Jakob Ivarsson 10403ae87c Add PeerConnection option to configure minimum audio jitter buffer delay. ...

Note that this value will override the minimum delay that is used for audio/video sync.

Bug: webrtc:10053
Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33
Reviewed-on: https://webrtc-review.googlesource.com/c/112121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25805}

2018-11-27 19:49:48 +00:00

..

test

Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric

2018-11-26 09:32:35 +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

Delete class ChannelReceiveProxy.

2018-11-16 09:56:54 +00:00

audio_receive_stream.cc

Add PeerConnection option to configure minimum audio jitter buffer delay.

2018-11-27 19:49:48 +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

Delete ChannelSend::RegisterTransport, replacing by construction argument

2018-11-26 13:08:41 +00:00

audio_send_stream.cc

Delete ChannelSend::RegisterTransport, replacing by construction argument

2018-11-26 13:08:41 +00:00

audio_send_stream.h

Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric

2018-11-26 09:32:35 +00:00

audio_state_unittest.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

audio_state.cc

Remove clang:find_bad_constructs suppression from call:call.

2018-08-29 11:57:00 +00:00

audio_state.h

Remove clang:find_bad_constructs suppression from call:call.

2018-08-29 11:57:00 +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

Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric

2018-11-26 09:32:35 +00:00

channel_receive.cc

Add PeerConnection option to configure minimum audio jitter buffer delay.

2018-11-27 19:49:48 +00:00

channel_receive.h

Add PeerConnection option to configure minimum audio jitter buffer delay.

2018-11-27 19:49:48 +00:00

channel_send.cc

Delete ChannelSend::RegisterTransport, replacing by construction argument

2018-11-26 13:08:41 +00:00

channel_send.h

Delete ChannelSend::RegisterTransport, replacing by construction argument

2018-11-26 13:08:41 +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

Delete ChannelSend::RegisterTransport, replacing by construction argument

2018-11-26 13:08:41 +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

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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