While converting the aggregated (stap-a) packet transform packet
framing input into an annex-b framing copy, the two loops (both the
required size calculation and the stap-a-to-annex-b copy) may
over-read the input buffer.
In both buffers, `nalu_ptr` follows the input (stap-a) buffer, which
is located in `data`, and whose length is `data_size`. Buffer is read
until `nalu_ptr` reaches the end of the buffer. Issues is that the 5th
line in the loop:
```
uint16_t segment_length = nalu_ptr[0] << 8 | nalu_ptr[1];
```
This line accesses `nalu_ptr[1]`, which needs to be protected in
the loop condition. Let's assume `data_size = 4`, and that we restart
the loop with `nalu_ptr = data + 3`. The condition of the loop does
hold (`nalu_ptr = data + 3 < data + data_size`), but the 5th line
will access to `data[3+1] = data[4]`, which is an over-read.
Tested:
```
$ ninja -C out/Default
$ out/Default/modules_unittests --gtest_filter=PacketBuffer*:H264*:RtpPacketizerH264Test*:VideoRtpDepacketizerH264Test*:TestH264SpsPpsTracker* --logs
...
[ PASSED ] 97 tests.
```
Change-Id: I8b8aaf7d12b0bb154430b8922f099cd49e684762
Bug: webrtc:11698
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177140
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31561}
Together with RtpDepacketizer refactoring that would reduce
number of memcpy while handling an rtp packet
Bug: webrtc:11152
Change-Id: I6f4e09c93af5e2a9314967a15eac8ced57ec712e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161087
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29985}
This reverts commit c98ff2eff0.
Reason for revert: breaks decoding of H264 RTP streams where M bit is set in AUD packets.
Original change's description:
> Reset end-of-frame flag in non-VCL packet.
>
> Bug: chromium:999807
> Change-Id: I28309d2fda16842e620e499cb9e77ec771827b8c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157160
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29560}
TBR=philipel@webrtc.org,ssilkin@webrtc.org,philipel@chromium.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:999807
Change-Id: I8d6bcf4c00197b00d279b9e53a11652d3e61171b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158204
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29600}
The values are available as part of the RTPVideoHeader member.
Bug: None
Change-Id: I832fffc449929badec3796d7096c9cdc0d43d344
Reviewed-on: https://webrtc-review.googlesource.com/c/123234
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26773}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I74cb86c29cebb69dd22083718f1446f18f705cd4
Reviewed-on: https://webrtc-review.googlesource.com/95883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24483}
Downstream projects have been updated, so this can now be relanded.
This is a revert (and rebase) of: https://webrtc-review.googlesource.com/c/src/+/88820
Bug: none
Change-Id: I424664ddef7aeebd3c6c94ae67c7f70a342dc9a4
Reviewed-on: https://webrtc-review.googlesource.com/92082
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24181}
This CL is in preparation to change the RTPVideoTypeHeader into an absl::variant.
Bug: none
Change-Id: I1672d866df0395f3417d8e278cc67f017ab0ff98
Reviewed-on: https://webrtc-review.googlesource.com/87261
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23856}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script from modules with parameters
'pacing video_coding congestion_controller remote_bitrate_estimator':
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118
Reviewed-on: https://webrtc-review.googlesource.com/83900
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23640}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/modules/video_coding/h264_sps_pps_tracker.cc (Browse further)
