webrtc_m130/pc/channel_unittest.cc

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/*
* Copyright 2009 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 <cstdint>
#include <memory>
#include <utility>
#include "absl/memory/memory.h"
#include "api/array_view.h"
#include "api/audio_options.h"
#include "api/rtp_parameters.h"
#include "media/base/codec.h"
#include "media/base/fake_media_engine.h"
#include "media/base/fake_rtp.h"
#include "media/base/media_channel.h"
#include "p2p/base/candidate_pair_interface.h"
#include "p2p/base/fake_dtls_transport.h"
#include "p2p/base/fake_packet_transport.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "pc/channel.h"
#include "pc/dtls_srtp_transport.h"
#include "pc/jsep_transport.h"
#include "pc/rtp_transport.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/buffer.h"
#include "rtc_base/byte_order.h"
#include "rtc_base/checks.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/ssl_identity.h"
#include "test/gtest.h"
using cricket::DtlsTransportInternal;
using cricket::FakeVoiceMediaChannel;
using cricket::StreamParams;
using webrtc::RtpTransceiverDirection;
using webrtc::SdpType;
namespace {
const cricket::AudioCodec kPcmuCodec(0, "PCMU", 64000, 8000, 1);
const cricket::AudioCodec kPcmaCodec(8, "PCMA", 64000, 8000, 1);
const cricket::AudioCodec kIsacCodec(103, "ISAC", 40000, 16000, 1);
const cricket::VideoCodec kH264Codec(97, "H264");
const cricket::VideoCodec kH264SvcCodec(99, "H264-SVC");
const cricket::DataCodec kGoogleDataCodec(101, "google-data");
const uint32_t kSsrc1 = 0x1111;
const uint32_t kSsrc2 = 0x2222;
const uint32_t kSsrc3 = 0x3333;
const uint32_t kSsrc4 = 0x4444;
const int kAudioPts[] = {0, 8};
const int kVideoPts[] = {97, 99};
enum class NetworkIsWorker { Yes, No };
} // namespace
Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) Reason for revert: This CL just landed: https://codereview.chromium.org/1323243006/ Which fixes the FYI bots for the original CL, and breaks them for this revert. Original issue's description: > Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) > > Reason for revert: > This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step. > > Original issue's description: > > TransportController refactoring. > > > > Getting rid of TransportProxy, and in its place adding a > > TransportController class which will facilitate access to and manage > > the lifetimes of Transports. These Transports will now be accessed > > solely from the worker thread, simplifying their implementation. > > > > This refactoring also pulls Transport-related code out of BaseSession. > > Which means that BaseChannels will now rely on the TransportController > > interface to create channels, rather than BaseSession. > > > > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83 > > Cr-Commit-Position: refs/heads/master@{#10022} > > TBR=pthatcher@webrtc.org,deadbeef@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > > Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c > Cr-Commit-Position: refs/heads/master@{#10024} TBR=pthatcher@webrtc.org,torbjorng@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1361773005 Cr-Commit-Position: refs/heads/master@{#10036}
2015-09-23 11:50:27 -07:00
template <class ChannelT,
class MediaChannelT,
class ContentT,
class CodecT,
class MediaInfoT,
class OptionsT>
class Traits {
public:
typedef ChannelT Channel;
typedef MediaChannelT MediaChannel;
typedef ContentT Content;
typedef CodecT Codec;
typedef MediaInfoT MediaInfo;
typedef OptionsT Options;
};
class VoiceTraits : public Traits<cricket::VoiceChannel,
cricket::FakeVoiceMediaChannel,
cricket::AudioContentDescription,
cricket::AudioCodec,
cricket::VoiceMediaInfo,
Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) Reason for revert: This CL just landed: https://codereview.chromium.org/1323243006/ Which fixes the FYI bots for the original CL, and breaks them for this revert. Original issue's description: > Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) > > Reason for revert: > This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step. > > Original issue's description: > > TransportController refactoring. > > > > Getting rid of TransportProxy, and in its place adding a > > TransportController class which will facilitate access to and manage > > the lifetimes of Transports. These Transports will now be accessed > > solely from the worker thread, simplifying their implementation. > > > > This refactoring also pulls Transport-related code out of BaseSession. > > Which means that BaseChannels will now rely on the TransportController > > interface to create channels, rather than BaseSession. > > > > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83 > > Cr-Commit-Position: refs/heads/master@{#10022} > > TBR=pthatcher@webrtc.org,deadbeef@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > > Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c > Cr-Commit-Position: refs/heads/master@{#10024} TBR=pthatcher@webrtc.org,torbjorng@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1361773005 Cr-Commit-Position: refs/heads/master@{#10036}
2015-09-23 11:50:27 -07:00
cricket::AudioOptions> {};
class VideoTraits : public Traits<cricket::VideoChannel,
cricket::FakeVideoMediaChannel,
cricket::VideoContentDescription,
cricket::VideoCodec,
cricket::VideoMediaInfo,
Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) Reason for revert: This CL just landed: https://codereview.chromium.org/1323243006/ Which fixes the FYI bots for the original CL, and breaks them for this revert. Original issue's description: > Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) > > Reason for revert: > This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step. > > Original issue's description: > > TransportController refactoring. > > > > Getting rid of TransportProxy, and in its place adding a > > TransportController class which will facilitate access to and manage > > the lifetimes of Transports. These Transports will now be accessed > > solely from the worker thread, simplifying their implementation. > > > > This refactoring also pulls Transport-related code out of BaseSession. > > Which means that BaseChannels will now rely on the TransportController > > interface to create channels, rather than BaseSession. > > > > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83 > > Cr-Commit-Position: refs/heads/master@{#10022} > > TBR=pthatcher@webrtc.org,deadbeef@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > > Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c > Cr-Commit-Position: refs/heads/master@{#10024} TBR=pthatcher@webrtc.org,torbjorng@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1361773005 Cr-Commit-Position: refs/heads/master@{#10036}
2015-09-23 11:50:27 -07:00
cricket::VideoOptions> {};
class DataTraits : public Traits<cricket::RtpDataChannel,
cricket::FakeDataMediaChannel,
cricket::DataContentDescription,
cricket::DataCodec,
cricket::DataMediaInfo,
Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) Reason for revert: This CL just landed: https://codereview.chromium.org/1323243006/ Which fixes the FYI bots for the original CL, and breaks them for this revert. Original issue's description: > Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) > > Reason for revert: > This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step. > > Original issue's description: > > TransportController refactoring. > > > > Getting rid of TransportProxy, and in its place adding a > > TransportController class which will facilitate access to and manage > > the lifetimes of Transports. These Transports will now be accessed > > solely from the worker thread, simplifying their implementation. > > > > This refactoring also pulls Transport-related code out of BaseSession. > > Which means that BaseChannels will now rely on the TransportController > > interface to create channels, rather than BaseSession. > > > > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83 > > Cr-Commit-Position: refs/heads/master@{#10022} > > TBR=pthatcher@webrtc.org,deadbeef@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > > Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c > Cr-Commit-Position: refs/heads/master@{#10024} TBR=pthatcher@webrtc.org,torbjorng@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1361773005 Cr-Commit-Position: refs/heads/master@{#10036}
2015-09-23 11:50:27 -07:00
cricket::DataOptions> {};
// Base class for Voice/Video/RtpDataChannel tests
template <class T>
class ChannelTest : public testing::Test, public sigslot::has_slots<> {
public:
enum Flags {
RTCP_MUX = 0x1,
SSRC_MUX = 0x8,
DTLS = 0x10,
// Use BaseChannel with PacketTransportInternal rather than
// DtlsTransportInternal.
RAW_PACKET_TRANSPORT = 0x20,
};
ChannelTest(bool verify_playout,
rtc::ArrayView<const uint8_t> rtp_data,
rtc::ArrayView<const uint8_t> rtcp_data,
NetworkIsWorker network_is_worker)
: verify_playout_(verify_playout),
rtp_packet_(rtp_data.data(), rtp_data.size()),
rtcp_packet_(rtcp_data.data(), rtcp_data.size()) {
if (network_is_worker == NetworkIsWorker::Yes) {
network_thread_ = rtc::Thread::Current();
} else {
network_thread_keeper_ = rtc::Thread::Create();
network_thread_keeper_->SetName("Network", nullptr);
network_thread_ = network_thread_keeper_.get();
}
}
void CreateChannels(int flags1, int flags2) {
CreateChannels(absl::make_unique<typename T::MediaChannel>(
nullptr, typename T::Options()),
absl::make_unique<typename T::MediaChannel>(
nullptr, typename T::Options()),
flags1, flags2);
}
void CreateChannels(std::unique_ptr<typename T::MediaChannel> ch1,
std::unique_ptr<typename T::MediaChannel> ch2,
int flags1,
int flags2) {
// Network thread is started in CreateChannels, to allow the test to
// configure a fake clock before any threads are spawned and attempt to
// access the time.
if (network_thread_keeper_) {
network_thread_keeper_->Start();
}
// Make sure if using raw packet transports, they're used for both
// channels.
RTC_DCHECK_EQ(flags1 & RAW_PACKET_TRANSPORT, flags2 & RAW_PACKET_TRANSPORT);
rtc::Thread* worker_thread = rtc::Thread::Current();
media_channel1_ = ch1.get();
media_channel2_ = ch2.get();
rtc::PacketTransportInternal* rtp1 = nullptr;
rtc::PacketTransportInternal* rtcp1 = nullptr;
rtc::PacketTransportInternal* rtp2 = nullptr;
rtc::PacketTransportInternal* rtcp2 = nullptr;
// Based on flags, create fake DTLS or raw packet transports.
if (flags1 & RAW_PACKET_TRANSPORT) {
fake_rtp_packet_transport1_.reset(
new rtc::FakePacketTransport("channel1_rtp"));
rtp1 = fake_rtp_packet_transport1_.get();
if (!(flags1 & RTCP_MUX)) {
fake_rtcp_packet_transport1_.reset(
new rtc::FakePacketTransport("channel1_rtcp"));
rtcp1 = fake_rtcp_packet_transport1_.get();
}
} else {
// Confirmed to work with KT_RSA and KT_ECDSA.
fake_rtp_dtls_transport1_.reset(new cricket::FakeDtlsTransport(
"channel1", cricket::ICE_CANDIDATE_COMPONENT_RTP));
rtp1 = fake_rtp_dtls_transport1_.get();
if (!(flags1 & RTCP_MUX)) {
fake_rtcp_dtls_transport1_.reset(new cricket::FakeDtlsTransport(
"channel1", cricket::ICE_CANDIDATE_COMPONENT_RTCP));
rtcp1 = fake_rtcp_dtls_transport1_.get();
}
if (flags1 & DTLS) {
auto cert1 =
rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT)));
fake_rtp_dtls_transport1_->SetLocalCertificate(cert1);
if (fake_rtcp_dtls_transport1_) {
fake_rtcp_dtls_transport1_->SetLocalCertificate(cert1);
}
}
}
// Based on flags, create fake DTLS or raw packet transports.
if (flags2 & RAW_PACKET_TRANSPORT) {
fake_rtp_packet_transport2_.reset(
new rtc::FakePacketTransport("channel2_rtp"));
rtp2 = fake_rtp_packet_transport2_.get();
if (!(flags2 & RTCP_MUX)) {
fake_rtcp_packet_transport2_.reset(
new rtc::FakePacketTransport("channel2_rtcp"));
rtcp2 = fake_rtcp_packet_transport2_.get();
}
} else {
// Confirmed to work with KT_RSA and KT_ECDSA.
fake_rtp_dtls_transport2_.reset(new cricket::FakeDtlsTransport(
"channel2", cricket::ICE_CANDIDATE_COMPONENT_RTP));
rtp2 = fake_rtp_dtls_transport2_.get();
if (!(flags2 & RTCP_MUX)) {
fake_rtcp_dtls_transport2_.reset(new cricket::FakeDtlsTransport(
"channel2", cricket::ICE_CANDIDATE_COMPONENT_RTCP));
rtcp2 = fake_rtcp_dtls_transport2_.get();
}
if (flags2 & DTLS) {
auto cert2 =
rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
rtc::SSLIdentity::Generate("session2", rtc::KT_DEFAULT)));
fake_rtp_dtls_transport2_->SetLocalCertificate(cert2);
if (fake_rtcp_dtls_transport2_) {
fake_rtcp_dtls_transport2_->SetLocalCertificate(cert2);
}
}
}
rtp_transport1_ = CreateRtpTransportBasedOnFlags(
fake_rtp_packet_transport1_.get(), fake_rtcp_packet_transport1_.get(),
fake_rtp_dtls_transport1_.get(), fake_rtcp_dtls_transport1_.get(),
flags1);
rtp_transport2_ = CreateRtpTransportBasedOnFlags(
fake_rtp_packet_transport2_.get(), fake_rtcp_packet_transport2_.get(),
fake_rtp_dtls_transport2_.get(), fake_rtcp_dtls_transport2_.get(),
flags2);
channel1_ = CreateChannel(worker_thread, network_thread_, &media_engine_,
std::move(ch1), rtp_transport1_.get(), flags1);
channel2_ = CreateChannel(worker_thread, network_thread_, &media_engine_,
std::move(ch2), rtp_transport2_.get(), flags2);
channel1_->SignalRtcpMuxFullyActive.connect(
this, &ChannelTest<T>::OnRtcpMuxFullyActive1);
channel2_->SignalRtcpMuxFullyActive.connect(
this, &ChannelTest<T>::OnRtcpMuxFullyActive2);
CreateContent(flags1, kPcmuCodec, kH264Codec, &local_media_content1_);
CreateContent(flags2, kPcmuCodec, kH264Codec, &local_media_content2_);
CopyContent(local_media_content1_, &remote_media_content1_);
CopyContent(local_media_content2_, &remote_media_content2_);
// Add stream information (SSRC) to the local content but not to the remote
// content. This means that we per default know the SSRC of what we send but
// not what we receive.
AddLegacyStreamInContent(kSsrc1, flags1, &local_media_content1_);
AddLegacyStreamInContent(kSsrc2, flags2, &local_media_content2_);
// If SSRC_MUX is used we also need to know the SSRC of the incoming stream.
if (flags1 & SSRC_MUX) {
AddLegacyStreamInContent(kSsrc1, flags1, &remote_media_content1_);
}
if (flags2 & SSRC_MUX) {
AddLegacyStreamInContent(kSsrc2, flags2, &remote_media_content2_);
}
}
std::unique_ptr<typename T::Channel> CreateChannel(
rtc::Thread* worker_thread,
rtc::Thread* network_thread,
Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) Reason for revert: This CL just landed: https://codereview.chromium.org/1323243006/ Which fixes the FYI bots for the original CL, and breaks them for this revert. Original issue's description: > Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) > > Reason for revert: > This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step. > > Original issue's description: > > TransportController refactoring. > > > > Getting rid of TransportProxy, and in its place adding a > > TransportController class which will facilitate access to and manage > > the lifetimes of Transports. These Transports will now be accessed > > solely from the worker thread, simplifying their implementation. > > > > This refactoring also pulls Transport-related code out of BaseSession. > > Which means that BaseChannels will now rely on the TransportController > > interface to create channels, rather than BaseSession. > > > > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83 > > Cr-Commit-Position: refs/heads/master@{#10022} > > TBR=pthatcher@webrtc.org,deadbeef@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > > Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c > Cr-Commit-Position: refs/heads/master@{#10024} TBR=pthatcher@webrtc.org,torbjorng@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1361773005 Cr-Commit-Position: refs/heads/master@{#10036}
2015-09-23 11:50:27 -07:00
cricket::MediaEngineInterface* engine,
std::unique_ptr<typename T::MediaChannel> ch,
webrtc::RtpTransportInternal* rtp_transport,
int flags) {
rtc::Thread* signaling_thread = rtc::Thread::Current();
auto channel = absl::make_unique<typename T::Channel>(
worker_thread, network_thread, signaling_thread, engine, std::move(ch),
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class that only handles SRTP configuration to a more generic structure that can be used and extended for all per peer connection CryptoOptions that can be on a given PeerConnection. Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be accessed as crypto_options.srtp.whatever_option_name. This is more inline with other structures we have in WebRTC such as VideoConfig. As additional features are added over time this will allow the structure to remain compartmentalized and concerned components can only request a subset of the overall configuration structure e.g: void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); In addition to this it made little sense for sslstreamadapter.h to hold all Srtp related configuration options. The header has become loo large and takes on too many responsibilities and spilting this up will lead to more maintainable code going forward. This will be used in a future CL to enable configuration options for the newly supported Frame Crypto. Reland Fix: - cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional root level configuration. - peerconnectionfactory - If this optional is set will now overwrite the underyling value. This along with the other field will be deprecated once dependent projects are updated. TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org Bug: webrtc:9681 Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d Reviewed-on: https://webrtc-review.googlesource.com/c/105560 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25135}
2018-10-11 15:33:17 -07:00
cricket::CN_AUDIO, (flags & DTLS) != 0, webrtc::CryptoOptions());
Reland "Reland "Propagate media transport to media channel."" This is a reland of da65ed2adcfa57ff3288ce01c1602c973fcab00d Original change's description: > Reland "Propagate media transport to media channel." > > This reverts commit 37cf2455a420124b341ad06ac27fa3c4dbd29d3c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Propagate media transport to media channel." > > > > This reverts commit 8c16f745ab92cb6d305283e87fa8a661ae500ce4. > > > > Reason for revert: Breaks downstream project > > > > Original change's description: > > > Propagate media transport to media channel. > > > > > > 1. Pass media transport factory to JSEP transport controller. > > > 2. Pass media transport to voice media channel. > > > 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel. > > > > > > Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71 > > > Bug: webrtc:9719 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/105542 > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > > Reviewed-by: Peter Slatala <psla@webrtc.org> > > > Commit-Queue: Anton Sukhanov <sukhanov@google.com> > > > Cr-Commit-Position: refs/heads/master@{#25152} > > > > TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:9719 > > Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1 > > Reviewed-on: https://webrtc-review.googlesource.com/c/105840 > > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25154} > > TBR=steveanton@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com > > Change-Id: I505ff3451eae81573531faef155ff35d7f894022 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9719 > Reviewed-on: https://webrtc-review.googlesource.com/c/106500 > Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> > Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25220} Bug: webrtc:9719 Tbr: Steve Anton <steveanton@webrtc.org> Tbr: Niels Moller <nisse@webrtc.org> Change-Id: Ib45691ba8be9abb89ff8c6dac1861bdf59be4c8d Reviewed-on: https://webrtc-review.googlesource.com/c/106561 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25240}
2018-10-17 13:15:42 -07:00
channel->Init_w(rtp_transport, /*media_transport=*/nullptr);
return channel;
}
std::unique_ptr<webrtc::RtpTransportInternal> CreateRtpTransportBasedOnFlags(
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport,
DtlsTransportInternal* rtp_dtls_transport,
DtlsTransportInternal* rtcp_dtls_transport,
int flags) {
if (flags & RTCP_MUX) {
rtcp_packet_transport = nullptr;
rtcp_dtls_transport = nullptr;
}
if (flags & DTLS) {
return CreateDtlsSrtpTransport(rtp_dtls_transport, rtcp_dtls_transport);
} else {
if (flags & RAW_PACKET_TRANSPORT) {
return CreateUnencryptedTransport(rtp_packet_transport,
rtcp_packet_transport);
} else {
return CreateUnencryptedTransport(rtp_dtls_transport,
rtcp_dtls_transport);
}
}
}
std::unique_ptr<webrtc::RtpTransport> CreateUnencryptedTransport(
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport) {
auto rtp_transport = absl::make_unique<webrtc::RtpTransport>(
rtcp_packet_transport == nullptr);
rtp_transport->SetRtpPacketTransport(rtp_packet_transport);
if (rtcp_packet_transport) {
rtp_transport->SetRtcpPacketTransport(rtcp_packet_transport);
}
return rtp_transport;
}
std::unique_ptr<webrtc::DtlsSrtpTransport> CreateDtlsSrtpTransport(
cricket::DtlsTransportInternal* rtp_dtls_transport,
cricket::DtlsTransportInternal* rtcp_dtls_transport) {
auto dtls_srtp_transport = absl::make_unique<webrtc::DtlsSrtpTransport>(
rtcp_dtls_transport == nullptr);
dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport,
rtcp_dtls_transport);
return dtls_srtp_transport;
}
void ConnectFakeTransports() {
network_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
bool asymmetric = false;
// Depending on test flags, could be using DTLS or raw packet transport.
if (fake_rtp_dtls_transport1_ && fake_rtp_dtls_transport2_) {
fake_rtp_dtls_transport1_->SetDestination(
fake_rtp_dtls_transport2_.get(), asymmetric);
}
if (fake_rtcp_dtls_transport1_ && fake_rtcp_dtls_transport2_) {
fake_rtcp_dtls_transport1_->SetDestination(
fake_rtcp_dtls_transport2_.get(), asymmetric);
}
if (fake_rtp_packet_transport1_ && fake_rtp_packet_transport2_) {
fake_rtp_packet_transport1_->SetDestination(
fake_rtp_packet_transport2_.get(), asymmetric);
}
if (fake_rtcp_packet_transport1_ && fake_rtcp_packet_transport2_) {
fake_rtcp_packet_transport1_->SetDestination(
fake_rtcp_packet_transport2_.get(), asymmetric);
}
});
}
bool SendInitiate() {
bool result = channel1_->SetLocalContent(&local_media_content1_,
SdpType::kOffer, NULL);
if (result) {
channel1_->Enable(true);
result = channel2_->SetRemoteContent(&remote_media_content1_,
SdpType::kOffer, NULL);
if (result) {
ConnectFakeTransports();
result = channel2_->SetLocalContent(&local_media_content2_,
SdpType::kAnswer, NULL);
}
}
return result;
}
bool SendAccept() {
channel2_->Enable(true);
return channel1_->SetRemoteContent(&remote_media_content2_,
SdpType::kAnswer, NULL);
}
bool SendOffer() {
bool result = channel1_->SetLocalContent(&local_media_content1_,
SdpType::kOffer, NULL);
if (result) {
channel1_->Enable(true);
result = channel2_->SetRemoteContent(&remote_media_content1_,
SdpType::kOffer, NULL);
}
return result;
}
bool SendProvisionalAnswer() {
bool result = channel2_->SetLocalContent(&local_media_content2_,
SdpType::kPrAnswer, NULL);
if (result) {
channel2_->Enable(true);
result = channel1_->SetRemoteContent(&remote_media_content2_,
SdpType::kPrAnswer, NULL);
ConnectFakeTransports();
}
return result;
}
bool SendFinalAnswer() {
bool result = channel2_->SetLocalContent(&local_media_content2_,
SdpType::kAnswer, NULL);
if (result)
result = channel1_->SetRemoteContent(&remote_media_content2_,
SdpType::kAnswer, NULL);
return result;
}
bool Terminate() {
channel1_.reset();
channel2_.reset();
fake_rtp_dtls_transport1_.reset();
fake_rtcp_dtls_transport1_.reset();
fake_rtp_dtls_transport2_.reset();
fake_rtcp_dtls_transport2_.reset();
fake_rtp_packet_transport1_.reset();
fake_rtcp_packet_transport1_.reset();
fake_rtp_packet_transport2_.reset();
fake_rtcp_packet_transport2_.reset();
if (network_thread_keeper_) {
network_thread_keeper_.reset();
}
return true;
}
void SendRtp1() {
media_channel1_->SendRtp(rtp_packet_.data(), rtp_packet_.size(),
rtc::PacketOptions());
}
void SendRtp2() {
media_channel2_->SendRtp(rtp_packet_.data(), rtp_packet_.size(),
rtc::PacketOptions());
}
void SendRtcp1() {
media_channel1_->SendRtcp(rtcp_packet_.data(), rtcp_packet_.size());
}
void SendRtcp2() {
media_channel2_->SendRtcp(rtcp_packet_.data(), rtcp_packet_.size());
}
// Methods to send custom data.
void SendCustomRtp1(uint32_t ssrc, int sequence_number, int pl_type = -1) {
rtc::Buffer data = CreateRtpData(ssrc, sequence_number, pl_type);
media_channel1_->SendRtp(data.data(), data.size(), rtc::PacketOptions());
}
void SendCustomRtp2(uint32_t ssrc, int sequence_number, int pl_type = -1) {
rtc::Buffer data = CreateRtpData(ssrc, sequence_number, pl_type);
media_channel2_->SendRtp(data.data(), data.size(), rtc::PacketOptions());
}
void SendCustomRtcp1(uint32_t ssrc) {
rtc::Buffer data = CreateRtcpData(ssrc);
media_channel1_->SendRtcp(data.data(), data.size());
}
void SendCustomRtcp2(uint32_t ssrc) {
rtc::Buffer data = CreateRtcpData(ssrc);
media_channel2_->SendRtcp(data.data(), data.size());
}
bool CheckRtp1() {
return media_channel1_->CheckRtp(rtp_packet_.data(), rtp_packet_.size());
}
bool CheckRtp2() {
return media_channel2_->CheckRtp(rtp_packet_.data(), rtp_packet_.size());
}
bool CheckRtcp1() {
return media_channel1_->CheckRtcp(rtcp_packet_.data(), rtcp_packet_.size());
}
bool CheckRtcp2() {
return media_channel2_->CheckRtcp(rtcp_packet_.data(), rtcp_packet_.size());
}
// Methods to check custom data.
bool CheckCustomRtp1(uint32_t ssrc, int sequence_number, int pl_type = -1) {
rtc::Buffer data = CreateRtpData(ssrc, sequence_number, pl_type);
return media_channel1_->CheckRtp(data.data(), data.size());
}
bool CheckCustomRtp2(uint32_t ssrc, int sequence_number, int pl_type = -1) {
rtc::Buffer data = CreateRtpData(ssrc, sequence_number, pl_type);
return media_channel2_->CheckRtp(data.data(), data.size());
}
bool CheckCustomRtcp1(uint32_t ssrc) {
rtc::Buffer data = CreateRtcpData(ssrc);
return media_channel1_->CheckRtcp(data.data(), data.size());
}
bool CheckCustomRtcp2(uint32_t ssrc) {
rtc::Buffer data = CreateRtcpData(ssrc);
return media_channel2_->CheckRtcp(data.data(), data.size());
}
rtc::Buffer CreateRtpData(uint32_t ssrc, int sequence_number, int pl_type) {
rtc::Buffer data(rtp_packet_.data(), rtp_packet_.size());
// Set SSRC in the rtp packet copy.
rtc::SetBE32(data.data() + 8, ssrc);
rtc::SetBE16(data.data() + 2, sequence_number);
if (pl_type >= 0) {
rtc::Set8(data.data(), 1, static_cast<uint8_t>(pl_type));
}
return data;
}
rtc::Buffer CreateRtcpData(uint32_t ssrc) {
rtc::Buffer data(rtcp_packet_.data(), rtcp_packet_.size());
// Set SSRC in the rtcp packet copy.
rtc::SetBE32(data.data() + 4, ssrc);
return data;
}
bool CheckNoRtp1() { return media_channel1_->CheckNoRtp(); }
bool CheckNoRtp2() { return media_channel2_->CheckNoRtp(); }
bool CheckNoRtcp1() { return media_channel1_->CheckNoRtcp(); }
bool CheckNoRtcp2() { return media_channel2_->CheckNoRtcp(); }
void CreateContent(int flags,
const cricket::AudioCodec& audio_codec,
const cricket::VideoCodec& video_codec,
typename T::Content* content) {
// overridden in specialized classes
}
void CopyContent(const typename T::Content& source,
typename T::Content* content) {
// overridden in specialized classes
}
// Creates a MediaContent with one stream.
// kPcmuCodec is used as audio codec and kH264Codec is used as video codec.
typename T::Content* CreateMediaContentWithStream(uint32_t ssrc) {
typename T::Content* content = new typename T::Content();
CreateContent(0, kPcmuCodec, kH264Codec, content);
AddLegacyStreamInContent(ssrc, 0, content);
return content;
}
// Will manage the lifetime of a CallThread, making sure it's
// destroyed before this object goes out of scope.
class ScopedCallThread {
public:
template <class FunctorT>
explicit ScopedCallThread(FunctorT&& functor)
: thread_(rtc::Thread::Create()),
task_(new rtc::FunctorMessageHandler<void, FunctorT>(
std::forward<FunctorT>(functor))) {
thread_->Start();
thread_->Post(RTC_FROM_HERE, task_.get());
}
~ScopedCallThread() { thread_->Stop(); }
rtc::Thread* thread() { return thread_.get(); }
private:
std::unique_ptr<rtc::Thread> thread_;
std::unique_ptr<rtc::MessageHandler> task_;
};
bool CodecMatches(const typename T::Codec& c1, const typename T::Codec& c2) {
return false; // overridden in specialized classes
}
void OnRtcpMuxFullyActive1(const std::string&) {
rtcp_mux_activated_callbacks1_++;
}
void OnRtcpMuxFullyActive2(const std::string&) {
rtcp_mux_activated_callbacks2_++;
}
cricket::CandidatePairInterface* last_selected_candidate_pair() {
return last_selected_candidate_pair_;
}
void AddLegacyStreamInContent(uint32_t ssrc,
int flags,
typename T::Content* content) {
// Base implementation.
}
// Tests that can be used by derived classes.
// Basic sanity check.
void TestInit() {
CreateChannels(0, 0);
EXPECT_FALSE(channel1_->srtp_active());
EXPECT_FALSE(media_channel1_->sending());
if (verify_playout_) {
EXPECT_FALSE(media_channel1_->playout());
}
EXPECT_TRUE(media_channel1_->codecs().empty());
EXPECT_TRUE(media_channel1_->recv_streams().empty());
EXPECT_TRUE(media_channel1_->rtp_packets().empty());
EXPECT_TRUE(media_channel1_->rtcp_packets().empty());
}
// Test that SetLocalContent and SetRemoteContent properly configure
// the codecs.
void TestSetContents() {
CreateChannels(0, 0);
typename T::Content content;
CreateContent(0, kPcmuCodec, kH264Codec, &content);
EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer, NULL));
EXPECT_EQ(0U, media_channel1_->codecs().size());
EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, NULL));
ASSERT_EQ(1U, media_channel1_->codecs().size());
EXPECT_TRUE(
CodecMatches(content.codecs()[0], media_channel1_->codecs()[0]));
}
// Test that SetLocalContent and SetRemoteContent properly configure
// extmap-allow-mixed.
void TestSetContentsExtmapAllowMixedCaller(bool offer, bool answer) {
// For a caller, SetLocalContent() is called first with an offer and next
// SetRemoteContent() is called with the answer.
CreateChannels(0, 0);
typename T::Content content;
CreateContent(0, kPcmuCodec, kH264Codec, &content);
auto offer_enum = offer ? (T::Content::kSession) : (T::Content::kNo);
auto answer_enum = answer ? (T::Content::kSession) : (T::Content::kNo);
content.set_extmap_allow_mixed_enum(offer_enum);
EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer, NULL));
content.set_extmap_allow_mixed_enum(answer_enum);
EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, NULL));
EXPECT_EQ(answer, media_channel1_->ExtmapAllowMixed());
}
void TestSetContentsExtmapAllowMixedCallee(bool offer, bool answer) {
// For a callee, SetRemoteContent() is called first with an offer and next
// SetLocalContent() is called with the answer.
CreateChannels(0, 0);
typename T::Content content;
CreateContent(0, kPcmuCodec, kH264Codec, &content);
auto offer_enum = offer ? (T::Content::kSession) : (T::Content::kNo);
auto answer_enum = answer ? (T::Content::kSession) : (T::Content::kNo);
content.set_extmap_allow_mixed_enum(offer_enum);
EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kOffer, NULL));
content.set_extmap_allow_mixed_enum(answer_enum);
EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kAnswer, NULL));
EXPECT_EQ(answer, media_channel1_->ExtmapAllowMixed());
}
// Test that SetLocalContent and SetRemoteContent properly deals
// with an empty offer.
void TestSetContentsNullOffer() {
CreateChannels(0, 0);
typename T::Content content;
EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer, NULL));
CreateContent(0, kPcmuCodec, kH264Codec, &content);
EXPECT_EQ(0U, media_channel1_->codecs().size());
EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, NULL));
ASSERT_EQ(1U, media_channel1_->codecs().size());
EXPECT_TRUE(
CodecMatches(content.codecs()[0], media_channel1_->codecs()[0]));
}
// Test that SetLocalContent and SetRemoteContent properly set RTCP
// mux.
void TestSetContentsRtcpMux() {
CreateChannels(0, 0);
typename T::Content content;
CreateContent(0, kPcmuCodec, kH264Codec, &content);
// Both sides agree on mux. Should no longer be a separate RTCP channel.
content.set_rtcp_mux(true);
EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer, NULL));
EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, NULL));
// Only initiator supports mux. Should still have a separate RTCP channel.
EXPECT_TRUE(channel2_->SetLocalContent(&content, SdpType::kOffer, NULL));
content.set_rtcp_mux(false);
EXPECT_TRUE(channel2_->SetRemoteContent(&content, SdpType::kAnswer, NULL));
}
// Test that SetLocalContent and SetRemoteContent properly set RTCP
// mux when a provisional answer is received.
void TestSetContentsRtcpMuxWithPrAnswer() {
CreateChannels(0, 0);
typename T::Content content;
CreateContent(0, kPcmuCodec, kH264Codec, &content);
content.set_rtcp_mux(true);
EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer, NULL));
EXPECT_TRUE(
channel1_->SetRemoteContent(&content, SdpType::kPrAnswer, NULL));
// Both sides agree on mux. Should signal RTCP mux as fully activated.
EXPECT_EQ(0, rtcp_mux_activated_callbacks1_);
EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, NULL));
EXPECT_EQ(1, rtcp_mux_activated_callbacks1_);
// Only initiator supports mux. Should still have a separate RTCP channel.
EXPECT_TRUE(channel2_->SetLocalContent(&content, SdpType::kOffer, NULL));
content.set_rtcp_mux(false);
EXPECT_TRUE(
channel2_->SetRemoteContent(&content, SdpType::kPrAnswer, NULL));
EXPECT_TRUE(channel2_->SetRemoteContent(&content, SdpType::kAnswer, NULL));
EXPECT_EQ(0, rtcp_mux_activated_callbacks2_);
}
// Test that SetLocalContent and SetRemoteContent properly
// handles adding and removing StreamParams when the action is a full
// SdpType::kOffer / SdpType::kAnswer.
void TestChangeStreamParamsInContent() {
cricket::StreamParams stream1;
stream1.groupid = "group1";
stream1.id = "stream1";
stream1.ssrcs.push_back(kSsrc1);
stream1.cname = "stream1_cname";
cricket::StreamParams stream2;
stream2.groupid = "group1";
stream2.id = "stream2";
stream2.ssrcs.push_back(kSsrc2);
stream2.cname = "stream2_cname";
// Setup a call where channel 1 send |stream1| to channel 2.
CreateChannels(0, 0);
typename T::Content content1;
CreateContent(0, kPcmuCodec, kH264Codec, &content1);
content1.AddStream(stream1);
EXPECT_TRUE(channel1_->SetLocalContent(&content1, SdpType::kOffer, NULL));
EXPECT_TRUE(channel1_->Enable(true));
EXPECT_EQ(1u, media_channel1_->send_streams().size());
EXPECT_TRUE(channel2_->SetRemoteContent(&content1, SdpType::kOffer, NULL));
EXPECT_EQ(1u, media_channel2_->recv_streams().size());
ConnectFakeTransports();
// Channel 2 do not send anything.
typename T::Content content2;
CreateContent(0, kPcmuCodec, kH264Codec, &content2);
EXPECT_TRUE(channel1_->SetRemoteContent(&content2, SdpType::kAnswer, NULL));
EXPECT_EQ(0u, media_channel1_->recv_streams().size());
EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kAnswer, NULL));
EXPECT_TRUE(channel2_->Enable(true));
EXPECT_EQ(0u, media_channel2_->send_streams().size());
SendCustomRtp1(kSsrc1, 0);
WaitForThreads();
EXPECT_TRUE(CheckCustomRtp2(kSsrc1, 0));
// Let channel 2 update the content by sending |stream2| and enable SRTP.
typename T::Content content3;
CreateContent(0, kPcmuCodec, kH264Codec, &content3);
content3.AddStream(stream2);
EXPECT_TRUE(channel2_->SetLocalContent(&content3, SdpType::kOffer, NULL));
ASSERT_EQ(1u, media_channel2_->send_streams().size());
EXPECT_EQ(stream2, media_channel2_->send_streams()[0]);
EXPECT_TRUE(channel1_->SetRemoteContent(&content3, SdpType::kOffer, NULL));
ASSERT_EQ(1u, media_channel1_->recv_streams().size());
EXPECT_EQ(stream2, media_channel1_->recv_streams()[0]);
// Channel 1 replies but stop sending stream1.
typename T::Content content4;
CreateContent(0, kPcmuCodec, kH264Codec, &content4);
EXPECT_TRUE(channel1_->SetLocalContent(&content4, SdpType::kAnswer, NULL));
EXPECT_EQ(0u, media_channel1_->send_streams().size());
EXPECT_TRUE(channel2_->SetRemoteContent(&content4, SdpType::kAnswer, NULL));
EXPECT_EQ(0u, media_channel2_->recv_streams().size());
SendCustomRtp2(kSsrc2, 0);
WaitForThreads();
EXPECT_TRUE(CheckCustomRtp1(kSsrc2, 0));
}
// Test that we only start playout and sending at the right times.
void TestPlayoutAndSendingStates() {
CreateChannels(0, 0);
if (verify_playout_) {
EXPECT_FALSE(media_channel1_->playout());
}
EXPECT_FALSE(media_channel1_->sending());
if (verify_playout_) {
EXPECT_FALSE(media_channel2_->playout());
}
EXPECT_FALSE(media_channel2_->sending());
EXPECT_TRUE(channel1_->Enable(true));
if (verify_playout_) {
EXPECT_FALSE(media_channel1_->playout());
}
EXPECT_FALSE(media_channel1_->sending());
EXPECT_TRUE(channel1_->SetLocalContent(&local_media_content1_,
SdpType::kOffer, NULL));
if (verify_playout_) {
EXPECT_TRUE(media_channel1_->playout());
}
EXPECT_FALSE(media_channel1_->sending());
EXPECT_TRUE(channel2_->SetRemoteContent(&local_media_content1_,
SdpType::kOffer, NULL));
if (verify_playout_) {
EXPECT_FALSE(media_channel2_->playout());
}
EXPECT_FALSE(media_channel2_->sending());
EXPECT_TRUE(channel2_->SetLocalContent(&local_media_content2_,
SdpType::kAnswer, NULL));
if (verify_playout_) {
EXPECT_FALSE(media_channel2_->playout());
}
EXPECT_FALSE(media_channel2_->sending());
ConnectFakeTransports();
if (verify_playout_) {
EXPECT_TRUE(media_channel1_->playout());
}
EXPECT_FALSE(media_channel1_->sending());
if (verify_playout_) {
EXPECT_FALSE(media_channel2_->playout());
}
EXPECT_FALSE(media_channel2_->sending());
EXPECT_TRUE(channel2_->Enable(true));
if (verify_playout_) {
EXPECT_TRUE(media_channel2_->playout());
}
EXPECT_TRUE(media_channel2_->sending());
EXPECT_TRUE(channel1_->SetRemoteContent(&local_media_content2_,
SdpType::kAnswer, NULL));
if (verify_playout_) {
EXPECT_TRUE(media_channel1_->playout());
}
EXPECT_TRUE(media_channel1_->sending());
}
// Test that changing the MediaContentDirection in the local and remote
// session description start playout and sending at the right time.
void TestMediaContentDirection() {
CreateChannels(0, 0);
typename T::Content content1;
CreateContent(0, kPcmuCodec, kH264Codec, &content1);
typename T::Content content2;
CreateContent(0, kPcmuCodec, kH264Codec, &content2);
// Set |content2| to be InActive.
content2.set_direction(RtpTransceiverDirection::kInactive);
EXPECT_TRUE(channel1_->Enable(true));
EXPECT_TRUE(channel2_->Enable(true));
if (verify_playout_) {
EXPECT_FALSE(media_channel1_->playout());
}
EXPECT_FALSE(media_channel1_->sending());
if (verify_playout_) {
EXPECT_FALSE(media_channel2_->playout());
}
EXPECT_FALSE(media_channel2_->sending());
EXPECT_TRUE(channel1_->SetLocalContent(&content1, SdpType::kOffer, NULL));
EXPECT_TRUE(channel2_->SetRemoteContent(&content1, SdpType::kOffer, NULL));
EXPECT_TRUE(
channel2_->SetLocalContent(&content2, SdpType::kPrAnswer, NULL));
EXPECT_TRUE(
channel1_->SetRemoteContent(&content2, SdpType::kPrAnswer, NULL));
ConnectFakeTransports();
if (verify_playout_) {
EXPECT_TRUE(media_channel1_->playout());
}
EXPECT_FALSE(media_channel1_->sending()); // remote InActive
if (verify_playout_) {
EXPECT_FALSE(media_channel2_->playout()); // local InActive
}
EXPECT_FALSE(media_channel2_->sending()); // local InActive
// Update |content2| to be RecvOnly.
content2.set_direction(RtpTransceiverDirection::kRecvOnly);
EXPECT_TRUE(
channel2_->SetLocalContent(&content2, SdpType::kPrAnswer, NULL));
EXPECT_TRUE(
channel1_->SetRemoteContent(&content2, SdpType::kPrAnswer, NULL));
if (verify_playout_) {
EXPECT_TRUE(media_channel1_->playout());
}
EXPECT_TRUE(media_channel1_->sending());
if (verify_playout_) {
EXPECT_TRUE(media_channel2_->playout()); // local RecvOnly
}
EXPECT_FALSE(media_channel2_->sending()); // local RecvOnly
// Update |content2| to be SendRecv.
content2.set_direction(RtpTransceiverDirection::kSendRecv);
EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kAnswer, NULL));
EXPECT_TRUE(channel1_->SetRemoteContent(&content2, SdpType::kAnswer, NULL));
if (verify_playout_) {
EXPECT_TRUE(media_channel1_->playout());
}
EXPECT_TRUE(media_channel1_->sending());
if (verify_playout_) {
EXPECT_TRUE(media_channel2_->playout());
}
EXPECT_TRUE(media_channel2_->sending());
}
// Tests that when the transport channel signals a candidate pair change
// event, the media channel will receive a call on the network route change.
void TestNetworkRouteChanges() {
static constexpr uint16_t kLocalNetId = 1;
static constexpr uint16_t kRemoteNetId = 2;
static constexpr int kLastPacketId = 100;
// Ipv4(20) + UDP(8).
static constexpr int kTransportOverheadPerPacket = 28;
static constexpr int kSrtpOverheadPerPacket = 10;
CreateChannels(DTLS, DTLS);
SendInitiate();
typename T::MediaChannel* media_channel1 =
static_cast<typename T::MediaChannel*>(channel1_->media_channel());
ASSERT_TRUE(media_channel1);
// Need to wait for the threads before calling
// |set_num_network_route_changes| because the network route would be set
// when creating the channel.
WaitForThreads();
media_channel1->set_num_network_route_changes(0);
network_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
rtc::NetworkRoute network_route;
// The transport channel becomes disconnected.
fake_rtp_dtls_transport1_->ice_transport()->SignalNetworkRouteChanged(
absl::optional<rtc::NetworkRoute>(network_route));
});
WaitForThreads();
EXPECT_EQ(1, media_channel1->num_network_route_changes());
EXPECT_FALSE(media_channel1->last_network_route().connected);
media_channel1->set_num_network_route_changes(0);
network_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
rtc::NetworkRoute 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;
// The transport channel becomes connected.
fake_rtp_dtls_transport1_->ice_transport()->SignalNetworkRouteChanged(
absl::optional<rtc::NetworkRoute>(network_route));
});
WaitForThreads();
EXPECT_EQ(1, media_channel1->num_network_route_changes());
EXPECT_TRUE(media_channel1->last_network_route().connected);
EXPECT_EQ(kLocalNetId,
media_channel1->last_network_route().local_network_id);
EXPECT_EQ(kRemoteNetId,
media_channel1->last_network_route().remote_network_id);
EXPECT_EQ(kLastPacketId,
media_channel1->last_network_route().last_sent_packet_id);
EXPECT_EQ(kTransportOverheadPerPacket + kSrtpOverheadPerPacket,
media_channel1->transport_overhead_per_packet());
}
// Test setting up a call.
void TestCallSetup() {
CreateChannels(0, 0);
EXPECT_FALSE(channel1_->srtp_active());
EXPECT_TRUE(SendInitiate());
if (verify_playout_) {
EXPECT_TRUE(media_channel1_->playout());
}
EXPECT_FALSE(media_channel1_->sending());
EXPECT_TRUE(SendAccept());
EXPECT_FALSE(channel1_->srtp_active());
EXPECT_TRUE(media_channel1_->sending());
EXPECT_EQ(1U, media_channel1_->codecs().size());
if (verify_playout_) {
EXPECT_TRUE(media_channel2_->playout());
}
EXPECT_TRUE(media_channel2_->sending());
EXPECT_EQ(1U, media_channel2_->codecs().size());
}
// Test that we don't crash if packets are sent during call teardown
// when RTCP mux is enabled. This is a regression test against a specific
// race condition that would only occur when a RTCP packet was sent during
// teardown of a channel on which RTCP mux was enabled.
void TestCallTeardownRtcpMux() {
class LastWordMediaChannel : public T::MediaChannel {
public:
LastWordMediaChannel() : T::MediaChannel(NULL, typename T::Options()) {}
~LastWordMediaChannel() {
T::MediaChannel::SendRtp(kPcmuFrame, sizeof(kPcmuFrame),
rtc::PacketOptions());
T::MediaChannel::SendRtcp(kRtcpReport, sizeof(kRtcpReport));
}
};
CreateChannels(absl::make_unique<LastWordMediaChannel>(),
absl::make_unique<LastWordMediaChannel>(), RTCP_MUX,
RTCP_MUX);
EXPECT_TRUE(SendInitiate());
EXPECT_TRUE(SendAccept());
EXPECT_TRUE(Terminate());
}
// Send voice RTP data to the other side and ensure it gets there.
void SendRtpToRtp() {
CreateChannels(RTCP_MUX, RTCP_MUX);
EXPECT_TRUE(SendInitiate());
EXPECT_TRUE(SendAccept());
EXPECT_EQ(nullptr, channel1_->rtcp_packet_transport());
EXPECT_EQ(nullptr, channel2_->rtcp_packet_transport());
SendRtp1();
SendRtp2();
WaitForThreads();
EXPECT_TRUE(CheckRtp1());
EXPECT_TRUE(CheckRtp2());
EXPECT_TRUE(CheckNoRtp1());
EXPECT_TRUE(CheckNoRtp2());
}
void TestDeinit() {
CreateChannels(0, 0);
EXPECT_TRUE(SendInitiate());
EXPECT_TRUE(SendAccept());
SendRtp1();
SendRtp2();
SendRtcp1();
SendRtcp2();
// Do not wait, destroy channels.
channel1_.reset(nullptr);
channel2_.reset(nullptr);
}
// Check that RTCP can be transmitted between both sides.
void SendRtcpToRtcp() {
CreateChannels(0, 0);
EXPECT_TRUE(SendInitiate());
EXPECT_TRUE(SendAccept());
EXPECT_NE(nullptr, channel1_->rtcp_packet_transport());
EXPECT_NE(nullptr, channel2_->rtcp_packet_transport());
SendRtcp1();
SendRtcp2();
WaitForThreads();
EXPECT_TRUE(CheckRtcp1());
EXPECT_TRUE(CheckRtcp2());
EXPECT_TRUE(CheckNoRtcp1());
EXPECT_TRUE(CheckNoRtcp2());
}
void SendDtlsSrtpToDtlsSrtp(int flags1, int flags2) {
CreateChannels(flags1 | DTLS, flags2 | DTLS);
EXPECT_FALSE(channel1_->srtp_active());
EXPECT_FALSE(channel2_->srtp_active());
EXPECT_TRUE(SendInitiate());
WaitForThreads();
EXPECT_TRUE(channel1_->writable());
EXPECT_TRUE(channel2_->writable());
EXPECT_TRUE(SendAccept());
EXPECT_TRUE(channel1_->srtp_active());
EXPECT_TRUE(channel2_->srtp_active());
SendRtp1();
SendRtp2();
SendRtcp1();
SendRtcp2();
WaitForThreads();
EXPECT_TRUE(CheckRtp1());
EXPECT_TRUE(CheckRtp2());
EXPECT_TRUE(CheckNoRtp1());
EXPECT_TRUE(CheckNoRtp2());
EXPECT_TRUE(CheckRtcp1());
EXPECT_TRUE(CheckRtcp2());
EXPECT_TRUE(CheckNoRtcp1());
EXPECT_TRUE(CheckNoRtcp2());
}
// Test that we can send and receive early media when a provisional answer is
// sent and received. The test uses SRTP, RTCP mux and SSRC mux.
void SendEarlyMediaUsingRtcpMuxSrtp() {
int sequence_number1_1 = 0, sequence_number2_2 = 0;
CreateChannels(SSRC_MUX | RTCP_MUX | DTLS, SSRC_MUX | RTCP_MUX | DTLS);
EXPECT_TRUE(SendOffer());
EXPECT_TRUE(SendProvisionalAnswer());
EXPECT_TRUE(channel1_->srtp_active());
EXPECT_TRUE(channel2_->srtp_active());
EXPECT_EQ(nullptr, channel1_->rtcp_packet_transport());
EXPECT_EQ(nullptr, channel2_->rtcp_packet_transport());
WaitForThreads(); // Wait for 'sending' flag go through network thread.
SendCustomRtcp1(kSsrc1);
SendCustomRtp1(kSsrc1, ++sequence_number1_1);
WaitForThreads();
EXPECT_TRUE(CheckCustomRtcp2(kSsrc1));
EXPECT_TRUE(CheckCustomRtp2(kSsrc1, sequence_number1_1));
// Send packets from callee and verify that it is received.
SendCustomRtcp2(kSsrc2);
SendCustomRtp2(kSsrc2, ++sequence_number2_2);
WaitForThreads();
EXPECT_TRUE(CheckCustomRtcp1(kSsrc2));
EXPECT_TRUE(CheckCustomRtp1(kSsrc2, sequence_number2_2));
// Complete call setup and ensure everything is still OK.
EXPECT_TRUE(SendFinalAnswer());
EXPECT_TRUE(channel1_->srtp_active());
EXPECT_TRUE(channel2_->srtp_active());
SendCustomRtcp1(kSsrc1);
SendCustomRtp1(kSsrc1, ++sequence_number1_1);
SendCustomRtcp2(kSsrc2);
SendCustomRtp2(kSsrc2, ++sequence_number2_2);
WaitForThreads();
EXPECT_TRUE(CheckCustomRtcp2(kSsrc1));
EXPECT_TRUE(CheckCustomRtp2(kSsrc1, sequence_number1_1));
EXPECT_TRUE(CheckCustomRtcp1(kSsrc2));
EXPECT_TRUE(CheckCustomRtp1(kSsrc2, sequence_number2_2));
}
// Test that we properly send RTP without SRTP from a thread.
void SendRtpToRtpOnThread() {
CreateChannels(0, 0);
EXPECT_TRUE(SendInitiate());
EXPECT_TRUE(SendAccept());
ScopedCallThread send_rtp1([this] { SendRtp1(); });
ScopedCallThread send_rtp2([this] { SendRtp2(); });
ScopedCallThread send_rtcp1([this] { SendRtcp1(); });
ScopedCallThread send_rtcp2([this] { SendRtcp2(); });
rtc::Thread* involved_threads[] = {send_rtp1.thread(), send_rtp2.thread(),
send_rtcp1.thread(),
send_rtcp2.thread()};
WaitForThreads(involved_threads);
EXPECT_TRUE(CheckRtp1());
EXPECT_TRUE(CheckRtp2());
EXPECT_TRUE(CheckNoRtp1());
EXPECT_TRUE(CheckNoRtp2());
EXPECT_TRUE(CheckRtcp1());
EXPECT_TRUE(CheckRtcp2());
EXPECT_TRUE(CheckNoRtcp1());
EXPECT_TRUE(CheckNoRtcp2());
}
// Test that the mediachannel retains its sending state after the transport
// becomes non-writable.
void SendWithWritabilityLoss() {
CreateChannels(RTCP_MUX, RTCP_MUX);
EXPECT_TRUE(SendInitiate());
EXPECT_TRUE(SendAccept());
EXPECT_EQ(nullptr, channel1_->rtcp_packet_transport());
EXPECT_EQ(nullptr, channel2_->rtcp_packet_transport());
SendRtp1();
SendRtp2();
WaitForThreads();
EXPECT_TRUE(CheckRtp1());
EXPECT_TRUE(CheckRtp2());
EXPECT_TRUE(CheckNoRtp1());
EXPECT_TRUE(CheckNoRtp2());
// Lose writability, which should fail.
network_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
fake_rtp_dtls_transport1_->SetWritable(false);
});
SendRtp1();
SendRtp2();
WaitForThreads();
EXPECT_TRUE(CheckRtp1());
EXPECT_TRUE(CheckNoRtp2());
// Regain writability
network_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
fake_rtp_dtls_transport1_->SetWritable(true);
});
EXPECT_TRUE(media_channel1_->sending());
SendRtp1();
SendRtp2();
WaitForThreads();
EXPECT_TRUE(CheckRtp1());
EXPECT_TRUE(CheckRtp2());
EXPECT_TRUE(CheckNoRtp1());
EXPECT_TRUE(CheckNoRtp2());
// Lose writability completely
network_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
bool asymmetric = true;
fake_rtp_dtls_transport1_->SetDestination(nullptr, asymmetric);
});
EXPECT_TRUE(media_channel1_->sending());
// Should fail also.
SendRtp1();
SendRtp2();
WaitForThreads();
EXPECT_TRUE(CheckRtp1());
EXPECT_TRUE(CheckNoRtp2());
EXPECT_TRUE(CheckNoRtp1());
// Gain writability back
network_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
bool asymmetric = true;
fake_rtp_dtls_transport1_->SetDestination(fake_rtp_dtls_transport2_.get(),
asymmetric);
});
EXPECT_TRUE(media_channel1_->sending());
SendRtp1();
SendRtp2();
WaitForThreads();
EXPECT_TRUE(CheckRtp1());
EXPECT_TRUE(CheckRtp2());
EXPECT_TRUE(CheckNoRtp1());
EXPECT_TRUE(CheckNoRtp2());
}
void SendBundleToBundle(const int* pl_types,
int len,
bool rtcp_mux,
bool secure) {
ASSERT_EQ(2, len);
int sequence_number1_1 = 0, sequence_number2_2 = 0;
// Only pl_type1 was added to the bundle filter for both |channel1_|
// and |channel2_|.
int pl_type1 = pl_types[0];
int pl_type2 = pl_types[1];
int flags = SSRC_MUX;
if (secure)
flags |= DTLS;
if (rtcp_mux) {
flags |= RTCP_MUX;
}
CreateChannels(flags, flags);
EXPECT_TRUE(SendInitiate());
EXPECT_TRUE(SendAccept());
// Both channels can receive pl_type1 only.
SendCustomRtp1(kSsrc1, ++sequence_number1_1, pl_type1);
SendCustomRtp2(kSsrc2, ++sequence_number2_2, pl_type1);
WaitForThreads();
EXPECT_TRUE(CheckCustomRtp2(kSsrc1, sequence_number1_1, pl_type1));
EXPECT_TRUE(CheckCustomRtp1(kSsrc2, sequence_number2_2, pl_type1));
EXPECT_TRUE(CheckNoRtp1());
EXPECT_TRUE(CheckNoRtp2());
SendCustomRtp1(kSsrc3, ++sequence_number1_1, pl_type2);
SendCustomRtp2(kSsrc4, ++sequence_number2_2, pl_type2);
WaitForThreads();
EXPECT_FALSE(CheckCustomRtp2(kSsrc3, sequence_number1_1, pl_type2));
EXPECT_FALSE(CheckCustomRtp1(kSsrc4, sequence_number2_2, pl_type2));
// RTCP test
SendCustomRtcp1(kSsrc1);
SendCustomRtcp2(kSsrc2);
WaitForThreads();
EXPECT_TRUE(CheckCustomRtcp1(kSsrc2));
EXPECT_TRUE(CheckNoRtcp1());
EXPECT_TRUE(CheckCustomRtcp2(kSsrc1));
EXPECT_TRUE(CheckNoRtcp2());
SendCustomRtcp1(kSsrc2);
SendCustomRtcp2(kSsrc1);
WaitForThreads();
// Bundle filter shouldn't filter out any RTCP.
EXPECT_TRUE(CheckCustomRtcp1(kSsrc1));
EXPECT_TRUE(CheckCustomRtcp2(kSsrc2));
}
void TestSetContentFailure() {
CreateChannels(0, 0);
std::string err;
std::unique_ptr<typename T::Content> content(
CreateMediaContentWithStream(1));
media_channel1_->set_fail_set_recv_codecs(true);
EXPECT_FALSE(
channel1_->SetLocalContent(content.get(), SdpType::kOffer, &err));
EXPECT_FALSE(
channel1_->SetLocalContent(content.get(), SdpType::kAnswer, &err));
media_channel1_->set_fail_set_send_codecs(true);
EXPECT_FALSE(
channel1_->SetRemoteContent(content.get(), SdpType::kOffer, &err));
media_channel1_->set_fail_set_send_codecs(true);
EXPECT_FALSE(
channel1_->SetRemoteContent(content.get(), SdpType::kAnswer, &err));
}
void TestSendTwoOffers() {
CreateChannels(0, 0);
std::string err;
std::unique_ptr<typename T::Content> content1(
CreateMediaContentWithStream(1));
EXPECT_TRUE(
channel1_->SetLocalContent(content1.get(), SdpType::kOffer, &err));
EXPECT_TRUE(media_channel1_->HasSendStream(1));
std::unique_ptr<typename T::Content> content2(
CreateMediaContentWithStream(2));
EXPECT_TRUE(
channel1_->SetLocalContent(content2.get(), SdpType::kOffer, &err));
EXPECT_FALSE(media_channel1_->HasSendStream(1));
EXPECT_TRUE(media_channel1_->HasSendStream(2));
}
void TestReceiveTwoOffers() {
CreateChannels(0, 0);
std::string err;
std::unique_ptr<typename T::Content> content1(
CreateMediaContentWithStream(1));
EXPECT_TRUE(
channel1_->SetRemoteContent(content1.get(), SdpType::kOffer, &err));
EXPECT_TRUE(media_channel1_->HasRecvStream(1));
std::unique_ptr<typename T::Content> content2(
CreateMediaContentWithStream(2));
EXPECT_TRUE(
channel1_->SetRemoteContent(content2.get(), SdpType::kOffer, &err));
EXPECT_FALSE(media_channel1_->HasRecvStream(1));
EXPECT_TRUE(media_channel1_->HasRecvStream(2));
}
void TestSendPrAnswer() {
CreateChannels(0, 0);
std::string err;
// Receive offer
std::unique_ptr<typename T::Content> content1(
CreateMediaContentWithStream(1));
EXPECT_TRUE(
channel1_->SetRemoteContent(content1.get(), SdpType::kOffer, &err));
EXPECT_TRUE(media_channel1_->HasRecvStream(1));
// Send PR answer
std::unique_ptr<typename T::Content> content2(
CreateMediaContentWithStream(2));
EXPECT_TRUE(
channel1_->SetLocalContent(content2.get(), SdpType::kPrAnswer, &err));
EXPECT_TRUE(media_channel1_->HasRecvStream(1));
EXPECT_TRUE(media_channel1_->HasSendStream(2));
// Send answer
std::unique_ptr<typename T::Content> content3(
CreateMediaContentWithStream(3));
EXPECT_TRUE(
channel1_->SetLocalContent(content3.get(), SdpType::kAnswer, &err));
EXPECT_TRUE(media_channel1_->HasRecvStream(1));
EXPECT_FALSE(media_channel1_->HasSendStream(2));
EXPECT_TRUE(media_channel1_->HasSendStream(3));
}
void TestReceivePrAnswer() {
CreateChannels(0, 0);
std::string err;
// Send offer
std::unique_ptr<typename T::Content> content1(
CreateMediaContentWithStream(1));
EXPECT_TRUE(
channel1_->SetLocalContent(content1.get(), SdpType::kOffer, &err));
EXPECT_TRUE(media_channel1_->HasSendStream(1));
// Receive PR answer
std::unique_ptr<typename T::Content> content2(
CreateMediaContentWithStream(2));
EXPECT_TRUE(
channel1_->SetRemoteContent(content2.get(), SdpType::kPrAnswer, &err));
EXPECT_TRUE(media_channel1_->HasSendStream(1));
EXPECT_TRUE(media_channel1_->HasRecvStream(2));
// Receive answer
std::unique_ptr<typename T::Content> content3(
CreateMediaContentWithStream(3));
EXPECT_TRUE(
channel1_->SetRemoteContent(content3.get(), SdpType::kAnswer, &err));
EXPECT_TRUE(media_channel1_->HasSendStream(1));
EXPECT_FALSE(media_channel1_->HasRecvStream(2));
EXPECT_TRUE(media_channel1_->HasRecvStream(3));
}
void TestFlushRtcp() {
CreateChannels(0, 0);
EXPECT_TRUE(SendInitiate());
EXPECT_TRUE(SendAccept());
EXPECT_NE(nullptr, channel1_->rtcp_packet_transport());
EXPECT_NE(nullptr, channel2_->rtcp_packet_transport());
// Send RTCP1 from a different thread.
ScopedCallThread send_rtcp([this] { SendRtcp1(); });
// The sending message is only posted. channel2_ should be empty.
EXPECT_TRUE(CheckNoRtcp2());
rtc::Thread* wait_for[] = {send_rtcp.thread()};
WaitForThreads(wait_for); // Ensure rtcp was posted
// When channel1_ is deleted, the RTCP packet should be sent out to
// channel2_.
channel1_.reset();
WaitForThreads();
EXPECT_TRUE(CheckRtcp2());
}
void TestOnTransportReadyToSend() {
CreateChannels(0, 0);
EXPECT_FALSE(media_channel1_->ready_to_send());
channel1_->OnTransportReadyToSend(true);
WaitForThreads();
EXPECT_TRUE(media_channel1_->ready_to_send());
channel1_->OnTransportReadyToSend(false);
WaitForThreads();
EXPECT_FALSE(media_channel1_->ready_to_send());
}
bool SetRemoteContentWithBitrateLimit(int remote_limit) {
typename T::Content content;
CreateContent(0, kPcmuCodec, kH264Codec, &content);
content.set_bandwidth(remote_limit);
return channel1_->SetRemoteContent(&content, SdpType::kOffer, NULL);
}
webrtc::RtpParameters BitrateLimitedParameters(absl::optional<int> limit) {
webrtc::RtpParameters parameters;
webrtc::RtpEncodingParameters encoding;
encoding.max_bitrate_bps = std::move(limit);
parameters.encodings.push_back(encoding);
return parameters;
}
void VerifyMaxBitrate(const webrtc::RtpParameters& parameters,
absl::optional<int> expected_bitrate) {
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_EQ(expected_bitrate, parameters.encodings[0].max_bitrate_bps);
}
void DefaultMaxBitrateIsUnlimited() {
CreateChannels(0, 0);
EXPECT_TRUE(channel1_->SetLocalContent(&local_media_content1_,
SdpType::kOffer, NULL));
EXPECT_EQ(media_channel1_->max_bps(), -1);
VerifyMaxBitrate(media_channel1_->GetRtpSendParameters(kSsrc1),
absl::nullopt);
}
// Test that when a channel gets new RtpTransport with a call to
// |SetRtpTransport|, the socket options from the old RtpTransport is merged
// with the options on the new one.
// For example, audio and video may use separate socket options, but initially
// be unbundled, then later become bundled. When this happens, their preferred
// socket options should be merged to the underlying transport they share.
void SocketOptionsMergedOnSetTransport() {
constexpr int kSndBufSize = 4000;
constexpr int kRcvBufSize = 8000;
CreateChannels(DTLS, DTLS);
channel1_->SetOption(cricket::BaseChannel::ST_RTP,
rtc::Socket::Option::OPT_SNDBUF, kSndBufSize);
channel2_->SetOption(cricket::BaseChannel::ST_RTP,
rtc::Socket::Option::OPT_RCVBUF, kRcvBufSize);
new_rtp_transport_ = CreateDtlsSrtpTransport(
static_cast<DtlsTransportInternal*>(channel2_->rtp_packet_transport()),
static_cast<DtlsTransportInternal*>(
channel2_->rtcp_packet_transport()));
channel1_->SetRtpTransport(new_rtp_transport_.get());
int option_val;
ASSERT_TRUE(
static_cast<DtlsTransportInternal*>(channel1_->rtp_packet_transport())
->GetOption(rtc::Socket::Option::OPT_SNDBUF, &option_val));
EXPECT_EQ(kSndBufSize, option_val);
ASSERT_TRUE(
static_cast<DtlsTransportInternal*>(channel1_->rtp_packet_transport())
->GetOption(rtc::Socket::Option::OPT_RCVBUF, &option_val));
EXPECT_EQ(kRcvBufSize, option_val);
}
protected:
void WaitForThreads() { WaitForThreads(rtc::ArrayView<rtc::Thread*>()); }
static void ProcessThreadQueue(rtc::Thread* thread) {
RTC_DCHECK(thread->IsCurrent());
while (!thread->empty()) {
thread->ProcessMessages(0);
}
}
void WaitForThreads(rtc::ArrayView<rtc::Thread*> threads) {
// |threads| and current thread post packets to network thread.
for (rtc::Thread* thread : threads) {
thread->Invoke<void>(RTC_FROM_HERE,
[thread] { ProcessThreadQueue(thread); });
}
ProcessThreadQueue(rtc::Thread::Current());
// Network thread move them around and post back to worker = current thread.
if (!network_thread_->IsCurrent()) {
network_thread_->Invoke<void>(
RTC_FROM_HERE, [this] { ProcessThreadQueue(network_thread_); });
}
// Worker thread = current Thread process received messages.
ProcessThreadQueue(rtc::Thread::Current());
}
// TODO(pbos): Remove playout from all media channels and let renderers mute
// themselves.
const bool verify_playout_;
std::unique_ptr<rtc::Thread> network_thread_keeper_;
rtc::Thread* network_thread_;
std::unique_ptr<cricket::FakeDtlsTransport> fake_rtp_dtls_transport1_;
std::unique_ptr<cricket::FakeDtlsTransport> fake_rtcp_dtls_transport1_;
std::unique_ptr<cricket::FakeDtlsTransport> fake_rtp_dtls_transport2_;
std::unique_ptr<cricket::FakeDtlsTransport> fake_rtcp_dtls_transport2_;
std::unique_ptr<rtc::FakePacketTransport> fake_rtp_packet_transport1_;
std::unique_ptr<rtc::FakePacketTransport> fake_rtcp_packet_transport1_;
std::unique_ptr<rtc::FakePacketTransport> fake_rtp_packet_transport2_;
std::unique_ptr<rtc::FakePacketTransport> fake_rtcp_packet_transport2_;
std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport1_;
std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport2_;
std::unique_ptr<webrtc::RtpTransportInternal> new_rtp_transport_;
cricket::FakeMediaEngine media_engine_;
// The media channels are owned by the voice channel objects below.
typename T::MediaChannel* media_channel1_ = nullptr;
typename T::MediaChannel* media_channel2_ = nullptr;
std::unique_ptr<typename T::Channel> channel1_;
std::unique_ptr<typename T::Channel> channel2_;
typename T::Content local_media_content1_;
typename T::Content local_media_content2_;
typename T::Content remote_media_content1_;
typename T::Content remote_media_content2_;
// The RTP and RTCP packets to send in the tests.
rtc::Buffer rtp_packet_;
rtc::Buffer rtcp_packet_;
int rtcp_mux_activated_callbacks1_ = 0;
int rtcp_mux_activated_callbacks2_ = 0;
cricket::CandidatePairInterface* last_selected_candidate_pair_;
};
template <>
void ChannelTest<VoiceTraits>::CreateContent(
int flags,
const cricket::AudioCodec& audio_codec,
const cricket::VideoCodec& video_codec,
cricket::AudioContentDescription* audio) {
audio->AddCodec(audio_codec);
audio->set_rtcp_mux((flags & RTCP_MUX) != 0);
}
template <>
void ChannelTest<VoiceTraits>::CopyContent(
const cricket::AudioContentDescription& source,
cricket::AudioContentDescription* audio) {
*audio = source;
}
template <>
bool ChannelTest<VoiceTraits>::CodecMatches(const cricket::AudioCodec& c1,
const cricket::AudioCodec& c2) {
return c1.name == c2.name && c1.clockrate == c2.clockrate &&
c1.bitrate == c2.bitrate && c1.channels == c2.channels;
}
template <>
void ChannelTest<VoiceTraits>::AddLegacyStreamInContent(
uint32_t ssrc,
int flags,
cricket::AudioContentDescription* audio) {
audio->AddLegacyStream(ssrc);
}
class VoiceChannelSingleThreadTest : public ChannelTest<VoiceTraits> {
public:
typedef ChannelTest<VoiceTraits> Base;
VoiceChannelSingleThreadTest()
: Base(true, kPcmuFrame, kRtcpReport, NetworkIsWorker::Yes) {}
};
class VoiceChannelDoubleThreadTest : public ChannelTest<VoiceTraits> {
public:
typedef ChannelTest<VoiceTraits> Base;
VoiceChannelDoubleThreadTest()
: Base(true, kPcmuFrame, kRtcpReport, NetworkIsWorker::No) {}
};
class VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest
: public ChannelTest<VoiceTraits> {
public:
typedef ChannelTest<VoiceTraits> Base;
VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest()
: Base(true,
kPcmuFrameWithExtensions,
kRtcpReport,
NetworkIsWorker::Yes) {}
};
class VoiceChannelWithEncryptedRtpHeaderExtensionsDoubleThreadTest
: public ChannelTest<VoiceTraits> {
public:
typedef ChannelTest<VoiceTraits> Base;
VoiceChannelWithEncryptedRtpHeaderExtensionsDoubleThreadTest()
: Base(true, kPcmuFrameWithExtensions, kRtcpReport, NetworkIsWorker::No) {
}
};
// override to add NULL parameter
Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) Reason for revert: This CL just landed: https://codereview.chromium.org/1323243006/ Which fixes the FYI bots for the original CL, and breaks them for this revert. Original issue's description: > Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) > > Reason for revert: > This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step. > > Original issue's description: > > TransportController refactoring. > > > > Getting rid of TransportProxy, and in its place adding a > > TransportController class which will facilitate access to and manage > > the lifetimes of Transports. These Transports will now be accessed > > solely from the worker thread, simplifying their implementation. > > > > This refactoring also pulls Transport-related code out of BaseSession. > > Which means that BaseChannels will now rely on the TransportController > > interface to create channels, rather than BaseSession. > > > > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83 > > Cr-Commit-Position: refs/heads/master@{#10022} > > TBR=pthatcher@webrtc.org,deadbeef@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > > Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c > Cr-Commit-Position: refs/heads/master@{#10024} TBR=pthatcher@webrtc.org,torbjorng@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1361773005 Cr-Commit-Position: refs/heads/master@{#10036}
2015-09-23 11:50:27 -07:00
template <>
std::unique_ptr<cricket::VideoChannel> ChannelTest<VideoTraits>::CreateChannel(
rtc::Thread* worker_thread,
rtc::Thread* network_thread,
Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) Reason for revert: This CL just landed: https://codereview.chromium.org/1323243006/ Which fixes the FYI bots for the original CL, and breaks them for this revert. Original issue's description: > Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) > > Reason for revert: > This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step. > > Original issue's description: > > TransportController refactoring. > > > > Getting rid of TransportProxy, and in its place adding a > > TransportController class which will facilitate access to and manage > > the lifetimes of Transports. These Transports will now be accessed > > solely from the worker thread, simplifying their implementation. > > > > This refactoring also pulls Transport-related code out of BaseSession. > > Which means that BaseChannels will now rely on the TransportController > > interface to create channels, rather than BaseSession. > > > > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83 > > Cr-Commit-Position: refs/heads/master@{#10022} > > TBR=pthatcher@webrtc.org,deadbeef@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > > Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c > Cr-Commit-Position: refs/heads/master@{#10024} TBR=pthatcher@webrtc.org,torbjorng@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1361773005 Cr-Commit-Position: refs/heads/master@{#10036}
2015-09-23 11:50:27 -07:00
cricket::MediaEngineInterface* engine,
std::unique_ptr<cricket::FakeVideoMediaChannel> ch,
webrtc::RtpTransportInternal* rtp_transport,
int flags) {
rtc::Thread* signaling_thread = rtc::Thread::Current();
auto channel = absl::make_unique<cricket::VideoChannel>(
worker_thread, network_thread, signaling_thread, std::move(ch),
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class that only handles SRTP configuration to a more generic structure that can be used and extended for all per peer connection CryptoOptions that can be on a given PeerConnection. Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be accessed as crypto_options.srtp.whatever_option_name. This is more inline with other structures we have in WebRTC such as VideoConfig. As additional features are added over time this will allow the structure to remain compartmentalized and concerned components can only request a subset of the overall configuration structure e.g: void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); In addition to this it made little sense for sslstreamadapter.h to hold all Srtp related configuration options. The header has become loo large and takes on too many responsibilities and spilting this up will lead to more maintainable code going forward. This will be used in a future CL to enable configuration options for the newly supported Frame Crypto. Reland Fix: - cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional root level configuration. - peerconnectionfactory - If this optional is set will now overwrite the underyling value. This along with the other field will be deprecated once dependent projects are updated. TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org Bug: webrtc:9681 Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d Reviewed-on: https://webrtc-review.googlesource.com/c/105560 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25135}
2018-10-11 15:33:17 -07:00
cricket::CN_VIDEO, (flags & DTLS) != 0, webrtc::CryptoOptions());
Reland "Reland "Propagate media transport to media channel."" This is a reland of da65ed2adcfa57ff3288ce01c1602c973fcab00d Original change's description: > Reland "Propagate media transport to media channel." > > This reverts commit 37cf2455a420124b341ad06ac27fa3c4dbd29d3c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Propagate media transport to media channel." > > > > This reverts commit 8c16f745ab92cb6d305283e87fa8a661ae500ce4. > > > > Reason for revert: Breaks downstream project > > > > Original change's description: > > > Propagate media transport to media channel. > > > > > > 1. Pass media transport factory to JSEP transport controller. > > > 2. Pass media transport to voice media channel. > > > 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel. > > > > > > Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71 > > > Bug: webrtc:9719 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/105542 > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > > Reviewed-by: Peter Slatala <psla@webrtc.org> > > > Commit-Queue: Anton Sukhanov <sukhanov@google.com> > > > Cr-Commit-Position: refs/heads/master@{#25152} > > > > TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:9719 > > Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1 > > Reviewed-on: https://webrtc-review.googlesource.com/c/105840 > > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25154} > > TBR=steveanton@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com > > Change-Id: I505ff3451eae81573531faef155ff35d7f894022 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9719 > Reviewed-on: https://webrtc-review.googlesource.com/c/106500 > Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> > Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25220} Bug: webrtc:9719 Tbr: Steve Anton <steveanton@webrtc.org> Tbr: Niels Moller <nisse@webrtc.org> Change-Id: Ib45691ba8be9abb89ff8c6dac1861bdf59be4c8d Reviewed-on: https://webrtc-review.googlesource.com/c/106561 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25240}
2018-10-17 13:15:42 -07:00
channel->Init_w(rtp_transport, /*media_transport=*/nullptr);
return channel;
}
template <>
void ChannelTest<VideoTraits>::CreateContent(
int flags,
const cricket::AudioCodec& audio_codec,
const cricket::VideoCodec& video_codec,
cricket::VideoContentDescription* video) {
video->AddCodec(video_codec);
video->set_rtcp_mux((flags & RTCP_MUX) != 0);
}
template <>
void ChannelTest<VideoTraits>::CopyContent(
const cricket::VideoContentDescription& source,
cricket::VideoContentDescription* video) {
*video = source;
}
template <>
bool ChannelTest<VideoTraits>::CodecMatches(const cricket::VideoCodec& c1,
const cricket::VideoCodec& c2) {
return c1.name == c2.name;
}
template <>
void ChannelTest<VideoTraits>::AddLegacyStreamInContent(
uint32_t ssrc,
int flags,
cricket::VideoContentDescription* video) {
video->AddLegacyStream(ssrc);
}
class VideoChannelSingleThreadTest : public ChannelTest<VideoTraits> {
public:
typedef ChannelTest<VideoTraits> Base;
VideoChannelSingleThreadTest()
: Base(false, kH264Packet, kRtcpReport, NetworkIsWorker::Yes) {}
};
class VideoChannelDoubleThreadTest : public ChannelTest<VideoTraits> {
public:
typedef ChannelTest<VideoTraits> Base;
VideoChannelDoubleThreadTest()
: Base(false, kH264Packet, kRtcpReport, NetworkIsWorker::No) {}
};
TEST_F(VoiceChannelSingleThreadTest, TestInit) {
Base::TestInit();
EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
EXPECT_TRUE(media_channel1_->dtmf_info_queue().empty());
}
TEST_F(VoiceChannelSingleThreadTest, TestDeinit) {
Base::TestDeinit();
}
TEST_F(VoiceChannelSingleThreadTest, TestSetContents) {
Base::TestSetContents();
}
TEST_F(VoiceChannelSingleThreadTest, TestSetContentsExtmapAllowMixedAsCaller) {
Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/true);
}
TEST_F(VoiceChannelSingleThreadTest,
TestSetContentsExtmapAllowMixedNotSupportedAsCaller) {
Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/false);
}
TEST_F(VoiceChannelSingleThreadTest, TestSetContentsExtmapAllowMixedAsCallee) {
Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/true);
}
TEST_F(VoiceChannelSingleThreadTest,
TestSetContentsExtmapAllowMixedNotSupportedAsCallee) {
Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/false);
}
TEST_F(VoiceChannelSingleThreadTest, TestSetContentsNullOffer) {
Base::TestSetContentsNullOffer();
}
TEST_F(VoiceChannelSingleThreadTest, TestSetContentsRtcpMux) {
Base::TestSetContentsRtcpMux();
}
TEST_F(VoiceChannelSingleThreadTest, TestSetContentsRtcpMuxWithPrAnswer) {
Base::TestSetContentsRtcpMux();
}
TEST_F(VoiceChannelSingleThreadTest, TestChangeStreamParamsInContent) {
Base::TestChangeStreamParamsInContent();
}
TEST_F(VoiceChannelSingleThreadTest, TestPlayoutAndSendingStates) {
Base::TestPlayoutAndSendingStates();
}
TEST_F(VoiceChannelSingleThreadTest, TestMediaContentDirection) {
Base::TestMediaContentDirection();
}
TEST_F(VoiceChannelSingleThreadTest, TestNetworkRouteChanges) {
Base::TestNetworkRouteChanges();
}
TEST_F(VoiceChannelSingleThreadTest, TestCallSetup) {
Base::TestCallSetup();
}
TEST_F(VoiceChannelSingleThreadTest, TestCallTeardownRtcpMux) {
Base::TestCallTeardownRtcpMux();
}
TEST_F(VoiceChannelSingleThreadTest, SendRtpToRtp) {
Base::SendRtpToRtp();
}
TEST_F(VoiceChannelSingleThreadTest, SendRtcpToRtcp) {
Base::SendRtcpToRtcp();
}
TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtp) {
Base::SendDtlsSrtpToDtlsSrtp(0, 0);
}
TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
Base::SendDtlsSrtpToDtlsSrtp(RTCP_MUX, RTCP_MUX);
}
TEST_F(VoiceChannelSingleThreadTest, SendEarlyMediaUsingRtcpMuxSrtp) {
Base::SendEarlyMediaUsingRtcpMuxSrtp();
}
TEST_F(VoiceChannelSingleThreadTest, SendRtpToRtpOnThread) {
Base::SendRtpToRtpOnThread();
}
TEST_F(VoiceChannelSingleThreadTest, SendWithWritabilityLoss) {
Base::SendWithWritabilityLoss();
}
TEST_F(VoiceChannelSingleThreadTest, TestSetContentFailure) {
Base::TestSetContentFailure();
}
TEST_F(VoiceChannelSingleThreadTest, TestSendTwoOffers) {
Base::TestSendTwoOffers();
}
TEST_F(VoiceChannelSingleThreadTest, TestReceiveTwoOffers) {
Base::TestReceiveTwoOffers();
}
TEST_F(VoiceChannelSingleThreadTest, TestSendPrAnswer) {
Base::TestSendPrAnswer();
}
TEST_F(VoiceChannelSingleThreadTest, TestReceivePrAnswer) {
Base::TestReceivePrAnswer();
}
TEST_F(VoiceChannelSingleThreadTest, TestFlushRtcp) {
Base::TestFlushRtcp();
}
TEST_F(VoiceChannelSingleThreadTest, TestOnTransportReadyToSend) {
Base::TestOnTransportReadyToSend();
}
TEST_F(VoiceChannelSingleThreadTest, SendBundleToBundle) {
Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, false);
}
TEST_F(VoiceChannelSingleThreadTest, SendBundleToBundleSecure) {
Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, true);
}
TEST_F(VoiceChannelSingleThreadTest, SendBundleToBundleWithRtcpMux) {
Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, false);
}
TEST_F(VoiceChannelSingleThreadTest, SendBundleToBundleWithRtcpMuxSecure) {
Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, true);
}
TEST_F(VoiceChannelSingleThreadTest, DefaultMaxBitrateIsUnlimited) {
Base::DefaultMaxBitrateIsUnlimited();
}
TEST_F(VoiceChannelSingleThreadTest, SocketOptionsMergedOnSetTransport) {
Base::SocketOptionsMergedOnSetTransport();
}
// VoiceChannelDoubleThreadTest
TEST_F(VoiceChannelDoubleThreadTest, TestInit) {
Base::TestInit();
EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
EXPECT_TRUE(media_channel1_->dtmf_info_queue().empty());
}
TEST_F(VoiceChannelDoubleThreadTest, TestDeinit) {
Base::TestDeinit();
}
TEST_F(VoiceChannelDoubleThreadTest, TestSetContents) {
Base::TestSetContents();
}
TEST_F(VoiceChannelDoubleThreadTest, TestSetContentsExtmapAllowMixedAsCaller) {
Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/true);
}
TEST_F(VoiceChannelDoubleThreadTest,
TestSetContentsExtmapAllowMixedNotSupportedAsCaller) {
Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/false);
}
TEST_F(VoiceChannelDoubleThreadTest, TestSetContentsExtmapAllowMixedAsCallee) {
Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/true);
}
TEST_F(VoiceChannelDoubleThreadTest,
TestSetContentsExtmapAllowMixedNotSupportedAsCallee) {
Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/false);
}
TEST_F(VoiceChannelDoubleThreadTest, TestSetContentsNullOffer) {
Base::TestSetContentsNullOffer();
}
TEST_F(VoiceChannelDoubleThreadTest, TestSetContentsRtcpMux) {
Base::TestSetContentsRtcpMux();
}
TEST_F(VoiceChannelDoubleThreadTest, TestSetContentsRtcpMuxWithPrAnswer) {
Base::TestSetContentsRtcpMux();
}
TEST_F(VoiceChannelDoubleThreadTest, TestChangeStreamParamsInContent) {
Base::TestChangeStreamParamsInContent();
}
TEST_F(VoiceChannelDoubleThreadTest, TestPlayoutAndSendingStates) {
Base::TestPlayoutAndSendingStates();
}
TEST_F(VoiceChannelDoubleThreadTest, TestMediaContentDirection) {
Base::TestMediaContentDirection();
}
TEST_F(VoiceChannelDoubleThreadTest, TestNetworkRouteChanges) {
Base::TestNetworkRouteChanges();
}
TEST_F(VoiceChannelDoubleThreadTest, TestCallSetup) {
Base::TestCallSetup();
}
TEST_F(VoiceChannelDoubleThreadTest, TestCallTeardownRtcpMux) {
Base::TestCallTeardownRtcpMux();
}
TEST_F(VoiceChannelDoubleThreadTest, SendRtpToRtp) {
Base::SendRtpToRtp();
}
TEST_F(VoiceChannelDoubleThreadTest, SendRtcpToRtcp) {
Base::SendRtcpToRtcp();
}
TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtp) {
Base::SendDtlsSrtpToDtlsSrtp(0, 0);
}
TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
Base::SendDtlsSrtpToDtlsSrtp(RTCP_MUX, RTCP_MUX);
}
TEST_F(VoiceChannelDoubleThreadTest, SendEarlyMediaUsingRtcpMuxSrtp) {
Base::SendEarlyMediaUsingRtcpMuxSrtp();
}
TEST_F(VoiceChannelDoubleThreadTest, SendRtpToRtpOnThread) {
Base::SendRtpToRtpOnThread();
}
TEST_F(VoiceChannelDoubleThreadTest, SendWithWritabilityLoss) {
Base::SendWithWritabilityLoss();
}
TEST_F(VoiceChannelDoubleThreadTest, TestSetContentFailure) {
Base::TestSetContentFailure();
}
TEST_F(VoiceChannelDoubleThreadTest, TestSendTwoOffers) {
Base::TestSendTwoOffers();
}
TEST_F(VoiceChannelDoubleThreadTest, TestReceiveTwoOffers) {
Base::TestReceiveTwoOffers();
}
TEST_F(VoiceChannelDoubleThreadTest, TestSendPrAnswer) {
Base::TestSendPrAnswer();
}
TEST_F(VoiceChannelDoubleThreadTest, TestReceivePrAnswer) {
Base::TestReceivePrAnswer();
}
TEST_F(VoiceChannelDoubleThreadTest, TestFlushRtcp) {
Base::TestFlushRtcp();
}
TEST_F(VoiceChannelDoubleThreadTest, TestOnTransportReadyToSend) {
Base::TestOnTransportReadyToSend();
}
TEST_F(VoiceChannelDoubleThreadTest, SendBundleToBundle) {
Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, false);
}
TEST_F(VoiceChannelDoubleThreadTest, SendBundleToBundleSecure) {
Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, true);
}
TEST_F(VoiceChannelDoubleThreadTest, SendBundleToBundleWithRtcpMux) {
Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, false);
}
TEST_F(VoiceChannelDoubleThreadTest, SendBundleToBundleWithRtcpMuxSecure) {
Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, true);
}
TEST_F(VoiceChannelDoubleThreadTest, DefaultMaxBitrateIsUnlimited) {
Base::DefaultMaxBitrateIsUnlimited();
}
TEST_F(VoiceChannelDoubleThreadTest, SocketOptionsMergedOnSetTransport) {
Base::SocketOptionsMergedOnSetTransport();
}
// VideoChannelSingleThreadTest
TEST_F(VideoChannelSingleThreadTest, TestInit) {
Base::TestInit();
}
TEST_F(VideoChannelSingleThreadTest, TestDeinit) {
Base::TestDeinit();
}
TEST_F(VideoChannelSingleThreadTest, TestSetContents) {
Base::TestSetContents();
}
TEST_F(VideoChannelSingleThreadTest, TestSetContentsExtmapAllowMixedAsCaller) {
Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/true);
}
TEST_F(VideoChannelSingleThreadTest,
TestSetContentsExtmapAllowMixedNotSupportedAsCaller) {
Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/false);
}
TEST_F(VideoChannelSingleThreadTest, TestSetContentsExtmapAllowMixedAsCallee) {
Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/true);
}
TEST_F(VideoChannelSingleThreadTest,
TestSetContentsExtmapAllowMixedNotSupportedAsCallee) {
Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/false);
}
TEST_F(VideoChannelSingleThreadTest, TestSetContentsNullOffer) {
Base::TestSetContentsNullOffer();
}
TEST_F(VideoChannelSingleThreadTest, TestSetContentsRtcpMux) {
Base::TestSetContentsRtcpMux();
}
TEST_F(VideoChannelSingleThreadTest, TestSetContentsRtcpMuxWithPrAnswer) {
Base::TestSetContentsRtcpMux();
}
TEST_F(VideoChannelSingleThreadTest, TestChangeStreamParamsInContent) {
Base::TestChangeStreamParamsInContent();
}
TEST_F(VideoChannelSingleThreadTest, TestPlayoutAndSendingStates) {
Base::TestPlayoutAndSendingStates();
}
TEST_F(VideoChannelSingleThreadTest, TestMediaContentDirection) {
Base::TestMediaContentDirection();
}
TEST_F(VideoChannelSingleThreadTest, TestNetworkRouteChanges) {
Base::TestNetworkRouteChanges();
}
TEST_F(VideoChannelSingleThreadTest, TestCallSetup) {
Base::TestCallSetup();
}
TEST_F(VideoChannelSingleThreadTest, TestCallTeardownRtcpMux) {
Base::TestCallTeardownRtcpMux();
}
TEST_F(VideoChannelSingleThreadTest, SendRtpToRtp) {
Base::SendRtpToRtp();
}
TEST_F(VideoChannelSingleThreadTest, SendRtcpToRtcp) {
Base::SendRtcpToRtcp();
}
TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtp) {
Base::SendDtlsSrtpToDtlsSrtp(0, 0);
}
TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
Base::SendDtlsSrtpToDtlsSrtp(RTCP_MUX, RTCP_MUX);
}
TEST_F(VideoChannelSingleThreadTest, SendEarlyMediaUsingRtcpMuxSrtp) {
Base::SendEarlyMediaUsingRtcpMuxSrtp();
}
TEST_F(VideoChannelSingleThreadTest, SendRtpToRtpOnThread) {
Base::SendRtpToRtpOnThread();
}
TEST_F(VideoChannelSingleThreadTest, SendWithWritabilityLoss) {
Base::SendWithWritabilityLoss();
}
TEST_F(VideoChannelSingleThreadTest, TestSetContentFailure) {
Base::TestSetContentFailure();
}
TEST_F(VideoChannelSingleThreadTest, TestSendTwoOffers) {
Base::TestSendTwoOffers();
}
TEST_F(VideoChannelSingleThreadTest, TestReceiveTwoOffers) {
Base::TestReceiveTwoOffers();
}
TEST_F(VideoChannelSingleThreadTest, TestSendPrAnswer) {
Base::TestSendPrAnswer();
}
TEST_F(VideoChannelSingleThreadTest, TestReceivePrAnswer) {
Base::TestReceivePrAnswer();
}
TEST_F(VideoChannelSingleThreadTest, TestFlushRtcp) {
Base::TestFlushRtcp();
}
TEST_F(VideoChannelSingleThreadTest, SendBundleToBundle) {
Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), false, false);
}
TEST_F(VideoChannelSingleThreadTest, SendBundleToBundleSecure) {
Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), false, true);
}
TEST_F(VideoChannelSingleThreadTest, SendBundleToBundleWithRtcpMux) {
Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), true, false);
}
TEST_F(VideoChannelSingleThreadTest, SendBundleToBundleWithRtcpMuxSecure) {
Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), true, true);
}
TEST_F(VideoChannelSingleThreadTest, TestOnTransportReadyToSend) {
Base::TestOnTransportReadyToSend();
}
TEST_F(VideoChannelSingleThreadTest, DefaultMaxBitrateIsUnlimited) {
Base::DefaultMaxBitrateIsUnlimited();
}
TEST_F(VideoChannelSingleThreadTest, SocketOptionsMergedOnSetTransport) {
Base::SocketOptionsMergedOnSetTransport();
}
// VideoChannelDoubleThreadTest
TEST_F(VideoChannelDoubleThreadTest, TestInit) {
Base::TestInit();
}
TEST_F(VideoChannelDoubleThreadTest, TestDeinit) {
Base::TestDeinit();
}
TEST_F(VideoChannelDoubleThreadTest, TestSetContents) {
Base::TestSetContents();
}
TEST_F(VideoChannelDoubleThreadTest, TestSetContentsExtmapAllowMixedAsCaller) {
Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/true);
}
TEST_F(VideoChannelDoubleThreadTest,
TestSetContentsExtmapAllowMixedNotSupportedAsCaller) {
Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/false);
}
TEST_F(VideoChannelDoubleThreadTest, TestSetContentsExtmapAllowMixedAsCallee) {
Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/true);
}
TEST_F(VideoChannelDoubleThreadTest,
TestSetContentsExtmapAllowMixedNotSupportedAsCallee) {
Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/false);
}
TEST_F(VideoChannelDoubleThreadTest, TestSetContentsNullOffer) {
Base::TestSetContentsNullOffer();
}
TEST_F(VideoChannelDoubleThreadTest, TestSetContentsRtcpMux) {
Base::TestSetContentsRtcpMux();
}
TEST_F(VideoChannelDoubleThreadTest, TestSetContentsRtcpMuxWithPrAnswer) {
Base::TestSetContentsRtcpMux();
}
TEST_F(VideoChannelDoubleThreadTest, TestChangeStreamParamsInContent) {
Base::TestChangeStreamParamsInContent();
}
TEST_F(VideoChannelDoubleThreadTest, TestPlayoutAndSendingStates) {
Base::TestPlayoutAndSendingStates();
}
TEST_F(VideoChannelDoubleThreadTest, TestMediaContentDirection) {
Base::TestMediaContentDirection();
}
TEST_F(VideoChannelDoubleThreadTest, TestNetworkRouteChanges) {
Base::TestNetworkRouteChanges();
}
TEST_F(VideoChannelDoubleThreadTest, TestCallSetup) {
Base::TestCallSetup();
}
TEST_F(VideoChannelDoubleThreadTest, TestCallTeardownRtcpMux) {
Base::TestCallTeardownRtcpMux();
}
TEST_F(VideoChannelDoubleThreadTest, SendRtpToRtp) {
Base::SendRtpToRtp();
}
TEST_F(VideoChannelDoubleThreadTest, SendRtcpToRtcp) {
Base::SendRtcpToRtcp();
}
TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtp) {
Base::SendDtlsSrtpToDtlsSrtp(0, 0);
}
TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
Base::SendDtlsSrtpToDtlsSrtp(RTCP_MUX, RTCP_MUX);
}
TEST_F(VideoChannelDoubleThreadTest, SendEarlyMediaUsingRtcpMuxSrtp) {
Base::SendEarlyMediaUsingRtcpMuxSrtp();
}
TEST_F(VideoChannelDoubleThreadTest, SendRtpToRtpOnThread) {
Base::SendRtpToRtpOnThread();
}
TEST_F(VideoChannelDoubleThreadTest, SendWithWritabilityLoss) {
Base::SendWithWritabilityLoss();
}
TEST_F(VideoChannelDoubleThreadTest, TestSetContentFailure) {
Base::TestSetContentFailure();
}
TEST_F(VideoChannelDoubleThreadTest, TestSendTwoOffers) {
Base::TestSendTwoOffers();
}
TEST_F(VideoChannelDoubleThreadTest, TestReceiveTwoOffers) {
Base::TestReceiveTwoOffers();
}
TEST_F(VideoChannelDoubleThreadTest, TestSendPrAnswer) {
Base::TestSendPrAnswer();
}
TEST_F(VideoChannelDoubleThreadTest, TestReceivePrAnswer) {
Base::TestReceivePrAnswer();
}
TEST_F(VideoChannelDoubleThreadTest, TestFlushRtcp) {
Base::TestFlushRtcp();
}
TEST_F(VideoChannelDoubleThreadTest, SendBundleToBundle) {
Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), false, false);
}
TEST_F(VideoChannelDoubleThreadTest, SendBundleToBundleSecure) {
Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), false, true);
}
TEST_F(VideoChannelDoubleThreadTest, SendBundleToBundleWithRtcpMux) {
Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), true, false);
}
TEST_F(VideoChannelDoubleThreadTest, SendBundleToBundleWithRtcpMuxSecure) {
Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), true, true);
}
TEST_F(VideoChannelDoubleThreadTest, TestOnTransportReadyToSend) {
Base::TestOnTransportReadyToSend();
}
TEST_F(VideoChannelDoubleThreadTest, DefaultMaxBitrateIsUnlimited) {
Base::DefaultMaxBitrateIsUnlimited();
}
TEST_F(VideoChannelDoubleThreadTest, SocketOptionsMergedOnSetTransport) {
Base::SocketOptionsMergedOnSetTransport();
}
// RtpDataChannelSingleThreadTest
class RtpDataChannelSingleThreadTest : public ChannelTest<DataTraits> {
public:
typedef ChannelTest<DataTraits> Base;
RtpDataChannelSingleThreadTest()
: Base(true, kDataPacket, kRtcpReport, NetworkIsWorker::Yes) {}
};
// RtpDataChannelDoubleThreadTest
class RtpDataChannelDoubleThreadTest : public ChannelTest<DataTraits> {
public:
typedef ChannelTest<DataTraits> Base;
RtpDataChannelDoubleThreadTest()
: Base(true, kDataPacket, kRtcpReport, NetworkIsWorker::No) {}
};
// Override to avoid engine channel parameter.
template <>
std::unique_ptr<cricket::RtpDataChannel> ChannelTest<DataTraits>::CreateChannel(
rtc::Thread* worker_thread,
rtc::Thread* network_thread,
cricket::MediaEngineInterface* engine,
std::unique_ptr<cricket::FakeDataMediaChannel> ch,
webrtc::RtpTransportInternal* rtp_transport,
int flags) {
rtc::Thread* signaling_thread = rtc::Thread::Current();
auto channel = absl::make_unique<cricket::RtpDataChannel>(
worker_thread, network_thread, signaling_thread, std::move(ch),
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class that only handles SRTP configuration to a more generic structure that can be used and extended for all per peer connection CryptoOptions that can be on a given PeerConnection. Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be accessed as crypto_options.srtp.whatever_option_name. This is more inline with other structures we have in WebRTC such as VideoConfig. As additional features are added over time this will allow the structure to remain compartmentalized and concerned components can only request a subset of the overall configuration structure e.g: void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); In addition to this it made little sense for sslstreamadapter.h to hold all Srtp related configuration options. The header has become loo large and takes on too many responsibilities and spilting this up will lead to more maintainable code going forward. This will be used in a future CL to enable configuration options for the newly supported Frame Crypto. Reland Fix: - cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional root level configuration. - peerconnectionfactory - If this optional is set will now overwrite the underyling value. This along with the other field will be deprecated once dependent projects are updated. TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org Bug: webrtc:9681 Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d Reviewed-on: https://webrtc-review.googlesource.com/c/105560 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25135}
2018-10-11 15:33:17 -07:00
cricket::CN_DATA, (flags & DTLS) != 0, webrtc::CryptoOptions());
channel->Init_w(rtp_transport);
return channel;
}
template <>
void ChannelTest<DataTraits>::CreateContent(
int flags,
const cricket::AudioCodec& audio_codec,
const cricket::VideoCodec& video_codec,
cricket::DataContentDescription* data) {
data->AddCodec(kGoogleDataCodec);
data->set_rtcp_mux((flags & RTCP_MUX) != 0);
}
template <>
void ChannelTest<DataTraits>::CopyContent(
const cricket::DataContentDescription& source,
cricket::DataContentDescription* data) {
*data = source;
}
template <>
bool ChannelTest<DataTraits>::CodecMatches(const cricket::DataCodec& c1,
const cricket::DataCodec& c2) {
return c1.name == c2.name;
}
template <>
void ChannelTest<DataTraits>::AddLegacyStreamInContent(
uint32_t ssrc,
int flags,
cricket::DataContentDescription* data) {
data->AddLegacyStream(ssrc);
}
TEST_F(RtpDataChannelSingleThreadTest, TestInit) {
Base::TestInit();
EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
}
TEST_F(RtpDataChannelSingleThreadTest, TestDeinit) {
Base::TestDeinit();
}
TEST_F(RtpDataChannelSingleThreadTest, TestSetContents) {
Base::TestSetContents();
}
TEST_F(RtpDataChannelSingleThreadTest, TestSetContentsNullOffer) {
Base::TestSetContentsNullOffer();
}
TEST_F(RtpDataChannelSingleThreadTest, TestSetContentsRtcpMux) {
Base::TestSetContentsRtcpMux();
}
TEST_F(RtpDataChannelSingleThreadTest, TestChangeStreamParamsInContent) {
Base::TestChangeStreamParamsInContent();
}
TEST_F(RtpDataChannelSingleThreadTest, TestPlayoutAndSendingStates) {
Base::TestPlayoutAndSendingStates();
}
TEST_F(RtpDataChannelSingleThreadTest, TestMediaContentDirection) {
Base::TestMediaContentDirection();
}
TEST_F(RtpDataChannelSingleThreadTest, TestCallSetup) {
Base::TestCallSetup();
}
TEST_F(RtpDataChannelSingleThreadTest, TestCallTeardownRtcpMux) {
Base::TestCallTeardownRtcpMux();
}
TEST_F(RtpDataChannelSingleThreadTest, TestOnTransportReadyToSend) {
Base::TestOnTransportReadyToSend();
}
TEST_F(RtpDataChannelSingleThreadTest, SendRtpToRtp) {
Base::SendRtpToRtp();
}
TEST_F(RtpDataChannelSingleThreadTest, SendRtcpToRtcp) {
Base::SendRtcpToRtcp();
}
TEST_F(RtpDataChannelSingleThreadTest, SendRtpToRtpOnThread) {
Base::SendRtpToRtpOnThread();
}
TEST_F(RtpDataChannelSingleThreadTest, SendWithWritabilityLoss) {
Base::SendWithWritabilityLoss();
}
TEST_F(RtpDataChannelSingleThreadTest, SocketOptionsMergedOnSetTransport) {
Base::SocketOptionsMergedOnSetTransport();
}
TEST_F(RtpDataChannelSingleThreadTest, TestSendData) {
CreateChannels(0, 0);
EXPECT_TRUE(SendInitiate());
EXPECT_TRUE(SendAccept());
cricket::SendDataParams params;
params.ssrc = 42;
unsigned char data[] = {'f', 'o', 'o'};
rtc::CopyOnWriteBuffer payload(data, 3);
cricket::SendDataResult result;
ASSERT_TRUE(media_channel1_->SendData(params, payload, &result));
EXPECT_EQ(params.ssrc, media_channel1_->last_sent_data_params().ssrc);
EXPECT_EQ("foo", media_channel1_->last_sent_data());
}
TEST_F(RtpDataChannelDoubleThreadTest, TestInit) {
Base::TestInit();
EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
}
TEST_F(RtpDataChannelDoubleThreadTest, TestDeinit) {
Base::TestDeinit();
}
TEST_F(RtpDataChannelDoubleThreadTest, TestSetContents) {
Base::TestSetContents();
}
TEST_F(RtpDataChannelDoubleThreadTest, TestSetContentsNullOffer) {
Base::TestSetContentsNullOffer();
}
TEST_F(RtpDataChannelDoubleThreadTest, TestSetContentsRtcpMux) {
Base::TestSetContentsRtcpMux();
}
TEST_F(RtpDataChannelDoubleThreadTest, TestChangeStreamParamsInContent) {
Base::TestChangeStreamParamsInContent();
}
TEST_F(RtpDataChannelDoubleThreadTest, TestPlayoutAndSendingStates) {
Base::TestPlayoutAndSendingStates();
}
TEST_F(RtpDataChannelDoubleThreadTest, TestMediaContentDirection) {
Base::TestMediaContentDirection();
}
TEST_F(RtpDataChannelDoubleThreadTest, TestCallSetup) {
Base::TestCallSetup();
}
TEST_F(RtpDataChannelDoubleThreadTest, TestCallTeardownRtcpMux) {
Base::TestCallTeardownRtcpMux();
}
TEST_F(RtpDataChannelDoubleThreadTest, TestOnTransportReadyToSend) {
Base::TestOnTransportReadyToSend();
}
TEST_F(RtpDataChannelDoubleThreadTest, SendRtpToRtp) {
Base::SendRtpToRtp();
}
TEST_F(RtpDataChannelDoubleThreadTest, SendRtcpToRtcp) {
Base::SendRtcpToRtcp();
}
TEST_F(RtpDataChannelDoubleThreadTest, SendRtpToRtpOnThread) {
Base::SendRtpToRtpOnThread();
}
TEST_F(RtpDataChannelDoubleThreadTest, SendWithWritabilityLoss) {
Base::SendWithWritabilityLoss();
}
TEST_F(RtpDataChannelDoubleThreadTest, SocketOptionsMergedOnSetTransport) {
Base::SocketOptionsMergedOnSetTransport();
}
TEST_F(RtpDataChannelDoubleThreadTest, TestSendData) {
CreateChannels(0, 0);
EXPECT_TRUE(SendInitiate());
EXPECT_TRUE(SendAccept());
cricket::SendDataParams params;
params.ssrc = 42;
unsigned char data[] = {'f', 'o', 'o'};
rtc::CopyOnWriteBuffer payload(data, 3);
cricket::SendDataResult result;
ASSERT_TRUE(media_channel1_->SendData(params, payload, &result));
EXPECT_EQ(params.ssrc, media_channel1_->last_sent_data_params().ssrc);
EXPECT_EQ("foo", media_channel1_->last_sent_data());
}
// TODO(pthatcher): TestSetReceiver?