This class will interface RTPSenderVideo with the underlying
erasure code. It is functionally similar to ProducerFec
(to be renamed UlpfecGenerator). In fact, the FlexfecSender is a
friend of ProducerFec, and reuses most of its implementation.
Besides the fact that FlexfecSender outputs FlexFEC packets,
the main difference with ProducerFec is that FlexfecSender
allocates RTP sequence numbers, whereas ProducerFec does not
do this for the RED-encapsulated ULPFEC packets.
This class is split as interface/implementation, since it will
be owned by VideoSendStream initially. Further along, it may be
owned by PacedSender.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2441613002
Cr-Commit-Position: refs/heads/master@{#14922}
There is no need for it to be an interface.
In this CL, I also took the opportunity to make two small fixes:
- remove the 'flexfec_' prefix from some member variables
- remove unnecessary use of a stringstream object
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2471073003
Cr-Commit-Position: refs/heads/master@{#14919}
This is a proposal for a new RTCP message. Feel free to comment on the
message structure, selected type ids etc, as well as code for
serialization/deserialization. Once we agree on this, I'll continue
with wiring it up in the actual rtcp sender and receiver.
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2306873003
Cr-Commit-Position: refs/heads/master@{#14867}
Separate the null implementation from rtp_rtcp_defines.h, and follow chromium style guide for virtual functions.
BUG=webrtc:163
Review-Url: https://codereview.webrtc.org/2400993002
Cr-Commit-Position: refs/heads/master@{#14738}
This class is split in interface/implementation classes, since it
will be referenced from the Call level. Its purpose is to interface
the erasure code decoder with a new class FlexfecReceiveStream
(for received packets), as well as with the main RTP pipeline (for
recovered packets).
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2392663006
Cr-Commit-Position: refs/heads/master@{#14594}
Add classes that can read and finalize FlexFEC headers.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2269903002
Cr-Commit-Position: refs/heads/master@{#14469}
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).
After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()
See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.
NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.
BUG=webrtc:6410, chromium:630755
Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
- Split out reading/writing of FEC headers to classes separate
from ForwardErrorCorrection. This makes ForwardErrorCorrection
oblivious to what FEC header scheme is used, and lets it focus on
encoding/decoding the FEC payloads.
- Add unit tests for FEC header readers/writers.
- Split ForwardErrorCorrection::XorPackets into XorHeaders and
XorPayloads and reuse these functions for both encoding and
decoding.
- Rename AttemptRecover -> AttemptRecovery in ForwardErrorCorrection.
BUG=webrtc:5654
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2260803002 .
Cr-Commit-Position: refs/heads/master@{#14316}
This patch enables bwe related variable logging to the command line.
This is useful to test congestion control algorithm over real networks.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2296253002
Cr-Commit-Position: refs/heads/master@{#14209}
Remove common_inherited_config from the targets and add it to the
template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
Remove common_config from the targets' config and add
it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
Remove //build/config/sanitizers:deps as a dependency for
all rtc_executable targets and add it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2308553002
Cr-Commit-Position: refs/heads/master@{#14048}
Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.
These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target
Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.
BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
When the sanitizer bots are switched to GN, this needs to be included as a dependency so that the executables can be compiled.
BUG=webrtc:6215
NOTRY=True
Review-Url: https://codereview.webrtc.org/2250893003
Cr-Commit-Position: refs/heads/master@{#13829}
Reason for revert:
Upstream fixes in place, should be OK now.
Original issue's description:
> Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ )
>
> 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: 5fc59e810b
>
> TBR=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=
>
> Committed: https://crrev.com/e5dd44101eca485f5ad12e5f7ce6f6b0d204116b
> Cr-Commit-Position: refs/heads/master@{#13417}
TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=
Review-Url: https://codereview.webrtc.org/2146013002
Cr-Commit-Position: refs/heads/master@{#13465}
Reason for revert:
It keeps breaking upstream.
Original issue's description:
> Reland Issue 2061423003: Refactor NACK bitrate allocation
>
> This is a reland of https://codereview.webrtc.org/2061423003/
> Which was reverted in https://codereview.webrtc.org/2131913003/
>
> The reason for the revert was that some upstream code used
> RtpSender::SetTargetBitrate(). I've added that back as a no-op until we
> it's been brought up to date.
>
> TBR=tommi@webrtc.org
>
> Committed: 05ce4ae31fTBR=tommi@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2130423002
Cr-Commit-Position: refs/heads/master@{#13419}
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 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}
The VUI part an SPS may specify max_num_reorder_frames and
max_dec_frame_buffering. These may cause a decoder to buffer a number
of frame prior allowing decode, leading to delays, even if no frames
using such references (ie B-frames) are sent.
Because of this we update any SPS block emitted by the encoder.
Also, a bunch of refactoring of H264-related code to reduce code
duplication.
BUG=
Review-Url: https://codereview.webrtc.org/1979443004
Cr-Commit-Position: refs/heads/master@{#13010}
This makes the GN configurations easier to read.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2020343003
Cr-Commit-Position: refs/heads/master@{#13006}
this class replace and extend RTCPUtility::RtcpCommonHeader structure and RTCPUtility::RtcpParseCommonHeader function.
In addition to header fields, payload pointer is stored because rtcp header without payload is rarely useful.
Sample usage can be checked in 'RTCP Parser sketched' CL: https://codereview.webrtc.org/1555683002/
BUG=webrtc:5260
R=asapersson@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1575413002 .
Cr-Commit-Position: refs/heads/master@{#11999}
CompactNtpIntervalToMs renamed to CompactNtpRttToMs and handle special cases:
large values consider negative/invalid and result in value of 1.
0 result consider too small and increases to 1.
BUG=590996
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1763823003 .
Cr-Commit-Position: refs/heads/master@{#11928}
std::vector<rtcp::TmmbItem> will replace TMMBRSet class for storage, processing and preparing TMBBR/TMMBN
(i.e. this TmmbItem replaces Timber structure introduced in https://codereview.webrtc.org/1474693002 )
Previous structures store bitrate in kbps. TmmbItem use bps removing need to regularly divide and multiply by 1000.
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1576223003
Cr-Commit-Position: refs/heads/master@{#11491}
Class renamed to indicated use of the rtcp::Empty class: it can only be used as container for other rtcp packets.
All tests that use Append function moved from rtcp_packet_unittest, even if they did not use Empty class.
This is because there is plan to make RtcpPacket class lighter by moving Append functionality to CompoundPacket class.
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1582503002
Cr-Commit-Position: refs/heads/master@{#11234}
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}
