webrtc/modules/audio_device/android
henrika 36b3179312 Removes flaky thread checker in AudioDeviceBuffer.
This CL removes a set of DCHECKs in AudioDeviceBuffer (ADB) where the goal has been
to ensure that some methods are called on one and the same native I/O thread.
The implementation of the ADB is platform independent but the underlying (driving)
audio components differ between platforms. This combination has shown to generate complex
corner cases such as:

- OS dependent I/O-thread(s) changes while audio is active
- OS dependent audio device changes and it leads to restart of native I/O threads
- Start/Stop of audio has different timing depending on platform and possibly also usage of
JNI and/or emulators.

To summarize: the gain of maintaining the current strict thread checking (in Debug mode)
is not worth all the efforts trying to resolve complex dynamic cases where the native
I/O threads changes ID.

TBR=glaznev

Bug: b/115385789
Change-Id: I681c89adec497a18b97d2a40421c04ea218fd919
Reviewed-on: https://webrtc-review.googlesource.com/100200
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24723}
2018-09-13 11:41:52 +00:00
..
java/src/org/webrtc/voiceengine Remove redundant initializers from WebRTC Java code. 2018-09-11 09:58:10 +00:00
aaudio_player.cc Removes flaky thread checker in AudioDeviceBuffer. 2018-09-13 11:41:52 +00:00
aaudio_player.h Add support of AAudio in native WebRTC on Android O and above 2018-03-16 10:20:27 +00:00
aaudio_recorder.cc Removes flaky thread checker in AudioDeviceBuffer. 2018-09-13 11:41:52 +00:00
aaudio_recorder.h Add support of AAudio in native WebRTC on Android O and above 2018-03-16 10:20:27 +00:00
aaudio_wrapper.cc Add support of AAudio in native WebRTC on Android O and above 2018-03-16 10:20:27 +00:00
aaudio_wrapper.h Add support of AAudio in native WebRTC on Android O and above 2018-03-16 10:20:27 +00:00
audio_common.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_device_template.h Removes usage of AGC APIs in the ADM. 2017-12-13 16:32:21 +00:00
audio_device_unittest.cc Relands Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC 2018-09-05 10:24:35 +00:00
audio_manager_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_manager.cc Force alignment of JVM called functions. 2018-03-23 10:20:55 +00:00
audio_manager.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_record_jni.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_record_jni.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_track_jni.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_track_jni.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
build_info.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
build_info.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ensure_initialized.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ensure_initialized.h
opensles_common.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
opensles_common.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
opensles_player.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
opensles_player.h FineAudioBuffer now uses 16-bit audio samples to match the AudioDeviceBuffer. 2018-04-19 12:20:28 +00:00
opensles_recorder.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
opensles_recorder.h FineAudioBuffer now uses 16-bit audio samples to match the AudioDeviceBuffer. 2018-04-19 12:20:28 +00:00