This is the first step towards removing this function from the interface. Once all implementers of the interface remove their implementations this can be removed.
Bug: webrtc:8572
Change-Id: Ia8f7f1b6949a482787df67b193d4cf999142e06e
Reviewed-on: https://webrtc-review.googlesource.com/27620
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21271}
This patch adds a RelayPortFactoryInterface that allows
for custom relay ports. The factor is added as optional argument
to BasicPortAlloctor. If none is provided a default implementation
that mimics existing behavior is created.
The patch also adds 2 stun functions, namely to copy a
StunAttribute and to remove StunAttribute's from a StunMessage.
Bug: webrtc:8640
Change-Id: I59bd51f0f5e2f8c187dff9fcf003a24c35ed037f
Reviewed-on: https://webrtc-review.googlesource.com/32600
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21267}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
When GetSources is called with non-existing ssrc, it will log the
error and return an empty RtpSource list instead of hitting the DCHECK.
Bug: chromium:793699
Change-Id: I30bebb657de32f87f9c82920fa0b19403893791f
Reviewed-on: https://webrtc-review.googlesource.com/32860
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21258}
This will require a manual roll to downstream projects, since
the //test:perf_test target was introduced.
This is a reland of 10a8e7a9b5261a7e3ce19900ba3511be3b5911f8
Original change's description:
> iOS: Save perf results under Documents/perf_result.json
>
> TBR=henrika@webrtc.org
>
> Bug: webrtc:7156
> Change-Id: Ib00992cce0007e0b5c9274340df1a892f810b0c5
> Reviewed-on: https://webrtc-review.googlesource.com/29202
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21244}
TBR=henrika@webrtc.org, phoglund@webrtc.org
No-Try: true
Bug: webrtc:7156
Change-Id: Iecdb108f605fd1c98acde4d50ab4f5a7b5f6bfaf
Reviewed-on: https://webrtc-review.googlesource.com/32761
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21252}
in particular change bitrate type to int64_t to follow style guide.
With an extra interface it will allow to add both RtpRtcp module
and RtcpTransceiver as feedback sender to PacketRouter
Bug: webrtc:8239
Change-Id: I9ea265686d7cd2d709f0b42e8a983ebe1790a6ba
Reviewed-on: https://webrtc-review.googlesource.com/32302
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21250}
Currently |bw_resolutions_disabled| is set per VP8EncoderImpl instance and reported via
OnEncodedImage callback.
Instead move logic to SendStatisticsProxy to determine if resolution is bw limited or not based
on info that is reported to SendStatisticsProxy::OnEncodedImage.
Bug: webrtc:8643
Change-Id: I6c148e3507a0f04a793775b9f84ce54028b64d0f
Reviewed-on: https://webrtc-review.googlesource.com/31460
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21249}
FFMpeg H264 decoder uses YUVJ420 when video_full_range_flag=1 in
bitstream.
Information about color range might be useful for color converter
and renderer. But currently there is no way to extract it from
the wrapper.
Bug: webrtc:8185
Change-Id: Ifd1113f0eee3d7b5906d0cefbc29b4a1061262f6
Reviewed-on: https://webrtc-review.googlesource.com/32000
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21246}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
Reviewed-on: https://webrtc-review.googlesource.com/31940
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21245}
This is a reland of the tests added in the reverted cl
https://webrtc-review.googlesource.com/c/src/+/23720, with
expectations relaxed to make tests pass also with the current (old)
estimator.
Bug: webrtc:8504
Change-Id: I69fd8cc7e87e05b24be75b146f1cac91c5f96f46
Reviewed-on: https://webrtc-review.googlesource.com/30142
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21243}
This tool is used downstream, so we want to christen rtc_tools as
a kind of api dir for tools. Tools in other locations should be
considered off limits.
I chose rtc_tools because video_quality_toolchain is already there,
which is also used downstream.
Bug: None
Change-Id: I234d874c8a590ca7413357ecda26b16d9b399836
Reviewed-on: https://webrtc-review.googlesource.com/32340
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21236}
The problem was that the encoder was feeded with frames that had 0 as
a timestamp. This confused the encoder. H264 high profile support
clause was also wrong and is corrected.
Bug: webrtc:8601
Change-Id: Ic5a893b4b7573e694f865b63620843b2c9aa489f
Reviewed-on: https://webrtc-review.googlesource.com/32300
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21234}
1. Added WEBRTC_FUCHSIA define.
2. Added PlatformThreadId typedef for Fuchsia.
3. Updated ifdefs for _strnicmp()/strncasecmd(), so _strnicmp()
is used on all platforms
3. Updated ifdefs in clock.cc to avoid invalid assumption that
POSIX = LINUX || MAC .
Bug: chromium:750940
Change-Id: Id7aa98e017f467bcebb78a0b298ba91655502072
Reviewed-on: https://webrtc-review.googlesource.com/31641
Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21233}
When calling CGDisplayStreamCreate(properties = nullptr) this
causes kCGDisplayStreamShowCursor to default to kCFBooleanTrue.
This CL set it to false always as it was assumed. Also if true
this causes some lags when moving the mouse pointer on the capture
side and in any case webrtc::MouseCursorMonitorMac already implements
a custom way to capture the mouse. Which appears to be more efficient
in this usecase.
Bug: webrtc:8625
Change-Id: Id0fae38fa47503d87d1890213706149762fa67fb
Reviewed-on: https://webrtc-review.googlesource.com/30902
Commit-Queue: Julien Isorce <julien.isorce@chromium.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21231}
Concealment is never used in WebRTC since we never feed decoders with
broken bitstream. If so, there is no need to evaluate concealment
quality.
But if we still want to evaluate it then the tests should be
redesigned: recovery frames should be generated with reasonable
interval and quality thresholds should be set to acceptable level.
Bug: webrtc:8524
Change-Id: Ie7197e0a5a88aafcb3b2698185edcb43b71fae3b
Reviewed-on: https://webrtc-review.googlesource.com/32303
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21230}
In order to support WinUWP platform, all main(..) routines must be normalized to the formal int main(int argc, char* argv[]) form. A platform wrapper main is auto-created linking against the default main(...). This can only work if the linkage is exactly matching the proper formal definition and not a loosely defined main(...) alternative.
Bug: webrtc:8608
Change-Id: I606663aaea7df1792c7c5636279617b8926fa5cc
Reviewed-on: https://webrtc-review.googlesource.com/28721
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21229}
Use std::fstream instead of rtc::File to write perf results.
On Android, when I use rtc::File, the results are not written for some reason.
Also rename the flag to '--chartjson_result_file'.
Bug: webrtc:8566
Change-Id: I32215e2233e18690c41050dfd35ac77e01d11f35
Reviewed-on: https://webrtc-review.googlesource.com/32001
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21225}
Example of an error:
** Presubmit ERRORS **
public_deps is not allowed in WebRTC BUILD.gn files because it doesn't map well to downstream build systems.
Used in: call/BUILD.gn (line 12).
Bug: webrtc:8603
Change-Id: I3b785b295ffb018cb9dfc2e14ae816d3e5e29a69
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/30262
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21224}
