This CL moves the responsibility for demuxing from FakeNetworkPipe
to DirectTransport. This makes the interface for FakeNetworkPipe more
consistent. It exposes fewer different interfaces for different usages.
It also means that any time degradations applied to the packets due in
FakeNetworkPipe in tests will now be propagated to Call in a more
realistic manner. Previously the time was set to uninitialized which
meant that Call filled in values based on the system clock.
Bug: webrtc:9054
Change-Id: Ie534062f5ae9ad992c06b19e43804138a35702f0
Reviewed-on: https://webrtc-review.googlesource.com/64260
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23017}
This is used to allow mocking in tests in later CLs.
Bug: None
Change-Id: Id610471efb4a86c903530585dd4ee2fa1d1ea5bc
Reviewed-on: https://webrtc-review.googlesource.com/70880
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22941}
This CL adds reporting of per packet feedback availability from Call
via RtpTransportControllerSend to SendSideCongestionController.
This is part of a series of CLs tracking the transport feedback status
of the streams known to BitrateAllocator and reporting the status to
the congestion controller.
Bug: webrtc:8415
Change-Id: I20b3dbb4a027c46476bc2d2bc875374bff05609a
Reviewed-on: https://webrtc-review.googlesource.com/63220
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22566}
Removing functions that has been removed from
RtpTransportControllerSendInterface from
MockRtpTransportControllerSend.
Deleted functions: GetPacerModule, GetModule,
SetTransportOverhead and AvailableBandwidth.
Bug: webrtc:8415
Change-Id: I24d460bd18d57966e3b333ce0c234c3e3dc19a9a
Reviewed-on: https://webrtc-review.googlesource.com/62762
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22489}
This is especially useful in Chrome, allowing use to emulate network
conditions in incoming or outgoing media without the need for platform
specific tools or hacks. It also doesn't interfere with the rest of the
network traffic.
Also includes some refactorings.
TBR=stefan@webrtc.org, philipel@webrtc.org
Originally reviewed on: https://webrtc-review.googlesource.com/33013
Bug: webrtc:8910
Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0
Reviewed-on: https://webrtc-review.googlesource.com/61782
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22430}
This reverts commit 31a12c557dcd84a31f9c3f2d8858d9646c2a3135.
Reason for revert: Breaks downstream project.
Original change's description:
> Add ability to emulate degraded network in Call via field trial
>
> This is especially useful in Chrome, allowing use to emulate network
> conditions in incoming or outgoing media without the need for platform
> specific tools or hacks. It also doesn't interfere with the rest of the
> network traffic.
>
> Also includes some refactorings.
>
> Bug: webrtc:8910
> Change-Id: I2656a2d4218acbe7f8ffd669de19a02275735438
> Reviewed-on: https://webrtc-review.googlesource.com/33013
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22418}
TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org
Change-Id: I22bda6da01c2ff5abd6f408c5ee9e4fba21294f2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8910
Reviewed-on: https://webrtc-review.googlesource.com/61700
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22419}
This is especially useful in Chrome, allowing use to emulate network
conditions in incoming or outgoing media without the need for platform
specific tools or hacks. It also doesn't interfere with the rest of the
network traffic.
Also includes some refactorings.
Bug: webrtc:8910
Change-Id: I2656a2d4218acbe7f8ffd669de19a02275735438
Reviewed-on: https://webrtc-review.googlesource.com/33013
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22418}
Removing the Synchronous call AvailableBandwidth from the
RtpTransportControllerSend interface. The bandwidth estimate is
provided trough a new interface that communicates with a struct
making it easier to add parameters in the future.
This prepares for removing locking behavior in
SendSideCongestionController that exists just to support this feature.
To keep backwards compatibility with the old
SendSideCongestionController, the struct TargetTransferRate
is constructed in RtpTransportControllerSend. This step can be
removed in the future when the old SendSideCongestionController
is deprecated.
Bug: webrtc:8415
Change-Id: I06f64a89848157de412901c989650d1ecf35246b
Reviewed-on: https://webrtc-review.googlesource.com/60800
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22387}
Ownership of the retransmission rate limiter for video is moved
from send side congestion controller to Call. This is to reduce the
interface on the rtp transport controller send.
Bug: webrtc:8415
Change-Id: Ie9c7317400a9eb61a3c8325b9e527844ffc13769
Reviewed-on: https://webrtc-review.googlesource.com/58745
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22254}
DeRegisterNetworkObserver is not used, since
RtpTransportControllerSend owns the thread on which
SendSideCongestionController runs it would not be safe to allow it to be
called from outside. Deregistration should be done by destroying
RtpTransportControllerSend.
In the future, the RegisterObserver functions should be removed as well,
in favor of providing the observer in the constructor. This requires
breaking a circular dependency between RtpTransportControllerSend and
Call.
Bug: webrtc:8415
Change-Id: Ifeb4c5d4a41e4d8419994b3146980bdaaf9cd6a9
Reviewed-on: https://webrtc-review.googlesource.com/58098
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22249}
Added dependencies used by MockRtpTransportControllerSend to its header
file. The mocked interface can't be used properly without those.
This prepares for later CLs utilizing the mock.
Bug: None
Change-Id: I5f8ca04032ad09810240f7c034cc628d700dbedb
Reviewed-on: https://webrtc-review.googlesource.com/56181
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22154}
Moving usage of bitrate configuration related interface from Call
interface to the corresponding methods in the RtpSendTransportController
interface.
SetBitrateConfig was replaced with SetSdpBitrateParameters
SetBitrateConfigMask was replaced with SetClientBitratePreferences
OnNetworkRouteChanged was replaced with OnNetworkRouteChanged
This makes it more clear that RtpSendTransportController owns bitrate
configuration and fits a longer term ambition to reduce the scope of
the Call class.
Bug: webrtc:8415
Change-Id: I6d04eaad22a54ecd5ed60096e01689b0c67e9c65
Reviewed-on: https://webrtc-review.googlesource.com/54365
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22131}
TransmitMixer's functionality is moved into the AudioTransportProxy
owned by AudioState. This removes the need for an AudioTransport
implementation in VoEBaseImpl, which means that the proxy is no longer
a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl.
In the short term, AudioState needs to know which AudioDeviceModule is
used, so it is added in AudioState::Config. AudioTransportImpl needs
to know which AudioSendStream:s are currently enabled to send, so
AudioState maintains a map of them, which is reduced into a simple
vector for AudioTransportImpl.
To encode and transmit audio,
AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame)
is introduced, which is used in both the Chromium and standalone use
cases. This removes the need for two different instances of
voe::Channel::ProcessAndEncodeAudio(), so there is now only one,
taking an AudioFrame as argument. Callers need to allocate their own
AudioFrame:s, which is wasteful but not a regression since this was
already happening in the voe::Channel functions.
Most of the logic changed resides in
AudioTransportImpl::RecordedDataIsAvailable(), where two strange
things were found:
1. The clock drift parameter was ineffective since
apm->echo_cancellation()->enable_drift_compensation(false) is
called during initialization.
2. The output parameter 'new_mic_volume' was never set - instead it
was returned as a result, causing the ADM to never update the
analog mic gain
(https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100).
Besides this, tests are updated, and some dead code is removed which
was found in the process.
Bug: webrtc:4690, webrtc:8591
Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2
Reviewed-on: https://webrtc-review.googlesource.com/26681
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21301}
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}
