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April 2015

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ffmpeg-devel.log.20150429
by burek 30 Apr '15

30 Apr '15

[00:08:37 CEST] <drwx> 12:41 < drwx> hi, could anyone look at this bug? https://trac.ffmpeg.org/ticket/2975 [00:08:40 CEST] <drwx> 12:41 < drwx> also, https://trac.ffmpeg.org/ticket/4487 looks like a duplicate [01:14:47 CEST] <penthief> Do you want a bug report containing something like the following http://paste.lisp.org/display/147622 ? (Incorrect resolution with x11grab causes a segfault when switching windows.) [01:16:15 CEST] <penthief> I can also chuck you a 7 meg core dump if you tell me where to put it. [01:18:12 CEST] <penthief> Possibly introduced with a6674d2e7771dbf7a4a5556f5e126be83cadac96 (Aug 24 2014). [03:41:24 CEST] <cone-188> ffmpeg 03Vittorio Giovara 07master:f5ba67ee1342: flacenc: Move a scratch buffer to struct used by the function [03:41:24 CEST] <cone-188> ffmpeg 03Michael Niedermayer 07master:77010a41fef0: Merge commit 'f5ba67ee1342b7741200ff637fc3ea3387b68a1b' [03:56:27 CEST] <cone-188> ffmpeg 03Vittorio Giovara 07master:3919a4572690: jpeglsenc: Check memory allocations [03:56:28 CEST] <cone-188> ffmpeg 03Michael Niedermayer 07master:27801500eed3: Merge commit '3919a4572690894d0a7ad4170d699c52b4748194' [04:03:25 CEST] <cone-188> ffmpeg 03Vittorio Giovara 07master:c3418201307c: jpeglsenc: Mark codec as init-thread-safe and init-cleanup [04:03:26 CEST] <cone-188> ffmpeg 03Michael Niedermayer 07master:966a57b6e918: Merge commit 'c3418201307c6b7a2b3e3be10f33ab4a20a32c3b' [04:11:08 CEST] <cone-188> ffmpeg 03Vittorio Giovara 07master:59dfc2b0c89f: h263: Drop commented-out code and a related otherwise unused function [04:11:09 CEST] <cone-188> ffmpeg 03Michael Niedermayer 07master:78f50e7eb694: Merge commit '59dfc2b0c89f35c5805abee72c2ae4ee7af8e98d' [04:22:49 CEST] <cone-188> ffmpeg 03Vittorio Giovara 07master:0c69164f451c: h263: Convert function to macro [04:22:50 CEST] <cone-188> ffmpeg 03Michael Niedermayer 07master:f6c524a9399f: Merge commit '0c69164f451cc9ca6ce9d6e7568083e2776bc845' [05:46:40 CEST] <cone-188> ffmpeg 03John Högberg 07master:82de8d71118f: mpegts: Update the PSI/SI table only if the version change [05:46:41 CEST] <cone-188> ffmpeg 03Michael Niedermayer 07master:12f7253d0ea1: Merge commit '82de8d71118f4eafd6a43e9ea9169bd411793798' [05:46:42 CEST] <cone-188> ffmpeg 03Michael Niedermayer 07master:639781492684: avformat/mpegts: reset last_version on seeking [05:56:12 CEST] <cone-188> ffmpeg 03Luca Barbato 07master:0ac8ff618c5e: avresample: Reallocate the internal buffer to the correct size [05:56:13 CEST] <cone-188> ffmpeg 03Michael Niedermayer 07master:215430e049b7: Merge commit '0ac8ff618c5e6d878c547a8877e714ed728950ce' [11:37:09 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:27b0e6ebfd47: h264: drop needs_realloc [11:37:10 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:0d5e615c1058: Merge commit '27b0e6ebfd47b0c11156c18b90fa8c571f0f60c3' [11:46:04 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:9d33bab583a8: h264: drop H264Context.ouputed_poc [11:46:05 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:e8a460672f7e: Merge commit '9d33bab583a82cf12286c65258a29c6888e1ff98' [11:55:52 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:a6cd154463be: h264: move the DPB init/uninit to init_context()/free_context() [11:55:53 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:cd63252f1ff6: Merge commit 'a6cd154463bea7eb56d28192db4c8c6d83f67fd7' [12:20:08 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:54986d6db911: h264: move context reinit lower down in update_thread_context() [12:20:09 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:580c86925ddf: Merge commit '54986d6db9116ef9704b1ce8414ffdb7f1ca127f' [13:24:36 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:ee62b364be0c: h264: eliminate ff_h264_set_parameter_from_sps() [13:24:37 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:ab7ad5e6a1b8: Merge commit 'ee62b364be0c30cba83b5ff10a3ca8c3e866ade6' [13:33:36 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:11f024ef0ab9: h264: move freeing the escaped RBSP buffer to free_context() [13:33:37 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:d3a5dba81b3a: Merge commit '11f024ef0ab923ed8680fc35a087d576e549c849' [13:41:23 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:440e8dd374b7: h264: drop a comment that carries no useful information [13:41:24 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:1aa3a0f457c3: Merge commit '440e8dd374b732c48d564d9f1bb0ec3b1b786fb9' [13:48:31 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:29dcde2b8755: h264: copy nal_length_size in update_thread_context [13:48:32 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:16b90d0bcbca: Merge commit '29dcde2b87559b72957454a7e9bcd5227d8057ba' [13:55:53 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:c9ccbc7333ed: h264: update the current bit depth/chroma type in init_slice_header() [13:55:54 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:51f7ba9ceec0: Merge commit 'c9ccbc7333eddd025ebbde5cc4f27d68a950c623' [14:02:52 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:8a66fd40260b: h264: drop the reinit parameter from init_slice_header() [14:02:53 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:cc87493f0654: Merge commit '8a66fd40260b7aae6226d68c4dbad43b05a8e524' [15:22:26 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:88c612e3a4d5: h264: merge the init and reinit paths in update_thread_context() [15:22:27 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:bd7dce594ea7: Merge commit '88c612e3a4d5b584e2d9f6e2541de78d67bdfb9f' [15:22:28 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:4a95cd5d9b4b: avcodec/h264_slice: Also copy x264_build in ff_h264_update_thread_context() [15:36:36 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:7a4f74eed51f: h264: embed the DPB in the context [15:36:37 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:07a79cf8694a: Merge commit '7a4f74eed51f914e9bbfebaffd4a92ac6791f819' [16:16:02 CEST] <saste> kierank, michaelni, do you know why there is no multi-program support in the mpeg-ts muxer? [16:16:33 CEST] <saste> see http://trac.ffmpeg.org/ticket/4525 [16:48:26 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:a0f2946068c6: h264: use properly allocated AVFrames [16:48:27 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:36f862e04c2a: Merge commit 'a0f2946068c62e18cb05ac25c0df3d86077251a6' [17:00:15 CEST] <cone-503> ffmpeg 03Anton Khirnov 07master:a09c499a2e25: rawenc: properly allocate coded_frame [17:00:16 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:50e46b7f0dca: Merge commit 'a09c499a2e25ded575cb9f37528384402be86ad8' [17:19:32 CEST] <cone-503> ffmpeg 03foo86 07master:e80b2b9c8171: avcodec/dca_parser: Extend DTS core sync word and fix existing check [17:26:07 CEST] <j-b> So, what's the status/mess with DCA? [17:26:43 CEST] <ubitux> stall game? [17:26:43 CEST] <Compn> i think they are arguing about some encoder patch [17:27:33 CEST] <Daemon404> "just use foo86's library" [17:28:03 CEST] <j-b> I mean, there is a new DTS-HD decoder in libav, but was it merged here? There was different code here for DTS-HD too, right? And there is the foo86 decoder, too, now? [17:28:23 CEST] <Daemon404> j-b, the foo86 one is the only one with a fixed point dca core implementation [17:28:27 CEST] <Daemon404> i.e. actually lossless [17:29:22 CEST] <j-b> Daemon404: ok. The libav one was merged in FFmpeg or? [17:29:57 CEST] <Daemon404> hell if know [17:30:18 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:8a710ddc7c11: avcodec/libx264: Remove cleanup code, its uneeded since FF_CODEC_CAP_INIT_CLEANUP [17:32:24 CEST] <Compn> its chaos! [17:32:25 CEST] <Compn> :P [17:44:09 CEST] <wm4> j-b: it was merged, and disabled by default [17:44:31 CEST] <wm4> at least AFAIK [17:44:53 CEST] <wm4> and the libdcadec wrapper exists in both libs too now [17:45:16 CEST] <j-b> wm4: disabled, because of innaccuracy? [17:45:38 CEST] <j-b> what about X96 ? [17:45:39 CEST] <wm4> and I think there were other problems with it too? maybe nevcairiel knows details [17:45:48 CEST] <wm4> no idea [17:46:07 CEST] <wm4> if you want some feature, I'd say foo86 is likely to implement it in his lib [18:27:53 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:14e9a20083c9: avformat/mpegts: Also parse the FMC descriptor if the codec has not been identified yet [18:29:44 CEST] <nevcairiel> j-b: wm4: libav disabled it as well by default on my suggestion, to avoid regressions in decoding, which do exist, and X96 is not supported [18:30:12 CEST] <nevcairiel> dcadec from foo86 supports all of the things in perfect accuracy however, the only missing feature is DTS Express, aka. LBR [18:32:43 CEST] <kierank> That is a separate codec [18:34:19 CEST] <nevcairiel> well it uses the same extension header, and presumably shares similar coding techniques [18:35:57 CEST] <kierank> It's different totally [18:36:11 CEST] <kierank> It's like e-ac3 vs ac-3 [18:36:20 CEST] <kierank> But it is marketed like it is similar [18:36:31 CEST] <nevcairiel> which share a lot of code in avcodec? :d [18:36:49 CEST] <kierank> Yes share code because of headers etc [18:36:56 CEST] <kierank> But it's a bigger gap [18:37:01 CEST] <j-b> nevcairiel: regressions? how can there be regressions? [18:37:12 CEST] <j-b> I mean, the core is the core, no? [18:37:30 CEST] <nevcairiel> j-b: trying to parse HD parts, failing, causing odd behavior, like switching between 5.1 and 7.1 decoding every other audio frame [18:39:51 CEST] <j-b> nevcairiel: do you have samples for such fails? [18:40:19 CEST] <nevcairiel> somewhere, i gave them to the libav people when it was first merged [18:54:04 CEST] <rcombs> any particular reason why libavformat/tls.c doesn't verify certs by default? [19:01:19 CEST] <wm4> because it just breaks shit [19:01:48 CEST] <rcombs> what kind of shit [19:01:52 CEST] <rcombs> (maybe someone should fix that) [19:02:43 CEST] <BtbN> I think enablding it would break a bunch of stuff. [19:03:17 CEST] <BtbN> And cause a bunch of angry people, because their self-signed certs suddenly don't work anymore. [19:03:51 CEST] <rcombs> there's an AVOption to disable it [19:04:04 CEST] <BtbN> Yes, but it breaks existing setups. [19:04:17 CEST] <BtbN> changing defaults allways does [19:04:32 CEST] <rcombs> biggest issue I see is that it doesn't seem to pull in the system cert bundle by default [19:05:23 CEST] <wm4> rcombs: maybe the main reason is because nobody wants to fight these crappy tls libs [19:06:10 CEST] <rcombs> it's not like self-signed cert users haven't had to disable verification in every other piece of software they use (or actually add their certs to their cert bundle, in which case this isn't a problem if that's used [which it isn't but should be]) [19:10:41 CEST] <rcombs> hmm, curl (which does this pretty reliably cross-platform) uses OSX's and Windows's built-in TLS facilities (Secure Transport and Schannel), and on Linux the default CA bundle for OpenSSL/GNUTLS and all the other libs they support is a configure option [19:16:36 CEST] <wm4> make libavformat use libcurl [19:18:20 CEST] <rcombs> I've seen that done, actually [19:18:53 CEST] <rcombs> quite some time ago, before lavf's protocol support was as stable [19:19:09 CEST] <rcombs> (wait a sec why is avio part of lavfnever mind) [19:20:37 CEST] <nevcairiel> i dont bundle any tls/ssl support in my windows builds right now because openssl is a license nightmare, and gnutls is terrible to build on windows [19:20:54 CEST] <nevcairiel> i was meaning to see if building a schannel native-windows th ing would be as terrible as people claim.. [19:25:19 CEST] <rcombs> Secure Transport does let you provide your own read/write callbacks, so tls.c could continue to use lavf's tcp interface as usual [19:25:24 CEST] <rcombs> hmm, I might actually do this [20:00:55 CEST] <ramiro> are aac frames always standalone? (as in they don't depend on data from previous frames) [20:22:10 CEST] <cone-503> ffmpeg 03Michael Niedermayer 07master:828759ed9afa: avcodec/cook: Remove unneeded #if [20:25:36 CEST] <michaelni> about ssl certs, everything should check them by default IMHO, its more secure [20:28:34 CEST] <wm4> lol [20:28:36 CEST] <wm4> "secure" [20:28:48 CEST] <wm4> a remote server can make ffmpeg connect anywhere you want [20:34:16 CEST] <rcombs> wm4: http://suptg.thisisnotatrueending.com/archive/24232601/images/1365921834628… [20:36:59 CEST] <wm4> at least http redirects and things like MOV reference files and HLS (which open further requests) [20:44:53 CEST] <rcombs> well that's the server you asked for (which could be verified) directing you somewhere else [23:23:53 CEST] <cone-837> ffmpeg 03Michael Niedermayer 07master:f8fa392a2edb: avfilter/vf_vignette: force per frame evaluation if per frame variables are used [00:00:00 CEST] --- Thu Apr 30 2015

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ffmpeg.log.20150429
by burek 30 Apr '15

30 Apr '15

[00:57:14 CEST] <poisond> hi :) Are the available codec parameters for nvenc_h265 documented somewhere? [00:58:01 CEST] <poisond> are they the same as for x265? [01:07:17 CEST] <BtbN> poisond, there is some command to list the options. [01:09:25 CEST] <BtbN> poisond, ffmpeg -h full shoult list them, i forgot what the command was to display only the help for one codec. [01:09:28 CEST] <BtbN> *d [01:12:06 CEST] <poisond> I'm not seeing it :/ [01:17:28 CEST] <BtbN> Does it list nvenc in the -encoders output? [01:18:25 CEST] <poisond> yeah, nvenc and nvenc_h265. But I'm looking for a list of supported options of those encoders. The ones from libx265 don't work [01:18:45 CEST] <poisond> like -profile [01:18:48 CEST] <BtbN> Yes, -h full should have it [01:19:55 CEST] <poisond> aaaaah, sorry. already past midnight and I should read more carefully ^^ [01:20:40 CEST] <BtbN> They are roughly inspired by the x264 options, but not the same. [01:23:19 CEST] <poisond> wow, transcoding at 80 FPS, definately better than ~7FPS :) [01:23:40 CEST] <BtbN> Don't expect too much from the quality though. [01:24:02 CEST] <poisond> :( [01:24:51 CEST] <BtbN> Even x265 is currently still beaten by x264, depending on the settings. [01:25:14 CEST] <trodis_> tell me what is the best codec + format to stream audio in realtime [01:25:23 CEST] <trodis_> with ffpmeg and ffserver [01:25:23 CEST] <BtbN> opus. [01:25:31 CEST] <trodis_> opus is full of delay [01:25:36 CEST] <trodis_> i cant get opus to work without delays [01:25:38 CEST] <BtbN> Use it in low delay mode then. [01:25:47 CEST] <trodis_> how can u pls pls pls show me how [01:25:55 CEST] <trodis_> im struggling days with this problems [01:26:03 CEST] <BtbN> No idea, never used it myself. But i know it exists, various voice-com-tools like mumble and teamspeak use it. [01:26:04 CEST] <kepstin> even in default mode the delay should only be in the 25msish range... [01:26:22 CEST] <poisond> there's -tier and -profile, both with main and high parameters. Are those just duplicates? [01:26:30 CEST] <BtbN> Nope [01:26:36 CEST] <BtbN> there are tiers and profiles for h265 [01:26:48 CEST] <kepstin> if you're getting delay with opus, it's almost certainly due to buffering somewhere, and it would be a problem regardless of codec. [01:26:57 CEST] <poisond> BtbN: ok, thanks :) [01:27:00 CEST] <BtbN> nvenc only supports exactly one profile, so there is no point in messing with that option. It supports diffrents tiers though. [01:27:04 CEST] <BtbN> For h265 [01:27:10 CEST] <BtbN> For h264 is supports all profiles. [01:28:23 CEST] <poisond> BtbN: well, at least with libx265 I've gotten some astonishing results size-wise [01:28:47 CEST] <BtbN> iirc it's already better for 4K stuff [01:29:56 CEST] <trodis_> im getting a > 5 sec. delay with libopus + ogg [01:30:06 CEST] <trodis_> let me show u my ffserver.conf [01:30:11 CEST] <kepstin> trodis_, with what streaming protocol and what player? [01:30:18 CEST] <trodis_> http and ffplay [01:30:20 CEST] <kepstin> it's most likely to be player buffering, really :/ [01:30:27 CEST] <BtbN> propably just your player buffering. [01:30:29 CEST] <trodis_> http://ix.io/ib5 [01:30:44 CEST] <trodis_> ffmpeg -f pulse -i default http://192.168.1.138:8090/feed1.ffm [01:30:51 CEST] <trodis_> ffplay -nodisp -fflags nobuffer http://192.168.1.138:8090/test.ogg [01:31:26 CEST] <kepstin> if you want really low delay, you should probably be looking at rtp or something rather than http streaming. [01:32:10 CEST] <trodis_> can u configure my ffserver.conf so i can use rtp [01:32:22 CEST] <trodis_> i tried a couple of times with the documentation [01:32:26 CEST] <trodis_> and failed -.- [01:33:20 CEST] <BtbN> What are you trying to do that a few seconds delay are that bad? [01:35:07 CEST] <trodis_> i want to broadcast microphone input, i need a system for interpreting in another language. Right now i have 20 Raspberry pis and one raspberry pi is running ffserver, the interpreter should talk into the microphone and ffmpeg captures the input from the mic and sends it to the ffserver. And the Raspberry pi's should play that [01:35:33 CEST] <trodis_> by connecting to ffserver [01:36:36 CEST] <BtbN> yes, RTP or some other udp multicast thing is propably what you want. [01:37:03 CEST] <trodis_> how does rtp work, can u explain a little, do i need ffserver for rtp? [01:37:34 CEST] <BtbN> https://trac.ffmpeg.org/wiki/StreamingGuide#StreamingasimpleRTPaudiostreamf… [01:38:30 CEST] <kepstin> that's a unicast rtp stream, so it'll only send to one client. gets more complex to send it out to multiple clients. [01:38:53 CEST] <trodis_> yeah that is the whole point using multiple clients [01:39:05 CEST] <trodis_> i would love to use mumble, but mumble does not offer a cli client [01:39:24 CEST] <trodis_> and i need a cli low ressource client, because the raspberry pis are my clients [01:39:29 CEST] <kepstin> if you're on a closed, private network multicast could be an option. [01:39:38 CEST] <trodis_> yeah it is [01:39:55 CEST] <trodis_> i will setup a local lan just for this purpose its a public conference [01:40:06 CEST] <kepstin> in which it's pretty much the same thing, except you send to a multicast address instead of a single client address [01:40:10 CEST] <trodis_> so one WLAN Router with 300Mbit should be able to handel 20 raspberry pis [01:40:22 CEST] <kepstin> or broadcast address even [01:40:32 CEST] <BtbN> wlan isn't multicast capable, as each clients has a seperate encrypted channel. [01:41:14 CEST] <trodis_> how would you handle this kind of problem [01:41:30 CEST] <BtbN> I wouldn't use wlan. [01:41:55 CEST] <trodis_> yeah for cable i had to use a switch [01:42:02 CEST] <BtbN> exactly. [01:42:04 CEST] <trodis_> its hard to hide all thoes things [02:27:09 CEST] <[2]Matoro> hello, is anyone here? [02:31:00 CEST] <bofh> Hi! I have an mp3 file, I want to get MP4 from this mp4 file and a single image [02:31:06 CEST] <bofh> ffmpeg -y -loop 1 -i ololo.jpg -i devzen.mp3 -r 30 -b:v 2500k -shortest devzen.mp4 [02:31:09 CEST] <bofh> doesn [02:31:11 CEST] <bofh> 't work [02:31:20 CEST] <bofh> Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height [02:33:45 CEST] <trodis_> try -b:v 250k [02:34:35 CEST] <[2]Matoro> hey [02:35:08 CEST] <[2]Matoro> just a quick question for anyone who's familiar with AVI, how would you go about adding ttf fonts that an ASS script uses? [02:35:21 CEST] <bofh> nevermind, specifying -s 540x480 fixed the error [02:35:32 CEST] <bofh> I still wonder why I couln't get dimensions from jpeg [02:55:29 CEST] <Freakshow> hello folks& does anyone have any experience with the thread_message_queue message present in v2.6.1 and above? [02:55:42 CEST] <Freakshow> Ive raised it&and it keeps suggesting I raise it [02:56:16 CEST] <Freakshow> Im currently using an input from an MPTS unicast source [07:33:24 CEST] <jookiyaya> can you guys tell the difference between ac3 and dts on DVD-video? [09:35:09 CEST] <pickled> if i want to output a file to something like x.tmp (instead of x.mp4) how would i do that? [09:35:42 CEST] <pickled> but the .tmp data is mp4 of course. [09:36:00 CEST] <relaxed> pickled: -f mp4 x.tmp [09:36:07 CEST] <pickled> thank you! [10:44:01 CEST] <kagami_> Hi guys. Does anybody know why VPX_IMG_FMT_HIGHBITDEPTH cannot be enabled via configure? It's used to allow highdepth profiles of VP9. [10:51:45 CEST] <relaxed> kagami_: did you compile libvpx with --enable-vp9-highbitdepth ? [10:58:17 CEST] <kagami_> relaxed: I got this error message: --enable-vp9-highbitdepth [10:58:23 CEST] <kagami_> I'm using latest commit from git [10:58:35 CEST] <kagami_> oops [10:58:44 CEST] <kagami_> this one: >Unknown option "--enable-vp9-highbitdepth". [11:02:50 CEST] <relaxed> no, I meant libvpx's configure [11:10:24 CEST] <kagami_> relaxed: recompiled libvpx and ffmpeg, now it works. Many thanks! [11:11:12 CEST] <relaxed> you're welcome [16:01:51 CEST] <RobotsOnDrugs> are there any good denoisers in ffmpeg? [16:02:10 CEST] <RobotsOnDrugs> afaict, hqdn3d seems to be the best one in ffmpeg, but it's not that great [16:02:35 CEST] <trodis_> hi how can i be able to establish low delay connection within this scenario: ffmpeg <-> ffserver <-> 20 ffplay clients [16:03:26 CEST] <trodis_> i tried libopus and ogg as the container, but the delay is > 5 sec. i turned off the buffering on the ffplay client sides but still no improvement [16:03:39 CEST] <trodis_> and i activated the Option application lowdelay for libopus [16:03:41 CEST] <trodis_> still nothing [16:04:01 CEST] <trodis_> the weird part is, when i use libmp3lame + mp3 format, the delay is only 1 sec [16:10:24 CEST] <rsevero> In ffserver, can I have 2 streams with different video and/or audio characteristics being feed from the same feed? [16:28:52 CEST] <djems54> Hi, I had a problem when recording a MP4 file. Now when I want to open it or convert it I have this message: [16:29:23 CEST] <djems54> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x2568360] moov atom not found 20150425-2108-5-4-1001.mp4: Invalid data found when processing input [16:29:54 CEST] <djems54> Does anyone help me to fix it ? [16:30:10 CEST] <djems54> I work on a Debian server [16:30:31 CEST] <djems54> ffmpeg version 2.6.git Copyright (c) 2000-2015 the FFmpeg developer [16:41:14 CEST] <BtbN> The process writing the file was interrupted before it could finish writing the header, which is at the end of an mp4 file. [16:41:19 CEST] <BtbN> Without the moov atom, the file is useless. [16:41:49 CEST] <BtbN> Don't use mp4 for recording, it's a horrible container. [16:43:33 CEST] <albru123> This might be an AviSynth question rather than ffmpeg, but is ffdshow used by default if I use DirectShowSource() in AviSynth script? I'm encountering some kind of a bug, leading to skipping of first few frames in the video (like 50 or so) [16:50:03 CEST] <djems54> @BtbN : yes, the process was interrupted [16:50:23 CEST] <BtbN> yes, the file is lost then. [16:51:01 CEST] <djems54> What would be the best alternative as container ? [16:51:31 CEST] <djems54> there is no way to recover the file? [16:51:34 CEST] <BtbN> flv, mkv [16:51:50 CEST] <djems54> Ok, MKV with the same codec [16:51:51 CEST] <BtbN> Also, don't interrupt the process [17:01:44 CEST] <djems54> there is no way to recover the existing file? [17:04:34 CEST] <c_14> djems54: you can try https://github.com/ponchio/untrunc . May or may not work. [17:04:36 CEST] <hendry> is there a ffmpeg one liner for creating a HLS or DASH stream from v4l? http://stackoverflow.com/q/29935731/4534 [17:07:08 CEST] <djems54> c_14: i will try your link, thanks [17:11:24 CEST] <djems54> untrunc doesnt work :( [18:06:57 CEST] <inetd50> hi [18:07:30 CEST] <inetd50> im trying to stream a video file using ffmpeg, but is there a way to loop the file? [18:09:29 CEST] <mccoy_slack> while /bin/true; do .... ffmpeg stuf... ; done [18:10:01 CEST] <c_14> Not perfect since ffmpeg will close and reopen the output stream. [18:11:44 CEST] <mccoy_slack> true [18:13:24 CEST] <average> hey [18:13:35 CEST] <average> any planet.ffmpeg.org or something like that ? [18:14:08 CEST] <c_14> hmm? [18:14:28 CEST] <average> c_14: planet.debian.org for example has news about debian [18:14:28 CEST] <inetd50> c_14 exactly.. i want it to be smooth loop [18:14:45 CEST] <inetd50> c_14: how come ffmpeg doesnt allow looping out of the box [18:15:04 CEST] <inetd50> i cannot find anything on the internet that allow me to loop seemlessly [18:15:11 CEST] <c_14> inetd50: nobody implemented it yet [18:15:24 CEST] <inetd50> dam [18:15:36 CEST] <inetd50> can it be done outside of ffmpeg? [18:15:41 CEST] <inetd50> using other tools [18:16:09 CEST] <inetd50> like ffplay or some other way then pipe to ffmpeg [18:16:17 CEST] <c_14> average: there's a news section on https://ffmpeg.org/index.html#news [18:16:39 CEST] <c_14> inetd50: you could create a really long concatfile and use the concat demuxer [18:16:53 CEST] <c_14> It won't be infinite, but you could easily make it n years long. [18:17:12 CEST] <average> c_14: yeah but that's just news from ffmpeg's team [18:17:20 CEST] <average> c_14: i'm also interested in an RSS aggregator about ffmpeg [18:17:32 CEST] <average> where it would collect what people around the web say and do with ffmpeg [18:17:52 CEST] <c_14> inetd50: You might even be able to make it infinite. Not sure if the demuxer cares if the file is finite. [18:17:54 CEST] <inetd50> inetd50: i tried the concat demuxer...but the output file doesnt seem to work properly with my hardware encoder :| [18:17:55 CEST] <c_14> average: nothing I know of [18:18:40 CEST] <average> ok [18:18:44 CEST] <inetd50> c_14: any sample cli you can recommend? [18:19:23 CEST] <c_14> inetd50: hmm? [18:19:35 CEST] <c_14> example cli for what? [18:20:25 CEST] <c_14> s/example/sample/ [18:23:08 CEST] <inetd50> c_14: concat demuxer... [18:24:27 CEST] <c_14> https://trac.ffmpeg.org/wiki/Concatenate [18:24:42 CEST] <c_14> Look at the examples for generating the file with loops. [18:25:05 CEST] <average> what ffmpeg blogs do you guys read though ? [18:25:10 CEST] <c_14> Just that instead of find or for file in; you'd either add the same file n times [19:55:10 CEST] <kathy_> hello, i want to convert a 10hr youtube video to audio. using youtube-dl i can extract audio after download. Is there a way to use ffmpeg and do it as it streams [19:57:35 CEST] <sfan5> kathy_: you can tell youtube-dl to download only the audio [19:58:12 CEST] <jonet> I'm trying to use libarchive along with ffmpeg to read metadata from compressed files, never used ffmpeg library before so I'm very fresh with this. [19:58:54 CEST] <kathy_> sfan5: i think that is part of post processing only though. AFAIK. so a 10 hr long video might take a lot of time that way [19:59:30 CEST] <sfan5> i don't think it's part of the post processing [19:59:38 CEST] <jonet> I'm trying to use a AVIOContext with some methods to read and seek forward in the archive which seems to work fine since the read function is clearly executed, but after a few reads it segfaults. [20:00:18 CEST] <jonet> Also, I can't really find ... Well, any documentation for libav* at all, except from the autogenerated API docs. [20:02:02 CEST] <jonet> Here's my code; https://gist.github.com/wolfhechel/860e557ef26ef46b7c0a [20:02:06 CEST] <sfan5> kathy_: when i pass -f 141 to youtube-dl it shows a smaller filesize than downloading the whole video [20:03:57 CEST] <kathy_> sfan5: https://github.com/rg3/youtube-dl/blob/master/README.md#readme (it says extract audio -x in post processing) [20:04:40 CEST] <sfan5> kathy_: thats a different option, -f 141 only downloads the audio, -x seems to extract it after downloading everything [20:04:54 CEST] <kathy_> sfan5: what is 141 for? [20:05:11 CEST] <sfan5> kathy_: 141 m4a audio only DASH audio 255k , m4a_dash container, aac @256k (44100Hz), 9.08MiB [20:05:21 CEST] <sfan5> (-F flags shows the available formats) [20:05:59 CEST] <kathy_> sfan5: thanks! if i want to convert this to ogg format with ffmpeg while it streams? [20:09:56 CEST] <sfan5> kathy_: youtube-dl -q -f 141 "VIDEO_URL" -o /dev/stdout | ffmpeg -i - -c:a libvorbis -q:a 4 -y out.ogg [20:09:59 CEST] <sfan5> that works for me [20:10:29 CEST] <sfan5> the only problem is that ffmpeg waits for more input after the audio ends [20:12:56 CEST] <sfan5> kathy_: this one works better: youtube-dl -f 141 "video_url" -o - | ffmpeg -i - -c:a libvorbis -q:a 4 -y out.ogg [20:13:11 CEST] <sfan5> youtube-dl will throw an error after it finishes [20:13:18 CEST] <sfan5> that's fine as long as the audio file is ok [20:15:46 CEST] <kathy_> sfan5: since the audio is huge.(10 hrs) is there a way to download it in chunks. manually give start and finish times . [20:16:32 CEST] <sfan5> no idea, i don't know of any way to do that [20:16:54 CEST] <kathy_> sfan5: ok . thanks ! :) [20:18:46 CEST] <PieterMX> Hi [20:20:10 CEST] <PieterMX> Anyone with experience in HLS configuration for ffserver? [20:23:00 CEST] <PieterMX> It seems I can't set hls_time etc., and the codec is giving me: [20:23:01 CEST] <PieterMX> Wed Apr 29 20:16:53 2015 [hls @ 0x312c3f0]failed to rename file .tmp to [20:23:04 CEST] <maqr> PieterMX: i'm currently fighting with HLS, but i'm not using ffserver [20:23:33 CEST] <PieterMX> hi maqr, using ffmpeg directly? [20:23:37 CEST] <maqr> PieterMX: yeah [20:23:44 CEST] <maqr> PieterMX: you on osx? [20:23:49 CEST] <PieterMX> debian [20:23:58 CEST] <PieterMX> or at least the server is :) [20:24:10 CEST] <maqr> PieterMX: got osx available? you'll need mediastreamvalidator [20:24:17 CEST] <maqr> i'm not sure if it's available on any other platforms [20:24:25 CEST] <maqr> PieterMX: i'll gist what i'm doing for you to take a look at though, sec [20:25:00 CEST] <PieterMX> apple's mediastreamvalidator? [20:26:00 CEST] <maqr> PieterMX: yeah [20:26:08 CEST] <maqr> PieterMX: this is how i'm testing https://gist.github.com/anonymous/f5c824696e9213025fa3 [20:26:15 CEST] <PieterMX> I have OS X over here [20:26:47 CEST] <PieterMX> lets see if my apple dev account is still active [20:27:10 CEST] <maqr> PieterMX: i have an input.mp4, that hls.m3u8, and my transcode.sh in the directory... it runs the ffmpeg command, spawns a test http server from the directory, then tries to validate it with mediastreamvalidator [20:27:39 CEST] <maqr> PieterMX: my goal is to get the validator to pass, and i'm in a lot better shape than yesterday, but the bitrates are still wrong [20:27:44 CEST] <PieterMX> mediastreamvalidator is not installed by default is it? [20:27:53 CEST] <maqr> nope [20:28:24 CEST] <maqr> PieterMX: https://developer.apple.com/streaming/ link on the right, but you need to log in to get it [20:28:52 CEST] <PieterMX> I'm signed in but cant find the download :S [20:29:01 CEST] <PieterMX> it says "no records" lol [20:29:30 CEST] <maqr> PieterMX: i can host the dmg for you if you want [20:29:53 CEST] <maqr> PieterMX: you shouldn't run untrusted binaries but https://dl.dropboxusercontent.com/u/19471040/streamingtools_beta194_signed.… [20:29:58 CEST] <maqr> i think the dmg is signed somehow [20:30:01 CEST] <maqr> or maybe you can google the hash or something [20:30:15 CEST] <maqr> i get so frustrated with apple [20:30:41 CEST] <PieterMX> cant do sorry, should be somwhere on that site let me look [20:31:24 CEST] <maqr> PieterMX: does this work for you? http://adcdownload.apple.com/QuickTime/http_live_streaming_tools/streamingt… [20:31:39 CEST] <maqr> maybe try it with a browser that's logged in to somewhere else on apple's dev site [20:31:42 CEST] <PieterMX> You may be a member of a program that does not include this page as a benefit. [20:31:43 CEST] <PieterMX> lol [20:31:58 CEST] <maqr> :/ [20:33:17 CEST] <maqr> PieterMX: you need that tool either way, but are you doing live HLS or serving static videos as HLS? [20:33:46 CEST] <PieterMX> live [20:34:04 CEST] <maqr> ah, i'm trying to do static, i guess we have different problems then, i'm stuck on wrong bitrates coming out of the transcoding [20:34:16 CEST] <maqr> you still need to validate your live stream with the validator though [20:44:05 CEST] <jonet> Actually nevermind, I was reading really old material. Once I just simplified the whole thing with the avio_read example it went smooth as silk. [20:48:59 CEST] <maqr> does it ever make sense for bufsize to be smaller than the target rate? [20:49:06 CEST] <maqr> i've read the docs, but i'm still not very clear on what bufsize actually is [20:49:46 CEST] <Mavrik> Size of player's buffer [20:50:10 CEST] <Mavrik> And it makes sense if the players buffer holds less than 1 sec video [20:50:52 CEST] <maqr> Mavrik: that still doesn't make sense to me... does a smaller bufsize mean more headers or something? [20:51:08 CEST] <maqr> like are they actually separate chunks inside of the TS files? [20:51:12 CEST] <Mavrik> No. [20:51:39 CEST] <Mavrik> Encoder keeps track of how the buffer is filled while encoding and will overshoot maxrate if it has to [20:51:52 CEST] <Mavrik> And then drop under maxrate to make sure player's buffer refills [20:52:10 CEST] <maqr> yeah, i'm overshooting now and failing HLS validation... so it sounds like i need a smaller bufsize [20:52:30 CEST] <PieterMX> hmmz, cant find the source of "[hls @ 0x313b6f0]failed to rename file .tmp to " (in an attempt to analyze) [20:52:31 CEST] <Mavrik> It's... not that simple [20:52:32 CEST] <maqr> Mavrik: what's the smallest possible bufsize? [20:53:35 CEST] <Mavrik> Hrmf, no idea. [20:53:59 CEST] <Mavrik> What is "HLS" validation you're talking about? [20:54:12 CEST] <maqr> Mavrik: apple's mediastreamvalidator [20:54:17 CEST] <Mavrik> Since having an accurate bufsize will increase quality [20:54:23 CEST] <Mavrik> So I dunno what kind of dumb software are you using [20:54:24 CEST] <maqr> hmm [20:54:27 CEST] <maqr> apple's dumb software [20:54:46 CEST] <maqr> Mavrik: does greater bufsize increase quality? [20:54:49 CEST] <maqr> i'm not sure what you mean by 'accurate' [20:54:57 CEST] <maqr> or maybe the right quesiton is, how can i find the accurate bufsize for my stream? [20:55:08 CEST] <PieterMX> what is the framerate? [20:55:13 CEST] <PieterMX> i mean [20:55:14 CEST] <PieterMX> bitrate [20:55:27 CEST] <Mavrik> maqr, uh, was I unclear? [20:55:35 CEST] <Mavrik> maqr, bufsize -> set PLAYER's buffer size [20:55:41 CEST] <maqr> Mavrik: maybe, or i'm dense... [20:55:48 CEST] <Mavrik> set accurately -> set it to value your video player uses [20:55:53 CEST] <Mavrik> that is how much video it'll buffer [20:55:53 CEST] <maqr> OH [20:55:59 CEST] <Mavrik> it can be smaller though :) [20:56:07 CEST] <Mavrik> but if it's larger you can get stuttering [20:56:34 CEST] <Mavrik> maqr, for starters setting it to the same value as maxrate is usually good enough, since pretty much all players will buffer at least a second [20:56:36 CEST] <maqr> Mavrik: i think it's related to hls_time then [20:57:06 CEST] <maqr> err, wait, i guess they're separate [20:57:10 CEST] <PieterMX> which cant be set in ffserver somehow :S [20:58:24 CEST] <PieterMX> error is part of libavformat [20:58:31 CEST] <maqr> Mavrik: https://gist.github.com/anonymous/4440431a27196dabd85e [20:58:49 CEST] <maqr> Mavrik: if i use a bufsize shorter than 1 sec, it seems to reduce some of these segment bandwidth warnings [20:59:04 CEST] <maqr> but that doesn't make sense to me, because i have an HLS time of 10 sec [21:00:02 CEST] <maqr> Mavrik: i've already trimmed off 10% from the target bitrate because of TS overhead [21:00:30 CEST] <maqr> Mavrik: i also don't think it's a coincidence that 24 is the last TS file and that's where the warning is [21:01:36 CEST] <maqr> ya, that last TS file is super small, maybe i should just ignore this warning [21:02:23 CEST] <Mavrik> hmm [21:02:36 CEST] <maqr> the last TS file is under 2 seconds, which i bet is why there's a warning here, because maybe apple's tool is validating wrong [21:03:13 CEST] <maqr> Mavrik: what's also surprising to me though is that apple's tool reports bitrates that are higher than the maxrate [21:03:20 CEST] <maqr> Segment bitrate: Average: 1.53 Mbits/sec, Max: 1.95 Mbits/sec [21:03:21 CEST] <Mavrik> yeah, because it probably is [21:03:26 CEST] <Mavrik> but if your segment is 2 seconds [21:03:34 CEST] <Mavrik> then x264 probably won't even stabilize it's bitrate enough [21:03:40 CEST] <maqr> and that's with me setting 1350k [21:03:54 CEST] <maqr> Mavrik: i thought maxrate was the max rate... [21:05:01 CEST] <maqr> "Set maximum bitrate tolerance (in bits/s). Requires bufsize to be set. (from INT_MIN to INT_MAX)" [21:05:19 CEST] <maqr> i bet if i cut the bufsize down to half a second, i won't get these warnings [21:05:43 CEST] <maqr> but i don't really understand why [21:05:44 CEST] <maqr> trying it now [21:08:13 CEST] <maqr> Mavrik: yeah, check this out, half-second bufsize, no warnings: https://gist.github.com/anonymous/6f0aa2fd5f7a115899a6 [21:08:58 CEST] <maqr> i don't understand why though... even the segment in question was almost 2 seconds, and the previous bufsize was 1 second [21:09:12 CEST] <maqr> maybe it couldn't fill a whole second buffer and somehow that made the rate go over [21:10:13 CEST] <maqr> i bet that's what's happening [21:10:48 CEST] <Mavrik> As I said, your segments are probably too small for reliable bitrate setting. [21:11:18 CEST] <maqr> yeah, i think there's always going to be a warning on the last one [21:13:12 CEST] <maqr> i think i'm understanding this now :) [21:14:51 CEST] <maqr> Mavrik: the only thing that's still really unclear to me is why the 264k average bitrate is farther from the target bitrate than the others [21:16:45 CEST] <Mavrik> the difference between I-frame and other P/B frames is more pronounced at lower bitrates [21:17:13 CEST] <maqr> Mavrik: is there a setting i should tweak to deal with that or just lower my bitrate until i get closer to the target? [21:17:13 CEST] <Mavrik> and x264 doesn't have enough time to compensate [21:17:37 CEST] <Mavrik> Honestly you should probably encode full video to each quality and slice it afterwards [21:17:42 CEST] <Mavrik> or do it directly [21:17:50 CEST] <Mavrik> not transcode individual segments [21:18:06 CEST] <maqr> wait, i thought that's what it was doing [21:18:14 CEST] <maqr> that must be what it's doing [21:18:43 CEST] <maqr> it's the built-in HLS stuff that's splitting out to multiple TS files, i'm not actually sure how it works [21:19:21 CEST] <maqr> "Output #1, hls, to 'ts/264k-400x300.m3u8'", i think it should be doing the transcode first then splitting it though, it knows about the whole m3u8 upfront [21:19:51 CEST] <kepstin-laptop> maqr, you're splitting based on time, and using a vbv buffer of ~1s? [21:20:12 CEST] <kepstin-laptop> if so, the bitrate for any particular segment will be slightly more or less than the average, nothing really wrong with that. [21:20:23 CEST] <maqr> kepstin-laptop: yeah [21:20:27 CEST] <maqr> hmm [21:20:38 CEST] <maqr> kepstin-laptop: for a particular segment yeah, but for the whole stream it should be close to the target [21:20:59 CEST] <kepstin-laptop> yeah. [21:21:03 CEST] <Mavrik> maqr, hrmf, you're right, I overlooked the cmd line :) [21:21:28 CEST] <maqr> i'm trying again with a bufsize of 10*bitrate, because my hls_time is 10 [21:21:54 CEST] <maqr> i would think that would make each TS file about the right size [21:22:03 CEST] <maqr> *about the right rate [21:22:56 CEST] <kepstin-laptop> that basically means that you're expecting the player to buffer 10*bitrate amount of stuff before starting to play the video, and can sustain a download speed of bitrate. [21:23:10 CEST] <kepstin-laptop> (modulo overheads and whatnot) [21:23:33 CEST] <maqr> kepstin-laptop: yeah, that's probably not what i want in production, but i'd think it would at least get all the files to the right rate [21:23:38 CEST] <maqr> hrm, and it doesn't [21:23:45 CEST] <kepstin-laptop> that will probably give you more variation [21:23:59 CEST] <maqr> it does, yeah [21:24:01 CEST] <kepstin-laptop> the bigger the vbv buffer, the more the encoder can move bits around between frames [21:24:18 CEST] <kepstin-laptop> which might move them between segments if you're splitting only based on time. [21:24:25 CEST] <kepstin-laptop> that's expected and normal. [21:24:31 CEST] <maqr> kepstin-laptop: 1 second seems like it should be a reasonable amount then, given that my TS target sizes are 10s (even though they're actually all over the place) [21:25:16 CEST] <kepstin-laptop> target duration, not target size :) [21:25:40 CEST] <maqr> heh, sorry, the terminology is hard :) [21:26:12 CEST] <kepstin-laptop> since x264 doesn't know about where the boundaries of the segments are, the vbv setting is allowing it to borrow bits from previous or next segments. [21:26:33 CEST] <kepstin-laptop> either way, as long as the player maintains a download speed of at least bitrate+overheads, it'll be fine. [21:26:51 CEST] <maqr> kepstin-laptop: that part makes sense, yeah, but how does it decide where to split the TS files? they mostly seem to be between 6 and 12 seconds [21:27:16 CEST] <kepstin-laptop> maqr, hmm. it might be trying to split them to start on a keyframe? [21:27:31 CEST] <maqr> how often are keyframes usually? [21:27:49 CEST] <kepstin-laptop> how ever often you set them to, and more often if a scene change is detected. [21:27:57 CEST] <maqr> oh, right, that makes sense [21:28:51 CEST] <kepstin-laptop> i dunno if that's what the splitter is actually doing tho, i haven't looked closely at it [21:29:44 CEST] <maqr> also, does it make sense to set -b:v to the same as maxrate? [21:30:00 CEST] <maqr> in my head that makes sense, because i'm thinking of -b:v as the target bitrate and i don't want to exceed the maxrate [21:30:12 CEST] <maqr> but maybe there's a pitfall there [21:30:14 CEST] <kepstin-laptop> maqr, yes, I think that makes sense. [21:30:45 CEST] <kepstin-laptop> (there's some special cases where you might want maxrate to be higher - but that's mostly limited to authoring to optical disks) [21:32:37 CEST] <kepstin-laptop> (e.g. "to fit on the disk, I have to encode to 2mbit average, and the disk drive can read at max 5mbit/s") [21:32:58 CEST] <maqr> oh, that make sense [21:33:18 CEST] <maqr> kepstin-laptop: i'm running my transcodes again with more sensible settings, i'll post the output in a minute when it's done [21:36:27 CEST] <maqr> kepstin-laptop: check this out, it's damn close https://gist.github.com/anonymous/e1ee814cda77da4c0466 [21:36:58 CEST] <maqr> for the 1500k video + 128k audio target, i'm averaging 1.69mbit instead of 1.63mbit, so i think that's good [21:37:12 CEST] <maqr> and i can ignore the warning on the last TS file because if it was padded out to the full 10 seconds, i'm sure that warning wouldn't be there [21:37:20 CEST] <maqr> same with the 728k stream [21:37:31 CEST] <maqr> the 264k one though... i need to adjust something to bring that down [21:37:51 CEST] <maqr> "Segment bitrate: Average: 314.38 kbits/sec (exceeds target by 19.08 %), Max: 369.39 kbits/sec (exceeds target by 39.92 %)" [21:38:12 CEST] <maqr> i could take 20% off the target bitrate, but that seems a bit brutish... i'm guessing there's some other config setting i can tweak to get closer to the target? [21:38:54 CEST] <kepstin-laptop> hard to say. your bitrate is so low that a fairly small difference is a big percentage. [21:39:20 CEST] <maqr> yeah [21:39:38 CEST] <maqr> but it seems like it's consistently higher across the whole stream [21:40:04 CEST] <maqr> if i reduce it to 160k instead of 200k, i bet it's going to be totally fine, but i don't really understand why that's a fix [21:40:56 CEST] <kepstin-laptop> your bufsize is 1500k, and each segment is only 250kish [21:41:22 CEST] <kepstin-laptop> so the encoder basically expects the decoder to be holding 6 segments in buffer at a time, hmm. [21:41:26 CEST] <maqr> kepstin-laptop: bufsize is 200k for the 200k stream [21:41:35 CEST] <maqr> "-s 400x300 -c:v libx264 -preset:v medium -level:v 3.0 -profile:v baseline -b:v 200k -maxrate 200k -bufsize 200k -framerate 12 -b:a 64k -ar 44100 -ac 2 -c:a libfdk_aac -profile:a aac_he_v2 -hls_time 10 -hls_list_size 0 ts/264k-400x300.m3u8 " [21:41:54 CEST] <kepstin-laptop> oh, hmm. [21:42:11 CEST] <kepstin-laptop> you're only getting the warnings on the small stream, i didn't realize there were multiples there. [21:42:21 CEST] <maqr> ah, yeah, sorry if i was unclear [21:42:45 CEST] <maqr> there's a 64k audio only stream, a 200k video stream, a 600k video stream, and a 1500k video stream [21:42:56 CEST] <maqr> the 200k video also has 64k audio [21:43:08 CEST] <maqr> the 600k and 1500k videos have 128k audio [21:43:36 CEST] <kepstin-laptop> hmm. encoding to 200k bitrate, there's ~50k overhead, I'd expect that to work. [21:44:03 CEST] <kepstin-laptop> it's averaging 314 over the entire stream rather than the desired 264. [21:44:26 CEST] <maqr> kepstin-laptop: yeah, and the 64k-audio-only stream seems fine, so i think it's not having a problem with the audio side [21:44:30 CEST] <maqr> i think it's a problem on the video side [21:45:26 CEST] <kepstin-laptop> oh, there's audio and video in the same stream? [21:45:50 CEST] <kepstin-laptop> 200 video + 64 audio + 56 overhead = 314 [21:45:54 CEST] <kepstin-laptop> there's the math. [21:46:05 CEST] <kepstin-laptop> well, not exactly, but it's close [21:46:41 CEST] <maqr> kepstin-laptop: i think the overhead is broken out separately though? [21:46:48 CEST] <maqr> ohhh, i see what you mean [21:47:09 CEST] <kepstin-laptop> the "Average segment structural overhead: 53.13 kbits/sec (16.90 %)" means that you only have 210kbit/s total to use for audio+video if you want the final file size to hit the target [21:47:32 CEST] <maqr> kepstin-laptop: so should i reduce my target rates by the size of the overhead? [21:48:02 CEST] <maqr> even for the other streams, i bet that would get me closer to the target rates [21:48:38 CEST] <maqr> so i can take 10% off the 1500k stream, 12% off the 600k stream, and 17% off the 200k stream, and i should be good? [21:49:17 CEST] <kepstin-laptop> maqr, something around that should be good, yeah [21:49:25 CEST] <kepstin-laptop> man, ts has a lot of overhead :/ [21:49:54 CEST] <maqr> yeah, i read that's because it was meant for dvd players and stuff [21:50:03 CEST] <maqr> and people are petitioning apple to let HLS use something besides TS [21:52:19 CEST] <kepstin-laptop> is your plan to have seamless switching between different bitrates in the player? [21:52:24 CEST] <maqr> kepstin-laptop: yeah [21:52:27 CEST] <kepstin-laptop> because right now that won't work [21:52:46 CEST] <maqr> kepstin-laptop: why? they're all the same audio samplerate [21:53:00 CEST] <maqr> which i thought is what made that seemless, but i might be missing something [21:53:02 CEST] <kepstin-laptop> to do that, i think (haven't confirmed this) that you need the video keyframes in matching positions in each video stream. [21:53:20 CEST] <maqr> hmm [21:53:24 CEST] <kepstin-laptop> right now, x264 is independently deciding on the position in each stream. [21:53:45 CEST] <maqr> kepstin-laptop: i've read a lot about this and haven't stumbled on that gem yet... any idea where you saw that? [21:54:20 CEST] <maqr> kepstin-laptop: i guess i could just set a keyframe interval somehow, and because they're all the same length, that might be fine? [21:54:29 CEST] <kepstin-laptop> i think it depends on the player; but if you're switching quality by downloading e.g. segment 10 at one quality then segment 11 at a different quality, and the segment times don't match exactly, there'll be a jump. [21:54:59 CEST] <maqr> kepstin-laptop: i thought the iOS player would take care of that for me, but that's one of the things i need to test today, so we'll find out :) [21:55:02 CEST] <kepstin-laptop> maqr, you'd have to set a keyframe interval, and disable x264's automatic scene change keyframes. [21:55:17 CEST] <maqr> kepstin-laptop: any idea what parameters those are so i'm looking up the right thing? [21:55:17 CEST] <kepstin-laptop> be worth testing to see; i'm not sure either way. haven't done this myself :) [21:56:34 CEST] <maqr> kepstin-laptop: btw, reducing by the overhead % on each stream seemd to work well: https://gist.github.com/anonymous/842444bdd9da2bb833a0 [21:56:37 CEST] <kepstin-laptop> I think setting the ffmpeg -g option (to e.g. 10*framerate) and the "no-scenecut" x264 option (via -x264opts) should do it. [21:57:05 CEST] <maqr> ok, thanks, i'll try that if it doesn't switch cleanly [21:57:24 CEST] <kepstin-laptop> actually, the "-sc_threshold 0" ffmpeg option might work there. [21:57:40 CEST] <maqr> kepstin-laptop: oh, but i bet only the video would jump, because there's also an audio-only stream, and there was a note about how sample rates need to be the same to prevent an audible skip when changing quality levels [21:57:57 CEST] <maqr> and the aac streams are still the same, no keyframes there (afaik, anyway) [21:58:05 CEST] <kepstin-laptop> maqr, right, but if the hls segments are being cut based on video keyframes... [21:58:54 CEST] <maqr> oh, right [21:58:54 CEST] <maqr> hmm [21:59:16 CEST] <maqr> figuring out how to test this is a bit tricky too [21:59:16 CEST] <maqr> heh [22:00:37 CEST] <kepstin-laptop> yeah, since you're dealing with software you don't really control on a locked down device :/ [22:00:47 CEST] <maqr> yeah, super frustrating [22:00:47 CEST] <maqr> heh [22:01:28 CEST] <kepstin-laptop> if you have it connected to wifi on a router running a full linux like openwrt, you might be able to play with the wifi router to apply rate limiting on the connection. [22:01:59 CEST] <maqr> that's actually a great suggestion, i have dd-wrt on my router [22:02:13 CEST] <maqr> i was going to try throttling with Charles Proxy, but i think that's a different kind of throttling that involves buffering somehow [22:02:17 CEST] Action: kepstin-laptop did that on his open "guest" network at home; and kind of pities the people who get barely faster than dialup speeds when they try to use it. [22:02:36 CEST] <maqr> heh [22:03:21 CEST] <kepstin-laptop> you'd want to use a separate router for this testing that's not actually running your real internet connection tho :) [22:03:23 CEST] <maqr> kepstin-laptop: for the m3u8, they say bandwidth is "The value is a decimal-integer of bits per second. It must be an upper bound of the overall bitrate of each media file, calculated to include container overhead, that appears or will appear in the playlist." [22:03:52 CEST] <maqr> so for the 264k one i have BANDWIDTH=264000 [22:04:22 CEST] <kepstin-laptop> hmm, fun, an upper bound that applies to each segment individually :/ [22:04:23 CEST] <maqr> i wonder if "overall bitrate, calculated to include container overhead" means that i'm ok with "Segment bitrate: Average: 278.11 kbits/sec, Max: 321.02 kbits/sec (exceeds target by 21.60 %)" [22:04:35 CEST] <maqr> well, it says "overall bitrate", so maybe that means average... [22:04:40 CEST] <maqr> maybe it's an upperbound on an average [22:04:41 CEST] <maqr> because apple [22:05:00 CEST] <maqr> that's from https://developer.apple.com/library/ios/technotes/tn2288/_index.html if you're curious [22:06:02 CEST] <maqr> i think the wording is really ambiguous [22:06:11 CEST] <kepstin-laptop> "calculated to include container overhead" means that you have to calculate the container overhead in when picking media bitrates :) [22:06:26 CEST] <maqr> the RFC draft is a bit clearer: "It MUST be an upper bound of the overall bitrate of each media file, calculated to include container overhead, that appears or will appear in the Playlist. [22:06:34 CEST] <maqr> well, i'm way off then [22:06:56 CEST] <maqr> kepstin-laptop: it sounds like i actually need ffmpeg to be 100% sure it never goes over the max bitrate? somehow? [22:07:11 CEST] <kepstin-laptop> i'd expect in most cases, an individual segment being over that amount will actually work... as long as the connection speed is slightly higher than the requested stream bitrate [22:07:27 CEST] <maqr> yeah, quite likely [22:07:30 CEST] <kepstin-laptop> which is why ffmpeg is using a 10% tolerance by default [22:07:45 CEST] <maqr> kepstin-laptop: where? [22:07:54 CEST] <kepstin-laptop> "WARNING: Media segment bandwidth exceeded target playlist bandwidth by 18 % (312.10 kbits/sec vs.264.00 kbits/sec, limit is 10 %)." [22:08:04 CEST] <maqr> oh no, that's apple's validator tool output, not ffmpeg [22:08:04 CEST] <kepstin-laptop> it doesn't show that message if the difference is <10% [22:08:13 CEST] <kepstin-laptop> oh, interesting. [22:08:28 CEST] <maqr> i got that by doing "mediastreamvalidator http://localhost:8090/hls.m3u8" [22:08:38 CEST] <kepstin-laptop> so apple's validator tool is basically saying that this isn't really a "MUST" ;) [22:08:52 CEST] <maqr> kepstin-laptop: here's what i'm actually doing https://gist.github.com/anonymous/3ca617420e5c3b1abd4a [22:08:59 CEST] <maqr> yeah, it seems that way [22:09:18 CEST] <maqr> but i'm still having a hard time understanding why i can't force ffmpeg to honor a max rate [22:09:40 CEST] <maqr> maybe i could if i did two-passes? [22:10:08 CEST] <kepstin-laptop> but yeah, if your average overall stream bitrate (including overhead, which you have to calculate yourself) is the m3u8 bitrate, and no segment is >10% above the m3u8 bitrate, you're fine [22:10:26 CEST] <kepstin-laptop> so just include the overhead when calculating video and audio bitrate and you're set for that. [22:11:02 CEST] <kepstin-laptop> dunno if two-pass would help here - i dunno how it interacts with using a vbv buffer. [22:11:16 CEST] <maqr> yeah, this is probably good enough [22:11:28 CEST] <kepstin-laptop> in normal mode, x264 two-pass works by the first pass calculates a crf, then second pass does a crf encode. [22:11:46 CEST] <maqr> kepstin-laptop: i haven't tried CRF yet, but that could also be an option here... [22:11:51 CEST] <kepstin-laptop> it's designed to *maximize* the bitrate variation, to keep quality constant [22:11:56 CEST] <maqr> though i'm not sure what i'd set the rate factor to in this case [22:11:59 CEST] <kepstin-laptop> the opposite of what you want here [22:12:07 CEST] <maqr> yeah, true [22:12:11 CEST] <kepstin-laptop> you want the bitrate to stay constant, and quality to vary [22:12:24 CEST] <maqr> kepstin-laptop: maybe i should set a minrate too? [22:12:31 CEST] <maqr> would that somehow force it into a more constant mode? [22:12:43 CEST] <kepstin-laptop> nah, that would just make it waste bits [22:13:04 CEST] <kepstin-laptop> only useful if for some reason you have a minimum transmission unit size, i think [22:13:28 CEST] <maqr> yeah, that's what i'd think too [22:13:31 CEST] <kepstin-laptop> i dunno if you need it for some types of satellite or radio broadcasting? [22:13:33 CEST] <kepstin-laptop> maybe? [22:14:16 CEST] <kepstin-laptop> for online video, if it doesn't need the full bitrate then you just saved someone a couple bucks on their cell phone bill, right? :) [22:14:32 CEST] <maqr> kepstin-laptop: yeah, that's what i'd think too, unless setting it makes ffmpeg go more strict with staying at the requested bitrate [22:14:35 CEST] <maqr> i'm testing it now to be sure [22:15:19 CEST] <kepstin-laptop> I'd expect it not to make much difference at that low of a bitrate, since x264 is probably using as much as it can at all times anyways. [22:16:22 CEST] <maqr> kepstin-laptop: yeah, pretty much no difference [22:17:46 CEST] <maqr> kepstin-laptop: ok, i'm gonna go grab lunch, then figure out how this theoretically correct-ish HLS (though not really draft RFC compliant) actually works on a real iOS device [22:17:51 CEST] <maqr> thanks so much for your help, much appreciated [22:19:20 CEST] <maqr> i'll let you know it goes [22:20:33 CEST] <kepstin-laptop> cool, i'd be curious to know [22:33:44 CEST] <rikai> Hm, can anyone tell me why adding subtitles to a video might cause the audio to desync? I can encode this video fine, but the moment i add subs i get weird pauses in the audio... [22:47:23 CEST] <__jack__> rikai: encoding, you mean muxing right ? the process is fast sure ? [22:49:22 CEST] <rikai> I suppose so, yeah. :) [22:51:06 CEST] <__jack__> can you pastebin the line & execution ? [22:52:13 CEST] <SpeakerToMeat> Hello. [22:52:43 CEST] <rikai> Ah, hm... interesting... if i use -c:s copy, it works fine. But if i let ffmpeg do what i want with the subtitle stream it messes things up. [22:52:48 CEST] <rikai> Hold, i'll post both up. [22:53:56 CEST] <SpeakerToMeat> Question, can one mux (remux, copy) ProRes 4444 video using ffmpeg (on linux)? the codec doesn't need to be understood since it's not being recompressed, but the codec id should be set correctly on the quicktime destination file for the track. [23:00:08 CEST] <rikai> Ah no, i was mistaken, still desynced. [23:00:24 CEST] <rikai> __jack__: http://pastie.org/pastes/10121019/text?key=su7aygntwrqkmwllfldlmw [23:08:58 CEST] <rikai> The warning about global headers appears to be irrelevant, because it happens even if i don't copy streams. I can only assume it's tied to the subs somehow. I've actually been trying to figure out how to burn the subs into the video directly to bypass the issue of weird sub desyncs, but i can't seem to get a grasp on that. Actually, if i seek in the video it syncs properly, so it MUST be tied to that... [23:23:10 CEST] <rikai> I also can not seem to burn in subs no matter what i do. Despite "ffmpeg -i punchline-intro2.mkv -vf ass="punchline-intro3.ssa" punchline-intro17.mkv" telling me "Added subtitle file: 'punchline-intro3.ssa' (36 styles, 987 events)" the final output still has no subs at all.. :/ [23:25:14 CEST] <LeoLeo> Hi ffmpeg team, may I ask you about potential bug in latest ffmpeg builds? If I try this movie https://drive.google.com/file/d/0BygbyP4_qEiTWmNUZFprS3MtVjg/view?usp=shari… with mplayer for linux or native ffmpeg player, I receive next error each time I trying to seek forward or back: [mp3float @ 016f4a40]Header missing . Could you please help me, how to avoid this? Is this a bug of ffmpeg? [23:36:00 CEST] <maqr> kepstin-laptop: heh, this does an interesting thing in practice [23:36:51 CEST] <maqr> let me see if i can get logs to explain it [23:39:24 CEST] <maqr> kepstin-laptop: it starts with the first thing in the m3u8 always... then it tries to request the quality level that it thinks is best, but it buffers pretty far ahead [23:40:34 CEST] <maqr> kepstin-laptop: https://gist.github.com/maqr/ffb5800dba3f9e16ba8c you can see the broken pipe where it disconnected, then you can see it alternating between 64k audio only and 246k audio/video [23:40:42 CEST] <maqr> but when it switches, the audio is continuous [23:40:50 CEST] <maqr> i'm going to try between two video quality levels now to see if the video is also continuous [23:41:29 CEST] <kepstin-laptop> huh, be interesting to see. [23:46:30 CEST] <foonix> i have possible ffprobe+hevc prob, i see coded_picture_number=0 for every frame, is this unimplemented for hevc or is that a hevc feature? [23:58:24 CEST] <maqr> kepstin-laptop: ok, so on ios8 at least, what happens is that it switches quality levels really far out, maybe 20 or 30 seconds out, it gets those files, and then it's a seemless transition in both audio and video [23:58:33 CEST] <maqr> kepstin-laptop: it seems to have all the data it needs for the overlapping period and the switch is clean [00:00:00 CEST] --- Thu Apr 30 2015

1 0

ffmpeg-devel.log.20150428
by burek 29 Apr '15

29 Apr '15

[00:24:13 CEST] <cone-328> ffmpeg 03Michael Niedermayer 07master:0e7444f6e400: avcodec/hevc: Remove current_sps [00:55:11 CEST] <BtbN> How do i propperly put a comment on top of a git email, without it beeing part of the commit message? Last time i tried it stripped the message entirely. [00:56:48 CEST] <wm4> BtbN: format-patch --cover-letter [00:56:56 CEST] <wm4> or something close to it [01:16:32 CEST] <BtbN> wm4, looks good, thanks [02:45:16 CEST] <cone-328> ffmpeg 03Vignesh Venkatasubramanian 07master:8ca170a78ab6: libavformat/webm_chunk: Write first audio packet. [03:32:52 CEST] <cone-328> ffmpeg 03Andreas Cadhalpun 07master:464c49155ce7: apedec: set s->samples only when init_frame_decoder succeeded [04:00:09 CEST] <cone-328> ffmpeg 03Vignesh Venkatasubramanian 07master:b410c69fb666: webmdashenc: Fix UTCTiming Element [08:39:26 CEST] <anshul_mahe> ubitux: in ass format isn't \N is treated as newline, or I have to add newline(\n\r) with \N [08:41:04 CEST] <ubitux> \N is for presentation, \r\n is for output [08:48:14 CEST] <ubitux> i wonder how you ended up with such output [08:48:18 CEST] <ubitux> (does it even work?) [08:48:25 CEST] <ubitux> can you share the .ts? [08:48:31 CEST] <anshul_mahe> ubitux:I really dont get what u mean, but is it related to changing code of closed caption, or some more ffmpeg option needed to produce output [08:48:59 CEST] <ubitux> you should have one "Dialogue" per line (physical/file) [08:49:16 CEST] <ubitux> \N is part of the markup; it produces a line return at the rendering [08:49:24 CEST] <ubitux> it's 2 different thing [08:49:26 CEST] <ubitux> +s [08:49:58 CEST] <anshul_mahe> I use this file http://gsocdev.ccextractor.org/~anshul/test_video/Closedcaption_rollup.ts [08:50:18 CEST] <ubitux> currently you have multiple "Dialogue" on the same line (at least on the mailing list) [08:50:31 CEST] <ubitux> so technically, it just won't work at all [08:51:00 CEST] <anshul_mahe> yes, I usually use the rendered output by ass [08:51:34 CEST] <anshul_mahe> how would decoder know that they are going to be rendered or just used as it is [08:52:22 CEST] <ubitux> ? [08:52:30 CEST] <ubitux> that ass can't be rendered correctly [08:52:43 CEST] <ubitux> you produce a broken output [08:52:50 CEST] <ubitux> it can't work [08:53:44 CEST] <anshul_mahe> may be, I am not sure. can you try that ts file and produce srt output. it show things fine. [08:53:46 CEST] <ubitux> http://pastie.org/pastes/10117649/text [08:53:51 CEST] <ubitux> you need to produce something like this [08:53:53 CEST] <ubitux> no [08:53:58 CEST] <ubitux> it's broken, period. [08:54:08 CEST] <ubitux> this generated ass is invalid [08:54:53 CEST] <ubitux> but i wonder how you ended up with this [08:55:00 CEST] <ubitux> the api shouldn't allow you to generate such file [08:55:07 CEST] <ubitux> unless you are abusing it in a very nasty way [08:55:39 CEST] <ubitux> anyway, gtg afk [08:57:09 CEST] <anshul_mahe> ok, I would try, it looks like i have to add \N in between dialogue. at same screen. but at end of screen actual \n\r should be used [08:59:04 CEST] <anshul_mahe> thanks, for help [11:36:24 CEST] <saste> are you aware of any mpeg ts analyzer available on linux? [11:36:46 CEST] <saste> possibly something which can be automated? [12:26:33 CEST] <cone-437> ffmpeg 03Maneesh Gupta 07master:cf234552b83a: OpenCL: Avoid potential buffer overflow in cmdutils_opencl.c [12:29:11 CEST] <kierank> saste: dvbsnoop [12:29:21 CEST] <kierank> lglinskih_: congratulations [12:30:18 CEST] <lglinskih_> kierank: thank you ^_^ [12:30:40 CEST] <saste> kierank, thanks [12:40:51 CEST] <cone-437> ffmpeg 03Maneesh Gupta 07master:91305c60261c: OpenCL: Fix ABI incompatibility issues [12:41:18 CEST] <drwx> hi, could anyone look at this bug? https://trac.ffmpeg.org/ticket/2975 [12:41:26 CEST] <drwx> also, https://trac.ffmpeg.org/ticket/4487 looks like a duplicate [12:45:24 CEST] <wm4> drwx: yeah, that looks somewhat serious [12:47:59 CEST] <drwx> wm4: yeah because of this i had to remux a lot of files that i encoded, never realised why the audio was slightly OOS [12:52:26 CEST] <cone-437> ffmpeg 03Andreas Cadhalpun 07master:699341d647f7: apedec: prevent out of array writes in decode_array_0000 [13:50:46 CEST] <alexvf> hi [13:50:49 CEST] <alexvf> how is a player using libavformat supposed to know the teletext page of a teletext subtitle stream? does it need to parse codec extradata or is there a more civilized way? [14:11:22 CEST] <alexvf> which is the right channel to discuss libavformat usage? [16:11:23 CEST] <cone-437> ffmpeg 03Michael Niedermayer 07master:e2877bdf3862: ffmpeg: remove incorrect network deinit [16:59:21 CEST] <HSKW> hello! i need help to watch/record an .ts segment (3seconds durations) with same name every time [17:44:38 CEST] <Daemon404> https://www.dropbox.com/s/rkznx34diznb0lb/watgcc.png?dl=0 <-- lolgcc [17:46:34 CEST] <nevcairiel> so it works thist ime? :p [17:47:01 CEST] <Daemon404> yep! [17:47:04 CEST] <Daemon404> (i think( [17:47:54 CEST] <nevcairiel> building with warnings-as-errors is always risky if new compilers show up =p [17:48:04 CEST] <Daemon404> g 42 [17:48:06 CEST] <iive> hey, wasn't there a gcc x.y.0 version that didn't miscompile ffmpeg? [17:48:27 CEST] <nevcairiel> 4.8 i think [17:48:30 CEST] <Daemon404> thats why they skipped .0 this time. [17:48:32 CEST] <Daemon404> ;) [17:48:41 CEST] <nevcairiel> their tarball is called 5.1.0 [17:48:53 CEST] <HSKW> hello! i need help to watch/record an .ts segment (3seconds durations) with same name every time [17:49:10 CEST] <Daemon404> nevcairiel, damn. [17:49:50 CEST] <nevcairiel> whats with everyone and inventing ways to quickly inflate versino numbers [17:51:27 CEST] <BtbN> inflating version numbers do help a bit to motivate users to upgrade faster. [17:51:43 CEST] <Daemon404> users dont upgrade gcc [17:51:46 CEST] <Daemon404> diestros do [17:51:48 CEST] <Daemon404> distros* [17:51:58 CEST] <BtbN> Yeah, for gcc it's quite pointless [17:52:26 CEST] <nevcairiel> chrome and firefox mostly auto-update, so its equally pointless, well, having any kind of version number kinda is in that case [17:55:52 CEST] <jamrial> no, it was 4.9.0 [17:56:07 CEST] <nevcairiel> didnt 4.9 have the flac trouble [17:56:11 CEST] <jamrial> yeah [17:56:42 CEST] <jamrial> afaik it was golomb [17:56:53 CEST] <jamrial> flac and other codecs were affected [17:57:09 CEST] <nevcairiel> then 4.9.0 isnt a version that didnt miscompile ffmpeg =p [17:57:39 CEST] <jamrial> oh, haha. misread iive's question :p [17:57:42 CEST] <nevcairiel> i think 4.8.0 fared pretty well, but i dont usually use .0 versions just to be safe [17:58:02 CEST] <Daemon404> i heard LTO in 5.1 may actually not crash gcc [17:58:05 CEST] <jamrial> gcc 5.1.0 passes fate on both linux and mingw-w64 here [17:58:08 CEST] <Daemon404> and, may, stay with me here [17:58:12 CEST] <Daemon404> make the code *actually* faster [17:58:13 CEST] <Daemon404> maybe. [17:58:13 CEST] <nevcairiel> i still didnt bother to setup 5.1 [17:58:34 CEST] <nevcairiel> some early tests with x264 and x265 seemed to indicate gcc 5 produces slower code [17:58:40 CEST] <Daemon404> rofl [18:00:07 CEST] <iive> well, it is known that gcc optimizes for newest cpu and older cpu settings kind of produce worse code. [18:00:39 CEST] <nevcairiel> i think tests were done on a ivy bridge cpu, so hardly the oldest model [18:01:01 CEST] <jamrial> starting with 4.9, mtune=generic (the default) optimizes for latest intel cpus rather than last century's [19:41:32 CEST] <cone-437> ffmpeg 03foo86 07master:191dd219d9c9: avformat/dtsdec: Extend DTS core sync word [19:43:20 CEST] <Daemon404> O_o [20:03:35 CEST] <rcombs> O_o [20:04:58 CEST] <wm4> you're O_o'ing at the DTS commit? [20:05:44 CEST] <rcombs> I was [20:35:47 CEST] <cone-437> ffmpeg 03Michael Niedermayer 07master:5bebd13d2d86: avcodec/vqavideo: Use ff_tlog() for byte and opcode tracing [20:55:45 CEST] <cone-437> ffmpeg 03Andreas Cadhalpun 07master:6621105877ce: nutdec: check for negative frame rate in decode_info_header [22:34:48 CEST] <cone-437> ffmpeg 03Andreas Cadhalpun 07master:3ff1af2b0db7: nutdec: check chapter creation in decode_info_header [22:54:10 CEST] <cone-437> ffmpeg 03Andreas Cadhalpun 07master:361702660d2c: nutdec: fix memleaks on error in nut_read_header [23:23:12 CEST] <cone-437> ffmpeg 03Andreas Cadhalpun 07master:7c24ca1bda2d: nutdec: fix illegal count check in decode_main_header [00:00:00 CEST] --- Wed Apr 29 2015

1 0

ffmpeg.log.20150428
by burek 29 Apr '15

29 Apr '15

[09:53:47 CEST] <termos> I'm having some issues when I get unordered pts frames into my filter graph, the buffer_sink is just filled up waiting for the correct pts frames, and my program can crash. Doesn't happen often, but I want to avoid it crashing. Can I somehow sanely stop the process when the buffer_sink buffer is too large? [10:14:17 CEST] <omnivibe> Question... I have a video player that I wrote in Actionscript/Flash and I'm thinking of porting it to C... looking at ffplay as the basis, is that the most basic starting point for current state of the art- seems like a lot of options that won't be necessary for me? (i.e. I need it to play *fast* and in full screen, take advantage of hw accelleration/gpu/ etc.- but I control the format it will be able to play, and can restrict it to only [10:14:18 CEST] <omnivibe> h264/x264 in various profiles)) [10:14:49 CEST] <omnivibe> Similarly, how does the LGPL license affect that? I guess it means I can compile ffmpeg libraries, but can't actually use ffplay since I won't be sharing the actual player source? [11:16:40 CEST] <dreamon> how can I merge 2 (same format) file to one file -> ffmpeg -i "concat:movie1.mkv|movie2.mkv" -f mpeg -c copy output.mkv -> could not found concat:movie1.mkv|movie2.mkv [11:16:58 CEST] <dreamon> is that syntax right? [11:18:02 CEST] <techtopia> it never works for mkv's for me, you have to transcode to .ts then concat then transcode result back to mkv [11:18:13 CEST] <techtopia> you could just do [11:18:50 CEST] <techtopia> mkvmerge -o output.mkv movie1.mkv + movie2.mkv [11:20:32 CEST] <dreamon> techtopia, Thank you. That makes sense. [11:21:15 CEST] <techtopia> np [11:33:25 CEST] <dreamon> techtopia, I merged both files. length is gotten bigger,but, is only playing the length on first file size instead longer hmm.. [11:43:08 CEST] <techtopia> are both mkv's the same spec [11:43:22 CEST] <techtopia> same resolution / framerate / encoding settings [11:43:44 CEST] <techtopia> it not you might need to re-encode them both with the same settings [12:20:48 CEST] <drwx> ffmpeg -i in.264 out.mkv; mediainfo -f out.mkv | grep Delay [12:20:51 CEST] <drwx> Delay : 83ms [12:20:55 CEST] <drwx> why does it do that? [12:21:00 CEST] <drwx> i've always found it pretty annoying [12:32:13 CEST] <drwx> ok so this is a bug https://trac.ffmpeg.org/ticket/2975 [12:34:22 CEST] <usbpc> Hi everyone. I tried using ffmpeg to convert an avi file (lagarith lossles codec) and uncompressed audio to h.264 using x264. The command I used is: ffmpeg -i "./lagarith.avi" -c:v libx264 -preset slow -crf 21.5 -c:a flac -compression_level 12 ./output.mkv The Problem is that the video has artefacts (I think they are called that) and in the console it repeats Output more bytes than length with numbers in brackets behind. Video: http://yo [12:55:50 CEST] <alexvf> hi [12:56:31 CEST] <alexvf> what is the best way of getting the teletext descriptors from a teletext stream using ffmpeg? [12:57:03 CEST] <alexvf> i cannot find a way apart from reading codec extradata, is it the expected way? [12:58:00 CEST] <alexvf> from what i see, mpegts.c only parses language from teletext descriptors and copies the rest of the descriptors to extradata [13:22:20 CEST] <alexvf> how is a player using libavformat supposed to know the teletext page of a teletext subtitle stream? does it need to parse codec extradata or is there a more civilized way? [14:10:54 CEST] <usbpc> Sorry, my PC crashed. Did someone answer my Question while I was away? [14:11:12 CEST] <quellen> no [14:11:48 CEST] <quellen> what is your question? [14:12:26 CEST] <usbpc> I have to rewrite it, wait a sec- [14:12:36 CEST] <quellen> i have a question too [14:13:51 CEST] <quellen> when ffmpeg converts a TS file (DTT noisy recorded), the audio is out of sync: http://pastebin.com/uuFSQzfh [14:14:44 CEST] <quellen> audio in original TS is ok [14:15:12 CEST] <usbpc> I tried to convert an avi (lagarith lossless codec) to h.264 usin x264. But the output looks like this: https://youtu.be/GQlqTJj6qrI Command I used: ffmpeg -i "./lagarith.avi" -c:v libx264 -preset slow -crf 21.5 -c:a flac -compression_level 12 ./output.mkv [14:16:14 CEST] <quellen> the output seems ok, what is the problem? [14:16:15 CEST] <usbpc> While it is converting the Console outputs: Output more bytes than length a lot. but otherwise its normal. [14:16:26 CEST] <usbpc> quellen: keep watching [14:18:02 CEST] <usbpc> Or watch there: https://www.youtube.com/watch?v=GQlqTJj6qrI&feature=youtu.be&t=145 sometimes something flashes on screen like raw data or something. Hard to describe. [14:18:29 CEST] <quellen> when i convert with ffmpeg i use these options: [14:18:30 CEST] <quellen> ffmpeg -y -i "original.vob" -f avi -r 25 -vcodec libxvid -vtag XVID -maxrate 1800k -b 1500k -qmin 3 -qmax 5 -bufsize 4096 -mbd 2 -bf 2 -trellis 1 -flags +aic -cmp 2 -subcmp 2 -g 300 -acodec copy "output.avi" [14:20:28 CEST] <techtopia> use crf 21 or 22 [14:20:28 CEST] <usbpc> I just saw in the title that debian is experimantal, I'll try the conversion on Windows again. [14:20:32 CEST] <techtopia> not 21.5 [14:20:35 CEST] <techtopia> there is no 21.5 [14:21:29 CEST] <usbpc> techtopia: okay, but why does it work with x264 using MeGUI? [14:22:11 CEST] <techtopia> it shouldnt work i guess it just ignores it and doesn't use crf at all [14:22:15 CEST] <techtopia> you can check with mediainfo [14:22:38 CEST] <quellen> if you need a ffmpeg gui, install winff [14:22:43 CEST] <usbpc> Okay I'll check. [14:23:18 CEST] <techtopia> what type of content is it [14:23:36 CEST] <usbpc> I don't need a GUI, I want to use my old PC as an automated rendering station. [14:23:50 CEST] <usbpc> And that PC is running as a Debian server. [14:26:03 CEST] <usbpc> media info tells me crf=21.5 when using MeGUI [14:26:54 CEST] <usbpc> techtopia: Content is Gaming. [14:27:13 CEST] <usbpc> In this case Metro: Last Light Redux [14:28:38 CEST] <techtopia> you would be better with a higher crf then probably 23 [14:30:03 CEST] <usbpc> Thats possible, but I don't think that that is the reason my video has these glitches [14:31:37 CEST] <juzzlin_> hi, is it possible to automatically make input jpg sizes even as it seems that encoding to x265 fails on odd image sizes? [14:32:46 CEST] <techtopia> usbpc can you do a open the source ts in mediainfo [14:33:01 CEST] <techtopia> select the view mode "text" and paste the text to pastebin [14:33:06 CEST] <techtopia> juzzlin, what do you mean [14:33:30 CEST] <usbpc> source ts? [14:33:38 CEST] <techtopia> oh it was an avi [14:33:41 CEST] <quellen> noone helps me [14:33:53 CEST] <techtopia> "./lagarith.avi" [14:34:05 CEST] <usbpc> okay, wait a sec [14:34:18 CEST] <techtopia> quellen you can get an app called delay cut to fix the sync [14:34:47 CEST] <techtopia> delaycut_1212_exe.zip have a google for it [14:35:26 CEST] <juzzlin_> techtopia: I get "x265 [error]: Picture height must be an integer multiple of the specified chroma subsampling" on certain input image sixes. [14:35:30 CEST] <quellen> nothing for Linux? [14:35:41 CEST] <usbpc> There the tect output: http://pastebin.com/wj7AntdU [14:35:48 CEST] <techtopia> their probably is but not that i know of quellen [14:35:54 CEST] <techtopia> im on windows [14:36:11 CEST] <quellen> i try to launch it with wine [14:36:20 CEST] <quellen> thx [14:38:21 CEST] <quellen> it works. i must extract audio from TS? [14:39:48 CEST] <quellen> what command i must use? ffmpeg -i original.TS output.mp3 ? [14:40:19 CEST] <usbpc> Dosen't work, still a glitch in that Video. But better but I used the wrong video file, I'll try with the right one again [14:40:51 CEST] <techtopia> usbpc try this [14:40:57 CEST] <techtopia> ffmpeg -y -i lagarith.avi -sws_flags spline -sn -s 1920x1080 -c:v libx264 -preset slow -profile:v high -level 4.1 -crf 23 -r 60 -x264opts colormatrix=bt709 -refs 4 -acodec copy output.mkv [14:41:18 CEST] <techtopia> yeah quellen demux it [14:41:46 CEST] <techtopia> ffmpeg -i input.ts -acodec copy -vn output.ac3 [14:41:53 CEST] <techtopia> ffmpeg -i input.ts -acodec copy -vn output.mp3 ** [14:41:55 CEST] <quellen> i think the problem is on video too, not only in audio [14:41:59 CEST] <quellen> ok [14:42:09 CEST] <techtopia> do "ffmpeg -i input.ts" [14:42:15 CEST] <techtopia> then you can see what the audio is [14:42:21 CEST] <techtopia> then demux it to the right extension [14:42:27 CEST] <usbpc> okay its rendering.... [14:42:50 CEST] <quellen> Stream #0:1[0x5fa]: Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, mono, s16p, 112 kb/s [14:42:54 CEST] <quellen> mp2? [14:42:57 CEST] <techtopia> yes [14:43:04 CEST] <techtopia> ffmpeg -i input.ts -acodec copy -vn output.mp2 [14:43:09 CEST] <quellen> ;-) [14:45:07 CEST] <quellen> in delaycut what i must do? only PROCESS? [14:45:28 CEST] <quellen> CRC errors is set to SILENCE [14:46:22 CEST] <quellen> http://pastie.org/10118147 [14:47:15 CEST] <techtopia> :( doesent look good [14:47:25 CEST] <techtopia> not sure whats causing the errors [14:47:37 CEST] <quellen> bad signal of DTT [14:47:39 CEST] <techtopia> i've used it before to fix sync issues with ac3 files and it worked [14:49:21 CEST] <techtopia> hmm is the audio ahead or behind the video? [14:50:30 CEST] <techtopia> either way work out the delay in ms [14:50:43 CEST] <techtopia> generate a silent mp2 that ammount of ms long [14:50:53 CEST] <usbpc> Nope Video still gliched. Command output: http://pastebin.com/Gupd7u3P [14:52:06 CEST] <techtopia> if audio is ahead of video do "copy /b silence.mp2 + sourceaudio.mp2 audiofix.mp2 [14:52:36 CEST] <techtopia> if audio is behind video do "copy /b sourceaudio.mp2 + silence.mp2 audiofix.mp2" [14:52:49 CEST] <techtopia> then mux audiofix.mp2 with encoded video [14:53:58 CEST] <techtopia> you sure the errors are not present in the source usbpc [14:54:35 CEST] <usbpc> pretty unless copying via ftp created errors. [14:54:36 CEST] <quellen> i will try [14:55:39 CEST] <usbpc> I'll try rendering it with ffmpeg locally on windows to see if it is a debian bug. [15:01:19 CEST] <quellen> techtopia, what do you mean for "audio ahead or behind the video"? [15:01:39 CEST] <quellen> Stream #0:0[0x4f6]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv), 480x576 [SAR 8:5 DAR 4:3], max. 15000 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc [15:01:39 CEST] <quellen> Stream #0:1[0x5fa]: Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, mono, s16p, 112 kb/s [15:03:31 CEST] <usbpc> Video is fine when rendering local on windows. [15:04:18 CEST] <quellen> how can generate a silent mp2? [15:05:21 CEST] <quellen> see you later [15:15:28 CEST] <usbpc> techtopia: I downloaded a newer version for debian but still gliched video. Command: ./ffmpeg-2.6.2-64bit-static/ffmpeg -y -i lagarith.avi -sws_flags spline -sn -s 1920x1080 -c:v libx264 -preset slow -profile:v high -level 4.1 -crf 23 -r 60 -x264opts colormatrix=bt709 -refs 4 -acodec copy output.mkv Output: http://pastebin.com/8vfBWHHC [15:28:45 CEST] <usbpc> techtopia: Okay it was the source file. It got corupted somehow while transfering to my old PC via FTP. Strange but thanks a lot! [15:49:39 CEST] <sybariten> oi videoers [15:49:44 CEST] <sybariten> failing with the multiple image input thing [15:50:37 CEST] <sybariten> images are called BLACK_PACKSHOT_00001.tif up until BLACK_PACKSHOT_00242.tif .... whats the ffmpeg syntax then? [15:56:16 CEST] <usbpc> try this: ffmpeg -framerate 1/5 -i BLACK_PACKSHOT_%03d.png -c:v libx264 -r 30 -pix_fmt yuv420p out.mp4 [15:58:32 CEST] <usbpc> sybariten, but I have no idea if it works, just looked it up on google. [16:06:27 CEST] <sybariten> yeah sorry should have gotten back here earlier.... in the case of for instance five numbers, like in this case... it's %05d [16:14:13 CEST] <usbpc> sybariten, so you got it? [16:21:01 CEST] <sybariten> yeah [16:21:03 CEST] <sybariten> thanks! [16:48:21 CEST] <albru123> Hello, I'm trying to encode an AviSynth script (mkv video with subtitles), but I keep getting "Could not write header for output file #0" error. Can someone tell me what's wrong? Here are some basic information: http://hastebin.com/lucofayuze.tex [16:55:29 CEST] <_deivor> Hi.. I'm having some static audio issues while using fdk-aac and this static is not there if I use mp2 (mp2 for rtp, fdk-aac for rtmp).. is this a known issue? [16:55:46 CEST] <_deivor> as in constant static interference, [16:57:03 CEST] <HSKW> hello! [16:57:19 CEST] <HSKW> i need help to watch an continuous .ts segment file [16:57:52 CEST] <HSKW> i dont know how to watch/record an .ts segment (3seconds durations) with same name every time [17:05:15 CEST] <_deivor> why is it the same name? [17:05:44 CEST] <_deivor> you could set up a watchdog to restart the process when the file is modified [17:06:36 CEST] <HSKW> i dont know why the .ts use the same name every segment [17:07:22 CEST] <HSKW> i must concatenate every segment with same name... like an live streaming with HLS system but use every segment the same .ts name [17:07:45 CEST] <HSKW> if i play with ffplay i can watch only few seconds and than stops... [17:21:02 CEST] <_deivor> HSKW can you try another player? [17:21:31 CEST] <usbpc> albru123, It seems that you use an Invalid argument. I don't know ffmpeg nor AviSynth very well but you should check your command and AviSynth script for typos. [17:21:56 CEST] <_deivor> I don't know how a player handles or even should handle that; but what you CAN do is set up a watchdog that immediately moves that file to another, when it is created [17:21:57 CEST] <HSKW> _deivor: i cant use another player... [17:22:28 CEST] <HSKW> its use HLS, only ffdshow can help me [17:22:36 CEST] <_deivor> mpv/vlc play hls [17:22:51 CEST] <HSKW> yes vlc same result, only play 3 seconds [17:23:15 CEST] <HSKW> regards: i dont have the m3u8, only the segmented .ts url [17:23:59 CEST] <HSKW> i must reload/concatenate the same .ts segment every 4 seconds [17:25:35 CEST] <_deivor> you are essentially destroying the video.. the player most likely won't handle that gracefully [17:25:49 CEST] <_deivor> running on linux or windows? [17:26:14 CEST] <HSKW> windows at the moment [17:33:12 CEST] <albru123> usbpc: It was maybe due to AviSynth and MP4 combination [17:33:14 CEST] <HSKW> _deivor: did you want some example .ts file? [17:33:26 CEST] <albru123> It rendered well with MKV, but the sound was disroted [17:35:16 CEST] <usbpc> albru123, In the original video the sound is okay? [17:35:51 CEST] <albru123> usbpc: Yep, I'm wondering how it gets the sound anyway, if it's processed by avisynth [17:36:01 CEST] <albru123> maybe some DirectShow bug [17:36:05 CEST] <albru123> or wrong settings [17:36:41 CEST] <usbpc> you could just extract the audio from the original and then into your output file. [17:37:58 CEST] <albru123> yeah, I was googling how to mux it directly in ffmpeg, but I don't know if that's possible [17:38:43 CEST] <usbpc> It is I know that, but I don't know the command. I'll try to find out. [17:43:47 CEST] <usbpc> albru123, try this command: ffmpeg -i C:\Users\User\Desktop\input.mkv -i output.mkv -c copy -map 0:a:0 -map 1:v:0 output1.mkv [17:44:47 CEST] <albru123> usbpc: What's output.mkv? o.O [17:45:02 CEST] <usbpc> your previous output file? [17:45:06 CEST] <albru123> oh [17:45:27 CEST] <albru123> didn't thought about this solution [17:45:38 CEST] <usbpc> no problem [17:46:14 CEST] <usbpc> I hope it works. [17:49:35 CEST] <albru123> usbpc: Yes, it worked well :) [17:49:55 CEST] <usbpc> Nice. [17:50:07 CEST] <albru123> But still useless, as previous step desynchronized audio and video for some reason xD [17:51:17 CEST] <usbpc> strange [17:51:22 CEST] <albru123> usbpc: Maybe the source video uses variable framerate, I'm not quite sure. I know it was mentioned somewhere, to set constant framerate when loading video in AviSynth, but I skipped that [17:53:39 CEST] <usbpc> albru123, I don't know [17:54:14 CEST] <usbpc> albru123, I don't really know ffmpeg. But I know how to use google. ;) [17:56:13 CEST] <albru123> usbpc: Hmm, basically I'm just trying to hardsub subtitles to a video, I'll try google as you do :D [18:03:16 CEST] <usbpc> albru123, maybe try without avisynth: ffmpeg -i C:\Users\User\Desktop\input.mkv -vf "ass=C:\Users\User\Desktop\Subs\Makai\input.ass" output.mkv [18:03:53 CEST] <kepstin-laptop> you probably want to use the 'subtitles' filter rather than 'ass' filter so it can pull fonts from the file. [18:04:43 CEST] <usbpc> kepstin-laptop, could you post the command for that? [18:04:55 CEST] <c_14> https://trac.ffmpeg.org/wiki/HowToBurnSubtitlesIntoVideo [18:05:03 CEST] <kepstin-laptop> exactly the same as what you just posted, but replace "ass" with "subtitles"... [18:05:05 CEST] <albru123> usbpc: You need to setup fontconfig for that, and that's really annoying [18:06:21 CEST] <HSKW> ffmpeg or ffplay to loop play an url? [18:06:30 CEST] <usbpc> kepstin-laptop, and then it just works? [18:06:56 CEST] <albru123> I also need perfect ass effects support, that's why I'm doing this thing all the time [18:07:01 CEST] <kepstin-laptop> usbpc, it should, yeah. [18:07:19 CEST] <kepstin-laptop> albru123, ah, so you're using vsfilter in preference to libass then :/ [18:07:56 CEST] <albru123> kepstin-laptop: Exactly, but for some reason that avisynth script desynchronized audio and video :P [18:08:12 CEST] <albru123> and disroted sound, weird [18:08:18 CEST] Action: kepstin-laptop doesn't have a windows box, so he hasn't ever seem most subtitles rendered with anything other than libass anyways. [18:10:11 CEST] <albru123> I used Handbrake, but it doesn't support those ass effects very well [18:13:43 CEST] <usbpc> albru123, could you just first burn the subtitles in the video without sound and then add sound? [18:14:19 CEST] <albru123> usbpc: Maybe, I'll try it, but if it changes the framerate somehow, it is worthless [18:14:33 CEST] <usbpc> albru123, true [18:14:56 CEST] <albru123> kepstin-laptop: Tried it with subtitles filter, I know why I haven't used it. It just kept throwing "Past duration 0.XXXXX too large" [18:20:39 CEST] <albru123> Trying with ConvertFPS=true, I hope it'll work ... [18:20:53 CEST] <HSKW> how to loop a video on ffplay? [18:29:16 CEST] <albru123> Looks like it works fine now and it also copied the audio stream even without setting -c:a copy [18:30:07 CEST] <usbpc> albru123, thats nice. [18:30:30 CEST] <albru123> now just pack it inside MP4 somehow [18:31:55 CEST] <usbpc> albru123, i think you need to convert the audio for that [18:32:50 CEST] <albru123> usbpc: Probably ... gotta get libfdk_aac to work first [18:33:47 CEST] <albru123> In case the stream copy really won't work [18:34:28 CEST] <albru123> Oh, it didn't copied the stream, it encoded it as vorbis [18:34:56 CEST] <usbpc> So it worked? [18:36:08 CEST] <usbpc> Yea mp4 only allows following audio codecs: MPEG-4 Part 3, AAC, MP3, MP2, MP1 [18:37:06 CEST] <albru123> usbpc: Yeah probably, it's still encoding [18:37:41 CEST] <usbpc> if not just convert mannually [18:38:56 CEST] <albru123> Yeah, I just don't won't to do it, because the source audio track is perfectly fine :/ [18:42:37 CEST] <usbpc> albru123, yea it seems to be raw wav or something like that (pcm_f32le) [18:43:14 CEST] <usbpc> but why do you need a mp4 your source is mkv too? [18:43:29 CEST] <albru123> usbpc: I'm pretty sure it's aac, but AviSynth is doing strange things with it maybe [18:44:17 CEST] <usbpc> maybe [18:44:43 CEST] <usbpc> we'll see when its done encoding. [18:44:49 CEST] <albru123> usbpc: It's not really neccessary, but it's some kind of convention that our group uses. If it's not easily possible to make mp4 from it, I'll just leave it as it is [22:47:12 CEST] <penthief> Where do I look when I see "minor_code:4 major_code:130." [22:48:37 CEST] <penthief> I'm interested to know where the error codes are documented, but this is the complete error: [x11grab @ 0x1db35e0] Cannot get the image data event_error: response_type:0 error_code:8 sequence:9 resource_id:50331648 minor_code:4 major_code:130. [22:49:06 CEST] <penthief> Then it cored: Process screencapture segmentation fault (core dumped) [22:49:53 CEST] <penthief> ffmpeg version 2.5.4-1 built with gcc 4.9.2 (Ubuntu 4.9.2-10ubuntu7) [22:51:08 CEST] <iive> penthief: i suspect that these are x11 codes [22:51:17 CEST] <penthief> oh, sorry [22:51:25 CEST] <penthief> and thanks [22:52:15 CEST] <penthief> Is the crash acceptable, what with x mibehaving? [22:53:06 CEST] <iive> depending on your definition of crash. exiting with error is OK, segfault is not. [22:53:59 CEST] <penthief> I'll look at raising a bug then, 3 segfaults. [22:55:01 CEST] <iive> btw, these numbers are decimals, in case you want to look them up. [22:55:02 CEST] <penthief> is there a doc for raising bugs? [22:55:25 CEST] <penthief> decimals? [22:55:36 CEST] <iive> the minor/major_code [22:56:06 CEST] Action: penthief notices the segfault occurs when changing window. [22:56:39 CEST] <iive> sometimes hexidecimal numbers are used. they should be prefixed with 0x, but it is not automatic. [22:57:00 CEST] <penthief> gotcha [23:04:12 CEST] <penthief> (the screen resolution was wrong, hence the error, but the subsequent segfault probably wants raising, not sure...) [23:04:43 CEST] <penthief> "incorrect resolution with x11grab causes a segfault when switching windows" [00:00:00 CEST] --- Wed Apr 29 2015

1 0

ffmpeg-devel.log.20150427
by burek 28 Apr '15

28 Apr '15

[02:27:44 CEST] <cone-408> ffmpeg 03ZhouXiaoyong 07master:0ace686ae854: avcodec: optimize mathops for Loongson-3 v1 [03:43:06 CEST] <ScHlAuChi> hi, does this channel cover ffmpg2 too? [03:57:40 CEST] <ScHlAuChi> So does this channel cover ffmpeg2 development too? [03:59:20 CEST] <jamrial> unless you're talking about ffmpeg 2.x releases, i'm fairly sure no [03:59:55 CEST] <ScHlAuChi> nope i do mean ffmpegsource2 - but guess not then :) [04:02:21 CEST] <ScHlAuChi> thx anyway :) [05:06:10 CEST] <prelude2004c> hey can anyone help me with a multiast problem? [05:06:16 CEST] <prelude2004c> i can't seem to figure it out [05:08:28 CEST] <cone-408> ffmpeg 03Michael Niedermayer 07master:46778ab212e2: avformat/aviobuf: Do not resize the buffer in ffio_ensure_seekback() if there is no read function [07:00:44 CEST] <ubitux> wm4: yes& :( [07:17:03 CEST] <fluke_> Hello. I am developing videotoolbox hwaccel for iOS. I want to know where can I find hwaccel architecture documents/slides. [10:22:36 CEST] <ffmpeg-devel285> Hello [10:24:17 CEST] <ffmpeg-devel285> I tried convert HEVC Raw video to MP4 (with codec copy) on ffmpeg [10:25:11 CEST] <ffmpeg-devel285> and convert MP4 to yuv (codec copy) on ffmpeg [10:25:49 CEST] <ffmpeg-devel285> compared yuv file and original raw video [10:26:01 CEST] <ffmpeg-devel285> I found they different [10:26:17 CEST] <ffmpeg-devel285> different hex value [10:27:10 CEST] <ffmpeg-devel285> Is it normal? [10:29:32 CEST] <JEEBsv> uhhh [10:29:37 CEST] <JEEBsv> > to yuv > codec copy [10:29:42 CEST] <JEEBsv> u wot m8t [10:30:10 CEST] <ffmpeg-devel285> I want purely copy stream when convert RAW to MP4 [10:30:22 CEST] <JEEBsv> yes, but see your line from mp4 to yuv :P [10:30:31 CEST] <JEEBsv> anyways, just pastebin your command lines [10:30:36 CEST] <JEEBsv> and link here [10:30:47 CEST] <ffmpeg-devel285> ok plz wait.. [10:35:18 CEST] <ffmpeg-devel285> http://pastebin.com/n9HnhYWt [10:36:35 CEST] <ffmpeg-devel285> My problem is some data change when convert RAW to MP4 [10:41:46 CEST] <BBB> JEEBsv: when he says raw yuv, Im assuming he means annexb hevc [10:42:02 CEST] <nevcairiel> how is that raw yuv [10:42:35 CEST] <nevcairiel> and some data needs to change when you convert raw hevc to mp4, since it uses a different syntax [10:44:16 CEST] <wm4> ffmpeg-devel285: #ffmpeg [10:53:58 CEST] <ffmpeg-devel285> hm.. so, that's normal? [11:21:09 CEST] <mop> hi all, [11:26:42 CEST] <mop> I'm trying to cross compile ffmpeg on ubuntu for wind with mingw , I'm also trying to use https://github.com/dlfcn-win32/dlfcn-win32 but the test seems not works also if I set all flags to specify the includes path, ideas ? [11:27:29 CEST] <nevcairiel> why would you need that for ffmpeg [11:28:25 CEST] <wm4> mop: I wouldn't use dlfcn for anything [11:28:32 CEST] <wm4> I mean [11:28:34 CEST] <wm4> dlfcn-win32 [11:29:14 CEST] <ubitux> fluke_: i don't think you'll find much doc about them but you can look at the already existing implementations [11:29:33 CEST] <ubitux> i'm very much interested in the outcome of this btw [11:29:40 CEST] <mop> I opened a pull request to be able to load filters dynamically so I need it (or an equivalent) to accomplish this task [11:29:53 CEST] <mop> it works fine with linux [11:30:05 CEST] <wm4> you can't load filters dynamically because that'd require use of private APIs [11:30:09 CEST] <wm4> so there's no ABI [11:30:13 CEST] <mop> https://github.com/FFmpeg/FFmpeg/pull/134 [11:30:24 CEST] <wm4> and dlfcn-win32 is not thread-safe [11:30:25 CEST] <nevcairiel> windows has dll loading functions too [11:31:02 CEST] <mop> wm4, my official intent is to be more quick to develop new filters to be proposed/integrated into ffmpeg [11:31:25 CEST] <mop> so I just changed the policy in case of [11:31:32 CEST] <wm4> that's a very bad reason [11:31:37 CEST] <wm4> just use the ffmpeg build system correctly [11:31:52 CEST] <nevcairiel> its not exactly hard to just build avfilter with your new filter integrated, if thats your only reason [11:31:57 CEST] <wm4> I wouldn't mind user-addable filters, but IMO the time is not ready yet [11:32:15 CEST] <ubitux> yes i think the api needs more changes before opening it again [11:32:16 CEST] <wm4> because of API and ABI issues [11:32:34 CEST] <wm4> right, it was once open [11:32:35 CEST] <ubitux> they are more important shortcomings to fix before opening it [11:32:43 CEST] <ubitux> and opening it will prevent fixing them anytime soon [11:33:01 CEST] <ubitux> or at best slow it down by a factor of several years [11:33:31 CEST] <mop> wm4, this is the reason because I changed my original plans to add this feature as "users feature" and move to a more strict developers usage purpose [11:33:39 CEST] Action: wm4 is still dreaming of a light-weight ffmpeg filter framework [11:33:59 CEST] <wm4> mop: what's the problem with developing with the current situation? [11:34:45 CEST] <mop> wm4, I cannot have my own filters as separated repo [11:35:06 CEST] <mop> (for example) [11:35:07 CEST] <wm4> and? [11:35:27 CEST] <wm4> currently there are no user filters, so that's a given [11:35:52 CEST] <mop> I'm developing my own [11:36:00 CEST] <wm4> and dynamic loading doesn't change anything about the API and ABI issues [11:36:07 CEST] <wm4> mop: but you can? [11:36:14 CEST] <mop> with my fix yes [11:36:30 CEST] <wm4> you don't need your "fix" [11:36:58 CEST] <wm4> what's the difference between using a separate repo and a branch in your ffmpeg git checkout? [11:37:04 CEST] <mop> and how can I use my own filter without it ? [11:37:38 CEST] <mop> wm4, the weight and the time to compile it, for example [11:37:57 CEST] <saste> wm4: the general idea is this, you enable a configure flag allowing to export the symbols [11:38:07 CEST] <saste> this would be only enabled explicitly [11:38:28 CEST] <saste> of course in case the internal ABI changes, the plugin filters will become useless [11:38:33 CEST] <wm4> mop: ffmpeg has an incremental build system, so changing your filter doesn't require full recompilation [11:39:28 CEST] <mop> wm4, I already know it, but I strongly prefer have a repo with just file needed by my functionalities and not the whole project [11:39:45 CEST] <mop> is thousands time more simple to read and understand it [11:40:11 CEST] <wm4> uh, you just don't understand how git works [11:40:14 CEST] <mop> in any case, it is a feature turned off as default and just for developers [11:40:22 CEST] <wm4> it's useless [11:40:29 CEST] <wm4> it doesn't make anything easier [11:40:43 CEST] <wm4> and it creates the false impression that libavfilter actually can do plugins [11:41:31 CEST] <mop> it can, with instability of interfaces, but it can [11:42:19 CEST] <wm4> which in turn means it absolutely has no value [11:43:09 CEST] <mop> I respect your opinion but I disagree and I found more simple develop my filter in this way [11:43:50 CEST] <mop> in any case, this will be the steps that you must performs when the interface will be stable enough [11:44:20 CEST] <wm4> I doubt it [11:44:27 CEST] <nevcairiel> just know that you are on your own then [11:44:32 CEST] <wm4> for one, it shouldn't just load such things globally [11:44:57 CEST] <wm4> so the final result will look differently anyway [11:45:33 CEST] <mop> wm4, what you mean with load such things globally ? [11:45:57 CEST] <mop> my patch search for specific symbol and load it as filter, no other things [11:46:14 CEST] <wm4> loading and adding them to a global list [11:47:39 CEST] <mop> ah , so, also if the interface will be stable, third party filters will be loaded in a different manner ? [11:48:54 CEST] <wm4> at least I'd hope so [11:50:53 CEST] <mop> what is the schema that you have in mind ? [11:52:28 CEST] <wm4> plugins should be registered to some kind of context, not globally [11:59:45 CEST] <mop> in this way you will manage filters in different ways depending if are built-in or dynamic, what is the plus in doing it ? [12:00:01 CEST] <fluke_> @ubitux thank you. I have already implemented a videotoolbox hwaccel/decoder follow the vda implementation. I still want to explain the details to people, so I need to figure out more details for myself first. I don't understand the whole architecture much. [12:00:35 CEST] <ubitux> oh it's already done? is this public somewhere? [12:00:41 CEST] <ubitux> ok bye. [12:01:24 CEST] <ubitux> fluke_: 12:00 <@ubitux> oh it's already done? is this public somewhere? [12:01:50 CEST] <ubitux> you can add something like doc/writing-hwaccel.txt or something [12:02:07 CEST] <ubitux> feel free to ask question on the mailing list if you don't get any here [12:02:23 CEST] <ubitux> currently, appart from the doxy and the code itself, you won't have much [12:02:40 CEST] <ubitux> (any [answer]) [12:03:05 CEST] <fluke_> It's not public for now. I work for company. But I plan to open it in monthes. [12:03:58 CEST] <ubitux> other companies are interested in such thing; if you don't release it soon enough, you might face competiting implementations [12:04:07 CEST] <ubitux> and your work might not be upstreamed [12:04:11 CEST] <ubitux> damn. [12:04:28 CEST] <ubitux> 12:03 <@ubitux> other companies are interested in such thing; if you don't release it soon enough, you might face competiting implementations [12:04:30 CEST] <ubitux> 12:03 <-- fluke_ (~fluke(a)209.9.128.43) has quit (Read error: Connection reset by peer) [12:04:32 CEST] <ubitux> 12:04 <@ubitux> and your work might not be upstreamed [12:04:34 CEST] <ubitux> 12:04 <@ubitux> damn. [12:04:36 CEST] <ubitux> fluke_ ^ [12:04:38 CEST] <ubitux> (you should fix your connection...) [12:04:57 CEST] <fluke_> What is hwaccel v2? [12:05:29 CEST] <fluke_> freenode connection is not stable from my place. [12:05:52 CEST] <mop> wm4, I read above, please [12:06:47 CEST] <wm4> ubitux: you know there's an old videotoolbox patch somewhere on the ML? [12:07:11 CEST] <ubitux> wm4: ah? i missed that [12:07:39 CEST] <wm4> <mop> in this way you will manage filters in different ways depending if are built-in or dynamic, what is the plus in doing it ? <- different API users in the same process shouldn't clash... consider different plugins with the same name [12:08:24 CEST] <ubitux> wm4: oh indeed... (http://ffmpeg.org/pipermail/ffmpeg-devel/2012-September/130716.html) [12:10:28 CEST] <ubitux> and nothing since then? :( [12:11:39 CEST] <mop> scoping the name not implies change the context necessarly [13:14:53 CEST] <cone-653> ffmpeg 03Rong Yan 07master:603c839398f8: swscale/ppc/swscale_altivec.c: POWER LE support in yuv2planeX_8() delete macro GET_VF() it was wrong [13:14:54 CEST] <cone-653> ffmpeg 03Rong Yan 07master:8a03d428f175: avcodec/ppc/fdctdsp: POWER LE support in ff_fdct_altivec() delete macros VEC_FMERGEH() VEC_FMERGEL(), they where wrong [13:23:44 CEST] <fffan> I have a problem with ffmpeg mp4 muxing, Can anybody help me. [13:38:27 CEST] <fffan> hi, j-b, are you online? [13:39:32 CEST] <j-b> maybe [13:39:56 CEST] <fffan> I have a ffmpeg build(with fdkaac, without faac) [13:40:42 CEST] <fffan> and use x264 and faac to genrate media data, and using ffmpeg to mux them to a mp4 file. [13:40:51 CEST] <kierank> so when does outreachy get announced [13:41:03 CEST] <fffan> but my ffplay can't play audio from the mp4 [13:41:36 CEST] <fffan> my question is : can a ffmpeg(without faac) muxing faac data into a mp4 file? [13:41:43 CEST] <fffan> vlc is just ok. [13:42:11 CEST] <fffan> my ffplay warns " aacDecoder_DecodeFrame failed" [13:42:29 CEST] <fffan> my ffplay contains fdkaac, contains not faac. [13:42:55 CEST] <fffan> since my old ffmpeg build(with faac) is just ok. [13:43:03 CEST] <fffan> it bothers me. [13:44:31 CEST] <fffan> could you give me some tip? [13:44:55 CEST] <sfan5> fffan: ffmpeg should be able to mux and decode any aac whether encoded by faac, ffmpeg aac, libvo_aacenc or any other lib [13:45:11 CEST] <sfan5> it could also help if you could pastebin the complete output somewhere [13:45:43 CEST] <fffan> so, why? why fdkaac tell me aacDecoder_DecodeFrame failed? but vlc is just ok. [13:46:02 CEST] <fffan> ok [13:46:08 CEST] <fffan> I'll paste. [13:49:03 CEST] <fffan> https://gist.github.com/rtmpnewbie/a7e45fa145942a4ee2e8 [13:49:26 CEST] <fffan> sorry, I can't access pastebin, I just use github. [13:51:23 CEST] <fffan> sfan5, is github gist ok? [13:52:04 CEST] <sfan5> looks like you're using libav [13:52:13 CEST] <sfan5> the /topic says "uestions about using FFmpeg or developing with libav* libs should be asked in #ffmpeg" [13:52:19 CEST] <sfan5> Questions* [13:52:45 CEST] <fffan> ok [13:59:59 CEST] <Compn> fffan : depends if the aac parser was enabled at build :p [14:01:43 CEST] <fffan> ? [14:02:19 CEST] <fffan> my ffplay(without faac) can play any media file except my own. [14:03:00 CEST] <fffan> my code works ok under my old ffmpeg build. [14:03:11 CEST] <Compn> hang on lets talk in #ffmpeg [14:03:16 CEST] <fffan> ok [14:22:13 CEST] <cone-653> ffmpeg 03ZhouXiaoyong 07master:4a7b50679db3: makefile: add Loongson-3 support [14:43:51 CEST] <ubitux> so mmh audio frames have no timestamps? [14:49:19 CEST] <Daemon404> ubitux, define frame [14:49:33 CEST] <ubitux> i must be doing something wrong [14:49:43 CEST] <Daemon404> audio packets sure do.. [14:50:10 CEST] <ubitux> yeah that seems to be the case with ffmpeg, so i'm probably doing something wrong [14:50:55 CEST] <vibr> kierank, to mimic the behavior of ffmpeg -f framecrc, i dont' have to reencode the decoded frames to rawvideo first, do i? [14:52:43 CEST] <vibr> kierank, is it enough to decode the frames and then write them with the AVOutputformat set to framecrc? [14:54:00 CEST] <ubitux> mmh ok, best_effort_timestamp doesn't work with audio [14:54:06 CEST] <ubitux> is this on purpose? [14:55:12 CEST] <wm4> ubitux: how does it not work? [14:55:20 CEST] <wm4> oh let me guess, partial packet consumption? [14:55:57 CEST] <ubitux> no idea, i get nopts with best_effort_ts, but i actually have a (probably) good value with f->pts [14:57:03 CEST] <vibr> as kierank isn't her [14:57:11 CEST] <vibr> e does anyone else know? [15:01:37 CEST] <vibr> lglinskih, do you know when kierank is going to be here? [15:06:00 CEST] <lglinskih_> vibr: no =( [15:07:09 CEST] <ubitux> vibr: decoded frame means raw [15:10:28 CEST] <wm4> ubitux: broken decoder? what decoder? [15:10:52 CEST] <ubitux> flac [15:10:54 CEST] <vibr> ubitux, so can i write the decoded frame with av_write_frame directly? [15:11:25 CEST] <vibr> i thought i had to re-wrap it into a packet somehow [15:11:34 CEST] <ubitux> vibr: idk what you are trying to do [15:11:49 CEST] <vibr> ubitux: trying to mimic ffmpeg -f framecrc [15:12:22 CEST] <ubitux> just run a CRC over the frame->data[0..N] planes of that decoded frame? [15:12:48 CEST] <ubitux> ah mmh yeah format, sorry mmh [15:12:59 CEST] <ubitux> lemme check what's going on here [15:14:11 CEST] <vibr> i found out that setting "framecrc" as the third argument to avformat_alloc_output_context2 results in using that muxer (framecrc) [15:14:41 CEST] <vibr> don't know if that's the correct way to set it [15:14:52 CEST] <ubitux> yeah encode it with rawvideo to get a "packet"; you can look how it's done in lavc/rawenc.c [15:15:14 CEST] <vibr> thanks, i'll have a look [15:15:23 CEST] <ubitux> which is basically doing a call to av_image_copy_to_buffer() in the end [15:15:35 CEST] <ubitux> so... crc(av_image_copy_to_buffer(frame)) [15:15:39 CEST] <ubitux> i would guess [15:15:55 CEST] <ubitux> or you can pass it through the muxer to get packets if you prefer [15:16:13 CEST] <ubitux> i don't exactly know what's your goal (if it's testing the whole chain or actually having the crc itself) [15:17:17 CEST] <vibr> the goal is reproducing the crcs files found in http://git.videolan.org/?p=ffmpeg.git;a=tree;f=tests/ref/fate;h=3c25ec16d76… [15:18:40 CEST] <vibr> so for an input file from fate-suite, say SVA_Base_B.264, i want to get the corresponding crc file (seq-3341-7_seq-3342-5-24bit.flac in this example) [15:19:44 CEST] <ubitux> full chain: encode to rawvideo codec, mux to rawvideo format, run crc on packet->data [15:19:52 CEST] <ubitux> short path: crc(av_image_copy_to_buffer(frame)) [15:20:10 CEST] <ubitux> afaict. [15:20:17 CEST] <ubitux> i let you find the necessary glue for that [15:20:48 CEST] <vibr> hm, but muxing to framecrc format seemed to do most of it [15:21:15 CEST] <vibr> i'd only have to decode first, i suppose [15:22:32 CEST] <ubitux> mux to framecrc sorry [15:23:21 CEST] <vibr> when muxing to framecrc i don't need to run crc on packet->data, do i? [15:23:31 CEST] <ubitux> yes, i don't know what you want [15:23:44 CEST] <ubitux> if you want the crc itself, you should just take the short path [15:23:51 CEST] <ubitux> if you want the same output, use framecrc, so long path [15:24:05 CEST] <vibr> yes, the later (long path) [15:24:56 CEST] <vibr> i need the complete file as in test/ref/fate [15:25:49 CEST] <vibr> ok, and i definitely have to encode to rawvideo after decoding, i can't leave out that step? [15:28:46 CEST] <vibr> ubitux,thanks so far, i'll try and see [15:30:14 CEST] <ubitux> encoding is mostly free [15:30:22 CEST] <ubitux> it's calling av_image_copy_to_buffer to create a packet [15:30:41 CEST] <ubitux> the muxer only takes packet, so you can emulate that part if you want, but that's probably saner to just call the encoder [15:31:14 CEST] <vibr> ok, thanks a lot [16:02:22 CEST] <prelude2004c> hey, anyone around this morning? [17:51:34 CEST] <ubitux> anyone has any plan to support MediaCodec in ffmpeg? (the android stuff) [17:59:39 CEST] <BtbN> Doesn't MediaCodec use ffmpeg? Or is that something else? [18:01:08 CEST] <nevcairiel> how its implemented depends on the device [18:01:14 CEST] <nevcairiel> but it also has OMX hardware codecs behind it [18:01:57 CEST] <av500> depends [18:01:59 CEST] <av500> some HW, some SW [18:04:50 CEST] <prelude2004c> hey guys.. can anyone give me a hand.. i am not sure where the problem is... i am using pimd to get the multicast stream from one ethernet to another.. then i am using openvpn to create a basic tunnel with no encryption. The servers are talking and the client server can see the multicas stream. It seems when i pull one stream it starts ok but looses packets sometimes.. then i pull more streams ( program id's ) and it starts to colla [18:05:03 CEST] <av500> pse [18:06:16 CEST] <ubitux> prelude2004c: you message is truncated [18:06:19 CEST] <ubitux> +r [18:06:32 CEST] <ubitux> "starts to colla<truncated>" [18:06:44 CEST] <cone-653> ffmpeg 03Alexis Ballier 07master:6fe2afaa7127: Adapt libavcodec/tableprint_vlc.h to the av_dlog -> ff_dlog change. [18:12:19 CEST] <BtbN> And please don't ask the same question in both channels at the same time. Specialy when it's not entirely ffmpeg related and even less ffmpeg dev related. [18:26:37 CEST] <cone-653> ffmpeg 03Clément BSsch 07master:11d29277d648: tests/subtitles: fix microdvd crlf in ref [18:26:38 CEST] <cone-653> ffmpeg 03Clément BSsch 07master:2879a4ecb9c3: tests: make sure subtitles tests are run with a rawdiff [18:35:53 CEST] <anshul_mahe> I am trying to make ref file for fate [18:36:18 CEST] <anshul_mahe> I am having this error http://pastebin.com/hML6khFa [18:36:28 CEST] <anshul_mahe> is this related to ffmpeg-devel [18:36:55 CEST] <anshul_mahe> basically my sample have lavc version in it [18:37:33 CEST] <anshul_mahe> while when I do it from fate it does not have lavc version, how to produce sample without lavc version [18:49:48 CEST] <anshul_mahe> I got it thanks, cmd was in log file there [18:53:10 CEST] <anshul_mahe> for making it work, I wrote 2 lines.http://pastebin.com/x49UZqLv [18:53:54 CEST] <anshul_mahe> but first line look wrong to me [18:54:22 CEST] <anshul_mahe> can any past expirence with writing fate test review it [18:59:52 CEST] <anshul_mahe> ok I will find the meaing of 3 words, no worry. sorry for noise [19:29:05 CEST] <Compn> lol [19:29:52 CEST] <nevcairiel> that guy doesnt seem to be very smart [19:37:09 CEST] <kierank> send a dmca [20:14:06 CEST] <vibr> kierank, can i ask a question? [20:17:59 CEST] <kierank> vibr: yes [20:21:41 CEST] <vibr> avcodec_open2 returns something < 0 and i don't know where to look for the reason [20:22:01 CEST] <vibr> i call avcodec to set up the encoder using rawvideo codec [20:22:57 CEST] <vibr> do i have to initialise something else before calling avcodec_open2? [20:23:32 CEST] <vibr> for the rawvideo encoder, that is? [20:25:15 CEST] <vibr> greater picture: i set up a new formatcontext (for output) with avformat_alloc_output_context2, create a stream with avformat_new_stream, look for the codec (rawvideo) with avcodec_find_encoder [20:25:25 CEST] <vibr> and so far everything's fine [20:25:39 CEST] <vibr> then i call avcodec_open2, and it returns something < 0 [20:26:19 CEST] <vibr> so my guess is, that i have to initialise/setup something before calling call avcodec_open2, but i don't know what and how? [20:29:57 CEST] <BtbN> What exactly does it return? [20:34:22 CEST] <vibr> BtbN, it returns 1 [20:34:29 CEST] <vibr> erm, no [20:37:54 CEST] <vibr> BtbN, it returns -22 [20:39:21 CEST] <BtbN> av_make_error_string can turn that into a usefull error message. [20:40:13 CEST] <BtbN> Or av_strerror [20:41:06 CEST] <BtbN> 22 is "Invalid Argument" iirc, so some argument you passed, or forgot to pass, is invalid. [20:43:56 CEST] <BtbN> Did you look at some examples, like http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/examples/decoding_encodi… [20:44:25 CEST] <BtbN> Make sure you call avcodec_register_all first [20:50:44 CEST] <vibr> BtbN, yes, i looked at the examples [20:50:59 CEST] <vibr> the only one which is applicable for my case is transcoding.c [20:51:19 CEST] <vibr> i called avcodec_register_all [20:51:46 CEST] <vibr> so in transcoding.c there's some initialisation before avcodec_open2 is called [20:52:22 CEST] <BtbN> yeah, selecting the codec and parameters [20:52:35 CEST] <vibr> i mean these bits: enc_ctx->height = dec_ctx->height [20:52:37 CEST] <vibr> and so on [20:52:39 CEST] <vibr> 5 lines [20:53:24 CEST] <vibr> line 128-134 [20:54:12 CEST] <vibr> but the comment for this part says: "we transcode to same properties" [20:54:22 CEST] <vibr> and i'm not sure when this applies [20:54:37 CEST] <BtbN> What do you want to do? [20:54:52 CEST] <vibr> in my case a transcode from some video codec to rawvideo [20:54:56 CEST] <vibr> -a i [20:55:23 CEST] <BtbN> so basicaly just decoding? [20:55:48 CEST] <vibr> the chain is: decode, encode (as rawvideo), mux with framecrc [20:55:49 CEST] <BtbN> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/examples/transcoding.c#l… that function does most of the stuff needed for that [20:56:53 CEST] <vibr> line 149?? my source is 3 days old [20:57:11 CEST] <BtbN> hm? [20:57:32 CEST] <vibr> line 149 only contains a bracket ;) [20:57:41 CEST] <BtbN> l49 [20:57:46 CEST] <BtbN> not 149 [20:57:59 CEST] <BtbN> just click the link, will take you right to the function [20:58:35 CEST] <vibr> i see, the link [20:58:47 CEST] <vibr> your link is open_input_file [20:58:59 CEST] <vibr> that's not my problem, this part works fine [20:59:26 CEST] <vibr> open_input_file initialises the decoder [20:59:35 CEST] <vibr> i have problems with the encoder [20:59:54 CEST] <vibr> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/examples/transcoding.c#l… [21:00:10 CEST] <vibr> <- corresponding part in transcoding.c [21:00:49 CEST] <vibr> as i said, they set up these bits: http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/examples/transcoding.c#l… [21:00:57 CEST] <vibr> five lines from that [21:01:05 CEST] <vibr> but i'm not sure it applies to my case [21:01:29 CEST] <vibr> because of this comment http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/examples/transcoding.c#l… [21:16:22 CEST] <vibr> how can i find out what kind of initialisation has to be done for the rawvideo encoder before calling avcodec_open2? [21:16:30 CEST] <vibr> i don't know where to look [21:18:48 CEST] <vibr> kierank, different topic: i'm late for the outreachy application, as you can see [21:19:14 CEST] <vibr> but i'm still trying to finish that task [21:19:15 CEST] <Compn> stick around , devs will be back later :) [21:19:31 CEST] <vibr> ok :) [21:19:32 CEST] <Compn> or ask on the list [21:20:45 CEST] <vibr> you mean ffmpeg-devel? [21:31:08 CEST] <wm4> <vibr> how can i find out what kind of initialisation has to be done for the rawvideo encoder before calling avcodec_open2? [21:31:15 CEST] <wm4> hm, at least width, height, and pixel format [21:31:24 CEST] <wm4> pixel format or codec tag [21:31:33 CEST] <wm4> but I don't remember exactly [21:35:29 CEST] <vibr> wm4, thanks [21:35:46 CEST] <vibr> and how do i know the correct settings? [21:36:08 CEST] <vibr> i the example code here [21:36:27 CEST] <vibr> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/examples/transcoding.c#l… [21:37:21 CEST] <vibr> they just re-use the values from input-formatcontext->stream->codeccontext [21:37:35 CEST] <vibr> but i'm not sure when i can re-use these values [22:09:41 CEST] <wm4> vibr: it generally depends on the codec [22:09:49 CEST] <wm4> normally, there are not a lot of fields [22:10:27 CEST] <wm4> here's what mpv does (it manually copies fields from the demuxer codec context to the decoder codec context): http://git.srsfckn.biz/mpv/tree/common/av_common.c#n48 [22:13:20 CEST] <vibr> wm4, thanks, i'll check this tomorrow - too tired now :) [00:00:00 CEST] --- Tue Apr 28 2015

1 0

ffmpeg.log.20150427
by burek 28 Apr '15

28 Apr '15

[04:41:30 CEST] <prelude2004c> hey guys.. i have a weird problem.. [04:41:47 CEST] <prelude2004c> so i have pimd running over openvpn and i am getting the multicast signal on a remote server just fine [04:41:57 CEST] <prelude2004c> when i run ffmpeg and encode one channel .. all is well [04:42:31 CEST] <prelude2004c> when i do ffmpeg -i 239.255.x.x for multiple program ID's on that system i start getting artifacts on all the channels. The odd part is pimd has the multicast program there already so for ffmpeg its local [04:42:42 CEST] <prelude2004c> why would pulling the syignal locally more than once cause issues ? doens't make sense [05:06:25 CEST] <prelude2004c> anyone? [05:06:50 CEST] <trodis_> dude this channel is dead [05:06:56 CEST] <prelude2004c> yup [05:46:12 CEST] <fluke_> Hello. I am developing videotoolbox hwaccel for iOS. I want to know where can I find hwaccel architecture documents/slides. [08:46:11 CEST] <fffan> if a ffmpeg build doesn't contain faac, can I use it to muxing x264 data and faac data into mp4? [09:25:03 CEST] <t4nk926> Hello [09:25:37 CEST] <t4nk926> Can I get raw video data from mp4 file on ffmpeg? [09:32:32 CEST] <selsta> t4nk926: ffmpeg -i input.mp4 -c:v copy -an video.h264 [10:10:38 CEST] <t4nk926> thanks:) [10:46:59 CEST] <siberian> Hi can any one help me solve issue why i cant install manually ffmpeg [10:47:16 CEST] <siberian> im running ./configure and getting an error something to do with gcc compiler [10:47:28 CEST] <spaam> what dist do you use? [10:48:09 CEST] <spaam> i guess you forgot to install the gcc compiler. [10:49:31 CEST] <siberian> Linux 2.6.30.4 i686 [10:50:28 CEST] <mop> morning guys [10:51:06 CEST] <spaam> siberian: debian? ubuntu? arch? [10:51:23 CEST] <spaam> siberian: opensuse? fedora? [10:51:38 CEST] <spaam> siberian: centos? [10:51:40 CEST] <siberian> what command to get all info [10:51:45 CEST] <siberian> of the system [10:51:51 CEST] <mop> I'm trying to cross compile ffmpeg with mingw under ubuntu but a test for an additional library fails also after I installed it correctly under a path specified with the --incpath option of configure [10:51:54 CEST] <mop> any idea ? [10:52:29 CEST] <spaam> siberian: try: cat /etc/os-release [10:53:03 CEST] <siberian> spaam: no such file [10:53:18 CEST] <siberian> its a special network unit running linux spaam [10:53:30 CEST] <siberian> apt-get also dont work [10:53:39 CEST] <siberian> apt- cache dont work [10:53:46 CEST] <siberian> saying no such command [10:54:58 CEST] <spaam> cat /proc/version ? [10:55:05 CEST] <siberian> Linux version 2.6.30.4 (root(a)vm-ncl-buildbot-slave-01.internal) (gcc version 4.1.2) #2 SMP Fri Jun 20 21:16:44 BST 2014 [10:56:38 CEST] <ffmpeg-devel285> Hi [10:56:49 CEST] <spaam> siberian: i have no idea what dist that is. [10:57:25 CEST] <ffmpeg-devel285> I have question [10:57:43 CEST] <ffmpeg-devel285> I was converted H265 RAW to MP4 [10:57:49 CEST] <ffmpeg-devel285> with codec copy [10:58:16 CEST] <ffmpeg-devel285> I compared MP4 and H265 on hexviewer [10:58:16 CEST] <siberian> me2 [10:58:36 CEST] <siberian> spaam what can i do if the linux is very restrictive here? [10:58:36 CEST] <ffmpeg-devel285> I found some data changed [10:59:06 CEST] <ffmpeg-devel285> http://imgur.com/VBBwNgm [10:59:28 CEST] <ffmpeg-devel285> I understand about change data on header and tail [10:59:37 CEST] <ffmpeg-devel285> but why body part changed? [11:23:42 CEST] <Venti> ffmpeg-devel285: raw byte stream needs to have start code prefixes escaped while mp4 doesn't.. In that example though, I think the actual start code prefixes have been replaced by sizes of the data boxes [11:26:59 CEST] <ffmpeg-devel285> ok... [11:27:03 CEST] <ffmpeg-devel285> thanks [11:29:03 CEST] <ffmpeg-devel285> and.. why can't play video at convert mp4 (hevc) to yuv(hevc) with codec copy option ? [11:30:18 CEST] <ffmpeg-devel285> Do not have any information on mediainfo after convert to yuv (hevc) [11:30:59 CEST] <ffmpeg-devel285> I think it's bug... [11:31:10 CEST] <Venti> I don't understand what you mean.. what is the command you use? [11:32:01 CEST] <ffmpeg-devel285> ffmpeg -i "ex.mp4" -vcodec copy -an "ex.yuv" & comp.bat [11:32:05 CEST] <ffmpeg-devel285> ffmpeg -i "ex.mp4" -vcodec copy -an "ex.yuv" [11:36:17 CEST] <Mavrik> That's because mp4 does not have SPS/PSS in the bitstream [11:36:17 CEST] <Mavrik> And if you want to convert that to a format without headers you'll have to convert it to so-called Annex-B bitstream [11:36:17 CEST] <Mavrik> there's a bitsteram filter for that, h264_mp4toannexb or whatsit [11:36:17 CEST] <Venti> there is no yuv (hevc).. you either want raw yuv or hevc annex B [11:37:03 CEST] <ffmpeg-devel285> how to convert to hevc annex B? [13:53:10 CEST] <fffan> I have a mp4 muxing problem can anybody help me? [13:53:11 CEST] <fffan> https://gist.github.com/rtmpnewbie/a7e45fa145942a4ee2e8 [14:03:27 CEST] <Compn> fffan : ok, how are you creating the mp4 file? with what command line or program ? your own program ? [14:03:51 CEST] <fffan> using code to generate https://gist.github.com/rtmpnewbie/a7e45fa145942a4ee2e8 [14:04:16 CEST] <fffan> old ffmpeg build(with faac) works ok [14:05:01 CEST] <fffan> I have two encode thread one use x264 encode and the other one use faac(directly). [14:05:24 CEST] <fffan> then using MP4RecorderSink to mux mp4. [14:11:53 CEST] <Compn> an old ffmpeg decodes it but not newer ffmpeg ? [14:12:07 CEST] <fffan> yes [14:12:08 CEST] <Compn> can you upload a sample mp4 that exhibits this problem, we'll see if its your ffmpeg or maybe a new bug... [14:13:08 CEST] <fffan> configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --able-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libdplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrw--enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsc [14:13:09 CEST] <fffan> hroedinge--enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enle-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis -enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable- bx265 --enable-libxavs --enable-libxvid --enable-lzma --enable-decklink --enab -zlib [14:15:50 CEST] <fffan> [libfdk_aac @ 00000000007125e0] aacDecoder_DecodeFrame() failed: 4006 [14:21:35 CEST] <fffan> ok [14:27:47 CEST] <fffan> 128.199.75.70/test.mp4 [14:27:54 CEST] <fffan> http://128.199.75.70/test.mp4 [14:28:01 CEST] <fffan> the sample h.264 file. [14:28:06 CEST] <fffan> mp4 file [14:32:22 CEST] <Compn> audio works here [14:33:05 CEST] <Compn> ffmpeg has an internal decoder which should work [14:33:11 CEST] <Compn> hmm [14:34:22 CEST] <fffan> but not works for my new ffplay build [14:34:59 CEST] <fffan> and my new ffplay build can play any movie files in my computer [14:35:12 CEST] <Compn> recompile and disable libfdk_aac [14:35:57 CEST] <fffan> maybe it should work it I disable libfdk_aac, but I need libfdk_aac [14:36:19 CEST] <Compn> i'm trying to see if its possible to choose the decoder with ffmpeg/ffplay [14:38:39 CEST] <fffan> but my new ffplay can play the same aac(encode by faac) from MediaCoder. [14:44:17 CEST] <Compn> so you are making some kind of rtmp program ? :) [14:44:34 CEST] <Compn> sorry i dont know how to fix it :\ [14:46:13 CEST] <fffan> not rtmp, thanks. [14:46:20 CEST] <fffan> I 'll try anther way. [18:00:16 CEST] <MadTBone> when using -ss on an input file which contains an audio stream and a inter-frame compressed video stream, and where the value of -ss is *not* an integeral multiple of the video frame rate, will the input file's audio begin exactly at the specified time, or will it stay time aligned with its video stream? For example, if I have a 25fps source, and do ffmpeg -ss 0:00:00.60 -i sourcefile out.mp4, the start time is in the middle of the second frame. [18:04:07 CEST] <c_14> If you reencode, cuts are frame-perfect. [18:04:55 CEST] <prelude2004c> hey guys.. can anyone give me a hand.. i am not sure where the problem is... i am using pimd to get the multicast stream from one ethernet to another.. then i am using openvpn to create a basic tunnel with no encryption. The servers are talking and the client server can see the multicas stream. It seems when i pull one stream it starts ok but looses packets sometimes.. then i pull more streams ( pr [18:04:56 CEST] <prelude2004c> ogram id's ) and it starts to collapse the entire thing. Like there is some bw limit ... weird .. anyone? [18:07:37 CEST] <BtbN> iirc OpenVPN is not multicast capable, so it forwards multicast messages to all VPN clients. [18:07:51 CEST] <BtbN> So if you have a lot of clients, you are propably just overwhelming your network [18:35:50 CEST] <Almondon> To copy (-c copy) a section of a video from an online source (-i https://eg.com/example.mp4) will ffmpeg need to download the entirety of the video up to the the -to section to give me my output? Would it make a difference how much was downloaded if the -ss was put before the -i instead of after? [18:40:49 CEST] <BtbN> Almondon, as it's mp4, ffmpeg will have to download the entire file, as the file header is at the end of the file, unless it has been moved to the front. [18:43:43 CEST] <Almondon> Thank you kindly [19:02:57 CEST] <xahdica> Hey guys, where do I submit GPL violation? Videostream for Google Chromecast is in clear violation and refuses to send me the code they use static linking, GPL and non-free packages.. I requested they send me code to see how they transcode on the fly through Chrome but they refuse to answer emails.. [19:03:46 CEST] <xahdica> They're making money, 1mil+ users and are just hoarding it... [19:05:11 CEST] <xahdica> I tried to submit a Google Webstore violation report, but as I'm not the license holder I can't report it. [19:12:14 CEST] <c_14> Make an issue on trac [19:12:40 CEST] <c_14> https://trac.ffmpeg.org/ [19:26:21 CEST] <xahdica> Captcha might be on drugs. "CAPTCHA failed to handle original request" [19:29:15 CEST] <Compn> lol [19:29:25 CEST] <Compn> xahdica : chromecast code is out there i'm pretty sure [19:30:00 CEST] <Compn> oh videostream [19:30:02 CEST] <Compn> for chromecast [19:30:03 CEST] <Compn> i see.e [19:31:32 CEST] <Compn> xahdica : you try asking dem on twitter ? [19:34:07 CEST] <xahdica> I did on email, yeah [19:34:53 CEST] <quellen> i have a great problem [19:35:07 CEST] <quellen> someone speaks italian? [19:37:03 CEST] <quellen> i recorded a movie from DTT (.TS format) but sometimes the signal is noisy. and when i convert it with FFMPEG the audio goes out of sync :-( [19:39:24 CEST] <quellen> it's a unsolvable problem? [19:39:29 CEST] <quellen> http://pastie.org/10116714 [19:41:39 CEST] <siberian> Can someone help me troubleshoot? why do i get an error when trying to install ffmpeg with ./configure [19:42:15 CEST] <siberian> gcc is unable to create executable file [19:42:56 CEST] <quellen> what distro do you use? [20:47:55 CEST] <quellen> full log: http://pastebin.com/pNJcqY8B [20:48:10 CEST] <quellen> error starts at 00:04:34.80 (signa DTT noisy) [20:49:18 CEST] <quellen> waht can i do? [20:49:57 CEST] <quellen> from 00:04:34.80 the audio is out of sync. (in the orignal file the audio is ok) [21:07:17 CEST] <cowai> Will be I able to split a source file into multiple segments, and then render each segment to target format on separate machines, and then concat/merge them together as a final file (for load balancing) ? [21:07:39 CEST] <cowai> Is there any problem with that? [21:07:51 CEST] <quellen> O_O [21:07:59 CEST] <cowai> I mean do I need to think about keyframes and all that? [21:10:08 CEST] <cowai> quellen: What? [21:10:43 CEST] <quellen> it's too hard [21:11:02 CEST] <c_14> Not really. [21:11:14 CEST] <c_14> You do need to make sure you cut at keyframes [21:11:50 CEST] <c_14> It'd probably also be good not to cut the audio, but mux that out separately. [21:12:22 CEST] <cowai> Good idea, [21:12:36 CEST] <cowai> I tried once, and there was stutters at each cut [21:14:01 CEST] <cowai> ffmpeg -i source.mp4 -acodec copy -f segment -segment_time 60 -c copy -reset_timestamps 1 input_segment%d.mp4 [21:14:01 CEST] <cowai> Does this look okay for splitting? [21:14:52 CEST] <c_14> You'll probably want to probe the source to find the exact timestamps of all the keyframes and cut on those. [21:15:00 CEST] <c_14> Or else you'll get duplicate/dropped frames. [21:16:08 CEST] <quellen> @_@ [21:17:06 CEST] <cowai> c_14: Do you know of an easier option? [21:18:03 CEST] <c_14> There are some programs around that do this sort of thing already. [21:18:41 CEST] <cowai> segmenting or the whole thing? [21:18:53 CEST] <c_14> The whole thing. [21:19:59 CEST] <cowai> Do you know of any open source? [21:20:55 CEST] <cowai> https://github.com/dlasley/encode-cluster [21:21:03 CEST] <cowai> this looks like something I could use [21:22:10 CEST] <cowai> The main output formats I am going to use is webm and h264/mp4 [23:12:33 CEST] <squeegily> ffmpeg -i [UTW]_Fate_Zero_-_03_[BD][h264-720p_AC3][C9AEDDFB].mkv -ss 00:00:07 -vframes 1 -f rawvideo -pixel_format rgb24 - [23:12:49 CEST] <squeegily> Why does that still use yuv420p10le? [23:14:52 CEST] <squeegily> I still get: Output #0, rawvideo, to 'pipe:': [&] Stream #0:0(jpn): Video: rawvideo (Y3[11][10] / 0xA0B3359), yuv420p10le, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 23.98 fps, 23.98 tbn, 23.98 tbc (default) [23:15:33 CEST] <kepstin-laptop> squeegily, because the option name is "-pix_fmt" not "-pixel_format" [23:16:19 CEST] <kepstin-laptop> (I think -pixel_format is an avoption for something, which is why it's not rejected outright, but -pix_fmt is the one you want here) [23:17:04 CEST] <squeegily> Ahh, thanks! [23:18:14 CEST] <kepstin-laptop> hmm. yeah, it's a bit confusing. you need to use the "-pixel_format" input option to tell the rawvideo input what format the file contains, but you use -pix_fmt to tell ffmpeg what format to send to the raw video output. [23:21:15 CEST] <kepstin-laptop> oh, huh, fate zero bds are out now? [23:22:21 CEST] <kepstin-laptop> (that's a bit off-topic tho ;) [23:24:05 CEST] <JEEBsv> have been for quite a while [23:24:28 CEST] <JEEBsv> I mean the first box came out some time after the first cour ended [23:27:48 CEST] <squeegily> I only watch really old stuff because that way all the episodes are out, there's a consensus on the status of the various fansubbing groups, etc [23:28:39 CEST] <squeegily> Waiting for a show to release an episode AND for the fansub group to get around to it is a pain.. and then if your group drops it halfway through you have to find a new one [00:00:00 CEST] --- Tue Apr 28 2015

1 0

ffmpeg-devel.log.20150426
by burek 27 Apr '15

27 Apr '15

[01:17:43 CEST] <cone-443> ffmpeg 03Michael Niedermayer 07master:d52a8bf15817: avcodec/h264_cabac: Use ff_tlog() for block level information [01:17:44 CEST] <cone-443> ffmpeg 03Michael Niedermayer 07master:7b94a2f4b183: avcodec/ituh263dec: Use ff_tlog() for block level information [01:32:37 CEST] <cone-443> ffmpeg 03Rico Tzschichholz 07master:5aed1d4240d4: vaapi: Fix -Wimplicit-function-declaration for ff_dlog() usage [01:32:38 CEST] <cone-443> ffmpeg 03Michael Niedermayer 07master:fa2d3b6dbe0e: Merge commit '5aed1d4240d411533c102eec6944aa925b7f4964' [13:02:47 CEST] <cone-408> ffmpeg 03Michael Niedermayer 07master:febbb3cbd395: avcodec/h264_slice: Change a few asserts to av_assert* [13:02:47 CEST] <cone-408> ffmpeg 03Michael Niedermayer 07master:0b13f42238c0: avcodec/interplayvideo: Use ff_tlog() for block level decode tracing [13:55:10 CEST] <cone-408> ffmpeg 03Michael Niedermayer 07master:e06dde52d75a: avcodec/dvbsubdec: Do not return a value from a function returning void [14:08:55 CEST] Action: Daemon404 starwes at the the latest b44 lulz [16:45:22 CEST] <cone-408> ffmpeg 03Michael Niedermayer 07master:71043016384a: avcodec/rv10: Use ff_tlog() for block level tracing [16:45:23 CEST] <cone-408> ffmpeg 03Matteo Nastasi (mop) 07master:9c3c7b457925: avutil/log: more complete description of av_log function [18:00:52 CEST] <cone-408> ffmpeg 03Michael Niedermayer 07release/2.4:de7b74d2544d: Changelog, fix typos [18:28:47 CEST] <cone-408> ffmpeg 03Matteo Nastasi (mop) 07n2.4.9:HEAD: avutil/log: more complete description of av_log function [19:55:19 CEST] <ubitux> > --disable-sdl --disable-libxcb --disable-dxva2 --disable-vaapi --disable-vda --disable-vdpau --disable-xlib [19:55:26 CEST] <ubitux> dammit this autodetect is driving me mad [19:55:36 CEST] <ubitux> 5th attemp to get a ffmpeg not linking to x11 [20:00:27 CEST] <ubitux> and now to jack & friends ;_; [20:02:16 CEST] <BtbN> disable-dxva2? oO [20:03:32 CEST] <ubitux> yeah this one is probably useless, i just picked every hwaccel listed and disabled them manually [20:03:43 CEST] <ubitux> (seems --disable-hwaccels wasn't enough) [20:10:09 CEST] <Compn> ubitux : disable ffplay , thats what enables x11 ? :P [20:10:17 CEST] <Compn> oh no, thats for sdl [20:10:27 CEST] <Compn> ffmpeg uses x11 for desktop capture i think [20:11:32 CEST] <wm4> getting ffmpeg not to link sdl is also quite the adventure [21:48:25 CEST] <cone-408> ffmpeg 03Michael Niedermayer 07master:81e40c26e176: avcodec/ffv1enc: Ensure that bits per raw sample is valid [21:48:26 CEST] <cone-408> ffmpeg 03Michael Niedermayer 07master:4b8a81949644: avcodec/ffv1enc: set bits_per_raw_sample for 8bit based formats [22:17:52 CEST] <michaelni> ubitux, your subtitle tree passes on linux but fails on ming (so dont commit yet) [22:18:03 CEST] <michaelni> mingw [22:18:09 CEST] <ubitux> ok [22:18:15 CEST] <nevcairiel> everyone hates windows =( [22:26:00 CEST] <ubitux> it's fine, there is an easy fix [22:32:19 CEST] <ubitux> michaelni: mmh wait, will resend a new patch [22:32:21 CEST] <ubitux> ignore last one [22:34:55 CEST] <ubitux> here you go [23:52:20 CEST] <wm4> ubitux: hm, another srt sub with a {y:i} tag in it [23:52:30 CEST] <wm4> is that a microdvd tag? [00:00:00 CEST] --- Mon Apr 27 2015

1 0

ffmpeg.log.20150426
by burek 27 Apr '15

27 Apr '15

[02:03:34 CEST] <kyleogrg> hey [02:03:43 CEST] <kyleogrg> curious about doing 25p to 24p [02:03:54 CEST] <kyleogrg> what's the most common way to do this? [02:09:50 CEST] <BtbN> Why do you want to do that? [02:10:59 CEST] <kyleogrg> curiousity, mostly [02:13:37 CEST] <kyleogrg> I downloaded some 25p video, and it made me wonder if the standard process is to slow down the video to 24p [02:25:48 CEST] <BtbN> Why would you want to do that? [02:25:57 CEST] <BtbN> Is it running to fast, and actualy intended to be 24 fps? [02:26:43 CEST] <kyleogrg> no, nothing's wrong with it [02:27:05 CEST] <kyleogrg> just wondering what someone would do IF they wanted to make it 24p [02:47:20 CEST] <prelude2004c> hi everyone.. anyone here familiar with pimd ? [03:17:12 CEST] <c_14> kyleogrg: either drop frames or slow the video down. Not much else you can do [03:19:07 CEST] <kyleogrg> c_14: sure [03:33:07 CEST] <kyleogrg> if i'm going from 60i to 60p (as opposed to 60i to 30p), should i have any special deinterlace settings? [03:33:25 CEST] <kyleogrg> or is there no difference, just whatever deinterlacer i want to use [03:37:40 CEST] <c_14> I think you need to tell the deinterlacer to output a frame per field instead of a frame per frame. For yadif this is with mode=1 or 3 iirc [03:38:42 CEST] <kyleogrg> ok, that's what i thought. so to do 60i to 60p, is it sufficient to do yadif (one field) and then set -r 59.94? is that all? [03:39:13 CEST] <c_14> you shouldn't even have to set the -r [03:39:34 CEST] <kyleogrg> ok, that yadif setting alone will do it... [03:40:09 CEST] <kyleogrg> you think mcdeint could be useful? [03:41:59 CEST] <kyleogrg> that is, is it just as useful for 60i to 60p as it is for 60i to 30p.... [03:43:05 CEST] <c_14> Depends on the amount of motion probably. [03:45:21 CEST] <kyleogrg> hmm, well, it can't hurt, right? the only downside would be unnecessary slowness? [03:49:04 CEST] <c_14> Depends on the content. Test on a sample of the video. [03:49:11 CEST] <kyleogrg> ok, thanks a lot [04:18:29 CEST] <hi117> what is that bar that goes down a survelance video called and how do i get rid of it? [06:02:03 CEST] <kyleogrg> hey [06:02:19 CEST] <kyleogrg> i'm trying to do a batch two-pass x264 encode [06:02:42 CEST] <kyleogrg> Here's my command for one-pass so far. (it takes all the AVIs in a folder) how can i adjust it? [06:02:45 CEST] <kyleogrg> for %%a in ("*.avi") do ffmpeg.exe -i "%%a" -y -t 00:00:10 -vf "yadif=1:1:0,mcdeint=0:1:1,scale=iw*sar:ih" -sws_flags lanczos -pix_fmt yuv420p -movflags +faststart -c:v libx264 -preset veryslow -c:a libfdk_aac -vbr 4 "%%~na.mp4" [06:03:05 CEST] <kyleogrg> ignore the -t [06:10:41 CEST] <kyleogrg> Or, simply put, how can I do TWO commands (Windows) for every AVI, instead of the one command pasted above? [06:17:50 CEST] <kyleogrg> never mind... [07:00:30 CEST] <unknownloner> Hello. I'm trying to record video and audio from my desktop, and the audio has a strange desync where it thinks more time has elapsed than really has. When I play the video with mplayer I see that the audio timestamp runs far ahead of the video timestamp. while recording the "time" value is far greater than the actual time I've been recording, matching the final audio timstamp. Example mplayer output: A: [07:00:33 CEST] <unknownloner> 66.9 V: 16.8 A-V: 50.097. Anything anyone can suggest based on that or should I pastebin my entire command? [07:02:28 CEST] <unknownloner> Looks like it's related to my somewhat strange conversion process so I'll paste the whole thing [07:03:56 CEST] <unknownloner> http://pastie.org/10114231 [07:04:41 CEST] <unknownloner> Yes there's a lack of -r in the conversion command, I've tried with and without it in there and it doesn't make a difference [07:05:56 CEST] <unknownloner> When I use the 'record()' function in there on it's own and write to /dev/null the Time number in the progress display is correct, so it's only after it goes through the FLV encoding that the time is off [07:06:16 CEST] <unknownloner> Also I've noticed I use -f flv and save to an mp4, whatever the case mplayer figures it out and plays it so w/e [07:28:36 CEST] <JamJams> Is there a way to change the Encoder metadata tag when remuxing? [07:28:54 CEST] <JamJams> I tried -metadata encoder="test" but it still writes "Lavf56.30.100" [09:58:17 CEST] <techtopia> anyone know how i can work out the location of an .ismv file on a http server [09:59:42 CEST] <techtopia> like the manifest file is here [09:59:44 CEST] <techtopia> http://lb-ngx.p-01.streamgate.nl/smartvod/video/videoland/138182_TL_Zwarte-… [10:00:16 CEST] <techtopia> so the ismv file should be in the same dir [10:00:26 CEST] <techtopia> but with a diffrent name [10:00:35 CEST] <techtopia> anyone got any ideas how i can find it [11:39:03 CEST] <mariangabriel> Hey guys, I got a problem when I want to concatenate 15 videos. Console http://pastebin.com/e336YYAt and the result https://www.youtube.com/watch?v=DItmfKIrtPQ&t=19s [11:39:09 CEST] <mariangabriel> wird pixels [11:39:13 CEST] <mariangabriel> wierd [11:45:50 CEST] <mariangabriel> console dump when I want to add a background http://pastebin.com/ctc1ZsEx . The errors start to 19s [17:40:54 CEST] <DeadSix27> for testing purposes what would be the way, to under windows, pipe from ffmpeg, to ffplay? [17:41:32 CEST] <DeadSix27> (if there is one) [17:41:33 CEST] <BtbN> The exact same way you'd do it everywhere else? [17:41:39 CEST] <jarainf> ^this [17:43:00 CEST] <DeadSix27> well, i have no idea how to do it anywhere else [17:43:56 CEST] <DeadSix27> prob a combination of dev/stdout and the "|" syntax, but, under windows?.. idk. [17:45:04 CEST] <Anoia> balh | frob [17:45:15 CEST] <c_14> 'pipe:1 | ffplay -' probably [17:45:20 CEST] <Anoia> stdout of balh will be attached to stdin of frob [17:45:42 CEST] <Anoia> I odn;t think - is standardised on Windows to be stdout/stdin [17:45:48 CEST] <DeadSix27> note: on windows [17:45:53 CEST] <DeadSix27> ye. [17:46:03 CEST] <DeadSix27> the | syntay for sure doesnt work [17:46:07 CEST] <DeadSix27> syntax* [17:46:13 CEST] <BtbN> the | syntax for sure does work. [17:46:36 CEST] <c_14> 'pipe:1 | ffplay pipe:0' then [17:46:53 CEST] <DeadSix27> hm [17:47:23 CEST] <DeadSix27> uh [17:47:31 CEST] <DeadSix27> i should definitely work more with the windows console [17:47:36 CEST] <DeadSix27> c_14: thanks that worked [17:48:14 CEST] <BtbN> ffmpeg -i something.m4a -c copy -f flv - | ffplay - [17:48:17 CEST] <BtbN> works fine for me. [17:48:36 CEST] <DeadSix27> didnt work for me BtbN [17:48:39 CEST] <DeadSix27> but the c_14 method worked [17:48:57 CEST] <BtbN> I'm running that right now, it definitely does work, with exactly that command, in the windows cmd.exe [17:48:58 CEST] <DeadSix27> (maybe i mistyped it the first time idk, but the pipe:1 & pipe:0 method did the trick. [17:49:11 CEST] <squeegily> Is there an "auto passthrough"? I want to write a shell script: ffmpeg -i "$1" -vf "subtitles='$1'" -c:a copy_if_possible_otherwise_idgaf "$2" [17:49:58 CEST] <DeadSix27> BtbN: like i assumed, the console was relatively small, so the "-" behind ffplay was right after a linebreak, and i wasnt sure if its there or not, and it probably wasnt the first few times [17:50:20 CEST] <c_14> squeegily: no [17:50:35 CEST] <c_14> And as soon as you add a video filter you won't be able to copy anyway. [17:50:51 CEST] <squeegily> You can still audio copy [17:51:03 CEST] <squeegily> Saves a tiny bit of CPU and audio quality [17:51:25 CEST] <squeegily> When transcoding fansubs for the PS3, I usually use c:a copy since the audio is usually AAC [17:51:33 CEST] <c_14> Oh, didn't see that :a [17:52:07 CEST] <c_14> The only way would be to use a scripting language of some sort. [18:19:22 CEST] <jookiyaya> why does so many dvds have black bars on all sides? like this http://www.imagebam.com/image/eab05d406028359 [18:21:26 CEST] <BtbN> Because TVs overscan anyway, and that way the codec can compress the actual image a little more efficiently. [18:22:31 CEST] <jookiyaya> btbn but not all dvds have blackbars [18:22:32 CEST] <Anoia> hangover form the analogue times :| [18:23:11 CEST] <jookiyaya> btbn what if i buy bluray for same title, would it have black bars? [18:23:31 CEST] <BtbN> If the producers feel like doing it. [18:23:43 CEST] <Anoia> i'd have thought it's down to the production company [18:23:52 CEST] <BtbN> There is no technical reason for it with digital video. [18:24:06 CEST] <Anoia> it seems stupid to take high res video, down samepl it a bit for TV to upsamepl it again [18:24:32 CEST] <jookiyaya> have you ever seen blackbars for bluray video? [19:22:25 CEST] <retard> uh, aren't black bars around the image super bad for codec compression? [19:22:35 CEST] <retard> with mpeg2 anyway [19:27:29 CEST] <squeegily> Ugh my "HDTV" is 1366x768 and overscans freaking everything [19:27:48 CEST] <squeegily> 720p gets scaled up so it can crop ~20px off the edges [19:28:11 CEST] <RobotsOnDrugs> unless your tv is really that terrible, you should be able to make it display properly [19:28:51 CEST] <RobotsOnDrugs> probably an option in the picture settings somewhere [19:29:25 CEST] <squeegily> Regrettably, there is nothing I can do.. I've scoured the menus and the manual and tweaked all the options [19:30:05 CEST] <squeegily> Windows XP is the only OS I've seen that actually detects 1366x768 as an available option [19:30:33 CEST] <squeegily> It's probably an OS-level workaround since XP's era overlaps with the "HD-Ready TV" era [19:30:47 CEST] <squeegily> It was convenient as fuqq [19:46:24 CEST] <myth_> hi [19:50:06 CEST] <myth_> For a project I a video with "~/bin/ffmpeg -y -i my.avi -codec copy -map 0 -f segment out\%07d.nut", encode part with "~/bin/ffmpeg -y -i <my segment001> -vcodec libx264 -vb 1000k -acodec copy -f mp4 001.ts" then for segment002... etc. When I concat the result it works fine until I let the "acodec copy". If I re-encode the audio, I can hear the split [19:51:02 CEST] <myth_> there is a something to avoid this noise ? [19:54:51 CEST] <techtopia> yes [19:55:05 CEST] <techtopia> demux the audio [19:55:48 CEST] <techtopia> and "copy /b" all the audio segments into a single file, then mux it back in to the concatted video file [19:56:17 CEST] <c_14> Just mux the video over from the original file? [20:55:28 CEST] <retard> squeegily resolution like that is a function of driver support [20:55:46 CEST] <retard> it's not a part of the os at all [22:38:22 CEST] <JoeDawson> Hey, is anyone about to answer a question or two? [22:39:47 CEST] <c_14> Just ask, if someone can help you they will. [22:41:09 CEST] <JoeDawson> Thanks - my first is regarding visualizations. Is this possible, bottom left? https://www.youtube.com/watch?v=fEx1k2zW6SA [22:42:51 CEST] <JoeDawson> My second question would be, if not possible - how about a single horizontal line? The width of the video. I've done some looking around from google and on the ffmpeg docs, but I'm yet to come across something at least similar to what I want. [22:44:20 CEST] <c_14> 1: If you have that as a video with alpha, the overlay filter should work afaik. [22:46:03 CEST] <c_14> With single horizontal line do you mean something like showwaves? [22:48:33 CEST] <JoeDawson> With the first question, I'd ideally like it to be created depending on the audio I pass to it - it might not be clear in that video as the visualization is pretty small, but it's matched to the audio. WIth the single line, showwaves looks like it may be what I'm after - can it be tweaked to look similar to the video above? [22:49:45 CEST] <JoeDawson> Just come across this actually re: showwaves - going to have a little play now; http://looooooooop.blog35.fc2.com/blog-entry-1098.html [23:00:40 CEST] <WhiteBunny> Hello everyone, i have a question about .m2v file how can i re-encode it to a bigger size ? [23:00:58 CEST] <c_14> You have a file and you want to make it bigger? [23:02:00 CEST] <WhiteBunny> The thing is i have a DVD9 source and i want to make it to a DVD5 source without unwanted materials like FR audio and stuff like that :D [23:02:40 CEST] <c_14> Isn't DVD9 bigger than DVD5? [23:03:00 CEST] <WhiteBunny> Yes it is [23:03:18 CEST] <BtbN> iirc dvd9 is double layer, dvd5 single layer. So a double layer dvd has double capacity. [23:03:21 CEST] <sfan5> re-encoding MPEG2 video to make it bigger won't make the quality any better [23:03:29 CEST] <c_14> So wouldn't you want to make it smaller, not larger? [23:04:17 CEST] <WhiteBunny> But after i have remove all unwanted materials and remove all unwanted audio stream i have a DVD size of 3.37gb :/ [23:05:32 CEST] <c_14> Making it bigger won't increase quality (it'll actually make it worse). Just mux/write what you have. [23:05:33 CEST] <BtbN> so you are well below the limit of a single layer dvd [23:08:14 CEST] <WhiteBunny> c_14 if i say it this way, i want to know how the ppl from scene are doing when they take a DVD9 source and re-encoding it to a DVD5 source with only English audio stream and have blank/stripp all Extras and warnings [23:12:05 CEST] <WhiteBunny> This release as a sample -> The.Interview.MULTISUBS.PAL.DVDR-WILDER i find the DVD9 source that they work on and i tried to stripp all unwanted materials and audio streams and i get a final dvd with English and Italian audio + An video sample of 50mb to a final size of 4.36gb :/ [23:13:05 CEST] <flurant> are you saying that the scene encode is larger? [23:14:29 CEST] <WhiteBunny> i dont know, but something are they doing to make it to a finally size of a release to 4.42 - 4.46GB :/ [23:14:42 CEST] <flurant> how do you know they are doing that if you dont know the size of it [23:17:51 CEST] <WhiteBunny> i can easy download the release and demux .m2v file and see how large it is and compare to the .m2v file i get [23:19:27 CEST] <c_14> Making the file bigger is easier, there's just no real reason to do so as it doesn't (positively) affect the quality. [23:20:21 CEST] <trodis_> what causes this 5 second delay when i use ffmpeg -> ffserver <-> ffplay [23:20:54 CEST] <c_14> buffering presumably. Either in the encoder, the muxer, or ffserver or ffplay [23:20:58 CEST] <trodis_> on ffplay side i turned off buffering with -fflag nobuffer on the server side i set the preroll to 1 [23:21:47 CEST] <trodis_> can u help me reduce this as much as possible, im just streaming audio [23:22:16 CEST] <trodis_> im encoding with the opus codec on the ffmpeg side [23:23:10 CEST] <c_14> try -application lowdelay ? [23:25:19 CEST] <trodis_> the werid part ist when i use libmp3lame the delay is reduced to 1 sec. [23:25:27 CEST] <trodis_> i thought opus was made for low latency [23:26:04 CEST] <c_14> Are you using opus or libopus ? [23:26:10 CEST] <trodis_> libopus [23:26:20 CEST] <c_14> Did you try adding -application lowdelay? [23:26:32 CEST] <trodis_> let me try [23:29:29 CEST] <trodis_> yeah i tried [23:29:34 CEST] <trodis_> still not improvement [23:29:38 CEST] <trodis_> with libopus [23:30:15 CEST] <trodis_> http://pastebin.com/bnVDyvUS ffmpeg [23:30:27 CEST] <trodis_> http://ix.io/i7U [23:30:32 CEST] <trodis_> my server conf [23:32:01 CEST] <c_14> Oh, adding -application lowdelay there won't actually do anything [23:32:18 CEST] <trodis_> where do i need to add it [23:32:23 CEST] <c_14> I think you need to add AVOptionAudio application lowdelay to ffserver.conf [23:33:51 CEST] <c_14> In the stream block [23:34:19 CEST] <trodis_> yeah i did now ffplay is not playing anything [23:45:41 CEST] <c_14> ok, I tested on my system. The 5 seconds of delay are because of the ogg format [23:45:50 CEST] <c_14> If I switch to matroska the delay is much lower [23:46:06 CEST] <trodis_> yeah thats what i noticed with libmp3lame [23:46:16 CEST] <trodis_> hmm any suggestion? [23:46:26 CEST] <trodis_> can i change the format [23:46:26 CEST] <c_14> Change the stream to test.mkv [23:46:31 CEST] <trodis_> ok [23:46:34 CEST] <c_14> Assuming your audio player can handle matroska [23:46:43 CEST] <trodis_> ffplay should be able to [23:46:55 CEST] <trodis_> for long term im going to use mplayer or mpc [23:51:03 CEST] <c_14> Don't know if there are any other containers that can hold opus... [23:53:35 CEST] <trodis_> even when i use mkv the delay is the same [23:53:45 CEST] <trodis_> i dont get it srsly so frustrating [23:53:53 CEST] <c_14> For me mkv has 4s less delay. [23:54:05 CEST] <trodis_> can i see ur ffserver.conf pls [23:54:09 CEST] <c_14> I was opening a raw tcp socket with ffmpeg though [23:54:32 CEST] <c_14> `ffmpeg -re -i foo -c:a libopus -application lowdelay -f oga tcp://127.0.0.1:5555?listen' [23:54:51 CEST] <trodis_> ah ok [23:54:54 CEST] <c_14> eh -f matroska [23:55:14 CEST] <c_14> Then open tcp://127.0.0.1:5555 and check the delay [23:55:16 CEST] <trodis_> what is the -re option for [23:56:46 CEST] <c_14> Reads the input in "real-time". You won't need it if you're using a live source like a microphone or something. [00:00:00 CEST] --- Mon Apr 27 2015

1 0

ffmpeg-devel.log.20150425
by burek 26 Apr '15

26 Apr '15

[00:49:50 CEST] <Compn> kierank : not seeing any mention of x264 and geovision ... might be worth looking into further [01:01:46 CEST] <j-b> I love iolations [01:01:52 CEST] <j-b> +v [01:11:48 CEST] <Prelude2004c> hey guys.. anyone know how to copy a trasnport stream from one place to anther without touching it ? everytime i do -c copy it shows up on the other side as just 1 program and it says " service01 " etc etc [02:41:10 CEST] <cone-359> ffmpeg 03Niklesh 07master:0ec3abeb8d9b: avcodec/movtextdec: Decoding of Bold-Italic_Underlined styles for 3gpp timed text subtitles [04:29:09 CEST] <Timothy_Gu> Prelude2004c: file a ticket [04:37:52 CEST] <Timothy_Gu> ls [04:37:55 CEST] <Timothy_Gu> oops [04:38:11 CEST] <Timothy_Gu> WHY DO I ALWAYS MESS UP WITH IRSSI [04:48:06 CEST] <cone-359> ffmpeg 03Michael Niedermayer 07master:75a730a22b76: avformat/format: Add generic trace av_log in av_probe_input_format3() [05:01:37 CEST] <highgod> Hi, all. I met a problem on compiling the latest ffmpeg trunk on mingw 32 [05:08:26 CEST] <highgod> is there any other websit like pastebin? because I can't use pastebin here [05:08:42 CEST] <highgod> I want to paste the compile error [05:09:18 CEST] <atomnuker> highgod: http://paste.debian.net/ works for me [05:14:55 CEST] <highgod> http://paste.debian.net/168761/ [05:15:06 CEST] <highgod> atomnuker: thanks [05:34:57 CEST] <jamrial> that's not an ffmpeg problem. looks like a cygwin one [05:38:50 CEST] <cone-359> ffmpeg 03Michael Niedermayer 07master:ee8ce211ead0: libavcodec/snowdec: Mark as FF_CODEC_CAP_INIT_THREADSAFE and FF_CODEC_CAP_INIT_CLEANUP [05:54:22 CEST] <highgod> but the previous version is correct [05:54:35 CEST] <highgod> I make again and the compilation passed [06:13:50 CEST] <Compn> rm -rf Timothy_Gu [06:13:53 CEST] <Compn> doh :P [12:59:27 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07master:d49210788b08: avcodec/snowenc: Mark as FF_CODEC_CAP_INIT_THREADSAFE and FF_CODEC_CAP_INIT_CLEANUP [13:41:00 CEST] <cone-793> ffmpeg 03ZhouXiaoyong 07master:6c5ef7f65732: avcodec/mips: disable assembly not supported for Loongson-3 [14:10:07 CEST] <cone-793> ffmpeg 03ZhouXiaoyong 07master:663321ed50ce: configure: add Loongson-3 support [15:05:57 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07master:a3110beebd78: avutil/softfloat: document av_int2sf() [15:25:14 CEST] <Prelude2004c> hey guys.. anyone know how to copy a trasnport stream from one place to anther without touching it ? everytime i do -c copy it shows up on the other side as just 1 program and it says " service01 " etc etc.. i just want to take the multicast from one place to another [15:28:03 CEST] <kierank> Prelude2004c: multicat [15:39:44 CEST] <ubitux> http://www.loc.gov/standards/iso639-2/ISO-639-2_utf-8.txt leaving that here, might be useful some day [15:51:57 CEST] <Prelude2004c> multicat ? netcat i tried [15:52:03 CEST] <Prelude2004c> multicat is different ? [15:52:06 CEST] <Prelude2004c> will def. try that [15:55:49 CEST] <Prelude2004c> hum.. i dont get it [17:04:49 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:8c832e3cc05c: avformat/img2dec: do not rewind custom io buffers [17:04:50 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:81e4b6f11a01: avformat/utils: Ensure that AVFMT_FLAG_CUSTOM_IO is set before use [17:04:51 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:d7071efae4a2: avformat/utils: avoid discarded streams in av_find_default_stream_index() [17:04:52 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:27993da8e789: avcodec/h264: Fix race between slices where one overwrites data from the next [17:04:53 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:7493c54ad075: avcodec/h264: finish previous slices before switching to single thread mode [17:04:54 CEST] <cone-793> ffmpeg 03Timothy Gu 07release/2.4:155e0e9fd964: tests: Fix test name for pixfmts tests(cherry picked from commit e1ee0521a698809ed216e9e5c11bd2bbb466ed04) [17:04:55 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:e617d6f98844: avcodec/h264_slice: Dont reset mb_aff_frame per slice [17:04:56 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:4191e01e9906: avcodec/h264: reset the counts in the correct context [17:04:57 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:2adcdf50844a: avcodec/aacdec: Fix storing state before PCE decode [17:04:58 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:9c4b09c4bdfa: avcodec/h264: Be more tolerant to changing pps id between slices [17:04:59 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:cf80856cbc79: avcodec/h264_ps: Move truncation check from VUI to SPS [17:05:00 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:fd90005b4619: avcodec/h264: Do not fail with randomly truncated VUIs [17:05:01 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:72ddcb2459fa: swresample: Allow reinitialization without ever setting channel counts [17:05:02 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:6a87a152e834: swresample: Allow reinitialization without ever setting channel layouts (cherry picked from commit 80a28c7509a11114e1aea5b208d56c6646d69c07) [17:05:03 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:115961acc15c: swresample: Check channel layouts and channels against each other and print human readable error messages [17:05:04 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:164ababc6238: swresample/swresample-test: Randomly wipe out channel counts [17:05:05 CEST] <cone-793> ffmpeg 03Andreas Cadhalpun 07release/2.4:65b257b21c7f: msrledec: use signed pixel_ptr in msrle_decode_pal4 [17:05:06 CEST] <cone-793> ffmpeg 03Andreas Cadhalpun 07release/2.4:d2ff2c9dd324: aacdec: consistently use avctx for logging in decode_eld_specific_config [17:05:07 CEST] <cone-793> ffmpeg 03Andreas Cadhalpun 07release/2.4:cfbfe6cd4d5a: aasc: return correct buffer size from aasc_decode_frame [17:05:08 CEST] <cone-793> ffmpeg 03Andreas Cadhalpun 07release/2.4:6458ee5af8f9: aacpsy: avoid psy_band->threshold becoming NaN [17:05:09 CEST] <cone-793> ffmpeg 03Andreas Cadhalpun 07release/2.4:6b1c8797f464: ac3: validate end in ff_ac3_bit_alloc_calc_mask [17:05:10 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:b28fab2ebe39: avcodec/atrac3plusdsp: fix on stack alignment [17:05:11 CEST] <cone-793> ffmpeg 03Andreas Cadhalpun 07release/2.4:1a8b4158c574: alsdec: ensure channel reordering is reversible [17:05:12 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:4c20249c86e1: avcodec/alsdec: Use av_mallocz_array() for chan_data to ensure the arrays never contain random data [17:05:13 CEST] <cone-793> ffmpeg 03Andreas Cadhalpun 07release/2.4:3bd7aa46601a: alsdec: validate time diff index [17:05:14 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:3fd66f7f4589: avutil/pca: Check for av_malloc* failures [17:05:15 CEST] <cone-793> ffmpeg 03Rainer Hochecker 07release/2.4:07a7f384217d: h264: avoid unnecessary calls to get_format [17:05:16 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:7572cf7fdcc9: avcodec/h264: Fail for invalid mixed IDR / non IDR frames in slice threading mode [17:05:17 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:1deebf3b225d: avcodec/h264_refs: Do not set reference to things which dont exist [17:05:18 CEST] <cone-793> ffmpeg 03Andreas Cadhalpun 07release/2.4:8b9cfdc9270a: alsdec: check sample pointer range in revert_channel_correlation [17:05:19 CEST] <cone-793> ffmpeg 03Andreas Cadhalpun 07release/2.4:f7a6a6d59174: mpeg4videodec: only allow a positive length [17:05:20 CEST] <cone-793> ffmpeg 03Andreas Cadhalpun 07release/2.4:9a9379ff7fc3: bink: check vst->index_entries before using it [17:05:21 CEST] <cone-793> ffmpeg 03Vittorio Giovara 07release/2.4:ee3d4e2b0981: lavf: Reset global flag on deinit [17:05:22 CEST] <cone-793> ffmpeg 03Andreas Cadhalpun 07release/2.4:b16a6c670998: alsdec: only adapt order for positive max_order [17:05:23 CEST] <cone-793> ffmpeg 03Andreas Cadhalpun 07release/2.4:dfe37f2be202: alac: reject rice_limit 0 if compression is used [17:05:24 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:f6dd6b4fd28f: tests/fate-run: do not attempt to parse tiny_psnrs output if it failed [17:16:51 CEST] <cone-793> ffmpeg 03Michael Niedermayer 07release/2.4:73af011c7929: Update for FFmpeg 2.4.9 release [19:31:34 CEST] <Timothy_1u> michaelni: still doesn't work: http://fatebeta.ffmpeg.org/report/armv7l-panda-gcc4.4-armv4t-n2.4/201504242… [19:40:12 CEST] <michaelni> maybe i hadnt fixed that when it run, it says "19 hours ago" [20:22:13 CEST] <cone-861> ffmpeg 03Michael Niedermayer 07master:f4c311fef4d2: avcodec/mpeg12dec: Change mb decode tracing stuff to use ff_tlog() [20:38:32 CEST] <cone-861> ffmpeg 03Michael Niedermayer 07master:9d3357391cec: avcodec/mpeg12dec: Make ff_dlog() messages more clear [21:17:17 CEST] <vibr> kierank, i can't find in the source code what ffmpeg does when it's called with -f framecrc, i.e. what library functions (API functions?) it calls to compute and write the Adler 32 CRCs? [21:19:36 CEST] <jamrial> vibr: see libavformat/framecrcenc.c and libavutil/adler32.c [21:21:39 CEST] <vibr> jamrial, thanks, these are not part of the API, are they? (i conclude that from the naming conventions?) [21:21:59 CEST] <jamrial> av_adler32_* are [21:25:26 CEST] <vibr> av_adler32_update is the only one i see [21:25:38 CEST] <vibr> but i would need to mimick what ffmpeg -f framecrc does [21:25:50 CEST] <vibr> that is, call the framecrc muxer [21:26:25 CEST] <vibr> and i didn't find how to do that [21:28:05 CEST] <vibr> -mimick +mimic :) [21:28:50 CEST] <vibr> erm, to be more precise: call the framecrc muxer via the API [21:29:08 CEST] <vibr> ideally [00:00:00 CEST] --- Sun Apr 26 2015

1 0

ffmpeg.log.20150425
by burek 26 Apr '15

26 Apr '15

[01:09:29 CEST] <Prelude2004c> hey guys.. anyone know how to copy a trasnport stream from one place to anther without touching it ? everytime i do -c copy it shows up on the other side as just 1 program and it says " service01 " etc etc [01:09:42 CEST] <Prelude2004c> i want to just copy it and that's t [01:09:42 CEST] <Prelude2004c> it [02:42:01 CEST] <rex____> hi [02:42:06 CEST] <rex____> i have big videos [02:42:20 CEST] <rex____> i have big videos and want to convert many to hq sound. [02:42:28 CEST] <rex____> Where do i start from here or google [07:13:57 CEST] <techtopia> hello all [07:14:41 CEST] <techtopia> im trying to encode some video that is encoded at 29.97 fps and the video has four frames clean then two frames blended throughout [07:15:11 CEST] <techtopia> how can i remove those blended frames [07:15:20 CEST] <techtopia> or fix them so they are not blended [07:15:32 CEST] <techtopia> im not used to ntsc video [07:28:45 CEST] <shevy> guys... I have from a university lecture [07:28:53 CEST] <shevy> several "podcast" downloads; these files are stored in .m4v [07:29:04 CEST] <shevy> is this the same as .mp4? let me run ffmpeg to analyze... [07:29:34 CEST] <shevy> ffmpeg says that the video stream is `h264` and the audio stream is `aac`. [07:29:44 CEST] <shevy> I wonder if I could just rename .m4v to .mp4 [07:31:44 CEST] <techtopia> is it a fragment [07:31:48 CEST] <techtopia> or a complete video [07:32:20 CEST] <techtopia> it's basically mp4 but with protection [07:32:33 CEST] <techtopia> normaly they are fragmented and served in fragments [07:32:41 CEST] <shevy> aha [07:33:04 CEST] <shevy> they seem to work fine as standalones; each .m4v here was a lecture from another day [07:33:04 CEST] <techtopia> mp4 is just a container btw [07:33:08 CEST] <techtopia> the video is h264 [07:33:16 CEST] <shevy> I kind of want to just merge them all together into one huge long 10 hours lecture [07:33:22 CEST] <shevy> ok techtopia [07:33:26 CEST] <techtopia> if they play just keep them as m4v no point to rename them [07:33:37 CEST] <techtopia> unless you want to re encode them to x264 or somthing [07:33:54 CEST] <techtopia> oh [07:34:00 CEST] <techtopia> then you would concat them [07:34:09 CEST] <shevy> nah, the video quality does not matter anyway, I am mostly interested in the audio... though the teacher also shows some .pdf files on the beamer, so it's nice to retain some quality to decipher these [07:34:12 CEST] <techtopia> it's kinda awkward with ffmpeg [07:34:35 CEST] <shevy> yeah, it's the "concat:" thing right? that's ok too, I trust ffmpeg :) [07:34:35 CEST] <techtopia> what i do is use ffmpeg to encode video to either mp4 or mkv depending on hd or sd [07:34:48 CEST] <techtopia> then i use mkvmerge gui to merge them into a solid file for hd [07:34:58 CEST] <techtopia> or mp4box to merge them into a solid file for mp4 [07:35:33 CEST] <techtopia> so use ffmpeg encode them all out to x264 in an mp4 container [07:35:59 CEST] <techtopia> at the end, use mp4box to demux all the mp4's. then use mp4box to join all the video files [07:36:56 CEST] <techtopia> for the audio just do copy /b from a command line like "copy /b audio01.mp2 + audio02.mp2 + audio03.mp2 audio_out.mp2" [07:37:14 CEST] <techtopia> then use mp4box to mux the merged audio and video files [07:37:18 CEST] <techtopia> it's the best way [07:38:22 CEST] <techtopia> might be easier to you know just create a playlist for your media player :p [12:30:43 CEST] <micechal> Has anyone ever used ffmpeg to create captchas? [12:31:07 CEST] <micechal> How good of an idea is that? [15:24:55 CEST] <Prelude2004c> hey guys.. anyone know how to copy a trasnport stream from one place to anther without touching it ? everytime i do -c copy it shows up on the other side as just 1 program and it says " service01 " etc etc.. i just want to take the multicast from one place to another [17:06:35 CEST] <vladmariang> any guy here? [17:06:38 CEST] <vladmariang> hi [17:07:32 CEST] <vladmariang> I want to put a background back on the image and I use ffmpeg -i 1.mp4 -i background.jpg \-filter_complex "[0:v][1:v] overlay=0:0:enable='between(t,0,20)'" \-pix_fmt yuv420p -c:a copy \output.mp4 [17:07:50 CEST] <vladmariang> but is a problem with this command [17:08:08 CEST] <vladmariang> the video is under the image [17:08:45 CEST] <c_14> swap 0:v with 1:v [17:12:53 CEST] <vladmariang> [1:v][0:v] ? [17:13:38 CEST] <c_14> yep [17:14:45 CEST] <vladmariang> hmm, same output [17:15:52 CEST] <c_14> [1:v][0:v]overlay should put the picture on the bottom [17:32:55 CEST] <vladmariang> c_14: Do you know a command for make that? [17:33:08 CEST] <c_14> hmm? [17:36:49 CEST] <vladmariang> for put a background and center the video [17:38:46 CEST] <c_14> ffmpeg -i video -i background -filter_complex '[1:v][0:v]overlay=x=(W-w)/2:y=(H-h)/2[v]' -map '[v]' out.mkv [17:41:29 CEST] <vladmariang> Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height [17:42:44 CEST] <vladmariang> http://pastebin.com/kbg0JuCE [17:43:47 CEST] <c_14> The complete console output is usually helpful. Not just the error message. [17:45:28 CEST] <vladmariang> http://pastebin.com/8zQcE5ND [17:47:51 CEST] <c_14> ffmpeg -i video -i background -filter_complex '[1:v]scale=trunc(iw/2)*2:-2[tmp];[tmp][0:v]overlay=x=(W-w)/2:y=(H-h)/2[v]' -map '[v]' out.mkv [17:51:43 CEST] <trodis> im trying to stream opus from my pc over udp to the ffserver, the streaming looks ok, but i cant play it [17:52:04 CEST] <trodis> this is how i stream to the ffserver: ffmpeg -re -f pulse -i default -c:a libvorbis -b:a 64k -f mpegts udp://192.168.1.138:554 [17:52:34 CEST] <trodis> this is my ffserver.conf: http://ix.io/ew8 [17:52:52 CEST] <sfan5> ummm... [17:52:52 CEST] <trodis> and this is how i try to play it: mplayer udp://192.168.1.138:554 [17:53:17 CEST] <trodis> i mean im trying to stream with the opus codec [17:53:35 CEST] <sfan5> trodis: ffmpeg will ignore -c:a libvorbis and -b:a 64k [17:53:53 CEST] <trodis> because he takes the config from the server? [17:53:56 CEST] <sfan5> yes [17:54:09 CEST] <trodis> ok i guessed that because thats the "output" [17:54:19 CEST] <trodis> so will encode to what the output should be [17:54:38 CEST] <trodis> but anyway i have no clue how to configure my ffserver so i can stream using udp [17:54:42 CEST] <sfan5> also I don't think ffserver takes any input on udp [17:55:16 CEST] <trodis> what i want to achieve is how i can stream audio with latency as low as possible [17:55:41 CEST] <sfan5> trodis: try ffmpeg -re -f pulse -i default http://192.168.1.138/feed1.ffm [17:55:59 CEST] <vladmariang> @c_14 It works with png background but the video has < 1 second duration [17:56:16 CEST] <sfan5> trodis: http://192.168.1.138:8080/feed1.ffm i mean [17:56:20 CEST] <vladmariang> with mp4 format [17:56:38 CEST] <vladmariang> output [17:56:50 CEST] <c_14> vladmariang: add -loop 1 before the -i for the background [17:57:22 CEST] <c_14> and add eof_action=endall to the overlay [17:58:12 CEST] <c_14> that or shortest=1 [17:58:18 CEST] <sfan5> trodis: i just googled something and it seems that RTSPPort is for outputting streams but not for taking input [17:58:24 CEST] <trodis> sfan5: now i get connection refused [17:59:06 CEST] <trodis> [tcp @ 0x7fe0696fb720] Connection to tcp://192.168.1.138:80 failed: Connection refused [17:59:25 CEST] <sfan5> :8080 not :80 [17:59:29 CEST] <trodis> [tcp @ 0x7f75e4c89760] Connection to tcp://192.168.1.138:8080 failed: Connection refused [17:59:55 CEST] <sfan5> which ip does ffserver have? [18:00:14 CEST] <trodis> 192.168.1.138 [18:00:37 CEST] <sfan5> should've read your config.. [18:00:39 CEST] <sfan5> try :8090 [18:00:53 CEST] <trodis> ah shit sry my bad [18:01:00 CEST] <sfan5> nah, my fault [18:01:12 CEST] <trodis> i have so many raspberry pis here i was confused with the ports [18:01:51 CEST] <trodis> ok now im streaming but which protocol im i using i guess its tcp ? [18:01:54 CEST] <sfan5> yes [18:02:09 CEST] <sfan5> i don't think your can stream to ffserver using anything other than tcp [18:02:23 CEST] <trodis> ok that answered my question [18:02:27 CEST] <c_14> Well, technically it's http over tcp [18:02:28 CEST] <trodis> so whats the deal with rtp protocoll [18:02:56 CEST] <sfan5> mplayer http://192.168.1.138:8090/test.ogg should work [18:03:16 CEST] <sfan5> i don't know how rtsp works but i'd guess it's rtsp://192.168.1.138:554/test.ogg [18:04:30 CEST] <trodis> than whats the deal with all the udp stuff [18:04:36 CEST] <trodis> https://trac.ffmpeg.org/wiki/StreamingGuide#Pointtopointstreaming [18:05:07 CEST] <vladmariang> c_14: It works with -loop 1 but the console keeps running infinitely and no sound for the video [18:05:30 CEST] <c_14> like I said, add :shortest=1 to the options for the overlay filter [18:05:40 CEST] <c_14> And add -map 0:a as an output option [18:05:40 CEST] <vladmariang> I need to press control+c to stop [18:05:41 CEST] <sfan5> trodis: rtsp/udp is when you want to stream from one computer to another [18:05:50 CEST] <trodis> so without server? [18:05:58 CEST] <c_14> vladmariang: you can also press q to stop, but that should fix itself once you add the :shortest=1 [18:06:10 CEST] <sfan5> trodis: the usecase for ffserver is: you have one video and want to deliver it (in possibly mulitple sizes & formats) to many computers [18:06:18 CEST] <sfan5> nope, no ffserver needed for point-to-point streaming [18:06:46 CEST] <trodis> and my usecase is, i have a microphone and want to broadcast in LAN to 20 raspberry pis [18:06:59 CEST] <trodis> with latency as low as possible, like Mumble or TS [18:07:41 CEST] <sfan5> sounds like you need multicast [18:07:46 CEST] <trodis> so i thought i could use ffmpeg to capture mic input and send it to the ffserver, and the 20 raspberry pis would connect to the ffserver and listen to what im saying [18:07:55 CEST] <sfan5> that would work too [18:08:52 CEST] <trodis> yeah but multicast is hard to maintain, the speak should not be bothered during broadcast when something goes wrong, i guess having a sever gives me more controlle [18:13:19 CEST] <vladmariang> c_14: '[v]' is an argument? [18:13:39 CEST] <c_14> It's a filterpad [18:13:41 CEST] <__jack__> multicast is just a less-stupid broadcast [18:14:03 CEST] <vladmariang> c_14: -map 0:a '[v]' out.mp4 [18:14:16 CEST] <c_14> -map 0:a -map '[v]' [18:16:47 CEST] <vladmariang> c_14: For 1080p I need to create the image in this format? [18:17:06 CEST] <c_14> that or scale it first [18:18:34 CEST] <vladmariang> c_14: because I want to make full hd and to remove the black borders from the video http://i.imgur.com/LuwmiuB.png [18:19:00 CEST] <vladmariang> c_14: for make a video like https://www.youtube.com/watch?v=mPrwsHrsodA [18:19:24 CEST] <trodis> sfan5: how can i reduce the latency when im using ffmpeg to stream to the ffserver, on the client side im using ffplay -fflags nobuffer or mplayer -nocache [18:20:23 CEST] <c_14> vladmariang: if the video already has black borders, you'll need to crop. Otherwise just overlaying the video onto a larger background should work. [18:25:48 CEST] <sfan5> trodis: no idea, maybe a lower FileMaxSize in ffserver.conf [18:28:33 CEST] <vladmariang> c_14: the video is 720x720 [18:30:08 CEST] <vladmariang> c_14: I think I need to increase height [18:30:45 CEST] <vladmariang> c_14: https://www.youtube.com/watch?v=AFmRxUO4W5w [18:32:06 CEST] <c_14> First, you'll want to crop off those black bars. Then you'll want to scale the height of the video up to the height of the overlay, or scale the overlay down to the height of the video. [18:35:52 CEST] <vladmariang> c_14: the problem with the black borders is from the video. [18:38:19 CEST] <c_14> Which is why I suggested cropping it. [18:39:03 CEST] <vladmariang> c_14: Now the video is ok, but he is very small [18:39:14 CEST] <c_14> Which is why I suggested scaling. [18:40:21 CEST] <vladmariang> Haha, how [18:40:34 CEST] <c_14> What's your current command? [18:41:04 CEST] <vladmariang> http://pastebin.com/kMrhy2sz [18:41:38 CEST] <c_14> Did you already crop 1.mp4 [18:42:08 CEST] <vladmariang> Yes, https://www.youtube.com/watch?v=GBjiXFNvjvs [18:43:17 CEST] <jaggz> how can I scale an audio rate to get its final length to match that of a video? [18:43:34 CEST] <c_14> ffmpeg -i 1.mp4 -loop 1 -i background.jpg -filter_complex '[1:v]scale=trunc(iw/2)*2:-2[tmp];[0:v]scale=-1:1080[tmp0];[tmp][tmp0]overlay=shortest=1:x=(W-w)/2:y=(H-h)/2[v]' -strict -2 -map 0:a -map '[v]' out.mp4 [18:43:42 CEST] <c_14> ^ vladmariang [18:44:03 CEST] <jaggz> I used a screen recorder that, for some reason, recorded the audio 4 seconds shorter (faster) over 2 hours [18:44:23 CEST] <jaggz> it saves the video in a separate file from the audio [18:44:29 CEST] <vladmariang> c_14: guy, you are awesome [18:44:40 CEST] <c_14> jaggz: -af atempo=video_length/audio_length (should do it) [18:45:05 CEST] <jaggz> beautiful, thank you :) [18:45:25 CEST] <jaggz> put that after the -i audio and before the output filename? [18:45:44 CEST] <c_14> yeah, you'll have to actually probe the input files to get the durations though [18:46:31 CEST] <vladmariang> c_14: You are the creator of the ffmpeg? [18:47:04 CEST] <c_14> vladmariang: Nah [18:47:19 CEST] <jaggz> ffmpeg shows it nicely I believe? Video 01:54:52.94 audio 01:54:48.86 [18:47:47 CEST] <c_14> yeah. you'll probably want to convert that to seconds though so you can divide easily [18:49:37 CEST] Action: jaggz changes script from bash to perl to parse the output easier [18:50:25 CEST] Action: c_14 actually wrote a bash script once to parse HH:MM:SS.MS. [18:50:29 CEST] <c_14> It's great (not really). [18:51:17 CEST] <vladmariang> c_14: There is another method for add a text to a video instead subtitle? [18:51:27 CEST] <c_14> drawtext ? [18:51:38 CEST] <c_14> https://ffmpeg.org/ffmpeg-filters.html#drawtext-1 [18:51:41 CEST] <vladmariang> c_14: Something like https://www.youtube.com/watch?v=55uBuHQzSA0 [18:52:03 CEST] <c_14> drawtext should be able to do everything except for that spectrum display [18:52:28 CEST] <jaggz> c_14, :) bash's expansion capabilities makes things easier nowadays, but still... [19:02:17 CEST] <jaggz> c_14: in testing, I'm skipping some intro material in the video where I can't tell if it's in sync.. is that too complicated to handle along with the rate issue? [19:02:19 CEST] <jaggz> ffmpeg.exe -y -ss $start -i "$1" -ss $start -i "$2" -c:v libxvid -b:v 100k -t 00:00:30 "$3" [19:03:04 CEST] <c_14> Just subtract $start from both durations? [19:03:04 CEST] <jaggz> $1 is the video, $2 audio.. figured I'd skip 30 minutes in (-ss "00:30:00") and run 30 seconds .. it's a lecture [19:03:18 CEST] <c_14> though [19:03:30 CEST] <c_14> hmm [19:04:20 CEST] <jaggz> it was just to save time while figuring out the other stuff.. I could just process from the beginning.. skipping that amount of time in both files might be an issue with the different rates... [19:04:36 CEST] <c_14> You'd probably have to reencode the audio with the atempo filter, then test with that file [19:04:48 CEST] <c_14> Reencoding only the audio shouldn't take long. [19:05:19 CEST] <c_14> Skipping in the video shouldn't be an issue, but if you skip in the audio and then try applying the rate filter on that, it'll be stretched incorrectly. [19:05:24 CEST] <jaggz> hey that's an interesting idea [19:06:31 CEST] <jaggz> and fast [19:06:46 CEST] <jaggz> oh you said that :) [19:07:02 CEST] <vladmariang> c_14: There is possibility to draw a text like subtitle, for a specified time? [19:07:03 CEST] <jaggz> are you carbon_14? [19:07:11 CEST] <jaggz> ignore me.. off topic.. [19:07:35 CEST] <vladmariang> c_14: And for use like text.txt [19:07:49 CEST] <vladmariang> Because I want to generate dynamically [19:08:09 CEST] <vladmariang> and save to a file and load [19:08:24 CEST] <c_14> Use the textfile option of the drawtext filter [19:09:56 CEST] <randomdue> hello [19:10:52 CEST] <jaggz> Oops.. it's shorter than it used to be. Original audio duration: 01:54:48.86, new: 01:54:44.81 [19:11:04 CEST] <randomdue> an ntsc video at 29.97fps has clean and blended frames in the patern 3:2:3:2:2 [19:11:27 CEST] <randomdue> which is what happens when the convert 25 fps pal for broadcast in ntsc areas [19:11:40 CEST] <randomdue> how can i correctly restore it to 25fps [19:11:41 CEST] <c_14> jaggz: eh, right. invert the video_duration and audio_duration. A positive atempo speeds up the audio [19:11:55 CEST] <jaggz> not that I like experimenting without knowing.. but I'll make it atempo = alen / vlen instead [19:13:02 CEST] <jaggz> randomdue: what's 3:2:3:2:2 indicate? (just for my own knowledge -- I'm unfamiliar with this) [19:15:31 CEST] <jaggz> c_14, perfect. lengths match to 00.01s [19:15:42 CEST] <vladmariang> c_14: Error opening filters! My ffmpeg is very clear, without any filter [19:16:48 CEST] <c_14> randomdue: if the pattern is exact, you can probably use the new detelecine filter. Otherwise you'll need to use one of the other ivtc filters like pullup or fieldmatch/decimate [19:17:07 CEST] <vladmariang> c_14: http://pastebin.com/kFR4T9iK [19:17:27 CEST] <jaggz> c_14, thanks for helping; me and others. [19:18:25 CEST] <c_14> vladmariang: you need an ffmpeg compiled with --enable-libfreetype [19:24:46 CEST] <vladmariang> c_14: There is a ffmpeg file already compiled with those things? [19:24:54 CEST] <c_14> http://johnvansickle.com/ffmpeg/ [19:30:49 CEST] <randomdue> [18:13:02] <jaggz> [19:31:09 CEST] <randomdue> pal 25fps video, broadcast at 29.97 fps [19:31:27 CEST] <randomdue> causes every 5th and 6th frame to be blended [19:31:39 CEST] <randomdue> im trying to remove the blends and restore it to 25fps [20:16:08 CEST] <vladmariang> c_14: Thanks :X [20:16:11 CEST] <vladmariang> for all [20:49:53 CEST] <mariangabriel> c_14: There is a way to reduce the time when I want to add a background? [20:50:05 CEST] <mariangabriel> frame= 447 fps=6.0 q=28.0 size= 8164kB time=00:00:17.83 bitrate=3748.8kbits/ [20:50:18 CEST] <mariangabriel> because I wait for every frame [20:50:34 CEST] <c_14> hmm? [20:51:14 CEST] <mariangabriel> http://pastebin.com/e53BNkcP [20:52:29 CEST] <mariangabriel> how ca I reduce the compilation time [20:52:32 CEST] <mariangabriel> can* [20:53:51 CEST] <mariangabriel> I think is impossible [20:56:03 CEST] <__jack__> mariangabriel: compilation ? you mean: encoding ? [20:56:09 CEST] <mariangabriel> yes [20:56:18 CEST] <mariangabriel> the process for adding background [20:57:43 CEST] <vlops> kierank, i can't find in the source code what ffmpeg does when it's called with -f framecrc, i.e. what library functions (API functions?) it calls to compute and write the Adler 32 CRCs? [21:09:20 CEST] <c_14> mariangabriel: you can add -preset fast/veryfast/ultrafast [21:11:38 CEST] <mariangabriel> this preset decrease the quality > [21:11:39 CEST] <mariangabriel> ? [21:12:10 CEST] <c_14> In your case it should mainly just make the output larger. [21:18:00 CEST] <Sneakyghost> Gday everyone. I would need help with converting some raw file containing an image to a windows viewable output. Can this be asked here? [21:20:02 CEST] <c_14> sure [21:21:31 CEST] <Sneakyghost> splash screen partition from a HTC One M9 doesnt convert using a command line that used to work on the predecessor phone M8, I cannot figure out the error. The two phones are very similar and the raw image actually shouldn't have changed [21:21:51 CEST] <Sneakyghost> ffmpeg -f rawvideo -pix_fmt rgb565 -s 1080x1920 -i mmcblk0p15 -f image2 splash1.png this worked on the M8 [21:22:20 CEST] <Sneakyghost> but on M9, it returns: [image2 @ 02fa82d0] Could not get frame filename from pattern av_interleaved_write_frame(): Input/output error [21:23:09 CEST] <c_14> try with -frames:v 1 [21:23:33 CEST] <Sneakyghost> before input or output? [21:24:51 CEST] <c_14> output [21:26:33 CEST] <Sneakyghost> unrecognized option frames:v [21:28:52 CEST] <c_14> try -vframes 1 [21:31:07 CEST] <Sneakyghost> man that did it [21:31:26 CEST] <Sneakyghost> whats the vframe option do? [21:31:45 CEST] <Sneakyghost> is it in the documentation page? [21:32:17 CEST] <Sneakyghost> Oh i see it [21:32:23 CEST] <Sneakyghost> it specifies the number of frames [21:32:26 CEST] <Sneakyghost> just that? [21:32:35 CEST] <Sneakyghost> I wonder why it worked on the previous phone's raw image [21:32:44 CEST] <Sneakyghost> without specifiying number of frames [21:33:23 CEST] <c_14> Probably because the new one looked like it had more frames or something. It might also just be a different version of ffmpeg. [21:34:18 CEST] <Sneakyghost> totally overwhelmed with ffmpeg. Thank you so much for helping me. I shall creep back to my cave now. [00:00:00 CEST] --- Sun Apr 26 2015

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