Files
platform-external-webrtc/api/transport/rtp/BUILD.gn
Ruslan Burakov d51cc7bd71 Add absolute capture time property to rtp sources.
This part of the effort to implement A/V sync metric.

Bug: webrtc:10739
Change-Id: I4adba1b99b37b31868168e37d9aa8e03f8ea6d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159886
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ruslan Burakov <kuddai@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29849}
2019-11-20 18:50:45 +00:00

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# Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../../webrtc.gni")
rtc_source_set("rtp_source") {
visibility = [ "*" ]
sources = [
"rtp_source.h",
]
deps = [
"../../../api:rtp_headers",
"../../../rtc_base:checks",
"//third_party/abseil-cpp/absl/types:optional",
]
}