Most notably, will return "INVALID_MODIFICATION" if a field in the
configuration was modified and modification of that field isn't supported.
Also changing RTCError to a class that wraps an enum type, because it will
eventually need to hold other information (like SDP line number), to match
the RTCError that was recently added to the spec:
https://github.com/w3c/webrtc-pc/pull/850
BUG=webrtc:6916
Review-Url: https://codereview.webrtc.org/2587133004
Cr-Original-Commit-Position: refs/heads/master@{#15777}
Committed: 7a5fa6cd61
Review-Url: https://codereview.webrtc.org/2587133004
Cr-Commit-Position: refs/heads/master@{#16016}
Trivial change that allows users to call MicrophoneVolumeIsAvailable()
(and get a valid result) on Android without crashing.
TBR=henrik.lundin
BUG=NONE
Review-Url: https://codereview.webrtc.org/2620243003
Cr-Commit-Position: refs/heads/master@{#16013}
This cl was produced by
git grep -l 'ASSERT(false)' |\
xargs -n1 sed -i 's/ASSERT(false)/RTC_NOTREACHED()/'
followed by additional includes of base/checks.h in affected files,
git cl format to adjust spacing in webrtc/base/transformadapter.cc.
Finally, to make presubmit happy, one unnamed TODO marker was deleted
in that file.
This is a step towards deletion of base/common.h.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2625003003
Cr-Commit-Position: refs/heads/master@{#16009}
Deps have rolled to 1.6, and since no one noticed that the old code path
was broken and wouldn't even compile, I assume no one is using it.
I therefore deem it time to clean away all these nasty ifdefs.
("const kNalHeaderSizeAllocation = 50;" doesn't declare a type)
BUG=chromium:614970
Review-Url: https://codereview.webrtc.org/2622233002
Cr-Commit-Position: refs/heads/master@{#16008}
Avoids confusion about the meaning of "incoming".
BUG=webrtc:6897
Review-Url: https://codereview.webrtc.org/2624073003
Cr-Commit-Position: refs/heads/master@{#16007}
In this CL:
- Removed unused variable |last_seq_num_|.
- Fixed bug where a new incomplete frame was detected as a complete frame.
- Added fuzzer to video_coding::PacketBuffer.
BUG=chromium:677101
Review-Url: https://codereview.webrtc.org/2613833003
Cr-Commit-Position: refs/heads/master@{#16003}
Also enable Intel HW Vp8 encoder by default in AppRTCMobile.
BUG=webrtc:6683
Review-Url: https://codereview.webrtc.org/2614373004
Cr-Commit-Position: refs/heads/master@{#16002}
This lets the RTP code be unaware of lower layers, and the
SetTransportOverhead method is deleted from RTPSender and RtpRtcp.
Instead, that method is added to CongestionController and
TransportFeedbackAdapter, where it is more appropriate.
BUG=wertc:6847
Review-Url: https://codereview.webrtc.org/2589743002
Cr-Commit-Position: refs/heads/master@{#15995}
The purpose is to be able to add field trials in Java code.
BUG=webrtc:6683
Review-Url: https://codereview.webrtc.org/2621003002
Cr-Commit-Position: refs/heads/master@{#15994}
Moves webrtc/common_video/rotation.h and parts of
webrtc/common_video/include/video_frame_buffer.h and
webrtc/video_frame.h, and adds to a new GN target api:video_frame_api.
BUG=webrtc:5880
Review-Url: https://codereview.webrtc.org/2517173004
Cr-Commit-Position: refs/heads/master@{#15993}
Since packets can be received out of order NACKs could be triggered before any
packet was actually dropped. This cause the test to never drop packets which in
turn caused the block of code which set the |observation_complete_| event to
never execute.
BUG=webrtc:2845
Review-Url: https://codereview.webrtc.org/2613443002
Cr-Commit-Position: refs/heads/master@{#15990}
Add RTC_DEPRACATed anonymous unions to not break downstream projects.
Orignal issue's description:
> commit 0ad21111fcc57a7e978edba3c4263f0062d7f9ff
> Author: danilchap <danilchap@webrtc.org>
> Date: Mon Dec 19 09:36:33 2016 -0800
>
> Revert of Rename RTPVideoHeader.isFirstPacket to
> .is_first_packet_in_frame. (patchset #1 id:1 of
> https://codereview.webrtc.org/2574943003/ )
>
> Reason for revert:
> breaks downstream project.
>
> Can you make this change in a compatible way using anonymous
> union:
> union {
> bool is_first_packet_in_frame;
> RTC_DEPRECATED bool isFirstPacket;
> };
> (unfortunetly this this treak breaks braced initialization in
> rtp_rtcp_impl_unittest.cc,
> so that should be rewritting in a more classic way)
>
> Original issue's description:
> > Rename RTPVideoHeader.isFirstPacket to
> > .is_first_packet_in_frame.
> >
> > Name should represent the actual meaning.
> >
> > BUG=None
> >
> > Review-Url: https://codereview.webrtc.org/2574943003
> > Cr-Commit-Position: refs/heads/master@{#15684}
> > Committed:
> > efde908380
>
> TBR=stefan@webrtc.org,sprang@webrtc.org,johan@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/2589783003
> Cr-Commit-Position: refs/heads/master@{#15686}
>
BUG=None
Review-Url: https://codereview.webrtc.org/2614503002
Cr-Commit-Position: refs/heads/master@{#15987}
External dependencies have been updated to use webrtc/sdk/android
instead, and we can remove the remaining files in webrtc/api/android.
BUG=webrtc:5882
TBR=tommi
Review-Url: https://codereview.webrtc.org/2628553003
Cr-Commit-Position: refs/heads/master@{#15985}
Left shifting of negative integers is undefined behavior, and should be prevented. This CL fixes one such instance in the Levinson Durbin function.
BUG=chromium:675349
Review-Url: https://codereview.webrtc.org/2621693002
Cr-Commit-Position: refs/heads/master@{#15984}
This used to be updated with the reserved capacity of the buffer,
not the actual portion in use.
BUG=webrtc:6034
Review-Url: https://codereview.webrtc.org/2620653005
Cr-Commit-Position: refs/heads/master@{#15982}
Reason for revert:
Speculative revert.
Linux memcheck bot started failing a lot at the time of this cl. Doesn't look related at first glance, but we don't have another lead yet.
Original issue's description:
> Fix BitrateProber to match the requested bitrate more precisely
>
> Previously BirateProber was calculating delay between probes based on
> the size of the previous probe. Because of that the actual sent bitrate
> can deviate greatly from the target value. With this change it uses
> total number of bytes in the cluster to estimate delay before each
> probe.
>
> BUG=webrtc:6952
>
> Review-Url: https://codereview.webrtc.org/2613543003
> Cr-Commit-Position: refs/heads/master@{#15971}
> Committed: 599c5011e7TBR=philipel@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6952
Review-Url: https://codereview.webrtc.org/2626473004
Cr-Commit-Position: refs/heads/master@{#15979}
Because Voice Engine was the only user.
(We have tried to land this many times before. I'm hoping that this
time all external dependencies on these files will really be gone.)
BUG=none
Review-Url: https://codereview.webrtc.org/2622493002
Cr-Commit-Position: refs/heads/master@{#15978}
The intention of SetConfiguration is that it modifies the configuration,
while keeping the constraints passed into CreatePeerConnection. Right
now that's now happening. See bug for more explanation.
BUG=webrtc:6942
Review-Url: https://codereview.webrtc.org/2603653002
Cr-Commit-Position: refs/heads/master@{#15974}
The BaseChannel code is geared around RTP; the presence of media engines,
send and receive streams, SRTP, SDP directional attribute negotiation, etc.
It doesn't make sense to use it for SCTP as well. This separation should make
future work both on BaseChannel and the SCTP code paths easier.
SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
doesn't get confused with webrtc::DataChannel any more.
Beyond just moving code around, some consequences of this CL:
- We'll now stop using the worker thread for SCTP. Packets will be
processed right on the network thread instead.
- The SDP directional attribute is ignored, as it's supposed to be.
BUG=None
Review-Url: https://codereview.webrtc.org/2564333002
Cr-Original-Commit-Position: refs/heads/master@{#15906}
Committed: 67b3bbe639
Review-Url: https://codereview.webrtc.org/2564333002
Cr-Commit-Position: refs/heads/master@{#15973}
Previously BirateProber was calculating delay between probes based on
the size of the previous probe. Because of that the actual sent bitrate
can deviate greatly from the target value. With this change it uses
total number of bytes in the cluster to estimate delay before each
probe.
BUG=webrtc:6952
Review-Url: https://codereview.webrtc.org/2613543003
Cr-Commit-Position: refs/heads/master@{#15971}
webrtcvideoengine2.cc uses a field for parameters_, and doesn't empty
out the current state in functions like SetCodec. In the case of
internal_source, SetCodec only set it for external encoders, which
means that in a switch from an internal-source external encoder to an
internal encoder, the internal_source bit would stay set.
(It's plausible that there are other places that are also unsafe and we
just don't notice because codec switches are uncommon in most usage)
In combination with https://codereview.webrtc.org/2574183002/,
generic_encoder.cc now creates 1x1 uninitialized frames as fake frames
for internal_source keyframe requests. The vp8 software encoder doesn't
deal correctly with frames of resolutions that don't match the
configured resolution (besides a DCHECK) and no longer throws these
away (they used to be 0x0 frames), so this results in the VP8
encoder creating a keyframe of the configured send codec size by reading
random memory off the end of the fake I420 frame. This could either
cause crashes or encoding junk data, depending on where the allocation
was.
BUG=webrtc:6957
Review-Url: https://codereview.webrtc.org/2617003003
Cr-Commit-Position: refs/heads/master@{#15969}
