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}
Avoids hitting a DCHECK in AudioRecordJni::OnCacheDirectBufferAddress()
when first init attempt has failed and we try again.
Bug: b/69434512
Change-Id: I4396ba22981d9258d6d72188bad66104255f19cf
Reviewed-on: https://webrtc-review.googlesource.com/31842
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21218}
This CL modifies the refactored render buffering scheme in AEC3
so that:
-A non-causal state can never occur which means that situations with
nonrecoverable echo should not occur.
-For a stable audio pipeline with a predefined API call jitter,
render overruns and underruns can never occur.
Bug: webrtc:8629,chromium:793305
Change-Id: I06ba1c368f92db95274090b08475dd02dbb85145
Reviewed-on: https://webrtc-review.googlesource.com/29861
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21215}
The experimental function that scales the histogram of inter-arrival times in NetEq suffered from an overflow bug. This caused unexpected increases in the calculated target level.
Bug: webrtc:8381
Change-Id: I2af4d22119fdc684b3cac838c9b317959af17a1f
Reviewed-on: https://webrtc-review.googlesource.com/30261
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21213}
- Defines stereo codec case, similar to RTX, that adds stereo codec to the SDP
negotiation. The underlying codec's payload type is similarly defined by "apt".
- If this negotiation is successful, codec name is included in sdp line via
"acn".
- Adds codec setting initializers for these specific stereo cases.
- Introduces new Stereo*Factory classes as optional convenience wrappers that
inserts stereo codec to the existing set of supported codecs on demand.
This CL is the step 5 for adding alpha channel support over the wire in webrtc.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: Ie12c56c8fcf7934e216135d73af33adec5248f76
Reviewed-on: https://webrtc-review.googlesource.com/22901
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21210}
Using fully qualified paths to include libyuv headers allows WebRTC to
avoid to rely on the //third_party/libyuv:libyuv_config target to
set the -I compiler flag.
Today some WebRTC targets depend on //third_party/libyuv only to
include //third_party/libyuv:libyuv_config but with fully qualified
paths this should not be needed anymore.
A follow-up CL will remove //third_party/libyuv from some targets that
don't need it because they are not including libyuv headers.
Bug: webrtc:8605
Change-Id: Icec707ca761aaf2ea8088e7f7a05ddde0de2619a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28220
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21209}
The computation (x-127) << 8 is undefined for x < 127.
This CL replaces the shift with a multiplication: (x-127) * (1 << 8)
Bug: chromium:793201
Change-Id: I38b40bd88300208a0bfbbd8fe144b0a5b51a48ed
Reviewed-on: https://webrtc-review.googlesource.com/31800
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21205}
Arguably we should add a few more targets, for instance a utility
target, but I tried to create as few targets as possible here based on
the current usage.
Bug: webrtc:6828
Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8
Reviewed-on: https://webrtc-review.googlesource.com/28301
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21199}
The definition of this field in RFC 3550 says that under certain
conditions it may have a negative value. This change exposes that
property in the WebRTC API.
Bug: webrtc:8626
Change-Id: I4ee249da045dcee940db66ebd915268a97fc13db
Reviewed-on: https://webrtc-review.googlesource.com/31260
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21159}
Instead of modifying the API, we'll add a new function to return
the true value, and have a shim that returns what other code expects.
> This reverts commit 4c34f435db2b921b82b8be19ee5c1746f46cb188.
>
> Reason for revert: Broke internal projects. Type mismatch.
>
> Original change's description:
> > Make RTCP cumulative_lost be a signed value
> >
> > This is formally defined as a signed 24-bit value in RFC 3550 section 6.4.1.
> > See RFC 3550 Appendix A.3 for the reason why it may turn negative.
> >
> > Noticed on discuss-webrtc mailing list.
> >
> > BUG=webrtc:8626
> >
> > Change-Id: I7317f73e9490a876e8445bd3d6b66095ce53ca0a
> > Reviewed-on: https://webrtc-review.googlesource.com/30901
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21142}
>
> TBR=stefan@webrtc.org,hta@webrtc.org
>
> Change-Id: I544f7979d584cfb72a2d0d526f4fef84aebeecb3
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8626
> Reviewed-on: https://webrtc-review.googlesource.com/31040
> Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21144}
Change-Id: I95c8c248f4f85c4d1aa2a47424d8c4d954d4ae7a
Bug: webrtc:8626
Reviewed-on: https://webrtc-review.googlesource.com/31220
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21154}
This reverts commit 4c34f435db2b921b82b8be19ee5c1746f46cb188.
Reason for revert: Broke internal projects. Type mismatch.
Original change's description:
> Make RTCP cumulative_lost be a signed value
>
> This is formally defined as a signed 24-bit value in RFC 3550 section 6.4.1.
> See RFC 3550 Appendix A.3 for the reason why it may turn negative.
>
> Noticed on discuss-webrtc mailing list.
>
> BUG=webrtc:8626
>
> Change-Id: I7317f73e9490a876e8445bd3d6b66095ce53ca0a
> Reviewed-on: https://webrtc-review.googlesource.com/30901
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21142}
TBR=stefan@webrtc.org,hta@webrtc.org
Change-Id: I544f7979d584cfb72a2d0d526f4fef84aebeecb3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8626
Reviewed-on: https://webrtc-review.googlesource.com/31040
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21144}
This is formally defined as a signed 24-bit value in RFC 3550 section 6.4.1.
See RFC 3550 Appendix A.3 for the reason why it may turn negative.
Noticed on discuss-webrtc mailing list.
BUG=webrtc:8626
Change-Id: I7317f73e9490a876e8445bd3d6b66095ce53ca0a
Reviewed-on: https://webrtc-review.googlesource.com/30901
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21142}
In the ClearTo function we risked removing one packet too many from the set of
|missing_packets_|, and in the case of H264 this would cause us to create incomplete
delta frames. This is turn disrupt the stream and a keyframe request has to be made.
Bug: webrtc:8536
Change-Id: Ie7ccd5f1631a4cf3bd463878d5b0fe746744ec23
Reviewed-on: https://webrtc-review.googlesource.com/30141
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21119}
This CL reverts the changes introduced that handles echoes in AEC3.
The revert is done to match the behavior which is in M63.
Bug: webrtc:8615,chromium:792346
Change-Id: I128ccb17dc359c7889a701a2faaaf06be40f86dd
Reviewed-on: https://webrtc-review.googlesource.com/30140
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21117}
This adds a command line flag to generate a python visualization script from neteq_rtpplay.
Bug: webrtc:8614
Change-Id: Ia6f10d7ff0abac6fdbe9302e7f97a8a12a5bb65b
Reviewed-on: https://webrtc-review.googlesource.com/29940
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21116}
WebRTC standalone tests show 24% speed up on foreman_cif, 16% speed up
on a 240p clip and 11% speed up on Bridge_r180.
Bug: None
Change-Id: I433b7a8841bd9df2402575f72dd1984cc5e011a9
Reviewed-on: https://webrtc-review.googlesource.com/29260
Commit-Queue: Jerome Jiang <jianj@google.com>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21096}
Alr state is now logged by the pacer. To avoid confusion,
loopback tools will now create two separate rtc event
logs for sender and receiver calls.
Bug: webrtc:8287, webrtc:8588
Change-Id: Ib3e47d109c3a65a7ed069b9a613e6a08fe6a2f30
Reviewed-on: https://webrtc-review.googlesource.com/26880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21084}
There's no downstream use of kFileFormatCompressedFile,
kFileFormatPreencodedFile or kFileFormatPcm48kHzFile.
Bug: None
Change-Id: I66cbe71151472d6348515a2432a280acbc3bbf85
Reviewed-on: https://webrtc-review.googlesource.com/28040
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21078}
A lot of WebRTC targets were depending on //third_party/libyuv using
public_deps instead of deps. This causes issues because a the
inclusion of libyuv headers is not declared to the build system and
this creates hidden dependencies that put the modularity of the project
at risk.
Bug: webrtc:8603
Change-Id: Ide0ceb84eb5640ae664dc782f3a722b55c3b601a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28120
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21039}
In particular, PlayoutStereoData and StartPlayingAudioFile. This also
eliminates the dependency on system_wrappers FileWrapper.
Bug: None
Change-Id: I61df1eea1ad5f5035e36c8229febbf3668808f65
Reviewed-on: https://webrtc-review.googlesource.com/28121
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21038}
This will also cause us to use the new Android HardwareVideoEncoder,
instead of the deprecated MediaCodecVideoEncoderFactory. Unfortunately,
the new HW encoder does not seem to work as good as the old (or the new
encoder is more strict with return values or something). I don't think
it adds much value to continue testing the deprecated encoder, so I
filed a bug for fixing the new encoder, and in this CL I disabled the
tests on Android. I want to remove as many places as possible where we
use the old WebRtcVideoEncoderFactory interface, because it makes it
more difficult to migrate to the new interface.
Bug: webrtc:7925
Change-Id: If8e34752148a5e5139944d2dfbe7e231fe58aeb9
Reviewed-on: https://webrtc-review.googlesource.com/27540
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21037}
This reverts commit 0af8370cb38b0b0f35f4ed4ec4237d0e6c7d59da.
Reason for revert: Breaks downstream
Original change's description:
> Fix all circular deps in audio_processing (but one).
>
> Arguably we should add a few more targets, for instance a utility
> target, but I tried to create as few targets as possible here based on
> the current usage.
>
> Bug: webrtc:6828
> Change-Id: If2740de2e4374eeae64b3d7599a52bb051594c6a
> Reviewed-on: https://webrtc-review.googlesource.com/28020
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21025}
TBR=phoglund@webrtc.org,peah@webrtc.org
Change-Id: I423f027f6919cf4eb44b4e08c7cb38f0506ad0d7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6828
Reviewed-on: https://webrtc-review.googlesource.com/28940
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21027}
Arguably we should add a few more targets, for instance a utility
target, but I tried to create as few targets as possible here based on
the current usage.
Bug: webrtc:6828
Change-Id: If2740de2e4374eeae64b3d7599a52bb051594c6a
Reviewed-on: https://webrtc-review.googlesource.com/28020
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21025}
This CL centralizes the render buffering in AEC3 so that all render
buffers are updated and synchronized/aligned with the render alignment
buffer.
Bug: webrtc:8597, chromium:790905
Change-Id: I8a94e5c1f27316b6100b420eec9652ea31c1a91d
Reviewed-on: https://webrtc-review.googlesource.com/25680
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20989}
