Jakob Ivarsson 53eae87bf8 Add PeerConnection option to enable RTX handling in the audio jitter buffer. ... Bug: webrtc:10178 Change-Id: I70abce0c7b74124d2b1978d9a5eb8216b6233d1a Reviewed-on: https://webrtc-review.googlesource.com/c/116784 Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26203}		2019-01-10 16:28:43 +00:00
..
test	Include absl/memory/memory.h if absl::make_unique is used	2019-01-08 20:08:32 +00:00
utility	Delete unneeded includes of common_types.h and gn deps on webrtc_common.	2018-11-20 16:28:39 +00:00
audio_level.cc	Reformat the WebRTC code base	2018-06-19 14:00:39 +00:00
audio_level.h	Delete AudioMonitor and related code.	2018-01-30 09:48:29 +00:00
audio_receive_stream_unittest.cc	Remove CodecInst pt.2	2018-12-17 10:33:55 +00:00
audio_receive_stream.cc	Add PeerConnection option to enable RTX handling in the audio jitter buffer.	2019-01-10 16:28:43 +00:00
audio_receive_stream.h	Delete class ChannelSendProxy	2018-11-19 10:17:13 +00:00
audio_send_stream_tests.cc	Reformat the WebRTC code base	2018-06-19 14:00:39 +00:00
audio_send_stream_unittest.cc	Simulcast work to enable RID mux.	2018-12-21 20:59:23 +00:00
audio_send_stream.cc	Simulcast work to enable RID mux.	2018-12-21 20:59:23 +00:00
audio_send_stream.h	Simulcast work to enable RID mux.	2018-12-21 20:59:23 +00:00
audio_state_unittest.cc	Update refcounting of AudioState to use rtc::RefCountedObject	2019-01-09 13:49:19 +00:00
audio_state.cc	Update refcounting of AudioState to use rtc::RefCountedObject	2019-01-09 13:49:19 +00:00
audio_state.h	Update refcounting of AudioState to use rtc::RefCountedObject	2019-01-09 13:49:19 +00:00
audio_transport_impl.cc	[Cleanup] Add missing #include. Remove useless ones.	2018-10-23 11:32:56 +00:00
audio_transport_impl.h	Reformat the WebRTC code base	2018-06-19 14:00:39 +00:00
BUILD.gn	Include absl/memory/memory.h if absl::make_unique is used	2019-01-08 20:08:32 +00:00
channel_receive.cc	Add PeerConnection option to enable RTX handling in the audio jitter buffer.	2019-01-10 16:28:43 +00:00
channel_receive.h	Add PeerConnection option to enable RTX handling in the audio jitter buffer.	2019-01-10 16:28:43 +00:00
channel_send.cc	Simulcast work to enable RID mux.	2018-12-21 20:59:23 +00:00
channel_send.h	Simulcast work to enable RID mux.	2018-12-21 20:59:23 +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	Simulcast work to enable RID mux.	2018-12-21 20:59:23 +00:00
null_audio_poller.cc	[Cleanup] Add missing #include. Remove useless ones.	2018-10-23 11:32:56 +00:00
null_audio_poller.h	[Cleanup] Add missing #include. Remove useless ones.	2018-10-23 11:32:56 +00:00
OWNERS	Moving src/webrtc into src/.	2017-09-15 04:25:06 +00:00
remix_resample_unittest.cc	Reformat the WebRTC code base	2018-06-19 14:00:39 +00:00
remix_resample.cc	[Cleanup] Add missing #include. Remove useless ones.	2018-10-23 11:32:56 +00:00
remix_resample.h	Remove dependencies on modules:module_api from AudioProcessing.	2018-04-12 22:05:27 +00:00
transport_feedback_packet_loss_tracker_unittest.cc	Cleanup modules_common_types	2018-09-18 08:08:33 +00:00
transport_feedback_packet_loss_tracker.cc	[Cleanup] Add missing #include. Remove useless ones.	2018-10-23 11:32:56 +00:00
transport_feedback_packet_loss_tracker.h	Remove clang:find_bad_constructs suppression from call:call.	2018-08-29 11:57:00 +00:00