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platform-external-webrtc/audio
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Alex Narest bbbe4e1a15 Better handle target audio bitrate allocation. 
Do not update target audio bitrate if WebRTC-Audio-SendSideBwe-For-Video is enabled but other side does not support TWCC

Bug: webrtc:8243
Change-Id: I6c3c4f223dc5168d726996324717d7ba9ec96e6c
Reviewed-on: https://webrtc-review.googlesource.com/88440
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23963}
2018-07-13 09:06:06 +00:00
..
test
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
audio_send_stream.cc
Better handle target audio bitrate allocation.
2018-07-13 09:06:06 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
channel.h
Remove StreamStatistician::IsPacketInOrder
2018-06-28 08:44:40 +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
…
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