We use ArrayView in our public API, so its header should be in
webrtc/api/.
BUG=none
Review-Url: https://codereview.webrtc.org/3007763002
Cr-Commit-Position: refs/heads/master@{#19658}
The SrtpTransport takes the SRTP responsibilities from the BaseChannel
and SrtpFilter. SrtpTransport is now responsible for setting the crypto
keys, protecting and unprotecting the packets. SrtpTransport doesn't know
if the keys are from SDES or DTLS handshake.
BaseChannel is now only responsible setting the offer/answer for SDES
or extracting the key from DtlsTransport and configuring the
SrtpTransport.
SrtpFilter is used by BaseChannel as a helper for SDES negotiation.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2997983002
Cr-Commit-Position: refs/heads/master@{#19636}
This is the first step towards supporting multiple audio/video
channels in PeerConnection/WebRtcSession. For now, there can only
be 0 or 1 channels in the vector. This adds the framework so that
all the other code that assumes a single audio/video channel can
be transitioned one-by-one to multiple channels.
Bug: webrtc:8183
Change-Id: I6456af32d6e3adf7eb83e281e43253ea973c4eb9
Reviewed-on: https://chromium-review.googlesource.com/644222
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19615}
Also clarify the ownership of created channels.
Bug: None
Change-Id: I9cbaec177069d27da2b3b48b93af48f705243b4b
Reviewed-on: https://chromium-review.googlesource.com/643950
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19612}
directives in our DEPS files are not needed anymore.
Includes from webrtc/rtc_base are also whitelisted in webrtc/DEPS
so we don't have to whitelist it in all the others DEPS files.
BUG=webrtc:7634
NOTRY=True
Review-Url: https://codereview.webrtc.org/3006583002
Cr-Commit-Position: refs/heads/master@{#19601}
Replacing the RTC_DCHECK with an if condition so that the method GetOptionsForAnswer
won't crash when there is no remote description. WebRtcSessionDescriptionFactory
will handle the null remote description.
TBR=deadbeef@webrtc.org
BUG=chromium:757830
Review-Url: https://codereview.webrtc.org/3006723002
Cr-Commit-Position: refs/heads/master@{#19589}
Will remove default implementations as well once landed and removed
in Chrome as well.
These two AudioDeviceModule APIs are removed:
int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight)
int32_t WaveOutVolume(uint16_t* volumeLeft, uint16_t* volumeRight) const
BUG=webrtc:7306
Review-Url: https://codereview.webrtc.org/3006793002
Cr-Commit-Position: refs/heads/master@{#19581}
The following fields are logged:
- error, if there was an error.
- elapsed time in milliseconds
BUG=webrtc:8100
Review-Url: https://codereview.webrtc.org/2996933003
Cr-Commit-Position: refs/heads/master@{#19574}
Adds two new stats to RTCMediaStreamTrackStats:
* totalSamplesReceived is the total number of samples received on
the audio channel and includes real and synthetic samples.
* concealedSamples is the total number of synthetic samples
received on the audio channel used to conceal packet loss.
Bug: webrtc:8076
Change-Id: I36e9828525ec341490cf3310a972b56a8b443667
Reviewed-on: https://chromium-review.googlesource.com/615902
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19506}
This test doesn't actually send/receive any audio; it's only testing
the interface layer. But the fact that it was creating/destroying real
audio devices repeatedly caused problems when tests were run in
parallel. So switching to a fake audio device solves this.
BUG=webrtc:7806
Review-Url: https://codereview.webrtc.org/2997383002
Cr-Commit-Position: refs/heads/master@{#19472}
Maximum of interframe delay is calculated over moving window in
ReceiveStatistics proxy now and reported via GetStats. Name of a metric
is also changed.
BUG=none
Review-Url: https://codereview.webrtc.org/2995143002
Cr-Commit-Position: refs/heads/master@{#19463}
no-unused-lambda-capture was suppressed, but it's been decided as desireable to stop suppressing it. This CL fixes places in the code that trigger it.
1. Some unnecessary captures removed.
2. s/constexpr/const when capturing a float by value - this is good enough to stop the error.
3. Complete removal of the constexpr/const-modifier for int-types as a workaround.
BUG=webrtc:7133
Review-Url: https://codereview.webrtc.org/3005433002
Cr-Commit-Position: refs/heads/master@{#19462}
This layer takes in a simplified "options" struct and the current local description,
and generates a new offer/answer. Previously the options struct assumed there would
only be one media description per media type (audio/video), but it now supports
N number of audio/video descriptions.
The |add_legacy_stream| options is removed from the mediasession.cc/.h
in this CL.
The next step is to add the ability for PeerConnection/WebRtcSession to create
"options" to represent multiple RtpTransceivers, and apply the Unified Plan
descriptions correctly. Right now, only Plan B descriptions will be
generated in unit tests.
BUG=chromium:465349
Review-Url: https://codereview.webrtc.org/2991693002
Cr-Original-Commit-Position: refs/heads/master@{#19343}
Committed: a77e6bbd30
Review-Url: https://codereview.webrtc.org/2991693002
Cr-Commit-Position: refs/heads/master@{#19394}
Reason for revert:
BUG=webrtc:8108: breaks Clang build.
Original issue's description:
> Adding support for Unified Plan offer/answer negotiation to the mediasession layer.
>
> This layer takes in a simplified "options" struct and the current local description,
> and generates a new offer/answer. Previously the options struct assumed there would
> only be one media description per media type (audio/video), but it now supports
> N number of audio/video descriptions.
>
> The |add_legacy_stream| options is removed from the mediasession.cc/.h
> in this CL.
>
> The next step is to add the ability for PeerConnection/WebRtcSession to create
> "options" to represent multiple RtpTransceivers, and apply the Unified Plan
> descriptions correctly. Right now, only Plan B descriptions will be
> generated in unit tests.
>
> BUG=chromium:465349
>
> Review-Url: https://codereview.webrtc.org/2991693002
> Cr-Commit-Position: refs/heads/master@{#19343}
> Committed: a77e6bbd30TBR=deadbeef@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:465349
Review-Url: https://codereview.webrtc.org/3001083002
Cr-Commit-Position: refs/heads/master@{#19384}
Reason for revert:
It breaks a downstream project.
Original issue's description:
> Trace the stats report as JSON instead of each stat separately.
>
> Trace the whole report as a string instead of each field on it's own. And test that the traces collected are valid.
>
> R=tommi@webrtc.org, hbos@webrtc.org
> BUG=chromium:653087
>
> Review-Url: https://codereview.webrtc.org/2986453002
> Cr-Commit-Position: refs/heads/master@{#19341}
> Committed: 80c65780e6TBR=hbos@webrtc.org,tommi@webrtc.org,ehmaldonado@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:653087
Review-Url: https://codereview.webrtc.org/3001683002
Cr-Commit-Position: refs/heads/master@{#19344}
This layer takes in a simplified "options" struct and the current local description,
and generates a new offer/answer. Previously the options struct assumed there would
only be one media description per media type (audio/video), but it now supports
N number of audio/video descriptions.
The |add_legacy_stream| options is removed from the mediasession.cc/.h
in this CL.
The next step is to add the ability for PeerConnection/WebRtcSession to create
"options" to represent multiple RtpTransceivers, and apply the Unified Plan
descriptions correctly. Right now, only Plan B descriptions will be
generated in unit tests.
BUG=chromium:465349
Review-Url: https://codereview.webrtc.org/2991693002
Cr-Commit-Position: refs/heads/master@{#19343}
That way, the debug printout will tell us which of x and y that was false.
BUG=none
Review-Url: https://codereview.webrtc.org/2988153003
Cr-Commit-Position: refs/heads/master@{#19297}
[This CL is work in progress.]
Wire up the rtp keep-alive in webrtc::Call::Config using new
SetRtpTransportParameters() method on RtpTransportInterface.
BUG=webrtc:7907
Review-Url: https://codereview.webrtc.org/2981513002
Cr-Commit-Position: refs/heads/master@{#19287}
Follow up to https://codereview.webrtc.org/2989243002/.
It turns out that "b=AS:-1" was being used to mean "no bandwidth limit",
even though just omitting "b=AS" completely will do that. So we should
treat this as a soft error for now, and give applications time to
transition to doing the standard thing.
BUG=chromium:675361
Review-Url: https://codereview.webrtc.org/2995463002
Cr-Commit-Position: refs/heads/master@{#19244}
It doesn't make sense to have a negative RTP session bandwidth; RFC3550
doesn't define any meaning for this. So just treat it as invalid SDP.
BUG=chromium:675361
Review-Url: https://codereview.webrtc.org/2989243002
Cr-Commit-Position: refs/heads/master@{#19221}
Added documentation of thread expectations for video tracks and sources to the API.
Originally landed as patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/.
Patchset 1 is the originall cl.
Patschet 2 is modified so that VideoTrackInterface::AddSink and RemoveSink have a default implementation.
BUG=none
Review-Url: https://codereview.webrtc.org/2989113002
Cr-Commit-Position: refs/heads/master@{#19195}
This allows an application to easily override the default limit
(currently 5).
Also adding a test that covers more of the
PeerConnection<->PortAllocator interaction.
BUG=webrtc:7703
Review-Url: https://codereview.webrtc.org/2985653003
Cr-Commit-Position: refs/heads/master@{#19160}
There is an inconsistency in behavior of PeerConnection.
When I remove track from PeerConnection observer->OnRenegotiationNeeded is called, however if I remove track from MediaStream then there is no notification to renegotiate.
This patch adds missing OnRenegotiationNeeded calls.
BUG=webrtc:7966
Review-Url: https://codereview.webrtc.org/2977493002
Cr-Commit-Position: refs/heads/master@{#19125}
SrtpTransport currently just delegates everything to RtpTransport.
Also makes BaseChannel::rtp_transport_ an RtpTransportInternal and constructs an SrtpTransport if srtp is required.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2981013002
Cr-Commit-Position: refs/heads/master@{#19095}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
This reverts commit 26d5e2e2809558148dc1e977ec1bc8318a2047bc.
Reverted originally because it dependend on a CL which was reverted. That CL has been reinstated in: https://chromium-review.googlesource.com/#/c/572070/
Bug: webrtc:7969
Change-Id: I404c3a42ad447312d981646dca0aa4cf0ec3134e
Reviewed-on: https://chromium-review.googlesource.com/572403
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19034}
Use rtc::SystemTimeNanos() instead of std::random_device() for PRNG seed
to avoid crashing when /dev/urandom is unavailable.
This reverts commit 3beb20720db349f651c2c04970c45b1b171c025c.
Bug: webrtc:7969
Change-Id: I5ed58a789939ee4caa99ac3abf9cab18e3e19c69
Reviewed-on: https://chromium-review.googlesource.com/572070
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19033}
This is a partial re-land. The change doesn't make the default Thread ctor protected anymore but it does mark it as deprecated and updates all use of it in WebRTC.
Original issue's description:
Make the default ctor of rtc::Thread, protected.
The goal is to force use of Thread::Create or Thread::CreateWithSocketServer.
The default constructor constructs a 'default' socket server, which is usually a 'physical' socket server, but not always. Not every instance of Thread actually needs to have network support, so it's better to have this be explicit instead of unknowingly instantiate one.
BUG=none
Review-Url: https://codereview.webrtc.org/2977953002
Cr-Commit-Position: refs/heads/master@{#19031}
This reverts commit aa41f0cfa64ece911ae2ecee83fc3190d4a42935.
Reason for revert:
Apparently, use of std::random_device() causes chromium on Linux to fail with this error:
terminating with uncaught exception of type std::__1::system_error: random_device failed to open /dev/urandom: Operation not permitted
Link to bot with failure:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Tester/builds/37563
Original change's description:
> API for periodically regathering ICE candidates
>
> Adds to the RTCConfiguration `ice_regather_interval_range` which, when
> set, specifies the randomized delay between automatic runs of ICE
> regathering. The regathering will occur on all networks and re-use the
> existing ICE ufrag/password. New connections are established once the
> candidates come back and WebRTC will automatically switch to the new
> connection that corresponds to the currently selected connection.
>
> Bug: webrtc:7969
> Change-Id: I6bbf5439a48e285f704aed9f408631cba038c82b
> Reviewed-on: https://chromium-review.googlesource.com/562505
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18978}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,steveanton@webrtc.org
No-Try: true
Bug: webrtc:7969
Change-Id: I86ef99e9f1070d3ac265398831317b68f562c614
Reviewed-on: https://chromium-review.googlesource.com/571008
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19024}
Check that ice_regather_interval_range is set only when continual
regathering is also set.
Bug: webrtc:7969
Change-Id: Ifcfeee744d817cf00914418d7e682f11528faf05
Reviewed-on: https://chromium-review.googlesource.com/569358
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19009}
Reason for revert:
Break projects.
Original issue's description:
> Make the default ctor of rtc::Thread, protected.
> The goal is to force use of Thread::Create or Thread::CreateWithSocketServer.
>
> The default constructor constructs a 'default' socket server, which is usually a 'physical' socket server, but not always. Not every instance of Thread actually needs to have network support, so it's better to have this be explicit instead of unknowingly instantiate one.
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2981623002
> Cr-Commit-Position: refs/heads/master@{#19001}
> Committed: a8a3515997TBR=kthelgason@webrtc.org,tommi@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/2979963002
Cr-Commit-Position: refs/heads/master@{#19003}
The goal is to force use of Thread::Create or Thread::CreateWithSocketServer.
The default constructor constructs a 'default' socket server, which is usually a 'physical' socket server, but not always. Not every instance of Thread actually needs to have network support, so it's better to have this be explicit instead of unknowingly instantiate one.
BUG=none
Review-Url: https://codereview.webrtc.org/2981623002
Cr-Commit-Position: refs/heads/master@{#19001}
streams.
When addTrack/removeTrack is used instead of addStream/removeStream
we an end up with tracks that are not contained within any local or
remote stream.
If all track IDs are not mapped when we produce RTCRTPStreamStats
we'll hit a DCHECK.
BUG=chromium:741638
Review-Url: https://codereview.webrtc.org/2978793002
Cr-Commit-Position: refs/heads/master@{#18991}
