This is one step towards separation of send-side and receive-side
processing.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2740163002
Cr-Commit-Position: refs/heads/master@{#17306}
Previosly it supported up to only 15 chunks which is a limit for csrcs in an rtp packet.
BUG=None
Review-Url: https://codereview.webrtc.org/2758533002
Cr-Commit-Position: refs/heads/master@{#17274}
This was a trivial delegation wrapper, with only a single use.
BUG=None
Review-Url: https://codereview.webrtc.org/2741413003
Cr-Commit-Position: refs/heads/master@{#17205}
CurrentNtp return time by taking two output parameters by reference
(also breaks style guide)
CurrentNtpTime treat ntp time as single entity and returns it using NtpTime structure.
(making interface clearer)
BUG=None
Review-Url: https://codereview.webrtc.org/2733823002
Cr-Commit-Position: refs/heads/master@{#17088}
Patchset 1 is patchset #5 id:80001 of https://codereview.webrtc.org/2717983003/
Patchset 2 fix call_perf_test dep on fake_audio_device.
This reverts commit 985371bda999c6db51286586c5850d2ff58f3511.
Original cl description:
Move fake_audio_device to its own target.
The purpose is to make it usefull for test targets that does not need or can use test_common.
For some reason this also triggered override issues in rtp module tests that are fixed in the same cl.
BUG=none
TBR=kjellander@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2718363002
Cr-Commit-Position: refs/heads/master@{#16922}
Reason for revert:
Breaks build DEPS.
Original issue's description:
> Move fake_audio_device to its own target.
> The purpose is to make it usefull for test targets that does not need or can use test_common.
>
> For some reason this also triggered override issues in rtp module tests that are fixed in the same cl.
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2717983003
> Cr-Commit-Position: refs/heads/master@{#16889}
> Committed: 03d850ddf9TBR=ehmaldonado@webrtc.org,danilchap@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=none
Review-Url: https://codereview.webrtc.org/2718083003
Cr-Commit-Position: refs/heads/master@{#16890}
The purpose is to make it usefull for test targets that does not need or can use test_common.
For some reason this also triggered override issues in rtp module tests that are fixed in the same cl.
BUG=none
Review-Url: https://codereview.webrtc.org/2717983003
Cr-Commit-Position: refs/heads/master@{#16889}
In this CL:
- Add message BweProbeCluster and BweProbeResult to rtc_event_log.proto.
- Add corresponding log functions to RtcEventLog.
- Add optional field |probe_cluster_id| to RtpPacket message and added
an overload function to log with this information.
- Propagate the probe_cluster_id to where RTP packets are logged.
BUG=webrtc:6984
Review-Url: https://codereview.webrtc.org/2666533002
Cr-Commit-Position: refs/heads/master@{#16857}
In order to not make this CL too large I have broken it down into at least two
steps. Previous CL: https://codereview.chromium.org/2628563003/
webrtc::PacedSender::Process <--- previous CL start here
webrtc::PacedSender::SendPacket
webrtc::PacketRouter::TimeToSendPacket
webrtc::ModuleRtpRtcpImpl::TimeToSendPacket <--- previous CL end here, this Cl start here
webrtc::RTPSender::TimeToSendPacket
webrtc::RTPSender::PrepareAndSendPacket
webrtc::RTPSender::AddPacketToTransportFeedback
webrtc::TransportFeedbackAdapter::AddPacket
webrtc::SendTimeHistory::AddAndRemoveOld <--- this CL end here
BUG=webrtc:6822
Review-Url: https://codereview.webrtc.org/2708873003
Cr-Commit-Position: refs/heads/master@{#16796}
This cl protects the access to the max_packet_size_, without fixing
the underlying race; the value is simply copied to a local variable,
whose value might be stale when used.
BUG=webrtc:7189
Review-Url: https://codereview.webrtc.org/2704263003
Cr-Commit-Position: refs/heads/master@{#16754}
by creating it on accepted tmmbr/tmmbn rtcp messages
rather on sender/receiver reports.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2702373002
Cr-Commit-Position: refs/heads/master@{#16748}
together with related functions and variables
to stress it is used for Tmmbr only.
This is explicitly pure rename CL with no functional changes.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2707763004
Cr-Commit-Position: refs/heads/master@{#16720}
and the method RTPSender::GenerateNewSSRC.
It's now mandatory for higher layers to call SetSSRC, RTPSender
no longer allocates any ssrc by default.
BUG=webrtc:4306,webrtc:6887
Review-Url: https://codereview.webrtc.org/2644303002
Cr-Commit-Position: refs/heads/master@{#16670}
This a pre-step for improving perfomance of the RTCPReceiver
- rest of the ReceiveInfo is tmmbr related and
can be handled only when tmmbr is explicitly enabled.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2681003003
Cr-Commit-Position: refs/heads/master@{#16667}
In order to not make this CL too large I have broken it down into at least two steps. In this CL we only propagate the pacing information part of the way:
webrtc::PacedSender::Process <--- propagate from here
webrtc::PacedSender::SendPacket
webrtc::PacketRouter::TimeToSendPacket
webrtc::ModuleRtpRtcpImpl::TimeToSendPacket <--- to here
webrtc::RTPSender::TimeToSendPacket
webrtc::RTPSender::PrepareAndSendPacket
webrtc::RTPSender::AddPacketToTransportFeedback
webrtc::TransportFeedbackAdapter::AddPacket
webrtc::SendTimeHistory::AddAndRemoveOld <--- goal is to propagte it here
BUG=webrtc:6822
Review-Url: https://codereview.webrtc.org/2628563003
Cr-Commit-Position: refs/heads/master@{#16664}
This avoids redoing RTP header parsing already done in Call, for video.
The next step is to convert other types of receive streams, i.e.,
audio and flexfec, to use a compatible OnRtpPacket method. We can then
introduce a shared base interface, and simplify media-independent
receive processing in Call.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2681673004
Cr-Commit-Position: refs/heads/master@{#16583}
Since the only used class is RTCPUtilitiy::NackStats,
rename it to RtcpNackStats and move it into dedicated file.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2680183004
Cr-Commit-Position: refs/heads/master@{#16515}
That would simplify their usage in tests where perfomance is not critical.
BUG=None
Review-Url: https://codereview.webrtc.org/2675713005
Cr-Commit-Position: refs/heads/master@{#16461}
Also fixes a bug where RTCP transport feedback was sent even though RTCP was disabled.
May affect perf numbers since the behavior of the send-side BWE differs a lot from the recv-side BWE.
BUG=webrtc:7111
Review-Url: https://codereview.webrtc.org/2669413003
Cr-Commit-Position: refs/heads/master@{#16451}
Useful if bitrate probing is to be used with audio streams.
BUG=webrtc:7043
Review-Url: https://codereview.webrtc.org/2652893004
Cr-Commit-Position: refs/heads/master@{#16404}
It combines and simplify use of GetStatusVector and GetReceiveDeltas accesors.
Replace use of all GetStatusVector/GetReceiveDeltasUs
BUG=None
Review-Url: https://codereview.webrtc.org/2633923003
Cr-Commit-Position: refs/heads/master@{#16139}
This CL adds an RTP module to FlexfecReceiveStreamImpl, and wires it up
to send RTCP RRs. It further makes some methods take const refs instead
of values, to make it more clear where packet copies are made. This
change reduces the number of copies by one, for the case when media
packets are added to the FlexFEC receiver.
The end-to-end test is modified to check for RTCP RRs being sent.
Part of this modification involves some indentation changes, and the
diff thus looks bigger than it logically is.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2625633003
Cr-Commit-Position: refs/heads/master@{#16106}
Removed const_cast while creating rtcp packet.
This way manually created packet is as good as parsed packet and can be used in tests directly.
To archive this, changed the way class stores deltas and their sizes:
encoded chunks are stored directly for all but last chunk simplifying rtcp packet creation.
deltas stored together with sequence_number that would allow to simplify reading them from the parsed packet.
Fixed test for maximum received packets.
BUG=None
Review-Url: https://codereview.webrtc.org/2616343003
Cr-Commit-Position: refs/heads/master@{#16091}
This lets the RTP code be unaware of lower layers, and the
SetTransportOverhead method is deleted from RTPSender and RtpRtcp.
Instead, that method is added to CongestionController and
TransportFeedbackAdapter, where it is more appropriate.
BUG=wertc:6847
Review-Url: https://codereview.webrtc.org/2589743002
Cr-Commit-Position: refs/heads/master@{#15995}
Moves webrtc/common_video/rotation.h and parts of
webrtc/common_video/include/video_frame_buffer.h and
webrtc/video_frame.h, and adds to a new GN target api:video_frame_api.
BUG=webrtc:5880
Review-Url: https://codereview.webrtc.org/2517173004
Cr-Commit-Position: refs/heads/master@{#15993}
Add RTC_DEPRACATed anonymous unions to not break downstream projects.
Orignal issue's description:
> commit 0ad21111fcc57a7e978edba3c4263f0062d7f9ff
> Author: danilchap <danilchap@webrtc.org>
> Date: Mon Dec 19 09:36:33 2016 -0800
>
> Revert of Rename RTPVideoHeader.isFirstPacket to
> .is_first_packet_in_frame. (patchset #1 id:1 of
> https://codereview.webrtc.org/2574943003/ )
>
> Reason for revert:
> breaks downstream project.
>
> Can you make this change in a compatible way using anonymous
> union:
> union {
> bool is_first_packet_in_frame;
> RTC_DEPRECATED bool isFirstPacket;
> };
> (unfortunetly this this treak breaks braced initialization in
> rtp_rtcp_impl_unittest.cc,
> so that should be rewritting in a more classic way)
>
> Original issue's description:
> > Rename RTPVideoHeader.isFirstPacket to
> > .is_first_packet_in_frame.
> >
> > Name should represent the actual meaning.
> >
> > BUG=None
> >
> > Review-Url: https://codereview.webrtc.org/2574943003
> > Cr-Commit-Position: refs/heads/master@{#15684}
> > Committed:
> > efde908380
>
> TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days
> ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2589783003
> Cr-Commit-Position: refs/heads/master@{#15686}
>
BUG=None
Review-Url: https://codereview.webrtc.org/2614503002
Cr-Commit-Position: refs/heads/master@{#15987}
Removing the DCHECK due to (sometimes) failing voe_auto_test.
Long-term, this DCHECK should be readded. Before that can happen,
the SSRC in the RTPSender should be made immutable.
TESTED=No failures when running third_party/gtest-parallel/gtest-parallel --repeat=5000 --gtest_filter="VolumeTest.ManualInputMutingMutesMicrophone" out/Debug/voe_auto_test.
BUG=webrtc:6887
Review-Url: https://codereview.webrtc.org/2610873002
Cr-Commit-Position: refs/heads/master@{#15962}
Reason for revert:
Doing a reland where systeminfo.cc includes basictypes.h so that CPU_X86 etc. are defined when they are checked/used.
Original issue's description:
> Revert of Replace basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2604043002/ )
>
> Reason for revert:
> Very likely cause of Chromium import bot breakage (unused function '__cpuid'), TBD why.
>
> Original issue's description:
> > Replace basictypes.h with stdint.h for int_t types.
> >
> > Removes basictypes.h for types that only makes use of it for fixed-size-int
> > typedefs and replaces it with stdint.h.
> >
> > BUG=webrtc:6853
> > R=tommi@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2604043002
> > Cr-Commit-Position: refs/heads/master@{#15867}
> > Committed: 7fd1a75300
>
> TBR=tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6853
>
> Review-Url: https://codereview.webrtc.org/2603203003
> Cr-Commit-Position: refs/heads/master@{#15869}
> Committed: 7eb0e23bcf
BUG=webrtc:6853
TBR=tommi@webrtc.org
Review-Url: https://codereview.webrtc.org/2609783002
Cr-Commit-Position: refs/heads/master@{#15873}
