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Unified Diff: webrtc/modules/pacing/include/packet_router.h

Issue 1247293002: Add support for transport wide sequence numbers (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase, again Created 5 years, 4 months ago
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Index: webrtc/modules/pacing/include/packet_router.h
diff --git a/webrtc/modules/pacing/include/packet_router.h b/webrtc/modules/pacing/include/packet_router.h
index c1b332a6bfc6b1ad6e7af0bea989b0803a648c89..e7d630eb2feecaa1943fdbffdf785110c359428b 100644
--- a/webrtc/modules/pacing/include/packet_router.h
+++ b/webrtc/modules/pacing/include/packet_router.h
@@ -14,6 +14,7 @@
#include <list>
#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
@@ -21,10 +22,7 @@
namespace webrtc {
-class CriticalSectionWrapper;
-class RTPFragmentationHeader;
class RtpRtcp;
-struct RTPVideoHeader;
// PacketRouter routes outgoing data to the correct sending RTP module, based
// on the simulcast layer in RTPVideoHeader.
@@ -44,14 +42,15 @@ class PacketRouter : public PacedSender::Callback {
size_t TimeToSendPadding(size_t bytes) override;
- private:
- // TODO(holmer): When the new video API has launched, remove crit_ and
- // assume rtp_modules_ will never change during a call. We should then also
- // switch rtp_modules_ to a map from ssrc to rtp module.
- rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+ void SetTransportWideSequenceNumber(uint16_t sequence_number);
+ uint16_t AllocateSequenceNumber();
+ private:
+ rtc::CriticalSection modules_lock_;
// Map from ssrc to sending rtp module.
- std::list<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_.get());
+ std::list<RtpRtcp*> rtp_modules_ GUARDED_BY(modules_lock_);
+
+ volatile int transport_seq_;
DISALLOW_COPY_AND_ASSIGN(PacketRouter);
};
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