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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

42
pc/rtptransport_unittest.cc
View File

@ -197,37 +197,49 @@ TEST(RtpTransportTest, SetRtcpTransportWithNetworkRouteChanged) {
EXPECT_FALSE(observer.network_route());
}
class SignalCounter : public sigslot::has_slots<> {
public:
explicit SignalCounter(RtpTransport* transport) {
transport->SignalReadyToSend.connect(this, &SignalCounter::OnReadyToSend);
}
int count() const { return count_; }
void OnReadyToSend(bool ready) { ++count_; }
private:
int count_ = 0;
};
TEST(RtpTransportTest, ChangingReadyToSendStateOnlySignalsWhenChanged) {
RtpTransport transport(kMuxEnabled);
TransportObserver observer(&transport);
SignalCounter observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetWritable(true);
// State changes, so we should signal.
transport.SetRtpPacketTransport(&fake_rtp);
EXPECT_EQ(observer.ready_to_send_signal_count(), 1);
EXPECT_EQ(observer.count(), 1);
// State does not change, so we should not signal.
transport.SetRtpPacketTransport(&fake_rtp);
EXPECT_EQ(observer.ready_to_send_signal_count(), 1);
EXPECT_EQ(observer.count(), 1);
// State does not change, so we should not signal.
transport.SetRtcpMuxEnabled(true);
EXPECT_EQ(observer.ready_to_send_signal_count(), 1);
EXPECT_EQ(observer.count(), 1);
// State changes, so we should signal.
transport.SetRtcpMuxEnabled(false);
EXPECT_EQ(observer.ready_to_send_signal_count(), 2);
EXPECT_EQ(observer.count(), 2);
}
// Test that SignalPacketReceived fires with rtcp=true when a RTCP packet is
// received.
TEST(RtpTransportTest, SignalDemuxedRtcp) {
RtpTransport transport(kMuxDisabled);
SignalPacketReceivedCounter observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetDestination(&fake_rtp, true);
transport.SetRtpPacketTransport(&fake_rtp);
TransportObserver observer(&transport);
// An rtcp packet.
const char data[] = {0, 73, 0, 0};
@ -247,15 +259,11 @@ static const int kRtpLen = 12;
// handled payload type is received.
TEST(RtpTransportTest, SignalHandledRtpPayloadType) {
RtpTransport transport(kMuxDisabled);
SignalPacketReceivedCounter observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetDestination(&fake_rtp, true);
// Disable the encryption to allow raw RTP data.
transport.SetRtpPacketTransport(&fake_rtp);
TransportObserver observer(&transport);
RtpDemuxerCriteria demuxer_criteria;
// Add a handled payload type.
demuxer_criteria.payload_types = {0x11};
transport.RegisterRtpDemuxerSink(demuxer_criteria, &observer);
transport.AddHandledPayloadType(0x11);
// An rtp packet.
const rtc::PacketOptions options;
@ -264,22 +272,16 @@ TEST(RtpTransportTest, SignalHandledRtpPayloadType) {
fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags);
EXPECT_EQ(1, observer.rtp_count());
EXPECT_EQ(0, observer.rtcp_count());
// Remove the sink before destroying the transport.
transport.UnregisterRtpDemuxerSink(&observer);
}
// Test that SignalPacketReceived does not fire when a RTP packet with an
// unhandled payload type is received.
TEST(RtpTransportTest, DontSignalUnhandledRtpPayloadType) {
RtpTransport transport(kMuxDisabled);
SignalPacketReceivedCounter observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetDestination(&fake_rtp, true);
transport.SetRtpPacketTransport(&fake_rtp);
TransportObserver observer(&transport);
RtpDemuxerCriteria demuxer_criteria;
// Add an unhandled payload type.
demuxer_criteria.payload_types = {0x12};
transport.RegisterRtpDemuxerSink(demuxer_criteria, &observer);
const rtc::PacketOptions options;
const int flags = 0;
@ -287,8 +289,6 @@ TEST(RtpTransportTest, DontSignalUnhandledRtpPayloadType) {
fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags);
EXPECT_EQ(0, observer.rtp_count());
EXPECT_EQ(0, observer.rtcp_count());
// Remove the sink before destroying the transport.
transport.UnregisterRtpDemuxerSink(&observer);
}
} // namespace webrtc