* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks
All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1534193008
Cr-Commit-Position: refs/heads/master@{#11191}
Renames CreateFakeNativeHandleFrame to FakeNativeHandle::CreateFrame and
moves into test.gyp target 'fake_video_frames' which contains previous
frame_generator target.
Removes unused warnings from includers of
webrtc/test/fake_texture_frame.h which did not use the function above.
BUG=webrtc:5398
R=kjellander@webrtc.orgTBR=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1554223002 .
Cr-Commit-Position: refs/heads/master@{#11149}
The issue occurred when deserializing and then serializing a rejected
content description, which doesn't have the ICE ufrag/pwd in the first
place.
BUG=webrtc:5105
Review URL: https://codereview.webrtc.org/1534363002
Cr-Commit-Position: refs/heads/master@{#11134}
VideoChannel::GetScreencastMaxPixels and VideoChannel::GetScreencastFps.
Unused in webrtc, also unused in everything indexed by google and chromium code search. With the exception of the magicflute plugin, which I'm told doesn't matter.
Review URL: https://codereview.webrtc.org/1532133002
Cr-Commit-Position: refs/heads/master@{#11108}
This will allow an app to create senders with the same stream id,
without SDP munging.
Review URL: https://codereview.webrtc.org/1538673002
Cr-Commit-Position: refs/heads/master@{#11092}
For applications with a strict filesize limit for debug files,
I added an option to specify a maximum filesize for AEC dumps. An
existing unit test is extended to check that the feature works as
advertised.
BUG=webrtc:4741
TBR=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1413483003
Cr-Commit-Position: refs/heads/master@{#11081}
This regression was introduced by CL 1505573002 to support remote fingerprint update. What happened is that during PrAnswer, we incorrectly do not apply bundle. However, the channel has become writable at that time. When Answer comes, we still reset the srtp_filter but since the channel has been writable, the new SRTP context has never been applied.
We're making sure that we could always apply SRTP context even when channel has been writable. We'll address the issue that bundle should apply even in PrAnswer in a different CL.
BUG=568734
Review URL: https://codereview.webrtc.org/1532543003
Cr-Commit-Position: refs/heads/master@{#11075}
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
On the receiving side, if a candidate arrives with an old ufrag, it will be dropped. If it contains a new frag that has never seen before, it will hold the ufrag and create connections, although those connections are not pingable until the ICE credentials are received.
This could avoid a bunch of ICE generation issues.
BUG=webrtc:5138,webrt:5292
Review URL: https://codereview.webrtc.org/1498993002
Cr-Commit-Position: refs/heads/master@{#11060}
using the wrong sample rate for the render signal.
The CL is basically a partial revert of the related changes done on
output_mixer.cc in the CL https://codereview.webrtc.org/1234463003.
The CL also reverts the removal of the input_sample_rate_hz() method
that was removed as part of the CL
https://codereview.webrtc.org/1379123002 (as it was at that point
no longer used).
It should be noted that this CL turns off the effect of the
IntelligibilityEnhancer when the AudioFrame AudioProcessing APIs are
used. While it may be possible to solve that by adding upsampling after
the API call, that approach was discarded due to that:
-That would add extra processing in the echo path, leading to possible
AEC performance reduction.
-That would add extra complexity for the mobile case.
-That would only patch the intelligibility enhancer operation as the
proper way to do such an operation is within APM.
-The intelligibility enhancer is not active by default anywhere.
BUG=webrtc:5237
Review URL: https://codereview.webrtc.org/1525173002
Cr-Commit-Position: refs/heads/master@{#11045}
If a MediaStream is added to a PeerConnection, and later a track
is added to the MediaStream, a new RtpSender will now be created for
that track, and it will appear in subsequent offers.
Similarly, removed tracks will remove RtpSenders.
BUG=webrtc:5265
Review URL: https://codereview.webrtc.org/1507973003
Cr-Commit-Position: refs/heads/master@{#11040}
This CL:
* Abstracts the functions in GlRectDrawer to an interface.
* Adds viewport location as argument to the draw() functions, because this information may be needed by some shaders. This also moves the responsibility of calling GLES20.glViewport() to the drawer.
* Moves uploadYuvData() into a separate helper class.
* Adds new SurfaceViewRenderer.init() function and new VideoRendererGui.create() function that takes a custom drawer as argument. Each YuvImageRenderer in VideoRendererGui now has their own drawer instead of a common one.
BUG=b/25694445
R=nisse@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1520243003 .
Cr-Commit-Position: refs/heads/master@{#11031}
Additionally:
* Moving all implementation inside RemoteAudioTrack into AudioTrack and remove RemoteAudioTrack.
* AddSink/RemoveSink are now on all audio sources (like they are for video sources).
While doing this I found that some of our tests are broken :) and fixed them. They were broken because AudioTrack didn't previously do much such as updating its state.
BUG=chromium:569526
Review URL: https://codereview.webrtc.org/1522903002
Cr-Commit-Position: refs/heads/master@{#11026}
Original issue's description:
> Revert of Free SCTP data channels asynchronously in PeerConnection. (patchset #3 id:40001 of https://codereview.webrtc.org/1492383002/ )
>
> Reason for revert:
> Breaks WebrtcTransportTest.DataStream, due to different rtc::Thread implementation on Chromium.
>
> Original issue's description:
> > Free SCTP data channels asynchronously in PeerConnection.
> >
> > This is needed so that the data channel isn't deleted while one of its
> > own methods is on the call stack.
> >
> > BUG=565048
> >
> > Committed: https://crrev.com/386869247f28e72a00307a1b5c92465eea343ad2
> > Cr-Commit-Position: refs/heads/master@{#10923}
>
> TBR=pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=565048
>
> Committed: https://crrev.com/a1f567ae9012a8de573b5bde492dd9ca0d17f137
> Cr-Commit-Position: refs/heads/master@{#10977}
BUG=565048
Review URL: https://codereview.webrtc.org/1516943002
Cr-Commit-Position: refs/heads/master@{#11015}
The input resolution (output from video_adapter) can be further scaled down or higher video layer(s) can be dropped due to bitrate constraints.
BUG=webrtc:4112
Review URL: https://codereview.webrtc.org/1502173002
Cr-Commit-Position: refs/heads/master@{#11006}
Ie, rotation is applied in C++ in the VideoFrameFactory is apply_rotation_ is set. If not, rotation is sent in RTP.
BUG=webrtc:4993
R=nisse@chromium.org
Review URL: https://codereview.webrtc.org/1493913007 .
Cr-Commit-Position: refs/heads/master@{#10986}
Reason for revert:
Breaks WebrtcTransportTest.DataStream, due to different rtc::Thread implementation on Chromium.
Original issue's description:
> Free SCTP data channels asynchronously in PeerConnection.
>
> This is needed so that the data channel isn't deleted while one of its
> own methods is on the call stack.
>
> BUG=565048
>
> Committed: https://crrev.com/386869247f28e72a00307a1b5c92465eea343ad2
> Cr-Commit-Position: refs/heads/master@{#10923}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=565048
Review URL: https://codereview.webrtc.org/1513143003
Cr-Commit-Position: refs/heads/master@{#10977}
