This changes PeerConnection to allow sending and receiving data channel
messages over the media transport. If |use_media_transport_for_data_channels|
is set, PeerConnection will use a DCT_MEDIA_TRANSPORT mode for data
channels.
DCT_MEDIA_TRANSPORT acts exactly like DCT_SCTP within the data channel
and peer connection layers. On the transport layer, it uses the media
transport instead of SCTP. It appears as an RTP data channel in SDP
(just as media over media-transport appears as RTP in SDP).
Bug: webrtc:9719
Change-Id: I6a90142bd3f43668479c825ed02689dcd0d58b78
Reviewed-on: https://webrtc-review.googlesource.com/c/109740
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25575}
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782
This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.
Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.
One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.
Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
The SetChannelParameters function was used when WebRTC supported decoding
with errors, which we no longer do.
This cleanup CL is related to the work tracked by 9946.
Bug: webrtc:9946
Change-Id: Id2d5ed23031388f890c42651bfbe5f79eda701e5
Reviewed-on: https://webrtc-review.googlesource.com/c/108861
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25505}
This enables PeerConnection tests to use LoopbackMediaTransport to test
data-channel-over-media-transport code.
Also changes LoopbackMediaTransport to invoke callbacks asynchronously.
This is more accurate, as these callbacks are triggered by network
events. The caller should not block while the callback executes.
Since LoopbackMediaTransport is used for testing, it provides a
FlushAsyncInvokes() method which may be used to ensure that callbacks
occur deterministically (eg. before checking that data has been
received).
Bug: webrtc:9719
Change-Id: Ib8ea9aebf4a0ad3d5934a6fe4ab33432c68523fd
Tbr: stefan@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/109060
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25489}
IceConnected state (transport state) now includes the state of the
MediaTransport.
This is a first change of two. Second change will add state change
signals to the PeerConnectionInterface informing separately about
ice+media transport vs ice+dtls.
Bug: webrtc:9719
Change-Id: I5731530073e8f26dfc8b188778d268b815da7052
Reviewed-on: https://webrtc-review.googlesource.com/c/108901
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25473}
This change deletes the default implementations of state and data
channel methods (SetMediaTransportStateCallback, SendData, CloseChannel,
and SetDataSink). It adds stub implementations to LoopbackMediaTransport
and FakeMediaTransport.
Bug: webrtc:9719
Change-Id: I49b7780c055b552330546b460c2e79ce8df81833
Reviewed-on: https://webrtc-review.googlesource.com/c/108940
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25457}
This allows to use secure, end to end communication if SDES cryptos are
passed. MediaTransport can use a derived key to secure its own
communication.
Bug: webrtc:9719
Change-Id: If1a20b136b3b4af0cb24f10b52fc5ce1eb31daa2
Reviewed-on: https://webrtc-review.googlesource.com/c/108504
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25452}
Clients of media_transport_interface need the ability to monitor BWE
estimates, and this change adds a TargetBitrate observer to the media
transport interface.
Bug: webrtc:9719
Change-Id: I90ebbf684c6f269e0c3cd58428010cfa511cc970
Reviewed-on: https://webrtc-review.googlesource.com/c/108106
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25415}
There was a suggestion in a previous CL to add an end to end test case to
prevent future regressions. I have enabled this by adding two fakes that
perform fake encryption and enabling an end to end test with VP8 and the
GenericDescriptor.
Bug: webrtc:9927
Change-Id: Icf96eeed541ada1e0579eb81b6f87a46d1c43d96
Reviewed-on: https://webrtc-review.googlesource.com/c/108020
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25399}
This change corrects a potential race condition when updating a FrameEncryptor
for the audio send channel. If a FrameEncryptor is set on an active audio
stream it is possible for the current FrameEncryptor attached to the audio channel to be deallocated due to
the FrameEncryptors reference count reaching zero before the new FrameEncryptor is set on the
channel.
To address this issue the ChannelSend is now holds a scoped_reftptr<FrameEncryptor>
to only allow deallocation when it is actually set on the encoder queue.
ChannelSend is unique in this respect as the Audio Receiver a long with the
Video Sender and Video Receiver streams all recreate themselves when they have
a configuration change. ChannelSend instead reconfigures itself using the
existing channel object.
Added Seth as TBR as this only introduces mocks.
TBR=shampson@webrtc.org
Bug: webrtc:9907
Change-Id: Ibf391dc9cecdbed1874e0252ff5c2cb92a5c64f4
Reviewed-on: https://webrtc-review.googlesource.com/c/107664
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25374}
These two files were using absl::make_unique without #including the
header that declares it.
Bug: None
Change-Id: I03019c9a7e06370631680b474d04dd33716b0fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/107041
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25266}
This is a reland of da65ed2adcfa57ff3288ce01c1602c973fcab00d
Original change's description:
> Reland "Propagate media transport to media channel."
>
> This reverts commit 37cf2455a420124b341ad06ac27fa3c4dbd29d3c.
>
> Reason for revert: <INSERT REASONING HERE>
>
> Original change's description:
> > Revert "Propagate media transport to media channel."
> >
> > This reverts commit 8c16f745ab92cb6d305283e87fa8a661ae500ce4.
> >
> > Reason for revert: Breaks downstream project
> >
> > Original change's description:
> > > Propagate media transport to media channel.
> > >
> > > 1. Pass media transport factory to JSEP transport controller.
> > > 2. Pass media transport to voice media channel.
> > > 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel.
> > >
> > > Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71
> > > Bug: webrtc:9719
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/105542
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > > Reviewed-by: Peter Slatala <psla@webrtc.org>
> > > Commit-Queue: Anton Sukhanov <sukhanov@google.com>
> > > Cr-Commit-Position: refs/heads/master@{#25152}
> >
> > TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com
> >
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> >
> > Bug: webrtc:9719
> > Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1
> > Reviewed-on: https://webrtc-review.googlesource.com/c/105840
> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25154}
>
> TBR=steveanton@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com
>
> Change-Id: I505ff3451eae81573531faef155ff35d7f894022
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9719
> Reviewed-on: https://webrtc-review.googlesource.com/c/106500
> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
> Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25220}
Bug: webrtc:9719
Tbr: Steve Anton <steveanton@webrtc.org>
Tbr: Niels Moller <nisse@webrtc.org>
Change-Id: Ib45691ba8be9abb89ff8c6dac1861bdf59be4c8d
Reviewed-on: https://webrtc-review.googlesource.com/c/106561
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25240}
This reverts commit 37cf2455a420124b341ad06ac27fa3c4dbd29d3c.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Propagate media transport to media channel."
>
> This reverts commit 8c16f745ab92cb6d305283e87fa8a661ae500ce4.
>
> Reason for revert: Breaks downstream project
>
> Original change's description:
> > Propagate media transport to media channel.
> >
> > 1. Pass media transport factory to JSEP transport controller.
> > 2. Pass media transport to voice media channel.
> > 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel.
> >
> > Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71
> > Bug: webrtc:9719
> > Reviewed-on: https://webrtc-review.googlesource.com/c/105542
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Peter Slatala <psla@webrtc.org>
> > Commit-Queue: Anton Sukhanov <sukhanov@google.com>
> > Cr-Commit-Position: refs/heads/master@{#25152}
>
> TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:9719
> Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1
> Reviewed-on: https://webrtc-review.googlesource.com/c/105840
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25154}
TBR=steveanton@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com
Change-Id: I505ff3451eae81573531faef155ff35d7f894022
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/106500
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25220}
Only used for output filename nowadays. Previously, it was used for
selecting the codec implementation. That is now done by injecting
the appropriate codec factory.
Bug: webrtc:9317
Change-Id: Ia2bf28f7df165fb65410ecd1f5d646ee6604e1be
Reviewed-on: https://webrtc-review.googlesource.com/c/106023
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25204}
These files uses absl::WrapUnique or absl::make_unique without including
absl/memory/memory.h. They used to include it indirectly via some other
headers, but in C++17 mode, we need to include it explicitly.
Bug: chromium:752720
Change-Id: Ic9a85a4844a71f8b8786c071f18d5b9cc301c26b
Reviewed-on: https://webrtc-review.googlesource.com/c/105880
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Taiju Tsuiki <tzik@chromium.org>
Cr-Commit-Position: refs/heads/master@{#25192}
Move MockVideoDecoder from
modules/video_coding/include/mock/mock_video_codec_interface.h
to
api/test/mock_video_decoder.h
The mock encoder has already moved:
https://webrtc-review.googlesource.com/c/src/+/105620
Keeping the old header until downstream projects have been updated.
Bug: webrtc:9722
Change-Id: I4bc849173a04813064212f17761876695ca3fed4
Reviewed-on: https://webrtc-review.googlesource.com/c/105900
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25170}
This reverts commit 8c16f745ab92cb6d305283e87fa8a661ae500ce4.
Reason for revert: Breaks downstream project
Original change's description:
> Propagate media transport to media channel.
>
> 1. Pass media transport factory to JSEP transport controller.
> 2. Pass media transport to voice media channel.
> 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel.
>
> Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71
> Bug: webrtc:9719
> Reviewed-on: https://webrtc-review.googlesource.com/c/105542
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Peter Slatala <psla@webrtc.org>
> Commit-Queue: Anton Sukhanov <sukhanov@google.com>
> Cr-Commit-Position: refs/heads/master@{#25152}
TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9719
Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1
Reviewed-on: https://webrtc-review.googlesource.com/c/105840
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25154}
1. Pass media transport factory to JSEP transport controller.
2. Pass media transport to voice media channel.
3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel.
Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/105542
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Anton Sukhanov <sukhanov@google.com>
Cr-Commit-Position: refs/heads/master@{#25152}
Basic integration of media_transport in JSepTransportController.
- Creates media_transport if media transport factory provided in jsep transport controller configuration.
- Unittest that makes sure media_transport is created with correct caller or callee setting.
- Added fake_media_transport, for now simple implementation which only stores caller/callee, but in the future fake media transport will be expanded to pass frames between two fake media_transports, which will enable audio / video integration tests.
NEXT STEPS: Once integration of media_transport with PeerConnection (https://webrtc-review.googlesource.com/c/src/+/103860) lands, we can start passing media transport factory from peer connection to jsep transport controller.
NOTE: Includes missing include change from https://webrtc-review.googlesource.com/c/src/+/103540 (otherwise this change will not compile)
Bug: webrtc:9719
Change-Id: I1e8a521beab445aa9f7ea93c8d7a537dc137d11c
Reviewed-on: https://webrtc-review.googlesource.com/c/104400
Commit-Queue: Anton Sukhanov <sukhanov@google.com>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25096}
It is done to better show what for this class exists and also restore
correspondence between config and interface, that is implemented by
configurable object.
Bug: webrtc:9630
Change-Id: I28456d1c792d67d9b2a405c8599054137a5d596a
Reviewed-on: https://webrtc-review.googlesource.com/c/104003
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25041}
Add FakeVp8Decoder that parse width and height from the payload.
Add unit test for testing that width and height is set when decoding frames.
Bug: none
Change-Id: Ifbfff4f62f99625309ce0ef21cf89c76448769d8
Reviewed-on: https://webrtc-review.googlesource.com/c/103140
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25038}
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}
This name will better describe what implementation should do and how
users will interact with this class. The real simulation is done
by FakeNetworkPipe and this class is just operates with packages
metadata, so it is more about describing behavior.
Bug: webrtc:9630
Change-Id: I00977e6be0ca84e7c233b4c35f0677e8263e4382
Reviewed-on: https://webrtc-review.googlesource.com/c/95944
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24989}
Some important NetEq information was not available in NetEqState, which
meant it was not available on the API. This CL adds additional
information.
Bug: webrtc:9667
Change-Id: I702707c7d60472f488047d48fb286f839c5608dc
Reviewed-on: https://webrtc-review.googlesource.com/c/102300
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24985}
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}
This reverts commit 3f4a4fad8cd661309ff5d9a631e89518f32e7c5e.
Reason for revert: Breaking internal tests
Original change's description:
> Added field trial WebRTC-GenericDescriptor for the new generic descriptor.
>
> Also parameterized tests to test the new generic descriptor and
> added --generic_descriptor flag to loopback tests.
>
> Bug: webrtc:9361
> Change-Id: I2b76889606fe2e81249ecdebb0b7b61151682485
> Reviewed-on: https://webrtc-review.googlesource.com/101900
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24835}
TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org
Change-Id: I4d4714a9f4ab0e95adf0f4130bc1a932efc448fa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9361
Reviewed-on: https://webrtc-review.googlesource.com/101940
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24839}
Also parameterized tests to test the new generic descriptor and
added --generic_descriptor flag to loopback tests.
Bug: webrtc:9361
Change-Id: I2b76889606fe2e81249ecdebb0b7b61151682485
Reviewed-on: https://webrtc-review.googlesource.com/101900
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24835}
Replace calls to .str() which copies with .Release which moves in cases where that's safe.
This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"
Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
It is problematic to set field trials more than once, so to avoid running into problems, this functionality has been placed in the main function of neteq_rtpplay.
Bug: webrtc:9667
Change-Id: Ib9b9990f30a1715b50889dbfc4d74787bcbe5dae
Reviewed-on: https://webrtc-review.googlesource.com/98541
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24673}
This move constructor causes downstream issues, so it needs to be removed for now.
Bug: webrtc:9667
Change-Id: Ic15bfdf6b392a95e05bf75bc2c1dd32ce132d32b
Reviewed-on: https://webrtc-review.googlesource.com/99121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24672}
This CL adds more useful information to NetEqState, and implements setting action_times_ms, which can be used to get a better idea of what actually happened during a timestep.
Bug: webrtc:9667
Change-Id: I789a3e1ad852066fdf4e9b4c96b8fb6033dacb27
Reviewed-on: https://webrtc-review.googlesource.com/98163
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24643}
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.
The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.
Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
