Alex Narest bcf91808a2 Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream. ...

This will allow experimenting with audio bitrate allocation in video calls without increasing transport overhead.

Bug: webrtc:8243
Change-Id: If961780921d53bdce95b68c26641df6875509c1f
Reviewed-on: https://webrtc-review.googlesource.com/84501
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23755}

2018-06-27 10:33:40 +00:00

..

test

Reformat the WebRTC code base

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

utility

Reformat the WebRTC code base

2018-06-19 14:00:39 +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

Reformat the WebRTC code base

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

audio_receive_stream.cc

Reformat the WebRTC code base

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

audio_receive_stream.h

Replace rtc::Optional with absl::optional in audio, call and video

2018-06-15 12:09:49 +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

Reformat the WebRTC code base

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

audio_send_stream.cc

Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.

2018-06-27 10:33:40 +00:00

audio_send_stream.h

Reformat the WebRTC code base

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

audio_state_unittest.cc

Reformat the WebRTC code base

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

audio_state.cc

Reformat the WebRTC code base

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

audio_state.h

Reformat the WebRTC code base

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

audio_transport_impl.cc

Reformat the WebRTC code base

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

audio_transport_impl.h

Reformat the WebRTC code base

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

BUILD.gn

Replace rtc::Optional with absl::optional in audio, call and video

2018-06-15 12:09:49 +00:00

channel_proxy.cc

Reformat the WebRTC code base

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

channel_proxy.h

Reformat the WebRTC code base

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

channel.cc

Update packetsLost and jitter stats any time a packet is received.

2018-06-25 23:56:39 +00:00

channel.h

Replace rtc::Optional with absl::optional in audio, call and video

2018-06-15 12:09:49 +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

Reformat the WebRTC code base

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

null_audio_poller.cc

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)

2017-11-01 11:04:26 +00:00

null_audio_poller.h

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)

2017-11-01 11:04:26 +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

Reformat the WebRTC code base

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

remix_resample.h

Remove dependencies on modules:module_api from AudioProcessing.

2018-04-12 22:05:27 +00:00

time_interval_unittest.cc

Replacing rtc::TimeDelta with webrtc::TimeDelta.

2018-05-08 13:22:53 +00:00

time_interval.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

time_interval.h

Replace rtc::Optional with absl::optional in audio, call and video

2018-06-15 12:09:49 +00:00

transport_feedback_packet_loss_tracker_unittest.cc

Reformat the WebRTC code base

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

transport_feedback_packet_loss_tracker.cc

Replace rtc::Optional with absl::optional in audio, call and video

2018-06-15 12:09:49 +00:00

transport_feedback_packet_loss_tracker.h

Replace rtc::Optional with absl::optional in audio, call and video

2018-06-15 12:09:49 +00:00