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

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/source/rtp_format.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format.h b/webrtc/modules/rtp_rtcp/source/rtp_format.h
index 351949924810e5a927cf9e168fcbba6895d0ca30..8cad6a916d0531115049067dbb1d20eb2c57bd04 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format.h
@@ -50,6 +50,10 @@ class RtpPacketizer {
virtual std::string ToString() = 0;
};
+// TODO(sprang): Update the depacketizer to return a std::unqie_ptr with a copy
+// of the parsed payload, rather than just a pointer into the incoming buffer.
+// This way we can move some parsing out from the jitter buffer into here, and
+// the jitter buffer can just store that pointer rather than doing a copy there.
class RtpDepacketizer {
public:
struct ParsedPayload {
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