[issue2127] compilation with --disable-optimizations fails on x86_32 for transpose4x4()

2011-01-03 Thread Carl Eugen Hoyos

Carl Eugen Hoyos ceho...@rainbow.studorg.tuwien.ac.at added the comment:

Fixed by Daniel Kang in r26204.

--
status: open - closed
substatus: open - fixed
topic: +avcodec


FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/issue2127



[issue2479] ffmpeg crashes on mp4 file with invalid headers

2011-01-03 Thread Carl Eugen Hoyos

Carl Eugen Hoyos ceho...@rainbow.studorg.tuwien.ac.at added the comment:

Sample moved to /samples/ffmpeg-bugs/roundup/issue2479.

--
priority: normal - important
substatus: open - reproduced


FFmpeg issue tracker iss...@roundup.ffmpeg.org
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[issue2481] Support AAC LTP

2011-01-03 Thread Carl Eugen Hoyos

New submission from Carl Eugen Hoyos ceho...@rainbow.studorg.tuwien.ac.at:

The sample from issue 963 contains two AAC LTP streams that libavcodec cannot
decode, but faad (and mplayer with the original sample and -ac faad) can.
See also http://wiki.multimedia.cx/index.php?title=AAC#MPEG-4_AAC_LTP
I will upload the results of mplayer -demuxer lavf -aid 16/17 -dumpaudio to
samples/ffmpeg-bugs/roundup/issue. They can be decoded fine with faad (the
executable), but not mplayer or ffmpeg-0.5 with libfaad.

See stream 0.17 and 0.18:
$ ffmpeg -i \[CCCP\]_Mega_Audio_Test.mkv
FFmpeg version SVN-r26202, Copyright (c) 2000-2011 the FFmpeg developers
  built on Jan  3 2011 12:49:22 with gcc 4.4.5
  configuration: --disable-optimizations --cc='/usr/local/gcc-4.4.5/bin/gcc 
-m32'
  libavutil 50.36. 0 / 50.36. 0
  libavcore  0.16. 0 /  0.16. 0
  libavcodec52.101. 0 / 52.101. 0
  libavformat   52.92. 0 / 52.92. 0
  libavdevice   52. 2. 2 / 52. 2. 2
  libavfilter1.72. 0 /  1.72. 0
  libswscale 0.12. 0 /  0.12. 0
[aac @ 0x8b691d0] Audio object type 4 is not supported.
[aac @ 0x8b6afa0] Audio object type 4 is not supported.
[aac @ 0x8b79cc0] Audio object type 4 is not supported.
[aac @ 0x8b7bae0] Audio object type 4 is not supported.
[mpeg4 @ 0x8b3ddd0] Invalid and inefficient vfw-avi packed B frames detected
[aac @ 0x8b691d0] Audio object type 4 is not supported.
Last message repeated 7 times
[aac @ 0x8b6afa0] Audio object type 4 is not supported.
Last message repeated 7 times
...
[aac @ 0x8b79cc0] Audio object type 4 is not supported.
Last message repeated 7 times
[aac @ 0x8b7bae0] Audio object type 4 is not supported.
Last message repeated 7 times
[matroska,webm @ 0x8b3a470] Estimating duration from bitrate, this may be 
inaccurate
Input #0, matroska,webm, from '[CCCP]_Mega_Audio_Test.mkv':
  Metadata:
title   : Macross Plus - Isamu's Preflight Check
  Duration: 00:01:12.10, start: 0.00, bitrate: 6208 kb/s
Stream #0.0(eng): Video: mpeg4, yuv420p, 656x464 [PAR 1:1 DAR 41:29], PAR
116:123 DAR 4:3, 23.98 fps, 23.98 tbr, 1k tbn, 23.98 tbc
Metadata:
  title   : Macross Plus - Isamu's Preflight Check
Stream #0.1(eng): Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s
Metadata:
  title   : AC3 6ch Audio
Stream #0.2(jpn): Audio: ac3, 48000 Hz, stereo, s16, 192 kb/s
Metadata:
  title   : AC3 2ch Audio
Stream #0.3(eng): Audio: dca, 48000 Hz, 5.1, s16, 1536 kb/s
Metadata:
  title   : DTS 1536kbps 6ch Audio
Stream #0.4(jpn): Audio: dca, 48000 Hz, 5.0, s16, 1536 kb/s
Metadata:
  title   : DTS 1536kbps 2ch Audio
Stream #0.5(eng): Audio: dca, 48000 Hz, 5.1, s16, 768 kb/s
Metadata:
  title   : DTS 768kbps 6ch Audio
Stream #0.6(jpn): Audio: dca, 48000 Hz, 5.0, s16, 768 kb/s
Metadata:
  title   : DTS 768kbps 2ch Audio
Stream #0.7(jpn): Audio: mp1, 48000 Hz, 2 channels, s16, 192 kb/s
Metadata:
  title   : MPEG-1 Layer 1 2ch Audio
Stream #0.8(jpn): Audio: mp2, 48000 Hz, 2 channels, s16, 128 kb/s
Metadata:
  title   : MPEG-1 Layer 2 2ch Audio
Stream #0.9(eng): Audio: mp3, 48000 Hz, 2 channels, s16, 192 kb/s
Metadata:
  title   : MPEG-1 Layer 3 6ch Audio
Stream #0.10(jpn): Audio: mp3, 48000 Hz, 2 channels, s16, 128 kb/s
Metadata:
  title   : MPEG-1 Layer 3 2ch Audio
Stream #0.11(jpn): Audio: mp1, 24000 Hz, 2 channels, s16, 128 kb/s
Metadata:
  title   : MPEG-2 LSF Layer 1 2ch Audio
Stream #0.12(jpn): Audio: mp2, 24000 Hz, 2 channels, s16, 96 kb/s
Metadata:
  title   : MPEG-2 LSF Layer 2 2ch Audio
Stream #0.13(jpn): Audio: mp3, 24000 Hz, 2 channels, s16, 64 kb/s
Metadata:
  title   : MPEG-2 LSF Layer 3 2ch Audio
Stream #0.14(jpn): Audio: mp3, 12000 Hz, 2 channels, s16, 32 kb/s
Metadata:
  title   : MPEG-2.5 Layer 3 2ch Audio
Stream #0.15(eng): Audio: aac, 48000 Hz, 5.1, s16
Metadata:
  title   : MPEG-2 AAC LC 6ch Audio
Stream #0.16(jpn): Audio: aac, 48000 Hz, stereo, s16
Metadata:
  title   : MPEG-2 AAC LC 2ch Audio
Stream #0.17(eng): Audio: aac, 0 channels
Metadata:
  title   : MPEG-2 AAC LTP 6ch Audio
Stream #0.18(jpn): Audio: aac, 0 channels
Metadata:
  title   : MPEG-2 AAC LTP 2ch Audio
Stream #0.19(eng): Audio: aac, 48000 Hz, 5.1, s16
Metadata:
  title   : MPEG-2 AAC Main 6ch Audio
Stream #0.20(jpn): Audio: aac, 48000 Hz, stereo, s16
Metadata:
  title   : MPEG-2 AAC Main 2ch Audio
Stream #0.21(eng): Audio: aac, 48000 Hz, 5.1, s16
Metadata:
  title   : MPEG-4 AAC LC 6ch Audio
Stream #0.22(jpn): Audio: aac, 48000 Hz, stereo, s16
Metadata:
  title   : MPEG-4 AAC LC 2ch Audio
Stream #0.23(eng): Audio: aac, 48000 Hz, 5.1, s16
Metadata:
  

[issue2481] Support AAC LTP

2011-01-03 Thread Carl Eugen Hoyos

Carl Eugen Hoyos ceho...@rainbow.studorg.tuwien.ac.at added the comment:

Fix priority, LTP2.aac and LTP6.aac uploaded to
samples/ffmpeg-bugs/roundup/issue2481.

--
priority: normal - wish


FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/issue2481



[issue2375] SageTV violates the LGPL

2011-01-03 Thread Carl Eugen Hoyos

Carl Eugen Hoyos ceho...@rainbow.studorg.tuwien.ac.at added the comment:

SageTV was informed about the (L)GPL conformance problems, but was not willing
to collaborate, so I added them to shame.


FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/issue2375



[issue2482] AVFrame.key_frame not set correctly for AVCHD files from Panasonic GH1

2011-01-03 Thread Robert Krüger

New submission from Robert Krüger krue...@signal7.de:

When decoding the file gh1-720p-50-6-seconds.mts, uploaded for issue 2437, the
field key_frame of the decoded AVFrames is only set to 1 for the first frame and
then never again, although the file contains several frames which are decoded
from only one I slice.

For the file gh1-1080p-25-psf-6-seconds.mts it also never reports an I frame
after the first one, which is probably not correct either but there it is not
that obvious.

--
messages: 13185
priority: normal
status: new
substatus: new
title: AVFrame.key_frame not set correctly for AVCHD files from Panasonic GH1
type: bug


FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/issue2482



[issue2478] MMSH shows full metadata, RTSP does not

2011-01-03 Thread Ronald S. Bultje

Ronald S. Bultje rsbul...@gmail.com added the comment:

Input #0, rtsp, from 'rtsp://djxmmx.net/Rap-RnB':
  Metadata:
genre   : Rap
WMFSDKVersion   : 11.0.6002.18049
WMFSDKNeeded: 0.0.0.
IsVBR   : 0
album   : Double Trouble
track   : 4
WM/Year : 2007
title   : U Aint Gotta Ask
artist  : Al Fatz  Chip Tha Ripper
copyright   : 
comment : 
  Duration: 00:03:06.64, start: 123.965000, bitrate: 128 kb/s
Stream #0.0: Audio: wmapro, 44100 Hz, stereo, flt, 128 kb/s
Stream #0.1: Data: [0][0][0][0] / 0x
 127.58 A-V:  0.000 s:0.0 aq=   94KB vq=0KB sq=0B f=0/0


FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/issue2478
Index: libavformat/rtpdec_asf.c
===
--- libavformat/rtpdec_asf.c(revision 26130)
+++ libavformat/rtpdec_asf.c(working copy)
@@ -96,6 +96,7 @@
 int ret = 0;
 if (av_strstart(p, pgmpu:data:application/vnd.ms.wms-hdr.asfv1;base64,, 
p)) {
 ByteIOContext pb;
+AVMetadataTag *t;
 RTSPState *rt = s-priv_data;
 int len = strlen(p) * 6 / 8;
 char *buf = av_mallocz(len);
@@ -112,6 +113,9 @@
 ret = av_open_input_stream(rt-asf_ctx, pb, , asf_demuxer, NULL);
 if (ret  0)
 return ret;
+while ((t = av_metadata_get(rt-asf_ctx-metadata, , t,
+AV_METADATA_IGNORE_SUFFIX)))
+av_metadata_set2(s-metadata, t-key, t-value, 0);
 rt-asf_pb_pos = url_ftell(pb);
 av_free(buf);
 rt-asf_ctx-pb = NULL;


[issue2478] MMSH shows full metadata, RTSP does not

2011-01-03 Thread Ronald S. Bultje

Ronald S. Bultje rsbul...@gmail.com added the comment:

does the metadata update in this radio station in WMP, btw? If so, we 
should try supporting that also...


FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/issue2478



[issue2478] MMSH shows full metadata, RTSP does not

2011-01-03 Thread Ronald S. Bultje

Ronald S. Bultje rsbul...@gmail.com added the comment:

At the end of each song, the server sends us this:

line='SET_PARAMETER rtsp://djxmmx.net:554/Rap-RnB RTSP/1.0'=0/0   
line='Content-Type: application/x-wms-extension-cmd'
line='X-Notice: 2101 End-of-Stream Reached'
line='RTP-Info: url=rtsp://djxmmx.net:554/Rap-RnB/audio;Seq=58258'
line='X-Playlist-Gen-Id: 81157'
line='Content-Length: 71'
line='Date: Mon, 03 Jan 2011 19:07:30 GMT'
line='CSeq: 1'
line='User-Agent: WMServer/9.5.6001.18281'
line=''

Apparently we have to open a new stream then, which we don't do - so we 
get silence. Would be interesting to support...


FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/issue2478



[issue2483] Mix Multiple Audio Inputs

2011-01-03 Thread HoboZero

New submission from HoboZero amial...@hotmail.com:

Based on dozens of posts on the subject (for example 
http://lists.mplayerhq.hu/pipermail/ffmpeg-user/2009-March/019580.html), 
it is not currently possible to mix down multiple audio tracks into a 
single audio track, say wav or mp3.  All of the forum entries on the 
subject either included a video track and multiple audio tracks or ended 
in the poster abandoning ffmpeg to mix pure audio.  

When multiple inputs are used, the result is garbled sound:
ffmpeg -y -i 1.wav -i 2.wav  -ac 1 output.mp3 -newaudio

If this is possible, it is not clear how to do so from the 
documentation. 

Thank you.

--
messages: 13189
priority: normal
status: new
substatus: new
title: Mix Multiple Audio Inputs
type: feature_request


FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/issue2483



[issue2483] Mix Multiple Audio Inputs

2011-01-03 Thread Carl Eugen Hoyos

Carl Eugen Hoyos ceho...@rainbow.studorg.tuwien.ac.at added the comment:

Sorry for my ignorance, but what should output.mp3 contain in your example?
(And what effect should -newaudio have? And why is -ac 1 needed?)

--
status: new - open
substatus: new - needs_more_info


FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/issue2483



[issue2471] ffmpeg crashes on mkv files with invalid time stamps

2011-01-03 Thread Aurelien Jacobs

Aurelien Jacobs au...@gnuage.org added the comment:

fixed in r26207.

--
status: open - closed
substatus: reproduced - fixed


FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/issue2471



[issue2484] WMS Compatibility Suggestions

2011-01-03 Thread Dennis

New submission from Dennis dfrati...@hotmail.com:

I purpose a few changes to the way FFmpeg handles WMS 
content. I purpose these for transparent 
compatibility with the Microsoft way, as I 
understand it. Your comments and thoughts are 
encouraged.

First, there should be a rollover protocol called: 
MMS (ex., mms://domain.tld/PubPoint). This rollover 
protocol should try to access the PubPoint in this 
order: RTSP-UDP -- RTSP-TCP -- MMSH (HTTP, see #2). 
I did not include pure MMS in the failover protocol 
scheme because Microsoft has depreciated it in Server 
2003 and it is not even present in Server 2008. I 
think leaving it in MMST is probaly best.

Second, the MMSH protocol should either be included 
in the regular HTTP protocol or there should be some 
sort of failover in HTTP to MMSH (perhaps auto 
detection or hook). I suggest this because Microsoft 
says to write WMS over HTTP URL's as: 
http://domain.tld:port/PubPoint when making an ASX or 
accessing the stream in Windows media player. keeping 
just http would help with compatibility.

Last, I don't believe FFmpeg fully supports ASX 
files. When I try my ASX generated with WMS 2008:
ffmpeg -i http://www.djxmmx.net/content/wms/Rap-
RnB.asx
FFmpeg version SVN-r26188, Copyright (c) 2000-2011 
the FFmpeg developers
  built on Jan  2 2011 04:10:51 with gcc 4.4.2
  configuration: --enable-gpl --enable-version3 --
enable-libgsm --enable-libvorb
is --enable-libtheora --enable-libspeex --enable-
libmp3lame --enable-libopenjpeg
 --enable-libschroedinger --enable-libopencore_amrwb -
-enable-libopencore_amrnb
--enable-libvpx --disable-decoder=libvpx --arch=x86 --
enable-runtime-cpudetect -
-enable-libxvid --enable-libx264 --enable-librtmp --
extra-libs='-lrtmp -lpolarss
l -lws2_32 -lwinmm' --target-os=mingw32 --enable-
avisynth --enable-w32threads --
cross-prefix=i686-mingw32- --cc='ccache i686-mingw32-
gcc' --enable-memalign-hack

  libavutil 50.36. 0 / 50.36. 0
  libavcore  0.16. 0 /  0.16. 0
  libavcodec52.101. 0 / 52.101. 0
  libavformat   52.91. 0 / 52.91. 0
  libavdevice   52. 2. 2 / 52. 2. 2
  libavfilter1.72. 0 /  1.72. 0
  libswscale 0.12. 0 /  0.12. 0
http://www.djxmmx.net/content/wms/Rap-RnB.asx: 
Invalid data found when processin
g input

The ASX is also suppose to merge the metadata in the 
stream. For example, if there is no copyright present 
in the stream metadata and I add the 
Copyright/Copyright tag in the ASX, it should 
display the copyright in the end content.

That's pretty much all I can think of right now.

Once again, these are only suggestions to help make 
FFmpeg the most complete and compliant. Let me know 
what you think. Thanks.

--
messages: 13192
priority: normal
status: new
substatus: new
title: WMS Compatibility Suggestions
type: feature_request


FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/issue2484



[issue2475] ffmpeg fails assertion on audio files with invalid sample rates

2011-01-03 Thread Daniel Kang

Daniel Kang daniel.d.k...@gmail.com added the comment:

I have uploaded a third sample to /MPlayer/incoming/wav_assertion_fail2. The
patch fixes it.


FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/issue2475