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henrika 45c136b579 Adds support for Bluetooth headsets to the iOS audio layer. 
This patch also also ensures that audio is restored after an incoming
GSM call.

BUG=webrtc:5058, webrtc:5012
TEST=Manual tests using modified AppRTCDemo and three different BT headsets

Review URL: https://codereview.webrtc.org/1401963002

Cr-Commit-Position: refs/heads/master@{#10354}
2015-10-21 11:12:01 +00:00
..
audio
Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
2015-10-20 13:41:06 +00:00
base
Introduce rtc::Maybe<T>, which either contains a T or not.
2015-10-21 10:44:17 +00:00
build
Suppress libyuv::TestCpuFlag races.
2015-10-20 11:37:41 +00:00
call
Remove system_wrappers/interface/trace_event.h
2015-10-21 06:00:57 +00:00
common_audio
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common_video
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examples
Create network change notifier and pass the event to NetworkManager
2015-10-19 16:39:38 +00:00
libjingle
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modules
Adds support for Bluetooth headsets to the iOS audio layer.
2015-10-21 11:12:01 +00:00
p2p
Revert of Add experiment on weak ping delay during call set up time (patchset #4 id:60001 of https://codereview.webrtc.org/1411883002/ )
2015-10-21 08:07:32 +00:00
sound
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system_wrappers
Add stats for average QP per frame for VP8 (for received video streams):
2015-10-21 06:55:32 +00:00
test
Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
2015-10-20 13:41:06 +00:00
tools
…
video
Add stats for average QP per frame for VP8 (for received video streams):
2015-10-21 06:55:32 +00:00
video_engine
Remove system_wrappers/interface/trace_event.h
2015-10-21 06:00:57 +00:00
voice_engine
Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
2015-10-20 13:41:06 +00:00
.gitignore
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audio_receive_stream.h
Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
2015-10-20 13:41:06 +00:00
audio_send_stream.h
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
BUILD.gn
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call.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
codereview.settings
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common_types.cc
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common_types.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
common.gyp
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common.h
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config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
…
engine_configurations.h
…
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
libjingle_examples.gyp
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
LICENSE
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LICENSE_THIRD_PARTY
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OWNERS
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PATENTS
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PRESUBMIT.py
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README.chromium
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rtc_unittests.isolate
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stream.h
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supplement.gypi
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transport.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
typedefs.h
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video_decoder.h
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video_encoder.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
video_engine_tests.isolate
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video_frame.h
Add stats for average QP per frame for VP8 (for received video streams):
2015-10-21 06:55:32 +00:00
video_receive_stream.h
…
video_renderer.h
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video_send_stream.h
…
webrtc_examples.gyp
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webrtc_perf_tests.isolate
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webrtc_tests.gypi
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
webrtc.gyp
…

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.