- add test that checks that the computed VAD probability is within
tolerance *1
- speed-up some tests by reducing the input length and skipping frames
- remove unused code in test_utils
- fix some comments
*1: RnnVadTest::RnnBitExactness is replaced by
RnnVadTest::RnnVadProbabilityWithinTolerance
Bug: webrtc:10480
Change-Id: I19332d06eacffbbe671bf7749ff4c92798bdc55c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133910
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27803}
This CL ensures that the AECm is only created when needed.
The changes in the CL are bitexact when running AECm via
audioproc_f
The CL also corrects an issue where there is a risk for
AEC2 to not be correctly setup when the sample rate
changes inbetween activations.
Bug: webrtc:8671
Change-Id: Id3b33e20969b1543e28c885d47495246cfbe549d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134216
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27800}
https://webrtc-review.googlesource.com/c/src/+/119121 added two calls to set the observed overhead. Both SetupSendCodec() and ReconfigureSendCodec() update the encoder's overhead. However, these calls happen before RTP has issued any callbacks to set the overhead, so they tell the encoder that the overhead is zero.
This change checks whether the overhead has been set to a non-zero value before each of the new calls and adds a DCHECK to quickly catch future cases which attempt to set overhead to zero.
Bug: webrtc:10150
Change-Id: Ieb3345ecfcda1cf25538d5d424383df17a71b4a2
TBR: solenberg@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134260
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27793}
This reverts commit 0870c70b0471c3bae16ad9a6732d812ee25446dd.
Reason for revert: Failed to handle lost packets.
Original change's description:
> Reland "Improving robustness of feedback matching code in event log parser."
>
> This is a reland of a1e4fbb25371867349a0c2ed6ba62224735a2ec7
>
> Original change's description:
> > Improving robustness of feedback matching code in event log parser.
> >
> > Removes the dependency on TransportFeedbackAdapter thereby removing
> > some of the complexity that came with it, in particular, we don't fill
> > in missing packets. This makes the code easier to debug and avoids some
> > confusing logging that's not relevant for the parser.
> >
> > Bug: webrtc:9883
> > Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27739}
>
> Bug: webrtc:9883
> Change-Id: I460d0c576626614fb4ce2c3d5e3ddbb5d1c122cf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134106
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27763}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9883
Change-Id: Ibcfc4f7425fe202d86f0c3a33de51e605dc17c04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134312
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27790}
It appears unused everywhere. It will be deleted in a followup cl.
Bug: webrtc:6471
Change-Id: Ief992db6e52aee3cf1bc77ffd659ffbc072672ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134212
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27787}
In this CL:
- Assign frame IDs so that simulcast streams share one frame ID space.
- Added a CodecBufferUsage class that represent how a particular buffer
was used (updated, referenced or both).
- Calculate frame dependencies based on the CodecBufferUsage information.
Bug: webrtc:10342
Change-Id: I4ed5ad703f9376a7d995c04bb757c7d214865ddb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131287
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27784}
This CL extends the supported runtime settings in
APM to also comprise the AGC2 fixed gain.
The CL was originally created by Adam Whiteside.
Bug: webrtc:10574
Change-Id: I79b3d6501f1e202b66a9b6018f8a493a56b01f62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27782}
This CL adds a new metric to NetEq, which logs whenever a loss
concealment event has lasted longer than 150 ms (an "interruption").
The number of such events, as well as the sum length of them, is kept
in a SampleCounter, which can be queried at any time.
Any initial PLC at the beginning of a call, before the first packet is
decoded, is ignored.
Unit tests and piping to neteq_rtpplay are included.
Bug: webrtc:10549
Change-Id: I8a224a34254c47c74317617f420f6de997232d88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132796
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27781}
Compiling without BoringSSL fails since g_use_time_callback_for_testing
is defined inside a OPENSSL_IS_BORINGSSL block.
Bug: webrtc:10160
Change-Id: I25c27fa8ed128a50aa855db2012026c97954b91b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134226
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27779}
Follows https://webrtc-review.googlesource.com/c/src/+/129768 closely.
Adds an ENCODER and sets it up to parse SDP config for multistream
opus.
E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"
Bug: webrtc:8649
Change-Id: I3fc341e76f5c41dab0243cf65f6461e4c3d9d67d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132001
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27775}
Use the MockVideoEncoder from the api target in
encoder_simulcast_proxy_unittest, rather than a custom MockEncoder.
This also prevents issue when new SetRates() is made pure virtual.
Bug: webrtc:10481
Change-Id: I72469803c00f7014eeac5b9321d1e0d716fa245d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134211
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27774}
This CL ensures that the AEC2 is only created when needed.
The changes in the CL are bitexact when running AEC2 via
audioproc_f
Bug: webrtc:8671
Change-Id: I5f6d33e45a7031c69ac53098781635c415668e49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129740
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27772}
Removing functionality to choose congestion controller implementation,
using injection instead. Also cleaning up some related functionality
that's no longer needed, such as the injection of event logs into the
factory.
Bug: webrtc:9883
Change-Id: Ia528005625430ae31a15bc88881e2d4ac6ad1d42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133890
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27768}
8-bit H264 HDR content is not rendered correctly in Chrome on Windows.
This is a temporary fix that converts the 8-bit buffer to a 10-bit
buffer if the color space indicates that the buffer should be
rendered as HDR.
Bug: webrtc:10575
Change-Id: I106612ec489c6371fa774424a4cf07d9bad40fc3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134040
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27766}
This reverts commit c9a2c5e93aa51606916e6728454bcff26bb75f79.
Reason for revert: Breaks downstream test
Original change's description:
> Reland "Copy video frames metadata between encoded and plain frames in one place"
>
> Reland with fixes: Do not remove extra metadata copies in software decoders as some downstream projects assumes these fields are copied by the encoders.
>
> Currently some video frames metadata like rotation or ntp timestamps are
> copied in every encoder and decoder separately. This CL makes copying to
> happen at a single place for send or receive side. This will make it
> easier to add new metadata in the future.
>
> Also, added some missing tests.
>
> Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
>
> Bug: webrtc:10460
> Change-Id: I8e49589bf75f406e2b5ddee34882de0faedbd09a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134102
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27756}
TBR=ilnik@webrtc.org,sprang@webrtc.org,kron@webrtc.org,artit@webrtc.org
Change-Id: I34cc563ec6383735c2a76a6f45a72a7726b74421
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134204
Reviewed-by: Artem Titarenko <artit@google.com>
Commit-Queue: Artem Titarenko <artit@google.com>
Cr-Commit-Position: refs/heads/master@{#27765}
This is a reland of a1e4fbb25371867349a0c2ed6ba62224735a2ec7
Original change's description:
> Improving robustness of feedback matching code in event log parser.
>
> Removes the dependency on TransportFeedbackAdapter thereby removing
> some of the complexity that came with it, in particular, we don't fill
> in missing packets. This makes the code easier to debug and avoids some
> confusing logging that's not relevant for the parser.
>
> Bug: webrtc:9883
> Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27739}
Bug: webrtc:9883
Change-Id: I460d0c576626614fb4ce2c3d5e3ddbb5d1c122cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134106
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27763}
