This reverts commit 7bcd2a98be.
Reason for revert: peerconnection_unittests fails on downstream test runner.
Original change's description:
> Reland "Optimize execution time of RTPSender::UpdateDelayStatistics"
>
> The reland has a lot of additional DCHECKS for easier debugging,
> so in debug builds it will actually be a ~2x slowdown compared to the old code.
> The excessive DCHECKS should be removed in a followup CL.
>
> Bug: webrtc:9439
> Change-Id: I493de337bf20c998aa32c2532212cac85c5517fb
> Reviewed-on: https://webrtc-review.googlesource.com/96641
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24501}
TBR=terelius@webrtc.org,asapersson@webrtc.org,philipel@webrtc.org
Change-Id: Ia48444d2a7647cf826ef93b4720f6d7ff9a712c3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9439
Reviewed-on: https://webrtc-review.googlesource.com/96960
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24502}
The reland has a lot of additional DCHECKS for easier debugging,
so in debug builds it will actually be a ~2x slowdown compared to the old code.
The excessive DCHECKS should be removed in a followup CL.
Bug: webrtc:9439
Change-Id: I493de337bf20c998aa32c2532212cac85c5517fb
Reviewed-on: https://webrtc-review.googlesource.com/96641
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24501}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated using script:
#!/bin/bash
dir=modules/rtp_rtcp
find $dir -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $dir -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc
Reviewed-on: https://webrtc-review.googlesource.com/83584
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23624}
They have been disabled by default for years, and should have been made redundant by the event logs.
Bug: webrtc:8982
Change-Id: I491923cbc93378d28f5166d24756b335619d9c12
Reviewed-on: https://webrtc-review.googlesource.com/82800
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23598}
bugs.webrtc.org/8439 introduces application data that could e.g. contain
timestamps. We would like to take different actions for this data
depending on whether this is the first time a packet is being sent.
Bug: webrtc:8906
Change-Id: Ib370d76beec2960d961bf44391930faa4b193479
Reviewed-on: https://webrtc-review.googlesource.com/77643
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Petter Strandmark <strandmark@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23426}
This removes the optimization that would stop sending the MID RTP
header extension when an RTCP report block is received. The old
implementation was not flexible enough for the API, and making
those changes is too involved at this time as we need this to work
now to unblock other work.
Bug: webrtc:4050
Change-Id: I099f8e9047a40993d93bcda9164eb82fdf810387
Reviewed-on: https://webrtc-review.googlesource.com/67192
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22776}
This CL adds the ability to configure RTPSender to include the
MID header extension when sending packets. The MID will be
included on every packet at the start of the stream until an RTCP
acknoledgment is received for that SSRC at which point it will
stop being included. The MID will be included on regular RTP
streams as well as RTX streams.
Bug: webrtc:4050
Change-Id: Ie27ebee1cd00a67f2b931f5363788f523e3e684f
Reviewed-on: https://webrtc-review.googlesource.com/60582
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22574}
This is a reland of 6328d7cbbc
Original change's description:
> Rework rtp packet history
>
> This CL rewrites the history from the ground up, but keeps the logic
> (mostly) intact. It does however lay the groundwork for adding a new
> mode where TransportFeedback messages can be used to remove packets
> from the history as we know the remote end has received them.
>
> This should both reduce memory usage and make the payload based padding
> a little more likely to be useful.
>
> My tests show a reduction of ca 500-800kB reduction in memory usage per
> rtp module. So with simulcast and/or fec this will increase. Lossy
> links and long RTT will use more memory.
>
> I've also slightly update the interface to make usage with/without
> pacer less unintuitive, and avoid making a copy of the entire RTP
> packet just to find the ssrc and sequence number to put into the pacer.
>
> The more aggressive culling is not enabled by default. I will
> wire that up in a follow-up CL, as there's some interface refactoring
> required.
>
> Bug: webrtc:8975
> Change-Id: I0c1bb528f32eeed0fb276b4ae77ae3235656980f
> Reviewed-on: https://webrtc-review.googlesource.com/59441
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22347}
Bug: webrtc:8975
Change-Id: I162cb9a1eccddf567bdda7285f8296dc2f005503
Reviewed-on: https://webrtc-review.googlesource.com/60900
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#22356}
Reviewed-on: https://webrtc-review.googlesource.com/61661
Cr-Commit-Position: refs/heads/master@{#22438}
This reverts commit 7bb37b884b.
Reason for revert: Breaks downstream projects
Original change's description:
> Reland "Rework rtp packet history"
>
> This is a reland of 6328d7cbbc
>
> Original change's description:
> > Rework rtp packet history
> >
> > This CL rewrites the history from the ground up, but keeps the logic
> > (mostly) intact. It does however lay the groundwork for adding a new
> > mode where TransportFeedback messages can be used to remove packets
> > from the history as we know the remote end has received them.
> >
> > This should both reduce memory usage and make the payload based padding
> > a little more likely to be useful.
> >
> > My tests show a reduction of ca 500-800kB reduction in memory usage per
> > rtp module. So with simulcast and/or fec this will increase. Lossy
> > links and long RTT will use more memory.
> >
> > I've also slightly update the interface to make usage with/without
> > pacer less unintuitive, and avoid making a copy of the entire RTP
> > packet just to find the ssrc and sequence number to put into the pacer.
> >
> > The more aggressive culling is not enabled by default. I will
> > wire that up in a follow-up CL, as there's some interface refactoring
> > required.
> >
> > Bug: webrtc:8975
> > Change-Id: I0c1bb528f32eeed0fb276b4ae77ae3235656980f
> > Reviewed-on: https://webrtc-review.googlesource.com/59441
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22347}
>
> Bug: webrtc:8975
> Change-Id: Ibbdbcc3c13bd58d994ad66f789a95ef9bd9bc19b
> Reviewed-on: https://webrtc-review.googlesource.com/60900
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22356}
TBR=danilchap@webrtc.org,sprang@webrtc.org
Change-Id: Id698f5dbba6f9f871f37501d056e2b8463ebae50
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8975
Reviewed-on: https://webrtc-review.googlesource.com/61020
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22358}
This is a reland of 6328d7cbbc
Original change's description:
> Rework rtp packet history
>
> This CL rewrites the history from the ground up, but keeps the logic
> (mostly) intact. It does however lay the groundwork for adding a new
> mode where TransportFeedback messages can be used to remove packets
> from the history as we know the remote end has received them.
>
> This should both reduce memory usage and make the payload based padding
> a little more likely to be useful.
>
> My tests show a reduction of ca 500-800kB reduction in memory usage per
> rtp module. So with simulcast and/or fec this will increase. Lossy
> links and long RTT will use more memory.
>
> I've also slightly update the interface to make usage with/without
> pacer less unintuitive, and avoid making a copy of the entire RTP
> packet just to find the ssrc and sequence number to put into the pacer.
>
> The more aggressive culling is not enabled by default. I will
> wire that up in a follow-up CL, as there's some interface refactoring
> required.
>
> Bug: webrtc:8975
> Change-Id: I0c1bb528f32eeed0fb276b4ae77ae3235656980f
> Reviewed-on: https://webrtc-review.googlesource.com/59441
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22347}
Bug: webrtc:8975
Change-Id: Ibbdbcc3c13bd58d994ad66f789a95ef9bd9bc19b
Reviewed-on: https://webrtc-review.googlesource.com/60900
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22356}
This reverts commit 6328d7cbbc.
Reason for revert: Breaks downstream build, due to use of std::pair constructor that some compilers appear to not support yet. See comment.
Original change's description:
> Rework rtp packet history
>
> This CL rewrites the history from the ground up, but keeps the logic
> (mostly) intact. It does however lay the groundwork for adding a new
> mode where TransportFeedback messages can be used to remove packets
> from the history as we know the remote end has received them.
>
> This should both reduce memory usage and make the payload based padding
> a little more likely to be useful.
>
> My tests show a reduction of ca 500-800kB reduction in memory usage per
> rtp module. So with simulcast and/or fec this will increase. Lossy
> links and long RTT will use more memory.
>
> I've also slightly update the interface to make usage with/without
> pacer less unintuitive, and avoid making a copy of the entire RTP
> packet just to find the ssrc and sequence number to put into the pacer.
>
> The more aggressive culling is not enabled by default. I will
> wire that up in a follow-up CL, as there's some interface refactoring
> required.
>
> Bug: webrtc:8975
> Change-Id: I0c1bb528f32eeed0fb276b4ae77ae3235656980f
> Reviewed-on: https://webrtc-review.googlesource.com/59441
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22347}
TBR=danilchap@webrtc.org,sprang@webrtc.org
Change-Id: I2fa7efc7d008c56f7a8f77bc9958c19119f69de8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8975
Reviewed-on: https://webrtc-review.googlesource.com/60880
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22350}
This CL rewrites the history from the ground up, but keeps the logic
(mostly) intact. It does however lay the groundwork for adding a new
mode where TransportFeedback messages can be used to remove packets
from the history as we know the remote end has received them.
This should both reduce memory usage and make the payload based padding
a little more likely to be useful.
My tests show a reduction of ca 500-800kB reduction in memory usage per
rtp module. So with simulcast and/or fec this will increase. Lossy
links and long RTT will use more memory.
I've also slightly update the interface to make usage with/without
pacer less unintuitive, and avoid making a copy of the entire RTP
packet just to find the ssrc and sequence number to put into the pacer.
The more aggressive culling is not enabled by default. I will
wire that up in a follow-up CL, as there's some interface refactoring
required.
Bug: webrtc:8975
Change-Id: I0c1bb528f32eeed0fb276b4ae77ae3235656980f
Reviewed-on: https://webrtc-review.googlesource.com/59441
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22347}
Pass pointer to application_data from RtpPacketToSend arriving via RtpSender::SendToNetwork through to Transport::SendRtp, in PacketOptions.
Bug: webrtc:8906
Change-Id: Ie75013ed472710f4efcfbcc160e46a6119a1f41d
Reviewed-on: https://webrtc-review.googlesource.com/55600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dino Radaković <dinor@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22174}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=danilchap@webrtc.org
Bug: None
Change-Id: Ib4694d183f04d675f2ea66d39661fdffb2a984f1
Reviewed-on: https://webrtc-review.googlesource.com/23604
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20846}
Specifically, I'm moving
safe_compare.h
safe_conversions.h
safe_minmax.h
They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.
BUG=webrtc:8445
Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
The RTPFragmentationHeader was used when sending audio using RED
for loss protection. This feature has been deprecated and
gradually removed. This cl removes remnants of support from
the RTP send path.
Bug: webrtc:6471
Change-Id: Ia1249047b09c16f79498827f74c2ce07aa38b8f7
Reviewed-on: https://webrtc-review.googlesource.com/16427
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20473}
This is needed in the general case, now that we aim to support codecs
other than those built-in to WebRTC.
BUG=webrtc:8159
Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706
Reviewed-on: https://webrtc-review.googlesource.com/5380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20136}
RtcEventLogImpl no longer hard-codes the way encoding is done. It now relies on RtcEventEncoder for it. This gives two benefits:
1. We can decide between the current encoding and the new encoding (which is still WIP) without code duplication (no need for RtcEventLogImplNew).
2. Encoding is done only when the event needs to be written to a file. This both avoids unnecessary encoding of events which don't end up getting written to a file, as well as is useful for the new, delta-based encoding, which is stateful.
BUG=webrtc:8111
Change-Id: I9517132e5f96b8059002a66fde8d42d3a678c3bb
Reviewed-on: https://webrtc-review.googlesource.com/1365
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20118}
Change LogIncomingRtcpPacket and LogOutgoingRtcpPacket to take ArrayView<uint8_t>.
Split LogSessionAndReadBack into three functions and create class to share state between them.
Split VerifyRtpEvent into one incoming and one outgoing version.
Originally uploaded as https://codereview.webrtc.org/2997973002/
Bug: webrtc:8111
Change-Id: I22bdc35163bef60bc8293679226b19e41e8f49b3
Reviewed-on: https://webrtc-review.googlesource.com/5020
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20063}
Stop using PayloadUnion's public member variables, since a future CL
will make them private.
BUG=webrtc:8159
Change-Id: Ia3dada56be7ef00ed80f3733209b18c178a36561
Reviewed-on: https://webrtc-review.googlesource.com/4380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20027}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
Making a long chain of interface changes to route a CodecSpecificInfo
struct from the video encoder function to the RTPSenderVideo. This
will be used to convey information needed by the RTP packetizer when
building the RTP headers.
Review URL: http://webrtc-codereview.appspot.com/56001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@140 4adac7df-926f-26a2-2b94-8c16560cd09d
