blob: a39e3e2457093c9c069674068f0ef70debbf3e71 [file] [log] [blame]
/*
* Copyright 2018 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 <memory>
#include "api/test/simulated_network.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/rtp_rtcp/source/rtp_util.h"
#include "rtc_base/task_queue_for_test.h"
#include "test/call_test.h"
#include "test/gtest.h"
#include "test/rtcp_packet_parser.h"
namespace webrtc {
class SsrcEndToEndTest : public test::CallTest {
protected:
void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
};
TEST_F(SsrcEndToEndTest, ReceiverUsesLocalSsrc) {
class SyncRtcpObserver : public test::EndToEndTest {
public:
SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
EXPECT_EQ(kReceiverLocalVideoSsrc, parser.sender_ssrc());
observation_complete_.Set();
return SEND_PACKET;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out while waiting for a receiver RTCP packet to be sent.";
}
} test;
RunBaseTest(&test);
}
TEST_F(SsrcEndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
class PacketInputObserver : public PacketReceiver {
public:
explicit PacketInputObserver(PacketReceiver* receiver)
: receiver_(receiver) {}
bool Wait() { return delivered_packet_.Wait(kDefaultTimeoutMs); }
private:
DeliveryStatus DeliverPacket(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override {
if (IsRtcpPacket(packet)) {
return receiver_->DeliverPacket(media_type, std::move(packet),
packet_time_us);
}
DeliveryStatus delivery_status = receiver_->DeliverPacket(
media_type, std::move(packet), packet_time_us);
EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status);
delivered_packet_.Set();
return delivery_status;
}
PacketReceiver* receiver_;
rtc::Event delivered_packet_;
};
std::unique_ptr<test::DirectTransport> send_transport;
std::unique_ptr<test::DirectTransport> receive_transport;
std::unique_ptr<PacketInputObserver> input_observer;
SendTask(
RTC_FROM_HERE, task_queue(),
[this, &send_transport, &receive_transport, &input_observer]() {
CreateCalls();
send_transport = std::make_unique<test::DirectTransport>(
task_queue(),
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())),
sender_call_.get(), payload_type_map_);
receive_transport = std::make_unique<test::DirectTransport>(
task_queue(),
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())),
receiver_call_.get(), payload_type_map_);
input_observer =
std::make_unique<PacketInputObserver>(receiver_call_->Receiver());
send_transport->SetReceiver(input_observer.get());
receive_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, 0, 0, send_transport.get());
CreateMatchingReceiveConfigs(receive_transport.get());
CreateVideoStreams();
CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
Start();
receiver_call_->DestroyVideoReceiveStream(video_receive_streams_[0]);
video_receive_streams_.clear();
});
// Wait() waits for a received packet.
EXPECT_TRUE(input_observer->Wait());
SendTask(RTC_FROM_HERE, task_queue(),
[this, &send_transport, &receive_transport]() {
Stop();
DestroyStreams();
send_transport.reset();
receive_transport.reset();
DestroyCalls();
});
}
void SsrcEndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
bool send_single_ssrc_first) {
class SendsSetSsrcs : public test::EndToEndTest {
public:
SendsSetSsrcs(const uint32_t* ssrcs,
size_t num_ssrcs,
bool send_single_ssrc_first,
TaskQueueBase* task_queue)
: EndToEndTest(kDefaultTimeoutMs),
num_ssrcs_(num_ssrcs),
send_single_ssrc_first_(send_single_ssrc_first),
ssrcs_to_observe_(num_ssrcs),
expect_single_ssrc_(send_single_ssrc_first),
send_stream_(nullptr),
task_queue_(task_queue) {
for (size_t i = 0; i < num_ssrcs; ++i)
valid_ssrcs_[ssrcs[i]] = true;
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
EXPECT_TRUE(valid_ssrcs_[rtp_packet.Ssrc()])
<< "Received unknown SSRC: " << rtp_packet.Ssrc();
if (!valid_ssrcs_[rtp_packet.Ssrc()])
observation_complete_.Set();
if (!is_observed_[rtp_packet.Ssrc()]) {
is_observed_[rtp_packet.Ssrc()] = true;
--ssrcs_to_observe_;
if (expect_single_ssrc_) {
expect_single_ssrc_ = false;
observation_complete_.Set();
}
}
if (ssrcs_to_observe_ == 0)
observation_complete_.Set();
return SEND_PACKET;
}
size_t GetNumVideoStreams() const override { return num_ssrcs_; }
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
encoder_config->max_bitrate_bps = 50000;
for (auto& layer : encoder_config->simulcast_layers) {
layer.min_bitrate_bps = 10000;
layer.target_bitrate_bps = 15000;
layer.max_bitrate_bps = 20000;
}
video_encoder_config_all_streams_ = encoder_config->Copy();
if (send_single_ssrc_first_)
encoder_config->number_of_streams = 1;
}
void OnVideoStreamsCreated(VideoSendStream* send_stream,
const std::vector<VideoReceiveStreamInterface*>&
receive_streams) override {
send_stream_ = send_stream;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for "
<< (send_single_ssrc_first_ ? "first SSRC."
: "SSRCs.");
if (send_single_ssrc_first_) {
// Set full simulcast and continue with the rest of the SSRCs.
SendTask(RTC_FROM_HERE, task_queue_, [&]() {
send_stream_->ReconfigureVideoEncoder(
std::move(video_encoder_config_all_streams_));
});
EXPECT_TRUE(Wait()) << "Timed out while waiting on additional SSRCs.";
}
}
private:
std::map<uint32_t, bool> valid_ssrcs_;
std::map<uint32_t, bool> is_observed_;
const size_t num_ssrcs_;
const bool send_single_ssrc_first_;
size_t ssrcs_to_observe_;
bool expect_single_ssrc_;
VideoSendStream* send_stream_;
VideoEncoderConfig video_encoder_config_all_streams_;
TaskQueueBase* task_queue_;
} test(kVideoSendSsrcs, num_ssrcs, send_single_ssrc_first, task_queue());
RunBaseTest(&test);
}
TEST_F(SsrcEndToEndTest, SendsSetSsrc) {
TestSendsSetSsrcs(1, false);
}
TEST_F(SsrcEndToEndTest, SendsSetSimulcastSsrcs) {
TestSendsSetSsrcs(kNumSimulcastStreams, false);
}
TEST_F(SsrcEndToEndTest, CanSwitchToUseAllSsrcs) {
TestSendsSetSsrcs(kNumSimulcastStreams, true);
}
TEST_F(SsrcEndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
class ObserveRedundantPayloads : public test::EndToEndTest {
public:
ObserveRedundantPayloads()
: EndToEndTest(kDefaultTimeoutMs),
ssrcs_to_observe_(kNumSimulcastStreams) {
for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
registered_rtx_ssrc_[kSendRtxSsrcs[i]] = true;
}
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
if (!registered_rtx_ssrc_[rtp_packet.Ssrc()])
return SEND_PACKET;
const bool packet_is_redundant_payload = rtp_packet.payload_size() > 0;
if (!packet_is_redundant_payload)
return SEND_PACKET;
if (!observed_redundant_retransmission_[rtp_packet.Ssrc()]) {
observed_redundant_retransmission_[rtp_packet.Ssrc()] = true;
if (--ssrcs_to_observe_ == 0)
observation_complete_.Set();
}
return SEND_PACKET;
}
size_t GetNumVideoStreams() const override { return kNumSimulcastStreams; }
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
encoder_config->max_bitrate_bps = 50000;
for (auto& layer : encoder_config->simulcast_layers) {
layer.min_bitrate_bps = 10000;
layer.target_bitrate_bps = 15000;
layer.max_bitrate_bps = 20000;
}
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
for (size_t i = 0; i < kNumSimulcastStreams; ++i)
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
// Significantly higher than max bitrates for all video streams -> forcing
// padding to trigger redundant padding on all RTX SSRCs.
encoder_config->min_transmit_bitrate_bps = 100000;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out while waiting for redundant payloads on all SSRCs.";
}
private:
size_t ssrcs_to_observe_;
std::map<uint32_t, bool> observed_redundant_retransmission_;
std::map<uint32_t, bool> registered_rtx_ssrc_;
} test;
RunBaseTest(&test);
}
} // namespace webrtc