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webrtc/audio
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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.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_receive_stream_unittest.cc Delete class ChannelReceiveProxy. 2018-11-16 09:56:54 +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_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_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_state_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Moving src/webrtc into src/. 2017-09-15 04:25:06 +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
remix_resample_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +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
transport_feedback_packet_loss_tracker_unittest.cc Cleanup modules_common_types 2018-09-18 08:08:33 +00:00