Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1397123003
Cr-Commit-Position: refs/heads/master@{#10307}
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index db0791f..c6f4b6b 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -12,6 +12,8 @@
sources = [
"audio_receive_stream.cc",
"audio_receive_stream.h",
+ "audio_send_stream.cc",
+ "audio_send_stream.h",
]
configs += [ "..:common_config" ]