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Revert "Replace BundleFilter with RtpDemuxer in RtpTransport."

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This reverts commit ea8b62a3e74fe91cd6bf66304839cd5677880a4e.

Reason for revert: Broke chromium tests.
Original change's description:
> Replace BundleFilter with RtpDemuxer in RtpTransport.
> 
> BundleFilter is replaced by RtpDemuxer in RtpTransport for payload
> type-based demuxing. RtpTransport will support MID-based demuxing later.
> 
> Each BaseChannel has its own RTP demuxing criteria and when connecting
> to the RtpTransport, BaseChannel will register itself as a demuxer sink.
> 
> The inheritance model is changed. New inheritance chain:
> DtlsSrtpTransport->SrtpTransport->RtpTranpsort
> 
> NOTE:
> When RTCP packets are received, Call::DeliverRtcp will be called for
> multiple times (webrtc:9035) which is an existing issue. With this CL,
> it will become more of a problem and should be fixed.
> 
> Bug: webrtc:8587
> Change-Id: I1d8a00443bd4bcbacc56e5e19b7294205cdc38f0
> Reviewed-on: https://webrtc-review.googlesource.com/61360
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22613}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org

Change-Id: If245da9d1ce970ac8dab7f45015e9b268a5dbcbd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8587
Reviewed-on: https://webrtc-review.googlesource.com/64860
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22614}
This commit is contained in:
Zhi Huang
2018-03-27 00:09:01 +00:00
committed by Commit Bot
parent ea8b62a3e7
commit 97d5e5b32c
28 changed files with 713 additions and 591 deletions
Download Patch File Download Diff File Expand all files Collapse all files

30
pc/channel.h
View File

@ -23,7 +23,6 @@
#include "api/rtpreceiverinterface.h"
#include "api/videosinkinterface.h"
#include "api/videosourceinterface.h"
#include "call/rtp_packet_sink_interface.h"
#include "media/base/mediachannel.h"
#include "media/base/mediaengine.h"
#include "media/base/streamparams.h"
@ -70,10 +69,9 @@ class MediaContentDescription;
// vtable, and the media channel's thread using BaseChannel as the
// NetworkInterface.
class BaseChannel : public rtc::MessageHandler,
public sigslot::has_slots<>,
public MediaChannel::NetworkInterface,
public webrtc::RtpPacketSinkInterface {
class BaseChannel
: public rtc::MessageHandler, public sigslot::has_slots<>,
public MediaChannel::NetworkInterface {
public:
// If |srtp_required| is true, the channel will not send or receive any
// RTP/RTCP packets without using SRTP (either using SDES or DTLS-SRTP).
@ -195,8 +193,10 @@ class BaseChannel : public rtc::MessageHandler,
virtual cricket::MediaType media_type() = 0;
// RtpPacketSinkInterface overrides.
void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;
// Public for testing.
// TODO(zstein): Remove this once channels register themselves with
// an RtpTransport in a more explicit way.
bool HandlesPayloadType(int payload_type) const;
// Used by the RTCStatsCollector tests to set the transport name without
// creating RtpTransports.
@ -264,10 +264,12 @@ class BaseChannel : public rtc::MessageHandler,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options);
void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
void HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
// TODO(zstein): packet can be const once the RtpTransport handles protection.
void OnPacketReceived(bool rtcp,
const rtc::CopyOnWriteBuffer& packet,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
void ProcessPacket(bool rtcp,
const rtc::CopyOnWriteBuffer& packet,
@ -358,11 +360,6 @@ class BaseChannel : public rtc::MessageHandler,
void AddHandledPayloadType(int payload_type);
void UpdateRtpHeaderExtensionMap(
const RtpHeaderExtensions& header_extensions);
bool RegisterRtpDemuxerSink();
private:
void ConnectToRtpTransport();
void DisconnectFromRtpTransport();
@ -442,9 +439,6 @@ class BaseChannel : public rtc::MessageHandler,
// The cached encrypted header extension IDs.
rtc::Optional<std::vector<int>> cached_send_extension_ids_;
rtc::Optional<std::vector<int>> cached_recv_extension_ids_;
bool encryption_disabled_ = false;
webrtc::RtpDemuxerCriteria demuxer_criteria_;
};
// VoiceChannel is a specialization that adds support for early media, DTMF,