If SW H264 is not supported and a client tries to create such a
decoder from InternalDecoderFactory, we currently crash. This CL
changes so that we log an error and return null from CreateDecoder()
instead.
Bug: webrtc:7925
Change-Id: I0c495f62dae25ac0bf4931c02527eb9977db3d92
Reviewed-on: https://webrtc-review.googlesource.com/92395
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24220}
This adds new constexpr create function for DataSize, DataRate,
TimeDelta and Timestamp. The names are capitalized to mirror the
naming scheme of the previously constexpr methods (Zero and
Infinity create functions). They are also kept longer since they
are not expected to be used in complex expressions.
Bug: webrtc:9574
Change-Id: I5950548718675050fc5d66699de295455c310861
Reviewed-on: https://webrtc-review.googlesource.com/91161
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24218}
Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.
To make it work, I had to add support for the "ptime" parameter to the
L16 codec.
Bug: webrtc:8396
Change-Id: I3869422882611d2eed65d6c849ea7cd3ad6bd126
Reviewed-on: https://webrtc-review.googlesource.com/87423
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24217}
* More canonical and efficient 'move to front'.
* Don't use 'new' when value semantic is fine.
* Simplify flow (remove One-off private method).
* Remove dead code.
Bug: webrtc:9575
Change-Id: Ie6a3c4e3d5e2342e77e54fd59fffa05f6e5f9ebe
Reviewed-on: https://webrtc-review.googlesource.com/92802
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24215}
This is a quick fix to ensure that we don't wrap the value. A proper
solution would be to ensure that the packets_lost field is signed and
handled as signed at all places it's used.
Bug: webrtc:9598
Change-Id: I3622f2a61aa3af57db6292ef4c0a8e97c4833aa4
Reviewed-on: https://webrtc-review.googlesource.com/92881
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24212}
We both frequentyly work on this code and much of it is intertwined with
pc/ and p2p/ code which we have OWNERs for already.
NOTRY=True
Bug: None
Change-Id: If56ebca6ef44cf9b7837e8d4bc3afa367a5d5216
Reviewed-on: https://webrtc-review.googlesource.com/90084
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24211}
This dependency has been copied from Chromium.
Bug: webrtc:9594
Change-Id: Ida86d73a39ffa14c92dcfd4783d95e08857b3da5
Reviewed-on: https://webrtc-review.googlesource.com/92397
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24210}
Allows enabling audio for RunWithAnalyzer method, and prints out audio jitterbuffer performance stats. Also fixes for RunWithRenderer when enabling audio (seg-faulted).
Bug: b/112299470
Change-Id: Ic7c0de1c455891f38cca317001c6c216e82f6ec3
Reviewed-on: https://webrtc-review.googlesource.com/92800
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24208}
This makes the constructor and the unchecked create functions
constexpr on the unit classes Timestamp, TimeDelta, Datarate and
DataSize. This allows using the units in constexpr constants.
Unchecked access methods are made constexpr as well. Making them
usable in static asserts.
Constexpr create functions for checked construction is added in
a separate CL.
Bug: webrtc:9574
Change-Id: I605ae2e8572195dbb2078c283056208be0f43333
Reviewed-on: https://webrtc-review.googlesource.com/91160
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24206}
Replaced by a int64_t representing time in us.
Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}
This CL adds a factory for creating a GoogCC network controller that
can be used without RTCP specific messages. This prepares for enabling
use of other underlying protocols as long as they can provide per
packet feedback.
Bug: None
Change-Id: I6671181949d97abd18843d0f4edf75040cc3f007
Reviewed-on: https://webrtc-review.googlesource.com/84583
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24198}
This CL removes a check in RoundRobinPacketQueue::FinalizePop. This
check will trigger if a the pause state is changed in PacedSender while
a packet is sent. This is a rare occurrence but would yield flaky
behavior. The check should not be required for the code to function
since the paused state is not read in FinalizePop other than for this
check.
Bug: webrtc:9586
Change-Id: Ib9476168eb637dc2f9710d0592bed92c4b03dacb
Reviewed-on: https://webrtc-review.googlesource.com/92090
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24197}
Also delete related code in RtpReceiverAudio, RtpReceiverVideo and
RtpPayloadRegistry.
Only intended change in behavior is that packets with unknown payload
types are not discarded at this level of the stack. They are discarded
higher up, in Channel::ReceivePacket (audio) and
RtpVideoStreamReceiver::ReceivePacket (video).
Bug: webrtc:8995
Change-Id: I807997120bb40a95b0575c55db6e20a0cac651bf
Reviewed-on: https://webrtc-review.googlesource.com/92087
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24196}
This reverts commit fad2aa23b406ca5d85b8aa9ab891f2067e51c782.
Reason for revert: There seems to be a mismatch with Chrome's default for VP8.
Original change's description:
> Extract color space from Vp8 decoder
>
> Makes use of ColorSpace class to extract info from Vp8 stream.
>
> Bug: webrtc:9522
> Change-Id: Id9d46eeea5497c4da31db27bfcf2743612ae4157
> Reviewed-on: https://webrtc-review.googlesource.com/90183
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24086}
TBR=sprang@webrtc.org,emircan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9522
Change-Id: Ie589963159c9e7ccbc52bf3fdfcbc383656a4ca9
Reviewed-on: https://webrtc-review.googlesource.com/92500
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24191}
Fixes a target name that was missed in the last CL.
TBR=phoglund
Bug: webrtc:9588
No-Try: True
Change-Id: I704325666b758cee7eb080f8628fc839ab89831d
Reviewed-on: https://webrtc-review.googlesource.com/92389
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24188}
For example, currently "resources/audio_coding/F02_tlm10.OUT20.sha1" would have been ignored by the pattern "**/*.OUT*".
No-Try: True
Bug: None
Change-Id: I91243a301950485cb61d5f72a00af08372ec7951
Reviewed-on: https://webrtc-review.googlesource.com/92085
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24185}
This matches Chromium pattern of naming instrumentation test apks with
a name ending in _test_apk. The old naming confuses generate_gradle.py.
Renames:
- AppRTCMobileTest
-> AppRTCMobile_test_apk
- AppRTCMobileTestStubbedVideoIO
-> AppRTCMobile_stubbed_video_io_test_apk
- libjingle_peerconnection_android_unittest
-> android_instrumentation_test_apk
Bug: webrtc:9588
TBR: phoglund
Change-Id: Idb82dc4bd089bc7c90e9373f7c3d572f9fd2d95a
Reviewed-on: https://webrtc-review.googlesource.com/92380
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24184}
FEATURE_ENABLE_VOICEMAIL is never used.
FEATURE_ENABLE_PSTN is used in //third_party/libjingle_xmpp but WebRTC
doesn't depend on it, so it is reasonable to assume that no one needs
to define it.
Bug: webrtc:9564
Change-Id: Idfb04081f497ef52fc5c140ffb82fa2dc7b9824d
Reviewed-on: https://webrtc-review.googlesource.com/92081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24182}
This also rolls up //base in DEPS, because it needs to be landed together with
54f759310c
Bug: chromium:867475
Change-Id: I5792cb0610d2df46a9368fd3b1846583aa134b38
Reviewed-on: https://webrtc-review.googlesource.com/90404
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24180}
The factory is plumbed down to P2PTransportChannel and will eventually
be used to resolve hostnames. Uses of PacketSocketFacotry::CreateAsyncResolver
will eventually be migrated to use this factory instead.
Bug: webrtc:4165
Change-Id: I1c48b2ffb8649609a831eba291f67ce544bb10eb
Reviewed-on: https://webrtc-review.googlesource.com/91300
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24176}
This changes makes the usage of the new probe controller reflect how the
old probe controller was used. That is probes are now sent as soon as
they are generated. This is to avoid regressions in performance doe to
the timing of the sent probes.
Bug: chromium:868776
Change-Id: I722585689258c9b01e8f1dc47249b284a05a2793
Reviewed-on: https://webrtc-review.googlesource.com/91441
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24175}
RTCNativeAudioSessionDelegateAdapter has a raw pointer to AudioDeviceIOS,
and receives callbacks from RTCAudioSession and forwards them to AudioDeviceIOS.
During teardown of these components the situation can occur that the dtor for
AudioDeviceIOS has been called but the ObjC runtime has not yet dealloced
RTCNativeAudioSessionDelegateAdapter, so it's still receiving callbacks while
the pointer it keeps to AudioDeviceIOS has been invalidated.
This occasionally triggers a crash when WebRTC is shutting down.
The fix in this CL is to make sure to deregister the adapter from RTCAudioSession
_before_ the dtor for AudioDeviceIOS returns.
Bug: webrtc:9523
Change-Id: Ica85420d76efc63940472bc43e3ec71d16036ccf
Reviewed-on: https://webrtc-review.googlesource.com/90245
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24173}
