webrtc/call/BUILD.gn - Issue 2867943003: New class RtpDemuxer and RtpPacketSinkInterface, use in Call.

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Unified Diff: webrtc/call/BUILD.gn

Issue 2867943003: New class RtpDemuxer and RtpPacketSinkInterface, use in Call. (Closed)

Patch Set: Fix use-after-free crash in av sync. 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",

]