Alessio Bazzica 7c19a706b0 Audio Processing Module: add play-out audio device runtime information ... Add a runtime setting that notifies play-out audio device changes. The payload is a pair indicating a device id and its maximum play-out volume. kPlayoutVolumeChange is now forwarded not only to capture, but also render (required by render_pre_processor). Bug: webrtc:10608 Change-Id: I8997c207422c1dcd1d53775397d6290939ef3db8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159002 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29725}		2019-11-07 13:33:09 +00:00
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aec_dump_factory.h	Change StartAecDump methods to work with FILE* and FileWrapper	2019-06-11 13:43:36 +00:00
aec_dump_impl.cc	Audio Processing Module: add play-out audio device runtime information	2019-11-07 13:33:09 +00:00
aec_dump_impl.h	Delete almost all includes of platform_file.h	2019-06-28 07:30:15 +00:00
aec_dump_integration_test.cc	Use std::make_unique instead of absl::make_unique.	2019-09-17 15:47:29 +00:00
aec_dump_unittest.cc	Format almost everything.	2019-07-08 13:45:15 +00:00
BUILD.gn	Move rtc_base/ignore_wundef.h to its own target.	2019-10-19 10:50:36 +00:00
capture_stream_info.cc	Add FixedGainController and move GainController2 in APM.	2018-02-16 10:56:38 +00:00
capture_stream_info.h	Remove dependencies on modules:module_api from AudioProcessing.	2018-04-12 22:05:27 +00:00
mock_aec_dump.cc	Reformat the WebRTC code base	2018-06-19 14:00:39 +00:00
mock_aec_dump.h	Store RuntimeSetting in Aec Dumps.	2018-09-10 11:40:28 +00:00
null_aec_dump_factory.cc	Change StartAecDump methods to work with FILE* and FileWrapper	2019-06-11 13:43:36 +00:00
write_to_file_task.cc	Remove webrtc::ProtoString.	2019-02-16 11:11:45 +00:00
write_to_file_task.h	Delete almost all includes of platform_file.h	2019-06-28 07:30:15 +00:00