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3 commits

Author SHA1 Message Date
Markus Handell
6287280d64 Migrate audio/ to use webrtc::Mutex 
Bug: webrtc:11567
Change-Id: Ic6a753f09aafb508690f4b8dadd4c99433fcfeb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176741
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31635}
 2020-07-06 14:21:38 +00:00
Philipp Hancke
b9d468573a insertable streams: include rtp_timestamp offset for audio 
includes the (random) rtp start offset in the timestamp passed to the frame transformer callback

Bug: chromium:1069278
Change-Id: I7d10130404d93df7cee3b8f87a0b780801a51415
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173329
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31067}
 2020-04-14 14:54:07 +00:00
Marina Ciocea
65674d83e1 Transform encoded frames in ChannelSend. 
This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: I75444283ddb7f8db742687b497bf532c6dda47be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171871
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30952}
 2020-03-31 21:59:26 +00:00