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webrtc_m130/audio
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Alejandro Luebs c77108446e Remove RTC_NOTREACHED from audio_send_stream when ANA didn't work 
Bug: None
Change-Id: Id3181827470aba8e486073380911db5873c7dd0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169800
Commit-Queue: Alejandro Luebs <aluebs@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30730}
2020-03-09 17:44:42 +00:00
..
test
Use newer version of TimeDelta and TimeStamp factories in webrtc
2020-02-10 12:21:17 +00:00
utility
Reformat GN files.
2020-01-21 12:13:11 +00:00
audio_level.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
audio_level.h
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
2019-07-04 08:13:45 +00:00
audio_receive_stream_unittest.cc
Delete media transport integration.
2019-11-26 19:19:36 +00:00
audio_receive_stream.cc
Add periodic logging of sync delays.
2020-02-11 09:43:49 +00:00
audio_receive_stream.h
Add periodic logging of sync delays.
2020-02-11 09:43:49 +00:00
audio_send_stream_tests.cc
Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing.
2020-01-07 13:02:52 +00:00
audio_send_stream_unittest.cc
Make sure that the audio stream is allocated with the correct overhead.
2020-03-06 14:49:37 +00:00
audio_send_stream.cc
Remove RTC_NOTREACHED from audio_send_stream when ANA didn't work
2020-03-09 17:44:42 +00:00
audio_send_stream.h
Ready to support of absolute capture timestamp header extension.
2020-03-06 13:16:29 +00:00
audio_state_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
audio_state.cc
Revert "Inlines NullAudioPoller functionality into AudioState class."
2020-01-30 18:14:11 +00:00
audio_state.h
Revert "Inlines NullAudioPoller functionality into AudioState class."
2020-01-30 18:14:11 +00:00
audio_transport_impl.cc
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 18:31:30 +00:00
audio_transport_impl.h
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 18:31:30 +00:00
BUILD.gn
Refactoring mock_transport to be used separately
2020-02-21 17:02:52 +00:00
channel_receive.cc
Only update the current time of a played out frame if a new frame is played out.
2020-02-25 12:48:32 +00:00
channel_receive.h
Delete media transport integration.
2019-11-26 19:19:36 +00:00
channel_send.cc
Reland "Send absolute capture time through audio coding module."
2020-01-27 13:18:27 +00:00
channel_send.h
Cleanup: Using RtpRtcp directly from AudioSendStream
2020-01-15 19:01:50 +00:00
conversion.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
DEPS
Move remaining traces of VoiceEngine
2018-01-17 13:27:47 +00:00
mock_voe_channel_proxy.h
Cleanup: Using RtpRtcp directly from AudioSendStream
2020-01-15 19:01:50 +00:00
null_audio_poller.cc
Revert "Inlines NullAudioPoller functionality into AudioState class."
2020-01-30 18:14:11 +00:00
null_audio_poller.h
Revert "Inlines NullAudioPoller functionality into AudioState class."
2020-01-30 18:14:11 +00:00
OWNERS
Remove wildcard ownership for build files.
2020-02-19 14:05:46 +00:00
remix_resample_unittest.cc
Add RTC_ prefix to non-standard format specifier macro "PRIdNS"
2019-08-07 13:36:05 +00:00
remix_resample.cc
Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.""
2019-07-24 16:47:13 +00:00
remix_resample.h
Remove dependencies on modules:module_api from AudioProcessing.
2018-04-12 22:05:27 +00:00