henrika f1239b5405 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}
2018-09-25 14:28:10 +00:00
.gn
2018-08-13 13:54:05 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2018-07-23 15:28:48 +00:00
2018-09-04 11:14:07 +00:00
2018-07-23 15:28:48 +00:00
2017-09-15 04:25:06 +00:00
2018-02-23 10:34:16 +00:00

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.

Development

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.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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