Artem Titov ac9365ed64 Set safe values to prevent possible sigsegv while using AudioTransport, add doc ...

Bug: webrtc:8946
Change-Id: Ica066a05905894fba6ba24e45af46b0d5951b5d5
Reviewed-on: https://webrtc-review.googlesource.com/65040
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22652}

2018-03-28 15:05:26 +00:00

..

android

Annotate rest of WebRTC with @Nullable.

2018-03-28 08:30:06 +00:00

dummy

Move file_wrapper.h to rtc_base/system/

2018-03-23 11:17:15 +00:00

include

Set safe values to prevent possible sigsegv while using AudioTransport, add doc

2018-03-28 15:05:26 +00:00

ios

Add support of AAudio in native WebRTC on Android O and above

2018-03-16 10:20:27 +00:00

linux

Remove unused members from AudioDeviceBuffer

2018-01-16 10:20:32 +00:00

mac

Remove unused members from AudioDeviceBuffer

2018-01-16 10:20:32 +00:00

win

Lowercase windows includes in win audio device.

2018-03-27 05:00:40 +00:00

audio_device_buffer.cc

Add support of AAudio in native WebRTC on Android O and above

2018-03-16 10:20:27 +00:00

audio_device_buffer.h

Delete unneeded includes of wav_file.h and file_wrapper.h.

2018-03-20 15:59:27 +00:00

audio_device_config.h

Don't build windows core audio if using dummy file devices.

2017-10-27 12:53:34 +00:00

audio_device_data_observer.cc

Set safe values to prevent possible sigsegv while using AudioTransport, add doc

2018-03-28 15:05:26 +00:00

audio_device_generic.cc

Removes Set/GetLoudspeakerStatus APIs from the ADM.

2017-11-16 19:44:24 +00:00

audio_device_generic.h

Removes usage of AGC APIs in the ADM.

2017-12-13 16:32:21 +00:00

audio_device_impl.cc

Add support of AAudio in native WebRTC on Android O and above

2018-03-16 10:20:27 +00:00

audio_device_impl.h

Removes usage of AGC APIs in the ADM.

2017-12-13 16:32:21 +00:00

audio_device_unittest.cc

Minor improvements in ADM unittest for Windows.

2018-03-20 11:57:35 +00:00

BUILD.gn

Move file_wrapper.h to rtc_base/system/

2018-03-23 11:17:15 +00:00

DEPS

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

fine_audio_buffer_unittest.cc

Add support of AAudio in native WebRTC on Android O and above

2018-03-16 10:20:27 +00:00

fine_audio_buffer.cc

Add support of AAudio in native WebRTC on Android O and above

2018-03-16 10:20:27 +00:00

fine_audio_buffer.h

Add support of AAudio in native WebRTC on Android O and above

2018-03-16 10:20:27 +00:00

mock_audio_device_buffer.h

Remove unused members from AudioDeviceBuffer

2018-01-16 10:20:32 +00:00

OWNERS

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