* qatar/master:
vc1: use an enum for Frame Coding Mode
doc: cleanup filter section
indeo3: error out if no motion vector is set.
x86inc: Flag shufps as an floating-point instruction for the AVX emulation code.
mpegaudio: do not use init_static_data() for initializing tables.
musepack: fix signed shift overflow in mpc_read_packet()
mov: Make format string match variable type.
wmavoice: Make format string match variable type.
vc1: select interlaced scan table by FCM element
Generalize RIFF INFO tag support; support reading INFO tag in wav
pthread: track thread existence in a separate variable.
Conflicts:
doc/filters.texi
libavcodec/pthread.c
libavformat/avi.c
libavformat/riff.c
libavformat/riff.h
libavformat/wav.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Using an unsigned variable avoids problems with overflows.
There is further no need for a 64-bit intermediate here.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master: (42 commits)
swscale: fix signed overflow in yuv2mono_X_c_template
snow: fix integer overflows
svq1enc: remove stale altivec-related hack
snow: fix signed overflow in byte to 32-bit replication
adx: rename ff_adx_decode_header() to avpriv_adx_decode_header()
avformat: add CRI ADX format demuxer
adx: add an ADX parser.
adx: move header decoding to ADX common code
adx: calculate the number of blocks in a packet
adx: define and use 2 new macro constants BLOCK_SIZE and BLOCK_SAMPLES
adx: check for unsupported ADX formats
adx: simplify encoding by using put_sbits()
adx: calculate correct LPC coeffs
adx: use 12-bit coefficients instead of 14-bit to avoid integer overflow
adx: simplify adx_decode() by using get_sbits() to read residual samples
adx: fix the data offset parsing in adx_decode_header()
adx: remove unneeded post-decode channel interleaving
adx: validate header values
adx: cosmetics: general pretty-printing and comment clean-up
adx: remove useless comments
...
Conflicts:
Changelog
libavcodec/cook.c
libavcodec/fraps.c
libavcodec/nuv.c
libavcodec/pthread.c
libavcodec/version.h
libavformat/Makefile
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This simplifies the decoder so it doesn't have to process an in-packet header
or handle arbitrary-sized packets. It also fixes decoding of files with large
headers.
Also reduce verbosity for the unsupported stream message, use
an AVFormatContext for av_log and and print the tag of the
unknown stream.
Improves ticket #672.
* qatar/master:
libavutil: add utility functions to simplify allocation of audio buffers.
libavutil: add planar sample formats and av_sample_fmt_is_planar()
avconv: fix segfault at EOF with delayed pictures
pcmdec: remove unneeded resetting of samples pointer
avconv: remove a now unused parameter from output_packet().
avconv: formatting fixes in output_packet()
avconv: declare some variables in blocks where they are used
avconv: use the same behavior when decoding audio/video/subs
bethsoftvideo: return proper consumed size for palette packets.
cdg: skip packets that don't contain a cdg command.
crcenc: add flags
avconv: use vsync 0 for AVFMT_NOTIMESTAMPS formats.
tiffenc: add a private option for selecting compression algorithm
md5enc: add flags
ARM: remove needless .text/.align directives
Conflicts:
doc/APIchanges
libavcodec/tiffenc.c
libavutil/avutil.h
libavutil/samplefmt.c
libavutil/samplefmt.h
tests/ref/fate/bethsoft-vid
tests/ref/fate/cdgraphics
tests/ref/fate/film-cvid-pcm-stereo-8bit
tests/ref/fate/mpeg2-field-enc
tests/ref/fate/nuv
tests/ref/fate/tiertex-seq
tests/ref/fate/tscc-32bit
tests/ref/fate/vmnc-32bit
Merged-by: Michael Niedermayer <michaelni@gmx.at>
AVFMT_NOTIMESTAMPS for crc, as it ignores the timestamps.
AVFMT_VARIABLE_FPS for framecrc, as it prints dts.
Many FATE changes, because avconv is no longer duplicating frames in
those tests.
Also added -vsync 0 for some tests to prevent avconv from dropping
frames until it can be fixed more properly.
AVFMT_NOTIMESTAMPS for md5, as it ignores the timestamps.
AVFMT_VARIABLE_FPS for framemd5, as it prints dts.
-vsync 0 for the vp8 test is needed because with vsync 2 the timestamp
guessing code gets confused by an altref frame that is never displayed
and drops a frame later.
* qatar/master: (22 commits)
aacdec: Fix PS in ADTS.
avconv: Consistently use PIX_FMT_NONE.
dsputil: use cpuflags in x86 emu_edge_core
dsputil: use movups instead of movdqu in ff_emu_edge_core_sse()
wma: initialize prev_block_len_bits, next_block_len_bits, and block_len_bits.
mov: Remove some redundant and obsolete comments.
Add libavutil/mathematics.h #includes for INFINITY
doxy: structure libavformat groups
doxy: introduce an empty structure in libavcodec
doxy: provide a start page and document libavutil
doxy: cleanup pixfmt.h
regtest: split video encode/decode tests into individual targets
ARM: add explicit .arch and .fpu directives to asm.S
pthread: do not touch has_b_frames
avconv: cleanup the transcoding loop in output_packet().
avconv: split subtitle transcoding out of output_packet().
avconv: split video transcoding out of output_packet().
avconv: split audio transcoding out of output_packet().
avconv: reindent.
avconv: move streamcopy-only code out of decoding loop.
...
Conflicts:
avconv.c
libavcodec/aaccoder.c
libavcodec/pthread.c
libavcodec/version.h
libavutil/audioconvert.h
libavutil/avutil.h
libavutil/mem.h
tests/ref/vsynth1/dv
tests/ref/vsynth1/mpeg2thread
tests/ref/vsynth2/dv
tests/ref/vsynth2/mpeg2thread
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Adding the thread count in frame level multithreading to has_b_frames
as an additional delay causes more problems than it solves.
For example inconsistent behaviour during timestamp calculation in
libavformat.
Thread count and frame level multithreading are both set by the user.
If the additional delay caused by frame level multithreading needs
to be considered in the calling code it has all information to take
it into account.
Should it become necessary to calculate a maximum delay inside
libavcodec it should be exported as its own field and not reusing
an existing field.
Based on a patch by Michael Niedermayer.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Some sample IFF ACBM files can be found here:
http://aminet.net/package/dev/basic/ABdemos
Thanks to Peter Ross for his help with this patch.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* shariman/wmall: (24 commits)
Clean-up
dump_int_buffer() to dump samples from a buffer
Implement revert_cdlms()
Doxy for reset_codec()
Store transient state and position of transient area
Implement use_high_update_speed() and use_normal_update_speed()
Initialize num_logged_tiles and remove unnecessary codes
Log index for each line of output
Log tile size
Output decoded residues
Replace placeholders with actual calls to clear_codec_buffers() and reset_codec()
Implement lms_update()
Implement lms_predict()
Implement reset_codec()
Add missing syntax elements to WmallDecodeCtx
Add .recent syntax element to cdlms struct
Implement clear_codec_buffers()
Add buffers to context necessary for reverting cdmls and mclms filter
Use avpriv_copy_bits() instead of ff_copy_bits()
Cosmetics
...
Conflicts:
libavcodec/wmalosslessdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
configure: add check for w32threads to enable it automatically
rtmp: do not hardcode invoke numbers
cinepack: return non-generic errors
fate-lavf-ts: use -mpegts_transport_stream_id option.
Add an APIchanges entry and a minor bump for avio changes.
avio: Mark the old interrupt callback mechanism as deprecated
avplay: Set the new interrupt callback
avconv: Set new interrupt callbacks for all AVFormatContexts, use avio_open2() everywhere
cinepak: remove redundant coordinate checks
cinepak: check strip_size
cinepak, simplify, use AV_RB24()
cinepak: simplify, use FFMIN()
cinepak: Fix division by zero, ask for sample if encoded_buf_size is 0
applehttp: Fix seeking in streams not starting at DTS=0
http: Don't use the normal http proxy mechanism for https
tls: Handle connection via a http proxy
http: Reorder two code blocks
http: Add a new protocol for opening connections via http proxies
http: Split out the non-chunked buffer reading part from http_read
segafilm: add support for raw videos
...
Conflicts:
avconv.c
configure
doc/APIchanges
libavcodec/cinepak.c
libavformat/applehttp.c
libavformat/version.h
tests/lavf-regression.sh
tests/ref/lavf/ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Note: FCPublish/FCUnpublish are adobe server specific and not described
in the rtmp specification. Some servers might not cope with them at
all.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The Apple HTTP Live Streaming demuxer's implementation of
seeking searches for the MPEG TS segment which contains the
requested timestamp. In its current implementation it assumes
that the first segment will start from 0.
But, MPEG TS streams do not necessarily start with timestamp
(near) 0, causing seeking to fail for those streams.
This also occurs when using live streaming of HTTP Live Streams.
In this case sliding playlists may be used, which means that in
that case only the last x encoded segments are stored, the earlier
segments get deleted from disk and removed from the playlist.
Because of this, when starting playback of a stream in the middle
of such a broadcast, the initial segment fetched after parsing
the m3u8 playlist will not start from timestamp (near) 0, causing
(the admittedly limited live) seeking to fail.
This patch changes this demuxers seeking implementation to use
the initial DTS as an offset for searching the segments containing
the requested timestamp.
Signed-off-by: Martin Storsjö <martin@martin.st>
The tls protocol handles connections via proxies internally.
With TLS/SSL, the peer verification requires that the client
speaks directly with the server, since the proxy doesn't have
the remote server's private key.
Signed-off-by: Martin Storsjö <martin@martin.st>
This opens a plain TCP connection through the proxy via the
CONNECT HTTP method. Normally, this is allowed for connections
on port 443, but can in general be used to allow connections
to any port (depending on proxy configuration), and could thus
be used to tunnel any TCP connection via a HTTP proxy.
Signed-off-by: Martin Storsjö <martin@martin.st>
RTCP timestamps are only necessary to synchronize time between
multiple streams. For a single stream, the RTP packet timestamp
provides more reliable timing. As a result, single-stream RTP
sessions should now have accurate and monotonic PTS.
Signed-off-by: Martin Storsjö <martin@martin.st>
