It's still valid SDP so just clamp it at INT_MAX.
BUG=chromium:648071
Review-Url: https://codereview.webrtc.org/2571073002
Cr-Commit-Position: refs/heads/master@{#15582}
An external audio mixer will be passed from PeerConnectionFactory to
AudioTransportProxy.
This CL has rewritten based on reverted CL
https://codereview.chromium.org/2539213003/
The only difference is that
static MediaEngineInterface* Create(
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
audio_decoder_factory,
WebRtcVideoEncoderFactory* video_encoder_factory,
WebRtcVideoDecoderFactory* video_decoder_factory);
in media/engine/webrtcmediaengine.h is kept in this CL instead of
replaced for backward compatibility.
BUG=webrtc:6457
Review-Url: https://codereview.webrtc.org/2570993002
Cr-Commit-Position: refs/heads/master@{#15580}
"Crypto required" is a property of the PeerConnection of construction
time; it has nothing to do with SDP. So I'm moving it out of
MediaContentDescription and putting it in the BaseChannel constructor
instead. This is more intuitive, and provides the added assurance that
"secure_required_" can't be flipped from "true" to "false".
BUG=None
Review-Url: https://codereview.webrtc.org/2537343003
Cr-Commit-Position: refs/heads/master@{#15579}
Also make supported protocols explicit in check.
Fix inconsistency where TLS_PROTOCOL_NAME was not exported.
BUG=webrtc:6885
Review-Url: https://codereview.webrtc.org/2570803003
Cr-Commit-Position: refs/heads/master@{#15577}
The improvement is mainly to extrapolate missing samples so that when querying the output, it assumes the input to continue even if no actual new samples are added.
The new implementation does not rely on base/exp_filter any longer. This is because it would be a bit cumbersome. base/exp_filter does pre-extrapolate, i.e., it assumes the all missing samples since the last sample equals the new sample.
BUG=webrtc:6443
Review-Url: https://codereview.webrtc.org/2551363002
Cr-Commit-Position: refs/heads/master@{#15575}
This ensures that the video_frame_buffer method never can return a
null pointer.
BUG=webrtc:6591
Review-Url: https://codereview.webrtc.org/2541863002
Cr-Commit-Position: refs/heads/master@{#15574}
Add functionality for automatic rolling of individual DEPS
entries. This will make it possible to move away from the
links created by setup_links.py to real DEPS entries.
There are two kinds of such deps we intend to use:
1. Third party dependencies used by both WebRTC and Chromium:
those are rolled to the same revision as the Chromium DEPS file of the revision passed
to the script
2. Chromium sub-directories needed for WebRTC (mainly for //build and BUILD.gn
files of third_party deps): those are rolled to the HEAD revision
Notice that the latter kind could be rolled ahead of the chromium_revision,
but generally these should be close (and if it passes the bots, we don't really mind).
The new functionality can coexist with the old one, to
enable a smooth transition (not everything needs to change at once).
Some of the updating logic was inspired by
https://cs.chromium.org/chromium/build/scripts/slave/recipes/v8/auto_roll_v8_deps.py
Add extensive tests for the logic of figuring out deps changes,
including mocking git ls-remote call to keep the tests hermetic.
BUG=webrtc:5006
NOTRY=True
Review-Url: https://codereview.webrtc.org/2570603003
Cr-Commit-Position: refs/heads/master@{#15570}
AddCodec represents better what this function actually does.
BUG=None
Review-Url: https://codereview.webrtc.org/2573593003
Cr-Commit-Position: refs/heads/master@{#15565}
The Chromium mock implementation implements the new GetStats API, so we
can remove this default implementation.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2566143002
Cr-Commit-Position: refs/heads/master@{#15563}
This CL doesn't start *using* a=bundle-only; it just adds support for
parsing it. We need to do this first, because otherwise old versions of
WebRTC will interpret a zero port value as a rejected m= section.
BUG=webrtc:4674
Review-Url: https://codereview.webrtc.org/2562183002
Cr-Commit-Position: refs/heads/master@{#15558}
Reason for revert:
A interface change broke some downstream code in google3.
Original issue's description:
> Support external audio mixer in webrtc.
>
> An external audio mixer will be passed from PeerConnectionFactory to
> AudioTransportProxy.
>
> BUG=webrtc:6457
>
> Committed: https://crrev.com/f6bcac59e88c3be5ffd73bbb1098f2d82ee316a1
> Cr-Commit-Position: refs/heads/master@{#15556}
TBR=solenberg@webrtc.org,aleloi@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6457
Review-Url: https://codereview.webrtc.org/2562333003
Cr-Commit-Position: refs/heads/master@{#15557}
An external audio mixer will be passed from PeerConnectionFactory to
AudioTransportProxy.
BUG=webrtc:6457
Review-Url: https://codereview.webrtc.org/2539213003
Cr-Commit-Position: refs/heads/master@{#15556}
It was using a non-WebRTC-named header guard, which could conflict with
other similarly named/intended headers.
BUG=None
NO_DEPENDENCY_CHECKS=true
Review-Url: https://codereview.webrtc.org/2548113002
Cr-Commit-Position: refs/heads/master@{#15554}
SimulcastEncoderAdapter calls Release() on a failed sub-encoder init,
but Release only knows how to clean up encoders that have registered
stream info. Since failed ones don't register, they aren't currently
cleaned up.
BUG=None
Review-Url: https://codereview.webrtc.org/2544003005
Cr-Commit-Position: refs/heads/master@{#15553}
Was added for video initially, but not for audio.
BUG=webrtc:6862
Review-Url: https://codereview.webrtc.org/2568553002
Cr-Commit-Position: refs/heads/master@{#15552}
The packetization parts of this class are accessed from the
encoder thread, which might change under different occasions.
The use of a sequenced task checker here is thus incorrect, since
that requires the access to always be on the same thread, whenever
a task queue is not used.
The access to the instantiated object of this class, at least when
it comes to the RTP packetization parts, is however synchronized
using the lock in PayloadRouter::OnEncodedImage. We can therefore
safely remove the sequenced task checker.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2562983002
Cr-Commit-Position: refs/heads/master@{#15549}
The file was aldready pruned down to the point where it only included
webrtc/typedefs.h. Therefore, all includes of
voice_engine_configurations.h are replaced with typedefs.h, except on
two occasions where it was obvously not needed.
BUG=webrtc:6506
Review-Url: https://codereview.webrtc.org/2553583002
Cr-Commit-Position: refs/heads/master@{#15547}
adaptReason in webrtcvideoengine2.h only defines NONE=0, CPU=1 and BANDWIDTH=2 so &0x4 can not happen anymore.
This was probably never implemented in videoengine2
BUG=webrtc:6870
Review-Url: https://codereview.webrtc.org/1887773002
Cr-Commit-Position: refs/heads/master@{#15546}
The changes here are the same as in https://codereview.webrtc.org/2523993002/:
- reduce packet loss from 50% to 5%, to allow the BWE to ramp up better.
- Do not wait for 100 packets to be sent before letting the test pass.
BUG=webrtc:6851
Review-Url: https://codereview.webrtc.org/2558303002
Cr-Commit-Position: refs/heads/master@{#15542}
This also deletes unused features of the video_capturer interface, the classes
VideoCaptureFeedBack, VideoCaptureEncodeInterface and related methods,
and the module id which used to be passed as an argument to the
VideoCaptureDataCallback.
In theory the module id could have been used to let a single
VideoCaptureDataCallback serve several capturers, and demultiplex
on the id, but in practice, it was unused. With this change, it is
required to use a separate VideoSinkInterface for each capturer.
BUG=webrtc:6789
Review-Url: https://codereview.webrtc.org/2534553002
Cr-Commit-Position: refs/heads/master@{#15540}
