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Unified Diff: webrtc/call/BUILD.gn

Issue 2867943003: New class RtpDemuxer and RtpPacketSinkInterface, use in Call. (Closed)
Patch Set: Fix Flexfec recursion issue in a different way. Created 3 years, 7 months ago
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« no previous file with comments | « webrtc/audio/audio_receive_stream.h ('k') | webrtc/call/call.cc » ('j') | webrtc/modules/rtp_rtcp/source/flexfec_receiver.cc » ('J')
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Index: webrtc/call/BUILD.gn
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 1f29e618221a4cdae82fa46280c69bfc9b56ea2d..0d87b2f9f764875dc791c6661bec7feeeddd0c3f 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/modules/rtp_rtcp/source/flexfec_receiver.cc » ('J')

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