* qatar/master: (21 commits)
build: simplify commands for clean target
swscale: split swscale.c in unscaled and generic conversion routines.
swscale: cosmetics.
swscale: integrate (literally) swscale_template.c in swscale.c.
swscale: split out x86/swscale_template.c from swscale.c.
swscale: enable hScale_altivec_real.
swscale: split out ppc _template.c files from main swscale.c.
swscale: remove indirections in ppc/swscale_template.c.
swscale: split out unscaled altivec YUV converters in their own file.
mpegvideoenc: fix multislice fate tests with threading disabled.
mpegts: Wrap #ifdef DEBUG and av_hex_dump_log() combination in a macro.
build: Simplify texi2html invocation through the --output option.
Mark some variables with av_unused
Replace avcodec_get_pix_fmt_name() by av_get_pix_fmt_name().
svq3: Check negative mb_type to fix potential crash.
svq3: Move svq3-specific fields to their own context.
rawdec: initialize return value to 0.
Remove unused get_psnr() prototype
rawdec: don't leak option strings.
bktr: get default framerate from video standard.
...
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Most of these variables are only used in av_dlog statements, some
are required but not used by other macros.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master: (33 commits)
rtpdec_qdm2: Don't try to parse data packet if no configuration is received
ac3enc: put the counting of stereo rematrixing bits in the same place to make the code easier to understand.
ac3enc: clean up count_frame_bits() and count_frame_bits_fixed()
mpegvideo: make FF_DEBUG_DCT_COEFF output coeffs via av_log() instead of just via AVFrame.
srtdec: make sure we don't write past the end of buffer
wmaenc: improve channel count and bitrate error handling in encode_init()
matroskaenc: make sure we don't produce invalid file with no codec ID
matroskadec: check that pointers were initialized before accessing them
lavf: fix function name in compute_pkt_fields2 av_dlog message
lavf: fix av_find_best_stream when providing a wanted stream.
lavf: fix av_find_best_stream when decoder_ret is given and using a related stream.
ffmpeg: factorize quality calculation
tiff: add support for SamplesPerPixel tag in tiff_decode_tag()
tiff: Prefer enum TiffCompr over int for TiffContext.compr.
mov: Support edit list atom version 1.
configure: Enable libpostproc automatically if GPL code is enabled.
Cosmetics: fix prototypes in oggdec
oggdec: fix memleak with continuous streams.
matroskaenc: add missing new line in av_log() call
dnxhdenc: add AVClass in private context.
...
swscale changes largely rewritten by me or replaced by baptsites due to lots of bugs in ronalds code.
Above code is also just in case its not obvios to a large extended duplicates that where cherry picked
from ffmpeg.
Conflicts:
configure
ffmpeg.c
libavformat/matroskaenc.c
libavutil/pixfmt.h
libswscale/ppc/swscale_template.c
libswscale/swscale.c
libswscale/swscale_template.c
libswscale/utils.c
libswscale/x86/swscale_template.c
tests/fate/h264.mak
tests/ref/lavfi/pixdesc_le
tests/ref/lavfi/pixfmts_copy_le
tests/ref/lavfi/pixfmts_null_le
tests/ref/lavfi/pixfmts_scale_le
tests/ref/lavfi/pixfmts_vflip_le
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Adding _POSIX_C_SOURCE to CPPFLAGS globally produces all sorts of problems
since it causes certain system functions to be hidden on some (BSD) systems.
The solution is to only add the flag on systems that really require it, i.e.
glibc-based ones.
This change makes BSD systems compile out-of-the-box without the need for
adding specific flags manually. It also allows dropping a number of flags
set manually on a file-per-file basis, but were only present to work around
breakage introduced by the presence of _POSIX_C_SOURCE.
Also add _XOPEN_SOURCE to CPPFLAGS for glibc systems. We use XSI extensions
in several places already, so it is preferable to define it globally instead
of littering source files with individual #defines only needed for glibc.
* qatar/master: (22 commits)
ac3enc: move extract_exponents inner loop to ac3dsp
avio: deprecate url_get_filename().
avio: deprecate url_max_packet_size().
avio: make url_get_file_handle() internal.
avio: make url_filesize() internal.
avio: make url_close() internal.
avio: make url_seek() internal.
avio: cosmetics, move AVSEEK_SIZE/FORCE declarations together
avio: make url_write() internal.
avio: make url_read_complete() internal.
avio: make url_read() internal.
avio: make url_open() internal.
avio: make url_connect internal.
avio: make url_alloc internal.
applehttp: Merge two for loops
applehttp: Restructure the demuxer to use a custom AVIOContext
applehttp: Move finished and target_duration to the variant struct
aacenc: reduce the number of loop index variables
avio: deprecate url_open_protocol
avio: deprecate url_poll and URLPollEntry
...
Conflicts:
libavformat/applehttp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: fix partial run when no samples path is specified
ARM: NEON fixed-point forward MDCT
ARM: NEON fixed-point FFT
lavf: bump minor version and add an APIChanges entry for avio changes
avio: simplify url_open_dyn_buf_internal by using avio_alloc_context()
avio: make url_fdopen internal.
avio: make url_open_dyn_packet_buf internal.
avio: avio_ prefix for url_close_dyn_buf
avio: avio_ prefix for url_open_dyn_buf
avio: introduce an AVIOContext.seekable field
ac3enc: use generic fixed-point mdct
lavfi: add fade filter
Change yadif to not use out of picture lines.
lavc: deprecate AVCodecContext.antialias_algo
lavc: mark mb_qmin/mb_qmax for removal on next major bump.
Conflicts:
doc/filters.texi
libavcodec/ac3enc_fixed.h
libavcodec/ac3enc_float.h
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/vf_fade.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It doesn't look fit to be a part of the public API.
Adding a temporary hack to ffserver to be able to use it, should be
cleaned up when somebody is up for it.
In the name of consistency:
put_byte -> avio_w8
put_<type> -> avio_w<type>
put_buffer -> avio_write
put_nbyte will be made private
put_tag will be merged with avio_put_str
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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
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
This also reverts SVN rev 26016, which incorrectly overwrote the time base
with 90 kHz for all streams, regardless of what was set by the SDP parsing.
The stream that triggered the fix in 26016 still works after this commit.
Originally committed as revision 26022 to svn://svn.ffmpeg.org/ffmpeg/trunk
The generic default is 0/0 and that obviously triggers once the value is used.
Originally committed as revision 26016 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes cases where the RTP time base and the sample rate of the stream
differ. Previously, the AVStream time_base was unconditionally set to
the sample rate (which initially was set to one value when parsing the
rtpmap field in the SDP, but later overridden by an a=SampleRate field).
Additionally, this makes the code actually use the stream time base set
in rtpmap for video codecs, instead of hardcoding it to always be 90 kHz.
Originally committed as revision 25908 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes playback of a RealRTSP/MP3 URL from the RTSP samples on
MultimediaWiki.
Originally committed as revision 25906 to svn://svn.ffmpeg.org/ffmpeg/trunk
This indicates that there was no error that needs to be reported to the
caller, so we can move on to parse the next packet immediately, if
available. The only error code that ff_mpegts_parse_packet can return
indicates that there was no packet to return from the provided data, for
which it returns -1.
Originally committed as revision 25496 to svn://svn.ffmpeg.org/ffmpeg/trunk
It may have returned a negative number for an error (e.g. AVERROR(EAGAIN),
if more data is required for it to be able to return a complete packet).
Originally committed as revision 25458 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes roundup issue 2270.
Patch by Robert Schlabbach, robert_s at gmx dot net
Originally committed as revision 25372 to svn://svn.ffmpeg.org/ffmpeg/trunk
Do the same change for ff_rdt_parse_packet, too, to keep the interfaces
similar.
Originally committed as revision 25289 to svn://svn.ffmpeg.org/ffmpeg/trunk
Patch by Josh Allmann, joshua dot allmann at gmail, original code
by Ronald S Bultje.
Originally committed as revision 24236 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows very large value strings, needed for xiph extradata.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23859 to svn://svn.ffmpeg.org/ffmpeg/trunk
This will be used for cleaning up code that is common among RTP depacketizers.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23847 to svn://svn.ffmpeg.org/ffmpeg/trunk
If these aren't reset, the timestamps make a huge jump when the next RTCP
is received.
Originally committed as revision 22918 to svn://svn.ffmpeg.org/ffmpeg/trunk
In order to sync RTP streams that get their initial RTCP timestamp at
different times, propagate the NTP timestamp of the first RTCP packet
to all other streams.
This makes the timestamps of returned packets start at (near) zero instead
of at any random offset.
Originally committed as revision 22917 to svn://svn.ffmpeg.org/ffmpeg/trunk
the Vorbis / theora depacketizers.
Patch by Josh Allmann <joshua DOT allmann AT gmail DOT com>.
Originally committed as revision 22765 to svn://svn.ffmpeg.org/ffmpeg/trunk
but doesn't actually do that. What's worse, it creates timestamp adjustments
that are different per stream within a session, leading to a/v sync issues.
See discussion in thread "[FFmpeg-devel] rtp streaming x264+audio issues (and
some ideas to fix them)". Patch suggested by Luca Abeni <lucabe72 email it>.
Originally committed as revision 21857 to svn://svn.ffmpeg.org/ffmpeg/trunk
what e.g. RealPlayer does. This allows proper port forwarding setup in NAT-
based environments.
Patch by Martin Storsjö <$firstname at $firstname dot st>.
Originally committed as revision 21856 to svn://svn.ffmpeg.org/ffmpeg/trunk
qualification task, see "RTP/Vorbis payload implementation (GSoC qual
task)" thread on mailinglist.
Originally committed as revision 18509 to svn://svn.ffmpeg.org/ffmpeg/trunk
SDP line handler that parses the streamID in the SDP so that ASF stream
data can be matched to their respective streams in the RTSP demuxer. See
"[PATCH] RTSP-MS 12/15: ASF payload support" thread on mailinglist.
Originally committed as revision 18061 to svn://svn.ffmpeg.org/ffmpeg/trunk
payload handlers take care of that themselves at their own option. What this
patch really does is "fix" a bug in MS-RTSP protocol where incoming packets
are always coming in over the connection (UDP) or interleave-id (TCP) of
the stream-id of the first ASF packet in the RTP packet. However, RTP packets
may contain multiple ASF packets (and usually do, from what I can see), and
therefore this leads to playback bugs. The intended stream-id per ASF packet
is given in the respective ASF packet header. The ASF demuxer will correctly
read this and set pkt->stream_index, but since the "stream" parameter can
not be known to rtpdec.c or any of the RTP/RTSP code, the "st" parameter
in all these functions is basically invalid. Therefore, using st->id as
pkt->stream_index leads to various playback bugs. The result of this patch
is that pkt->stream_index is left untouched for RTP/ASF (and possibly for
other payloads that have similar behaviour).
The patch was discussed in the "[PATCH] rtpdec.c: don't overwrite
pkt->stream_index in finalize_packet()" thread on the mailinglist.
Originally committed as revision 17767 to svn://svn.ffmpeg.org/ffmpeg/trunk
called for all packets with an internal handler function but only for
non-first packets from dynamic payload parse_packet() handlers. This patch
fixes that. Bug was noticed by Luca in "[PATCH] rtpdec.c: don't overwrite
pkt->stream_index in finalize_packet()" thread.
Originally committed as revision 17764 to svn://svn.ffmpeg.org/ffmpeg/trunk
under review. See "[FFmpeg-devel] RTP mark bit not passed to parse_packet"
thread on mailinglist.
Originally committed as revision 17616 to svn://svn.ffmpeg.org/ffmpeg/trunk
to the parse_packet() function pointer in RTPDynamicProtocolHandlers. This
allows these functions to peek back and retrieve values from the demuxer's
context (or RTSPState). The ASF/RTP payload parser will use this to be able
to parse SDP values (which occur even before the payload ID is given), store
them in the RTSPState and then retrieve them while parsing payload data. See
"[PATCH] RTSP-MS 13/15: add RTSP demuxer AVFormatContext to parse_packet()
function pointer (was: transport context)" mailinglist thread.
Originally committed as revision 17015 to svn://svn.ffmpeg.org/ffmpeg/trunk
(and thus preparing for the introduction of RDTDemuxContext) in a next patch.
See discussion in "RDT/Realmedia patches #2" thread on ML.
Originally committed as revision 15542 to svn://svn.ffmpeg.org/ffmpeg/trunk
not use RTPDemuxContext, but rather take a pointer to the payload context
directly. This allows using payload handlers regardless over the transport
over which they were sent, and prepares for the introduction of a future
RDTDemuxContext. See discussion in "RDT/Realmedia patches #2" thread on ML.
Originally committed as revision 15541 to svn://svn.ffmpeg.org/ffmpeg/trunk
The feature_tests.h header from Sun systems (Solaris/OpenSolaris) will abort
the build if _XOPEN_SOURCE is defined to 500, and C99 is requested (as well
as POSIX.1-2001), and will only accept it to be defined to 600.
inspired by a patch from Diego Pettenò, flameeyes gmail com
Originally committed as revision 15460 to svn://svn.ffmpeg.org/ffmpeg/trunk
it so that I can use it in rdt.c as well. See discussion in "Realmedia patch"
thread on ML.
Originally committed as revision 15233 to svn://svn.ffmpeg.org/ffmpeg/trunk
(correctly compute the presentation times based on the RTP timestamps
and the RTCP SR packets)
Originally committed as revision 14045 to svn://svn.ffmpeg.org/ffmpeg/trunk
This can be used later by RDT to get the flags from the RTP packet and
use that for the RealMedia packet (such as whether this RTP packet
represents a keyframe or not). For discussion, see "[PATCH] Realmedia
/ RTSP (RDT)".
Originally committed as revision 11557 to svn://svn.ffmpeg.org/ffmpeg/trunk
