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webrtc_m130/modules
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Rasmus Brandt 69008a8718 Avoid div-by-zero in VideoCodecTest stats calculation.
Bug: webrtc:10400
Change-Id: I82b1e86cc8f7d1547fc4863c08c0f8ab82801ac4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128086
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27145}
2019-03-15 13:15:02 +00:00
..
audio_coding
Increase the default maximum jitter buffer size to 200 packets.
2019-03-15 10:25:11 +00:00
audio_device
Reland "Replacing rtc::Thread with task queue for TestAudioDeviceModule."
2019-03-13 09:01:05 +00:00
audio_mixer
…
audio_processing
Qualify cmath functions.
2019-03-15 07:31:59 +00:00
bitrate_controller
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congestion_controller
Revert "Log an error if the RTT is negative"
2019-03-15 13:11:24 +00:00
desktop_capture
…
include
Delete VCMPacket constructor with WebRtcRTPHeader
2019-03-12 15:18:41 +00:00
pacing
…
remote_bitrate_estimator
Qualify cmath functions.
2019-03-14 13:09:34 +00:00
rtp_rtcp
…
third_party
…
utility
…
video_capture
Remove dependency on DirectShow baseclasses (streams.h from the winsdk_samples directory).
2019-03-12 11:52:20 +00:00
video_coding
Avoid div-by-zero in VideoCodecTest stats calculation.
2019-03-15 13:15:02 +00:00
video_processing
…
BUILD.gn
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module_common_types_unittest.cc
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OWNERS
…
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