Files
platform-external-webrtc/webrtc/call/rtp_demuxer.h
eladalon a52722fac4 Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
Reason for revert:
About to fix problem and reland.

Original issue's description:
> Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ )
>
> Reason for revert:
> Breaks Chromium FYI bots.
>
> The problem is in the BUILD.gn file.
>
> Sample failure:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829
>
> Sample logs:
> use_goma = true
> """ to /b/c/b/Linux_Builder/src/out/Release/args.gn.
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
>   -> returned 1
> ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file.
>     "//webrtc/base:rtc_base_approved",
>     ^--------------------------------
>
> Original issue's description:
> > Create RtcpDemuxer. Capabilities:
> > 1. Demux RTCP messages according to the sender-SSRC.
> > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP).
> > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks").
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2943693003
> > Cr-Commit-Position: refs/heads/master@{#18763}
> > Committed: cb83bdf01f
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2957763002
> Cr-Commit-Position: refs/heads/master@{#18764}
> Committed: 0e7e7869e7

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2960623002
Cr-Commit-Position: refs/heads/master@{#18768}
2017-06-26 18:23:54 +00:00

99 lines
3.8 KiB
C++

/*
* Copyright (c) 2017 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.
*/
#ifndef WEBRTC_CALL_RTP_DEMUXER_H_
#define WEBRTC_CALL_RTP_DEMUXER_H_
#include <map>
#include <set>
#include <string>
#include <vector>
namespace webrtc {
class RsidResolutionObserver;
class RtpPacketReceived;
class RtpPacketSinkInterface;
// This class represents the RTP demuxing, for a single RTP session (i.e., one
// ssrc space, see RFC 7656). It isn't thread aware, leaving responsibility of
// multithreading issues to the user of this class.
// TODO(nisse): Should be extended to also do MID-based demux and payload-type
// demux.
class RtpDemuxer {
public:
RtpDemuxer();
~RtpDemuxer();
// Registers a sink. The same sink can be registered for multiple ssrcs, and
// the same ssrc can have multiple sinks. Null pointer is not allowed.
void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink);
// Registers a sink's association to an RSID. Null pointer is not allowed.
void AddSink(const std::string& rsid, RtpPacketSinkInterface* sink);
// Removes a sink. Return value reports if anything was actually removed.
// Null pointer is not allowed.
bool RemoveSink(const RtpPacketSinkInterface* sink);
// Returns true if at least one matching sink was found.
bool OnRtpPacket(const RtpPacketReceived& packet);
// Allows other objects to be notified when RSID-SSRC associations are
// resolved by this object.
void RegisterRsidResolutionObserver(RsidResolutionObserver* observer);
// Undo a previous RegisterRsidResolutionObserver().
void DeregisterRsidResolutionObserver(const RsidResolutionObserver* observer);
private:
// Records a sink<->SSRC association. This can happen by explicit
// configuration by AddSink(ssrc...), or by inferred configuration from an
// RSID-based configuration which is resolved to an SSRC upon
// packet reception.
void RecordSsrcToSinkAssociation(uint32_t ssrc, RtpPacketSinkInterface* sink);
// When a new packet arrives, we attempt to resolve extra associations.
void ResolveAssociations(const RtpPacketReceived& packet);
// Find the associations of RSID to SSRCs.
void ResolveRsidToSsrcAssociations(const RtpPacketReceived& packet);
// Notify observers of the resolution of an RSID to an SSRC.
void NotifyObserversOfRsidResolution(const std::string& rsid, uint32_t ssrc);
// This records the association SSRCs to sinks. Other associations, such
// as by RSID, also end up here once the RSID, etc., is resolved to an SSRC.
std::multimap<uint32_t, RtpPacketSinkInterface*> sinks_;
// A sink may be associated with an RSID - RTP Stream ID. This tag has a
// one-to-one association with an SSRC, but that SSRC is not yet known.
// When it becomes known, the association of the sink to the RSID is deleted
// from this container, and moved into |sinks_|.
std::multimap<std::string, RtpPacketSinkInterface*> rsid_sinks_;
// Iterating over |rsid_sinks_| for each incoming and performing multiple
// string comparisons is of non-trivial cost. To avoid this cost, we only
// check RSIDs for the first packet on each incoming SSRC stream.
// (If RSID associations are added later, we check again.)
std::set<uint32_t> processed_ssrcs_;
// Avoid an attack that would create excessive logging.
bool logged_max_processed_ssrcs_exceeded_ = false;
// Observers which will be notified when an RSID association to an SSRC is
// resolved by this object.
std::vector<RsidResolutionObserver*> rsid_resolution_observers_;
};
} // namespace webrtc
#endif // WEBRTC_CALL_RTP_DEMUXER_H_