Memory frames are now expected to be owned by the mixing participants.
Review-Url: https://codereview.webrtc.org/2127763002
Cr-Commit-Position: refs/heads/master@{#13554}
In Android AppRTC Demo, there was a bug where toggling disable hardware
AGC/NS would not be reflected into the summary of those settings. This
change fixes this issue.
Review-Url: https://codereview.webrtc.org/2184223003
Cr-Commit-Position: refs/heads/master@{#13551}
OutputMixer and AudioConferenceMixer communicated via a callback. OutputMixer implemented an AudioMixerOutputReceiver interface, which defines the callback function NewMixedAudio. This has been removed and replaced by a simple function in the new mixer. The audio frame with mixed audio is now copied one time less. I have also removed one forward declaration.
Review-Url: https://codereview.webrtc.org/2111293003
Cr-Commit-Position: refs/heads/master@{#13550}
Add IsClosed check when excuting some functions so that they can return early if the PeerConnection is closed.
The observer will not be called after the PeerConnection is closed.
BUG=webrtc:5861
Review-Url: https://codereview.webrtc.org/1975453002
Cr-Commit-Position: refs/heads/master@{#13544}
Apparently, a class will fail with VerifyError if it contains catch
statements with an Exception from a newer API, even if the code is never
executed. This happens only on Android versions before 4.4.2 and is a
bug. See https://code.google.com/p/android/issues/detail?id=209129 for
more info.
BUG=b/30376736
Review-Url: https://codereview.webrtc.org/2185833003
Cr-Commit-Position: refs/heads/master@{#13542}
Updated the sources in audio_processing:audioproc_test_utils to match the configuration on
"webrtc/modules/audio_processing/audio_processing_tests.gypi"
Removed audio_buffer_tools from modules_unittests to match the gyp file.
BUG=webrtc:6041
Review-Url: https://codereview.webrtc.org/2178963002
Cr-Commit-Position: refs/heads/master@{#13541}
There is a bug on pre LMR1 devices that only support legacy
implementation of camera2 API where aspect ratio of the camera is
incorrect if the output format doesn't match the aspect ratio of the
sensor array. On these devices, we want to disable the output formats that
have different aspect ratio.
Review-Url: https://codereview.webrtc.org/2181803003
Cr-Commit-Position: refs/heads/master@{#13538}
It can fail in some real circumstances, such as when IDs are exhausted
or you explicitly try to create one with an already-used ID.
Review-Url: https://codereview.webrtc.org/2181933002
Cr-Commit-Position: refs/heads/master@{#13535}
These tests will be reenabled and fixed after Opus 1.1.3 has landed in
Chromium and is rolled into WebRTC.
BUG=
Review-Url: https://codereview.webrtc.org/2185673002
Cr-Commit-Position: refs/heads/master@{#13534}
maximum allowed sized raised from limited by physical udp packet size to
limited by theoritical maximum rtcp packet size.
BUG=webrtc:5260
R=åsapersson
Review-Url: https://codereview.webrtc.org/1998633002
Cr-Commit-Position: refs/heads/master@{#13532}
This is a first CL wiring up AudioSendStream to BitrateAllocator. This
is still experimental and there is a test added for the audio only case,
combined audio video variable bitrate test cases will be added as a
follow up.
BUG=5079
Review-Url: https://codereview.webrtc.org/2165743003
Cr-Commit-Position: refs/heads/master@{#13527}
The iOS H264 video toolbox encoder is currently undershooting the
intended bitrate. This seems to be caused by the data rate limit
property. This CL increases the data rate limit to a set
percentage above the intended bitrate to avoid undershooting. The
AverageBitRate property is still set to the intended bitrate, which
keeps it from overshooting the intended bitrate.
BUG=b/28713684
Review-Url: https://codereview.webrtc.org/2177873003
Cr-Commit-Position: refs/heads/master@{#13526}
to be aware about rare situation where packet resend before sent:
Expectations reduced by validating frame was rendered after or before last
packet for that frame was dropped.
BUG=webrtc:5540
Review-Url: https://codereview.webrtc.org/2180903002
Cr-Commit-Position: refs/heads/master@{#13523}
Test was expecting no rtx packet before dropped packet.
Because of prober there might be some non-padding rtx packets before nack.
Those checks removed, test primary expectations are unaffected.
BUG=webrtc:5540
R=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2180843002
Cr-Commit-Position: refs/heads/master@{#13522}
This CL fixes a crash that could happen when JSON event tracing is
shutting down. The cause of the crash was the fact that the logger
thread function was returning 'true', causing the platform thread to run
it repeatedly even though that wasn't the intention.
Usually the EventLogger::Stop() function would set the event requesting
the logging thread to clean up and close the file, and then immediately
call PlatformThread::Stop() which would stop the outer loop. The Log()
function would only run once and everything behaves as expected.
However, if a context switch happens between the shutdown_event_.Set()
and logging_thread_.Stop() calls in EventLogger::Stop(), the logger
thread function would close the file and exit the Log() method, while
PlatformThread will rerun it again. So the Log() function runs twice,
and the second time output_file_ is NULL which either causes the DCHECK
to fail (in debug builds) or the fprintf() to crash with SIGSEGV (in
release builds).
The fix simply changes the return value of the thread function to false
so it never runs twice.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2168283002 .
Cr-Commit-Position: refs/heads/master@{#13510}
