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June 2015

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ffmpeg-devel.log.20150619
by burek 20 Jun '15

20 Jun '15

[00:12:02 CEST] <philipl> ubitux: thanks [00:13:41 CEST] <cone-489> ffmpeg 03Andreas Cadhalpun 07master:d216b9debd98: vp9: don't retain NULL as segmentation_map [01:49:00 CEST] <cone-489> ffmpeg 03Andreas Cadhalpun 07master:590743101dc9: postproc: fix unaligned access [02:26:50 CEST] <cone-489> ffmpeg 03Michael Niedermayer 07master:76cc8582fd6f: avcodec/fft-test: fix type of cpuflags [02:26:51 CEST] <cone-489> ffmpeg 03Michael Niedermayer 07master:d8a227f2319a: avutil/adler32: Fix data type in test code [02:26:52 CEST] <cone-489> ffmpeg 03Michael Niedermayer 07master:fdbad24090e1: avutil/aes: Fix types [02:26:53 CEST] <cone-489> ffmpeg 03Michael Niedermayer 07master:2c881c7d45bc: avutil/avstring: Fix undefined shift [02:26:54 CEST] <cone-489> ffmpeg 03Michael Niedermayer 07master:d1dce1c51728: avutil/crc: Fix type of p table so its content fits without overflwoing [05:33:40 CEST] <cone-661> ffmpeg 03Michael Niedermayer 07master:bb3703a4c306: avcodec/j2kenc: YUV should not use the ICT [08:31:15 CEST] <rcombs> durandal_1707: so, it turns out BFSTM files always have little-endian ADPC tables, even when the whole rest of the file is big-endian [08:32:48 CEST] <durandal_1707> good [09:50:16 CEST] <rcombs> durandal_1707: do you have a sample without an ADPC chunk? [09:51:06 CEST] <durandal_1707> rcombs: unfortunately no [09:51:23 CEST] <rcombs> because I've just made a lavc change that would make those really easy to support [09:51:44 CEST] <durandal_1707> nice [09:53:10 CEST] <rcombs> https://gist.github.com/2ee9b4bb33f5568031db <-- look sane? [10:07:01 CEST] <rcombs> now I'm curious who actually cared about x265 on POWER [10:08:16 CEST] <durandal_1707> rcombs: looks good to me [10:09:09 CEST] <rcombs> cool, I'll send along with a few revisions to my existing patches on the topic [10:11:23 CEST] <rcombs> oh, that reminds me, did anyone else have an opinion on wm4's suggestion of using AVStream extradata for loop points, instead of adding members? [10:11:25 CEST] <rcombs> seemed sane enough to me [10:12:02 CEST] <wm4> feels better not to litter the context with even more fields [10:12:12 CEST] <wm4> side data is slightly more organized [10:12:28 CEST] <rcombs> or, yeah, side data [10:12:33 CEST] <durandal_1707> well I'm not fan of loop points so I will leave it to others [10:14:43 CEST] <rcombs> wm4: also, do you think it would make more sense to require the demuxer (or, in the muxing case, user) to specify both points, then? [10:15:01 CEST] <rcombs> (I'm imagining a struct with 2 int64_t's) [10:15:04 CEST] <wm4> not sure what you mean [10:15:16 CEST] <wm4> but yes, a single side data for both values [10:15:45 CEST] <rcombs> right now, AV_NOPTS_VALUE can be specified for the end loop point to mean "the duration of the file" [10:16:04 CEST] <cone-563> ffmpeg 03George Boyle 07release/2.7:a9b600cf3995: avcodec/flacenc: Fix Invalid Rice order [10:16:07 CEST] <wm4> this could remain as-is [10:16:31 CEST] <rcombs> alright [10:17:57 CEST] <rcombs> though, waitaminute [10:18:35 CEST] <rcombs> right now I have it on AVFormatContext because every format that supports this that I'm aware of specifies it at the container level, rather than the stream level [10:18:51 CEST] <rcombs> and it doesn't seem like it would make sense to have separate loop points for each stream [10:19:29 CEST] <rcombs> so how would that work at the AVStream level if the file had multiple streams [10:19:29 CEST] <wm4> these files are single-stream anyway, aren't they? [10:20:42 CEST] <rcombs> the formats with native support that I'm aware of are B[RFC]STM and AIFF, which are single-stream, yes [10:21:23 CEST] <rcombs> so I guess it works for the formats I'm aware of [11:06:07 CEST] <cone-563> ffmpeg 03George Boyle 07release/2.6:d9dd0cbe632e: avcodec/flacenc: Fix Invalid Rice order [11:06:08 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.6:7e5a298bce17: avcodec/hevc_ps: Only discard overread VPS if a previous is available [11:06:09 CEST] <cone-563> ffmpeg 03banastasov 07release/2.6:df408d213ad1: avcodec/dvbsubdec: Fix buf_size check in dvbsub_parse_display_definition_segment() [11:06:10 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.6:a6bdfd9b46bb: avcodec/x86/h264_weight: handle weight1=128 [11:06:11 CEST] <cone-563> ffmpeg 03Paul B Mahol 07release/2.6:81d923f14bed: avcodec/exr: fix crash caused by merge [11:06:12 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.6:ecd3cce0ec75: avfilter/x86/vf_hqdn3d: Fix register types [11:06:13 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.6:e2054a8b55ee: avcodec/mpegvideo: Factor ff_mpv_reallocate_putbitbuffer() out [11:06:14 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.6:678a189e5dd0: avcodec/mjpegenc_common: Use ff_mpv_reallocate_putbitbuffer() [11:06:15 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.6:8c5d196f5026: avformat/mov: Mark avio context of decompressed atoms as seekable [11:19:16 CEST] <cone-563> ffmpeg 03Paul B Mahol 07release/2.5:f621ae6a6dd8: avcodec/exr: fix crash caused by merge [11:19:17 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.5:e29f68cf9e33: avcodec/x86/h264_weight: handle weight1=128 [11:19:18 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.5:2d55d3483b8e: avcodec/hevc_ps: Only discard overread VPS if a previous is available [11:19:19 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.5:d4c7a1ecb8ce: avformat/mov: Mark avio context of decompressed atoms as seekable [11:19:20 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.5:2e210e19b156: avcodec/mpegvideo: Factor ff_mpv_reallocate_putbitbuffer() out [11:19:21 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.5:11f597ed8486: avcodec/mjpegenc_common: Use ff_mpv_reallocate_putbitbuffer() [11:19:22 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.5:e9502ee26ae1: avfilter/x86/vf_hqdn3d: Fix register types [11:29:40 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.4:65d61a4bbbb9: avfilter/x86/vf_hqdn3d: Fix register types [11:29:41 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.4:57d094e3e399: avformat/mov: Mark avio context of decompressed atoms as seekable [11:29:42 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.4:bd28de1b4df5: avcodec/hevc_ps: Only discard overread VPS if a previous is available [11:29:43 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.4:ff02eeafd8b3: avcodec/x86/h264_weight: handle weight1=128 [11:29:44 CEST] <cone-563> ffmpeg 03Paul B Mahol 07release/2.4:1a4cb3b29cda: avcodec/exr: fix crash caused by merge [11:35:46 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.3:86c85b6f1a8e: avfilter/x86/vf_hqdn3d: Fix register types [11:35:47 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.3:7d97587b1ae7: avformat/mov: Mark avio context of decompressed atoms as seekable [11:35:47 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.3:e1afd6ed6075: avcodec/x86/h264_weight: handle weight1=128 [11:35:48 CEST] <cone-563> ffmpeg 03Paul B Mahol 07release/2.3:17d9157c8229: avcodec/exr: fix crash caused by merge [11:42:25 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.2:85d0c7355b61: avfilter/x86/vf_hqdn3d: Fix register types [11:42:26 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.2:59a1e0cba062: avformat/mov: Mark avio context of decompressed atoms as seekable [11:47:00 CEST] <durandal_1707> michaelni: thx [11:48:51 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.1:c77d226f7116: avfilter/x86/vf_hqdn3d: Fix register types [11:48:52 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.1:a444857db318: avformat/mov: Mark avio context of decompressed atoms as seekable [11:48:53 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.1:e0537e49a45e: avcodec/x86/h264_weight: handle weight1=128 [11:54:14 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.0:4a2be7027f7e: avfilter/x86/vf_hqdn3d: Fix register types [11:59:01 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/1.2:1218e9d73dc3: avfilter/x86/vf_hqdn3d: Fix register types [11:59:02 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/1.2:581bfd9c6f3a: avcodec/x86/h264_weight: handle weight1=128 [13:17:48 CEST] <cone-563> ffmpeg 03James Almer 07master:a9af9da6311b: library.mak: Workaround SDL redefining main and breaking fate tests on mingw [13:30:46 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.7:8749b83e0be1: avcodec/libx264: Avoid reconfig on equivalent aspect ratios [13:59:44 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07master:d277b05c518a: avcodec/fic: Add skip_cursor AVOption [14:18:23 CEST] <BBB> so if I want to use special LDFLAGS for one test binary [14:18:25 CEST] <BBB> how do I do that? [14:18:37 CEST] <BBB> do we support something like TINY_SSIM_LDFLAGS = .. ? [14:34:01 CEST] <ubitux> tiny_ssim$(EXESUF): LDFLAGS += ... [14:34:09 CEST] <ubitux> ah, he left [14:35:27 CEST] <Daemon404> he waited 7 whole minutes. [15:13:27 CEST] <cone-563> ffmpeg 03Shivraj Patil 07master:ee3ef5fda2f1: avcodec/mips: MSA (MIPS-SIMD-Arch) optimizations for hpel functions [16:10:43 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07master:52e02a9e59f8: avutil/avstring: Do not print NULL [17:41:27 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07master:8edc17b639c4: avcodec/dpxenc: implement write16/32 as functions [18:18:54 CEST] <durandal_1707> rcombs: sending patches? [18:49:32 CEST] <cone-563> ffmpeg 03wm4 07release/2.7:93d076b4fde1: avformat: clarify what package needs to be compiled with SSL support [18:49:33 CEST] <cone-563> ffmpeg 03Andreas Cadhalpun 07release/2.7:250cf2d4dae0: vp9: don't retain NULL as segmentation_map [18:49:34 CEST] <cone-563> ffmpeg 03Andreas Cadhalpun 07release/2.7:bea4894d0c42: postproc: fix unaligned access [18:49:35 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.7:23614d09d511: avutil/avstring: Do not print NULL [18:49:36 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.7:157dd527008c: avcodec/dpxenc: implement write16/32 as functions [18:51:49 CEST] <BBB> no comments on my tiny_ssim_avf? :( [18:51:58 CEST] <BBB> I was hoping somebody would pick it up and actually include it [19:03:04 CEST] <durandal_1707> does it crash? [19:10:14 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07release/2.7:7cbceb16ade6: Update for 2.7.1 [19:16:39 CEST] <BBB> durandal_1707: no, it just reads two files and outputs a ssim on stdout, pretty boring stuff [19:16:52 CEST] <BBB> fairly similar to tiny_ssim I s'pose [19:18:01 CEST] <durandal_1707> what if files have different dimensions? [19:22:46 CEST] <durandal_1707> michaelni: how that code in dpxenc could crash? [19:25:06 CEST] <BBB> I suppose it should error (same for colorspaces), but I didnt bother to test/implement [19:25:18 CEST] <BBB> it could also support raw yuv with private options but I didnt do that either [19:25:24 CEST] <BBB> long todo as you can see [19:26:52 CEST] <durandal_1707> my plan is to add lavfi one with slice threading [20:07:35 CEST] <michaelni> durandal_1707, one of the arguments was doing ++ [20:08:59 CEST] <lglinskih> kierank: now my test calculates crc of a frames, but two last frames are always lost. av_read_frame() just doesn't give me two last packets. I don't know why this is happening. [20:20:55 CEST] <BBB> lglinskih: do you flush? [20:21:14 CEST] <BBB> lglinskih: as in, do you call av_decode_videoN() with NULL? [20:24:45 CEST] <lglinskih> BBB: no, no, explain it to me please ) [20:25:28 CEST] <BBB> https://ffmpeg.org/pipermail/ffmpeg-devel/2015-June/174340.html [20:25:34 CEST] <BBB> look for the function read_frame() [20:25:50 CEST] <BBB> it calls av_read_frame and then calls avcodec_decode_video2 [20:26:04 CEST] <BBB> then if av_read_frame fails (because of end of file), it calls avcodec_decode_video2 with NULL [20:26:11 CEST] <BBB> this flushes the internal queue of packets [21:06:11 CEST] <thrwaway> anybody feel like helping a newbie like myself to hunt down an issue very similar to https://trac.ffmpeg.org/ticket/2882 (ffmpeg can't save h264 stream from v4l2 webcam into file without recoding)? see the problem on a rapsberry pi with pi camera module. same problem on ffmpeg 1.2.12, ffmpeg 2.6.1 and git master. currently experimenting with st->need_parsing = AVSTREAM_PARSE_FULL vs AVSTREAM_PARSE_H [21:06:17 CEST] <thrwaway> EADERS. [21:24:34 CEST] <lglinskih> BBB: thank you!! it works ^_^ [21:24:59 CEST] <BBB> lglinskih: good! [21:25:27 CEST] <BBB> thrwaway: sounds like you need parsing yes [21:30:35 CEST] <anoop_r> how to crop 10 pixels from the left, 30 pixels from the right 20 pixels from the top and 5 pixels from the bottom borders of a video using ffmpeg [21:31:26 CEST] <Compn> anoop_r : ask in #ffmpeg [21:34:32 CEST] <cone-563> ffmpeg 03Andreas Cadhalpun 07release/2.7:d89cd16afa97: matroskadec: check audio sample rate [21:34:33 CEST] <cone-563> ffmpeg 03Andreas Cadhalpun 07release/2.7:973e67e5d2d4: matroskadec: validate audio channels and bitdepth [21:34:33 CEST] <cone-563> ffmpeg 03Andreas Cadhalpun 07release/2.7:73e7fe8e648a: configure: make makeinfo_html check more robust [21:34:35 CEST] <cone-563> ffmpeg 03Andreas Cadhalpun 07release/2.7:620197d1ffea: doc: avoid incorrect phrase 'allows to' [22:05:44 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07master:9a2fc170b10e: avcodec/j2kenc: Support writing JP2 encapsulation [22:05:45 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07master:7e2bdea764a5: avcodec/j2kenc: Use JP2 format by default [22:11:59 CEST] <thrwaway> BBB: Chaning AVSTREAM_PARSE_HEADERS to AVSTREAM_PARSE_FULL in libavdevice/v4l2.c:v4l2_read_header() indeed seem to fix issue for me. Here's a writeup with logs -> https://pastebin.mozilla.org/8837436 [22:17:21 CEST] <wm4> michaelni: can we have the 2.7.1 release already? [22:17:30 CEST] <wm4> the complaints are starting to pile up [22:36:21 CEST] <ubitux> BBB: 14:34 <ubitux> tiny_ssim$(EXESUF): LDFLAGS += ... [22:36:57 CEST] <BBB> ooh nice, ty [22:58:55 CEST] <cone-563> ffmpeg 03Michael Niedermayer 07n2.7.1:HEAD: avcodec/j2kenc: Use JP2 format by default [00:00:00 CEST] --- Sat Jun 20 2015

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ffmpeg.log.20150619
by burek 20 Jun '15

20 Jun '15

[03:29:55 CEST] <zachary> ffplay "rtmp://url/app/name live=1": Detected librtmp style URL parameters, these aren't supported by the libavformat [03:30:09 CEST] <zachary> What is wrong with this message? [03:47:09 CEST] <c_14> Well, I think the 'the' shouldn't be there. But that's grammatical nitpicking. [03:47:47 CEST] <c_14> As to why the error is occurring, probably because you're using url syntax supported by librtmp, but not by the internel rtmp protocol. [03:48:29 CEST] <zachary> c_14: Yes. The extra parameter `live=1` is not supported by ffmpeg [03:48:46 CEST] <zachary> c_14: how do i enable librtmp? [03:48:50 CEST] <c_14> --enable-librtmp [03:50:09 CEST] <zachary> c_14: hmm. i will try that. [05:36:45 CEST] <haasn> The HEVC MSA optimizations added in d6d98237 and 271195f8 are much slower for me than the ported OpenHEVC intrinsics that were present in nevcairiel's branch. Will this situation improve significantly, or should I un-apply those patches and re-apply the OpenHEVC ones if I want fast playback of HEVC on my machine? [05:44:13 CEST] <haasn> Ah, don't even need to unapply those, I can just cherry-pick the OpenHEVC stuff [10:51:13 CEST] <gurpartap1> I am facing problem while saving stream from ip camera and capturing audio from laptop microphone.there are synchronization issues in it? Please somebody help. [10:52:25 CEST] <gurpartap1> when I move close to mic the lag is different and when I am away from mic, lag is different.How to solve this problem....please help.. [10:53:11 CEST] <gurpartap1> I am fixing with -itsoffset 00:00:0.8 while close to mic ..Is there any other method [10:56:39 CEST] <BtbN> You mean when you move physicaly closer to the microphone, the audio/video offset changes? oO [10:56:44 CEST] <BtbN> That doesn't make much sense [11:13:50 CEST] <NoobSaibot> lol maybe he's moving to and fro faster than the speed of sound? I bet that would cause some clicks... [11:14:41 CEST] <jcay> only sonic boom [11:15:07 CEST] <relaxed> sonic boom mic [11:16:13 CEST] <gurpartap1> yes, when I move close to mic audio is coming before video and vice versa [11:16:44 CEST] <NoobSaibot> move more slowly and audio will line up. #relativity [11:17:13 CEST] <NoobSaibot> really, that's weird, i'm not sure what's happening with your hardware, grupartap1 [11:17:18 CEST] <BtbN> We're talking about a few cm, or do you mean something like moving a few 100m away? oO [11:19:40 CEST] <jcay> i think its related to audio intensity, not relativistic effects [11:20:43 CEST] <BtbN> Or the camera does weird stuff with proximity detection [11:22:29 CEST] <jcay> can you monitor the sound levels? it should never exceed but be close to 0db [11:23:08 CEST] <jcay> maybe you also have a booster on [11:25:46 CEST] <gurpartap1> Is there any way to synchronize audio and video with timestamp values.... [11:26:31 CEST] <BtbN> That's what's happening by default, if there are any timestamps [11:27:35 CEST] <gurpartap1> There is another situation: When I am grabbing my desktop and recording with inbuilt mic it is in sync .....But problem arises with external mic... [11:30:28 CEST] <NoobSaibot> is your cable 300,000km long? [11:30:32 CEST] <gurpartap1> BtbN: you mean ffmpeg by default synchronizes using timestamps of both audio and video? [11:31:02 CEST] <BtbN> Of course it does. The problem is getting usefull timestampf [11:31:04 CEST] <BtbN> *s [11:32:27 CEST] <gurpartap1> haha NoobSaibot. No a little less than that! [11:34:45 CEST] <gurpartap1> Is there any way I can extract these timestamps? [11:37:14 CEST] <gurpartap1> BtbN: what -vsync and -async options exactly do? [11:38:10 CEST] <BtbN> It decides how exactly audio/video it synced up [11:38:28 CEST] <BtbN> dropping frames, stretching the audio, or whatever [11:38:58 CEST] <BtbN> Your problem is your device giving you wrong timestamps, or even none at all [11:39:17 CEST] <gurpartap1> how can I check this? [11:40:35 CEST] <gurpartap1> BtbN: While doing recording debug is showing : [AVIOContext @ 0x1b4f460] Statistics: 60 seeks, 155 writeouts [11:41:25 CEST] <gurpartap1> can you look at debug info of my commandline.... [11:41:43 CEST] <BtbN> There isn't realy much you can do if your device is behaving that weird [11:44:43 CEST] <gurpartap1> BtbN: I am using ffmpeg -f alsa -ac 1 -itsoffset 00:00:00 -i default -i rtsp://192.168.0.127 -acodec libfaac -vcodec copy -s:v:0 600x480 -v debug -y -map 0:0 -map 1:0 output.mp4 [11:45:01 CEST] <gurpartap1> Do you suggest something? [12:04:59 CEST] <relaxed> gurpartap1: see if it happens when you copy the audio [12:05:43 CEST] <relaxed> also try encoding both streams [13:34:53 CEST] <hardikj> hi I am want a ffmpeg destop capture for a low cpu machine, after lots of experiment I found out mpeg2 to be the best choice when using low cpu [13:35:16 CEST] <hardikj> command I am using is ffmpeg -y -framerate 10 -f gdigrab -i desktop -vcodec mpeg2video -qscale:v 15 -r 8 -pix_fmt yuv420p test.mpg [13:35:30 CEST] <hardikj> is there any way I can improve it further? [13:36:07 CEST] <hardikj> will using using minrate or maxrate help me? [14:07:56 CEST] <DHE> hardikj: -qscale forces a certain image quality, loosely speaking. you don't normally mix and match them. but a smaller number will give better image quality. [14:08:40 CEST] <hardikj> will omitting qscale will improve cpu usage? [14:08:55 CEST] <hardikj> I am not much concerned with video quality [14:09:04 CEST] <hardikj> a medium quality will work too [14:09:11 CEST] <hardikj> DHE ^^ [14:11:12 CEST] <DHE> I rather doubt it [14:11:26 CEST] <spl33n> hello all i try to read this stream https://www.passageenseine.org/live.m3u8 with mpv , i have this issue [ffmpeg] tls: unable to negociate TLS/SSL session failed to open https..... [14:11:48 CEST] <DHE> typically you specify a bitrate, or you specify the quantizer parameter. you're doing the latter. [14:24:52 CEST] <sj> ji [14:25:04 CEST] <sj> hi can someone help me with my ffmpeg code [14:25:26 CEST] <sj> which i am running via php web page on the ffmpeg ubuntu server [14:28:14 CEST] <rds123> Hi. How do you set the deadline in VP8 to realtime using libavcodec? I have AVCodecContext for my VP8 encoder but I'm not sure what to set if I want the 'deadline' to 'realtime'. Using the command line interface you set '-deadline realtime'. [14:29:34 CEST] <gurpartap1> relaxed: My problem is solved to some extent but when video packets are dropped the audio lags behind... [14:29:40 CEST] <rcombs> rds123: it's an AVOption [14:32:22 CEST] <gurpartap1> relaxed: how can I sync them so that when video packets dropped then audio packets should also dropped.. [14:32:36 CEST] <gurpartap1> Is there any option? [14:39:47 CEST] <rds123> rcombs: So if I just do av_opt_set_from_string(myCodecContext, "deadline=realtime", NULL, "=",":"); that should the deadline to realtime? [14:41:03 CEST] <rcombs> rds123: "av_opt_set" is probably a better choice for your use-case [14:41:05 CEST] <rcombs> but yes [14:42:35 CEST] <rds123> rds123: Oh yes, I didn't see that one. Thank you [15:34:50 CEST] <hardikj> DHE, do you have any idea on how to capture raw stream from gdigrab [15:35:02 CEST] <hardikj> I think that should not take any cpu at all [15:35:11 CEST] <hardikj> I can deal with the file size [15:35:38 CEST] <DHE> no, I don't use windows [15:35:42 CEST] <DHE> for me it's x11grab [15:36:00 CEST] <BtbN> i don't think you know how massive unencoded video gets [15:36:51 CEST] <BtbN> 1080p is ~8MB per frame, so over 450MB/s of data [15:36:57 CEST] <BtbN> you need a damn good SSD to store that [15:38:56 CEST] <DHE> mpeg2 is usually a good choice. motion jpeg might be worth investigating... [15:39:13 CEST] <hardikj> BtbN, I am aiming to send the rawdata to another system and re-encode there. the systems where I will be running ffmpeg screen capture have low cpu and I dont want to intercept the ops running there [15:40:06 CEST] <hardikj> so I wont store the files anywhere, I will stream it to a cluster of machines and then send it to s3 after reencoding [15:40:39 CEST] <BtbN> so you have a 10Gb/s connection between them? [15:41:14 CEST] <BtbN> cause that raw data is way too much for a normal gigabit lan [15:41:20 CEST] <hardikj> I will capture low quality video, maybe? [15:41:43 CEST] <BtbN> Then why not use some cheap lossless codec? [15:41:51 CEST] <BtbN> Propably more efficient than sending that much data [15:42:21 CEST] <hardikj> I tried mpeg2 it is using 30% of my cpu, which makes the tasks running on the machine slow [15:42:39 CEST] <BtbN> mpeg2 is everything but lossless [15:42:55 CEST] <hardikj> I can afford 10-15% cpu usage, not more than that [15:43:06 CEST] <BtbN> utvideo or h264 with a high bitrate/actualy lossless [15:43:11 CEST] <BtbN> Might even use a hardware encoder [15:43:55 CEST] <hardikj> BtbN, you mean utvideo will be better than mpeg2 in terms of cpu usage? [15:44:06 CEST] <BtbN> no idea, try it? [15:44:30 CEST] <hardikj> I already tried h264, mpeg2 use low cpu [15:44:31 CEST] <hardikj> ok,thanks for the suggestion [15:44:31 CEST] <hardikj> let me try utvideo [15:45:38 CEST] <hardikj> BtbN, btw is there any way to capture low quality video and send its raw package using ffmpeg? just in case utvideo didn't work? [15:45:50 CEST] <BtbN> sure, just don't encode them [15:46:10 CEST] <BtbN> But i wouldn't be surprised of scaling it down uses more cpu than some cheap lossless codec [15:54:12 CEST] <rngz> Hello [15:56:42 CEST] <rngz> I am trying to build FFmpeg with libx264 using the MSVC compiler for Windows 64-bit. "CC=cl ./configure --prefix=../../build --disable-cli --cross-prefix=x86_64-w64-mingw32- --host=x86_64-w64-mingw32 --enable-static --enable-win32thread" builds a libx264.lib file, but FFmpeg configuration is complaining that it can't find libx264. [15:57:22 CEST] <hardikj> BtbN, ffmpeg -y -framerate 8 -f gdigrab -i desktop -vcodec utvideo -r 8 -pix_fmt yuv420p output.avi [15:57:33 CEST] <hardikj> takes 30% CPU [15:57:35 CEST] <hardikj> :( [16:04:12 CEST] <ldiamond1> hardikj: seems like you're encoding a video. [16:04:44 CEST] <ldiamond1> I'm not familiar with that command, you're grabbing 8 frames per s from your desktop? [16:06:05 CEST] <ldiamond1> I'd be surprised that ffmpeg actually needs all that processing power for that. [16:06:12 CEST] <ldiamond1> maybe try running it through nice? [16:10:06 CEST] <ldiamond1> Now my question... Is it possible to change ffmpeg "settings" during a live stream? More specifically, I add an overlay of my webcam on the stream from my desktop, I want to be able to move the camera around without stopping and restarting ffmpeg. [16:11:18 CEST] <JEEBsv> rngz: how are you trying to make sure ffmpeg finds libx264? pkg-config? [16:13:43 CEST] <rngz> JEEBsv: I'm just setting the --extra-ldflags and --extra-cflags in the ffmpeg configure, I've also tried adding the path to the x264 lib/headers to $LIB and $INCLUDE [16:14:55 CEST] <JEEBsv> also that x264 configure line sure looks like mingw-w64 rather than MSVC with the mingw-w64 stuff in there... [16:15:12 CEST] <JEEBsv> rngz: not even sure how well the x264 pkg-config files would work, of course... [16:15:21 CEST] <JEEBsv> what's the result with config.log? [16:15:24 CEST] <rngz> I had to mess with that because I was having trouble getting x264 to compile with MSVC [16:15:47 CEST] <JEEBsv> uhh, which MSVC are you using? I think 2013 update 2 or so is where the official support was added [16:16:00 CEST] <rngz> 2013 update 4 [16:16:07 CEST] <JEEBsv> ok, that should in theory work [16:16:32 CEST] <JEEBsv> also mixing static mingw-w64 libs with MSVC can be "fun" [16:16:32 CEST] <rngz> my current ffmpeg configure looks like it's failing without creating a config.log [16:16:47 CEST] <JEEBsv> then it got pretty much nowhere :P [16:16:49 CEST] <rngz> yes, I am certainly having fun [16:17:03 CEST] <JEEBsv> because it should start writing to the log file right away [16:18:19 CEST] <rngz> Running "CC=cl ./configure --prefix=../../build --disable-cli --enable-static --enable-win32thread" results in basically "Unknown system, edit the configure" [16:18:28 CEST] <rngz> (on libx264) [16:19:48 CEST] <JEEBsv> if you have the MSVC tools in PATH a basic configure should work... http://git.videolan.org/?p=x264.git;a=commit;h=6fbbb5b0c05a1d95cbd6efa7f018… [16:19:55 CEST] <JEEBsv> you should see config.log there as well [16:21:25 CEST] <rngz> "CC=cl ./configure" results in a 1 line config.log, "DIED: Unknown system , edit the configure" [16:21:38 CEST] <JEEBsv> how old x264? [16:21:39 CEST] <rngz> cloned the latest master from git.videolan.org... [16:23:04 CEST] <rngz> I'm doing this using MSYS2... not sure if that's a problem [16:23:17 CEST] <JEEBsv> msys2 is probably the least retarded of the msys family [16:23:40 CEST] <JEEBsv> but yeah, I just had an old msys setup with the MSVC stuff in PATH and it worked for me [16:24:32 CEST] <JEEBsv> oh... [16:24:39 CEST] Action: JEEBsv checks the configure script [16:28:28 CEST] <rngz> it looks like my $host_os is just empty... [16:31:00 CEST] <JEEBsv> see what config.guess outputs? [16:31:03 CEST] <rngz> ./config.guess is unable to guess the system type [16:31:07 CEST] <JEEBsv> yeah [16:33:22 CEST] <rngz> does msys do something different, and can I just hand jam the values it returns if you run it? [16:34:34 CEST] <JEEBsv> hmm [16:36:01 CEST] <JEEBsv> ooh [16:36:35 CEST] <JEEBsv> I guess you have an msys compiler as the one without a prefix? [16:36:56 CEST] <JEEBsv> which is correct technically I'd say [16:37:00 CEST] <rngz> ... idk? [16:37:33 CEST] <JEEBsv> gcc -v [16:37:37 CEST] <JEEBsv> seee what target that is for [16:37:44 CEST] <chungy> rngz: the ./configure script won't work for MSVC [16:37:53 CEST] <JEEBsv> chungy: oh it does [16:38:00 CEST] <JEEBsv> also at this point we're at the wrong channel [16:38:06 CEST] <JEEBsv> since we're starting off at x264 :P [16:38:11 CEST] <chungy> it's geared towards making builds that run on compilers that are like POSIX (gcc, clang, tcc, etc) [16:38:21 CEST] <JEEBsv> chungy: http://fate.ffmpeg.org/report.cgi?time=20150619060858&slot=x86_64-msvc12-wi… [16:38:28 CEST] <JEEBsv> it has support for MSVC just fine [16:38:32 CEST] <JEEBsv> at least the sane versions [16:38:58 CEST] <JEEBsv> there's even a semi-new part in the documentation regarding it [16:40:54 CEST] <chungy> hmm. taking a look, it seems it's not actually an autoconf script :o [16:40:55 CEST] <rngz> Yeah, I've built ffmpeg with --toolchain=msvc just fine, it's just this part about building libx264 and linking it into the ffmpeg build that I'm stuck on [16:41:01 CEST] <chungy> i wasn't aware of that [16:41:22 CEST] <JEEBsv> chungy: also Daemon404 has noted me that even autotools supports it these days, but I have never made sure :P [16:42:01 CEST] <rngz> I'll move to #x264 until I get that part resolved [16:42:06 CEST] <JEEBsv> yeah [17:03:12 CEST] <BtbN> ldiamond1, no [17:03:16 CEST] <BtbN> use OBS for stuff like that [17:11:49 CEST] <JEEBsv> rngz: after you've cleaned and clean-installed libx264 to your prefix, take a look if you can configure ffmpeg without any libraries first with an MSVC configuration a la the one on that FATE run I linked [17:12:07 CEST] <JEEBsv> also I remember that --extra-*flags with MSVC were wonky [17:12:19 CEST] <JEEBsv> I think I was using some environment variable instead in MSVC builds [17:12:34 CEST] <rngz> I can configure it without libx264 just fine [17:12:55 CEST] <rngz> and build [17:12:59 CEST] <rngz> with msvc no problem [17:14:06 CEST] <JEEBsv> ok, so yeah... I have a hunch that it could be similar issue like I had with --extra-*flags [17:14:18 CEST] <rngz> you might be on to something with having to set environmental variables manually.. I just don't know which ones [17:14:39 CEST] <JEEBsv> try adding your libx264 prefix's lib and include in front of the LIB and INCLUDE env variables [17:14:52 CEST] <JEEBsv> like export LIB="C:\JEEB\msys\msvc-zlib\lib;${LIB}" [17:15:09 CEST] <JEEBsv> and export INCLUDE="C:\JEEB\msys\msvc-zlib\include;${INCLUDE}" [17:15:33 CEST] <rngz> OH [17:15:36 CEST] <JEEBsv> this is for zlib, which you probably guessed [17:15:46 CEST] <rngz> do those paths have to be C:\xyz format? [17:15:50 CEST] <JEEBsv> yes [17:15:53 CEST] <rngz> because I've been using their paths in the msys env [17:16:11 CEST] <rngz> like /home/user/x264/build/include or whatever [17:16:13 CEST] <JEEBsv> msys(2) does some translation of those, but I have no idea if it's enough [17:16:16 CEST] <rngz> ahhhh, that might be it [17:16:25 CEST] <JEEBsv> and I just remember --extra-*flags having some wonky issues [17:16:32 CEST] <JEEBsv> because those params were the ones I tried first, of course [17:18:37 CEST] <rngz> bah, still not found [17:19:05 CEST] <JEEBsv> config.log :P [17:19:20 CEST] <rngz> empty... something is calling the require() function in the configure script to error out [17:19:42 CEST] <JEEBsv> it shouldn't empty config.log [17:19:49 CEST] <rngz> there are some variables in that function, name: libx264, header: x264.h, func: x264_encoder_encode [17:20:55 CEST] <rngz> I'll have to come back to this unfortunately... oh well. Thanks for digging in this far with me. [17:22:22 CEST] <JEEBsv> I would guess that building ffmpeg with MSVC together with x264 isn't something that's often excersized ;) [17:22:32 CEST] <JEEBsv> since those people generally want to use ffmpeg under LGPL [17:48:02 CEST] <JEEBsv> rngz: I get a config.log created just fine [17:48:34 CEST] <JEEBsv> http://up-cat.net/p/1b150780 [17:53:41 CEST] <JEEBsv> and by copying libx264.lib => x264.lib [17:53:43 CEST] <JEEBsv> it built [17:54:13 CEST] <JEEBsv> so yeah, as useless as it is, I now have ffmpeg with libx264 built in built with MSVC [17:55:13 CEST] <JEEBsv> and lo and behold [17:55:17 CEST] <JEEBsv> it even works [18:38:50 CEST] <thrwaway> I'm having issues streaming H264 from a Raspberry Pi camera using the v4l2 device in ffmpeg. It seems like the H264 decoder gets fed mostly junk data. It's similar to the report at https://trac.ffmpeg.org/ticket/2882 (ffmpeg can't save h264 stream from v4l2 webcam into file without recoding), which is marked as closed. [18:39:26 CEST] <thrwaway> I've verified the problem is present in ffmpeg 1.2.12, 2.6.1 and git master. [18:43:07 CEST] <thrwaway> here's full output from ffmpeg showing the issue: https://pastebin.mozilla.org/8837411 [18:45:50 CEST] <proserpine-> how can i save the output from blackdetect filter to a file? command line is ffmpeg -i video.mp4 -vf blackdetect=d=0.1:pix_th=0.1 -f rawvideo -y nul [18:46:56 CEST] <proserpine-> replacing nul by out.txt produces garbage obviously [18:48:56 CEST] <proserpine-> nvm found the solution, redirect stderr to stdout and grep for blackdetect [18:48:59 CEST] <BtbN> 2>somefile [21:12:54 CEST] <anoop_r> how to resize video without maintaing aspect ratio using ffmpeg ? [21:13:12 CEST] <c_14> -vf scale=someheight:somewidth [21:14:47 CEST] <anoop_r> but this maintaing display aspect ratio [21:16:13 CEST] <c_14> -vf scale=someheight:somewidth,setsar=1 [21:17:32 CEST] <anoop_r> thanks c_14 [21:24:27 CEST] <anoop_r> how to crop 10 pixels from the left, 30 pixels from the right 20 pixels from the top and 5 pixels from the bottom borders of a video using ffmpeg [21:36:00 CEST] Last message repeated 1 time(s). [21:36:29 CEST] <anoop_r> crop=in_w-2*10:in_h-2*20 this will crop 10 pixels from the left and right borders, and 20 pixels from the top and bottom borders [21:36:46 CEST] <anoop_r> but how to crop from all sides with different values [22:02:50 CEST] <feliwir> hey, ffmpeg recognizes some vp6 files as aac files [22:10:35 CEST] <feliwir> how long does it take a pr to be accepted? (or can i even do pr's? [22:10:39 CEST] <Lokie> hello is there a way to get a similar output from my static ffmpeg? http://johnvansickle.com/ffmpeg/ [22:10:54 CEST] <Lokie> -version or -buildconfig doesn't seem to be exactly similar [22:13:09 CEST] <durandal_1707> feliwir: you can open bug reports [22:13:38 CEST] <feliwir> durandal_1707, wouldn't it be faster to fix it myself? [22:19:32 CEST] <durandal_1707> if you know it and post a patch [23:54:15 CEST] Action: _DragonsLordKERN is away: AWAY [00:00:00 CEST] --- Sat Jun 20 2015

1 0

ffmpeg-devel.log.20150618
by burek 19 Jun '15

19 Jun '15

[01:02:37 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07master:3a6a8f6ee1cb: avcodec/ffv1enc: fix bps for >8bit yuv when not explicitly set [02:51:44 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07master:08326e73a234: avcodec/jpeg2000dec: Continue decoding reslevels even when empty ones are encountered [04:54:50 CEST] <rcombs> anyone know if Paul B Mahol (onemda) is on Freenode? [04:58:00 CEST] <jamrial> yes, he's durandal_1707 [04:59:25 CEST] <rcombs> ah [04:59:26 CEST] <rcombs> durandal_1707: ping [05:00:09 CEST] <durandal_1707> listening [05:02:13 CEST] <rcombs> so, apparently the 'SEEK' chunk in BFSTM is the equivalent to the ADPC chunk in BRSTM [05:02:58 CEST] <rcombs> so I've removed a bunch of the extradata and related code from my BFSTM demuxer in favor of using that, but apparently I'm doing something quite wrong here [05:03:51 CEST] <rcombs> I was hearing some small popping sounds at what seem to be packet boundaries before that change; now they've just gotten worse instead of going away like I was hoping they would [05:04:04 CEST] <rcombs> mind taking a look? [05:05:25 CEST] <rcombs> https://gist.github.com/d7cacaccf741c8eebfeb <-- also I added little-endian support for BCSTM (the 3DS variant [the only difference is endianness and the magic string]) [05:07:08 CEST] <durandal_1707> there are brstm files with adpcm but without adpc chunk, they are currently unsupported. I think bit-exact seeking is not possible at all even with adpc chunk [05:08:06 CEST] <rcombs> vgstream doesn't seem to use it at all [05:09:04 CEST] <durandal_1707> you hearing clicks when seeking or always? [05:09:10 CEST] <rcombs> always [05:09:49 CEST] <rcombs> not present in BRSTMs; before I made the SEEK change I assumed it was just part of the track as they were pretty quiet [05:10:04 CEST] <rcombs> but after the change I'm pretty sure it's an error on my part [05:12:24 CEST] <rcombs> samples at http://109.163.233.168/Wii%20U/Splatoon.rar [05:25:09 CEST] <durandal_1707> perhaps seek table is used only when seeking, but that doesn't explain quiter clicks [05:25:48 CEST] <rcombs> yeah :/ [05:26:14 CEST] <durandal_1707> I would try listening vgmstream decode and compare [05:29:03 CEST] <rcombs> good plan [05:30:06 CEST] <durandal_1707> maybe there is bcstm muxer code somewhere on web [05:30:49 CEST] <durandal_1707> I'm sure encountered brstm one [05:34:11 CEST] <jamrial> if bcstm is only used on WiiU games then i doubt it. there's no homebrew/modding scene for that console yet afaik, unlike the Wii [05:34:22 CEST] <jamrial> so a muxer is pointless for the time being [05:35:55 CEST] <rcombs> jamrial: BCSTM is for 3DS [05:36:00 CEST] <rcombs> which does have a modding community [05:36:03 CEST] <jamrial> oh, i see [05:36:32 CEST] <rcombs> and AFAICT the only difference between BCSTM (3DS) and BFSTM (WiiU) is endianness and the magic string [05:37:12 CEST] <rcombs> might actually change it to only 1 demuxer for both of those [05:38:13 CEST] <rcombs> durandal_1707: hmm, vgmstream's output (piped to ffplay) sounds fine [05:40:04 CEST] <durandal_1707> maybe their decoder does something different [05:40:40 CEST] <rcombs> only appreciable difference I've seen is a ` + 1024` [05:43:31 CEST] <rcombs> or& wait [05:43:35 CEST] <durandal_1707> I think it doesn't use table at all for decoding, maybe that code caused clicks [05:43:37 CEST] Action: rcombs traces things [05:44:24 CEST] <rcombs> they do use the table from the INFO header; just not the initialization values from ADPC/SEEK [05:46:20 CEST] <rcombs> they have better support for sample counts not divisible by 14, which I'm planning on fixing once I'm done with this stuff [05:46:55 CEST] <rcombs> (just consuming the extra data silenced an error but now that I look closer it does indeed leave a few samples undecoded) [05:50:06 CEST] <rcombs> there _is_ some extra data at the end of each channel in the last block that should be ignored [05:50:14 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07master:d1050d995061: ffmpeg: Free last_frame instead of just unref [05:50:20 CEST] <durandal_1707> I would compare adpc with seek of same music from brawlcustommusic as they allow downloading in brstm, bfstm and bcstm [05:53:49 CEST] <rcombs> looks like they always write SEEK tables entirely composed of 0s [05:55:13 CEST] <durandal_1707> hmm so vgmstream doesnt have quiter clicks? [05:55:49 CEST] <rcombs> lemme see [05:56:30 CEST] <rcombs> with ffmpeg on the brawlcustommusic track I hear quiet clicks in both BFSTM and BRSTM [05:57:13 CEST] <rcombs> none with vgmstream [05:59:32 CEST] <rcombs> lemme try commenting the initialization code [06:02:54 CEST] <rcombs> :/ I think it's something other than the ADPCM initialization [06:03:28 CEST] <rcombs> with the initial samples (from the seek table) ignored, I'm still hearing quiet clicks in brawlcustommusic's BFSTM, but not their BRSTM [06:24:42 CEST] <philipl> ubitux: Can you review Niklesh's latest changes? [06:25:14 CEST] <philipl> I've been going over them with him by email so I think they're decent at this point, but extra eyes would be a lot of help. [06:27:13 CEST] <rcombs> oh, btw, I actually do have a 10-channel sample [06:27:24 CEST] <rcombs> it's quite obnoxious to actually play [06:27:49 CEST] <rcombs> have to manually specify a channel layout [06:29:53 CEST] <rcombs> and I think it's actually 5 variations on a 2-channel track [07:42:59 CEST] <rcombs> durandal_170: turns out I was missing an extra 24-byte offset to the data start position that BFSTM has with ADPCM_THP for some reason [07:57:23 CEST] <jamrial> rcombs: coeff tables maybe? [07:59:08 CEST] <jamrial> ast (another format that supports that adpcm codec) has 24bytes per packet/frame storing that [08:20:59 CEST] <rcombs> jamrial: it's just once, not per-packet [08:21:06 CEST] <rcombs> and the coefficient tables are elsewhere [08:22:05 CEST] <rcombs> maybe it's meant to store initial state values, but I read those out of the SEEK chunk now, and they're always 0 for the first packet in the samples I have [09:57:29 CEST] <rcombs> michaelni: could you add http://puu.sh/iti0d/bcf75b6960.bcstm at fate-suite/bfstm/loz-mm-mikau.bcstm [10:08:43 CEST] <rcombs> http://puu.sh/itijr/a27e2aab46.png why yes, Adobe, that _is_ the full set of your products that I'd be interested in right now [10:23:26 CEST] <nevcairiel> it knows you so well [10:31:25 CEST] <ubitux> philipl: ok [10:37:40 CEST] <wm4> " [10:37:40 CEST] <wm4> Could you add multithreading to FLAC encoding? Maybe, openCL would be help common users too. flaCCL, FlaCuda? are multithreaded and have better compression than that one of ffmpeg. Could you include them into ffmpeg?" [10:37:42 CEST] <wm4> pain and misery [10:38:46 CEST] <nevcairiel> does flac use inter-compression, or is every audio frame independent? [10:38:47 CEST] <BtbN> A CUDA based encoder provides better compression, wat? [10:39:18 CEST] <nevcairiel> if its independent, frame based encoding is probably even possible without too much of a hassle [10:39:27 CEST] <nevcairiel> frame based mt-encoding that is [10:39:28 CEST] <thardin> what, they have one huge wav they need flac encoded or something? [10:39:32 CEST] <wm4> I know we actually have multithreaded decoding [10:39:36 CEST] <wm4> for a few lossless codecs [10:39:38 CEST] <wm4> durandal_170: ping [10:40:03 CEST] <nevcairiel> we also have the frame thread encoder, which only does intra-only image formats [10:40:09 CEST] <nevcairiel> but no reason it couldnt possibly do audio as well [10:40:17 CEST] <thardin> because otherwise someone needs to tell them about xargs -P [10:40:37 CEST] <thardin> or parallell [10:41:06 CEST] <BtbN> Is the flac algorithm even easily parallelizable? [10:41:26 CEST] <nevcairiel> well these flac opencl and cuda encoders are an actual thing [10:41:26 CEST] <durandal_170> wm4: pong [10:41:29 CEST] <nevcairiel> so maybe? [10:41:43 CEST] <BtbN> Only Cuda, can't find an OpenCL one [10:42:06 CEST] <BtbN> Oh, FlaCuda uses OpenCL [10:42:09 CEST] <nevcairiel> BtbN: http://www.cuetools.net/wiki/FLACCL [10:42:12 CEST] <nevcairiel> they renamed it later [10:44:16 CEST] <nevcairiel> not sure how accurate these speed graphs still are, flac got a bunch of SIMD since then [10:44:24 CEST] <nevcairiel> reference flac that is [11:00:52 CEST] <durandal_170> flac and wavpack frame mt is possible, but I haven't yet looked how easy it would be to do [11:08:56 CEST] <durandal_170> rcombs: are you writing patches for bfstm? [11:09:12 CEST] <rcombs> durandal_170: just sent a pile of them [11:12:12 CEST] <rcombs> clicking is fixed, seeking works, loop points work, BCSTM decoding works [11:31:56 CEST] <rcombs> durandal_170: http://puu.sh/itlH7/4c27fbd954.bfstm [11:35:19 CEST] <rcombs> durandal_170: and there are indeed files which the block size change fixes [11:35:25 CEST] <michaelni> rcombs, added bfstm/loz-mm-mikau.bcstm [11:35:32 CEST] <rcombs> michaelni: thanks [11:39:58 CEST] <durandal_170> rcombs: but overflow is possible so block size check need updating [11:40:07 CEST] <rcombs> yup, tweaking now [12:22:50 CEST] <nevcairiel> 1 hour flac has been encoding for days on him? what kind of crap system is that? =P [12:24:03 CEST] <xtemp09> 12-core Intel Xeon cluster [12:24:25 CEST] <nevcairiel> every core running at 1mhz? [12:24:34 CEST] <nevcairiel> encoding an 1-hour flac takes minutes at best [12:24:48 CEST] <xtemp09> ffmpeg -i source.wav -lpc_passes 6 -lpc_type 3 -compression_level 9 -exact_rice_parameters 1 -multi_dim_quant 1 output.flac [12:24:53 CEST] <xtemp09> that was the command [12:25:01 CEST] <xtemp09> source.wav ~1 Gb [12:25:09 CEST] <xtemp09> ~60 minutes long [12:26:01 CEST] <nevcairiel> channels? sample rate? [12:26:05 CEST] <xtemp09> 2.8 GHz [12:26:07 CEST] <xtemp09> Xeon [12:26:22 CEST] <xtemp09> I'm not sure [12:26:34 CEST] <xtemp09> about the channels [12:34:01 CEST] <xtemp09> and sampling rate. [12:37:14 CEST] <wm4> "Trac thinks your submission might be Spam." [12:38:11 CEST] <nevcairiel> did you try to troll again [12:38:55 CEST] <wm4> successfully, fflogger is just slow [12:49:21 CEST] <cone-489> ffmpeg 03Shivraj Patil 07master:98eb1ac90127: avcodec/mips: MSA (MIPS-SIMD-Arch) optimizations for qpel functions [14:19:13 CEST] <rcombs> nobody apply my patch series just yet, I've got a few more changes to make [14:52:29 CEST] <cone-489> ffmpeg 03Michael Niedermayer 07master:7b1c03aa744f: avcodec/libx264: Avoid reconfig on equivalent aspect ratios [14:59:58 CEST] <durandal_170> rcombs: clicks with bfstm that was not previously present [15:00:28 CEST] <rcombs> durandal_170: yeah, it's due to reading the seek table :/ [15:00:57 CEST] <rcombs> so now I'm trying to work what the difference between the seek table and reality is [15:01:15 CEST] <rcombs> funny part is, the ones in my BCSTM samples work fine [15:02:30 CEST] <rcombs> there are some differences between the calculated values and the ones in the seek table for those as well, but they're minor [15:03:24 CEST] <rcombs> whereas with the BFSTM samples they seem to be completely out-of-sync [15:05:09 CEST] <rcombs> durandal_170: I'm using this to examine the differences between the calculated values and the ones from the seek/adpc chunk: https://gist.github.com/8ca74ca061dc8196d0f8 [15:08:27 CEST] <ubitux> i still have weird shit while trying to use ff vda; probably on my side, but it seems to be decoding the same frames all over again randomly (i have about 5 differents data checksum, appearing randomly) [15:08:37 CEST] <ubitux> i think i disabled the threading properly, but maybe not [15:09:15 CEST] <ubitux> i guess this symptom doesn't say anything to anyone? [15:15:21 CEST] <ubitux> the pixbuf addresses changes though [15:34:53 CEST] <ubitux> ok that's because i requested rgb32 [15:35:01 CEST] <ubitux> somehow it works with yuv [15:35:14 CEST] <ubitux> maybe something is broken in ff, i need to check [15:42:06 CEST] <nevcairiel> ff itself only requests native preferred formats [15:42:15 CEST] <nevcairiel> Might be rgb problem. [15:49:28 CEST] <ubitux> yeah [15:49:38 CEST] <ubitux> it probably doesn't handle properly the !planar ones [15:49:44 CEST] <ubitux> i'll investigate [15:50:01 CEST] <ubitux> anyway, got it working :p [17:51:10 CEST] <wm4> michaelni: does this ring a bell? "mjpeg: unsupported coding type (ca)" [17:51:14 CEST] <wm4> michaelni: do you want a sample? [17:51:50 CEST] <wm4> michaelni: well, if you do, it's here https://www.dropbox.com/s/ho62m5py3kqk4km/01%20-%20Cider%20Time.mp3?dl=1 [17:51:56 CEST] <wm4> (yep, it's the cover art picture) [17:52:14 CEST] Action: Daemon404 barfg [17:57:50 CEST] <alexvf> hi [17:58:55 CEST] <alexvf> is there a way of calculating the expected audio dts/pts that does not need sample_rate and frame_size? [17:59:44 CEST] <alexvf> i have a ts lacking pts for some audio packets and i don't have sample_rate or frame_size (they are equal to 0) [18:01:10 CEST] <nevcairiel> Audio is trivial, after decoding you will know the exact length of the frame anyway [18:01:21 CEST] <nevcairiel> So just interpolate with that [18:01:54 CEST] <alexvf> but how, if i don't have sample_rate? [18:04:44 CEST] <wm4> do it after decoding [18:05:14 CEST] <nevcairiel> If you don't have sample rate after decoding then your audio is not usable for anything [18:05:42 CEST] <kierank> it's like a video without knowing the resolution [18:08:00 CEST] <alexvf> mmm, i understand [18:08:25 CEST] <alexvf> the problem is that i need to interpolate in the demux [18:08:37 CEST] <alexvf> and i would only get that information in the dcoder [18:09:06 CEST] <kierank> then buffer one frame [18:09:09 CEST] <wm4> why do you need to? [18:09:51 CEST] <alexvf> one frame is not enough, in the sample video i have the pattern is 1 audio packet with pts, 5 without pts [18:10:10 CEST] <kierank> it's an mpeg-ts? [18:10:17 CEST] <alexvf> yep [18:10:29 CEST] <kierank> you need to use a parser for that [18:10:43 CEST] <kierank> that's what ffmpeg does [18:12:02 CEST] <alexvf> do you mean the ffmpeg program? [18:12:20 CEST] <alexvf> because i'm already using libavformat and libavcodec [18:12:44 CEST] <Daemon404> if you are decoding anyway, why do you need packet timestamps [18:12:48 CEST] <Daemon404> get the frame timestamp [18:15:29 CEST] <alexvf> Daemon404: i guess i could, and now i'm wondering why do i need it in the demux ... [18:15:52 CEST] <cone-489> ffmpeg 03Michael Niedermayer 07master:57078e4d255a: avcodec/hevc_ps: Only discard overread VPS if a previous is available [18:26:45 CEST] <wm4> is there an avio discard protocol? [18:27:12 CEST] <wm4> similar to writing to /dev/null [18:34:38 CEST] <alexvf> Daemon404: the frame pts has no value, is that possible? [18:39:21 CEST] <kierank> then you have to wait for one with a valid pts [18:40:06 CEST] <Daemon404> there is also av best effort timestamp [18:40:07 CEST] <Daemon404> or w/e [18:40:09 CEST] <Daemon404> this is what teh cli uses. [18:41:29 CEST] Action: alexvf wonders how vlc plays this damn file ... [18:41:44 CEST] <wm4> uh [18:41:49 CEST] <wm4> missing audio timestamps are not hard [18:42:01 CEST] <wm4> and in this case, you interpolate the missing PTS [18:42:04 CEST] <wm4> after decoding [18:42:15 CEST] <wm4> which is trivial, because you have the samplerate and sample count [18:43:45 CEST] <alexvf> then, why libavcodec cannot do it? i'm getting no pts even after decoding and it supposedly has all that info [18:45:29 CEST] <kierank> mpegts allows this [18:46:24 CEST] <alexvf> i read somewhere that it allows intervals of 700ms could it be? [18:46:28 CEST] <Daemon404> im thinking he wants to know why lavc doesnt interpolate for him. [18:46:53 CEST] <alexvf> but i never found such a file [18:48:21 CEST] <cone-489> ffmpeg 03Michael Niedermayer 07master:622f1468c92c: avformat/id3v2: detect PNG by header instead of mime [18:48:49 CEST] <wm4> alexvf: libavcodec doesn't mess with timestamps, and shouldn'r [18:48:53 CEST] <wm4> *shouldn't [18:49:37 CEST] <alexvf> ok, it's fine but i'm definetely doing something wrong because even av_frame_get_best_effort_timestamp() return no pts [18:49:45 CEST] <wm4> michaelni: can you apply this patch I wrote for stripping spaces from id3v2 cover art picture descriptions? a user keeps bothering me about it [18:49:47 CEST] <Daemon404> wm4, yeah that's lavf's job [18:49:48 CEST] Action: Daemon404 runs [18:49:53 CEST] Action: wm4 stabs Daemon404 [18:50:09 CEST] <wm4> alexvf: no, it's not wrong [18:50:12 CEST] <wm4> alexvf: that's your job [18:50:21 CEST] <wm4> it comes with libavcodec being a low level API and all [18:51:16 CEST] <alexvf> wm4: yep, i meant i may be doing it wrong, because vlc can play this file and my program can't ;) [18:52:04 CEST] <wm4> alexvf: well, just interpolate the timestamps and you should be fine [18:52:36 CEST] <alexvf> time to dive into vlc code, thank you for the help guys [18:55:28 CEST] <kierank> vlc has it's own demux [18:55:33 CEST] <kierank> and parser/framer [18:55:36 CEST] <kierank> that does all that [18:55:38 CEST] <wm4> I'm not sure if the vlc code is the best to learn from, its lavc usage tends to be overcomplicated [18:56:16 CEST] <wm4> looking at ffplay.c might be better, although ffplay.c is awful (all over the place) [19:07:37 CEST] <lglinskih> kierank: I'm stuck on the stage of computing framecrc of a frame. Muxer framecrc works with a packet, I have a frame. Is it better to write my own function which works with frames? [19:14:30 CEST] <michaelni> lglinskih, you dont need to implement something bitexactly identical to framecrc [19:14:52 CEST] <michaelni> any reasonable checksum should be fine [19:16:37 CEST] <cone-489> ffmpeg 03wm4 07master:d4007d176313: id3v2: strip trailing spaces from APIC tag [19:18:05 CEST] <michaelni> lglinskih, you can use av_image_get_linesize() to find the number of "active" bytes in a line for an image [19:23:21 CEST] <michaelni> lglinskih, it might be easier to use av_image_copy_to_buffer() then av_image_get_linesize() [19:24:30 CEST] <michaelni> that should give you a byte array that you can then easily apply any checksum function too [19:24:33 CEST] <michaelni> to [19:28:27 CEST] <lglinskih> michaelni: ok, thanks! and is it okay to use av_adler32_update for checksum? [19:28:37 CEST] <michaelni> should be ok [19:34:39 CEST] <wm4> michaelni: heh, thanks [21:05:19 CEST] <wm4> michaelni: can you make a fast 2.7.1 release with my gnutls fix for disconnecting correctly included? [21:05:43 CEST] <wm4> this breaks anything that wants to seek with https and if gnutls is used [21:12:01 CEST] <michaelni> i need to do a 2.2 release,ill look at 2.7 after that [21:13:42 CEST] <michaelni> wm4, the commit to port is 2222f419da99ef85d49ab04e7e15b76612f4d054 ? [21:14:26 CEST] <wm4> michaelni: correct [21:17:08 CEST] <cone-489> ffmpeg 03wm4 07n2.2.16:HEAD: id3v2: strip trailing spaces from APIC tag [21:27:16 CEST] <jamrial> michaelni: softfloat.h is not an installed header. is that intended? [21:27:23 CEST] <jamrial> all its functions use the av_ prefix [21:33:22 CEST] <michaelni> if people want to use it from outside or want it to be available to the outside then it can be made an installed header [21:33:35 CEST] <michaelni> i think its not installed atm as nothing uses it [21:40:16 CEST] <cone-489> ffmpeg 03James Almer 07release/2.7:3b6aeb148b1d: avformat/singlejpeg: fix standalone compilation [22:18:20 CEST] <cone-489> ffmpeg 03Michael Niedermayer 07release/2.7:965f96c5ed44: ffmpeg_opt: Check for localtime() failure [22:18:21 CEST] <cone-489> ffmpeg 03Andreas Cadhalpun 07release/2.7:ed5041143edb: configure: only disable VSX for !ppc64el [22:18:22 CEST] <cone-489> ffmpeg 03Andreas Cadhalpun 07release/2.7:265db4154097: doc: fix spelling errors [22:18:23 CEST] <cone-489> ffmpeg 03Andreas Cadhalpun 07release/2.7:3ba55ea4aec2: h264: er: Copy from the previous reference only if compatible [22:18:24 CEST] <cone-489> ffmpeg 03Andreas Cadhalpun 07release/2.7:8a24ccfee384: examples/demuxing_decoding: use properties from frame instead of video_dec_ctx [22:18:25 CEST] <cone-489> ffmpeg 03wm4 07release/2.7:9473e5a05d2c: avio: fix potential crashes when combining ffio_ensure_seekback + crc [22:18:26 CEST] <cone-489> ffmpeg 03Michael Niedermayer 07release/2.7:08fadda68aeb: avcodec/ffv1enc: fix bps for >8bit yuv when not explicitly set [22:18:27 CEST] <cone-489> ffmpeg 03Michael Niedermayer 07release/2.7:e04ae11fa06a: ffmpeg: Free last_frame instead of just unref [22:18:28 CEST] <cone-489> ffmpeg 03Michael Niedermayer 07release/2.7:6432f8826dd1: avcodec/hevc_ps: Only discard overread VPS if a previous is available [22:18:29 CEST] <cone-489> ffmpeg 03wm4 07release/2.7:c3c8365dbd3b: tls_gnutls: fix hang on disconnection [22:34:54 CEST] <kierank> lglinskih: sorry was on a train [22:34:58 CEST] <kierank> but i think michaelni helped you [22:38:23 CEST] <BBB> is anyone interested in a patch that changes tiny_ssim to become lavf based? [22:38:38 CEST] <BBB> its a bit hacky so its not exactly ready, but maybe others want to finish it up [22:38:55 CEST] <BBB> (i was getting fed up with having to use pipes every time I want to compare two files that are not raw yuv) [22:39:36 CEST] <durandal_1707> You mean lavfi based? [22:39:41 CEST] <BBB> no, lavf [22:39:57 CEST] <BBB> it doesnt fopen() two yuv inputs [22:40:06 CEST] <BBB> it avformat_open_input()s two inputs [22:40:15 CEST] <BBB> is basically what the patch does [22:40:16 CEST] <durandal_1707> hack hack hack [22:40:20 CEST] <BBB> why? [22:40:28 CEST] <BBB> open two files, read/decode frames, ssim them [22:40:31 CEST] <BBB> hows that hack? [22:41:06 CEST] <durandal_1707> it should be in lavfi. [22:41:13 CEST] <BBB> what part? [22:41:15 CEST] <BBB> and why? [22:41:58 CEST] <durandal_1707> it duplicates read/decode code [22:42:26 CEST] <BBB> it uses public api [22:42:29 CEST] <BBB> thats a hack nowaday? [22:42:44 CEST] <nevcairiel> Don't listen to him [22:43:23 CEST] <nevcairiel> Can you make it stay compatible with old calling syntax? [22:45:10 CEST] <BBB> it used to be tiny_ssim a.yuv b.yuv widthxheight [22:45:17 CEST] <BBB> now its tiny_ssim a.y4m b.mkv [22:45:27 CEST] <BBB> (an widthxheight is implied by the inputs) [22:45:37 CEST] <BBB> so the syntax is a little different [22:45:48 CEST] <BBB> I didnt add support for specifying raw yuv parameters yet, probably should [22:46:01 CEST] <BBB> (like tiny_ssim -size withxheight a.yuv b.mkv) [22:46:45 CEST] <nevcairiel> Just trying to preserve syntax for peoples scripts might be useful [22:46:47 CEST] <nevcairiel> So that previous commands for yuv files still wofk [22:46:54 CEST] <BBB> I might just make it a new utility then [22:46:56 CEST] <BBB> like tiny_ssim_2 [22:47:07 CEST] <durandal_1707> You will need to maintain it whenever API changes [22:47:31 CEST] <nevcairiel> It's a good check for not breaking public api too often :P [22:47:38 CEST] <durandal_1707> With lavfi you leave it to others [22:47:48 CEST] <nevcairiel> It doesn't belong in lavfi [22:48:08 CEST] <durandal_1707> It does. [22:48:44 CEST] <BBB> I dont even know what you guys are talking about [22:48:47 CEST] <BBB> what belongs in lavfi? [22:48:53 CEST] <BBB> calculating the ssim between two inputs? [22:48:56 CEST] <durandal_1707> But I will not object lavf one [22:49:29 CEST] <nevcairiel> Putting it there would be the hack, instead of creating a couple lines of standalone tool for testing purposes [22:50:19 CEST] <durandal_1707> use ffmpeg for everything [22:50:52 CEST] <jamrial> changing tiny_ssim to support opening formats that are not raw yuv is certainly lavf territory, not lavfi [22:52:58 CEST] <durandal_1707> qt-fastdstart is slower than ffmpeg [22:55:36 CEST] <durandal_1707> do tell^ [22:55:55 CEST] <wm4> lol ffmpeg devs afraid of using their own API, because the API could change [22:56:17 CEST] <durandal_1707> huh [22:57:15 CEST] <gchristensen> Hi there, it is possible I could donate infrastructure to ffmpeg, including build infrastructure farms, or websites, etc. I'm pretty flexible. it would be on behalf of the company I work for. Who should I talk to, what would be most beneficial, etc ..? [22:57:27 CEST] <gchristensen> website hosting* [22:57:54 CEST] <nevcairiel> Like I said, don't listen to that guy, "use ffmpeg for everything" is what's wrong woth any devs attitude :p [22:58:13 CEST] <nevcairiel> with many* [22:58:36 CEST] <durandal_1707> you are nightmare [22:58:45 CEST] <gchristensen> my co has a fairly advanced and capable build farm infrastructure which might be particularly interesting, but again -- just looking to contribute. [22:59:20 CEST] <durandal_1707> michaelni ^ [23:00:48 CEST] <ubitux> gchristensen: first easy thing would be to provide fate instances (see http://fate.ffmpeg.org/ and https://ffmpeg.org/fate.html) with various exotic configuration/hw/toolchain/... [23:01:34 CEST] <ubitux> gchristensen: then i think we looked at moving to a new server a while back; even though it seems mostly sysadmin blocked, michaelni can probably tell you more about this [23:01:44 CEST] <ubitux> or beastd, but he isn't connected right now [23:01:51 CEST] <ubitux> and thanks a lot for the proposition [23:02:05 CEST] <michaelni> most stuff is already moved to the new server [23:03:04 CEST] <michaelni> faster/more powerfull computers for developers who want/need more powerfull hw might be useful [23:03:27 CEST] <michaelni> also we need a slave name server that isnt in .hu [23:03:31 CEST] <ubitux> BBB: i agree with durandal_1707 that you should probably just abstract all the formats & codecs layers by just adding a filter; i think there is even a vf_psnr already [23:04:26 CEST] <BBB> Im still not sure I understand how I would use it [23:04:34 CEST] <ubitux> just like vf_psnr? [23:04:39 CEST] <BBB> like, right now, I have this tool thats a few lines of change and I call it as tiny_ssim a.y4m b.mkv [23:04:42 CEST] <BBB> how would vf_ssim work? [23:04:45 CEST] <BBB> I really have no idea [23:04:53 CEST] <nevcairiel> lavfi sucks for such stuff [23:04:57 CEST] <ubitux> https://ffmpeg.org/ffmpeg-filters.html#psnr [23:05:27 CEST] <nevcairiel> The syntax to use it would be a 100x more complicated than tiny_ssim now [23:05:51 CEST] <ubitux> yes maybe the interface is not great [23:05:53 CEST] <wm4> heh [23:06:24 CEST] <durandal_1707> pick whatever you prefer I plan to add vf_swim anyway, haters gona hate [23:06:31 CEST] <nevcairiel> Learn from linux and produce small usable tools [23:06:41 CEST] <ubitux> like systemd? [23:06:59 CEST] Action: JEEBsv quickly hides the go-systemd tab from his browser [23:07:07 CEST] <durandal_1707> :) [23:08:09 CEST] <durandal_1707> Stupid android spellchecker,can it be disabled? [23:10:44 CEST] <cone-489> ffmpeg 03wm4 07master:f6c3f1ed6048: avformat: clarify what package needs to be compiled with SSL support [23:11:00 CEST] <gchristensen> sorry for being AFK for a minute there. so 1. I can provide fate instances. how do those get triggered, or would it be on a cron, or would it be on commit? what config /hw / toolchain would be considered exotic? this would be very easy for me to set up.I may be able to provide access to an AWS account with a limited budget that ffmpeg could do anything they wanted (budget-limited) which could include [23:11:02 CEST] <gchristensen> dev machines or slave DNS [23:11:06 CEST] <gchristensen> ubitux / michaelni ^ [23:11:38 CEST] <jamrial> durandal_1707: in "language and input" or such, i guess [23:12:01 CEST] <ubitux> gchristensen: the trigger is generally up to the user (see the 2nd link for how to setup) [23:12:12 CEST] <ubitux> gchristensen: exotic is something you can't find on the first link [23:12:45 CEST] <ubitux> so, weird compiler, weird arch, weird OS, ... [23:13:11 CEST] <ubitux> or just weird ffmpeg configuration, new crazy static analyzer, efficient thread analysis tool, ... [23:13:13 CEST] <ubitux> that kind of stuff [23:13:18 CEST] <jamrial> or not necessarely weird but not currently covered [23:13:53 CEST] <jamrial> like for example, old compiler versions tend to be dropped pretty fast. there are almost no gcc 4.9 clients ever since you moved to gcc 5 :p [23:14:24 CEST] <ubitux> :) [23:14:35 CEST] <ubitux> bleeding edge ftw [23:14:36 CEST] <gchristensen> so this all seems fine, I can run it across a variety of configurations. is there any obvious, pressing configuration that would be a great first start? [23:15:57 CEST] <ubitux> i can't think of any right now... maybe we can ask cehoyos, he generally suggests some build configuration [23:16:07 CEST] <jamrial> hi kurosu [23:16:32 CEST] <ubitux> gchristensen: you might want to send a mail to ffmpeg-devel to reach more people [23:17:43 CEST] <kurosu> Hi James & all [23:17:47 CEST] <gchristensen> can-do [23:21:00 CEST] <michaelni> tickets related to fate: (that is also some missing fate clients), also feel free to add more tickets if you know of other missing cases https://trac.ffmpeg.org/query?status=new&status=open&status=reopened&keywor… [23:23:48 CEST] <jamrial> i'm personally running a gcc trunk client to catch compiler bugs early and avoid another fiasco like gcc 4.9.0 [23:26:01 CEST] <nevcairiel> f'ing SDL, why does it force building in GUI mode just by linking in its libraries [23:26:46 CEST] <wm4> just don't link sdl [23:27:01 CEST] <nevcairiel> thats the easy way out =p [23:27:10 CEST] <nevcairiel> but its annoying if you want ffplay to test things [23:27:22 CEST] <jamrial> it's the "-Dmain=SDL_main" addition to CFLAGS what breaks things [23:29:21 CEST] <jamrial> doesn't seem to break ffmpeg or ffprobe, but it does break the stuff inside the tools/ folder and the "-test" executables from lavc/lavu [00:00:00 CEST] --- Fri Jun 19 2015

1 0

ffmpeg.log.20150618
by burek 19 Jun '15

19 Jun '15

[00:01:27 CEST] <Nolski> ahh i see [02:10:06 CEST] <podman> hey guys, anyone know what would cause me to get an error like this? "Input picture width (640) is greater than stride (0)" [05:01:14 CEST] <Tecan> not sure why but there is issues converting wma files, sounds staticy [05:01:25 CEST] <Tecan> vlc playes them nice [05:01:59 CEST] <durandal_1707> sample, what codec? [05:02:19 CEST] <Tecan> ffmpeg -y -i 01a01.wma -f wav out.wav [05:02:28 CEST] <Tecan> i was trying to play some mozart one sec [05:04:12 CEST] <Tecan> trying to compile mplayer to see if that works [05:05:03 CEST] <kepstin> in theory, mplayer, vlc, and ffmpeg command line tool are all using the same decoder from ffmpeg, maybe just different versions of it [05:06:39 CEST] <chungy> yeah they should all be using libavcodec for it, although depending on a few things (mainly your OS), you might be using a Libav version instead of FFmpeg. [05:10:30 CEST] <Tecan> http://www.xup.in/dl,14782458/01a01.wma.7z/ <- educational use only [05:10:38 CEST] <Tecan> delete after testing ;) [05:13:48 CEST] <chungy> are you able to provide another file to demonstrate what it should decode to (wav or flac would be best) [05:14:12 CEST] <Tecan> with the static ? [05:14:36 CEST] <chungy> the normal decoding, without static I should assume [05:14:51 CEST] <Tecan> its probably just too high of volume [05:15:03 CEST] <Tecan> or the gain is too high ? [05:15:25 CEST] <Tecan> the normal decode has the static, one sec [05:16:02 CEST] <chungy> doesn't seem staticy for me, with ffmpeg 0.7 [05:16:17 CEST] <chungy> 2.7, I mean. [05:16:41 CEST] <Tecan> ffmpeg version N-71229-gacbb904 [05:16:52 CEST] <Tecan> wonder how old it is [05:19:47 CEST] <chungy> I kind of wish more typical git describes were used, yeah... [05:20:07 CEST] <chungy> $ git describe N-71229-gacbb904 [05:20:07 CEST] <chungy> n2.7-dev-740-gacbb904 [05:20:14 CEST] <chungy> a bit before the final 2.7 release. [05:20:30 CEST] <c_14> Apr 2, This year [05:22:21 CEST] <chungy> Tecan: random suggestion since I had it happened to me recently, make sure your system volume isn't set over 100% :P [05:22:44 CEST] <chungy> I accidentally had mine set to like 140% recently and almost thought my speakers were going bad because everything was clipping [05:23:06 CEST] <chungy> still don't know how that happened accidentally, haha [09:19:33 CEST] <rweichler> eff eff em peg [10:01:05 CEST] <damnlie> hiii, do you guys have documentation about how can i set up a separate ffmeg server, i have a separate webserver (where apache en this sort of things are running) and a separate staging server. I never used ffmeg so im a bit lost [10:18:51 CEST] <BtbN> ffmpeg server? [10:18:56 CEST] <BtbN> you mean ffserver? [10:20:04 CEST] <damnlie> i think so :) my coworker installed the whole thing from git so i even do not find the config files... :/ [10:27:51 CEST] <damnlie> http://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu#ffmpeg i found this but im not get what de path are? [10:42:44 CEST] <Polochon_street> hello [10:43:28 CEST] <Polochon_street> does anyone know why sometimes ffprobe shows duplicate titles with semicolon? [10:43:44 CEST] <Polochon_street> lik this: [10:43:47 CEST] <Polochon_street> TITLE : Controlling Crowds;Controlling Crowds [10:44:38 CEST] <Polochon_street> (but maybe this is a dev question?) [13:35:06 CEST] <gurparta> I am facing problem while saving stream from ip camera and capturing audio from laptop microphone.there are synchronization issues in it? Please somebody help. [13:36:07 CEST] <gurparta> this is commandline: ffmpeg -f alsa -ac 2 -async 1 -i default -i rtsp://192.168.0.127 acodec libfaac -v "debug" -y -vcodec copy output.mp4 [13:37:39 CEST] <gurparta> Whiskey: this is commandline: ffmpeg -f alsa -ac 2 -async 1 -i default -i rtsp://192.168.0.127 acodec libfaac -v "debug" -y -vcodec copy output.mp4 [13:41:03 CEST] <gurparta> relaxed: this is commandline: ffmpeg -f alsa -ac 2 -async 1 -i default -i rtsp://192.168.0.127 acodec libfaac -v "debug" -y -vcodec copy output.mp4 [15:49:35 CEST] <proserpine-> hi. is there a way for ffmpeg to cut out frames where the video is still? i'm trying to edit very long camera footage but it's quite tedious. [15:53:28 CEST] <saste> proserpine-, select filter, check the scene_change parameter [15:54:12 CEST] <proserpine-> saste: excellent, exactly what i needed. thank you [16:00:42 CEST] <ads34123423> Quick question, is an mpeg transport stream containing an H.264 encoded video stream likely to contain H.264 NAL units complete with start codes (0x000001), or would they be redundant and not used? [16:48:26 CEST] <Mavrik> afaik TS streams have to use AnnexB format [20:22:39 CEST] <ldiamond1> I want to use ffmpeg for twitch.tv streaming. I want to add a webcam overlay to the stream. So far it's all easy and there's plenty of examples online. However, I want to know if it's possible to send commands to the ffmpeg process to "alter" options, such that I can move the webcam overlay around and remove/add it on the fly (for example if it's obstructing the content). [20:37:12 CEST] <daan_> Hello! I have ffmpeg installed on my Raspberry Pi but my terminal gives me as output: Failed to set value '[1:v]scale=1920:1080,setdar=16/9,boxblur=10[bg];[1:v]scale=1080:1080[fg];[bg][fg]overlay=(W-w)/2:(H-h)/2[v]' for option 'filter_complex' how can i fix it? [20:49:04 CEST] <lucascastro> I'm trying to setup ffserver, but when I stream my /dev/video0 to the server, on server show /feed1.ffm not found [21:22:35 CEST] <BW^-> ffmpeg -i 00058.MTS jake.mts -pass 3 -b:v 4000k -acodec copy [21:22:55 CEST] <BW^-> for audio input it says: " Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 256 kb/s" [21:23:17 CEST] <BW^-> for the output though it says " Stream #0:1 -> #0:1 (ac3 (native) -> mp2 (native))" [21:23:24 CEST] <BW^-> and "Stream #0:1: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s" [21:23:29 CEST] <BW^-> so that clearly looks like recoding to me!? [21:24:50 CEST] <BW^-> hmm with the latest FFMPEG, I have a "Stream #0:0: Video: mpeg2video, yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 50 fps, 90k tbn, 50 tbc" 26383kb/sec input stream that takes ~150MB per minute [21:25:00 CEST] <BW^-> I just want to get it down to maybe 15-30MB per minute [21:25:15 CEST] <BW^-> what codec do you suggest for preserving maximum quality? [21:25:50 CEST] <BW^-> um.. wait.. no this is my input stream: " Stream #0:0[0x1011]: Video: h264 (High) (HDPR / 0x52504448), yuv420p, 1920x1 080 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc" [21:26:24 CEST] <BW^-> now i just did "-pass 3 -b:v 4000k" and that generates a less than superb output [21:27:41 CEST] <BW^-> aha let's try "-c:v libx264 -crf 22 " [21:34:08 CEST] <BW^-> :D [21:34:10 CEST] <BW^-> yey was perfet! [21:34:18 CEST] <BW^-> ok this is good enough so no need to go into details - thanks :)) [22:29:38 CEST] <anoop_r> hi [22:30:31 CEST] <anoop_r> any plans to include Apple AAC support for ffmpeg [22:37:20 CEST] <durandal_1707> anoop_r: what? Is it open source? [22:37:30 CEST] <JEEBsv> nope [22:38:26 CEST] <durandal_1707> then answer is simply NO [22:38:31 CEST] <JEEBsv> it has an API, to use which you have to install the API parts of QT. It is not GPL compatible anywhere else than on OS X (because there QT is a system component), and I don't see it being really better than fdk-aac [22:40:02 CEST] <anoop_r> thanks [22:50:01 CEST] <gchristensen> Hi, if someone wanted to donate infrastructure, is that something which would be useful? [22:50:47 CEST] <durandal_1707> Absolutely yes [22:50:58 CEST] <gchristensen> what would be beneficial, and who do I talk to? [22:51:44 CEST] <durandal_1707> Ask in -devel [23:16:15 CEST] <anoop_r> any way to add FAAC to ffmpeg [23:22:27 CEST] <durandal_1707> decoder? [23:38:01 CEST] <anoop_r> encoder [23:42:41 CEST] <durandal_1707> IIRC it is already there [23:57:20 CEST] <anoop_r> anyway to add QAAC to ffmpeg [00:00:00 CEST] --- Fri Jun 19 2015

1 0

ffmpeg-devel.log.20150617
by burek 18 Jun '15

18 Jun '15

[00:02:25 CEST] <jamrial> wm4: you mean it should use avpriv_atomic_ptr_cas()? [00:03:06 CEST] <wm4> jamrial: it should use pthread_once [00:04:29 CEST] <jamrial> why? the atomic compare and swap function exists exactly for this purpose [00:07:16 CEST] <wm4> jamrial: I suppose doing ~260 atomic ops on init isn't that expensive [00:07:22 CEST] <jamrial> there's also apparently no wrapper for pthread_once using w32threads [00:07:32 CEST] <wm4> but it sure as hell is roundabout as hell [00:07:49 CEST] <wm4> yes, because fuck everything including ffmpeg and microsoft [00:07:57 CEST] <wm4> let's just use racy code [00:09:03 CEST] <wm4> michaelni: ok, ffio_ensure_seekback() and checksum_ptr (whatever the ridiculous crap this is) interact badly [00:09:15 CEST] <wm4> why is aviobuf doing checksums [00:09:28 CEST] <wm4> and in such a fragile way [00:12:02 CEST] <nevcairiel> it would be possible to emulate pthread_once for windows without too much trouble, using the atomics, but I find pthread_once in itself an equally ugly solution [00:12:50 CEST] <jamrial> wm4, nevcairiel: there seems to be this https://msdn.microsoft.com/en-us/library/windows/desktop/ms683493%28v=vs.85… but it's Vista+ only [00:12:57 CEST] <wm4> jamrial: yep [00:13:10 CEST] <wm4> I'm confident it could be emulated on xp too [00:13:17 CEST] <wm4> but what about OS/2? [00:13:23 CEST] <jamrial> using atomics as nev said, yeah [00:15:09 CEST] <jamrial> does anyone care about OS/2? [00:15:37 CEST] <wm4> so we just break it and that-one-guy has to deal with it? [00:15:40 CEST] <nevcairiel> apparently at least one person that keeps fixing it when we break it [00:16:02 CEST] <nevcairiel> if we build a pre-vista layer, he could probably copy that without too much pain [00:35:14 CEST] <wm4> rc0mbs: what the heck is bfstm anyway? [00:35:31 CEST] <rc0mbs> (hi I'm here under my usual nick) [00:35:35 CEST] <rc0mbs> oh wait I'm not [00:35:44 CEST] <rcombs> welp [00:35:55 CEST] <rcombs> wm4: WiiU sound format [00:36:41 CEST] <wm4> ah [00:37:38 CEST] <nevcairiel> why cant modern game consoles just use normal ordinary formats [00:38:01 CEST] <nevcairiel> PC games usually manage to these days [00:42:47 CEST] <BtbN> To make it harder to use their assets [00:43:07 CEST] <wm4> nevcairiel: ordinary formats like bink? [00:43:22 CEST] <nevcairiel> is that really used by new games anymore? [00:43:31 CEST] <wm4> (I hope not) [00:43:56 CEST] <wm4> but last time I talked with a gamedev about bink, he defended it [00:46:19 CEST] <cone-928> ffmpeg 03Michael Niedermayer 07master:1125c71fd761: avcodec/jpeg2000dwt: assert that mod == 0 for encoding [00:46:20 CEST] <cone-928> ffmpeg 03Michael Niedermayer 07master:f140a99f8b31: swscale/output: Factorize rgb48 and 64bit code [00:46:21 CEST] <cone-928> ffmpeg 03Michael Niedermayer 07master:e29d99614969: swscale/output: Add rgba64/rgb48/bgra64/bgr48 output functions with full chroma interpolation [00:47:04 CEST] <wm4> michaelni: that was quick [01:26:06 CEST] <cone-928> ffmpeg 03wm4 07master:dc87758775e2: avio: fix potential crashes when combining ffio_ensure_seekback + crc [01:58:07 CEST] <cehoyos> -Fyr-: No, I would have set the status to "open" and "reproduced" if I had. [04:01:30 CEST] <fatter> hello [04:01:53 CEST] <fatter> WINDRES libswresample/swresampleres.o [04:01:55 CEST] <fatter> LD libswresample/swresample-1.dll [04:01:56 CEST] <fatter> sed -e "s/ @[^ ]*//" libswresample/swresample-1.orig.def > libswresample/swresample-1.def; lib.exe /machine:i386 /def:libswresample/swresample-1.def /out:libswresample/swresample.lib [04:01:58 CEST] <fatter> Microsoft (R) Library Manager Version 10.00.40219.01 [04:01:59 CEST] <fatter> Copyright (C) Microsoft Corporation. All rights reserved. [04:02:01 CEST] <fatter> c(ú libswresample/swresample.lib ùa libswresample/swresample.exp [04:02:02 CEST] <fatter> LD libavcodec/avcodec-56.dll [04:02:04 CEST] <fatter> gcc.exe: error: libavcodec/moti: No such file or directory [04:02:05 CEST] <fatter> library.mak:106: recipe for target 'libavcodec/avcodec-56.dll' failed [04:02:07 CEST] <fatter> make: *** [libavcodec/avcodec-56.dll] Error 1 [04:02:08 CEST] <fatter> 09:53:50 Build Finished (took 10m:26s.193ms) [04:02:24 CEST] <fatter> what's up [04:02:56 CEST] <fatter> i used msys2 and mingw32 and eclipse [04:56:37 CEST] <Compn> fatter : what ver make ? [04:57:00 CEST] Action: Compn sleeps [05:00:06 CEST] <fatter> sorry , i have removed msys2 and reinstall it and retry it [05:36:09 CEST] <philipl> wm4: Re bink: I assumed it was because the consoles didn't include any codec licenses for devs. And bink was cheaper/easier than getting an h.264 licence. [05:37:18 CEST] <rcombs> Nintendo outright bought a (shitty) video codec company [05:38:05 CEST] <rcombs> for audio I guess ADPCM is light on the baby CPU and has very predictable performance [05:39:20 CEST] <philipl> Good ol' Nintendo. [05:39:47 CEST] <philipl> I assume the xbone and ps4 have AMD's decoder hardware and API support, but maybe I'm too optimistic. [05:39:52 CEST] <jamrial> most games nowadays use pcm. light on the CPU and blu-rays have plenty of space anyway [05:40:17 CEST] <philipl> unreal engine pushes people towards vorbis it seems. [05:40:32 CEST] <rcombs> jamrial: Nintendo games (and the system menu and such) use their own special-snowflake ADPCM, but close enough [05:41:11 CEST] <jamrial> i'm talking about ps4/xbox/PC (when they are multiplatform) [05:41:40 CEST] <jamrial> since both the ps4 and xbox cpus are shit, those cycles they save by not decoding some lossy or lossless codec seem to be very valuable [05:42:14 CEST] <rcombs> yup [05:42:48 CEST] <rcombs> the WiiU seems to actually have an H.264 decoder, judging by it playing web video [05:43:04 CEST] <rcombs> and probably a hardware encoder as well, judging by Mario Kart 8 being able to upload to YouTube [05:46:01 CEST] <philipl> https://github.com/hcs64/ww2ogg [05:46:11 CEST] <philipl> Why would they do that? [05:46:33 CEST] <philipl> Vorbis but not vorbis. [08:49:48 CEST] <j-b> 'matin [09:11:00 CEST] <kierank> philipl: iirc vorbis codebooks are quite large [09:29:04 CEST] <durandal_1707> j-b: ffmpeg supports sevc/evrc [09:40:52 CEST] <Fyr> guys, if you click "Open+New Tickets", you will see 1093 bugs, part of them was opened 3-2 years and modified 3-2 years ago. is ffmpeg obsolete? [09:42:39 CEST] <wm4> Fyr: yes it is [09:42:54 CEST] <wm4> we're just working on it for the lack of something better tto do [09:43:06 CEST] <Fyr> ='( [09:43:32 CEST] <rcombs> who do I throw new fate samples at [09:43:55 CEST] <kierank> Fyr: interesting troll [09:44:25 CEST] <Fyr> guys, you can check that. [09:44:44 CEST] <Fyr> the majority of the bugs was opened 3-2 years ago. [09:46:43 CEST] <thardin> bugs or feature requests? [09:47:02 CEST] <Fyr> tickets [09:47:06 CEST] <thardin> bugs or feature requests? [09:47:19 CEST] <Fyr> defect [09:47:43 CEST] <Fyr> only a few of enhancement [09:51:15 CEST] <thardin> seems to mostly be "ffmpeg doesn't work exactly like $TOOL" [09:51:28 CEST] <thardin> heh, opus tickets [09:52:12 CEST] <BtbN> Some automatic ticket closer and bumper would indeed be nice [09:52:28 CEST] <BtbN> i'm sure most of these tickets have become pointless [09:53:01 CEST] <Fyr> I posted a bug and see no movement for two days. =)) [09:53:10 CEST] <Fyr> even no confirmation.=/ [09:55:19 CEST] <j-b> durandal_1707: lol. Seriously? [10:03:53 CEST] <durandal_1707> Fyr: what bug? [10:04:07 CEST] <Fyr> https://trac.ffmpeg.org/ticket/4622#ticket [10:04:29 CEST] <Fyr> ffmpeg doesn't add ReplayGain into m4a container. [10:04:48 CEST] <durandal_1707> j-b: seriously, vlc have 2 years old bug report [10:04:57 CEST] <kierank> all big projects have old bug reports [10:05:03 CEST] <kierank> fyr [10:05:10 CEST] <Fyr> ={ [10:06:18 CEST] <durandal_1707> Fyr: Perhaps it could be done via lavfi filter [10:06:28 CEST] <Fyr> maybe [10:06:54 CEST] <Fyr> but why does ffmpeg forgets to write ReplayGain? [10:07:11 CEST] <Fyr> it shows RG in output. [10:07:26 CEST] <Fyr> but doesn't add it to metadata. [10:07:35 CEST] <BtbN> Because someone forgot to implement that/there is a bug breaking it. [10:07:37 CEST] <durandal_1707> rcombs: ask michaelni [10:07:57 CEST] <rcombs> durandal_1707: I looked it up; turns out the correct answer is "provide a link in your commit message" [10:08:30 CEST] <rcombs> BtbN: may as well say "because it doesn't" [10:08:40 CEST] <rcombs> BtbN: which& is the only real answer to that question :P [10:08:53 CEST] <durandal_1707> you could also send it [10:09:14 CEST] <BtbN> Isn't m4a just renamed mov/mp4? [10:09:41 CEST] <Fyr> it is [10:09:59 CEST] <BtbN> Yeah, not surprised if random stuff breaks with that container. [10:10:40 CEST] <durandal_1707> perhaps if you donate some money first [10:11:03 CEST] <Fyr> perhaps after fixing the bug. [10:11:04 CEST] <Fyr> =)) [10:11:45 CEST] <durandal_1707> in my experiance people then just dissapear [10:13:00 CEST] <Fyr> appreciation and gratefulness of the users are not just enough to do a pleasing to God business? [10:13:39 CEST] <Fyr> xD [10:14:44 CEST] <BtbN> putting a bounty on a bug definitely helps ;) [10:14:59 CEST] <durandal_1707> there are people that steal from us [10:15:10 CEST] <Fyr> it sounds like a bribe. =) [10:15:10 CEST] <BtbN> There even is a service for that, and i have seen it used a few times in the ffmpeg trac [10:15:38 CEST] <BtbN> You want some people to do something for you, in their free time. And you want them to do it faster than usual. [11:23:31 CEST] <ubitux> wm4: it seems vda actually does the expected convert [11:23:50 CEST] <ubitux> at least requesting rgb32 seems to work (didn't check output yet though) [11:24:20 CEST] <wm4> interesting [11:26:40 CEST] <ubitux> btw, any idea how much frames can be kept cached? [11:58:24 CEST] <wm4> ubitux: what do you mean? [12:19:40 CEST] <cbsrobot> michaelni: thanks a lot for the full "chroma interpolation" ! [13:32:05 CEST] <cone-315> ffmpeg 03Rodger Combs 07master:bb42a7d4d401: lavf/brstm: add support for BFSTM files [14:13:41 CEST] <cone-315> ffmpeg 03Shivraj Patil 07master:63eaf529bcfa: avcodec/mips: MSA (MIPS-SIMD-Arch) optimizations for H263 lpf functions [14:21:53 CEST] <ubitux> wm4: like, how much CVPixelBufferRef can be stored? [14:25:39 CEST] <wm4> ubitux: I'd just assume as much as there's GPU memory [14:25:51 CEST] <ubitux> ok [14:25:51 CEST] <wm4> I don't know, I didn't ever have trouble with this [14:26:05 CEST] <ubitux> your frame queue is large? [14:26:24 CEST] <wm4> no, maybe 2 or 3 frames [14:26:30 CEST] <ubitux> ok [14:44:32 CEST] <JEEBsv> hmm, was the default audio bit rate also 200kbps? [15:15:46 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07master:9841024402e8: avcodec/jpeg2000dwt: More special cases to handle singularities in 9/7 decode [15:29:22 CEST] <JEEBsv> do I understand correctly that setting -maxrate doesn't try to collerate it between streams, random encoders and muxers just take the value of rc_max_rate and use it? [15:29:35 CEST] <JEEBsv> at least that's how it looks like with a quick git grep [15:33:47 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07master:7c3f7e6ed185: avcodec/jpeg2000dwt: Support 9/7 singularity cases on the encoder side [20:50:59 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07master:2687a51a3fcf: avcodec/jpeg2000dec: Drop unneeded rlevel variable [20:51:00 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07master:e6190045b3a1: avcodec/mqcenc: Add ff_mqc_flush_to() [20:51:01 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07master:4624656797b6: avcodec/j2kenc: Properly flush the end of the truncated AC stream [20:59:58 CEST] <cone-315> ffmpeg 03Vittorio Giovara 07master:d8d124eebcf2: aic: Improve error reporting [20:59:59 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07master:98b95e746668: Merge commit 'd8d124eebcf2ec1f6a1936b12a4f00a48f08e85c' [21:12:14 CEST] <cone-315> ffmpeg 03James Almer 07master:1e79d5c6e73a: libwebpenc: use WebPMemoryWriterClear() [21:12:15 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07master:23c22d37d201: Merge commit '1e79d5c6e73ad131f9395f337b58a2b59ee04c1b' [22:09:52 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:47936e1a5c87: swresample: Check the return value of resampler->init() [22:09:53 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:b71657324d0c: libopenjpegenc: add NULL check for img before accessing it [22:09:54 CEST] <cone-315> ffmpeg 03Ganesh Ajjanagadde 07release/2.2:0fee46e2dbc0: swresample/dither: check memory allocation [22:09:55 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:f33adf1b5345: avformat/mxfenc: Accept MXF D-10 with 49.999840 Mbit/sec [22:09:56 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:41e3d5bc3aef: Revert "avformat/rtpenc: check av_packet_get_side_data() return, fix null ptr dereference" [22:09:57 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:b97fb80db9e6: swresample/swresample: Cleanup on init failure. [22:09:58 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:eb021638a116: avcodec/atrac3plusdec: consume only as many bytes as available [22:09:59 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:c03c2f23b229: avcodec/alsdec: Check for overread [22:10:00 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:57c99f8fe4d6: avcodec/adpcm: Check for overreads [22:10:01 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:78ef1be8e12e: avformat/ffmdec: Check ffio_set_buf_size() return value [22:10:02 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:4cb077793d00: avcodec/jpeg2000dec: Check that coords match before applying ICT [22:10:03 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:4348a241ed53: avcodec/x86/h264_weight: handle weight1=128 [22:10:04 CEST] <cone-315> ffmpeg 03Simon Thelen 07release/2.2:b2d7a0840534: libavutil/channel_layout: Correctly return layout when channel specification ends with a trailing 'c'. [22:10:05 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:4a5a6fe6ba1a: avdevice/lavfi: do not rescale AV_NOPTS_VALUE in lavfi_read_packet() [22:10:06 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:d6bde7be5f34: takdec: ensure chan2 is a valid channel index [22:10:07 CEST] <cone-315> ffmpeg 03Deliang Fu 07release/2.2:557aa4c6b32a: avformat: Fix bug in parse_rps for HEVC. [22:10:08 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:8bb21c031074: ffmpeg_opt: Check for localtime() failure [22:10:09 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:de8e63ca8215: mov: abort on EOF in ff_mov_read_chan [22:10:10 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:ebf0a8fbda1f: vp9: change type of tile_size from unsigned to int64_t [22:10:11 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:fabb394a2dd0: vp8: change mv_{min,max}.{x,y} type to int [22:10:12 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:f87d76e659e0: sonic: set avctx->channels in sonic_decode_init [22:10:13 CEST] <cone-315> ffmpeg 03wm4 07release/2.2:539603e8770f: avio: fix potential crashes when combining ffio_ensure_seekback + crc [22:10:14 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:d29153591299: diracdec: check that block length is valid [22:10:15 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:3a0c421b5669: aacsbr: break infinite loop in sbr_hf_calc_npatches [22:10:16 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:9f9af1dd04fb: libavutil/mem: use size_t for the length in av_strdup() [22:10:17 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:0b6c8bb26c62: avutil/dict: Use size_t for appending strings [22:10:18 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:d27b9d3bd1d2: avformat/vorbiscomment: Check entry length in ff_vorbiscomment_write() [22:10:19 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:f76b54fba781: avutil/avstring: Use size_t in av_strlcatf() [22:10:20 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:17ad9778a9a3: avformat/url: Use size_t for len from strlen() [22:10:21 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:404db2b99eca: avformat/subtitles: Use size_t for len [22:10:22 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:6bb62d64d240: tools/graph2dot: use larger data types than int for array/string sizes [22:10:23 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:0452e3e41cd0: avformat/rtpdec_xiph: Check upper bound on len in xiph_handle_packet() [22:10:25 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:b8060bc217e4: avformat/rtpenc_jpeg: Check remaining buffer size for SOS [22:10:25 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:3062689f3766: avformat/nutdec: Fix use of uinitialized value [22:10:26 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:ed578c91bfdf: avformat/matroskadec: Cleanup error handling for bz2 & zlib [22:10:27 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:55a9a373590b: avformat/hevc: Check cpb_cnt_minus1 [22:10:28 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:06fda5bef349: avformat/hevc: Check num_negative_pics and num_positive_pics [22:10:29 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:14d69655b782: aacdec: don't return frames without data [22:10:30 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:e1b98555b9aa: avcodec/vqavideo: Check chunk size [22:10:31 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:6d6acef9fb83: avcodec/hevc_sei: Check num_sps_ids_minus1 value [22:10:32 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:eed8ccd9eb3e: avcodec/hevc: Check offset_len [22:10:33 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:87d2c6105df2: avcodec/hevc: Check num_entry_point_offsets [22:10:34 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:a2e183aa0552: avcodec/hevc_ps: Check vps_num_hrd_parameters [22:10:35 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:003b006c8564: avcodec/hevc_ps: Explicitly check num_tile_* for negative values [22:10:36 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:046439d6084e: avcodec/jpeg2000dec: fix boolean operator [22:10:37 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:611791e77e94: avcodec/mjpegdec: fix len computation in ff_mjpeg_decode_dqt() [22:10:38 CEST] <cone-315> ffmpeg 03Rainer Hochecker 07release/2.2:02c51e05640c: swr: fix alignment issue caused by 8ch sse functions [22:10:39 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:05a3a3e5b0d9: avcodec/mpeg4audio: add some padding/alignment to MAX_PCE_SIZE [22:10:40 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:e52d66c647e6: avcodec/cavsdec: Check esc_code [22:10:41 CEST] <cone-315> ffmpeg 03wm4 07release/2.2:f06168f7dcd2: hevc: make avcodec_decode_video2() fail if get_format() fails [22:10:42 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:d9194389049b: avcodec/dcadec: Check nchans [22:10:43 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:b259c7e89101: avcodec/dcadec: Check subsubframes [22:10:44 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:f8f24d5dbdf9: ffmpeg_opt: Fix -timestamp parsing [22:10:45 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:cd489b8f6469: avcodec/proresdec2: Reset slice_count on deallocation [22:10:46 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:9f885733d6a9: avcodec/shorten: Fix code depending on signed overflow behavior [22:10:47 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:16af12a807ce: avcodec/shorten: Check skip_bytes() [22:10:48 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:47db54cef679: avcodec/shorten: More complete pred_order check [22:10:49 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:49a759fb3fe9: avcodec/smvjpegdec: check avcodec_decode_video2() return code [22:10:50 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:3a43c712aebc: avcodec/sonic: check memory allocations [22:10:51 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:3344a28787a7: avcodec/sonic: More completely check sample_rate_index and channels [22:10:52 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:3edb74db3bc9: avcodec/dcadec: Check scale table index [22:10:53 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:0122e38cb19b: avcodec/dcadec: Check active_bands [22:10:54 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:ed6d7696db31: avcodec/libtheoraenc: Check for av_malloc failure [22:10:55 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:9cee1c4ac7cd: avcodec/hevc: Fix typo in num_entry_point_offsets check [22:10:56 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:c8676f9daf9d: avcodec/golomb: get_ur_golomb_jpegls: Fix reading huge k values [22:10:57 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:6ad365467097: avformat/nutdec: Return error on EOF from get_str() [22:10:58 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:8cfc0fdb23e1: avformat/nutdec: Fix recovery when immedeately after seeking a failure happens [22:10:59 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:fe1e6b919d8d: avformat/nutdec: Check X in 2nd branch of index reading [22:11:00 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:9414358f1586: nutdec: fix infinite resync loops [22:11:01 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:4d625f270a59: nutdec: stop skipping bytes at EOF [22:11:02 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:dfd1d487301b: nutdec: abort if EOF is reached in decode_info_header/read_sm_data [22:11:03 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:9d6d4eb42542: avcodec/put_bits: Update size_in_bits in set_put_bits_buffer_size() [22:11:04 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:4f09968865a0: avcodec/mpegvideo_enc: Update the buffer size as more slices are merged [22:11:05 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:2b499b694002: avcodec/put_bits: Assert that there is enough space left in skip_put_bytes() [22:11:06 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:249021eb2336: avcodec/bitstream: Assert that there is enough space left in avpriv_copy_bits() [22:11:07 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:f494b906e344: avformat/utils: Ensure that AVFMT_FLAG_CUSTOM_IO is set before use [22:11:08 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:e96e0e1333c8: mpeg4videodec: only allow a positive length [22:11:09 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:940b9484dcde: bink: check vst->index_entries before using it [22:11:10 CEST] <cone-315> ffmpeg 03Vittorio Giovara 07release/2.2:cd5a832b0aab: lavf: Reset global flag on deinit [22:11:11 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:fe8a763bc706: tests/fate-run: do not attempt to parse tiny_psnrs output if it failed [22:11:12 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:d9ff9b8ab0b5: apedec: set s->samples only when init_frame_decoder succeeded [22:11:13 CEST] <cone-315> ffmpeg 03Maneesh Gupta 07release/2.2:25daf27484ea: OpenCL: Avoid potential buffer overflow in cmdutils_opencl.c [22:11:14 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:02b434044caa: ffmpeg: remove incorrect network deinit [22:11:15 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:b548a6208eed: nutdec: check for negative frame rate in decode_info_header [22:11:16 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:ffdd93fe79b7: nutdec: fix illegal count check in decode_main_header [22:11:17 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:45ea5ba0686c: avcodec/wavpack: Check L/R values before use to avoid harmless integer overflow and undefined behavior in fate [22:11:18 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:1f5eeed43ba4: matroskadec: use uint64_t instead of int for index_scale [22:11:19 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:80cd5abf7fdb: avcodec/ffv1dec: Check chroma shift parameters [22:11:20 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:7d9725187d1a: matroskadec: check s->streams[k] before using it [22:11:21 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:4f0d1af68ad3: avformat/matroskadec: Use tracks[k]->stream instead of s->streams[k] [22:11:22 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:a64102e25fb9: diracdec: prevent overflow in data_unit_size check [22:11:23 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:b946f849bb92: diracdec: avoid overflow of bytes*8 in decode_lowdelay [22:11:24 CEST] <cone-315> ffmpeg 03Andreas Cadhalpun 07release/2.2:c9c5bd8c89b6: diracdec: check if reference could not be allocated [22:11:25 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:cea2106fb2e1: avcodec/h264_slice: More complete cleanup in h264_slice_header_init() [22:20:13 CEST] <durandal_1707> michaelni: backport exr fixes? [22:21:46 CEST] <michaelni> durandal_1707, which git commit hash ? [22:22:19 CEST] <cone-315> ffmpeg 03Michael Niedermayer 07release/2.2:051cd7dc5f42: Update for 2.2.16 [22:23:07 CEST] <michaelni> btw you can cherry pick commits to release branches too if you see anythng else that should be backported [22:28:12 CEST] <durandal_1707> michaelni: last commit to exr, I'm on phone [22:41:50 CEST] <michaelni> durandal_1707, that one says its a regression since 95582b5c, 95582b5c is in 2.3 but not 2.2 [22:44:20 CEST] <durandal_1707> michaelni: ok, backport to where it is sane [22:44:36 CEST] <michaelni> ok [23:52:17 CEST] <cone-315> ffmpeg 03George Boyle 07master:47cadf4d1fa0: avcodec/flacenc: Regression test for ticket #4628 [00:00:00 CEST] --- Thu Jun 18 2015

1 0

ffmpeg.log.20150617
by burek 18 Jun '15

18 Jun '15

[00:17:57 CEST] Action: _DragonsLordKERN is away: AWAY [00:18:21 CEST] <_DragonsLord_> Good night to everyone [00:23:06 CEST] <Brandoney> ? [00:23:15 CEST] <Brandoney> Greetings bots. I want to capture 2 video cameras and live stream to 8-16 Internet users via http with a linux server. Is ffmpeg and ffserver the right choice for me? [00:36:29 CEST] <Brandoney> ffmpeg doesn't work on linux. [00:40:34 CEST] <Nuc1eoN> hi, I would like to demux an xvid video file from an avi container [00:40:55 CEST] <Nuc1eoN> would "mpeg" be the native output format? [00:41:16 CEST] <Nuc1eoN> actually I thought it would be ".xvid" [00:46:49 CEST] <dine909> Brandoney you serious? of course it works on linux [00:47:26 CEST] <Brandoney> Lol. No, not serious. Just lonely. [00:47:29 CEST] <Brandoney> Greetings bots. I want to capture 2 video cameras and live stream to 8-16 Internet users via http with a linux server. Is ffmpeg and ffserver the right choice for me? [00:48:27 CEST] <Anoia> what are they being viewed in? [00:48:37 CEST] <c_14> You can probably get ffserver to work for you, but something like hls or nginx-rtmp might be easier. [00:48:41 CEST] <Brandoney> Web browser. [00:48:44 CEST] <Anoia> http allows html5 video [00:49:00 CEST] <Anoia> flash players can pull RTSP or RTMP [00:49:30 CEST] <Brandoney> Will each simultaneous user have to use a different URL? [00:50:00 CEST] <Brandoney> So like a flash player embedded in my webpage/ [00:53:32 CEST] <Anoia> that's what we use [00:55:27 CEST] <Brandoney> Ok. I haven't heard of nginx-rtmp, but I'll check it out. I have used vlc years ago for multicast over LAN, but I need it over Internet. [00:56:34 CEST] <Brandoney> I read up on HLC but could not figure out how it's implemented. [00:57:05 CEST] <Brandoney> HLC = HLS [00:57:48 CEST] <Brandoney> Uh, I guess I should mention that I need to support smartphones including iphone. [01:00:17 CEST] <DHE> HLS is a text file containing a list of .ts files which play back when concatenated [01:00:40 CEST] <DHE> ffmpeg also supports HLS production for itself with a "-f hls" output format. then you could serve content with any ordinary http server [01:01:12 CEST] <DHE> I've played with it, it works but you really do need to check the -hls_* family of options and decide what you're up to [01:01:54 CEST] <Brandoney> So I guess I'm looking for camera -> hardware capture to h.264 -> ffmpeg -> (transcode to something) -> ffserver with copy -> HLS [01:04:16 CEST] <Brandoney> And it looks like I'll need some javascript to support IE? [01:09:54 CEST] <Nolski> Brandoney: pretty sure IE supports HTML5 video streaming [01:15:13 CEST] <Nolski> Can you concat a section onto itself? [01:15:40 CEST] <Nolski> I'm trying to cut a portion of the video out and concat it upon itself a number of times to loop a section of video [01:19:55 CEST] <pzich> you should be able to pass in the video as an input multiple times, or reference 0:v etc. multiple times [01:21:45 CEST] <Nolski> pzich: I think I might be doing it wrong then. Here is what I'm trying http://fpaste.org/232847/34496877/ [01:28:45 CEST] <Nuc1eoN> hi, I would like to demux an xvid video file from an avi container [01:28:47 CEST] <Nuc1eoN> would "mpeg" be the native output format? [01:28:57 CEST] <Nuc1eoN> actually I thought it would be ".xvid" [01:31:34 CEST] <Brandoney> Hmm, so I can use an mpeg capture card and then I just need to mux instead of transcode to get my mpeg-ts for HLS. [01:31:50 CEST] <Brandoney> Thanks for the help guys. You saved me countless hours. [01:35:49 CEST] <pzich> Nolski: hmm, I just ran `ffmpeg -i in.mp4 -filter_complex '[0:v]trim=start=10:end=20[v];[v][v]concat[out]' -map '[out]' out.mp4` and it seems to have worked [01:36:05 CEST] <Nolski> wat [01:36:13 CEST] <Nolski> hmmm [01:37:21 CEST] <Nolski> huh that works for me as well [01:38:37 CEST] <Nolski> Not sure why it didn't before. So I assume if I wanted to loop twice, I'd just do [out][v]concat[out] right? [01:38:41 CEST] <Nolski> at the end? [01:39:10 CEST] <pzich> I think you'll need a different input and output name, but that seems like it should work [01:42:20 CEST] Action: Nolski prays that numbers work as names [04:35:44 CEST] <xxashxx> how can i build ffmpeg for android? [04:36:14 CEST] <xxashxx> is there someone to answer my question? [05:26:57 CEST] <koz_> If I want to extract subtitles from a video that has them, what format should I put them into? [05:27:07 CEST] <koz_> I want to be able to read and edit them, obviously. [05:30:31 CEST] <kepstin> koz_, if you want a simple format to read and edit as text, use srt. If you want a format that preserves styling but it a bit more complex to edit (although it's well-supported by tools), use ass. [05:30:48 CEST] <kepstin> if they're image subs off a dvd or bluray, well, you're out of luck :/ [05:31:11 CEST] <kepstin> (gonna have to use external software to OCR them) [05:39:46 CEST] <koz_> kepstin: Thanks very much! How would I use ffmpeg to extract srt or ass? [05:49:57 CEST] <koz_> It says that there are 2 PGS text streams. [05:50:09 CEST] <koz_> Of which I assume one is the subtitle track. [05:51:25 CEST] <kepstin> pgs is bluray image subtitles [05:51:44 CEST] <kepstin> not text :) [05:51:59 CEST] <koz_> FFFFFFFF. [05:52:02 CEST] <koz_> Oh well. [05:52:13 CEST] <koz_> Looks like I'll need to look for a subtitle track. [05:52:20 CEST] <koz_> Thanks for all your help! [09:50:27 CEST] <pandb> how do i rectify something like this: [mp2 @ 0xc79c20] Specified sample_fmt is not supported. [10:00:55 CEST] <hardikj> hi, I am trying to run ffmpeg and libx264 with profile baseline but it does not accept both profile and profile:v [10:01:00 CEST] <hardikj> command I am using is ffmpeg -y -framerate 8 -f dshow [10:01:00 CEST] <hardikj> -f gdigrab -i "desktop" -c:v libx264 -vprofile baseline -crf 0 -preset ultrafast [10:01:00 CEST] <hardikj> -threads 0 temp.mkv [10:01:25 CEST] <relaxed> pandb: look at 'ffmpeg -h encoder=mp2' and set a valid rate with -r:a [10:02:13 CEST] <relaxed> pandb: well, I think. pastebin your command and all output if that doesn't work. [10:02:20 CEST] <hardikj> it gives me error saying incorrect options, the error is due to "vprofile flag" i am using [10:02:24 CEST] <hardikj> what am i missing ? [10:03:15 CEST] <relaxed> hardikj: you're trying baseline and lossless (-crf 0), which isn't possible [10:03:37 CEST] <hardikj> relaxed, I even tried it with crf 35 [10:04:15 CEST] <relaxed> oh, just use -profile [10:04:51 CEST] <hardikj> I removed the crf option tried "profile", "profile:v", "vprofile", no luck :( [10:04:54 CEST] <relaxed> or -profile:v [10:05:12 CEST] <relaxed> pastebin your command and output [10:05:34 CEST] <hardikj> ok [10:07:31 CEST] <hardikj> relaxed, http://paste.ubuntu.com/11729330/ [10:08:09 CEST] <hardikj> same output with profile [10:08:14 CEST] <hardikj> and profile:v [10:08:28 CEST] <relaxed> oh, add -pix_fmt yuv420p [10:08:53 CEST] <relaxed> the real error is, "x264 [error]: baseline profile doesn't support 4:4:4" [10:09:23 CEST] <hardikj> ah, nice catch [10:10:29 CEST] <hardikj> relaxed, it worked thanks [10:10:36 CEST] <hardikj> what does pix_fmt means? [10:12:08 CEST] <relaxed> pandb: forget what I said earlier, you want -sample_fmt s16 [10:13:15 CEST] <relaxed> hardikj: it sets the pixel format [10:13:26 CEST] <hardikj> ok, cool [10:19:24 CEST] <hardikj> relaxed, will using baseline profile help me reducing CPU power while screen capturing? [10:20:07 CEST] <relaxed> -preset ultrafast will [10:21:33 CEST] <hardikj> hmm, I am already using that, also is there libx264 a good choice if I am concerned about cpu usage, quality is not what I am aiming for [10:21:54 CEST] <relaxed> did you read https://trac.ffmpeg.org/wiki/Capture/Desktop ? [11:23:58 CEST] <Guest40450> BtbN: I want to log pts_time [11:47:58 CEST] <bigzed> Hello guys, when I decode raw video in ffmpeg like this `ffmpeg -f rawvideo -pix_fmt <fmt> -i - -y out.mp4`. Which pix_fmt is the correct one for PixelType 16Bit RGB Planar and PixelType 16Bit RGB interleaved? [11:54:11 CEST] <durandal_1707> bigzed: what endianess? [11:55:35 CEST] <durandal_1707> if le: gbrp16le and rgb48le, if no alpha [11:55:54 CEST] <bigzed> I'm not sure. The input is RED footage decode via there SDK into `rawvideo` and their documentation doesn't say anything about endianess [11:56:45 CEST] <bigzed> They offer to choose the `VideoPixelType` e.g.: PixelType_16Bit_RGB_Planar, PixelType_16Bit_RGB_Interleaved, PixelType_8Bit_BGRA_Interleaved, PixelType_10Bit_DPX_MethodB, PixelType_12Bit_BGR_Interleaved,.... [11:57:54 CEST] <bigzed> ok, I'll try these :) [13:55:34 CEST] <bigzed> durandal_1707, how do I shuffle planes? [14:02:44 CEST] <durandal_1707> there is filter with [14:02:57 CEST] <durandal_1707> same name [14:10:16 CEST] <bigzed> thanks, shuffling planes fixed it for me. Now the colors are finally the same :) [14:15:23 CEST] <kubast2> Hey ,how can I[on windows] convert multiply files[9 files in the same folder] dash audio container[youtube-dl] to m4a[aac ,as 9 files in a diffrent folder] ? [14:19:15 CEST] <sp00x> kubast2: see: for %f in (..) do ffmpeg .. and then use %~nf etc vars they're documented somewhere [14:19:50 CEST] <sp00x> kubast2: for /? [14:32:56 CEST] <Anoia> hand [14:33:03 CEST] <Anoia> erm, wrong window [15:00:26 CEST] <bigzed> durandal_1707, I still have a question left about the pixel_formats. I tried the shuffleplanes with jpg's and it worked as expected. But as soon as I try to convert the whole video instead of just a single frame into mp4 it crashes because the filter is not aplyable. Probably because mp4 needs yuv... as a pixfmt [15:00:57 CEST] <bigzed> is it possible to tell ffmpeg to use the rawvideo, remapp the channels and use the remapped channels to create a yuv... pix fmt? [15:02:41 CEST] <durandal_1707> crashes? [15:03:12 CEST] <bigzed> crashing is not the correct term. It renders an error: Cannot map between a subsampled chroma plane and a luma or alpha plane. Failed to configure input pad on Parsed_shuffleplanes_3 [15:04:25 CEST] <durandal_1707> hmm pastebin whole command and output [15:04:38 CEST] <bigzed> sure [15:05:45 CEST] <bigzed> https://gist.github.com/bigzed/b587ab4ffca2d0948166 [15:09:55 CEST] <durandal_1707> remove last pixfmt yuv420p and use format filter: format=yuv420p after shuffleplanes filter [15:11:49 CEST] <bigzed> Still fails with the same error https://gist.github.com/bigzed/b587ab4ffca2d0948166#file-snd-output [15:12:24 CEST] <bigzed> :( [15:16:30 CEST] <bigzed> I removed all unecessary stuff from the command and now it works. Thanks so far. I will investigate it further :) [15:21:57 CEST] <bigzed> The shuffleplanes and format filter have to be the first 2 filters [15:22:02 CEST] <bigzed> then it works [15:22:37 CEST] <durandal_1707> Yes I was just planed to write that [15:33:01 CEST] <XwZ> hi, I have an issue, I try to extract images from an avi file from 04:00 to 05:30 so I use : ffmpeg -t 00:01:30 -ss 00:04:00 -i my.avi images/img_%05d.pgm but I have only black images [15:33:28 CEST] <XwZ> my avi file info are : Stream #0:0: Video: rawvideo, pal8, 2048x2048, 838957 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc [15:36:20 CEST] <relaxed> XwZ: did you try with -t and -ss after the input? [15:36:31 CEST] <XwZ> nope, I will [15:39:26 CEST] <relaxed> if that doesn't work pastebin your command and output [15:42:16 CEST] <XwZ> it "works" better but I have bunch of this http://pastebin.com/gVYbS076 and the result is http://pastebin.com/D8QqypAJ [15:45:12 CEST] <relaxed> works better? [15:45:58 CEST] <XwZ> it seems it try to reach the 4th minutes but nothing dumped [15:46:46 CEST] <relaxed> "if that doesn't work pastebin your command and output" [16:03:10 CEST] <XwZ> relaxed: ok well, now it works ... thank you :) [16:08:46 CEST] <lucascastro> Hey, I'm trying setup ffserver, but when I stream, the server return File '/feed1.ffm' not found [16:09:01 CEST] <lucascastro> can someone help me ? [16:35:32 CEST] <ValdikSS> Hello guys. Is Intel QSV is available in Linux? [17:43:39 CEST] <Timster> Hey, guys - I have a problem with ffmpeg reencoding a Livestream and changing Keyframe Interval / gop values. My forum post is explaining it in detail - http://ffmpeg.gusari.org/viewtopic.php?f=11&t=2164 - does somebody have an idea what could cause this? :/ [18:07:30 CEST] <DHE> so you want the re-encoded version to have its GOP settings match what the inputs? [18:08:26 CEST] <jbruggeman> Hi. I'm messing around with the ffmpeg assembly, and i keep getting an error 'libavcodec/x86/ac3dsp.d:1: *** missing separator. Stop.' [18:09:07 CEST] <jbruggeman> I see this after it builds during a 'make install' as we as for an incremental build [18:09:11 CEST] <sq6rr> Hi. I have two video inputs (BT878) and one output http. how switch input on the fly. [18:10:05 CEST] <DHE> jbruggeman: sounds like a tab symbol got turned into spaces [18:10:20 CEST] <DHE> or something else ended up mangling the .d file [18:10:46 CEST] <jbruggeman> hmmm, is there a specification for a .d file? [18:10:59 CEST] <jbruggeman> mine just has a path inside it to an asm file [18:11:50 CEST] <DHE> they're Makefile definitions [18:12:07 CEST] <DHE> naming a file and its build dependencies, typically a list of #include files involved [18:12:45 CEST] <jbruggeman> Cool. Where do those get generated? Is it through ./configure? [18:12:47 CEST] <Timster> DHE: Yes, the inputs are 40fps and Keyframe Interval of 2 seconds. I thought that fixing it with -r 40 and -g 80 in ffmpeg might help, but unfortunately it didn't. [18:19:39 CEST] <Timster> DHE: You think that copying fps and gop from the inputs might help? [18:22:27 CEST] <DHE> Timster: I gave no advice, I was just asking for clarification [18:24:15 CEST] <Timster> DHE: Thanks anyways. [18:24:30 CEST] <Timster> Does anybody have an idea regarding this? [18:26:25 CEST] <feliwir> where can i ask generel questions about video decoding? [18:39:14 CEST] <Nolski> I'm having a bit trouble trimming sections of a video (along with audio) and concating them to create loops of video. My output is here. https://gist.github.com/Nolski/2a9176392842ab688a1a [18:39:48 CEST] <Nolski> The video has an audio track associated with it so I figured [0:a] and atrim would work just fine. If I remove the audio trimming, it seems to work [19:05:55 CEST] <savah> hi guys [19:06:04 CEST] <savah> anyone has used ffmpeg on ios ? [19:15:26 CEST] <feliwir> savah, vlc has an ios app and they use ffmpeg [19:16:01 CEST] <savah> is it legal ? [19:16:21 CEST] <savah> since i've read a lot of stuff about it being illegal [19:16:36 CEST] <Nolski> savah: yes vlc is totally illegal [19:16:47 CEST] <feliwir> of course it is otherwise it wouldn't be in the appstore [19:17:35 CEST] <Nolski> Why is looping sections of video in ffmpeg so freaking impossible. [19:18:17 CEST] <feliwir> someone here is an vp6 expert by any chance? [21:19:34 CEST] <mkoskar> hi all, I'm trying to figure out why following video doesn't stream: ffplay 'https://people.gnome.org/~carlosg/Screencast%20from%2006-17-2015%2007:09:01…' [21:20:28 CEST] <mkoskar> there is some TLS warning generated though, certificate seems ok, I'm able to download it with other standard tools like e.g., curl [21:22:06 CEST] <c_14> Works for me. [21:22:34 CEST] <mkoskar> O_O exact command as I've posted? [21:23:46 CEST] <c_14> Well, I used ffmpeg -i url -c copy out.webm because I don't have ffplay, but ye. [21:23:58 CEST] <mkoskar> I got [tls @ 0x7f2e18004b80] A TLS warning alert has been received.=0/0 ... followed by line ending with Input/output error [21:24:05 CEST] <BtbN> does your ffmpeg support tls? [21:24:28 CEST] <c_14> If it wasn't built with ssl support you'd get a different error. [21:24:51 CEST] <BtbN> Hm, does it know any root certs then? [21:25:29 CEST] <mkoskar> BtbN: I did pastebin ... https://ptpb.pw/G0t8 [21:27:21 CEST] <mkoskar> I admit I don't know much about ffmpeg, ... I assume --enable-gnutls is OK, for reference it's current ArchLinux version on updated system [21:27:26 CEST] <_DragonsLord_> hello everybody [21:27:31 CEST] Action: _DragonsLord_ is back (gone 21:09:34) [21:28:33 CEST] <mkoskar> brb [21:43:01 CEST] <mkoskar> BtbN: I've passed -cafile /etc/ssl/cert.pem ... that being ArchLinux's CA bundle, so I think that wouldn't be an issue [21:51:23 CEST] <DRAWNBOX> does anyone know how I could reduce the file size of a gif? [22:01:07 CEST] <Nolski> DRAWNBOX: make it smaller [22:01:17 CEST] <Nolski> DRAWNBOX: make it have less colors [22:02:02 CEST] <mkoskar> c_14: you said it is working for you ... could you please pastebin: ffmpeg -loglevel trace -i https://people.gnome.org/\~carlosg/Screencast%20from%2006-17-2015%2007:09:0… -c copy out.webm [22:02:24 CEST] <mkoskar> c_14: I'm filling a bug report for Arch and it would be helpful I think [22:03:13 CEST] <c_14> Sure, but I'll limit it to the first 10s with -t 10 [22:04:26 CEST] <c_14> http://pb.c-14.de/t/kng.CIg1Ly [22:04:56 CEST] <mkoskar> c_14: will do, thanks a bunch ;) [22:12:46 CEST] <feliwir> i ask again? Someone here familiar with the vp6 codec? [22:17:24 CEST] <feliwir> i don't see why this https://github.com/Fluffiest/splayer/blob/3482b0ffff02528a774d1888523bd3e61… doesn't work for me [23:04:32 CEST] <feliwir> can anyone please look my vp6 decoder? [23:08:12 CEST] <durandal_1707> this is ffmpeg channel and not random decoder support channel [23:53:43 CEST] <Nolski> I'm having a bit trouble trimming sections of a video (along with audio) and concating them to create loops of video. My output is here. https://gist.github.com/Nolski/2a9176392842ab688a1a [23:53:50 CEST] <Nolski> The video has an audio track associated with it so I figured [0:a] and atrim would work just fine. If I remove the audio trimming, it seems to work [23:54:30 CEST] <Nolski> is ffmpeg meant to do these sorts of thibgs or should i look for another tool? [23:55:08 CEST] <c_14> Nolski: try zeroing the pts for each of the segments [23:55:17 CEST] <c_14> eh [23:55:20 CEST] <c_14> and your concats are messed up [23:55:47 CEST] <c_14> try [v][a][v][a]concat=n=2:v=1:a=1[v1][a1] [23:56:00 CEST] <c_14> Though you might have to run the v/a through a split first [23:56:01 CEST] <c_14> not sure though [23:57:41 CEST] <Nolski> woah, uh... could you explain that to me a bit? why are there 2 [v][a]s? [23:58:35 CEST] <c_14> Same reason you had [v][v] and [a][a] before [00:00:00 CEST] --- Thu Jun 18 2015

1 0

ffmpeg-devel.log.20150616
by burek 17 Jun '15

17 Jun '15

[00:28:53 CEST] <philipl> what are the chances the new nvidia driver release fixes any of my problems? Low. [03:53:55 CEST] <cone-906> ffmpeg 03Christian Suloway 07master:907ac20aa293: avformat/hlsenc: added HLS encryption [07:32:59 CEST] <philipl> Hey everyone. I've been thinking about how we (mechanically) do code reviews. At my paid job, we use review board for reviews, and I find it a far more pleasant experience than doing reviews by email. It makes it much easier to do inline comments, to track whether fix requests are acknowledged, and whether a change is committed. [07:33:31 CEST] <philipl> I think it's really great software for doing reviews, and we'd probably benefit a lot from using it, or a similar tool. [07:34:00 CEST] <philipl> It can send review requests and replies to email so there would still be a record in the mailing list archives. [07:34:05 CEST] <philipl> https://www.reviewboard.org/ [07:34:08 CEST] <philipl> Just a thought. [07:35:19 CEST] <BBB> Im partially with that idea [07:35:28 CEST] <BBB> (I personally like gerrit but I guess I dont care what we use) [07:35:53 CEST] <BBB> having email notifications is obviously useful, but reviewboards add some extra stuff to reviews imo [07:36:14 CEST] <BBB> like maintaining comments between iterations, easy ways to see diffs between diffs, etc. [07:36:24 CEST] <philipl> Yeah. Lots of good stuff. [07:36:56 CEST] <philipl> Gerrit has strong workflow features like pre-checkin staging and tests, but reviewboard is a far better tool for actually reviewing code. [07:40:01 CEST] <BBB> sounds good to me then :) [07:41:51 CEST] Action: BBB goes sleep [08:39:08 CEST] <j-b> 'morning [09:10:47 CEST] <durandal_1707> good morning [12:23:09 CEST] <cone-928> ffmpeg 03James Zern 07master:202188a26c1a: doc/muxers.texi: properly insert {}s in example [13:03:30 CEST] <cone-928> ffmpeg 03George Boyle 07master:2469ed32c81e: avcodec/flacenc: Fix Invalid Rice order [13:29:21 CEST] <cone-928> ffmpeg 03Michael Niedermayer 07master:cf11fd632eef: avformat/mxfdec: Change version byte for JPEG2000 to match mxf.c and RP224v12 [14:10:34 CEST] <ubitux> int32_t fmt = 'avc1' [14:10:38 CEST] <ubitux> damn, this is valid code? [14:11:35 CEST] <nevcairiel> Surprisingly yes [14:18:30 CEST] <__gb__> dates back from apple's fourcc codes for quicktime [14:18:56 CEST] <__gb__> imho, not standard C anyway [14:20:51 CEST] <__gb__> err, not specific to quicktime, but resource names -- pff very old things :) [14:21:06 CEST] <__gb__> circa mac/68k [16:02:12 CEST] <ubitux> is av_vda_default_init(fmt_ctx->streams[X]->codec) supposed to work out of the box or i missed a step? [16:03:39 CEST] <ubitux> i don't see how tweaking the avctx is going to change anything, so i'm a bit confused at why it fails badly (the classic err -12473) [16:13:24 CEST] <wm4> uh [16:13:33 CEST] <wm4> don't use the demuxer codec context for decoding [16:14:10 CEST] <wm4> but other than that, and implementing get_format, yes [16:14:25 CEST] <nevcairiel> vda is like the easiest to get going [16:14:29 CEST] <wm4> (you call av_vda_default_init() from get_format) [16:14:40 CEST] <wm4> did I mention yet that the hwaccel API sucks? especially get_format [16:15:01 CEST] <nevcairiel> get_format aint too bad, its just not very intuitive [16:15:38 CEST] <wm4> it could be as simple as having to select a h264/vda-hwaccel specific decoder [16:15:45 CEST] <nevcairiel> For a long time I didn't even init in get_format, i did that before the _open call, but I changed it some time ago because it works better with streams that change format [16:16:13 CEST] <nevcairiel> well this allows you to bail out of hwaccel when format goes incompat [16:16:28 CEST] <nevcairiel> although of course that could be implemented some other way [16:16:48 CEST] <nevcairiel> i do not like falling back to software in the same context anyway [16:16:54 CEST] <wm4> the decode call could just return an appropriate status code and return all unconsumed packets or something [16:19:05 CEST] <ubitux> wm4: ah? i need to create another context? [16:19:36 CEST] <wm4> ubitux: of course [16:19:44 CEST] <nevcairiel> you can use the demuxing context, but its deprecated and not recommended [16:19:53 CEST] <nevcairiel> so yeah, better start on the right path [16:19:57 CEST] <ubitux> mmh; is there some doc about that? [16:20:08 CEST] <ubitux> i'm still very much confused on how to use that api [16:20:13 CEST] <wm4> haha [16:20:16 CEST] <ubitux> :( [16:20:22 CEST] <Daemon404> >own devs cant handle api [16:20:26 CEST] <wm4> "ffmpeg dev attempts to use ffmpeg API" [16:20:39 CEST] <wm4> makers great banter [16:20:43 CEST] <wm4> *makes [16:20:53 CEST] <ubitux> :D [16:21:09 CEST] <wm4> ubitux: the current VDA API is approximately infinitely better than the old one [16:21:28 CEST] <ubitux> no doubt about it, but i'm unable to make a distinction between the two anyway [16:21:45 CEST] <ubitux> why do you need 2 decoding context btw? [16:21:57 CEST] <wm4> because what happens with midstream parameter changes? [16:22:05 CEST] <wm4> then you'd have 4 sets of parameters [16:22:26 CEST] <nevcairiel> its not like its hard to setup the new one, there is an API to copy it [16:22:36 CEST] <ubitux> nevcairiel: where? [16:22:43 CEST] <wm4> the user-visible ones in AVCodecContext (1 for decoding, 1 for demuxing), and 2 internal ones (1 for the decoder thread, 1 for the state behind the packet queue) [16:23:15 CEST] <kierank> 3:20 PM <"wm4> makers great banter [16:23:19 CEST] <kierank> archbishop of banterbury [16:23:23 CEST] <kierank> bantersaurus rex [16:23:42 CEST] <nevcairiel> ubitux: pDecodingContext = avcodec_alloc_context3(NULL); avcodec_copy_context(pDecodingContext, fmt_ctx->streams[X]->codec); [16:24:30 CEST] <ubitux> ok ok; but how is that necessary to make av_vda_default_init() not fail? [16:24:41 CEST] <nevcairiel> dunno [16:24:46 CEST] <nevcairiel> what does it say [16:24:50 CEST] <nevcairiel> it should log an output when it fails [16:25:42 CEST] <ubitux> it fails just like if i had no video output (same failure as when i'm in ssh iirc), while ffmpeg works (in the same environment) [16:25:47 CEST] <ubitux> gonna paste the exact error, just a sec [16:26:36 CEST] <ubitux> vtDecompressionDuctCreate signalled err=-8973 (err) (Could not select and open decoder instance) at /SourceCache/CoreMedia_frameworks/CoreMedia-1562.235/Sources/VideoToolbox/VTDecompressionSession.c line 1181 [16:26:38 CEST] <ubitux> [NULL @ 0x7fec2b006600] Cannot initialize VDA -12473 [16:27:22 CEST] <wm4> that happens to me too with some videos [16:27:29 CEST] <wm4> or if another application uses VDA [16:27:32 CEST] <wm4> (like Safari) [16:27:44 CEST] <ubitux> the same videos work with ffmpeg [16:28:19 CEST] <nevcairiel> are you sure ffmpeg uses vda and doesnt just swallow the error? [16:29:08 CEST] <ubitux> pretty sure it does, but i can recheck [16:31:01 CEST] <nevcairiel> maybe the app needs to do some magic to init something? I havent actually used it outside of ffmepg yet [16:31:06 CEST] <nevcairiel> although i really should already [16:31:10 CEST] <wm4> I had lots of trouble figuring out whether ffmpeg actually used hwaccel [16:31:19 CEST] <wm4> actually, I think I could never confirm it [16:31:39 CEST] <nevcairiel> you can really tell by the output format [16:31:52 CEST] <nevcairiel> its usually nv12 for sane hwaccels or UYVY422 for vda [16:32:38 CEST] <wm4> yeah, but where does it print this format? [16:32:55 CEST] <nevcairiel> it usually mentions that when it auto-inserts the scaling filter to convert it [16:34:24 CEST] <ubitux> arg, i'm an idiot, i wasn't using the same video input [16:57:46 CEST] <Fyr> how long does it take for developers to confirm and fix a simple bug? [16:57:54 CEST] <Fyr> *the simplest [16:59:32 CEST] <wm4> 1 minute [16:59:45 CEST] <ubitux> wm4 is accepting the challenge [17:00:02 CEST] <ubitux> so mmh, vda related; cv_pix_fmt_type is not always honored? [17:00:03 CEST] <wm4> a bug can be easily reclassified as "not simple" [17:00:22 CEST] <wm4> ubitux: dunno, I suppose it depends on the hw support? [17:03:39 CEST] <ubitux> :( [17:05:12 CEST] <wm4> ubitux: what are you trying to achieve? [17:05:49 CEST] <ubitux> i was just requesting rgb32 [17:06:02 CEST] <cone-928> ffmpeg 03Christian Suloway 07master:7e871d6999fd: avformat/hlsenc: removed empty/unused print_encryption_tag function [17:06:03 CEST] <cone-928> ffmpeg 03Christian Suloway 07master:918abc19ff74: MAINTAINERS: add myself as hls encryption maintainer [17:06:07 CEST] <ubitux> and expecting the same pix fmt in CVPixelBufferGetPixelFormatType(frame->data[3]) in the get_frame callback [17:06:25 CEST] <ubitux> kCVPixelFormatType_32BGRA to be more specific [17:09:53 CEST] <ubitux> well i guess i'm derping again [17:12:49 CEST] <wm4> you expect it to decode to rgba? [17:13:01 CEST] <wm4> I know packed yuv and (on newer systems) nv12 are possible [17:13:26 CEST] <ubitux> i was assuming it was capable of converting [17:14:24 CEST] <ubitux> but the problem is elsewhere anyway [17:54:25 CEST] <Daemon404> innocent soul trying to suggest a code review tool on the ML [17:54:59 CEST] <kierank> yeah [17:56:13 CEST] <thardin> have they missed that that's what the ML is for? [17:57:06 CEST] <Daemon404> just the usual "ML is crude, i prefer web based stuff" thing [17:57:56 CEST] <nevcairiel> use a web mail client [17:57:59 CEST] Action: nevcairiel ducks [18:27:38 CEST] <philipl> innocent soul is me [18:28:21 CEST] <philipl> I'm used to the cumudgeny response. [18:50:12 CEST] <cone-928> ffmpeg 03Michael Niedermayer 07master:36241c405541: avcodec/jpeg2000dec: Fix term_cnt check [18:50:14 CEST] <cone-928> ffmpeg 03Michael Niedermayer 07master:20657e05bc26: avcodec/jpeg2000dec: Improve end check [20:49:51 CEST] <cone-928> ffmpeg 03Michael Niedermayer 07master:8294ec67b5b1: avcodec/jpeg2000dwt: add special cases for handling length 1 in 5/3 decode [20:49:52 CEST] <cone-928> ffmpeg 03Michael Niedermayer 07master:34121ca18734: avcodec/jpeg2000dwt: Fix 9/7 IDWT for small sizes [21:19:06 CEST] <cone-928> ffmpeg 03Paul B Mahol 07master:e0dac4030692: avcodec/codec_desc: rename smv video codec name to smvjpeg to match decoder name [21:59:39 CEST] <cone-928> ffmpeg 03Andreas Cadhalpun 07master:5b76c82fd7a5: matroskadec: check audio sample rate [21:59:40 CEST] <cone-928> ffmpeg 03Andreas Cadhalpun 07master:92e79a2f7bf2: matroskadec: validate audio channels and bitdepth [21:59:41 CEST] <cone-928> ffmpeg 03Andreas Cadhalpun 07master:72654526e448: configure: make makeinfo_html check more robust [21:59:42 CEST] <cone-928> ffmpeg 03Andreas Cadhalpun 07master:28efeb6502df: doc: avoid incorrect phrase 'allows to' [22:21:35 CEST] <wm4> michaelni: apparently I'm still getting "full chroma interpolation for destination format 'rgb48be' not yet implemented" [22:21:42 CEST] <wm4> michaelni: what does it mean? [22:21:46 CEST] <wm4> (ffmpeg 2.7 and git) [23:01:07 CEST] <durandal_1707> this faasoft shift uses ffmpeg with its "own" 4gv decoder [23:01:29 CEST] <durandal_1707> *shit [23:11:48 CEST] <michaelni> wm4, it means theres a code path missing ... wasnt aware of that being the case for rgb48be [23:12:34 CEST] <wm4> (it's needed when writing 16 bit PNGs) [23:37:04 CEST] <wm4> av_crc_get_table() isn't thread-safe under some configurations [23:37:06 CEST] <wm4> hate [00:00:00 CEST] --- Wed Jun 17 2015

1 0

ffmpeg.log.20150616
by burek 17 Jun '15

17 Jun '15

[00:05:43 CEST] <podman> so what happened to ffmpeg.org? [00:12:24 CEST] <Nolski> Hey all, I'm a bit new to ffmpeg and I'm trying to skip or cut out sections of a clip (like play 0-30 and 45-end but not 30-45) [00:14:00 CEST] <Nolski> I found an example of a command which seems to demonstrate it by using trim and concat but it seems to trim in two spots and I was wondering if someone could help break it down for me. http://fpaste.org/232361/40640214/ [00:14:58 CEST] <techtopia> your better off cutting before hand with another app then encoding a solid file with ffmpeg [00:15:15 CEST] <techtopia> but you can use -ss to mark the start point of the encode [00:15:24 CEST] <techtopia> and -t to specify a duration [00:18:59 CEST] <Nolski> techtopia: Unfortunatly the entire file has to be processed with ffmpeg. The application I'm writing is using ffmpeg to encode video edits from a web editor into a flat video file [00:21:03 CEST] <Nolski> From the snippet it looks like it seeks to 30 seconds (trim=duration=30) skips to 40 seconds (trim=start=40) but I'm not really sure what STARTPTS is for [00:21:08 CEST] <pzich> is that trim and concat command not working, or you're just trying to understand it? [00:21:22 CEST] <Nolski> pzich: I'm just trying to understand it [00:21:41 CEST] <Nolski> (it works) [00:22:33 CEST] <Nolski> Oh also, I'm not sure if anyone here is hosting ffmpeg.org but it seems that the website is down (connecting from California in the US) [00:22:52 CEST] <pzich> and how familiar are you with the details of video encoding? the PTS is a technical thing that you don't necessarily need to understand [00:23:04 CEST] <pzich> yeah, I think there are ongoing issues with the site ATM [00:24:11 CEST] <Nolski> pzich: just the basics. [00:25:30 CEST] <pzich> it's probably not worth getting into the details, but basically in this case you need it when you're not starting at the start, e.g. seeking with a start time in trim [00:27:10 CEST] <Nolski> Ahhh I see. [00:29:06 CEST] <Nolski> pzich: so on line 2, it says trim=duration=30 but it looks like it only cuts out 10 seconds (starting at the 30 second mark). Is that line just seeking to the 30 second mark? [00:31:57 CEST] <pzich> it's trimming from 0 to 30 seconds (and keeping it) [00:32:11 CEST] <pzich> it's cutting out 10 seconds because the next trim starts at 40 and 40 - 30 = 10 [00:34:20 CEST] <Nolski> Ooooh! I get it! [00:34:35 CEST] <Nolski> drawing this stuff out on paper helps. Thanks pzich! [00:35:14 CEST] <pzich> no problem [01:00:30 CEST] <Zeranoe> Has anyone experimented with livestreaming HTML5 video with FFmpeg? [01:48:15 CEST] <Jekar> Evening all! So, I am completely new to using ffmpeg and so far it's been fairly straightforward. I was just looking for some advice to reduce lag/frame drop when grabbing audio/video from games. I'm using the following command: [01:48:39 CEST] <Jekar> ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1024x768 -i :0.0 -acodec pcm_s16le -vcodec libx264 -preset ultrafast -crf 0 -threads 0 (my-output).mkv [02:36:56 CEST] <voip_> hello guys [02:37:51 CEST] <voip_> did you try use ffmpeg on Rasbery pi board as live stream transcoder h264? [03:23:42 CEST] <zachary> What is the difference between `-r` and `-framerate`? [03:23:54 CEST] <zachary> I cannot find information on the official site. [03:38:08 CEST] <roboman2444> i think one of them is related to input fps [03:39:24 CEST] <c_14> Both are/can be. -r is the general option to override input fps detection, -framerate is an option specific to several demuxers that sets the fps [03:42:21 CEST] <zachary> ambiguous [03:42:36 CEST] <zachary> why the official site don't give a explanation [03:42:44 CEST] <c_14> Use -framerate for the demuxers that support it, otherwise use -r [03:44:09 CEST] <c_14> >As an input option, ignore any timestamps stored in the file and instead generate timestamps assuming constant frame rate fps. This is not the same as the -framerate option used for some input formats like image2 or v4l2 (it used to be the same in older versions of FFmpeg). If in doubt use -framerate instead of the input option -r [03:44:32 CEST] <kyleogrg> hello [03:45:00 CEST] <kyleogrg> I have a question, probably simple, about an interlaced/IVTC DVD. [03:45:11 CEST] <kyleogrg> It's a PAL-region DVD. [03:45:46 CEST] <kyleogrg> I frame-stepped through a small portion in VLC and came up with this pattern (1 = looks interlaced, 0 = doesn't): [03:45:50 CEST] <kyleogrg> 10000000000000011111111111100000000000001111111111110 [03:46:20 CEST] <Kuukunen> zachary: ffmpeg -h full gives you something [03:46:25 CEST] <kyleogrg> What is this pattern? [03:48:51 CEST] <zachary> Kuukunen: so for input device, use -framerate is a secure option [03:49:21 CEST] <Kuukunen> zachary: for the demuxers that have it [03:49:24 CEST] <Kuukunen> I guess [03:50:50 CEST] <kyleogrg> Could this pattern be a 2:2 pulldown? [03:52:19 CEST] <kepstin> kyleogrg, looks like a 25:24 pulldown to me [03:52:29 CEST] <kepstin> a 2:2 pulldown looks like constant progressive video [03:52:55 CEST] <kyleogrg> Yes. I have several PAL DVDs, and they look progressive, except this one. [03:53:12 CEST] <kepstin> this looks like they tried to change the video to 50 fields per second at the same speed, rather than speed up from 24 to 25fps like most films. [03:53:13 CEST] <kyleogrg> You're saying maybe this was converted from 25 to 24? [03:53:21 CEST] <kepstin> from 24 to 25 [03:53:38 CEST] <kyleogrg> hmm. this is a double-sided dvd -- the other side is ntsc [03:54:20 CEST] <kyleogrg> i know you can IVTC with NTSC [03:54:24 CEST] <kepstin> then rip the ntsc one, it might be easier. probably a standard 3:2 pulldown [03:54:40 CEST] <kepstin> (use the 'pullup' or 'filmdint' filters with default options to reverse that) [03:55:02 CEST] <kyleogrg> can the 25:24 pulldown be undone? [03:55:31 CEST] <kepstin> sure, but you'll have to play around with the detelecining filters to convince them to do it. [03:55:39 CEST] <kyleogrg> sure [03:55:55 CEST] <kepstin> and it's not something i've done before, living in north america as i do ;) [03:56:07 CEST] <kyleogrg> haha so do i [03:58:44 CEST] <kyleogrg> what's the format for specifying a custom detelecine pattern (in handbrake, for example)? [04:38:56 CEST] <zachary> Cannot grab x11 Desktop: http://pastebin.com/4gXu48L6 [04:39:08 CEST] <zachary> which part should I modify my screen [04:39:26 CEST] <zachary> which parameters should I modify [04:50:05 CEST] <Quixxx> Is a lower crf higher quality? [04:55:12 CEST] <Devrim> yes [05:04:21 CEST] <chungy> zachary: check your audio input [05:04:35 CEST] <zachary> my audio input is OK. [05:04:54 CEST] <chungy> ffmpeg isn't finding it ok [05:05:25 CEST] <chungy> double-check "ffmpeg -sinks" to see what you have. looks like you have wrong options there. [05:05:31 CEST] <zachary> chungy: when I remove the `scale=w=iw/2:h=ih/2` part, it works now. [05:06:05 CEST] <zachary> chungy: what -sinks does? [05:06:15 CEST] <chungy> list the sinks on your system [05:06:28 CEST] <Quixxx> I don't have any sinks, my PC is air cooled [05:06:34 CEST] <Quixxx> c: [05:07:01 CEST] <chungy> also 1362x766 sounds like you're working around an ffmpeg bug with x11grab :P [05:07:27 CEST] <zachary> chungy: what does it mean? My screen resolution is 1366*768 [05:07:48 CEST] <zachary> chungy: so i specify the video_size to 1362*766 [05:08:23 CEST] <zachary> chungy: the small reduction is that x11grab cannot find position 0,0 on screen. [05:08:29 CEST] <chungy> Right; the tip of the release/2.6 branch has a fix for that. [05:09:16 CEST] <zachary> chungy: you are right. If I change 1362*768 to 1280*720, it also works. [07:27:06 CEST] <apaul> I am using ffmpeg commandline to extract pts_time and save them in log.txt file.I want to know how can I append current system time in front of each pts_time value. [07:27:36 CEST] <apaul> here is command: ffmpeg -i rtsp://192.168.0.127 -vstats_file vstats.log -map 0:0 -map 0:1 -s 640x360 -vcodec libx264 -g 60 -vb 500000 -filter:v "showinfo" -strict experimental -an -y -f segment -segment_time 20 "part-%03d.ts" 2>&1 | grep pts_time | awk '{print $8}'| tee log.txt | grep pts_time [07:57:25 CEST] <apaul> Anoia: I am using ffmpeg commandline to extract pts_time and save them in log.txt file.I want to know how can I append current system time in front of each pts_time value. [07:57:26 CEST] <apaul> <apaul> here is command: ffmpeg -i rtsp://192.168.0.127 -vstats_file vstats.log -map 0:0 -map 0:1 -s 640x360 -vcodec libx264 -g 60 -vb 500000 -filter:v "showinfo" -strict experimental -an -y -f segment -segment_time 20 "part-%03d.ts" 2>&1 | grep pts_time | awk '{print $8}'| tee log.txt | grep pts_time [08:17:52 CEST] <roboman2444> im using libavcodec, and the only input format it will allow me to do is YUV420P [08:18:12 CEST] <roboman2444> can i use BGRA or other as an input pixel format when encoding with h264 or mpeg1video? [09:10:21 CEST] <durandal_1707> for rgba use libx264rgb encoder [09:47:41 CEST] <termos> is sample size the same as AVCodecContext::frame_size in libffmpeg? [09:54:46 CEST] <termos> oh, it's of course specified by the sample format [09:55:42 CEST] <baadf00d> Hello. I'm trying to sync multiple video streams from different rtsp servers by the wallclock time. I'm using libVLC to play the video, and it doesn't support reading creation_time metadata. Is there any way to save the start time of the video in some other place [09:55:54 CEST] <baadf00d> filename for example [09:56:33 CEST] <baadf00d> I found this https://ffmpeg.org/pipermail/ffmpeg-user/2014-March/020393.html But it didnt' seem to work [10:01:22 CEST] <baadf00d> One thing i'm wondering, how can i verify that my file has the correct value in start_time_realtime? http://ffmpeg.org/doxygen/trunk/structAVFormatContext.html#aa5ddb5cee1df28f… [10:01:41 CEST] <baadf00d> I didn't find a way to see if ffprobe gave it somehow [10:06:24 CEST] <Anoia> apaul: your question is unrelated to ffmpeg, you just need to append the time int he awk string, but they'll all have the time you run the command [10:09:14 CEST] <ac_slater_> curious, what happens if I give a mux (output) context a stream with a weird timebase? Or, rather, what if I have a container like MPEGTS and I have a stream with 25/1 and another with 72/1 ? [10:33:37 CEST] <anshul_> I am using yuv file in input, ffmpeg gives warning Past duration 0.639992 too large [10:34:32 CEST] <anshul_> I suppose there are no timestamp in my yuv file, then what is sense of pastduration warning [10:46:39 CEST] <anshul_> this warning come when i give x264opts level=4.0 [10:49:14 CEST] <anshul_> sry i was wrong those warning are when I give -r 25 [13:50:51 CEST] <feliwir> hey, i asked this a couple days ago already, but does someone here still have libvp62 on his pc by any chance? [13:58:17 CEST] <xtemp09> guys, I've posted a bug (something incredibly simple). when should I expect it be fixed? I looked through open bugs and lost my courage. =( [14:07:02 CEST] <apaul> how to get output filename from OutputFile structure in ffmpeg.c code [14:07:49 CEST] <apaul> I am using OutputFile *file ; file->ctx->filename ; [14:41:54 CEST] <anshul_> apaul: what is it you trying to do [14:46:11 CEST] <anshul_> in ffmpeg if i use ultrafast h264 preset then in H264 sps it says that profile is Main [14:47:47 CEST] <anshul_> since in utlrafast dct8x8 is not selected and in h264 while making sps it check for dct8x8 or i_cqm_preset [14:48:12 CEST] <BtbN> x264 ultrafast should result in contstrained baseline [15:25:36 CEST] <anshul_> BtbN: thanks [15:29:22 CEST] <anshul_> BtbN: is it ok to use level 4.1 and constrained baseline if my target device where video would play is ios [15:29:56 CEST] <anshul_> since they say that u should use High profile with level 4.1 [15:30:57 CEST] <BtbN> Is that combination even legal? [15:32:50 CEST] <anshul_> till now I was using profile high and level 4.1 with preset ultrafast, but in sps of output profile was not high. [15:33:06 CEST] <anshul_> Even I have the same question is that combination legal [15:33:15 CEST] <BtbN> ultrafast implies contrained baseline, as you turn off requires features for high and main [15:33:21 CEST] <BtbN> *d [15:34:10 CEST] <anshul_> do I need get rid of utlrafast if I have to make high and level4.1 [15:34:38 CEST] <BtbN> There's no point in that combination [15:36:07 CEST] <anshul_> BtbN: can u elaborate? should I get rid of high profile or ultrafast [15:36:18 CEST] <BtbN> Whatever you want [15:38:18 CEST] <anshul_> i choose high profile since I selected 4.1 level, and apple peoples say we support 4.1 with high profile, ultrafast I chose since I want to encode videos faster [15:38:51 CEST] <BtbN> ultrafast disabled most features, to be fast [15:38:58 CEST] <BtbN> so it ends up at constrained baseline [15:39:02 CEST] <BtbN> and basicaly mpeg2 quality [15:39:25 CEST] <anshul_> Would there be any problem on ios player with level 4.1 and constrained baseline [15:39:32 CEST] <BtbN> ... [15:40:16 CEST] <anshul_> I dont have access to all kind of decoders given by ios [15:40:32 CEST] <anshul_> to test, so just want to be on safe side [15:40:51 CEST] <BtbN> I just told you that i don't think that it even is a valid combination. [15:57:01 CEST] <anshul_> BtbN: thanks again [16:28:27 CEST] <Fyr> guys, is it possible to process audio multithreadedly? I have a large FLAC (~1 Gb) that I want to recompress. I have 12-cored Xeon cluster, but ffmepg uses only 100% of the cluster. ='( [16:32:07 CEST] <Fyr> I've converted into WAV and now I am trying to convert into FLAC back with turned all the compress options on. [16:34:04 CEST] <c_14> Fyr: according to the help output, the flac encoder doesn't thread [16:34:17 CEST] <Fyr> ='( [16:34:49 CEST] <Fyr> is it feasible to have it multithreaded? [16:34:57 CEST] <Fyr> is libflac support multiple threads? [16:35:08 CEST] <Fyr> *does [16:35:30 CEST] <c_14> According to the libflac api docs it does not thread either. [16:35:42 CEST] <Fyr> it's too sad. [16:36:00 CEST] <c_14> There was apparently a proof of concept on the Flac-dev ml [16:38:39 CEST] <c_14> The patches 404 of course. [16:40:52 CEST] <spectromas> what does ffmpeg do if I don't specify a -crf value? I can't find any info about what level is used instead, the default 23? [16:41:33 CEST] <c_14> libx264 defaults to a crf of 23, yes [16:41:53 CEST] <spectromas> ok thank you [16:42:08 CEST] <Fyr> I hope Xiph will buy a multicored computer to their employees. [16:42:17 CEST] <c_14> Fyr: I found copies of the patches that don't 404 along with the relevant blog post. http://www.akalin.cx/parallelizing-flac-encoding if you're interested. Probably a lot of work to get it working 100% though. [16:42:54 CEST] <Fyr> oh [16:43:09 CEST] <Fyr> too old libFLAC =( [17:13:08 CEST] <Netek> is there anyway I can run ffmpeg in background on ubuntu? [17:14:15 CEST] <Fyr> screen/tmux [17:14:23 CEST] <Fyr> use tmux, Netek. [17:15:00 CEST] <hardikj> hi, I am using 'ffmpeg -framerate 8 -f gdigrab -vcodec libx264 -crf 0 -preset ultrafast output.mkv' to capture my screen, is there any way to reduce the cpu usage [17:15:17 CEST] <hardikj> does the output format like ".mkv" affect the cpu usage? [17:15:20 CEST] <Fyr> hw encoding? [17:15:52 CEST] <Fyr> hardikj, have you tried to recomile it with dxva2 enabled? or nvenc? [17:16:59 CEST] <hardikj> Fyr, thanks for quick response, by recompling you mean running again on same output [17:17:20 CEST] <Fyr> no, I meant recompilation. [17:17:43 CEST] <Fyr> http://its.ffmpeg.org/wiki/CompilationGuide [17:18:07 CEST] <hardikj> ah, nice [17:18:17 CEST] <hardikj> let me try that :) [17:18:56 CEST] <damnlie> hi, may i have a question, im running ffmpeg N-72812 on debian 8.1 now im coming a cros with a problem Unknown encoder 'libx264' [17:19:25 CEST] <Fyr> if I were you, I would use something more convenient than ffmpeg, for example - OBS. [17:19:32 CEST] <Fyr> hardikj [17:20:22 CEST] <Fyr> damnlie, have downloaded ffmpeg from the website or compiled it on your own? [17:20:33 CEST] <damnlie> from git [17:20:47 CEST] <Fyr> did you compile it with libx264? [17:21:40 CEST] <hardikj> Fyr, I am looking for a tool that will be used to record selenium test running on 1000 windows/mac box. For Mac box ffmpeg works out of the box [17:21:49 CEST] <trosh> ( ./configure ... --enable-libx264 ... ) [17:22:04 CEST] <hardikj> but some window box have 1 core and 2 gb ram [17:22:11 CEST] <Fyr> hardikj, OBS is the right choice. [17:22:27 CEST] <hardikj> Fyr, ok will look into it [17:22:30 CEST] <hardikj> thanks for help [17:32:17 CEST] <durandal_1707> Fur: ffmpeg does multithreaded decoding [17:32:55 CEST] <durandal_1707> Fyr ^ [17:33:17 CEST] <Fyr> cool [17:33:33 CEST] <Fyr> how to use it when converting into FLAC? [17:33:53 CEST] <durandal_1707> encoding could be simple to add it too [17:35:05 CEST] <Fyr> great! how do I send a feature request to Xiph? [17:35:08 CEST] <durandal_1707> I mean multithreaded encoding for audio is not done but should be simple to implement [17:35:50 CEST] <durandal_1707> I m talking about ffmpeg and not about xiph [17:36:10 CEST] <Fyr> but does ffmpeg use libflac? [17:39:05 CEST] <durandal_1707> nope [17:41:54 CEST] <wk-work> is there any way to make ffmpeg read from disk faster? i'm trying to seek into a 74gb file and it seems to be taking a while [17:47:31 CEST] <durandal_1707> Faster disk? [17:51:08 CEST] <Netek> wk-work get a ssd hard drive :) [17:51:39 CEST] <wk-work> it's not the disk [17:51:42 CEST] <Netek> its like you are asking can you make a ford fiesta faster :D [17:51:52 CEST] <Netek> when you want speeds of a ferrari ;) [17:52:08 CEST] <wk-work> the disk reads as fast as up to 60 mb / s and it's only using 20 [17:52:23 CEST] <Netek> check cpu and ram processes [17:53:08 CEST] <wk-work> only usiong 20% of one core, ram not an issue [17:53:09 CEST] <wk-work> i already checked this [17:53:55 CEST] <Netek> no idea [17:58:20 CEST] <Nolski> wk-work: are you -ss before -i? [17:58:34 CEST] <wk-work> Nolski: nope [17:59:23 CEST] <Nolski> wk-work: you should probably do that. If you seek after -i it is more accurate but much slower [18:00:52 CEST] <Nolski> wk-work: the first answer here explains it a bit better http://superuser.com/questions/138331/using-ffmpeg-to-cut-up-video [18:00:53 CEST] <wk-work> Nolski: didn't think about that, thanks, it was *instant* now [18:01:48 CEST] <Nolski> ;) no problem wk-work [18:04:48 CEST] <sfan5> wk-work: iirc -ss <where you want to seek> -i input file -ss 0 will seek to the exact time but way faster [18:06:02 CEST] <c_14> You don't need that second -ss 0 [18:06:12 CEST] <c_14> -ss before -i is accurate by default (in relatively modern versions of ffmpe) [18:06:14 CEST] <c_14> +g [18:16:03 CEST] <Netek> ffmpeg -re -i "rtmp://127.0.0.1/itv_effectivestreaming/live live=1" -hls_time 10 -hls_wrap 7 -hls_list_size 3 -f hls itv.m3u8s when using this command could somebody give me a better understanding what this error means [18:16:27 CEST] <Netek> Past duration 0.619987 too largeN/A time=00:00:03.04 bitrate=N/A dup=46 drop=0 [18:19:02 CEST] <hardikj> Fyr, you around? [18:19:13 CEST] <hardikj> seems like OBS use ffmpeg in background? [18:19:29 CEST] <hardikj> ain't it better to use ffmpeg directly? [18:21:14 CEST] <Netek> hardikj? [18:22:24 CEST] <hardikj> Netek, Fyr recommended me to use OBS instead of ffmpeg, since I am using ffmpeg for screen recording of selenium tests using this command 'ffmpeg -framerate 8 -f gdigrab -vcodec libx264 -crf 0 -preset ultrafast output.mkv' [18:22:33 CEST] <Netek> ahh ok [18:22:34 CEST] <hardikj> but, this is taking lot of cpu [18:22:46 CEST] <hardikj> I wanted to reduce the cpu consumption [18:22:56 CEST] <hardikj> do you have any suggestions? [18:23:41 CEST] <Netek> hardikj no I am sorry, I am learning ffmpeg myself [18:23:49 CEST] <hardikj> np [18:40:40 CEST] <Fyr> hardikj, OBS can use hardware perks. [18:41:12 CEST] <Fyr> for example, it uses NVENC when grabbing. [18:41:32 CEST] <Fyr> maybe, it will use ATI's hardware tools to grab. [18:42:03 CEST] <Fyr> it requires 0% of CPU workload on my computer. [18:42:11 CEST] <Fyr> it all goes through the videocard. [18:43:04 CEST] <hardikj> Fyr, notice the tool would be running on virtual machines so no videocard [18:43:15 CEST] <hardikj> that's why I selected ffmpeg [18:43:34 CEST] <Fyr> VMs have graphic card too. [18:45:15 CEST] <Nolski> Hey, I'm trying to split a video using this command http://fpaste.org/232678/43447301/ but I'm getting the error "No such filter: ''" [18:45:33 CEST] <Nolski> Am I just formatting the filter wrong? It looks okay to me [18:47:42 CEST] <durandal_1707> pb! Nolski [18:48:58 CEST] <Nolski> durandal_1707: I did use pastebin? [18:49:24 CEST] <durandal_1707> but not full output [18:49:56 CEST] <Nolski> Ah, right [18:49:58 CEST] <Nolski> http://fpaste.org/232682/47338014/ [18:50:00 CEST] <Nolski> durandal_1707: ^ [18:54:01 CEST] <durandal_1707> hmm, try removing last ; [18:54:23 CEST] <Nolski> lol wow [18:54:26 CEST] <Nolski> durandal_1707__ [18:54:29 CEST] <Nolski> durandal_1707++ [18:54:33 CEST] <Nolski> yup [18:54:36 CEST] <Nolski> good catch [18:57:34 CEST] <hardikj> Fyr,I just confirmed unfortunately our VM dont :( [18:57:48 CEST] <hardikj> so i guess only option is to recompile ffmpeg [18:57:53 CEST] <Fyr> ok [19:10:13 CEST] <Nolski> Has anyone had issues with running ffmpeg commands through the python subprocess module? [19:11:21 CEST] <Nolski> I'm getting an error for a command that works when manually run in bash but not python subprocess.call() [19:11:22 CEST] <Nolski> http://fpaste.org/232690/47420414/ [19:25:39 CEST] <jcay> Nolski escape ' with \' maybe? [19:28:00 CEST] <chungy> don't have the ' in Python's subprocess call at all [19:28:06 CEST] <chungy> you don't need to do shell escaping in it [19:33:16 CEST] <Bombo> i got a video.mp4 with metadata 90 (it's rotated) so i recoded it with -vf rotate=90 but the result looks like 45°... [19:33:26 CEST] <Bombo> and the metadata ist still there... [19:34:14 CEST] <Nolski> ahh that's right [19:34:15 CEST] <pzich> use transpose instead? [19:34:22 CEST] <Nolski> thanks chungy and jcay [19:35:32 CEST] <pzich> you may also need to strip the rotation metadata off after actually rotating it with something like -metadata:s:v:0 rotate=0 [19:39:29 CEST] <Bombo> pzich: ok that worked, thx :) [19:39:39 CEST] <Bombo> but, the creation_time is stripped too [19:40:02 CEST] <Bombo> can i keep that? [19:44:21 CEST] <hardikj> Fyr, sorry for the ping, but is there any other way instead of hardware accelaration to improve cpu usage? [19:46:05 CEST] <Fyr> recompilation with optimization options, required for your CPU. [19:46:27 CEST] <pzich> Bombo: not sure, you may be able to set it with the metadata flag, but I think it's intentionally set to the time you created the encode [19:47:23 CEST] <Bombo> got it: metadata:s:v:0 rotate=0 -map_metadata 0:s:0 [19:47:24 CEST] <Bombo> ;) [19:50:08 CEST] <Fyr> hardikj, if there is no option available, you can try other converters. for example, Handbrake. [19:50:25 CEST] <Fyr> maybe, it will use the hardware acceleration you need. [19:50:53 CEST] <hardikj> Fyr, ok, thanks once again :) [19:51:43 CEST] <Bombo> hmmm now it would be cool if i could say 'just rotate or transpose video if there is a rotate tag in metadata' [19:52:16 CEST] <Bombo> i guess i'll have to find that out before encoding, with bash magic [19:57:45 CEST] <c_14> Bombo: git ffmpeg should do that automagically [20:04:05 CEST] <Bombo> c_14: hmmm it keeps the Metadata: rotate : 90 and adds Side data:displaymatrix: rotation of -90.00 degrees [20:04:43 CEST] <Bombo> keeps the "Metadata: rotate : 90" adds "Side data:displaymatrix: rotation of -90.00 degrees" [20:05:11 CEST] <Bombo> old mplayer doesn't seem to read the side data [20:05:18 CEST] <Bombo> mpv does [20:05:49 CEST] <Bombo> vlc does too [20:06:50 CEST] <Bombo> so ffmpeg doesn't 'unrotate' but it sets just another tag for players to unrotate at display [20:07:41 CEST] <c_14> What build did you try? [20:07:54 CEST] <c_14> (From when)? [20:08:34 CEST] <c_14> According to a1acae0409700e1abe62cbc900321fcffc9d971d at least, the metadata should be removed when autorotated. [20:09:27 CEST] <Bombo> ffmpeg version git-2015-04-06-f79cd98 [20:10:07 CEST] <c_14> That's too old. [20:10:25 CEST] <Bombo> ok ;) [20:32:57 CEST] <Nolski> Hey all, sorry for asking so many questions. I'm trying to render some text over a video but only during a specific time. I figured if I seek to that spot and then render the text it would work fine but it looks like this command (http://fpaste.org/232725/47940214/) still just renders the text over the whole video [20:33:19 CEST] <c_14> just use the enable option to the drawtext filter [20:36:24 CEST] <c_14> If it's not listed in the docs with the filter, it's somewhere at the bottom because it's a general option that applies for several filters. [20:39:21 CEST] <Nolski> I don't see any enable option listed on the page (even not under drawtext). There are 66 occurances of the word "enable" though so I might just be missing it... [20:40:00 CEST] <Nolski> found it [20:40:23 CEST] <Nolski> is there docs on how the option works? [20:40:34 CEST] <Nolski> s/is/are/ [20:41:06 CEST] <c_14> https://ffmpeg.org/ffmpeg-filters.html#Timeline-editing [20:41:09 CEST] <c_14> That was it. [20:41:35 CEST] <c_14> All the ifs/gts etc are documented in ffmpeg-utils [20:45:08 CEST] <Nolski> awesome [20:45:24 CEST] <Nolski> Thanks c_14 [20:51:46 CEST] <Bombo> c_14: ack, ffmpeg version git-2015-06-16-20657e0 ;) [20:52:24 CEST] <Bombo> it simply autorotates it, with no parameters... and metadata is gone [20:53:36 CEST] <c_14> just like magic [20:53:44 CEST] <Bombo> (and with just -map_metadata 0:s:0 the creation_time survives too) [21:09:48 CEST] <spectromas> can anyone tell me what the problem is here? https://bpaste.net/show/fe7df1b84aa7 [21:26:36 CEST] <Bombo> spectromas: what version of ffmpeg do you use? (update it!) ;) [21:28:22 CEST] <spectromas> Bombo: I have 2.7, that's the current version isn't it? [21:28:43 CEST] <spectromas> maybe I'm missing a useflag? [21:28:57 CEST] <podman> if i include yadif=0:-1:0 as a filter, will it always try to deinterlace or will it detect if the source is interlaced and only deinterlace if it is? [21:31:21 CEST] <Bombo> spectromas: quite so, i just compiled the git version... [21:36:01 CEST] <spectromas> oh oops I think I was trying to save somewhere without permission [21:36:10 CEST] <spectromas> what a fool [21:38:49 CEST] <Bombo> hehe shit happens, and a bad error message [21:39:34 CEST] <Bombo> i found my xgrab script: ffmpeg -f x11grab -s ${size} -r 30 -i :0.0${area} -sameq -crf 0 -profile:v ultrafast -f alsa -i pulse -acodec pcm_s16le output.mkv [21:39:47 CEST] <Bombo> worked with a old version [21:40:09 CEST] <Bombo> -profile:v doesnt... [21:40:16 CEST] <Bombo> Option profile:v (set profile) cannot be applied to input file pulse -- you are trying to apply an input option to an output file or vice versa. Move this option before the file it belongs to. [21:41:40 CEST] <spectromas> that looks handy. I do this so seldom that I forget every time [21:43:28 CEST] <Bombo> Option profile:v (set profile) cannot be applied to input file pulse [21:43:37 CEST] <Bombo> crap orders [21:49:18 CEST] <Bombo> ffmpeg -y -f x11grab -i :0.0${area} -f alsa -i pulse -c:v libx264 -crf 0 -profile:v ultrafast -r 30 -vf scale=${size} -c:a pcm_s16le out.mkv [21:49:23 CEST] <Bombo> almost got it ;) [21:49:35 CEST] <Bombo> Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height [21:56:04 CEST] <voip_> hello guys [21:56:58 CEST] <voip_> may i use rasberi pi board for h264 live stream transcodig ? [21:58:36 CEST] <luckcolors> hello [21:58:42 CEST] <Fyr> hey [21:58:57 CEST] <luckcolors> can someone help me? [21:59:39 CEST] <Fyr> it depends.. [22:00:33 CEST] <luckcolors> i'm using ffmpeg to download and save an m3u8 playlist [22:00:57 CEST] <luckcolors> but ffmpeg save the download in a mp4 file [22:01:27 CEST] <luckcolors> is it possible to just download the single .ts files of the playlist? [22:01:48 CEST] <luckcolors> and not join them in a mp4 file [22:02:27 CEST] <Fyr> that m3u8 file on pastebin + the command you are using. [22:02:41 CEST] <luckcolors> a sec [22:03:05 CEST] <Bombo> spectromas: http://bombo.jpe.gs/xgrab.sh ;) [22:04:51 CEST] <Bombo> does work now with git version, just one minor issue 'Guessed Channel Layout for Input Stream #1.0 : stereo' although i included '-ac 2' ... [22:05:45 CEST] <Bombo> voip_: it's probably too slow, as encoding is done in cpu, maybe Pi2 or a mod with faster cpu [22:07:24 CEST] <voip_> Bombo, yes i menat Pi2 model with faster CPU, so did you tryffmpeg on rasbery ? [22:07:48 CEST] <spectromas> thanks a lot Bombo :) [22:08:35 CEST] <luckcolors> fyr here's the truncated version of the file http://pastebin.com/uuuiBpmR [22:09:19 CEST] <luckcolors> i don't know the command with i start ffmpeg because i'm using youtube-dl [22:09:36 CEST] <Bombo> voip_: no i got Pi1, definitely too slow i didn't even try ;) [22:09:50 CEST] <Fyr> youtube-dl automatically uses ffmpeg to compund the files. [22:09:55 CEST] <voip_> Bombo, :) [22:10:06 CEST] <Bombo> spectromas: np [22:10:08 CEST] <Fyr> luckcolors, what command do you use with youtube-dl? [22:10:43 CEST] <luckcolors> youtube-dl http://www.twitch.tv/twitch [22:10:51 CEST] <luckcolors> no wait [22:11:56 CEST] <luckcolors> youtube-dl http://www.twitch.tv/twitch/v/6244102 [22:12:03 CEST] <luckcolors> this is the command i use [22:12:10 CEST] <Bombo> yes let's dl the internet ;) [22:12:57 CEST] <luckcolors> yeah :) [22:13:05 CEST] <Fyr> luckcolors, youtube-dl is downloading mp4 file. [22:13:22 CEST] <Bombo> isn't twitch live streaming? [22:13:23 CEST] <Fyr> it's the problem of youtube-dl and its parameters, not ffmpeg. [22:14:09 CEST] <luckcolors> youtube-dl just gets the stream metadata (output file name + m3u8 url) and the passes the execution to ffmpeg [22:15:09 CEST] <luckcolors> infact i cannot even download without ffmpeg [22:16:17 CEST] <luckcolors> fyr which parameters i can give to ffmpeg so it cdan download just the .ts files? [22:16:34 CEST] <Bombo> luckcolors: but yt-dl sets parameters for ffmpeg [22:17:05 CEST] <Fyr> luckcolors, http://pastebin.com/TubuZe4z [22:17:26 CEST] <Fyr> youtube-dl offers only m4a. [22:18:57 CEST] <Bombo> maybe twitch does only offer m4a [22:19:03 CEST] <luckcolors> http://vod.ak.hls.ttvnw.net/v1/AUTH_system/vods_e869/twitch_14883844112_257… [22:19:16 CEST] <Fyr> I see. [22:19:28 CEST] <Fyr> but youtube-dl can see only mp4 files. [22:19:35 CEST] <Fyr> ask, #youtube-dl [22:19:54 CEST] <luckcolors> ok [22:22:41 CEST] <luckcolors> can someone remove the twitch link that i posted from the logs? :P [22:23:02 CEST] <Fyr> easily [22:23:10 CEST] <Fyr> call KGB, they will do the trick. [22:23:15 CEST] <luckcolors> thanks [22:30:16 CEST] <luckcolors> thanks fyr for your help bye [22:30:21 CEST] <Fyr> ok [23:22:05 CEST] <dine909> hi there - i'm looking for some advice, i need to make a video (of undetermined length) crossfade into itself over 5 seconds at the end (for looped playback), i've been trying with -filter_complex using an example of something similar i found, but i'm under pressure and the syntax is making me sick, is there an easier way to achieve the same result? [23:24:04 CEST] <durandal_1707> what filters you used [23:26:16 CEST] <dine909> http://hastebin.com/xinulesaxa.sh so far [23:27:02 CEST] <dine909> but thats totally wrong [23:28:01 CEST] <dine909> i just changed an example i found to use the same file in 2 streams, i know it can work i just cant find the right brain door to set the correct length and fade points relatively using PTS [23:30:46 CEST] <dine909> for reference, that example came from http://superuser.com/questions/778762/crossfade-between-2-videos-using-ffmp… [23:34:26 CEST] <dine909> what does fade=t=out mean ? what is t? [23:35:08 CEST] <Nolski> dine909: normally when I see t it means to specifiy the input in seconds [23:35:17 CEST] <Nolski> (I may be wrong though) [23:35:48 CEST] <Nolski> dine909: https://ffmpeg.org/ffmpeg-filters.html#fade [23:35:55 CEST] <Nolski> It means type [23:36:24 CEST] <dine909> ah i see, thanks [23:36:33 CEST] <Nolski> np [23:37:12 CEST] <dine909> so fade=t=out:st=4:d=1:alpha=1 [23:37:25 CEST] <dine909> is a fadeout at 4s lasting 1s? [23:37:58 CEST] <Nolski> dine909: yup, I think so [23:38:18 CEST] <Nolski> It'd fade to black [23:39:38 CEST] <dine909> now, how could i set st to be length-5s [23:39:54 CEST] <Nolski> dine909: you want the duration of the fade to be 5 seconds? [23:40:00 CEST] <Nolski> you just set d=5 [23:40:18 CEST] <dine909> yes, but 5s before the end of the clip [23:40:25 CEST] <dine909> given i don't know how long that might be [23:40:42 CEST] <Nolski> dine909: you find out how long the clip is and then you subtract 5 seconds from that [23:40:46 CEST] <dine909> ie, fade any input to black at the end [23:40:47 CEST] <dine909> yes [23:41:02 CEST] <Nolski> It'd be a step separate from the fade [23:41:09 CEST] <dine909> thats my question is ENDPTS a thing? [23:42:15 CEST] <Nolski> not sure [23:47:43 CEST] <Nolski> Is it possible to loop a section of a video? [00:00:00 CEST] --- Wed Jun 17 2015

1 0

ffmpeg-devel.log.20150615
by burek 16 Jun '15

16 Jun '15

[00:51:00 CEST] <ac_slater_> hey all. This might not be the best place to ask but #ffmpeg is kinda dead today. Currently, I use `find_stream_info` on my clients to open RTSP and RTP streams. Does libavformat/codec provide a simple way to "pre-fill" some context(s) so that I dont have to use `find_stream_info`. I really only use ffmpeg/libavformat for demuxing, but it would be nice to pre-fill things [00:52:26 CEST] <ac_slater_> oh dang, I just saw the topic. Disregard this question if it's too of topic [00:53:37 CEST] <Compn> hehe [00:53:53 CEST] <Compn> no idea [00:53:58 CEST] <Compn> wait for someone that knows api [00:57:28 CEST] <ac_slater_> find_stream_info is so crytic [00:57:31 CEST] <ac_slater_> cryptic * [01:42:36 CEST] <cone-586> ffmpeg 03wm4 07master:74ea1167d91c: tls_gnutls: fix hang on disconnection [01:42:37 CEST] <cone-586> ffmpeg 03Michael Niedermayer 07master:dd9400930a30: Merge commit '74ea1167d91ccb2e1f2943efa030f2c278b598be' [01:50:47 CEST] <lglinskih> kierank: as I understand my test should open some test video, decode it and after that I should compare result-file of framecrc with some existing file. Is that right? [01:51:06 CEST] <cone-586> ffmpeg 03Michael Niedermayer 07master:5ef578d03a55: avcodec/jpeg2000dec: Also include remaining length in "Block length" error message [01:52:05 CEST] <kierank> lglinskih: yes some known framecrc [01:52:06 CEST] <kierank> as in fate [02:01:04 CEST] <lglinskih> And about framecrc: you told me that I should write my own, but I don't understand how should it differ from framecrcenc.c =/ [02:23:20 CEST] <lglinskih> kierank: And about framecrc: you told me that I should write my own, but I don't understand how should it differ from framecrcenc.c =/ [02:26:18 CEST] <kierank> it shouldnt' differ [02:26:26 CEST] <kierank> but it's nice to have a totally independent test suite [02:26:29 CEST] <kierank> it's not a big deal though [02:44:04 CEST] <lglinskih> kierank: "totally independent" is about test suite for API? [02:44:42 CEST] <kierank> it's nice to have a separate framecrc in the same way that it is nice to test decoders separately from ffmpeg.c [02:44:55 CEST] <kierank> but in reality it's very unlikely that someone will break the framecrc code [02:45:05 CEST] <lglinskih> =) [02:45:26 CEST] <kierank> however it is likely that someone can break decoding from API but still works in ffmpeg.c [02:48:18 CEST] <cone-586> ffmpeg 03Michael Niedermayer 07master:4bfdd967a6b2: avcodec/jpeg2000dec: Use <0 instead of != 0 for error checking [02:48:19 CEST] <cone-586> ffmpeg 03Michael Niedermayer 07master:a58f1bcc4cbe: avcodec/jpeg2000dec: Skip SOP [03:11:05 CEST] <cone-586> ffmpeg 03Michael Niedermayer 07master:5b0f55aab9d5: avcodec/jpeg2000dec: Check reslevelno in RPCL [04:14:23 CEST] <kierank> michaelni: can you perhaps document that magic number? [04:14:29 CEST] <kierank> and why you skip six [05:55:47 CEST] <jamrial> kierank: apparently, FF91 = SoC marker. 0004 = lenght of marker. following two bytes = packet sequence number [07:46:39 CEST] <ubitux> durandal_1707: first draft @ https://github.com/ubitux/FFmpeg/compare/selectivecolor [07:46:53 CEST] <ubitux> i'll try to submit something this or next week [10:34:49 CEST] <durandal_1707> news on the web page are still not updated [12:35:38 CEST] <dockheas23> Hi, my name is George. I sent a mail to the mailing list last week mentioning that I'd like to try to learn about and make some contributions to the project. I think I may have found something that might be interesting to look into, and was wondering if I could ask a couple of questions about it? [12:37:27 CEST] <dockheas23> I spent some time getting to grips with the testing system, and as part of that I think I may have found a bug in the flac encoding implementation. [12:39:10 CEST] <dockheas23> Basically, I have a wav sample file that, when encoded and decoded, doesn't produce an exact match of the original, and decoding produces an "Invalid rice parameter" warning. [12:41:14 CEST] <dockheas23> My first question is, should I raise this as a bug on trac? [12:42:05 CEST] <dockheas23> And secondly, do you see any problem with me attempting to fix it? [12:43:07 CEST] <durandal_1707> yes and no [12:43:49 CEST] <durandal_1707> feel free to report and fix it [12:43:52 CEST] <dockheas23> Great. Thank you! [12:44:56 CEST] <cehoyos> dockheas23: Which version of FFmpeg did you test? [12:45:21 CEST] <dockheas23> The latest commit on master (as of yesterday) [12:45:36 CEST] <cehoyos> Could you point me to the wav file? [12:45:43 CEST] <cehoyos> The issue definitely sounds important. [12:45:53 CEST] <dockheas23> It also behaved the same on my Arch distribution's version. [12:46:22 CEST] <dockheas23> It was a file I created myself, while messing around with some of the audio filters. [12:46:29 CEST] <dockheas23> I can attach it to the bug report. [12:46:56 CEST] <cehoyos> Please make it available (one way or other)! [12:47:08 CEST] <dockheas23> Will do! [13:15:18 CEST] <dockheas23> "trimmedchorus.wav" on ticket #4628 is the sample I've been working with. [13:15:42 CEST] <cehoyos> Dockheas23: Please do not provide an excerpt of the console output, always provide the command line that allows to reproduce the issue together with the complete, uncut console output. [13:16:31 CEST] <cehoyos> And generally, do not attach output files: There are many examples of tickets where this lead to immense confusion. [13:22:19 CEST] <dockheas23> Ok, thanks. I'll add the console output to the ticket. Cannot see an option to remove that attachment, but I'll know next time! [13:35:07 CEST] <cehoyos> Dockheas23: I have done the bisect: Will you look at the bug? [13:40:21 CEST] <dockheas23> Yes, I'll give it a shot and, either way, I'll let you know how I'm getting on by the end of the day (c. 18:00 UTC+1). [13:41:19 CEST] <cehoyos> Don't worry, the question was just if someone should send an email to Reimar right now. [13:42:54 CEST] <cone-822> ffmpeg 03Michael Niedermayer 07master:9ba5fe7f3d9f: avcodec/jpeg2000dec: Remove redundant check [13:42:54 CEST] <cone-822> ffmpeg 03Michael Niedermayer 07master:4ec14ce121df: avcodec/jpeg2000dec: Improve readability of SOP check [13:43:30 CEST] <dockheas23> @cehoyes: Ok, will I email him? [13:44:22 CEST] <cehoyos> Not if you can fix the issue... [13:50:14 CEST] <cehoyos> michaelni: I tested file5.jp2 from the reference files and it decodes bitexact wrt jasper and (a version of) libopenjpeg but the image looks different (too dark) compared to the reference outpu tjp2_5.tif [13:50:22 CEST] <cehoyos> jasper shows the following warning: [13:50:49 CEST] <cehoyos> ICC Profile CS 52474220 warning: inaccurate color [13:51:23 CEST] <cehoyos> Is there a missing feature in FFmpeg (and the other j2k decoders) or is this impossible (or is the reference output wrong)? [13:51:42 CEST] <cehoyos> impossible to get the same output as the reference tif image. [13:53:53 CEST] <cehoyos> jasper shows the same warning for file7.jp2 which looks ugly with FFmpeg but since it is yuv420p and the tif reference (that looks not ugly this way) is rgb, so I guess it is difficult to get bitexact output. [13:53:55 CEST] <wm4> does jasper apply icc profiles? [13:54:02 CEST] <cehoyos> I guess not. [13:54:17 CEST] <dockheas23> Ok, now I understand! Given some time I'm sure I can get to the bottom of it, but I cannot say just yet that I know exactly what's wrong. I was going to use it as an opportunity to get familiar with the code/algorithm. But, if you consider this to be an important bug that should be fixed asap, then it's probably best to send it to someone who already knows it well. [13:54:20 CEST] <cehoyos> (It is bitexact with FFmpeg, not with the reference and it prints a warning) [13:56:10 CEST] <cehoyos> dockheas23: Note that further tests with the flac encoder and decoder (I guess you could revert the offending patch locally) are very welcome! [13:56:39 CEST] <cehoyos> Ideally, they should include a second decoding pass with flac: We have recently fixed such an issue but others may exist. [13:57:00 CEST] <cehoyos> We have recently fixed an issue where FFmpeg could decode its own files but flac could not. [14:00:38 CEST] <dockheas23> Ok, I can do that. Just to clarify, do you mean tests involving the Xiph flac implementation? And if so, do you mean incorporating them into the fate test suite? [14:04:12 CEST] <cehoyos> No, no new tests for fate: But since you have done tests with flac already, I wanted to suggest something similar. This was assuming you used a script to find the issue you reported. [14:04:38 CEST] <cehoyos> But since the offending patch cannot be easily reverted, my idea was probably not a good one. [14:05:01 CEST] <dockheas23> Yeah, I just tried a revert there and it wasn't too pretty [14:05:54 CEST] <cehoyos> I still don't know what parts of FFmpeg you have already worked with... [14:06:52 CEST] <cehoyos> Or what parts you are particularly interested in and / or have some knowledge about the subject. [14:21:04 CEST] <dockheas23> In terms of the code, I have not worked on any of it before. My idea was to begin by trying to write some tests to build up some knowledge about it, which was how I came across this bug. In terms of my interest, I have a general interest in media encoding, and container formats in particular, but not a whole lot of knowledge yet. I'd like to improve that. [14:29:32 CEST] <kierank> you could work with ludmila on the api test suite [14:34:59 CEST] <dockheas23> That sounds good. [14:36:37 CEST] <dockheas23> How would I go about getting started on it? [14:52:52 CEST] <kierank> dockheas23: erm [14:52:57 CEST] <kierank> have a look at "make fate" [14:53:17 CEST] <kierank> basically at the moment all the tests are based on ffmpeg.c [14:53:24 CEST] <kierank> but that doesn't test API users [14:53:34 CEST] <kierank> and it happens sometimes that API users have bugs that aren't covered in ffmpeg.c [15:01:35 CEST] <dockheas23> Ok, thanks. How about I drop ludmila an email about it? [15:07:01 CEST] <kierank> yes and cc me in [15:07:04 CEST] <kierank> kierank(a)obe.tv [15:07:50 CEST] <dockheas23> Will do, thanks. Do you have an email for Ludmila? [15:09:08 CEST] <kierank> lglinskih(a)gmail.com [15:09:28 CEST] <dockheas23> Great. Thanks a lot! [16:10:49 CEST] <Compn> >gstreamer bugs [18:20:02 CEST] <BBB> Daemon404: poke [18:20:06 CEST] <Daemon404> ? [18:20:34 CEST] <BBB> I saw this one a while ago: http://www.mesclado.com/smpte-forum-2015-future-proofing-media-production-p… [18:20:51 CEST] <Daemon404> i assure you i do not think SMPTE is anything usefil [18:20:56 CEST] <BBB> so youre quoted as saying hevc needs to improve, Im assuming you mean s/hevc/x265/, right? (or any encoder, for that matter) [18:21:01 CEST] <Daemon404> yes [18:21:10 CEST] <Daemon404> "quoted" [18:21:13 CEST] <Daemon404> paraphrased. [18:21:13 CEST] <BBB> can you define improve? is it just speed of encoding? [18:21:21 CEST] <BBB> (and degradation of quality for improvement of speed) [18:21:26 CEST] <BBB> or something else also? [18:21:32 CEST] <nevcairiel> i find it funny that Daemon404 is somehow the SMTPE guy around here now =p [18:21:43 CEST] <BBB> (Im just making sure I got your quote correct if I decide to quote your quote) [18:21:49 CEST] <Daemon404> x265 placebo only ties or marginally beats x264 if at all [18:21:55 CEST] <Daemon404> for an order of magnitude slower encoding [18:22:01 CEST] <Daemon404> and its the best of a bad showing [18:22:22 CEST] <Daemon404> nevcairiel, BBB's former coworker asked me to show up. [18:22:24 CEST] <Daemon404> and they paid. [18:22:58 CEST] <nevcairiel> BBB: from my own experience, if you try to compress visually lossless at decent bitrates, the differences between h264 and h265 are very slim, only if you start going down in bitrate and accept loss to some extent, h265 eventually gets better .. so personally, I think they could also do better in high quality scenarios [18:23:16 CEST] <nevcairiel> s/h265/x265/ [18:23:19 CEST] <Daemon404> x265 is actually worse at transparent bitrates. [18:23:25 CEST] <nevcairiel> and x265 is much slower in the process [18:28:30 CEST] <BBB> ok will quote as such [18:31:25 CEST] <cone-822> ffmpeg 03Michael Niedermayer 07master:021351f246b1: avcodec/mqcdec: set raw flag at the begin of ff_mqc_initdec() [18:31:26 CEST] <cone-822> ffmpeg 03Michael Niedermayer 07master:96bbbebaf951: avcodec/jpeg2000dec: Fix Vertically causal context formation [18:38:51 CEST] <cone-822> ffmpeg 03Andreas Cadhalpun 07master:fdc64a104410: h264: er: Copy from the previous reference only if compatible [18:38:53 CEST] <cone-822> ffmpeg 03Andreas Cadhalpun 07master:dd6c8575dbc8: examples/demuxing_decoding: use properties from frame instead of video_dec_ctx [19:25:19 CEST] <cone-822> ffmpeg 03Paul B Mahol 07master:d107413f1caf: doc/filters: add one more compand example found in the wild [20:44:54 CEST] <wm4> ==5942== Use of uninitialised value of size 4 [20:44:55 CEST] <wm4> ==5942== at 0x4A6E16E: av_crc (crc.c:370) [20:44:55 CEST] <wm4> ==5942== by 0x5C4F7CB: ff_crcA001_update (aviobuf.c:506) [20:44:57 CEST] <wm4> huh [20:45:12 CEST] <wm4> (from mp3_read_packet) [21:04:50 CEST] <cone-822> ffmpeg 03Michael Niedermayer 07master:2819aeb0f3c2: avcodec/jpeg2000dec: Omit mqc reinit after the last pass [21:10:42 CEST] <cone-822> ffmpeg 03James Zern 07master:8ce321f0aaa9: encoders.texi: update libvpx documentation [21:29:58 CEST] <cone-822> ffmpeg 03Vittorio Giovara 07master:07b2db81d06e: riff: Add MNM4 FourCC as mpeg4 [21:29:59 CEST] <cone-822> ffmpeg 03Michael Niedermayer 07master:c5fc48fdfb76: Merge commit '07b2db81d06e1cd6b1718d3e2dd7a42e8bccf8c0' [21:37:22 CEST] <cone-822> ffmpeg 03James Zern 07master:e6c71385f001: libvpxenc: cosmetics: reindent after 2c70436 [21:37:23 CEST] <cone-822> ffmpeg 03James Zern 07master:a829040c431a: libvpxenc: remove stray '\'s [21:51:41 CEST] <cone-822> ffmpeg 03Luca Barbato 07master:24ad3ac6a3e2: nut: Drop doxygen markers [21:51:42 CEST] <cone-822> ffmpeg 03Michael Niedermayer 07master:efcf8cfa48c8: Merge commit '24ad3ac6a3e20350214e6c3f7a931635f264ae07' [22:10:32 CEST] <cone-822> ffmpeg 03Luca Barbato 07master:03ca6d70df19: x264: Factor out the reconfiguration code [22:10:33 CEST] <cone-822> ffmpeg 03Michael Niedermayer 07master:4ed3a01d717f: Merge commit '03ca6d70df192125a772dadd01acfe3905aa653f' [22:29:31 CEST] <cone-822> ffmpeg 03Luca Barbato 07master:9af7e8045e3e: lavc: Clarify the behaviour of dimension and format context fields [22:29:32 CEST] <cone-822> ffmpeg 03Michael Niedermayer 07master:20526f2e2f58: Merge commit '9af7e8045e3e63ab39adedae9a7c11b1c410af26' [22:40:31 CEST] <cone-822> ffmpeg 03Andreas Cadhalpun 07master:a4fbd55d6e03: h264: er: Copy from the previous reference only if compatible [22:40:33 CEST] <cone-822> ffmpeg 03Michael Niedermayer 07master:12e36a3dfdc6: Merge commit 'a4fbd55d6e03eabdbecc3b7892ec09eb8062d066' [23:08:27 CEST] <anoop_r> how to disable help in ffmpeg.exe [23:13:52 CEST] <kierank> ahahaha [23:16:39 CEST] <anoop_r> like ffmpeg -h shows command line help i want to disable it [23:18:02 CEST] <cone-822> ffmpeg 03Paul B Mahol 07master:eb85060b84d8: avfilter/af_afade: add couple of more curves [23:27:55 CEST] <cone-822> ffmpeg 03Michael Niedermayer 07master:fefe04259ae9: avcodec/jpeg2000dec: increase tile part limit to 32 [23:34:25 CEST] <anoop_r> how to disable help in ffmpeg.exe [00:00:00 CEST] --- Tue Jun 16 2015

1 0

ffmpeg.log.20150615
by burek 16 Jun '15

16 Jun '15

[00:49:04 CEST] <ac_slater_> I guess I can take the context, look at it in a debugger, and start fulling in structures from that info [01:24:58 CEST] <jY> i have the following stream coming into my server and I need to pretty much send that to nginx-rtmp and I'm having an issue working out the details of how to use ffmpeg to relay it.. [01:24:59 CEST] <jY> https://gist.github.com/mzupan/75025a4dcd54258398a4 [01:27:21 CEST] <jY> i've tried ffmpeg -y -f mpegts -i udp://0.0.0.0:1935 -c:v copy -map 0:0 -f mpegts -rtmp_buffer 100 rtmp://127.0.0.1/live/test [01:27:27 CEST] <jY> but i just get conversion fails [02:04:23 CEST] <Fusl> is there a way to let ffmpeg generate the mandelbrot source with multiple threads? i'm having issues with generating a video with "mandelbrot=s=4096x2160 ", it's only using one of the cpu cores which slows down the ffmpeg process completely [02:10:16 CEST] <Quixxx> Can someone help me build my flags for the encode I want to do? I need to take a 1080i25 file to 720p50, two pass, 4800k bitrate, 'slower' preset if that's a thing, and a good deinterlace filter. [02:46:43 CEST] <kepstin> Quixxx, do you mean i50 to p25? [02:47:11 CEST] <Quixxx> I guess it depends on how you measure the interlaced. 50 half frames every second [02:47:22 CEST] <Quixxx> But the resulting file is p50. [02:47:33 CEST] <kepstin> interlaced is measured in fields per second, progressive in frames per second. [02:49:30 CEST] <kepstin> so probably something like ffmpeg -i inputfile -vf yadif=1,scale=1280:720 -c:a copy -c:v libx264 -preset veryslow -b:v 4800k -pass 1 outputfile [02:49:34 CEST] <kepstin> and repeat with pass=2 [02:50:20 CEST] <kepstin> adjust the "preset" setting to medium, slow, slower, veryslow, etc. according to taste. [03:29:21 CEST] <Quixxx> kepstin, and the input and output files will be the same for the second pass, correct? [03:31:46 CEST] <kepstin> yeah, the only thing to change between the two passes is the -pass parameter. [03:32:10 CEST] <kepstin> (the output file of the first pass doesn't actually matter, you can send it to /dev/null or whatever) [03:32:43 CEST] <Quixxx> Alrighty, I'll have to wait for the encodes to finish [03:33:02 CEST] <Quixxx> also, what about -threads? I stuck -threads 4 in there for the first pass but I don't think it matters until the second. [03:33:20 CEST] <kepstin> if you don't put in a -threads option, it picks a default suitable for your machine [03:33:45 CEST] <Quixxx> I'm used to using webm for retards, I think it sticks -threads 1 in by default. [03:43:41 CEST] <Quixxx> >-pass 2 wants me to overwright [03:43:43 CEST] <Quixxx> write [03:43:45 CEST] <Quixxx> Yes, right? [03:43:55 CEST] <kepstin> yeah [03:44:21 CEST] <Quixxx> Wait [03:44:25 CEST] <Quixxx> The file doesn't look that bad right now [03:44:56 CEST] <Quixxx> Deinterlace is a smidge blurry I think. Ah well. [03:45:17 CEST] <kepstin> the output of the first pass is encoded to your requested bitrate using single pass rate control [03:45:40 CEST] <kepstin> which means it's not able to balance bitrate usage between complex vs. simple sections of video [03:45:46 CEST] <kepstin> that's what the second pass does. [03:45:54 CEST] <Quixxx> So, the first pass file is larger? [03:46:07 CEST] <kepstin> no, since they're both being encoded at the same bitrate [03:46:16 CEST] <Quixxx> uhh [03:46:23 CEST] <Quixxx> I'll just compare both of them when the encode is done, I guess. [03:46:47 CEST] <kepstin> the second pass should look better in places that are hard to encode, because it's able to use higher bitrate in hard spots and less bitrate in easy spots [03:47:00 CEST] <kepstin> instead of the 1-pass mode, where it keeps the bitrate steady through the file. [03:48:05 CEST] <Quixxx> It looked like it was changing a bit, 4800-5000 or so [03:48:21 CEST] <kepstin> of course, the only reason to use 2-pass mode is if you're targetting a specific final file size [03:49:04 CEST] <kepstin> if you just want a good quality output and don't care as much about file size variation, you should do a 1-pass encode setting the "-crf" option instead of using bitrate [03:49:25 CEST] <Quixxx> I might do that for personal stuff [03:49:43 CEST] <Quixxx> I've got a lot of stuff to encode but this poor 2500k isn't fast enough [03:49:50 CEST] <Quixxx> I need me some 14 core xeons [03:50:14 CEST] <kepstin> "-crf" effectively gives you a constant-quality encode, where parts get more or less bitrate to maintain the same visual quality throughout. [03:50:43 CEST] <Quixxx> Most of the sites I use are constant bitrate, I'm not sure if that's for quality reasons or otherwise. [03:50:50 CEST] <Quixxx> I think -crf you usually use... 20? [04:13:35 CEST] <techtopia> why two pass quixx [04:14:51 CEST] <techtopia> sounds like you are encoding european transport streams heh [04:15:10 CEST] <techtopia> to get 50fps progressive out of the 25fps interlaced source [04:15:15 CEST] <techtopia> you need to bob deinterlace [04:15:25 CEST] <techtopia> you can do it with -vf yadif [04:15:51 CEST] <kepstin> techtopia, yeah, I already gave Quixxx the yadif options to do that. [04:16:08 CEST] <techtopia> cool :) [04:16:19 CEST] <Quixxx> Is there a way to use QTGMC with ffmpeg? [04:16:28 CEST] <techtopia> what is that? [04:16:46 CEST] <Quixxx> A deinterlace filter, I've been told it's gud [04:16:52 CEST] <techtopia> use yadif [04:16:55 CEST] <kepstin> looks like an avisynth deinterlacer plugin [04:17:23 CEST] <kepstin> you're downsampling from 1080i to 720p anyways, so the deinterlacer quality doesn't matter as much. [04:17:33 CEST] <Quixxx> Ah [04:17:44 CEST] <Quixxx> Yeah the files I've seen it referenced for were SD>SD VHS rips [06:07:39 CEST] <jY> i'm converting a mpeg stream via udp.. every now and then i get a "Conversion failed!" and ffmpeg stopps [06:07:51 CEST] <jY> is there anyway to ignore that and stay running [07:05:15 CEST] <Abbott> #edmproduction [07:05:26 CEST] <Abbott> whoops [07:26:39 CEST] <gurpartap> I am saving a rtsp stream to an avi file. Is there any way that I can set the maximum file size limit and keep saving my stream in ordered video files? [08:32:58 CEST] <gurpartap> I am using ffmpeg commandline to produce segmented output...Here is command ffmpeg -i rtsp://192.168.0.127 -f segment -segment_time 60 "test-%03d.avi" but it is adding extra 1 minute with still frames. [09:24:18 CEST] <karty> Hello guys . I'm running ffmpeg with vstats_file option. It gives time value in milliseconds for one type of file and in seconds for another [09:24:41 CEST] <karty> is there a option to tweak this ? [09:24:58 CEST] <karty> to get just seconds or just milli seconds accrosss all files or formats [09:26:00 CEST] Last message repeated 1 time(s). [09:26:15 CEST] <karty> Hello guys . I'm running ffmpeg with vstats_file option. It gives time value in milliseconds for one type of file and in seconds for another is there a option to tweak this ? to get just seconds or just milli seconds accrosss all files or formats. [12:04:53 CEST] <antiatom> Hello, I would like to take advantage of multithread VP9 encoding in libvpx 1.4.0 [12:05:06 CEST] <antiatom> Do I need a specific version of FFmpeg in order to use this feature? [12:06:50 CEST] <gurpartap> I am using ffmpeg -i rtsp://192.168.0.127 -map 0:0 -map 0:1 -s 640x360 -vcodec libx264 -g 60 -vb 500000 -strict experimental -vf copy -an -y -f segment -segment_time 20 "part-%03d.ts" -f mpegts udp://127.0.0.1:1234 to make video chunks of 20 seconds and restream it through udp.But sometimes packets are missing which are shown in ffmpeg log... [12:07:25 CEST] <relaxed> antiatom: Try the latest git version, http://johnvansickle.com/ffmpeg/ [12:07:57 CEST] <durandal_1707> why you are using vf filter? [12:08:43 CEST] <antiatom> relaxed: So version 2.7 does not support this? [12:11:36 CEST] <relaxed> try it [12:13:25 CEST] <gurpartap> :durandal_1707 are you asking me? [12:16:32 CEST] <durandal_1707> yes [12:17:29 CEST] <gurpartap> I am using this link....http://stackoverflow.com/questions/10588001/how-to-record-udp-strea… [12:17:48 CEST] <gurpartap> I remove the -vf and it is working fine..... [12:18:24 CEST] <gurpartap> but tell me how can I restream it through rtp at same time [12:40:54 CEST] <gurpartap> how can I extract timestamp of video coming from camera and save it in log file... [12:43:06 CEST] <Anoia> gurpartap: from what point of view? [12:43:28 CEST] <Anoia> as code that has an RTSP receiver and saves H.264, or reading the saved data? [12:43:49 CEST] <Anoia> or even motion JPEG.. [12:43:50 CEST] <gurpartap> I want to extract pts_time from live stream... [12:44:11 CEST] <gurpartap> I am recieving through rtsp protocol [12:44:31 CEST] <gurpartap> ffmpeg -i input.mp4 -t 10 -filter:v "fps=fps=25, showinfo" -f null - 2>&1 grep pts_time | awk '{print $6}' | cut -d: -f2 [12:44:49 CEST] <gurpartap> this I am using for saved video [12:45:16 CEST] <gurpartap> Anoia: this I am using for saved video [12:45:55 CEST] <Anoia> I have no idea form the ffmpeg command line [12:46:31 CEST] <gurpartap> Anoia: ok....thanks for help [12:46:34 CEST] <Anoia> it's a feature of the RTSP protocol rather than the video format, and I use livemedia (in code) to do RTSP [12:53:33 CEST] <gormed> hi, some ffmepg dev here? [13:08:16 CEST] <durandal_1707> yes [13:14:53 CEST] <gormed> okay, my question is here: http://pastebin.com/5W1DrmSg [13:22:49 CEST] <gormed> if you need additional info, just say it :) [15:15:53 CEST] <gormed> Hey, just wanted to bump my question about using ffmpeg together with Unreal Engine 4: http://pastebin.com/5W1DrmSg [15:37:37 CEST] <trosh> hello all [15:38:32 CEST] <trosh> i'm converting a video, and i'm trying to change the audio framerate. i get a message saying [avi @ 0x1c1b140] New audio stream 0:2 at pos:3241480 and DTS:1.3035s [15:38:53 CEST] <trosh> but then when i go ffmpeg -i on the file i just created i can't see any 0:2 stream [15:39:00 CEST] <trosh> does this make sense ? [15:44:53 CEST] <trosh> ok ! [15:44:54 CEST] <trosh> thanks [15:47:05 CEST] <trosh> http://pastebin.com/xvQRMy09 [15:47:07 CEST] <trosh> this good ? [15:48:47 CEST] <chungy> Yes; the info message about the new stream is only about the input file, the output will still have only the streams that were originally detected/specified. [15:49:21 CEST] <trosh> so there is a new audio stream detected at 1.3s ? [15:49:36 CEST] <trosh> if i do ffmpeg -i on the original file, i only have 0:0 and 0:1 [15:49:42 CEST] <chungy> try doing "ffprobe -analyzeduration 5000000 segoyann.avi" and see if it detects 0:2 [15:50:12 CEST] <c_14> If you want the other stream as well, you'll at very least need -map 0 (or an equivalent that maps all the streams you want) [15:50:35 CEST] <trosh> (for more info, this is DV, and it starts with a test screen like frame for 1.3s) [15:52:10 CEST] <trosh> ffprobe give me only 0:0 and 0:1 (it's instant mind you) [15:52:36 CEST] <c_14> Try increasing -probesize as well. [15:53:43 CEST] <trosh> nothing new :/ [15:55:18 CEST] <trosh> if i -ss at 1.3, could it make any difference ? [16:01:20 CEST] <c_14> it might [17:32:03 CEST] <DHE> I'm running ffmpeg (git) and trying to use a filter_complex for multiple outputs with semi-similar filters. I wrote: ~/ffmpeg/ffmpeg_g -filter_complex '[0:v]pp=fd[deint];[deint]scale=w=1280:h=720[720p];[deint]scale=w=848:h=480[480p]' -i ..... [17:32:19 CEST] <DHE> but get the error: Stream specifier 'deint' in filtergraph description [0:v]pp=fd[deint];[deint]scale=w=1280:h=720[720p];[deint]scale=w=848:h=480[480p] matches no streams. [17:33:00 CEST] <DHE> The idea is that I do the deinterlace once, then scale as needed for each output. I may also do further filtering but I'm stuck here first [17:37:46 CEST] <relaxed> DHE: filters go after the input [17:38:07 CEST] <DHE> I tried that... same error [17:39:41 CEST] <relaxed> you want multiple streams in one output? [17:40:25 CEST] <DHE> several output files [17:40:45 CEST] <DHE> -map [720p] output1.mp4 -map [480p] output4.mp4 [17:41:05 CEST] <DHE> plus audio, obviously [17:41:19 CEST] <DHE> there's a fine line between being specific and flooding the channel.... [17:42:03 CEST] <relaxed> are you trying, ffmpeg -i input -filter:v 720p_filter 720p_output.mkv -filter:v 480p_filter 480p_output.mkv [17:43:09 CEST] <DHE> I just figured it out. I need a 'split' filter to make two outputs because ffmpeg doesn't allow the same output to be used multiple times in the filtergraph... [17:43:17 CEST] <relaxed> ah [17:43:52 CEST] <relaxed> can you pastebin the command that worked? I'm interested to see it. [17:45:18 CEST] <DHE> http://pastebin.com/wgejVhKL [17:45:26 CEST] <DHE> a bit simplified, removed all the codec options, etc [17:45:37 CEST] <relaxed> thanks [17:46:19 CEST] <DHE> looks like it works... [17:50:40 CEST] <relaxed> DHE: yadif is considered the best deinterlacer, fyi [17:52:33 CEST] <DHE> hmm... noted [18:51:45 CEST] <jc0n> I can ask for elp in here no? [18:51:48 CEST] <jc0n> *help [18:53:07 CEST] <jc0n> becuase I want to stream one video input to one file and two audio inputs to another. [18:53:22 CEST] <jc0n> using the amix filter for the second file [18:59:22 CEST] <JEEBsv> ffmpeg -i input -i input [things for the first output] output1 [things for the second output] output2 [18:59:48 CEST] <JEEBsv> I should have put input1 and input2 there because that is how you specify multiple inputs :P [19:00:14 CEST] <jcay> jc0n: sure, just be patient, the experts are busy :-) [19:00:29 CEST] <zhanshan> hi [19:00:29 CEST] <jc0n> jcay: thank you :) [19:01:20 CEST] <zhanshan> someone was friendly enough to help me to batch rename image files in a folder in order to make ffmpeg put them together in the originally intended way [19:01:27 CEST] <zhanshan> but I forgot the command line.. [19:01:47 CEST] <zhanshan> the files are i.e. fk1_daan_full0001.png fk1_daan_full31730.png [19:02:03 CEST] <zhanshan> so digits differ between 4 and 5 [19:04:17 CEST] <zhanshan> what I want to do is make them the same length in order to call the command: ffmpeg -f image2 -start_number 0000001 -i 'fk1_daan_full%05d.png' -i 'audio' -pix_fmt yuv420p -c:v libx264 -crf 18 -c:a flac 'out' [19:04:37 CEST] <zhanshan> or is that possible like this already [19:04:49 CEST] <zhanshan> start-frame would be of course 0001 [19:06:22 CEST] <zhanshan> the command to rename was something like: $ rename 's/([0-9]{4})/0$1/' * [19:06:45 CEST] <zhanshan> but I have absolutely no clue how these commands work. I can't understand them [19:07:48 CEST] <jc0n> when recording to one file, this is the stream mapping: http://hastebin.com/edayesobos.coffee [19:08:01 CEST] <zhanshan> I think I don't even need the command to use %05d [19:08:40 CEST] <relaxed> zhanshan: try, ffmpeg -pattern_type glob -i *png [19:12:19 CEST] <zhanshan> relaxed one time I had an issue resulting in a video where all frames between 1000 and 9999 were mixed up in every 11th frame [19:12:30 CEST] <zhanshan> that's why I wanted to try to avoid it [19:12:39 CEST] <zhanshan> because it's 68000 frames [19:13:42 CEST] <Netek> hi all, I am currently using ffmpeg for live streaming rtmp. My client forwards me the stream to my server and my server distributes this stream online, I wish to convert the rtmp live stream to mp4 to be displayed in html5 for any device. Is this possible? [19:17:39 CEST] <BtbN> mp4 is not streamable [19:18:16 CEST] <Netek> BtbN ok what would be recommended stream? I managed to convert to m3u8 but this is not compatible for desktop browsers [19:18:33 CEST] <BtbN> you can't easily livestream to html5 [19:18:46 CEST] <BtbN> Not in a way that works in more than one or two browsers. [19:19:12 CEST] <anoop_r> hi [19:19:20 CEST] <BtbN> Unless you use flash, rtmp is just fine then [19:19:24 CEST] <Netek> BtbN is there no way of using a fallback system? ie if m3u8 does not load then fall to rtmp stream? [19:19:33 CEST] <Netek> sorry my english is not perfect [19:19:33 CEST] <BtbN> html5 can't play rtmp [19:19:50 CEST] <anoop_r> sombody help me to correct this command line http://pastebin.com/9HAFqt3Q [19:19:54 CEST] <BtbN> And html5 can't play m3u8 (hls), except in Safari on OSX [19:19:58 CEST] <Netek> yes I understand that but can I not convert it to another format or html5? [19:20:06 CEST] <Netek> or/for [19:20:13 CEST] <BtbN> There is no common format for html5 [19:20:25 CEST] <Netek> hmm ok [19:20:31 CEST] <anoop_r> i don't know the corret format to put parametes with -x265-params [19:20:44 CEST] <BtbN> Safari on OSX plays HLS, Chrome(And anything Chromium based) plays DASH, IE plays DASH, Firefox doesn't support it at all. [19:21:07 CEST] <Netek> ok interesting [19:21:50 CEST] <Netek> so DASH would be what I would require to get somewhere at least [19:22:09 CEST] <BtbN> you also need a huge chunk of JavaScript, as no browser supports it natively [19:22:40 CEST] <zhanshan> relaxed not working! [19:22:53 CEST] <zhanshan> after frame 999 every eleventh frame is wrong [19:23:00 CEST] <Netek> BtbN this is something I need to learn, we have a client base that uses rtmp streaming services from us but now they have requested for multiple browser support [19:23:06 CEST] <zhanshan> it's a kind of stroboscope effect [19:23:23 CEST] <BtbN> Well, every streaming site still uses Flash for their live streams... [19:23:39 CEST] <Netek> orginally we did rtmp stream in browser using jwplayer with a for mobile viewers link directly the m3u8 [19:23:52 CEST] <Netek> android and apple does not support flash [19:24:00 CEST] <Netek> so I need to see if I can find something [19:24:13 CEST] <BtbN> There is no satisfiying solution. [19:24:23 CEST] <BtbN> Every Browser wants to push its own standard [19:24:33 CEST] <DHE> I find HLS works on Chrome for Android... [19:24:48 CEST] <BtbN> Yes, HLS works on Chrome for Android [19:24:54 CEST] <Netek> DHE yes HLS is not an issue but once you try to access on desktop it does not [19:24:55 CEST] <BtbN> but it's candidate for removal [19:25:04 CEST] <BtbN> In favor of DASH [19:25:13 CEST] <kepstin-laptop> firefox was supposed to support the mediasource extensions needed for dash as of v38, but I dunno what's happened there... [19:25:30 CEST] <BtbN> They implemented what was needed to get Youtube WebM working, and then stopped [19:26:23 CEST] <relaxed> zhanshan: create links and verify they're right with "ls -l", x=1; for i in *png; do counter=$(printf %05d $x); ln -s "$i" temp-"$counter".png; x=$(($x+1)); done [19:29:07 CEST] <relaxed> zhanshan: if so, ffmpeg -framerate $framerate -i temp-%05d.png ... [19:35:44 CEST] <zhanshan> relaxed: sorry I don't understand [19:36:09 CEST] <zhanshan> the following command seems to produce the proper video: $ ffmpeg -f image2 -start_number 00001 -i fk1_daan_full%05d.png -i 'FK-#1_daan_Accuracy1024s.flac' -pix_fmt yuv420p -c:v libx264 -crf 18 -c:a flac '002_FK-#1_daan.mkv' [19:36:15 CEST] <zhanshan> however [19:36:39 CEST] <zhanshan> I don't understand why [19:36:46 CEST] <Netek> I think the only solution would be to have rtmp and hls streaming at the same time and using a browser check on website, if browser is a mobile device show 1 player else show other browser [19:37:02 CEST] <Netek> player [19:37:07 CEST] <zhanshan> because fk1_daan_full033011.png for example has six digits, right? [19:37:26 CEST] <pzich> it appears to [19:37:58 CEST] <zhanshan> thanks pzich! [19:38:22 CEST] <pzich> do you want to reformat all of the files to pad to 6 digits? [19:41:09 CEST] <zhanshan> actually I'm looking for a command like that [19:41:33 CEST] <zhanshan> maybe even remove everything except the digits and the .png [19:41:35 CEST] <pzich> are all of your files of the format "fk1_daan_full<number>.png"? and you're using bash? [19:42:00 CEST] <zhanshan> get rid of this useless excess of name-space [19:42:10 CEST] <zhanshan> yes pzich! [19:42:26 CEST] <zhanshan> wait actually I'm using xfce4-terminal 0.6.3 [19:42:33 CEST] <zhanshan> is that a bash? [19:42:48 CEST] <zhanshan> I'm always confused by the different forms of command lines [19:43:40 CEST] <kepstin-laptop> the terminal (the program that draws letters on the screen) and the shell (the program that interprets what you type and runs commands) are two different things [19:43:49 CEST] <kepstin-laptop> you're probably running bash inside your xfce-terminal [19:44:55 CEST] <zhanshan> thanks kepstin-laptop! [19:45:25 CEST] <zhanshan> now I got it, so easy [19:45:48 CEST] <anoop_r> somebody help me with this http://pastebin.com/9HAFqt3Q [19:46:08 CEST] <zhanshan> so with my command %05d.png as expected ffmpeg only rendered until frame number 9999 of 68700 [19:46:17 CEST] <zhanshan> so it's a failure [19:48:32 CEST] <pzich> so just to check, all of the files start with "fk1_daan_full", have a series of numbers, then end in ".png"? [19:48:41 CEST] <pzich> and you'd like just 000001.png etc.? [19:49:41 CEST] <pzich> Try running this, do these renames look right to you? for F in *.png; do N="${F/fk1_daan_full}"; N="${N/.png}"; echo $F `printf %06d.png $N`; done [19:52:59 CEST] <zhanshan> pzich I'm gonna check on a few files first, thanks! [19:53:52 CEST] <pzich> You'll need to change that last "echo" to "mv" if it looks right, right now it's just printing out the old and new names [19:54:51 CEST] <zhanshan> pzich: bash: printf: 068001: invalid octal number [19:55:24 CEST] <pzich> heh, ok, just a sec [19:56:45 CEST] <anoop_r> -x265-params level=85:bitrate=746 [19:57:06 CEST] <anoop_r> when i use it like this bit rate changes but level not changing [19:57:15 CEST] <anoop_r> why ? [20:03:28 CEST] <pzich> zhanshan: try for F in *.png; do N="${F/fk1_daan_full}"; N="${N/.png}"; N=$((10#$N)); echo $F `printf %06d.png $N`; done [20:16:32 CEST] <jc0n> jcay: ready yet? [20:16:56 CEST] <zhanshan> pzich seems to work, thanks! [20:17:12 CEST] <pzich> cool [20:19:38 CEST] <zhanshan_eats> yeah 68k files now.... [20:19:56 CEST] <zhanshan_eats> l8er! [20:55:23 CEST] <HebusLeTroll> Hello, anyone knows what interpollation is used when using for example -vf "scale=1280:720" argument ? [21:02:37 CEST] <rsevero> I have several USB AV capture dongles being recorded by ffmpeg. Today one of them stoped working. I have already changed the USB dongle, upgraded ffmpeg and the kernel and even changed the computer that does the recording but still doesn't work. ffmpeg doesn't present any error. It just stays halted until I interrupt it with CTRL-C. You can see it's output here: http://pastebin.com/PunerWAK Any ideas? [21:03:55 CEST] <BtbN> Multiple capture dongles on the same bus? [21:04:03 CEST] <BtbN> Or each one on a seperate usb controler? [21:04:11 CEST] <rsevero> No, just one per USB host/bus. [21:05:05 CEST] <BtbN> You might be just running out of bandwidth then [21:05:25 CEST] <rsevero> How if I only have one dongle per SUB bus? [21:05:31 CEST] <rsevero> USB bus [21:05:41 CEST] <BtbN> So you do have them on seperate controlers? [21:06:04 CEST] <BtbN> In that case, does anything else work on that controler? Did you try swapping them around? [21:06:36 CEST] <rsevero> Take a look yourself: http://pastebin.com/ubAZaS9L [21:06:44 CEST] <rsevero> Yes, I swaped them around. [21:06:52 CEST] <rsevero> Just this one doesn't work. [21:07:20 CEST] <rsevero> It's as if ffmpeg doesn't want to record this particular TV channel anymore. [21:07:22 CEST] <anoop_r> anybody alive here to help me ? [21:07:36 CEST] <BtbN> Maybe it just broke? [21:08:25 CEST] <rsevero> I got a new USB dongle and replaced it. Still the same results. The results I pasted are with the new dongle. In fact I tried 5 different dongles already. [21:08:30 CEST] <anoop_r> http://pastebin.com/9HAFqt3Q [21:08:43 CEST] <anoop_r> please correct this command line [21:10:54 CEST] <anoop_r> http://pastebin.com/9HAFqt3Q [21:11:31 CEST] <anoop_r> this is not a script its an ffmpeg command line with x265 [21:11:44 CEST] <anoop_r> i need help with x265params [21:11:59 CEST] <rsevero> anoop_r: But it's missing the ffmpeg output. [21:12:18 CEST] <anoop_r> ok [21:13:50 CEST] <relaxed> why on earth would you need to set that many options? [21:14:38 CEST] <anoop_r> i am making a gui [21:14:46 CEST] <anoop_r> http://pastebin.com/KX2p3mRV [21:16:03 CEST] <anoop_r> any where can i get a reference for -x265-params [21:16:09 CEST] <relaxed> I hope you read https://trac.ffmpeg.org/wiki/Encode/H.265 [21:16:18 CEST] <relaxed> http://x265.readthedocs.org/en/latest/cli.html [21:24:57 CEST] <delarita> Hello, it is a message for dericed. I would like to use your code to overlay a x264 10bit 422 video encoded from ProResHQ to begin test comparison. I modify the level of grey from 128 to 512 as i was guessing that 10bit eqaul to 1024 grey level. But i don't have a result that match what i expect. I am going to post the video to be watch. [21:39:33 CEST] <anoop_r> -x264-params level=5.1:crf=23:tune=animation notworking [21:40:14 CEST] <bencc> when I cut and concatenate mp4, can I do it without transcoding? [21:40:15 CEST] <FDS_> quote the params [21:40:21 CEST] <bencc> I mean, cut in time [21:40:40 CEST] <FDS_> @anoop_r [21:41:38 CEST] <anoop_r> ffmpeg -i Input.mkv -c:v libx264 -preset veryslow -x264-params level=5.1:crf=23:tune=Animation -c:a copy output.mkv [21:41:39 CEST] <anoop_r> this is full params [21:41:55 CEST] <anoop_r> you mean to use like tune="animation" [21:42:13 CEST] <FDS_> ffmpeg -i Input.mkv -c:v libx264 -preset veryslow -x264-params level=5.1:crf=23:tune=Animation -c:a copy output.mkv [21:49:21 CEST] <Kuukunen> also, it should be -tune animation [21:49:36 CEST] <anoop_r> ya [21:49:48 CEST] <anoop_r> trying with that [21:49:51 CEST] <Kuukunen> (instructed him elsewhere, but mentioning here) [21:50:24 CEST] <anoop_r> @Kuukunen can you help me with x265-parmas [21:50:37 CEST] <anoop_r> some of the params not working [21:50:52 CEST] <anoop_r> no-constrained-intra:rect:no-amp:early-skip:fast-intra:no-cu-lossless:tskip-fast:b-pyramid: [21:52:00 CEST] <FDS_> what is not working ? [21:53:10 CEST] <Kuukunen> anoop_r: you still didn't pastebin your console output :P [21:53:20 CEST] <anoop_r> http://pastebin.com/KX2p3mRV [21:53:27 CEST] <anoop_r> see that [21:53:46 CEST] <anoop_r> if put all the options it will not work means it will not change [21:58:02 CEST] <zhanshan> pzich the files look integer to me [21:58:04 CEST] <zhanshan> thanks! [22:05:45 CEST] <Netek> is it normal to be using large cpu resources for converting rtmp to hls? [22:07:40 CEST] <BtbN> If you re-encode in the process, yes. [22:07:53 CEST] <Netek> ok [22:08:11 CEST] <BtbN> you most likely don't want to do that though [22:08:19 CEST] <Netek> well I managed to get it working to some standard [22:08:35 CEST] <Netek> BtbN if I need both to be live? Can you please explain [22:08:46 CEST] <BtbN> Why encode it a second time? Just stream-copy [22:08:59 CEST] <Netek> oh yes [22:08:59 CEST] <BtbN> Or use nginx-rtmp, which can output hls, dash and rtmp at the same time. [22:09:32 CEST] <Netek> hmm, a few people told me about nginx-rtmp but my linux friends keep pushing me to use ffmpeg direct [22:09:44 CEST] <Netek> they keep telling me dont use nginx-rtmp [22:10:31 CEST] <BtbN> Why not? [22:10:37 CEST] <Netek> no idea [22:10:46 CEST] <BtbN> It's the best streaming server around, if you don't want to pay for stuff like wowza [22:10:51 CEST] <Netek> I see [22:11:12 CEST] <Netek> I will have a look [22:11:29 CEST] <Netek> will this also reduce power consumption on server? [22:11:43 CEST] <BtbN> Why would it? [22:12:14 CEST] <Netek> hah just a silly question, I guess I deserve a silly response. sorry :) [22:12:17 CEST] <Netek> no reason why not [22:12:32 CEST] <BtbN> Just don't pointlessly re-encode [22:12:38 CEST] <BtbN> a stream copy needs basicaly no resources [22:12:50 CEST] <Netek> ok [22:40:43 CEST] <jbruggeman> Hi. The contribute link is broken on the ffmpeg.org webpage [23:06:11 CEST] <anoop_r> how to disable help in ffmpeg.exe [23:07:29 CEST] <anoop_r> like ffmpeg -h shows command line help [23:07:34 CEST] <anoop_r> i want to disable it [23:20:18 CEST] <Anoia> anoop_r: for what puroposes? [23:20:27 CEST] <Anoia> just don;t type ffmpeg -h [23:20:36 CEST] <Anoia> if youw ant to remove it, you have the source [00:00:00 CEST] --- Tue Jun 16 2015

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