3 Commits

Author	SHA1	Message	Date
henrika	86d907cffd	Refactor the AudioDevice for iOS and improve the performance and stability ... This CL contains major modifications of the audio output parts for WebRTC on iOS: - general code cleanup - improves thread handling (added thread checks, remove critical section, atomic ops etc.) - reduces loopback latency of iPhone 6 from ~90ms to ~60ms ;-) - improves selection of audio parameters on iOS - reduces complexity by removing complex and redundant delay estimates - now instead uses fixed delay estimates if for some reason the SW EAC must be used - adds AudioFineBuffer to compensate for differences in native output buffer size and the 10ms size used by WebRTC. Same class as is used today on Android and we have unit tests for this class (the old code was buggy and we have several issue reports of crashes related to it) Similar improvements will be done for the recording sid as well in a separate CL. I will also add support for 48kHz in an upcoming CL since that will improve Opus performance. BUG=webrtc:4796,webrtc:4817,webrtc:4954, webrtc:4212 TEST=AppRTC demo and iOS modules_unittests using --gtest_filter=AudioDevice* R=pbos@webrtc.org, tkchin@webrtc.org Review URL: https://codereview.webrtc.org/1254883002 . Cr-Commit-Position: refs/heads/master@{#9875}	2015-09-07 14:10:10 +00:00
Peter Kasting	dce40cf804	Update a ton of audio code to use size_t more correctly and in general reduce ... use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}	2015-08-24 21:52:45 +00:00
henrike@webrtc.org	82f014aa0b	OpenSL (not default): Enables low latency audio on Android. ... BUG=1669 R=andrew@webrtc.org, fischman@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2032004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-09-10 18:24:07 +00:00