If an annex b bitstream is muxed into mov, the actual written
sample is reformatted to mp4 syntax before writing.
Currently, the RTP hints that copy data from the normal video
track, where the payload data might be offset compared to the
original sample that the RTP hinting used (when 3 byte
annex b startcodes have been converted into 4 byte mp4 format
startcodes).
Signed-off-by: Martin Storsjö <martin@martin.st>
It had become dead code when code was added to avoid
exporting audio and video codec id as metadata.
Untested due to lack of sample.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
isom: sort and pretty-print codec_movaudio_tags[]
isom: remove pointless comments in codec_movaudio_tags[]
isom: remove commented-out tag for vorbis
movenc: write 'chan' tag for AC-3 in MOV
mov: add support for reading and writing the 'chan' tag
audioconvert: add some additional channel and channel layout macros
audioconvert: change 7.1 "wide" layout to use side surround channels
movenc: simplify handling of pcm vs. adpcm vs. other compressed codecs
doc: update documentation to use avconv
doc: update demuxers section
doc: extend external library coverage
doc: split platform specific information
doc: port the git-howto to texinfo
doc: provide fallback css and customize @float
doc: document fate in a texinfo
doxy: change hue value to match our green
Conflicts:
doc/fate.txt
doc/ffserver.texi
doc/general.texi
doc/muxers.texi
doc/protocols.texi
doc/t2h.init
libavformat/isom.c
libavformat/mov.c
libavutil/avutil.h
tests/ref/acodec/pcm_s16be
tests/ref/acodec/pcm_s24be
tests/ref/acodec/pcm_s32be
tests/ref/acodec/pcm_s8
tests/ref/lavf/mov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This implements reading the tag in the demuxer and adds support for writing it
in the muxer. Some example channel layout tables for muxing are included for
ac3, aac, and alac, but they are not utilized yet.
Use Sound Sample Description Version 2 for all MOV files.
Updated FATE references accordingly.
Note that ADPCM is treated as compressed audio in version 2.
* qatar/master:
cljr: K&R cosmetics
cljr: return a more sensible value when encountering invalid headers
cljr: drop unnecessary emms_c() calls without MMX code
cljr: remove useless casts
cljr: group encode/decode parts under single ifdefs
cljr: remove stray semicolon
cljr: add missing return statement in decode_end()
doc: add pulseaudio to the input list
avconv: remove unsubstantiated comment
shorten: avoid abort() on unknown audio types
cljr: add encoder
build: merge lists of HTML documentation targets
tests/examples: Mark some variables only used within their files as static.
tests/tools/examples: Replace direct exit() calls by return.
x86 cpuid: set vendor union members separately
cljr: release picture at end of decoding
rv40: NEON optimised rv40 qpel motion compensation
Conflicts:
doc/examples/muxing.c
libavcodec/cljr.c
libavcodec/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* tjoppen/opatom_demuxing_and_seeking:
mxfdec: Index table driven demuxing and seeking
mxfdec: Compute packet offsets properly
mxfdec: Use MaterialPackage - Track - TrackID instead of the system_item hack
mxfdec: Parse more values in PartitionPack
mxfdec: Parse TemporalOffsets
mxfdec: av_dlog():ify 'no corresponding source package found'
mxfdec: Compute essence container offsets and lengths into mxf->partitions
mxfdec: Make mxf->partitions sorted by offset
mxfdec: Parse ThisPartition
mxfdec: Speed up metadata and index parsing
mxfdec: Make sure DataDefinition is consistent between material track and source track
mxfdec: Add EssenceContainer UL found in 0001GL00.MXF.A1.mxf_opatom.mxf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
These values include KAGSize, HeaderByteCount and IndexByteCount.
The length of the pack itself is also stored, and KAGSize is sanity checked.
The FATE sample has KAGSize == 0, which is adjusted to 512.
Other bad KAGSizes are set to 1.
The information is relevant, but under normal circumstances
it raises far too many false alarms.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
drawtext: remove typo
pcm-mpeg: implement new audio decoding api
w32thread: port fixes to pthread_cond_broadcast() from x264.
doc: add editor configuration section with Vim and Emacs settings
dxva2.h: include d3d9.h to define LPDIRECT3DSURFACE9
avformat/utils: Drop unused goto label.
doxygen: Replace '\' by '@' in Doxygen markup tags.
cosmetics: drop some completely pointless parentheses
cljr: simplify CLJRContext
drawtext: introduce rand(min, max)
drawtext: introduce explicit draw/hide variable
rtmp: Use nb_invokes for all invoke commands
Conflicts:
libavcodec/mpegvideo.c
libavfilter/vf_drawtext.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Specifically, this means parsing as before until we run into essence.
At that point we seek to the footer and parse until EOF. After that we start
seeking backward to the previous partition and parse that until we run into
essence or the next partition. This procedure is repeated until we encounter
the last partition we parsed in the forward direction.
The end result of all this is that large essence containers aren't needlessly
parsed. This speeds up parsing large files a lot.
704af3e29c broke publishing
of rtmp streams, at least publishing to Wowza servers.
This changes all invoke commands to use nb_invokes.
Signed-off-by: Martin Storsjö <martin@martin.st>
Its checked a few lines below too.
The only difference is that empty atoms with size=0 will now get parsed too.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The computed size doesn't contain the header size because it's already
skipped by incrementing total_size, but then it's skipped again in the
last line. The atom comes out 8 bytes short and the function
mov_read_chan() aborts the whole parsing process. I think the computed
size should be atom.size - total_size + 8.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mov: Don't av_malloc(0).
avconv: only allocate 1 AVFrame per input stream
avconv: fix memleaks due to not freeing the AVFrame for audio
h264-fate: remove -strict 1 except where necessary (mr4/5-tandberg).
misc Doxygen markup improvements
doxygen: eliminate Qt-style doxygen syntax
g722: Add a regression test for muxing/demuxing in wav
g722: Change bits per sample to 4
g722dec: Signal skipping the lower bits via AVOptions instead of bits_per_coded_sample
api-example: update to use avcodec_decode_audio4()
avplay: use avcodec_decode_audio4()
avplay: use a separate buffer for playing silence
avformat: use avcodec_decode_audio4() in avformat_find_stream_info()
avconv: use avcodec_decode_audio4() instead of avcodec_decode_audio3()
mov: Allow empty stts atom.
doc: document preferred Doxygen syntax and make patcheck detect it
Conflicts:
avconv.c
ffplay.c
libavcodec/mlpdec.c
libavcodec/version.h
libavformat/mov.c
tests/codec-regression.sh
tests/fate/h264.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
malloc() is allowed to return NULL when zero is the argument. This
causes us to think malloc has failed and return AVERROR(ENOMEM). In
addition OS X malloc() returns an unfreeable non-NULL pointer for size
zero when alignment is greater than 16.
instead of when the 2nd stream has been found.
This isnt ideal as we will likely still like before miss a data stream.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
aac_latm: reconfigure decoder on audio specific config changes
latmdec: fix audio specific config parsing
Add avcodec_decode_audio4().
avcodec: change number of plane pointers from 4 to 8 at next major bump.
Update developers documentation with coding conventions.
svq1dec: avoid undefined get_bits(0) call
ARM: h264dsp_neon cosmetics
ARM: make some NEON macros reusable
Do not memcpy raw video frames when using null muxer
fate: update asf seektest
vp8: flush buffers on size changes.
doc: improve general documentation for MacOSX
asf: use packet dts as approximation of pts
asf: do not call av_read_frame
rtsp: Initialize the media_type_mask in the rtp guessing demuxer
Cleaned up alacenc.c
Conflicts:
doc/APIchanges
doc/developer.texi
libavcodec/8svx.c
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/nellymoserdec.c
libavcodec/tta.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/wmadec.c
libavformat/asfdec.c
tests/ref/seek/lavf_asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Pass the correct size in bits to mpeg4audio_get_config and add a flag
to disable parsing of the sync extension when the size is not known.
Latm with AudioMuxVersion 0 does not specify the size of the audio
specific config. Data after the audio specific config can be
misinterpreted as sync extension resulting in random and wrong configs.
Commit 035af99 made avconv always call an encoder when using the
null muxer. While useful for 2-pass encodes, it inadvertently
caused an extra memcpy of raw frames when decoding only.
This hack restores the old behaviour when only decoding while
allowing use of the null muxer with encoded streams as well.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The media_type_mask is initialized via AVOptions for the
rtsp and sdp demuxers, but it isn't available as an option
for the rtp guessing demuxer (since it doesn't really make
sense there). Therefore, it must be manually initialized
instead, since a zero value means no media types at all
are accepted.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (25 commits)
rtpenc: Add support for G726 audio
rtpdec: Interpret the different G726 names as bits_per_coded_sample
rtpenc: Change rtp_send_samples to handle sample sizes other than even bytes
rtpenc: Cast a rescaling parameter to int64_t
h264: cap max has_b_frames at MAX_DELAYED_PIC_COUNT - 1.
ARM: fix indentation in ff_dsputil_init_neon()
ARM: NEON put/avg_pixels8/16 cosmetics
ARM: add remaining NEON avg_pixels8/16 functions
ARM: clean up NEON put/avg_pixels macros
fate: split acodec-pcm into individual tests
swscale: #include "libavutil/mathematics.h"
pmpdec: don't use deprecated av_set_pts_info.
rv34: align temporary block of "dct" coefs
Add PlayStation Portable PMP format demuxer
proto: Realign struct initializers
proto: Use .priv_data_size to allocate the private context
mmsh: Properly clean up if the second ffurl_alloc failed
rtmp: Clean up properly if the handshake failed
md5proto: Remove the get_file_handle function
applehttpproto: Use the close function if the open function fails
...
Conflicts:
libavcodec/vble.c
libavformat/mmsh.c
libavformat/pmpdec.c
libavformat/udp.c
tests/ref/acodec/pcm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When skipping over the extended header, take into account
that the size field has already been read. The extended header
also takes up space, so adjust total header length accordingly.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
For the standardized 8 kHz sample rate, this works exactly the same.
For nonstandard sample rates, the different predefined G726
names (G726-16, G726-24, G726-32, G726-40) are interpreted as an
indication of the bits per coded sample, even though their
actual bitrates aren't what the name specifies.
This feels more sane than using free-form names for nonstandard
sample rate/bitrate combinations, e.g like G726-22, G726-33
for 11025 Hz.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids overflow if frame_size is over 2147, since both
frame_size and AV_TIME_BASE are plain integers.
Signed-off-by: Martin Storsjö <martin@martin.st>
Not yet complete, for demuxing AAC the AAC header must be generated
manually.
Possibly the decoder could accept the header as extradata to simplify
this.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This simplifies the open functions by avoiding one function
call that needs error checking, reducing the amount of
extra bulk code.
Signed-off-by: Martin Storsjö <martin@martin.st>
This string will be passed to ff_http_auth_create_response
even if no proxy is used, resulting in reading uninitialized
memory. The other auth string is always initialized by
av_url_split.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
rtpdec: Templatize the code for different g726 bitrate variants
rv40: move loop filter to rv34dsp context
lavf: make av_set_pts_info private.
rtpdec: Add support for G726 audio
rtpdec: Add an init function that can do custom codec context initialization
avconv: make copy_tb on by default.
matroskadec: don't set codec timebase.
rmdec: don't set codec timebase.
avconv: compute next_pts from input packet duration when possible.
lavf: estimate frame duration from r_frame_rate.
avconv: update InputStream.pts in the streamcopy case.
Conflicts:
avconv.c
libavdevice/alsa-audio-dec.c
libavdevice/bktr.c
libavdevice/fbdev.c
libavdevice/libdc1394.c
libavdevice/oss_audio.c
libavdevice/v4l.c
libavdevice/v4l2.c
libavdevice/vfwcap.c
libavdevice/x11grab.c
libavformat/au.c
libavformat/eacdata.c
libavformat/flvdec.c
libavformat/mpegts.c
libavformat/mxfenc.c
libavformat/rtpdec_g726.c
libavformat/wtv.c
libavformat/xmv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This requires using a separate init function, since there
isn't necessarily any fmtp lines for this codec, so
parse_sdp_a_line won't be called. Incorporating it with the
alloc function wouldn't do either, since it is called before
the full rtpmap line is parsed (where the sample rate is
extracted).
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
adtsenc: Check frame size.
txd: Fix order of operations.
APIchanges: fill in some blanks
timer: fix misspelling of "decicycles"
Eliminate pointless 0/NULL initializers in AVCodec and similar declarations.
indeo3: cosmetics
md5proto: Fix order of operations.
dca: Replace oversized unused get_bits() with skip_bits_long().
Conflicts:
doc/APIchanges
libavformat/mmsh.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
Our ac3 code chain can handle it fine.
More ideal would be to write a demuxer that actually extracts what can be from the additional
headers and uses it for whatever it can be used for.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
TLSv1 is compatible with SSLv3, so this doesn't change much
in terms of compatibility. By explicitly using TLSv1, OpenSSL
sends the server name indication (SNI) header, which we
already set using SSL_set_tlsext_host_name (earlier, this
didn't have any effect).
SNI allows servers to serve SSL content for different host
names with separate certificates on one single port (vhosts).
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
mpegaudiodec: Don't use a nonexistent log context for av_dlog
avformat: Accept the ISO8601 separate format as input, too
avformat: Interpret times in ff_iso8601_to_unix_time as UTC
avutil: Add av_timegm as a public function
cinepak: Add another special case so that it can handle the following file:
lagarith: add some RGBA decoding support
lagarith: Add correct line prediction for RGB
Conflicts:
doc/APIchanges
libavcodec/cinepak.c
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes the function accept the format of creation_time
as output by demuxers (e.g. the mov demuxer), making the
creation timestamp stay intact if transcoding.
Signed-off-by: Martin Storsjö <martin@martin.st>
This function is used in muxers for parsing the 'creation_time'
metadata key, for converting it to a time value.
This makes it match the behaviour of the exported 'creation_time'
metadata from demuxers, where it is in UTC, too.
Signed-off-by: Martin Storsjö <martin@martin.st>
Converting to double before the multiplication rather than after
avoids an integer overflow in some cases.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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.
* qatar/master:
binkvideo: simplify and remove invalid shifts
pulse: compute frame_duration once and fix it
lavf: simplify format_child_class_next()
hwaccel: OS X Video Decoder Acceleration (VDA) support.
doc: add support for an optional navigation bar in texi2html pages
Conflicts:
configure
libavcodec/Makefile
libavcodec/allcodecs.c
libavcodec/vda.c
libavcodec/vda.h
libavcodec/vda_h264.c
libavcodec/vda_internal.h
libavcodec/version.h
libavformat/options.c
libavutil/avutil.h
libavutil/pixfmt.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
