[issue716] IMC decoding broken
New submission from Benjamin Larsson [EMAIL PROTECTED]: Something broke decoding since it was commited. What revision that did it is unknown. -- messages: 3651 priority: normal status: new substatus: new title: IMC decoding broken type: bug __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue716 __
[issue715] channel mask lookup in mkv files
New submission from Benjamin Larsson [EMAIL PROTECTED]: avcodec_get_channel_layout_string() is always called before the proper channel_mask is set for ftp://upload.mplayerhq.hu/incoming/DTS-48-24-4.0/Edward.Scissorhands.x264.DTS-4.0-48_24.mkv And when converting this to a wav file the wrong channel mask is set. ./ffmpeg -y -i Edward.Scissorhands.x264.DTS-4.0-48_24.mkv out.wav FFmpeg version SVN-r15829, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --disable-optimizations --enable-debug=3 --disable-mmx libavutil 49.12. 0 / 49.12. 0 libavcodec52. 3. 0 / 52. 3. 0 libavformat 52.23. 1 / 52.23. 1 libavdevice 52. 1. 0 / 52. 1. 0 built on Nov 15 2008 19:48:08, gcc: 4.3.2 Input #0, matroska, from 'Edward.Scissorhands.x264.DTS-4.0-48_24.mkv': Duration: 00:01:16.99, start: 0.00, bitrate: N/A Stream #0.0(eng): Video: h264, yuv420p, 1280x688, 23.98 tb(r) Stream #0.1: Audio: dca, 48000 Hz, quad, s16 Output #0, wav, to 'out.wav': Stream #0.0: Audio: pcm_s16le, 48000 Hz, quad, s16, 3072 kb/s Stream mapping: Stream #0.1 - #0.0 Press [q] to stop encoding size=4004kB time=10.68 bitrate=3072.0kbits/s video:0kB audio:4004kB global headers:0kB muxing overhead 0.001073% ./ffmpeg -i out.wav FFmpeg version SVN-r15829, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --disable-optimizations --enable-debug=3 --disable-mmx libavutil 49.12. 0 / 49.12. 0 libavcodec52. 3. 0 / 52. 3. 0 libavformat 52.23. 1 / 52.23. 1 libavdevice 52. 1. 0 / 52. 1. 0 built on Nov 15 2008 19:48:08, gcc: 4.3.2 Input #0, wav, from 'out.wav': Duration: 00:00:10.67, bitrate: 3071 kb/s Stream #0.0: Audio: pcm_s16le, 48000 Hz, quad, s16, 3072 kb/s The proper order for this file is: CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_BACK_CENTER, /// 4, C + L + R+ S Extracting the dts core with: ./ffmpeg -y -i Edward.Scissorhands.x264.DTS-4.0-48_24.mkv -acodec copy edward.dts and ./ffmpeg -y -i edward.dts out2.wav FFmpeg version SVN-r15829, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --disable-optimizations --enable-debug=3 --disable-mmx libavutil 49.12. 0 / 49.12. 0 libavcodec52. 3. 0 / 52. 3. 0 libavformat 52.23. 1 / 52.23. 1 libavdevice 52. 1. 0 / 52. 1. 0 built on Nov 15 2008 19:48:08, gcc: 4.3.2 Input #0, dts, from 'edward.dts': Duration: 00:01:15.67, bitrate: 1535 kb/s Stream #0.0: Audio: dca, 48000 Hz, 4 channels (FL|FR|FC|BC), s16, 1536 kb/s Output #0, wav, to 'out2.wav': Stream #0.0: Audio: pcm_s16le, 48000 Hz, quad, s16, 3072 kb/s Stream mapping: Stream #0.0 - #0.0 Press [q] to stop encoding size= 28884kB time=77.02 bitrate=3072.0kbits/s video:0kB audio:28884kB global headers:0kB muxing overhead 0.000149% so here the right channels are extracted but the output is still wrong. IE (FL|FR|FC|BC)-QUAD -- messages: 3648 priority: normal status: new substatus: new title: channel mask lookup in mkv files type: bug __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue715 __
[issue390] visualhub / isquint ffmpeg source?
Benjamin Larsson [EMAIL PROTECTED] added the comment: compn wrote: compn [EMAIL PROTECTED] added the comment: http://www.techspansion.com/visualhub/download.php isquint seems to have killed visualhub. the new project is open source i'm not sure if this bug is still valid. if a company stops distributing (but has already distributed source) of a gpl/lgpl project, do they still have to provide source? :P I think we should close this, they are providing source for the complete project (or will in the future). MvH Benjamin Larsson __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue390 __
[issue48] Play DVD-Audio files (DD Lossless/MLP decoder)
Benjamin Larsson [EMAIL PROTECTED] added the comment: We have a decoder for this format now. -- status: open - closed substatus: open - implemented _ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue48 _
[issue303] x264 2pass log file
Benjamin Larsson [EMAIL PROTECTED] added the comment: No answer for a while - closed. -- status: open - closed substatus: open - invalid __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue303 __
[issue412] Commit 11635 breaks MKV decoding
Benjamin Larsson [EMAIL PROTECTED] added the comment: This looks fixed to me, reopen if I'm wrong. -- status: open - closed substatus: reproduced - fixed __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue412 __
[issue485] Can't play an WMV file with 4CC: WVP2
Benjamin Larsson [EMAIL PROTECTED] added the comment: Move it to the right place. -- priority: normal - wish __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue485 __
[issue626] Support BMP 8 bits
Benjamin Larsson [EMAIL PROTECTED] added the comment: mat wrote: mat [EMAIL PROTECTED] added the comment: thanks work fine for me. With don't you submit/commit it ? Who knows how those Ukrainian guys reason. You could post it as a patch on the mailinglist. MvH Benjamin Larsson __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue626 __
[issue539] ffmpeg stalls when trying to play udp stream from an unavaible adress.
Benjamin Larsson [EMAIL PROTECTED] added the comment: Sebastian Jansson wrote: New submission from Sebastian Jansson [EMAIL PROTECTED]: I'm using ffmpeg as a lib to receive an udp stream via multicast. That works fine when there is a stream. But when I try to connect to an unavailable ipadress:port it just locks and wait for an connection. Same effect under Linux: gdb --args ffplay_g udp://235.2.2.2:5000 GNU gdb 6.8-debian Copyright (C) 2008 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as i486-linux-gnu... (gdb) run Starting program: /home/banan/soc/aacenc/ffmpeg/ffplay_g udp://235.2.2.2:5000 [Thread debugging using libthread_db enabled] FFplay version SVN-r14260, Copyright (c) 2003-2008 Fabrice Bellard, et al. configuration: libavutil version: 49.7.0 libavcodec version: 51.60.0 libavformat version: 52.17.0 libavdevice version: 52.0.0 built on Jul 17 2008 10:41:51, gcc: 4.2.3 (Ubuntu 4.2.3-2ubuntu7) [New Thread 0xb7ba26b0 (LWP 12387)] [New Thread 0xb78abb90 (LWP 12390)] [New Thread 0xb7023b90 (LWP 12391)] [Locked state, I pressed Ctrl-C] Program received signal SIGINT, Interrupt. [Switching to Thread 0xb7ba26b0 (LWP 12387)] 0xb7f40402 in __kernel_vsyscall () (gdb) bt #0 0xb7f40402 in __kernel_vsyscall () #1 0xb7d97cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7eae788 in SDL_Delay () from /usr/lib/libSDL-1.2.so.0 #3 0xb7e5c09b in SDL_WaitEvent () from /usr/lib/libSDL-1.2.so.0 #4 0x08061aaf in main (argc=0, argv=0x1bf0bd) at ffplay.c:2246 (gdb) quit The program is running. Exit anyway? (y or n) y I think the bt is from the wrong thread. MvH Benjamin Larsson -- title: ffmpeg stalls when trying to play udp stream from an unavaible adress. - ffmpeg stalls when trying to play udp stream from an unavaible adress. __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue539 __
[issue495] Nellymoser FLV audio out of sync (encodes faster than video)
Benjamin Larsson [EMAIL PROTECTED] added the comment: louislouis wrote: louislouis [EMAIL PROTECTED] added the comment: Hi, I think when the source flv was being recorded, the model had some microphone issues, so for the first minute there is silence. Rest of the video has audio nonetheless. This has happened twice now, where if the model has microphone issues at the start of recording, the source FLV will not play nice in FFmpeg. Unfortunately I've deleted the 1st problematic FLV. Also I'm not sure what kind of microphone issues they're having at the start. I guess they're not plugged in or muted or something. Anyway, let me know if you need any further info. Many Thanks Louis Ok, the issue then is that the decoder doesn't get any frames to decode for a specific duration and no silence is added instead. Thus the result is a unsynchronized result. This issue should be a dupe of another ticket and I don't know how to resolve this. FFmpeg needs to fill the stream with silence when there is a hole between the pts's. MvH Benjamin Larsson __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue495 __
[issue495] Nellymoser FLV audio out of sync (encodes faster than video)
Benjamin Larsson [EMAIL PROTECTED] added the comment: louislouis wrote: louislouis [EMAIL PROTECTED] added the comment: The FLV's (input and output) plays without any problems in all players. It's just the sound that screws up during converting. The FLV is generated using Wowza Media Server (http://www.wowzamedia.com/) . It's similar to Flash Media Server. Here is a link to the input flv (130.0 MB): http://76.76.14.14/archives/archiveTemp/chipmunk.flv Output FLV (4.MB) http://76.76.14.14/archives/archiveTemp/chipmunk2.flv The command used to generate the output FLV: ffmpeg -ss 0 -t 120 - i /var/www/html/archives/archiveTemp/chipmunk.flv -r 7 -async 1 /var/www/html/archives/archiveTemp/chipmunk2.flv Sorry the input file is so big. Let me know if you need any further info. Many Thanks Louis Hi, is there silence in the audio stream sometimes ? MvH Benjamin Larsson __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue495 __
[issue484] Rhozet Carbon Coder violates LGPL
Benjamin Larsson [EMAIL PROTECTED] added the comment: Diego Biurrun wrote: Diego Biurrun [EMAIL PROTECTED] added the comment: Can you try looking for some of the GPL parts of FFmpeg? Maybe they have enabled them. Looks like no GPL code in there. MvH Benjamin Larsson __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue484 __
[issue442] parameter for getting information about files
New submission from Benjamin Larsson [EMAIL PROTECTED]: Currently people are parsing the output of ffmpeg to extract info about media files. FFmpeg should add an -identify parameter that should output the file parameters in a future safe format. For example in xml something like this could be the output. containermov/container videostreammpeg-4/videostream fps25/fps audostreamac3/audostream samplerate44100/samplerate This would be nice for those who otherwise parse the output from -i. -- messages: 1976 nosy: banan priority: wish status: new substatus: new title: parameter for getting information about files type: feature_request __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue442 __
[issue382] DVDFab Decrypter Platinum fails to provide source and licensing information
Benjamin Larsson [EMAIL PROTECTED] added the comment: Vitor wrote: [...] Install fine and kind of run in wine. [EMAIL PROTECTED]:~$ strings codecs.dll (...) This software is derived from the GNU GPL XviD codec (1.1.0). Your software distributor has to give access to its source code. (...) swScaler: slices start in the middle! swScaler: internal error %s - %s converter hSwScaler: reducing / aligning filtersize %d - %d swScaler: %s is not supported as input format swScaler: %s is not supported as output format swScaler: %dx%d - %dx%d is invalid scaling dimension SwScaler: using unscaled %s - %s special converter SwScaler: output Width is not a multiple of 32 - no MMX2 scaler SwScaler: FAST_BILINEAR scaler, SwScaler: BILINEAR scaler, SwScaler: BICUBIC scaler, SwScaler: Experimental scaler, SwScaler: Nearest Neighbor / POINT scaler, SwScaler: Area Averageing scaler, SwScaler: luma BICUBIC / chroma BILINEAR scaler, (...) By the way, wouldn't it be nice to add to ffmpeg a phrase like the one from XviD if compiled with --enable-gpl? -Vitor Hi, I would like to have that also. But if we do it we should obfuscate it somehow so you just can't search the source for the string. And preferably we should have it in several places with different obfuscations. MvH Benjamin Larsson -- title: DVDFab Decrypter Platinum fails to provide source and licensing information - DVDFab Decrypter Platinum fails to provide source and licensing information __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue382 __
[issue351] Glitch when encoding to mp2
Benjamin Larsson [EMAIL PROTECTED] added the comment: When encoding frame nr 151 from glitch.wav something causes the encoded stream to be corrupted. -- status: new - open substatus: new - reproduced __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue351 __
[issue81] dca encoder (forward qmf)
Benjamin Larsson [EMAIL PROTECTED] added the comment: Hi, Alexander E. Patrakov wrote: Alexander E. Patrakov [EMAIL PROTECTED] added the comment: I have implemented the forward QMF for the non-perfect reconstruction case. Here is how I came to the implementation: 1) Noted that the subband synthesis (section C.6 of the specification) is just some linear transformation of input to output with a 512-sample history, so that linear algebra applies. Correct, it's a linear system. 2) Calculated the 512-sample response of the reference implementation to subband samples of the form: (0.0 ... 0.0 1.0 0.0 ... 0.0) followed by all-zero subband samples, for all 32 possible positions of 1.0. Concatenated the results into what's called [2] below. 3) Noticed that the convolution of [2] with the time-reversed response to the (1.0 0.0 ... 0.0) vector (i.e., the standard way to build a FIR filter designed to catch something) responds only to the first part of [2]. The same procedure somewhat works in higher subbands, too, but also catches unwanted noise in the neighbouring subbands. 4) Noticed that the response of the reference decoder implementation to (0.0 ... 0.0 -1.0 1.0 0.0 ... 0.0) with 1.0 in the i-th subband has a simple _approximate_ (with 0.25% error) relation to its response to (1.0 0.0 ... 0.0) (just multiply by a cosine with the appropriate frequency). The idea is that the relation is intended to be exact. The idea proved to be true - is I calculate convolutions with such signals (instead of the catching idea from step 3), the accuracy of forward + reverse QMF is greatly improved. That sounds like the description of a cosine modulated pseudo QMF. The analysis and synthesis filters should have this relation a(n)=s(L-1-n) where L is the length of the filter. If that is correct the forward transform is the same as the reverse but we need to reverse the filter taps. (Talking about the libavcodec/dca.c code.) 5) The rest is just some algebra that gives some speedup. Attached is a standalone proof-of-concept integer-only pseudo-encoder that takes a CD-quality wav file and produces a stereo DTS file of the same bitrate by throwing away high subbands and quantizing the rest with 16 bits. Note: accuracy of reverse + forward QMF test is improved a bit if you replace 105372028 with 105372029 in unqmf-int.c. Nice work! It will take some time for me before I have time to look at this properly though. As for the license of this code: I grant the ffmpeg project (but, currently, not anybody else) the permission to reuse it under GPL version 2 or later. In the future, the license of my code will be changed to GPL with some additional rights or to LGPL (not decided yet). I hope I can convince you to release the transform code in the LGPL license. It would be a valuable contribution to FFmpeg. MvH Benjamin Larsson _ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue81 _
[issue304] SigSEGV in libavcodec
Benjamin Larsson [EMAIL PROTECTED] added the comment: Hi, I tried the sample with ffmpeg and ffplay and it worked as it should, the problem must be in transcode. Please report this to the transcode developers or reproduce with ffmpeg/ffplay. -- status: open - closed substatus: needs_more_info - invalid __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue304 __
[issue257] mplayer fails to decode an ape stream
Benjamin Larsson [EMAIL PROTECTED] added the comment: Hi, can you reproduce this with ffmpeg/ffplay and attach the sample here ? __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue257 __
[issue258] converting 3gp to yuv4mpegpipe rawvideo fails
Benjamin Larsson [EMAIL PROTECTED] added the comment: yuv4mpegpipe issues confirmed. Output for yuv4mpegpipe: [EMAIL PROTECTED]:~/svnffmpeg/trunk$ ./ffmpeg -i ~/samples/MOV00159.3GP -f yuv4mpegpipe test.yuv FFmpeg version SVN-r11316, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: libavutil version: 49.6.0 libavcodec version: 51.49.0 libavformat version: 52.3.0 built on Dec 26 2007 17:22:30, gcc: 4.1.3 20070929 (prerelease) (Ubuntu 4.1.2-16ubuntu2) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/banan/samples/MOV00159.3GP': Duration: 00:00:10.6, start: 0.00, bitrate: 92 kb/s Stream #0.0(jpn): Video: h263, yuv420p, 176x144 [PAR 12:11 DAR 4:3], 29.97 tb(r) Stream #0.1(jpn): Audio: samr / 0x726D6173, 8000 Hz, mono File 'test.yuv' already exists. Overwrite ? [y/N] y Output #0, yuv4mpegpipe, to 'test.yuv': Stream #0.0(jpn): Video: rawvideo, yuv420p, 176x144 [PAR 12:11 DAR 4:3], q=2-31, 200 kb/s, 29.97 tb(c) Stream mapping: Stream #0.0 - #0.0 Press [q] to stop encoding error, non monotone timestamps 18018 = 18018 av_interleaved_write_frame(): Error while opening file And when I play the sample with ffplay it plays in correct speed but when I converted it to yuv4mpegpipe with an older ffmpeg the speed was different. -- substatus: needs_more_info - reproduced __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue258 __
[issue304] SigSEGV in libavcodec
Benjamin Larsson [EMAIL PROTECTED] added the comment: Hi, supply a sample and the command line used to reproduce the crash. -- status: new - open substatus: new - needs_more_info __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue304 __
[issue278] DTS 48/24 4.0 channels is not decoded properly
Benjamin Larsson [EMAIL PROTECTED] added the comment: Channel layout and channel order support for libavcodec. __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue278 __
[issue301] FFmpeg fails to build with 2.95
New submission from Benjamin Larsson [EMAIL PROTECTED]: i386/vc1dsp_mmx.c: In function `vc1_put_ver_16b_shift2_mmx': i386/vc1dsp_mmx.c:104: more than 10 operands in `asm' i386/vc1dsp_mmx.c: In function `vc1_put_shift2_mmx': i386/vc1dsp_mmx.c:200: more than 10 operands in `asm' i386/vc1dsp_mmx.c: In function `vc1_put_shift1_mmx': i386/vc1dsp_mmx.c:385: more than 10 operands in `asm' i386/vc1dsp_mmx.c: In function `vc1_put_ver_16b_shift1_mmx': i386/vc1dsp_mmx.c:386: more than 10 operands in `asm' i386/vc1dsp_mmx.c: In function `vc1_put_hor_16b_shift1_mmx': i386/vc1dsp_mmx.c:387: more than 10 operands in `asm' i386/vc1dsp_mmx.c: In function `vc1_put_shift3_mmx': i386/vc1dsp_mmx.c:390: more than 10 operands in `asm' i386/vc1dsp_mmx.c: In function `vc1_put_ver_16b_shift3_mmx': i386/vc1dsp_mmx.c:391: more than 10 operands in `asm' i386/vc1dsp_mmx.c: In function `vc1_put_hor_16b_shift3_mmx': i386/vc1dsp_mmx.c:392: more than 10 operands in `asm' make[1]: *** [i386/vc1dsp_mmx.o] Error 1 make[1]: Leaving directory `/home/banan/ffmpeg/libavcodec' make: *** [lib] Error 2 -- messages: 1296 nosy: banan priority: normal status: new substatus: new title: FFmpeg fails to build with 2.95 type: bug __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue301 __
[issue270] Transcoding FLV - MP4 (et al.): Audio Sample Rate for NellyMoser FLV file is misread
Benjamin Larsson [EMAIL PROTECTED] added the comment: Ok, sorry for wasting all your time. I took a better look now and the codec id number is FLV_CODECID_NELLYMOSER_8HZ_MONO. So the attached patches are wrong. The correct way to do this is to check if the codec id is FLV_CODECID_NELLYMOSER_8HZ_MONO and if so never reset the samplerate even if the metadata for it exist. An if (!acodec-sample_rate) before switch((int)num_val) should be the best way to solve this. __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue270 __
[issue77] swf muxer doesnt like adpcm_swf encoded audio
Benjamin Larsson [EMAIL PROTECTED] added the comment: Move issue. -- priority: normal - wish _ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue77 _
[issue278] DTS 48/24 4.0 channels is not decoded properly
Benjamin Larsson [EMAIL PROTECTED] added the comment: A solution for ffmpeg is in the works, it will take a couple months before it is ready. -- assignedto: - banan nosy: +banan __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue278 __
[issue270] Transcoding FLV - MP4 (et al.): Audio Sample Rate for NellyMoser FLV file is misread
Benjamin Larsson [EMAIL PROTECTED] added the comment: Can you try this patch and see if it fixes the problem ? __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue270 __Index: libavformat/flvdec.c === --- libavformat/flvdec.c (revision 11254) +++ libavformat/flvdec.c (working copy) @@ -183,7 +183,10 @@ case 44000: acodec-sample_rate = 44100 ; break; case 22000: acodec-sample_rate = 22050 ; break; case 11000: acodec-sample_rate = 11025 ; break; -case 5000 : acodec-sample_rate = 5512 ; break; +case 5000 : acodec-sample_rate = 5512 ; +if (acodec-codec_id == CODEC_ID_NELLYMOSER) +acodec-sample_rate = 8000; +break; default : acodec-sample_rate = num_val; } }
[issue271] Nellymoser Bug with silences transcoding
Benjamin Larsson [EMAIL PROTECTED] added the comment: The problem is in the flv demuxer. Not in the decoder. -- nosy: -banan __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue271 __
[issue58] Flash adpcm regressions test doesn't exist
Benjamin Larsson [EMAIL PROTECTED] added the comment: Added in revision 11160. -- status: open - closed substatus: open - implemented _ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue58 _
[issue271] Nellymosser Bug with silences transcoding
Benjamin Larsson [EMAIL PROTECTED] added the comment: De Cock Xavier wrote: New submission from De Cock Xavier [EMAIL PROTECTED]: I use the nellymosser decoder and it works quite well except in one case. It seems that this codec does not allocate datas for white spaces. So when i encode files with no sound in part of the video, it packs all part of the video sound at the beginning of the video. If i have a 30 sec video with a blank running from second 8 to 16, So, something like that: 15 10 15 20 25 30 ||||||| Video:vvv Sound: yy The recoded video with: /usr/bin/ffmpeg -i in.flv -acodec libmp3lame -ar 44100 -ab 32 -r 30 -f flv -y out.flv Will have those datas 15 10 15 20 25 30 ||||||| Video:vvv Sound:yy Furthermore, the playback of the generated flv seems to have problems. (hanging, slow down so on) I have no idea on how to fix this, I've put a sample file on the FTP there: /MPlayer/incoming/nellymosser_silence_bug -- messages: 1106 nosy: void priority: normal status: new substatus: new title: Nellymosser Bug with silences transcoding type: bug Hi, does it work with this decoder: http://code.google.com/p/nelly2pcm/downloads/list MvH Benjamin Larsson __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue271 __
[issue270] Transcoding FLV - MP4 (et al.): Audio Sample Rate for NellyMoser FLV file is misread
Benjamin Larsson [EMAIL PROTECTED] added the comment: Per Thomsen wrote: New submission from Per Thomsen [EMAIL PROTECTED]: When transcoding an flv file (generated by Red5) to mp4 and 3gp, the audio plays about 30% slow in the transcoded file. $ ffmpeg -i 2-1195540996717.flv -r 15 -acodec libfaac -vcodec mpeg4 -ar 16000 2-1195540996717.mp4 FFmpeg version SVN-r11067, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --enable-libmp3lame --enable-libfaad --enable-libfaac --enable-libx264 --enable-gpl --disable-vhook libavutil version: 49.5.0 libavcodec version: 51.48.0 libavformat version: 51.19.0 built on Nov 19 2007 22:39:14, gcc: 4.1.2 20070626 (Red Hat 4.1.2-13) Seems stream 0 codec frame rate differs from container frame rate: 1000.00 (1000/1) - 50.92 (611/12) Input #0, flv, from '2-1195540996717.flv': Duration: 00:00:14.7, start: 0.00, bitrate: N/A Stream #0.0: Video: flv, yuv420p, 160x120 [PAR 0:1 DAR 0:1], 50.92 tb(r) Stream #0.1: Audio: nellymoser, 5500 Hz, mono Output #0, mp4, to '2-1195540996717.mp4': Stream #0.0: Video: mpeg4, yuv420p, 160x120 [PAR 0:1 DAR 0:1], q=2-31, 200 kb/s, 15.00 tb(c) Stream #0.1: Audio: libfaac, 16000 Hz, mono, 64 kb/s Stream mapping: Stream #0.0 - #0.0 Stream #0.1 - #0.1 Press [q] to stop encoding frame= 219 fps= 0 q=2.0 Lsize= 325kB time=14.6 bitrate= 182.5kbits/s video:120kB audio:63kB global headers:0kB muxing overhead 77.204278% Input stream 0.1 is found to be 5500Hz sample rate, but that is not correct, according to nelly2pcm: $ nelly2pcm 2-1195540996717.flv audio.pcm mono Nellymoser stream with 16-bit samples at 8kHz Most likely this is a bad flv mux by Red5. Pleasy supply a sample. (No you didn't attach the sample.) MvH Benjamin Larsson -- title: Transcoding FLV - MP4 (et al.): Audio Sample Rate for NellyMoser FLV file is misread - Transcoding FLV - MP4 (et al.): Audio Sample Rate for NellyMoser FLV file is misread __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue270 __
[issue237] Nellymoser conversion results in 0 kB audio stream
Benjamin Larsson [EMAIL PROTECTED] added the comment: Issue fixed in svn. Thanks for the bug report. -- status: open - closed substatus: open - fixed __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue237 __
[issue189] segfault in put_h264_qpel16_mc00_mmx2
Benjamin Larsson [EMAIL PROTECTED] added the comment: heydowns wrote: heydowns [EMAIL PROTECTED] added the comment: The null pointer causing the failure is last_picture.data (or next, presumably). Looks like its seeking to a B-type picture, but doesn't have last/next pictures to work from. last_picture_ptr, next_picture_ptr are non null. Looking at mpeg12 decoder, flush nulls out last and next picture pointers. H264 decoder doesn't (it doesn't use these internall). The attached fixes it here. Since h264 is using mpegvideo's picture management and such, seems reasonable to call mpegvideo's flush as well. Confirmed working on my setup. (Ubuntu amd64). MvH Benjamin Larsson __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue189 __
[issue224] Get rid of av_mallocz_static usage
New submission from Benjamin Larsson [EMAIL PROTECTED]: libavcodec/mpeg12.c libavcodec/utils.c libavcodec/mpegvideo.c Contains code that uses av_mallocz_static, find a way to not use it and remove this depricated function. -- messages: 858 nosy: banan priority: minor status: new substatus: new title: Get rid of av_mallocz_static usage type: feature_request __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue224 __
[issue188] segfault in ff_h264_weight_16x8_mmx2()
Benjamin Larsson [EMAIL PROTECTED] added the comment: Yannick Gingras wrote: New submission from Yannick Gingras [EMAIL PROTECTED]: Mencoder from svn segfaults on a .mkv file. The core contains this stack trace: You have to reproduce it with ffmpeg/ffplay. MvH Benjamin Larsson __ FFmpeg issue tracker [EMAIL PROTECTED] https://roundup.mplayerhq.hu/roundup/ffmpeg/issue188 __
[issue48] Play DVD-Audio files (DD Lossless/MLP decoder)
-> [issue603] high-resolution video partial black shown with ffplay ffmpeg-issues -- Thread -- -- Date -- <!-- google_ad_client = "pub-7266757337600734"; google_alternate_ad_url = "http://www.mail-archive.com/blank.png"; google_ad_width = 160; google_ad_height = 600; google_ad_format = "160x600_as"; google_ad_channel = "8427791634"; google_color_border = "FF"; google_color_bg = "FF"; google_color_link = "006792"; google_color_url = "006792"; google_color_text = "00"; //--> [issue48] Play DVD-Audio files (DD Lossless/MLP decoder) Benjamin Larsson Reply via email to
