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

Issue 1979443004: Add H264 bitstream rewriting to limit frame reordering marker in header (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed comments Created 4 years, 7 months ago
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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..11fbea8655f566bffacdf666a9755ea7ce2146e0 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -32,8 +32,6 @@ source_set("rtp_rtcp") {
"source/forward_error_correction.h",
"source/forward_error_correction_internal.cc",
"source/forward_error_correction_internal.h",
- "source/h264_sps_parser.cc",
- "source/h264_sps_parser.h",
"source/mock/mock_rtp_payload_strategy.h",
"source/packet_loss_stats.cc",
"source/packet_loss_stats.h",
@@ -167,6 +165,7 @@ source_set("rtp_rtcp") {
deps = [
"../..:webrtc_common",
+ "../../common_video",
"../../system_wrappers",
"../remote_bitrate_estimator",
]
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