For use when send-side bandwidth estimation is enabled.
Receive times need to be captured, buffered and then sent using
TransportFeedback RTCP messaged back to the send side.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1290813008
Cr-Commit-Position: refs/heads/master@{#9898}
Handling the case when encoder drops only the higher layer.
Added options to screenshare loopback test to discard high temporal or spatial layers (to view the lower layers).
Review URL: https://codereview.webrtc.org/1287643002
Cr-Commit-Position: refs/heads/master@{#9883}
Value was incorrectly truncated to 16 bits when serializing the message.
Fixed, with added regression tests.
BUG=
Review URL: https://codereview.webrtc.org/1294393002
Cr-Commit-Position: refs/heads/master@{#9858}
Pitfalls:
* Left shift of signed integer has undefined behavior
* Right-shift of signed integer has platform-specific behavior is value is negative
* Cast from unsigned to signed has undefined behavior if value is negative
BUG=webrtc:4824
Review URL: https://codereview.webrtc.org/1226993003
Cr-Commit-Position: refs/heads/master@{#9854}
Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.
BUG=webrtc:4311
Review URL: https://codereview.webrtc.org/1247293002
Cr-Commit-Position: refs/heads/master@{#9670}
Also fixed an arithmetic issue where a 0 0 3 at the end of the rbsp would include the 3 (that's not a legal bitstream anyway, so it probably wasn't a real bug, but it was incorrect).
This maintains the underflow fix from an earlier CL (https://codereview.webrtc.org/1219493004/). The overflow fix is virtually impossible to hit (hence no unit tests), but is there for strict correctness.
BUG=
Review URL: https://codereview.webrtc.org/1226203002
Cr-Commit-Position: refs/heads/master@{#9581}
Adds a class used to classify whether packet loss events are a single packet or multiple packets as well as how many packets have been lost. Also exposes a new function in the RtpRtcp interface to retrieve these statistics.
BUG=
Review URL: https://codereview.webrtc.org/1198853004
Cr-Commit-Position: refs/heads/master@{#9568}
Bugs found by manual inspection of code, not by fuzzing or packet
replays. At least one of them confirmed by local fuzzing.
BUG=chromium:496094, webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1182793002
Cr-Commit-Position: refs/heads/master@{#9542}
Prevents OOB reads on truncated FU-A NAL units, StapA headers and past
truncation just after StapA headers.
BUG=webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1218023003
Cr-Commit-Position: refs/heads/master@{#9522}
If the buffer becomes full an OnPacketReady callback will be used to
send the packets created so far. On success the buffer can be reused.
The same callback will be called when the last packet has beed created.
Also made some changes to RawPacket. Buffer will now be heap-allocated
rather than (potentially) stack-allocated, but on the plus side it can
now be allocted with variable size and also avoids one memcpy.
BUG=
patch from issue 56429004 at patchset 160001 (http://crrev.com/56429004#ps160001)
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1165113002
Cr-Commit-Position: refs/heads/master@{#9390}
The main purpose of this CL is to clean up RTCPSender::PrepareRTCP, but
it has quite a few ramifications. Notable changes:
* Removed the rtcpPacketTypeFlags bit vector and don't assume
RTCPPacketType values have a single unique bit set. This will allow
making this an enum class once rtcp_receiver has been overhauled.
* Flags are now stored in a map that is a member of the class. This
meant we could remove some bool flags (eg send_remb_) which was
previously masked into rtcpPacketTypeFlags and then masked out again
when testing if a remb packet should be sent.
* Make all build methods, eg. BuildREMB(), have the same signature.
An RtcpContext struct was introduced for this purpose. This allowed
the use of a map from RTCPPacketType to method pointer. Instead of
18 consecutive if-statements, there is now a single loop.
The context class also allowed some simplifications in the build
methods themselves.
* A few minor simplifications and cleanups.
The next step is to gradually replace the builder methods with the
builders from the new RtcpPacket classes.
BUG=2450
R=asapersson@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48329004
Cr-Commit-Position: refs/heads/master@{#9166}
This CL makes two changes to rtc::Buffer that have had to wait for
Chromium's use of it to be modernized:
1. Change default return type of rtc::Buffer::data() from char* to
uint8_t*. uint8_t is a more natural type for bytes, and won't
accidentally convert to a string. (Chromium previously expected
the default return type to be char, which is why
rtc::Buffer::data() initially got char as default return type in
9478437f, but that's been fixed now.)
2. Stop accepting void* inputs in constructors and methods. While
this is convenient, it's also dangerous since any pointer type
will implicitly convert to void*.
(This was previously committed (9e1a6d7c) but had to be reverted
(cbf09274) because Chromium on Android wasn't quite ready for it).
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47109004
Cr-Commit-Position: refs/heads/master@{#9132}
