We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
rand() usage replaced with new Random class, which also makes it clearer what interval random number is in.
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1519503002
Cr-Commit-Position: refs/heads/master@{#11019}
rtcp_utility, rtp_utility, tmmbr_help, rtcp_receiver, rtcp_receiver_help are explicetly excluded from the cleanup becaues there are short plans (or cls) to do a deeper cleaning there.
BUG=webrtc:5277
R=pbos@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1512493002
Cr-Commit-Position: refs/heads/master@{#10966}
Created a simple unit test for the new random number generator. (It mostly tests
that the generated numbers are consistent with the intended distribution, e.g. uniform.
It is not a comprehensive test of the quality of the random numbers.)
Several assertions in OveruseDetectorTest seem to depend on the exact sequence of random numbers. I updated those numbers to work with the new PRNG.
Compute the standard deviation of the expected result in TestReorderFilter instead of passing an uncertainty parameter.
BUG=webrtc:5177
Review URL: https://codereview.webrtc.org/1457023002
Cr-Commit-Position: refs/heads/master@{#10965}
This CL makes sure no RTCP SR is sent before there is a valid timestamp
to set in the SR, based on the first sent media packet.
BUG=webrtc:1600
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1506103006 .
Cr-Commit-Position: refs/heads/master@{#10964}
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.
BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1479023002 .
Cr-Commit-Position: refs/heads/master@{#10909}
Created rtcp::Psfb abstract class between rtcp::Pli and rtcp::RtcpPacket to hold common data for Feedback Message.
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1446513002
Cr-Commit-Position: refs/heads/master@{#10823}
to match name given in the RFC5450
private member renamed to inter_arrival_jitters_ for the same reason.
rtcp::ExtendedJitterReport moved into own file
accessors and Parse function added
to make class usable for parsing packet
Review URL: https://codereview.webrtc.org/1434213004
Cr-Commit-Position: refs/heads/master@{#10636}
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.
BUG=None
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1405023016
Cr-Commit-Position: refs/heads/master@{#10594}
The defines still in use was only used in single files, so they were
moved to these specific cc-files.
Review URL: https://codereview.webrtc.org/1411573007
Cr-Commit-Position: refs/heads/master@{#10539}
Reason for revert:
Breaks bot.
Original issue's description:
> Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile. Max p_diff is 8 bits.
>
> Change type of number of reference pictures (size_t -> uint8_t). Max is 2 bits.
>
> Size of WebRtcRTPHeader: 4352 -> 1784 bytes.
>
> BUG=webrtc:5144, chromium:500602
>
> Committed: https://crrev.com/81c5c7f8157f767747bd97419eb0a589207354cf
> Cr-Commit-Position: refs/heads/master@{#10504}
TBR=stefan@webrtc.org,mflodman@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5144, chromium:500602
Review URL: https://codereview.webrtc.org/1423493005
Cr-Commit-Position: refs/heads/master@{#10508}
Change type of number of reference pictures (size_t -> uint8_t). Max is 2 bits.
Size of WebRtcRTPHeader: 4352 -> 1784 bytes.
BUG=webrtc:5144, chromium:500602
Review URL: https://codereview.webrtc.org/1427253002
Cr-Commit-Position: refs/heads/master@{#10504}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
Required a bit of refactoring to make it possible to pass a Call to DirectTransport on construction. This also lead to me having to remove the shared lock between PacketTransport and RtpRtcpObserver. Now RtpRtcpObserver has a SetTransports method instead of a SetReceivers method.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1419193002
Cr-Commit-Position: refs/heads/master@{#10430}
Matches the include order in webrtc/base/criticalsection.h and makes use
of winsock2.h instead of winsock.h for consistency.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1407053008
Cr-Commit-Position: refs/heads/master@{#10389}
We don't allow more than one retransmission within one RTT, but the RTT
estimate might be off. Reasonably, the remote end will not send a NACK
until the packet after has been received - so always resend on first
request.
Review URL: https://codereview.webrtc.org/1414563003
Cr-Commit-Position: refs/heads/master@{#10362}
