stack will be removed soon in a separate CL. Constraints will not be supported
in the new implementation. Apps can request a format directly and the closest
supported format will be selected.
Changes needed from the apps:
1. Use the new createVideoSource without constraints.
2. Call startCapture manually.
3. Don't call videoSource.stop/restart, use startCapture/stopCapture instead.
R=magjed@webrtc.orgTBR=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/2127893002 .
Cr-Commit-Position: refs/heads/master@{#13504}
Now it check if rtp timestamp can be calculating instead of checking number of rtp packets. This way it works for reconfigured streams too.
It also moved deeper into rtcp_sender class to prevent SR no matter the reason it need to be genereated. This way it prevents creating compound rtcp packets that have to start with Sender Report and Sender Reports as response to (mostly theoretical) sr-request rtcp packet.
BUG=webrtc:1600
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1639253007 .
Cr-Commit-Position: refs/heads/master@{#13503}
Retransmissions are supposed to be sent before normal packets by the pacer, but the current implementation will only use it if the second packet is a retransmission and the first packet is not. It misses the case where the first packet is retransmission and the second packet is not.
This CL fixes the comparator and adds a unit test.
Also changed the SendAndExpectPacket function to propagate the retransmission flag to the expectations. Previously, all packets were expected to be normal packets.
BUG=webrtc:6124
Review-Url: https://codereview.webrtc.org/2156063004
Cr-Commit-Position: refs/heads/master@{#13502}
and use the preparsed headers to plot the network delay changes.
This is the first of several CLs that clean up the visualization
tool to make it easier to add new metrics.
Review-Url: https://codereview.webrtc.org/2145153002
Cr-Commit-Position: refs/heads/master@{#13499}
The added unittest triggers this CHECK:
433ed06800/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc (146)
It happens because the unwrap of the sequence number fails if the unwrappers last sequence number is small, but the newly added sequence number is large (greater than last seq num + 2^15), and therefore should have been interpreted as a reordering and a backwards wrap. Since that would mean the sequence number returned from the unwrapper would be negative, it simply returns the original sequence number instead. This causes problems later where the wrap is correctly handled, and everything breaks.
The real solution should be to correctly handle wraps, but to prevent the crash this is a reasonable workaround for now.
BUG=
Review-Url: https://codereview.webrtc.org/2157843002
Cr-Commit-Position: refs/heads/master@{#13496}
We thought we could safely remove this, but older versions of Chrome
don't do role conflict resolution properly, so it's actually not safe
to yet.
BUG=628676
Review-Url: https://codereview.webrtc.org/2152963003
Cr-Commit-Position: refs/heads/master@{#13492}
Logging when a candidate is gathered or the gathering state or a
Port changes. This will make it easier to identify problems related
to candidate gathering.
Review-Url: https://codereview.webrtc.org/2122373004
Cr-Commit-Position: refs/heads/master@{#13490}
After https://codereview.webrtc.org/1827263002, audio devices are no
longer (ever) initialized if they return true from
RecordingIsInitialized. Since this was left as "return true;" for
file_audio_device, the recording buffer was never set up correctly, and
the audio buffer would assert when called (in debug) and FileAudioDevice
would cause memory corruption (in release).
BUG=
Review-Url: https://codereview.webrtc.org/2116003003
Cr-Commit-Position: refs/heads/master@{#13489}
Necessary when compiling this file on a case-sensitive file system.
BUG=chromium:495204,chromium:617318
Review-Url: https://codereview.webrtc.org/2145373004
Cr-Commit-Position: refs/heads/master@{#13488}
In order to correctly determine the references of a frame when using Vp9
with GOF one has to wait for all frames on the lower temporal layers
to make sure no up-switch point is missed.
This patch fix a bug where upon receiving a frame the RtpFrameReferenceFinder
would try to add missing frame for a group with a not yet knows scalability
structure.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2127073002
Cr-Commit-Position: refs/heads/master@{#13487}
Reason for revert:
Nope, breaks chromium
Original issue's description:
> Fix inconsistent setting of the rtc_include_unittests flag.
>
> On advice from brettw, se discussion here:
> https://codereview.webrtc.org/2149543002/
>
> The problem was we were setting the flag to both false and true,
> and the the true happened to win out for WebRTC checkouts and
> false for Chromium checkouts. This change should make this
> mechanic more obvious.
>
> This change _should_ have no effect downstream.
>
> Doing tbr to see if we can get the chromium import back into a good state today.
>
> TBR=tommi@webrtc.org
>
> Committed: https://crrev.com/9d34148714773339a4e8396bd28aceb571554d36
> Cr-Commit-Position: refs/heads/master@{#13484}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2150293002
Cr-Commit-Position: refs/heads/master@{#13486}
After disabling all of these, the rest of the tests pass, at least on
my Nexus 7 device.
BUG=webrtc:4364
Review-Url: https://codereview.webrtc.org/2151823002
Cr-Commit-Position: refs/heads/master@{#13485}
On advice from brettw, se discussion here:
https://codereview.webrtc.org/2149543002/
The problem was we were setting the flag to both false and true,
and the the true happened to win out for WebRTC checkouts and
false for Chromium checkouts. This change should make this
mechanic more obvious.
This change _should_ have no effect downstream.
Doing tbr to see if we can get the chromium import back into a good state today.
TBR=tommi@webrtc.org
Review-Url: https://codereview.webrtc.org/2154693002
Cr-Commit-Position: refs/heads/master@{#13484}
This it to avoid requiring targets that include header files that in turn use SequenceTaskedChecker, to also have to define the macros needed by TaskQueue.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2145393003
Cr-Commit-Position: refs/heads/master@{#13482}
If the currently selected connection becomes not receiving and if a backup connection
becomes strong first, we will not switch the connection until X milliseconds is passed
but the selected connection is still not receiving and the backup connection is still receiving. This will prevent the connection switching from happening too frequently.
BUG=
Review-Url: https://codereview.webrtc.org/2143653005
Cr-Commit-Position: refs/heads/master@{#13480}
It causes an asan initialization-order-fiasco in trying to read the
names of other globally constructed data:
==21449==ERROR: AddressSanitizer: initialization-order-fiasco on address 0x7f6f297bc5e8 at pc 0x7f6f26b332a7 bp 0x7ffd479f8cb0 sp 0x7ffd479f8ca8
READ of size 8 at 0x7f6f297bc5e8 thread T0
#0 0x7f6f26b332a6 in name
webrtc/base/flags.h:83:38
#1 0x7f6f26b332a6 in Lookup
webrtc/base/flags.cc:133
#2 0x7f6f26b332a6 in rtc::FlagList::Register(rtc::Flag*)
webrtc/base/flags.cc:260
#3 0x7f6f2529972b in __cxx_global_var_init.1
BUG=
Review-Url: https://codereview.webrtc.org/2110963004
Cr-Commit-Position: refs/heads/master@{#13479}
This cl clean up unused forward declaration of cricket::AudioFrame and cricket::Renderer.
cricket::VideoFrame is used in the interface and thus the header file is included.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2145613006
Cr-Commit-Position: refs/heads/master@{#13477}
This cl is in preparation for https://codereview.webrtc.org/2060403002/ Add task queue to Call.
In the coming cl the video_sender, and i420_buffer_pool will be used on a task queue and therefore SequencedTaskChecker is needed instead of a ThreadChecker.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2149553002
Cr-Commit-Position: refs/heads/master@{#13474}
Merge with the latest webrtc native code.
Remove deprecated function Connect() in QuicTransportChannel.
Fix the compiling issue and broken unit tests by adding the network thread to QUIC related classes.
Review-Url: https://codereview.webrtc.org/2089553002
Cr-Commit-Position: refs/heads/master@{#13472}
Reason for revert:
Reverting because description was inaccurate. Will reland after updating description.
Original issue's description:
> Modify PeerConnection for end-to-end QuicDataChannel usage
>
> To allow end-to-end QuicDataChannel usage with a
> PeerConnection, RTCConfiguration has been modified to
> include a boolean for whether to do QUIC, since negotiation of
> QUIC is not implemented. If one peer does QUIC, then it will be
> assumed that the other peer must do QUIC or the connection
> will fail.
>
> PeerConnection has been modified to create data channels of type
> QuicDataChannel when the peer wants to do QUIC.
>
> WebRtcSession has been modified to use a QuicTransportChannel
> instead of a DtlsTransportChannelWrapper/DataChannel
> when QUIC should be used.
>
> Modification of previous in-flight CL: https://codereview.chromium.org/1844803002/
>
> Committed: https://crrev.com/36c8d69ce188102ae6fd48c371cf1518f08698fb
> Cr-Commit-Position: refs/heads/master@{#13470}
TBR=pthatcher@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2146133002
Cr-Commit-Position: refs/heads/master@{#13471}
To allow end-to-end QuicDataChannel usage with a
PeerConnection, RTCConfiguration has been modified to
include a boolean for whether to do QUIC, since negotiation of
QUIC is not implemented. If one peer does QUIC, then it will be
assumed that the other peer must do QUIC or the connection
will fail.
PeerConnection has been modified to create data channels of type
QuicDataChannel when the peer wants to do QUIC.
WebRtcSession has been modified to use a QuicTransportChannel
instead of a DtlsTransportChannelWrapper/DataChannel
when QUIC should be used.
Modification of previous in-flight CL: https://codereview.chromium.org/1844803002/
Review-Url: https://codereview.webrtc.org/2089553002
Cr-Commit-Position: refs/heads/master@{#13470}
The DtlsTransportChannel wasn't properly handling the
P2PTransportChannel becoming writable before receiving a remote
fingerprint; it wasn't starting the DTLS handshake whenever this
happened.
Review-Url: https://codereview.webrtc.org/2140283002
Cr-Commit-Position: refs/heads/master@{#13469}
This updates how we initialize timer events to be the same way as we initialize non-timer events.
BUG=
Review-Url: https://codereview.webrtc.org/2147973002
Cr-Commit-Position: refs/heads/master@{#13468}
The decode thread should be stopped before triggering shutdown of the
video receiver, so that the decoder doesn't try to insert a new frame
while the jitter buffer is being shut down.
BUG=webrtc:6102
Review-Url: https://codereview.webrtc.org/2146883002
Cr-Commit-Position: refs/heads/master@{#13467}
Reason for revert:
Upstream fixes in place, should be OK now.
Original issue's description:
> Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ )
>
> Reason for revert:
> Breaks upstream code.
>
> Original issue's description:
> > Refactor NACK bitrate allocation
> >
> > Nack bitrate allocation should not be done on a per-rtp-module basis,
> > but rather shared bitrate pool per call. This CL moves allocation to the
> > pacer and cleans up a bunch if bitrate stats handling.
> >
> > BUG=
> > R=danilchap@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
> >
> > Committed: 5fc59e810b
>
> TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/e5dd44101eca485f5ad12e5f7ce6f6b0d204116b
> Cr-Commit-Position: refs/heads/master@{#13417}
TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=
Review-Url: https://codereview.webrtc.org/2146013002
Cr-Commit-Position: refs/heads/master@{#13465}
Reason for revert:
Looks like things are still breaking upstream... :(
Original issue's description:
> Reland of Adds data logging in native AudioDeviceBuffer class (patchset #1 id:1 of https://codereview.webrtc.org/2141413002/ )
>
> Reason for revert:
> Will make one more try since we have now confirmed that our TaskQueue tests works on Android. Let's hope for the best...
>
> Original issue's description:
> > Revert of Adds data logging in native AudioDeviceBuffer class (patchset #1 id:1 of https://codereview.webrtc.org/2138403003/ )
> >
> > Reason for revert:
> > Reverting again since it might have caused this issue:
> >
> > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus9%29/builds/13622/steps/content_browsertests/logs/stdio
> >
> > Original issue's description:
> > > Reland of Adds data logging in native AudioDeviceBuffer class (patchset #1 id:1 of https://codereview.webrtc.org/2139233002/ )
> > >
> > > Reason for revert:
> > > My original patch broke things that are now fixed by https://codereview.webrtc.org/2141193002/.
> > >
> > > Hence I am relanding my original change.
> > >
> > > Original issue's description:
> > > > Revert of Adds data logging in native AudioDeviceBuffer class (patchset #10 id:180001 of https://codereview.webrtc.org/2132613002/ )
> > > >
> > > > Reason for revert:
> > > > Seems to break things upstream.
> > > >
> > > > Original issue's description:
> > > > > Adds data logging in native AudioDeviceBuffer class.
> > > > >
> > > > > Goal is to provide periodic logging of most essential audio parameters
> > > > > for playout and recording sides. It will allow us to track if the native audio layer is working as intended.
> > > > >
> > > > > BUG=NONE
> > > > >
> > > > > Committed: https://crrev.com/348e411dd27e6dbe9b84b27ce46e9b7c657c1eae
> > > > > Cr-Commit-Position: refs/heads/master@{#13440}
> > > >
> > > > TBR=stefan@webrtc.org,henrika@webrtc.org
> > > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > > NOPRESUBMIT=true
> > > > NOTREECHECKS=true
> > > > NOTRY=true
> > > > BUG=NONE
> > > >
> > > > Committed: https://crrev.com/025aa94ccb85e4c6fe20a3fecdac5d27ec9ba3da
> > > > Cr-Commit-Position: refs/heads/master@{#13441}
> > >
> > > TBR=stefan@webrtc.org,sprang@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=NONE
> > >
> > > Committed: https://crrev.com/dd2fdecc78c50377d10ec98b41179acde9218ee7
> > > Cr-Commit-Position: refs/heads/master@{#13455}
> >
> > TBR=stefan@webrtc.org,sprang@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=NONE
> >
> > Committed: https://crrev.com/5dd941e5a5ccde541d9b40a1df379ed59c5fab5c
> > Cr-Commit-Position: refs/heads/master@{#13457}
>
> TBR=stefan@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=NONE
>
> Committed: https://crrev.com/b201da3fab5efc048a4341f39293d2dcf27b2eec
> Cr-Commit-Position: refs/heads/master@{#13462}
TBR=stefan@webrtc.org,henrika@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=NONE
Review-Url: https://codereview.webrtc.org/2148623004
Cr-Commit-Position: refs/heads/master@{#13464}
Plot graphs to python code. Generate packet, playout, sequence number and total bitrate graphs.
Add bitrate and latency plots. Generate multiple figures, and control which ones are used through command line flags.
Committed: https://crrev.com/a478786ef1513790194792010f766324a469db4d
Review-Url: https://codereview.webrtc.org/1995523002
Cr-Original-Commit-Position: refs/heads/master@{#13443}
Cr-Commit-Position: refs/heads/master@{#13463}
Reason for revert:
Will make one more try since we have now confirmed that our TaskQueue tests works on Android. Let's hope for the best...
Original issue's description:
> Revert of Adds data logging in native AudioDeviceBuffer class (patchset #1 id:1 of https://codereview.webrtc.org/2138403003/ )
>
> Reason for revert:
> Reverting again since it might have caused this issue:
>
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus9%29/builds/13622/steps/content_browsertests/logs/stdio
>
> Original issue's description:
> > Reland of Adds data logging in native AudioDeviceBuffer class (patchset #1 id:1 of https://codereview.webrtc.org/2139233002/ )
> >
> > Reason for revert:
> > My original patch broke things that are now fixed by https://codereview.webrtc.org/2141193002/.
> >
> > Hence I am relanding my original change.
> >
> > Original issue's description:
> > > Revert of Adds data logging in native AudioDeviceBuffer class (patchset #10 id:180001 of https://codereview.webrtc.org/2132613002/ )
> > >
> > > Reason for revert:
> > > Seems to break things upstream.
> > >
> > > Original issue's description:
> > > > Adds data logging in native AudioDeviceBuffer class.
> > > >
> > > > Goal is to provide periodic logging of most essential audio parameters
> > > > for playout and recording sides. It will allow us to track if the native audio layer is working as intended.
> > > >
> > > > BUG=NONE
> > > >
> > > > Committed: https://crrev.com/348e411dd27e6dbe9b84b27ce46e9b7c657c1eae
> > > > Cr-Commit-Position: refs/heads/master@{#13440}
> > >
> > > TBR=stefan@webrtc.org,henrika@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=NONE
> > >
> > > Committed: https://crrev.com/025aa94ccb85e4c6fe20a3fecdac5d27ec9ba3da
> > > Cr-Commit-Position: refs/heads/master@{#13441}
> >
> > TBR=stefan@webrtc.org,sprang@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=NONE
> >
> > Committed: https://crrev.com/dd2fdecc78c50377d10ec98b41179acde9218ee7
> > Cr-Commit-Position: refs/heads/master@{#13455}
>
> TBR=stefan@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=NONE
>
> Committed: https://crrev.com/5dd941e5a5ccde541d9b40a1df379ed59c5fab5c
> Cr-Commit-Position: refs/heads/master@{#13457}
TBR=stefan@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=NONE
Review-Url: https://codereview.webrtc.org/2146853003
Cr-Commit-Position: refs/heads/master@{#13462}
Don't update histograms until we're sure process threads won't call into
the instance being destructed, trying to update stats.
BUG=webrtc:6103
Review-Url: https://codereview.webrtc.org/2151433002
Cr-Commit-Position: refs/heads/master@{#13461}
Reason for revert:
For some reason, payload_type_mapper.cc is not being picked up in Chrome builds, leading to undefined references. Reverting while investigating.
Original issue's description:
> WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs.
>
> Changed WebRtcVoiceEngine to present receive codecs from the formats
> provided by its decoder factory. Added supported formats to
> BuiltinAudioDecoderFactory. Added helper functions for creating some
> simple decoder factories for mocking.
>
> Created a PayloadTypeMapper for assigning payload types to formats. I
> think we'll eventually want to use this further up, or possibly in
> both the audio and video sides. It would be best if the engines didn't
> have to talk payload types at all, but it might be more difficult to
> get around when payload types depend on each-other, like the RTX
> codecs for video.
>
> This CL also includes some changes to rtc::Optional. I've put them in
> a separate CL that should (or should not) land first, making these
> changes void.
> See: https://codereview.webrtc.org/2072713002/
>
> BUG=webrtc:5805
>
> Committed: https://crrev.com/95eb1ba0db79d8fd134ae61b0a24648598684e8a
> Cr-Commit-Position: refs/heads/master@{#13459}
TBR=ivoc@webrtc.org,tina.legrand@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2151453002
Cr-Commit-Position: refs/heads/master@{#13460}
Changed WebRtcVoiceEngine to present receive codecs from the formats
provided by its decoder factory. Added supported formats to
BuiltinAudioDecoderFactory. Added helper functions for creating some
simple decoder factories for mocking.
Created a PayloadTypeMapper for assigning payload types to formats. I
think we'll eventually want to use this further up, or possibly in
both the audio and video sides. It would be best if the engines didn't
have to talk payload types at all, but it might be more difficult to
get around when payload types depend on each-other, like the RTX
codecs for video.
This CL also includes some changes to rtc::Optional. I've put them in
a separate CL that should (or should not) land first, making these
changes void.
See: https://codereview.webrtc.org/2072713002/
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2072753002
Cr-Commit-Position: refs/heads/master@{#13459}
