Currently, AVStream contains an embedded AVCodecContext instance, which
is used by demuxers to export stream parameters to the caller and by
muxers to receive stream parameters from the caller. It is also used
internally as the codec context that is passed to parsers.
In addition, it is also widely used by the callers as the decoding (when
demuxer) or encoding (when muxing) context, though this has been
officially discouraged since Libav 11.
There are multiple important problems with this approach:
- the fields in AVCodecContext are in general one of
* stream parameters
* codec options
* codec state
However, it's not clear which ones are which. It is consequently
unclear which fields are a demuxer allowed to set or a muxer allowed to
read. This leads to erratic behaviour depending on whether decoding or
encoding is being performed or not (and whether it uses the AVStream
embedded codec context).
- various synchronization issues arising from the fact that the same
context is used by several different APIs (muxers/demuxers,
parsers, bitstream filters and encoders/decoders) simultaneously, with
there being no clear rules for who can modify what and the different
processes being typically delayed with respect to each other.
- avformat_find_stream_info() making it necessary to support opening
and closing a single codec context multiple times, thus
complicating the semantics of freeing various allocated objects in the
codec context.
Those problems are resolved by replacing the AVStream embedded codec
context with a newly added AVCodecParameters instance, which stores only
the stream parameters exported by the demuxers or read by the muxers.
* commit '2e814d0329aded98c811d0502839618f08642685':
rtpenc: Simplify code by introducing a macro for rescaling NTP timestamps
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
bgmc: Fix av_malloc checks in ff_bgmc_init()
rtp: set the payload type as stream id
Conflicts:
libavformat/rtpenc_chain.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '1cd432e167b1a80853760c89a33606e2b5f229c2':
configure: fix libcdio check
rtsp: Allow setting the reordering buffer size via an AVOption
rtsp: Vertically align a constant definition
rtp: Update the check for distinguishing between RTP and RTCP
aac: fix build with hardcoded tables
fate: dependencies for screen codec tests
riff: Move functions around to be covered by appropriate #ifdefs
Conflicts:
configure
tests/fate/screen.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
cosmetics: Consistently use C-style comments with multiple inclusion guards
anm: fix a few Doxygen comments
misc typo and wording fixes
attributes: add av_noreturn
attributes: drop pointless define guards
configure: do not disable av_always_inline with --enable-small
flvdec: initial stream switch support
avplay: fix write on freed memory for rawvideo
snow: remove a VLA used for edge emulation
x86: lavfi: fix gradfun/yadif build with mmx/sse disabled
snow: remove the runs[] VLA.
snow: Check mallocs at init
flacdec: remove redundant setting of avctx->sample_fmt
Conflicts:
ffplay.c
libavcodec/h264.c
libavcodec/snow.c
libavcodec/snow.h
libavcodec/snowdec.c
libavcodec/snowenc.c
libavformat/flvdec.c
libavutil/attributes.h
tools/patcheck
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
shorten: Use separate pointers for the allocated memory for decoded samples.
atrac3: Fix crash in tonal component decoding.
ws_snd1: Fix wrong samples counts.
movenc: Don't set a default sample duration when creating ismv
rtp: Factorize the check for distinguishing RTCP packets from RTP
golomb: avoid infinite loop on all-zero input (or end of buffer).
bethsoftvid: synchronize video timestamps with audio sample rate
bethsoftvid: add audio stream only after getting the first audio packet
bethsoftvid: Set video packet duration instead of accumulating pts.
bethsoftvid: set packet key frame flag for audio and I-frame video packets.
bethsoftvid: fix read_packet() return codes.
bethsoftvid: pass palette in side data instead of in a separate packet.
sdp: Ignore RTCP packets when autodetecting RTP streams
proresenc: initialise 'sign' variable
mpegaudio: replace memcpy by SIMD code
vc1: prevent using last_frame as a reference for I/P first frame.
Conflicts:
libavcodec/atrac3.c
libavcodec/golomb.h
libavcodec/shorten.c
libavcodec/ws-snd1.c
tests/ref/fate/bethsoft-vid
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
x86: cabac: replace explicit memory references with "m" operands
avplay: don't request a stereo downmix
wmapro: use av_float2int()
lavc: avoid invalid memcpy() in avcodec_default_release_buffer()
lavu: replace int/float punning functions
lavfi: install libavfilter/vsrc_buffer.h
Remove extraneous semicolons
sdp: Restore the original mp4 format h264 extradata if converted
rtpenc: Add support for mp4 format h264
rtpenc: Simplify code by introducing a separate end pointer
movenc: Use the actual converted sample for RTP hinting
Fix a bunch of common typos.
Conflicts:
doc/developer.texi
doc/eval.texi
doc/filters.texi
doc/protocols.texi
ffmpeg.c
ffplay.c
libavcodec/mpegvideo.h
libavcodec/x86/cabac.h
libavfilter/Makefile
libavformat/avformat.h
libavformat/cafdec.c
libavformat/flvdec.c
libavformat/flvenc.c
libavformat/gxfenc.c
libavformat/img2.c
libavformat/movenc.c
libavformat/mpegts.c
libavformat/rtpenc_h264.c
libavformat/utils.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (21 commits)
fate: allow testing with libavfilter disabled
x86: XOP/FMA4 CPU detection support
ws_snd: misc cosmetic clean-ups
ws_snd: remove the 2-bit ADPCM table and just subtract 2 instead.
ws_snd: use memcpy() and memset() instead of loops
ws_snd: use samples pointer for loop termination instead of a separate iterator variable.
ws_snd: make sure number of channels is 1
ws_snd: add some checks to prevent buffer overread or overwrite.
ws_snd: decode to AV_SAMPLE_FMT_U8 instead of S16.
flacdec: fix buffer size checking in get_metadata_size()
rtp: Simplify ff_rtp_get_payload_type
rtpenc: Add a payload type private option
rtp: Correct ff_rtp_get_payload_type documentation
avconv: replace all fprintf() by av_log().
avconv: change av_log verbosity from ERROR to FATAL for fatal errors.
cmdutils: replace fprintf() by av_log()
avtools: parse loglevel before all the other options.
oggdec: add support for Xiph's CELT codec
sol: return error if av_get_packet() fails.
cosmetics: reindent and pretty-print
...
Conflicts:
avconv.c
cmdutils.c
libavcodec/avcodec.h
libavcodec/version.h
libavformat/oggparsecelt.c
libavformat/utils.c
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Specifying the payload type is useful when the type number has
already been negotiated before creating the stream, for example
in SIP protocol.
Signed-off-by: Martin Storsjö <martin@martin.st>
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
Consistently apply this rule: the guard name is obtained from the
filename by stripping the leading "lib", converting '/' and '.' to
'_' and uppercasing the resulting name. Guard names in the root
directory have to be prefixed by "FFMPEG_".
Originally committed as revision 15120 to svn://svn.ffmpeg.org/ffmpeg/trunk
Log:
Add missing header #includes.
Policy violation (change not approved by maintainer)
and while discussions where ongoing and no consensus has been reached.
Originally committed as revision 14500 to svn://svn.ffmpeg.org/ffmpeg/trunk
Anton Khirnov