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Unified Diff: webrtc/call/BUILD.gn

Issue 2867943003: New class RtpDemuxer and RtpPacketSinkInterface, use in Call. (Closed)
Patch Set: Address danil's comments. Created 3 years, 7 months ago
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Index: webrtc/call/BUILD.gn
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 826ef65873f008458e8a9726c9169552e630b8f6..2e5ac31b9d7d1afb764dd962c2bd831dc7ce7305 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -16,6 +16,7 @@ rtc_source_set("call_interfaces") {
"audio_state.h",
"call.h",
"flexfec_receive_stream.h",
+ "rtp_demuxer.h",
"rtp_transport_controller_send_interface.h",
"syncable.cc",
"syncable.h",
@@ -38,6 +39,7 @@ rtc_static_library("call") {
"call.cc",
"flexfec_receive_stream_impl.cc",
"flexfec_receive_stream_impl.h",
+ "rtp_demuxer.cc",
"rtp_transport_controller_send.cc",
"rtp_transport_controller_send.h",
]
« no previous file with comments | « webrtc/audio/audio_receive_stream.h ('k') | webrtc/call/call.cc » ('j') | webrtc/call/call.cc » ('J')

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