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platform-external-webrtc/audio
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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
…
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