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webrtc/audio
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henrika 90bace0958 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API 
(this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)

This SetAudioPlayout method lets applications disable audio playout while
still processing incoming audio data and generating statistics on the
received audio.

This may be useful if the application wants to set up media flows as
soon as possible, but isn't ready to play audio yet. Currently, native
applications don't have any API point to control this, unless they
completely implement their own AudioDeviceModule.

The SetAudioRecording works in a similar fashion but for the recorded
audio. One difference is that calling SetAudioRecording(false) does not
keep any audio processing alive.

TBR=solenberg

Bug: webrtc:7313
Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
Reviewed-on: https://webrtc-review.googlesource.com/16180
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20499}
2017-10-31 12:35:42 +00:00
..
test Fix flag name in low_bandwidth_audio_test.py 2017-10-04 17:26:14 +00:00
utility Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
audio_receive_stream.cc New method RtpReceiver::GetLatestTimestamps. 2017-10-03 16:14:29 +00:00
audio_receive_stream.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_receive_stream_unittest.cc Added RTCMediaStreamTrackStats.jitterBufferDelay for audio 2017-10-02 10:47:00 +00:00
audio_send_stream.cc Reland of BWE allocation strategy 2017-10-20 10:16:15 +00:00
audio_send_stream.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_send_stream_tests.cc Remove voe_auto_test and add new tests to cover the missing cases. 2017-09-15 16:56:08 +00:00
audio_send_stream_unittest.cc Remove voe::Statistics. 2017-09-29 13:00:28 +00:00
audio_state.cc Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API 2017-10-31 12:35:42 +00:00
audio_state.h Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API 2017-10-31 12:35:42 +00:00
audio_state_unittest.cc Remove the VoiceEngineObserver callback interface. 2017-09-26 16:35:01 +00:00
audio_transport_proxy.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_transport_proxy.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
BUILD.gn Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API 2017-10-31 12:35:42 +00:00
conversion.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
null_audio_poller.cc Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API 2017-10-31 12:35:42 +00:00
null_audio_poller.h Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API 2017-10-31 12:35:42 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
scoped_voe_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
time_interval.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
time_interval.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
time_interval_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00