The method for looking up track ID by SSRC was never updated for
Unified Plan so it only looked at the first audio section and the
first video section.
This CL changes the method to look through all audio and video
media sections rather than just the first.
Bug: chromium:906988
Change-Id: Ie79e6162b2bd24b8ac9e983b5fa7360c96f030da
Reviewed-on: https://webrtc-review.googlesource.com/c/112223
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25833}
Auxiliary threads (worker, network) are still active
while PeerConnection is destructed, leading to race condition
in tests such as:
* RTCStatsIntegrationTest.GetStatsFromCaller
* RTCStatsIntegrationTest.GetsStatsWhileDestroyingPeerConnection
This CL prevents the conflict to happen by explicitly
closing the PeerConnection.
Bug: webrtc:9847
Change-Id: I40880bb9b193201711031b8c4563c6bbd4983c71
Reviewed-on: https://webrtc-review.googlesource.com/c/104820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25801}
This CL moves webrtc::CreatePeerConnectionFactory definitions out of
pc:create_pc_factory and merges it with its declaration in the api/
directory.
In order to avoid circular dependencies a new build target is created:
* api:create_peerconnection_factory
Bug: webrtc:9862
Change-Id: Ie215c94460cba026f5bf7d11c9a5aa03792064af
Reviewed-on: https://webrtc-review.googlesource.com/c/111186
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25744}
Fixing bug that allows a channel to be set on a stopped transceiver.
This CL contains the following refactoring:
1. Extracted ChannelInterface from BaseChannel
2. Unified SetXxxMediaChannel (Voice, Video) into SetMediaChannel
Bug: webrtc:9932
Change-Id: I2fbf00c823b7848ad4f2acb6e80b1b58ac45ee38
Reviewed-on: https://webrtc-review.googlesource.com/c/110564
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25641}
Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
that only handles SRTP configuration to a more generic structure that can be
used and extended for all per peer connection CryptoOptions that can be on a
given PeerConnection.
Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
accessed as crypto_options.srtp.whatever_option_name. This is more inline with
other structures we have in WebRTC such as VideoConfig. As additional features
are added over time this will allow the structure to remain compartmentalized
and concerned components can only request a subset of the overall configuration
structure e.g:
void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
In addition to this it made little sense for sslstreamadapter.h to hold all
Srtp related configuration options. The header has become loo large and takes on
too many responsibilities and spilting this up will lead to more maintainable
code going forward.
This will be used in a future CL to enable configuration options for the newly
supported Frame Crypto.
Reland Fix:
- cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional
root level configuration.
- peerconnectionfactory - If this optional is set will now overwrite the
underyling value.
This along with the other field will be deprecated once dependent projects
are updated.
TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org
Bug: webrtc:9681
Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d
Reviewed-on: https://webrtc-review.googlesource.com/c/105560
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25135}
This reverts commit ac2f3d14e45398930bc35ff05ed7a3b9b617d328.
Reason for revert: Breaks downstream project
Original change's description:
> Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h
>
> Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
> that only handles SRTP configuration to a more generic structure that can be
> used and extended for all per peer connection CryptoOptions that can be on a
> given PeerConnection.
>
> Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
> accessed as crypto_options.srtp.whatever_option_name. This is more inline with
> other structures we have in WebRTC such as VideoConfig. As additional features
> are added over time this will allow the structure to remain compartmentalized
> and concerned components can only request a subset of the overall configuration
> structure e.g:
>
> void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
>
> In addition to this it made little sense for sslstreamadapter.h to hold all
> Srtp related configuration options. The header has become loo large and takes on
> too many responsibilities and spilting this up will lead to more maintainable
> code going forward.
>
> This will be used in a future CL to enable configuration options for the newly
> supported Frame Crypto.
>
> Change-Id: I99d1be36740c59548c8e62db52d68d738649707f
> Bug: webrtc:9681
> Reviewed-on: https://webrtc-review.googlesource.com/c/105180
> Reviewed-by: Emad Omara <emadomara@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25130}
TBR=steveanton@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
Bug: webrtc:9681
Change-Id: Ib0075c477c951b540d4deecb3b0cf8cf86ba0fff
Reviewed-on: https://webrtc-review.googlesource.com/c/105541
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25133}
Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
that only handles SRTP configuration to a more generic structure that can be
used and extended for all per peer connection CryptoOptions that can be on a
given PeerConnection.
Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
accessed as crypto_options.srtp.whatever_option_name. This is more inline with
other structures we have in WebRTC such as VideoConfig. As additional features
are added over time this will allow the structure to remain compartmentalized
and concerned components can only request a subset of the overall configuration
structure e.g:
void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
In addition to this it made little sense for sslstreamadapter.h to hold all
Srtp related configuration options. The header has become loo large and takes on
too many responsibilities and spilting this up will lead to more maintainable
code going forward.
This will be used in a future CL to enable configuration options for the newly
supported Frame Crypto.
Change-Id: I99d1be36740c59548c8e62db52d68d738649707f
Bug: webrtc:9681
Reviewed-on: https://webrtc-review.googlesource.com/c/105180
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25130}
This will later allow simulcast to be set up without any SDP
manipulation. Currently limited to only one layer as the SDP
generated is not spec compliant and more work is required
to support simulcast.
Initial encoding parameters are deferred and applied when the ssrc
is set on the sender. This allows parameters to be changed before
negotiation is completed.
Bug: webrtc:7600
Change-Id: I0a31cd1c2bfc72ebb61ce0fa4fa69d87e3d8b74d
Reviewed-on: https://webrtc-review.googlesource.com/95488
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24917}
RtpPacket::UpdateDelayStatistics was previously optimized with several
sanity checks added. These sanity checks caused many of the unit tests
in peerconnection_integration_unittests to fail and the CL was therefore
reverted. This CL contains the sanity checks along with patches so that
the unit tests pass.
Bug: webrtc:9439
Change-Id: Ia5f5e8b125e5f3f4b79d433e2282901143530a25
Reviewed-on: https://webrtc-review.googlesource.com/99802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24813}
This change injects the FrameEncryptorInterface and the FrameDecryptorInterface
into the RtpSenderInterface and RtpReceiverInterface respectively. This is the
second stage of the injection. In a follow up CL non owning pointers to these
values will be passed down into the media channel.
This change also updates the corresponding mock files.
Bug: webrtc:9681
Change-Id: I964084fc270e10af9d1127979e713493e6fbba7d
Reviewed-on: https://webrtc-review.googlesource.com/96625
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24489}
The usage of MetricsObserverInterface to log metrics has been replaced
by RTC_HISTOGRAM_* macros in WebRTC.
Bug: webrtc:9409
Change-Id: I67df74a18942ac7ea4227e4affdf84f06258a287
Reviewed-on: https://webrtc-review.googlesource.com/86780
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24048}
This is an reland of 6f5b0f920af08d66e6b77ee4f91ade5797145368
Relanded after speculative revert without any changes.
TBR=ilnik@webrtc.org
Original change's description:
> Remove rtc::Optional alias and api:optional target
>
> Update left-overs where old target still was used.
>
> Bug: webrtc:9078
> Change-Id: I2162c928091fc4ff1dea33a3f03adbe47207d206
> Reviewed-on: https://webrtc-review.googlesource.com/84740
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23913}
Bug: webrtc:9078
Change-Id: Ia33c6438253c6ec49f45d938e8a3607b51c418be
Reviewed-on: https://webrtc-review.googlesource.com/88160
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23941}
Generated using gmock_gen.py with some editing.
This mock doesn't seem to be used by unittest in webrtc, but we need to use it in downstream unittests.
Bug: None
Change-Id: Ia7904ffdd22f3d16fe5fd515fa68833817b44481
Reviewed-on: https://webrtc-review.googlesource.com/85780
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23900}
This test creates a one way audio and video call, allows for bandwidth
estimation to ramp up and then runs the call for 10 seconds. The
average bandwidth estimate over this time is recorded as a perf metric.
This is done at the PeerConnection level with the intention to catch
regressions related to ICE configurations. Stats are taken from
PeerConnection for BWE, and the network simulation is done with a
VirtualSocketServer.
Bug: webrtc:7668
Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db
Reviewed-on: https://webrtc-review.googlesource.com/78361
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23758}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'pc'
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: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12
Reviewed-on: https://webrtc-review.googlesource.com/84181
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23669}
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}
The proper closing procedure is:
1. Alice resets outgoing stream.
2. Bob receives incoming stream reset, resets his outgoing stream.
3. Alice receives incoming stream reset; channel closed!
4. Bob receives acknowledgement of reset; channel closed!
https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.7
However, up until now we've been sending both an incoming and outgoing reset
from the side initiating the closing procedure, and doing nothing on the remote
side.
This means that if you call "Close" and the remote endpoint is using an old
version of WebRTC, the channel's state will be stuck at "closing" since the
remote endpoint won't send a reset. Which is already what happens when Firefox
is talking to Chrome.
This CL also fixes an issue where the DataChannel's state prematurely went to
"closed" before the closing procedure was complete. Which could result in a
new DataChannel attempting to re-use the ID and failing.
TBR=magjed@webrtc.org
Bug: chromium:449934, webrtc:4453
Change-Id: Ic1ba813e46538c6c65868961aae6a9780d68a5e2
Reviewed-on: https://webrtc-review.googlesource.com/79061
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23478}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
Previous code had a FakePeriodicVideoSource and a
VideoTrackSource, where the latter is reference counted and
outlives the former. That results in potential races when
RemoveSink is called on the VideoTrackSource after the
FakePeriodicVideoSource is destroyed, with a complicated sequence
to do correct shutdown.
The new class, FakePeriodicVideoTrackSource, owns a
FakePeriodicVideoSource, and they get the same lifetime.
Bug: webrtc:6353
Change-Id: Ic33b393e00a31fa28893dce2018948d3f90e0a9e
Reviewed-on: https://webrtc-review.googlesource.com/76961
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23320}
This is a reland of 5faf36ef3c582350fba5ef97a3549e440d81a283
The issue in Chrome has been fixed and this should be safe to reland.
TBR=deadbeef
Original change's description:
> Implement RtpParameters.transaction_id for PC RtpSenderInterface
>
> The transaction_id field should be refreshed for every getParameters()
> call and checked at each setParameters() call.
> This also checks that getParameters() was ever called to return a proper
> error code.
>
> Bug: webrtc:7580
> Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
> Reviewed-on: https://webrtc-review.googlesource.com/70820
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23120}
Bug: webrtc:7580
Change-Id: Iabd41fb21afdf452c039d5513824ae334f8d1d3f
Reviewed-on: https://webrtc-review.googlesource.com/76980
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23247}
Replaces both BitrateConstraintsMask and
PeerConnectionInterface::BitrateParameters. The latter is kept
temporarily for backwards compatibility.
Bug: None
Change-Id: Ibe1d043f2a76e56ff67809774e9c0f5e0ec9e00f
Reviewed-on: https://webrtc-review.googlesource.com/74020
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23148}
The transaction_id field should be refreshed for every getParameters()
call and checked at each setParameters() call.
This also checks that getParameters() was ever called to return a proper
error code.
Bug: webrtc:7580
Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
Reviewed-on: https://webrtc-review.googlesource.com/70820
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23120}
To prepare for making the software codecs optional and injectable, these
codec factories provide a way to pass in identical factories as were the
default old behaviour.
Bug: webrtc:7925
Change-Id: I0c70fa3c56c999e9d1af6e172eff2fbba849e921
Reviewed-on: https://webrtc-review.googlesource.com/71162
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23096}
This profile will now not be used unless the application explicitly
sets the flag in CryptoOptions to true. As a result, an 80-bit
authentication tag will be used instead of a 32-bit one. See bug for
more details.
Bug: webrtc:7670
Change-Id: I7c0a118fd7b1e7aac23b9eb8717099f055de0441
Reviewed-on: https://webrtc-review.googlesource.com/66600
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22757}
This CL removes PushCaptureData(), which is unused.
The reason I'm removing it is since this method is cauing chromium-style
violations for all files that includes
modules/audio_device/include/audio_device_defines.h, and it's annoying
to suppress it everywhere.
Bug: webrtc:8659
Change-Id: I9133d05259075d8e8ec89b764be934f37b5fa77e
Reviewed-on: https://webrtc-review.googlesource.com/66404
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22717}
