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Add support for transport wide sequence numbers

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Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.

BUG=webrtc:4311

Review URL: https://codereview.webrtc.org/1247293002

Cr-Commit-Position: refs/heads/master@{#9670}
This commit is contained in:
sprang
2015-08-03 04:38:41 -07:00
committed by Commit bot
parent d67a219bec
commit 867fb5224e
39 changed files with 818 additions and 334 deletions
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17
webrtc/modules/pacing/include/packet_router.h
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@ -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);
};