Add a method to DtlsSrtpTransport to cache the RTP Absolute Send Time
extension id. The method would be called when using DTLS-SRTP with
external authentication.
TBR=pthatcher@webrtc.org
Bug: webrtc:7013
Change-Id: Ie9edb9382cbb4cf43eea5da3030991a0d20293a5
Reviewed-on: https://webrtc-review.googlesource.com/27260
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20947}
This is a reland of 20f2133d5dbd1591b89425b24db3b1e09fbcf0b1
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}
TBR=danilchap@webrtc.org, sprang@webrtc.org, niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: If8f0c7e6e3a2a704f19161f0e8bf1880906e7fe0
Reviewed-on: https://webrtc-review.googlesource.com/27160
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20946}
This was only used to ensure compatibility with external projects.
Now that everyone has moved to RtpTransceiverDirection, these
methods are redundant.
Bug: webrtc:8558
Change-Id: Iff5a8d13f9a4300d06902fa0441ceaeebf6809a2
Reviewed-on: https://webrtc-review.googlesource.com/24746
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20945}
This reverts commit 94f39301067b9fbf820100cbd4018aad3a32cc52.
Reason for revert: Broke internal builds
Original change's description:
> Removes deprecated ADM APIs (reland).
>
> Usage should now be removed and this change can be relanded.
> It was reverted here: https://webrtc-review.googlesource.com/c/src/+/25320
>
> TBR=solenberg
>
> Bug: webrtc:7306
> Change-Id: I1afea773eff51bf5ec80711f0d7753ac0b7be77b
> Reviewed-on: https://webrtc-review.googlesource.com/27000
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20936}
TBR=henrika@webrtc.org
Change-Id: If91ff815fa69f7c36b0531e295f553a8c4a95590
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7306
Reviewed-on: https://webrtc-review.googlesource.com/27221
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20944}
The DtlsSrtpTransport takes the reponsiblity of setting up DTLS-SRTP from
the BaseChannel.
The BaseChannel doesn't handle the signals from the P2P layer transport anymore.
The RtpTransport handles the signals from the PacketTransportInternal and the
DtlsSrtpTransport handles the DTLS-specific signals and determines when to extract
the keys and setting the parameters.
In channel_unittests.cc, call from DTLS to SDES is expected to fail since the
fallback from DTLS to SDES is not supported.
Bug: webrtc:7013
Change-Id: I0a54e017986f5a8ae9710e79643a4651bef3c38f
Reviewed-on: https://webrtc-review.googlesource.com/24702
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20941}
This is a minor change to generated Python code used for testing the echo likelihood metric.
Bug: webrtc:8573
Change-Id: Ifb2438fdd36c3ade8cd830df0d05917af0f77dec
Reviewed-on: https://webrtc-review.googlesource.com/26281
Commit-Queue: Daniel Johansson <dajo@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20939}
For uniformity. Uniformity is nice.
Bug: none
Change-Id: I3156c4db1f6f261ba035cf95b632fd413c8afc2a
Reviewed-on: https://webrtc-review.googlesource.com/25482
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20937}
Usage should now be removed and this change can be relanded.
It was reverted here: https://webrtc-review.googlesource.com/c/src/+/25320
TBR=solenberg
Bug: webrtc:7306
Change-Id: I1afea773eff51bf5ec80711f0d7753ac0b7be77b
Reviewed-on: https://webrtc-review.googlesource.com/27000
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20936}
This makes it visible that there are no side effects and no dependency on BitrateAllocator.
Bug: None
Change-Id: I3d54ea545e694ae8303860114ddb3ce7569ecb14
Reviewed-on: https://webrtc-review.googlesource.com/26920
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20933}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=mbonadei@webrtc.org
Bug: None
Change-Id: Iec336d342414dc68b59ba4b4623fdf768f6fb655
Reviewed-on: https://webrtc-review.googlesource.com/23602
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20932}
This reverts commit 20f2133d5dbd1591b89425b24db3b1e09fbcf0b1.
Reason for revert: Breaks downstream project.
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}
TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org,emircan@webrtc.org
Change-Id: I57f3172ca3c60a84537d577a574dc8018e12d634
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/26940
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20931}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=sprang@webrtc.org
Bug: None
Change-Id: I50d25d6174486928963c2e98455587a8a9f0bee6
Reviewed-on: https://webrtc-review.googlesource.com/23616
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20930}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
Bug: None
Change-Id: Ia65be19b24c93db360a313f82a84bfae1a49bf2d
Reviewed-on: https://webrtc-review.googlesource.com/23605
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20929}
This reverts commit aede67a199ae0552074bfec4bb03cc9a6a5fba0f.
Reason for revert: Causes error:
JNI ERROR (app bug): local reference table overflow (max=512)'
Original change's description:
> Android: Generate JNI code for stats
>
> This CL also unifies the functions for converting from C++ to Java, and
> generates the boiler plate for converting C++ vectors to Java arrays.
>
> Bug: webrtc:8278
> Change-Id: I262e9162beae8a64ba0e8b6a27e1081207b03961
> Reviewed-on: https://webrtc-review.googlesource.com/26020
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20918}
TBR=magjed@webrtc.org,sakal@webrtc.org
Change-Id: Ieb26ed8577bd489a4dd4f7542d16a7d0e11f409f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8278
Reviewed-on: https://webrtc-review.googlesource.com/26900
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20926}
The UMA metric will log the same information that goes into the
googPreferredJitterBufferMs stat.
Bug: webrtc:8488
Change-Id: I4e4e1e362dd42377105d52d2c4cd49c1ecb1a90d
Reviewed-on: https://webrtc-review.googlesource.com/26740
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20923}
This is causing compilation issues in a chromium cl because of type conflicts.
BUG=none
TBR=henrikg@webrtc.org
Tbr-ing to fix build issue upstream and because there's no code change.
Change-Id: Ia34ae3844fe3f57f047cb44422fa591f752b7bda
Reviewed-on: https://webrtc-review.googlesource.com/26680
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20921}
- Defines CodecSpecificInfoStereo that carries stereo specific header info from
encoded image.
- Defines RTPVideoHeaderStereo that carries the above info to packetizer,
see module_common_types.h.
- Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
header.
- Uses new data containers in StereoAdapter classes.
This CL is the step 3 for adding alpha channel support over the wire in webrtc.
See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
Reviewed-on: https://webrtc-review.googlesource.com/22900
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20920}
This CL also unifies the functions for converting from C++ to Java, and
generates the boiler plate for converting C++ vectors to Java arrays.
Bug: webrtc:8278
Change-Id: I262e9162beae8a64ba0e8b6a27e1081207b03961
Reviewed-on: https://webrtc-review.googlesource.com/26020
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20918}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=tommi@webrtc.org
Bug: None
Change-Id: I0ca1b624859a6561e227480b7dac8c254d26ad57
Reviewed-on: https://webrtc-review.googlesource.com/23562
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20916}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=tina.legrand@webrtc.org
Bug: None
Change-Id: Iea6f04db7c1f92fe9da2c855bb60ad2f70c371d3
Reviewed-on: https://webrtc-review.googlesource.com/23615
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20915}
The random square generator produces unrealistically complex frames in
some situations, leading to frames > 250kb even at max QP. This leads to
unmanageably long transmission delays.
Bug: None
Change-Id: I8f5a33d52fb5efa03de97e529ad598b75511f679
Reviewed-on: https://webrtc-review.googlesource.com/23561
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20912}
class to pass packet to RtcpPacketParser
This helpers make tests setup cleaner and
makes explicit expectation on number of packets passed to the transport.
Bug: webrtc:8239
Change-Id: I2d5975be59327cee440e87dbd0701b93514c9726
Reviewed-on: https://webrtc-review.googlesource.com/22460
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20911}
This tests NetEq with a stream encoded with Opus using it's internal
DTX/CNG.
Also adding a new resource file which is encoded using Opus with DTX.
Bug: webrtc:8488
Change-Id: Icfba5bc5dc7f9c9d0e637a90f4df674e8ba40358
Reviewed-on: https://webrtc-review.googlesource.com/26028
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20905}
The root cause of the flakiness is unknown, the possible issue is that the
console window running the test case is hidden or minimized. So this change
adds a SetWindow(SW_MAXIMIZE) to ensure the console window is showing.
I have run the tests against win_asan for hundreds times during the
thanksgiving. So far, no flakiness were caught.
Bug: webrtc:8568
Change-Id: Ib2c93e9bd511257213254bdaa0079c14ea50f3e4
Reviewed-on: https://webrtc-review.googlesource.com/25286
Reviewed-by: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20902}
To make desktopCapture autotest in chromium more meaningful, it's better
to creake fake capturer of each capture type. Here we can reuse the
FakeDesktopCapturer for window capture in chromium.
Bug: chromium:699201
Change-Id: Icbe134d99cbd4980bf27fe74c1c629a1469836ea
Reviewed-on: https://webrtc-review.googlesource.com/26360
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20901}
This change adds |transceiver_direction()| and
|set_transceiver_direction()| to MediaContentDescription so that
external users can switch off of MediaContentDirection.
This deprecates the use of |direction()| and |set_direction()|
for external users. Once everyone has moved off of those methods,
the signiture will change to return/set RtpTransceiverDirection.
Then external users can move back to these methods with the new
signature.
Bug: webrtc:8558
Change-Id: I7e3ba289d3a0ac738b364b0388621cc3e7bcf5d3
Reviewed-on: https://webrtc-review.googlesource.com/24743
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20900}
