This commit also disables the async fate test, because it
used internal APIs in a non-kosher way, which no longer
exists.
* commit '2758cdedfb7ac61f8b5e4861f99218b6fd43491d':
lavf: reorganize URLProtocols
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Instead of a linked list constructed at av_register_all(), store them
in a constant array of pointers.
Since no registration is necessary now, this removes some global state
from lavf. This will also allow the urlprotocol layer caller to limit
the available protocols in a simple and flexible way in the following
commits.
This implementation does not support TLS listen sockets and loading
CA/Certs from files.
The Windows API does not support loading PEM certs, and would either
require a manual loader or instead be limited to loading Windows PFX
certificates
TLS listen sockets would have to be implemented quite separately, as many
of the APIs are different for server-mode (as opposed to client mode).
* commit 'b08569a23948db107e5e6175cd4c695427d5339d':
lavf: Replace the ASF demuxer
Conflicts:
Changelog
libavformat/asf.h
libavformat/asfdec.c
libavformat/version.h
tests/ref/fate/wmv8-drm-nodec
tests/ref/seek/lavf-asf
The rewritten demuxer is placed in a new file, the current demuxer is
left as default. Carl has tested both and the one working better is
default.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Move the OpenSSL and GnuTLS implementations to their own files. Other
than the connection code (including options) and some boilerplate, no
code is actually shared.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Move the OpenSSL and GnuTLS implementations to their own files. Other
than the connection code (including options) and some boilerplate, no
code is actually shared.
Signed-off-by: Martin Storsjö <martin@martin.st>
This patch adds support for WebM Live Muxing by adding a new WebM
Chunk muxer. It writes out live WebM Chunks which can be used for
playback using Live DASH Clients.
Please see muxers.texi for sample usage.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Additionally, update some documentation with support for APNG
Signed-off-by: Donny Yang <work@kota.moe>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Since this structurally is quite different from normal RTP
(multiple streams are muxed into one single mpegts stream,
which is packetized into one single RTP session), it is kept
as a separate muxer.
Since this structurally also behaves differently than normal
RTP, all of the other muxers that do chained RTP muxing
(rtsp, sap, mp4) would need to be updated similarly to handle
this - in particular, creating one single rtp_mpegts muxer
for the whole presentation instead of one rtp muxer per stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is mostly to serve as a reference example on how to segment
the output from the mp4 muxer, capable of writing the segment
list in four different ways:
- SegmentTemplate with SegmentTimeline
- SegmentTemplate with implicit segments
- SegmentList with individual files
- SegmentList with one single file per track, and byte ranges
The muxer is able to serve live content (with optional windowing)
or create a static segmented MPD.
In advanced cases, users will probably want to do the segmenting
in their own application code.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '051aadeed104ecbe8ee4850ec2d7e5394f5e1ccd':
ogg: Provide aliases for Speex, Opus and audio-only ogg
Conflicts:
Changelog
libavformat/oggenc.c
libavformat/version.h
See: 2ccc6ff03a
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This patch adds the ability to generate WebM DASH manifest XML using
ffmpeg. A sample command line would be as follows:
ffmpeg \
-f webm_dash_manifest -i video1.webm \
-f webm_dash_manifest -i video2.webm \
-f webm_dash_manifest -i audio1.webm \
-f webm_dash_manifest -i audio2.webm \
-map 0 -map 1 -map 2 -map 3 \
-c copy \
-f webm_dash_manifest \
-adaptation_sets “id=0,streams=0,1 id=1,streams=2,3” \
manifest.xml
It works by exporting necessary fields as metadata tags in matroskadec
and use those values to write the appropriate XML fields as per the WebM
DASH Specification [1]. Some ideas are adopted from webm-tools project
[2].
[1]
https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification
[2]
https://chromium.googlesource.com/webm/webm-tools/+/master/webm_dash_manifest/
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It has not been properly maintained for years and there is little hope
of that changing in the future.
It appears simpler to write a new replacement from scratch than
unbreaking it.
This was added in 9b07a2dc02 as an ABI hack to allow older
code built with lavf 52 to register protocols even if the size
of the URLProtocol struct was increased. Later, registering
protocols from outside of lavf was removed and this workaround
isn't needed any longer since lavf 53.
This removes an unchecked malloc and a memory leak for the cases
when this workaround actually was used - which it hasn't since
lavf 53.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'feeafb4adabd5c17de1738ed9962e40892b20edb':
lavf: do not export av_register_{rtp,rdt}_dynamic_payload_handlers from shared objects
Merged-by: Michael Niedermayer <michaelni@gmx.at>
F4V is Adobe's mp4/iso media variant, with the most significant
addition/change being supporting other flash codecs than just
aac/h264.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is admittedly kind of pointless since usually -f image2pipe
can be used for the purpose, but this is more user-friendly.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* commit '2f3bada63e57345329c4f9b48e9b81b5cfc03d05':
lavf: Add a protocol for SRTP encryption/decryption
rtsp: Support decryption of SRTP signalled via RFC 4568 (SDES)
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is mostly useful for encryption together with the RTP muxer,
but could also be set up as IO towards the peer with the SDP
demuxer with custom IO.
Signed-off-by: Martin Storsjö <martin@martin.st>
The function find_things() in configure is confused by component
registration calls as part of multiline macros defining combined
component registration. Coalesce those macros into one line to
work around the issue.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Signed-off-by: Martin Storsjö <martin@martin.st>
Note that the linebreaks text codec option (but not the feature) has
been removed; its main goal was to allow demuxers to configure the text
decoder (and not meant to be used by users), but the AVOption are not a
viable solution. This is solved differently in this commit.
Gif demuxer is capable of extracting multiple frames from gif file.
In conjunction with gif decoder it implements support for reading
animated gifs.
Demuxer has two options available to user: default_delay and min_delay.
These options are for protection from too rapid gif animations. In practice
it is standard approach to slow down rendering of this kind of gifs. If you try to
play gif with delay between frames of one hundredth of second (100fps) using
one of major web browsers, you get significantly slower playback,
around 10 fps. This is because browser detects that delay value is less than some
threshold (usually 2 hundredths of second) and reset it to default value (usually 10
hundredths of second, which corresponds to 10fps). Manipulating these options user
can achieve the same effect during conversion to some video format. Otherwise user
can set them to not protect from rapid animations at all.
The other case when these options necessary is for gif images encoded according to
gif87a standard since prior to gif89a there was no delay information included in file.
Bump lavf minor version.
Signed-off-by: Vitaliy E Sugrobov <vsugrob@hotmail.com>
* commit '07584eaf4a95db3f11d3bc411f9786932829e82b':
mpegts: check substreams before discarding
Add a smooth streaming segmenter muxer
file: Add an avoption for disabling truncating existing files on open
img2dec: always close AVIOContexts
rtpdec_jpeg: Error out on other unsupported type values as well
rtpdec_jpeg: Disallow using the reserved q values
rtpdec_jpeg: Fold the default qtables case into an existing if statement
rtpdec_jpeg: Store and reuse old qtables for q values 128-254
rtpdec_jpeg: Simplify the calculation of the number of qtables
rtpdec_jpeg: Add more comments about the fields in the SOF0 section
rtpdec_jpeg: Clarify where the subsampling magic numbers come from
rtpdec_jpeg: Don't use a bitstream writer for the EOI marker
rtpdec_jpeg: Don't needlessly use a bitstream writer for the header
rtpdec_jpeg: Simplify writing of the jpeg header
rtpdec_jpeg: Merge two if statements
rtpdec_jpeg: Write the DHT section properly
Conflicts:
libavformat/Makefile
libavformat/allformats.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This muxer splits the output from the ismv muxer into individual
files, in realtime.
The same can also be done by the standalone tool ismindex, but this
muxer is needed for doing it in realtime (especially for live
streams that need extra handling for updating the lookahead fields
in the fragment headers).
Using this muxer, one can deliver live smooth streaming from a
normal static file web server. (Using ismindex, one can deliver
premade smooth streaming files from a static file web server,
or prepare files for serving with IIS.)
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Michael Bradshaw <mbradshaw@sorensonmedia.com>
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
v410dec: Implement explode mode support
zerocodec: fix direct rendering.
wav: init st to NULL to avoid a false-positive warning.
wavpack: set bits_per_raw_sample for S32 samples to properly identify 24-bit
h264: refactor NAL decode loop
RTMPTE protocol support
RTMPE protocol support
rtmp: Add ff_rtmp_calc_digest_pos()
rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
swscale: add missing HAVE_INLINE_ASM check.
lavfi: place x86 inline assembly under HAVE_INLINE_ASM.
vc1: Add a test for interlaced field pictures
swscale: Mark all init functions as av_cold
swscale: x86: Drop pointless _mmx suffix from filenames
lavf: use conditional notation for default codec in muxer declarations.
swscale: place inline assembly bilinear scaler under HAVE_INLINE_ASM.
dsputil: ppc: cosmetics: pretty-print
dsputil: x86: add SHUFFLE_MASK_W macro
configure: respect CC_O setting in check_cc
Conflicts:
Changelog
configure
libavcodec/v410dec.c
libavcodec/zerocodec.c
libavformat/asfenc.c
libavformat/version.h
libswscale/utils.c
libswscale/x86/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPE protocol implementation uses ffrtmpcrypt:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
libopenjpeg: introduce encoding support
libopenjpeg: rename decoder source file.
RTMPTS protocol support
RTMPS protocol support
avconv: print an error message when demuxing fails.
tscc2: DCT output should not be clipped
rtmp: Rename rtmphttp to ffrtmphttp
Conflicts:
Changelog
configure
doc/general.texi
libavcodec/libopenjpegenc.c
libavcodec/version.h
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This simplifies usage for segment streaming formats with no global
headers, tipically MPEG 2 transport stream "ts" files.
The seg class duplication is required in order to avoid an infinite loop
in libavformat/utils.c:format_child_next_class().
* qatar/master: (24 commits)
flvdec: remove incomplete, disabled seeking code
mem: add support for _aligned_malloc() as found on Windows
lavc: Extend the documentation for avcodec_init_packet
flvdec: remove incomplete, disabled seeking code
http: replace atoll() with strtoll()
mpegts: remove unused/incomplete/broken seeking code
af_amix: allow float planar sample format as input
af_amix: use AVFloatDSPContext.vector_fmac_scalar()
float_dsp: add x86-optimized functions for vector_fmac_scalar()
float_dsp: Move vector_fmac_scalar() from libavcodec to libavutil
lavr: Add x86-optimized function for flt to s32 conversion
lavr: Add x86-optimized function for flt to s16 conversion
lavr: Add x86-optimized functions for s32 to flt conversion
lavr: Add x86-optimized functions for s32 to s16 conversion
lavr: Add x86-optimized functions for s16 to flt conversion
lavr: Add x86-optimized function for s16 to s32 conversion
rtpenc: Support packetizing iLBC
rtpdec: Add a depacketizer for iLBC
Implement the iLBC storage file format
mov: Support muxing/demuxing iLBC
...
Conflicts:
Changelog
configure
libavcodec/avcodec.h
libavcodec/dsputil.c
libavcodec/version.h
libavformat/movenc.c
libavformat/mpegts.c
libavformat/version.h
libavutil/mem.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavr: add x86-optimized functions for mixing 1-to-2 s16p with flt coeffs
lavr: add x86-optimized functions for mixing 1-to-2 fltp with flt coeffs
Add Dolby/DPLII downmix support to libavresample
vorbisdec: replace div/mod in loop with a counter
fate: vorbis: add 5.1 surround test
rtpenc: Allow requesting H264 RTP packetization mode 0
configure: Sort the library listings in the help text alphabetically
dwt: remove variable-length arrays
RTMPT protocol support
http: Properly handle chunked transfer-encoding for replies to post data
http: Fail reading if the connection has gone away
amr: Mark an array const
amr: More space cleanup
rtpenc: Fix memory leaks in the muxer open function
Conflicts:
Changelog
configure
doc/APIchanges
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPT protocol implementation uses rtmphttp:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (27 commits)
libxvid: Give more suitable names to libxvid-related files.
libxvid: Separate libxvid encoder from libxvid rate control code.
jpeglsdec: Remove write-only variable in ff_jpegls_decode_lse().
fate: cosmetics: lowercase some comments
fate: Give more consistent names to some RealVideo/RealAudio tests.
lavfi: add avfilter_get_audio_buffer_ref_from_arrays().
lavfi: add extended_data to AVFilterBuffer.
lavc: check that extended_data is properly set in avcodec_encode_audio2().
lavc: pad last audio frame with silence when needed.
samplefmt: add a function for filling a buffer with silence.
samplefmt: add a function for copying audio samples.
lavr: do not try to copy to uninitialized output audio data.
lavr: make avresample_read() with NULL output discard samples.
fate: split idroq audio and video into separate tests
fate: improve dependencies
fate: add convenient shorthands for ea-vp6, libavcodec, libavutil tests
fate: split some combined tests into separate audio and video tests
fate: fix dependencies for probe tests
mips: intreadwrite: fix inline asm for gcc 4.8
mips: intreadwrite: remove unnecessary inline asm
...
Conflicts:
cmdutils.h
configure
doc/APIchanges
doc/filters.texi
ffmpeg.c
ffplay.c
libavcodec/internal.h
libavcodec/jpeglsdec.c
libavcodec/libschroedingerdec.c
libavcodec/libxvid.c
libavcodec/libxvid_rc.c
libavcodec/utils.c
libavcodec/version.h
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/buffersink.h
tests/Makefile
tests/fate/aac.mak
tests/fate/audio.mak
tests/fate/demux.mak
tests/fate/ea.mak
tests/fate/image.mak
tests/fate/libavutil.mak
tests/fate/lossless-audio.mak
tests/fate/lossless-video.mak
tests/fate/microsoft.mak
tests/fate/qt.mak
tests/fate/real.mak
tests/fate/screen.mak
tests/fate/video.mak
tests/fate/voice.mak
tests/fate/vqf.mak
tests/ref/fate/ea-mad
tests/ref/fate/ea-tqi
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (21 commits)
CDXL demuxer and decoder
hls: Re-add legacy applehttp name to preserve interface compatibility.
hlsproto: Rename the functions and context
hlsproto: Encourage users to try the hls demuxer instead of the proto
doc: Move the hls protocol section into the right place
libavformat: Rename the applehttp protocol to hls
hls: Rename the functions and context
libavformat: Rename the applehttp demuxer to hls
rtpdec: Support H263 in RFC 2190 format
rv30: check block type validity
ttadec: CRC checking
movenc: Support muxing VC1
avconv: Don't split out inline sequence headers when stream copying VC1
rv34: handle size changes during frame multithreading
rv40: prevent undefined signed overflow in rv40_loop_filter()
rv34: use AVERROR return values in ff_rv34_decode_frame()
rv34: use uint16_t for RV34DecContext.deblock_coefs
librtmp: Add "lib" prefix to librtmp URLProtocol declarations.
movenc: Use defines instead of hardcoded numbers for RTCP types
smjpegdec: implement seeking
...
Conflicts:
Changelog
doc/general.texi
libavcodec/avcodec.h
libavcodec/rv30.c
libavcodec/tta.c
libavcodec/version.h
libavformat/Makefile
libavformat/allformats.c
libavformat/version.h
libswscale/x86/swscale_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Keep the old protocol name around for backwards compatibility
until the next bump.
Deprecate the method of implicitly assuming the nested protocol.
For applehttp://server/path, it might have felt logical, but
supporting hls://server/path isn't quite as intuitive. Therefore
only support hls+http://server/path from now on.
Using this protocol at all is discouraged, since the hls demuxer
is more complete and fits into the architecture better. There
have been cases where the protocol implementation worked better
than the demuxer, but this should no longer be the case.
Signed-off-by: Martin Storsjö <martin@martin.st>
When this demuxer was created, there didn't seem to be any
consensus of a common short name for this protocol. Now
the consensus seems to be to call it hls.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows easily differentiating between both implementations within the build
system and combining the native implementation for plain RTMP with librtmp for
the RTMPE, RTMPS, RTMPT, RTMPTE protocol variants.
* qatar/master:
rtpdec: Use 4 byte startcodes for H.264
matroskadec: Mark variable as av_unused.
Move some conditionally used variables into the block where they are used.
Drop some completely unnecessary av_unused attributes.
swscale: Remove unused variable alpMmxFilter.
Drop unnecessary av_uninit attributes from some variable declarations.
movenc: Support muxing wmapro in ismv/isma
mpegtsenc: Add an AVOption for forcing a new PAT/PMT/SDT to be written
swscale: move YUV2PACKED16WRAPPER() macro down to where it is used.
swscale: handle gray16 as a "planar" YUV format (Y-only, of course).
swscale: use yuv2packed1() functions for unscaled chroma also.
swscale: fix incorrect chroma bias in yuv2rgb48_1_c().
swscale: fix invalid memory accesses in yuvpacked1() functions.
Move PS2 MMI code below the mips subdirectory, where it belongs.
mips: Move MMI function declarations to a header.
build: Set correct dependencies for rtmp* protocols implemented by librtmp.
Conflicts:
libavcodec/ac3enc_template.c
libavformat/mpegtsenc.c
libswscale/output.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
wma: Clip WMA1 and WMA2 frame length to 11 bits.
movenc: Don't require frame_size to be set for modes other than mov
doc: Update APIchanges with info on muxer flushing
movenc: Reindent a block
tools: Remove some unnecessary #undefs.
rv20: prevent calling ff_h263_decode_mba() with unset height/width
tools: K&R reformatting cosmetics
Ignore generated aviocat and ismindex tools.
build: Automatically include architecture-specific library Makefile snippets.
indeo5: prevent null pointer dereference on broken files
pktdumper: Use usleep instead of sleep
cosmetics: Remove some unnecessary block braces.
Drop unnecessary prefix from *sink* variable and struct names.
Add a tool for creating smooth streaming manifests
movdec: Calculate an average bit rate for fragmented streams, too
movenc: Write the sample rate instead of time scale in the stsd atom
movenc: Add a separate ismv/isma (smooth streaming) muxer
movenc: Allow the caller to decide on fragmentation
libavformat: Add a flag for muxers that support write_packet(NULL) for flushing
movenc: Add support for writing fragmented mov files
...
Conflicts:
Changelog
cmdutils.c
cmdutils.h
doc/APIchanges
ffmpeg.c
ffplay.c
libavfilter/Makefile
libavformat/Makefile
libavformat/avformat.h
libavformat/movenc.c
libavformat/movenc.h
libavformat/version.h
tools/graph2dot.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rv34: add NEON rv34_idct_add
rv34: 1-pass inter MB reconstruction
add SMJPEG muxer
avformat: split out common SMJPEG code
pictordec: Use bytestream2 functions
avconv: use avcodec_encode_audio2()
pcmenc: use AVCodec.encode2()
avcodec: bump minor version and add APIChanges for the new audio encoding API
avcodec: Add avcodec_encode_audio2() as replacement for avcodec_encode_audio()
avcodec: add a public function, avcodec_fill_audio_frame().
rv34: Intra 16x16 handling
rv34: Inter/intra MB code split
Conflicts:
Changelog
libavcodec/avcodec.h
libavcodec/pictordec.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/rv34dsp.asm
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (46 commits)
mtv: Make sure audio_subsegments is not 0
v4l2: use V4L2_FMT_FLAG_EMULATED only if it is defined
avconv: add symbolic names for -vsync parameters
flvdec: Fix compiler warning for uninitialized variables
rtsp: Fix compiler warning for uninitialized variable
ulti: convert to new bytestream API.
swscale: Use standard multiple inclusion guards in ppc/ header files.
Place some START_TIMER invocations in separate blocks.
v4l2: list available formats
v4l2: set the proper codec_tag
v4l2: refactor device_open
v4l2: simplify away io_method
v4l2: cosmetics
v4l2: uniform and format options
v4l2: do not force interlaced mode
avio: exit early in fill_buffer without read_packet
vc1dec: fix invalid memory access for small video dimensions
rv34: fix invalid memory access for small video dimensions
rv34: joint coefficient decoding and dequantization
avplay: Don't call avio_set_interrupt_cb(NULL)
...
Conflicts:
Changelog
avconv.c
doc/APIchanges
doc/indevs.texi
libavcodec/adxenc.c
libavcodec/dnxhdenc.c
libavcodec/h264.c
libavdevice/v4l2.c
libavformat/flvdec.c
libavformat/mtv.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
APIchanges: fill in revision for AVFrame.age deprecation
avcodec: deprecate AVFrame.age
4xm: remove unneeded check for remaining unused data.
lavf: force threads to 1 in avformat_find_stream_info()
swscale: fix overflows in vertical scaling at top/bottom edges.
lavf: add OpenMG audio muxer.
omadec: split data that will be used in the muxer to a separate file.
lavf: rename oma.c -> omadec.c
tmv decoder: set correct pix_fmt
Conflicts:
Changelog
doc/APIchanges
libavcodec/mpegvideo.c
libavcodec/version.h
libavformat/oma.c
libavformat/version.h
libswscale/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
