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Unified Diff: webrtc/modules/rtp_rtcp/source/rtcp_packet.h

Issue 1338203003: Wire up send-side bandwidth estimation. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 3 months ago
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Index: webrtc/modules/rtp_rtcp/source/rtcp_packet.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
index c5432de9d6030544f4d6ea91405950d1712cbce6..a30ff8a17ba10e3dca3786908afefda23b105f7c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
@@ -91,6 +91,9 @@ class RtcpPacket {
size_t max_length,
PacketReadyCallback* callback) const;
+ // Size of this packet in bytes (including headers, excluding nested packets).
+ virtual size_t BlockLength() const = 0;
+
protected:
RtcpPacket() {}
@@ -109,7 +112,6 @@ class RtcpPacket {
size_t* index,
RtcpPacket::PacketReadyCallback* callback) const;
- virtual size_t BlockLength() const = 0;
size_t HeaderLength() const;
static const size_t kHeaderLength = 4;

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