This is similar to an existing check for the second-last frame
from 062421e3.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids a potential division by zero.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Some files have the duration set to -1 in the mdhd atom, more
or less legitimately. (We produce such files ourselves, for the
initial duration in fragmented mp4 files.)
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Also return a proper error code.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Also pass on any returned error code.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This more closely corresponds to the usage of the field.
Its usage here is unrelated to the channel ID.
Signed-off-by: Martin Storsjö <martin@martin.st>
Channel 4 is typically used by the Flash player to transmit
audio, channel 6 for video, and various stream-specific invokes
get sent over channel 8, which is designated the source channel.
This more closely matches the behavior of the Flash player,
including the transmission of play requests over channel 8.
Signed-off-by: Martin Storsjö <martin@martin.st>
Sending non-monotonic packets (e.g. when the audio and video
streams are monotonic within themselves but not muxed
monotonically) will lead to negative values the RTMP timestamp
field (where timestamps are transmitted only as deltas for each
channel), and this delta can end up being incorrectly written as
a large unsigned number.
Signed-off-by: Martin Storsjö <martin@martin.st>
Since 596e5d4783, this is not necessary anymore. It also allows to
actually disable the flushing, improving write performance (but
possibly giving worse latency in real-time streaming).
Signed-off-by: Martin Storsjö <martin@martin.st>
This is enabled by default and can be disabled with
"-fflags -flush_packets".
Inspired by a patch from Nicolas George <nicolas.george@normalesup.org>.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously the wrong buffer pointer was checked, when buf
instead of *buf was checked. But checking the return value
instead is even better.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
If idx equaled num_coeffs - 1 on entry to the loop, the previous
check failed to break out of the loop.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
If we really want to support parameter changes, they need to be
signalled along with the AVPackets as parameter change side data,
not just changing the AVCodecContext parameters when a packet
is demuxed (since there may be other earlier packets yet undecoded).
Something similar was already done for the sample rate in 0883109b2,
but some parameters were left changeable.
This avoids having to recheck the channel count for validity for
each decoded frame in (ad)pcm decoders, unless the decoders
explicitly say that they accept parameter changes.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Limit the size to INT_MAX/2 (for simplicity) to be sure that
size + BYTES_PER_FRAME_RECORD won't overflow.
Also factorize other existing error return paths.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
The volatile is not required here, and prevents a miscompilation with GCC
4.8.1 when building on x86 with --cpu=i686
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
The sample buffering logic does not take into account that the blocksize
could change. Reset the buffer if the channel configuration changes,
since if there are leftover samples, it is most likely a broken or
misconcatenated stream. This could lead to negative numbers for
missing_samples during decoding.
Thanks to Michael Niedermeyer for pointing these out.
Limit the size to INT_MAX/2 (for simplicity) to be sure that
size + FF_INPUT_BUFFER_PADDING_SIZE won't overflow.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Also don't pointlessly set the buffer size to 1 after copying
one packet.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
The seektable is required for filling in ape->frames[i].pos
further down.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
