webrtc_m130/pc/rtptransport_unittest.cc

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/*
* Copyright 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.
*/
#include <string>
#include <utility>
#include "p2p/base/fakepackettransport.h"
#include "pc/rtptransport.h"
#include "pc/rtptransporttestutil.h"
#include "rtc_base/gunit.h"
namespace webrtc {
constexpr bool kMuxDisabled = false;
constexpr bool kMuxEnabled = true;
constexpr uint16_t kLocalNetId = 1;
constexpr uint16_t kRemoteNetId = 2;
constexpr int kLastPacketId = 100;
constexpr int kTransportOverheadPerPacket = 28; // Ipv4(20) + UDP(8).
TEST(RtpTransportTest, SetRtcpParametersCantDisableRtcpMux) {
RtpTransport transport(kMuxDisabled);
RtpTransportParameters params;
transport.SetParameters(params);
params.rtcp.mux = false;
EXPECT_FALSE(transport.SetParameters(params).ok());
}
TEST(RtpTransportTest, SetRtcpParametersEmptyCnameUsesExisting) {
static const char kName[] = "name";
RtpTransport transport(kMuxDisabled);
RtpTransportParameters params_with_name;
params_with_name.rtcp.cname = kName;
transport.SetParameters(params_with_name);
EXPECT_EQ(transport.GetParameters().rtcp.cname, kName);
RtpTransportParameters params_without_name;
transport.SetParameters(params_without_name);
EXPECT_EQ(transport.GetParameters().rtcp.cname, kName);
}
TEST(RtpTransportTest, SetRtpTransportKeepAliveNotSupported) {
// Tests that we warn users that keep-alive isn't supported yet.
// TODO(sprang): Wire up keep-alive and remove this test.
RtpTransport transport(kMuxDisabled);
RtpTransportParameters params;
params.keepalive.timeout_interval_ms = 1;
auto result = transport.SetParameters(params);
EXPECT_FALSE(result.ok());
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
}
class SignalObserver : public sigslot::has_slots<> {
public:
explicit SignalObserver(RtpTransport* transport) {
transport_ = transport;
transport->SignalReadyToSend.connect(this, &SignalObserver::OnReadyToSend);
transport->SignalNetworkRouteChanged.connect(
this, &SignalObserver::OnNetworkRouteChanged);
if (transport->rtp_packet_transport()) {
transport->rtp_packet_transport()->SignalSentPacket.connect(
this, &SignalObserver::OnSentPacket);
}
if (transport->rtcp_packet_transport()) {
transport->rtcp_packet_transport()->SignalSentPacket.connect(
this, &SignalObserver::OnSentPacket);
}
}
bool ready() const { return ready_; }
void OnReadyToSend(bool ready) { ready_ = ready; }
rtc::Optional<rtc::NetworkRoute> network_route() { return network_route_; }
void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route) {
network_route_ = std::move(network_route);
}
void OnSentPacket(rtc::PacketTransportInternal* packet_transport,
const rtc::SentPacket& sent_packet) {
if (packet_transport == transport_->rtp_packet_transport()) {
rtp_transport_sent_count_++;
} else {
ASSERT_EQ(transport_->rtcp_packet_transport(), packet_transport);
rtcp_transport_sent_count_++;
}
}
int rtp_transport_sent_count() { return rtp_transport_sent_count_; }
int rtcp_transport_sent_count() { return rtcp_transport_sent_count_; }
private:
int rtp_transport_sent_count_ = 0;
int rtcp_transport_sent_count_ = 0;
RtpTransport* transport_ = nullptr;
bool ready_ = false;
rtc::Optional<rtc::NetworkRoute> network_route_;
};
TEST(RtpTransportTest, SettingRtcpAndRtpSignalsReady) {
RtpTransport transport(kMuxDisabled);
SignalObserver observer(&transport);
rtc::FakePacketTransport fake_rtcp("fake_rtcp");
fake_rtcp.SetWritable(true);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetWritable(true);
transport.SetRtcpPacketTransport(&fake_rtcp); // rtcp ready
EXPECT_FALSE(observer.ready());
transport.SetRtpPacketTransport(&fake_rtp); // rtp ready
EXPECT_TRUE(observer.ready());
}
TEST(RtpTransportTest, SettingRtpAndRtcpSignalsReady) {
RtpTransport transport(kMuxDisabled);
SignalObserver observer(&transport);
rtc::FakePacketTransport fake_rtcp("fake_rtcp");
fake_rtcp.SetWritable(true);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetWritable(true);
transport.SetRtpPacketTransport(&fake_rtp); // rtp ready
EXPECT_FALSE(observer.ready());
transport.SetRtcpPacketTransport(&fake_rtcp); // rtcp ready
EXPECT_TRUE(observer.ready());
}
TEST(RtpTransportTest, SettingRtpWithRtcpMuxEnabledSignalsReady) {
RtpTransport transport(kMuxEnabled);
SignalObserver observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetWritable(true);
transport.SetRtpPacketTransport(&fake_rtp); // rtp ready
EXPECT_TRUE(observer.ready());
}
TEST(RtpTransportTest, DisablingRtcpMuxSignalsNotReady) {
RtpTransport transport(kMuxEnabled);
SignalObserver observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetWritable(true);
transport.SetRtpPacketTransport(&fake_rtp); // rtp ready
EXPECT_TRUE(observer.ready());
transport.SetRtcpMuxEnabled(false);
EXPECT_FALSE(observer.ready());
}
TEST(RtpTransportTest, EnablingRtcpMuxSignalsReady) {
RtpTransport transport(kMuxDisabled);
SignalObserver observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetWritable(true);
transport.SetRtpPacketTransport(&fake_rtp); // rtp ready
EXPECT_FALSE(observer.ready());
transport.SetRtcpMuxEnabled(true);
EXPECT_TRUE(observer.ready());
}
// Tests the SignalNetworkRoute is fired when setting a packet transport.
TEST(RtpTransportTest, SetRtpTransportWithNetworkRouteChanged) {
RtpTransport transport(kMuxDisabled);
SignalObserver observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
EXPECT_FALSE(observer.network_route());
rtc::NetworkRoute network_route;
// Set a non-null RTP transport with a new network route.
network_route.connected = true;
network_route.local_network_id = kLocalNetId;
network_route.remote_network_id = kRemoteNetId;
network_route.last_sent_packet_id = kLastPacketId;
network_route.packet_overhead = kTransportOverheadPerPacket;
fake_rtp.SetNetworkRoute(rtc::Optional<rtc::NetworkRoute>(network_route));
transport.SetRtpPacketTransport(&fake_rtp);
ASSERT_TRUE(observer.network_route());
EXPECT_EQ(network_route, *(observer.network_route()));
EXPECT_EQ(kTransportOverheadPerPacket,
observer.network_route()->packet_overhead);
EXPECT_EQ(kLastPacketId, observer.network_route()->last_sent_packet_id);
// Set a null RTP transport.
transport.SetRtpPacketTransport(nullptr);
EXPECT_FALSE(observer.network_route());
}
TEST(RtpTransportTest, SetRtcpTransportWithNetworkRouteChanged) {
RtpTransport transport(kMuxDisabled);
SignalObserver observer(&transport);
rtc::FakePacketTransport fake_rtcp("fake_rtcp");
EXPECT_FALSE(observer.network_route());
rtc::NetworkRoute network_route;
// Set a non-null RTCP transport with a new network route.
network_route.connected = true;
network_route.local_network_id = kLocalNetId;
network_route.remote_network_id = kRemoteNetId;
network_route.last_sent_packet_id = kLastPacketId;
network_route.packet_overhead = kTransportOverheadPerPacket;
fake_rtcp.SetNetworkRoute(rtc::Optional<rtc::NetworkRoute>(network_route));
transport.SetRtcpPacketTransport(&fake_rtcp);
ASSERT_TRUE(observer.network_route());
EXPECT_EQ(network_route, *(observer.network_route()));
EXPECT_EQ(kTransportOverheadPerPacket,
observer.network_route()->packet_overhead);
EXPECT_EQ(kLastPacketId, observer.network_route()->last_sent_packet_id);
// Set a null RTCP transport.
transport.SetRtcpPacketTransport(nullptr);
EXPECT_FALSE(observer.network_route());
}
// Test that RTCP packets are sent over correct transport based on the RTCP-mux
// status.
TEST(RtpTransportTest, RtcpPacketSentOverCorrectTransport) {
// If the RTCP-mux is not enabled, RTCP packets are expected to be sent over
// the RtcpPacketTransport.
RtpTransport transport(kMuxDisabled);
rtc::FakePacketTransport fake_rtcp("fake_rtcp");
rtc::FakePacketTransport fake_rtp("fake_rtp");
transport.SetRtcpPacketTransport(&fake_rtcp); // rtcp ready
transport.SetRtpPacketTransport(&fake_rtp); // rtp ready
SignalObserver observer(&transport);
fake_rtp.SetDestination(&fake_rtp, true);
fake_rtcp.SetDestination(&fake_rtcp, true);
rtc::CopyOnWriteBuffer packet;
EXPECT_TRUE(transport.SendRtcpPacket(&packet, rtc::PacketOptions(), 0));
EXPECT_EQ(1, observer.rtcp_transport_sent_count());
// The RTCP packets are expected to be sent over RtpPacketTransport if
// RTCP-mux is enabled.
transport.SetRtcpMuxEnabled(true);
EXPECT_TRUE(transport.SendRtcpPacket(&packet, rtc::PacketOptions(), 0));
EXPECT_EQ(1, observer.rtp_transport_sent_count());
}
Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" This reverts commit 27f3bf512827b483f9e0c67ce76362d83faa1950. Reason for revert: Broken internal project. Original change's description: > Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." > > This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." > > > > 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} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8587 > Reviewed-on: https://webrtc-review.googlesource.com/64862 > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22615} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8587 Change-Id: I694ce9a039ed52c5961cdc0cba57587bed4cbde4 Reviewed-on: https://webrtc-review.googlesource.com/65381 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22665}
2018-03-29 00:08:03 +00:00
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);
Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" This reverts commit 27f3bf512827b483f9e0c67ce76362d83faa1950. Reason for revert: Broken internal project. Original change's description: > Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." > > This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." > > > > 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} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8587 > Reviewed-on: https://webrtc-review.googlesource.com/64862 > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22615} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8587 Change-Id: I694ce9a039ed52c5961cdc0cba57587bed4cbde4 Reviewed-on: https://webrtc-review.googlesource.com/65381 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22665}
2018-03-29 00:08:03 +00:00
SignalCounter observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetWritable(true);
// State changes, so we should signal.
transport.SetRtpPacketTransport(&fake_rtp);
Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" This reverts commit 27f3bf512827b483f9e0c67ce76362d83faa1950. Reason for revert: Broken internal project. Original change's description: > Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." > > This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." > > > > 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} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8587 > Reviewed-on: https://webrtc-review.googlesource.com/64862 > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22615} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8587 Change-Id: I694ce9a039ed52c5961cdc0cba57587bed4cbde4 Reviewed-on: https://webrtc-review.googlesource.com/65381 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22665}
2018-03-29 00:08:03 +00:00
EXPECT_EQ(observer.count(), 1);
// State does not change, so we should not signal.
transport.SetRtpPacketTransport(&fake_rtp);
Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" This reverts commit 27f3bf512827b483f9e0c67ce76362d83faa1950. Reason for revert: Broken internal project. Original change's description: > Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." > > This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." > > > > 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} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8587 > Reviewed-on: https://webrtc-review.googlesource.com/64862 > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22615} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8587 Change-Id: I694ce9a039ed52c5961cdc0cba57587bed4cbde4 Reviewed-on: https://webrtc-review.googlesource.com/65381 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22665}
2018-03-29 00:08:03 +00:00
EXPECT_EQ(observer.count(), 1);
// State does not change, so we should not signal.
transport.SetRtcpMuxEnabled(true);
Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" This reverts commit 27f3bf512827b483f9e0c67ce76362d83faa1950. Reason for revert: Broken internal project. Original change's description: > Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." > > This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." > > > > 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} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8587 > Reviewed-on: https://webrtc-review.googlesource.com/64862 > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22615} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8587 Change-Id: I694ce9a039ed52c5961cdc0cba57587bed4cbde4 Reviewed-on: https://webrtc-review.googlesource.com/65381 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22665}
2018-03-29 00:08:03 +00:00
EXPECT_EQ(observer.count(), 1);
// State changes, so we should signal.
transport.SetRtcpMuxEnabled(false);
Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" This reverts commit 27f3bf512827b483f9e0c67ce76362d83faa1950. Reason for revert: Broken internal project. Original change's description: > Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." > > This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." > > > > 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} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8587 > Reviewed-on: https://webrtc-review.googlesource.com/64862 > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22615} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8587 Change-Id: I694ce9a039ed52c5961cdc0cba57587bed4cbde4 Reviewed-on: https://webrtc-review.googlesource.com/65381 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22665}
2018-03-29 00:08:03 +00:00
EXPECT_EQ(observer.count(), 2);
}
// Test that SignalPacketReceived fires with rtcp=true when a RTCP packet is
// received.
TEST(RtpTransportTest, SignalDemuxedRtcp) {
RtpTransport transport(kMuxDisabled);
Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" This reverts commit 27f3bf512827b483f9e0c67ce76362d83faa1950. Reason for revert: Broken internal project. Original change's description: > Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." > > This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." > > > > 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} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8587 > Reviewed-on: https://webrtc-review.googlesource.com/64862 > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22615} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8587 Change-Id: I694ce9a039ed52c5961cdc0cba57587bed4cbde4 Reviewed-on: https://webrtc-review.googlesource.com/65381 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22665}
2018-03-29 00:08:03 +00:00
SignalPacketReceivedCounter observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetDestination(&fake_rtp, true);
transport.SetRtpPacketTransport(&fake_rtp);
// An rtcp packet.
const char data[] = {0, 73, 0, 0};
const int len = 4;
const rtc::PacketOptions options;
const int flags = 0;
fake_rtp.SendPacket(data, len, options, flags);
EXPECT_EQ(0, observer.rtp_count());
EXPECT_EQ(1, observer.rtcp_count());
}
static const unsigned char kRtpData[] = {0x80, 0x11, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0};
static const int kRtpLen = 12;
// Test that SignalPacketReceived fires with rtcp=false when a RTP packet with a
// handled payload type is received.
TEST(RtpTransportTest, SignalHandledRtpPayloadType) {
RtpTransport transport(kMuxDisabled);
Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" This reverts commit 27f3bf512827b483f9e0c67ce76362d83faa1950. Reason for revert: Broken internal project. Original change's description: > Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." > > This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." > > > > 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} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8587 > Reviewed-on: https://webrtc-review.googlesource.com/64862 > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22615} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8587 Change-Id: I694ce9a039ed52c5961cdc0cba57587bed4cbde4 Reviewed-on: https://webrtc-review.googlesource.com/65381 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22665}
2018-03-29 00:08:03 +00:00
SignalPacketReceivedCounter observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetDestination(&fake_rtp, true);
transport.SetRtpPacketTransport(&fake_rtp);
// An rtp packet.
const rtc::PacketOptions options;
const int flags = 0;
rtc::Buffer rtp_data(kRtpData, kRtpLen);
fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags);
EXPECT_EQ(1, observer.rtp_count());
EXPECT_EQ(0, observer.rtcp_count());
}
} // namespace webrtc