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webrtc_m130/modules
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Hanna Silen c69188d15a AudioProcessingImpl: Add input volume unit tests
Bug: webrtc:7494
Change-Id: I5a32359cacfb7cd6b610ae13b95f92283c761362
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275500
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38132}
2022-09-20 15:29:59 +00:00
..
async_audio_processing
…
audio_coding
RtpPacketInfo: new ctor + deprecated ctors clean-up
2022-09-20 08:58:38 +00:00
audio_device
In android aaudio wrappers use threads through TaskQueue interface
2022-09-05 11:10:21 +00:00
audio_mixer
RtpPacketInfo: new ctor + deprecated ctors clean-up
2022-09-20 08:58:38 +00:00
audio_processing
AudioProcessingImpl: Add input volume unit tests
2022-09-20 15:29:59 +00:00
congestion_controller
Add field trial to not probe if estimates are larger that max needed.
2022-09-20 07:55:49 +00:00
desktop_capture
Make ScreenCastPortal::CaptureSourceType private
2022-09-15 02:21:08 +00:00
include
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pacing
Make it possible to set the packet size needed to trigger a probe.
2022-09-15 10:13:57 +00:00
remote_bitrate_estimator
…
rtp_rtcp
Surface local_capture_clock_offset from RtpSource
2022-09-20 12:51:22 +00:00
third_party
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utility
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video_capture
…
video_coding
Remove forward declares
2022-09-16 13:33:08 +00:00
video_processing
…
BUILD.gn
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module_common_types_unittest.cc
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