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January 2012

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ffmpeg-devel.log.20120105
by burek 06 Jan '12

06 Jan '12

[00:27] <pasteeater> that's what i told the op too [00:48] <pasteeater> another spam account on trac: matildaharris [00:48] <pasteeater> she's a witch! [00:51] <ubitux> buuuurrrnnn [01:00] <michaelni> pasteeater, how do u detect trac spamers ? [01:01] <pasteeater> when i delete shit like "Phuket property" from the main page and then i look at page history. [01:03] <michaelni> humpf, user deleted [01:03] <pasteeater> thanks [01:05] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r1171d938af 10ffmpeg/libavformat/utils.c: (log message trimmed) [01:05] <CIA-101> ffmpeg: lavf: revert commit bb99ae3ae924c942a634bec7711ec7ee11c38eb9 [01:05] <CIA-101> ffmpeg: Author: Michael Niedermayer <michaelni(a)gmx.at> [01:05] <CIA-101> ffmpeg: Date: Thu Nov 3 22:38:10 2011 +0100 [01:05] <CIA-101> ffmpeg: lavf: fix null pointer dereference in rdt [01:05] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:05] <CIA-101> ffmpeg: This is no longer needed and causes various problems with RTSP [01:05] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * rad8e3304f7 10ffmpeg/libavformat/utils.c: [01:05] <CIA-101> ffmpeg: lavf: use avg_frame_rate and packet number to exit find_stream_info [01:05] <CIA-101> ffmpeg: qatar does this too but clobbers the AVPacket.duration by approximate [01:05] <CIA-101> ffmpeg: values. [01:05] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:05] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * rc4eec85a1f 10ffmpeg/libavformat/rmdec.c: [01:05] <CIA-101> ffmpeg: Revert "rmdec: Avoid allocating huge packets" [01:05] <CIA-101> ffmpeg: This reverts commit 66f71f3b5e2e7b6e5049bd9831c371e16aff8a53. [01:05] <CIA-101> ffmpeg: This causes regressions with RDT. [01:13] <CIA-101> ffmpeg: 03Oana Stratulat 07release/0.9 * r32b62c7d6a 10ffmpeg/libavformat/electronicarts.c: [01:13] <CIA-101> ffmpeg: electronicarts: Fix division by zero. Fixes Ticket #793 [01:13] <CIA-101> ffmpeg: Signed-off-by: Oana Stratulat <oanaandreeastratulat(a)gmail.com> [01:13] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:13] <CIA-101> ffmpeg: (cherry picked from commit 0d3a51e5d279dd2a56c81ba7a81a70128c5a7545) [01:13] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:13] <CIA-101> ffmpeg: 03Michael Niedermayer 07release/0.9 * r9f9b6388a3 10ffmpeg/libavformat/electronicarts.c: [01:13] <CIA-101> ffmpeg: electronicarts: Pass error through ea_read_header(). [01:13] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:13] <CIA-101> ffmpeg: (cherry picked from commit f37174bc1913d534913e8825f1609a637cc4b364) [01:13] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:13] <CIA-101> ffmpeg: 03Michael Niedermayer 07release/0.9 * r66569b375c 10ffmpeg/libavcodec/mjpegdec.c: [01:13] <CIA-101> ffmpeg: jpegdec: Fix vlc table check for progressive jpegs. [01:13] <CIA-101> ffmpeg: Fixes Ticket834 [01:13] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:13] <CIA-101> ffmpeg: (cherry picked from commit 15b219fae9da1691dfb264f51637805e1ca63d1a) [01:13] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:13] <CIA-101> ffmpeg: 03Chris Evans 07release/0.9 * r49b8709870 10ffmpeg/libavformat/matroskadec.c: (log message trimmed) [01:13] <CIA-101> ffmpeg: mkv: Fix a bug where a pointer was cached to an array that might later move due to [01:13] <CIA-101> ffmpeg: a realloc() [01:13] <CIA-101> ffmpeg: BUG=100492 [01:13] <CIA-101> ffmpeg: Review URL: http://codereview.chromium.org/8366004 [01:13] <CIA-101> ffmpeg: Fixes: 1 of 2 for CVE-2011-3893 [01:13] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:13] <CIA-101> ffmpeg: 03Chris Evans 07release/0.9 * r6a3fed8749 10ffmpeg/libavcodec/vorbis.c: (log message trimmed) [01:13] <CIA-101> ffmpeg: ogg: Avoid the possibility to read out-of-bounds of a static global array in Vorbis [01:13] <CIA-101> ffmpeg: Review URL: http://codereview.chromium.org/8365014 [01:13] <CIA-101> ffmpeg: This fixes 25% of CVE-2011-3893 [01:13] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:13] <CIA-101> (55 lines omitted) [01:29] <OanaStratulat> michaelni: how can i get the latest src files without making clone again? [01:33] <ubitux> like i said, you put your changes in a branch, go back to master, and run git pull [01:38] <OanaStratulat> ubitux: have some time to discuss about https://ffmpeg.org/trac/ffmpeg/ticket/599 ? [01:45] <OanaStratulat> michaelni: ping [01:57] <ubitux> i'm not sure this will be easy to fix [01:58] <OanaStratulat> ubitux: why is that? [01:58] <OanaStratulat> ubitux: how is the duration calculated ? [01:59] <ubitux> i guess the duration is written in the container (mov) but it should be read (calculated?) from the ts in the codec? [01:59] <ubitux> i'm not familiar with that stuff at all [02:00] <OanaStratulat> i cannot find another easy bug on trac :( [02:00] <OanaStratulat> suppose i must create one and then solve it [02:05] <michaelni> OanaStratulat, yes thats an option, easy bugs tend to be fixed quickly [02:06] <michaelni> about git, it was git reset --hard origin [02:06] <OanaStratulat> so back to the drawing board, zzufing files [02:06] <OanaStratulat> yes [02:06] <OanaStratulat> found that [02:07] <kierank> just don't write the duration in the mov [02:08] <kierank> i mean don't use it for anything important [02:09] <OanaStratulat> michaelni: can you give me an idea on how to automatically test some files with zzuf ? [02:10] <OanaStratulat> michaelni: not to input each file [02:19] <CIA-101> ffmpeg: 03Jonathan Nieder 07master * r96d8e18a98 10ffmpeg/doc/APIchanges: [02:19] <CIA-101> ffmpeg: APIchanges: mention cleaned up header includes [02:19] <CIA-101> ffmpeg: Fixes bug 188. [02:19] <CIA-101> ffmpeg: Signed-off-by: Jonathan Nieder <jrnieder(a)gmail.com> [02:19] <CIA-101> ffmpeg: Signed-off-by: Janne Grunau <janne-libav(a)jannau.net> [02:19] <CIA-101> ffmpeg: 03Justin Ruggles 07master * rc98c6e18d8 10ffmpeg/libavcodec/adxenc.c: adxenc: remove unneeded log message [02:19] <CIA-101> ffmpeg: 03Justin Ruggles 07master * ra85ab8ad45 10ffmpeg/libavcodec/adxenc.c: adxenc: cosmetics: pretty-printing [02:19] <CIA-101> ffmpeg: 03Justin Ruggles 07master * re3d4f59f95 10ffmpeg/libavcodec/adxenc.c: adxenc: remove unneeded comments [02:19] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r25edfc88e3 10ffmpeg/libavcodec/adxenc.c: adxenc: change some data types [02:19] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r656e606cae 10ffmpeg/libavcodec/adxenc.c: adxenc: use BLOCK_SIZE and BLOCK_SAMPLES macros [02:19] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r6c117bd8e0 10ffmpeg/libavcodec/adxenc.c: [02:19] <CIA-101> ffmpeg: adxenc: remove unneeded loops [02:19] <CIA-101> ffmpeg: avctx->frame_size is 32, so that is how many samples we process per call. [02:19] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r613668210b 10ffmpeg/libavcodec/adxenc.c: adxenc: avoid stereo deinterleaving [02:19] <CIA-101> ffmpeg: 03Justin Ruggles 07master * rf1be41c63d 10ffmpeg/libavcodec/adxenc.c: adxenc: use a loop to encode each channel [02:19] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r1fb47728cd 10ffmpeg/libavcodec/adxenc.c: [02:19] <CIA-101> ffmpeg: adxenc: use bytestream functions for header writing. [02:19] <CIA-101> ffmpeg: 03Ronald S. Bultje 07master * r400ba1d735 10ffmpeg/libavcodec/h264.c: [02:19] <CIA-101> ffmpeg: h264: return index in buffer on end-of-sequence. [02:19] <CIA-101> ffmpeg: Fixes hangs if the last packet contains an end-of-sequence NAL unit, [02:19] <CIA-101> ffmpeg: bug 158. [02:19] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r3e57573fce 10ffmpeg/tests/ (codec-regression.sh ref/acodec/adpcm_adx): fate: add ADX encoding/decoding test [02:19] <CIA-101> ffmpeg: 03Diego Biurrun 07master * r54e68fb3b8 10ffmpeg/libavcodec/ (dnxhdenc.c h264.c): [02:19] <CIA-101> ffmpeg: Merge some declarations and initializations. [02:19] <CIA-101> ffmpeg: This fixes compilation failures related to START_TIMER/STOP_TIMER macros and [02:19] <CIA-101> ffmpeg: -Werror=declaration-after-statement. START_TIMER declares variables and thus [02:19] <CIA-101> ffmpeg: may not be placed after statements outside of a new block. [02:19] <CIA-101> ffmpeg: 03Diego Biurrun 07master * r9dec55748c 10ffmpeg/libavcodec/ (cabac.c cabac.h): cabac: drop unused and disabled get_cabac_u() / get_cabac_ueg() functions [02:19] <CIA-101> ffmpeg: 03Diego Biurrun 07master * re451c26c5f 10ffmpeg/libavcodec/ (cabac.c cabac.h): cabac: drop unused STRICT_LIMITS code branch [02:19] <CIA-101> ffmpeg: 03Ronald S. Bultje 07master * r6b3995ccd1 10ffmpeg/libswscale/swscale.c: swscale: remove obsolete comment. [02:19] <CIA-101> ffmpeg: 03Ronald S. Bultje 07master * r6ea64339c5 10ffmpeg/libswscale/ (Makefile x86/output.asm x86/scale.asm): [02:19] <CIA-101> ffmpeg: v4l2: simplify away io_method [02:19] <CIA-101> (14 lines omitted) [02:19] <CIA-101> ffmpeg: 03Anton Khirnov 07master * re8c04f6240 10ffmpeg/ (avconv.c doc/avconv.texi): avconv: add symbolic names for -vsync parameters [02:19] <CIA-101> ffmpeg: 03Ronald S. Bultje 07master * rf910dbcdb0 10ffmpeg/libswscale/Makefile: swscale: add MMX files to MMX-OBJS instead of OBJS. [02:19] <CIA-101> ffmpeg: 03Justin Ruggles 07master * rcc40c056d0 10ffmpeg/libavcodec/adxenc.c: adxenc: log an error message and return AVERROR(EINVAL) for invalid channels [02:19] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r7f83db3124 10ffmpeg/: (log message trimmed) [02:19] <CIA-101> ffmpeg: Merge remote-tracking branch 'qatar/master' [02:19] <CIA-101> ffmpeg: * qatar/master: (46 commits) [02:19] <CIA-101> ffmpeg: mtv: Make sure audio_subsegments is not 0 [02:19] <CIA-101> ffmpeg: v4l2: use V4L2_FMT_FLAG_EMULATED only if it is defined [02:19] <CIA-101> ffmpeg: avconv: add symbolic names for -vsync parameters [02:19] <CIA-101> ffmpeg: flvdec: Fix compiler warning for uninitialized variables [02:19] <CIA-101> ffmpeg: 03Shitiz Garg 07master * rfeb15cee5e 10ffmpeg/libavformat/mtv.c: [02:19] <CIA-101> ffmpeg: mtv: Make sure audio_subsegments is not 0 [02:19] <CIA-101> ffmpeg: audio_subsegments would be 0 and cause floating point exceptions [02:19] <CIA-101> ffmpeg: Fixes bugzilla #144 [02:19] <CIA-101> ffmpeg: Signed-off-by: Justin Ruggles <justin.ruggles(a)gmail.com> [02:19] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r779ef255e6 10ffmpeg/ (5 files in 3 dirs): adx: add an adx muxer [02:19] <CIA-101> ffmpeg: 03Janne Grunau 07master * rf13a9ca906 10ffmpeg/libavdevice/v4l2.c: [02:19] <CIA-101> ffmpeg: v4l2: use V4L2_FMT_FLAG_EMULATED only if it is defined [02:19] <CIA-101> ffmpeg: V4L2_FMT_FLAG_EMULATED was added in 2.6.32. [02:19] <CIA-101> ffmpeg: 03Ronald S. Bultje 07master * r9ea3501d87 10ffmpeg/libswscale/ (swscale.c utils.c): [02:19] <CIA-101> ffmpeg: swscale: don't show full-chroma-int warning for non-RGB output. [02:19] <CIA-101> ffmpeg: Non-RGB output always uses full chroma interpolation. [02:19] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r754ebd1a5b 10ffmpeg/libavcodec/adxenc.c: adxenc: check output buffer size before writing [02:19] <CIA-101> ffmpeg: 03Jean First 07master * r2df73eefb4 10ffmpeg/libavformat/flvdec.c: [02:19] <CIA-101> ffmpeg: flvdec: Fix compiler warning for uninitialized variables [02:19] <CIA-101> ffmpeg: These can't be used uninitialized in practice, but the [02:19] <CIA-101> ffmpeg: compiler doesn't realize it. [02:20] <CIA-101> (35 lines omitted) [02:22] <michaelni> OanaStratulat, you can write a shell script to run zzuf ove many files [02:23] <OanaStratulat> michaelni: and how can i do when i have a crash to make a file with the output? [02:23] <OanaStratulat> michaelni: not with all the files even if they succeeded the tests [02:31] <michaelni> zzuf has a different exit code depending on what happened [02:32] <OanaStratulat> anyway 0 is for all tests passed i think no? [02:33] <ubitux> don't we have trac issues for windows fate instance? :) [02:34] <OanaStratulat> michaelni: do you use osx ? [02:37] <michaelni> no osx here [02:38] <OanaStratulat> i have a question not related to ffmpeg [02:38] <OanaStratulat> i have downloaded zzuf and i have a patch how can i apply it ? [02:38] <OanaStratulat> the patch fixes http://www.google.ro/search?aq=f&ix=hca&sourceid=chrome&ie=UTF-8&q=zzuf+mem… [02:43] <michaelni> if you have the zzuf source, "patch -p(some small number) <patchfile" should work [02:44] <michaelni> you can try with --dry-run to see if it would work [02:47] <OanaStratulat> ok [02:47] <OanaStratulat> thx [02:47] <OanaStratulat> anyway i`m closing the night here [02:47] <OanaStratulat> good night everybody [03:04] <CIA-101> ffmpeg: 03John Brooks 07release/0.9 * rfc9c5ad9ea 10ffmpeg/libavcodec/rv34.c: (log message trimmed) [03:04] <CIA-101> ffmpeg: rv34: fix invalid memory access for small video dimensions [03:04] <CIA-101> ffmpeg: For small video dimensions calculations of the upper bound for pixel [03:04] <CIA-101> ffmpeg: access may result in negative value. Using an unsigned comparison [03:04] <CIA-101> ffmpeg: works only if the greater operand is non-negative. This is fixed by [03:04] <CIA-101> ffmpeg: doing edge emulation explicitly for such conditions. [03:04] <CIA-101> ffmpeg: Signed-off-by: Kostya Shishkov <kostya.shishkov(a)gmail.com> [03:04] <CIA-101> ffmpeg: 03Michael Niedermayer 07release/0.9 * r601e648458 10ffmpeg/ (Doxyfile RELEASE VERSION): [03:04] <CIA-101> ffmpeg: Update for 0.9.1 [03:04] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:04] <CIA-101> ffmpeg: 03John Brooks 07release/0.9 * re540446f6c 10ffmpeg/libavcodec/vc1dec.c: (log message trimmed) [03:04] <CIA-101> ffmpeg: vc1dec: fix invalid memory access for small video dimensions [03:04] <CIA-101> ffmpeg: For small video dimensions, these calculations of the upper bound [03:04] <CIA-101> ffmpeg: for pixel access may have a negative result. Using an unsigned [03:04] <CIA-101> ffmpeg: comparison to bound a potentially negative value only works if [03:04] <CIA-101> ffmpeg: the greater operand is non-negative. Fixed by doing edge emulation [03:04] <CIA-101> ffmpeg: when the upper bound is probably negative, everywhere that this [06:41] <chrisballinger> hey guys [06:42] <chrisballinger> does anyone have any experience with the libavformat code for decoding Core Audio (.caf) files that contain AAC? [06:42] <chrisballinger> http://www.hydrogenaudio.org/forums/index.php?showtopic=89889 [06:43] <chrisballinger> sorry if this is the wrong channel but i figure i'd find a developer here who was perhaps somewhat familiar with that specific code [11:36] <silverrocker> I have some questions concerning Google Summer of Code and ffmpeg project proposals. Is there someone specific I need to go to with my questions? [11:38] <av500> you are a tad late [11:38] <av500> or very early [11:40] <silverrocker> av500: hah, well actaully I'm only partially intressted in Google SoC. I have my bachelor thesis coming up next month and I need to decide on what I'm going to do specifically. I have great intressed in multimedia compression and everything around that and my would-be-mentor suggested that I would take a look at the list of SoC to get some idea's. [11:41] <silverrocker> Also, the idea would be to contribute to the ffmpeg project but this would be my first time doing something like that so I don't really have a clue how to jump into the ffmpeg code which is actaully why I am here. [11:46] <av500> git clone it [11:52] <silverrocker> av500, But where do I start when I cloned it [11:53] <av500> ./configure [11:53] <av500> make [11:53] <av500> where to start depends on what you want to do [11:54] <av500> you are looking at: http://wiki.multimedia.cx/index.php?title=FFmpeg_/_Libav_Summer_Of_Code_2011 [11:54] <av500> i guess [11:56] <silverrocker> av500, I have looked at that list but it is still unclear to me [11:58] <av500> what is your background? [12:01] <av500> silverrocker: also, I would advise you find a project that your mentor can really mentor you with [12:01] <av500> you should not rely on IRC to do that for you [12:02] <av500> if your would be mentor is new to ffmpeg, I would not do it [12:04] <silverrocker> av500, I'm currently in my 3rd bachelor year of computer science and I have had some video compression, math and of course C (and C++). I think these are the most relevant for what I would be doing [12:08] <av500> thats fine [12:08] <av500> still you need a mentor that is not IRC [12:08] <av500> IRC will mentor you for GSOC [12:09] <av500> I see a lot of students ending up in IRC with various projects where the mentor is unable to help [12:09] <av500> most don't go far in the end [12:30] <OanaStratulat> michaelni: ping [12:33] <michaelni> OanaStratulat, pong [12:33] <OanaStratulat> i have found a bug, can you check it if i supply the file? [12:34] <michaelni> sure [12:34] <OanaStratulat> ok just a moment [12:36] <OanaStratulat> michaelni: http://ptp-roman.org/crash4XM here is the file, when i valgrind it i found seg11 [12:38] <michaelni> Invalid read of size 8 here [12:39] <michaelni> so yes, its a bug [12:39] <OanaStratulat> do you fink it is hard to fix? [12:41] <michaelni> probably alot easier than the mov muxer one [12:41] <michaelni> but i didnt look very carefull [12:41] <OanaStratulat> think [12:41] <silverrocker> av500, Thank you for the information. My promotor has experience with the code of ffmpeg and has done research related to that so I think I'm ok there but I'm not sure if I'm adequate for the task really. [12:41] <av500> ok [12:42] <av500> well, he should be able to tell you if you can do the task or not [12:42] <av500> your mentor is somebody we know? [12:42] <silverrocker> av500, ok thanks I will have to ask him for more information then before jumping in [12:43] <OanaStratulat> michaelni: i see in gdb that the error is in memmove$VARIANT$sse42 () [12:43] <OanaStratulat> but it does not supply any .c file [12:47] <michaelni> ==15701== at 0x4C2A327: memcpy (mc_replace_strmem.c:635) [12:47] <michaelni> ==15701== by 0x84DE21: pcm_encode_frame (pcm.c:171) [12:48] <OanaStratulat> michaelni: an i check if the var i want to memcpy are assigned? [12:48] <OanaStratulat> can* [12:56] <michaelni> OanaStratulat, not sure i understand you [12:57] <OanaStratulat> michaelni: the segfault is because i want to access a part of memory that is not assigned to my program no? [13:01] <michaelni> yes [13:01] <OanaStratulat> michaelni: how can i verify that those parts of memory are assigned to my program ? [13:02] <michaelni> access them, valgrind will tell ;) [13:03] <OanaStratulat> michaelni: put i want to check them and if they are unassigned return an AVERR, like the bug with FPE i have fixed [13:04] <OanaStratulat> michaelni: i want to implement this check in ffmpeg not to use external programs like valgrind [13:06] <michaelni> its not easy to check that, also there must be something wrong if the part you read is not "assigned" to you [13:20] <michaelni> maybe you should look at what parameters get passed into the pcm_encode_frame() function ? [13:21] <michaelni> are the pointers non null, the size sane ? [13:43] <ubitux> 13:26:17 <OanaStratulat> http://pastebin.com/W8BePW8k [13:43] <ubitux> 13:26:28 <OanaStratulat> this is my output [13:43] <ubitux> the encode is fed with an invalid buf_size [13:43] <ubitux> do_audio_out in ffmpeg.c has a strange size value [13:44] <ubitux> afaik this is where OanaStratulat is looking for [13:44] <OanaStratulat> yes i am looking in that file [13:49] <OanaStratulat> as i see in do_audio_out audio_buf_size; is my buf size no? [13:49] <ubitux> first, did you confirm the size_out value is also wrong when calling avcodec_encode_audio in ffmpeg.c? [13:51] <OanaStratulat> yes [13:51] <OanaStratulat> size is also too big [13:51] <OanaStratulat> and then going negative [13:52] <ubitux> you mean size at the beginning of do_audio_out? [13:52] <OanaStratulat> http://pastebin.com/5bwqtRFC [13:52] <ubitux> int size = decoded_frame->nb_samples * dec->channels * isize; [13:52] -OanaStratulat:#ffmpeg-devel- ret = avcodec_encode_audio(enc, audio_out, size_out,(short *)buftmp); [13:52] <ubitux> if size is too big, check the 3 values here [13:52] <OanaStratulat> the size there [13:52] <OanaStratulat> i checked [13:53] <ubitux> which one of the 3 values is wrong then? [13:53] <OanaStratulat> wait what size are you talking about from avcodec_encode_audio ? [13:53] <ubitux> "size" [13:53] <OanaStratulat> or the int size you gave me? [13:54] <ubitux> it seems there is a size_out = size [13:54] <ubitux> so look at size at the beginning of the function [13:54] <ubitux> and check if indeed size is wrong, and if so, which on the 3 values is wrong (nb samples? channels? or isize?) [13:55] <OanaStratulat> wrong meaning very big or going negative ? [13:55] <ubitux> i guess very big is not fine either [13:55] <OanaStratulat> 20052 5884 46300 are good values for size [13:55] <OanaStratulat> ? [13:55] <ubitux> i guess yes [13:55] <OanaStratulat> ok [13:56] <OanaStratulat> at the end i have size : 268454888 [13:56] <OanaStratulat> this is bad [13:56] <av500> not on your bank account [13:56] <ubitux> thanks for your help av500 :D [13:56] <ubitux> OanaStratulat: then look at int size = decoded_frame->nb_samples * dec->channels * isize; [13:56] <ubitux> and check the 3 variables here [13:56] <ubitux> one or more may be wrong [13:57] <OanaStratulat> yes [13:57] <OanaStratulat> nb_samples what type is it? [13:57] <av500> the number of samples for frame [13:57] <av500> per frame [13:57] <ubitux> OanaStratulat: %d should be fine to print all of them [13:59] -OanaStratulat:#ffmpeg-devel- size : 268454888 [13:59] -OanaStratulat:#ffmpeg-devel- nb_samples : 67113722 [13:59] -OanaStratulat:#ffmpeg-devel- dec->channels : 2 [13:59] -OanaStratulat:#ffmpeg-devel- isize : 2 [13:59] <OanaStratulat> at the end ^ [14:00] <OanaStratulat> so nb_samples is too big [14:00] <OanaStratulat> in another output nb_sampels is 4850 and values like this [14:00] <ubitux> yes, so the decode of the frame gave an invalid number of samples :p [14:00] <ubitux> not sure where this is defined& [14:01] <OanaStratulat> mhmh [14:01] <OanaStratulat> so don`t know where should i check next ? [14:02] <ubitux> not yet precisely, i'm discovering the code in the same time ;) [14:02] <OanaStratulat> aa ok [14:03] <ubitux> avcodec_decode_audio4 will set the number of samples (obviously) [14:03] <ubitux> now you have to check in the decode audio callback why it is set like this [14:03] <OanaStratulat> as i see transcode_audio calls do_audio_out [14:07] <ubitux> i guess the frame content is set in libavcodec/4xm.c [14:08] <ubitux> but i don't know where the number of samples is specified& [14:11] <OanaStratulat> michaelni: any idea for this? [14:11] <av500> OanaStratulat: its called "staring at code" [14:11] <av500> this value is being set somewhere [14:11] <av500> add some printfs [14:22] <OanaStratulat> ubitux: isn`t avcodec_decode_audio4 defined in avcodec ? [14:26] <ubitux> avcodec_decode_audio4 should call the libavcodec/4xm.c decode callback [14:27] <ubitux> OanaStratulat: why are you asking about avcodec? [14:27] <OanaStratulat> i see that avcodec_decode_audio4 is in avcodec.h [14:28] <ubitux> yes the prototype is here, the code is in libavcodec/utils.c [14:28] <michaelni> the size_out passed to avcodec_encode_audio() is negative [14:29] <ubitux> michaelni: yes, but this is because the number of samples decoded is invalid [14:29] <ubitux> afaiu [14:29] <michaelni> it should not become negative ever [14:30] <OanaStratulat> so when the size_out is negative it should return invalid data ? [14:30] <michaelni> id say whatever makes it negative is buggy [14:32] <OanaStratulat> so size out is made by size_out = swr_convert(...) [14:33] <OanaStratulat> is there where it is initialized ? [14:43] <OanaStratulat> as i see it size_out is being made by int swr_convert [15:33] <OanaStratulat> michaelni: ping [15:35] <michaelni> OanaStratulat, pong [15:36] <OanaStratulat> regarding the bug, as i see it size_out is being made by int swr_convert [15:36] <michaelni> is it negative there ? [15:36] <OanaStratulat> am i right? [15:37] <OanaStratulat> yes size_out goes negative then segafult [15:37] <michaelni> where does it go negative ? [15:38] <OanaStratulat> http://pastebin.com/H8HPq5e9 [15:38] <OanaStratulat> it goes like 10 loops then it gives a negative value [15:39] <michaelni> which assignment puts a negative value in it ? [15:39] <OanaStratulat> http://pastebin.com/w9B8W5qx this is the output [15:39] <OanaStratulat> let me av_log in different places and i`ll tell you [15:39] <michaelni> ok [15:43] <OanaStratulat> michaelni: if (coded_bps) size_out = size_out *coded_bps after this it goes negative [15:45] <OanaStratulat> michaelni: int how big can it contain ? [15:46] <OanaStratulat> if i av_log before the sizeout=size_out*coded_bps /8 it is positive and if i av_log after is negative [15:46] <OanaStratulat> and coded_bps is positive [15:48] <michaelni> you found where the bug is, now you have to fix it :) [15:49] <OanaStratulat> if i make size_out bigger [15:49] <OanaStratulat> change its type ? [15:49] <OanaStratulat> is a good approach ? [15:51] <OanaStratulat> i have changed it to int64_t [15:51] <OanaStratulat> and now it does not crash anymore [15:55] <OanaStratulat> michaelni: ^ [16:46] <michaelni> OanaStratulat, making size_out int64_t should work, yes [16:46] <OanaStratulat> is a good solution for this bug? [16:47] <OanaStratulat> michaelni: http://google-melange.appspot.com/gci/task/view/google/gci2011/7175252 please accept the claim [16:48] <OanaStratulat> michaelni: so i must file the bug 1st no? [16:50] <michaelni> yes, please open a bug report [16:50] <michaelni> claim accepted [16:50] <OanaStratulat> thx [16:56] <OanaStratulat> michaelni: bug filed and patch submitted, http://google-melange.appspot.com/gci/task/view/google/gci2011/7175252 marked as completed [17:03] <OanaStratulat> michaelni: do you have time to review now ? [17:10] <michaelni> OanaStratulat, did you test the patch ? [17:10] <OanaStratulat> yes [17:10] <OanaStratulat> it does not crash anymore [17:10] <michaelni> also the indention of the first line looks messed uü [17:10] <michaelni> up [17:11] <OanaStratulat> can you change that locally? [17:11] <OanaStratulat> or how do i amend ? [17:11] <OanaStratulat> git commit -amend ? [17:13] <michaelni> git commit --amend -a [17:14] <OanaStratulat> ok patch modified on trac [17:15] <michaelni> looks ok [17:16] <OanaStratulat> michaelni: http://google-melange.appspot.com/gci/task/view/google/gci2011/7180407 please accept [17:16] <OanaStratulat> if needed can you extend my deadline ? [17:21] <OanaStratulat> ubitux: michaelni what does [avi @ 0x7f7f5c03d800] Too deeply nested ODML indexes mean ? [17:25] <michaelni> OanaStratulat, i can extend as much as needed [17:25] <OanaStratulat> michaelni: can you extend with 24 hours because 2morrow i am a bit busy [17:25] <OanaStratulat> or you need some work to be submitted 1st ? [17:26] <OanaStratulat> i have found a hang with a dxtory file [17:26] <av500> OanaStratulat: it means the AVI file is beyond hope [17:26] <michaelni> extended [17:29] <OanaStratulat> av500: meaning it cannot be opened ? [17:29] <av500> what file is that? [17:29] <OanaStratulat> a dxtory file [17:29] <av500> what are you doing? [17:29] <av500> whatever that is [17:30] <av500> ah ok [17:30] <av500> so its likely HUGE and therefore uses ODML [17:31] <OanaStratulat> av500: ptp-roman.org/dxtory_crash here is the file [17:31] <av500> #define MAX_ODML_DEPTH 1000 [17:31] <av500> 1000 nested indexes? [17:32] <av500> that files is only 1.3MB [17:32] <OanaStratulat> it is a fuzzed file [17:32] <av500> ah [17:32] <av500> so what do you expet? [17:32] <av500> expect? [17:32] <OanaStratulat> but i wanted it to crash, to have a bug to fix.. [17:33] <OanaStratulat> but i did not understand the output [17:33] <av500> well, I would check the return value of read_braindead_odml_indx [17:33] <av500> inside read_braindead_odml_indx() [17:34] <OanaStratulat> thx, ok i`m leaving see you later [17:36] <ubitux> that was quick ;) [17:54] <mkoch> hiho! sorry if it's a bit off on this channel. is it possible to compile an universal binary from ffmpeg on os x lion? I need the static libs in a plugin which should also be compatible with both 32 and 64bit archs [18:16] <j-b> bcoudurier: hi! [18:16] <j-b> bcoudurier: http://mailman.videolan.org/pipermail/vlc-devel/2012-January/084719.html [18:28] <bcoudurier> hey j-b [18:28] <bcoudurier> happy new year [18:28] <j-b> happy new year to you the same [18:28] <j-b> bcoudurier: you might get interested for an invite to the above... or maybe not [18:29] <bcoudurier> count me in [18:31] <j-b> bcoudurier: mail me or courmisch then :) [18:32] <bcoudurier> ok [20:37] <ubitux> bcoudurier: are you ok with the current timecode patchset for now? i will think of a better public api in lavu ASAP; but i'd like to get done with the most of the features so we can move to a cleaner api then [20:41] <bcoudurier> it's ok but this should have been done earlier [20:41] <CIA-101> ffmpeg: 03Marton Balint 07master * r5d94f28e9e 10ffmpeg/ffplay.c: [20:41] <CIA-101> ffmpeg: ffplay: honor SDL_AUDIO_CHANNELS and make sure to use SDL supported number of audio channels [20:41] <CIA-101> ffmpeg: Fixes ticket #838. [20:41] <CIA-101> ffmpeg: Signed-off-by: Marton Balint <cus(a)passwd.hu> [20:41] <CIA-101> ffmpeg: 03Marton Balint 07master * r8ee77fc8d4 10ffmpeg/ffplay.c: [20:41] <CIA-101> ffmpeg: ffplay: fix invalid wanted_channel_layout calculation [20:41] <CIA-101> ffmpeg: Signed-off-by: Marton Balint <cus(a)passwd.hu> [20:41] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r4d5bbbadd7 10ffmpeg/: [20:41] <CIA-101> ffmpeg: Merge remote-tracking branch 'cus/stable' [20:41] <CIA-101> ffmpeg: * cus/stable: [20:41] <CIA-101> ffmpeg: ffplay: fix invalid wanted_channel_layout calculation [20:41] <CIA-101> ffmpeg: ffplay: honor SDL_AUDIO_CHANNELS and make sure to use SDL supported number of audio channels [20:41] <CIA-101> ffmpeg: Merged-by: Michael Niedermayer <michaelni(a)gmx.at> [20:41] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r02026d08a9 10ffmpeg/libavcodec/libvpxenc.c: [20:41] <CIA-101> ffmpeg: libvpxenc: Remove duplicate AVOption constants. [20:41] <CIA-101> ffmpeg: Found-by: j(a)v2v.cc [20:41] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:41] <ubitux> bcoudurier: ok thank you :) [20:59] <CIA-101> ffmpeg: 03ami_stuff 07master * r871e2f4fac 10ffmpeg/libavcodec/pnm.c: [20:59] <CIA-101> ffmpeg: Support 16bpp grayscale pam decoding. [20:59] <CIA-101> ffmpeg: Fixes ticket #881. [20:59] <CIA-101> ffmpeg: 03ami_stuff 07master * ra44b63f6c8 10ffmpeg/libavcodec/pnm.c: [20:59] <CIA-101> ffmpeg: Support 48bpp pam decoding. [20:59] <CIA-101> ffmpeg: Fixes ticket #882. [21:00] <CIA-101> ffmpeg: 03Clément BSsch 07master * r2aa7375a98 10ffmpeg/libavcodec/ (timecode.c timecode.h version.h): timecode: add avpriv_check_timecode_rate(). [21:00] <CIA-101> ffmpeg: 03Clément BSsch 07master * radc278788b 10ffmpeg/libavformat/mov.c: mov: annotate read values in tmcd track. [21:00] <CIA-101> ffmpeg: 03Clément BSsch 07master * r35da85562d 10ffmpeg/ (Changelog libavformat/mov.c): mov: support timecode extraction. [21:14] <profoX`> hi.. does anyone know a good IRC channel or forum for generic question about video encoding? (currently developing small video encoder/decoder for a small project which isn't related to ffmpeg, but I don't know any good resources) [21:34] <ubitux> profoX`: doom9 [21:35] <ubitux> animemusicvideos.org too maybe [21:37] <profoX`> ubitux: yea I was looking at doom9 but I couldn't really find a good place to post my question.. perhaps "New and alternative video codecs" I suppose.. thanks for the answer [21:53] <CIA-101> ffmpeg: 03Clément BSsch 07master * r96fc1f0733 10ffmpeg/libavfilter/af_volume.c: volume: remove duplicated condition. [22:25] <CIA-101> ffmpeg: 03Reimar Döffinger 07master * r1d0d63052b 10ffmpeg/libavcodec/sipr.c: (log message trimmed) [22:25] <CIA-101> ffmpeg: sipr: fall back to setting mode based on bit_rate. [22:25] <CIA-101> ffmpeg: Not all applications (e.g. MPlayer) set block_align, and [22:25] <CIA-101> ffmpeg: when using a different demuxer it might not even be [22:25] <CIA-101> ffmpeg: easily available. [22:25] <CIA-101> ffmpeg: So fall back to selecting mode based on bit rate as before [22:25] <CIA-101> ffmpeg: if block_align has not useful value. [22:25] <CIA-101> ffmpeg: 03Reimar Döffinger 07master * r4278023240 10ffmpeg/libavcodec/vqavideo.c: [22:25] <CIA-101> ffmpeg: Check for overread in vqa video decoder. [22:25] <CIA-101> ffmpeg: This issue was discovered while decoding the FATE sample vqa/ws_snd.vqa. [22:25] <CIA-101> ffmpeg: For some unknown reason only audio decoding is tested by FATE for that file, [22:25] <CIA-101> ffmpeg: but not video. [22:25] <CIA-101> ffmpeg: Signed-off-by: Reimar Döffinger <Reimar.Doeffinger(a)gmx.de> [22:49] <durandal_1707> is -atag -vtag etc meant to be useful at all? it doesn't work with avi,mkv,mov [23:02] <bcoudurier> it does work with mov [23:02] <OanaStratulat> michaelni: ping [23:02] <michaelni> durandal_1707, it should be usefull and working, if not open a bug report ill fix it [23:02] <michaelni> OanaStratulat, pong [23:03] <OanaStratulat> michaelni: ptp-roman.org/dxtory_crash this files makes a hang in ffmpeg [23:04] <OanaStratulat> michaelni: can it be considered a bug? [23:07] <michaelni> ffmpeg -i ~/videos/dxtory_crash -f null - [23:07] <michaelni> does not hang nor crash here [23:07] <OanaStratulat> what is your output ? [23:08] <OanaStratulat> can you pastebin it? [23:11] <michaelni> http://pastebin.com/v1egeAtb [23:12] <OanaStratulat> michaelni: i get [avi @ 0x7f7f5c03d800] Too deeply nested ODML indexes mean [23:12] <OanaStratulat> now cloning again the ffmpeg [23:12] <OanaStratulat> maybe some fixes were sent when i was away [23:18] <OanaStratulat> michaelni: it hang on my ffmpeg and it is cloned right now.. [23:18] <OanaStratulat> ubitux: ping [23:18] <ubitux> yup? [23:19] <OanaStratulat> ptp-roman.org/dxtory_crash this files makes a hang in ffmpeg, can you confirm ? [23:22] <ubitux> no issue there [23:22] <OanaStratulat> mhmh [23:22] <OanaStratulat> my ffmpeg goes wrong with it [23:23] <OanaStratulat> av500: confirmed it this afternoon [23:23] <ubitux> valgrind notices nothing with ./ffmpeg -i dxtory_crash -f null - [23:24] <OanaStratulat> ubitux: try to download the file again [23:24] <OanaStratulat> and check with it [23:24] <OanaStratulat> i have modified it [23:25] <ubitux> same [23:25] <OanaStratulat> ok so need to find another one... [23:25] <OanaStratulat> thx anyway [23:51] <cbsrobot> ubitux: ping [23:52] <ubitux> cbsrobot: yup? [23:52] <cbsrobot> whas the difference between "frame number format" and QT timecode ? [23:53] <OanaStratulat> ubitux: can you check ptp-roman.org/crash.mp4 if fails for you ? [23:55] <ubitux> cbsrobot: http://developer.apple.com/library/mac/documentation/QuickTime/QTFF/qtff.pdf [23:55] <ubitux> cbsrobot: check page 163 for QT timecode [23:55] <ubitux> frame number format is just the frame number written on 32 bytes [23:56] <ubitux> OanaStratulat: yes, i have an assert there [23:56] <OanaStratulat> ubitux: can you say in more normal terms what an assert is ? [23:56] <ubitux> "it should not happen, but let's check anyway" [23:57] <ubitux> and actually here, it happens, so it doesn't crash but miserably fail [23:58] <ubitux> michaelni: don't forget to update RELEASE :p [23:58] <OanaStratulat> can it be considered a bug? [23:58] <ubitux> michaelni: you should write a release checklist ;) [23:58] <ubitux> OanaStratulat: yes [23:59] <OanaStratulat> when i bt in gdb [23:59] <OanaStratulat> i get 3 lines, the last line is where the error started? [00:00] --- Fri Jan 6 2012

1 0

ffmpeg.log.20120105
by burek 06 Jan '12

06 Jan '12

[01:21] <Freakshow> curiously I'm seeing this in my log output [01:21] <Freakshow> h264_mp4toannexb failed for stream 0, codec libx264: Invalid data found when processing input [01:21] <Freakshow> with this cmd line [01:22] <Freakshow> ffmpeg -i "/usr/local/bin/sample.mp4" -threads 0 -re -vcodec libx264 -profile high -preset medium -g 30 -acodec libfaac -ab 128000 -ar 48000 -ac 2 -vbsf h264_mp4toannexb -f flv [01:22] <Freakshow> shooting to rtmp:// [01:22] <Freakshow> the sample.mp4 is from wowzamediaserver-2.2.4 [01:24] <Freakshow> google doesn't really give me much info on that log output [01:27] <Freakshow> is there something wrong with the command line that I'm using that would produce that wonky output? [01:57] <maximumHAX> probably a stupid question.. but when using ffmpeg to add text overlay why does the 1st part of video look great, then 5 sec into it degrades, compute performance? or setting? [01:59] <maximumHAX> in and out video is same format.. no settings passed except input video output name and -vf drawtext [03:17] <relaxed> Freakshow: I think you only need "-vbsf h264_mp4toannexb" if you're copying from mp4. Did you try without it? [04:49] <Soul_keeper> Does -vb not work with libvpx encoding ? [05:02] <Soul_keeper> appears that ffmpeg encodes to about 2x the specified bitrate [05:25] <kcm1700> ffmpeg 0.9.1 released? [05:33] <noisymouse> hi, how can I export a video as jpeg with the number of each frame added? [05:36] <drv> noisymouse: http://ffmpeg.org/faq.html#How-do-I-encode-movie-to-single-pictures_003f [05:42] <noisymouse> ok, I guess I can use that and have the image printer add the frame number/file name [05:42] <noisymouse> I'd like the image itself to have the frame number embedded [05:45] <relaxed> noisymouse: it could be done with two passes but I'm not sure about one pass. [05:57] <noisymouse> two passes with ffmpeg? [05:59] <Soul_keeper> I'm getting slight a/v sync issues with the following: ffmpeg -i dvd-01.vob -threads 8 -f webm -vcodec libvpx -aspect 16:9 -vb 400k -r 30000/1001 -acodec libvorbis -ab 192k -y test.webm [05:59] <Soul_keeper> any recommendations ? [06:00] <Soul_keeper> seems that the audio is a few seconds ahead of the video [06:01] <relaxed> noisymouse: Yes. Output all the frames to %06d.png, then run a for loop extracting the number from the filename and writing it to the frame. [06:02] <noisymouse> ok, do you have a link for how to do the for loop? [06:02] <relaxed> No, I just thought of it. [06:03] <noisymouse> ok np [06:06] <relaxed> something like, for i in *png; do ffmpeg -i $i -vf "drawtext=... text=${i%.*}" anotherdir/$i [06:07] <relaxed> ; done [06:07] <relaxed> ^^^^ should go on the end [06:10] <shifter1> I have problem with ffmpeg and input buffering [06:10] <relaxed> Soul_keeper: why are you changing the framerate? [06:11] <shifter1> is there any way to increase the amount of input buffering ffmpeg can do? [06:16] <Soul_keeper> relaxed, it appeared to be auto changing it to 60fps, yet the original appears to be 25.80 fps [06:16] <Soul_keeper> is 25.80 fps a common standard ? [06:16] <relaxed> no, 25 would be pal [06:17] <Soul_keeper> Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480 [SAR 32:27 DAR 16:9], 8500 kb/s, 25.80 fps, 59.94 tbr, 90k tbn, 59.94 tbc [06:17] <Soul_keeper> that is what ffmpeg spits out during the transcoding [06:20] <Soul_keeper> i'll try without forcing it [06:41] <chrisballinger> hey guys [06:44] <chrisballinger> does anyone have any experience with the libavformat code for decoding Core Audio (.caf) files that contain AAC? [06:44] <chrisballinger> http://www.hydrogenaudio.org/forums/index.php?showtopic=89889 [06:46] <relaxed> chrisballinger: ffmpeg can decode it if that's what you're wondering [06:46] <chrisballinger> oh no i have a specific problem with a malformed file [06:47] <chrisballinger> [caf @ 0x7fbee284ac00] Missing packet table. It is required when block size or frame size are variable. [06:47] <chrisballinger> It is because the file didn't get "closed" properly when the recording was stopped [06:48] <chrisballinger> so it is missing the packet table chunk: http://developer.apple.com/library/mac/#documentation/MusicAudio/Reference/… [06:49] <chrisballinger> the AAC contained inside the CAF is CBR, so hopefully that would help us reclaim the lost data [06:49] <relaxed> There are some audio devs that hang out in #libav, try asking there. [06:49] <chrisballinger> by rewriting a new packet table [06:49] <chrisballinger> cool thanks! [06:57] <darkstarbyte> Is there a way to cut parts of a video out without reconverting it? [06:59] <relaxed> darkstarbyte: look at mkvmerge's --split [07:18] <darkstarbyte> Anything a little more advanced? [07:19] <darkstarbyte> That does not have all of the stuff I need [07:38] <sam1> hi guys,i use ffmpeg to convert *.srt to *.ass but the last text always on top of video. [08:05] <Skaag> I'm trying to crop a 16:9 clip to 4:3 [08:05] <Skaag> (I don't mind losing the left & right sides) [08:06] <Skaag> I tried setting the dimensions, and setting the -aspect to 16:9 [08:06] <Skaag> (doesn't work) [08:14] <Skaag> how do I set the source aspect to something other than 1:1 [09:07] <kcm1700> it'd be better if topic is changed from 0.9 to 0.9.1 maybe? [11:33] <silverrocker> I have some questions concerning Google Summer of Code and ffmpeg project proposals. Where should I go with my questions [11:35] <Mavrik> silverrocker, mailing lists most likely [11:35] <Mavrik> -dev? [11:35] <silverrocker> just noticed that there was an ffmpeg-devel channel :) I will go there thanks [13:22] <iOOO9> i'm trying to get my head around this... what is mpeg-4? if something tells me they've sent me a mpeg-4 file, what does that actually say? [13:24] <Soul_keeper> iOOO9, http://en.wikipedia.org/wiki/MPEG-4 [13:27] <iOOO9> Soul_keeper: didn't make me much saner. but from what i understand now it can be many things. a container format is one of them [13:27] <keeperofdakeys> iOOO9: there are many parts to MPEG4 [13:28] <keeperofdakeys> there is the container, and three video specifications [13:28] <Kuukunen> (or more, depends on who's counting) [13:28] <keeperofdakeys> yeah, but what you actually encounter [13:28] <keeperofdakeys> and not hidden away [13:29] <Kuukunen> there's bunch of profiles :P [13:30] <Kuukunen> iOOO9: usually it means it's something like .avi or .mp4 file with XviD or DivX (MPEG-4 ASP) [13:30] <keeperofdakeys> "The key parts to be aware of are MPEG-4 part 2 (including Advanced Simple Profile, used by codecs such as DivX, Xvid, Nero Digital and 3ivx and by Quicktime 6) and MPEG-4 part 10 (MPEG-4 AVC/H.264 or Advanced Video Coding, used by the x264 encoder, by Nero Digital AVC, by Quicktime 7, and by high-definition video media like Blu-ray Disc)." [13:30] <keeperofdakeys> of course H264 is usually never called mpeg4 [13:31] <keeperofdakeys> some later forms of WMV are based on mpeg4 too [13:31] <Kuukunen> iOOO9: but if they're stupid, it might as well be something totally random renamed to something.mp4 :P [13:32] <iOOO9> so part 10 seems to be h.264. but is there anything that specifies the combination of a audio + video codec? [13:32] <iOOO9> like aac and h.264 [13:32] <Kuukunen> keeperofdakeys: the other ones aren't really "hidden", they're just not used [13:32] <iOOO9> Kuukunen: not used? you mean the other parts except for part 2 and 10? [13:33] <keeperofdakeys> Kuukunen: or, oblivous to normal users [13:33] <Kuukunen> iOOO9: well MPEG-4 Part 14 is the "mp4" container... [13:33] <Kuukunen> iOOO9: the profiles are included in Part 2 [13:33] <Kuukunen> "MPEG-4 Part 2 has approximately 21 profiles, including profiles called Simple, Advanced Simple, Main, Core, Advanced Coding Efficiency, Advanced Real Time Simple, etc" [13:33] <keeperofdakeys> actually, more not used now I look at them [13:33] <iOOO9> Kuukunen: and those profiles can be applied to both audio and video? [13:33] <iOOO9> Kuukunen: or are those only for video? [13:34] <keeperofdakeys> iOOO9: AAC is part 3 [13:34] <Kuukunen> iOOO9: those are video, but don't worry about them :P [13:34] <keeperofdakeys> look at the wikipedia link, it clearly lists the parts [13:34] <iOOO9> keeperofdakeys: yes [13:34] <Kuukunen> and in case you're wondering: no, most of the weird parts are not really used [13:35] <iOOO9> :) [13:35] <keeperofdakeys> iOOO9: as for your original question, I guess you can say mpeg4 is usually mpeg4 video, not h264 video, mpeg2 video, or other types [13:35] <keeperofdakeys> and depending on whether it uses b frames depends on whether it is simple profile or advanced simple profile [13:35] <iOOO9> so when divx or xvid where written, they were written according to the mpeg-4 standard? [13:36] <keeperofdakeys> iOOO9: divx used to be based on mpeg4 part 2, simple profile [13:36] <keeperofdakeys> then some of the devs took it closed source, and others made xvid as an open source project [13:36] <keeperofdakeys> and they both made codecs based on advanced simple profile (divx 5+) [13:37] <keeperofdakeys> now divx 7 is h264, but divx isn't that popular anymore [13:37] <keeperofdakeys> oh, I forgot to say divx used to be an opensource project [13:39] <iOOO9> keeperofdakeys: divx 7 IS h.264? how can it be? doesn't h.264 have license that restricts that? [13:40] <keeperofdakeys> iOOO9: the point is you pay money to use that license [13:40] <iOOO9> keeperofdakeys: so that's why it is no longer open source? [13:40] <keeperofdakeys> iOOO9: no, that occured long ago [13:40] <iOOO9> ok [13:44] <keeperofdakeys> ah, looks like divx has a project called "opendivx", but they stopped it, so people forked into xvid [13:44] <keeperofdakeys> that was around version 4, which was only simple profile [13:48] <Kuukunen> (and just in case you were wondering, there never was DivX 2, and the DivX 3 was just a hack and was never really fully MPEG-4 compatible anyway) [13:48] <Kuukunen> at least afaik =P [13:59] <iOOO9> am i right if i say that mpeg-4 opens for more variants of implementations than mpeg-2? [14:00] <iOOO9> for video and audio stream in a container format [14:00] <cbreak> mpeg-4 defines one container format based on QuickTime's Mov [14:01] <cbreak> and a video codec with a bunch of versions [14:03] <iOOO9> cbreak: and with a bunch of versions you mean divx, xvid etc [14:03] <iOOO9> too [14:03] <cbreak> I mean mpeg4 asp, mpeg4 avc [14:03] <iOOO9> cbreak: and what does mpeg-2 offer compared to that [14:03] <cbreak> mpeg4 also has levels [14:04] <iOOO9> cbreak: ah, ok. and xvid/divx are built according to asp or avc [14:04] <cbreak> with hardware decoders supporting less features than software decoders [14:04] <cbreak> those limit the maximal bit rate, the number of reference frames, consecutive b frames [14:04] <bigzed> Hello, I'm trying to demux some AVID media files with ffmpeg but actually I'm kind of lost and need a hint how to do it because I always get a file without a stream as an output [14:04] <cbreak> the size of the frame and a whole bunch of other things [14:04] <bigzed> here is a pastebin http://pastebin.com/ZnhbqvLL [14:05] <bigzed> The whole idea is just to remove the AVID mxf container and get the stream out of the avid file [14:05] <bigzed> has anyone a hint? [14:07] <iOOO9> from a preservation and archival point of view mpeg-2 still seems to be the recommended format with the motivation that possible implementations are relatively more constrained. [14:07] <iOOO9> does that seem logical to you people? [14:17] <keeperofdakeys> iOOO9: mpeg4 and h264 are quite well standardised [14:17] <keeperofdakeys> it just happens there are 'spinoffs' [14:18] <indicator> is there anything in particular that would make the following have very bad audio quality? ffmpeg -loop_input -aq 1 -ab 256k -ar 22050 -qscale 1 -s 720x480 -b 300k -r 10 -i bwp.jpg -i Music/530.ogg -shortest bwp.mp4 [14:19] <keeperofdakeys> but h264 approaches reasonable-sized lossless video, so in that respect it is probably prefered [14:34] <iOOO9> keeperofdakeys: does ffmpeg offer support of http://en.wikipedia.org/wiki/ProRes_422 as an input format? [14:35] <keeperofdakeys> iOOO9: I have no idea, check ffmpeg -formats [14:35] <keeperofdakeys> D is decode, E is encode [14:35] <iOOO9> thanks [14:35] <iOOO9> and thanks for all the great explanations :) [15:15] <cbreak> for archival, I'd use h.264, it just has better quality than alternatives [15:15] <cbreak> of course, if you want lossless, there are others [15:26] <iOOO9> are mpeg-4 profiles and parts the same thing? [15:42] <cryptopsy> in ffprobe some_vbr.mp3 , what do each kb/s mean ? [15:42] <cryptopsy> Duration: 00:11:19.23, start: 0.000000, bitrate: 252 kb/s [15:42] <cryptopsy> Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16, 320 kb/s [15:57] <Freakshow> relaxed: no I did not try that, let me give that a shot... [15:57] <Freakshow> oh, not here [15:58] <relaxed> I'm here [15:58] <Freakshow> irssi is acting strange. ok, yes... I'll give that a go [15:58] <Freakshow> thanks! [16:12] <simon_> hello there, does anybody has experience with ffmpeg on iOS and muxing / streaming the arc from the apple encoder? [16:12] <simon_> *arc = aac [16:17] <simon_> we are muxing an h264 and an aac stream into an mpeg-ts stream, but we get lots of errors regarding the audio stream at the receiving end, even the PID from the audio stream is going crazy [17:43] <mkoch> hiho! is it possible to compile an universal binary from ffmpeg on os x lion? I need the static libs in a plugin which should also be compatible with both 32 and 64bit archs [18:04] <tjholowaychuk> yo. I need to extract the audio from an flv and add it to an mp4 [18:04] <tjholowaychuk> i currently have two commands [18:04] <tjholowaychuk> is it possible to do with one? [18:27] <bigzed> Is it possible to encode DNX HD 120 and DNX HD 185 with ffmpeg? [18:28] <bigzed> And is it possible to decode DNX HD 120 and DNX HD 185 with ffmpeg, because I was successfull for DNX HD 60, 90 and 90x but not for other formats [18:29] <bcoudurier> yes and yes [18:31] <sangcn> Hello, Based on ffmpeg, Can we create the same program with http://www.photodvdcreator.com/ ? [18:41] <lavluda> is it possible to send/stream a file through rtmp (red5 server) [18:43] <relaxed> sangcn: yes [18:43] <bigzed> bcoudurier: So, how would you decode it? Because I have some Avid Mediafiles here with those codecs and I'm trying to get a mpeg4 out of these. Can you help me with that? Here the command I just up until know, which succeeds with codecs below 120 http://pastebin.com/rHUvdxJD [18:44] <bigzed> Oh, I forgot input is 'something.mxf' [18:45] <sangcn> relaxed: thanks [19:00] <mkoch> is it possible that the configure script is broken? if I set --arch=x86_32 the script ignores it and in config.mak ARCH_X86_64=yes is the active one [19:00] <bigzed> bcoudurier: Isn't it exactly this issue http://roundup.libav.org/issue2160? Because I'm trying to convert MXF files, too. [19:03] <cbreak> mkoch: that's an invalid architecture string [19:03] <mkoch> cbreak: what are the valid ones then? [19:03] <cbreak> mkoch: maybe you want to use i386? [19:03] <cbreak> read the config script :D [19:04] <mkoch> well, the best would be to have universal binary on my mac os x lion [19:04] <cbreak> not sure if it supports universal [19:04] <cbreak> but if you're on a mac, there are really no reasons to use 32bit code unless you have deps you can't recompile [19:05] <mkoch> cbreak: that's what I did, "x86_32" is in the ARCH_LIST [19:05] <mkoch> cbreak: I need to use the lib in a browser plugin and chrome doesn't support 64bit plugins yet :( [19:06] <cbreak> if you compile with gcc and give it -arch x86_64 -arch i386, you get a fat binary [19:06] <cbreak> but I think ffmpeg has lots of special stuff that needs more [19:06] <cbreak> so you can try to compile the libs for the architectures separately [19:06] <cbreak> and then stitch them together with lipo [19:08] <mkoch> cbreak: that's what I wanted to try but for that I need the 32bit version first. so why is x86_32 an invalid string? it's in ARCH_LIST (line 1113 in the latest version in the configure script)? [19:08] <cbreak> use i386 [19:09] <cbreak> it's one of the standard gcc architectures [19:09] <mkoch> trying... [19:10] <mkoch> no luck... [19:10] <mkoch> still the 64bit flag is active in config.mak [19:11] <cbreak> does it matter? [19:12] <cbreak> is the result 64 bit? [19:12] <mkoch> sure, the linke complains that it can't find any symbols for the selected architecture (which is i386) [19:14] <mkoch> (how can I find out the arch that a lib is build for form the .a file?) [19:14] <cbreak> ask lipo or file [19:14] <cbreak> lipo -info file [19:14] <cbreak> or file file [19:15] <cbreak> you can also check the .o files the compiler made [19:15] <mkoch> thx [19:16] <mkoch> ../ffmpeg/libavcodec/inverse.o: Mach-O 64-bit object x86_64 [19:16] <mkoch> so yes, the linker has right :( [19:18] <mkoch> cbreak: any other idea maybe? [19:18] <cbreak> trying it here [19:18] <cbreak> the ffmpeg git is huge [19:18] <mkoch> thanks! [19:20] <cbreak> I am on 10.6 though [19:21] <mkoch> (mine is 10.7.1 but i hope it doesn't matter) [19:23] <cbreak> hmm... it's still 64 bit here too, with --arch [19:25] <mkoch> re [19:25] <cbreak> maybe you can get something to work with -m32 or so in CFLAGS and LDFLAGS [19:26] <cbreak> have to cook now, but the -m32 should give you 32 bit builds... but it won't affect ffmpeg's self-recognition, so it might have 64 bit asm enabled and fail in spectacular ways or so... [19:26] <cbreak> I'll try some more later [19:26] <mkoch> cbreak: may I ask you to throw me a mail if you find out any useful? [19:26] <mkoch> and thanks for the help so far [19:28] <mkoch> I sent my address in priv [19:28] <mkoch> I must leave now too, thx and bye [19:31] <setuid> I'm trying to convert xmas video from my FLiP camera to something I can email/send/Dropbox to my girlfriend. The videos are coming out -larger- than the original source. [19:31] <setuid> I'm using ffmpeg -acodec mp2 -sameq [19:31] <setuid> Is there some other combination of options which will give me great quality but smaller file size, and be compatible with most Windows/Mac players? [19:40] <relaxed> setuid: ffmpeg -i input -vcodec libx264 -preset veryslow -threads 0 -crf 18 -acodec libvo_aacenc -ab 128k output.mp4 [19:40] <relaxed> If it complains about libvo_aacenc, try libfaac [19:47] <setuid> giving that a go now [19:48] <setuid> It doesn't like -preset or libx264 [19:48] <setuid> I'm using: FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. [19:49] Action: setuid pulls an update and tries that [19:50] <relaxed> yeah, builds from 2009 aren't going to work. [20:00] <maujhsn> Anybody in this chat an expert with FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1.2? [20:00] <maujhsn> smukherjee [20:01] <maujhsn> crytopsy [20:03] <pasteeater> that's ancient [20:03] <pasteeater> maujhsn: what's your question? [20:05] <maujhsn> pasteeater great hearing from you again! [20:06] <maujhsn> pasteeater I am using ffmpeg on a liveUSB! I get some interesting results! [20:07] <pasteeater> i never used one of those before [20:08] <maujhsn> pasteeater my question : I am having a hard time finding the source directory so that I can re-compile ffmpeg! [20:09] <maujhsn> pasteeater the liveUSB experience is very interesting! [20:09] <pasteeater> did you already download the source? [20:11] <maujhsn> pasteeater I downloaded ffmpeg from my synaptic package manager! [20:11] <maujhsn> pasteeater FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1.2 [20:11] <pasteeater> do you want to compile recent ffmpeg or the old ffmpeg source that ubuntu provides? [20:12] <maujhsn> pasteeater the old version that installed on my liveUSB. [20:13] <setuid> relaxed, latest version in Ubuntu is: FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1.2, Copyright (c) 2000-2009 Fabrice Bellard, et al. [20:13] <setuid> I just updated [20:14] <maujhsn> setuid I had no idea that this was the latest version! [20:14] <pasteeater> maujhsn: i think you can just: apt-get source ffmpeg [20:14] <pasteeater> setuid: that may be the latest ubuntu provides for your ubuntu version [20:15] <maujhsn> pasteeater what will this command do? [20:15] <pasteeater> it will download the ffmpeg source that ubuntu used to create their ffmpeg package [20:16] <pasteeater> setuid: but it's not the latest ffmpeg version and 0.5 isn't the latest that ubuntu provides for recent ubuntu [20:16] <pasteeater> see topic in this channel for ubuntu ffmpeg compile guide if you want the newest you can get [20:17] <maujhsn> pasteeater I will check that out as well! [20:18] <setuid> pasteeater, sure is... [20:19] <setuid> relaxed, I'm pulling from git, building from HEAD now [20:21] <maujhsn> pasteeater I am currently downloading kdenlive the video editor to a new liveUSB, and ffmpeg along with various other packages! [20:22] <maujhsn> pasteeater It is going to be interesting to see how this experiment works out! [20:31] <maujhsn> pasteeater Single photo to a 10 second .mpg movie is there a better command than this: ffmpeg -loop_input -i cab.png -r 29.97 -t 00:00:15 -qscale 4 cab_A.mpg? [20:33] <maujhsn> setuid any thoughts about my query? [20:38] <pasteeater> maujhsn: that's 15 seconds [20:39] <pasteeater> ffmpeg -loop_input -r 30000/1001 -i input -t 10 -qscale 4 output.mpg [20:39] <pasteeater> or -r ntsc [20:40] <maujhsn> pasteeater kdenlive & ffmpeg downloaded successfully onto my liveUSB! Question Why would I need to go sudo apt-get source ffmpeg? [20:40] <pasteeater> you won't. i thought you wanted the source files for some reason [20:40] <maujhsn> pasteeater "Your the best"! [20:41] <pasteeater> your example will, by default, read the input file at 25fps and your output will be 29.97. better to tell it to read input at your desired output fps. [20:41] <setuid> ERROR: libx264 version must be >= 0.118. [20:41] <setuid> yuck [20:42] <pasteeater> setuid: http://ubuntuforums.org/showthread.php?t=786095 [20:42] Action: setuid fetches more source [20:42] <pasteeater> step-by-step, copy and paste instructions [20:42] <pasteeater> there is a guide for each suported ubuntu version [20:42] <maujhsn> pasteeater The only packages that I need to download are the medubuntu non-free extras! [20:51] <maujhsn> pasteeater In the previous example -r ntsc, can the ntsc-dv, ntsc-dvd switches apply as well? [20:53] <pasteeater> not for -r, those are for -target [20:55] <maujhsn> pasteeater "Ouch! Your right!" [21:18] <defaultro> will this command make a proper mpg file? cat orig1.mpg orig2.mpg | ffmpeg -f mpeg -i - -vcodec copy -acodec copy merged.mpg [21:20] <ubitux> it should; you also have ffmpeg concat: protocol [21:21] <ubitux> ffmpeg -i concat:orig1.mpg,orig2.mpg -c copy merged.mpg should do the trick [21:21] <defaultro> ok [21:21] <defaultro> i'll use yours instead [21:21] <defaultro> thanks [21:35] <JetJaguarXP> Is there an option in FFmpeg to convert multiple files into one file? [21:38] <JetJaguarXP> And I'm not talking about the joining of files. I mean encode multiple files directly into one media file. [21:39] <ubitux> you mean multiple a/v streams? [21:39] <JetJaguarXP> No. [21:40] <ubitux> the same video stream composed of multiple ones then? [21:40] <ubitux> scale+overlay filters? [21:41] <JetJaguarXP> I have to convert DVD's created by a court reporter to MPEG 1. FFmpeg will read the VOB's on the DVD but there are multiple VOBs. I want to convert them to one MPEG 1 file. [21:41] <maujhsn> ubitux I going to experiment with that concat command! [21:42] <ubitux> JetJaguarXP: then why are you telling it's not about joining them?... [21:43] <JetJaguarXP> Because I'm not talking about using CAT to join files. I'm wondering if FFmpeg can actually encode multiple source files into one output file. [21:44] <maujhsn> ubitux Can you point to documentation that gives details about what you have just stated? [21:45] <JetJaguarXP> http://ffmpeg.org/faq.html#How-can-I-join-video-files_003f [21:45] <ubitux> JetJaguarXP: concat: then? [21:45] <ubitux> maujhsn: yes just a second [21:46] <ubitux> maujhsn, JetJaguarXP http://ffmpeg.org/ffmpeg.html#concat [21:47] <JetJaguarXP> Is that just for ffplay? [21:47] <ubitux> no [21:47] <ubitux> protocols are supported by all the tools [21:48] <maujhsn> ubitux Thanks! [21:48] <maujhsn> jetjaguarxp Thanks! [21:49] <ubitux> it's strange to see 3 random ppl asking the exact same thing at the same moment [21:50] <maujhsn> ubitux "Silly question what is ppl? [21:50] <ubitux> people [21:52] <JetJaguarXP> Ok. I'll give that a shot. [21:52] <maujhsn> ubitux OK! [21:53] <setuid> relaxed, Seems to be working now [21:54] <setuid> rebuilt everything from source [21:54] <setuid> it didn't like libvo_aacenc, but libfaac works [21:56] <JetJaguarXP> ubitux So... /usr/bin/ffmpeg -i concat <files> or take out the -i ? [21:57] <ubitux> ffmpeg -i concat:a.vob\|b.vob\|c.vob [21:57] <JetJaguarXP> Cool thanks. [21:57] <ubitux> defaultro: sorry btw, the sep is |, not a comma [22:06] <maujhsn> I am extending this invitation to all of you in chat today! to subscribe to http://www.youtube.com/telesismedia "Practical applications of ffmpeg! [22:07] <maujhsn> Learning ffmpeg is a continual work in progress! [22:12] <JetJaguarXP> ubitux concat worked like a charm [22:15] <maujhsn> JetJaguarXP When using the "ffmpeg -i concat:a.vob\|b.vob\|c.vob" command does it have to be in /usr/bin/ffmpeg? [22:16] <maujhsn> '/usr/bin/ffmpeg" [22:16] <maujhsn> "/usr/bin/ffmpeg" correction! [22:25] <EOF-sensei> why is it that ffmpeg -V doesn't return the version of libpostproc? [22:27] <EOF-sensei> is this normal or am I doing something wrong at compile time? [22:38] <JetJaguarXP> maujhsn I just posted it that way. It doesn't have to be. [22:40] <maujhsn> JetJaguarXP "Thats good to know! I might still use your conventions!" [22:41] <cbreak> mkoch: ./configure --arch=i386 --extra-cflags=-m32 --extra-ldflags=-m32 [22:41] <cbreak> hmm... [22:49] <setuid> These vids are still converting to the same size as their original [23:04] <setuid> How do I shrink the aspect ratio of a given video when converting? -aspect 0.75 isn't doing what the docs claim it should [23:04] <relaxed> setuid: you can scale the video down [23:05] <blauefuesse> hi. If i was to convert a .flv file to .avi with this script: ffmpeg -i INPUT.flv -sameq OUTPUT.avi ,then burn it to a CD, will i be able to watch it on a Televison DVD-Player ? [23:08] <setuid> relaxed, with -vf scale, I see that, but what if I don't know the size of the original, to be able to do the math to get it to 0.75% of that size? Can I supply one value, like -vf scale=900? [23:08] <setuid> I don't want to squash the aspect ratio [23:09] <relaxed> setuid: try this, ffmpeg -i input -vf scale=640:-1 .... [23:10] <setuid> Ok, that worked [23:37] <praedo> hello [23:38] <praedo> i need help with presets [23:38] <praedo> in one computer it works and not in the other [23:38] <praedo> i get this error: [23:38] <praedo> File for preset 'ultrafast' not found [23:38] <praedo> and on the comptuer where it works, i have the presets in /usr/local/share/ffmpeg [23:38] <praedo> i would say the ffmpeg package installed is the same [23:38] <praedo> i'm running archlinux [23:39] <praedo> what should i do to have presets working there? [23:54] <praedo> i have ffmpeg-git [23:56] <sacarasc> praedo: -preset ultrafast rather than -vpre ultrafast [23:59] <maujhsn> praedo -preset ultrafast has worked for me more often than not! [00:00] --- Fri Jan 6 2012

1 0

ffmpeg-devel.log.20120104
by burek 05 Jan '12

05 Jan '12

[00:02] <pasteeater> ubitux: i've seen a few users asking for a silence detector. [00:02] <saste> ubitux: uhmm.. just two notes, more parsable "noise" intervals, also the tolerance threshold specification may be more flexible [00:02] <saste> like it may support db and power values [00:03] <saste> and yes there is a feature request for it in trac [00:06] <ubitux> oh really? [00:06] <ubitux> nice [00:06] <ubitux> ok i'll try to come up with something better then [00:06] <saste> #454 [00:07] <ubitux> thx [00:07] <ubitux> arh and i need to do somthing with the subtitles too [00:08] <ubitux> saste: did you reply to Nicolas about his suggestion for lavfi? [00:08] <ubitux> can't remember the exact proposition [00:08] <saste> ubitux: ouch, not yet [00:08] <ubitux> i'm really not familiar enough with lavfi to make big decision [00:09] <ubitux> so i'm lazily waiting for you guys to come up with good theory/architectural ideas :p [00:10] <ubitux> ok got the mail, this one: http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/2011-December/117936.html [00:10] <ubitux> pasteeater: what is the usage behind? [00:11] <ubitux> the only one i see atm is for monitoring an audio live stream and detect cut/inactivity [00:12] <ubitux> or maybe secret tracks or similar on a CD& ? [00:12] <OanaStratulat> michaelni: ping [00:14] <pasteeater> ubitux: i don't recall any particular reason why it was wanted. [00:16] Action: durandal_1707 wants nice visualization filter [00:17] <ubitux> ah yeah the visualization filter, need to get done with the spectrum one [00:17] <ubitux> too much fun to enjoy, i think i'm going to overflow [00:17] <durandal_1707> visualization filterS [00:17] <michaelni> OanaStratulat, pong [00:18] <OanaStratulat> michaelni: i want to take a FFMPEG bug closing task do you have time to discuss about it ? [00:18] <ubitux> durandal_1707: i don't if saste started and want to finish the mylkist wrapper, but i think it could be a nice contribution ;) [00:19] <OanaStratulat> michaelni: be right back in 2 minutes, rebooting my system [00:20] <OanaStratulat> michaelni: i`m here. i`m now making the FFMPEG build [00:22] <saste> ubitux: what's this milkyst thing? i tried with projectm, i have a local patch but still can't compile [00:23] <saste> it's c++, no easy to make ffmpeg digest it [00:25] <ubitux> milkymist/milkdrop [00:25] <durandal_1707> projectM doesn't even work on FreeBSD last time i tried [00:25] <ubitux> yes that's the third name [00:25] <ubitux> can't remember even one correctly :p [00:26] <ubitux> anyway, afaik, it's the "reference" for visualizations [00:26] <ubitux> saste: don't we have already various c++ wrapper? [00:28] <saste> ubitux: yes something related to android [00:29] <ubitux> utvideo too afaik [00:29] <saste> main problem with c++ is that it doesn't support designated init [00:29] <ubitux> yes i don't like it at all [00:29] <saste> but that's not a problem which can't be sorted out with a few hacks here and there [00:39] <durandal_1707> is there any specification for dv format? [00:39] <JEEBsv> yes, the SMPTE stuff IIRC [00:40] <JEEBsv> those are lolpriced tho so you'll have to first find the correct spec number and then try your luck at your chinese searching skills [00:41] <CIA-101> ffmpeg: 03Stefano Sabatini 07release/0.9 * r103cb461b0 10ffmpeg/libavfilter/vf_drawtext.c: [00:41] <CIA-101> ffmpeg: vf_drawtext: select YUV color for drawbox() in case YUV colorspace is used [00:41] <CIA-101> ffmpeg: Fix box alpha rendering when a YUV colorspace is selected, in particular [00:41] <CIA-101> ffmpeg: fix trac ticket #763. [00:41] <CIA-101> ffmpeg: (cherry picked from commit 1858a5c25ee1ccb8089339d20138116a46cc6c77) [00:44] <ubitux> mmh i though we had a "dB" volume parsing helper [00:44] <durandal_1707> JEEBsv: libdv? [00:44] <ubitux> seems there is only some code af volume& i thought we had sth similar in af pan [00:44] <JEEBsv> durandal_1707: you asked for a _specification_ [00:45] <JEEBsv> of course there can be _implementations_ that can be easier to get a hold of [00:45] <durandal_1707> well i'm trying to fix #217 and need _specifications_ [00:45] <JEEBsv> it's worth checking out the multimedia.cx wiki as well [00:46] <JEEBsv> durandal_1707: good luck then, you'll just have to find the spec number and search for the pdf on your favourite search engine :3 [00:46] <JEEBsv> unless smpte or whatever has made it free [01:09] <CIA-101> ffmpeg: 03Stefano Sabatini 07master * r90347dabc8 10ffmpeg/ffprobe.c: [01:09] <CIA-101> ffmpeg: ffprobe: change order of operations in probe_file() [01:09] <CIA-101> ffmpeg: This is required by a pending patch, also fixes a memleak due to the [01:09] <CIA-101> ffmpeg: writer context not being closed in case of open_input_file() failure. [01:09] <CIA-101> ffmpeg: 03Stefano Sabatini 07master * r4c5c0609c1 10ffmpeg/ffprobe.c: ffprobe: reindent after the last commit [01:09] <CIA-101> ffmpeg: 03Stefano Sabatini 07master * rc972f91d5b 10ffmpeg/ffprobe.c: ffprobe: prefer av_log to fprintf(stderr) [01:09] <CIA-101> ffmpeg: 03Stefano Sabatini 07master * r55ed19bf3b 10ffmpeg/ffprobe.c: ffprobe: remove unused variable in show_format() [01:09] <CIA-101> ffmpeg: 03Stefano Sabatini 07master * rd6da16dca5 10ffmpeg/ (Changelog doc/ffprobe.texi doc/ffprobe.xsd ffprobe.c): ffprobe: add support to option -show_error [01:39] <CIA-101> ffmpeg: 03Diego Biurrun 07master * r418f066f8c 10ffmpeg/libavcodec/snow.c: snow: Restore three mistakenly removed casts. [01:39] <CIA-101> ffmpeg: 03Daniel Verkamp 07master * rb73ec05473 10ffmpeg/libavutil/common.h: [01:39] <CIA-101> ffmpeg: Add av_popcount64 [01:39] <CIA-101> ffmpeg: Signed-off-by: Janne Grunau <janne-libav(a)jannau.net> [01:39] <CIA-101> ffmpeg: 03Daniel Verkamp 07master * r02e8f4272c 10ffmpeg/ (configure libavcodec/pthread.c): [01:39] <CIA-101> ffmpeg: win32: detect number of CPUs using affinity [01:39] <CIA-101> ffmpeg: Signed-off-by: Janne Grunau <janne-libav(a)jannau.net> [01:39] <CIA-101> ffmpeg: 03Anton Khirnov 07master * r212fd3a1f1 10ffmpeg/libavformat/utils.c: [01:39] <CIA-101> ffmpeg: lavf: flush decoders in avformat_find_stream_info(). [01:39] <CIA-101> ffmpeg: This means that avformat_find_stream_info() now works properly with [01:39] <CIA-101> ffmpeg: multiple threads, so there's no need to force thread count to 1. [01:39] <CIA-101> ffmpeg: 03Aneesh Dogra 07master * r2e70b2de58 10ffmpeg/libavcodec/dct.c: [01:39] <CIA-101> ffmpeg: dct: K&R formatting cosmetics [01:39] <CIA-101> ffmpeg: Signed-off-by: Diego Biurrun <diego(a)biurrun.de> [01:39] <CIA-101> ffmpeg: 03Shitiz Garg 07master * rf37b4efe03 10ffmpeg/libavcodec/dca.c: [01:39] <CIA-101> ffmpeg: dca: K&R formatting cosmetics [01:39] <CIA-101> ffmpeg: Signed-off-by: Diego Biurrun <diego(a)biurrun.de> [01:39] <CIA-101> ffmpeg: 03Janne Grunau 07master * rc04a954da6 10ffmpeg/ (doc/APIchanges libavutil/avutil.h): [01:39] <CIA-101> ffmpeg: lavu: bump lavu minor for av_popcount64 [01:39] <CIA-101> ffmpeg: cmdutils: K&R reformatting cosmetics [01:39] <CIA-101> ffmpeg: Signed-off-by: Diego Biurrun <diego(a)biurrun.de> [01:39] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r4f1a787744 10ffmpeg/libavcodec/adxdec.c: [01:39] <CIA-101> ffmpeg: adxdec: Do not require extradata. [01:39] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:39] <CIA-101> ffmpeg: Signed-off-by: Justin Ruggles <justin.ruggles(a)gmail.com> [01:39] <CIA-101> ffmpeg: 03Justin Ruggles 07master * re49d21306a 10ffmpeg/libavcodec/alac.c: [01:39] <CIA-101> ffmpeg: alacdec: implement the 2-pass prediction type. [01:39] <CIA-101> ffmpeg: The reference encoder does not generate any streams using this, but the [01:39] <CIA-101> ffmpeg: reference decoder can handle it, so we should as well. [01:39] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * rb024209b1f 10ffmpeg/libavcodec/adx_parser.c: [01:40] <CIA-101> ffmpeg: adx_parser: rewrite. [01:40] <CIA-101> ffmpeg: The previous code ended in multiple different infinite [01:40] <CIA-101> ffmpeg: loops. See stl_ten_1_big.sfd as example with and without zzuf [01:40] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [01:40] <CIA-101> ffmpeg: Signed-off-by: Justin Ruggles <justin.ruggles(a)gmail.com> [01:40] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r6fd075f180 10ffmpeg/libavcodec/adx.c: adxdec: Validate channel count to fix a division by zero. [01:40] <CIA-101> ffmpeg: 03Justin Ruggles 07master * rbb63475aec 10ffmpeg/libavcodec/alacenc.c: [01:40] <CIA-101> ffmpeg: alacenc: implement the 2-pass prediction type. [01:40] <CIA-101> ffmpeg: This isn't used by the reference encoder, but it is supported by the decoder. [01:40] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r149e1b0468 10ffmpeg/libavcodec/alacenc.c: alacenc: do not generate invalid multi-channel ALAC files [01:41] <CIA-101> (10 lines omitted) [02:30] <durandal_1707> isn't AVrp just reversed AJA Kona? [02:49] <CIA-101> ffmpeg: 03Oana Stratulat 07master * r0d3a51e5d2 10ffmpeg/libavformat/electronicarts.c: [02:49] <CIA-101> ffmpeg: electronicarts: Fix division by zero. Fixes Ticket #793 [02:49] <CIA-101> ffmpeg: Signed-off-by: Oana Stratulat <oanaandreeastratulat(a)gmail.com> [02:49] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:31] <stpere> hi [04:31] <stpere> I've sent a patch to ffmpeg-devel@.. [04:51] <michaelni> stpere, do you have a sample als file for testing ? [04:51] <michaelni> from a quick look the patch makes sense [04:52] <stpere> well, I don't have a file honestly [04:53] <stpere> and maybe the logic isn't perfect nor good.. [04:53] <stpere> but in any cases, if it did fall into this loop, I can't see how it could play. [04:53] <stpere> :) [04:53] <stpere> I'll try to find one [04:54] <michaelni> ill mail the author of alsdec [04:55] <stpere> thanks! [05:04] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r6072a19b4f 10ffmpeg/libavformat/utils.c: [05:04] <CIA-101> ffmpeg: lavf: Fix try_decode_frame() so it doesnt loop infinitely. [05:04] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [05:04] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * rf37174bc19 10ffmpeg/libavformat/electronicarts.c: [05:04] <CIA-101> ffmpeg: electronicarts: Pass error through ea_read_header(). [05:04] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [05:23] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r15b219fae9 10ffmpeg/libavcodec/mjpegdec.c: [05:23] <CIA-101> ffmpeg: jpegdec: Fix vlc table check for progressive jpegs. [05:23] <CIA-101> ffmpeg: Fixes Ticket834 [05:23] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [09:52] <OanaStratulat> michaelni: ping [10:59] <Tjoppen> am I stupid, or does aacenc's output really not work with mov nor mp4? [10:59] <Tjoppen> "0.0kbits/malformated aac bitstream, use -absf aac_adtstoasc" yet if I add that it still fails [11:00] <av500> one should use AAC only inside its native RealVideo container.... [11:02] <Tjoppen> bah, I'll use libmp3lame [11:02] <OanaStratulat> michaelni: ping [13:19] Last message repeated 1 time(s). [13:21] <CIA-101> ffmpeg: 03Clément BSsch 07master * r837126568c 10ffmpeg/doc/filters.texi: doc: use @code{} for --{enable,disable}-options. [13:21] <CIA-101> ffmpeg: 03Clément BSsch 07master * rdc7ad85c40 10ffmpeg/doc/ (7 files): doc: use @command{} for commands. [14:47] <OanaStratulat> michaelni: ping [15:21] Last message repeated 1 time(s). [15:21] <av500> you have a cron job? [15:22] <OanaStratulat> av500: the question is for me? [15:27] <av500> yes [15:27] <av500> i guess he will see himself highlighted, so no need to repeat that all the time [15:38] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07master * r2754514787 10ffmpeg/ (10 files in 4 dirs): [15:38] <CIA-101> ffmpeg: v308 Quicktim Uncompressed 4:4:4 encoder and decoder. [15:38] <CIA-101> ffmpeg: Reviewed-by: Derek Buitenhuis [15:38] <CIA-101> ffmpeg: Reviewed-by: Paul B Mahol [15:38] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07master * r64e4f4836a 10ffmpeg/ (8 files in 4 dirs): [15:38] <CIA-101> ffmpeg: Add decoder for Avid 1:1 10-bit RGB Packer (AVrp). [15:38] <CIA-101> ffmpeg: Fixes ticket #525. [15:38] <CIA-101> ffmpeg: Reviewed-by: Paul B Mahol [15:38] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07master * r17edc370b2 10ffmpeg/ (10 files in 4 dirs): [15:38] <CIA-101> ffmpeg: yuv4 libquicktime packed 4:2:0 encoder and decoder. [15:38] <CIA-101> ffmpeg: Reviewed-by: Derek Buitenhuis [15:38] <CIA-101> ffmpeg: Reviewed-by: Paul B Mahol [16:18] <michaelni> OanaStratulat, pong [16:48] <OanaStratulat> michaelni: i want to work on http://ffmpeg.org/trac/ffmpeg/ticket/236 [16:49] <OanaStratulat> michaelni: the valgrind says that FPE is at mathematics.c at line 91, at return (a * b + r)/c; [16:50] <michaelni> OanaStratulat, best talk to baptiste, he knows the mov muxer best [16:50] <OanaStratulat> michaelni: what nick? [16:52] <michaelni> OanaStratulat: bcoudurier [16:57] <OanaStratulat> michaelni: can`t you look into this because he is not online and 2morrow i do not think i`ll be online [17:06] <OanaStratulat> michaelni: i have emailed him regarding this issue [18:24] <ubitux> < OanaStratulat> michaelni: i want to work on http://ffmpeg.org/trac/ffmpeg/ticket/236 // that issue looks similar to http://ffmpeg.org/trac/ffmpeg/ticket/810 [18:25] <OanaStratulat> ubitux: so what bug should i try to close 1st? [18:26] <ubitux> it's not the same bug, but it looks like a similar mistake is made [18:26] <ubitux> like a timebase not specified under some condition [18:27] <ubitux> i didn't look much, just pointing out potential ideas [18:27] <ubitux> i don't think it matters the one you may close first [18:38] <michaelni> ubitux, if you could mentor oana, that would be great, iam a bit busy atm sadly ..., but ill close the task when you say its ok [18:38] <OanaStratulat> ubitux: i will be online in like 3-4 hours, now i`m in town [18:38] <OanaStratulat> ubitux: is that ok? [19:05] <ubitux> michaelni: i'm not really familiar with what she's ("she", right?) trying to fix, or i'd already fixed it [19:05] <ubitux> i'm not sure i'll be of much help [19:05] <ubitux> but sure i can help if i know, like i do most of the time [19:15] <michaelni> ubitux, thx, (i need to look into some security issues ...) [19:50] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07master * r84ce58faf5 10ffmpeg/libavcodec/ (v308dec.c v308enc.c yuv4dec.c yuv4enc.c): [19:50] <CIA-101> ffmpeg: Replace deprecated FF_I_TYPE with AV_PICTURE_TYPE_I in v308 and yuv4. [19:50] <CIA-101> ffmpeg: Found-by: Paul B Mahol [19:50] <CIA-101> ffmpeg: 03Andrew Ryan 07master * r4452d58c72 10ffmpeg/libavformat/mov.c: Insert missing newline at end of error log message in mov demuxer. [20:35] <OanaStratulat> ubitux: ping [20:35] <ubitux> hi OanaStratulat [20:35] <ubitux> sorry i wasn't around when you asked [20:35] <OanaStratulat> ubitux: do you have time to discuss about the bug i wan to fix? [20:35] <OanaStratulat> want* [20:35] <ubitux> sure go ahead [20:35] <OanaStratulat> have you run the file and the command on your build? [20:35] <ubitux> but i'll need to look more at the source of the issue to be of any help [20:36] <OanaStratulat> i have found the source of the problem, i tink [20:36] <OanaStratulat> think [20:36] <ubitux> give me a few minutes [20:36] <OanaStratulat> valgrind says that the problem is at mathematics.c line 93 [20:37] <ubitux> yes but the function is ok [20:37] <OanaStratulat> that function is called in movenc.c 1st in the mov_write_mvhd_tag function [20:37] <OanaStratulat> and then the mov_write_mvhd_tag function is called in mov_write_moov_tag [20:38] <OanaStratulat> and mov_write_moov_tag is called in update_first_fragment [20:38] <OanaStratulat> everything begins in update_first_fragment i think [20:47] <OanaStratulat> ubitux: am i right or thinking it wrong ? [20:54] <ubitux> OanaStratulat: i'm available in a minute, sorry :) [20:54] <OanaStratulat> ubitux: ok ok no prob, ping me when ready [20:57] <ubitux> ok so [20:58] <OanaStratulat> yes [20:58] <ubitux> doesn't crash here, and detect no error mmmh [20:59] <OanaStratulat> it does not crash on your build? [20:59] <ubitux> yep, can't reproduce atm [20:59] <OanaStratulat> running the latest src from git? [20:59] <ubitux> yes, just updated a few minutes ago [21:00] <OanaStratulat> how can i update the src ? [21:00] <OanaStratulat> not to clone it again [21:00] <OanaStratulat> git rebase ? [21:00] <OanaStratulat> the bug is reproduced with ffmpeg -i fcp_export8.mov -dcodec copy -map 0:2 -an -vn out.mov [21:00] <OanaStratulat> ubitux: using this ? [21:00] <ubitux> ah, right [21:00] <ubitux> this one crash ok [21:01] <CIA-101> ffmpeg: 03Philippe Saint-Pierre 07master * rf0f2babca2 10ffmpeg/libavcodec/alsdec.c: [21:01] <CIA-101> ffmpeg: Fix possible infinite loop decoding als. [21:01] <CIA-101> ffmpeg: Reviewed-by: Thilo Borgmann [21:01] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07release/0.9 * r9f561ec398 10ffmpeg/libavcodec/wavpack.c: [21:01] <CIA-101> ffmpeg: Revert "wavpack: Clip samples after shifting" [21:01] <CIA-101> ffmpeg: This reverts commit 8d055e9079f151f13d34e8e04f4aa7ca0273c448. [21:01] <CIA-101> ffmpeg: The original commit introduced ticket #871, do not import the problem [21:01] <CIA-101> ffmpeg: into the release branch. [21:02] <ubitux> and i have this backtrace in valgrind: http://pastie.org/3126857 [21:03] <OanaStratulat> how do i backtrace in valgrind? [21:03] <ubitux> just run it with valgrind ./ffmpeg_g -i ... [21:04] <ubitux> i'm not sure if it's as efficient as gdb [21:04] <ubitux> it should point out the same issue [21:04] <OanaStratulat> i used gdb [21:04] <ubitux> i'm concerned about the warning: "Ignoring attempt to set invalid timebase for st:0" [21:04] <funman> valgrind --db-attach=yes [21:05] <ubitux> i'm wondering if the timebase is not set to zero, and thus doesn't provoke the crash [21:05] <ubitux> just like what's happening with the mxf bug [21:06] <OanaStratulat> mhmh [21:07] <OanaStratulat> so what do you think is the bug coming from the functions i have enumerated ? [21:10] <ubitux> yes sure that's basically what's the gdb backtrace is saying [21:10] <ubitux> the concern is now to understand why does this final function gets a zero in the rationnal variable (which leads to the floating exception) [21:11] <OanaStratulat> i have traced the functions until update_first_fragment [21:12] <ubitux> #3 update_first_fragment (s=0x1457560) at libavformat/movenc.c:2619 [21:12] <ubitux> #4 flush_cluster_buffer (s=0x1457560) at libavformat/movenc.c:2216 [21:12] <ubitux> #5 0x00000000004f3a97 in mov_write_trailer (s=<optimized out>) at libavformat/movenc.c:2643 [21:12] <ubitux> this are the previous steps [21:12] <ubitux> what's interesting to know is where the rationnal variable leading to zero is initialized (or isn't) [21:13] <ubitux> and if it isn't initialized, where should it be (basically you have to check where it is set to something different than zero in the normal case, and figure out why it doesn't in the crash scenario) [21:13] <OanaStratulat> i do not think that is initialized i do not see anything with it [21:16] <ubitux> can you confirm mov->tracks[0].timescale=0 just before the crash? [21:16] <OanaStratulat> yes [21:17] <OanaStratulat> done av_log and it is 0 [21:17] <ubitux> al'right [21:18] <ubitux> so, check where the timescale is normaly initialized [21:18] <OanaStratulat> track->timescale = MOV_TIMESCALE; :2409 [21:18] <OanaStratulat> there [21:19] <OanaStratulat> and the timescale is initialized at track->timescale = st->codec->time_base.den; :2526 [21:20] <OanaStratulat> ubitux: food time, be right back approximatively 10 minutes [21:34] <OanaStratulat> ubitux: back [21:35] <ubitux> al'right [21:35] <ubitux> about your git question earlier; do you have some local changes or not? [21:35] <OanaStratulat> yes i have some [21:35] <ubitux> also, are you in the master branch? (default) [21:35] <OanaStratulat> yes in the master branch [21:35] <ubitux> are the changes committed? [21:36] <OanaStratulat> no [21:36] <ubitux> ok so just a few files modified [21:36] <ubitux> but no local commits with these changes [21:36] <ubitux> right? [21:36] <OanaStratulat> yes [21:37] <ubitux> i suggest you to save these changes in a commit in a separate branch: [21:37] <ubitux> git checkout -b dev # this will create a branch "dev", the same as the current one (master) [21:37] <ubitux> git add/commit your stuff [21:37] <OanaStratulat> ok [21:37] <OanaStratulat> and then switch to the master branch again no [21:38] <ubitux> that's right [21:38] <OanaStratulat> ok, now with the bug [21:38] <ubitux> then you can git pull [21:38] <OanaStratulat> have you looked at those lines i pinned out? [21:38] <ubitux> to update your master branch [21:38] <ubitux> yes i had a look [21:38] <ubitux> a notice multiple places where it is supposed to be initialized [21:38] <OanaStratulat> are those the places where the timescale is being worked with ? [21:39] <ubitux> the main one is imho in mov_write_header() (but i may be wrong) [21:39] <ubitux> i'd personnaly check if mov_write_header() is called [21:39] <ubitux> if you look closely at the function you'll see the timescale being initialized according to the different type of track/stream [21:40] <OanaStratulat> yes, that i wanted to tell you now [21:40] <ubitux> and i guess it ins't in the particular case you are studying [21:40] <ubitux> you may want to try something in that direction [21:40] <ubitux> i really don't know if it's the correct thing to do, just pointing out what i'm quickly observing [21:41] <OanaStratulat> mov_write_header is called [21:41] <OanaStratulat> i have av_log when it starts and it is called once then the crash [21:42] <ubitux> do you see the loop on each stream? with the timescale assignment in various cases? [21:42] <OanaStratulat> please be more explicit [21:43] <OanaStratulat> http://pastebin.com/UTnSytuq [21:43] <OanaStratulat> this is my output [21:43] <OanaStratulat> the mov_write_header is called then it says Ignoring attempt to set invalid timebase for st:0 [21:43] <ubitux> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavformat/movenc.c;hb=HEAD… [21:44] <ubitux> look at this [21:44] <ubitux> and look at the various if (... == AVMEDIA_TYPE...) [21:44] <ubitux> it's basically looping over all the track to write AFAIU [21:45] <ubitux> and for each type of track (audio, video, subtitles, ...) [21:45] <ubitux> it sets a particular timescale [21:45] <OanaStratulat> mhmh a bit collapsing over me [21:45] <OanaStratulat> 1st of all what is the timescale ? [21:47] <ubitux> try to print when it's audio, video, etc :) [21:47] <ubitux> it's related to the rate [21:47] <ubitux> sorry i don't know your C level at all, or what you already did [21:47] <ubitux> and how much i'm supposed to help you ;) [21:47] <ubitux> also, it's the first time i'm opening this file so well& [21:48] <OanaStratulat> my C level is not very high, i`m studying at school c+ [21:48] <OanaStratulat> c++* [21:49] <ubitux> if/while/for is ok? functions ok? pointers are ok? structures? allocation? [21:49] <OanaStratulat> allocation not so good [21:49] <OanaStratulat> the rest are ok [21:49] <ubitux> ok [21:50] <ubitux> so well, [21:50] <ubitux> you have a main loop: for(i=0; i<s->nb_streams; i++){ ... } [21:50] <ubitux> which is looping over each stream: [21:50] <ubitux> AVStream *st= s->streams[i]; [21:50] <OanaStratulat> yes [21:50] <ubitux> the stream has an associated codec, and thus "type" [21:51] <ubitux> video, audio, subtitles, data, & [21:51] <ubitux> if you scroll down a bit, you'll see: [21:51] <ubitux> the following pattern: [21:51] <ubitux> if type is video: [misc stuff] stream->timescale = .... [21:51] <ubitux> else if type is audio: [misc stuff] stream->timescale = ... [21:51] <ubitux> etc [21:52] <OanaStratulat> st->codec->codec_type == AVMEDIA_TYPE_VIDEO [21:52] <OanaStratulat> yes [21:52] <ubitux> right [21:52] <OanaStratulat> video audio and subtitle [21:52] <ubitux> that's right [21:52] <ubitux> then what is your command trying to do? [21:53] <ubitux> "-dcodec copy -map 0:2" [21:53] <OanaStratulat> this i do not know [21:53] <ubitux> data codec, copy it, and map it [21:53] <OanaStratulat> -map is just to decode only streams from 0 :2 ? [21:53] <OanaStratulat> data codec meaning raw data ? [21:53] <ubitux> no it means stream with a data codec, you have to copy them [21:54] <ubitux> data codec are for instance for various information track [21:54] <ubitux> doesn't matter really [21:54] <OanaStratulat> ok [21:54] <ubitux> but well it seems you're trying to map a data stream [21:54] <OanaStratulat> what does the mapping is all about ? [21:54] <ubitux> can you see a check for this kind of codec? [21:55] <ubitux> it just indicates what's your taking from the input [21:55] <ubitux> and putting/mapping on the output [21:55] <OanaStratulat> please have in mind that is my 1st task like this and i`m not very familiar with the terminology [21:55] <ubitux> yes sure [21:56] <ubitux> don't take what i'm saying for absolute truth either, i'm not that much familiar either, and like everyone i make mistakes :) [21:56] <ubitux> anyway [21:56] <ubitux> you're trying to map a stream of type "data" [21:56] <OanaStratulat> so somewhere here the timescale is initialized with 0 you would say? [21:56] <OanaStratulat> and you want to see if it is for audio or video ? [21:56] <ubitux> init to zero... or isn't at all [21:56] <OanaStratulat> which one of them ? [21:57] <ubitux> i don't know, look at the code :) [21:57] <ubitux> it seems audio and video have there respective check [21:57] <ubitux> subtitles too [21:57] <ubitux> can you see the check for data? [21:58] <OanaStratulat> there is no check for data [21:58] <OanaStratulat> only audio video and sub [21:59] <OanaStratulat> it must be as the same format st->codec->codec_type == data ? [22:00] <ubitux> if you don't see any check for data maybe that means the timescale is not set for data [22:00] <ubitux> which actually makes sense in case of a "data" track but well [22:00] <ubitux> you could try to see what happens when you set one [22:01] <OanaStratulat> st->codec->codec_type == AVMEDIA_TYPE_DATA ? [22:01] <ubitux> remember: i'm just giving testing hints, not a solution :p [22:01] <OanaStratulat> yes yes [22:01] <ubitux> should be, check the various types [22:02] <ubitux> (the list is in libavutil/avutil.h) [22:02] <OanaStratulat> ok [22:03] <OanaStratulat> AVMEDIA_TYPE_DATA is [22:07] <OanaStratulat> ubitux: i have done if(st->codec->codec_type == AVMEDIA_TYPE_VIDEO)av_log(0,0,"\n\ndata\n\n\n"); and there is no output.. [22:07] <OanaStratulat> sorry wait [22:07] <OanaStratulat> my mistake [22:07] <OanaStratulat> yes it is data strea [22:07] <OanaStratulat> stream [22:09] <OanaStratulat> ubitux: ^ [22:09] <ubitux> yes sure so you confirmed what i was expecting [22:09] <OanaStratulat> the check for data is not there no? [22:10] <ubitux> i don't know where it's supposed to be if any [22:10] <ubitux> looking at the various other muxers, i don't see such check [22:10] <OanaStratulat> isn`t it normal to be here ? [22:10] <OanaStratulat> but maybe if i use the other muxers with the data codec like this they will bug too? [22:11] <OanaStratulat> if there is no check [22:11] <ubitux> well it makes sense not to set a timescale for data [22:11] <ubitux> but i'm not sure then why it is actually used [22:11] <OanaStratulat> so in the end what should i do ? [22:12] <ubitux> good question, i'm trying to figure out [22:12] <ubitux> the interesting thing is that it also happens in the MXF muxer [22:12] <ubitux> and maybe more [22:12] <ubitux> but it looks like a error reproduced each time to me ;) [22:13] <OanaStratulat> so it is in more muxers this error [22:13] <ubitux> as i mentionned, you have a very similar issue in the MXF (see the trac i pasted a while ago) [22:13] <ubitux> if you want to compare [22:13] <ubitux> http://ffmpeg.org/trac/ffmpeg/ticket/810 [22:14] <ubitux> for now, maybe someone else can enlight us about why it's done like this and what could be done [22:14] <ubitux> i never look at any muxer [22:14] <CIA-101> ffmpeg: 03Chris Evans 07master * rf35e037c93 10ffmpeg/libavformat/matroskadec.c: [22:14] <CIA-101> ffmpeg: mkv: Fix a bug where a pointer was cached to an array that might later move due to [22:14] <CIA-101> ffmpeg: a realloc() [22:14] <CIA-101> ffmpeg: BUG=100492 [22:14] <CIA-101> ffmpeg: Review URL: http://codereview.chromium.org/8366004 [22:14] <CIA-101> ffmpeg: Fixes: 1 of 2 for CVE-2011-3893 [22:14] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:14] <ubitux> i don't really know how it's supposed to work [22:14] <CIA-101> ffmpeg: 03Chris Evans 07master * r7149fce2ca 10ffmpeg/libavcodec/vorbis.c: [22:14] <CIA-101> ffmpeg: ogg: Avoid the possibility to read out-of-bounds of a static global array in Vorbis [22:14] <CIA-101> ffmpeg: decoding. [22:14] <CIA-101> ffmpeg: BUG=100543 [22:14] <OanaStratulat> it`s the same functions that has the 0 var [22:14] <CIA-101> ffmpeg: Review URL: http://codereview.chromium.org/8365014 [22:14] <CIA-101> ffmpeg: This fixes 25% of CVE-2011-3893 [22:14] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:14] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r6fcf2bb8af 10ffmpeg/libavcodec/vorbis.c: [22:14] <CIA-101> ffmpeg: vorbis: Fix last quarter of CVE-2011-3893 [22:14] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:14] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r405e99bdfd 10ffmpeg/libavcodec/vorbisdec.c: [22:14] <CIA-101> ffmpeg: vorbisdec: Make sure blocksize is not set to an invalid value. [22:14] <OanaStratulat> michaelni: ping [22:15] <CIA-101> ffmpeg: This possibly makes part of the CVE-2011-3895 fix unneeded. [22:15] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:15] <CIA-101> ffmpeg: 03Chris Evans 07master * rf74ce3a60d 10ffmpeg/libavcodec/vorbisdec.c: [22:15] <CIA-101> ffmpeg: vorbis: An additional defense in the Vorbis codec. [22:15] <CIA-101> ffmpeg: BUG=101458 [22:15] <CIA-101> ffmpeg: Review URL: http://codereview.chromium.org/8414025 [22:15] <CIA-101> ffmpeg: Fixes second part of CVE-2011-3895 [22:15] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:16] <OanaStratulat> ubitux: if you do not know i don`t know either, i`m waiting for someone to give an idea or something [22:17] <ubitux> have you tried to check for the zero ? [22:17] <OanaStratulat> in the video and audio? [22:17] <OanaStratulat> if it is 0? [22:17] <ubitux> i mean just if (xxx != 0) the_function_which_crash_when_its_zero() [22:17] <ubitux> av_rescale_rnd iirc [22:18] <ubitux> the resulting output file may be invalid, but it may avoid the crash [22:18] <OanaStratulat> ok try now [22:18] <ubitux> you could start looking that way [22:21] <ubitux> another solution is to try setting timescale to another value (let's say 1) in case it's a data stream [22:22] <OanaStratulat> if i check where timescale is used and use the function when the result is !=0 i get a segfault 11 [22:22] <OanaStratulat> so it`s not good doing like this [22:23] <ubitux> well of course there might be some more side effect to control [22:25] <OanaStratulat> ubitux: and doing if(st->codec->codec_type == AVMEDIA_TYPE_DATA){track->timescale=1;av_log(0,0,"\n\ndata\n\n\n");} also gives a segfault [22:25] <ubitux> interesting [22:25] <ubitux> since no one seems to be up for any hint you could try to look why it segfault [22:26] <ubitux> michaelni: if you have any hint btw& ;) [22:27] <ubitux> michaelni: the issue is that both mov & mxf muxer do not set any timescale for data streams, leading to floating exception since there is a need of ts all around [22:27] <OanaStratulat> http://pastebin.com/1zYB1NPg [22:27] <OanaStratulat> the valgrind [22:27] <ubitux> michaelni: so i'm not sure about trying to set a fake timescale, check everytime if it is set, or something else [22:28] <ubitux> OanaStratulat: run ffmpeg_g instead of ffmpeg [22:28] <OanaStratulat> ok [22:28] <ubitux> it will give more debug details [22:28] <OanaStratulat> but it is on ffmpeg_g [22:28] <OanaStratulat> the valgrind [22:28] <ubitux> mmh ok strange [22:28] <ubitux> heh indeed& [22:28] <ubitux> looks like serious corruption earlier [22:29] <ubitux> by 0x27... by 0x3... these are not valid addresses [22:29] <ubitux> looks like the stack is trashed somewhere i guess [22:30] <ubitux> i'll let you look by yourself for a while, i have some stuff to get done right now [22:30] <OanaStratulat> ok, ping me when back? [22:30] <ubitux> feel free to ask for any specific question though [22:30] <ubitux> i'm still there, just busy doing some stuff [22:31] <OanaStratulat> ok [22:41] <OanaStratulat> ubitux: tried with different timescales and with all it`s a segfault 11 [22:41] <OanaStratulat> ubitux: i`m trying to think that this bug may be out of my hand [22:42] <durandal_1707> is failing to read timestamp in nut muxed with ffmpeg worth reporting? [23:03] <michaelni> durandal_1707, everything is worth reporting [23:03] <michaelni> OanaStratulat, ubitux the bug is not trivial ... [23:03] <OanaStratulat> michaelni: so hard work here as i see [23:03] <cbsrobot> OanaStratulat: I tried a bit aswell [23:04] <cbsrobot> but no luck sofar [23:04] <cbsrobot> there is a default timesacel in the movenc.h [23:04] <cbsrobot> #define MOV_TIMESCALE 1000 [23:05] <cbsrobot> I tried to set the AVMEDIA_TYPE_DATA to this timescale but it did not work [23:05] <durandal_1707> michaelni: demuxer reports "no index at the end" [23:05] <OanaStratulat> yes tried as well with that timescale but it gives a segfault 11 [23:05] <cbsrobot> maybe some other default value is missing [23:06] <michaelni> durandal_1707, please open a ticket on trac, i dont have time atm :( [23:15] <OanaStratulat> michaelni: i will unclaim this task and claim one for a bug in libav, if i find another bug in ffmpeg which i can repair i will claim it again. sorry for this inconvenience [23:29] <durandal_1707> hmm, #844 should be closed, i definetely see content [23:30] <durandal_1707> nvm, perhaps op wants more than just title in rss [23:48] <ubitux> durandal_1707: extracting the content reliably is not easy [23:49] <ubitux> i don't think it's worth trying to do it though [00:00] --- Thu Jan 5 2012

1 0

ffmpeg.log.20120104
by burek 05 Jan '12

05 Jan '12

[00:04] <upvar> hey i'm trying to do a slice of an old episode of startrek and the output file (i only want 10-20 seconds) has some audio lag of about a second [00:04] <upvar> tried a variety of things, not sure what i can do to get the best sync [00:06] <pasteeater> Amok time. [01:25] <acerimmer> command line question: need to (1) convert .bmp image sequence to video and (2) overlay a logo on the video. Prefer to do this in one pass as multiple renders degrade video. [01:27] <pasteeater> acerimmer: two things to get you started: http://ffmpeg.org/faq.html#How-do-I-encode-single-pictures-into-movies_003f [01:27] <pasteeater> http://ffmpeg.org/libavfilter.html#overlay-1 [01:29] <acerimmer> pasteeater: thanks. [01:32] <acerimmer> I've successfully rendered a .mp4 from .bmp image sequence. Then rendered the mp4 AGAIN to overlay a .png logo. .mp4 is now visibly degraded. How do I render both image sequence and logo in a single pass? [01:34] <pasteeater> ffmpeg -i input%d.bmp -vf overlay="your over stuff goes here" -c:v libx264 -preset medium -crf 22 output.mp4 [01:42] <acerimmer> pasteeater: You Da Man! I'll try it. Thanks! [01:47] <pasteeater> your syntax can vary depending on how old your ffmpeg is [01:48] <acerimmer> pasteeater: seems to be missing a step - I need the .bmp's to render at 24 fps. [01:51] <acerimmer> pasteeater: http://pastebin.com/14K4Aawk for the present two step proces [01:54] Action: newl hasn't seen a bmp in 15 years [01:56] <acerimmer> newl: used Star Wars Crawl Text Creator on my ihackintosh to create the video intro. It only outputs .bmp [02:13] <pasteeater> acerimmer: then add '-r 24' before -i [02:14] <pasteeater> you don't need to do this in two steps [02:15] <acerimmer> pasteeater: that was my hope [03:00] <tlvince> I have ten images names 0{0..9}.jpg, using "ffmpeg -f image2 -r 5 -i %02d.jpg -s 640x480 output.mp4" seems to use only half the images? [03:02] <newl> half the images ie 5 or half the image in the image? is the image really there - check with feh [03:04] <tlvince> newl: half the images ie 5 [03:04] <tlvince> they show up in feh [03:05] <newl> do the script using ls instead of ffmpeg until you figure out the problem [03:05] <newl> pastbin the exact script [03:11] <tlvince> newl: https://gist.github.com/1558087 [03:12] <xxthink> are there some audio de-noise filter? [03:13] <tlvince> xxthink: not that I've specified [03:14] <newl> for i in *.jpg; do [03:14] <newl> if [ $i == \*.jpg ]; then [03:14] <newl> echo no jpg files [03:14] <newl> else [03:14] <newl> { [03:14] <newl> echo ffmpeg command using $i for files [03:14] <newl> ffmpeg command using $i for files [03:14] <newl> } [03:14] <newl> fi [03:14] <newl> done ^^ tlvince [03:16] <xxthink> tlvince: you mean there are not audio denoise filter in ffmpeg now? [03:28] <metvik> can ffmpeg normalise audio or do dynamic range compression? if not, is there any plan to implement this? [03:56] <SteveTheCat> I don't see x264 in my ffmpeg -codecs [03:57] <SteveTheCat> but i put --enable-libx264 [03:57] <SteveTheCat> in the ./configure [03:58] <newl> you don't see 113: D V D h264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 [03:58] <newl> 133: EV libx264 libx264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 [04:00] <SteveTheCat> newl I don't see libx264 [04:00] <SteveTheCat> but I do see h264 [04:02] <mystica555_> SteveTheCat: you do have the libraryt installed, right? [04:02] <mystica555_> *library [04:02] <SteveTheCat> yes [04:03] <mystica555_> did the ./configure script error on any of the libx264 tests? [04:03] <newl> yea check the config.log to see what happened when it looked for the libx264 [04:03] <SteveTheCat> i used --extra-cflags=-I/usr/local/customvlc/x264/include --extra-ldflags=-L/usr/local/customvlc/x264/lib [04:03] <SteveTheCat> i can't find any search results for missing [04:04] <SteveTheCat> I don't seen any errors for libx264 http://pastebin.com/DSzjXkzj [04:04] <SteveTheCat> see* [04:07] <mystica555_> hmmm [04:09] <SteveTheCat> what repos could be conflicting with this? [04:09] <SteveTheCat> it's probably some repo i have [04:11] <mystica555_> not certain; what i usually end up doing is putting all my third-party (locally done) stuff all in /usr/local; by default i find ./configure without a --prefix moves everything there. you may want to try this with your additional libraries; then, configuring ffmpeg to use /usr/local as its prefix should find your dependencies [04:13] <stpere> hi! [04:13] <stpere> quick question [04:13] <stpere> http://ffmpeg.org/doxygen/trunk/alsdec_8c-source.html#l01009 [04:13] <stpere> wouldn't zero_remaining() would contain an infinite loop? :P [04:18] <SteveTheCat> --prefix works for libs too? [04:21] <newl> SteveTheCat: prefix is just where it is to be installed - so yes [04:22] <newl> SteveTheCat: does the entries in /etc/ld.so.conf or cpp -v show where your custom include and libs for x264 are? [04:23] <SteveTheCat> i thought --extra-cflags=-I/usr/local/customvlc/x264/include --extra-ldflags=-L/usr/local/customvlc/x264/lib is okay [04:23] <mystica555_> SteveTheCat: --prefix i believe sets the entire prefix for default code searching of anything being compiled/installed [04:23] <SteveTheCat> i'll try it out now [04:24] <newl> mystica555_: just for install [04:24] <mystica555_> only install? [04:24] <mystica555_> hrmf [04:24] <SteveTheCat> :( [04:24] <newl> if not just for install then i've been mislead for 15 years ;) [04:24] <SteveTheCat> but honestly i think a repo is interferring [04:24] <mystica555_> so it doesnt make any bearing on where includes are? [04:24] <mystica555_> or where the compilation process finds said includes? [04:25] <newl> mystica555_: do a search on the system for the libs you have locate/slocate? [04:25] <mystica555_> yes [04:25] <mystica555_> although, im not the one with the unfindable libs problem today [04:26] <newl> lol /s/ mystica555_ / SteveTheCat ... unfindable libs problem 'TODAY' ? ;) [04:26] <mystica555_> ive had a few in the past due to pkg-config not finding its files ;) [04:27] <SteveTheCat> but i'm not getting an unfindable x264 lib [04:27] <mystica555_> well the includes for the local version seem to not be found.. [04:28] <maximumHAX> how can I get vhooks? do I have to compile ffmpeg a certain way.. I remember on slackware I had vhook/drawtext.so on gentoo I don't what happened? [04:28] <mystica555_> sort of similar, but i didnt use the best wording [04:29] <maximumHAX> seems my fedora machine has no drawtext.so either [04:30] <newl> that might be a devel lib - do you have the dev lib package installed [04:30] <maximumHAX> on gentoo I believe all would be dev.. but fedora no.. just ffmpeg [04:32] <newl> the so is a 'dev' link to the 'real' lib used in compiling - not running the app [04:33] <SteveTheCat> libavdevice.so.53: cannot open shared object file: No such file or directory [04:33] <SteveTheCat> now we have progresss my friends [04:34] <newl> that lib is created when you compile ffmpeg [04:34] <SteveTheCat> well apparently it works when i used yum install for libav [04:35] <SteveTheCat> and now i have x264 after deleting some repos and installing libav and compiling [04:38] <newl> magic [04:42] <aca20031> offtopic - for burning dvds for use on standard american dvd players - PAL or NTSC? [04:44] <newl> america is ntsc - but where are you getting the choice? [04:45] <aca20031> nero [04:45] <mystica555_> all of north america, parts of south america, japan, a few other countries [04:45] <mystica555_> use ntsc [04:45] <aca20031> dvd refuses to play in dvd player, windows media players the menu then says Unknown DVD when you make a choice :( [04:45] <aca20031> I was hoping that was the issue [04:45] <aca20031> apparently its not [04:47] <newl> aca20031: growisofs is kinda easy from command line - and ntsc vs pal ain't your problem [04:47] <aca20031> yeah well i need a program that creates a decent dvd menu that i can customize, took me all day to find that nero works fine (encore: failed, roxio: failed) [04:47] <aca20031> finally get nero, make my menus and dvd covers, and the damn thing wont play [04:48] <aca20031> freaking rediculous [04:48] <newl> try mplayer dvd:// [04:49] <aca20031> i kinda need it to play on a dvd player, if i just wanted to view it on a computer i wouldnt be burning :P [04:49] <aca20031> thanks though [04:49] <mystica555_> well if mplayer barfs on it, perhaps nero is doing something reallly wrong [04:50] <aca20031> it plays in my ps3 [04:50] <aca20031> windows media player plays the menu, but the film, and a dvd player doesnt play shit lol [04:50] <aca20031> its rediculous [04:50] <aca20031> feel like i live in bizzaro world [04:50] <newl> is it encrypted? libdvdcss needed? [04:50] <aca20031> erm, im not sure why nero would encrypt anything [05:30] <SteveTheCat> i'm currently converting with ffmpeg [05:30] <SteveTheCat> does time=01:53:55.42 mean 1 hour and 53 minutes left? [05:59] <Skaag> how do I find out what qmin and qmax mean when I encode a clip with libx264? [06:03] <defaultro> cat it then grep x264 [06:04] <Skaag> I'm experimenting with those params, to achieve better encoding quality [06:04] <Skaag> right now I've set qmin to 8 and qmax to 10 [06:05] <Skaag> I wonder if there's a tool for Mac that analyzes mp4 files to show bit rate behavior in VBR streams [06:05] <Skaag> I think I saw something like that for Windows [06:31] <Skaag> can ffmpeg do auto key framing on scene change? [06:49] <Skaag> ok I found the scene cut parameter [08:42] <severnaya> hi, i'm trying to compile ffmpeg but it keeps breaking with the error: libmpcodecs/vd_ffmpeg.c:833:10: error: 'AVCodecContext' has no member named 'hurry_up'. and then vd_ffmpeg.o fails to build [08:43] <severnaya> can anyone point me in the right direction? [09:44] <simon> hello, someone here? i have a question regarding MPEG-TS muxing [09:51] <sprayd> hello michaelni [09:54] <sprayd> i hope you remember this task michaelni http://www.google-melange.com/gci/task/view/google/gci2011/7185251 [09:58] <sprayd> i never got reply to the letters i sent to the iso authorities [12:55] <MrChrisDruif> Hello everyone, just a quick question: how do you use the verbosity level with ffmpeg? Is it -v verbose or what should the proper syntax be [12:57] <MrChrisDruif> The question I asked is for smile4ever, who just joined this channel [12:57] <smile4ever> :) [13:10] <MrChrisDruif> No one? [13:12] <smile4ever> please help me :) [15:36] <egoc> hi. i'm trying to encode an avi video to mkv or mp4 but i keep getting this error. can anyone help? http://pastebin.com/BuDDWrbe [15:39] <relaxed> egoc: -ab 128k [15:40] <JEEB> yeah, you haven't set a bitrate for audio [15:40] <JEEB> also personally I would just use the highest crf that still looks good for you instead of an arbitrary bitrate [15:41] <egoc> i still get the same error [15:41] <relaxed> rm -bt 1000k [15:42] <egoc> relaxed: still not working [15:42] <JEEB> re-pastebin pl0x [15:43] <relaxed> I see "[libfaac @ 0x14ef700] libfaac doesn't support this output format!" but don't know what it's referring to. [15:43] <egoc> http://pastebin.com/WxYXf5sz [15:44] <j^> ffmpeg -h lists "ffmpeg -vf filter list" as a command to list filters [15:44] <j^> that does not work, anyone knows how to list the filters? [15:46] <egoc> that's what i get when i remove -bt and add -ab [15:46] <relaxed> egoc: It's tripping up on the audio. run, "dd if=input.avi of=sample.avi bs=5M count=1" and upload the sample somewhere. [15:47] <egoc> relaxed: where should i upload it? [15:48] <relaxed> egoc: get a dropbox account for 2gb free storage http://db.tt/cDWPLzP [15:52] <relaxed> j^: ffmpeg -filters [15:54] <egoc> relaxed: http://www.mediafire.com/download.php?9psd6fvukyc92r3 [15:58] <relaxed> egoc: Works here although I'm using libvo_aacenc instead of libfaac. Compiling a recent version ffmpeg will probably fix it. [16:11] <egoc> relaxed: still not working. how do i use libvo_aacenc? i get Unknown encoder 'libvo_aacenc' [16:12] <relaxed> It should be available through your distro. [16:13] <relaxed> Install the -dev pacakge and configure ffmpeg with --enable-libvo-aacenc, then use -acodec libvo_aacenc [17:30] <gtaylor> I'm doing a 2-pass encoding of an AVI that appears to be about 10 years old. The AVI file's audio/video are fine, but when I 2-pass encode it with h264 and libfaac to mp4, I get slowed down audio and a ton of static. [17:30] <gtaylor> Here is the output of ffmpeg -i and my encoding settings for the first and second pass, respectively: http://pastebin.com/XdD0XcR3 [17:31] <gtaylor> There are warnings about packets/times being invalid for the audio and video streams in ffmpeg -i. [17:40] <raven> hi [17:41] <silverrocker> Hi, I have a question for the ffmpeg community. I have my bachelor thesis comming up (which will start in about a month). I'm intressted in videocompression and things that are related to multimedia compression. I'm still quite unsure what my subject will be. I thought it might be a good place to ask here. I have read some of the SoC project proposals but still I feel quite lost. [17:45] <raven> 80gb raw dv file freezes at a point and ffplay is not controllable any more while time is still running. what could that be? other tools tell me about eof at this point [17:57] <raven> hi [18:16] <sgfgdf> hello, guys! i'm trying to convert vob file to mp4 or mov file, but when using ffmpeg -i file.vob -vcodec copy -acodec copy file.mp4 i got erros -- http://cl.ly/190X1Q2k2k3G113V3f0j . can onyone help, please? [18:24] <Nagy> will open_codec now automatically start threads? even if thread_count is set to 0? [18:25] <relaxed> sgfgdf: why would you want to copy it into mp4? [18:26] <sgfgdf> relaxed: i need to convert that vob to format that Final Cut Pro opens (which i hope mp4 is one of it). [18:27] <vivienschilis_> quick question why is there a ffpresets and presets folder? [18:27] <vivienschilis_> in the git [18:31] <relaxed> sgfgdf: change the filename from file.vob to file.mpeg and see if it will open [18:35] <sgfgdf> relaxed: no it doesn't want to open. [18:38] <relaxed> sgfgdf: ntsc or pal? [18:40] <relaxed> ffmpeg -i file.vob -target ntsc-dv output.dv (output will be HUGE) [18:42] <sgfgdf> relaxed: so what the .dv file means? [18:42] <relaxed> http://en.wikipedia.org/wiki/DV [18:47] <sgfgdf> relaxed: does only because of the uncompressed audio the file is going to be so big? [18:51] <sgfgdf> relaxed: anyway, probably it will work. thank you! [20:10] <_sascha> i, is it possible to switch audio channels (example switch audio channel 1 (german) with channel 2 (englisch)) so that the new channel 1 is english and channel 2 is german ? [20:24] <cbreak> _sascha: in a stereo audio track? [20:25] <cbreak> mplayer can do that with -af pan or -af channels [20:25] <cbreak> but it'll need reencoding of the audio [20:40] <relaxed> ls [20:40] <relaxed> sorry [20:43] <MrJones> hi [20:44] <MrJones> is it actually possible to replace a program's dynamically linked FFmpeg with another without issues? due to libavcodec/avconf.h [20:45] <MrJones> so will a program compiled with one particular set of FFmpeg features reflected through that header file actually recognise another FFmpeg dll with another featureset properly? [20:47] <pasteeater> _sascha: i think you can use the map option to arrange desired stream [20:51] <pasteeater> as in: ffmpeg -i input -map 0:0 -map 0:4 -map 0:1 -c copy output [20:54] <Freakshow> hmm... is -strict experimental still used by anyone? [20:55] <pasteeater> Freakshow: -acodec aac, -acodec vorbis, probably some more [20:58] <cbreak> MrJones: that's the point of dynamic libs :) As long as they are binary compatible, there's no problem from the pure linking perspective [20:59] <MrJones> cbreak: so are things like the available codecs and other stuff, despite surely being stored in that avconf.h aswell, still taken out of the runtime lib? [20:59] <JEEB> cbreak, you for get one thing [20:59] <JEEB> that works as long as the ABI is the same [20:59] <MrJones> yea, and do different options alter the ABI for FFmpeg? [20:59] <cbreak> the ABI is decided on by the compiler/operating system vendor [20:59] <cbreak> I included it in binary compatibility [20:59] <JEEB> and the application at hand >_> [21:00] <JEEB> if you have made something with ffmpeg 0.6 [21:00] <JEEB> it might not work as-is with ffmpeg 0.9 [21:00] <JEEB> :P [21:00] <MrJones> well it is surely obvious that FFmpeg has enough possibilities to ruin the ABI [21:00] <MrJones> which is why I am asking if it does with different options [21:00] <JEEB> anyways, not sure but it's mostly safe in case you're linking against a similar'ish version [21:00] <cbreak> options should not affect binary compatibility [21:00] <MrJones> so same version, vastly different options (like, LGPL with no output devices, no input devices, no protocols) will still be compatible with a GPL dll with output devices? [21:00] <cbreak> unless you do something stupid like microsoft [21:01] <MrJones> and be able to use them? [21:01] <cbreak> and change the size of types that you pass by value [21:02] <MrJones> e.g. AVCODEC_MAX_AUDIO_FRAME_SIZE [21:02] <MrJones> doesn't it change with the choice of audio codecs? [21:02] <MrJones> or is it always the worst case of all audio codecs I could have chosen [21:03] <JEEB> I would guess the latter [21:07] <MrJones> hm [21:08] <MrJones> the avconf.h file apparently contains only whether it is a big endian system or fast unaligned is available [21:08] <MrJones> so not terribly much :) [21:37] <_sascha> Hi, I got an error and dont know what to do [21:37] <_sascha> command: ffmpeg -i in.mkv -vcodec copy -acodec copy -scodec copy -y out.mkv [21:37] <_sascha> error: [matroska @ 0x2028f00] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 82 >= 42 [21:39] <relaxed> _sascha: are you using a recent version of ffmpeg? [21:41] <_sascha> yes... 0.9 from a ppa [21:46] <_sascha> relaxed: first I tried it with an older version, but an upgrade didn't solve the problem [21:51] <basilgohar> I see reference to -lameopts in some documentation in the ffmpeg source tree, but there does not appear to be any actual -lameopts parameter used, even when used in conjuction with -acodec libmp3lame. [21:51] <basilgohar> I'd like to use features like LAME's presets, which I believe are only accessible through -lametops. [21:51] <basilgohar> (opts [21:51] <basilgohar> Any ideas? [22:02] <pasteeater> basilgohar: the only private libmp3lame option i see is -reservoir [22:03] <basilgohar> pasteeater: Okay. But I believe there used to be -lameopts, and that is referenced all over the place, but it does not seem to be valid anymore. [22:04] <relaxed> _sascha: use mkvmerge [22:05] <pasteeater> basilgohar: ah, i see it's in libavfilter/libmpcodecs/help_mp.h [22:05] <pasteeater> must be a mencoder option [22:05] <_sascha> ok, thx [22:05] <basilgohar> pasteeater: Yes, I see it. I just did a git log -S and found ti was introduced only in documentation about MPLayer. [22:06] <basilgohar> So, it's not something ffmpeg itself has ever had. [22:06] <pasteeater> you could add a feature request on the bug tracker [22:07] <basilgohar> Okay. Thanks! [22:07] <basilgohar> ffmpeg handles tags nicely from FLAC to MP3, which is why I'm interested. [22:07] <pasteeater> what lame options are you interested in? [22:07] <basilgohar> Only because my cell phone and, it seems, many car audio head units only support MP3. [22:07] <basilgohar> pasteeater: Mainly -preset standard [22:08] <basilgohar> I don't think that's the same as just VBR encoding, so just using -aq whatever won't be the same. [22:10] <pasteeater> preset standard appears to be VBR [22:11] <pasteeater> md5sum of '-V2' and '--preset standard' are the same [22:11] <pasteeater> so i guess use -q:a 2 [22:12] <basilgohar> pasteeater: That's interesting...I'll have to compare the two and see for myself. [22:12] <basilgohar> It could be source-dependent, as well. [22:12] <basilgohar> I know, for example, -preset standard doesn't go below 128kbps, for example. [22:12] <basilgohar> At least, it once was like that. [00:00] --- Thu Jan 5 2012

1 0

ffmpeg-devel.log.20120103
by burek 04 Jan '12

04 Jan '12

[00:05] <durandal_1707> nice, chained oggs are not supported [00:17] <durandal_1707> does ffmpeg correctly mux flac in mov? [00:21] <Daemon404> .... sincewhen does mov even support flac? [00:22] <durandal_1707> ehehe, https://trac.videolan.org/vlc/ticket/5703 [00:22] <durandal_1707> since ffmpeg [00:23] Action: Daemon404 s/https/http/ [00:23] <Daemon404> im not sure supporting crap like that is a very good idea [00:24] <kierank> that's retarded [00:24] <j-b> beyond that [00:24] <j-b> completly stupid [00:24] <durandal_1707> than do not allow muxing random codec into mov [00:25] <kierank> i mean if flac was registered as allowed in mov then i guess it would be acceptable [00:25] <kierank> but making it work for fun is a bit silly [00:25] <durandal_1707> there is no flac in isom.c [00:33] <iive> i don't find the idea that bad, as long as the implementation is clean. It probably is the easiest way to pave the way for official flac acceptance. [00:34] <Daemon404> iive, i object to official flac accemptance [00:34] <Daemon404> the format and specs are flawed [00:34] <Daemon404> <.< [00:34] <iive> then, it is time to write version2. [00:35] <durandal_1707> but ms[241][172 is not right fourcc/whatever [00:35] <durandal_1707> copy one from riff? [00:35] <durandal_1707> btw, is flac allowed in avi? [00:35] <Daemon404> how about you proply fail [00:35] <Daemon404> and dont let them mux it in teh first place [00:36] <durandal_1707> qt specifications have partial tables [00:37] <Daemon404> hmm? [00:39] <durandal_1707> image/audio commpression formats and associated "compression type" : cvid,jpeg,smc,rle,... [00:40] <Daemon404> im not sure where this is located, and hpw it's relevant [00:40] <Daemon404> (to flac) [00:41] <durandal_1707> http://wiki.multimedia.cx/index.php?title=Mov [00:45] <kierank> there is an official list of formats allowed in mov [00:45] <kierank> forget where [00:48] <ffmpeguser> http://mango.blender.org/production/mango-synopsis/#comment-540 [00:53] <gnafu> XD [00:53] <gnafu> kierank: If you find a link, I'd love to have it. I was trying to find something like that the other day :-P. [00:55] <michaelni> btw, politics aside, ffplay plays the flac in mov fine [01:13] <durandal_1707> people still use mencoder [01:16] <JEEB> I think there was a system for you to specify your own subformat or whatever in mov/iso media? That would in theory make FLAC in it not be nonstandard >_> [01:16] <iive> yeh, it barely works, and yet, it is still one of the easiest programs to work with. [01:16] <JEEB> although I don't even remember the name of the site that lists those registered things [01:28] <durandal_1707> "In my system (through my ears and with my brain) ape sounds better than flac and almost indistinguishable compared to wav: ..." [01:30] <gnafu> wat [01:31] <durandal_1707> every time i visit HA i read something stupid [01:36] <gnafu> "If it hurts, don't do it!" [01:36] <gnafu> ;D [01:37] <gnafu> But maybe you're into that sort of thing. [01:44] <iive> durandal_1707: ape indeed sounds better than flac. I mean... what rimes with flac? ;) [01:51] <durandal_1707> :) [01:55] <drv> 8-track? [01:58] Action: iive explaining the joke. I was talking about the names of the codecs, not the output. The output is binary identical. [03:14] <CIA-101> ffmpeg: 03Anton Khirnov 07master * rc88d53199b 10ffmpeg/avconv.c: [03:14] <CIA-101> ffmpeg: avconv: remove the 'codec framerate is different from stream' warning [03:14] <CIA-101> ffmpeg: Codec timebase is not framerate and the situation when it's different [03:14] <CIA-101> ffmpeg: from AVStream.r_frame_rate is quite common, nothing to warn about. [03:14] <CIA-101> ffmpeg: 03Hendrik Leppkes 07master * r14165fe125 10ffmpeg/libavcodec/wavpack.c: [03:14] <CIA-101> ffmpeg: wavpack: determine sample_fmt before requesting a buffer [03:14] <CIA-101> ffmpeg: Signed-off-by: Justin Ruggles <justin.ruggles(a)gmail.com> [03:14] <CIA-101> ffmpeg: 03Janne Grunau 07master * raf9240cd3c 10ffmpeg/libavformat/mpegenc.c: [03:14] <CIA-101> ffmpeg: mpegenc: simplify muxrate calculation [03:14] <CIA-101> ffmpeg: The fate-h264-bsf-mp4toannexb failures were caused by an integer [03:14] <CIA-101> ffmpeg: overflow of the unneeded multiplication. [03:14] <CIA-101> ffmpeg: Inspired by patch by: Michael Niedermayer <michaelni(a)gmx.at> [03:14] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r03fafdec82 10ffmpeg/libavcodec/mpegaudiodec.c: [03:14] <CIA-101> ffmpeg: mpegaudiodec: skip all channels when skipping granules [03:14] <CIA-101> ffmpeg: Also fix calculation of new position when switching buffers. [03:14] <CIA-101> ffmpeg: This fixes "overread" error messages when seeking. [03:14] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r8893fbdf7a 10ffmpeg/libavcodec/bmv.c: bmv audio: implement new audio decoding API [03:14] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r24964f21e4 10ffmpeg/libavcodec/mjpegdec.c: [03:14] <CIA-101> ffmpeg: jpegdec: Only enable rgb mode when there are 3 components. [03:14] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:14] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r3a1867deac 10ffmpeg/libavcodec/wavpack.c: wavpack: fix typos in previous cosmetic clean-up commit [03:14] <CIA-101> ffmpeg: 03Reinhard Tartler 07master * r302371c055 10ffmpeg/doc/RELEASE_NOTES: [03:15] <CIA-101> ffmpeg: Release notes: mention cleaned up header includes [03:15] <CIA-101> ffmpeg: A rebuild in debian revealed that this broke compilation of a number packages: [03:15] <CIA-101> ffmpeg: http://bugs.debian.org/cgi-bin/pkgreport.cgi?users=siretart@tauware.de;tag=… [03:59] <CIA-101> ffmpeg: 03Jean First 07master * rf86c348d46 10ffmpeg/libavcodec/vp3.c: [03:59] <CIA-101> ffmpeg: vp3: remove redeclaration of vp3_decode_flush [03:59] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [03:59] <CIA-101> ffmpeg: Reviewed-by: Paul B Mahol <onemda(a)gmail.com> [03:59] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:59] <CIA-101> ffmpeg: 03Jean First 07master * r535705a489 10ffmpeg/ffmpeg.c: [03:59] <CIA-101> ffmpeg: ffmpeg: use avformat_close_input instead of av_close_input_file [03:59] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [03:59] <CIA-101> ffmpeg: Reviewed-by: Paul B Mahol <onemda(a)gmail.com> [03:59] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:59] <CIA-101> ffmpeg: 03Jean First 07master * rf2eb2e5c4f 10ffmpeg/libavformat/wtvdec.c: [03:59] <CIA-101> ffmpeg: wtvdec: use intfloat.h instead of intfloat_readwrite.h [03:59] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [03:59] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:59] <CIA-101> ffmpeg: 03Paul B Mahol 07master * r7099f36578 10ffmpeg/libavcodec/ (10 files): [04:00] <CIA-101> ffmpeg: 03Jean First 07master * r12bdc7b15e 10ffmpeg/ffmpeg.c: [04:00] <CIA-101> ffmpeg: ffmpeg: fix compiler warning for uninitialized variables [04:00] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [04:00] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [06:31] <CIA-101> ffmpeg: 03Michael Bradshaw 07master * r3f07ef1dff 10ffmpeg/libavcodec/libopenjpegdec.c: [06:31] <CIA-101> ffmpeg: Added RGBA and YUV440 decoding support for libopenjpeg decoder [06:31] <CIA-101> ffmpeg: Reviewed-by: Jean First <jeanfirst(a)gmail.com> [06:31] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [06:31] <CIA-101> ffmpeg: 03Michael Bradshaw 07master * rd1669e5fe3 10ffmpeg/libavcodec/libopenjpegdec.c: [06:31] <CIA-101> ffmpeg: Fixed openjpeg decoding bug with width/linesize issue [06:31] <CIA-101> ffmpeg: The original code wasn't taking into account the fact that linesize may not equal the frame's width. This is to correct that. [06:31] <CIA-101> ffmpeg: Signed-off-by: Michael Bradshaw <mbradshaw(a)sorensonmedia.com> [06:31] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [06:31] <CIA-101> ffmpeg: 03Paul B Mahol 07master * r69d766daa0 10ffmpeg/libavcodec/ (g723_1.c libaacplus.c nellymoserdec.c): [06:31] <CIA-101> ffmpeg: lavc: replace rest of deprecated SAMPLE_FMT_* with AV_SAMPLE_FMT_* [06:31] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [06:31] <CIA-101> ffmpeg: 03Michael Bradshaw 07master * r2ba3416362 10ffmpeg/libavcodec/libopenjpegdec.c: [06:31] <CIA-101> ffmpeg: Added yuva420p decoding support for libopenjpeg [06:31] <CIA-101> ffmpeg: Signed-off-by: Michael Bradshaw <mbradshaw(a)sorensonmedia.com> [06:31] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [10:19] <Granjow> Good morning! I've been trying to find the reason for the segfault in http://codepad.org/tyEekKMe for hours now but without success. The segfault occurs on line 128 in avcodec_encode_video(), but some symbols already disappear after the sws_scale() (line 89) [10:20] <Granjow> The file should compile with just the ffmpeg libraries, and is assumed to read frames from a video and export them as PNG. [10:21] <Granjow> Later I will maybe want to keep them in RAM or scale them or so. [10:22] <Granjow> But I cannot find the reason for the segfault; imho I've done everything correctly. [13:53] <CIA-101> ffmpeg: 03Michael Niedermayer 07release/0.9 * r1506becf36 10ffmpeg/libavcodec/h264.c: [13:53] <CIA-101> ffmpeg: h264: Print pass in NAL debug output [13:53] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:53] <CIA-101> ffmpeg: (cherry picked from commit 393253b4b3b454830b22c8e69acbe11e2684da7e) [13:53] <CIA-101> ffmpeg: 03Michael Niedermayer 07release/0.9 * r96545777d7 10ffmpeg/libavcodec/h264.c: [13:53] <CIA-101> ffmpeg: h264: add forgotten \n to "Invalid mix of idr and non-idr slices" [13:53] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:53] <CIA-101> ffmpeg: (cherry picked from commit ecb14b8af73b92e5a1be47c119d2f528ff402ebd) [13:53] <CIA-101> ffmpeg: 03Michael Niedermayer 07release/0.9 * r01ff5378f0 10ffmpeg/libavcodec/h264.c: [13:53] <CIA-101> ffmpeg: h264: reset picture->sync on allocation. [13:53] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:53] <CIA-101> ffmpeg: (cherry picked from commit 36fa781bd6b4e0f40abd31431af2d29d87e9b194) [13:53] <CIA-101> ffmpeg: 03Michael Niedermayer 07release/0.9 * r23ed56d307 10ffmpeg/libavcodec/h264_refs.c: [13:53] <CIA-101> ffmpeg: h264: Fix seeking in 00017.MTS [13:53] <CIA-101> ffmpeg: Fixes Ticket852 [13:53] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:53] <CIA-101> ffmpeg: (cherry picked from commit 9d9316ae34ea2a5e481ac15f31d42c87a8ef4380) [13:54] <CIA-101> ffmpeg: 03Piotr Bandurski 07master * r912e75169b 10ffmpeg/libavcodec/sunrast.c: [13:55] <CIA-101> ffmpeg: 03Michael Niedermayer 07release/0.9 * r2a442eb70f 10ffmpeg/libavcodec/ (h264.c h264_refs.c): [13:55] <CIA-101> ffmpeg: h264: Fix frame sync / random access handling. [13:55] <CIA-101> ffmpeg: It seems I have misunderstood the spec when I implemented this [13:55] <CIA-101> ffmpeg: originally. [13:55] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:55] <CIA-101> ffmpeg: (cherry picked from commit 9ba43e50efa2483fb3ca1e3f62a474db7dd3ac8d) [13:56] <CIA-101> ffmpeg: 03Michael Niedermayer 07release/0.9 * rb84683a906 10ffmpeg/libavcodec/h264.c: [13:56] <CIA-101> ffmpeg: h264: Fix recovery_frame initialization when recovery_frame is -1 [13:56] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [13:56] <CIA-101> ffmpeg: (cherry picked from commit fba6777363eb74d926aa0eed281a77ff152a3ad5) [13:56] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07release/0.9 * r980e4263bc 10ffmpeg/libavcodec/libopenjpegenc.c: (log message trimmed) [13:56] <CIA-101> ffmpeg: Add missing PIX_FMT_NONE to libopenjpegenc [13:56] <CIA-101> ffmpeg: fixes ticket #813 [13:56] <CIA-101> ffmpeg: Reviewed-by: Michael Niedermayer <michaelni(a)gmx.at> [13:56] <CIA-101> ffmpeg: Reviewed-by: Jean First <jeanfirst(a)gmail.com> [13:56] <CIA-101> ffmpeg: Reviewed-by: Paul B Mahol <onemda(a)gmail.com> [14:06] <CIA-101> ffmpeg: 03Hendrik Leppkes 07release/0.9 * ra46fa584c5 10ffmpeg/libavcodec/mlpdec.c: [14:06] <CIA-101> ffmpeg: mlpdec: fix channel order for wide 7.1 truehd layouts [14:06] <CIA-101> ffmpeg: (cherry picked from commit b9d8af03270223202c9a991655d3f75b5a8b56b5) [14:06] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07release/0.9 * r55e2dc7f77 10ffmpeg/libavcodec/mlpdec.c: [14:06] <CIA-101> ffmpeg: Fix channel order for some less common TrueHD layouts. [14:06] <CIA-101> ffmpeg: Reported and tested by TDiTP_ [14:06] <CIA-101> ffmpeg: Fixes ticket #779. [14:06] <CIA-101> ffmpeg: (cherry picked from commit 1af0ace3a4b8aa8c6240a99de3c40f495ab279e9) [16:56] <durandal_1707> what is 186 byte in avi? [17:02] <av500> wuz? [17:09] <durandal_1707> it should be depth [17:10] <av500> molim? [17:12] <durandal_1707> are you using google translate? [17:12] <av500> zasto? [17:13] <durandal_1707> you should talk in english [23:00] <durandal_1707> so bunch of old bugs need to be moved to trac? [23:03] <CIA-101> ffmpeg: 03Jean First 07master * refaf448888 10ffmpeg/libavformat/rtsp.c: [23:03] <CIA-101> ffmpeg: rtsp: Fix compiler warning for uninitialized variable. [23:03] <CIA-101> ffmpeg: Reviewed-by: Martin Storsjö [23:06] <durandal_1707> WTF [23:06] <durandal_1707> Option -pix_fmt is deprecated, use -pixel_format. [23:06] <durandal_1707> Option pixel_format not found. [23:08] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07master * r001567c767 10ffmpeg/libavcodec/y41penc.c: [23:08] <CIA-101> ffmpeg: Set bits_per_coded_sample when encoding Y41P. [23:08] <CIA-101> ffmpeg: This allows mplayer -vc bt411 to decode files [23:08] <CIA-101> ffmpeg: encoded with FFmpeg. [23:08] <CIA-101> ffmpeg: Reviewed-by: Paul B Mahol [23:17] <michaelni> -pixel_format / pix_fmt should work with inputs where it makes sense [23:18] <michaelni> the error message is really sick trash though [23:53] <CIA-101> ffmpeg: 03Mashiat Sarker Shakkhar 07master * r001a74ff53 10ffmpeg/libavcodec/wmalosslessdec.c: Fix AC filter buffers and AC filter reversion [23:53] <CIA-101> ffmpeg: 03Mashiat Sarker Shakkhar 07master * r8dc8dd9f6f 10ffmpeg/libavcodec/wmalosslessdec.c: Reset acfilter_prevvalues buffer in clear_codec_buffers() [23:53] <CIA-101> ffmpeg: 03Mashiat Sarker Shakkhar 07master * r3d84495128 10ffmpeg/libavcodec/wmalosslessdec.c: [23:53] <CIA-101> ffmpeg: Use quantizer value read from bitstream [23:53] <CIA-101> ffmpeg: Currently the code initializes quantizer even [23:53] <CIA-101> ffmpeg: before reading it from bitstream and uses that [23:53] <CIA-101> ffmpeg: to dequantize samples. [23:53] <CIA-101> ffmpeg: 03Mashiat Sarker Shakkhar 07master * rf7166f7d8d 10ffmpeg/libavcodec/wmalosslessdec.c: Suppress dumping of residues buffer [23:53] <CIA-101> ffmpeg: 03Mashiat Sarker Shakkhar 07master * r2083a10abf 10ffmpeg/libavcodec/wmalosslessdec.c: (log message trimmed) [23:53] <CIA-101> ffmpeg: Fix audio output [23:53] <CIA-101> ffmpeg: Up until now, the decoder didn't output anything [23:54] <CIA-101> ffmpeg: in the data buffer. This fixes all the issues [23:54] <CIA-101> ffmpeg: related to sample format, removes leftover code [23:54] <CIA-101> ffmpeg: and actually outputs some audio to data buffer. [23:54] <CIA-101> ffmpeg: With this, the only sample we have can be played. [23:54] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * rdde545c697 10ffmpeg/: (log message trimmed) [23:54] <CIA-101> ffmpeg: Merge remote-tracking branch 'shariman/wmall' [23:54] <CIA-101> ffmpeg: * shariman/wmall: [23:54] <CIA-101> ffmpeg: Fix audio output [23:54] <CIA-101> ffmpeg: Suppress dumping of residues buffer [23:54] <CIA-101> ffmpeg: Use quantizer value read from bitstream [23:54] <CIA-101> ffmpeg: Cosmetics: Remove two empty lines and realign some code [23:54] <CIA-101> ffmpeg: 03Mashiat Sarker Shakkhar 07master * ra6da23f146 10ffmpeg/libavcodec/wmalosslessdec.c: Cosmetics: Remove two empty lines and realign some code [23:55] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r6429269096 10ffmpeg/libavcodec/wmalosslessdec.c: [23:55] <CIA-101> ffmpeg: wmall: remove ;; [23:55] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [23:57] <ubitux> michaelni: i think e1a8164d is an important fix, and it seems needed for 0.9.x [23:57] <ubitux> ah my bad, seems to be ok [23:57] <ubitux> forget this [23:57] <ubitux> sorry [23:58] <ubitux> btw, just wrote a silencedetector audio filter [23:59] <ubitux> anyone interested in such thing? [23:59] <ubitux> saste: http://blackhole.ubitux.fr/0001-lavfi-WIP-add-audio-silencedetect-filter.pa… [23:59] <ubitux> think it's worth trying to finish it properly (handle sample rate changes, support more sample fmt, &) [23:59] <ubitux> +do you +? [00:00] --- Wed Jan 4 2012

1 0

ffmpeg.log.20120103
by burek 04 Jan '12

04 Jan '12

[00:02] <relaxed> works fine using avconv [00:02] <cbsrobot> works using ffmpeg too [00:03] <iiu7> Then what's wrong at my place? [00:03] <cbsrobot> compile it yourself [00:03] <cbsrobot> with latest git head [00:03] <iiu7> Yeah, I think I'll have to do that! [00:04] <iiu7> Thanks all! Great help! :) [00:54] <SteveTheCat> how do you change a video encoding of a file with ffmpeg? [00:59] <SteveTheCat> I'm trying to do ffmpeg -i file.avi -vcodec h264 OutputFile.avi but i'm getting file.avi: Invalid data found when processing input [01:11] <iiu7> I compiled ffmpeg from the latest source and everything is working now! Thanks again for all the help! [01:26] <SteveTheCat> how do you skip a stream in your file? [01:26] <SteveTheCat> i'm having problems converting a file that has vorb in it but I don't need that audio track [01:28] <cbsrobot> -an [01:29] <SteveTheCat> also how do you view what codecs the file is using? [01:30] <SteveTheCat> nevermind [01:30] <SteveTheCat> it didn't show before for some reason [01:41] <SteveTheCat> anyone know where i can grab a source of aacplus devel? [01:58] <dp_> is there a way to tell ffmpeg to create the output with the same name (save the extension) as the input file? [02:04] <pasteeater> dp_: no, but it can be done easily in bash [02:08] <dp_> pasteeater: that's what I figured. which means I'm going to have to figure out when rtorrent triggers the move properly [02:08] <pasteeater> SteveTheCat: here i think: http://tipok.org.ua/node/17 [02:08] <SteveTheCat> thanks pasteeater [02:09] <pasteeater> http://217.20.164.161/~tipok/aacplus/libaacplus-2.0.2.tar.gz [02:09] <pasteeater> also, -vcodec h264 is the name of the H.264 decoder. the encoder is libx264 [02:30] <SteveTheCat> I'm getting ./ffmpeg: error while loading shared libraries: libavdevice.so.53: cannot open shared object file: No such file or directory [02:30] <SteveTheCat> but i also compiled libav, how do i get ffmpeg to use my compiled libav instead of the older version i used yum install on? [02:48] <SteveTheCat> what libav would be compatible with ffmpeg 0.9? [02:48] <SteveTheCat> to compile with? [02:50] <pasteeater> https://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide [02:51] <pasteeater> or just trying running "ldconfig" and then "ffmpeg" again [02:51] <pasteeater> libav and ffmpeg will conflict if you install them in the same places [02:52] <pasteeater> ffmpeg won't "use" the libav fork. [02:52] <pasteeater> if that's what you're talking about [02:52] <SteveTheCat> this is what I get: # ffmpeg ffmpeg: error while loading shared libraries: libswresample.so.0: cannot open shared object file: No such file or directory [02:52] <pasteeater> did you run ldconfig? [02:53] <SteveTheCat> but i have libswscale.so.2.1.0 [02:53] <SteveTheCat> yes [02:53] <SteveTheCat> wrong file [02:53] <pasteeater> i don't know how you compiled anything, or if you're mixing repo stuff with compiled stuff [02:53] <SteveTheCat> it seems i really odn't have it [02:54] <SteveTheCat> i probably am mixing repo stuff with compiled stuff [02:56] <SteveTheCat> I deleted some of the repo stuff that was conflicting I believe but now I'm getting a different lib problem: ./ffmpeg: error while loading shared libraries: libpostproc.so.51: cannot open shared object file: No such file or directory [02:57] <SteveTheCat> is it becaues I didn't ./configure libav properly? [02:58] <pasteeater> i don't know. might want to ask in #libav if that's what you're using [02:58] <pasteeater> how did you configure it? [02:59] <SteveTheCat> like so: ./configure --enable-shared --prefix=/usr/local/customvlc/libav [03:12] <SteveTheCat> got it all working [03:12] <SteveTheCat> thanks for the advice pasteeater [03:13] <pasteeater> SteveTheCat: what did you do? [03:13] <SteveTheCat> got rid of repos that i made compilations of before, was conflicting with eachother [03:14] <SteveTheCat> and added the compiled ffmpef lib folder to ld.so.conf [03:14] <SteveTheCat> echo "/usr/local/customvlc/ffmpeg/lib/" >> /etc/ld.so.conf [03:19] <SteveTheCat> now i'm getting this error though trying to encode with h264 ./ffmpeg: relocation error: ./ffmpeg: symbol avformat_alloc_output_context2, version LIBAVFORMAT_53 not defined in file libavformat.so.53 with link time reference [06:15] <lluvia> is there support for splicing ogg videos? [06:16] <lluvia> files* [08:23] <Ginks> is there a way to do say "-aspect 4:3 -s *x480" [08:23] <Ginks> ? [08:26] <ubitux> -vf scale=-1:480 maybe [11:49] <swkide> Happy new year and hello! I am trying out the new segmenter feature of the ffmpeg 0.9 - Can anyone help me with the parameters or how to create segments? [12:40] <hellop> Hello. Do you think it would be possible to add audio from my microphone to an RTSP stream, then restream the RTSP stream with Audio and Video? [12:48] <hellop> Another question I've had, how do I determine the name of my microphone for ffmpeg input when I am using a new computer? For linux I have found: "-f oss -i /dev/dsp" But, what about Windows? I have seen commands where it was some long string specific to that computer. How do I discover the name? [13:05] <eightfold> hi. i wonder if ffmpeg manages to join two aac files? [13:14] <hellop> eightfold: I'm also looking for that answer. I've found how to add audio to a video stream, or replace the audio in a video stream. [13:22] <eightfold> hellop: i think that's is where i am right now, too :) [13:23] <hellop> You could also try mencoder, or mp4box to mux streams together. [13:23] <eightfold> hellop: let'see if anyone knows. the manual does not seem to be speaking of "join". [13:26] <hellop> I am able to get info about my camera using: ffmpeg -list_options true -f dshow -i video="Integrated Webcam" [13:27] <hellop> I did this by looking at capture devices in the VLC GUI. It shows "Microphone (Realtek High Definition Audio)", but that string doesn't work as an ffmpeg input. Any suggestions how to capture my mic in windows with ffmpeg? [13:52] <hellop> Ok, I was able to find my audio device using the VLC GUI by clcking "more options" in the capture screen, my mic is called: "Microphone (Realtek High Defini" and works in ffmpeg. [14:04] <hellop> sorta [15:20] <MrJones> how can I make an AVFormatContext use a custom AVIOContext? I have read a code example that simply sets AVFormatContext->pb and then uses it with avformat_open_input with an empty filename. is that the right way to do it? setting something on a struct directly seemed a bit hackish to me [17:32] <qubodup> hi [17:33] <qubodup> I have archlinux, vorbis installed (1.3.2) and ffmpeg 20111211-1 (ffmpeg version N-35681-g16abd68) [17:33] <qubodup> running the following line leads to [libvorbis @ 0x7fda84003e00] oggvorbis_encode_init: init_encoder failed [17:33] <qubodup> ffmpeg -f alsa -i pulse -f x11grab -acodec pcm_s16le -r 30 -s $CAPWIN_DIM -i :0.0+$CAPWIN_XPOS,$CAPWIN_YPOS -vcodec libvpx -vpre libvpx-720p -threads 0 capwin-$DATE.mkv -ab 96k [17:33] <qubodup> (with $CAPWIN_ variables being correct parameters) [17:41] <qubodup> if I remove -i pulse, it works, but no sound gets recorded [17:41] <kollapse> Hi. x264 problem. I get `File for preset 'libx264-fastfirstpass' not found` [17:42] <qubodup> kollapse: that's because the presets are missing. [17:42] <kollapse> Yep, how do I get them ? [17:42] <qubodup> kollapse: I don't know how to make or get compatible ones for recent versions [17:42] <kollapse> Well then how do I x264 encode :? [17:43] <qubodup> kollapse: my way is switching to other presets [17:43] <microchip_> kollapse: are you on a recent ffmpeg? [17:43] <kollapse> version 0.9 [17:43] <qubodup> kollapse: ls /usr/share/ffmpeg/ [17:43] <qubodup> kollapse: using a different preset might save your day [17:44] <microchip_> kollapse: use -preset <name_of_x264_preset> [17:44] <microchip_> kollapse: recent versions of ffmpeg use x264's presets directly [17:44] <kollapse> Not good. I only have libx264-ipod320.ffpreset and libx264-ipod640.ffpreset [17:44] <qubodup> same here for libx264 [17:44] <qubodup> libx264-ipod640.ffpreset is not bad [17:45] <qubodup> kollapse: you could probably edit the file to tweak the values you want (save as new file) [17:45] <kollapse> I'm not an x264 wizard. I just need the `fastfirstpass` and `hq` presets. No way to get them ? [17:46] <qubodup> kollapse: there is, but I'm afraid they don't work with the latest version [17:46] <JEEB> fast first pass is used by default by libx264 in case first pass is specified [17:46] <JEEB> (and slow-firstpass isn't set) [17:46] <kollapse> JEEB: and `hq` ? [17:47] <JEEB> whatever is of similar speed in libx264's internal presets [17:47] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [17:47] <JEEB> ^ a listing [17:48] <JEEB> everything but placebo doesn't set slow-firstpass so if you set first pass as-is you will get "fast first pass" settings [17:49] <vcs> hi, in the latest ffmpeg I get "unrecognized option 'newvideo' [17:49] <vcs> anyone know what that was replaced with [17:50] <JEEB> what did it do to begin with? [17:50] <vcs> added a new video stream (if the source file had multiple video streams for example, by default only 1 was added) [17:50] <vcs> so if i used -newvideo twice, 3 streams would be copied from the source [17:51] <JEEB> oh, nowadays you IIRC have better control on which streams get passed [17:51] <JEEB> -map was it? [17:51] <vcs> ill check it out, thanks [17:52] <relaxed> vcs: I don't think you need -newvideo. Use -map for each stream you want in the output. [17:53] <JEEB> http://ffmpeg.org/ffmpeg.html#Advanced-options <- first one in this spot is -map [17:53] <vcs> wow, the syntax makes sense now, thats great. [17:53] <vcs> thanks guys [17:53] <JEEB> aye [17:53] <JEEB> the capability of using libx264's defaults as well as internal presets/tunes, mapping system... [17:54] <JEEB> I used to be one of those people who wouldn't recommend ffmpeg for encoding because of this, but nowadays my mind has changed somewhat [17:54] <qubodup> mikobuntu: I don't understand how to set x264's presets in ffmpeg (if possible?) [17:55] <kollapse> Any way to sharpen using ffmpeg ? [17:55] <JEEB> qubodup, -preset after -vcodec [17:55] <JEEB> -vcodec libx264 -preset veryslow for example [17:55] <qubodup> JEEB: and then the name of the preset, which has to be in /usr/share/ffmpeg/ ? [17:55] <JEEB> doesn't have to be if you have a new enough ffmpeg/libx264 [17:55] <qubodup> 'x264' has no preset files installed [17:56] <JEEB> ffmpeg now can use libx264's internal presets :P [17:56] <JEEB> thus I linked you a listing of them a few dozen lines upwards [17:56] <qubodup> File for preset 'lossless_ultrafast' not found [17:56] <JEEB> lossless would be crf 0 with 8bit H.264 with libx264 [17:56] <qubodup> JEEB: yeah, an interesting link, but I don't understand how these exist [17:56] <JEEB> they exist in the library? [17:56] <qubodup> I guess internal means in-code/binary [17:57] <JEEB> they are given out by libx264 as libx264 itself needs a way to set different speed/compression levels easily enough :V [17:57] <qubodup> oh wait. that one doesn't exist :) [17:57] <JEEB> for lossless + fast you can use -vcodec libx264 -crf 0 -preset ultrafast ? [17:57] <qubodup> hm. using "ultrafast" also does not exist [17:58] <JEEB> are you sure you are applying it in the range of where video settings are? [17:58] <qubodup> oh [17:58] <qubodup> -crf 0 helped [17:58] <qubodup> thanks! [17:58] <qubodup> oh dog it works! [17:59] <JEEB> basically libx264 has two ways of setting lossless encoding, either crf 0 (that now works only with 8bit H.264) that keeps users out of the qp setting, and then qp 0 which works with all bit depths [17:59] <JEEB> I don't know how to set qp encoding with ffmpeg and I think you are using a 8bit H.264 (as 99% of the whole user population), so crf 0 seemed practical [18:00] <JEEB> also, the presets should work even without crf 0 [18:00] <JEEB> (libx264's internal ones) [18:00] <JEEB> as they set the defaults [18:00] <JEEB> you were probably having a PEBKAC [18:24] <vcs> wow, -map is the greatest thing to ever happen to ffmpeg [19:53] <SteveTheCat> I seem to be getting this error after I compiled ffmpeg [19:53] <SteveTheCat> ./ffmpeg: relocation error: ./ffmpeg: symbol avformat_alloc_output_context2, version LIBAVFORMAT_53 not defined in file libavformat.so.53 with link time reference [20:18] <aca20031> I have a h.264 in an mkv container, and id like to change its container to avi or something that adobe encore will recognize [20:18] <aca20031> apparently its not as simple as -i in.mkv out.avi ;) [20:18] <JEEB> you don't want it in avi [20:18] <JEEB> at least if it contains b-frames [20:19] <JEEB> -i derp.mkv -vcodec copy -an (or -acodec copy if you want the original audio) out.mp4 [20:19] <SteveTheCat> b-frames? [20:19] <aca20031> avi wmv or mxf are its suggested formats [20:20] <JEEB> aca20031, I'd be surprised if it couldn't take in mov or mp4 :P [20:20] <JEEB> and avi isn't exactly a container that properly supports B-frames [20:20] <SteveTheCat> move suck imo [20:20] <aca20031> im just going by the extensions when i select Video File from the dropdown of types [20:20] <aca20031> the actual problem is, is that i can import the .avi [20:20] <aca20031> but Encore only sees the audio track [20:20] <aca20031> not the video track [20:20] <aca20031> it think its an audio only file lol [20:20] <SteveTheCat> then it's a problem with the video encoding [20:20] Action: JEEB sighs [20:21] <SteveTheCat> did you have problems with importing h264 before? [20:21] <JEEB> SteveTheCat, as a container it's better than avi tho [20:21] <aca20031> this is my first time trying [20:21] <aca20031> it says it supports it though [20:21] <SteveTheCat> for the program you're importing with? [20:21] <aca20031> the video is: H264 - MPEG-4 AVC [20:21] <aca20031> according to VLC [20:21] <SteveTheCat> what OS you running? [20:21] <aca20031> windows 7 [20:21] <JEEB> nothing official takes in H.264 in AVI without extra vfw codecs, so no -- that's not how you do it [20:21] <JEEB> dang it [20:21] <JEEB> just try mov and mp4 :P [20:21] <JEEB> in the way I said [20:22] <aca20031> lol [20:22] <aca20031> im doing that now [20:22] Action: JEEB sighs [20:22] <SteveTheCat> go in command line and type vlc -l > modules.txt [20:22] <aca20031> just thought id point out the original problem while it transcodes [20:22] <SteveTheCat> vlc being where your vlc.exe is [20:22] <aca20031> just in case my assessment was wrong [20:22] <JEEB> aca20031, it shouldn't transcode >_> [20:22] <JEEB> -vcodec copy copies the stream [20:22] <aca20031> oh [20:22] <aca20031> well that explains why its done [20:22] <aca20031> let me try it ;) [20:23] <aca20031> hmm [20:23] <JEEB> of course personally I would dump the thing into ut video or ffvhuff or something in AVI, of course that'd be pretty huge [20:23] <aca20031> more than two audio channelsfor this type of encoding is not supported" [20:23] <JEEB> I don't think I anywhere said for you to re-encode the audio >_> [20:23] <aca20031> i didnt [20:23] <JEEB> <JEEB> -i derp.mkv -vcodec copy -an (or -acodec copy if you want the original audio) out.mp4 [20:23] <aca20031> i typed -acodec copy [20:23] <JEEB> hm [20:24] <aca20031> D:\Downloads\PE HD\Planet.Earth.S01E02.Mountains.1080p.BluRay.x264-CULTHD>ffmpeg [20:24] <aca20031> -i Mountains.mkv -vcodec copy -acodec copy out.mp4 [20:24] <JEEB> probably because the audio in it isn't supported in mp4 [20:24] <aca20031> further suggestion? [20:24] <JEEB> but test video first >_> [20:24] <JEEB> with -an [20:25] <aca20031> video works in vlc [20:25] <aca20031> actually [20:25] <JEEB> and that says nothing [20:26] <aca20031> audio works in vlc too [20:26] <aca20031> lol [20:26] <JEEB> and that says nothing [20:26] <aca20031> how would you like me to test it then [20:26] <aca20031> it wont import into encore at all so i cant really test there [20:26] <JEEB> -vcodec copy -an [20:26] <aca20031> mm [20:26] <JEEB> ok [20:26] <aca20031> oh [20:26] <JEEB> it doesn't read it? [20:26] <aca20031> i thought -an was the same as -acodec copy [20:26] <JEEB> nope [20:26] <JEEB> that's AudioNone [20:26] <aca20031> i misread your original statement [20:26] <aca20031> let me test that [20:26] <JEEB> also test both mov and mp4 [20:26] <JEEB> and if neither work blame the maker of the app >_> [20:27] <aca20031> blame adobe? :P [20:27] <JEEB> yeh [20:27] <aca20031> its a professional dvd creator, i imagine its as good as i can get [20:27] <JEEB> not really [20:27] <JEEB> you'd be surprised [20:27] <aca20031> im open to suggestions, i just want control enough to make a custom menu and burn more than one videos to one dvd [20:27] <aca20031> tried Roxio, it doesnt give you much control [20:27] <SteveTheCat> i don't understand if you're using windows, why not use something specialized for windows? [20:28] <aca20031> ..such as? [20:28] <aca20031> i did my googling [20:28] <JEEB> SteveTheCat, just stay silent for a moment [20:28] <SteveTheCat> here: [20:28] <JEEB> thank you [20:28] <aca20031> version with -an works [20:28] <JEEB> aca20031, if you want to make a menu it'd mean that you'd be making a normal DVD, thus re-encoding the thing in any case. Am I correct? [20:29] <JEEB> oh [20:29] <JEEB> great [20:29] <aca20031> trying .mov copy then i guess [20:29] <JEEB> with mov I think you can also decode the audio into raw pcm and mux it there [20:29] <JEEB> if nothing else works :P [20:29] <JEEB> as if it is something like DTS there's a rather big chance it won't be read [20:29] <aca20031> if i have to, the original mkv audio worked in encore and the video didnt [20:29] <aca20031> lol [20:30] <aca20031> i can combine them [20:30] <JEEB> sure [20:30] <JEEB> that works too [20:31] <aca20031> hmm [20:31] <JEEB> also, talking about DVD/BD mastering the last thing I tested was Scenarist, it was pretty good. Didn't do extra transcoding, just let me work with pre-done stuff if I had done things by the DVD/BD spec video and audio-wise. [20:31] <aca20031> cant import move, "Quicktime and/or the appropriate codec appears not the be installed" [20:31] <JEEB> Of course, scenarist is nothing I can personally afford >_> [20:32] <JEEB> lol [20:32] <aca20031> quicktime is definitely installed :( [20:32] <aca20031> s/move/mov/ [20:32] <JEEB> funny how they use QT for mov, while parsing mp4 by themselves :P [20:32] <JEEB> given the fact that the formats are very close [20:32] <JEEB> anyways, could it be that the editor is now 64bit or something? [20:33] <JEEB> because QT for Windows is 32bit only [20:33] <JEEB> (and apps of different bitness on windows can't link against each other etc.) [20:33] <aca20031> nah encore is 32 bit, only premiere and photoshop in the adobe CS5 is 64 bit [20:33] <JEEB> k [20:33] <JEEB> is the audio DTS or something? [20:33] <JEEB> because that could be only in QT Pro or whatever [20:34] <JEEB> so it's just easier to decode it with ffmpeg [20:34] <JEEB> and get out 16bit little endian pcm [20:34] <aca20031> Codec: A52 Audio (aka AC3) (a52 ) [20:34] <JEEB> oh [20:34] <aca20031> dunno what that is [20:34] <JEEB> well that's usually premium as well [20:34] <JEEB> lol [20:35] Action: aca20031 facepalms [20:35] <JEEB> (although less often than with DTS, because the licensing costs DTS wants are lulzy) [20:35] <JEEB> (never mind that AC3 is actually better than DTS format-wise) [20:36] <JEEB> so yeah, use whatever way you want to use to load that one in [20:36] <JEEB> heck, depending on the type of the AC3 track you might be able to put it out on a DVD as-is [20:36] <JEEB> but not sure if that adobe thingy will let you [20:36] <SteveTheCat> why am i getting Unknown encoder 'h264' when I compiled with --enable-libx264 ..... [20:36] <aca20031> still need encore to not refuse to import it, so i can make the dvd menu [20:36] <JEEB> SteveTheCat, h264 is the decoder, libx264 is libx264 [20:37] <JEEB> aca20031, just decode to lpcm 16bit little endian? [20:37] <JEEB> switch -acodec to it [20:37] <aca20031> yep yep [20:37] <JEEB> also, ac3 in mp4 is "new" (although was specified a few year(s) ago already, so I'm not surprised if something couldn't read it) [20:38] <SteveTheCat> ohhh [20:38] <SteveTheCat> i see JEEB [20:38] <SteveTheCat> Thanks [20:39] <aca20031> avi freezes encore, time to transcode the audio [20:39] Action: aca20031 slaps adobe [20:39] <SteveTheCat> I think I went through the trouble of compiling ffmpeg without reading the codecs lol [20:40] <JEEB> well, external libraries more often or not do have the notation of lib<something> ;) [20:40] <JEEB> so it should be rather understandable to try that first :) [20:40] <JEEB> but great if you got it to work [20:40] <JEEB> aca20031, for avi I bet it tries to use vfw [20:40] <JEEB> you do not want that [20:40] <SteveTheCat> I'm getting Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height for Stream #0:1 -> #0:1 (aac -> libvorbis) [20:40] <aca20031> just wanted to give it a try and see what it looked like [20:40] <aca20031> freezing is bad though [20:40] <JEEB> b-frames + avi + vfw = not a good combo [20:40] <SteveTheCat> when i'm not even transcoding the audio [20:41] <JEEB> SteveTheCat, you seemingly are [20:41] <JEEB> -acodec copy for audio copy [20:41] <SteveTheCat> oh, assumed it was on by default [20:41] <JEEB> nope [20:42] <SteveTheCat> on second thought, i only saw the h264 before compiling [20:42] <SteveTheCat> i don't think i saw lib264 [20:42] <aca20031> whats the -acodec for lpcm 16 bit LE? [20:42] <SteveTheCat> not sure what is the l before pcm means [20:42] <aca20031> pcm_s16le? [20:43] <JEEB> yeah [20:43] <JEEB> that should be it [20:43] <SteveTheCat> did you get my query aca20031? [20:43] <aca20031> yes steve, but my problem isnt a lack of a good converter [20:43] <SteveTheCat> alright [20:43] <aca20031> thank you though [20:43] <JEEB> SteveTheCat, http://wiki.multimedia.cx/index.php?title=PCM#Linear_PCM [20:44] <JEEB> the default type of PCM I guess as it has nothing written in the area >_> [20:45] <aca20031> ffs [20:45] <aca20031> audio track is freaking empty [20:45] <JEEB> <(^o^) [20:45] <SteveTheCat> did you use -an? [20:46] <SteveTheCat> just checking? :P [20:46] <aca20031> even in vlc :/ [20:46] <SteveTheCat> finding the right codec is where the fun is [20:46] <aca20031> D:\Downloads\PE HD\Planet.Earth.S01E02.Mountains.1080p.BluRay.x264-CULTHD>ffmpeg [20:46] <aca20031> -i Mountains.mkv -vcodec copy -acodec pcm_s16le out.mp4 [20:46] <aca20031> track 1: could not find tag, codec not currently supported in container [20:47] <JEEB> oh [20:47] <JEEB> derp [20:47] <JEEB> mp4 doesn't officially support LPCM [20:47] <JEEB> you'll have to output it to a separate wav or something [20:47] <JEEB> ffmpeg -i input.derp -vn -acodec pcm_s16le derp.wav [20:47] <JEEB> this should do [20:48] <aca20031> phew what a pain [20:48] <aca20031> lol [20:48] <SteveTheCat> can't you use a different audio encoding [20:48] <aca20031> gotta do this with 10 other ones too [20:48] <SteveTheCat> like mp3? [20:48] <JEEB> SteveTheCat, lossy -> lossy? [20:48] <JEEB> U MAD? [20:49] <SteveTheCat> :( [20:49] <SteveTheCat> i dunno [20:49] <SteveTheCat> wait [20:49] <JEEB> of course he will most probably be encoding it to lossy afterwards for the DVD [20:49] <JEEB> but that'd be just lossy->lossless->lossy [20:49] <aca20031> yeah exactly [20:49] <aca20031> lol [20:49] <JEEB> in your case it'd be lossy->lossy->lossy [20:49] <JEEB> 3x lossy encoding, ho [20:51] <aca20031> sigh [20:51] <aca20031> more than two audio channels not supported for this type of encoding :( [20:51] <JEEB> yeah, generally [20:51] <JEEB> I think if it was 5.1 AC3 I'd prolly just extract it out once [20:52] <JEEB> and then mux it when doing the final mux for the DVD [20:52] <JEEB> but I don't know if Adobe wants you to have such luxury [20:52] <SteveTheCat> wish me luck [20:52] <SteveTheCat> about to test if i can finally stream a converted video [20:52] <JEEB> if it's flv it should work, if it's mp4 you'd have to put the index part into the beginning [20:53] <JEEB> mpeg-ts should work as-is as well [20:54] <aca20031> so not worth this much bullshit [20:54] <JEEB> welcome to the world of "easy solutions" [20:55] <JEEB> oh [20:55] Action: JEEB wonders if that pcm in avi would work [20:55] <aca20031> media encoding and the web [20:55] <aca20031> 2 things that need to be wiped out [20:55] <JEEB> lol [20:55] <aca20031> and done again in one universal format [20:55] <JEEB> Never gonna happen because technology keeps leaping forward [20:55] <JEEB> not to mention that video-wise we pretty much are already set [20:56] <JEEB> what's limiting you is adobe, nothing else atm [20:56] <JEEB> no idea how it wants possible 5.1 audio [20:57] <JEEB> and what I was listing were different containers that SteveTheCat might be using -- some of them don't need anything special, other types like mp4 (and mov) need some special care [20:57] <JEEB> although there's a feature in mp4 that makes such unneeded [20:59] <aca20031> wish it would use vlc player to decode it :P [20:59] <aca20031> it never has these issues [21:05] <aca20031> meh, its being burned for my mother on a 27 inch stereo tv [21:05] Action: aca20031 goes lossless [21:05] <aca20031> lossy* [22:14] <Bizzeh> hi. im trying to build ffmpeg with mingw32 (4.2.1-sjlj), and i keep getting LOADS of "error: no previous prototype for..." errors [22:14] <Bizzeh> anyone any idea how i would fix this? [22:46] <Moral_> Good afternoon. I am trying to convert a file from mp4/flv to ust mp3. I use ffmpeg -i videofile output.mp3, however no matter what the bitrate of the audio in the video file, the mp3 is 128kb/s even if the audio bitrate is higher in the video file. I want the audio bitrate of the mp3 to match that of the video file. Any options? [22:47] <Bizzeh> also, is it a known bug that you cannot downsample 5.1 to stereo using the new build in aac encoder [23:03] <SteveTheCat> I seem to be getting 'FF_MM_MMX' undeclared (first use in this function) while compiling VLC but from /usr/local/customvlc/ffmpeg/include/libavcodec/avcodec.h:3986 [23:10] <pasteeater> Moral_: if the input is flv the audio is probably already mp3 [23:10] <Moral_> True, but for .mp4 it's not. [23:10] <pasteeater> use a pastebin site to show your ffmpeg command and the complete console output [23:11] <Moral_> one moment. Please. [23:16] <Moral_> http://pastebin.com/NkDsw47V [23:18] <Moral_> That may not be a good example, a previous video was telling me it was 152kb/s orig, and output was 128 [23:19] <Moral_> http://pastebin.com/LkDu1vG8 [23:19] <Moral_> there's what I was talking about earlier. [23:24] <Bizzeh> is there a build bug in ffmpeg at the minute? [23:24] <Bizzeh> even a basic build is giving me 100's of linking errors. loads of _ff_* undefined [23:28] <pasteeater> Moral_: if you do not declare a bitrate for the output ffmpeg will choose 128k by default. [23:30] <Moral_> pasteeater, is there a way to tell ffmpeg to use the audio bitrate of the original file? [23:30] <pasteeater> not with any ffmpeg option other than '-b:a', such as '-b:a 192k' [23:31] <pasteeater> so no, unless you use a script or something [23:31] <pasteeater> why do you want to match the bitrate? [23:32] <pasteeater> why not just use a constant quality option? [23:32] <pasteeater> ffmpeg -i input -c:a libmp3lame -q:a 4 output.mp3 [23:33] <pasteeater> which is about the same as using 'lame -V4' which will give VBR audio instead of CBR or ABR [23:33] <Moral_> I just wanted the best quality for each video [23:34] <pasteeater> if you must re-encode, and want "best quality", then use -q:a instead of -b:a [23:36] <pasteeater> with -b:a the output quality will be ambiguous, but with -q:a you will always have a consistent quality shared between your videos [23:36] <pasteeater> i mean audio files, not videos [00:00] --- Wed Jan 4 2012

1 0

ffmpeg-devel.log.20120102
by burek 03 Jan '12

03 Jan '12

[00:19] <durandal_1707> done [03:25] <CIA-101> ffmpeg: 03Janne Grunau 07master * r60d626650a 10ffmpeg/tests/ (12 files in 2 dirs): [03:25] <CIA-101> ffmpeg: fate: use rgb24 as output format for dfa tests [03:25] <CIA-101> ffmpeg: Palette is as supposed in native endianness. Converting the pal8 output [03:25] <CIA-101> ffmpeg: to rgb24 is thus necessary for identical CRCs on big and little endian [03:25] <CIA-101> ffmpeg: systems. [03:25] <CIA-101> ffmpeg: 03Aneesh Dogra 07master * rda7c65f0ce 10ffmpeg/avplay.c: [03:25] <CIA-101> ffmpeg: avplay: K&R cleanup [03:25] <CIA-101> ffmpeg: Signed-off-by: Anton Khirnov <anton(a)khirnov.net> [03:25] <CIA-101> ffmpeg: 03Janne Grunau 07master * rb6064d9a59 10ffmpeg/libavcodec/utils.c: threads: set thread_count to 1 when thread support is disabled [03:25] <CIA-101> ffmpeg: 03Janne Grunau 07master * rb12d217339 10ffmpeg/libavcodec/pthread.c: [03:25] <CIA-101> ffmpeg: threads: limit the number of automatic threads to MAX_AUTO_THREADS [03:25] <CIA-101> ffmpeg: The extra thread added in {frame_}*thread_init was not taken into [03:25] <CIA-101> ffmpeg: account. Explicitly sets thread_count to 1 if only one CPU core was [03:25] <CIA-101> ffmpeg: detected. Also fixes two typos in comments. [03:25] <CIA-101> ffmpeg: 03Janne Grunau 07master * rb4d44a45f9 10ffmpeg/libavcodec/ (avcodec.h libx264.c pthread.c version.h): [03:25] <CIA-101> ffmpeg: threads: introduce CODEC_CAP_AUTO_THREADS and add it to libx264 [03:25] <CIA-101> ffmpeg: Some external codecs have their own code to determine the best number [03:26] <CIA-101> ffmpeg: bfi: K&R cosmetics [03:26] <CIA-101> ffmpeg: Signed-off-by: Anton Khirnov <anton(a)khirnov.net> [03:26] <CIA-101> ffmpeg: 03Anton Khirnov 07master * rf15f02c204 10ffmpeg/avconv.c: avconv: make -frames work for all types of streams, not just video. [03:26] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r4121148388 10ffmpeg/: (log message trimmed) [03:26] <CIA-101> ffmpeg: Merge remote-tracking branch 'qatar/master' [03:26] <CIA-101> ffmpeg: * qatar/master: [03:26] <CIA-101> ffmpeg: avconv: make -frames work for all types of streams, not just video. [03:26] <CIA-101> ffmpeg: bfi: K&R cosmetics [03:26] <CIA-101> ffmpeg: bgmc: K&R cleanup [03:26] <CIA-101> (18 lines omitted) [04:03] <CIA-101> ffmpeg: 03Paul B Mahol 07master * rdfa77dead2 10ffmpeg/ (10 files in 4 dirs): [04:03] <CIA-101> ffmpeg: y41p encoder and decoder [04:03] <CIA-101> ffmpeg: y41p is a packed 12-bit 4:1:1 YUV format used by Brooktree. [04:03] <CIA-101> ffmpeg: Fixes issue 1123 / ticket #102. [04:03] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:03] <CIA-101> ffmpeg: 03Jean First 07master * rb103156235 10ffmpeg/libavcodec/j2kdec.c: [04:03] <CIA-101> ffmpeg: j2kdec: av_log formatting use %tx instead of %x [04:03] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [04:03] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:03] <CIA-101> ffmpeg: 03Jean First 07master * re2c1d3ec41 10ffmpeg/libavformat/flvdec.c: [04:04] <CIA-101> ffmpeg: flvdec: use av_uninit to silence compiler warnings [04:04] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [04:04] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [07:22] <ubitux> michaelni: i need to get the drawtext lavc dep patch done; i'll do it today, and will certainly commit it in the following days; please wait until then for the 0.9.1 [08:45] <ubitux> mmh, strange thing, i can't play anymore mp3 encoded with ffmpeg on my mp3 player [08:46] <ubitux> i wonder if it's an issue in lame [08:55] <ubitux> ok it seems to come from ffmpeg [09:02] <ubitux> it works with latest ffmpeg 0.8 [09:03] <ubitux> and outputs totally differs mmmh. [09:12] <ubitux> ok it seems 1fea432b7a6957b800006fd7c4bd9b4076b0cc06 is at fault [09:17] <ubitux> looks sane to me though... [09:24] <ubitux> it's indeed this commit; output differs after while they shouldn't [09:24] <kcm1700> hmm... [09:27] <ubitux> ah i know [09:27] <ubitux> header is modified in the loop [09:28] <ubitux> and reused after [09:28] <ubitux> thus, the mask need to be kept [09:28] <kcm1700> can 12th~15th bits in header already be set? [09:28] <ubitux> mmh or maybe not. [09:30] <ubitux> ok the "if (needed <= c.frame_size)" doesn't trigger the mask disabling [09:30] <ubitux> so the header shouldn't have its mask disabled [09:30] <kcm1700> aha. [09:35] <ubitux> didn't test it properly, but this should do the trick: http://pastie.org/3111074 [09:36] <ubitux> but now i wonder if 40176fc3149bc72c1309f93989a57aec2322e626 is still necessary [09:38] <ubitux> ok i confirm it fixes the playback issue [09:46] <ubitux> patch sent. [10:00] <kcm1700> how about or-ing after the loop? [10:00] <kcm1700> I like the simplified version. [10:01] <ubitux> i prefer the working one ;) [10:02] <ubitux> kcm1700: i thought about that, but the or-ing seems to depend on if it trigger the case or not [10:02] <ubitux> (the if needed...) [10:03] <ubitux> i let the ppl knowing what does this code mean decide :p [10:03] <ubitux> i don't want to assume wrong things [10:04] <kcm1700> indeed. [14:23] <ubitux> does anyone have mov samples with timecode tracks in "frame count" and in "mov" formats? [14:24] <ubitux> i added the timecode extraction from mov, but i have samples where the header indicates the timecode is in mov format while it is actually in "frame count" format [14:25] <ubitux> i have a hacky heuristic to fix the behaviour but it sucks a bit& [14:25] <ubitux> if i could get some more samples, that would be nice ;p [15:16] <ubitux> btw, any way to do some emphasis in texi? [15:17] <ubitux> @emph{}, ok. [15:35] <CIA-101> ffmpeg: 03Clément BSsch 07master * r940d3cc0b0 10ffmpeg/ (doc/filters.texi libavfilter/vf_drawtext.c): drawtext: fix hard dependency to lavc (timecode). [15:35] <CIA-101> ffmpeg: 03Clément BSsch 07master * re1a8164d62 10ffmpeg/libavformat/mp3enc.c: [15:35] <CIA-101> ffmpeg: mp3enc: fix playback with mp3 encoded files on some devices. [15:35] <CIA-101> ffmpeg: This partially reverts 1fea432b; header variable needs to keep its value [15:35] <CIA-101> ffmpeg: in case it triggers "if (needed <= c.frame_size)". [15:39] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * ra2a12e3358 10ffmpeg/libavcodec/vp3.c: [15:39] <CIA-101> ffmpeg: vp3: fix regression with mplayer-crash.ogv [15:39] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:19] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r0af34c4d2c 10ffmpeg/MAINTAINERS: [17:19] <CIA-101> ffmpeg: MAINTAINERS: update release maintainers [17:19] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:36] <CIA-101> ffmpeg: 03Michael Niedermayer 07release/0.7 * r1218f8ed49 10ffmpeg/libavcodec/vp3.c: [17:36] <CIA-101> ffmpeg: vp3: fix regression with mplayer-crash.ogv [17:36] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:36] <CIA-101> ffmpeg: (cherry picked from commit a2a12e3358c3bbdc0246ffc94973e58eba50ee30) [17:36] <CIA-101> ffmpeg: 03Michael Niedermayer 07release/0.8 * rc409ac5adc 10ffmpeg/libavcodec/vp3.c: [17:36] <CIA-101> ffmpeg: vp3: fix regression with mplayer-crash.ogv [17:36] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [17:36] <CIA-101> ffmpeg: (cherry picked from commit a2a12e3358c3bbdc0246ffc94973e58eba50ee30) [20:22] <CIA-101> ffmpeg: 03Michael Niedermayer 07release/0.7 * r870e74dc43 10ffmpeg/ (Doxyfile RELEASE VERSION): [20:22] <CIA-101> ffmpeg: Update for 0.7.10 [20:22] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [20:22] <CIA-101> ffmpeg: 03Michael Niedermayer 07release/0.8 * rcee1568ae1 10ffmpeg/ (Doxyfile RELEASE VERSION): [20:22] <CIA-101> ffmpeg: Update for 0.8.9 [20:22] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [22:47] <ubitux> bcoudurier: hi; what do you mean by "yet"? no app generates such media atm? [22:50] <bcoudurier> doesn't matter for now [22:50] <bcoudurier> once a sample is found, it will be added [22:50] <ubitux> ok& so basically i have to assume frame num format all the time? [22:51] <ubitux> sounds wrong to me but well, should do the trick for all the samples i have [22:52] <bcoudurier> as long as it doesn't fail it's fine it would just show a wrong timecode [22:53] <bcoudurier> same for the check framerate, if it's wrong it shouldn't fail demuxing, it should be fine though the ret value is not considered in your patch [22:54] <ubitux> yes return values in read_timecode_track are ignored [22:54] <ubitux> it just displays err/warn [23:10] <durandal_1707> why our libpostproc version is different from libav? latest mplayer2 configure fails because of this [00:00] --- Tue Jan 3 2012

1 0

ffmpeg.log.20120102
by burek 03 Jan '12

03 Jan '12

[00:57] <grap3_ap3> Please help me understand. Recorded .ogv file looks/sounds just fine. Converted to .mkv for YouTube upload & resolution of video gets munged. Text in terminal/browser appears to have been partially rendered. This pastebin has the command line input/output (http://pastebin.com/x7a8ZvSN) How can I improve the resolution issues during conversion? [03:50] <SteveTheCat> hey [03:51] <SteveTheCat> I'm trying to compile 0.5.2 [03:51] <SteveTheCat> and i'm getting problems with this configuration [03:51] <SteveTheCat> http://pastebin.com/b8zTW9CD [03:52] <sacarasc> Why are you trying to compile one so ancient? [03:53] <SteveTheCat> i'm following this forum post that says his stream is working with these builds [03:53] <SteveTheCat> http://ntp1.centos.org/modules/newbb/viewtopic.php?viewmode=flat&order=ASC&… [03:54] <SteveTheCat> basically having problems with using x264 in VLC [04:05] <SteveTheCat> anyone? [05:46] <metvik> is there anyway to extract subtitles from a dvb ts with ffmpeg? [05:47] <mystica555_> i'd bet there is, but as ive never done it, i can't explain how. ffmpeg does do many wonderful things however. [06:13] <upvar> does ffmpeg write to pipes other than stderr? i am trying to catch some errors with ffmpeg and they don't seem to be going to stderr [06:14] <upvar> getting a VBV underflow error and i want to terminate ffmpeg if that happens [06:14] <upvar> but it doesn't seem to be getting sent to the stderr like everything else [06:15] <mystica555_> is it perhaps going to stdout instead? [06:15] <upvar> nope [06:15] <upvar> i am reading stdout and stderr and everything seems to come from stderr [06:15] <upvar> [libx264 @ 0x122da60] VBV underflow (frame 1582, -92 bits) [06:15] <upvar> that's the kind of thing, so i assume the libx264 module is writing to some other pipe? [06:16] <upvar> i don't know enough about POSIX to grasp this properly [06:16] <relaxed> it should all go to stderr [06:17] <upvar> yeah it's totally not [06:18] <upvar> i have a wrapper script reading stderr and writing it to a file [06:18] <upvar> and those lines are not going through [06:18] <upvar> everything else is [06:18] <upvar> wonder if it's a buffering thing [06:20] <relaxed> How are you reading stderr? [06:20] <upvar> looks like it's a buffering thing [06:21] <upvar> using fgets() [06:25] <upvar> oh well waiting for the buffer to fill up sure beats ffmpeg hanging the box for 2 hours :) [06:42] <kcm1700> ffmpeg blahblah 2>&1 [06:42] <kcm1700> stderr to stdout redirection. [06:46] <upvar> yeah that's not what i was asking [06:46] <upvar> since i'm reading stderr directly with fgets [06:47] <upvar> this is why i was asking [06:47] <upvar> Note that in case stdin is associated with a terminal, there may also be input buffering in the terminal driver, entirely unrelated to stdio buffering. (Indeed, normally terminal input is line buffered in the kernel.) [06:47] <upvar> http://linux.about.com/library/cmd/blcmdl3_stderr.htm [09:22] <meero> Hi, how to mixdown video from stereou sound to mono "-ac 1" is not working with error like av_interleaved_write_frame(): Operation not permitted [13:35] <upvar> hey, i'm getting this warning from both libx264 and ogg on different input videos [13:35] <upvar> [ogg @ 0xd649e0] Frame rate very high for a muxer not effciciently supporting it. [13:35] <upvar> Please consider specifiying a lower framerate, a different muxer or -vsync 2 [13:35] <upvar> it looks like each time that error is mentioned the transcode takes forever or fucks out completely [13:36] <upvar> could i safely assume if that error comes up the video is bad and kill the ffmpeg process? [13:39] <Mavrik> hmm [13:39] <Mavrik> you can safely assume your framerate is wrongly set :) [13:39] <upvar> same typo on both of them too [13:40] <upvar> this is the command, no framerate specified /usr/bin/ffmpeg -i VID_20111231_171746.mp4 -acodec libvorbis -y -ss 0 -ac 2 -b 2000k 'video-hd.ogv' [13:44] <Mavrik> why aren't you setting any properties for the video? [13:44] <Mavrik> that just forces ffmpeg to guess (and it's pretty bad at it :P) [13:45] <MrJones> are there any precompiled LGPL ffmpeg dlls out there? [13:47] <upvar> Mavrik: back story, this company has given me a bunch of ffmpeg strings for different handheld devices, and i'm trying to batch convert a lot of their data for all these devices [13:47] <upvar> but i'm pretty sure these ffmpeg strings are crackheaded [13:47] <upvar> but short of getting a copy of all of the devices in question, not sure what i can do [14:03] <eightfold> isn't it possible to set -s (size) in ffmpeg to something other than the presets? [14:04] <Mavrik> it is. [14:04] <Mavrik> well... depends on the codec actually [14:04] <Mavrik> using scale= filter is the preferred way now though [14:12] <eightfold> Mavrik: okay, i want to use it with libx264. are you talking about https://ffmpeg.org/ffmpeg.html#scale ? [14:13] <Mavrik> eightfold, libx264 supports pretty much arbitrary resolution as long as it is divisible by 2 [14:13] <Mavrik> so -s 482x844 or -vf scale=482:844 should not be problematic [14:14] <eightfold> Mavrik: thanks. is it in any way possible to specify only width or height and "lock" the aspect ratio from the source file? [14:15] <Mavrik> eightfold, if you use the scale filter, yes [14:15] <Mavrik> -vf scale=-1:720 [14:15] <Mavrik> for 720p :) [14:15] <Mavrik> eightfold, check the documentation you linked :) [14:32] <eightfold> Mavrik: ah, great. i will :) [15:18] <fris> im having an issue compiling ffmpeg and h264 http://pastie.org/3112490 anyone have any ideas? [15:28] <relaxed> fris: are you using the latest git source from both? [15:35] <fris> using 0.9 and x264 0.120.x [15:38] <relaxed> use git source [15:39] <fris> ffmpeg? [15:39] <fris> or x264 [15:43] <relaxed> both [18:51] <hiexpo> morning all compiling ffmpeg and am getting and error libx264 not found and i have x264 installed [18:51] <Hyperi> Well apparently it can't find it [18:52] <Hyperi> Can you define the path to x264 with configure script ? [18:52] <hiexpo> is it looking for x264 in which i compiled or does it actually need libx264 [18:54] <hiexpo> i am looking in my packages and i have x264 and libx264-85 installed [19:10] <relaxed> it needs libx264 and a version recent to the ffmpeg version you're compiling. [19:11] <relaxed> download the latest source via git for both and you'll be good to go. [19:11] <hiexpo> relaxed, yes i got it [22:24] <dericed> I'm trying to create a quicktime file with pcm_s24be audio using the fourcc 'in24'. But -atag in24 has no effect. http://pastebin.com/nFyhDNpS Is there a way to force the codecid to in24? [22:25] <dericed> Looks like ffmbc respects "-atag in24" but ffmpeg doesn't. [22:40] <iiu7> I have a MKV movie with this set of streams: http://pastebin.com/ANgXtBHr, I want to convert the audio stream to mp3 instead of dts because it isn't support on my tv. [22:41] <iiu7> I have tried using this command: "ffmpeg -i foo.mkv -map 0:0 -map 0:1 -sameq -vcodec copy -acodec libmp3lame foo-new.mk" but I get the error: "Number of stream maps must match number of output streams" [22:41] <iiu7> What am I doing wrong? [22:42] <relaxed> iiu7: 1) never use -sameq 2) -acodec libmp3lame -aq 0 -ac 2 [22:42] <iiu7> why not sameq? [22:43] <iiu7> ah, because vcodec copy ofcourse [22:45] <iiu7> Alright, with the above change into this: "ffmpeg -i foo.mkv -vcodec copy -acodec libmp3lame -aq 0 -ac 2 foo-new.mkv", get this error: "Encoder (codec id 94210) not found for output stream #0.2" [22:46] <iiu7> I don't understand that error. [22:48] <relaxed> pastebin [22:51] <iiu7> http://pastebin.com/ghjWXTiP [22:52] <ubitux> you need to map only a & v streams, or specify a scodec for the subtitle stream [22:52] <ubitux> such as -scodec copy [22:54] <iiu7> ubitux, thanks! If I only wanted 1 subtitle stream how would I do that? Using -map 0:1 -map 0:2 -map 0:5 for example? [22:54] <ubitux> yes [22:55] <ubitux> -map also has some stream type specifiers [22:55] <ubitux> 'v' for video, 'a' for audio, 's' for subtitles, & check the documentation [22:55] <ubitux> you might find a simpler way [22:56] <iiu7> Thanks! [23:03] <iiu7> I'm now trying to strip the Chinese subs and keep the english ones only using this command: "ffmpeg -i foo.mkv -vcodec copy -acodec libmp3lame -ac 2 -map 0:0 -map 0:1 -map 0:2 -map 0:3 -map 0:4 foo-new.mkv" then I get the error "Number of stream maps must match number of output streams" [23:03] <iiu7> Maybe I haven't understood the streams part or how to map correctly. [23:11] <relaxed> iiu7: To keep things organized, use -map [stream] [encoding options for that stream] -map [stream 2] [etc...] [23:12] <relaxed> I only see 3 streams in your pastebin. [23:13] <iiu7> There are 5. [23:13] <iiu7> http://pastebin.com/sSP4dFmS [23:14] <iiu7> Actually 6, sorry [23:14] <iiu7> 1 video, 1 audio, and 4 subs [23:14] <relaxed> you only want english subs? [23:15] <iiu7> yes, I'm trying with this (reorganized) "ffmpeg -i foo.mkv -vcodec copy -acodec libmp3lame -map 0:0 -map 0:1 -map 0:2 -map 0:3 foo-new.mkv " [23:15] <iiu7> But this gives me: Numbers of streams.. [23:16] <relaxed> ffmpeg -i foo.mkv -map 0:0 -vcodec copy -map 0:1 -acodec libmp3lame -aq 0 -ac 1 -map 0:2 -scodec copy -map 0:3 -scodec copy output.mkv [23:18] <iiu7> Using the one you just make I still get "Number of stream maps must match number of output streams" [23:18] <iiu7> However I see your point about the organizing. [23:18] <relaxed> pastebin your command and all output [23:19] <iiu7> http://pastebin.com/gRfqYhJd [23:22] <cbsrobot> iiu7: what version is that ? [23:22] <cbsrobot> ffmpeg version , Copyright (c) 2000-2011 the FFmpeg developers [23:23] <iiu7> It's the one from debian-multimedia for squeeze. [23:23] <relaxed> hmm, that command should work [23:25] <relaxed> iiu7: try -map 0:v -vcodec copy -map 0:a -acodec libmp3lame -aq 0 -ac 1 -map 0:s:0 -scodec copy -map 0:s:1 -scodec copy output.mkv [23:27] <iiu7> Same error: Number of streams.. [23:28] <relaxed> this is the only thing it said, "Number of stream maps must match number of output streams"? You didn't omit anything? [23:29] <iiu7> yes, only that! [23:31] <relaxed> I'm out of ideas [23:32] <iiu7> relaxed, thanks for your time!! I'll just have to keep those subs in then. At least I got the conversion of the audio. [23:35] <iiu7> It does the same even if I use -scodec copy for all the subtitle stream. [23:35] <iiu7> Just to try. [23:38] <cbsrobot> iiu7: try: ffmpeg -i foo.mkv map 0:0 -map 0:1 -map 0:2 -vcodec copy -acodec libmp3lame -aq 0 -ac 2 output.mkv -scodec copy -newsubtitle [23:40] <iiu7> Same error: Number of streams.. [23:41] <cbsrobot> ffmpeg -i foo.mkv map 0:0 -map 0:1 -map 0:2 -vcodec copy -acodec libmp3lame -aq 0 -ac 2 -scodec copy output.mkv -newsubtitle [23:43] <iiu7> I tried that myself, that gives me "At least one output file must be specified". [23:43] <iiu7> From what I understand, the following command should work and just copy ALL subtitles. [23:43] <iiu7> ffmpeg -i foo.mkv -map 0:0 -vcodec copy -map 0:1 -acodec libmp3lame -aq 0 -ac 2 -map 0:2 -scodec copy -map 0:3 -scodec copy -map 0:4 -scodec copy -map 0:5 -scodec copy output.mkv [23:43] <iiu7> I tried this to make sure all streams are covered.. but.. this also gives me Number of streams.. [23:44] <iiu7> I don't get that! [23:47] <relaxed> iiu7: run, dd if=foo.mkv of=output.mkv bs=10M count=1 [23:48] <relaxed> and stick a sample up somewhere [23:51] <iiu7> Is there a place to stick it without registration? [23:51] <cbsrobot> megaupload [23:51] <cbsrobot> iiu7: last try - without reencoding: [23:51] <cbsrobot> ffmpeg -i foo.mkv -vcodec copy -map 0.0 -acodec copy -map 0.1 output.mkv -scodec copy -map 0.2 -newsubtitle [23:52] <iiu7> Number of streams.. :) [23:52] <cbsrobot> wtf [23:53] <iiu7> 1 sec, I'll upload the sample [23:54] <relaxed> iiu7: get a dropbox account for 2gb of free storage http://db.tt/cDWPLzP [23:54] <iiu7> I have another box posting something to usenet, so it's a bit slow in the upload.. 2 min [23:58] <iiu7> I made it smaller, only 1 MB, here: http://www.megaupload.com/?d=SPLV9RPC [00:00] --- Tue Jan 3 2012

1 0

ffmpeg-devel.log.20120101
by burek 02 Jan '12

02 Jan '12

[00:09] <michaelni> happy new year everyone ... :) [00:10] <microchip_> michaelni: for you too! [00:10] <microchip_> \o/ [00:32] <ubitux> :) [00:32] <ubitux> time to update: cmdutils.c:static const int this_year = 2011; [02:49] <CIA-101> ffmpeg: 03Shitiz Garg 07master * rc3513477bb 10ffmpeg/libavcodec/cook.c: [02:49] <CIA-101> ffmpeg: cook: K&R formatting cosmetics [02:49] <CIA-101> ffmpeg: Signed-off-by: Diego Biurrun <diego(a)biurrun.de> [02:49] <CIA-101> ffmpeg: 03Janne Grunau 07master * rc11b92b304 10ffmpeg/Changelog: [02:49] <CIA-101> ffmpeg: Changelog: restore version <next> header [02:49] <CIA-101> ffmpeg: Indeo 4 decoder and SMJPEG demuxer are not included in 0.8_beta1. Move [02:49] <CIA-101> ffmpeg: them to version <next>. [02:49] <CIA-101> ffmpeg: 03Janne Grunau 07master * rbcc7396065 10ffmpeg/ (configure libavcodec/pthread.c): [02:49] <CIA-101> ffmpeg: threads: add sysconf based number of CPUs detection [02:49] <CIA-101> ffmpeg: Can act as fallback and should work on a couple of Unix systems. [02:49] <CIA-101> ffmpeg: 03Justin Ruggles 07master * r481a7ff5bd 10ffmpeg/libavcodec/mpegaudiodec.c: mpegaudiodec: fix compilation when testing the unchecked bitstream reader [02:49] <CIA-101> ffmpeg: 03Ronald S. Bultje 07master * r464f26889c 10ffmpeg/libavcodec/mpegaudiodec.c: [02:49] <CIA-101> ffmpeg: mpegaudiodec: fix seeking. [02:49] <CIA-101> ffmpeg: The safe bitstream reader does not allow using skip_bits_long() to seek to a [02:49] <CIA-101> ffmpeg: point before the start of the buffer, which was needed by the mp3 decoder. [02:49] <CIA-101> ffmpeg: This change instead calculates the start point of the first valid granule and [02:49] <CIA-101> ffmpeg: skips to that position. [02:49] <CIA-101> ffmpeg: Signed-off-by: Justin Ruggles <justin.ruggles(a)gmail.com> [02:49] <CIA-101> ffmpeg: 03Oana Stratulat 07master * rfebd022228 10ffmpeg/tests/ (13 files in 3 dirs): [03:52] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r57581e4110 10ffmpeg/libavcodec/pthread.c: [03:52] <CIA-101> ffmpeg: pthread: dont autoenable frame threads with visualization [03:52] <CIA-101> ffmpeg: This combination doesnt work [03:52] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [03:52] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r490c97bdf5 10ffmpeg/libavformat/mpegenc.c: [03:52] <CIA-101> ffmpeg: mpegpsenc: Fix integer overflow of the muxrate calculation. [03:52] <CIA-101> ffmpeg: this should fix the failure of h264-bsf-mp4toannexb on freebsd/openbsd [03:52] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [04:44] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07master * r03d7d8faef 10ffmpeg/libavcodec/pamenc.c: Support gray8a pam encoding. [05:04] <CIA-101> ffmpeg: 03Carl Eugen Hoyos 07master * r8dc973e6d1 10ffmpeg/cmdutils.c: Update copyright year. [10:38] <Granjow> Why does this http://codepad.org/HruatjSg segfault in valgrind with an invalid write in avpicture_alloc on line 70? [10:40] <Granjow> Backtrace looks like this http://codepad.org/6PmmKl1h [10:44] <nevcairiel> you allocate a rgb24 buffer, but for the rgb stride you use width*4 .. shouldnt that be *3? 24-bit and all [10:46] <nevcairiel> (also, i would make sure that the linesize is at least 16-byte aligned) [10:46] <nevcairiel> but that may cause issues with writing the file, but huh [10:46] <nevcairiel> the first one is a clear mistake, imho [10:48] <Granjow> nevcairiel: Wow! Indeed, this fixed the invalid write. How can I ensure alignment? [10:49] <nevcairiel> there is the FFALIGN macro, however i don't know if PNG can deal with that. :) [10:50] <nevcairiel> probably should [10:50] <nevcairiel> just use FFALIGN(3*pCodecContext->width, 16) and see if everything still works [10:51] <nevcairiel> oh [10:51] <nevcairiel> need to do that when allocating the picture, too, of course [10:53] <nevcairiel> i would recommend to use av_image_alloc instead of avpicture_alloc, it lets you specify an alignment, and you don't need to do anything manually [10:56] <Granjow> Still works. At least the part that worked before :) http://codepad.org/UTRIChQa line 92; did I use it correctly? [10:56] <Granjow> I have an invalid read on line 135 now: http://codepad.org/9mBvQbYk but in my opinion I passed all parameters correctly. [10:57] <nevcairiel> yes, however your buffer might be too small now, you only allocate it for "width" but with the alignment it may use more [10:59] <Granjow> So I should use int rgbLines[4+2] approximately for the additional 16 bits? [11:04] <Granjow> Hm. Changed to av_picture_alloc now and the invalid read is gone; however I get: [IMGUTILS @ 0x7fefffdc0] Picture size 0x0 is invalid [NULL @ 0xce45f80] buffer smaller than minimum size [11:06] <nevcairiel> that would only happen if you give it width/height of 0 [11:10] <Granjow> Yes; I thought if I already had to give width/height to av_image_alloc it would also fill in those fields, but it did not so they were still 0. Again the same invalid read as I posted above. [11:16] <Granjow> sws_scale seems to break a lot :S When I watch pRgbFrame->data in the debugger, it disappears after this call. [11:24] <Granjow> If I got this correctly then I can use the linesizes created by av_image_alloc, right? Do I have any chance now to find out why sws_scale does not do what it should? http://codepad.org/tIjOD0GT line 100 [11:26] <Granjow> pFrame->data is filled in by avcodec_decode_video2, pFrame->linesize too, so nothing I could change there. pRgbFrame fields are filled by av_image_alloc. pCodecContext->height is the correct height too according to my debugger. [11:32] <nevcairiel> looks fine so far [11:33] <nevcairiel> one thing i noticed is that you seem to allocate a buffer further down but never free it (the "buf" buffer), but unless you let it run for a while and it runs out of memory, thats not the cause i guess :p [11:34] <Granjow> I currently don't even run it once because of the segfault :D [11:37] <Granjow> But you're right, I'm freeing it now. What else could be wrong? Is the swsContext bad? [11:37] <Granjow> Or my call to avcodec_decode_video2? [11:39] <Granjow> Do I need to av_image_alloc pFrame too? [11:53] <Granjow> Tjoppen: If you happen to be online -- Do you maybe have an idea how I can find out where the problem lies? I get a segfault on line 138 (avcodec_encode_video) here http://codepad.org/UmbptCkV, looks like this: http://codepad.org/A8sVNz1j [11:54] <Granjow> In my debugger some local variables already disappear after the sws_scale call. [12:11] <Granjow> :( [12:13] <nevcairiel> hm [12:14] <nevcairiel> dont you need to call avcodec_open on the encoder, too [12:14] <nevcairiel> you dont seem to do that [16:14] <cbsrobot> michaelni: are you around ? [16:19] <CIA-101> ffmpeg: 03Stefano Sabatini 07master * r42a8ac94d9 10ffmpeg/ (3 files in 2 dirs): [16:19] <CIA-101> ffmpeg: vf_tinterlace: implement interlace mode 5 [16:19] <CIA-101> ffmpeg: Allow creating interlaced bottom field first video. [16:40] <michaelni> cbsrobot, yes, how can i help ? [16:45] <CIA-101> ffmpeg: 03Jean First 07master * r2e0efa332b 10ffmpeg/libavfilter/af_astreamsync.c: [16:45] <CIA-101> ffmpeg: af_astreamsync: remove unused variable [16:45] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [16:49] <cbsrobot> michaelni: the flvdec patch i sent [16:49] <cbsrobot> do you mean to initialise like this : [16:49] <cbsrobot> int av_unused sample_rate = 0; [16:49] <cbsrobot> or [16:49] <cbsrobot> int av_unused sample_rat; [16:50] <cbsrobot> *rate; [16:50] <michaelni> av_uninit(sample_rate) [16:50] <michaelni> the unused was a typo, sorry [16:52] <cbsrobot> need to run ... [17:42] <durandal_1707> can we get more y41p samples? [17:44] <durandal_1707> i think that this codec just always encodes image bottom to up [17:57] <durandal_1707> is #322 so hard to fix? [21:48] <durandal_1707> michaelni: so what I should do with this flipping for Y41P? Apparently other software/hardware can output this format too (v4l2)... [21:52] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * recb14b8af7 10ffmpeg/libavcodec/h264.c: [21:52] <CIA-101> ffmpeg: h264: add forgotten \n to "Invalid mix of idr and non-idr slices" [21:52] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:52] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r393253b4b3 10ffmpeg/libavcodec/h264.c: [21:52] <CIA-101> ffmpeg: h264: Print pass in NAL debug output [21:52] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:52] <CIA-101> ffmpeg: 03Jean First 07master * r6bd8d61378 10ffmpeg/libavformat/mov.c: [21:52] <CIA-101> ffmpeg: mov: av_dlog use PRId64 instead of %ld [21:52] <CIA-101> ffmpeg: Signed-off-by: Jean First <jeanfirst(a)gmail.com> [21:52] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:52] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r36fa781bd6 10ffmpeg/libavcodec/h264.c: [21:52] <CIA-101> ffmpeg: h264: reset picture->sync on allocation. [21:52] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:52] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * rfba6777363 10ffmpeg/libavcodec/h264.c: [21:52] <CIA-101> ffmpeg: h264: Fix recovery_frame initialization when recovery_frame is -1 [21:52] <CIA-101> ffmpeg: Signed-off-by: Michael Niedermayer <michaelni(a)gmx.at> [21:52] <CIA-101> ffmpeg: 03Michael Niedermayer 07master * r9ba43e50ef 10ffmpeg/libavcodec/ (h264.c h264_refs.c): [21:57] <michaelni> durandal_1707, iam not sure i understood the fliping issue? [21:57] <michaelni> does some software output it top down and some bottom up ? [21:59] <michaelni> i thoight your latest patch fixed the flip issue ? [22:00] <durandal_1707> i dunno, i have only one sample, both xanim and bt20 do flipping [22:01] <durandal_1707> s/bt20/bt411 via mplayer [22:04] <michaelni> hmm, t [22:04] <michaelni> hmm, try setting codec_tag in the encoder [22:09] <durandal_1707> how that should help? avi muxer already write Y41P twice [22:10] <michaelni> does it write positive height ? [22:10] <durandal_1707> yes [22:10] <michaelni> hmm, the linked bug was speaking of negative .... [22:11] <durandal_1707> yeah, I just thought that was causing crash but apperently it is something else [22:11] <michaelni> could be all the docs are wrong and Y41P with positive height is actually fliped [22:11] <michaelni> someone with a windows box that supports it could test that [22:12] <durandal_1707> but sample which works have not flipped height [22:12] <michaelni> can i have that sample ? [22:13] <durandal_1707> http://ffmpeg.org/trac/ffmpeg/ticket/102 [23:13] <rahulr92> Hi [23:17] <rahulr92> Hi I am new to open source and would like to contribute to FFmpeg. I picked the following task from the Small FFmpeg Tasks page: Implement a Vivo demuxer. It corresponds to ticket #132 https://avcodec.org/trac/ffmpeg/ticket/132 .I have went through mplayer demux_viv.c code, but could use additional assistance and pointers. Any guidance will be greatly appreciated. Thanks. [23:19] <durandal_1707> rahulr92: there is already patch for that demuxer and it is 2 years old (i have synced it with master locally but it segfaults though) [23:21] <rahulr92> durandal_1707: Alright. So I think I will scout for another beginner level task. [23:21] <durandal_1707> you can still work on it [23:22] <durandal_1707> but it may be hard for begginer... [23:23] <rahulr92> Okay. So can you please direct me towards something suitable for a beginner. [23:24] <CIA-101> ffmpeg: 03Paul B Mahol 07master * rc0f7c574e9 10ffmpeg/libavcodec/ (j2kdec.c mjpegenc.c): Cosmetics: Replace -1 with PIX_FMT_NONE. [23:27] <rahulr92> How about Port missing demuxers from MPlayer? Are they similarly hard? [23:27] <rahulr92> TiVo? [23:28] <durandal_1707> well if you can get samples... [23:29] <durandal_1707> ah they are on usual location [23:29] <durandal_1707> whan is almost 1GB though [23:31] <rahulr92> Sorry, but I am totally new to this. Slight hand holding or reference would be of great help. [23:31] <rahulr92> samples? [23:31] <durandal_1707> ty file [23:33] <durandal_1707> http://wiki.multimedia.cx/index.php?title=Tivo [23:34] <rahulr92> On a differeent note, how do I create an account in the wiki page. [23:35] <rahulr92> Is it same for all wikimedia sites? [23:37] <durandal_1707> no you need to contact mike at multimedia.cx [23:38] <rahulr92> Alright. I will go through the documentation first and be back. Thanks a lot. [23:40] <rahulr92> Just I doubt, should I send a mail to mike(a)multimedia.cx requesting an account? [23:41] <michaelni> durandal_1707, xanim seems to unconditionally flip y41p [23:41] <michaelni> see AVI_Decode_Y41P() [23:42] <durandal_1707> michaelni: yes I'm aware of that, i'm more interested in encoders side of story :) [23:43] <durandal_1707> rahulr92: if you want to edit wiki than yes [23:43] <rahulr92> Ok. Thanks [23:45] <michaelni> its very likely the encoder will flip too [23:45] <michaelni> otherwise the existing sample is hard to explain [23:50] <durandal_1707> there is flipped y41p colorspace thread on doom9 forum [23:55] <michaelni> maybe theres a fliped and non fliped one [23:56] <michaelni> id say add support for the sample we have and let users find a sample that is the other way around [23:56] <michaelni> then we can figure out what the difference is and support it too [00:00] --- Mon Jan 2 2012

1 0

ffmpeg.log.20120101
by burek 02 Jan '12

02 Jan '12

[00:00] <defaultro> hey folks, check out this hdr test video I made. Of course, ffmpeg was used about 90% during the processing :) http://www.youtube.com/watch?v=s44o8MBfGg0 please let me know [00:05] <defaultro> btw, can ffmpeg do what twixtor does? [00:07] <pasteeater> ffmpeg can slow down or speed up, but by dropping or duping frames, and it can't do it on a "curve" (maybe programatically) like twixtor appears to be doing [00:08] <Granjow> pasteeater: no, but you can use my program for this [00:08] <Granjow> http://slowmovideo.granjow.net/index.html [00:08] <defaultro> ok [00:08] <Granjow> happy new year and so on, good night [00:09] <defaultro> WOW!!!!!!!!!! [00:09] <defaultro> That is amazing Granjow!!!!!! [00:10] <defaultro> Thanks for sharing :) Happy New Year too!!! [00:14] <pasteeater> defaultro: what is the input format from the camera? [00:16] <defaultro> mov file [00:16] <defaultro> let me ffmpeg it [00:16] <defaultro> Stream #0.0(eng): Video: h264, yuvj420p, 1920x1088, 71887 kb/s, 30 fps, 30 tbr, 3k tbn, 6k tbc [00:17] <defaultro> that's the format that Magic Lantern on my Canon 50D created [00:17] <defaultro> what do you think about my hdr test? does it look good? [00:21] <pasteeater> i would call it faux HDR because you are using one input with one "range". [00:21] <defaultro> yup [00:21] <defaultro> does it look bad? [00:22] <defaultro> magic lantern's real hdr implementation is based on iso. It records frames at iso 800 and iso 1600 [00:23] <defaultro> the recording is frame 1 at iso 800, frame 2 at iso 1600, frame 3 is at iso 800 and so on [00:23] <defaultro> magic lantern's hdr support on my camera is still buggy that's why I made that faux hdr solution [00:24] <defaultro> if there was a raw recording like when we take still pictures, that would be awesome [00:24] <pasteeater> the original looks better to me. [00:24] <defaultro> :D [00:25] <defaultro> it's dark, can't see the cabinet [00:25] <pasteeater> interesting experiment though [00:25] <defaultro> yup [00:25] <defaultro> it's my first time, I'll use a different tool next time [00:26] <pasteeater> i just installed CHDK on a SD940. i want to make some time lapse on a rail dolly. [00:26] <defaultro> awesome [00:26] <defaultro> i've been wanting to create a motorized dolly but I haven't started [00:27] <defaultro> i already have the stepper motor. Hopefully, I'll be able to start it next year [00:27] <pasteeater> arduino? [00:27] <defaultro> not arduino, i was told another one [00:27] <defaultro> i forgot [00:27] <defaultro> i'll have to look up from my bookmark [00:28] <defaultro> brb, need to grab something from CostCo [01:56] <qxt> Is there a reason that ffmpeg is only using on of my i7 2600k cores? [01:56] <qxt> one [01:57] <qxt> The cmd I am running is.. [01:57] <qxt> ffmpeg -i Drive.2011.BluRay.1080p.x264.DTS-LTT.mkv -pass 1 -vcodec libx264 -preset veryslow -b:v 1119k -f mp4 -an -y /dev/null && ffmpeg -i Drive.2011.BluRay.1080p.x264.DTS-LTT.mkv -pass 2 -vcodec libx264 -preset veryslow -b:v 1119k -acodec libfaac -ab 128k -ac 2 drive.mkv [01:59] <qxt> also I am wondering if using that -f mp4 is OK. I tried with -f mkv but I was got a msg saying that was a no go. [02:07] <qxt> Do I need to set threads manually in the above cmd? [02:31] <pasteeater> qxt: you can try '-threads 0' if you're using an older ffmpeg for it to auto choose an appropriate value, otherwise it's now default, IIRC [02:32] <pasteeater> you should use -f matroska [02:32] <pasteeater> but hopefully it won't care [02:33] <qxt> pasteeater I compiled it about a week ago on a debian machine. So I write the whole word matroska not just mkv? [02:33] <pasteeater> yeah. see 'ffmpeg -formats' for a list of available -f options [02:35] <qxt> there it is "E matroska Matroska file format" [02:37] <pasteeater> for 1st pass (on the very rare occasion i use it) i make the -f the same as the 2nd pass output format. [02:39] <qxt> pasteeater seems like it got sentimental and tossed out "Decoder (codec text) not found for input stream #0:2" when I used the -f matroska cmd ? [02:40] <pasteeater> use a pastebin site to show your ffmpeg command and the complete console output [02:40] <qxt> sure one sec [02:44] <qxt> pasteeater http://pastebin.com/A3hyKhxi [02:45] <pasteeater> you're applying -threads 0 as an input option, not an output option [02:45] <qxt> lol that might have something to do with it ! [02:45] <pasteeater> and your ffmpeg is probably new enough that you don't even need -threads 0 [02:46] <pasteeater> add -sn to ignore subtitles [02:46] <pasteeater> as an output option to both passes [02:47] <qxt> pasteeater OK I removed the thread 0 but I still got the error. Ill try the -sn. Looking at the error it looks like your right on the money. [02:50] <qxt> pasteeater coolio got rid of the error. Oddly I can not use more then one core though? Could it be that I am using veryslow? [02:54] <pasteeater> look at the ffmpeg output. see the x264 options. does it show threads=12? [02:54] <pasteeater> (assuming you've removed -threads 0) [02:55] <pasteeater> line 29 in your last paste is an example of the x264 option output [02:58] <qxt> Yeah I was just looking at that. Even without the -threads 0 I am getting this "threads=12 sliced_threads=0 nr=0 " so it seems to detect something at least. [02:58] <qxt> Just that I am only using about 20-25% of my cpu [02:59] <pasteeater> i don't know, but if it is showing "threads=12", which is the number that -threads 0 chose (i7 i'm guessing?) then you know it's not "user error". [03:00] <pasteeater> ohm duh, you mentioned i7 [03:01] <pasteeater> i have to go now. good luck with that. [03:02] <qxt> Yeah a SB i7 with HT. Thx for your help. All the best [04:47] <grap3_ap3> Please help me understand. Recorded .ogv file looks/sounds just fine. Converted to .mkv for YouTube upload. Resolution of video gets munged. Text in terminal/browser unreadable but bigger text (in word processor) while still showing pixelation, is better and actually can be read. This pastebin has the command line input/output (http://pastebin.com/x7a8ZvSN) How can I improve the resolution issues during conversion? [08:15] <llappy> so, I have THP files at 608x480 [08:16] <llappy> they're meant to be displayed in 16:9 by stretching, but I can't figure out how to change the aspect ratio without cropping [08:54] <llappy> never mind, figured it out [10:27] <upvar> i'm calling the ffmpeg binary from another process and with some inputs i get a spam of '[libx264 @ 0x25afa60] VBV underflow ...' and as a result the encoder doesn't seem to terminate, i can't really put a timeout on the encoding task because the input videos vary wildly and i can't guess how long a task should take, can i tell ffmpeg some how to terminate if something goes wrong? [12:49] <funyun> hi. i'm trying to convert a movie with DTS audio to AC3 audio with ffmpeg but i get this error http://pastebin.com/pDvMzisA . can anyone help me? [12:58] <cbsrobot> funyun: not sure if it's an audio problem - try to convert the audio without the video [12:58] <cbsrobot> like: [12:58] <cbsrobot> ffmpeg -i input.mkv -acodec ac3 -ab 640k -ac 6 -vn output.mkv [13:13] <funyun> cbsrobot: that works. but how do i get the audio and the vid together? [13:38] <cbsrobot> funyun: I'm not sure how to fix this: [matroska @ 0x14dd7c0] pts < dts in stream 0 [13:38] <cbsrobot> but try [13:38] <cbsrobot> ffmpeg -i input.mkv -vcodec copy -an output.mkv [13:39] <cbsrobot> and then you can try to merge the two files [13:39] <cbsrobot> or maybe you have to transcode the video stream aswell [13:39] <cbsrobot> It's just some wild guessing [14:19] <qxt> Is there a tool out there that will calculate the right bit rates needed for a predetermined size video? [14:37] <Agiofws> http://www.youtube.com/watch?v=kwhkU4-FI-U [14:44] <qxt> Agiofws "Robo bunny flock of birds" Nice video ... ? [14:46] <Agiofws> ? [15:38] <funyun> join mw3 [16:03] <qxt> How do I copy a subtitle stream from a video and save it as a srt file? [16:06] <Matty> If the subtitles are part of the video stream(part of the image) then there is no way. If they are separate stream then something like "ffmpeg -i video subs.srt". [16:08] <Matty> Unfortunetly this is broken for mkv files at the moment, not sure about other. [16:08] <qxt> Matty Yeah they are part of the video stream. Was hoping there might be a way to map the stream and dump it to a file... [16:08] <qxt> Matty but there would be issues about font type etc I suppose. [16:10] <grap3_ap3> Please help me understand. Recorded .ogv file looks/sounds just fine. Converted to .mkv for YouTube upload. Resolution of video gets munged. Text in terminal/browser unreadable but bigger text (in word processor) while still showing pixelation, is better and actually can be read. This pastebin has the command line input/output (http://pastebin.com/x7a8ZvSN) How can I improve the resolution issues during conversion? [16:36] <Mavrik> grap3_ap3, for starters, stop encoding a 3360x1040 video to just 1.5Mbit/s [16:37] <Mavrik> that kind of resolutions are usually encoded in something closer to 20-40Mbit/s [16:37] <grap3_ap3> Mavrik, thank you! Didn't do it intentionally, likely due to my dual monitor setup. [16:38] <Mavrik> second of all, set -vcodec libx264 -preset slow to explicitly set preset for h.264 encoding [16:38] <Mavrik> grap3_ap3, maybe you should downsample the video to 1080p or some such, Youtube is going to do this anyway [16:38] <Mavrik> and you won't have to wait for several GB of video to upload [16:39] <grap3_ap3> Mavrik, I'm a video noob, can you point me toward resources on learning how to do this downsampling? [16:39] <Mavrik> grap3_ap3, just tell ffmpeg to resize your video, e.g. "-vf scale=-1:1080" [16:39] <Mavrik> grap3_ap3, this will make small text blurry since you're actually resizing the actual picture [16:40] <Mavrik> grap3_ap3, so use something like "ffmpeg -i <yourile> -vcodec libx264 -preset slow -vf scale=-1:1080 -b:v 8000k -acodec libmp3lame out.mkv" [16:41] <grap3_ap3> Mavrik, thank you so much for your help. You confirmed what I was thinking since I started this project- that the recording resolution is the problem. [16:41] <Mavrik> grap3_ap3, play with the -b:v parameter for quality - higher is better quality + larger file [16:42] <grap3_ap3> Mavrik, trying your suggestion now, Thanks again! I really need clear text on this video so will look at ways to change the recording resolution without changing the system resolution- I thought I was combating that by selecting only the deisred recording area but apparently that was the first of several incorrect presumptions on my part here :( [16:43] <Mavrik> grap3_ap3, recording only part of the screen is your only option really - if you lower the resolution of video (while keeping screen resolution the same) it'll make the text blurry [16:43] <Mavrik> grap3_ap3, you can also use video editing tools to actually crop the video [16:44] <grap3_ap3> Mavrik, this was supposed to be a 20minute project- turned into 2 days so far because of this stupid encoding issue.. I'd like to avoid having to actually edit the vid if possible. \ [16:45] <Mavrik> well, is there really any reason for you to use such a high res? instead of switching you screen to 1280x720 or something like that? [16:45] <Mavrik> you'll have all kinds of problem getting video of 3300xsomething resolution around [16:46] <grap3_ap3> only because that's what fits best for this dual monitor setup. [16:47] <Mavrik> well, turn off one monitor, switch the other to lower res, record what you need to record [16:47] <Mavrik> that'll save you tons of trouble [16:47] <grap3_ap3> Guess I'm not sure why I didn't try that. [16:47] <grap3_ap3> I was really just thinking it would work as-is [16:48] <grap3_ap3> thanks for the help. [16:52] <qxt> Going to do a 2 pass video. Wanted to know if this is the right way to get the sound size in MB [16:52] <qxt> (128 [kilobits] * 6047 [time sec]) / 8192 = size MB [16:53] <qxt> Was thinking about the 8192 part mostly. [16:54] <Mavrik> ((kbit /s * time sec) / 8) / 1024 [16:54] <Mavrik> so yea, about right [16:55] <qxt> Mavrik nice thx [17:40] <qxt> Is there a cmd to get the time duration of a video? [17:40] <alesan> hi... I have this 5.1 audio in my movie, how can I correctly transform it in normal stereo? [17:40] <qxt> -acodec libfaac -ab 128k -ac 2 [17:41] <qxt> if you have a bunch of streams add -sn [17:41] <relaxed> qxt: ffprobe [17:41] <qxt> thx relaxed [17:43] <qxt> alesan if you just want to do this fast "ffmpeg -i input -vcodec copy -acodec libfaac -ab 128k -ac 2 output.mkv [17:44] <alesan> thanks! [17:44] <alesan> I'll try [17:44] <qxt> np [17:45] <alesan> so 128k is enough for a stereo signal? [17:46] <alesan> what is -sn for?? [17:47] <qxt> alesan that is in case you have a bunch of stream it will not toss an error. Try it without the -sn. If you get a warning then use -sn (streams no) [17:47] <alesan> well I do get that error [17:47] <alesan> with it, it works correctly it seems... let me try again [17:48] <qxt> alesan =) [17:48] <alesan> I am uploading a video of yesterday's party to my mom... [17:48] <alesan> I need to recompress it a bit, scale it down, and stereo sound is more than enough [17:49] <alesan> she could not attend my son's first birthday party :( [17:49] <qxt> alesan for a home video I find 128k to be just fine. Its up to you though. [17:52] <qxt> alesan what size do you want to compress your video to? [17:53] <alesan> I just want it to be more easily transferable with scp [17:53] <alesan> the original is... [17:54] <alesan> 190MB [17:54] <alesan> recompressing it with cfr 24, converting to 30fps and scaling to 1280x720, it's 11MB apparently [17:55] <alesan> for 64 seconds [17:56] <qxt> alesan Does it look OK? [17:56] <alesan> I think it does. [17:56] <alesan> it's that the original, 1920x1080p 60fps, with 5.1 audio, is just too big [17:57] <qxt> alesan I find about a crf of about 18-27 to be OK [17:57] <qxt> depends on what I am encoding. [17:57] <alesan> using 24 for now, maybe I should lower it a little [17:58] <alesan> but my mom does not care too much I suppose [17:58] <qxt> 5.1 audio on a home video to be sent over the net is silly. The sound cmd above is more appropriate. [18:00] <qxt> Unless you want to hear the maids baby crying in surround.. [18:07] <qxt> Is there a reason that pass 1 is not using threading - ffmpeg -i movie.mkv -pass 1 -vcodec libx264 -preset veryslow -b:v 1226.72134943k -f matroska -sn -an -y /dev/null [18:08] <Mavrik> it's just an analysis phase [18:08] <Mavrik> probably nothing to do in multiple threads [18:09] <qxt> Mavrik yeah this is just the analysis but it is only doing 18 fps on a Sandy Bridge i7 2600k [18:10] <qxt> Mavrik the next phase does thread though. [18:10] <Mavrik> that's normal. [18:11] <qxt> Mavrik thx good to know I did not fudge the cmd. [20:15] <aca20031> So if I want to convert to h.264 in a .mp4 do I do -vcodec h264, or -vcodec mp4 or something entirely different? [20:17] <relaxed> libx264 is a h264 encoder and mp4 is a container [20:18] <aca20031> so... [20:18] <aca20031> -f mp4 -vcodec h264? [20:18] <relaxed> so, ffmpeg -i input -vcodec libx264 -preset slow -crf 18 output.mp4 [20:19] <aca20031> ok, ffmpeg -formats doesnt list "libx264" - how does one list them? [20:19] <relaxed> because it's a codec "ffmpeg -codecs" [20:20] <aca20031> then what is h264 as a "format"? [20:20] <aca20031> sorry, im a bit slow on this media stuff. [20:20] <relaxed> h264 format is the raw video [20:21] Action: aca20031 has a headache [20:21] <relaxed> what is your goal? targeting a specific device? [20:22] <aca20031> yeah, i want to convert all my videos to h264 in mp4 so it doesnt have to be transcoded on the fly for the PS3 and future media clients [20:24] <relaxed> which version do you have? [20:24] <aca20031> of ps3? [20:24] <relaxed> no, ffmpeg [20:24] <aca20031> oh the latest built binary for windows, let me see [20:25] <aca20031> FFmpeg git-f514695 64-bit Shared (Latest) [20:25] <aca20031> ffmpeg N-36193-gf514695 [20:28] <relaxed> aca20031: this will create a 2 minute sample to test. ffmpeg -i input -vcodec libx264 -profile high -preset slow -threads 0 -crf 18 -level 42 -maxrate 25000k -bufsize 25000k -acodec libvo_aacenc -ab 192k -ac 2 output-sample.mp4 [20:28] <relaxed> oh, add "-t 120" after input [20:28] <aca20031> so very many switches [20:30] <relaxed> devices require specific settings [20:31] <aca20031> do you know of a better future proof format? I dont want the PS4 or the Xbox 720 to refuse to play them once i convert them :P [20:32] <relaxed> future devices will be able to play it. [20:33] <relaxed> you may need to limit refs, so if the above doesn't work try adding "-refs 3" [20:37] <relaxed> aca20031: there's handbrake if you ned something clicky http://handbrake.fr [20:37] <relaxed> s/ned/need/ [20:46] <aca20031> command lines will probably be faster, it just doesnt help that i dont fully understand the millions of different encodings vs codecs vs containers vs 'raw video' lol [20:47] <aca20031> nevermind that audio has its own crap [21:23] <defaultro> hey folks, does it matter which format we use? Is .mov better than .mts? [22:04] <qxt> Was wondering if there is a way to see if TurboBoost is running on my i7 2600. On Debian kenel 3.1.0-1-amd64, ffmpeg compiled a week ago. [22:06] <qxt> cat /proc/cpuinfo I hear does not show TurboBoost info [22:58] <linuxstony> Hello ladies and gentlemen - i am trying to build ffmpeg-git together with x264-git on arch linux. x264 complained about a missing libavformat.so.52 so i thought about recompiling ffmpeg from git together with the latest x264. Sadly ffmpeg does not build because of the following error: ERROR: libx264 version must be >= 0.118. and I should come here for help.The installed x264 version is "x264-git 20120101-1" if that is of [22:58] <linuxstony> any help. [22:59] <linuxstony> Does anyone have an advice for me - as i need to create some Savage2 videos that users are waiting for :) - and sorry if this is a noobish question - i tried google and the some boards to find a solution for that - sadly i didnt find anything that indicated something i could try. [23:25] <Matty> linuxstony: I do not know what is your problem, but ffmpeg depends on x264. So x264 should be build first and should not complain about missing ffmpeg while compiling. [00:00] --- Mon Jan 2 2012

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