libopenmpt 0.3 deprecates openmpt_module_create_from_memory() and
provides a replacement function openmpt_module_create_from_memory2().
Detecting libopenmpt 0.3 can be done at build time via the API
version macros provided by libopenmpt. libopenmpt 0.2 did not provide
all required macros, however libopenmpt documents the required #define
shims that can be safely added for libopenmpt 0.2.
Using openmpt_module_create_from_memory2() instead of
openmpt_module_create_from_memory() avoids the deprecation warning
when building ffmpeg with libopenmpt 0.3.
openmpt_module_create_from_memory2() provides more fine-grained error
reporting and in particular allows distinguishing out-of-memory from
input file parsing errors. Return appropriate ffmpeg errors
accordingly.
libopenmpt 0.3 is ABI and API compatible with applications built
against libopenmpt 0.2. Building ffmpeg with libopenmpt 0.2 is still
supported.
Signed-off-by: Jörn Heusipp <osmanx@problemloesungsmaschine.de>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Apple's AIFF protocol clearly states that each chucnk which is odd sized a padding should be added.
In the old version of aiffdec adding of padding was done in `get_meta`. And in case of unknown chunk name it was done in defalut case.
The new version has deleted the padding in default case and added padding adding after the switch.
But the new version didn't removed the padding adding in the `get_meta` function so in some cases padding was added twice which leaded to a bug.
Fixes: sample.aiff
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Since the function used to be exposed for ffserver's sake, it was renamed
to the same name as a system's aton function which caused a collision.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Certain AIFF files encode XA ADPCM compressed audio using a chunk
with the tag `APCM`. Aside from this custom chunk type, they're
otherwise standard AIFF files. I've only observed these files in the
Sega Saturn game Sonic Jam so far.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If the stream was aborted using the libavformat interrupt callback, we
don't want it to log the reconnect warning. (Exiting after logging this
warning worked well, so this is only for avoiding the ugly warning.)
It makes no sense to return an error after the first reconnect, and then
somehow resume the next time it's called. Usually this will lead to
demuxer errors. Make reconnecting block instead, until it has either
successfully reconnected, or given up.
Also make the wait reasonably interruptible. Since there is no mechanism
for this in the API, polling is the best we can do. This behaves roughly
the same as other interruptible network functions in libavformat.
(The original code would work if it returned AVERROR(EAGAIN) or so,
which would make retry_transfer_wrapper() repeat the read call. But I
think having an explicit loop for this is better anyway.)
I also snuck in a fix for reconnect_at_eof. It has to check for
AVERROR_EOF, not 0.
fix CID: 1426991
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
There is no POSIX error code for EOF - recv() signals EOF by simply
returning 0. But libavformat recently changed its conventions and
requires an explicit AVERROR_EOF, or it might get into an endless retry
loop, consuming 100% CPU while doing nothing.
pts_wrap_bits defaults to 33 (like MPEG), that causes valid
timestamps to be unwrapped and become invalid.
Inspired by a patch by Wu Zhiqiang <mymoeyard@gmail.com>.
Encrypted HLS segments have regular http:// urls, but open_input()
actually prefixes them with crypto+ before calling open_url(), so
they end up using the crypto protocol and not the http protocol.
This means invoking ff_http_do_new_request will fail, so we avoid
calling it in the first place. After the earlier http.c commit,
the failure results in a warning printed to the user. In earlier
versions, the failure would cause a segfault.
Signed-off-by: Aman Gupta <aman@tmm1.net>
This is required for AV playout from master.m3u8.
Otherwise master.m3u8 lists only video-only and/or audio-only streams.
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Some http/1.0 implementations, like python's SimpleHTTPServer, can only support one client connection at a time. Making a second request while the first is still connected leads to a deadlock.
This change enables multiple connections for http/1.1 servers only, which need to support keepalive by default and should have no problem with concurrent requests.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Can be used by the api user to figure out what http features the server supports based on the response received.
Signed-off-by: Aman Gupta <aman@tmm1.net>
This makes do_new_request fail early when dealing with a http/1.0 server, avoiding unnecessary "reconnecting" warnings shown to the user.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Fixes compile error when building with network or protocols disabled.
This code would never be reached (because the demuxer fails much earlier on http playlists or segments), so it doesn't matter much what we do here as long as it compiles.
Signed-off-by: Aman Gupta <aman@tmm1.net>
It's completely absurd that libavcodec would care about libavformat
locking, but it was there because the lock manager was in libavcodec.
This is more stright forward. Changes ABI, but we don't require ABI
compatibility currently.
Currently http end of chunk is signalled implicitly in hlsenc_io_open().
This mean playlists http writes would have to wait upto a segment duration to signal end of chunk causing delays.
This patch will fix that problem and improve performance.
AVERROR_EOF is an internal error which means the http socket is no longer
valid for new requests. It informs the caller that a new connection must
be established, and as such does not need to be surfaced to the user as
a warning.
Signed-off-by: Aman Gupta <aman@tmm1.net>
This fixes a deadlock when using the hls demuxer's new http_persistent feature
to stream a youtube live stream over HTTPS. The youtube servers are http/1.1
compliant, but return a "Connecton: close". Before this commit, the demuxer
would attempt to send a new request on the partially shutdown connection and
cause a deadlock in the tls protocol.
Signed-off-by: Aman Gupta <aman@tmm1.net>
This improves network throughput of the hls demuxer by avoiding
the latency introduced by downloading segments one at a time.
The problem is particularly noticable over high-latency network
connections: for instance, if RTT is 250ms, there will a 250ms idle
period between when one segment response is read and the next one
starts.
The obvious solution to this is to use HTTP pipelining, where a
second request can be sent (on the persistent http/1.1 connection)
before the first response is fully read. Unfortunately the way the
http protocol is implemented in avformat makes implementing pipleining
very complex.
Instead, this commit simulates pipelining using two separate persistent
http connections. This has the advantage of working independently of
the http_persistent option, and can be used with http/1.0 servers as
well. The pair of connections is swapped every time a new segment starts
downloading, and a request for the next segment is sent on the secondary
connection right away. This means the second response will be ready and
waiting by the time the current response is fully read.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
This teaches the HLS demuxer to use the HTTP protocols
multiple_requests=1 option, to take advantage of "Connection:
Keep-Alive" when downloading playlists and segments from the HLS server.
With the new option, you can avoid TCP connection and TLS negotiation
overhead, which is particularly beneficial when streaming via a
high-latency internet connection.
Similar to the http_persistent option recently implemented in hlsenc.c
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
This mimics logging that was added in 53e0d5d724 for security
purposes.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This will prevent improper use of ff_http_do_new_request() if the user
tries to send a request for a different host to a previously connected
persistent http/1.1 connection.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Karthick J <kjeyapal@akamai.com>
Before this patch persistent http connections would work only for media segments.
The playlists were still opening a new connection everytime.
This patch extends persistent http connections to playlists as well.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Since close is not called, during http persistent connection,
flush needs to be called so that output is written on time.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
With FCPX 10.4, Apple has expanded the set of colorspace, primaries,
and trc flags officially supported in QuickTime files. The expanded set
matches the codepoints used in ffmpeg and many other specs.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Every bitstream filter behaves as intended now, so there's no need to
wait for the first packet of every stream.
Signed-off-by: James Almer <jamrial@gmail.com>
The current edit unit cannot be reliably determined for the last packet of a
video stream, because we can't query the start offset of the next edit unit
from the index. This caused missing timestamps for the last video packet.
Therefore from now on, we allow setting the PTS even if we are not sure of the
current edit unit if mxf_set_current_edit_unit returned a specific failure, and
the assumed current edit unit is the last.
Fixes last packet timestamp of:
ffprobe -fflags nofillin -show_packets tests/data/lavf/lavf.mxf -select_streams v
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes ticket #6804. All of the ogg header and packet parsers may
return standard AVERROR codes; these return values should not be
treated as success.
Additionally changes oggparsevorbis, to not give up too early
with certain types of poorly muxed files.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
An invalid file may cause huge alloc. Delay expansion of ctts entries
until the number of samples is known in mov_build_index.
Fixes: 23
Found-by: zhao dongzhuo, AD-lab of Venustech
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fix suggested by Luca Barbato.
This was causing spurious EOFs when using -rtsp_transport udp, as
reported in https://bugzilla.libav.org/show_bug.cgi?id=1103
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This is to take full advantage of Common Media Application Format(CMAF).
Now server can generate one content and serve both HLS and DASH players.
Reviewed-by: Steven Liu <lq@onvideo.cn>
On some failure paths, the error code is not correctly set.
Signed-off-by: Pan Bian <bianpan2016@163.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In function mov_read_custom(), it returns 0 on the path that av_malloc()
returns a NULL pointer. 0 indicates success. An error code should be
assigned to ret.
Signed-off-by: Pan Bian <bianpan2016@163.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This patch fixes Bug #6868
Sometimes end_pts is getting initialized to audio stream's
first pts, while the duration is calculated based on video stream's pts.
In this patch the end_pts is initialized with the correct stream's first pts.
Reviewed-by: Steven Liu <lq@onvideo.cn>
Tested-by: beloko
The function avformat_new_stream() returns a NULL pointer on failure.
However, in function rtp_mpegts_write_header(), its return value is not
validated before it is dereferenced. Check the return value against NULL
to avoid potential NULL dereference.
Signed-off-by: Pan Bian <bianpan2016@163.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Prevents out of array accesses. Adressess ticket #6873
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Writes one set of field framing information for progressive streams and
two sets for interlaced streams. Fixes ticket #6383.
Unfortunately the OpenDML v1.02 document is not very specific on what
value to use for start_line when frame data is not coming from a
capturing device, so this is just using 0/1 depending on the field order
as a best-effort guess.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
2LL << (wrap_bits=64 - 1) does not fit in int64_t; change the
code to use a uint64_t (2ULL) and add an av_assert2() to
ensure wrap_bits <= 64.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In the event of ff_mov_read_stsd_entries() failure, sc->stsd_count
is not updated, even if the function allocates extradata memory.
Instead update the sc->stsd_count as entries are parsed so that
mov_read_close() can do the right thing.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Without this there can be multiple memory leaks for unrecognized
ogg streams.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In the DASHContext structure, min_seg_duration is declared as an int,
but the AVOption list claimed it was an INT64. Change the option list
to use the correct size, which should fix some initialization errors
seen on big-endian platforms.
Signed-off-by: James Cowgill <jcowgill@debian.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Potentially fixes:
https://bugs.chromium.org/p/chromium/issues/detail?id=786269#c1
In theory, the crash can be triggered by an invalid stream that has
either tfdt or trun outside of the moof
Reviewed-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Should be unconditionally freed at the end of mov_fix_index() in
case it hasn't been used during the fix up.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Reviewed-by: Sasi Inguva <isasi-at-google.com@ffmpeg.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This can reduce latency and increase throughput, particularly on high
latency networks.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Reviewed-by: Jeyapal, Karthick <kjeyapal@akamai.com>
The avc3 sample entry type is useful for adaptive streaming. It permits
parameter sets to be written inline in the video stream.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
After c2a8f0fcbe this can happen on normal edit lists starting on a B-frame.
Signed-off-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Currently if you use the multiple_requests=1 option and try to
receive a chunked-encoded response, http_buf_read() will hang forever.
After this patch, EOF is emulated once a 0-byte final chunk is
received by setting a new flag. This flag is reset in ff_http_do_new_request(),
which is used to make additional requests on the open socket.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Aman Gupta <aman@tmm1.net>
multiple_requests=1 is also set. Without an EOF to signal the end of
the last chunk, tls_read gets stuck forever trying to read more data
than is available. This occurs with the http protocol reproducibly,
because http.c always reads 4kb at a time, and the last chunk of an
http response is often much smaller.
After this commit, tls_read always returns any buffered plaintext
first before attempting to read more encrypted data off the
underlying tcp socket.
Signed-off-by: Rodger Combs <rodger.combs@gmail.com>
This works as expected on iOS, except for the ca_file feature which
is disabled because SecItemImport is not available.
Signed-off-by: Aman Gupta <aman@tmm1.net>
* commit '55fe72a841ba306370e68e86c88f34b4456aa4dd':
matroskadec: don't warn about unknown spherical medata when none is present
Merged-by: James Almer <jamrial@gmail.com>
* commit 'defe307fb22beca60a632e976ab97e5edd4aee25':
mov: move stsd finalization to an appropriate place
mov: Do not set stsd_count if mov_read_stsd() fails
mov: log and return early on non-positive stsd entry counts
See 8b43ee4054656feb641d
Merged-by: James Almer <jamrial@gmail.com>
It has no reason to be in a public header, even if defined as private.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
* commit '95ce02b35d3d1bb16111031df1d82a6e4d894d36':
rmdec: don't ignore the return value of av_get_packet()
See 65b83ce01b
Merged-by: James Almer <jamrial@gmail.com>
* commit '4d330da006fe48178a4c8047f06270925eaedf63':
os_support: Use HAVE_UWP instead of manually checking WINAPI_FAMILY
d3d11va: Check WINAPI_FAMILY instead of HAVE_LOADLIBRARY
lavf: Remove codec_tag from dashenc and smoothstreamingenc
hevc: Add support for alternative transfer characterics SEI
Merged-by: James Almer <jamrial@gmail.com>
Don't free the buffer allocated in ff_wms_parse_sdp_a_line() after
calling avformat_open_input(), as it may free it and replace it with
another one.
Should fix ticket #6808
Tested-by: Carl Eugen Hoyos <ceffmpeg@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
