Since the errno.h values don't match the error codes that winsock
returns, map the winsock error codes to the errno ones, to make
sure explicit checks against AVERROR(x) match.
Signed-off-by: Martin Storsjö <martin@martin.st>
Apparently this include is needed on some systems for building the
poll fallback (for the timeval struct for select?), but it isn't
available on all systems. Thus only include it if it exists.
Signed-off-by: Martin Storsjö <martin@martin.st>
This heaader is required for close() for sockets in network
code. For winsock, the equivalent function is defined in the
winsock2.h header.
This avoids having the HAVE_UNISTD_H in all files dealing with
raw sockets.
Signed-off-by: Martin Storsjö <martin@martin.st>
On MSVC, gmtime returns NULL for values outside of their supported
range (and these show up in our fate test). This doesn't seem
to affect the actual fate test result.
Signed-off-by: Martin Storsjö <martin@martin.st>
Currently if a pattern is given we look for up to the fifth file name in
the sequence. This option sets that limit to an arbitrary number.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This adds the capability to start counting file number from an arbitrary
integer.
This includes a few lines of trivial code from FFmpeg codebase.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Also use ff_neterrno() instead of errno directly (which doesn't work
on windows), for getting the error code.
Signed-off-by: Martin Storsjö <martin@martin.st>
getnameinfo doesn't set errno on failure, it returns an error code,
which should be handled by gai_strerror instead of the normal
strerror.
Signed-off-by: Martin Storsjö <martin@martin.st>
Rtmpt is effectively half duplex - the server can't return any
data unless we send a request (to which the server responds). If
we don't have any data to send currently, and the server didn't
return any data either, wait a little before doing the next request.
This avoids busy looping with idle posts with empty replies, while
waiting for more data from the server.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add a new option 'rtmp_flush_interval' that allows specifying the
number of packets to write before sending it off as a HTTP request.
This is mostly relevant for RTMPT - for plain RTMP, it only controls
how often we check the socket for incoming packets, which shouldn't
affect the performance in any noticeable way.
Signed-off-by: Martin Storsjö <martin@martin.st>
This seems to be the correct mode to send, according to the
original RTSP RFC, and matches the method RECORD which is
sent later when starting to send data.
Darwin Streaming Server works fine with either of them.
Signed-off-by: Martin Storsjö <martin@martin.st>
This requires all NAL units to fit within single RTP packets. It
doesn't change the actual packetization for packets that fit, but
errors out and gives a helpful hint if the NAL units would have to
be split, and signals the right packetization mode in the SDP.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
This can happen if doing a new request using the same socket,
but the new request failed, which clears the urlcontext.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add spaces around operators, fix brace placement and whitespace to
match K&R style, vertically align code, remove redundant != 0 and
convert x == 0 into !x, drop useless braces.
Signed-off-by: Martin Storsjö <martin@martin.st>
This can easily happen when the caller is using a custom AVIOContext.
Behave as if the filename was an empty string in this case.
CC: libav-stable@libav.org
This makes sure all incoming packets are read and handled (and reacted
to) while sending an FLV stream over RTMP to a server. If there were
enough incoming data to fill the TCP buffers, this could potentially
make things block at unexpected places. For the upcoming RTMPT support,
we need to consume all incoming data before we can send the next
request.
Signed-off-by: Martin Storsjö <martin@martin.st>
While there is no reason for starting a frame with anything else
than a Mode A packet, some senders seem to consistently use Mode B
packets for everything. This fixes depacketization of such streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
This factorizes existing code into a new function gen_buffer_time(),
which generates the client buffer time message and sends it to the
server.
Signed-off-by: Martin Storsjö <martin@martin.st>
Allow using connection parameters in order to append arbitrary
AMF data like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0" to the
Connect message. You can pass these parameters through the -rtmp_conn
option.
Signed-off-by: Martin Storsjö <martin@martin.st>
This adds the avoption mpegts_flags and converts the existing
resend_headers option into a flag, keeping the old option as
fallback for now.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This isn't required any longer, when the mpegts muxer uses it
as a proper chained muxer.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This removes the dependency on adts.c internals, and simplifies
adding other packetization formats.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes issues with opening http urls that have authentication
or redirects, introduced in commit e999b641.
Signed-off-by: Martin Storsjö <martin@martin.st>
avisynth is a non-unicode application and cannot accept UTF-8
characters. Therefore, the input filename should be converted to
the correct code page that it expects.
Signed-off-by: Martin Storsjö <martin@martin.st>
Introduce ff_http_do_new_request(), a new function which sends a new
HTTP request, reusing the existing connection to the server.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add a new AVOption 'multiple_requests', which indicates if we want
to use persistent connections (ie. Connection: keep-alive).
Signed-off-by: Martin Storsjö <martin@martin.st>
Some systems abuse the static payload types 35 or 36 (which
according to IANA are unassigned) for H264.
Signed-off-by: Martin Storsjö <martin@martin.st>
If an URLContext is passed in, its ownership is given to this
function, and is either owned by the returned AVFormatContext
on a successful return, or freed on failure.
Signed-off-by: Martin Storsjö <martin@martin.st>
The sample_rate variable is used for checks for audio format
changes at the end of the function.
This fixes cases where the sample rate was set from the codec
id by flv_set_audio_codec (as for nellymoser 8 kHz/16 kHz),
so the value set to last_sample_rate wasn't equal to sample_rate
at this point. This caused the demuxer otherwise reports a spurious
change to 5512 Hz and back to the correct one.
Updating channels in the same way is only done for consistency.
Currently, flv_set_audio_codec doesn't update that value.
Signed-off-by: Martin Storsjö <martin@martin.st>
tcp_shutdown() isn't needed at the moment, but is added for
consistency to explain how the function is supposed to be used.
Signed-off-by: Martin Storsjö <martin@martin.st>
If using the new -rtmp_app and -rtmp_playpath parameters,
one can in many cases set the main url to just rtmp://server/.
If the trailing slash is omitted, path is a string of zero length,
and using path+1 will end up reading uninitialized data.
Signed-off-by: Martin Storsjö <martin@martin.st>
According to the behaviour of librtmp, it is recommended to send this
message to the server after receiving the 'onBWDone' callback in order
to do bandwidth checking and improve compatibility with some servers.
The strtol() interface makes it difficult to use with
const-qualified pointers. With this change, although
the const is still lost, the compiler does not warn
about it.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Add/fix spacing, split long lines, align assignments where suitable.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Signed-off-by: Martin Storsjö <martin@martin.st>
Split long comments, move long comments at the end of lines to
separate lines above, fix vertical alignment, fix up comment style
(unify trailing dots - comments had a mix of 2, 3 or 4 dots, where
it would be just as good without them at all).
Signed-off-by: Martin Storsjö <martin@martin.st>
It is worth keeping instead of removing, in case reading this
bit becomes necessary at some later point.
Signed-off-by: Martin Storsjö <martin@martin.st>
Skip to parse fields for additional independent substreams and its
associated dependent substreams since libavcodec's E-AC-3 decoder does not
support them yet.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
This fixes crashes, where the demuxer could return 0 even
if the returned AVPacket isn't initialized at all. This
could happen if running into EOF or running out of probesize
with non-seekable sources.
Signed-off-by: Martin Storsjö <martin@martin.st>
The new incremental parser doesn't always clear prev_pkt,
however the packet queue is cleared when seeking. Which leads
to a use-after-free.
Verified using Valgrind.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Reduces the amount of upfront data required for cluster parsing
thus decreasing latency on seek and startup.
The change in the seek-lavf_mkv FATE test is due to incremental
parsing no longer reading as much data as the old parser and
thus not having that additional data to generate index entries
based on keyframes. Index entries are added correctly as the
file is parsed.
All FATE tests pass and Chrome has been using this patch for ~6
months without issue.
Currently incremental parsing is not supported for files with
SSA tracks since they require merging packets between clusters.
In this case the code falls back to non-incremental parsing.
Signed-off-by: Aaron Colwell <acolwell@chromium.org>
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The mpegts demuxer reads 5 KB at startup just for discovering
the packet size. Since the default avio buffer size is 32 KB,
the seek back to the start will in most cases be within the
avio buffer, and will in most cases succeed even if the actual
protocol isn't seekable.
This makes the demuxer startup faster/with less data when
reading data from a non-seekable input, by not skipping
the first few KB.
If it fails, don't warn if the protocol isn't seekable, making
it behave as before in the failure case.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fix this warning:
libavformat/aviobuf.c:663:20: warning: assignment discards qualifiers from pointer target type
Although this is a public header, it should remain source and
binary compatible.
Signed-off-by: Mans Rullgard <mans@mansr.com>
If a video track specifies a zero frame rate (invalid but occurs),
this results in a division by zero and subsequent undefined conversion
to integer. Setting the default duration from the frame rate only
if the latter is greater than zero avoids such problems.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This option is the stream identifier to play or to publish.
Sometimes the URL parser cannot determine the correct
playpath automatically, so it must be given explicitly
using this option (ie. -rtmp_playpath).
Signed-off-by: Martin Storsjö <martin@martin.st>
This option is the name of application to connect on the RTMP server.
Sometimes the URL parser cannot determine the app name automatically,
so it must be given explicitly using this option (ie. -rtmp_app).
Signed-off-by: Martin Storsjö <martin@martin.st>
Instead of allocating over the original, free first. MOVStreamContext
is zero initialized so no double free will occur. Same style as other
fixes for the same problem in this file.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
During error conditions matroska_parse_block may exit without
freeing the memory allocated for laces.
Found via valgrind: http://pastebin.com/E54k8QFU
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Share the formerly internal write_packet with the hinter and move the
fragment flush logic to the user facing one since it is not concerned
about movtrack-only streams.
Fixes bug #263
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids exposing a dummy AVStream which won't get any data
and which will make avformat_find_stream_info wait for info about
this stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
Searching for packet markers doesn't make sense for this use case,
where packets are fed one at a time to the demuxer.
This fixes playing back streams that have packets not starting
with the 0x82, 0x00, 0x00 marker.
Signed-off-by: Martin Storsjö <martin@martin.st>
It can take a long time before subtitles or data streams show up,
so we shouldn't wait for those before assuming we have all info
for streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also add missing trailing commas, break long codec_tag lines and
add spaces in codec_tag declarations.
Signed-off-by: Martin Storsjö <martin@martin.st>
The audio codecs property is composed by all values except
SUPPORT_SND_INTEL (0x0008) and SUPPORT_SND_UNUSED (0x0010) which are
unused.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The previous condition of 0 page size was wrong,
that would disable the mechanism for all frames at
a start of a page, thus some keyframes still would not
get their own granule.
The real problem is that header packets must not be flushed,
but they have (and must have) 0 granule and thus would
be detected as keyframes.
Add a separate parameter to mark header packets.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
When set, if an Ogg stream buffer has enough data, a page is made
instead of filling maximum-size pages. Using smaller pages results
smaller seek intervals at the expense of higher container overhead.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Version from vqa header does not dictate which sound chunks may
appear in file.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
This patch allows the user to force flushing of all queued packets
by calling av_interleaved_write_frame() with pkt set to NULL.
Signed-off-by: Jindrich Makovicka <jindrich.makovicka@nangu.tv>
Signed-off-by: Martin Storsjö <martin@martin.st>
This isn't exactly equivalent with the earlier code for codecs
other than H264 and VC1, but those are two only codecs supported
by this codepath anyway, and it simplifies it a bit.
Signed-off-by: Martin Storsjö <martin@martin.st>
Takes encoder delay into account by comparing first the coded page
duration with the calculated page duration. Handles last packet duration
if needed, also by comparing coded duration with calculated duration.
Also does better handling of timestamp generation for packets in the
first page for streamed ogg files where the start time is not
necessarily zero.
The other fragmentation options (frag_duration, frag_size and
frag_keyframe) are combined with OR, cutting fragments at the
first of the conditions being fulfilled.
Signed-off-by: Martin Storsjö <martin@martin.st>
Make the muxers/demuxers that use the field handle the default
-1 in the same way as 0.
This allows distinguishing an intentionally set 0 from the default
value where the user hasn't set it.
Signed-off-by: Martin Storsjö <martin@martin.st>
An obscure Japanese lossless video codec, originally intended
for use with a remote desktop application.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
The private option has not been part of any release yet (and
it is only of use in quite rare cases), so just remove it instead
of keeping it with deprecation warnings.
Signed-off-by: Martin Storsjö <martin@martin.st>
This was forgotten in the transition from av_open_input_file to
avformat_open_input, see 603b8bc2a1.
This doesn't change anything for the default case where the
option isn't set, since PROBE_BUF_MAX is 1048576 (which was
used as max probe size earlier) while the default value for
the probesize option is 5000000, which for the probe function
is clipped to PROBE_BUF_MAX anyway.
Signed-off-by: Martin Storsjö <martin@martin.st>
Jordi Ortiz
Anton Khirnov
Ronald S. Bultje
Martin Storsjö
Carl Eugen Hoyos
Mashiat Sarker Shakkhar
Mans Rullgard
Kostya Shishkov
Samuel Pitoiset
Janne Grunau
Alex Converse
Jindřich Makovička
Luca Barbato
Oka Motofumi
Dave Yeo
Diego Biurrun
James Zern
Sean McGovern
Ivan Kovtunov
Justin Ruggles
Joakim Plate
Yusuke Nakamura
Hendrik Leppkes
Dale Curtis
Michael Niedermayer
Asen Lekov
Raffaele Sena
Reimar Döffinger
Andres Gonzalez
Paul B Mahol
Jindrich Makovicka
Derek Buitenhuis
Nicolas George