webrtc/modules/rtp_rtcp/BUILD.gn - Issue 2007743003: Add sender controlled playout delay limits

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Unified Diff: webrtc/modules/rtp_rtcp/BUILD.gn

Issue 2007743003: Add sender controlled playout delay limits (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@cleanup_rtp_hdr_extensions

Patch Set: Remove notion of max and current header length Created 4 years, 7 months ago

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« webrtc/config.cc ('K') | « webrtc/modules/modules.gyp ('k') | webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h » ('j') | webrtc/modules/rtp_rtcp/source/playout_delay_oracle_unittest.cc » ('J')
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Index: webrtc/modules/rtp_rtcp/BUILD.gn

diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn

index b826fd1ea921b61668de1f6697975e1cd21e25f1..1e0950c6ad68adb672a2258515651ed293e8b886 100644

--- a/webrtc/modules/rtp_rtcp/BUILD.gn

+++ b/webrtc/modules/rtp_rtcp/BUILD.gn

@@ -37,6 +37,8 @@ source_set("rtp_rtcp") {

"source/mock/mock_rtp_payload_strategy.h",

"source/packet_loss_stats.cc",

"source/packet_loss_stats.h",

+ "source/playout_delay_oracle.cc",

+ "source/playout_delay_oracle.h",

"source/producer_fec.cc",

"source/producer_fec.h",

"source/receive_statistics_impl.cc",