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webrtc_m130/webrtc/audio
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minyue-webrtc 8de1826b6d Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor" 
BUG=b/63898232, b/64053465

Originally Reviewed-on: https://chromium-review.googlesource.com/584707

Reverted-on: https://chromium-review.googlesource.com/586268
Change-Id: I212b0c1e81a6ccd73b051e6728e601a8641463b8
Reviewed-on: https://chromium-review.googlesource.com/586328
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Michael T <tschumim@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19153}
2017-07-26 14:28:51 +00:00
..
test
Disable flaky NoBandwidthDropAfterDtx test.
2017-07-19 14:52:47 +00:00
utility
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
audio_receive_stream_unittest.cc
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
2017-07-14 19:17:49 +00:00
audio_receive_stream.cc
Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
2017-07-26 09:09:44 +00:00
audio_receive_stream.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
audio_send_stream_unittest.cc
Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
2017-07-26 14:28:51 +00:00
audio_send_stream.cc
Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
2017-07-26 14:28:51 +00:00
audio_send_stream.h
Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
2017-07-26 09:09:44 +00:00
audio_state_unittest.cc
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +00:00
audio_state.cc
External APM usage downstream dependency support cleanup
2017-07-07 11:25:11 +00:00
audio_state.h
External APM usage downstream dependency support cleanup
2017-07-07 11:25:11 +00:00
audio_transport_proxy.cc
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +00:00
audio_transport_proxy.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
BUILD.gn
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
conversion.h
…
DEPS
Injectable audio encoders: WebRtcVoiceEngine and company
2017-04-27 09:08:52 +00:00
OWNERS
…
scoped_voe_interface.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
time_interval_unittest.cc
Adds a histogram metric tracking for how long audio RTP packets are sent
2017-07-19 07:39:19 +00:00
time_interval.cc
Adds a histogram metric tracking for how long audio RTP packets are sent
2017-07-19 07:39:19 +00:00
time_interval.h
Adds a histogram metric tracking for how long audio RTP packets are sent
2017-07-19 07:39:19 +00:00