1. Bitrate allocation strategy unit test
2. Perf test determining minimal supported bitrate with and without strategy
Bug: webrtc:8243
Change-Id: Idf675fbadddb66c77b2582052d6497971eb99ad6
Reviewed-on: https://webrtc-review.googlesource.com/4880
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20886}
So that third party projects don't still have to implement it when they
switch over to the new signature.
Bug: webrtc:8473
Change-Id: I329814ad6e899def7bad97939e8643380a268f91
Reviewed-on: https://webrtc-review.googlesource.com/26022
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20885}
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: I2c1e50090e48c892fda74f1678231a7624cac9fa
Reviewed-on: https://webrtc-review.googlesource.com/23575
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20884}
For better consistency between the Objective-C API constant definitions
and the existing constants defined in the underlying core, re-use the
available video codec-name constants from cricket to define the peer
constants in the public API.
BUG=None
Change-Id: I8d5ddc2c1bd6670810fca1665aaf9a116620a34e
Reviewed-on: https://webrtc-review.googlesource.com/25360
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20883}
This reverts commit b54bc06079a66a379595068f6ea56d56a4fd95c0.
Reason for revert: Downstream projects should be fixed now
Original change's description:
> Let PrintResultMeanAndError accept a string.
>
> Some downstream projects still use it. I'll update them and then revert
> this change.
>
> Bug: webrtc:8566
> Change-Id: Ib4e56348c40a3645f3049382b47089ca6c675e96
> Reviewed-on: https://webrtc-review.googlesource.com/25841
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20872}
TBR=phoglund@webrtc.org,ehmaldonado@webrtc.org
Change-Id: I5d2a59cbfb6a148fc6c621a69fa23397ba2c6991
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8566
Reviewed-on: https://webrtc-review.googlesource.com/25920
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20880}
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: Iedebf4dc56a973306e7d7e7649525879808dc72b
Reviewed-on: https://webrtc-review.googlesource.com/23578
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20878}
The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM.
Bug: webrtc:8563, b/67926135
Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24
Reviewed-on: https://webrtc-review.googlesource.com/25621
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20877}
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=henrikg@webrtc.org
Bug: None
Change-Id: I8cb71d015813312ccfc1b6fc7a2a27f391aaf4d1
Reviewed-on: https://webrtc-review.googlesource.com/23614
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20876}
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=terelius@webrtc.org
Bug: None
Change-Id: I18b44f6c1bd3ccb4288807eee883b87afa7900f4
Reviewed-on: https://webrtc-review.googlesource.com/23569
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20875}
This test produces a consistent stream of false positive alerts, and I have been unable to make it more robust, despite several attempts. It also has never managed to catch a real regression, so I think it is better to remove it.
Bug: chromium:788318
Change-Id: I7e9731834f67af1ef2fa15a655e620bd64a4cfde
Reviewed-on: https://webrtc-review.googlesource.com/25824
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20874}
Some downstream projects still use it. I'll update them and then revert
this change.
Bug: webrtc:8566
Change-Id: Ib4e56348c40a3645f3049382b47089ca6c675e96
Reviewed-on: https://webrtc-review.googlesource.com/25841
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20872}
GetBandwidthObserver should be used instead as it exposes a smaller interface.
Bug: webrtc:8415
Change-Id: I29ca795657e205186d7ebd929e756038a294b5f7
Reviewed-on: https://webrtc-review.googlesource.com/23900
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20871}
The audio processing module reports the metrics 'echo return loss'
and 'echo return loss enhancement' for AEC3 via the old, soon to be
replaced, GetStatistics fuction.
Bug: webrtc:8533
Change-Id: I6b2286b5cdf8f20ebf14f82f1180f6bfb6c00c68
Reviewed-on: https://webrtc-review.googlesource.com/25642
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20869}
This new tool provides the similar functionality as the legacy tool, but
is implemented using less legacy helpers. It also replaces RTPtimeshift
and RTPchange. The most significant change versus the old RTPjitter tool
is that the new tool takes the timing data in the form of integers in a
text file (instead of the binary data file used by the old tool). This
should make it easier to create custom timing files when needed.
Bug: webrtc:2692
Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287
Reviewed-on: https://webrtc-review.googlesource.com/25700
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20868}
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=pthatcher@webrtc.org
Bug: None
Change-Id: Icdc7e9e4395eeac053483c69e53501e5aa222107
Reviewed-on: https://webrtc-review.googlesource.com/23567
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20867}
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.
TBR=pthatcher@webrtc.org
Bug: None
Change-Id: Idb0e5e038ad59a6021cb53ef10e8f6733b1da05b
Reviewed-on: https://webrtc-review.googlesource.com/23618
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20864}
Also improves ownership model by using std::unique_ptr in a couple of
places instead of raw pointers.
Bug: webrtc:8278
Change-Id: I0429ec3c416b5baa1ffa21dad71e0d64b004c446
Reviewed-on: https://webrtc-review.googlesource.com/25020
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20863}
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: Ibe15d2814074a4cf67de18d6e04540076f1a9dc9
Reviewed-on: https://webrtc-review.googlesource.com/23611
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20862}
This is not the case for fallback codecs.
Bug: webrtc:7925
Change-Id: I5039d8879923a2db1e7c08bb5640763dd20ea8e7
Reviewed-on: https://webrtc-review.googlesource.com/24863
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20861}
It's now possible to generate JNI code for constructors and enums
correctly. This CL cleans that up.
Bug: webrtc:8278,webrtc:8551,webrtc:8556
Change-Id: I2284a30139cbb186c80713eb6113eda5659c16ad
Reviewed-on: https://webrtc-review.googlesource.com/25622
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20860}
Previously, wrapped native codec instances would leak the native object
if it was never used. This change fixes it by changing getNative method
to createNative.
Also fixes "Video codec hardware acceleration" setting in AppRTCMobile.
Bug: webrtc:7925
Change-Id: I53f6dc1dd5e37dea8d14278423122dede17719c5
Reviewed-on: https://webrtc-review.googlesource.com/24881
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20859}
This CL will unblock the Chromium Roll while deprecated declarations
will be removed from the WebRTC codebase.
Bug: webrtc:8570
Change-Id: I55cf78040758369ce45176cf0a00df50a87eb972
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/25641
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20858}
This new tool provides the same functionality as the legacy tool, but it
is implemented using AudioCodingModule and AudioEncoder APIs instead of
the naked codecs.
Bug: webrtc:2692
Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc
Reviewed-on: https://webrtc-review.googlesource.com/24861
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20857}
The WEBRTC_TRACE facility and GetStaticInstance were deleted a while
ago, but corresponding tsan supressions were left around.
Bug: webrtc:5118
Change-Id: I00f04f0e4f20c17120df80cb43475f71c9d4a6ad
Reviewed-on: https://webrtc-review.googlesource.com/25240
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20856}
Description for changes from the original CL:
Calling legacy SRD, implemented using
SetRemoteDescriptionObserverAdapter wrapping the old observer, was
meant to have the exact same behavior as the legacy SRD implementation
which invokes the callbacks in a Post.
However, in the CL that landed and got reverted (PS1), the Adapter had
its own message handler, and callbacks would be invoked even if the PC
was destroyed.
In PS2 I've changed the Adapter to use the PeerConnection's message
handler. If the PC is destroyed, the callback will not be invoked.
This gives identical behavior to before this CL, and the legacy
behavior is covered by a new unittest.
I changed the adapter to be an implementation detail of
peerconnection.cc, therefor some stuff was moved, and the only tests
covering this is now in peerconnection_rtp_unittest.cc.
This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72
Original change's description:
> SetRemoteDescriptionObserverInterface added.
>
> The new observer replaced SetSessionDescriptionObserver for
> SetRemoteDescription. Unlike SetSessionDescriptionObserver,
> SetRemoteDescriptionObserverInterface is invoked synchronously so
> that the you can rely on the state of the PeerConnection to represent
> the result of the SetRemoteDescription call in the callback.
>
> The new observer succeeds or fails with an RTCError.
>
> This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack
> and SetSessionDescriptionObserver, with the benefit that all media
> object changes can be processed in a single callback by the application
> in a synchronous callback. This will help Chromium keep objects in-sync
> across layers and threads in a non-racy and straight-forward way, see
> design doc (Proposal 2):
> https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing
>
> An adapter for SetSessionDescriptionObserver is added to allow calling
> the old SetRemoteDescription signature and get the old behavior
> (OnSuccess/OnFailure callback in a Post) until third parties switch.
>
> Bug: webrtc:8473
> Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99
> Reviewed-on: https://webrtc-review.googlesource.com/17523
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20841}
TBR=pthatcher@webrtc.org
Bug: webrtc:8473
Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5
Reviewed-on: https://webrtc-review.googlesource.com/25640
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20854}
Unfortunately, H264 makes it non-trivial to compare video formats for
equality. For every video format besides H264 it's enough to look at the
name, but for H264, we need to dig into the parameters. This logic is
currently in several places, and this CL unifies it to one place.
Bug: webrtc:7925
Change-Id: I83a516b108d6b4d6792fd0bf1d24296916d9f7fa
Reviewed-on: https://webrtc-review.googlesource.com/25120
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20853}
This can be used to determine whether the bitrate of a given spatial and temporal layer has been set in the allocation, even if the value it's set to is zero.
GetBitrate still returns 0 if the queried layer does not have the bitrate set.
Bug: webrtc:8479
Change-Id: I1d982e211da9b052fcccdbf588b67da1a4550c60
Reviewed-on: https://webrtc-review.googlesource.com/17440
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Varga <erikvarga@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20852}
This reverts commit 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72.
Reason for revert: Third party project breaks due to use-after-free
in the callback. I suspect this is because the adapter is processing
the async callback instead of the pc, i.e. callback is called from
SetRemoteDescriptionObserverAdapter::OnMessage instead of from
PeerConnection::OnMessage. This makes it possible for the callback to
be invoked after the PC is destroyed.
I argue this is how it should be done, and that if you're using a raw
pointer in an async callback you're doing it wrong, but I will reland
this CL with the callback processed in PeerConnection::OnMessage
instead as to not change the behavior of the old SRD signature.
Original change's description:
> SetRemoteDescriptionObserverInterface added.
>
> The new observer replaced SetSessionDescriptionObserver for
> SetRemoteDescription. Unlike SetSessionDescriptionObserver,
> SetRemoteDescriptionObserverInterface is invoked synchronously so
> that the you can rely on the state of the PeerConnection to represent
> the result of the SetRemoteDescription call in the callback.
>
> The new observer succeeds or fails with an RTCError.
>
> This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack
> and SetSessionDescriptionObserver, with the benefit that all media
> object changes can be processed in a single callback by the application
> in a synchronous callback. This will help Chromium keep objects in-sync
> across layers and threads in a non-racy and straight-forward way, see
> design doc (Proposal 2):
> https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing
>
> An adapter for SetSessionDescriptionObserver is added to allow calling
> the old SetRemoteDescription signature and get the old behavior
> (OnSuccess/OnFailure callback in a Post) until third parties switch.
>
> Bug: webrtc:8473
> Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99
> Reviewed-on: https://webrtc-review.googlesource.com/17523
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20841}
TBR=hbos@webrtc.org,hta@webrtc.org,pthatcher@webrtc.org,guidou@webrtc.org
Change-Id: I715555e084d9aae49ee2a8831c70dc006dbdb74c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8473
Reviewed-on: https://webrtc-review.googlesource.com/25580
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20850}
Previously Java 8 bytecode was transpiled into Java 7 bytecode for AAR.
With this change we instead include Java 8 bytecode in the AAR.
This drops support for Java 7 but allows using Java 8 features such as
static methods in interfaces and default methods.
Bug: webrtc:8084, chromium:762967
Change-Id: I3c6ec9332c5612b7ed811a81957f25c94c80da70
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/21222
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20849}
It will be easier to make perf results output to JSON if the PrintResult*
functions receive doubles instead of strings.
I'll make follow-up CLs for PrintResultMeanAndError and PrintResultList.
Bug: webrtc:8566
Change-Id: I198e422a7bb8cd237c6364af98d2f67f0858452e
Reviewed-on: https://webrtc-review.googlesource.com/25300
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20848}
The current names are too long and make the stats screen in
AppRTCMobile unusable.
Bug: None
Change-Id: I5c4d0526e478ff45392f80a8bb22c2d97f63de17
Reviewed-on: https://webrtc-review.googlesource.com/25481
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20847}
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=danilchap@webrtc.org
Bug: None
Change-Id: Ib4694d183f04d675f2ea66d39661fdffb2a984f1
Reviewed-on: https://webrtc-review.googlesource.com/23604
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20846}
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=sakal@webrtc.org
Bug: None
Change-Id: Ib01eb9ef7b3b76c0f55890d7ea1b8dfcd30f0617
Reviewed-on: https://webrtc-review.googlesource.com/23607
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20845}
The APM-QA tool produces clean-speech + noise + echo mixes with the
--additive_noise_tracks_path, --test_data_generators,
--echo_path_simulator flags. From this CL, it automatically produces
compressed Numpy annotations for the mixes. Annotations are placed in
the same folder as the mixes with name '${basename}-annotations.npz'.
TBR=alessiob@webrtc.org
NOTRY=True
Bug: webrtc:7494
Change-Id: I71941a4283594ef813de3ed65be31623bce5d7de
Reviewed-on: https://webrtc-review.googlesource.com/24960
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20844}
Previously when creating an answer, we used the codecs from the remote list
to get correctly mapped codec id, but the side effect is that we also used
the remote codec parameters which is wrong.
With this change, we use the local codecs with local parameters and the
|NegotiateCodecs| method will take care of the codec id mapping.
Bug: webrtc:8550
Change-Id: Ib7b2fe5f6ed3a72c58cf5df8bf2c1d00476c141c
Reviewed-on: https://webrtc-review.googlesource.com/25285
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20843}
The new observer replaced SetSessionDescriptionObserver for
SetRemoteDescription. Unlike SetSessionDescriptionObserver,
SetRemoteDescriptionObserverInterface is invoked synchronously so
that the you can rely on the state of the PeerConnection to represent
the result of the SetRemoteDescription call in the callback.
The new observer succeeds or fails with an RTCError.
This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack
and SetSessionDescriptionObserver, with the benefit that all media
object changes can be processed in a single callback by the application
in a synchronous callback. This will help Chromium keep objects in-sync
across layers and threads in a non-racy and straight-forward way, see
design doc (Proposal 2):
https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing
An adapter for SetSessionDescriptionObserver is added to allow calling
the old SetRemoteDescription signature and get the old behavior
(OnSuccess/OnFailure callback in a Post) until third parties switch.
Bug: webrtc:8473
Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99
Reviewed-on: https://webrtc-review.googlesource.com/17523
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20841}
This type of check should instead be performed by the application/client.
If the app does not have mic permissions, construction of the AudioRecord
object will fail and the user will receive an error callback anyhow.
Bug: b/69434512
Change-Id: If1d7eff488f7c693697e048a567c17ed0c51f040
Reviewed-on: https://webrtc-review.googlesource.com/25261
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20839}
capture_time_ms_ is always 0 for frames received from the network. This
caused a bug because it was used as an unique identified.
Bug: b/68271454
Change-Id: Ic4417a52e61cf2b0cd796a89207a90b603a16590
Reviewed-on: https://webrtc-review.googlesource.com/24940
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20837}
