mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 14:50:39 +01:00

WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
214 lines
7.9 KiB
C++
214 lines
7.9 KiB
C++
/*
|
|
* Copyright 2008 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 "pc/channel_manager.h"
|
|
|
|
#include <memory>
|
|
|
|
#include "api/rtc_error.h"
|
|
#include "api/test/fake_media_transport.h"
|
|
#include "api/transport/media/media_transport_config.h"
|
|
#include "api/video/builtin_video_bitrate_allocator_factory.h"
|
|
#include "media/base/fake_media_engine.h"
|
|
#include "media/base/test_utils.h"
|
|
#include "media/engine/fake_webrtc_call.h"
|
|
#include "p2p/base/dtls_transport_internal.h"
|
|
#include "p2p/base/fake_dtls_transport.h"
|
|
#include "p2p/base/p2p_constants.h"
|
|
#include "p2p/base/packet_transport_internal.h"
|
|
#include "pc/dtls_srtp_transport.h"
|
|
#include "rtc_base/checks.h"
|
|
#include "rtc_base/thread.h"
|
|
#include "test/gtest.h"
|
|
|
|
namespace {
|
|
const bool kDefaultSrtpRequired = true;
|
|
}
|
|
|
|
namespace cricket {
|
|
|
|
static const AudioCodec kAudioCodecs[] = {
|
|
AudioCodec(97, "voice", 1, 2, 3),
|
|
AudioCodec(111, "OPUS", 48000, 32000, 2),
|
|
};
|
|
|
|
static const VideoCodec kVideoCodecs[] = {
|
|
VideoCodec(99, "H264"),
|
|
VideoCodec(100, "VP8"),
|
|
VideoCodec(96, "rtx"),
|
|
};
|
|
|
|
class ChannelManagerTest : public ::testing::Test {
|
|
protected:
|
|
ChannelManagerTest()
|
|
: network_(rtc::Thread::CreateWithSocketServer()),
|
|
worker_(rtc::Thread::Create()),
|
|
video_bitrate_allocator_factory_(
|
|
webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
|
|
fme_(new cricket::FakeMediaEngine()),
|
|
fdme_(new cricket::FakeDataEngine()),
|
|
cm_(new cricket::ChannelManager(
|
|
std::unique_ptr<MediaEngineInterface>(fme_),
|
|
std::unique_ptr<DataEngineInterface>(fdme_),
|
|
rtc::Thread::Current(),
|
|
rtc::Thread::Current())),
|
|
fake_call_() {
|
|
fme_->SetAudioCodecs(MAKE_VECTOR(kAudioCodecs));
|
|
fme_->SetVideoCodecs(MAKE_VECTOR(kVideoCodecs));
|
|
}
|
|
|
|
std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() {
|
|
rtp_dtls_transport_ = std::make_unique<FakeDtlsTransport>(
|
|
"fake_dtls_transport", cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
|
|
/*rtcp_mux_required=*/true);
|
|
dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport_.get(),
|
|
/*rtcp_dtls_transport=*/nullptr);
|
|
return dtls_srtp_transport;
|
|
}
|
|
|
|
std::unique_ptr<webrtc::MediaTransportInterface> CreateMediaTransport(
|
|
rtc::PacketTransportInternal* packet_transport) {
|
|
webrtc::MediaTransportSettings settings;
|
|
settings.is_caller = true;
|
|
auto media_transport_result =
|
|
fake_media_transport_factory_.CreateMediaTransport(
|
|
packet_transport, network_.get(),
|
|
/*is_caller=*/settings);
|
|
RTC_CHECK(media_transport_result.ok());
|
|
return media_transport_result.MoveValue();
|
|
}
|
|
|
|
void TestCreateDestroyChannels(
|
|
webrtc::RtpTransportInternal* rtp_transport,
|
|
webrtc::MediaTransportConfig media_transport_config) {
|
|
cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
|
|
&fake_call_, cricket::MediaConfig(), rtp_transport,
|
|
media_transport_config, rtc::Thread::Current(), cricket::CN_AUDIO,
|
|
kDefaultSrtpRequired, webrtc::CryptoOptions(), &ssrc_generator_,
|
|
AudioOptions());
|
|
EXPECT_TRUE(voice_channel != nullptr);
|
|
cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
|
|
&fake_call_, cricket::MediaConfig(), rtp_transport,
|
|
media_transport_config, rtc::Thread::Current(), cricket::CN_VIDEO,
|
|
kDefaultSrtpRequired, webrtc::CryptoOptions(), &ssrc_generator_,
|
|
VideoOptions(), video_bitrate_allocator_factory_.get());
|
|
EXPECT_TRUE(video_channel != nullptr);
|
|
cricket::RtpDataChannel* rtp_data_channel = cm_->CreateRtpDataChannel(
|
|
cricket::MediaConfig(), rtp_transport, rtc::Thread::Current(),
|
|
cricket::CN_DATA, kDefaultSrtpRequired, webrtc::CryptoOptions(),
|
|
&ssrc_generator_);
|
|
EXPECT_TRUE(rtp_data_channel != nullptr);
|
|
cm_->DestroyVideoChannel(video_channel);
|
|
cm_->DestroyVoiceChannel(voice_channel);
|
|
cm_->DestroyRtpDataChannel(rtp_data_channel);
|
|
cm_->Terminate();
|
|
}
|
|
|
|
std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport_;
|
|
std::unique_ptr<rtc::Thread> network_;
|
|
std::unique_ptr<rtc::Thread> worker_;
|
|
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
|
|
video_bitrate_allocator_factory_;
|
|
// |fme_| and |fdme_| are actually owned by |cm_|.
|
|
cricket::FakeMediaEngine* fme_;
|
|
cricket::FakeDataEngine* fdme_;
|
|
std::unique_ptr<cricket::ChannelManager> cm_;
|
|
cricket::FakeCall fake_call_;
|
|
webrtc::FakeMediaTransportFactory fake_media_transport_factory_;
|
|
rtc::UniqueRandomIdGenerator ssrc_generator_;
|
|
};
|
|
|
|
// Test that we startup/shutdown properly.
|
|
TEST_F(ChannelManagerTest, StartupShutdown) {
|
|
EXPECT_FALSE(cm_->initialized());
|
|
EXPECT_EQ(rtc::Thread::Current(), cm_->worker_thread());
|
|
EXPECT_TRUE(cm_->Init());
|
|
EXPECT_TRUE(cm_->initialized());
|
|
cm_->Terminate();
|
|
EXPECT_FALSE(cm_->initialized());
|
|
}
|
|
|
|
// Test that we startup/shutdown properly with a worker thread.
|
|
TEST_F(ChannelManagerTest, StartupShutdownOnThread) {
|
|
network_->Start();
|
|
worker_->Start();
|
|
EXPECT_FALSE(cm_->initialized());
|
|
EXPECT_EQ(rtc::Thread::Current(), cm_->worker_thread());
|
|
EXPECT_TRUE(cm_->set_network_thread(network_.get()));
|
|
EXPECT_EQ(network_.get(), cm_->network_thread());
|
|
EXPECT_TRUE(cm_->set_worker_thread(worker_.get()));
|
|
EXPECT_EQ(worker_.get(), cm_->worker_thread());
|
|
EXPECT_TRUE(cm_->Init());
|
|
EXPECT_TRUE(cm_->initialized());
|
|
// Setting the network or worker thread while initialized should fail.
|
|
EXPECT_FALSE(cm_->set_network_thread(rtc::Thread::Current()));
|
|
EXPECT_FALSE(cm_->set_worker_thread(rtc::Thread::Current()));
|
|
cm_->Terminate();
|
|
EXPECT_FALSE(cm_->initialized());
|
|
}
|
|
|
|
TEST_F(ChannelManagerTest, SetVideoRtxEnabled) {
|
|
std::vector<VideoCodec> codecs;
|
|
const VideoCodec rtx_codec(96, "rtx");
|
|
|
|
// By default RTX is disabled.
|
|
cm_->GetSupportedVideoCodecs(&codecs);
|
|
EXPECT_FALSE(ContainsMatchingCodec(codecs, rtx_codec));
|
|
|
|
// Enable and check.
|
|
EXPECT_TRUE(cm_->SetVideoRtxEnabled(true));
|
|
cm_->GetSupportedVideoCodecs(&codecs);
|
|
EXPECT_TRUE(ContainsMatchingCodec(codecs, rtx_codec));
|
|
|
|
// Disable and check.
|
|
EXPECT_TRUE(cm_->SetVideoRtxEnabled(false));
|
|
cm_->GetSupportedVideoCodecs(&codecs);
|
|
EXPECT_FALSE(ContainsMatchingCodec(codecs, rtx_codec));
|
|
|
|
// Cannot toggle rtx after initialization.
|
|
EXPECT_TRUE(cm_->Init());
|
|
EXPECT_FALSE(cm_->SetVideoRtxEnabled(true));
|
|
EXPECT_FALSE(cm_->SetVideoRtxEnabled(false));
|
|
|
|
// Can set again after terminate.
|
|
cm_->Terminate();
|
|
EXPECT_TRUE(cm_->SetVideoRtxEnabled(true));
|
|
cm_->GetSupportedVideoCodecs(&codecs);
|
|
EXPECT_TRUE(ContainsMatchingCodec(codecs, rtx_codec));
|
|
}
|
|
|
|
TEST_F(ChannelManagerTest, CreateDestroyChannels) {
|
|
EXPECT_TRUE(cm_->Init());
|
|
auto rtp_transport = CreateDtlsSrtpTransport();
|
|
TestCreateDestroyChannels(rtp_transport.get(),
|
|
webrtc::MediaTransportConfig());
|
|
}
|
|
|
|
TEST_F(ChannelManagerTest, CreateDestroyChannelsWithMediaTransport) {
|
|
EXPECT_TRUE(cm_->Init());
|
|
auto rtp_transport = CreateDtlsSrtpTransport();
|
|
auto media_transport = CreateMediaTransport(rtp_dtls_transport_.get());
|
|
TestCreateDestroyChannels(
|
|
rtp_transport.get(), webrtc::MediaTransportConfig(media_transport.get()));
|
|
}
|
|
|
|
TEST_F(ChannelManagerTest, CreateDestroyChannelsOnThread) {
|
|
network_->Start();
|
|
worker_->Start();
|
|
EXPECT_TRUE(cm_->set_worker_thread(worker_.get()));
|
|
EXPECT_TRUE(cm_->set_network_thread(network_.get()));
|
|
EXPECT_TRUE(cm_->Init());
|
|
auto rtp_transport = CreateDtlsSrtpTransport();
|
|
TestCreateDestroyChannels(rtp_transport.get(),
|
|
webrtc::MediaTransportConfig());
|
|
}
|
|
|
|
} // namespace cricket
|