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webrtc_m130/call/test
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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
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fake_network_pipe_unittest.cc
Moving demux from FakeNetworkPipe to DirectTransport.
2018-04-25 10:13:03 +00:00
mock_audio_send_stream.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mock_bitrate_allocator.h
Adds mock bitrate allocator.
2018-04-19 14:41:42 +00:00
mock_rtp_packet_sink_interface.h
…
mock_rtp_transport_controller_send.h
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