This change extends the definition of wired headset to also include USB
devices. The effect is that audio will now be routed to USB audio devices
when used in combination with AppRTCMobile.
BUG=webrtc:7931
Review-Url: https://codereview.webrtc.org/2971613003
Cr-Commit-Position: refs/heads/master@{#18889}
Original issue:
https://codereview.webrtc.org/2957073002/
Reason for reland:
Failed Android unit tests and failed Windows compile.
The tests seemed related at the time, but not after more consideration.
Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.
BUG=webrtc:7882
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2970793003
Cr-Commit-Position: refs/heads/master@{#18886}
This CL addresses the issue of echo leakage of low level
echoes by making the echo canceller more restrictive for
that scenario.
BUG=webrtc:7930
Review-Url: https://codereview.webrtc.org/2969943002
Cr-Commit-Position: refs/heads/master@{#18884}
Reason for revert:
The following, seemingly related, unit tests crash on Android32 (M Nexus5X).
org.webrtc.PeerConnectionTest#testCompleteSession
org.webrtc.PeerConnectionTest#testDataChannelOnlySession
A Windows build fails with a mysterious compile error.
Original issue's description:
> Add received audio/video call duration metrics based on packets.
>
> Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.
>
> BUG=webrtc:7882
>
> Review-Url: https://codereview.webrtc.org/2957073002
> Cr-Commit-Position: refs/heads/master@{#18881}
> Committed: 746749237aTBR=stefan@webrtc.org,aleloi@webrtc.org,asapersson@webrtc.org,holmer@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7882
Review-Url: https://codereview.webrtc.org/2972613002
Cr-Commit-Position: refs/heads/master@{#18882}
Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.
BUG=webrtc:7882
Review-Url: https://codereview.webrtc.org/2957073002
Cr-Commit-Position: refs/heads/master@{#18881}
The complexity test for the audio processing module have long proven
to give false alarms of complexity regressions for which no related
changes can be identified. Attempts to address that has improved the
that, but the tests do still give false alarms.
This CL deactivates the complexity tests until a better way of
testing this is available.
BUG=chromium:713507, webrtc:5846,webrtc:6685,webrtc:7712
Review-Url: https://codereview.webrtc.org/2897403006
Cr-Commit-Position: refs/heads/master@{#18879}
We'd like to remove all occurrences of WEBRTC_TRACE and delete the
macro! One logging mechanism is enough.
AudioConferenceMixer is scheduled for removal and is one of the
things tracked by bugs.webrtc.org/4690. The logging is changed to not
block webrtc:5118
NOTRY=True
Bug: webrtc:5118
Change-Id: Ibad1ae45e8af1ba5bbe253d4c693ecf9e7c422ac
Reviewed-on: https://chromium-review.googlesource.com/518172
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18876}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
Also adds some full stack test variants with the experiment enabled.
BUG=webrtc:7694
Review-Url: https://codereview.webrtc.org/2949553002
Cr-Commit-Position: refs/heads/master@{#18869}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
Changes since last attempt: Some system headers were moved back to their original location since on Windows compilation breaks otherwise.
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2966523003
Cr-Commit-Position: refs/heads/master@{#18868}
This CL finalizes the support for allowing an external
audio processing module to be used in a peerconnection.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2965703002
Cr-Commit-Position: refs/heads/master@{#18864}
The CL in https://codereview.webrtc.org/2918333002/ enabled
FecTest.FlexfecTest and also added a sequence number offset between
the FEC packets and the media packets. This was to simulate that the
sequence numbers were generated from different spaces, i.e., that they
belong to different SSRCs.
The test does not account for sequence number wraparound, which means
that it could fail when the sequence number offset realization was large.
This CL fixes the problem by ensuring that the offset always lies in
[0, 2^15].
This CL also fixes spelling of UlpfecTest.
BUG=webrtc:7912
TESTED=ninja -C out/Debug && third_party/gtest-parallel/gtest-parallel --gtest_filter="*Flexfec*" -r 1000 out/Debug/modules_tests
Review-Url: https://codereview.webrtc.org/2966753002
Cr-Commit-Position: refs/heads/master@{#18863}
This CL exposes the parameter for adjusting the AEC3 performance
for large rooms.
Bug: webrtc:7519
Review-Url: https://codereview.webrtc.org/2967603002
Cr-Commit-Position: refs/heads/master@{#18862}
We have a tweak preventing multiple deep-examinations of packets; packets with a given SSRC are only inspected deeply (RSID) once (only the first received packet). Once we move to many-to-one stream-to-sink associations, this becomes less useful, and is better removed.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2955373002
Cr-Commit-Position: refs/heads/master@{#18859}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2963273002
Cr-Commit-Position: refs/heads/master@{#18856}
(This got reverted because of a problem with the Opus encoder parts.
Re-landing without changes.)
BUG=webrtc:7837
Review-Url: https://codereview.webrtc.org/2950453002
Cr-Commit-Position: refs/heads/master@{#18855}
This was previously reverted, because external projects were using the
internal webrtc::AudioEncoderOpus class and broke when it was renamed.
This re-land avoids renaming it immediately, to give those projects
time to adapt. It also has to revert some of the changes I had made to the
Config struct, since that was also used by the same external projects.
BUG=webrtc:7831
Review-Url: https://codereview.webrtc.org/2948483002
Cr-Commit-Position: refs/heads/master@{#18852}
Due to the implementation of the analog AGC in the audio
processing module, the detection for the analog gain done in AEC3
fails on some platforms where there is no analog gain to control.
This CL removes that functionality until the AGC behavior has
been corrected.
Bug: webrtc:7910, chromium:738322
Change-Id: Ibdbe1e02252387dfd94b36ba7471f5c56ae27f48
Reviewed-on: https://chromium-review.googlesource.com/556040
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18850}
This CL loosens the requirement on the API jitter in APM
that can be tolerated without affecting the AEC3 performance.
BUG=webrtc:7911,chromium:738323
Review-Url: https://codereview.webrtc.org/2967493004
Cr-Commit-Position: refs/heads/master@{#18849}
Leaving compatibility script in webrtc/tools/compare_videos.py to
avoid breaking our video quality tests in Chromium.
Forwarding GN targets are left in webrtc/tools/BUILD.gn.
BUG=webrtc:7855
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2965593002
Cr-Commit-Position: refs/heads/master@{#18848}
Can be enabled by setting "enable_encrypted_rtp_header_extensions" in
"crypto_options" of "PeerConnectionFactoryInterface::Options" and will
only be used if both peers support it.
BUG=webrtc:3411
Review-Url: https://codereview.webrtc.org/2761143002
Cr-Commit-Position: refs/heads/master@{#18842}
Linking external ObjC / Swift apps fails when the app code is using any
of the new frame-buffer classes RTCI420Buffer, RTCMutableI420Buffer, or
RTCCVPixelBuffer. To fix, we need to add the appropriate attribute to
the classes (e.g. using the RTC_EXPORT macro).
BUG=None
Review-Url: https://codereview.webrtc.org/2961293002
Cr-Commit-Position: refs/heads/master@{#18840}
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
