The last in-tree call site recently disappeared, so they were unused.
BUG=webrtc:5922
Review-Url: https://codereview.webrtc.org/2066473002
Cr-Commit-Position: refs/heads/master@{#13751}
- Make more use of std::unique_ptr.
- Auto type deduction for iterator type names.
- More extensive comments.
- Variable renaming.
- Make ProducerFec::BuildRedPacket() static.
- Avoid dynamic allocation of ProducerFec::fec_.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2110763002
Cr-Commit-Position: refs/heads/master@{#13700}
This CL expands the test coverage by checking that the FEC can
handle reordered received media packets. Specifically, this checks
that |recovered_packets| is kept in sorted order.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2101253002
Cr-Commit-Position: refs/heads/master@{#13693}
This CL should have no changes to functionality.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2107703002
Cr-Commit-Position: refs/heads/master@{#13690}
This CL converts the ForwardErrorCorrection class to use std::unique_ptr
for memory management, rather than manually delete'ing allocated memory.
It further renames some data structures and types to distinguish between
generated FEC packets (i.e. coming from the encode operation) and
received FEC packets (i.e. coming in over the wire, intended for the
decode operation).
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2099243003
Cr-Commit-Position: refs/heads/master@{#13687}
it become static to clearly state input, output and temporary variables
style updated with minor improvments to actual algorithm.
Review-Url: https://codereview.webrtc.org/1989363006
Cr-Commit-Position: refs/heads/master@{#13652}
double check rtp_sender in sending mode when altering sequence_number
adjust test to skip validating timestamp on rtx streams
fix test by waiting for all 3 media streams instead of 3 out 6 media and rtx streams.
BUG=webrtc:4332
Review-Url: https://codereview.webrtc.org/2177523002
Cr-Commit-Position: refs/heads/master@{#13587}
- Renamed variables and some function to comply with style guide.
- Removed default argument values.
- Removed some dead code.
- Cleaned up comments formatting in rtp_rtcp.h
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2067673004 .
Cr-Commit-Position: refs/heads/master@{#13565}
This is an issue if the sequence numbers are to be used to compute packet loss statistics since it introduces gaps which are not related to loss.
Also making sure that the header extensions are properly guarded by the send crit sect.
Review-Url: https://codereview.webrtc.org/2190913002
Cr-Commit-Position: refs/heads/master@{#13557}
maximum allowed sized raised from limited by physical udp packet size to
limited by theoritical maximum rtcp packet size.
BUG=webrtc:5260
R=åsapersson
Review-Url: https://codereview.webrtc.org/1998633002
Cr-Commit-Position: refs/heads/master@{#13532}
Now it check if rtp timestamp can be calculating instead of checking number of rtp packets. This way it works for reconfigured streams too.
It also moved deeper into rtcp_sender class to prevent SR no matter the reason it need to be genereated. This way it prevents creating compound rtcp packets that have to start with Sender Report and Sender Reports as response to (mostly theoretical) sr-request rtcp packet.
BUG=webrtc:1600
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1639253007 .
Cr-Commit-Position: refs/heads/master@{#13503}
Reason for revert:
Breaks upstream code.
Original issue's description:
> Refactor NACK bitrate allocation
>
> Nack bitrate allocation should not be done on a per-rtp-module basis,
> but rather shared bitrate pool per call. This CL moves allocation to the
> pacer and cleans up a bunch if bitrate stats handling.
>
> BUG=
> R=danilchap@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
>
> Committed: 5fc59e810bTBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2131913003
Cr-Commit-Position: refs/heads/master@{#13417}
This CL mainly updates the FEC code to use more C++11 features and
to be more in line with the style guide. These changes should
have no impact on the functionality provided by the FEC.
Summary of style fixes:
- Use range-based for loops, where applicable.
- Use auto type deduction for iterator type names.
- Use RTC_DCHECK instead of assert.
- Rename FEC to Fec, where applicable.
- Update test_fec.cc to use variable_names rather than variableNames.
- Avoid redefining the PacketList types outside ForwardErrorCorrection.
Another minor change is that storage for the packet masks, as these
are generated, now is provided by a member variable, rather than
being dynamically allocated on every call to GenerateFec.
BUG=webrtc:5654
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2080553003 .
Cr-Commit-Position: refs/heads/master@{#13403}
A bug in the transpot feedback adapter causes new feedback message to
always start with a received packet. This makes it impossible for the
receiver to distinguish from actual dropped packets and dropped feedback
messages.
BUG=webrtc:6073
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2122863002 .
Cr-Commit-Position: refs/heads/master@{#13381}
Allows detecting large-enough audio packets as part of a probe,
speculative fix for a rampup-time regression in M50. These packets are
accounted on the send side when probing.
BUG=webrtc:5985
R=mflodman@webrtc.org, philipel@webrtc.org
Review URL: https://codereview.webrtc.org/2061193002 .
Cr-Commit-Position: refs/heads/master@{#13210}
It appears there the encode and send operation can happen over multiple
threads. Also, padding data itself may be sent on a different thread.
Remove thread checker and protect all data with crit_sect.
BUG=webrtc:5998
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2066863002
Cr-Commit-Position: refs/heads/master@{#13137}
to fix projects that has own copy of base/thread_annotation.h
R=åsapersson
NOTRY=true
Review-Url: https://codereview.webrtc.org/2048113002
Cr-Commit-Position: refs/heads/master@{#13073}
This CL adds support for an extension on RTP frames to allow the sender
to specify the minimum and maximum playout delay limits.
The receiver makes a best-effort attempt to keep the capture-to-render delay
within this range. This allows different types of application to specify
different end-to-end delay goals. For example gaming can support rendering
of frames as soon as received on receiver to minimize delay. A movie playback
application can specify a minimum playout delay to allow fixed buffering
in presence of network jitter.
There are no tests at this time and most of testing is done with chromium
webrtc prototype.
On chromoting performance tests, this extension helps bring down end-to-end
delay by about 150 ms on small frames.
BUG=webrtc:5895
Review-Url: https://codereview.webrtc.org/2007743003
Cr-Commit-Position: refs/heads/master@{#13059}
This shouldn't be needed, but because the receiver assumes RTX packets
contain RED if configured to receive them (due to an incompatibility
issue), we also have to make sure we send them for now.
BUG=webrtc:5675
Review-Url: https://codereview.webrtc.org/2033763002
Cr-Commit-Position: refs/heads/master@{#13024}
