This also moves the packet feedback tracking to RtpVideoSender.
Bug: webrtc:9517
Change-Id: Ifb1ff85051730108a0b0d1dd30f6f8595ad2af6e
Reviewed-on: https://webrtc-review.googlesource.com/c/95920
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25019}
This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into
the audio media path. If a FrameEncryptorInterface is set on an outgoing audio RTPSender
then each outgoing audio payload will first pass through the provided FrameEncryptor which
will have a chance to modify the payload contents for the purposes of encryption.
If a FrameDecryptorInterface is set on an incoming audio RtpReceiver then each incoming
audio payload will first pass through the provided FrameDecryptor which have a chance to
modify the payload contents for the purpose of decryption.
While AEAD is supported by the FrameDecryptor/FrameEncryptor interfaces this CL does not
use it and so it is left as null.
Bug: webrtc:9681
Change-Id: Ic383a9dce280528739f9d271357c2220e0a0dccf
Reviewed-on: https://webrtc-review.googlesource.com/c/101702
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25001}
The RtpReceiver class is no longer used. Together with it, delete
RTPPayloadRegistry, RtpReceiverStrategy, and the tests under
modules/rtp_rtcp/test/testAPI/.
Bug: webrtc:8995
Change-Id: Ia9924d2f0f4315914a0dce6b7375ebb3601a6f96
Reviewed-on: https://webrtc-review.googlesource.com/c/103503
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24968}
When bitrate is allocated to streams that does not have packet feedback,
the allocated bitrate should be included in the estimate. This was
previously only implemented for the old congestion controller and not
for the new task queue based version.
To make the behavior more robust, the responsibility for tracking this
is moved to BitrateAllocator where it's handled consistently for
multiple streams without feedback.
Bug: webrtc:9586, webrtc:8243
Change-Id: I8af7fec23e1bdc08cc61cf1b4ff10461c3711fb0
Reviewed-on: https://webrtc-review.googlesource.com/102681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24905}
Add new method OnRtpPacket, but leave
ReceiveStatisticsImpl::IncomingPacket and most of the implementation
unchanged. Deleting the old method and converting implementation from
RTPHeader to RtpPacketreceived is planned for a followup, after
downstream code is updated.
Bug: webrtc:7135, webrtc:8016
Change-Id: I697ec12804618859f8d69415622d1b957e1d0847
Reviewed-on: https://webrtc-review.googlesource.com/100104
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24889}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
Some of the AEC settings in WebRtcVoiceEngine agree with just about everywhere
else and have therefore been set in stone inside AEC2/AECM. A lot of routing
for those settings disappears.
The comfort noise setting is exposed in the API, so the flag for it will be
removed a PSA later.
Bug: webrtc:9535
Change-Id: I53816152415a9a069cea9520cec697b6bcfe0948
Reviewed-on: https://webrtc-review.googlesource.com/101622
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24863}
Uninline RTPFragmentaion functions
fix RTPFragmentation move constructor and assign operators (was recursive for win)
replace assert with rtc::dchecked_cast
Remove unused includes and dependencies.
Fix other targets that used those includes transitively instead of directly
Bug: None
Change-Id: I647cb1eda107dc7d87d25234095545bc2842fa40
Reviewed-on: https://webrtc-review.googlesource.com/100500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24759}
Return value always passed as the |retransmitted| argument to
ReceiveStatistics::IncomingPacket. The implementation of this method,
StreamStatisticianImpl::IncomingPacket, can call its own
IsRetransmitOfOldPacket, which is demoted to a private method.
Bug: webrtc:7135
Change-Id: I904db676738689c7a1db4caa588f70e64e3c357d
Reviewed-on: https://webrtc-review.googlesource.com/95649
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24494}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I74cb86c29cebb69dd22083718f1446f18f705cd4
Reviewed-on: https://webrtc-review.googlesource.com/95883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24483}
This reverts commit 84916937b70472715efe5682bc273e91c3a72695.
Reason for revert: breaking downstream projects.
Original change's description:
> Update packetsLost and jitter stats any time a packet is received.
>
> Before this CL, the packetsLost and jitter stats (as returned by
> GetStats, at the API level) were only being updated when an RTCP SR or
> RR is generated. According to the stats spec, "local" stats like this
> should be updated any time a packet is received.
>
> This CL also fixes some minor issues with the calculation of packetsLost
> (and fractionLost):
> * Packets weren't being count as lost if lost over a sequence number
> rollover.
> * Temporary periods of "negative" loss (caused by duplicate or out of
> order packets) weren't being accumulated into the cumulative loss
> counter. Example:
> Period 1: Received packets 1, 2, 4
> Loss over that period: 1 (expected 4 packets, got 3)
> Reported cumulative loss: 1
> Period 2: Received packets 3, 5
> Loss over that period: -1 (expected 1 packet, got 2)
> Reported cumulative loss: 1 (should be 0!)
>
> Landing with NOTRY because Android compile bots are broken for an
> unrelated reason.
> NOTRY=True
>
> Bug: webrtc:8804
> Change-Id: I840ba34de8957b1276f6bdaf93718f805629f5c8
> Reviewed-on: https://webrtc-review.googlesource.com/50020
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23731}
TBR=danilchap@webrtc.org,deadbeef@webrtc.org,ossu@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Landing with NOTRY because ios64_sim_ios10_dbg bot is broken.
Passing all other bots.
NOTRY=True
Bug: webrtc:8804
Change-Id: I07bd6b1206d5a8d211792ad392842f9ed6c505e9
Reviewed-on: https://webrtc-review.googlesource.com/95280
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24370}
Introduce SimulatedNetworkReceiverInterface and switch DirectTransport
on this interface. Also switch part of related users on
DefaultNetworkSimulationConfig.
This two changes united into single CL to prevent work duplication.
Most changes were done because of stop including fake_network_pipe.h
into direct_transport.h, so splitting this into 2 CLs will require
first fix all imports of fake_network_pipe.h and then replace them
on new API imports again.
Bug: webrtc:9630
Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6
Reviewed-on: https://webrtc-review.googlesource.com/94762
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24336}
For use in AudiReceiveStream, introduce a new method GetSyncInfo. This
change is analogous to https://webrtc-review.googlesource.com/91123,
doing the same for RtpVideoStreamReceiver. It's a preparation for
bypassing the RtpReceiver class.
Bug: webrtc:7135
Change-Id: I87c1c6f0a1f28b0baebe07c4181f6f0427afa314
Reviewed-on: https://webrtc-review.googlesource.com/93022
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24228}
Replaced by a int64_t representing time in us.
Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}
This is a reland of 06f66c72600e58438ba9caf9f523e00a519ef3c0
Original change's description:
> Removing unneeded dependency.
>
> The //audio build target does not depend on the
> builtin_audio_encoder_factory, this CL removes it from the dependency
> list in order to avoid to propagate symbols that are not supposed to
> be there.
>
> Bug: webrtc:9528
> Change-Id: Ib3868ee93f61057f61283faaa83e0633ebfdea90
> Reviewed-on: https://webrtc-review.googlesource.com/89002
> Reviewed-by: Yves Gerey <yvesg@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24019}
TBR: yvesg@webrtc.org
Bug: webrtc:9528
Change-Id: I696140d5255c659de69430a2c24fcb364eed3c19
Reviewed-on: https://webrtc-review.googlesource.com/89401
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24045}
This reverts commit 06f66c72600e58438ba9caf9f523e00a519ef3c0.
Reason for revert: Breaks downstream project.
Original change's description:
> Removing unneeded dependency.
>
> The //audio build target does not depend on the
> builtin_audio_encoder_factory, this CL removes it from the dependency
> list in order to avoid to propagate symbols that are not supposed to
> be there.
>
> Bug: webrtc:9528
> Change-Id: Ib3868ee93f61057f61283faaa83e0633ebfdea90
> Reviewed-on: https://webrtc-review.googlesource.com/89002
> Reviewed-by: Yves Gerey <yvesg@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24019}
TBR=mbonadei@webrtc.org,ossu@webrtc.org,yvesg@webrtc.org
Change-Id: Icf8f0ad4e7f5cce96fa1c0491a281ef2fd2e713f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9528
Reviewed-on: https://webrtc-review.googlesource.com/89400
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24023}
The //audio build target does not depend on the
builtin_audio_encoder_factory, this CL removes it from the dependency
list in order to avoid to propagate symbols that are not supposed to
be there.
Bug: webrtc:9528
Change-Id: Ib3868ee93f61057f61283faaa83e0633ebfdea90
Reviewed-on: https://webrtc-review.googlesource.com/89002
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24019}
Do not update target audio bitrate if WebRTC-Audio-SendSideBwe-For-Video is enabled but other side does not support TWCC
Bug: webrtc:8243
Change-Id: I6c3c4f223dc5168d726996324717d7ba9ec96e6c
Reviewed-on: https://webrtc-review.googlesource.com/88440
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23963}
this function is now only used in combination with StreamStatistician::IsRetransmitOfOldPacket
but IsRetransmitOfOldPacket internally checks if packet is in_order, thus making extra check unnecessary
In addition to making code simpler, removing this checks avoids
taking two extra CritSection on common code path of incoming rtp packet.
Bug: webrtc:8016
Change-Id: I050004e256b5698ce700e3416aa86b55f446a270
Reviewed-on: https://webrtc-review.googlesource.com/85361
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23762}
This will allow experimenting with audio bitrate allocation in video calls without increasing transport overhead.
Bug: webrtc:8243
Change-Id: If961780921d53bdce95b68c26641df6875509c1f
Reviewed-on: https://webrtc-review.googlesource.com/84501
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23755}
Before this CL, the packetsLost and jitter stats (as returned by
GetStats, at the API level) were only being updated when an RTCP SR or
RR is generated. According to the stats spec, "local" stats like this
should be updated any time a packet is received.
This CL also fixes some minor issues with the calculation of packetsLost
(and fractionLost):
* Packets weren't being count as lost if lost over a sequence number
rollover.
* Temporary periods of "negative" loss (caused by duplicate or out of
order packets) weren't being accumulated into the cumulative loss
counter. Example:
Period 1: Received packets 1, 2, 4
Loss over that period: 1 (expected 4 packets, got 3)
Reported cumulative loss: 1
Period 2: Received packets 3, 5
Loss over that period: -1 (expected 1 packet, got 2)
Reported cumulative loss: 1 (should be 0!)
Landing with NOTRY because Android compile bots are broken for an
unrelated reason.
NOTRY=True
Bug: webrtc:8804
Change-Id: I840ba34de8957b1276f6bdaf93718f805629f5c8
Reviewed-on: https://webrtc-review.googlesource.com/50020
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23731}
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 by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -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: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
This will avoid enabling TWCC for calls having WebRTC-Audio-SendSideBwe enabled on one side of the call but not on the other.
Currently the side supporting audio BWE indicates TWCC extension in SDP but the side that does not support will not. As the result the not supporting side will send TWCC but will not use it and the side supporting audio BWE will not send TWCC.
Bug: webrtc:8243
Change-Id: I4d59e78998982051004b8ad86c24b9be34fc095f
Reviewed-on: https://webrtc-review.googlesource.com/82803
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23583}
This removes the redundant type and replaces all usages. A slight change
in behavior is that we no longer get nanosecond resolution. This should
not matter since no current code requires nanosecond resolution.
Bug: webrtc:9155
Change-Id: I04334e08c686d95731621a6c8a7e40400d0ae3b2
Reviewed-on: https://webrtc-review.googlesource.com/71163
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23174}
This makes it easier to refactor the interface in upcoming CLs.
Bug: None
Change-Id: I67d0216e24f087294e95ac96f7278f302bf69832
Reviewed-on: https://webrtc-review.googlesource.com/71041
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22933}
- protobuf library
- file_wrapper.h
These appear to have been left behind during the AecDump refactoring.
After this CL, APM no longer depends on zlib by default! :)
Bug: webrtc:9139
Change-Id: I12a8df2a17a575515b9c07165825f0879c4e15eb
Reviewed-on: https://webrtc-review.googlesource.com/70762
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22923}
Commit bbf21a3fd617abb37567a86e65f8ba18b8d64eb2 ("Remove dependencies on
modules:module_api from AudioProcessing") causes the build to fail with
libstdc++ due to several files using memcpy(3) or memset(3) while relying on
string.h being included implicitly by other headers.
Bug: webrtc:9139
Change-Id: Ib73284962f8694d8bed0551968265bfd13cab967
Reviewed-on: https://webrtc-review.googlesource.com/70180
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Raphael Kubo da Costa (rakuco) <raphael.kubo.da.costa@intel.com>
Cr-Commit-Position: refs/heads/master@{#22895}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
