commit b8c775aeaf41d796a85402116e7353efb3fb2202
author Tim Na <natim@webrtc.org> Fri Jan 10 10:33:05 2020 -0800
committer Commit Bot <commit-bot@chromium.org> Mon Jan 13 18:31:30 2020 +0000
tree c86c3ad8552fbf743bda597b2e72fe66d40bcda3
parent 8234b92ba3aa454c7ed63a7d168f4cab8d3f439e

Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api

Bug: webrtc:11251
Change-Id: Id3b6ff1814931d8250c4aaac59e494521fbe93ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164560
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30238}

audio/BUILD.gn

diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index a6d7ed4..6f815f3 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -54,6 +54,7 @@
     "../api/rtc_event_log",
     "../api/task_queue",
     "../api/transport/rtp:rtp_source",
+    "../call:audio_sender_interface",
     "../call:bitrate_allocator",
     "../call:call_interfaces",
     "../call:rtp_interfaces",