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6784 Commits

Author SHA1 Message Date
Baptiste Coudurier
c3024f9e7e In mov demuxer, handle better wrong time scale, fix issue #2528 
Originally committed as revision 26311 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-11 21:06:37 +00:00
Baptiste Coudurier
f6253caf8b In mov demuxer, set r_frame_rate for cfr files 
Originally committed as revision 26310 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-11 19:58:00 +00:00
Daniel Kang
e048a9cab1 Do not crash for illegal sample size, fixes issue 2502. 
Patch by Daniel Kang, daniel.d.kang at gmail

Originally committed as revision 26309 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-11 14:08:45 +00:00
Stefano Sabatini
440d761e40 Clarify timestamps related error messages in compute_pkt_fields2(). 
Originally committed as revision 26308 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-11 12:17:27 +00:00
Peter Ross
1c4ac03530 electronicarts: prevent endless loop opportunity in process_audio_header_elements() 
Fixes issue2529.

Originally committed as revision 26307 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-11 02:25:12 +00:00
Peter Ross
74093bb593 revert r26302 
Originally committed as revision 26305 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-10 22:25:45 +00:00
Peter Ross
e19e051e56 electronicarts: prevent endless loop opportunity in process_audio_header_elements() 
Fixes issue2529.

Originally committed as revision 26302 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-10 12:56:50 +00:00
Peter Ross
42396c2e67 electronicarts: only apply audio sanity checks when audio stream is present 
Originally committed as revision 26301 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-10 12:51:45 +00:00
Daniel Kang
cb77dad724 perform sanity check on sample rate in electronicarts demuxer 
Fixes issue2525
Original patch by Daniel Kang, daniel.d.kang at gmail

Originally committed as revision 26298 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-10 11:38:38 +00:00
Daniel Kang
4da766ce65 perform sanity check on number of channels in electronicarts demuxer 
Fixes issue2514
Original patch by Daniel Kang, daniel.d.kang at gmail

Originally committed as revision 26296 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-10 11:02:07 +00:00
Martin Storsjö
a3b058b7ba rtsp: Properly fail if unable to open an input RTP port 
Originally committed as revision 26285 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-09 10:47:53 +00:00
Peter Ross
5a477e5960 fix indentation 
Originally committed as revision 26278 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-09 02:44:54 +00:00
Peter Ross
866009ea19 wtv: only process timestamp_guid chunks for streams that we know about 
Originally committed as revision 26277 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-09 02:42:16 +00:00
Peter Ross
a5a36a7970 wtv: do not repopulate codec information after we have seen data chunks 
Originally committed as revision 26276 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-09 02:35:06 +00:00
Peter Ross
bf2e54174e wtv: stop processing chunks if length is smaller than chunk header 
Originally committed as revision 26275 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-09 02:21:30 +00:00
Peter Ross
9372f31e03 wtv: fix typo 
Originally committed as revision 26274 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-09 02:13:58 +00:00
Peter Ross
50d83b2005 Add audio codec 0x1602 (AAC LATM) 
Originally committed as revision 26273 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-09 02:11:41 +00:00
Carl Eugen Hoyos
d267b339e4 Lagarith decoder by Nathan Caldwell, saintdev at gmail 
Originally committed as revision 26270 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-08 21:22:15 +00:00
Baptiste Coudurier
a2b7ed3274 In mov muxer, override codec tag for dv in mov, fix remuxing from avi 
Originally committed as revision 26257 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-07 19:56:31 +00:00
Baptiste Coudurier
10d8eac98d In mov muxer, override codec tag for h263 in mov, fix remuxing from 3gp 
Originally committed as revision 26255 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-07 19:55:08 +00:00
Martin Storsjö
a92c30d76e rtsp: Allow requesting of filtering of source packets 
If filtered, only packets from the right source address and port
are received.

To test, play back e.g. some mpeg4 video RTSP stream (where the
video stream is the first stream in the presentation) over UDP.
While receiving this stream, send another stream to the same port:
ffmpeg -re -i <whatever> -vcodec mpeg4 -an -f rtp
rtp://127.0.0.1:5000?localport=1234
Normally, the RTSP playback reports lots of errors at this point.

If the RTSP stream has the ?filter_src option enabled, these
interferring packets are ignored.

Originally committed as revision 26246 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-06 15:22:58 +00:00
Martin Storsjö
21a569f302 udp: Allow specifying the connect option in udp_set_remote_url, too 
If the remote address is updated later with this function, the caller
shouldn't set the connect option until in this call.

Originally committed as revision 26245 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-06 15:16:50 +00:00
Martin Storsjö
babd19ce2e rtpproto: Allow specifying the connect option, passed through to udp 
By calling connect on the UDP socket, only packets from the chosen
peer address and port are received on the socket. This is one
solution to issue 1688.

Originally committed as revision 26244 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-06 15:16:09 +00:00
Carl Eugen Hoyos
504530bfba Set blkalign to 3840 (maximum bytes per frame) for AC-3 in avi. 
Fixes playback for corner-cases like 32kHz 320kb.

Originally committed as revision 26242 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-06 13:30:32 +00:00
Martin Storsjö
79d482b108 rtpdec: Don't set RTP timestamps if they already are set by the depacketizer 
For MS-RTSP, we don't always get RTCP packets (never?), so the earlier
timestamping code never wrote anything into pkt->pts. The rtpdec_asf
depacketizer just sets the dts of the packet, so if the generic RTP
timestamping is used, too, we get inconsistent timestamps.

Therefore, skip the generic RTP timestamp algorithm if the depacketizer
already has set something.

This fixes "Invalid timestamps" warnings, present since SVN rev 26187.

Originally committed as revision 26241 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-06 11:33:06 +00:00
Daniel Kang
6cbce63650 Fix assertion fail on audio files with invalid sample rates, 
fixes issue 2475.

Patch by Daniel Kang, daniel.d.kang at gmail

Originally committed as revision 26240 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-06 03:30:19 +00:00
Stefano Sabatini
6bbdba08c2 Revert previous commit, as it was not meant to be pushed. 
Originally committed as revision 26239 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-06 01:11:57 +00:00
Stefano Sabatini
7820147e6f Issue more explicit error messages in compute_pkt_fields2(). 
Originally committed as revision 26238 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-06 01:01:21 +00:00
Stefano Sabatini
81bd411965 In av_close_input_stream(), flush the packet queue before to actually 
close the stream.

This way the flushed packets can still reference the still unclosed
format context.

In particular this fixes a spurious error issued when closing the
video4linux2 buffer in mmap_release_buffer(), which tries to access
the file descriptor of an already closed file.

Originally committed as revision 26237 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-06 01:01:14 +00:00
Martin Storsjö
29db7c3af4 rtsp: Parse RTP-Info headers 
Originally committed as revision 26236 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-05 21:23:42 +00:00
Martin Storsjö
4cb06874c7 Reindent 
Originally committed as revision 26235 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-05 21:23:12 +00:00
Martin Storsjö
91d96bd3c0 rtsp: Simplify code 
Originally committed as revision 26234 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-05 21:22:50 +00:00
Martin Storsjö
1726813f13 rtsp: Move resetting of rtpdec parameters to before sending the PLAY request 
Originally committed as revision 26233 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-05 21:22:15 +00:00
Baptiste Coudurier
ab04337464 In ogg muxer, correctly mux VFR streams, fix issue #2398 
Originally committed as revision 26229 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-05 19:32:45 +00:00
Baptiste Coudurier
5e2202d6f3 In mov demuxer, check that gmtime returns a valid value, fix crash, issue #2490 
Originally committed as revision 26228 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-05 19:21:04 +00:00
Baptiste Coudurier
4af7166fb4 In mov demuxer, check that stts data exists, fix crash, issue #2479 
Originally committed as revision 26227 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-05 19:14:44 +00:00
Anton Khirnov
14fa75eab4 lavf: rename meta.h->ffmeta.h for consistency. 
Originally committed as revision 26211 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-04 12:35:39 +00:00
Peter Ross
6780f48846 wtv: obtain codec information from stream2_guid chunks, if present 
Originally committed as revision 26208 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-04 11:38:29 +00:00
Peter Ross
17e33f662a wtv: display warning if scrambled stream is detected 
Originally committed as revision 26197 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-02 11:17:34 +00:00
Anssi Hannula
cf99e4aa00 Add AVOption support for muxers. 
Patch by Anssi Hannula, anssi d hannula a iki d fi

Originally committed as revision 26195 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-02 10:52:34 +00:00
Anssi Hannula
febd72be65 Use new function put_nbyte() to speed up padding. 
Patch by Anssi Hannula, anssi d hannula a iki d fi

Originally committed as revision 26194 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-02 10:46:50 +00:00
Anssi Hannula
17ee8f669f Add function put_nbyte() to speed up padding in SPDIF muxer. 
Patch by Anssi Hannula, anssi d hannula a iki d fi

Originally committed as revision 26193 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-02 10:45:07 +00:00
Martin Storsjö
d2995eb910 rtsp: Store the Content-Base header value straight to the target 
This avoids having a large temporary buffer in the struct used for
storing the rtsp reply headers.

Originally committed as revision 26192 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-02 10:11:12 +00:00
Martin Storsjö
77223c5388 rtsp: Pass the method name to ff_rtsp_parse_line 
Originally committed as revision 26191 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-02 10:10:12 +00:00
Martin Storsjö
acc9ed1450 rtsp: Pass RTSPState to ff_rtsp_parse_line, instead of HTTPAuthState 
This allows ff_rtsp_parse_line to do more changes directly in RTSPState
when parsing the reply, instead of having to store large amounts of
temporary data in RTSPMessageHeader.

Originally committed as revision 26190 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-02 10:07:56 +00:00
Martin Storsjö
3df54c6bf2 rtsp: Add a method parameter to ff_rtsp_read_reply 
Originally committed as revision 26189 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-02 10:06:21 +00:00
Daniel Kang
7f8ffc4efd Fix a floating point exception for invalid framerate, fixes issue 2470. 
Patch by Daniel Kang, daniel.d.kang at gmail

Originally committed as revision 26188 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-02 05:01:46 +00:00
Martin Storsjö
3a1cdcc798 rtpdec: Emit timestamps for packets before the first RTCP packet, too 
Emitted timestamps in each stream start from 0, for the first received
RTP packet. Once an RTCP packet is received, that one is used for
sync, emitting timestamps that fit seamlessly into the earlier ones.

Originally committed as revision 26187 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-01 22:27:16 +00:00
Peter Ross
773d892a31 move ff_get_bmp_header under CONFIG_DEMUXERS block 
Originally committed as revision 26182 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2011-01-01 03:24:10 +00:00
Carl Eugen Hoyos
f6bf6e511d Set blkalign to maximum framesize to allow playback on WMP (see issue 2455 and issue 2446). 
Originally committed as revision 26167 to svn://svn.ffmpeg.org/ffmpeg/trunk
 2010-12-30 14:21:14 +00:00