Applications can choose to decide when to give up control of the
AVAudioSession to WebRTC. Otherwise, behavior should be
unchanged.
Adds a toggle to AppRTCDemo so developers can see the different
paths.
BUG=
R=haysc@webrtc.org
Review URL: https://codereview.webrtc.org/1822543002 .
Cr-Commit-Position: refs/heads/master@{#12080}
Part 3 of refactor. Also:
- better weak pointer delegate storage + tests
- we now ignore route changes when we're interrupted
- fixed bug where preferred sample rate wasn't set if audio session
wasn't active
BUG=
Review URL: https://codereview.webrtc.org/1796983004
Cr-Commit-Position: refs/heads/master@{#12007}
In addition:
- Introduces RTCAudioSessionTest
- iOS/Mac gtests now have an autoreleasepool
- Moves ScopedAutoreleasePool to rtc_base_approved
- Introduces route change button in AppRTCDemo
BUG=webrtc:5649
Review URL: https://codereview.webrtc.org/1782363002
Cr-Commit-Position: refs/heads/master@{#11971}
Adds a loopback button that will connect to itself by simulating another client connection to the web socket server.
Adds an audio only mode switch.
BUG=
Review URL: https://codereview.webrtc.org/1334003002
Cr-Commit-Position: refs/heads/master@{#10153}
Significant changes:
- move the libjingle_examples.gyp file into webrtc directory.
- rename talk/examples/android to webrtc/examples/androidapp to avoid name conflicts.
- update paths in talk/libjingle_tests.gyp to point to webrtc directory for Objective-C test.
BUG=
R=pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1235563006 .
Cr-Commit-Position: refs/heads/master@{#9681}
