The Windows SDK in MSVC doesn't have mkdir, only _mkdir, and
MSDN says one should include direct.h to use it.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes building with DEBUG defined after the function was made
static and the prototype removed in d7f9786cbc.
Signed-off-by: Martin Storsjö <martin@martin.st>
This also makes sure the function pointers are stored via the same
union member as they are accessed via.
Signed-off-by: Martin Storsjö <martin@martin.st>
This C99 feature is generally not used in the libav codebase,
since it breaks building with some fringe compilers (GCC 2.95,
MSVC).
Signed-off-by: Martin Storsjö <martin@martin.st>
The way this bit is decoded was accidentally flipped in b70feb405,
leading to warnings "Encountered a bad or corrupted frame" for each
decoded frame.
Signed-off-by: Martin Storsjö <martin@martin.st>
Not all compilers support ssize_t (MSVC doesn't), and none of these
variables need to be larger than 32 bit.
Signed-off-by: Martin Storsjö <martin@martin.st>
These are normally initialized to AV_NOPTS_VALUE at the start
of avformat_find_stream_info, but if a new stream is found while
this function is running (e.g. like in mpegts), the newly added
AVStreams didn't have these values properly initalized, leading
to avformat_find_stream_info terminating too soon (when the
first timestamps are far from 0).
Signed-off-by: Martin Storsjö <martin@martin.st>
At this place, the normal way of initializing a struct works
fine, there's no need for a struct literal.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously, we returned any error code except AVERROR_EOF to the
caller - only if AVERROR_EOF or 0 was returned, we proceeded to
the next segment.
With some setups of web servers, using Connection: close in https
and GnuTLS, we don't get a clean error code at the end of segments.
In those cases, just proceed to the next segment.
Tested-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
OpenSSL returns 0 when the peer has closed the connection. GnuTLS
doesn't return that though, but returns
GNUTLS_E_UNEXPECTED_PACKET_LENGTH if the connection simply is closed
without a clean close notify packet.
Tested-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
The rtmpts protocol uses https implicitly, via the ffrtmphttp
protocol, but the ffrtmphttp protocol is also useable for plain
rtmpt without https, so the dependency needs to be added here instead.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is only used for checking for a certain library, but the code
doesn't need to know whether the function was found.
Signed-off-by: Martin Storsjö <martin@martin.st>
Even if linking directly to getaddrinfo, use our version of
gai_strerror instead of the system's version. Microsoft explicitly
documents that their version of gai_strerror is thread-unsafe.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids warnings if there already exists a definition.
This is the case on windows, where the getaddrinfo isn't available
and linked to (and we use our fallbacks instead, which actually
try to use the proper getaddrinfo version if found at runtime),
but gai_strerror still exists as a define.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is useful if a proper getaddrinfo is loaded dynamically on
windows, while using the fallback implementation of gai_strerror.
Signed-off-by: Martin Storsjö <martin@martin.st>
This was unnoticed on linux, since stdlib.h apparently includes
files declaring the pthread_mutex_t and pthread_cond_t types.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The SPLATB_REG macro already adds the 'd' suffix internally.
This fixes building on Win64, which has been broken since 878e66902.
This worked for unix, where r2 happened to be rdx in this case, which
with the first suffix rdxd was mapped to eax, and eaxd is defined back
to eax. On win64 however, r2 happened to be R8 in this case, and
R8d mapps to R8D just fine, but there's no mapping for R8Dd to anything.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids creating new AVStreams for them when switching between
different variants of them, since we can handle changes between
different sample rates of nellymoser within the same stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
If the output frame size is smaller than the input sample rate,
and the input stream time base corresponds exactly to the input
frame size (getting input packet timestamps like 0, 1, 2, 3, 4 etc),
the output timestamps from the filter will be like
0, 1, 2, 3, 4, 4, 5 ..., leadning to non-monotone timestamps later.
A concrete example is input mp3 data having frame sizes of 1152
samples, transcoded to aac with 1024 sample frames.
By setting the audio filter time base to the sample rate, we will
get sensible timestamps for all output packets, regardless of
the ratio between the input and output frame sizes.
Signed-off-by: Martin Storsjö <martin@martin.st>
For reading from normal files on disk, the queue limits for
demuxed data work fine, but for reading data from realtime
streams, they mean we're not reading from the input stream
at all once the queue limit has been reached. For TCP streams,
this means that writing to the socket from the peer side blocks
(potentially leading to the peer dropping data), and for UDP
streams it means that our kernel might drop data.
For some protocols/servers, the server initially sends a
large burst with data to fill client side buffers, but once
filled, we should keep reading to avoid dropping data.
For all realtime streams, it IMO makes sense to just buffer
as much as we get (rather in buffers in avplay.c than in
OS level buffers). With this option set, the input thread
should always be blocking waiting for more input data,
never sleeping waiting for the decoder to consume data.
Signed-off-by: Martin Storsjö <martin@martin.st>
In Smooth Streaming, the fragments are addressed by time, and
the manifest only stores one list of time offests for all streams,
so all streams need to have identical fragment offsets. Warn if
this isn't the case, so that the user can fix the files instead of
getting failures at runtime when the fragments can't be found.
Signed-off-by: Martin Storsjö <martin@martin.st>
These files use NAN/INFINITY but didn't include mathematics.h to get
the fallback definitions if the system lacks the macros.
Signed-off-by: Martin Storsjö <martin@martin.st>
ff_wma_init is used only by wmadec and wmaenc, and neither of them
can handle more than 2 channels.
This fixes crashes with invalid files.
Based on patch by Piotr Bandurski and Michael Niedermayer.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows passing the right options to encoders when there's more
than one encoder for a certain codec id.
Signed-off-by: Martin Storsjö <martin@martin.st>
The fallback function is a non-static function, we shouldn't be
defining non-static functions outside of the proper ff/av prefix
namespaces.
This is especially important for a function like poll, which
other parties (other libraries, or executables linking these
libraries) also might provide similar but incompatible fallbacks for.
Signed-off-by: Martin Storsjö <martin@martin.st>
We need to include winsock2.h here, to make sure we have the
real pollfd struct definition, if one exists, before defining the
fallback poll function.
Signed-off-by: Martin Storsjö <martin@martin.st>
The fds are unsigned integers in the windows definition of struct
sockfds. Due to this, the comparison if (fds[i].fd > n) was always
false.
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>
MSVC has isatty (in io.h), but not unistd.h. (isatty isn't called
at all for windows, since there's a special case block for that.)
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
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 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>
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 fixes issues with opening http urls that have authentication
or redirects, introduced in commit e999b641.
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>
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>
Martin Storsjö
