* qatar/master:
Fix parser: mark av_parser_parse() for removal on next major bump
swscale: postpone sws_getContext removal until next major bump.
fate: add AAC LATM test
mmst: get rid of deprecated AVERRORs
lxfdec: use AVERROR(ENOMEM) instead of deprecated AVERROR_NOMEM.
Reemove remaining uses of deprecated AVERROR_NOTSUPP.
REIMPLEMENTED in 2 lines of code: lavf: if id3v2 tag is present and all else fails, guess by file extension
Conflicts:
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Support the URL_FLAG_NONBLOCK semantic and uniform the protocol.
The quick retry loop is already part of retry_transfer_wrapper.
The polling routine is common to the network protocols:
udp, tcp and, once merged, sctp.
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.
* qatar/master:
psymodel: extend API to include PE and bit allocation.
avio: always compile dyn_buf functions
Remove unnecessary parameter from ff_thread_init() and fix behavior
Revert "aac_latm_dec: use aac context and aac m4ac"
configure: tell user if libva is enabled like the rest of external libs.
Add silence support for AV_SAMPLE_FMT_U8.
avio: make URL_PROTOCOL_FLAG_NESTED_SCHEME internal
avio: deprecate av_url_read_seek
avio: deprecate av_url_read_pause
ac3enc: NEON optimised extract_exponents
Conflicts:
libavcodec/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
This avoids issues where EOF at the end of the segment is given
the variant demuxer. Now the demuxers only see one single data
stream (as when using the applehttp protocol handler).
* 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.
As per issue2629, most 23.976fps matroska H.264 files are incorrectly
detected as 24fps, as the matroska timestamps usually have only
millisecond precision.
Fix that by doubling the amount of timestamps inspected for frame rate
for streams that have coarse time base. This also fixes 29.970 detection
in matroska.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Fixed-point MDCT with 32-bit unscaled output
lavc: deprecate rate_emu
lavc: mark hurry_up for removal on next major bump
parser: mark av_parser_parse() for removal on next major bump
lavc: add missing audioconvert includes
jvdec: don't use deprecated CODEC_TYPE_*/PKT_FLAG_KEY
Conflicts:
libavcodec/h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ac3enc: ARM optimised ac3_compute_matissa_size
ac3: armv6 optimised bit_alloc_calc_bap
fate: simplify fft test rules
avio: document avio_alloc_context.
lavf: make compute_chapters_end less picky.
sierravmd: fix Indeo3 videos
FFT: simplify fft8()
fate: add fixed-point fft/mdct tests
Fixed-point support in fft-test
ape: check that number of seektable entries is equal to number of frames
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In particular, now it assumes that
a) chapters are chronologically ordered
b) chapters have the same timebases
c) duration of the stream is known
and asserts if any of these is not met.
Make it properly deal with harsher conditions.
fixes issue2320
* newdev/master:
mpegts: propagate avio EOF in read_packet()
configure: Initial support for --target-os=symbian
Fixed-point FFT and MDCT
Include dependencies for test programs
ac3enc: simplify sym_quant()
flvdec: read index stored in the 'keyframes' tag.
mov: Add support for zero-sized stsc runs.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Allows distinguishing between EOF and IO error in read_packet return code.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* newdev/master:
rtsp: Use GET_PARAMETER for keep-alive for generic RTSP servers
mlp_parse.c: set AVCodecContext channel_layout
APIChanges: mark the place where 0.6 was branched.
avio: make get_checksum() internal.
avio: move ff_crc04C11DB7_update() from avio.h -> avio_internal.h
avio: make init_checksum() internal.
NOT MERGED Add MxPEG decoder
NOT MERGED Add support for picture_ptr field in MJpegDecodeContext
NOT MERGED Move MJPEG's input buffer preprocessing in separate public function
NOT MERGED Support reference picture defined by bitmask in MJPEG's SOS decoder
sndio bug fix
Merged-by: Michael Niedermayer <michaelni@gmx.at>
'keyframes' metatag is not part of the standard, it is just
convention to use such kind of metatag information for indexing.
Structure is following, it allows to have it inconsistent:
keyframes:
times (array):
time0 (num)
time1 (num)
time2 (num)
filepositions (array)
position0 (num)
position1 (num)
Signed-off-by: Anton Khirnov <anton@khirnov.net>
A zero sized stsc run doesn't make a lot of sense but the spec does not
prohibit them and MPlayer VLC demuxers support them.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
According to the RFC, GET_PARAMETER should be used for
this, and according to a report from Tim Ouellette,
OPTIONS doesn't work for keeping the connection alive for some
servers. Also, live555 uses GET_PARAMETER for this purpose.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* ffmpeg-mt/master:
Update todo. More items appeared...
Fix mdec
Duplicate: id3v1: change filesize to int64_t.
Duplicate: id3v1: Seek back to old position after reading.
Conflicts:
libavcodec/mpegvideo.c
libavcodec/snow.c
libavformat/id3v1.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
FFmpeg did not seek back to the original position, but to "0", making
reading a VBR tag impossible.
(issue 2645)
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* newdev/master:
ac3enc: avoid memcpy() of exponents and baps in EXP_REUSE case by using exponent reference blocks.
Chronomaster DFA decoder
DUPLICATE: framebuffer device demuxer
NOT MERGED: cosmetics: fix dashed line length after 070c5d0
http: header field names are case insensitive
Conflicts:
LICENSE
README
doc/indevs.texi
libavdevice/fbdev.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Amazon S3 sends header field names all lowercase.
This is actually acceptable according to the HTTP standard.
http://tools.ietf.org/html/rfc2616#section-4.2
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* newdev/master:
mov: set audio service type for AC-3 from bitstream mode in the 'dac3' atom.
Get audio_service_type for AC-3 based on bitstream mode in the AC-3 parser and decoder, and vice-versa for the AC-3 encoder.
Use audio_service_type to set stream disposition.
Add APIchanges entry for audio_service_type.
Add audio_service_type field to AVCodecContext for encoding and reporting of the service type in the audio bitstream.
configure: in check_ld, place new -l flags before existing ones
support @heading, @subheading, @subsubheading, and @subsubsection in texi2pod.pl
doc: update build system documentation
aacenc: indentation
aacenc: fix the side calculation in search_for_ms
vp8.c: rename EDGE_* to VP8_EDGE_*.
Conflicts:
doc/APIchanges
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/vp8.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* newdev/master:
matroskadec: set default duration for simple block
When building for MinGW32 disable strict ANSI compliancy.
ARM: fix ff_apply_window_int16_neon() prototype
configure: check for --as-needed support early
ARM: NEON optimised apply_window_int16()
ac3enc: NEON optimised shift functions
ac3enc: NEON optimised ac3_max_msb_abs_int16 and ac3_exponent_min
mpeg12.c: fix slice threading for mpeg2 field picture mode.
ffmetadec.c: fix compiler warnings.
configure: Don't explicitly disable ffplay or in/outdevices on dos
configure: Remove the explicit disabling of ffserver
configure: Add fork as a dependency to ffserver
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When a normal Block is parsed, duration is initialized to
AV_NOPTS_VALUE. If it is not changed, then the track's default
duration is used. But for SimpleBlock, duration is initialized to
0 instead of AV_NOPTS_VALUE. This is due to the difference in how
EBML_NEST vs EBML_PASS are processed. Setting duration to 0 leads
eventually to wrongly estimate the frame duration in util.c
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* newdev/master:
avio: make udp_set_remote_url/get_local_port internal.
asfdec: also subtract preroll when reading simple index object
matroskaenc: remove a variable that's unused after bc17bd9.
avio: cosmetics - nicer vertical alignment.
Remove unnecessary icc version checks
Disable 'attribute "foo" ignored' warnings from icc
rtsp: Don't use a locale dependent format string
Add xd55 codec tag for XDCAM HD422 720p25 CBR files.
configure: get libavcodec version from new version.h header
lavc: move the version macros to a new installed header.
matroskaenc: simplify get_aac_sample_rates by using ff_mpeg4audio_get_config
Do not use format string "%0.3f" for RTSP Range field.
Add apply_window_int16() to DSPContext with x86-optimized versions and use it in the ac3_fixed encoder.
Document usage of import libraries created by dlltool
configure: Set the correct lib target for arm/wince dlltool
fate: simplify regression-funcs.sh
fate: add support for multithread testing
Conflicts:
libavformat/rtspdec.c
libavutil/attributes.h
libavutil/internal.h
libavutil/mem.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* newdev/master: (33 commits)
Fix an infinite loop when RoQ encoded generated a frame with a size greater than the maximum valid size.
Add kbdwin.o to AC3 decoder
Detect byte-swapped AC-3 and support decoding it directly.
cosmetics: indentation
Always copy input data for AC3 decoder.
ac3enc: make sym_quant() branch-free
cosmetics: indentation
Add a CPU flag for the Atom processor.
id3v2: skip broken tags with invalid size
id3v2: don't explicitly skip padding
Make sure kbhit() is in conio.h
fate: update wmv8-drm reference
vc1: make P-frame deblock filter bit-exact.
configure: Add the -D parameter to the dlltool command
amr: Set the AVFMT_GENERIC_INDEX flag
amr: Set the pkt->pos field properly to the start of the packet
amr: Set the codec->bit_rate field based on the last packet
rtsp: Specify unicast for TCP interleaved streams, too
Set the correct target for mingw64 dlltool
applehttp: Change the variable for stream position in seconds into int64_t
...
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/ac3dec.c
libavformat/avio.h
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows libavformat to guess an estimated duration for
amr files.
For streams with varying bit rates (or with silence descriptors
or "no frame" blocks) the guess is, of course, inaccurate.
According to the RFC, the default is multicast if nothing is
specified, which doesn't make sense for TCP.
According to a bug report, some Axis camera models give a
"400 Bad Request" error if this is omitted.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
A similar variable for the total stream duration was changed to
int64_t in b79c3df088, due to overflows in some odd
streams.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
It's an evil hack that assumes an AVIOContext is always based on top of
an URLContext.
It's also not used anywhere.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Turns a comment into an av_dlog() instruction, also add a commented
issues.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 77f21ce464)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
Functions interrupted by url_interrupt_cb should not be restarted.
Therefore using AVERROR(EINTR) was wrong, as it did not allow to distinguish
when the underlying system call was interrupted and actually needed to be
restarted.
This fixes roundup issues 2657 and 2659 (ffplay not exiting for streamed
content).
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
In the event of overflow, the JV_PADDING state will avio_skip over
any overflow bytes (using JVFrame.total_size).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
In the main loop, stream_number is incremented after checking the stream type,
so the search usually will not find the wanted stream.
This patch eliminates the useless stream_number variable and introduces a new
one, called real_stream_index to store the real stream index of the current
stream, no matter if we are looping through all the streams or only the streams
of a program.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is used at least on some older DVB broadcasts for dual-mono audio
tracks.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 6a7e074eb9)
when writing and pressing q during encoding. Instead, check url_interrupt_cb
at the end.
Note that when a protocol is interrupted by url_interrupt_cb, some data may
be silently discarded: the protocol context is not suitable for anything
anymore.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 688c22e033)
AVIOContext.max_packet_size should be used directly instead.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit e8bb2e2439)
It's not used anywhere and doesn't look ver useful to be public.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 655e45e7df)
If this flag is set, the protocol can handle URLs where the
scheme is a nested scheme such as applehttp+file: - the protocol
can handle any URL where the first segment of the nested scheme
belongs to this protocol.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit 8f73c06077)
when writing and pressing q during encoding. Instead, check url_interrupt_cb
at the end.
Note that when a protocol is interrupted by url_interrupt_cb, some data may
be silently discarded: the protocol context is not suitable for anything
anymore.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This is a substitute for the url_fskip function that was deprecated by
commit 0300db8ad7. avio_fskip is provided to
improve demuxer code readability. It distinguishes the act of skipping over
unknown or irrelevant bytes from the standard avio_seek operation.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
If this flag is set, the protocol can handle URLs where the
scheme is a nested scheme such as applehttp+file: - the protocol
can handle any URL where the first segment of the nested scheme
belongs to this protocol.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This removes a fixme issue, by allowing the av_pkt_dump functions
to use the correct time base.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit 863c471638)
In most cases, s->buf_ptr will be equal to s->buf_end when
fill_buffer is called, but this may not always be the case, if
we're seeking forward by reading (permitted by the short seek
threshold).
If fill_buffer is writing to s->buf_ptr instead of s->buf_end (when
they aren't equal and s->buf_ptr is ahead of s->buffer), the data
between s->buf_ptr and s->buf_end is overwritten, leading to
inconsistent buffer content. This could return incorrect data if
later seeking back into the area before the current s->buf_ptr.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit e360ada2d1)
In most cases, s->buf_ptr will be equal to s->buf_end when
fill_buffer is called, but this may not always be the case, if
we're seeking forward by reading (permitted by the short seek
threshold).
If fill_buffer is writing to s->buf_ptr instead of s->buf_end (when
they aren't equal and s->buf_ptr is ahead of s->buffer), the data
between s->buf_ptr and s->buf_end is overwritten, leading to
inconsistent buffer content. This could return incorrect data if
later seeking back into the area before the current s->buf_ptr.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Map EAGAIN and EINTR from ff_neterrno to the normal AVERROR()
error codes. Provide fallback definitions of other errno.h network
errors, mapping them to the corresponding winsock errors.
This eases catching these error codes in common code, without having
to distinguish between FF_NETERRNO(EAGAIN) and AVERROR(EAGAIN).
This fixes roundup issue 2614, unbreaking blocking network IO on
windows.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 28c4741a66)
Also change the comments a bit since the FOURCCs aren't specific to Flip4Mac
and different ones are used for 720 versus 1080 lines.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 8f935b9271)
Map EAGAIN and EINTR from ff_neterrno to the normal AVERROR()
error codes. Provide fallback definitions of other errno.h network
errors, mapping them to the corresponding winsock errors.
This eases catching these error codes in common code, without having
to distinguish between FF_NETERRNO(EAGAIN) and AVERROR(EAGAIN).
This fixes roundup issue 2614, unbreaking blocking network IO on
windows.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Also change the comments a bit since the FOURCCs aren't specific to Flip4Mac
and different ones are used for 720 versus 1080 lines.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This adds the AAC codec to the list of audio codecs that results
in a PES stream_id of 0xc0 (audio stream).
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 6c065f0acd)
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>
(cherry picked from commit 77eb5504d3)
In the name of consistency:
get_byte -> avio_r8
get_<type> -> avio_r<type>
get_buffer -> avio_read
get_partial_buffer will be made private later
get_strz is left out becase I want to change it later to return
something useful.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit b7effd4e83)
This adds the AAC codec to the list of audio codecs that results
in a PES stream_id of 0xc0 (audio stream).
Signed-off-by: Mans Rullgard <mans@mansr.com>
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>
In the name of consistency:
get_byte -> avio_r8
get_<type> -> avio_r<type>
get_buffer -> avio_read
get_partial_buffer will be made private later
get_strz is left out becase I want to change it later to return
something useful.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Update libavformat/version.h and doc/APIChanges after renaming
init_put_byte() and ByteIOContext to ffio_init_context() (private)
and AVIOContext, (public), and deprecating the originals.
(cherry picked from commit d2bbf82e65)
init_put_byte should never be used outside of lavf, since
sizeof(AVIOContext) isn't part of public ABI.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit e731b8d872)
Allows playback of nonprimary audio streams in multiple bitrate sources,
such as mmsh://wmscr1.dr.dk/e02ch03m
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 74d6871d62)
Update libavformat/version.h and doc/APIChanges after renaming
init_put_byte() and ByteIOContext to ffio_init_context() (private)
and AVIOContext, (public), and deprecating the originals.
init_put_byte should never be used outside of lavf, since
sizeof(AVIOContext) isn't part of public ABI.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Allows playback of nonprimary audio streams in multiple bitrate sources,
such as mmsh://wmscr1.dr.dk/e02ch03m
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The new av_parse_time() is created in libavutil/parseutils.h, all the
internal functions used by parse_date are moved to
libavutil/parseutils.c and made static.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit f6c7375a17)
If udp_read_packet returns 0, rtsp_st isn't set and we shouldn't
treat it as a successfully received packet (which is counted and
possibly triggers a RTCP receiver report).
This fixes issue 2612.
(cherry picked from commit 2c35a6bde9)
The new av_parse_time() is created in libavutil/parseutils.h, all the
internal functions used by parse_date are moved to
libavutil/parseutils.c and made static.
Signed-off-by: Mans Rullgard <mans@mansr.com>
If udp_read_packet returns 0, rtsp_st isn't set and we shouldn't
treat it as a successfully received packet (which is counted and
possibly triggers a RTCP receiver report).
This fixes issue 2612.
The current implementation has a bug, it is returning the stream index
in the found program, and not the stream index in the list of all
streams. The attached patch fixes this issue.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 22ec6b738f)
Our poll implementation does not iterate over the pollfd array properly
while setting the revents.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit 9ac2085dbf)
The current implementation has a bug, it is returning the stream index
in the found program, and not the stream index in the list of all
streams. The attached patch fixes this issue.
Signed-off-by: Mans Rullgard <mans@mansr.com>
It is pretty hopeless that other considerable projects will adopt
libavutil alone in other projects. Projects that need small footprint
are better off with more specialized libraries such as gnulib or rather
just copy the necessary parts that they need. With this in mind, nobody
is helped by having libavutil and libavcore split. In order to ease
maintenance inside and around FFmpeg and to reduce confusion where to
put common code, avcore's functionality is merged (back) to avutil.
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
Fixes errors after a few minutes (first ping) when playing back
mmst://wm.bbc.co.uk/wms/bbc7coyopa/bbc7_-_friday_0430.wma
(cherry picked from commit 275189a2bd)
There is a check for HAVE_BIGENDIAN when outputting the IEC 61937
stream. On big-endian systems the payload data is not byteswapped,
causing in effect the outputted payload data to be in a different byte
order on big-endian than on little-endian systems.
However, the IEC 61937 preamble (and the final odd byte if present) is
always outputted in the same byte order. This means that on big-endian
systems the headers have a different byte order than the payload,
preventing useful use of the output.
Fix that by outputting the data in a format suitable for sending to an
audio device in S16LE format by default. Output as big-endian (S16BE)
is added as an AVOption. This makes the muxer output the same on all
archs by default.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 57f2c9aed9)
There is a check for HAVE_BIGENDIAN when outputting the IEC 61937
stream. On big-endian systems the payload data is not byteswapped,
causing in effect the outputted payload data to be in a different byte
order on big-endian than on little-endian systems.
However, the IEC 61937 preamble (and the final odd byte if present) is
always outputted in the same byte order. This means that on big-endian
systems the headers have a different byte order than the payload,
preventing useful use of the output.
Fix that by outputting the data in a format suitable for sending to an
audio device in S16LE format by default. Output as big-endian (S16BE)
is added as an AVOption. This makes the muxer output the same on all
archs by default.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
This is used for mapping AVStreams back to their corresponding
RTSPStream. Since d9c0510, the RTSPStream pointer isn't stored in
AVStream->priv_data any longer, breaking this mapping from AVStreams
to RTSPStreams.
Also, we don't need to clear the priv_data in rdt cleanup any longer,
since it isn't set to duplicate pointers.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit b2dd842d21)
it's not touched anywhere in ffmpeg, the code setting it was removed
over two years ago (e9b78eeba2).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit b0294c80d3)
This is used for mapping AVStreams back to their corresponding
RTSPStream. Since d9c0510, the RTSPStream pointer isn't stored in
AVStream->priv_data any longer, breaking this mapping from AVStreams
to RTSPStreams.
Also, we don't need to clear the priv_data in rdt cleanup any longer,
since it isn't set to duplicate pointers.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
it's not touched anywhere in ffmpeg, the code setting it was removed
over two years ago (e9b78eeba2).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This code will be later split out into a function which takes a 'size'
argument, so I'm keeping the name 'sizeX' here.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 0b1d291a71)
Only trivial splits are done here -- i.e. copy/paste + reindent +
missing variable declarations.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit c1fea23070)
Keep the original corner case behaviour, where reuse is enabled
for the case where no argument is given to the reuse url option.
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 00952be424)
This code will be later split out into a function which takes a 'size'
argument, so I'm keeping the name 'sizeX' here.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Keep the original corner case behaviour, where reuse is enabled
for the case where no argument is given to the reuse url option.
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
It is useful for applications that hand input data directly to lavf via
a ByteIOContext.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 3940caad02)
There is no need to pass the ByteIOContext via a pointer to a pointer
anymore.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit aad216fd7e)
Currently (since the data_offset fix) the ogg demuxer assumes that
after the first non-header packets in any stream no more header packets
will follow.
This is not guaranteed, so change the code back again to wait until it
has finished the headers for all streams before returning from ogg_get_headers.
This fixes issue 2428.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 6bd69e6ada)
This validate the length of a mkv element directly after reading
it.
This has the advantage that it is easy to add new limits and makes
it less likely to forget to add checks and also avoids issues like
bits of the length value above the first 32 being ignored because
the parsing functions only takes an int.
Previously discussed in the "mkv 0-byte integer parsing" thread.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 95ec3d4cac)
Change int64_t into a int, which caused this compiler warning:
libavformat/oggparseskeleton.c:64: warning: passing argument 2 of ‘av_reduce’ from incompatible pointer type
(cherry picked from commit 69ff149204)
Use avio functions instead of bytestream ones (also drops dependency on
lavc and removes a bunch of warnings).
Drop custom version of avio_get_str16 and use that instead.
Tested on mewmew-ssa.avi sample.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 47fdf00a77)
Makes playing QDMC files in MPlayer work when using the libavformat demuxer.
Problem was that the extradata was not passed from demuxer to decoder.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit ed19fafd48)
Currently (since the data_offset fix) the ogg demuxer assumes that
after the first non-header packets in any stream no more header packets
will follow.
This is not guaranteed, so change the code back again to wait until it
has finished the headers for all streams before returning from ogg_get_headers.
This fixes issue 2428.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This validate the length of a mkv element directly after reading
it.
This has the advantage that it is easy to add new limits and makes
it less likely to forget to add checks and also avoids issues like
bits of the length value above the first 32 being ignored because
the parsing functions only takes an int.
Previously discussed in the "mkv 0-byte integer parsing" thread.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Change int64_t into a int, which caused this compiler warning:
libavformat/oggparseskeleton.c:64: warning: passing argument 2 of ‘av_reduce’ from incompatible pointer type
Use avio functions instead of bytestream ones (also drops dependency on
lavc and removes a bunch of warnings).
Drop custom version of avio_get_str16 and use that instead.
Tested on mewmew-ssa.avi sample.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Makes playing QDMC files in MPlayer work when using the libavformat demuxer.
Problem was that the extradata was not passed from demuxer to decoder.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit ad3cffb68f)
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 90441276e4)
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit fe174fc8fc)
If required, the caller can do this itself. ff_write_chained rescales
timestamps as necessary, and all current callers of rtpenc_chain
use ff_write_chained, making this timebase copy unnecessary.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 397ffde115)
This function is useful for freeing data structures allocated by
muxers, which currently have to be freed manually by the caller.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit f124b087ee)
This avoids having the chained AVStream->codec point to the same
AVCodecContext owned by the outer AVStream. The downside is that
changes to the AVCodecContext made after calling av_write_header
cannot be detected automatically within the chained muxer.
This avoids having to manually unlink the chained AVStream->codec
by setting it to null before freeing the chained muxer via generic
freeing functions.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 1338dc0823)
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 51b317d2e9)
If required, the caller can do this itself. ff_write_chained rescales
timestamps as necessary, and all current callers of rtpenc_chain
use ff_write_chained, making this timebase copy unnecessary.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This function is useful for freeing data structures allocated by
muxers, which currently have to be freed manually by the caller.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This avoids having the chained AVStream->codec point to the same
AVCodecContext owned by the outer AVStream. The downside is that
changes to the AVCodecContext made after calling av_write_header
cannot be detected automatically within the chained muxer.
This avoids having to manually unlink the chained AVStream->codec
by setting it to null before freeing the chained muxer via generic
freeing functions.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Now the first argument is URLContext *h. However, the function logs to
LOG_CONTEXT, which is #defined as 's' for new lavf major versions.
Therefore, rename h -> s.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 9ad4c65f6f)
This fixes memory leaks in the RTSP muxer and RTP hinting in the
mov muxer present since SVN rev 25418.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit ce41c51b0c)
For mpegts in RTP, there isn't a direct mapping between RTSPStreams
and AVStreams, and the RTSPStream isn't ever stored in
AVStream->priv_data, which was earlier leaked. The fix for this
leak, in ea7f080749, lead to
double frees for other, normal RTP streams.
This patch avoids storing RTSPStreams in AVStream->priv_data, thus
avoiding the double free. The RTSPStreams are always available via
RTSPState->rtsp_streams anyway.
Tested with MS-RTSP, RealRTSP, DSS and mpegts/RTP.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit d9c0510e22)
The bumps are for adding version.h and avio_{get/put}_str functions in
lavf and making av_dlog public in lavu.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
Now the first argument is URLContext *h. However, the function logs to
LOG_CONTEXT, which is #defined as 's' for new lavf major versions.
Therefore, rename h -> s.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This fixes memory leaks in the RTSP muxer and RTP hinting in the
mov muxer present since SVN rev 25418.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
For mpegts in RTP, there isn't a direct mapping between RTSPStreams
and AVStreams, and the RTSPStream isn't ever stored in
AVStream->priv_data, which was earlier leaked. The fix for this
leak, in ea7f080749, lead to
double frees for other, normal RTP streams.
This patch avoids storing RTSPStreams in AVStream->priv_data, thus
avoiding the double free. The RTSPStreams are always available via
RTSPState->rtsp_streams anyway.
Tested with MS-RTSP, RealRTSP, DSS and mpegts/RTP.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The bumps are for adding version.h and avio_{get/put}_str functions in
lavf and making av_dlog public in lavu.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
The first part of the metadata, the "vendor" string, is required by
libvorbis, it will refuse to play when it is not available.
Also we do not currently parse that part into metadata so it would also
be lost if we removed it as well.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 8cb3c557a9)
The first part of the metadata, the "vendor" string, is required by
libvorbis, it will refuse to play when it is not available.
Also we do not currently parse that part into metadata so it would also
be lost if we removed it as well.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Around 01/28/11 18:56, Ronald S. Bultje scribbled:
> That patch is now merged, can you submit the update to muxers.texi?
> Then we'll apply the whole thing.
See attached. I hope the documentation is enough.
--
Georgi Chorbadzhiyski
http://georgi.unixsol.org/
From c236024b8254f5c2c45934c30fff390cb0e55a5e Mon Sep 17 00:00:00 2001
From: Georgi Chorbadzhiyski <gf@unixsol.org>
Date: Tue, 25 Jan 2011 13:09:17 +0200
Subject: [PATCH] mpegts: Replace defines in with AVOptions
This patch adds support for setting transport_stream_id,
original_network_id, service_id, pmt_start_pid and start_pid
in mpegts muxer.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 445996aa51)
Around 01/28/11 18:56, Ronald S. Bultje scribbled:
> That patch is now merged, can you submit the update to muxers.texi?
> Then we'll apply the whole thing.
See attached. I hope the documentation is enough.
--
Georgi Chorbadzhiyski
http://georgi.unixsol.org/
From c236024b8254f5c2c45934c30fff390cb0e55a5e Mon Sep 17 00:00:00 2001
From: Georgi Chorbadzhiyski <gf@unixsol.org>
Date: Tue, 25 Jan 2011 13:09:17 +0200
Subject: [PATCH] mpegts: Replace defines in with AVOptions
This patch adds support for setting transport_stream_id,
original_network_id, service_id, pmt_start_pid and start_pid
in mpegts muxer.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
$subject. Have used this for loopback testing with mpegts.c.
-- Peter
(A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
[2. text/x-diff; 0001-mpegtsenc-support-CODEC_ID_AAC_LATM.patch]
From 0f7f9db4b7da1793996af6dda84298507703759a Mon Sep 17 00:00:00 2001
From: Peter Ross <pross@xvid.org>
Date: Sun, 9 Jan 2011 09:45:50 +1100
Subject: [PATCH] mpegtsenc: support CODEC_ID_AAC_LATM
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 4d54df8e07)
poll() is only used by networking code, so the fallback should
only be built if networking is enabled. Also remove CONFIG_FFSERVER
condition from the declarations.
This should fix building on systems without poll(), broken
by a8475bbdb6.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 362d8f7d9e)
Add an error message in case the user requests to write more than one file
and the path does not contain a "%d" or "%0Nd" pattern.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 4fc9ff0ad6)
According to EN 300 468 section 3.1 (Definitions):
Unless otherwise specified within the present document all
"reserved_future_use" bits is set to "1".
This was not the case for SDT generation so this patch fixes it.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit a7827a17c6)
Avoids an assert when the sample rate is invalid and the timebase
is thus set to e.g. 1/0.
Sample file is http://samples.mplayerhq.hu/ogg/fuzzed-srate-crash.ogg
This is a quick fix for a crash, not a final solution.
Signed-off-by: Mans Rullgard <mans@mansr.com>
