| Commit message (Collapse) | Author | Age |
|
|
|
|
|
|
|
|
|
| |
streamed data
This should increase the effectiveness of ffio_ensure_seekback by reducing the
number of buffer reallocations and memmoves/memcpys because even a small
seekback window requires max_buffer_size+window_size buffer space.
Signed-off-by: Marton Balint <cus@passwd.hu>
|
|
|
|
| |
Signed-off-by: Marton Balint <cus@passwd.hu>
|
|
|
|
|
|
|
|
| |
not needed
Let's move unread data to the start of the old buffer instead.
Signed-off-by: Marton Balint <cus@passwd.hu>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
needed
Previously ffio_ensure_seekback never flushed the buffer, so successive
ffio_ensure_seekback calls were all respected. This could eventually cause
unlimited memory and CPU usage if a demuxer called ffio_ensure_seekback on all
it's read data.
Most demuxers however only rely on being able to seek back till the position of
the last ffio_ensure_seekback call, therefore we change the semantics of
ffio_ensure_seekback so that a new call can invalidate seek guarantees of the
old. In order to support some level of "nested" ffio_ensure_seekback calls, we
document that the function only invalidates the old window (and potentially
discards the already read data from the IO buffer), if the newly requested
window does not fit into the old one.
This way we limit the memory usage for ffio_ensure_seekback calls requesting
consecutive data windows.
Signed-off-by: Marton Balint <cus@passwd.hu>
|
|
|
|
|
|
|
|
| |
It was possible for the old code to seek back before the most recently read
data if start of a new multipart was across read boundaries. Now we read some
small sections multiple times to avoid this, but that is OK.
Signed-off-by: Marton Balint <cus@passwd.hu>
|
|
|
|
|
|
| |
requested probe buffer
Signed-off-by: Marton Balint <cus@passwd.hu>
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
The new buf_size was detemined too conservatively, maybe because of the
off-by-one issue which was fixed recently in fill_buffer. We can safely
substract 1 more from the new buffer size, because max_buffer_size space must
only be guaranteed when we are reading the last byte of the requested window.
Comparing the new buf_size against filled did not make a lot of sense, what
makes sense is that we want to reallocate the buffer if the new buf_size is
bigger than the old, therefore the change in the check.
Signed-off-by: Marton Balint <cus@passwd.hu>
|
|
|
|
|
|
|
|
| |
Existing code did not check if the requested seekback buffer is
already read entirely. In this case, nothing has to be done to guarantee
seekback.
Signed-off-by: Marton Balint <cus@passwd.hu>
|
|
|
|
|
|
|
| |
There was an off-by-one error when checking if the IO buffer still has enough
space till the end. One more byte can be safely written.
Signed-off-by: Marton Balint <cus@passwd.hu>
|
|
|
|
|
|
|
| |
There is one general rtsp connection plus two connections per stream (rtp/rtcp).
Reviewed-by: Zhao Zhili <zhilizhao@tencent.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
|
|
|
|
| |
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
|
|
|
|
| |
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
|
|
|
|
|
|
|
|
|
|
| |
This is currently safe here, because the effective lifetime of
adaptionset_lang is parse_manifest_adaptationset() (i.e. the pointer
gets overwritten each time on entry to the function and gets freed
before exiting the function), but it is nevertheless safer to reset the
pointer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
Use xmlFree instead of av_freep
snip from libxml2:
* xmlGetProp:
...
* Returns the attribute value or NULL if not found.
* It's up to the caller to free the memory with xmlFree().
According to libxml2, you are supposed to use xmlFree instead of free
on the pointer returned by it, and also using av_freep on Windows will
call _aligned_free instead of normal free, causing _aligned_free to raise
SIGTRAP and crashing ffmpeg and ffplay.
Signed-off-by: Christopher Degawa <ccom@randomderp.com>
|
| |
|
| |
|
|
|
|
| |
Signed-off-by: hwren <hwrenx@126.com>
|
|
|
|
| |
Signed-off-by: hwren <hwrenx@126.com>
|
|
|
|
|
|
| |
Add AVS2_* prefix to macro definitions to avoid confusion
Signed-off-by: hwren <hwrenx@126.com>
|
|
|
|
|
| |
Signed-off-by: hbj <hanbj@pku.edu.cn>
Signed-off-by: hwren <hwrenx@126.com>
|
|
|
|
| |
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
|
|
|
|
|
|
|
|
|
|
| |
The buffers used when fragmented output is enabled have up until now not
been freed in the deinit function; they leak e.g. if one errors out of
mov_write_trailer() before one reaches the point where they are normally
written out and freed. This can e.g. happen if allocating new vos_data
fails at the beginning of mov_write_trailer().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
|
|
|
|
|
|
|
|
| |
Otherwise the old data leaks whenever extradata needs to be rewritten
(e.g. when encoding FLAC with our encoder that sends an updated
extradata packet at the end).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
|
|
|
|
| |
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
|
|
|
|
|
|
| |
They are not used anywhere else.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
When remuxing an rtp hint stream (or any stream with the tag "rtp "),
the mov muxer treats this as one of the rtp hint tracks it creates
internally when ordered to do so; yet this track lacks the
AVFormatContext for the hinting rtp muxer, leading to segfaults in
mov_write_udta_sdp() if a "trak" atom is written for this stream; if not,
the stream's codecpar is freed by mov_free() as if the mov muxer owned
it (it does for the internally created "rtp " tracks), but without
resetting st->codecpar, leading to double-frees lateron. This commit
therefore ignores said tag which makes rtp hint streams unremuxable.
This fixes tickets #8181 and #8186.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
|
|
|
|
|
|
|
|
|
| |
Fixes: Infinite loop
Fixes: 26016/clusterfuzz-testcase-minimized-ffmpeg_IO_DEMUXER_fuzzer-6195663833137152
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
|
| |
|
|
|
|
| |
Improves decoding speed by 24x
|
|
|
|
|
|
|
| |
The generated text streams for chapters lack an AVStream since they
are but an internal concept within movenc.
Fixes #8910
|
|
|
|
|
|
|
|
| |
Fixes: signed integer overflow: 20 * 5184056935931942919 cannot be represented in type 'long'
Fixes: 25466/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-4798660247552000
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
|
|
|
|
|
|
|
|
| |
Fixes: signed integer overflow: 8224000000 * 1629552639 cannot be represented in type 'long'
Fixes: 24908/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-4658478506049536
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
|
|
|
|
|
|
|
| |
1. Remove the assumption that the message method is TEARDOWN.
2. Don't ignore the error code of ff_rtsp_parse_streaming_commands.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
In listen mode with UDP transport, once the sender has sent
the TEARDOWN and closed the connection, poll will indicate that
one can read from the connection (indicating that the socket has
reached EOF and should be closed by the receiver as well). In this
case, parse_rtsp_message won't try to parse the command (because
it's no longer in state STREAMING), but previously just returned
zero.
Prior to f6161fccf8c5720ceac1ed1df8ba60ff8fed69f5, this caused
udp_read_packet to return zero, which is treated as EOF by
read_packet. But after that commit, udp_read_packet would continue
if parse_rtsp_message didn't return an explicit error code.
To keep the original behaviour from before that commit, more
explicitly return an error in parse_rtsp_message when in the wrong
state.
Fixes: #8840
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
|
|
| |
Signed-off-by: James Almer <jamrial@gmail.com>
|
|
|
|
| |
Signed-off-by: James Almer <jamrial@gmail.com>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
There are two possible kinds of timecode tracks (with tag "tmcd") in the
mov muxer: Tracks created internally by the muxer and timecode tracks
sent by the user. If any of the latter exists, the former are
deactivated. The former all belong to another track, the source
track; the latter don't have a source track set, but the index of the
source track is initially zeroed by av_mallocz_array(). This is a
problem since 3d894db700cc1e360a7a75ab9ac8bf67ac6670a3: Said commit added
a function that calculates the duration of tracks and the duration of
timecode tracks is calculated by rescaling the duration (calculated by
the very same function) of the source track. This gives an infinite
recursion if the first track (the one that will be treated as source
track for all timecode tracks) is a timecode track itself, leading to a
stack overflow.
This commit fixes this by not using the nonexistent source track
when calculating the duration of timecode tracks not created internally
by the mov muxer.
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
|
|
|
|
|
| |
Reviewed-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
|
|
|
|
|
|
|
|
|
| |
Fixes: Infinite loop
Fixes: 25844/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5660803318153216
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
|
|
|
|
|
|
|
|
|
|
| |
Add AC-3/EAC-3 to allowed extensions file list.
From HTTP Live Streaming 2nd Edition draft-pantos-hls-rfc8216bis-07
section 3.1.3.Packed Audio, HLS demuxer need to support MP3/AC-3/EAC-3.
Reviewd-by: Steven Liu <liuqi05@kuaishou.com>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
|
| |
|
|
|
|
| |
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
certain circumstances
It seems that in files where the BASF block isn't first, v1.1 ASF streams are
allowed to be non-22050. Either this format is really inconsistent, or
FX Fighter and Croc just ignored the sample rate field, requiring the v1.1
restriction in the first place.
This bumps the version to 1.2 in these streams so they're not "corrected".
Found in Alien Odyssey games files in:
./GRAPHICS/COMMBUNK/{{COMADD1,COMM2_{1,2,3E},COMM3_{2,3,4,5,6}},FADE{1,2}}.BRP
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
|
|
|
|
| |
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
|
|
|
|
| |
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
|
|
|
|
|
|
| |
We can't actually use them though.
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
|
|
|
|
| |
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
|
|
|
|
|
| |
Reviewed-by: "mypopy@gmail.com" <mypopy@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
|
| |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
Since release 4.2, FFmpeg fails to detect the correct streams in an RTMP
stream that contains a |RtmpSampleAccess AMF object prior to the
onMetaData AMF object. In the debug log it would show "[flv] Unknown
type |RtmpSampleAccess".
This functionality broke in commit d7638d8dfc3c4ffd0dc18a64937a5a07ed67b354
as unknown metadata packets now result in an opaque data stream, and the
|RtmpSampleAccess packet was an "unknown" metadata packet type.
With this change the RTMP streams are correctly detected when there
is a |RtmpSampleAccess object prior to the onMetaData object.
Signed-off-by: Peter van der Spek <p.vanderspek@bluebillywig.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
|