: fix a null pointer dereference [21:48] <cone-225> ffmpeg.git 03James Almer 07master:22596383f3b8: avcodec/hevcdsp: optimize pixel comparison in sao_edge_filter [22:19] <cone-225> ffmpeg.git 03Christophe Gisquet 07master:89cb4995fa0c: x86: hevc_mc: save 1 gpr in epel filter loading [22:57] <cone-225> ffmpeg.git 03James Almer 07master:6e3651983c7c: avcodec/hevcdsp: add missing () to CMP macro [23:10] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:ced26a2c653b: avformat/tls: add () to protect macro arguments [23:10] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:37013fd018ae: swresample/swresample-test: Add () to protect uint_rand() argument [23:10] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:17a62cbcb9ee: avcodec/bmvvideo: Add () to protect NEXT_BYTE() argument [23:58] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:235782260da4: avcodec/me_cmp: add () to protect avg*s arguments [23:58] <cone-225> ffmpeg.git 03Michael Niedermayer 07master:0b638b83c66d: avformat/mpc8: Add () to protect MKMPCTAG()s arguments [00:00] --- Tue Feb 17 2015

1 0

ffmpeg.log.20150216
by burek 17 Feb '15

17 Feb '15

[01:26] <matham> I have been using the logic of av_image_fill_pointers to determine if the user passing plane buffers missed a plane for the format, as well as the required plane size. [01:27] <matham> However, the second plane is optional for PAL formats so, unless I make explicitly an exception for PAL, my code would return an error if the user didn't supply that plane. [01:28] <matham> Is pal the only format whose other planes are optional? Or do I have to make an exception for other formats? [04:03] <nell> hi [04:38] <wad> Hi guys. Over the past month, I've captured the contents of 34 mini DV tapes (home videos) to my hard drive. They are .avi files, 720x480. [04:38] <wad> Each file is about 13 GB in size. [04:38] <wad> I'm running Ubuntu Linux. [04:38] <wad> I'm now attempting to cut them up into smaller files, and transcode them into a less bulky file format. [04:39] <wad> I've tried using a few program: HandBrake, Pitivi, WinFF, OpenShot. [04:39] <wad> All of them except for OpenShot choke on the input files. They lock up or crash. [04:40] <wad> OpenShot seems to be working, so far. My question is: What format should I export the files as? [04:40] <wad> I'd like to be able to view them on PCs, mobile devices, and my old hacked XBox. [04:41] <wad> I think the framerate is 24.97 fps. [04:41] <wad> OpenShot gives me tons of export options, but I don't know which one to choose. [04:41] <wad> Any guidance would be welcome. [07:20] <YaMoonSun> How do I recommend suggestions to developers? It seems .iso files aren't being mapped, and I'd like to be able to make a proper rip without having to extract an .mkv first. [07:35] <relaxed> YaMoonSun: dvd iso? [07:35] <YaMoonSun> Yeah, it was extracted using dvd decrypter [07:38] <relaxed> ffmpeg can't read DVDs, just have dvd decrypter dump the title to a vob [07:38] <YaMoonSun> I dunno if it can, but will attempt. [07:39] <relaxed> or mount the DVD and use the vob as input for ffmpeg [07:39] <relaxed> (assuming it's the main title) [07:54] <YaMoonSun> Fingers crossed, thanks for the input [08:07] <YaMoonSun> So I did 'ffmpeg -i "Source.mp4" -c:v copy -c:a copy -ss 00:00:13 "Output.mp4" and now the music plays before the video appears in VLC, what do? [08:07] Action: YaMoonSun runs to get something to eat [08:08] <YaMoonSun> Also, do I need to use .mkv .ogg and .webm when supplying data to Archive.org? [08:08] <YaMoonSun> Implying, do the codecs need to be as free as the content? [08:33] <YaMoonSun> Could it be because I used checked the WebOptimization box in the first .mp4 and don't know how to manually optimize for web? [09:53] <YaMoonSun> I'm trying to take one part of a webm and put it into another webm, but my file keeps appearing corrupted. No video. [09:53] <YaMoonSun> http://pastebin.com/ayzRuqxB [09:58] <relaxed> No video using vlc? [10:00] <YaMoonSun> Yeah, not unless I do -c:v libvpx -b:v 1M [10:31] <YaMoonSun> Why would a site tell me my .gif that I made using ffmeg from a webm contains malicious content? =/ [10:38] <BtbN> "a site"? [10:40] <YaMoonSun> Worked fine on imgur, but 4chan refused it. [11:06] <yaronj> Hi, I was looking around to find the reason for the warning while running with the compiled FFmpeg: [11:06] <yaronj> "No accelerated colorspace conversion found from yuv420p to rgb24" [11:06] <yaronj> and found it's related to the reason of not usging the GPU. [11:06] <yaronj> did the following with no difference: [11:06] <yaronj> 1) compiled the FFmpeg with x264 [11:06] <yaronj> 2) added to the configure flags: --enable-neon [11:06] <yaronj> this warning become when calling: [11:06] <yaronj> sws_getContext with PIX_FMT_RGB24 where the source is YUV420P [11:06] <yaronj> compiled for iOS arm64 [11:22] <jpluijmers> Hi guys, I have been banging my head against this problem for days now and am at a complete loss. I have posted on several forums but no body seems to know a definitive answer [11:22] <jpluijmers> My command is: ffmpeg -y -i bunny-source.mp4 -i wm.png -filter_complex \"[1:v]scale=iw*0.3:-1[watermark]; \ [0:v][watermark]overlay=10:main_h-overlay_h-10[outv]" \ -map "[outv]" -map 0:a marked.mp4 [11:22] <jpluijmers> the output is: http://pastebin.com/Jn0krS7W [11:23] <jpluijmers> I am trying to get the watermark complex filter to scale percentage based on the width of the first input source. [11:23] <jpluijmers> But I cant seem to manage to get a valid (non-selfreferencing) token within the complex filter. [11:24] <jpluijmers> Any input at all would be greatly appreciated. [12:03] <jpluijmers> Hi guys, I have been banging my head against this problem for days now and am at a complete loss. I have posted on several forums but no body seems to know a definitive answer [12:03] <jpluijmers> My command is: ffmpeg -y -i bunny-source.mp4 -i wm.png -filter_complex \"[1:v]scale=iw*0.3:-1[watermark]; \ [0:v][watermark]overlay=10:main_h-overlay_h-10[outv]" \ -map "[outv]" -map 0:a marked.mp4 [12:03] <jpluijmers> the output is: http://pastebin.com/Jn0krS7W [12:03] <jpluijmers> I am trying to get the watermark complex filter to scale percentage based on the width of the first input source. [12:03] <jpluijmers> But I cant seem to manage to get a valid (non-selfreferencing) token within the complex filter. [12:03] <jpluijmers> Any input at all would be greatly appreciated. [12:13] <BtbN> Is ffmpeg capable of recovering a broken mp4? (Live recording, app crashed before writing the moov atom) [12:29] <Mavrik> BtbN, I doubt anything is really capable of recovering an MP4 without an index. [12:56] <synthecypher> Is there a way to skip files that haven't changed? [13:07] <BtbN> Mavrik, there are tools that can do it, but i'd like to do it with ffmpeg [13:42] <termos> I have some issues linking against my newly compiled FFmpeg: undefined reference to `lzma_stream_decoder' etc. Is this a known problem? I have libtiff installed [13:42] <BtbN> Is your ffmpeg a static library? [13:43] <Mavrik> termos, it works just like you intended [13:43] <Mavrik> you just don't understand how static linking works [13:43] <Mavrik> (you have to explicitly link objects for dependent libraries youself, LD won't do that for you) [13:49] <termos> you're right it's static, libav*.av files in my lib folder. I used the same configure arguments as before though, has something changed? [13:50] <BtbN> Nope, static libs don't have a dependency system. You have to take care of that manualy. [13:57] <termos> I'd prefer to dynamically link against FFmpeg though [13:58] <Mavrik> then link dynamically. You'll still have to properly link to all dependencies. [13:58] <Mavrik> Aren't you using CMake or any proper build system that does that for you/ [13:58] <Mavrik> _ [14:00] <BtbN> Well, with dynamic linkins the dll/so have dependencies [14:00] <termos> I'm using CMake where I specify include and lib directories, then I have a list of target_link_libraries with -lavcodec stc. [14:00] <termos> etc [14:03] <jpluijmers> Hi guys, I have been banging my head against this problem for days now and am at a complete loss. I have posted on several forums but no body seems to know a definitive answer [14:03] <jpluijmers> My command is: ffmpeg -y -i bunny-source.mp4 -i wm.png -filter_complex \"[1:v]scale=iw*0.3:-1[watermark]; \ [0:v][watermark]overlay=10:main_h-overlay_h-10[outv]" \ -map "[outv]" -map 0:a marked.mp4 [14:03] <jpluijmers> the output is: http://pastebin.com/Jn0krS7W [14:03] <jpluijmers> I am trying to get the watermark complex filter to scale percentage based on the width of the first input source. [14:03] <jpluijmers> But I cant seem to manage to get a valid (non-selfreferencing) token within the complex filter. [14:03] <jpluijmers> Any input at all would be greatly appreciated. [14:08] <termos> Okey I fixed the linking problem by install libzma-dev and adding "-llzma" to my linking, I guess this is a new dependency for FFmpeg that I didn't know about [14:09] <termos> Last time I compiled the library was about six months ago [14:14] <Mavrik> termos, well cmake should resolve all the -l's required for a dependency [16:48] <nslinux> hi [16:48] <nslinux> hello guys [16:49] <nslinux> i got this error when i try to compile the ffmpeg [16:49] <nslinux> any one can help plz [16:49] <nslinux> ERROR: libnut not found If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user(a)ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solve the problem. [17:26] <Mavrik> hrmf, on OS X VDA doesn't work... at all [17:26] <Mavrik> that's not cool [17:26] <nslinux> bad irc room [17:27] <ac_slater_> alright guys. I was here a few days ago. I have a custom encoder that gives me h264 data (I and P frames), but no timestamps. Any advice when generating PTS? [17:29] <ac_slater_> it doesnt even give me NAL boundaries sadly [17:31] <Mavrik> well, that sucks :) [17:31] <ac_slater_> yea it's worst case [17:36] <Mavrik> well [17:36] <Mavrik> Short answer: you can't. [17:37] <ac_slater_> cant? I can rig detect when a 'frame' is complete I guess. Assume I can. Assume I have a group of NALs that represent a frame. [17:38] <Mavrik> You're asking the same things you asked a few days ago. [17:38] <Mavrik> Since you can't know how the frame is ordered without the encoder telling you that. [17:39] <Mavrik> There's no way for you to extrapolate since NAL's don't have timing info. [17:40] <ac_slater_> Mavrik: right. The new info I have is, I can always get data in decode order. I can, infact, read() until EAGAIN, find the nal boundaries (in order), and get a 'frame' [17:44] <Mavrik> that still doesn't help you if you have B-frames. [17:44] <ac_slater_> I dont. I thought I did, but I only have I and P [17:45] <Mavrik> well then it's simple [17:45] <Mavrik> you have FPS [17:45] <Mavrik> and you know when you get a new frame [17:45] <Mavrik> so just increment PTS and DTS with appropriate difference [17:45] <Mavrik> everytime you get a frame [17:45] <ac_slater_> can they be incremented at the same rate? DTS and PTS that is? [18:36] <sg90> Hi, I'm trying to record a video and split it into segments every 30 seconds using the segment muxer. I am having issues forcing a new keyframe on the first frame on the new segment. Has anyone encountered this issue before? [18:44] <sleep> hi, is there a way to use the gdigrab module by using a pid, or something else unique, to find the window to record ? [18:45] <sleep> I don quite get why someone thought grabbing by title only is a good idea, tbh [18:45] <d00f_> is ffmpeg have a method to encode a low resolution high framerate video stream with a high resolution low framerate stream into a single high resolution output stream? [18:46] <d00f_> s/is/does/g [18:59] <ac_slater_> Mavrik: thanks for the help, I was able to get something working. It's kinda crazy [19:00] <Mavrik> ac_slater_, yes, they can be incremented at the same rate and probably should be the same if you have no B frames [19:00] <ac_slater_> Mavrik: that's what I thought. Thanks mate [21:07] <sybariten> evening [21:07] <sybariten> does an ffmpeg from 2012 typically not have -vf ? [21:07] <sybariten> trying to do -vf scale=iw/2:-1 [21:07] <sybariten> FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6) [21:08] <JEEB> ahaha [21:08] <JEEB> that's old, congratulations [21:08] <JEEB> like pre-fork stuff [21:08] <sybariten> ok [21:56] <d00f_> Hello - does anyone know of any resources that would help me live transcode a low resolution video stream with a high resolution stream together? The framerates between the two are different [21:57] <c_14> What do you mean, "transcode together" [21:57] <d00f_> "interleave" i think [21:57] <c_14> frame 'a' frame 'b' frame 'a' frame 'b' ? [21:58] <d00f_> yes, but the framerates are different as well [21:58] <d00f_> so it would be abbbbbbabbbbbbabbbbbbabbbbbb [21:59] <d00f_> its a high resolution camera with two feeds [22:00] <c_14> try the interleave filter, not sure if that'll work out of the box though [22:00] <ChocolateArmpits> Do they have to be different in the end? [22:02] <d00f_> no, they dont have to be different in the end. I am trying to cut down transmission size downstream [22:03] <d00f_> It might look cool too in the end [22:34] <voip_> Hello guys [22:35] <voip_> what is mean: Circular buffer overrun. To avoid, increase fifo_size URL option. To survive in such case, use overrun_nonfatal option [22:35] <voip_> ? [22:38] <rcombs> seems pretty self-explanatory to me [22:41] <voip_> rcombs, how to fix ? [22:41] <voip_> increase fifo_size URL option [22:54] <ubitux> sybariten: 0.6.x is probably the worst release ever made [22:54] <ubitux> why the hell would you use that? [22:57] <JEEB> > red hat 4.4.5 [22:57] <JEEB> seems like either centos or rhel [22:57] <JEEB> aka fucking awesomely old shit [22:57] <JEEB> (thankfully compilation isn't too hard in general) [00:00] --- Tue Feb 17 2015

1 0

ffmpeg-devel.log.20150215
by burek 16 Feb '15

16 Feb '15

[00:40] <aetasx> does the movie filter skip some of the format or stream testing that would otherwise happen if specified using -i ? [00:44] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:64e036980875: avutil/fixed_dsp: Use av_clip_int16() instead of the non optimized av_clip_int16_c() [00:44] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:7d60baa8d90e: avformat/matroskaenc: add allow_raw_vfw and disable it by default [01:39] <cone-657> ffmpeg.git 03Zhaoxiu Zeng 07master:2a72b1664d7b: avcodec/h264: use avpriv_find_start_code() in h264_split() [04:44] <kierank> lol --disable-yasm in broadcast lgpl violation [04:54] <cone-61> ffmpeg.git 03Zhaoxiu Zeng 07fatal: ambiguous argument 'refs/tags/n2.4.7': unknown revision or path not in the working tree. [04:54] <cone-61> Use '--' to separate paths from revisions [04:54] <cone-61> refs/tags/n2.4.7:HEAD: avcodec/h264: use avpriv_find_start_code() in h264_split() [12:03] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:1567ec43a3bd: avfilter/vsrc_mandelbrot: fix indention of mis-indented block [12:03] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:8123e19e61d7: avcodec/dcadec: Move remaining tables to dcadata.h [12:54] <cone-859> ffmpeg.git 03Anshul Maheshwari 07master:bbb063458514: ffmpeg_opt: Creating new data stream for output format context [14:07] <Muthu> hello Guys. [14:08] <Muthu> I need help, can someone help me to segmentation for iOS with m3u8 list [14:08] <Muthu> please help me [15:03] <cone-859> ffmpeg.git 03Thomas Volkert 07master:e4a6486c17dd: rtpdec: experimental VP9 depacketizer (draft 0) [15:55] <anshul_mahe> where to upload videos files, so that they get downloaded when we do make fate-resync [15:57] <compn> to fateserver [15:57] <compn> actually i dont know :) [15:59] <anshul_mahe> I am writing a test case, for that the video must be there on fate server, but I dont know how to put it and where/ [15:59] <anshul_mahe> I am writing a test case, for that the video must be there on fate server, but I dont know how to put it and where? [16:00] <anshul_mahe> Do we have to send mail to ffmeg-devel and tell the path, so that one who is previliged put it on server [16:16] <michaelni> anshul_mahe, yes, send mail with link and information on in which directory it needs to be put [16:52] <kierank> anshul_: you never explain how you test the mux is correct [17:10] <anshul_mahe> hello, I am trying to run fate again, its not working here is error message http://pastebin.com/fYjGUQm2 [17:11] <anshul_mahe> I did make fate-rsync and I am following https://ffmpeg.org/fate.html [17:11] <anshul_mahe> make fate is not working [17:13] <michaelni> make distclean and rerun configure [17:16] <anshul_mahe> ok [17:29] <cone-859> ffmpeg.git 03zhaoxiu.zeng 07master:38c619c1ed22: avcodec/vc1: cleanup and simplification [17:30] <durandal_1707> 3 filters left to 200th filter [17:33] <ubitux> i'm working on one currently [17:33] <ubitux> i hope to get it done in maybe 2 or 3 weeks [17:34] <durandal_1707> is it still secret what it does? [17:35] <ubitux> not really, i'm just implementing nlmeans [17:42] <ubitux> if someone is looking for filters to implement [17:42] <ubitux> http://hajo.me/blog/2014/12/28/how-surround-sound-for-headphones-works/ [17:42] <ubitux> this one might be interesting [17:43] <ubitux> also a generic FIR filter can be written relatively easily [18:59] <anshul_mahe> I was making a test case for closed caption, In output I get \N written instead of actual newline, here are my log http://pastebin.com/XDZGBckQ [19:02] <anshul_mahe> While developing closed caption it was required to \N instead of \n for ass subtitle, otherwise it wont work [19:02] <anshul_mahe> is that output that I am getting is fine in logs pasted above, or I have to modify it to display actual \n instead of \N [19:02] <cone-859> ffmpeg.git 03James Almer 07master:f7ed997a6ddd: x86/swr: make pack_8ch functions work with compilers without aligned stack [19:11] <anshul_mahe_> pasting that refrence log again http://pastebin.com/XDZGBckQ, [19:13] <kierank> anshul_mahe_: how are you testing the mpegts is scte35 compliant? [19:13] <anshul_mahe_> I have only one mpegts file given on that ticket [19:14] <anshul_mahe_> so I just belive it to be complient [19:14] <anshul_mahe_> *compliant [19:14] <kierank> the mpegts ffmpeg outputs... [19:15] <anshul_mahe_> In my implementation I do not give scte output in mpegts, I have not yet implemented that [19:16] <kierank> so what does ffmpeg do with the scte35 data? [19:17] <anshul_mahe_> In actual work that I have compleve providedted I change the name of segment in playlist file of hls, to what user ha [19:17] <anshul_mahe_> In actual work that I have complted I change the name of segment in playlist file of hls, to what user have provided [19:17] <anshul_mahe_> so it plays local adsegment instead of actual ad segment [19:18] <cone-859> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:8129ccec3a6a: ffserver: reorder HTTP fwd decls [19:18] <cone-859> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:50c1cac4324f: ffserver: break too-long lines [19:18] <cone-859> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:bd8e16f2e6f0: ffserver: decouple assignment and check [19:18] <cone-859> ffmpeg.git 03Reynaldo H. Verdejo Pinochet 07master:df4346c541ca: ffserver: fix wrong indentation [19:19] <anshul_mahe_> I dont have any fund to buy any proprietry tsreader or device which will tell that my scte is complient or not [19:23] <kierank> you should ask on the mailing list then [19:24] <anshul_mahe_> Do you have any suggestion on this situation [19:36] <cone-859> ffmpeg.git 03Thomas Volkert 07master:cbe508553ad1: avformat/rtpdec: DV depacketizer (RFC 6469) [19:42] <anshul_mahe_> kierank: i m going to bed, send me private mail if u have anything to say, bye [19:59] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:543e329a2670: avformat/version: Bump version for "avformat/rtpdec: DV depacketizer (RFC 6469)" [20:14] <cone-859> ffmpeg.git 03Anton Khirnov 07master:d3449ac93833: changelog: mention DXVA/HEVC and AAC ELD480 [20:15] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:8242946af20f: Merge commit 'd3449ac938333d59955360b6cb8e7bc24a9b792e' [20:34] <cone-859> ffmpeg.git 03Anton Khirnov 07master:e21d85309943: FATE: add a test for the SVQ1 header byte swapping [20:34] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:fe03456cdcc8: Merge commit 'e21d85309943a51b7808f5e01dd258b262e09148' [20:49] <cone-859> ffmpeg.git 03Luca Barbato 07master:c23ccaf98827: xcbgrab: Check for xcb-shape [20:49] <cone-859> ffmpeg.git 03Luca Barbato 07master:d40815a982c3: xcbgrab: Correctly make shm optional [20:49] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:7d56aab2de77: Merge commit 'c23ccaf98827cffa651cec3bfa5e57457dfc9dff' [20:49] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:3171ac209b04: Merge commit 'd40815a982c36ff9da09ae3edb5e3bd7d97b5d42' [20:57] <cone-859> ffmpeg.git 03Luca Barbato 07master:b8df0b71c525: rtp: h264: Move profile_level_id parsing to a function [20:57] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:3773b6edea6f: Merge commit 'b8df0b71c525e9fc9fbee790d093bae3aa62035c' [21:03] <cone-859> ffmpeg.git 03Luca Barbato 07master:a9a0b8d6c14e: rtp: h264: Move parse_sprop_parameter_sets parsing to a function [21:03] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:8f00d7f4b401: Merge commit 'a9a0b8d6c14ece1b4698c6ede9227aca980f6c5b' [21:09] <cone-859> ffmpeg.git 03Luca Barbato 07master:f0a87479960c: rtp: h264: Move STAP-A NAL parsing to a function [21:09] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:2d2c744f8faa: Merge commit 'f0a87479960ce000f23f2beaf474707797b4b0d0' [21:20] <cone-859> ffmpeg.git 03Luca Barbato 07master:3b33dfb23c50: rtp: h264: Move FU-A NAL parsing to a function [21:20] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:a848cf63dcc1: Merge commit '3b33dfb23c5022a0cb2a7b13878e4a3492a996c7' [21:28] <cone-859> ffmpeg.git 03Luca Barbato 07master:53367b34e115: rtp: h264: Drop the asserts [21:28] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:f3b6a8f554e2: Merge commit '53367b34e1156614e82ef7af888928f322566f88' [21:41] <cone-859> ffmpeg.git 03Federico Tomassetti 07master:b4cda0a999fa: ogg: check memory allocations [21:42] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:fa321cc0c8ed: Merge commit 'b4cda0a999facbddacb95861a67ff61074edc39b' [21:50] <cone-859> ffmpeg.git 03Luca Barbato 07master:e1a57cbb1c27: ape: Use the proper variable type [21:50] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:2870b86c2a9f: Merge commit 'e1a57cbb1c2752feb9315f748836dc44e58d2dc6' [22:03] <cone-859> ffmpeg.git 03Daniel Moran 07master:51ca3cb604a7: xcbgrab: Use the correct geometry for the region highlight [22:03] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:7b564a4cbb44: Merge commit '51ca3cb604a7585a7cff35d4b954794508955c19' [00:00] --- Mon Feb 16 2015

1 0

ffmpeg.log.20150215
by burek 16 Feb '15

16 Feb '15

[01:43] <zetaPRIME> Can CLI ffmpeg be fed a set of PNG frames directly through stdin without writing them to disk first? [01:45] <zetaPRIME> I'm trying to make a C#+monogame application that can encode rendered video in a way that works crossplatform and doesn't force my application to be under a particular license [01:48] <zetaPRIME> and as far as I can tell there are no libraries that can encode video that aren't full GPL themselves [01:50] <JEEB> well if you are fine with what libavcodec itself supports there's a relatively wide range of stuff that's under LGPL [01:50] <JEEB> it's mostly third party libs that are GPL, like libx264 [01:50] <zetaPRIME> so the best I've come up with is creating a png frameset and mixing the audio track with ffmpeg-sharp or something, then piping both into CLI [01:51] <zetaPRIME> I was thinking openh264 since it's mentioned on the supported codec list [01:51] <JEEB> uhh [01:51] <JEEB> I really wouldn't go that way [01:51] <JEEB> I mean, it depends on what exactly you want [01:51] <JEEB> because there's two use cases I can think of [01:51] <JEEB> one is lossless capture of the game [01:52] <JEEB> another is capture for live streaming [01:52] <JEEB> for number one there's plenty of alternatives within libavcodec [01:52] <zetaPRIME> it's not actually a game [01:52] <zetaPRIME> it's a sprite-based animation suite [01:52] <JEEB> so you most probably want to output the stuff lossless from the suite [01:53] <JEEB> in that case there's plenty of stuff in base libavcodec [01:53] <JEEB> ut video being one of those that has input plugins for most video editors for most OSs [01:53] <zetaPRIME> ut? [01:53] <JEEB> initials of the original developer [01:53] <JEEB> I made the LGPL implementation [01:53] <JEEB> in libavcodec [01:54] <JEEB> original was GPL [01:54] <zetaPRIME> well, codec use can be changed later if I'm going to be using ffmpeg [01:55] <JEEB> anyways, I think C interop should be relatively simple from C# [01:55] <zetaPRIME> the problem is giving it video, since ffmpeg-sharp doesn't do video encoding at all and it was last active in january 2012 [01:55] <JEEB> so wrapping lavf/lavc [01:55] <JEEB> should be quite simple [01:55] <JEEB> there are C examples on how to use the APIs in the examples dir [01:55] <zetaPRIME> I don't know how to do managed-to-native interaction at all [01:55] <JEEB> this is the time to learn then [01:56] <JEEB> it should be relatively simple, or at least I'd be surprised if it was hard :P [01:56] <zetaPRIME> well... [01:56] <JEEB> it gets less simple with C++ where there are different ABIs etc [01:56] <JEEB> while in C it should just work, more or less [01:56] <zetaPRIME> I can generate a set of png-format frames without issue [01:56] <zetaPRIME> the problem is just getting them into ffmpeg [01:57] <zetaPRIME> I could write to disk then give it the fileset, but I'd rather not depend that much on disk I/O [01:58] <JEEB> http://blogs.microsoft.co.il/sasha/2012/01/24/managed-unmanaged-interoperab… [01:58] <zetaPRIME> thus the initial question; can it be fed png data directly through stdin by any means? [01:58] <JEEB> I know you can feed ffmpeg stuff through stdin, but I'm definitely not sure about multiple pngs :P [01:58] <JEEB> http://stackoverflow.com/questions/9521916/wrapping-c-for-use-in-c-sharp/95… basically this series could be quite useful I think? [01:59] <JEEB> I've not done dotNET development too much since 2003 or so [01:59] <zetaPRIME> and pinvoke conceptually seems like an absolute minefield for crossplatform use [01:59] <zetaPRIME> like if I didn't care about crossplatform I could just use directshow and NAudio and call it a day, but I don't know if that would even work in wine+mono [01:59] <JEEB> well I'm just saying that it shouldn't be too hard and there should definitely be tools for doing it with mono or whatever [01:59] <m4t> zetaPRIME: if you're on linux and can't do stdin, try a fifo [02:00] <m4t> eg mkfifo myfifo [02:00] <JEEB> you should be the one better versed in this shit, not me [02:00] <m4t> ./pngprog > myfifo [02:00] <m4t> ffmpeg -i myfifo [02:00] <JEEB> http://www.mono-project.com/docs/advanced/pinvoke/ [02:00] <zetaPRIME> uhhhh [02:00] <JEEB> mono has this too it seems [02:01] <JEEB> it seems like the docs @ mono-project.com are pretty good :P [02:02] <zetaPRIME> m4t: ...not what I'm asking? I just need to know whether CLI ffmpeg can take any of the inputs that I'm capable of producing directly via stdin =/ [02:02] <JEEB> it can take shit in from stdin, yes [02:02] <JEEB> if it can take in the data you require from there, you would have to test [02:02] <JEEB> write a few pngs down and cat them to ffmpeg :P [02:03] <JEEB> see if the result is according to what you wanted [02:03] <JEEB> ffmpeg -i - -f png -c:v utvideo out.avi [02:03] <JEEB> for example [02:03] <JEEB> there was a multiple picture format as well, but I do not remember that without going to the documentation at 3am [02:04] <JEEB> also ugh [02:04] <JEEB> -f before -i [02:04] <JEEB> because it's for input, not output [02:04] <JEEB> my bad [02:07] <zetaPRIME> ...problem there is that I don't actually have a nonwindows system to test on [02:07] <zetaPRIME> http://stackoverflow.com/questions/13294919/can-you-stream-images-to-ffmpeg… though it looks like this is doing almost exactly what I want to do [02:08] <JEEB> well I'm pretty damn sure you can cat things on windows as well [02:08] <JEEB> if with nothing else then a cygwin or mingw cat binary will do it [02:08] <JEEB> http://superuser.com/a/434876 [02:08] <JEEB> there, I googled it for you [02:10] <JEEB> and yeah, image2pipe seemingly is the multiple pictures format [02:10] <JEEB> you will have to set -c png or so for png, though, I guess? [02:11] <JEEB> also you definitely don't want to use -c:v mpeg4, but instead -c:v utvideo or whatever other lossless format you like [02:11] <zetaPRIME> hmm. well, actually trying to implement this is going to have to wait until there's something actually rendering, though that shouldn't take too long once I have my most basic design elements laid out in code [02:11] <JEEB> or heck, if you don't distribute the ffmpeg binary yourself you might even get libx264 and friends :P [02:11] <JEEB> (or you can distribute an LGPL one and then the user can switch to a GPL one if he/she wants) [02:12] <zetaPRIME> or users can just replace the binary on their own--yeah [02:12] <JEEB> you just parse -codecs:v [02:12] <JEEB> and enable formats that you know are good enough and available [02:12] <zetaPRIME> ...also, I'm guessing stdio in c# is accessible as a Stream, in which case I can just use rendertarget2d.tostream directly~ [02:12] <JEEB> except I guess :v doesn't work with -codecs [02:13] <JEEB> oh well, I always try that and end up getting the full listing :P [02:13] <JEEB> because it just makes sense to be able to limit the output to specific types like you can do with -c [02:14] <zetaPRIME> ...now I just need to figure out how to do audio mixing in a way that ffmpeg-sharp can work with [02:14] <JEEB> oh right, audio :P [02:14] <JEEB> unless you want to do NUT muxing yourself [02:14] <JEEB> and feeding that from stdin [02:15] <JEEB> you either do separate file output, or you start poking the APIs in that case :P [02:15] <zetaPRIME> can ffmpeg just take an existing video container and insert an audio track after the fact? [02:15] <JEEB> yes [02:15] <JEEB> ffmpeg -i input1 -i input2 -map 0:v -map 1:a out.file [02:15] <JEEB> maps all video streams from input nr0 and all audio streams from input nr1 [02:15] <JEEB> and outputs them as out.file [02:16] <JEEB> of course you really don't want to do it like this without specifying output video/audio format :P [02:16] <JEEB> because by default you re-encode to whatever is the FFmpeg default [02:16] <JEEB> which is generally not what you want [02:16] <JEEB> -c copy should do stream copy for all tracsk [02:16] <JEEB> *tracks [02:17] <zetaPRIME> separate audio is perfectly fine for disk-first if I just ffmpeg-sharp it to an mp3 or ogg or what havest thou [02:17] <JEEB> you will want raw pcm or something lossless if output is a lossless intermediate [02:18] <JEEB> or aac or vorbis or opus if you want non-lossless [02:18] <zetaPRIME> audio being lossless is significantly less important than video, I'd say [02:19] <JEEB> yes, but I mean that if you want to do a lossless intermediate then both should be lossless [02:19] <JEEB> which is what you generally want when moving the thing to a video editor or whatever [02:20] <zetaPRIME> well, mp3 320-44k should be "good enough" in my experience [02:22] <zetaPRIME> though I suppose I could stdin a second time for the "ziptogether" [02:23] <JEEB> lossless audio is such a small thing after lossless video so I see no reason to not do it for the intermediate output mode or whatever [02:23] <JEEB> if someone wants to do a "final encode" of the thing right there, right now, then there the thing can be coded lossy [02:27] <zetaPRIME> yeah, I just don't want to have to write lossless twice in a row [02:28] <JEEB> well, you generally want two modes like that :P [02:29] <JEEB> one for an intermediate (for editing/input for multiple encodes), and one for "this is all I want" [02:31] <zetaPRIME> the final-output mode definitely has to be mp4 h264, because youtube [02:33] <JEEB> no [02:33] <JEEB> youtube actually takes pretty much everything and /always/ re-encodes [02:33] <JEEB> so in theory you want to input to it the best thing you can :P and while for bandwidth-limited people it will be exactly what you say, that's not the optimum thing [02:36] <zetaPRIME> well, mp4 h264 is definitely the least prone to youtube's occasional gray-video-syndrome from what I've heard [02:42] <JEEB> it's true that youtube doesn't often update it's FFmpeg, but I still contain that most stuff you can throw from current FFmpeg to it will work [02:42] <JEEB> ugh, early morning without sleep and it shows... "doesn't often update its FFmpeg" [02:42] <JEEB> also libx264 can do lossless as well so not like it's the worst thing to pick :P [07:50] <zetaPRIME> so it looks like I'll have to do all my actual pcm mixing in my own code on the CPU, and in order to actually play this live I'll have to render a buffer ahead and make very thorough abuse of monogame's creation of soundeffect objects from a RIFF-formated datastream [07:55] <JamJams> Hey is R3D supported in ffmpeg? I see it in the list however I get errors when I try to import an R3D file from my Camera. [14:02] <Muthu> hello [14:02] <Muthu> can anyone help to do segmenation for iphone using ffmpeg [14:04] <Muthu> please help me [16:53] <rsully> to use ffmpeg with any decent options (libfdk, etc) on ubuntu do I have to compile it myself? [16:55] <JEEBsv> yes [16:55] <JEEBsv> ffmpeg with fdk-aac linked is not distributable [16:56] <JEEBsv> same goes for faac and aacplus libraries [16:56] <JEEBsv> since all of them are license-wise incompatible with (l)gpl [16:57] <rsully> yep :/ [16:57] <rsully> is there a script that walks you through compilling and getting dependencies? or should i just follow the wiki guide [17:19] <rsully> is ffmpeg-snapshot master or the current stable release? [17:35] <__jack__> rsully: building ffmpeg is (really) easy [18:01] <wintellect_> What's the best way to get ffmpeg into LinuxMint? [19:00] <blez> how can I stream some video to html5 friendly webm? [19:32] <shevy> is there a long option for -y available? overwrite output files [19:32] <shevy> or perhaps put differently, how do I find out both short and long names to commandline arguments? [19:46] <__jack__> does not exist, said the man [19:47] <JEEB> if there was it'd be --yes :P [19:47] <JEEB> because that's what -y means [20:32] <shevy> lol [20:32] <shevy> ffmpeg -i foo.avi --yes --no [21:11] <Thaconut> Hi! I'm trying to use this script (https://gist.github.com/Taconut/f719bc7ac2b84fa335c7) to compile FFMPEG programmatically, but I've run across a problem: its taking up way too much space! Since I'm only using a small part of FFMPEG, most of the content there isn't required. How can I tweak this script to only include the following functionality?: [21:11] <Thaconut> Converting audio/webm and audio/mp4 to audio/mp3 [21:11] <Thaconut> Getting the length in seconds of a song with ffprobe [21:12] <Thaconut> Injecting ID3 tags into mp3s [21:12] <shevy> hehe [21:12] <shevy> quite a long list [21:12] <Thaconut> Yeah [21:13] <Thaconut> But FFMPEG is like a swiss army knife, and I only need the scissors :P [21:13] <Thaconut> Also I meant audio/mpeg not audio/mp3 [21:13] <Mavrik> well select only what you need in configure then. [21:14] <Mavrik> maybe don't use a copy-pasted script you don't understand ? [21:14] <shevy> Thaconut, when you do: "ffmpeg -i name_of_song.mp3" you get a long text result [21:14] <BtbN> that script doesn't even set where it wants to put stuff [21:14] <shevy> one line has this entry: "Duration: 00:07:04.93, start: 0.025057, bitrate: 128 kb/s" [21:14] <Thaconut> I know [21:14] <Thaconut> I'm using ffmpeg already [21:15] <shevy> the Duration part tells you how many seconds that is. Now you can use grep or a programming language to scan for these :::: entries there [21:15] <Thaconut> But my host doesn't allow me to just install it [21:15] <Thaconut> I already did that [21:15] <Thaconut> Regex worked fine [21:15] <Thaconut> But I need to add that functionality when `make`ing ffmpeg [21:15] <shevy> you want to modify the source of ffmpeg? [21:15] <Thaconut> No [21:16] <Thaconut> I just want to compile it with limited functionality in hopes that I can save space [21:16] <Mavrik> btw, for stream info, use ffprobe with JSON or other structured output [21:16] <Mavrik> don't parse human readable strings -_- [21:16] <Thaconut> Too late :/ [21:17] <Thaconut> But I will change that if I can get FFMPEG working on my production server. [21:17] <Thaconut> Thanks [21:17] <Thaconut> It seems as though configure allows me to enable/disable certain functionality [21:18] <Thaconut> But I'm not sure what I need to enable to get these features [21:18] <shevy> the power of GNU configure [21:18] <shevy> how do you know that "these features" are available [21:18] <Thaconut> Because I'm using them with a complete installation of FFMPEG on my development server [21:18] <Thaconut> and it works beautifully [21:19] <shevy> if you just do "ffmpeg" then you will see the compiled options [21:19] <shevy> configuration: --prefix=/usr --enable-gpl --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libxvid --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-version3 --enable-x11grab --extra-libs='"-ldl"' --disable-libopenjpeg [21:19] <shevy> oops sorry, that was too much, my bad [21:20] <Thaconut> So I'm guessing I need libmp3lame [21:20] <Mavrik> Thaconut, use "./configure --help" for all options, "ffmpeg -codecs" to see list of codecs, "ffmpeg -formats" to see list of formats [21:20] <Mavrik> and use those :) [21:20] <Thaconut> Cool! Thanks! [21:20] <Thaconut> That makes things easier [21:20] <shevy> libmp3lame is lame aka http://sourceforge.net/projects/lame/files/lame/3.99/lame-3.99.5.tar.gz [21:20] <Harzilein> shevy: if you mean that the line was truncated, it appeared up to "--disable-libopenjpeg", so i guess if that's not too much and you don't regularly paste 200 column lines, most support channels should be fine :) [21:21] <ubitux> Thaconut: eval $(ffprobe -v error -of flat=s=_ -show_entries format=duration in.mp3); echo $format_duration [21:21] <shevy> I mean I think it was too much, or at least on my IRC client, it appears as if I pasted like 5 lines in a row kind of :\ [21:21] <Thaconut> It looks nice and tidy on mine [21:21] <shevy> I have bad eyes so I use huge fonts! [21:21] <Harzilein> :D [21:22] <Thaconut> Can I convert to an mp3 without lame? [21:23] <shevy> hmm [21:23] <relaxed> no [21:23] <Thaconut> And I'm assuming that means I'm going to have to compile lame as well? [21:23] <relaxed> Thaconut: did you see http://johnvansickle.com/ffmpeg/ ? [21:24] <Thaconut> Wow! [21:24] <Thaconut> You just made my life infinitely easier! [21:24] <Thaconut> That's awesome! Thanks! [21:24] <relaxed> you're welcome [22:05] <ahop> Hi! Can ffmpeg convert .3gp to .mp3 ? [22:05] <ahop> Or do I need another tool like lame , etc. ? [22:05] <c_14> If ffmpeg can read the 3gp file, and has lame support built in, yes. [22:05] <ahop> Is there an easy CLI command to convert .3gp to .mp3 ? [22:06] <c_14> https://trac.ffmpeg.org/wiki/Encode/MP3 [22:06] <BtbN> Isn't 3gp a video format? [22:06] <c_14> look at the VBR encoding example [22:06] <c_14> 3gp is just a container iirc [22:06] <ahop> BtbN in fact I have .3ga but similar to 3gp [22:08] <ahop> I tried something like : ffmpeg -i input.wav -codec:a libmp3lame -qscale:a 2 output.mp3 [22:08] <ahop> but "Unrecognized option 'codec:a'" [22:08] <Mavrik> you're using ancient ffmpeg. [22:08] <Mavrik> if at all. [22:09] <c_14> Try with a static build [22:09] <c_14> http://johnvansickle.com/ffmpeg/ [22:09] <ahop> is there a single .exe file version ? [22:11] <ahop> if I want to have a single file version I should take this one ? : http://ffmpeg.zeranoe.com/builds/ **Static** ? [22:11] <c_14> ye [22:12] <ahop> ok [22:16] <ahop> ffmpeg -i *.wav -codec:a libmp3lame -qscale:a 2 (inputname).mp3 [22:16] <ahop> How can I do that ^^ (ie convert all .wav files in the folder) [22:18] <ahop> is it possible to do it with ffmpeg itself, or do I need to use a windows batch (i hate batch!) [22:20] <c_14> Just use a batch file (even if you hate it) or a scripting language of some sort. [22:21] <ahop> ok :) [22:23] <ahop> it works! thanks a lot [23:06] <acosonic> Hi everyone, does anyone know is there such player like splayer and eyecareplayer which filter brightness while playing videos? [23:11] <pzich> acosonic: I'm not sure exactly what you mean, but you could certainly apply a filter to adjust the brightness/contrast, or apply curves or levels [00:00] --- Mon Feb 16 2015

1 0

ffmpeg-devel.log.20150214
by burek 15 Feb '15

15 Feb '15

[00:42] <cone-366> ffmpeg.git 03zhaoxiu.zeng 07master:b11a187575f6: avcodec/golomb: simplify sign conversion [01:10] <cone-366> ffmpeg.git 03zhaoxiu.zeng 07master:ac7fc444eed5: avcodec/apedec: simplify sign conversion [06:23] <cone-237> ffmpeg.git 03James Almer 07release/2.4:a78b7c504a7b: x86/swr: add missing alignment check to pack_6ch functions [06:23] <cone-237> ffmpeg.git 03James Almer 07release/2.4:3c6350379214: avutil/opencl: don't include config.h [10:19] <ubitux> damn, lanczos is actually really great for downscaling [10:27] <ubitux> at least compared to bilinear [13:03] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:ad1549aec3ea: avformat/mov: print a warning if parsing udta failed [13:03] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:e2fb12b629e8: avformat/matroskaenc: fix handling of VFW style raw rgb [13:04] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:9ccc4eedd181: avformat/matroskaenc: Do not use native mode for raw RGB [13:28] <wm4> wait what, matroskaenc outputs vfw style shit? that'S fucking evil [13:29] <nevcairiel> for some codecs thats the only valid mode for matroska [13:30] <nevcairiel> ..and as long as it doesnt use bframes also not really a problem [13:30] <nevcairiel> it just becomes a problem when someone uses h264 in avi with all its evilness and muxes it as vfw in mkv :p [13:48] <cone-657> ffmpeg.git 03Paul B Mahol 07master:ba22295e76f0: lavc: deprecate VIMA decoder [14:18] <Mavrik> hmm [14:18] <Mavrik> [mpegts @ 0x2a27980] Invalid timestamps stream=1, pts=9399, dts=8589926186, size=1625 [14:18] <Mavrik> This looks like a bug in 2.5.3 - PTS/DTS rollover shouldn't count as invalid timestamp right? [14:20] <Mavrik> hrm, it looks like it's just a warning. [14:23] <rcombs> the RasPi2 plays 10-bit H.264!? [14:26] <wm4> ...not [14:33] <rcombs> rip [14:33] <thardin> the new ones does have a lot more horsepower [14:33] <rcombs> rcombs just didn't know what 8-bit H.264 decoders do when given 10-bit input [14:39] <BtbN> depending on the resolution, the CPU might be able to handle it [14:39] <BtbN> but i'm quite sure the hardware decoder doesn't [15:06] <cone-657> ffmpeg.git 03Zhaoxiu Zeng 07master:3b5ad8fbf77a: avcodec/parser: optimize ff_mpeg4video_split() [16:29] <wm4> avcodec_flush_buffers() behavior changed recently [16:29] <wm4> now the only way to make the decoder unref _all_ frames is closing the codec [16:29] <wm4> inconvenient [16:42] <durandal_1707> when exactly it changed? [16:45] <nevcairiel> wm4: that would be a bug [16:46] <nevcairiel> i dont think i noticed that particular problem myself [16:48] <wm4> nevcairiel: well, I already fixed my code; it was actually pretty simple [16:48] <wm4> but surely it could be considered a lavc bug [16:48] <wm4> it makes no sense to keep any state after flushing [16:48] <nevcairiel> closing the decoder can have other side effects [16:48] <wm4> I just know this: [16:48] <wm4> ==1802== by 0x4D60822: ff_h264_unref_picture (h264_picture.c:55) [16:48] <wm4> ==1802== by 0x4BB43F2: h264_decode_end (h264.c:1920) [16:48] <wm4> ==1802== by 0x4BD3B83: avcodec_close (utils.c:2841) [16:48] <nevcairiel> like, for example it forgets if the encoder was x264 for some workarounds [16:49] <wm4> (this is called right after a flush call) [16:49] <nevcairiel> I require a flush to free all surfaces when handling a resolution change in hwaccel [16:49] <nevcairiel> otherwise bad things start to happen [16:51] <wm4> good point [16:51] <wm4> normally, resolution changes should flush the decoder [16:51] <nevcairiel> maybe it works in that case because of that [16:54] <wm4> nevcairiel: do you have a small sample so I can test? [16:55] <nevcairiel> fate has one [16:55] <nevcairiel> h264/reinit-large_420_8-to-small_420_8.h264 [16:56] <wm4> ok, let's see [16:57] <wm4> nevcairiel: works fine with my code [17:01] <nevcairiel> my ffmpeg is a month old or so, but i see no problems there at least [17:01] <wm4> mine is from 4 days ago [17:02] <nevcairiel> i put a warning in my code a long time ago that checks if all hw surfaces have been free'ed after a flush on seek, i'll at least notice when it breaks [17:15] <cone-657> ffmpeg.git 03zhaoxiu.zeng 07master:a196e0c66dd4: avcodec/vc1_pred: few branchless optimizations [17:21] <wm4> ubitux: have you seen the ffplay subtitle patch yet? [17:24] <durandal_1707> wm4: why so many '?' [17:26] <wm4> each one asks "why" [17:27] <wm4> maybe my reaction was a bit exaggerated, but it's definitely an absurd patch [17:27] <Daemon404> the word youre looking for is inbred [17:47] <kierank> wtf [17:52] <ubitux> wm4: yes, i will object to it obviously [17:52] <ubitux> i was pretty sure a hate army would rise anyway before i do [17:55] <Daemon404> "based on a complaint from wm4" [17:55] <Daemon404> very nice. [17:57] <ubitux> don't we have a violation flag or something? [18:28] <Daemon404> ubitux, for trac? [18:28] <Daemon404> i thought os [18:28] <Daemon404> so [18:28] <ubitux> no sorry, i was thinking about the matroska thing [18:29] <ubitux> trac has a category on its own [18:29] <Daemon404> o [18:58] <cone-657> ffmpeg.git 03Christophe Gisquet 07master:b533949813cc: x86: hevc: remove a parameter to WP internals [19:25] <funman> fosdem video team needs some help to create composited videos [19:25] <funman> their FFmpeg script is not working as they'd want it to [19:25] <funman> they want to generate something like http://video-post.fosdem.org/janson_what_is_wrong_with_operating_systems.mp4 [19:27] <funman> Thulinma: hello Jaron, I will let you explain in details what's not working :) [19:28] <Thulinma> funman: Sure thing. Did you already explain the situation? [19:28] <funman> 19:25 < funman> fosdem video team needs some help to create composited videos [19:28] <funman> 19:25 < funman> their FFmpeg script is not working as they'd want it to [19:29] <funman> 19:25 < funman> they want to generate something like http://video-post.fosdem.org/janson_what_is_wrong_with_operating_systems.mp4 [19:29] <kierank> hi jaron [19:29] <Thulinma> kierank: Hey, kieran, how are you? [19:30] <ubitux> funman: sounds like basic overlay setup [19:30] <kierank> Thulinma: good thanks, you? [19:30] <Thulinma> This is the script we're using to generate the overlays: http://pastebin.com/PTKvfkAN [19:30] <Thulinma> kierank: Pretty good. I was sick last week, but all okay again now :) [19:30] <funman> ubitux: looks simple enough, i'd be interested to see how it can be done. AFAIU VLC can't do anything like that [19:31] <ubitux> Thulinma: what doesn't work? [19:31] <funman> nice filter setup [19:31] <Thulinma> The video is generated just fine, but there's an issue where the framesync buffer overflows and it drops frames from one or both of the overlayed videos. [19:31] <ubitux> also, i'd suggest to use -filter_complex_script at this point [19:32] <Thulinma> The filter setup was created by somebody from the core fosdem team - but I tweaked it a little. Filters isn't my thing, heh :) [19:32] <ubitux> can you pastebin the complete ffmpeg output? [19:33] <funman> it's -filter_complex not -filter_complex_script, is there a difference? [19:33] <Thulinma> Sure. One moment. It's alright if I interrupt it a few minutes in, right? Running an entire talk through this takes like an hour. [19:33] <ubitux> why -re? [19:33] <Thulinma> Oh, was an attempt to fix the overflow issue. it didn't make a difference. [19:33] <ubitux> yeah sure, just press 'q' [19:33] <funman> ah filter_complex_script reads from a file [19:36] <Thulinma> http://pastebin.com/eancZGWu <-- Here's the complete output. [19:36] <ubitux> why -r 50 btw? [19:37] <Thulinma> Also an attempt to fix the problem. [19:37] <Thulinma> It also didn't make any difference. [19:38] <Thulinma> (several people spend a few hours trying to get this to work right - there's some leftover attempts in the script, sorry about that) [19:38] <ubitux> mmh, it might be caching too much at the same time [19:38] <wm4> lol [19:38] <ubitux> reminds me of something [19:38] <wm4> also this belongs to #ffmpeg (I just want to be an ass) [19:39] <ubitux> Thulinma: make sure you try latest btw, 2.3 is starting to get old [19:39] <Thulinma> I could do that - I'm not doing this on my own system. I'm bound to the system that has all the video material stored. It's 1 TB, so it's not easily moved. [19:41] <ubitux> Thulinma: can you try to simplify the graph until the issue is not reproducible? [19:41] <ubitux> like, is it reproducible if you drop all the audio layer, etc [19:41] <ubitux> all the text stuff, .. [19:41] <Thulinma> Sure let me try that.... just a few minutes.... [19:45] <ubitux> also yeah, please drop all the -framerate, -r and -re [19:45] <ubitux> they look quite wrong [19:47] <ubitux> anyway, back to interesting stuff, i just made my colorquant go from 8.431s 3.394s by adding a stupid cache [19:47] <ubitux> so i'm going to push that stuff in a few hours [19:48] <Thulinma> Ok, dropping the audio, text and post/pre rolls - problem disappears. [19:48] <Thulinma> I'll see what makes it come back.... [19:48] <ubitux> just try to create the simplest reproducible case [19:49] <Thulinma> Working on it :) [19:49] <ubitux> Thulinma: you possibly want to try to simulate your sources with -f lavfi -i testsrc or something like that btw [19:49] <ubitux> so you can open a ticket without having to provide samples [19:50] <Thulinma> Good idea. First let me see what causes it in the first place... [19:50] <ubitux> :) [19:50] <ubitux> but this reminds me of a known issue though [19:51] <ubitux> some frames are probably stuck into the graph or something [19:51] <ubitux> i mean, it's a deja vu, so a real test case is welcome [19:52] <wm4> ubitux: so does your patch convert video to pal8 via a lavfi filter? [19:52] <ubitux> in 2 pass yes [19:53] <ubitux> first pass for the palette, then in a second pass you send that palette to another filter which is going to use it to output pal8 [19:53] <wm4> how is the data from pass 1 retrieved and handed to pass 2? [19:54] <ubitux> first pass output 1 16x16 image, which is the palette [19:55] <ubitux> then you have a filter taking 2 inputs, the palette and the input stream [19:55] <wm4> eh [19:56] <wm4> I understand the technical reasons, but that's pretty weird from a user POV [19:56] <ubitux> ffmpeg -i input.mkv -vf palettegen palette.png [19:56] <ubitux> ffmpeg -i input.mkv -i palette.png -lavfi paletteuse output.gif [19:56] <ubitux> looks like this in the simplest case [19:56] <ubitux> it gets a bit clumsy when you insert scales, trim and fps, but well. [19:56] <ubitux> it's cool to be able to see the palette, and not have any memory issue [19:59] <ubitux> Thulinma: btw, just a suggestion, since your background with text and logo etc looks static, you probably want to generate it in a first pass and re-use it for the overlay [19:59] <ubitux> it will be faster [19:59] <ubitux> (there is no point in calling 3x drawtext per frame) [19:59] <Thulinma> Yes that sounds like a good idea. How? :) [19:59] <Thulinma> I think I found the cause, by the way. [20:00] <Thulinma> The framerate of the preroll is what does it, from the looks of it. [20:00] <Thulinma> Still verifying all the combinaties to make sure..... but looks like that was it. [20:00] <ubitux> i wonder why this -framerate was there in the first place :p [20:00] <Thulinma> What would be the correct way to make that first image last 5 seconds? I'm sorry for the noob question there. [20:01] <ubitux> mmh [20:01] <Thulinma> To do that - make the single-image preroll 5 seconds long. [20:01] <ubitux> what are you trying to do exactly? [20:01] <Thulinma> (that was part of the script that I hadn't touched, heh) [20:01] <Thulinma> It's a single image with the FOSDEM logo, that needs to be shown for 5 seconds before the rest of the video starts. [20:02] <Thulinma> there's a postroll with all the sponsors, too. Same thing. [20:02] <ubitux> ooh ok mmh [20:02] <ubitux> [0:0] trim=end=5 [preroll] maybe? [20:02] <Thulinma> I'll try that. [20:03] <Thulinma> and then -loop 1, I assume? [20:04] <ubitux> yeah i guess [20:05] <Thulinma> That works :) [20:05] <Thulinma> Let me put the sound back in to verify that was indeed the problem... [20:08] <Thulinma> Ah, bah. That makes the problem come back. It's much less often now, though. [20:09] <ubitux> make sure you trim your infinite silence sources too [20:09] <ubitux> so they send EOF [20:09] <ubitux> aevalsrc=0:d=10 for instance [20:10] <ubitux> or with a trim filter [20:10] <ubitux> ah seems it's the case already, my bad [20:11] <cone-657> ffmpeg.git 03Gilles Chanteperdrix 07master:8ca098f4445c: avcocdec/mpegaudio_parser: add MP3 ADU headers parser [20:11] <cone-657> ffmpeg.git 03Gilles Chanteperdrix 07master:22470510d1f9: avformat/rtpdec_mpeg12: add robust MPEG audio depacketization (RFC 5219) [20:19] <Thulinma> Ok - it's caused by the compand filter. [20:19] <Thulinma> if I leave that out, it's perfect - if I add it back in, it breaks again. [20:21] <Thulinma> Looks like that filter wasn't set up right anyway... I'll yell at the guy that wrote that config :) [20:22] <Thulinma> Thanks for all the help! I think we got it covered from here on out. [20:22] <Thulinma> if not, I'll be back ;) [20:22] <cone-657> ffmpeg.git 03Gilles Chanteperdrix 07master:3eec775b211c: avformat/rtpdec_ac3: add AC3 RTP depacketization (RFC 4184) [20:30] <funman> Thulinma: nice [20:32] <Thulinma> funman: Yes! Just some tweaking left and we'll have the talks online! Whoohoo! [20:37] <cone-657> ffmpeg.git 03Himangi Saraogi 07master:7769be590c7a: vp56: Return meaningful error codes [20:37] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:4177f501b433: Merge commit '7769be590c7aeb2aad26ca723d105cf5203e33d2' [20:44] <funman> Thulinma: let me know so I can download what I want before the crowd arrives ^_^ [20:45] <Thulinma> We just have one minor issue to resolve before they can be put online... It looks like something in the VGA capture setup caused the timestamps in that stream to drift, and most talks don't line up correctly. :-( [20:45] <Thulinma> (slides seem to play back 3/5 faster than real time... which is a pain) [20:46] <cone-657> ffmpeg.git 03Paul B Mahol 07master:088dfd3ff15e: avcodec/dxtory: use init_get_bits8() [20:56] <cone-657> ffmpeg.git 03Clément BSsch 07master:adb9b235b6eb: avformat/gif: simplify gif_image_write_header() prototype [20:56] <cone-657> ffmpeg.git 03Clément BSsch 07master:5f9986f597e8: avformat/gif: use first packet palette as global for PAL8 [20:56] <cone-657> ffmpeg.git 03Clément BSsch 07master:3cab173e2327: avcodec/gif: support crop and transparency with PAL8 [20:56] <cone-657> ffmpeg.git 03Clément BSsch 07master:9b964690e399: avfilter: add palettegen filter [20:56] <cone-657> ffmpeg.git 03Clément BSsch 07master:bab4fcebb112: avfilter: add paletteuse filter [20:56] <cone-657> ffmpeg.git 03Clément BSsch 07master:4ab7eb0da232: avfilter: bump minor and Changelog document the new filters [21:01] <ubitux> alright, now back to equally fun stuff to do [21:14] <cone-657> ffmpeg.git 03Diego Biurrun 07master:daf8cf358a09: avformat: Don't anonymously typedef structs [21:14] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:a0fe1a25fa76: Merge commit 'daf8cf358a098a903d59adb6c0d0cc3262a8c93e' [21:26] <cone-657> ffmpeg.git 03Diego Biurrun 07master:7f9f771eac0d: avcodec: Don't anonymously typedef structs [21:26] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:a94eba6f0c06: Merge commit '7f9f771eac0d37a632e0ed9bd89961d57fcfb7e0' [21:35] <cone-657> ffmpeg.git 03Diego Biurrun 07master:bf704132a51f: Don't anonymously typedef structs [21:35] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:6998400c618f: Merge commit 'bf704132a51f5d838365158331d4e535e1df4c8e' [21:59] <rcombs> wow, the raspi's libc yells at you and abort()s if you free(NULL) [21:59] <rcombs> "double free or corruption (out)", it says [22:00] <JEEB> lol [22:06] <nevcairiel> what libc do they use? [22:13] <rcombs> oh, actually, I take that back [22:14] <rcombs> didn't realize that av_freep clears the input pointer before it frees it [22:25] <cone-657> ffmpeg.git 03Diego Biurrun 07master:b339019de4e5: dca: Split code for handling the EXSS extension off into a separate file [22:25] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:d8901c2f018e: Merge commit 'b339019de4e5f4d3c661bbdba98ae248ab77e2f0' [22:25] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:ad0be7038257: avcodec/dcaenc: rename DCA_SUBBANDS [22:27] <ubitux> jamrial: btw, i'll disconnect my laptop around march 20th [22:27] <ubitux> for about a month [22:27] <jamrial> ok [22:28] <ubitux> (traveling, and need a light one, even though the clickpad is going to drive me mad pretty quickly) [22:32] <cone-657> ffmpeg.git 03Diego Biurrun 07master:a96f51f29ac4: dca: Return more informative error codes [22:32] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:877c038a9db5: Merge commit 'a96f51f29ac4cd95650a8bcda6c3d5d87c6357fa' [22:39] <cone-657> ffmpeg.git 03Diego Biurrun 07master:8a213179aff0: dca: Remove trace debugging code [22:39] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:6c44dd6c6ee6: Merge commit '8a213179aff0174d81b3e889134a3b4f7d21f5c3' [22:46] <cone-657> ffmpeg.git 03Diego Biurrun 07master:2a9c6fae9279: dca: Move all tables into dcadata.h [22:46] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:d2223ec2bd36: Merge commit '2a9c6fae927964b5dd0b5d3d9292f5621bd21664' [23:00] <cone-657> ffmpeg.git 03Diego Biurrun 07master:36cf8eb4489f: mov: Fix compilation with DEBUG enabled [23:00] <cone-657> ffmpeg.git 03Diego Biurrun 07master:ecbcebde344c: vdpau: Adjust necessary #includes for vdpau_internal.h [23:00] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:dfa7cb646d20: Merge commit '36cf8eb4489f8709b5fb1cdb87e125ef53301c2f' [23:00] <cone-657> ffmpeg.git 03Michael Niedermayer 07master:e33b084ee986: Merge commit 'ecbcebde344c9eaeb8877ba2c5d32eb3af621e7f' [00:00] --- Sun Feb 15 2015

1 0

ffmpeg.log.20150214
by burek 15 Feb '15

15 Feb '15

[01:23] <YaMoonSun> Dunno what I did wrong. Works fine from m4a to aac http://pastebin.com/d1d43W8H [01:36] <YaMoonSun> Actually, now that I look, it converted one of the songs twice and mislabeled it. [01:38] <llogan> and aac_adtstosc is a typo [01:59] <YaMoonSun> Fuck, thanks, I'm swapping to batch files [02:46] <YaMoonSun> The last m4a in my .bat file refuses to copy the .aac - Says no output file specified. [03:04] <YaMoonSun> Nevermind, my dumb ass.. [03:04] Action: YaMoonSun shakes his head [05:45] <m4t> hey is there a way to have ffmpeg switch files every so often [05:45] <m4t> as in, rotate security cam footage from an rtsp steam [05:49] <m4t> i guess i could do -t <seconds>, but when i do that it just hangs after the specified number of seconds [06:48] <DaneC> Hello, would someone mind helping me with a newb problem? :) [06:50] <DaneC> I'm trying to convert an image (jpeg/png, whichever is easiest) to various pixel formats. Some of them work but others I get an error "unable to find suitable output format". How do I fix this? [07:22] <YaMoonSun> Shouldn't transcode from jpeg, use png as your source, just saying. [07:34] <hts-NOOB> hey anyone online who may be able to help with my dodgy setup? [07:34] <hts-NOOB> [NULL @ 00000000085d8ce0] Unable to find a suitable output format for 'audio=Tim [07:34] <hts-NOOB> eleah HD Capture' [07:40] <hts-NOOB> http://pastebin.com/m2M5HDXv [07:42] <hts-NOOB> also, dhsow log shows video source as "Video: rawvideo (YUY2 / 0x32595559), yuyv422, 720x480, 59.94", yet it should read as Pal settings [07:45] Action: YaMoonSun wishes he could [07:45] <hts-NOOB> thx man [07:46] Action: YaMoonSun didn't do anything though D= [07:46] <hts-NOOB> its most likely some error ive made [07:46] <YaMoonSun> Is dshow to capture your cam? [07:46] <hts-NOOB> hd source [07:46] <YaMoonSun> I know that feeling - I was messing up my .bat files earilier. [07:47] <hts-NOOB> the hdmi capture card, i cannot seem to get ffmpeg to detect the audio, so i plugged in analog and have got a step further [07:48] <hts-NOOB> the dshow filters have same name for audio and vidio [07:48] <YaMoonSun> Damn, nice, doing game capturing? [07:48] <hts-NOOB> yep [07:48] <hts-NOOB> trying to setup for my mate [07:49] <hts-NOOB> hes mad keen gamer, me not so much unfortunately [07:49] <YaMoonSun> Pfft, I wish I had more experience with such things. I mean, I'm a gamer too, but I've never had to use a capture card. I'd presume they come with software. [07:49] <hts-NOOB> my personal opinion, is the discrepency between the hdmi detected as ntsc, and i guess the analog will be 25 fps(pal) [07:50] <YaMoonSun> I don't know if you've every heard of OBS, but it may work also. [07:50] <hts-NOOB> the card has software and drivers(for the one i have anyhow) [07:51] <YaMoonSun> Ouch, what game are you trying to record [07:51] <hts-NOOB> kk thx. i tried vlc, and picture is awsome, but again, no audio detected [07:52] <hts-NOOB> oh wait, not sure ive tried vlc now that i have analog plugged in [07:53] <hts-NOOB> brb [07:53] Action: YaMoonSun wishes you luck [08:00] <hts-NOOB> lol, runs and crashes pc - complete lockup [08:00] <hts-NOOB> brb, need to reboot and update vlc [08:13] <YaMoonSun> That's fucked man. You ran it from VLC you said? [08:53] <hts-NOOB> YaMoonSun>, yeah, i can open to play with vlc on local, plays fine, select to stream via http, and lockup [09:06] <YaMoonSun> Neat. Ever look at OBS? [09:30] <hts-NOOB> not yet mate, sorry reading google [10:08] <hts-NOOB> bbl [14:38] <DaneC> Trying to convert png to various raw formats, but some of the raw pixel formats give me an "unsuitable output format" error, anything I can try? [19:15] <m4t> hey, when using -t parameter with an rtsp:// input, is it possible to have ffmpeg exit at the end of the specified time? [19:15] <m4t> because it just hangs. [20:02] <circ-user-Pcdjw> Hi, could anyone help me out by pointing out what's wrong in this command: http://pastebin.com/sySdtkWx [20:03] <circ-user-Pcdjw> I'm not very familiar with FFMPEG [20:04] <circ-user-Pcdjw> and am getting "Unable to find a suitable output format for 'fps=fps=1/600' fps=fps=1/600: Invalid argument" [20:13] <circ-user-Pcdjw> Anyone ... please ? [20:51] <c_14> get rid of that fps part, it's not valid in its current location [21:04] <sakujo> is "SDL support no" (and opencl disabled) bad for Kodi Home Theatre? [21:05] <JEEB> how would kodi use either of those :P [21:05] <JEEB> also I thought kodi used some crappy fork with random hacks in it? [21:06] <sakujo> it certainly might [21:06] <sakujo> they are very emotional about how distros ship libav as being gross and their ffmpegs being weird or whatnot [21:06] <sakujo> and only their ffmpeg being officially supported [21:07] <sakujo> the issue here is i have no clue what the simple directmedia layer does [21:07] <sakujo> or wait i do [21:07] <sakujo> it controls stuff on low level [21:07] <JEEB> well I'd consider their ffmpeg being the weird one :P [21:07] <sakujo> yeah i was just paraphrasing :p [21:08] <JEEB> also they loudly herped a derp when an old VDPAU thing was removed which was crap, and replaced with a better interface [21:08] <sakujo> hmm so it really would not matter if ffmpeg has SDL support or not i suppose [21:08] <JEEB> generally I have a very negative impression of the whole project [21:08] <sakujo> that is a shame [21:08] <JEEB> but not that I know of any better project in that space [21:08] <sakujo> it is very huge, i think it is inevitable that a lot of negative things come to light [21:08] <JEEB> so while it's shit, I guess you're stuck with it in that feature space [21:09] <sakujo> i would be lying if i did not get disappointed at least a few times, but something that huge cannot cater to every demographic [21:09] <JEEB> but yes, neither opencl or sdl support should have any relation to playback :P [21:09] <sakujo> at least not all the time [21:09] <JEEB> well, I'm more derping about the fact of how little care they give on the lavf/lavc/etc libraries' upstream, and that it's a very "ugh I do this now. ugh I am right" kind of project [21:09] <JEEB> :P [21:10] <JEEB> but I'm just a grumpy multimedia person [21:11] <sakujo> nah as a debian user i can relate [21:12] <sakujo> "now don't let me get started on debian" (/s) [21:13] <JEEB> well personally I can see the debian libav maintainer being quite good compared to some other things [21:13] <JEEB> I mean, he mostly had to fix dozens of packages to get newer versions in because other upstreams wouldn't do it [21:13] <JEEB> that's the reason why gentoo still has latest libav and ffmpeg versions hard-masked [21:14] <JEEB> whatever you might think of FFmpeg/Libav, I personally have no issues regarding how the Debian Libav maintainer did his job during updates for it [21:14] <JEEB> because it's a shitty job, having to update all the things to latest APIs [21:15] <JEEB> and once again, there's the gentoo case where that shit is still going on [21:15] <JEEB> (latest versions of FFmpeg/Libav being hardmasked) [21:17] <Mavrik> mhm, well libav API does change quite a bit [21:18] <JEEB> well, they do keep old APIs for at least one release [21:18] <JEEB> and it doesn't happen too often that a release happens there [21:18] <Mavrik> True, but remember, we're talking about projects that still link to ffmpeg 0.8 or smthn :D [21:18] <JEEB> well, yeah. old releases have old shit [21:19] <JEEB> debian generally is actually pretty fast with Libav [21:19] <JEEB> considering that it's a single person poking everything depending on it [21:19] <Mavrik> Talking to the guy who maintained Cinelerra for a while was rather fun. [21:19] <Mavrik> Until it affected his sanity too much [21:19] <JEEB> I think libav 9 was... half a year+ [21:19] <JEEB> yeah, I'm surprised the Debian Libav maintainer is keeping his cool [21:19] <JEEB> libav 10 was I think a bit less [21:20] <JEEB> I complained to the gentoo guy some time ago and he just ragequit [21:20] <JEEB> "nobody understands my issues and how this is not just my problem!" [21:20] <Mavrik> Personally I think linking to libav dynamically is just crazy [21:21] <JEEB> dynamic linking is ok, but at this point if you want to support some mostly unmaintained crap as a distro, you should distro multiple versions [21:21] <JEEB> but that seems to be a no-go [21:21] <JEEB> "we did that once for audacity, I don't want to do it again" [21:22] <Mavrik> meh, that sounds like the lesson every package management software has to learn at some point [21:22] <Mavrik> ALWAYS pin your dependencies to the version your software is compiled/released at [21:22] Action: Mavrik came from Go meetup where they learned that the hard way... again. [21:23] <JEEB> the package management software has supported multiple API versions of stuff for ages as long as the soname is set [21:23] <JEEB> but it seems like security people are hard against that [21:23] <JEEB> "the more shit we poke out the more shit we have to look out for here at security >:|" [22:13] <m4t> is there a way to minimize the long delay when starting the recording of an rtsp stream? [22:16] <m4t> hrm i guess it seems to still record even though it doesn't output the stream info [22:17] <BtbN> It's propably probing for codec infos [22:17] <BtbN> how long is the delay? [22:19] <m4t> 30s [22:19] <m4t> is it estimating stream duration? [22:20] <m4t> my main issues are with the '-t' option. both linux64 static builds hang at the end of duration [22:20] <m4t> and the debian wheezy package seems to just ignore the parameter. [22:21] <m4t> took 64s to start this time [22:23] <BtbN> Try to set a low -analyzeduration [22:24] <BtbN> But iirc default is 5 seconds [22:24] <BtbN> So 30 seconds is strange and propably a diffrent problem [22:24] <BtbN> what does it putput when you set the loglevel to debug? [22:24] <BtbN> *output [22:25] <m4t> we'll see in 5min :-) [22:26] <m4t> [rtsp @ 0x32d9160] Probe buffer size limit 5000000 reached [22:26] <m4t> then it starts [22:27] <m4t> -probesize does it [22:29] <m4t> yeah it still blows right past -timelimit [22:56] <m4t> i can't get debian avconv to convert this ulaw stream to aac either [22:56] <m4t> audio comes up 0kb [22:56] <m4t> libmp3lame works fine [23:02] <m4t> i'll just use the static release and prefix ffmpeg with a 'timeout -s INT <time>' [00:00] --- Sun Feb 15 2015

1 0

ffmpeg-devel.log.20150213
by burek 14 Feb '15

14 Feb '15

[00:11] <cone-118> ffmpeg.git 03Gilles Chanteperdrix 07master:afb0e5a810ae: avformat/rtsp: allow receiving subtitles via RTP [00:38] <cone-118> ffmpeg.git 03Gilles Chanteperdrix 07master:af940e6cb121: avformat/rtpdec: add T.140 RTP depacketization (RFC 4103) [00:59] <cone-118> ffmpeg.git 03Gilles Chanteperdrix 07master:c7ad1f562b0f: avformat/rtsp: parse lang attribute in SDP [01:23] <pmarques> hi everyone, I'm using ffprobe to parse metatags but it don't let me catch multiple genres or multiple artists [01:24] <pmarques> anyone know if there is some thoughts about support it or if it should be supported at all? [04:32] <cone-201> ffmpeg.git 03Michael Niedermayer 07master:849ad5175b79: avformat/mov: Heuristically detect raw udta [10:23] <cone-366> ffmpeg.git 03Paul B Mahol 07master:8bb489fab54a: fate: add wavpack encoder [12:30] <cone-366> ffmpeg.git 03Anton Khirnov 07master:faa8ffda2c51: doc/APIchanges: fill in missing dates and hashes [12:30] <cone-366> ffmpeg.git 03Michael Niedermayer 07master:c74b3983f99a: Merge commit 'faa8ffda2c513573733624784f0a7d0a4959d33e' [13:04] <cone-366> ffmpeg.git 03Peter Meerwald 07master:eea769df322f: hevc: Use generic av_clip function, not C implementation [13:04] <cone-366> ffmpeg.git 03Michael Niedermayer 07master:3e2714992b50: Merge commit 'eea769df322fac2601a96db195fa7dc8d12a8fbc' [14:02] <cone-366> ffmpeg.git 03zhaoxiu.zeng 07master:b39ac9d210df: avcodec/vc1: simplify find_next_marker() [14:49] <ahmed2> hello , I'm new to the codecs thing , and I search alot before asking but still confused , Can anyone explaine what ffmpeg is ? in a simple words I can understand ? [14:51] <ubitux> ahmed2: http://ffmpeg.org/ffmpeg.html#Description http://ffmpeg.org/ffmpeg.html#Detailed-description [14:57] <compn> ahmed2 : ffmpeg can convert videos from one format to another [14:58] <bryno> ... and A LOT more! [14:59] <compn> yep [15:00] <ahmed2> when I convert a video using handbrak ... what I'm actully using ? x264 or ffmpeg ? [15:01] <compn> both [15:01] <iive> ffmpeg is universal video processing program. It consists of libraries that can read and write most file formats (de/muxing), almost complete collection of decoders (including some obscure old video game ones), a have a number of encoders. [15:01] <iive> x264 is H.264 encoder, and good one at that. ffmpeg can use it externally. [15:01] <ahmed2> which program is the one who encode videos ? [15:01] <iive> or rather, use it as library. [15:02] <arwa> ubitux, Can you help me with the fft-domain filtering code? I tried doing a very basic thing, I am just taking fft of rows of the image and idft of it. The code is giving me seg fault. [15:02] <arwa> http://pastebin.com/UET2v3DN [15:02] <iive> a video file consists of audio and video, synchronized to match. x264 can do only video encoding, it doesn't even provide a decoder. [15:03] <ahmed2> so it's like : ffmpeg is gui for x264 ............ and handbrake is gui for ffmpeg [15:03] <ahmed2> ? [15:04] <iive> ffmpeg is definitely not gui. [15:05] <ahmed2> a command-line front end for x264 and other codecs ? [15:05] <ahmed2> <iive> [15:06] <iive> in case you didn't get it. ffmpeg contains its own set of decoders for everything. [15:07] <iive> it also have huge set of (de)muxers, allowing to read and write different file formats. [15:07] <BtbN> The ffmpeg cli tool is also just another application using the libav* libraries, which are the primary part of ffmpeg as a project. [15:07] <iive> it even contains support for different input methods (captures). [15:13] <ahmed2> this may sound stupid ... But can I use x264 instead of using ffmpeg ? because x264 is the best codec to encode videos into h264 format [15:14] <iive> ahmed2: x264 can use libav* libraries from ffmpeg/libav [15:14] <iive> otherwise you'd need to feed it raw frames and get video without sound. [15:16] <cone-366> ffmpeg.git 03zhaoxiu.zeng 07master:b4b9a64bdb61: avcodec/vc1: simplify vc1_split() [15:23] <ubitux> arwa: where does it segfault? use valgrind or gdb [15:32] <wm4> michaelni: someone captured mpv's dvb output, which libavformat reportedly crunched on for 5-8 seconds (with default settings); I don't know if the ts is entirely valid: http://srsfckn.biz/mpv-samples/dvb-s2-capture.ts [15:34] <nevcairiel> on normal file playback, you really dont notice the delays, its only on livestreams because you dont have all the data available, so it waits for it [15:36] <wm4> and on tvheadend it reportedly needs only 1-2 seconds [15:36] <nevcairiel> i have some hacks in my code that try to determine if its a streaming source, and then reduces analyzeduration to 1 second [15:37] <nevcairiel> it would be ideal if it just stopped after having a pmt and all the streams in it [15:41] <wm4> and with analyzeduration set to 1 second it still takes 5 seconds [15:43] <arwa> i tried using valgrind, but it is not showing the line number. :/ [15:45] <arwa> http://pastebin.com/G1mKB35R [15:45] <arwa> ubitux, ^ [15:46] <mateo`> arwa: maybe use ffmpeg_g instead of ffmpeg [15:47] <arwa> same error! [15:47] <arwa> btw what is the diff between ffmpeg and ffmpeg_g? [15:47] <ramiro> arwa: ffmpeg_g contains debug symbols [15:47] <ramiro> so that your backtrace actually contains names instead of ??? all over the place =) [15:48] <arwa> oh nice! [15:48] <compn> yeah those ??? are not helpful ;D [15:50] <arwa> yeah...its not helping :/ [15:50] <arwa> it is just showing that the error occured in filter_frame..! [15:51] <durandal_1707> which filter? [15:51] <ramiro> pastebin the new error [15:55] <cone-366> ffmpeg.git 03zhaoxiu.zeng 07master:e4ea84e12e3a: avcodec/wmv2: simplify cbp_table_index calculation [15:58] <arwa> fft filter [15:59] <arwa> http://pastebin.com/Vwaz21PN [16:00] <arwa> new error ^ [16:00] <ubitux> so what's line 72? [16:20] <eric_> Hello all! Is there a reason that avformat_free_context -> ff_free_stream does not free the intra_matrix, inter_matrix & rc_override fields? Where as avcodec_free_context does free these fields [16:21] <wm4> eric_: did you use the AVCodecContext from the AVStream for decoding? [16:22] <eric_> No, I'm debugging an encoding problem and was just stepping through and noticed this [16:22] <wm4> or for encoding [16:22] <wm4> in both cases, the AVCodecContext must be separate AFAIK [16:24] <eric_> avformat_new_stream creates a new context, which i use [16:24] <wm4> well, don't [16:25] <eric_> I saw the note about that in the V11 release notes, but thought using the avformat_new_stream one was ok now [16:25] <wm4> use avcodec_alloc_context3 and avcodec_copy_context [16:25] <wm4> (as far as I understood this) [16:26] <eric_> Yes - I'll change that now [16:26] <eric_> thanks! [16:26] <wm4> eric_: try running ffmpeg (the command line program), and see if it has the same leak [16:27] <eric_> I've not Valgrinded it yet, I just saw it doing a code read [16:29] <eric_> It seems it might be safer for ff_free_stream to call avcodec_free_context to reduce the code duplication [16:36] <Daemon404> nevcairiel, Unknown OS 'mingw64_nt-6.3'. [16:36] <Daemon404> did we ever decide on the 'fix'? [16:37] <nevcairiel> i guess just extend the mingw32 checks to cover mingw64 as well [16:37] <nevcairiel> and screw the MSYS* case [16:37] <Daemon404> nobody should be using that case [16:37] <Daemon404> i think [16:38] <nevcairiel> I just hate that its the default if you just call bash.exe [16:38] <nevcairiel> which a bunch of my scripts do [16:39] <Daemon404> like x264? ;) [16:41] <kierank> wm4: ts looks fine [16:42] <wm4> eric_: that's probably true... though the Libav development direction goes towards removing the AVCodecContext from AVStream entirely [16:42] <wm4> kierank: nice, so it's really just libavformat, probably [16:42] <ramiro> last time I tried msys2 it was considerably slower than msys. [16:43] <ramiro> why do the ffmpeg docs suggest using msys2? [16:43] <wm4> because msys is dead and broken or something [16:43] <ramiro> dead - ok, broken - really? [16:44] <wm4> unless it's 100% bug-free (which is not possible), yes [16:45] <ramiro> true, but I'd like to know what is so broken about that msys that it's not suitable for building ffmpeg [16:45] <wm4> oh, ffmpeg probably supports irt [16:46] <wm4> because ffmpeg supports anything no matter how broken or ancient it is [16:49] <ramiro> not true, BeOS is gone [16:50] <durandal_1707> michaelni: the piz bug is on any big endian not just mips? [16:50] <michaelni> could be, i only tested mips [17:08] <cone-366> ffmpeg.git 03zhaoxiu.zeng 07fatal: ambiguous argument 'refs/tags/n2.5.4': unknown revision or path not in the working tree. [17:08] <cone-366> Use '--' to separate paths from revisions [17:08] <cone-366> refs/tags/n2.5.4:HEAD: avcodec/wmv2: simplify cbp_table_index calculation [17:19] <durandal_1707> saste: why you removed lavfi from GSoC page? [17:35] <saste> durandal_1707, because I'm not going to mentor it [17:46] <Daemon404> now i remember why i dont use the official mingw-w64 toolchains... [17:46] <Daemon404> fricking winpthreads [17:47] <ramiro> what's wrong with them? [17:47] <nevcairiel> winpthreads is broken as fuck [17:49] <ramiro> more specifically... ? [17:50] <wm4> I'm interested in this too [17:51] <nevcairiel> i never bothered to investigate too deep, all I know is that it was slow and deadlocked occasionally [17:51] <nevcairiel> pthreads-w32 ftw [17:51] <nevcairiel> although i use w32threads for ffmpeg anyway [17:52] <ramiro> isn't winpthreads based on pthreads-w32? [17:52] <nevcairiel> dont think so [17:52] <nevcairiel> and even if i'm wrong, then they broke it :P [17:57] <ramiro> hm, apparently it isn't based on pthreads-w32... [17:58] <ramiro> anyways, nevcairiel, I'm sure kai tietz will be very helpful if you ever encounter a bug and report it to him [17:58] <nevcairiel> i just dont use it [18:23] <cone-366> ffmpeg.git 03zhaoxiu.zeng 07master:bc0a440e889a: avcodec/wmalosslessdec: change type of acfilter_coeffs from int64_t to int16_t [18:35] <cone-366> ffmpeg.git 03zhaoxiu.zeng 07master:80f13780051f: avcodec/wmalosslessdec: optimize sign operation [19:13] <aetasx> I dont have a problem too often with winpthreads but I have been known to bitch about it in the past [19:29] <wm4> this downstream list looks really useful [19:29] <wm4> michaelni: keep in mind that if you EOL some ffmpeg releases, the distros using it will probably update [19:54] <wm4> why does the h264 decoder print "Detected unsupported YCgCo colorspace." [19:54] <Daemon404> because it detected the colorspace is ycgco [19:54] <Daemon404> why else? [19:55] <wm4> why would it care? it exports the flag just fine [19:55] <Daemon404> i dont know [19:55] <wm4> of course libswscale barfs all over this, but the decoder should be fine [19:55] <Daemon404> does swscale support it? [19:55] <wm4> no [19:55] <Daemon404> yeah so it's just normal inbreeding [19:55] <Daemon404> ;) [19:55] <wm4> trollol [20:44] <cone-366> ffmpeg.git 03Luca Barbato 07release/2.4:891de4b27a07: log: Unbreak no-tty support on 256color terminals [20:44] <cone-366> ffmpeg.git 03Xiaohan Wang 07release/2.4:532c96a2158c: matroskadec: Fix read-after-free in matroska_read_seek() [20:44] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:9d1e775b8966: Merge commit '891de4b27a07b808839b9e873b6a886248c8fd6b' into release/2.4 [20:44] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:92595faab9f4: Merge commit '532c96a2158c04f265d750d54f2f103b8d9fe0ef' into release/2.4 [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:0a878d0c941c: doc/APIchanges: Fill in some more missing hash values [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:2029acb6679b: doc/APIchanges: Add av_find_best_pix_fmt_of_2() and av_get_pix_fmt_loss() [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:5080ab26d304: doc/APIchanges: fill in and correct some values [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:05a8114c7190: doc/APIchanges: fill in more missing hash values and dates [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:01b5e61845b8: avformat/utils: Fix number suffixes in tb_unreliable() [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:1ecce1c6a752: swresample/dither: Cleanup number suffixes [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:096fd2698ae9: avcodec/dxtory: Use LL instead of L number suffix [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:1497f355c7ea: avformat/matroskadec: Fix number suffixes [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:84e5b314f354: avformat/smacker: Fix number suffix [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:b8546aee8431: avformat/omadec: fix number suffix [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:8a16b27de9c3: avcodec/h264_cabac: use int instead of long for mbb_xy [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:a31fdcef55e4: avcodec/mpegvideo_enc: Fix number suffixes in rc_buffer_size calculation [21:00] <cone-366> ffmpeg.git 03wm4 07release/2.4:e2e835f017d6: avformat/tta: fix crash with corrupted files [21:00] <cone-366> ffmpeg.git 03wm4 07release/2.4:2515de3b1534: avformat/mpc8: fix hang with fuzzed file [21:00] <cone-366> ffmpeg.git 03wm4 07release/2.4:600c6ebc7d04: avformat/mpc8: fix broken pointer math [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:ee8e48d38677: avformat/mpc8: Use uint64_t in *_get_v() to avoid undefined behavior [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:3531ff8db319: avcodec/mjpegdec: Check escape sequence validity [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:08822122987f: avcodec/mjpegdec: Check number of components for JPEG-LS [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:8413ddcd39ee: avcodec/mpegvideo_motion: Fix gmc chroma dimensions [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:076b98c9b74c: swscale/utils: Limit filter shifting so as not to read from prior the array [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:0c125519ecc4: avformat/thp: Check av_get_packet() for failure not only for partial output [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:3e46e3a33c84: avcodec/h264_ps: More completely check the bit depths [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:724c79276ab2: avcodec/h264: Be more strict on rejecting pps_id changes [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:9dc8f4482985: avcodec/h264: Be more strict on rejecting pps/sps changes [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:1cc419eae8a1: avutil/opt: Fix types used to access AV_OPT_TYPE_PIXEL_FMT [21:00] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:b250375e77a5: avutil/opt: Fix type used to access AV_OPT_TYPE_SAMPLE_FMT [21:01] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:7b213e88b5e6: avcodec/h264_slice: Do not change frame_num after the first slice [21:01] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:2073ab266eef: avcodec/h264_slice: Check picture structure before setting the related fields [21:01] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:4d5beea7a182: avcodec/h264_slice: ignore SAR changes in slices after the first [21:01] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:6005f375aab6: avcodec/h264_slice: assert that reinit does not occur after the first slice [21:01] <cone-366> ffmpeg.git 03Carl Eugen Hoyos 07release/2.4:ca98c016cd44: lavc/aarch64: Do not use the neon horizontal chroma loop filter for H.264 4:2:2. (cherry picked from commit 4faea46bd906b3897018736208123aa36c3f45d5) [21:01] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:492818d724d9: avcodec/mjpegdec: Skip blocks which are outside the visible area [21:01] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:1a263f0dd96a: avcodec/arm/videodsp_armv5te: Fix linking failure with "g++ -shared -D__STDC_CONSTANT_MACROS -o test.so ... libavcodec.a" [21:01] <cone-366> ffmpeg.git 03wm4 07release/2.4:d705125b949b: qpeg: avoid pointless invalid memcpy() [21:01] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:43924a8e992f: avcodec/hevc: Fix handling of skipped_bytes() reallocation failures [21:01] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:416501da1aa7: avdevice: Use av_format_get_control_message_cb() [21:01] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:74c7273b5d5c: avfilter/vf_framepack: Check and update frame_rate [21:01] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:cb7d72ed1814: avcodec/flac_parser: fix handling EOF if no headers are found [21:29] <cone-366> ffmpeg.git 03Michael Niedermayer 07release/2.4:d66d5d61881f: Update for 2.4.7 [22:52] <rcombs> http://puu.sh/fSvgp/7b906ed651.png is this evil enough? [22:56] <ubitux> why not use ffprobe -of json? [22:57] <wm4> too easy [22:57] <Loriker> :o) [22:58] <rcombs> this involves an unhealthy quantity of C++ templates and large macros [22:59] <wm4> v8? [22:59] <rcombs> yup [23:00] <ubitux> all of this makes me very sad [23:00] <wm4> I'm wondering how hard it'd be to make a V8 vapoursynth wrapper [23:00] <rcombs> because that AVFormatContext isn't just a bunch of properties of the C struct copied into JS [23:00] <rcombs> the v8 objects are actually mapped directly to the underlying structs [23:01] <rcombs> writing to the metadata objects in JS calls av_dict_set [23:01] <wm4> exactly what we'd need for vapoursynth [23:01] <wm4> how hard was it? [23:02] <rcombs> the I/O functions are async and take place on a separate thread and call the callback when they finish [23:02] <wm4> wat [23:02] <rcombs> can't be blocking the main event loop :3 [23:03] <wm4> fucking node garbage [23:03] <rcombs> (callback is called with Null if everything went well and an Exception, whose text comes from av_err2str) [23:04] <rcombs> wm4: most of the hard parts had to do with finding good docs on v8, because if you google for v8 names you'll generally find Doxygen HTML from before an apparently-recent API overhaul [23:05] <wm4> so v8 has a sufficiently unstable api [23:05] <rcombs> I ended up just looking at the headers most of the time [23:05] <rcombs> oh, and it's not super well-tested, but as far as I'm aware it all GCs well [23:06] <wm4> maybe I'll give it a try [23:06] <wm4> hopefully won't end in blowing my brains out [23:07] <wm4> (a possibility with C++, templates, and google software) [23:07] <rcombs> the parent objects all keep references to their children, and their children get messages via weak references if their parents are deleted and stop accessing memory that belongs to them [23:07] <rcombs> (which effectively looks like becoming an empty object) [23:08] <rcombs> I started from https://github.com/OptimalBits/navcodec and ended up stripping out almost everything [23:09] <rcombs> because it was largely written badly, with the old v8 API, with what appears to be a poor understanding of both Node module conventions and lav* [23:09] <cone-366> ffmpeg.git 03zhaoxiu.zeng 07master:9e6198f0a457: avcodec/wmalosslessdec: simplify [23:09] <wm4> not sure if I'd write it as node module though [23:09] <wm4> or will js hipsters hate me if I do not [23:09] <rcombs> at some point I need to add some JS wrappers that make the functions more friendly and optionally return Promises instead of taking callbacks [23:12] <rcombs> (right now you see those empty args being passed because if I don't my code throws an exception yelling at me for not having proper args) [23:12] <rcombs> (because juggling args is not a fun thing to do with the v8 API) [23:13] <rcombs> (so I just went "fuck it, I'm enforcing everything as being correct and I can wrap it for prettiness in JS later") [23:14] <wm4> promises sound like a perfect abstraction for the vs frame callback [23:15] <rcombs> in my experience the best thing about promises is that they make it easy to avoid The Callback Pyramid Of Doom [23:16] <rcombs> by still having The Callback Pyramid Of Doom but abstracting it so you don't actually have to indent for each one [23:16] <wm4> meh [23:16] <wm4> why do badly designed things always get popular [23:17] <wm4> ok enough hating on node for a day; good night [23:17] <rcombs> doThing().then(function(ret){&return Promise(things);}).then(&).then(&) [23:18] <wm4> lol when will thry introduce monads [23:18] <rcombs> instead of doThing(function(ret){&moreThings(function(stuff){stillMoreThings(function(items){&})})}) [23:18] <rcombs> anything that ends in )})}) is not fun [23:20] <rcombs> I actually had this whole setup doing everything I wanted it to do hours ago [23:20] <rcombs> but then decided I should be a bit more thorough about cleaning up the existing code [23:21] <rcombs> now I'm at the point where I think I'll wipe out everything I haven't gone over (and don't currently need) and push it [23:24] <cone-366> ffmpeg.git 03zhaoxiu.zeng 07master:0073c8e34500: avcodec/apedec: move 'coeffs[256] and delay[256]' into, long_filter_high_3800 [00:00] --- Sat Feb 14 2015

1 0

ffmpeg.log.20150213
by burek 14 Feb '15

14 Feb '15

[00:00] <qxt> ofc that makes sense [00:00] <kepstin-laptop> just too many variables for a single fps value to have any meaning whatsoever. [00:31] <qxt> this is the official site right? ffmpeg.mplayerhq.hu [00:32] <qxt> not www.ffmpeg.org ? [00:37] <c_14> afaik they're both official [00:38] <c_14> ffmpeg.org is hosted by mplayer [00:55] <ac_slater_> hey all. I'm generating some video from an external encoder. I need to suck up the video and mux it to mpegts. I'm struggling to understand generating PTS/DTS. Is there a solid guide for this? [01:18] <pmarques> Hi [01:19] <pmarques> I'm using ffprobe to parse metatags but it don't let me catch multiple genres [01:20] <pmarques> anyone know if there is some thoughts about support it? [01:27] <DayDreamer> gnight [03:00] <k_sze[work]> Is it normal that when decoding packets from a .nut file containing H.264 video, the frames can come out of sequence (and duplicate from time to time)? [07:38] <ac_slater> hey all. I'm trying to manually fill some AVFrame structures. What's the allocation scheme? The docs say av_frame_alloc() is just for the structure, but how do I allocate the actual buffers? [07:41] <ac_slater> ah nevermind, apparently `av_frame_get_buffer()` [07:41] <ac_slater> though, that's really confusing [08:31] <k_sze[work]> Has anybody run into a weird problem with encoding or decoding where frame times start getting out of sequence after about 2^20 frames (about 9:42:32.53333 in terms of length, at 30 fps)? (I'm getting this with only a YUV422 video stream encoded as H.264, no audio, no subtitle, in a .nut file) [10:46] <Lirk> hi all [10:46] <Lirk> how I can to capture file with timestamp and then reencode it? [10:48] <Lirk> in one step [11:28] <Ders> Anyone here that can help me with something? [11:30] <iive> Ders: nobody can help you, [11:30] <iive> until you ask your question. (don't ask to ask, just ask) [11:32] <iive> and be patient... people are not jumping to read the channel on every new message. [11:32] <Ders> Well, I wanted to know if people were online [11:33] <Ders> So, I have raw audio data in s16le format and I want to it to be written to a file that I can listen to [11:33] <Ders> http://pastebin.com/P4JdMKgj [11:33] <Ders> that is my command I'm using [11:33] <Ders> now, I'm using stdin to write the data to the input [11:34] <Ders> this is how I write the data: http://pastebin.com/XCbRXyZj [11:35] <Ders> But when it has created the file it's really small and I can't play it with VLC [11:35] <iive> pcm is not small, -ac 1 means mono, so 88kBytes/s [11:36] <iive> can ffplay play it back? [11:36] <Ders> I haven't tried that and I don't think I have ffplay at hand. I'm only given the ffmpeg.exe [11:37] <iive> oh, windows [11:37] <Ders> yeah [11:37] <iive> ok, first think I find troubling is that you have 2 acodec's [11:37] <iive> i'm not sure that's how you specify the input format. [11:37] <Ders> which one do I have to remove? I guess the Pcm_s16le one? [11:38] <Ders> the input format is -f s16le [11:38] <Ders> it's a list of samples as short ints [11:41] <iive> -f s16le is correct. [11:42] <Ders> Yeah that's what I'm most confident of as well [11:44] <iive> well, the double acodec doesn't seem to cause any problem. [11:44] <iive> you line seems to work fine with existing wav file [11:45] <Ders> yeah, I tried that as well. [11:45] <Ders> That's why it's so weird that it's not working with manually stdin data [11:49] <Ders> I don't know if it's of any help but I tried writing the values to a .s16le file [11:49] <Ders> this is the result: http://pastebin.com/ydCsGT3y [11:49] <Ders> Normally it should only be numbers [11:49] <Ders> Maybe that's the problem? [11:53] <iive> Ders: this looks like hexadecimal humbers [11:55] <Ders> Yeah, I noticed that too. Let me post the code I used to generate those numbers [11:56] <Ders> http://pastebin.com/35mBxD6j [11:57] <Ders> oh, and I'm using fstream to write it to a file atm [11:57] <iive> bbl [13:36] <synthecy1her> How do I stop ffmpeg creating 'ffmpeg2pass-0.log' and 'ffmpeg2pass-0.log.mbtree' while doing two pass encoding? [13:38] <Mavrik> By not doing twopass encoding. [13:39] <Mavrik> Those files are kinda the integral point of 2-pass. [13:49] <hta> TEPMPDIR=$(mktemp -d); pushd $TEMPDIR; <encode>; popd; rm -rf $TEMPDIR [13:50] <hta> I guess... [16:40] <Ders> Does anyone know that when I do the following command, my output file doesn't have audio? [16:40] <Ders> http://pastebin.com/89PraDtG [16:41] <c_14> try -map 0 -map 1 [16:41] <Ders> will do [16:42] <kepstin-laptop> and you shouldn't need to set the format, codec, channels, etc. on a wav; ffmpeg will read that from the header. [16:42] <kepstin-laptop> (in fact, with those settings you'll get a brief bit of noise at the start as the wave header is interpreted as audio) [16:43] <Ders> oh ok, Well, the -map doesn't change anything [16:43] <Ders> for some reason VLC can't even find an audio stream [16:43] <kepstin-laptop> Ders: pastebin the ffmpeg output? [16:44] <Ders> I'll try. Normally I can't select the data from the output I have [16:48] <Ders> I can't copy it but I get lots of "packet too large, ignoring buffer limits to mux it" and "buffer underflow st=1 bufi=3939 size=4096" [16:48] <Ders> that's all I get [16:54] <Ders> Is it possible that I have to reencode my audio? I tried with pcm_alaw and that didn't give any errors but I still don't have sound [16:55] <kepstin-laptop> Ders: I'm interested in what ffmpeg prints out about the stream mapping at the start of the encode. [16:56] <Ders> yeah, I can see that now it doesn't from thousands of errors. Five me a second [17:00] <Ders> It's an image: http://i.imgur.com/15emTe5.png [17:00] <Ders> I couldn't copy the output to Pastebin [17:03] <durandal_1707> Ders: have you chacked that maybe it is VLC fault [17:05] <Ders> How would I do that? By the way. I read that the encoding of my wav file (s16le) isn't available to encode to an mpeg file? Is that right? Maybe I should reencode my audio first [17:05] <kepstin-laptop> Ders: your audio file and video file have vastly different lengths, which is what's causing the buffer underflow errors, and could cause playback errors [17:06] <kepstin-laptop> says you have a minute of audio and only 14 seconds of video [17:06] <Ders> So do I have to add the -shortest command? [17:07] <kepstin-laptop> yeah, try something like ffmpeg -i VIDEO_INPUT_FILE.mpeg -y -i AUDIO_INPUT_FILE.wav -vcodec copy -acodec pcm_s16le OUTPUT_FILE.mpeg [17:07] <kepstin-laptop> with -shortest as well [17:08] <Ders> ok. trying now [17:11] <Ders> nope. The properties of the file do say that there are 2 channels (although the original file is encoded for one channel) and that its audio sample rate is 44khz [17:15] <kepstin-laptop> Ders: are you getting output like "[mpeg @ 0x1bef080] packet too large, ignoring buffer limits to mux it" ? [17:15] <Ders> yeah [17:15] <kepstin-laptop> if so, then the output file is useless junk :/ [17:16] <Ders> haha :D and how do I solve that? [17:16] <smo_> hi, i use nodejs to pipe http video links thru ffmpeg, sometimes it play directly and sometimes i need like 10% of the file downloaded before the stream start start playing an idear why ? [17:16] <Ders> Oh I got it working :D [17:16] <Ders> hold on I'll post the commands [17:16] <kepstin-laptop> Ders: the issue is just that the unencoded audio looks like it's too big. encode it with mp3 or something. [17:17] <Ders> So, I had to reencode my pcm audio with -f vob to another .wav file. that was actually the problem [17:17] <Ders> That's everything :) [17:17] <kepstin-laptop> umm. the sentence you just wrote makes no sense. if you use -f vob, it's not a wav file [17:19] <kepstin-laptop> (it looks like unless you specify otherwise, ffmpeg encodes to mp2 audio when you use vob output) [17:19] <kepstin-laptop> and mp2 makes the sound packets small enough that they would fit. [17:24] <kepstin-laptop> your original command would have worked if you'd simply dropped the "-c:a" option completely (no copy, no pcm_whatever), since ffmpeg would have defaulted to mp2 audio then. [17:24] <Ders> Well. I create a wav file in the s16le format. But mpeg doesn't like that. So I had to first reencode the original audio file to another format [17:24] <Ders> and I did that by outputting the file with the -f vob command [17:24] <kepstin-laptop> no, ffmpeg could read the input fine. the issue was that it didn't fit into the output file unless encoded to a different format [17:24] <Ders> oh [17:25] <Ders> well, the -shortest command does not help with that. It still threw those errors of overflow when it wasn't reencoded [17:25] <kepstin-laptop> so either you have to encode the audio with something like ac3, mp3, mp3 - or use a different output container, like mkv [17:25] <Ders> ok! I'll keep that in mind! Thank you very much for the information :) [17:26] <kepstin-laptop> the shortest command would not help with that, no, the issue is that with pcm audio formats, each individual audio packet was bigger than the max size in the mpg format. [17:28] <Ders> oh ok. I understand now :) [17:30] <kepstin-laptop> and next time you run into issues, please figure out how to copy/paste text from the terminal window. It is possible to do in windows... [17:30] <Ders> yeah I know, I'll ask how to do it in the office (I can't use the regular Mark -> select -> Enter method) [17:31] <kepstin-laptop> it's weird tho, because I know that dvds (which use an mpeg format) can have pcm audio. [17:32] <Ders> Maybe I have to resize my buffer? [17:32] <kepstin-laptop> i dunno. [17:33] <kepstin-laptop> looks like ffmpeg does have a special codec 'pcm_dvd' for dvds, but it doesn't support encoding to it. [17:33] <Ders> Well, the buffer for the video encoding is set to some value. Maybe it's something to do with that [17:33] <Ders> hmm [17:42] <kepstin-laptop> i wonder if the ffmpeg mpeg muxer just doesn't support outputting pcm audio at all? [17:42] <kepstin-laptop> seems silly that it lets you try, but I guess ffmpeg lets you do lots of silly things. [17:43] <Ders> Yeah, that's what I was thinking as well. Problem is now that I'm creating two files. Could it be possible to reencode the original file while it's an input stream? [17:43] <Ders> ffmpeg is one silly beast. But when you got it working, man... so fast [17:44] <kepstin-laptop> hmm? there's no reason to create two files... [17:44] <kepstin-laptop> just drop the "-c:a copy" from your original command and ffmpeg will autoselect a working audio codec (probably mp2) [17:45] <kepstin-laptop> or you can manually pick libmp3lame or ac3 or whatever. [17:46] <Ders> I'll do some tests. At the moment it's not working at least. [17:46] <kepstin-laptop> or, if you don't really need mpg, switch to matroska (mkv) or something, which has more codec compatibility. [17:47] <Ders> ok! Thanks for all the help! Gotta go [17:47] <Ders> cya [17:50] <smo_> hi, i use nodejs to pipe http video links thru ffmpeg, sometimes it play directly and sometimes i need like 10% of the file downloaded before the stream start start playing an idear why ? [17:58] <c_14> mp4? [17:58] <smo_> me c_14 ? [17:58] <c_14> yes [17:59] <smo_> could be any type of files... [17:59] <smo_> mkv avi mp4 ts ... [17:59] <smo_> think it s related but not sure [18:00] <c_14> If it happens with mp4 files it's probably tho MOOV atom. [18:01] <smo_> i convert all the files in live so can i add it to all videos? [18:01] <c_14> hmm? [18:01] <smo_> the moov atom ? ( i convert with libx264) [18:02] <smo_> a avi files for exemple do not have atom right ? [18:03] <c_14> yes [19:23] <spaam> is there a standard yet for storing hevc/h265 in .mp4 yet ? [19:25] <jdsanders1> Hi folks - I converted a .mov file to .mp3, and ffmpeg reports the duration I expect, but when I play the file in itunes or quicktime player, the duration is doubled, and the second half is just dead air [19:25] <jdsanders1> does anybody know what's going on there? [19:26] <spaam> how did you do it? what is the commandline for ffmpeg [19:26] <JEEB> spaam, for about a year now [19:26] <JEEB> or actually almost two [19:27] <JEEB> FDIS is from around sept 2013 in my archives [19:27] <jdsanders1> spaam: as naively as possible: ffmpeg -i foo.mov foo.mp3 [19:29] <jdsanders1> just tried it with manually setting the duration to what i expect [19:29] <jdsanders1> with the same behavior [19:30] <spaam> JEEB: great [19:31] <spaam> jdsanders1: what does the output from ffmpeg say when you run that? pastebin the output [19:31] <JEEB> I made a post on matroska-devel back then to basically "standardize" the same kind of thing there [19:31] <datajerk> hi, looking for advice or searchable terms to help myself. I'd like to in a video at specific times highlight a circular region of the screen by dimming the rest. It would also be nice to fade the effect in and out [19:32] <spaam> JEEB: ahh nice :D [19:32] <JEEB> http://permalink.gmane.org/gmane.comp.multimedia.matroska.devel/4771 [19:32] <jdsanders1> spaam: thanks for your help! here's the output: http://pastebin.com/y9mytk2J [19:32] <saste> datajerk, at the moment this doesn't look possible directly, but you may found a clever combination of a pre-generated overlay background+overlay [19:33] <saste> datajerk, for overlay I mean the filter [19:34] <datajerk> yeah i can create a b/w image ahead of time and can overlay on the image, i do that now with inserting PPTX slides into my videos, just need to know what filters to look at to support some type of transparency and fade in/out of the image [19:40] <ac_slater> hey all I'm struggling to understand the relationship between PTS and DTS. Are there any good guides out there? [19:40] <JEEB> PTS is the Presentation Time Stamp [19:40] <Mavrik> well, there's bunch of documentation [19:40] <JEEB> and DTS is the Decoding Time Stamp [19:41] <Mavrik> but those really aren't that hard to get :) [19:41] <ac_slater> right, I get what they mean - by definition. I've read the FFMPEG docs 100 times. There are too many varying cases to really grasp it from nothing. [19:42] <ac_slater> So I guess my questions really come up when generating PTS and DTS [19:43] <ac_slater> I've seen a lot of things say "just set them to 0" ... and I dont get that [19:43] <Mavrik> uh [19:43] <Mavrik> you probably shouldn't generate them. [19:44] <Mavrik> PTS and DTS should be set by your encoder. [19:44] <ac_slater> ah right. Well, I have an encoder that doesnt really generate them for me [19:44] <Mavrik> You never should generate both unless you're doing something more advanced, but then I guess you'd get that. [19:44] <BtbN> well, at some point they have to gen generated. The encoder just passes them through. [19:44] <Mavrik> I somehow doubt you have an encoder that doesn't generate timestamps. [19:44] <Mavrik> BtbN, no, the encoder takes the timestamp of the frame [19:45] <Mavrik> BtbN, and sets PTS and DTS of output encoded content [19:45] <ac_slater> Mavrik: it's a homegrown h.264 encoder that lives in an FPGA. I have the full source and there is no timestamp (PTS or DTS) generation [19:45] <BtbN> so it passes the timestamp though. [19:45] <Mavrik> DTS is a concept that is relevant only for some formats. [19:45] <Mavrik> ac_slater, ok, does it have a delay/buffer? Or do you get encoded frame back immediately? [19:46] <Mavrik> ac_slater, which profile does it encode to? Does it use B-frames? [19:46] <ac_slater> I know it runs at a certain rate [19:46] <ac_slater> yea, b-frames and p-frames, and baseline [19:46] <ac_slater> level 3.1 [19:46] <JEEB> do you actually do it so that the decoded order is not the presentation order? [19:46] <JEEB> which is what is possible with b-pictures [19:47] <ac_slater> oh right. I remember now why DTS is important [19:47] <ac_slater> JEEB: reminded me [19:47] <ac_slater> thanks [19:47] <Mavrik> ac_slater, um [19:47] <Mavrik> Baseline does not allow B-frames. [19:47] <Mavrik> So it's either one or the other. [19:48] <ac_slater> interesting. one moment [19:48] <Mavrik> Again, you CANNOT do what you want before knowing how your encoder works. [19:48] <ac_slater> that makes sense [19:48] <JEEB> well usually you get pictures from the encoder in the decoding order, and then the encoder should somehow mark them with the coded picture's presentation timestamp [19:48] <Mavrik> Also, if your encoder doesn't give you timestamps of encoded content, you NEED to know how much delay it has :) [19:48] <Mavrik> mhm [19:49] <Mavrik> JEEB, either that or his encoder is so simple it doesn't generate B-frames and immediately returns a coded frame - and you just stamp it with same PTS/DTS [19:49] <Mavrik> but I haven't seen that around yet O.o [19:50] <ac_slater> this encoder isnt simple sadly - it's very huge haha. But yea, it has different modes. Currently I get it in `Main 3.1` with bframes [19:50] <Mavrik> well, for B-frames the encoder really has to tell you at least PTS [19:50] <ac_slater> and if it doesnt? [19:50] <JEEB> then you make it do so [19:51] <ac_slater> well thanks for the advice guys [19:51] <ac_slater> it's really helpfu [19:51] <ac_slater> helpful * [19:51] <Mavrik> ac_slater, well, you can't guess what's he giving you out [19:51] <Mavrik> ac_slater, so "it's not" is really not an option [19:51] <Mavrik> I guess you're missing part of API or somethin :) [19:52] <ac_slater> Mavrik: I guess so. I'll report back with that. [19:52] <JEEB> if it's with a constant frame rate all the time I guess in theory you could stick an annex b demuxer after it and try to have it generate something [19:52] <Mavrik> mhm [19:52] <Mavrik> but that kind of encoder design would be non-sensical really [19:53] <ac_slater> you're right it does give me PTS [19:53] <ac_slater> via wall clock :( [19:53] <Mavrik> that's ok then :) [19:55] <ac_slater> so I guess I get really confused when dealing with PTS on multiple levels... the source (encoder), and the muxer, in my case mpegts [19:58] <ac_slater> So assuming the encoder gives me a proper PTS. What if it has B-frames but doesnt give me DTS? [19:59] <JEEB> well as long as the decoder outputs things in decoding order you can just put a monotonically increasing DTS there [19:59] <ac_slater> with any start point? [20:00] <ac_slater> ie - non-relational to the PTS? [20:00] <JEEB> yup [20:00] <ac_slater> awesome [20:02] <Mavrik> ac_slater, yeah, the player only cares about difference of PTS/DTS not absolute values [20:02] <Mavrik> it only needs to know how long to wait until it has to display a frame or audio sample [20:02] <ac_slater> i seeee [20:02] <ac_slater> thanks guys [20:02] <Mavrik> that's why your PTS/DTS can wrap [20:03] <Mavrik> do make sure you handle that case ;) [20:03] Action: Mavrik grumbles something about half of ffmpeg not handling that case. [20:03] <ac_slater> Mavrik: I remember you telling me that last time. I made a note of it and realized that ffmpeg/libavformat has mechanisms to deal with that [20:04] <ac_slater> interesting [20:04] <ac_slater> how *do* you handle that? [20:05] <Mavrik> um, you just carry on :P [20:05] <ac_slater> ah ok :D [20:05] <Mavrik> but if you do stuff like reorder packets or something [20:05] <Mavrik> then it's nice to know that sometimes going from 33 bits -> 0 is OK ;) [20:06] <ac_slater> I see. Well damn. [20:06] <ac_slater> I'll deal with that when the time comes ;) [20:08] <ac_slater> so while I have you guys here. I'm trying to mux a mpegts container with multiple streams. These streams are strictly data and I wrote the encoder for the data. FFMPEG supports the type and the mpegts muxer recognizes it. It's KLVA. But since there is no ffmpeg encoder, I'm writing one. When creating an AVStream for this, how does one pick an arbitrary timebase for both the codec and stream? [20:10] <ac_slater> maybe a simple runnable code snippet is in order for this [20:13] <ac_slater> I guess this question can be summed up with: How do you pick a timebase? [20:13] <Mavrik> ac_slater, I'm pretty sure MPEG-TS wants all timebases to be 1/90000 [20:13] <Mavrik> that really depends on the format [20:13] <Mavrik> for some of them it can actually be arbitrary [20:14] <ac_slater> Mavrik: you're right about the 1/90000 [20:14] <Mavrik> but otherwise, people just use timebase so their PTS can go += 1 for each frame [20:14] <Mavrik> e.g. 1/25 for 25fps video, 1/44100 for 44100Hz audio, etc. [20:15] <ac_slater> interesting. I saw that too. But then I realized there are 2 timebases... one for the AVStream, and another for the output format [20:16] <Mavrik> yep [20:16] <Mavrik> that makes sense in a way :) [20:16] <ac_slater> it does really [20:16] <ac_slater> but when creating AVPackets to write, I cant tell how to scale the timebase [20:17] <Mavrik> um, I 'm 80% sure that av_write_packet expects the PTS and DTS to be in stream time_base [20:19] <ac_slater> ah ok. Interesting. [20:19] <Mavrik> and JEEB will slap me if I'm wrong :P [20:19] <ac_slater> this is actually pretty simple [20:20] <ac_slater> Mavrik: the docs for av_write_frame dont explicitly say [20:59] <ac_slater> last thing for bit. Most encoders take some estimated bitrate as an initialization argument. In my case, I'm determining how to set AVStream->codec->bitrate for something that is non-constant like h.264. Do you just sorta pick the upperbound? [21:05] <JEEB> ac_slater, I'm pretty sure the bit rate doesn't matter jack shit after the stuff has been encoded [21:05] <JEEB> only maxrate/bufsize should matter in case of mpeg-ts [21:06] <ac_slater> interesting. Thanks for the input [21:33] <ac_slater> JEEB: a bit ago you said ".. he could stick an annex b muxer in there to generate timestamps". Does this account for a case where the encoder is spitting out an annexb bitstream with no timestamps? [21:37] <ac_slater> JEEB: also, what exactly did you mean with maxrate/bufsize? Are these properties of avformatcontext? [23:00] <YaMoonSun> If I wanted to copy a raw .aac file into a .m4a file, what command would you recommend? [23:02] <YaMoonSun> http://pastebin.com/fbSZWAUS [23:11] Action: YaMoonSun waits patiently [23:12] <c_14> Did you try adding -bsf:a aac_adtstoasc ? [23:13] <YaMoonSun> Nope, what's that do? [23:13] <c_14> Shuffles some stuff around so that tha aac is mp4 compliant [23:14] <YaMoonSun> It came from an m4a, and I'm trying to put the same aac stream into another m4a without loss [23:15] <c_14> The filter should be lossless [23:17] <YaMoonSun> Oh? Nice, will give it a shot =/ [23:18] <YaMoonSun> Very nice, thank you very much =) [23:20] Action: YaMoonSun tips hat to c_14 [23:50] <YaMoonSun> Is it possible to convert an entire directory using one command withing making a .bat file? [23:54] <JEEBsv> you can possibly do a one-liner in cmd.exe but that would be the same as making a goddamn batch file [23:54] <JEEBsv> and would be much less readable [23:55] <YaMoonSun> Well fuck [23:55] <YaMoonSun> Cheers anyways [00:00] --- Sat Feb 14 2015

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