Implement AnalogGainStatsReporter and add it in AudioProcessingImpl.
This class computes statistics for analog gain updates and
periodically reports them into a histogram.
The added histograms for analog gain update statistics:
- WebRTC.Audio.ApmAnalogGainDecreaseRate
- WebRTC.Audio.ApmAnalogGainIncreaseRate
- WebRTC.Audio.ApmAnalogGainUpdateRate
- WebRTC.Audio.ApmAnalogGainDecreaseAverage
- WebRTC.Audio.ApmAnalogGainIncreaseAverage
- WebRTC.Audio.ApmAnalogGainUpdateAverage
The histograms are defined in
https://chromium-review.googlesource.com/c/chromium/src/+/3207987
Bug: webrtc:12774
Change-Id: I3c58d4bb3eb034a11c3f39ab8edb2bc67c5fd5e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234140
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35301}
The main change is to remove sockets from epoll if there are no
requested events, which happens when a socket is considered closed
(due to an error or otherwise). This prevents a busy loop when a socket
is an error condition where it will constantly be signaled, but not
deleted by higher level code.
Other related changes:
* Set DE_CLOSE on errors regardless of whether the socket is readable or
writable.
* Don't set DE_ACCEPT on errors.
* Handle getsockopt(SO_ERROR) errors.
* In IsDescriptorClosed:
* Retry recv on EINTR.
* Treat ECONNABORTED and EPIPE as errors.
Original patch contributed by andrey.semashev@gmail.com.
Bug: webrtc:11124
Change-Id: I67f33213efc1418b1ffc8f4867f606b7f8dc4ece
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235863
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35300}
Instead use `parameters_.config.rtp.ssrcs.size()` directly to make decisions about the number of temporal and spatial layer used.
Bug: none
Change-Id: Icba553178ae7fea281c2c67654c510228d9ab5b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237080
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35299}
Removes the ability to accept nonencrypted answers to encrypted offers.
Fixes some logic around bundled sessions and requirement for
transport parameters.
Bug: webrtc:11066
Change-Id: I56d8628d223614918a1e5260fdb8a117c8c02dbd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236344
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35298}
Remove analog_level_minimum and analog_level_maximum from
AudioProcessing GainController1 and replace their use with fixed
values 0 and 255, respectively.
Bug: webrtc:12774
Change-Id: Ia4bfe5ed43a65f1587ed67f36bfbb2966b6fdf26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235822
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35297}
Changes one preexisting enum-to-string function to use the new format.
Also changes the RTC_LOG macros that created collisions with ToString,
for tidiness, and documents the recommended function form.
Bug: webrtc:13272
Change-Id: Ic8bb54ed31402ba32675b142d796cf276ee78df5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235722
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35296}
This reverts commit 3b18208f13e85b356e61a95c0a261e9781403743
and is the third attempt at removing stun origin support
Bug: webrtc:12132
Change-Id: Ic41a6d011fb6239907a257cc4c81ec4d2923dc4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236260
Reviewed-by: Taylor Brandstetter <deadbeef@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35294}
The constants are being made private since no new code should use them.
However, the helper functions sill uses "AV1X" internally for backwards
compatibility.
Bug: webrtc:13166
Change-Id: I0a0cd46f31ca70bb7f395c9b1e9cdb202df11f6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236680
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35289}
Sanitizer builds are already slower than release builds, so removing
DCHECKs might allow for more coverage (less tests skipped because
of timing issues).
Bug: webrtc:13329
Change-Id: I5433f0e520b3ad3e463dea019f3b524a6034f1ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236583
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35285}
The sequences of threads entering the VideoStreamEncoder has been
unclear. Fix this by renaming the uninformational |main_queue_| to
|worker_queue_|, and introduce a new |network_queue_| which is set
on construction.
Bug: chromium:1255737
Change-Id: Ic4d3a5b8188b8cc98e60b72aee2c09c9afbc7356
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236523
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35283}
We need to use RTC_NOT_SANITIZE("cfi-icall") everywhere where we do
function typecasting, otherwise doing official Chrome builds will result
into crash.
Bug: chromium:1262535
Change-Id: If7358ccab6bd626e494b7ecd3077aa29502080c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236587
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35281}
RetransmissionQueue was growing too long (almost 1000 lines), and as
there is reason to believe that more changes are necessary in it for
performance reasons, the data structure that handles managing the
in-flight outstanding data has been extracted as a separate class with
its own test cases. What remains in RetransmissionQueue is that it holds
OutstandingData, fetch data from the SendQueue and manage all congestion
control variables and algorithms.
Bug: webrtc:12943
Change-Id: I46062a774e0e76b44e36c66f836b7d992508bf5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235980
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35279}
Networks tests were previously disabled if building in debug mode as
debug mode adds DCHECKs, and when DCHECKs are enabled, a lot of the
components in dcSCTP will add consistency checks, and they can be really
expensive to run in these network tests.
However, if running in with TSAN or MSAN sanitizers and with DCHECKs
enabled, they also take a long time.
Current run-time on my relatively fast CPU (with is_debug=false):
(no sanitizer) always_dcheck=false: 2.5s
(no sanitizer) always_dcheck=true: 31s
is_tsan=true, always_dcheck=false: 53s
is_tsan=true, always_dcheck=true: 5m50s <-- too slow
is_asan=true, always_dcheck=false: 13s
is_asan=true, always_dcheck=true: 47s
is_msan=true, always_dcheck=false: 35s
is_msan=true, always_dcheck=true: 1m53s <-- too slow
Note that buildbots may be much slower than my computer.
Bug: webrtc:12943
Change-Id: If044ee9936372d54c9899b4864156c9f680af0b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236581
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35276}
This ensures that the payload descriptor and potential generic
descriptors uses the same temporal layer.
Bug: b/200518293
Change-Id: I17e980b47fe6c814cb393fc459064576447aa27a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236520
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35275}
Holdback window can be specified as absolute time and in terms of packet
send times. Example:
WebRTC-TaskQueuePacer/Enabled,holdback_window:20ms,holdback_packet:3/
If current conditions have us running with 2000kbps pacing rate and
1250byte (10kbit) packets, each packet send time is 5ms.
The holdback window would then be min(20ms, 3*5ms) = 15ms.
The default is like before 1ms and packets no take into account when
TQ pacer is used, parameters have no effect with legacy process thread
pacer.
Bug: webrtc:10809
Change-Id: I800de05107e2d4df461eabaaf1ca04fb4c5de51e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233421
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35266}
A vector of which TSNs that were acked for each received SACK was
created, but only used in debug logs, which aren't enabled by default.
Removing them, as they don't add that much value and cost a bit
of performance.
Bug: webrtc:12943
Change-Id: Ice323cf46ca6e469fbbcf2a268ad67ca883bb2f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235985
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35265}
This field is unused within WebRTC, and doesn't seem to
be essential for any existing customers.
If this works well, it will be deprecated and removed.
Bug: none
Change-Id: I96d7485e4d094abfa6a8c3d1e6855834c13dedd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189680
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35263}
This is a reland of b141c162ee2ef88a7498ba8cb8bc852287f93ad2
Original change's description:
> Take out listen support from AsyncPacketSocket
>
> Moved to new interface class AsyncListenSocket.
>
> Bug: webrtc:13065
> Change-Id: Ib96ce154ba19979360ecd8144981d947ff5b8b18
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232607
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35234}
Bug: webrtc:13065
Change-Id: I88bebdd80ebe6bcf6ac635023924d79fbfb76813
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35260}
This is somewhat klugey, because it does the same checks at two
different layers in the stack, in different functions, which runs
the risk of making them out of sync. But it solves the immediate
problem.
Bug: chromium:1249753
Change-Id: I2ad96f0cc9499c15540ff6946a409b40df3e3925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235826
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35259}
Only affects turn server. Refactored to wrap sockets with SSLAdapter
after Accept, using the SSLAdapterFactory to hold needed configuration.
Bug: webrtc:13065
Change-Id: I5df65aad5728d8d40d95b22db6398a573ec7a36f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235823
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35258}
The ios bots are only interested in running ios_remoting_unittests,
so in order to avoid breaking when //ios/{web,chrome} requires new
version of Xcode, set `ios_build_chrome=false` to stop building
those targets (as they are not run/tested since they don't depend
on WebRTC).
Bug: webrtc:13222
Change-Id: Ib08044157d7ee9ea44a3c608310609cad99665b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sylvain Defresne <sdefresne@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35256}
This CL includes two changes that break bit-exactness, but that haven't
changed the way AGC2 behaves - the new behavior has been verified with
audioproc_f on a collection of AEC dumps and Wav files (42 recordings
in total).
1) The fixed digital controller can directly be initialized in the
`GainController2` ctor. Before, `SetGainFactor()` was called after the
creation of the object and that caused an initial ramp up lasting one
10 ms frame from -inf to 0 dB. As an effect of the new initialization,
the initial ramp up doesn't happen anymore.
2) In [1] the AGC2 VAD has been moved from the adaptive digital
controller into `GainController2`. In order to not break bit-exactness,
the VAD was placed after the fixed digital controller and before the
adaptive digital one. However, to reduce the chance of incorrect
estimation of the speech probability, the VAD should analyze the
audio before any digital processing is applied inside AGC2.
[1] https://webrtc-review.googlesource.com/c/src/+/234583
Bug: webrtc:7494
Change-Id: I9418229cbe537014fed8271c5550c3ce2bc88e26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235240
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35252}
