| commit | f1239b5405158e1e1cc9d6cc022b27b4acb2cebf | [log] [tgz] |
|---|---|---|
| author | henrika <henrika@webrtc.org> | Tue Sep 25 15:39:22 2018 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Sep 25 14:28:10 2018 +0000 |
| tree | d5a830cd9de5684477ba30e51f8a8c271cc29299 | |
| parent | f3119ef66d5c11f491adea1bac839af037819161 [diff] |
Fixes issue where WebRTC.Audio.RecordSampleRateOffsetInPercent can report 100% (part II) See https://webrtc-review.googlesource.com/c/src/+/100241 for part I. It can happen that recording fail to start but playout works. If that happens, we can log stats like this: 9-25 15:32:02.023 13903 13948 I audio_device_buffer.cc: (line 414): [REC : 10002msec, 48kHz] callbacks: 0, samples: 0, rate: 0, rate diff: 100%, level: 0 09-25 15:32:02.024 13903 13948 I audio_device_buffer.cc: (line 432): [PLAY: 10002msec, 48kHz] callbacks: 1002, samples: 480960, rate: 48086, rate diff: 0%, level: 0 09-25 15:32:12.028 13903 13948 I audio_device_buffer.cc: (line 414): [REC : 10005msec, 48kHz] callbacks: 0, samples: 0, rate: 0, rate diff: 100%, level: 0 09-25 15:32:12.028 13903 13948 I audio_device_buffer.cc: (line 432): [PLAY: 10005msec, 48kHz] callbacks: 1000, samples: 480000, rate: 47976, rate diff: 0%, level: 0 hence, we log invalid UMA stats for a rate offset of 100%. This change fixes the problem, and in the case above, we now instead log: 09-25 15:35:56.141 14116 14161 I audio_device_buffer.cc: (line 432): [PLAY: 10011msec, 48kHz] callbacks: 1002, samples: 480960, rate: 48043, rate diff: 0%, level: 0 09-25 15:36:06.151 14116 14161 I audio_device_buffer.cc: (line 432): [PLAY: 10010msec, 48kHz] callbacks: 1002, samples: 480960, rate: 48048, rate diff: 0%, level: 0 09-25 15:36:16.162 14116 14161 I audio_device_buffer.cc: (line 432): [PLAY: 10011msec, 48kHz] callbacks: 1001, samples: 480480, rate: 47995, rate diff: 0%, level: 0 09-25 15:36:26.173 14116 14161 I audio_device_buffer.cc: (line 432): [PLAY: 10011msec, 48kHz] callbacks: 1001, samples: 480480, rate: 47995, rate diff: 0%, level: 0 Bug: b/113648245 Change-Id: Ic8cb71ca049ef24bf68963a81f95d4e5c2282518 Reviewed-on: https://webrtc-review.googlesource.com/101881 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24828}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.