blob: 96598f5c6b06fc52a6bc000d52041b004d4871a5 [file] [log] [blame]
/*
* Copyright (c) 2004 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 "media/engine/webrtc_video_engine.h"
#include <algorithm>
#include <map>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_parameters.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/mock_video_bitrate_allocator.h"
#include "api/test/mock_video_bitrate_allocator_factory.h"
#include "api/test/mock_video_decoder_factory.h"
#include "api/test/mock_video_encoder_factory.h"
#include "api/test/video/function_video_decoder_factory.h"
#include "api/transport/field_trial_based_config.h"
#include "api/units/time_delta.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/i420_buffer.h"
#include "api/video/video_bitrate_allocation.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "api/video_codecs/h264_profile_level_id.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "call/flexfec_receive_stream.h"
#include "media/base/fake_frame_source.h"
#include "media/base/fake_network_interface.h"
#include "media/base/fake_video_renderer.h"
#include "media/base/media_constants.h"
#include "media/base/rtp_utils.h"
#include "media/base/test_utils.h"
#include "media/engine/fake_webrtc_call.h"
#include "media/engine/fake_webrtc_video_engine.h"
#include "media/engine/simulcast.h"
#include "media/engine/webrtc_voice_engine.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/event.h"
#include "rtc_base/experiments/min_video_bitrate_experiment.h"
#include "rtc_base/fake_clock.h"
#include "rtc_base/gunit.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/field_trial.h"
#include "test/fake_decoder.h"
#include "test/field_trial.h"
#include "test/frame_forwarder.h"
#include "test/gmock.h"
using ::testing::_;
using ::testing::Contains;
using ::testing::Each;
using ::testing::ElementsAreArray;
using ::testing::Eq;
using ::testing::Field;
using ::testing::IsEmpty;
using ::testing::Pair;
using ::testing::Return;
using ::testing::SizeIs;
using ::testing::StrNe;
using ::testing::Values;
using ::webrtc::BitrateConstraints;
using ::webrtc::RtpExtension;
using ::webrtc::RtpPacket;
namespace {
static const int kDefaultQpMax = 56;
static const uint8_t kRedRtxPayloadType = 125;
static const uint32_t kTimeout = 5000U;
static const uint32_t kSsrc = 1234u;
static const uint32_t kSsrcs4[] = {1, 2, 3, 4};
static const int kVideoWidth = 640;
static const int kVideoHeight = 360;
static const int kFramerate = 30;
static const uint32_t kSsrcs1[] = {1};
static const uint32_t kSsrcs3[] = {1, 2, 3};
static const uint32_t kRtxSsrcs1[] = {4};
static const uint32_t kFlexfecSsrc = 5;
static const uint32_t kIncomingUnsignalledSsrc = 0xC0FFEE;
static const int64_t kUnsignalledReceiveStreamCooldownMs = 500;
constexpr uint32_t kRtpHeaderSize = 12;
static const char kUnsupportedExtensionName[] =
"urn:ietf:params:rtp-hdrext:unsupported";
cricket::VideoCodec RemoveFeedbackParams(cricket::VideoCodec&& codec) {
codec.feedback_params = cricket::FeedbackParams();
return std::move(codec);
}
void VerifyCodecHasDefaultFeedbackParams(const cricket::VideoCodec& codec,
bool lntf_expected) {
EXPECT_EQ(lntf_expected,
codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamLntf, cricket::kParamValueEmpty)));
EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)));
EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamNack, cricket::kRtcpFbNackParamPli)));
EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamRemb, cricket::kParamValueEmpty)));
EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty)));
EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir)));
}
// Return true if any codec in `codecs` is an RTX codec with associated payload
// type `payload_type`.
bool HasRtxCodec(const std::vector<cricket::VideoCodec>& codecs,
int payload_type) {
for (const cricket::VideoCodec& codec : codecs) {
int associated_payload_type;
if (absl::EqualsIgnoreCase(codec.name.c_str(), "rtx") &&
codec.GetParam(cricket::kCodecParamAssociatedPayloadType,
&associated_payload_type) &&
associated_payload_type == payload_type) {
return true;
}
}
return false;
}
// TODO(nisse): Duplicated in call.cc.
const int* FindKeyByValue(const std::map<int, int>& m, int v) {
for (const auto& kv : m) {
if (kv.second == v)
return &kv.first;
}
return nullptr;
}
bool HasRtxReceiveAssociation(const webrtc::VideoReceiveStream::Config& config,
int payload_type) {
return FindKeyByValue(config.rtp.rtx_associated_payload_types,
payload_type) != nullptr;
}
// Check that there's an Rtx payload type for each decoder.
bool VerifyRtxReceiveAssociations(
const webrtc::VideoReceiveStream::Config& config) {
for (const auto& decoder : config.decoders) {
if (!HasRtxReceiveAssociation(config, decoder.payload_type))
return false;
}
return true;
}
rtc::scoped_refptr<webrtc::VideoFrameBuffer> CreateBlackFrameBuffer(
int width,
int height) {
rtc::scoped_refptr<webrtc::I420Buffer> buffer =
webrtc::I420Buffer::Create(width, height);
webrtc::I420Buffer::SetBlack(buffer);
return buffer;
}
void VerifySendStreamHasRtxTypes(const webrtc::VideoSendStream::Config& config,
const std::map<int, int>& rtx_types) {
std::map<int, int>::const_iterator it;
it = rtx_types.find(config.rtp.payload_type);
EXPECT_TRUE(it != rtx_types.end() &&
it->second == config.rtp.rtx.payload_type);
if (config.rtp.ulpfec.red_rtx_payload_type != -1) {
it = rtx_types.find(config.rtp.ulpfec.red_payload_type);
EXPECT_TRUE(it != rtx_types.end() &&
it->second == config.rtp.ulpfec.red_rtx_payload_type);
}
}
cricket::MediaConfig GetMediaConfig() {
cricket::MediaConfig media_config;
media_config.video.enable_cpu_adaptation = false;
return media_config;
}
// Values from GetMaxDefaultVideoBitrateKbps in webrtcvideoengine.cc.
int GetMaxDefaultBitrateBps(size_t width, size_t height) {
if (width * height <= 320 * 240) {
return 600000;
} else if (width * height <= 640 * 480) {
return 1700000;
} else if (width * height <= 960 * 540) {
return 2000000;
} else {
return 2500000;
}
}
class MockVideoSource : public rtc::VideoSourceInterface<webrtc::VideoFrame> {
public:
MOCK_METHOD(void,
AddOrUpdateSink,
(rtc::VideoSinkInterface<webrtc::VideoFrame> * sink,
const rtc::VideoSinkWants& wants),
(override));
MOCK_METHOD(void,
RemoveSink,
(rtc::VideoSinkInterface<webrtc::VideoFrame> * sink),
(override));
};
} // namespace
#define EXPECT_FRAME_WAIT(c, w, h, t) \
EXPECT_EQ_WAIT((c), renderer_.num_rendered_frames(), (t)); \
EXPECT_EQ((w), renderer_.width()); \
EXPECT_EQ((h), renderer_.height()); \
EXPECT_EQ(0, renderer_.errors());
#define EXPECT_FRAME_ON_RENDERER_WAIT(r, c, w, h, t) \
EXPECT_EQ_WAIT((c), (r).num_rendered_frames(), (t)); \
EXPECT_EQ((w), (r).width()); \
EXPECT_EQ((h), (r).height()); \
EXPECT_EQ(0, (r).errors());
namespace cricket {
class WebRtcVideoEngineTest : public ::testing::Test {
public:
WebRtcVideoEngineTest() : WebRtcVideoEngineTest("") {}
explicit WebRtcVideoEngineTest(const std::string& field_trials)
: override_field_trials_(
field_trials.empty()
? nullptr
: std::make_unique<webrtc::test::ScopedFieldTrials>(
field_trials)),
task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()),
call_(webrtc::Call::Create([&] {
webrtc::Call::Config call_config(&event_log_);
call_config.task_queue_factory = task_queue_factory_.get();
call_config.trials = &field_trials_;
return call_config;
}())),
encoder_factory_(new cricket::FakeWebRtcVideoEncoderFactory),
decoder_factory_(new cricket::FakeWebRtcVideoDecoderFactory),
video_bitrate_allocator_factory_(
webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
engine_(std::unique_ptr<cricket::FakeWebRtcVideoEncoderFactory>(
encoder_factory_),
std::unique_ptr<cricket::FakeWebRtcVideoDecoderFactory>(
decoder_factory_),
field_trials_) {
// Ensure fake clock doesn't return 0, which will cause some initializations
// fail inside RTP senders.
fake_clock_.AdvanceTime(webrtc::TimeDelta::Micros(1));
}
protected:
void AssignDefaultAptRtxTypes();
void AssignDefaultCodec();
// Find the index of the codec in the engine with the given name. The codec
// must be present.
size_t GetEngineCodecIndex(const std::string& name) const;
// Find the codec in the engine with the given name. The codec must be
// present.
cricket::VideoCodec GetEngineCodec(const std::string& name) const;
void AddSupportedVideoCodecType(const std::string& name);
VideoMediaChannel* SetSendParamsWithAllSupportedCodecs();
VideoMediaChannel* SetRecvParamsWithSupportedCodecs(
const std::vector<VideoCodec>& codecs);
void ExpectRtpCapabilitySupport(const char* uri, bool supported) const;
// Has to be the first one, so it is initialized before the call or there is a
// race condition in the clock access.
rtc::ScopedFakeClock fake_clock_;
std::unique_ptr<webrtc::test::ScopedFieldTrials> override_field_trials_;
webrtc::FieldTrialBasedConfig field_trials_;
webrtc::RtcEventLogNull event_log_;
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
// Used in WebRtcVideoEngineVoiceTest, but defined here so it's properly
// initialized when the constructor is called.
std::unique_ptr<webrtc::Call> call_;
cricket::FakeWebRtcVideoEncoderFactory* encoder_factory_;
cricket::FakeWebRtcVideoDecoderFactory* decoder_factory_;
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory_;
WebRtcVideoEngine engine_;
VideoCodec default_codec_;
std::map<int, int> default_apt_rtx_types_;
};
TEST_F(WebRtcVideoEngineTest, DefaultRtxCodecHasAssociatedPayloadTypeSet) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
AssignDefaultCodec();
std::vector<VideoCodec> engine_codecs = engine_.send_codecs();
for (size_t i = 0; i < engine_codecs.size(); ++i) {
if (engine_codecs[i].name != kRtxCodecName)
continue;
int associated_payload_type;
EXPECT_TRUE(engine_codecs[i].GetParam(kCodecParamAssociatedPayloadType,
&associated_payload_type));
EXPECT_EQ(default_codec_.id, associated_payload_type);
return;
}
FAIL() << "No RTX codec found among default codecs.";
}
TEST_F(WebRtcVideoEngineTest, SupportsTimestampOffsetHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kTimestampOffsetUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsAbsoluteSenderTimeHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kAbsSendTimeUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsTransportSequenceNumberHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kTransportSequenceNumberUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsVideoRotationHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kVideoRotationUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsPlayoutDelayHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kPlayoutDelayUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsVideoContentTypeHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kVideoContentTypeUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsVideoTimingHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kVideoTimingUri, true);
}
TEST_F(WebRtcVideoEngineTest, SupportsColorSpaceHeaderExtension) {
ExpectRtpCapabilitySupport(RtpExtension::kColorSpaceUri, true);
}
TEST_F(WebRtcVideoEngineTest, AdvertiseGenericDescriptor00) {
ExpectRtpCapabilitySupport(RtpExtension::kGenericFrameDescriptorUri00, false);
}
class WebRtcVideoEngineTestWithGenericDescriptor
: public WebRtcVideoEngineTest {
public:
WebRtcVideoEngineTestWithGenericDescriptor()
: WebRtcVideoEngineTest("WebRTC-GenericDescriptorAdvertised/Enabled/") {}
};
TEST_F(WebRtcVideoEngineTestWithGenericDescriptor,
AdvertiseGenericDescriptor00) {
ExpectRtpCapabilitySupport(RtpExtension::kGenericFrameDescriptorUri00, true);
}
class WebRtcVideoEngineTestWithDependencyDescriptor
: public WebRtcVideoEngineTest {
public:
WebRtcVideoEngineTestWithDependencyDescriptor()
: WebRtcVideoEngineTest(
"WebRTC-DependencyDescriptorAdvertised/Enabled/") {}
};
TEST_F(WebRtcVideoEngineTestWithDependencyDescriptor,
AdvertiseDependencyDescriptor) {
ExpectRtpCapabilitySupport(RtpExtension::kDependencyDescriptorUri, true);
}
TEST_F(WebRtcVideoEngineTest, AdvertiseVideoLayersAllocation) {
ExpectRtpCapabilitySupport(RtpExtension::kVideoLayersAllocationUri, false);
}
class WebRtcVideoEngineTestWithVideoLayersAllocation
: public WebRtcVideoEngineTest {
public:
WebRtcVideoEngineTestWithVideoLayersAllocation()
: WebRtcVideoEngineTest(
"WebRTC-VideoLayersAllocationAdvertised/Enabled/") {}
};
TEST_F(WebRtcVideoEngineTestWithVideoLayersAllocation,
AdvertiseVideoLayersAllocation) {
ExpectRtpCapabilitySupport(RtpExtension::kVideoLayersAllocationUri, true);
}
class WebRtcVideoFrameTrackingId : public WebRtcVideoEngineTest {
public:
WebRtcVideoFrameTrackingId()
: WebRtcVideoEngineTest(
"WebRTC-VideoFrameTrackingIdAdvertised/Enabled/") {}
};
TEST_F(WebRtcVideoFrameTrackingId, AdvertiseVideoFrameTrackingId) {
ExpectRtpCapabilitySupport(RtpExtension::kVideoFrameTrackingIdUri, true);
}
TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionBeforeCapturer) {
// Allocate the source first to prevent early destruction before channel's
// dtor is called.
::testing::NiceMock<MockVideoSource> video_source;
AddSupportedVideoCodecType("VP8");
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
// Add CVO extension.
const int id = 1;
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.extensions.push_back(
RtpExtension(RtpExtension::kVideoRotationUri, id));
EXPECT_TRUE(channel->SetSendParameters(parameters));
EXPECT_CALL(
video_source,
AddOrUpdateSink(_, Field(&rtc::VideoSinkWants::rotation_applied, false)));
// Set capturer.
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &video_source));
// Verify capturer has turned off applying rotation.
::testing::Mock::VerifyAndClear(&video_source);
// Verify removing header extension turns on applying rotation.
parameters.extensions.clear();
EXPECT_CALL(
video_source,
AddOrUpdateSink(_, Field(&rtc::VideoSinkWants::rotation_applied, true)));
EXPECT_TRUE(channel->SetSendParameters(parameters));
}
TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionBeforeAddSendStream) {
// Allocate the source first to prevent early destruction before channel's
// dtor is called.
::testing::NiceMock<MockVideoSource> video_source;
AddSupportedVideoCodecType("VP8");
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
// Add CVO extension.
const int id = 1;
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.extensions.push_back(
RtpExtension(RtpExtension::kVideoRotationUri, id));
EXPECT_TRUE(channel->SetSendParameters(parameters));
EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
// Set source.
EXPECT_CALL(
video_source,
AddOrUpdateSink(_, Field(&rtc::VideoSinkWants::rotation_applied, false)));
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &video_source));
}
TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionAfterCapturer) {
::testing::NiceMock<MockVideoSource> video_source;
AddSupportedVideoCodecType("VP8");
AddSupportedVideoCodecType("VP9");
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
// Set capturer.
EXPECT_CALL(
video_source,
AddOrUpdateSink(_, Field(&rtc::VideoSinkWants::rotation_applied, true)));
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &video_source));
// Verify capturer has turned on applying rotation.
::testing::Mock::VerifyAndClear(&video_source);
// Add CVO extension.
const int id = 1;
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
parameters.extensions.push_back(
RtpExtension(RtpExtension::kVideoRotationUri, id));
// Also remove the first codec to trigger a codec change as well.
parameters.codecs.erase(parameters.codecs.begin());
EXPECT_CALL(
video_source,
AddOrUpdateSink(_, Field(&rtc::VideoSinkWants::rotation_applied, false)));
EXPECT_TRUE(channel->SetSendParameters(parameters));
// Verify capturer has turned off applying rotation.
::testing::Mock::VerifyAndClear(&video_source);
// Verify removing header extension turns on applying rotation.
parameters.extensions.clear();
EXPECT_CALL(
video_source,
AddOrUpdateSink(_, Field(&rtc::VideoSinkWants::rotation_applied, true)));
EXPECT_TRUE(channel->SetSendParameters(parameters));
}
TEST_F(WebRtcVideoEngineTest, SetSendFailsBeforeSettingCodecs) {
AddSupportedVideoCodecType("VP8");
std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(123)));
EXPECT_FALSE(channel->SetSend(true))
<< "Channel should not start without codecs.";
EXPECT_TRUE(channel->SetSend(false))
<< "Channel should be stoppable even without set codecs.";
}
TEST_F(WebRtcVideoEngineTest, GetStatsWithoutSendCodecsSetDoesNotCrash) {
AddSupportedVideoCodecType("VP8");
std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(123)));
VideoMediaInfo info;
channel->GetStats(&info);
}
TEST_F(WebRtcVideoEngineTest, UseFactoryForVp8WhenSupported) {
AddSupportedVideoCodecType("VP8");
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
channel->OnReadyToSend(true);
EXPECT_TRUE(
channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
EXPECT_EQ(0, encoder_factory_->GetNumCreatedEncoders());
EXPECT_TRUE(channel->SetSend(true));
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Sending one frame will have allocate the encoder.
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1));
EXPECT_TRUE_WAIT(encoder_factory_->encoders()[0]->GetNumEncodedFrames() > 0,
kTimeout);
int num_created_encoders = encoder_factory_->GetNumCreatedEncoders();
EXPECT_EQ(num_created_encoders, 1);
// Setting codecs of the same type should not reallocate any encoders
// (expecting a no-op).
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel->SetSendParameters(parameters));
EXPECT_EQ(num_created_encoders, encoder_factory_->GetNumCreatedEncoders());
// Remove stream previously added to free the external encoder instance.
EXPECT_TRUE(channel->RemoveSendStream(kSsrc));
EXPECT_EQ(0u, encoder_factory_->encoders().size());
}
// Test that when an encoder factory supports H264, we add an RTX
// codec for it.
// TODO(deadbeef): This test should be updated if/when we start
// adding RTX codecs for unrecognized codec names.
TEST_F(WebRtcVideoEngineTest, RtxCodecAddedForH264Codec) {
using webrtc::H264Level;
using webrtc::H264Profile;
using webrtc::H264ProfileLevelId;
using webrtc::H264ProfileLevelIdToString;
webrtc::SdpVideoFormat h264_constrained_baseline("H264");
h264_constrained_baseline.parameters[kH264FmtpProfileLevelId] =
*H264ProfileLevelIdToString(H264ProfileLevelId(
H264Profile::kProfileConstrainedBaseline, H264Level::kLevel1));
webrtc::SdpVideoFormat h264_constrained_high("H264");
h264_constrained_high.parameters[kH264FmtpProfileLevelId] =
*H264ProfileLevelIdToString(H264ProfileLevelId(
H264Profile::kProfileConstrainedHigh, H264Level::kLevel1));
webrtc::SdpVideoFormat h264_high("H264");
h264_high.parameters[kH264FmtpProfileLevelId] = *H264ProfileLevelIdToString(
H264ProfileLevelId(H264Profile::kProfileHigh, H264Level::kLevel1));
encoder_factory_->AddSupportedVideoCodec(h264_constrained_baseline);
encoder_factory_->AddSupportedVideoCodec(h264_constrained_high);
encoder_factory_->AddSupportedVideoCodec(h264_high);
// First figure out what payload types the test codecs got assigned.
const std::vector<cricket::VideoCodec> codecs = engine_.send_codecs();
// Now search for RTX codecs for them. Expect that they all have associated
// RTX codecs.
EXPECT_TRUE(HasRtxCodec(
codecs,
FindMatchingCodec(codecs, cricket::VideoCodec(h264_constrained_baseline))
->id));
EXPECT_TRUE(HasRtxCodec(
codecs,
FindMatchingCodec(codecs, cricket::VideoCodec(h264_constrained_high))
->id));
EXPECT_TRUE(HasRtxCodec(
codecs, FindMatchingCodec(codecs, cricket::VideoCodec(h264_high))->id));
}
#if defined(RTC_ENABLE_VP9)
TEST_F(WebRtcVideoEngineTest, CanConstructDecoderForVp9EncoderFactory) {
AddSupportedVideoCodecType("VP9");
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
EXPECT_TRUE(
channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
}
#endif // defined(RTC_ENABLE_VP9)
TEST_F(WebRtcVideoEngineTest, PropagatesInputFrameTimestamp) {
AddSupportedVideoCodecType("VP8");
FakeCall* fake_call = new FakeCall();
call_.reset(fake_call);
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
EXPECT_TRUE(
channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 60);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder));
channel->SetSend(true);
FakeVideoSendStream* stream = fake_call->GetVideoSendStreams()[0];
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
int64_t last_timestamp = stream->GetLastTimestamp();
for (int i = 0; i < 10; i++) {
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
int64_t timestamp = stream->GetLastTimestamp();
int64_t interval = timestamp - last_timestamp;
// Precision changes from nanosecond to millisecond.
// Allow error to be no more than 1.
EXPECT_NEAR(cricket::VideoFormat::FpsToInterval(60) / 1E6, interval, 1);
last_timestamp = timestamp;
}
frame_forwarder.IncomingCapturedFrame(
frame_source.GetFrame(1280, 720, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / 30));
last_timestamp = stream->GetLastTimestamp();
for (int i = 0; i < 10; i++) {
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame(
1280, 720, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / 30));
int64_t timestamp = stream->GetLastTimestamp();
int64_t interval = timestamp - last_timestamp;
// Precision changes from nanosecond to millisecond.
// Allow error to be no more than 1.
EXPECT_NEAR(cricket::VideoFormat::FpsToInterval(30) / 1E6, interval, 1);
last_timestamp = timestamp;
}
// Remove stream previously added to free the external encoder instance.
EXPECT_TRUE(channel->RemoveSendStream(kSsrc));
}
void WebRtcVideoEngineTest::AssignDefaultAptRtxTypes() {
std::vector<VideoCodec> engine_codecs = engine_.send_codecs();
RTC_DCHECK(!engine_codecs.empty());
for (const cricket::VideoCodec& codec : engine_codecs) {
if (codec.name == "rtx") {
int associated_payload_type;
if (codec.GetParam(kCodecParamAssociatedPayloadType,
&associated_payload_type)) {
default_apt_rtx_types_[associated_payload_type] = codec.id;
}
}
}
}
void WebRtcVideoEngineTest::AssignDefaultCodec() {
std::vector<VideoCodec> engine_codecs = engine_.send_codecs();
RTC_DCHECK(!engine_codecs.empty());
bool codec_set = false;
for (const cricket::VideoCodec& codec : engine_codecs) {
if (!codec_set && codec.name != "rtx" && codec.name != "red" &&
codec.name != "ulpfec" && codec.name != "flexfec-03") {
default_codec_ = codec;
codec_set = true;
}
}
RTC_DCHECK(codec_set);
}
size_t WebRtcVideoEngineTest::GetEngineCodecIndex(
const std::string& name) const {
const std::vector<cricket::VideoCodec> codecs = engine_.send_codecs();
for (size_t i = 0; i < codecs.size(); ++i) {
const cricket::VideoCodec engine_codec = codecs[i];
if (!absl::EqualsIgnoreCase(name, engine_codec.name))
continue;
// The tests only use H264 Constrained Baseline. Make sure we don't return
// an internal H264 codec from the engine with a different H264 profile.
if (absl::EqualsIgnoreCase(name.c_str(), kH264CodecName)) {
const absl::optional<webrtc::H264ProfileLevelId> profile_level_id =
webrtc::ParseSdpForH264ProfileLevelId(engine_codec.params);
if (profile_level_id->profile !=
webrtc::H264Profile::kProfileConstrainedBaseline) {
continue;
}
}
return i;
}
// This point should never be reached.
ADD_FAILURE() << "Unrecognized codec name: " << name;
return -1;
}
cricket::VideoCodec WebRtcVideoEngineTest::GetEngineCodec(
const std::string& name) const {
return engine_.send_codecs()[GetEngineCodecIndex(name)];
}
void WebRtcVideoEngineTest::AddSupportedVideoCodecType(
const std::string& name) {
encoder_factory_->AddSupportedVideoCodecType(name);
decoder_factory_->AddSupportedVideoCodecType(name);
}
VideoMediaChannel*
WebRtcVideoEngineTest::SetSendParamsWithAllSupportedCodecs() {
VideoMediaChannel* channel = engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get());
cricket::VideoSendParameters parameters;
// We need to look up the codec in the engine to get the correct payload type.
for (const webrtc::SdpVideoFormat& format :
encoder_factory_->GetSupportedFormats()) {
cricket::VideoCodec engine_codec = GetEngineCodec(format.name);
if (!absl::c_linear_search(parameters.codecs, engine_codec)) {
parameters.codecs.push_back(engine_codec);
}
}
EXPECT_TRUE(channel->SetSendParameters(parameters));
return channel;
}
VideoMediaChannel* WebRtcVideoEngineTest::SetRecvParamsWithSupportedCodecs(
const std::vector<VideoCodec>& codecs) {
VideoMediaChannel* channel = engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get());
cricket::VideoRecvParameters parameters;
parameters.codecs = codecs;
EXPECT_TRUE(channel->SetRecvParameters(parameters));
return channel;
}
void WebRtcVideoEngineTest::ExpectRtpCapabilitySupport(const char* uri,
bool supported) const {
const std::vector<webrtc::RtpExtension> header_extensions =
GetDefaultEnabledRtpHeaderExtensions(engine_);
if (supported) {
EXPECT_THAT(header_extensions, Contains(Field(&RtpExtension::uri, uri)));
} else {
EXPECT_THAT(header_extensions, Each(Field(&RtpExtension::uri, StrNe(uri))));
}
}
TEST_F(WebRtcVideoEngineTest, UsesSimulcastAdapterForVp8Factories) {
AddSupportedVideoCodecType("VP8");
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
EXPECT_TRUE(channel->AddSendStream(CreateSimStreamParams("cname", ssrcs)));
EXPECT_TRUE(channel->SetSend(true));
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 60);
EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(2));
// Verify that encoders are configured for simulcast through adapter
// (increasing resolution and only configured to send one stream each).
int prev_width = -1;
for (size_t i = 0; i < encoder_factory_->encoders().size(); ++i) {
ASSERT_TRUE(encoder_factory_->encoders()[i]->WaitForInitEncode());
webrtc::VideoCodec codec_settings =
encoder_factory_->encoders()[i]->GetCodecSettings();
EXPECT_EQ(0, codec_settings.numberOfSimulcastStreams);
EXPECT_GT(codec_settings.width, prev_width);
prev_width = codec_settings.width;
}
EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), nullptr, nullptr));
channel.reset();
ASSERT_EQ(0u, encoder_factory_->encoders().size());
}
TEST_F(WebRtcVideoEngineTest, ChannelWithH264CanChangeToVp8) {
AddSupportedVideoCodecType("VP8");
AddSupportedVideoCodecType("H264");
// Frame source.
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("H264"));
EXPECT_TRUE(channel->SetSendParameters(parameters));
EXPECT_TRUE(
channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder));
// Sending one frame will have allocate the encoder.
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_EQ_WAIT(1u, encoder_factory_->encoders().size(), kTimeout);
cricket::VideoSendParameters new_parameters;
new_parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel->SetSendParameters(new_parameters));
// Sending one frame will switch encoder.
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
EXPECT_EQ_WAIT(1u, encoder_factory_->encoders().size(), kTimeout);
}
TEST_F(WebRtcVideoEngineTest,
UsesSimulcastAdapterForVp8WithCombinedVP8AndH264Factory) {
AddSupportedVideoCodecType("VP8");
AddSupportedVideoCodecType("H264");
std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel->SetSendParameters(parameters));
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
EXPECT_TRUE(channel->AddSendStream(CreateSimStreamParams("cname", ssrcs)));
EXPECT_TRUE(channel->SetSend(true));
// Send a fake frame, or else the media engine will configure the simulcast
// encoder adapter at a low-enough size that it'll only create a single
// encoder layer.
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(2));
ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode());
EXPECT_EQ(webrtc::kVideoCodecVP8,
encoder_factory_->encoders()[0]->GetCodecSettings().codecType);
channel.reset();
// Make sure DestroyVideoEncoder was called on the factory.
EXPECT_EQ(0u, encoder_factory_->encoders().size());
}
TEST_F(WebRtcVideoEngineTest,
DestroysNonSimulcastEncoderFromCombinedVP8AndH264Factory) {
AddSupportedVideoCodecType("VP8");
AddSupportedVideoCodecType("H264");
std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("H264"));
EXPECT_TRUE(channel->SetSendParameters(parameters));
EXPECT_TRUE(
channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
// Send a frame of 720p. This should trigger a "real" encoder initialization.
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1));
ASSERT_EQ(1u, encoder_factory_->encoders().size());
ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode());
EXPECT_EQ(webrtc::kVideoCodecH264,
encoder_factory_->encoders()[0]->GetCodecSettings().codecType);
channel.reset();
// Make sure DestroyVideoEncoder was called on the factory.
ASSERT_EQ(0u, encoder_factory_->encoders().size());
}
TEST_F(WebRtcVideoEngineTest, SimulcastEnabledForH264BehindFieldTrial) {
RTC_DCHECK(!override_field_trials_);
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-H264Simulcast/Enabled/");
AddSupportedVideoCodecType("H264");
std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("H264"));
EXPECT_TRUE(channel->SetSendParameters(parameters));
const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
EXPECT_TRUE(
channel->AddSendStream(cricket::CreateSimStreamParams("cname", ssrcs)));
// Send a frame of 720p. This should trigger a "real" encoder initialization.
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(channel->SetVideoSend(ssrcs[0], nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1));
ASSERT_EQ(1u, encoder_factory_->encoders().size());
FakeWebRtcVideoEncoder* encoder = encoder_factory_->encoders()[0];
ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode());
EXPECT_EQ(webrtc::kVideoCodecH264, encoder->GetCodecSettings().codecType);
EXPECT_LT(1u, encoder->GetCodecSettings().numberOfSimulcastStreams);
EXPECT_TRUE(channel->SetVideoSend(ssrcs[0], nullptr, nullptr));
}
// Test that FlexFEC is not supported as a send video codec by default.
// Only enabling field trial should allow advertising FlexFEC send codec.
TEST_F(WebRtcVideoEngineTest, Flexfec03SendCodecEnablesWithFieldTrial) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
auto flexfec = Field("name", &VideoCodec::name, "flexfec-03");
EXPECT_THAT(engine_.send_codecs(), Not(Contains(flexfec)));
RTC_DCHECK(!override_field_trials_);
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-FlexFEC-03-Advertised/Enabled/");
EXPECT_THAT(engine_.send_codecs(), Contains(flexfec));
}
// Test that FlexFEC is supported as a receive video codec by default.
// Disabling field trial should prevent advertising FlexFEC receive codec.
TEST_F(WebRtcVideoEngineTest, Flexfec03ReceiveCodecDisablesWithFieldTrial) {
decoder_factory_->AddSupportedVideoCodecType("VP8");
auto flexfec = Field("name", &VideoCodec::name, "flexfec-03");
EXPECT_THAT(engine_.recv_codecs(), Contains(flexfec));
RTC_DCHECK(!override_field_trials_);
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-FlexFEC-03-Advertised/Disabled/");
EXPECT_THAT(engine_.recv_codecs(), Not(Contains(flexfec)));
}
// Test that the FlexFEC "codec" gets assigned to the lower payload type range
TEST_F(WebRtcVideoEngineTest, Flexfec03LowerPayloadTypeRange) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
auto flexfec = Field("name", &VideoCodec::name, "flexfec-03");
// FlexFEC is active with field trial.
RTC_DCHECK(!override_field_trials_);
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-FlexFEC-03-Advertised/Enabled/");
auto send_codecs = engine_.send_codecs();
auto it = std::find_if(send_codecs.begin(), send_codecs.end(),
[](const cricket::VideoCodec& codec) {
return codec.name == "flexfec-03";
});
ASSERT_NE(it, send_codecs.end());
EXPECT_LE(35, it->id);
EXPECT_GE(65, it->id);
}
// Test that codecs are added in the order they are reported from the factory.
TEST_F(WebRtcVideoEngineTest, ReportSupportedCodecs) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
const char* kFakeCodecName = "FakeCodec";
encoder_factory_->AddSupportedVideoCodecType(kFakeCodecName);
// The last reported codec should appear after the first codec in the vector.
const size_t vp8_index = GetEngineCodecIndex("VP8");
const size_t fake_codec_index = GetEngineCodecIndex(kFakeCodecName);
EXPECT_LT(vp8_index, fake_codec_index);
}
// Test that a codec that was added after the engine was initialized
// does show up in the codec list after it was added.
TEST_F(WebRtcVideoEngineTest, ReportSupportedAddedCodec) {
const char* kFakeExternalCodecName1 = "FakeExternalCodec1";
const char* kFakeExternalCodecName2 = "FakeExternalCodec2";
// Set up external encoder factory with first codec, and initialize engine.
encoder_factory_->AddSupportedVideoCodecType(kFakeExternalCodecName1);
std::vector<cricket::VideoCodec> codecs_before(engine_.send_codecs());
// Add second codec.
encoder_factory_->AddSupportedVideoCodecType(kFakeExternalCodecName2);
std::vector<cricket::VideoCodec> codecs_after(engine_.send_codecs());
// The codec itself and RTX should have been added.
EXPECT_EQ(codecs_before.size() + 2, codecs_after.size());
// Check that both fake codecs are present and that the second fake codec
// appears after the first fake codec.
const size_t fake_codec_index1 = GetEngineCodecIndex(kFakeExternalCodecName1);
const size_t fake_codec_index2 = GetEngineCodecIndex(kFakeExternalCodecName2);
EXPECT_LT(fake_codec_index1, fake_codec_index2);
}
TEST_F(WebRtcVideoEngineTest, ReportRtxForExternalCodec) {
const char* kFakeCodecName = "FakeCodec";
encoder_factory_->AddSupportedVideoCodecType(kFakeCodecName);
const size_t fake_codec_index = GetEngineCodecIndex(kFakeCodecName);
EXPECT_EQ("rtx", engine_.send_codecs().at(fake_codec_index + 1).name);
}
TEST_F(WebRtcVideoEngineTest, RegisterDecodersIfSupported) {
AddSupportedVideoCodecType("VP8");
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
std::unique_ptr<VideoMediaChannel> channel(
SetRecvParamsWithSupportedCodecs(parameters.codecs));
EXPECT_TRUE(
channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
ASSERT_EQ(1u, decoder_factory_->decoders().size());
// Setting codecs of the same type should not reallocate the decoder.
EXPECT_TRUE(channel->SetRecvParameters(parameters));
EXPECT_EQ(1, decoder_factory_->GetNumCreatedDecoders());
// Remove stream previously added to free the external decoder instance.
EXPECT_TRUE(channel->RemoveRecvStream(kSsrc));
EXPECT_EQ(0u, decoder_factory_->decoders().size());
}
// Verifies that we can set up decoders.
TEST_F(WebRtcVideoEngineTest, RegisterH264DecoderIfSupported) {
// TODO(pbos): Do not assume that encoder/decoder support is symmetric. We
// can't even query the WebRtcVideoDecoderFactory for supported codecs.
// For now we add a FakeWebRtcVideoEncoderFactory to add H264 to supported
// codecs.
AddSupportedVideoCodecType("H264");
std::vector<cricket::VideoCodec> codecs;
codecs.push_back(GetEngineCodec("H264"));
std::unique_ptr<VideoMediaChannel> channel(
SetRecvParamsWithSupportedCodecs(codecs));
EXPECT_TRUE(
channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
ASSERT_EQ(1u, decoder_factory_->decoders().size());
}
// Tests when GetSources is called with non-existing ssrc, it will return an
// empty list of RtpSource without crashing.
TEST_F(WebRtcVideoEngineTest, GetSourcesWithNonExistingSsrc) {
// Setup an recv stream with `kSsrc`.
AddSupportedVideoCodecType("VP8");
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
std::unique_ptr<VideoMediaChannel> channel(
SetRecvParamsWithSupportedCodecs(parameters.codecs));
EXPECT_TRUE(
channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
// Call GetSources with |kSsrc + 1| which doesn't exist.
std::vector<webrtc::RtpSource> sources = channel->GetSources(kSsrc + 1);
EXPECT_EQ(0u, sources.size());
}
TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullFactories) {
std::unique_ptr<webrtc::VideoEncoderFactory> encoder_factory;
std::unique_ptr<webrtc::VideoDecoderFactory> decoder_factory;
webrtc::FieldTrialBasedConfig trials;
WebRtcVideoEngine engine(std::move(encoder_factory),
std::move(decoder_factory), trials);
EXPECT_EQ(0u, engine.send_codecs().size());
EXPECT_EQ(0u, engine.recv_codecs().size());
}
TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, EmptyFactories) {
// `engine` take ownership of the factories.
webrtc::MockVideoEncoderFactory* encoder_factory =
new webrtc::MockVideoEncoderFactory();
webrtc::MockVideoDecoderFactory* decoder_factory =
new webrtc::MockVideoDecoderFactory();
webrtc::FieldTrialBasedConfig trials;
WebRtcVideoEngine engine(
(std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)),
(std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)), trials);
// TODO(kron): Change to Times(1) once send and receive codecs are changed
// to be treated independently.
EXPECT_CALL(*encoder_factory, GetSupportedFormats()).Times(1);
EXPECT_EQ(0u, engine.send_codecs().size());
EXPECT_EQ(0u, engine.recv_codecs().size());
EXPECT_CALL(*encoder_factory, Die());
EXPECT_CALL(*decoder_factory, Die());
}
// Test full behavior in the video engine when video codec factories of the new
// type are injected supporting the single codec Vp8. Check the returned codecs
// from the engine and that we will create a Vp8 encoder and decoder using the
// new factories.
TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
// `engine` take ownership of the factories.
webrtc::MockVideoEncoderFactory* encoder_factory =
new webrtc::MockVideoEncoderFactory();
webrtc::MockVideoDecoderFactory* decoder_factory =
new webrtc::MockVideoDecoderFactory();
std::unique_ptr<webrtc::MockVideoBitrateAllocatorFactory>
rate_allocator_factory =
std::make_unique<webrtc::MockVideoBitrateAllocatorFactory>();
EXPECT_CALL(*rate_allocator_factory,
CreateVideoBitrateAllocator(Field(&webrtc::VideoCodec::codecType,
webrtc::kVideoCodecVP8)))
.WillOnce(
[] { return std::make_unique<webrtc::MockVideoBitrateAllocator>(); });
webrtc::FieldTrialBasedConfig trials;
WebRtcVideoEngine engine(
(std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)),
(std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)), trials);
const webrtc::SdpVideoFormat vp8_format("VP8");
const std::vector<webrtc::SdpVideoFormat> supported_formats = {vp8_format};
EXPECT_CALL(*encoder_factory, GetSupportedFormats())
.WillRepeatedly(Return(supported_formats));
EXPECT_CALL(*decoder_factory, GetSupportedFormats())
.WillRepeatedly(Return(supported_formats));
// Verify the codecs from the engine.
const std::vector<VideoCodec> engine_codecs = engine.send_codecs();
// Verify default codecs has been added correctly.
EXPECT_EQ(5u, engine_codecs.size());
EXPECT_EQ("VP8", engine_codecs.at(0).name);
// RTX codec for VP8.
EXPECT_EQ("rtx", engine_codecs.at(1).name);
int vp8_associated_payload;
EXPECT_TRUE(engine_codecs.at(1).GetParam(kCodecParamAssociatedPayloadType,
&vp8_associated_payload));
EXPECT_EQ(vp8_associated_payload, engine_codecs.at(0).id);
EXPECT_EQ(kRedCodecName, engine_codecs.at(2).name);
// RTX codec for RED.
EXPECT_EQ("rtx", engine_codecs.at(3).name);
int red_associated_payload;
EXPECT_TRUE(engine_codecs.at(3).GetParam(kCodecParamAssociatedPayloadType,
&red_associated_payload));
EXPECT_EQ(red_associated_payload, engine_codecs.at(2).id);
EXPECT_EQ(kUlpfecCodecName, engine_codecs.at(4).name);
int associated_payload_type;
EXPECT_TRUE(engine_codecs.at(1).GetParam(
cricket::kCodecParamAssociatedPayloadType, &associated_payload_type));
EXPECT_EQ(engine_codecs.at(0).id, associated_payload_type);
// Verify default parameters has been added to the VP8 codec.
VerifyCodecHasDefaultFeedbackParams(engine_codecs.at(0),
/*lntf_expected=*/false);
// Mock encoder creation. `engine` take ownership of the encoder.
webrtc::VideoEncoderFactory::CodecInfo codec_info;
codec_info.has_internal_source = false;
const webrtc::SdpVideoFormat format("VP8");
EXPECT_CALL(*encoder_factory, QueryVideoEncoder(format))
.WillRepeatedly(Return(codec_info));
rtc::Event encoder_created;
EXPECT_CALL(*encoder_factory, CreateVideoEncoder(format)).WillOnce([&] {
encoder_created.Set();
return std::make_unique<FakeWebRtcVideoEncoder>(nullptr);
});
// Mock decoder creation. `engine` take ownership of the decoder.
EXPECT_CALL(*decoder_factory, CreateVideoDecoder(format)).WillOnce([] {
return std::make_unique<FakeWebRtcVideoDecoder>(nullptr);
});
// Create a call.
webrtc::RtcEventLogNull event_log;
auto task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
webrtc::Call::Config call_config(&event_log);
webrtc::FieldTrialBasedConfig field_trials;
call_config.trials = &field_trials;
call_config.task_queue_factory = task_queue_factory.get();
const auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
// Create send channel.
const int send_ssrc = 123;
std::unique_ptr<VideoMediaChannel> send_channel(engine.CreateMediaChannel(
call.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
rate_allocator_factory.get()));
cricket::VideoSendParameters send_parameters;
send_parameters.codecs.push_back(engine_codecs.at(0));
EXPECT_TRUE(send_channel->SetSendParameters(send_parameters));
send_channel->OnReadyToSend(true);
EXPECT_TRUE(
send_channel->AddSendStream(StreamParams::CreateLegacy(send_ssrc)));
EXPECT_TRUE(send_channel->SetSend(true));
// Set capturer.
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(send_channel->SetVideoSend(send_ssrc, nullptr, &frame_forwarder));
// Sending one frame will allocate the encoder.
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
encoder_created.Wait(kTimeout);
// Create recv channel.
const int recv_ssrc = 321;
std::unique_ptr<VideoMediaChannel> recv_channel(engine.CreateMediaChannel(
call.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
rate_allocator_factory.get()));
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(engine_codecs.at(0));
EXPECT_TRUE(recv_channel->SetRecvParameters(recv_parameters));
EXPECT_TRUE(recv_channel->AddRecvStream(
cricket::StreamParams::CreateLegacy(recv_ssrc)));
// Remove streams previously added to free the encoder and decoder instance.
EXPECT_CALL(*encoder_factory, Die());
EXPECT_CALL(*decoder_factory, Die());
EXPECT_CALL(*rate_allocator_factory, Die());
EXPECT_TRUE(send_channel->RemoveSendStream(send_ssrc));
EXPECT_TRUE(recv_channel->RemoveRecvStream(recv_ssrc));
}
// Test behavior when decoder factory fails to create a decoder (returns null).
TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullDecoder) {
// `engine` take ownership of the factories.
webrtc::MockVideoEncoderFactory* encoder_factory =
new webrtc::MockVideoEncoderFactory();
webrtc::MockVideoDecoderFactory* decoder_factory =
new webrtc::MockVideoDecoderFactory();
std::unique_ptr<webrtc::MockVideoBitrateAllocatorFactory>
rate_allocator_factory =
std::make_unique<webrtc::MockVideoBitrateAllocatorFactory>();
webrtc::FieldTrialBasedConfig trials;
WebRtcVideoEngine engine(
(std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)),
(std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)), trials);
const webrtc::SdpVideoFormat vp8_format("VP8");
const std::vector<webrtc::SdpVideoFormat> supported_formats = {vp8_format};
EXPECT_CALL(*encoder_factory, GetSupportedFormats())
.WillRepeatedly(Return(supported_formats));
// Decoder creation fails.
EXPECT_CALL(*decoder_factory, CreateVideoDecoder).WillOnce([] {
return nullptr;
});
// Create a call.
webrtc::RtcEventLogNull event_log;
auto task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
webrtc::Call::Config call_config(&event_log);
webrtc::FieldTrialBasedConfig field_trials;
call_config.trials = &field_trials;
call_config.task_queue_factory = task_queue_factory.get();
const auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
// Create recv channel.
EXPECT_CALL(*decoder_factory, GetSupportedFormats())
.WillRepeatedly(::testing::Return(supported_formats));
const int recv_ssrc = 321;
std::unique_ptr<VideoMediaChannel> recv_channel(engine.CreateMediaChannel(
call.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
rate_allocator_factory.get()));
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(engine.recv_codecs().front());
EXPECT_TRUE(recv_channel->SetRecvParameters(recv_parameters));
EXPECT_TRUE(recv_channel->AddRecvStream(
cricket::StreamParams::CreateLegacy(recv_ssrc)));
// Remove streams previously added to free the encoder and decoder instance.
EXPECT_TRUE(recv_channel->RemoveRecvStream(recv_ssrc));
}
TEST_F(WebRtcVideoEngineTest, DISABLED_RecreatesEncoderOnContentTypeChange) {
encoder_factory_->AddSupportedVideoCodecType("VP8");
std::unique_ptr<FakeCall> fake_call(new FakeCall());
std::unique_ptr<VideoMediaChannel> channel(
SetSendParamsWithAllSupportedCodecs());
ASSERT_TRUE(
channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
cricket::VideoCodec codec = GetEngineCodec("VP8");
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
channel->OnReadyToSend(true);
channel->SetSend(true);
ASSERT_TRUE(channel->SetSendParameters(parameters));
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
VideoOptions options;
EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1));
EXPECT_EQ(webrtc::VideoCodecMode::kRealtimeVideo,
encoder_factory_->encoders().back()->GetCodecSettings().mode);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// No change in content type, keep current encoder.
EXPECT_EQ(1, encoder_factory_->GetNumCreatedEncoders());
options.is_screencast.emplace(true);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Change to screen content, recreate encoder. For the simulcast encoder
// adapter case, this will result in two calls since InitEncode triggers a
// a new instance.
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(2));
EXPECT_EQ(webrtc::VideoCodecMode::kScreensharing,
encoder_factory_->encoders().back()->GetCodecSettings().mode);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Still screen content, no need to update encoder.
EXPECT_EQ(2, encoder_factory_->GetNumCreatedEncoders());
options.is_screencast.emplace(false);
options.video_noise_reduction.emplace(false);
EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder));
// Change back to regular video content, update encoder. Also change
// a non `is_screencast` option just to verify it doesn't affect recreation.
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(3));
EXPECT_EQ(webrtc::VideoCodecMode::kRealtimeVideo,
encoder_factory_->encoders().back()->GetCodecSettings().mode);
// Remove stream previously added to free the external encoder instance.
EXPECT_TRUE(channel->RemoveSendStream(kSsrc));
EXPECT_EQ(0u, encoder_factory_->encoders().size());
}
class WebRtcVideoChannelEncodedFrameCallbackTest : public ::testing::Test {
protected:
webrtc::Call::Config GetCallConfig(
webrtc::RtcEventLogNull* event_log,
webrtc::TaskQueueFactory* task_queue_factory) {
webrtc::Call::Config call_config(event_log);
call_config.task_queue_factory = task_queue_factory;
call_config.trials = &field_trials_;
return call_config;
}
WebRtcVideoChannelEncodedFrameCallbackTest()
: task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()),
call_(absl::WrapUnique(webrtc::Call::Create(
GetCallConfig(&event_log_, task_queue_factory_.get())))),
video_bitrate_allocator_factory_(
webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
engine_(
webrtc::CreateBuiltinVideoEncoderFactory(),
std::make_unique<webrtc::test::FunctionVideoDecoderFactory>(
[]() { return std::make_unique<webrtc::test::FakeDecoder>(); },
kSdpVideoFormats),
field_trials_),
channel_(absl::WrapUnique(static_cast<cricket::WebRtcVideoChannel*>(
engine_.CreateMediaChannel(
call_.get(),
cricket::MediaConfig(),
cricket::VideoOptions(),
webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get())))) {
network_interface_.SetDestination(channel_.get());
channel_->SetInterface(&network_interface_);
cricket::VideoRecvParameters parameters;
parameters.codecs = engine_.recv_codecs();
channel_->SetRecvParameters(parameters);
}
~WebRtcVideoChannelEncodedFrameCallbackTest() override {
channel_->SetInterface(nullptr);
}
void DeliverKeyFrame(uint32_t ssrc) {
RtpPacket packet;
packet.SetMarker(true);
packet.SetPayloadType(96); // VP8
packet.SetSsrc(ssrc);
// VP8 Keyframe + 1 byte payload
uint8_t* buf_ptr = packet.AllocatePayload(11);
memset(buf_ptr, 0, 11); // Pass MSAN (don't care about bytes 1-9)
buf_ptr[0] = 0x10; // Partition ID 0 + beginning of partition.
constexpr unsigned width = 1080;
constexpr unsigned height = 720;
buf_ptr[6] = width & 255;
buf_ptr[7] = width >> 8;
buf_ptr[8] = height & 255;
buf_ptr[9] = height >> 8;
call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, packet.Buffer(),
/*packet_time_us=*/0);
}
void DeliverKeyFrameAndWait(uint32_t ssrc) {
DeliverKeyFrame(ssrc);
EXPECT_EQ_WAIT(1, renderer_.num_rendered_frames(), kTimeout);
EXPECT_EQ(0, renderer_.errors());
}
static const std::vector<webrtc::SdpVideoFormat> kSdpVideoFormats;
webrtc::FieldTrialBasedConfig field_trials_;
webrtc::RtcEventLogNull event_log_;
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
std::unique_ptr<webrtc::Call> call_;
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory_;
WebRtcVideoEngine engine_;
std::unique_ptr<WebRtcVideoChannel> channel_;
cricket::FakeNetworkInterface network_interface_;
cricket::FakeVideoRenderer renderer_;
};
const std::vector<webrtc::SdpVideoFormat>
WebRtcVideoChannelEncodedFrameCallbackTest::kSdpVideoFormats = {
webrtc::SdpVideoFormat("VP8")};
TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest,
SetEncodedFrameBufferFunction_DefaultStream) {
testing::MockFunction<void(const webrtc::RecordableEncodedFrame&)> callback;
EXPECT_CALL(callback, Call);
EXPECT_TRUE(channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kSsrc), /*is_default_stream=*/true));
channel_->SetRecordableEncodedFrameCallback(/*ssrc=*/0,
callback.AsStdFunction());
EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
DeliverKeyFrame(kSsrc);
EXPECT_EQ_WAIT(1, renderer_.num_rendered_frames(), kTimeout);
EXPECT_EQ(0, renderer_.errors());
channel_->RemoveRecvStream(kSsrc);
}
TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest,
SetEncodedFrameBufferFunction_MatchSsrcWithDefaultStream) {
testing::MockFunction<void(const webrtc::RecordableEncodedFrame&)> callback;
EXPECT_CALL(callback, Call);
EXPECT_TRUE(channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kSsrc), /*is_default_stream=*/true));
EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
channel_->SetRecordableEncodedFrameCallback(kSsrc, callback.AsStdFunction());
DeliverKeyFrame(kSsrc);
EXPECT_EQ_WAIT(1, renderer_.num_rendered_frames(), kTimeout);
EXPECT_EQ(0, renderer_.errors());
channel_->RemoveRecvStream(kSsrc);
}
TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest,
SetEncodedFrameBufferFunction_MatchSsrc) {
testing::MockFunction<void(const webrtc::RecordableEncodedFrame&)> callback;
EXPECT_CALL(callback, Call);
EXPECT_TRUE(channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kSsrc), /*is_default_stream=*/false));
EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
channel_->SetRecordableEncodedFrameCallback(kSsrc, callback.AsStdFunction());
DeliverKeyFrame(kSsrc);
EXPECT_EQ_WAIT(1, renderer_.num_rendered_frames(), kTimeout);
EXPECT_EQ(0, renderer_.errors());
channel_->RemoveRecvStream(kSsrc);
}
TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest,
SetEncodedFrameBufferFunction_MismatchSsrc) {
testing::StrictMock<
testing::MockFunction<void(const webrtc::RecordableEncodedFrame&)>>
callback;
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc + 1),
/*is_default_stream=*/false));
EXPECT_TRUE(channel_->SetSink(kSsrc + 1, &renderer_));
channel_->SetRecordableEncodedFrameCallback(kSsrc, callback.AsStdFunction());
DeliverKeyFrame(kSsrc); // Expected to not cause function to fire.
DeliverKeyFrameAndWait(kSsrc + 1);
channel_->RemoveRecvStream(kSsrc + 1);
}
TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest,
SetEncodedFrameBufferFunction_MismatchSsrcWithDefaultStream) {
testing::StrictMock<
testing::MockFunction<void(const webrtc::RecordableEncodedFrame&)>>
callback;
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc + 1),
/*is_default_stream=*/true));
EXPECT_TRUE(channel_->SetSink(kSsrc + 1, &renderer_));
channel_->SetRecordableEncodedFrameCallback(kSsrc, callback.AsStdFunction());
DeliverKeyFrame(kSsrc); // Expected to not cause function to fire.
DeliverKeyFrameAndWait(kSsrc + 1);
channel_->RemoveRecvStream(kSsrc + 1);
}
class WebRtcVideoChannelBaseTest : public ::testing::Test {
protected:
WebRtcVideoChannelBaseTest()
: task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()),
video_bitrate_allocator_factory_(
webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
engine_(webrtc::CreateBuiltinVideoEncoderFactory(),
webrtc::CreateBuiltinVideoDecoderFactory(),
field_trials_) {}
void SetUp() override {
// One testcase calls SetUp in a loop, only create call_ once.
if (!call_) {
webrtc::Call::Config call_config(&event_log_);
call_config.task_queue_factory = task_queue_factory_.get();
call_config.trials = &field_trials_;
call_.reset(webrtc::Call::Create(call_config));
}
cricket::MediaConfig media_config;
// Disabling cpu overuse detection actually disables quality scaling too; it
// implies DegradationPreference kMaintainResolution. Automatic scaling
// needs to be disabled, otherwise, tests which check the size of received
// frames become flaky.
media_config.video.enable_cpu_adaptation = false;
channel_.reset(
static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel(
call_.get(), media_config, cricket::VideoOptions(),
webrtc::CryptoOptions(), video_bitrate_allocator_factory_.get())));
channel_->OnReadyToSend(true);
EXPECT_TRUE(channel_.get() != NULL);
network_interface_.SetDestination(channel_.get());
channel_->SetInterface(&network_interface_);
cricket::VideoRecvParameters parameters;
parameters.codecs = engine_.send_codecs();
channel_->SetRecvParameters(parameters);
EXPECT_TRUE(channel_->AddSendStream(DefaultSendStreamParams()));
frame_forwarder_ = std::make_unique<webrtc::test::FrameForwarder>();
frame_source_ = std::make_unique<cricket::FakeFrameSource>(
640, 480, rtc::kNumMicrosecsPerSec / kFramerate);
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, frame_forwarder_.get()));
}
// Utility method to setup an additional stream to send and receive video.
// Used to test send and recv between two streams.
void SetUpSecondStream() {
SetUpSecondStreamWithNoRecv();
// Setup recv for second stream.
EXPECT_TRUE(channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kSsrc + 2)));
// Make the second renderer available for use by a new stream.
EXPECT_TRUE(channel_->SetSink(kSsrc + 2, &renderer2_));
}
// Setup an additional stream just to send video. Defer add recv stream.
// This is required if you want to test unsignalled recv of video rtp packets.
void SetUpSecondStreamWithNoRecv() {
// SetUp() already added kSsrc make sure duplicate SSRCs cant be added.
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
EXPECT_FALSE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
EXPECT_TRUE(channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kSsrc + 2)));
// We dont add recv for the second stream.
// Setup the receive and renderer for second stream after send.
frame_forwarder_2_ = std::make_unique<webrtc::test::FrameForwarder>();
EXPECT_TRUE(
channel_->SetVideoSend(kSsrc + 2, nullptr, frame_forwarder_2_.get()));
}
void TearDown() override {
channel_->SetInterface(nullptr);
channel_.reset();
}
void ResetTest() {
TearDown();
SetUp();
}
bool SetDefaultCodec() { return SetOneCodec(DefaultCodec()); }
bool SetOneCodec(const cricket::VideoCodec& codec) {
frame_source_ = std::make_unique<cricket::FakeFrameSource>(
kVideoWidth, kVideoHeight, rtc::kNumMicrosecsPerSec / kFramerate);
bool sending = channel_->sending();
bool success = SetSend(false);
if (success) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
success = channel_->SetSendParameters(parameters);
}
if (success) {
success = SetSend(sending);
}
return success;
}
bool SetSend(bool send) { return channel_->SetSend(send); }
void SendFrame() {
if (frame_forwarder_2_) {
frame_forwarder_2_->IncomingCapturedFrame(frame_source_->GetFrame());
}
frame_forwarder_->IncomingCapturedFrame(frame_source_->GetFrame());
}
bool WaitAndSendFrame(int wait_ms) {
bool ret = rtc::Thread::Current()->ProcessMessages(wait_ms);
SendFrame();
return ret;
}
int NumRtpBytes() { return network_interface_.NumRtpBytes(); }
int NumRtpBytes(uint32_t ssrc) {
return network_interface_.NumRtpBytes(ssrc);
}
int NumRtpPackets() { return network_interface_.NumRtpPackets(); }
int NumRtpPackets(uint32_t ssrc) {
return network_interface_.NumRtpPackets(ssrc);
}
int NumSentSsrcs() { return network_interface_.NumSentSsrcs(); }
rtc::CopyOnWriteBuffer GetRtpPacket(int index) {
return network_interface_.GetRtpPacket(index);
}
static int GetPayloadType(rtc::CopyOnWriteBuffer p) {
RtpPacket header;
EXPECT_TRUE(header.Parse(std::move(p)));
return header.PayloadType();
}
// Tests that we can send and receive frames.
void SendAndReceive(const cricket::VideoCodec& codec) {
EXPECT_TRUE(SetOneCodec(codec));
EXPECT_TRUE(SetSend(true));
channel_->SetDefaultSink(&renderer_);
EXPECT_EQ(0, renderer_.num_rendered_frames());
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_EQ(codec.id, GetPayloadType(GetRtpPacket(0)));
}
void SendReceiveManyAndGetStats(const cricket::VideoCodec& codec,
int duration_sec,
int fps) {
EXPECT_TRUE(SetOneCodec(codec));
EXPECT_TRUE(SetSend(true));
channel_->SetDefaultSink(&renderer_);
EXPECT_EQ(0, renderer_.num_rendered_frames());
for (int i = 0; i < duration_sec; ++i) {
for (int frame = 1; frame <= fps; ++frame) {
EXPECT_TRUE(WaitAndSendFrame(1000 / fps));
EXPECT_FRAME_WAIT(frame + i * fps, kVideoWidth, kVideoHeight, kTimeout);
}
}
EXPECT_EQ(codec.id, GetPayloadType(GetRtpPacket(0)));
}
cricket::VideoSenderInfo GetSenderStats(size_t i) {
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
return info.senders[i];
}
cricket::VideoReceiverInfo GetReceiverStats(size_t i) {
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
return info.receivers[i];
}
// Two streams one channel tests.
// Tests that we can send and receive frames.
void TwoStreamsSendAndReceive(const cricket::VideoCodec& codec) {
SetUpSecondStream();
// Test sending and receiving on first stream.
SendAndReceive(codec);
// Test sending and receiving on second stream.
EXPECT_EQ_WAIT(1, renderer2_.num_rendered_frames(), kTimeout);
EXPECT_GT(NumRtpPackets(), 0);
EXPECT_EQ(1, renderer2_.num_rendered_frames());
}
cricket::VideoCodec GetEngineCodec(const std::string& name) {
for (const cricket::VideoCodec& engine_codec : engine_.send_codecs()) {
if (absl::EqualsIgnoreCase(name, engine_codec.name))
return engine_codec;
}
// This point should never be reached.
ADD_FAILURE() << "Unrecognized codec name: " << name;
return cricket::VideoCodec();
}
cricket::VideoCodec DefaultCodec() { return GetEngineCodec("VP8"); }
cricket::StreamParams DefaultSendStreamParams() {
return cricket::StreamParams::CreateLegacy(kSsrc);
}
webrtc::RtcEventLogNull event_log_;
webrtc::FieldTrialBasedConfig field_trials_;
std::unique_ptr<webrtc::test::ScopedFieldTrials> override_field_trials_;
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
std::unique_ptr<webrtc::Call> call_;
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory_;
WebRtcVideoEngine engine_;
std::unique_ptr<cricket::FakeFrameSource> frame_source_;
std::unique_ptr<webrtc::test::FrameForwarder> frame_forwarder_;
std::unique_ptr<webrtc::test::FrameForwarder> frame_forwarder_2_;
std::unique_ptr<WebRtcVideoChannel> channel_;
cricket::FakeNetworkInterface network_interface_;
cricket::FakeVideoRenderer renderer_;
// Used by test cases where 2 streams are run on the same channel.
cricket::FakeVideoRenderer renderer2_;
};
// Test that SetSend works.
TEST_F(WebRtcVideoChannelBaseTest, SetSend) {
EXPECT_FALSE(channel_->sending());
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, frame_forwarder_.get()));
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_FALSE(channel_->sending());
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->sending());
SendFrame();
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
EXPECT_TRUE(SetSend(false));
EXPECT_FALSE(channel_->sending());
}
// Test that SetSend fails without codecs being set.
TEST_F(WebRtcVideoChannelBaseTest, SetSendWithoutCodecs) {
EXPECT_FALSE(channel_->sending());
EXPECT_FALSE(SetSend(true));
EXPECT_FALSE(channel_->sending());
}
// Test that we properly set the send and recv buffer sizes by the time
// SetSend is called.
TEST_F(WebRtcVideoChannelBaseTest, SetSendSetsTransportBufferSizes) {
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
EXPECT_EQ(256 * 1024, network_interface_.recvbuf_size());
}
// Test that we properly set the send and recv buffer sizes when overriding
// via field trials.
TEST_F(WebRtcVideoChannelBaseTest, OverridesRecvBufferSize) {
// Set field trial to override the default recv buffer size, and then re-run
// setup where the interface is created and configured.
const int kCustomRecvBufferSize = 123456;
RTC_DCHECK(!override_field_trials_);
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-IncreasedReceivebuffers/123456/");
ResetTest();
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
EXPECT_EQ(kCustomRecvBufferSize, network_interface_.recvbuf_size());
}
// Test that we properly set the send and recv buffer sizes when overriding
// via field trials with suffix.
TEST_F(WebRtcVideoChannelBaseTest, OverridesRecvBufferSizeWithSuffix) {
// Set field trial to override the default recv buffer size, and then re-run
// setup where the interface is created and configured.
const int kCustomRecvBufferSize = 123456;
RTC_DCHECK(!override_field_trials_);
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-IncreasedReceivebuffers/123456_Dogfood/");
ResetTest();
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
EXPECT_EQ(kCustomRecvBufferSize, network_interface_.recvbuf_size());
}
// Test that we properly set the send and recv buffer sizes when overriding
// via field trials that don't make any sense.
TEST_F(WebRtcVideoChannelBaseTest, InvalidRecvBufferSize) {
// Set bogus field trial values to override the default recv buffer size, and
// then re-run setup where the interface is created and configured. The
// default value should still be used.
const char* prev_field_trials = webrtc::field_trial::GetFieldTrialString();
std::string field_trial_string;
for (std::string group : {" ", "NotANumber", "-1", "0"}) {
std::string trial_string = "WebRTC-IncreasedReceivebuffers/";
trial_string += group;
trial_string += "/";
// Dear reader. Sorry for this... it's a bit of a mess.
// TODO(bugs.webrtc.org/12854): This test needs to be rewritten to not use
// ResetTest and changing global field trials in a loop.
TearDown();
// This is a hack to appease tsan. Because of the way the test is written
// active state within Call, including running task queues may race with
// the test changing the global field trial variable.
// This particular hack, pauses the transport controller TQ while we
// change the field trial.
rtc::TaskQueue* tq = call_->GetTransportControllerSend()->GetWorkerQueue();
rtc::Event waiting, resume;
tq->PostTask([&waiting, &resume]() {
waiting.Set();
resume.Wait(rtc::Event::kForever);
});
waiting.Wait(rtc::Event::kForever);
field_trial_string = std::move(trial_string);
webrtc::field_trial::InitFieldTrialsFromString(field_trial_string.c_str());
SetUp();
resume.Set();
// OK, now the test can carry on.
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
EXPECT_EQ(256 * 1024, network_interface_.recvbuf_size());
}
webrtc::field_trial::InitFieldTrialsFromString(prev_field_trials);
}
// Test that stats work properly for a 1-1 call.
TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
const int kDurationSec = 3;
const int kFps = 10;
SendReceiveManyAndGetStats(DefaultCodec(), kDurationSec, kFps);
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1U, info.senders.size());
// TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
// For webrtc, bytes_sent does not include the RTP header length.
EXPECT_EQ(info.senders[0].payload_bytes_sent,
NumRtpBytes() - kRtpHeaderSize * NumRtpPackets());
EXPECT_EQ(NumRtpPackets(), info.senders[0].packets_sent);
EXPECT_EQ(0.0, info.senders[0].fraction_lost);
ASSERT_TRUE(info.senders[0].codec_payload_type);
EXPECT_EQ(DefaultCodec().id, *info.senders[0].codec_payload_type);
EXPECT_EQ(0, info.senders[0].firs_rcvd);
EXPECT_EQ(0, info.senders[0].plis_rcvd);
EXPECT_EQ(0u, info.senders[0].nacks_rcvd);
EXPECT_EQ(kVideoWidth, info.senders[0].send_frame_width);
EXPECT_EQ(kVideoHeight, info.senders[0].send_frame_height);
EXPECT_GT(info.senders[0].framerate_input, 0);
EXPECT_GT(info.senders[0].framerate_sent, 0);
EXPECT_EQ(1U, info.send_codecs.count(DefaultCodec().id));
EXPECT_EQ(DefaultCodec().ToCodecParameters(),
info.send_codecs[DefaultCodec().id]);
ASSERT_EQ(1U, info.receivers.size());
EXPECT_EQ(1U, info.senders[0].ssrcs().size());
EXPECT_EQ(1U, info.receivers[0].ssrcs().size());
EXPECT_EQ(info.senders[0].ssrcs()[0], info.receivers[0].ssrcs()[0]);
ASSERT_TRUE(info.receivers[0].codec_payload_type);
EXPECT_EQ(DefaultCodec().id, *info.receivers[0].codec_payload_type);
EXPECT_EQ(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
info.receivers[0].payload_bytes_rcvd);
EXPECT_EQ(NumRtpPackets(), info.receivers[0].packets_rcvd);
EXPECT_EQ(0, info.receivers[0].packets_lost);
// TODO(asapersson): Not set for webrtc. Handle missing stats.
// EXPECT_EQ(0, info.receivers[0].packets_concealed);
EXPECT_EQ(0, info.receivers[0].firs_sent);
EXPECT_EQ(0, info.receivers[0].plis_sent);
EXPECT_EQ(0U, info.receivers[0].nacks_sent);
EXPECT_EQ(kVideoWidth, info.receivers[0].frame_width);
EXPECT_EQ(kVideoHeight, info.receivers[0].frame_height);
EXPECT_GT(info.receivers[0].framerate_rcvd, 0);
EXPECT_GT(info.receivers[0].framerate_decoded, 0);
EXPECT_GT(info.receivers[0].framerate_output, 0);
EXPECT_EQ(1U, info.receive_codecs.count(DefaultCodec().id));
EXPECT_EQ(DefaultCodec().ToCodecParameters(),
info.receive_codecs[DefaultCodec().id]);
}
// Test that stats work properly for a conf call with multiple recv streams.
TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
cricket::FakeVideoRenderer renderer1, renderer2;
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(DefaultCodec());
parameters.conference_mode = true;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
EXPECT_TRUE(channel_->SetSink(1, &renderer1));
EXPECT_TRUE(channel_->SetSink(2, &renderer2));
EXPECT_EQ(0, renderer1.num_rendered_frames());
EXPECT_EQ(0, renderer2.num_rendered_frames());
std::vector<uint32_t> ssrcs;
ssrcs.push_back(1);
ssrcs.push_back(2);
network_interface_.SetConferenceMode(true, ssrcs);
SendFrame();
EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kVideoWidth, kVideoHeight,
kTimeout);
EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kVideoWidth, kVideoHeight,
kTimeout);
EXPECT_TRUE(channel_->SetSend(false));
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1U, info.senders.size());
// TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
// For webrtc, bytes_sent does not include the RTP header length.
EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
GetSenderStats(0).payload_bytes_sent, kTimeout);
EXPECT_EQ_WAIT(NumRtpPackets(), GetSenderStats(0).packets_sent, kTimeout);
EXPECT_EQ(kVideoWidth, GetSenderStats(0).send_frame_width);
EXPECT_EQ(kVideoHeight, GetSenderStats(0).send_frame_height);
ASSERT_EQ(2U, info.receivers.size());
for (size_t i = 0; i < info.receivers.size(); ++i) {
EXPECT_EQ(1U, GetReceiverStats(i).ssrcs().size());
EXPECT_EQ(i + 1, GetReceiverStats(i).ssrcs()[0]);
EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
GetReceiverStats(i).payload_bytes_rcvd, kTimeout);
EXPECT_EQ_WAIT(NumRtpPackets(), GetReceiverStats(i).packets_rcvd, kTimeout);
EXPECT_EQ_WAIT(kVideoWidth, GetReceiverStats(i).frame_width, kTimeout);
EXPECT_EQ_WAIT(kVideoHeight, GetReceiverStats(i).frame_height, kTimeout);
}
}
// Test that stats work properly for a conf call with multiple send streams.
TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleSendStreams) {
// Normal setup; note that we set the SSRC explicitly to ensure that
// it will come first in the senders map.
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(DefaultCodec());
parameters.conference_mode = true;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
EXPECT_TRUE(SetSend(true));
SendFrame();
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
// Add an additional capturer, and hook up a renderer to receive it.
cricket::FakeVideoRenderer renderer2;
webrtc::test::FrameForwarder frame_forwarder;
const int kTestWidth = 160;
const int kTestHeight = 120;
cricket::FakeFrameSource frame_source(kTestWidth, kTestHeight,
rtc::kNumMicrosecsPerSec / 5);
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(5678)));
EXPECT_TRUE(channel_->SetVideoSend(5678, nullptr, &frame_forwarder));
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(5678)));
EXPECT_TRUE(channel_->SetSink(5678, &renderer2));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kTestWidth, kTestHeight,
kTimeout);
// Get stats, and make sure they are correct for two senders. We wait until
// the number of expected packets have been sent to avoid races where we
// check stats before it has been updated.
cricket::VideoMediaInfo info;
for (uint32_t i = 0; i < kTimeout; ++i) {
rtc::Thread::Current()->ProcessMessages(1);
EXPECT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(2U, info.senders.size());
if (info.senders[0].packets_sent + info.senders[1].packets_sent ==
NumRtpPackets()) {
// Stats have been updated for both sent frames, expectations can be
// checked now.
break;
}
}
EXPECT_EQ(NumRtpPackets(),
info.senders[0].packets_sent + info.senders[1].packets_sent)
<< "Timed out while waiting for packet counts for all sent packets.";
EXPECT_EQ(1U, info.senders[0].ssrcs().size());
EXPECT_EQ(1234U, info.senders[0].ssrcs()[0]);
EXPECT_EQ(kVideoWidth, info.senders[0].send_frame_width);
EXPECT_EQ(kVideoHeight, info.senders[0].send_frame_height);
EXPECT_EQ(1U, info.senders[1].ssrcs().size());
EXPECT_EQ(5678U, info.senders[1].ssrcs()[0]);
EXPECT_EQ(kTestWidth, info.senders[1].send_frame_width);
EXPECT_EQ(kTestHeight, info.senders[1].send_frame_height);
// The capturer must be unregistered here as it runs out of it's scope next.
channel_->SetVideoSend(5678, nullptr, nullptr);
}
// Test that we can set the bandwidth.
TEST_F(WebRtcVideoChannelBaseTest, SetSendBandwidth) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(DefaultCodec());
parameters.max_bandwidth_bps = -1; // <= 0 means unlimited.
EXPECT_TRUE(channel_->SetSendParameters(parameters));
parameters.max_bandwidth_bps = 128 * 1024;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
}
// Test that we can set the SSRC for the default send source.
TEST_F(WebRtcVideoChannelBaseTest, SetSendSsrc) {
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
SendFrame();
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
RtpPacket header;
EXPECT_TRUE(header.Parse(GetRtpPacket(0)));
EXPECT_EQ(kSsrc, header.Ssrc());
// Packets are being paced out, so these can mismatch between the first and
// second call to NumRtpPackets until pending packets are paced out.
EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(header.Ssrc()), kTimeout);
EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(header.Ssrc()), kTimeout);
EXPECT_EQ(1, NumSentSsrcs());
EXPECT_EQ(0, NumRtpPackets(kSsrc - 1));
EXPECT_EQ(0, NumRtpBytes(kSsrc - 1));
}
// Test that we can set the SSRC even after codecs are set.
TEST_F(WebRtcVideoChannelBaseTest, SetSendSsrcAfterSetCodecs) {
// Remove stream added in Setup.
EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(999)));
EXPECT_TRUE(channel_->SetVideoSend(999u, nullptr, frame_forwarder_.get()));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(WaitAndSendFrame(0));
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
RtpPacket header;
EXPECT_TRUE(header.Parse(GetRtpPacket(0)));
EXPECT_EQ(999u, header.Ssrc());
// Packets are being paced out, so these can mismatch between the first and
// second call to NumRtpPackets until pending packets are paced out.
EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(header.Ssrc()), kTimeout);
EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(header.Ssrc()), kTimeout);
EXPECT_EQ(1, NumSentSsrcs());
EXPECT_EQ(0, NumRtpPackets(kSsrc));
EXPECT_EQ(0, NumRtpBytes(kSsrc));
}
// Test that we can set the default video renderer before and after
// media is received.
TEST_F(WebRtcVideoChannelBaseTest, SetSink) {
RtpPacket packet;
packet.SetSsrc(kSsrc);
channel_->SetDefaultSink(NULL);
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(0, renderer_.num_rendered_frames());
channel_->OnPacketReceived(packet.Buffer(), /* packet_time_us */ -1);
channel_->SetDefaultSink(&renderer_);
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
}
// Tests setting up and configuring a send stream.
TEST_F(WebRtcVideoChannelBaseTest, AddRemoveSendStreams) {
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
channel_->SetDefaultSink(&renderer_);
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_GT(NumRtpPackets(), 0);
RtpPacket header;
size_t last_packet = NumRtpPackets() - 1;
EXPECT_TRUE(header.Parse(GetRtpPacket(static_cast<int>(last_packet))));
EXPECT_EQ(kSsrc, header.Ssrc());
// Remove the send stream that was added during Setup.
EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
int rtp_packets = NumRtpPackets();
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(789u)));
EXPECT_TRUE(channel_->SetVideoSend(789u, nullptr, frame_forwarder_.get()));
EXPECT_EQ(rtp_packets, NumRtpPackets());
// Wait 30ms to guarantee the engine does not drop the frame.
EXPECT_TRUE(WaitAndSendFrame(30));
EXPECT_TRUE_WAIT(NumRtpPackets() > rtp_packets, kTimeout);
last_packet = NumRtpPackets() - 1;
EXPECT_TRUE(header.Parse(GetRtpPacket(static_cast<int>(last_packet))));
EXPECT_EQ(789u, header.Ssrc());
}
// Tests the behavior of incoming streams in a conference scenario.
TEST_F(WebRtcVideoChannelBaseTest, SimulateConference) {
cricket::FakeVideoRenderer renderer1, renderer2;
EXPECT_TRUE(SetDefaultCodec());
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(DefaultCodec());
parameters.conference_mode = true;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
EXPECT_TRUE(channel_->SetSink(1, &renderer1));
EXPECT_TRUE(channel_->SetSink(2, &renderer2));
EXPECT_EQ(0, renderer1.num_rendered_frames());
EXPECT_EQ(0, renderer2.num_rendered_frames());
std::vector<uint32_t> ssrcs;
ssrcs.push_back(1);
ssrcs.push_back(2);
network_interface_.SetConferenceMode(true, ssrcs);
SendFrame();
EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kVideoWidth, kVideoHeight,
kTimeout);
EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kVideoWidth, kVideoHeight,
kTimeout);
EXPECT_EQ(DefaultCodec().id, GetPayloadType(GetRtpPacket(0)));
EXPECT_EQ(kVideoWidth, renderer1.width());
EXPECT_EQ(kVideoHeight, renderer1.height());
EXPECT_EQ(kVideoWidth, renderer2.width());
EXPECT_EQ(kVideoHeight, renderer2.height());
EXPECT_TRUE(channel_->RemoveRecvStream(2));
EXPECT_TRUE(channel_->RemoveRecvStream(1));
}
// Tests that we can add and remove capturers and frames are sent out properly
TEST_F(WebRtcVideoChannelBaseTest, DISABLED_AddRemoveCapturer) {
using cricket::FOURCC_I420;
using cricket::VideoCodec;
using cricket::VideoFormat;
using cricket::VideoOptions;
VideoCodec codec = DefaultCodec();
const int time_between_send_ms = VideoFormat::FpsToInterval(kFramerate);
EXPECT_TRUE(SetOneCodec(codec));
EXPECT_TRUE(SetSend(true));
channel_->SetDefaultSink(&renderer_);
EXPECT_EQ(0, renderer_.num_rendered_frames());
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(480, 360, rtc::kNumMicrosecsPerSec / 30,
rtc::kNumMicrosecsPerSec / 30);
// TODO(nisse): This testcase fails if we don't configure
// screencast. It's unclear why, I see nothing obvious in this
// test which is related to screencast logic.
VideoOptions video_options;
video_options.is_screencast = true;
channel_->SetVideoSend(kSsrc, &video_options, nullptr);
int captured_frames = 1;
for (int iterations = 0; iterations < 2; ++iterations) {
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, &frame_forwarder));
rtc::Thread::Current()->ProcessMessages(time_between_send_ms);
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
++captured_frames;
// Wait until frame of right size is captured.
EXPECT_TRUE_WAIT(renderer_.num_rendered_frames() >= captured_frames &&
480 == renderer_.width() &&
360 == renderer_.height() && !renderer_.black_frame(),
kTimeout);
EXPECT_GE(renderer_.num_rendered_frames(), captured_frames);
EXPECT_EQ(480, renderer_.width());
EXPECT_EQ(360, renderer_.height());
captured_frames = renderer_.num_rendered_frames() + 1;
EXPECT_FALSE(renderer_.black_frame());
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
// Make sure a black frame is generated within the specified timeout.
// The black frame should be the resolution of the previous frame to
// prevent expensive encoder reconfigurations.
EXPECT_TRUE_WAIT(renderer_.num_rendered_frames() >= captured_frames &&
480 == renderer_.width() &&
360 == renderer_.height() && renderer_.black_frame(),
kTimeout);
EXPECT_GE(renderer_.num_rendered_frames(), captured_frames);
EXPECT_EQ(480, renderer_.width());
EXPECT_EQ(360, renderer_.height());
EXPECT_TRUE(renderer_.black_frame());
// The black frame has the same timestamp as the next frame since it's
// timestamp is set to the last frame's timestamp + interval. WebRTC will
// not render a frame with the same timestamp so capture another frame
// with the frame capturer to increment the next frame's timestamp.
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
}
}
// Tests that if SetVideoSend is called with a NULL capturer after the
// capturer was already removed, the application doesn't crash (and no black
// frame is sent).
TEST_F(WebRtcVideoChannelBaseTest, RemoveCapturerWithoutAdd) {
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
channel_->SetDefaultSink(&renderer_);
EXPECT_EQ(0, renderer_.num_rendered_frames());
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
// Wait for one frame so they don't get dropped because we send frames too
// tightly.
rtc::Thread::Current()->ProcessMessages(30);
// Remove the capturer.
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
// No capturer was added, so this SetVideoSend shouldn't do anything.
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
rtc::Thread::Current()->ProcessMessages(300);
// Verify no more frames were sent.
EXPECT_EQ(1, renderer_.num_rendered_frames());
}
// Tests that we can add and remove capturer as unique sources.
TEST_F(WebRtcVideoChannelBaseTest, AddRemoveCapturerMultipleSources) {
// WebRTC implementation will drop frames if pushed to quickly. Wait the
// interval time to avoid that.
// WebRTC implementation will drop frames if pushed to quickly. Wait the
// interval time to avoid that.
// Set up the stream associated with the engine.
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
cricket::VideoFormat capture_format(
kVideoWidth, kVideoHeight,
cricket::VideoFormat::FpsToInterval(kFramerate), cricket::FOURCC_I420);
// Set up additional stream 1.
cricket::FakeVideoRenderer renderer1;
EXPECT_FALSE(channel_->SetSink(1, &renderer1));
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
EXPECT_TRUE(channel_->SetSink(1, &renderer1));
EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(1)));
webrtc::test::FrameForwarder frame_forwarder1;
cricket::FakeFrameSource frame_source(kVideoWidth, kVideoHeight,
rtc::kNumMicrosecsPerSec / kFramerate);
// Set up additional stream 2.
cricket::FakeVideoRenderer renderer2;
EXPECT_FALSE(channel_->SetSink(2, &renderer2));
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
EXPECT_TRUE(channel_->SetSink(2, &renderer2));
EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(2)));
webrtc::test::FrameForwarder frame_forwarder2;
// State for all the streams.
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
// A limitation in the lmi implementation requires that SetVideoSend() is
// called after SetOneCodec().
// TODO(hellner): this seems like an unnecessary constraint, fix it.
EXPECT_TRUE(channel_->SetVideoSend(1, nullptr, &frame_forwarder1));
EXPECT_TRUE(channel_->SetVideoSend(2, nullptr, &frame_forwarder2));
EXPECT_TRUE(SetSend(true));
// Test capturer associated with engine.
const int kTestWidth = 160;
const int kTestHeight = 120;
frame_forwarder1.IncomingCapturedFrame(frame_source.GetFrame(
kTestWidth, kTestHeight, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / kFramerate));
EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kTestWidth, kTestHeight,
kTimeout);
// Capture a frame with additional capturer2, frames should be received
frame_forwarder2.IncomingCapturedFrame(frame_source.GetFrame(
kTestWidth, kTestHeight, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / kFramerate));
EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kTestWidth, kTestHeight,
kTimeout);
// Successfully remove the capturer.
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
// The capturers must be unregistered here as it runs out of it's scope
// next.
EXPECT_TRUE(channel_->SetVideoSend(1, nullptr, nullptr));
EXPECT_TRUE(channel_->SetVideoSend(2, nullptr, nullptr));
}
// Tests empty StreamParams is rejected.
TEST_F(WebRtcVideoChannelBaseTest, RejectEmptyStreamParams) {
// Remove the send stream that was added during Setup.
EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
cricket::StreamParams empty;
EXPECT_FALSE(channel_->AddSendStream(empty));
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(789u)));
}
// Test that multiple send streams can be created and deleted properly.
TEST_F(WebRtcVideoChannelBaseTest, MultipleSendStreams) {
// Remove stream added in Setup. I.e. remove stream corresponding to default
// channel.
EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
const unsigned int kSsrcsSize = sizeof(kSsrcs4) / sizeof(kSsrcs4[0]);
for (unsigned int i = 0; i < kSsrcsSize; ++i) {
EXPECT_TRUE(channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
}
// Delete one of the non default channel streams, let the destructor delete
// the remaining ones.
EXPECT_TRUE(channel_->RemoveSendStream(kSsrcs4[kSsrcsSize - 1]));
// Stream should already be deleted.
EXPECT_FALSE(channel_->RemoveSendStream(kSsrcs4[kSsrcsSize - 1]));
}
TEST_F(WebRtcVideoChannelBaseTest, SendAndReceiveVp8Vga) {
SendAndReceive(GetEngineCodec("VP8"));
}
TEST_F(WebRtcVideoChannelBaseTest, SendAndReceiveVp8Qvga) {
SendAndReceive(GetEngineCodec("VP8"));
}
TEST_F(WebRtcVideoChannelBaseTest, SendAndReceiveVp8SvcQqvga) {
SendAndReceive(GetEngineCodec("VP8"));
}
TEST_F(WebRtcVideoChannelBaseTest, TwoStreamsSendAndReceive) {
// Set a high bitrate to not be downscaled by VP8 due to low initial start
// bitrates. This currently happens at <250k, and two streams sharing 300k
// initially will use QVGA instead of VGA.
// TODO(pbos): Set up the quality scaler so that both senders reliably start
// at QVGA, then verify that instead.
cricket::VideoCodec codec = GetEngineCodec("VP8");
codec.params[kCodecParamStartBitrate] = "1000000";
TwoStreamsSendAndReceive(codec);
}
#if defined(RTC_ENABLE_VP9)
TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderFallback) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP9"));
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
VideoCodec codec;
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_EQ("VP9", codec.name);
// RequestEncoderFallback will post a task to the worker thread (which is also
// the current thread), hence the ProcessMessages call.
channel_->RequestEncoderFallback();
rtc::Thread::Current()->ProcessMessages(30);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_EQ("VP8", codec.name);
// No other codec to fall back to, keep using VP8.
channel_->RequestEncoderFallback();
rtc::Thread::Current()->ProcessMessages(30);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_EQ("VP8", codec.name);
}
TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchWithConfig) {
const std::string kParam = "the-param";
const std::string kPing = "ping";
const std::string kPong = "pong";
cricket::VideoSendParameters parameters;
VideoCodec vp9 = GetEngineCodec("VP9");
vp9.params[kParam] = kPong;
parameters.codecs.push_back(vp9);
VideoCodec vp8 = GetEngineCodec("VP8");
vp8.params[kParam] = kPing;
parameters.codecs.push_back(vp8);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
channel_->SetVideoCodecSwitchingEnabled(true);
VideoCodec codec;
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP9"));
// RequestEncoderSwitch will post a task to the worker thread (which is also
// the current thread), hence the ProcessMessages call.
webrtc::EncoderSwitchRequestCallback::Config conf1{"VP8", kParam, kPing};
channel_->RequestEncoderSwitch(conf1);
rtc::Thread::Current()->ProcessMessages(30);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP8"));
EXPECT_THAT(codec.params, Contains(Pair(kParam, kPing)));
webrtc::EncoderSwitchRequestCallback::Config conf2{"VP9", kParam, kPong};
channel_->RequestEncoderSwitch(conf2);
rtc::Thread::Current()->ProcessMessages(30);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP9"));
EXPECT_THAT(codec.params, Contains(Pair(kParam, kPong)));
}
TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchIncorrectParam) {
const std::string kParam = "the-param";
const std::string kPing = "ping";
const std::string kPong = "pong";
cricket::VideoSendParameters parameters;
VideoCodec vp9 = GetEngineCodec("VP9");
vp9.params[kParam] = kPong;
parameters.codecs.push_back(vp9);
VideoCodec vp8 = GetEngineCodec("VP8");
vp8.params[kParam] = kPing;
parameters.codecs.push_back(vp8);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
channel_->SetVideoCodecSwitchingEnabled(true);
VideoCodec codec;
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP9"));
// RequestEncoderSwitch will post a task to the worker thread (which is also
// the current thread), hence the ProcessMessages call.
webrtc::EncoderSwitchRequestCallback::Config conf1{"VP8", kParam, kPing};
channel_->RequestEncoderSwitch(conf1);
rtc::Thread::Current()->ProcessMessages(30);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP8"));
EXPECT_THAT(codec.params, Contains(Pair(kParam, kPing)));
// Incorrect conf2.value, expect no codec switch.
webrtc::EncoderSwitchRequestCallback::Config conf2{"VP9", kParam, kPing};
channel_->RequestEncoderSwitch(conf2);
rtc::Thread::Current()->ProcessMessages(30);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP8"));
EXPECT_THAT(codec.params, Contains(Pair(kParam, kPing)));
}
TEST_F(WebRtcVideoChannelBaseTest,
RequestEncoderSwitchWithConfigBeforeEnabling) {
const std::string kParam = "the-param";
const std::string kPing = "ping";
const std::string kPong = "pong";
cricket::VideoSendParameters parameters;
VideoCodec vp9 = GetEngineCodec("VP9");
vp9.params[kParam] = kPong;
parameters.codecs.push_back(vp9);
VideoCodec vp8 = GetEngineCodec("VP8");
vp8.params[kParam] = kPing;
parameters.codecs.push_back(vp8);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
VideoCodec codec;
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP9"));
webrtc::EncoderSwitchRequestCallback::Config conf{"VP8", kParam, kPing};
channel_->RequestEncoderSwitch(conf);
// Enable codec switching after it has been requested.
channel_->SetVideoCodecSwitchingEnabled(true);
// RequestEncoderSwitch will post a task to the worker thread (which is also
// the current thread), hence the ProcessMessages call.
rtc::Thread::Current()->ProcessMessages(30);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_THAT(codec.name, Eq("VP8"));
EXPECT_THAT(codec.params, Contains(Pair(kParam, kPing)));
}
#endif // defined(RTC_ENABLE_VP9)
class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
public:
WebRtcVideoChannelTest() : WebRtcVideoChannelTest("") {}
explicit WebRtcVideoChannelTest(const char* field_trials)
: WebRtcVideoEngineTest(field_trials),
frame_source_(1280, 720, rtc::kNumMicrosecsPerSec / 30),
last_ssrc_(0) {}
void SetUp() override {
AddSupportedVideoCodecType("VP8");
AddSupportedVideoCodecType("VP9");
#if defined(WEBRTC_USE_H264)
AddSupportedVideoCodecType("H264");
#endif
fake_call_.reset(new FakeCall());
channel_.reset(engine_.CreateMediaChannel(
fake_call_.get(), GetMediaConfig(), VideoOptions(),
webrtc::CryptoOptions(), video_bitrate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
last_ssrc_ = 123;
send_parameters_.codecs = engine_.send_codecs();
recv_parameters_.codecs = engine_.recv_codecs();
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
}
void TearDown() override {
channel_->SetInterface(nullptr);
channel_ = nullptr;
}
void ResetTest() {
TearDown();
SetUp();
}
cricket::VideoCodec GetEngineCodec(const std::string& name) {
for (const cricket::VideoCodec& engine_codec : engine_.send_codecs()) {
if (absl::EqualsIgnoreCase(name, engine_codec.name))
return engine_codec;
}
// This point should never be reached.
ADD_FAILURE() << "Unrecognized codec name: " << name;
return cricket::VideoCodec();
}
cricket::VideoCodec DefaultCodec() { return GetEngineCodec("VP8"); }
// After receciving and processing the packet, enough time is advanced that
// the unsignalled receive stream cooldown is no longer in effect.
void ReceivePacketAndAdvanceTime(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
channel_->OnPacketReceived(packet, packet_time_us);
rtc::Thread::Current()->ProcessMessages(0);
fake_clock_.AdvanceTime(
webrtc::TimeDelta::Millis(kUnsignalledReceiveStreamCooldownMs));
}
protected:
FakeVideoSendStream* AddSendStream() {
return AddSendStream(StreamParams::CreateLegacy(++last_ssrc_));
}
FakeVideoSendStream* AddSendStream(const StreamParams& sp) {
size_t num_streams = fake_call_->GetVideoSendStreams().size();
EXPECT_TRUE(channel_->AddSendStream(sp));
std::vector<FakeVideoSendStream*> streams =
fake_call_->GetVideoSendStreams();
EXPECT_EQ(num_streams + 1, streams.size());
return streams[streams.size() - 1];
}
std::vector<FakeVideoSendStream*> GetFakeSendStreams() {
return fake_call_->GetVideoSendStreams();
}
FakeVideoReceiveStream* AddRecvStream() {
return AddRecvStream(StreamParams::CreateLegacy(++last_ssrc_));
}
FakeVideoReceiveStream* AddRecvStream(const StreamParams& sp) {
size_t num_streams = fake_call_->GetVideoReceiveStreams().size();
EXPECT_TRUE(channel_->AddRecvStream(sp));
std::vector<FakeVideoReceiveStream*> streams =
fake_call_->GetVideoReceiveStreams();
EXPECT_EQ(num_streams + 1, streams.size());
return streams[streams.size() - 1];
}
void SetSendCodecsShouldWorkForBitrates(const char* min_bitrate_kbps,
int expected_min_bitrate_bps,
const char* start_bitrate_kbps,
int expected_start_bitrate_bps,
const char* max_bitrate_kbps,
int expected_max_bitrate_bps) {
ExpectSetBitrateParameters(expected_min_bitrate_bps,
expected_start_bitrate_bps,
expected_max_bitrate_bps);
auto& codecs = send_parameters_.codecs;
codecs.clear();
codecs.push_back(GetEngineCodec("VP8"));
codecs[0].params[kCodecParamMinBitrate] = min_bitrate_kbps;
codecs[0].params[kCodecParamStartBitrate] = start_bitrate_kbps;
codecs[0].params[kCodecParamMaxBitrate] = max_bitrate_kbps;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
}
void ExpectSetBitrateParameters(int min_bitrate_bps,
int start_bitrate_bps,
int max_bitrate_bps) {
EXPECT_CALL(
*fake_call_->GetMockTransportControllerSend(),
SetSdpBitrateParameters(AllOf(
Field(&BitrateConstraints::min_bitrate_bps, min_bitrate_bps),
Field(&BitrateConstraints::start_bitrate_bps, start_bitrate_bps),
Field(&BitrateConstraints::max_bitrate_bps, max_bitrate_bps))));
}
void ExpectSetMaxBitrate(int max_bitrate_bps) {
EXPECT_CALL(*fake_call_->GetMockTransportControllerSend(),
SetSdpBitrateParameters(Field(
&BitrateConstraints::max_bitrate_bps, max_bitrate_bps)));
}
void TestExtmapAllowMixedCaller(bool extmap_allow_mixed) {
// For a caller, the answer will be applied in set remote description
// where SetSendParameters() is called.
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
send_parameters_.extmap_allow_mixed = extmap_allow_mixed;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
const webrtc::VideoSendStream::Config& config =
fake_call_->GetVideoSendStreams()[0]->GetConfig();
EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed);
}
void TestExtmapAllowMixedCallee(bool extmap_allow_mixed) {
// For a callee, the answer will be applied in set local description
// where SetExtmapAllowMixed() and AddSendStream() are called.
channel_->SetExtmapAllowMixed(extmap_allow_mixed);
EXPECT_TRUE(
channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
const webrtc::VideoSendStream::Config& config =
fake_call_->GetVideoSendStreams()[0]->GetConfig();
EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed);
}
void TestSetSendRtpHeaderExtensions(const std::string& ext_uri) {
// Enable extension.
const int id = 1;
cricket::VideoSendParameters parameters = send_parameters_;
parameters.extensions.push_back(RtpExtension(ext_uri, id));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(123));
// Verify the send extension id.
ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size());
EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id);
EXPECT_EQ(ext_uri, send_stream->GetConfig().rtp.extensions[0].uri);
// Verify call with same set of extensions returns true.
EXPECT_TRUE(channel_->SetSendParameters(parameters));
// Verify that SetSendRtpHeaderExtensions doesn't implicitly add them for
// receivers.
EXPECT_TRUE(AddRecvStream(cricket::StreamParams::CreateLegacy(123))
->GetConfig()
.rtp.extensions.empty());
// Verify that existing RTP header extensions can be removed.
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size());
send_stream = fake_call_->GetVideoSendStreams()[0];
EXPECT_TRUE(send_stream->GetConfig().rtp.extensions.empty());
// Verify that adding receive RTP header extensions adds them for existing
// streams.
EXPECT_TRUE(channel_->SetSendParameters(parameters));
send_stream = fake_call_->GetVideoSendStreams()[0];
ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size());
EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id);
EXPECT_EQ(ext_uri, send_stream->GetConfig().rtp.extensions[0].uri);
}
void TestSetRecvRtpHeaderExtensions(const std::string& ext_uri) {
// Enable extension.
const int id = 1;
cricket::VideoRecvParameters parameters = recv_parameters_;
parameters.extensions.push_back(RtpExtension(ext_uri, id));
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(123));
// Verify the recv extension id.
ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size());
EXPECT_EQ(id, recv_stream->GetConfig().rtp.extensions[0].id);
EXPECT_EQ(ext_uri, recv_stream->GetConfig().rtp.extensions[0].uri);
// Verify call with same set of extensions returns true.
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
// Verify that SetRecvRtpHeaderExtensions doesn't implicitly add them for
// senders.
EXPECT_TRUE(AddSendStream(cricket::StreamParams::CreateLegacy(123))
->GetConfig()
.rtp.extensions.empty());
// Verify that existing RTP header extensions can be removed.
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_TRUE(recv_stream->GetConfig().rtp.extensions.empty());
// Verify that adding receive RTP header extensions adds them for existing
// streams.
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size());
EXPECT_EQ(id, recv_stream->GetConfig().rtp.extensions[0].id);
EXPECT_EQ(ext_uri, recv_stream->GetConfig().rtp.extensions[0].uri);
}
void TestLossNotificationState(bool expect_lntf_enabled) {
AssignDefaultCodec();
VerifyCodecHasDefaultFeedbackParams(default_codec_, expect_lntf_enabled);
cricket::VideoSendParameters parameters;
parameters.codecs = engine_.send_codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(channel_->SetSend(true));
// Send side.
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(1));
EXPECT_EQ(send_stream->GetConfig().rtp.lntf.enabled, expect_lntf_enabled);
// Receiver side.
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(1));
EXPECT_EQ(recv_stream->GetConfig().rtp.lntf.enabled, expect_lntf_enabled);
}
void TestExtensionFilter(const std::vector<std::string>& extensions,
const std::string& expected_extension) {
cricket::VideoSendParameters parameters = send_parameters_;
int expected_id = -1;
int id = 1;
for (const std::string& extension : extensions) {
if (extension == expected_extension)
expected_id = id;
parameters.extensions.push_back(RtpExtension(extension, id++));
}
EXPECT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(123));
// Verify that only one of them has been set, and that it is the one with
// highest priority (transport sequence number).
ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size());
EXPECT_EQ(expected_id, send_stream->GetConfig().rtp.extensions[0].id);
EXPECT_EQ(expected_extension,
send_stream->GetConfig().rtp.extensions[0].uri);
}
void TestDegradationPreference(bool resolution_scaling_enabled,
bool fps_scaling_enabled);
void TestCpuAdaptation(bool enable_overuse, bool is_screenshare);
void TestReceiverLocalSsrcConfiguration(bool receiver_first);
void TestReceiveUnsignaledSsrcPacket(uint8_t payload_type,
bool expect_created_receive_stream);
FakeVideoSendStream* SetDenoisingOption(
uint32_t ssrc,
webrtc::test::FrameForwarder* frame_forwarder,
bool enabled) {
cricket::VideoOptions options;
options.video_noise_reduction = enabled;
EXPECT_TRUE(channel_->SetVideoSend(ssrc, &options, frame_forwarder));
// Options only take effect on the next frame.
frame_forwarder->IncomingCapturedFrame(frame_source_.GetFrame());
return fake_call_->GetVideoSendStreams().back();
}
FakeVideoSendStream* SetUpSimulcast(bool enabled, bool with_rtx) {
const int kRtxSsrcOffset = 0xDEADBEEF;
last_ssrc_ += 3;
std::vector<uint32_t> ssrcs;
std::vector<uint32_t> rtx_ssrcs;
uint32_t num_streams = enabled ? 3 : 1;
for (uint32_t i = 0; i < num_streams; ++i) {
uint32_t ssrc = last_ssrc_ + i;
ssrcs.push_back(ssrc);
if (with_rtx) {
rtx_ssrcs.push_back(ssrc + kRtxSsrcOffset);
}
}
if (with_rtx) {
return AddSendStream(
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs));
}
return AddSendStream(CreateSimStreamParams("cname", ssrcs));
}
int GetMaxEncoderBitrate() {
std::vector<FakeVideoSendStream*> streams =
fake_call_->GetVideoSendStreams();
EXPECT_EQ(1u, streams.size());
FakeVideoSendStream* stream = streams[streams.size() - 1];
EXPECT_EQ(1u, stream->GetEncoderConfig().number_of_streams);
return stream->GetVideoStreams()[0].max_bitrate_bps;
}
void SetAndExpectMaxBitrate(int global_max,
int stream_max,
int expected_encoder_bitrate) {
VideoSendParameters limited_send_params = send_parameters_;
limited_send_params.max_bandwidth_bps = global_max;
EXPECT_TRUE(channel_->SetSendParameters(limited_send_params));
webrtc::RtpParameters parameters =
channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
parameters.encodings[0].max_bitrate_bps = stream_max;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Read back the parameteres and verify they have the correct value
parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_EQ(stream_max, parameters.encodings[0].max_bitrate_bps);
// Verify that the new value propagated down to the encoder
EXPECT_EQ(expected_encoder_bitrate, GetMaxEncoderBitrate());
}
// Values from kSimulcastConfigs in simulcast.cc.
const std::vector<webrtc::VideoStream> GetSimulcastBitrates720p() const {
std::vector<webrtc::VideoStream> layers(3);
layers[0].min_bitrate_bps = 30000;
layers[0].target_bitrate_bps = 150000;
layers[0].max_bitrate_bps = 200000;
layers[1].min_bitrate_bps = 150000;
layers[1].target_bitrate_bps = 500000;
layers[1].max_bitrate_bps = 700000;
layers[2].min_bitrate_bps = 600000;
layers[2].target_bitrate_bps = 2500000;
layers[2].max_bitrate_bps = 2500000;
return layers;
}
cricket::FakeFrameSource frame_source_;
std::unique_ptr<FakeCall> fake_call_;
std::unique_ptr<VideoMediaChannel> channel_;
cricket::VideoSendParameters send_parameters_;
cricket::VideoRecvParameters recv_parameters_;
uint32_t last_ssrc_;
};
TEST_F(WebRtcVideoChannelTest, SetsSyncGroupFromSyncLabel) {
const uint32_t kVideoSsrc = 123;
const std::string kSyncLabel = "AvSyncLabel";
cricket::StreamParams sp = cricket::StreamParams::CreateLegacy(kVideoSsrc);
sp.set_stream_ids({kSyncLabel});
EXPECT_TRUE(channel_->AddRecvStream(sp));
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
EXPECT_EQ(kSyncLabel,
fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group)
<< "SyncGroup should be set based on sync_label";
}
TEST_F(WebRtcVideoChannelTest, RecvStreamWithSimAndRtx) {
cricket::VideoSendParameters parameters;
parameters.codecs = engine_.send_codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(channel_->SetSend(true));
parameters.conference_mode = true;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
// Send side.
const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1);
const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1);
FakeVideoSendStream* send_stream = AddSendStream(
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs));
ASSERT_EQ(rtx_ssrcs.size(), send_stream->GetConfig().rtp.rtx.ssrcs.size());
for (size_t i = 0; i < rtx_ssrcs.size(); ++i)
EXPECT_EQ(rtx_ssrcs[i], send_stream->GetConfig().rtp.rtx.ssrcs[i]);
// Receiver side.
FakeVideoReceiveStream* recv_stream = AddRecvStream(
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs));
EXPECT_FALSE(
recv_stream->GetConfig().rtp.rtx_associated_payload_types.empty());
EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
<< "RTX should be mapped for all decoders/payload types.";
EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
GetEngineCodec("red").id))
<< "RTX should be mapped for the RED payload type";
EXPECT_EQ(rtx_ssrcs[0], recv_stream->GetConfig().rtp.rtx_ssrc);
}
TEST_F(WebRtcVideoChannelTest, RecvStreamWithRtx) {
// Setup one channel with an associated RTX stream.
cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]);
FakeVideoReceiveStream* recv_stream = AddRecvStream(params);
EXPECT_EQ(kRtxSsrcs1[0], recv_stream->GetConfig().rtp.rtx_ssrc);
EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
<< "RTX should be mapped for all decoders/payload types.";
EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
GetEngineCodec("red").id))
<< "RTX should be mapped for the RED payload type";
}
TEST_F(WebRtcVideoChannelTest, RecvStreamNoRtx) {
// Setup one channel without an associated RTX stream.
cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
FakeVideoReceiveStream* recv_stream = AddRecvStream(params);
ASSERT_EQ(0U, recv_stream->GetConfig().rtp.rtx_ssrc);
}
// Test propagation of extmap allow mixed setting.
TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedAsCaller) {
TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/true);
}
TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedDisabledAsCaller) {
TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/false);
}
TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedAsCallee) {
TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/true);
}
TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedDisabledAsCallee) {
TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/false);
}
TEST_F(WebRtcVideoChannelTest, NoHeaderExtesionsByDefault) {
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcs1[0]));
ASSERT_TRUE(send_stream->GetConfig().rtp.extensions.empty());
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrcs1[0]));
ASSERT_TRUE(recv_stream->GetConfig().rtp.extensions.empty());
}
// Test support for RTP timestamp offset header extension.
TEST_F(WebRtcVideoChannelTest, SendRtpTimestampOffsetHeaderExtensions) {
TestSetSendRtpHeaderExtensions(RtpExtension::kTimestampOffsetUri);
}
TEST_F(WebRtcVideoChannelTest, RecvRtpTimestampOffsetHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(RtpExtension::kTimestampOffsetUri);
}
// Test support for absolute send time header extension.
TEST_F(WebRtcVideoChannelTest, SendAbsoluteSendTimeHeaderExtensions) {
TestSetSendRtpHeaderExtensions(RtpExtension::kAbsSendTimeUri);
}
TEST_F(WebRtcVideoChannelTest, RecvAbsoluteSendTimeHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(RtpExtension::kAbsSendTimeUri);
}
TEST_F(WebRtcVideoChannelTest, FiltersExtensionsPicksTransportSeqNum) {
webrtc::test::ScopedFieldTrials override_field_trials(
"WebRTC-FilterAbsSendTimeExtension/Enabled/");
// Enable three redundant extensions.
std::vector<std::string> extensions;
extensions.push_back(RtpExtension::kAbsSendTimeUri);
extensions.push_back(RtpExtension::kTimestampOffsetUri);
extensions.push_back(RtpExtension::kTransportSequenceNumberUri);
TestExtensionFilter(extensions, RtpExtension::kTransportSequenceNumberUri);
}
TEST_F(WebRtcVideoChannelTest, FiltersExtensionsPicksAbsSendTime) {
// Enable two redundant extensions.
std::vector<std::string> extensions;
extensions.push_back(RtpExtension::kAbsSendTimeUri);
extensions.push_back(RtpExtension::kTimestampOffsetUri);
TestExtensionFilter(extensions, RtpExtension::kAbsSendTimeUri);
}
// Test support for transport sequence number header extension.
TEST_F(WebRtcVideoChannelTest, SendTransportSequenceNumberHeaderExtensions) {
TestSetSendRtpHeaderExtensions(RtpExtension::kTransportSequenceNumberUri);
}
TEST_F(WebRtcVideoChannelTest, RecvTransportSequenceNumberHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(RtpExtension::kTransportSequenceNumberUri);
}
// Test support for video rotation header extension.
TEST_F(WebRtcVideoChannelTest, SendVideoRotationHeaderExtensions) {
TestSetSendRtpHeaderExtensions(RtpExtension::kVideoRotationUri);
}
TEST_F(WebRtcVideoChannelTest, RecvVideoRotationHeaderExtensions) {
TestSetRecvRtpHeaderExtensions(RtpExtension::kVideoRotationUri);
}
TEST_F(WebRtcVideoChannelTest, IdenticalSendExtensionsDoesntRecreateStream) {
const int kAbsSendTimeId = 1;
const int kVideoRotationId = 2;
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kVideoRotationUri, kVideoRotationId));
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(123));
EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
ASSERT_EQ(2u, send_stream->GetConfig().rtp.extensions.size());
// Setting the same extensions (even if in different order) shouldn't
// reallocate the stream.
absl::c_reverse(send_parameters_.extensions);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
// Setting different extensions should recreate the stream.
send_parameters_.extensions.resize(1);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_EQ(2, fake_call_->GetNumCreatedSendStreams());
}
TEST_F(WebRtcVideoChannelTest, IdenticalRecvExtensionsDoesntRecreateStream) {
const int kTOffsetId = 1;
const int kAbsSendTimeId = 2;
const int kVideoRotationId = 3;
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOffsetId));
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kVideoRotationUri, kVideoRotationId));
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(123));
EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
ASSERT_EQ(3u, recv_stream->GetConfig().rtp.extensions.size());
// Setting the same extensions (even if in different order) shouldn't
// reallocate the stream.
absl::c_reverse(recv_parameters_.extensions);
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
// Setting different extensions should not require the stream to be recreated.
recv_parameters_.extensions.resize(1);
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
}
TEST_F(WebRtcVideoChannelTest,
SetSendRtpHeaderExtensionsExcludeUnsupportedExtensions) {
const int kUnsupportedId = 1;
const int kTOffsetId = 2;
send_parameters_.extensions.push_back(
RtpExtension(kUnsupportedExtensionName, kUnsupportedId));
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOffsetId));
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(123));
// Only timestamp offset extension is set to send stream,
// unsupported rtp extension is ignored.
ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size());
EXPECT_STREQ(RtpExtension::kTimestampOffsetUri,
send_stream->GetConfig().rtp.extensions[0].uri.c_str());
}
TEST_F(WebRtcVideoChannelTest,
SetRecvRtpHeaderExtensionsExcludeUnsupportedExtensions) {
const int kUnsupportedId = 1;
const int kTOffsetId = 2;
recv_parameters_.extensions.push_back(
RtpExtension(kUnsupportedExtensionName, kUnsupportedId));
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOffsetId));
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(123));
// Only timestamp offset extension is set to receive stream,
// unsupported rtp extension is ignored.
ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size());
EXPECT_STREQ(RtpExtension::kTimestampOffsetUri,
recv_stream->GetConfig().rtp.extensions[0].uri.c_str());
}
TEST_F(WebRtcVideoChannelTest, SetSendRtpHeaderExtensionsRejectsIncorrectIds) {
const int kIncorrectIds[] = {-2, -1, 0, 15, 16};
for (size_t i = 0; i < arraysize(kIncorrectIds); ++i) {
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kIncorrectIds[i]));
EXPECT_FALSE(channel_->SetSendParameters(send_parameters_))
<< "Bad extension id '" << kIncorrectIds[i] << "' accepted.";
}
}
TEST_F(WebRtcVideoChannelTest, SetRecvRtpHeaderExtensionsRejectsIncorrectIds) {
const int kIncorrectIds[] = {-2, -1, 0, 15, 16};
for (size_t i = 0; i < arraysize(kIncorrectIds); ++i) {
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kIncorrectIds[i]));
EXPECT_FALSE(channel_->SetRecvParameters(recv_parameters_))
<< "Bad extension id '" << kIncorrectIds[i] << "' accepted.";
}
}
TEST_F(WebRtcVideoChannelTest, SetSendRtpHeaderExtensionsRejectsDuplicateIds) {
const int id = 1;
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, id));
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, id));
EXPECT_FALSE(channel_->SetSendParameters(send_parameters_));
// Duplicate entries are also not supported.
send_parameters_.extensions.clear();
send_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, id));
send_parameters_.extensions.push_back(send_parameters_.extensions.back());
EXPECT_FALSE(channel_->SetSendParameters(send_parameters_));
}
TEST_F(WebRtcVideoChannelTest, SetRecvRtpHeaderExtensionsRejectsDuplicateIds) {
const int id = 1;
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, id));
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, id));
EXPECT_FALSE(channel_->SetRecvParameters(recv_parameters_));
// Duplicate entries are also not supported.
recv_parameters_.extensions.clear();
recv_parameters_.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, id));
recv_parameters_.extensions.push_back(recv_parameters_.extensions.back());
EXPECT_FALSE(channel_->SetRecvParameters(recv_parameters_));
}
TEST_F(WebRtcVideoChannelTest, AddRecvStreamOnlyUsesOneReceiveStream) {
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
}
TEST_F(WebRtcVideoChannelTest, RtcpIsCompoundByDefault) {
FakeVideoReceiveStream* stream = AddRecvStream();
EXPECT_EQ(webrtc::RtcpMode::kCompound, stream->GetConfig().rtp.rtcp_mode);
}
TEST_F(WebRtcVideoChannelTest, TransportCcIsEnabledByDefault) {
FakeVideoReceiveStream* stream = AddRecvStream();
EXPECT_TRUE(stream->GetConfig().rtp.transport_cc);
}
TEST_F(WebRtcVideoChannelTest, TransportCcCanBeEnabledAndDisabled) {
FakeVideoReceiveStream* stream = AddRecvStream();
EXPECT_TRUE(stream->GetConfig().rtp.transport_cc);
// Verify that transport cc feedback is turned off when send(!) codecs without
// transport cc feedback are set.
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(RemoveFeedbackParams(GetEngineCodec("VP8")));
EXPECT_TRUE(parameters.codecs[0].feedback_params.params().empty());
EXPECT_TRUE(channel_->SetSendParameters(parameters));
stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_FALSE(stream->GetConfig().rtp.transport_cc);
// Verify that transport cc feedback is turned on when setting default codecs
// since the default codecs have transport cc feedback enabled.
parameters.codecs = engine_.send_codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_TRUE(stream->GetConfig().rtp.transport_cc);
}
TEST_F(WebRtcVideoChannelTest, LossNotificationIsDisabledByDefault) {
TestLossNotificationState(false);
}
TEST_F(WebRtcVideoChannelTest, LossNotificationIsEnabledByFieldTrial) {
RTC_DCHECK(!override_field_trials_);
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-RtcpLossNotification/Enabled/");
ResetTest();
TestLossNotificationState(true);
}
TEST_F(WebRtcVideoChannelTest, LossNotificationCanBeEnabledAndDisabled) {
RTC_DCHECK(!override_field_trials_);
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-RtcpLossNotification/Enabled/");
ResetTest();
AssignDefaultCodec();
VerifyCodecHasDefaultFeedbackParams(default_codec_, true);
{
cricket::VideoSendParameters parameters;
parameters.codecs = engine_.send_codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(channel_->SetSend(true));
}
// Start with LNTF enabled.
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(1));
ASSERT_TRUE(send_stream->GetConfig().rtp.lntf.enabled);
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(1));
ASSERT_TRUE(recv_stream->GetConfig().rtp.lntf.enabled);
// Verify that LNTF is turned off when send(!) codecs without LNTF are set.
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(RemoveFeedbackParams(GetEngineCodec("VP8")));
EXPECT_TRUE(parameters.codecs[0].feedback_params.params().empty());
EXPECT_TRUE(channel_->SetSendParameters(parameters));
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_FALSE(recv_stream->GetConfig().rtp.lntf.enabled);
send_stream = fake_call_->GetVideoSendStreams()[0];
EXPECT_FALSE(send_stream->GetConfig().rtp.lntf.enabled);
// Setting the default codecs again, including VP8, turns LNTF back on.
parameters.codecs = engine_.send_codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_TRUE(recv_stream->GetConfig().rtp.lntf.enabled);
send_stream = fake_call_->GetVideoSendStreams()[0];
EXPECT_TRUE(send_stream->GetConfig().rtp.lntf.enabled);
}
TEST_F(WebRtcVideoChannelTest, NackIsEnabledByDefault) {
AssignDefaultCodec();
VerifyCodecHasDefaultFeedbackParams(default_codec_, false);
cricket::VideoSendParameters parameters;
parameters.codecs = engine_.send_codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(channel_->SetSend(true));
// Send side.
FakeVideoSendStream* send_stream =
AddSendStream(cricket::StreamParams::CreateLegacy(1));
EXPECT_GT(send_stream->GetConfig().rtp.nack.rtp_history_ms, 0);
// Receiver side.
FakeVideoReceiveStream* recv_stream =
AddRecvStream(cricket::StreamParams::CreateLegacy(1));
EXPECT_GT(recv_stream->GetConfig().rtp.nack.rtp_history_ms, 0);
// Nack history size should match between sender and receiver.
EXPECT_EQ(send_stream->GetConfig().rtp.nack.rtp_history_ms,
recv_stream->GetConfig().rtp.nack.rtp_history_ms);
}
TEST_F(WebRtcVideoChannelTest, NackCanBeEnabledAndDisabled) {
FakeVideoSendStream* send_stream = AddSendStream();
FakeVideoReceiveStream* recv_stream = AddRecvStream();
EXPECT_GT(recv_stream->GetConfig().rtp.nack.rtp_history_ms, 0);
EXPECT_GT(send_stream->GetConfig().rtp.nack.rtp_history_ms, 0);
// Verify that NACK is turned off when send(!) codecs without NACK are set.
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(RemoveFeedbackParams(GetEngineCodec("VP8")));
EXPECT_TRUE(parameters.codecs[0].feedback_params.params().empty());
EXPECT_TRUE(channel_->SetSendParameters(parameters));
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(0, recv_stream->GetConfig().rtp.nack.rtp_history_ms);
send_stream = fake_call_->GetVideoSendStreams()[0];
EXPECT_EQ(0, send_stream->GetConfig().rtp.nack.rtp_history_ms);
// Verify that NACK is turned on when setting default codecs since the
// default codecs have NACK enabled.
parameters.codecs = engine_.send_codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_GT(recv_stream->GetConfig().rtp.nack.rtp_history_ms, 0);
send_stream = fake_call_->GetVideoSendStreams()[0];
EXPECT_GT(send_stream->GetConfig().rtp.nack.rtp_history_ms, 0);
}
// This test verifies that new frame sizes reconfigures encoders even though not
// (yet) sending. The purpose of this is to permit encoding as quickly as
// possible once we start sending. Likely the frames being input are from the
// same source that will be sent later, which just means that we're ready
// earlier.
TEST_F(WebRtcVideoChannelTest, ReconfiguresEncodersWhenNotSending) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
channel_->SetSend(false);
FakeVideoSendStream* stream = AddSendStream();
// No frames entered.
std::vector<webrtc::VideoStream> streams = stream->GetVideoStreams();
EXPECT_EQ(0u, streams[0].width);
EXPECT_EQ(0u, streams[0].height);
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Frame entered, should be reconfigured to new dimensions.
streams = stream->GetVideoStreams();
EXPECT_EQ(rtc::checked_cast<size_t>(1280), streams[0].width);
EXPECT_EQ(rtc::checked_cast<size_t>(720), streams[0].height);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, UsesCorrectSettingsForScreencast) {
static const int kScreenshareMinBitrateKbps = 800;
cricket::VideoCodec codec = GetEngineCodec("VP8");
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
VideoOptions min_bitrate_options;
min_bitrate_options.screencast_min_bitrate_kbps = kScreenshareMinBitrateKbps;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &min_bitrate_options,
&frame_forwarder));
EXPECT_TRUE(channel_->SetSend(true));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size());
FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames());
// Verify non-screencast settings.
webrtc::VideoEncoderConfig encoder_config =
send_stream->GetEncoderConfig().Copy();
EXPECT_EQ(webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo,
encoder_config.content_type);
std::vector<webrtc::VideoStream> streams = send_stream->GetVideoStreams();
EXPECT_EQ(rtc::checked_cast<size_t>(1280), streams.front().width);
EXPECT_EQ(rtc::checked_cast<size_t>(720), streams.front().height);
EXPECT_EQ(0, encoder_config.min_transmit_bitrate_bps)
<< "Non-screenshare shouldn't use min-transmit bitrate.";
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames());
VideoOptions screencast_options;
screencast_options.is_screencast = true;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &screencast_options,
&frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Send stream recreated after option change.
ASSERT_EQ(2, fake_call_->GetNumCreatedSendStreams());
send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames());
// Verify screencast settings.
encoder_config = send_stream->GetEncoderConfig().Copy();
EXPECT_EQ(webrtc::VideoEncoderConfig::ContentType::kScreen,
encoder_config.content_type);
EXPECT_EQ(kScreenshareMinBitrateKbps * 1000,
encoder_config.min_transmit_bitrate_bps);
streams = send_stream->GetVideoStreams();
EXPECT_EQ(rtc::checked_cast<size_t>(1280), streams.front().width);
EXPECT_EQ(rtc::checked_cast<size_t>(720), streams.front().height);
EXPECT_FALSE(streams[0].num_temporal_layers.has_value());
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
ConferenceModeScreencastConfiguresTemporalLayer) {
static const int kConferenceScreencastTemporalBitrateBps = 200 * 1000;
send_parameters_.conference_mode = true;
channel_->SetSendParameters(send_parameters_);
AddSendStream();
VideoOptions options;
options.is_screencast = true;
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
EXPECT_TRUE(channel_->SetSend(true));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size());
FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
webrtc::VideoEncoderConfig encoder_config =
send_stream->GetEncoderConfig().Copy();
// Verify screencast settings.
encoder_config = send_stream->GetEncoderConfig().Copy();
EXPECT_EQ(webrtc::VideoEncoderConfig::ContentType::kScreen,
encoder_config.content_type);
std::vector<webrtc::VideoStream> streams = send_stream->GetVideoStreams();
ASSERT_EQ(1u, streams.size());
ASSERT_EQ(2u, streams[0].num_temporal_layers);
EXPECT_EQ(kConferenceScreencastTemporalBitrateBps,
streams[0].target_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, SuspendBelowMinBitrateDisabledByDefault) {
FakeVideoSendStream* stream = AddSendStream();
EXPECT_FALSE(stream->GetConfig().suspend_below_min_bitrate);
}
TEST_F(WebRtcVideoChannelTest, SetMediaConfigSuspendBelowMinBitrate) {
MediaConfig media_config = GetMediaConfig();
media_config.video.suspend_below_min_bitrate = true;
channel_.reset(engine_.CreateMediaChannel(
fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
channel_->SetSendParameters(send_parameters_);
FakeVideoSendStream* stream = AddSendStream();
EXPECT_TRUE(stream->GetConfig().suspend_below_min_bitrate);
media_config.video.suspend_below_min_bitrate = false;
channel_.reset(engine_.CreateMediaChannel(
fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
channel_->SetSendParameters(send_parameters_);
stream = AddSendStream();
EXPECT_FALSE(stream->GetConfig().suspend_below_min_bitrate);
}
TEST_F(WebRtcVideoChannelTest, Vp8DenoisingEnabledByDefault) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoCodecVP8 vp8_settings;
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
EXPECT_TRUE(vp8_settings.denoisingOn);
}
TEST_F(WebRtcVideoChannelTest, VerifyVp8SpecificSettings) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
// Single-stream settings should apply with RTX as well (verifies that we
// check number of regular SSRCs and not StreamParams::ssrcs which contains
// both RTX and regular SSRCs).
FakeVideoSendStream* stream = SetUpSimulcast(false, true);
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
webrtc::VideoCodecVP8 vp8_settings;
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
EXPECT_TRUE(vp8_settings.denoisingOn)
<< "VP8 denoising should be on by default.";
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false);
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
EXPECT_FALSE(vp8_settings.denoisingOn);
EXPECT_TRUE(vp8_settings.automaticResizeOn);
EXPECT_TRUE(vp8_settings.frameDroppingOn);
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, true);
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
EXPECT_TRUE(vp8_settings.denoisingOn);
EXPECT_TRUE(vp8_settings.automaticResizeOn);
EXPECT_TRUE(vp8_settings.frameDroppingOn);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
stream = SetUpSimulcast(true, false);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
EXPECT_EQ(3u, stream->GetVideoStreams().size());
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
// Autmatic resize off when using simulcast.
EXPECT_FALSE(vp8_settings.automaticResizeOn);
EXPECT_TRUE(vp8_settings.frameDroppingOn);
// In screen-share mode, denoising is forced off.
VideoOptions options;
options.is_screencast = true;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false);
EXPECT_EQ(3u, stream->GetVideoStreams().size());
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
EXPECT_FALSE(vp8_settings.denoisingOn);
// Resizing and frame dropping always off for screen sharing.
EXPECT_FALSE(vp8_settings.automaticResizeOn);
EXPECT_FALSE(vp8_settings.frameDroppingOn);
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, true);
ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
EXPECT_FALSE(vp8_settings.denoisingOn);
EXPECT_FALSE(vp8_settings.automaticResizeOn);
EXPECT_FALSE(vp8_settings.frameDroppingOn);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
// Test that setting the same options doesn't result in the encoder being
// reconfigured.
TEST_F(WebRtcVideoChannelTest, SetIdenticalOptionsDoesntReconfigureEncoder) {
VideoOptions options;
webrtc::test::FrameForwarder frame_forwarder;
AddSendStream();
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Expect 1 reconfigurations at this point from the initial configuration.
EXPECT_EQ(1, send_stream->num_encoder_reconfigurations());
// Set the options one more time and expect no additional reconfigurations.
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
EXPECT_EQ(1, send_stream->num_encoder_reconfigurations());
// Change `options` and expect 2 reconfigurations.
options.video_noise_reduction = true;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
EXPECT_EQ(2, send_stream->num_encoder_reconfigurations());
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
class Vp9SettingsTest : public WebRtcVideoChannelTest {
public:
Vp9SettingsTest() : Vp9SettingsTest("") {}
explicit Vp9SettingsTest(const char* field_trials)
: WebRtcVideoChannelTest(field_trials) {
encoder_factory_->AddSupportedVideoCodecType("VP9");
}
virtual ~Vp9SettingsTest() {}
protected:
void TearDown() override {
// Remove references to encoder_factory_ since this will be destroyed
// before channel_ and engine_.
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
}
};
TEST_F(Vp9SettingsTest, VerifyVp9SpecificSettings) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP9"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = SetUpSimulcast(false, false);
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
webrtc::VideoCodecVP9 vp9_settings;
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
EXPECT_TRUE(vp9_settings.denoisingOn)
<< "VP9 denoising should be on by default.";
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false);
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
EXPECT_FALSE(vp9_settings.denoisingOn);
// Frame dropping always on for real time video.
EXPECT_TRUE(vp9_settings.frameDroppingOn);
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, true);
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
EXPECT_TRUE(vp9_settings.denoisingOn);
EXPECT_TRUE(vp9_settings.frameDroppingOn);
// In screen-share mode, denoising is forced off.
VideoOptions options;
options.is_screencast = true;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false);
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
EXPECT_FALSE(vp9_settings.denoisingOn);
// Frame dropping always on for screen sharing.
EXPECT_TRUE(vp9_settings.frameDroppingOn);
stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false);
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
EXPECT_FALSE(vp9_settings.denoisingOn);
EXPECT_TRUE(vp9_settings.frameDroppingOn);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(Vp9SettingsTest, MultipleSsrcsEnablesSvc) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP9"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
FakeVideoSendStream* stream =
AddSendStream(CreateSimStreamParams("cname", ssrcs));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
webrtc::VideoCodecVP9 vp9_settings;
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
const size_t kNumSpatialLayers = ssrcs.size();
const size_t kNumTemporalLayers = 3;
EXPECT_EQ(vp9_settings.numberOfSpatialLayers, kNumSpatialLayers);
EXPECT_EQ(vp9_settings.numberOfTemporalLayers, kNumTemporalLayers);
EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, nullptr));
}
TEST_F(Vp9SettingsTest, SvcModeCreatesSingleRtpStream) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP9"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
FakeVideoSendStream* stream =
AddSendStream(CreateSimStreamParams("cname", ssrcs));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
// Despite 3 ssrcs provided, single layer is used.
EXPECT_EQ(1u, config.rtp.ssrcs.size());
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
webrtc::VideoCodecVP9 vp9_settings;
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
const size_t kNumSpatialLayers = ssrcs.size();
EXPECT_EQ(vp9_settings.numberOfSpatialLayers, kNumSpatialLayers);
EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, nullptr));
}
TEST_F(Vp9SettingsTest, AllEncodingParametersCopied) {
cricket::VideoSendParameters send_parameters;
send_parameters.codecs.push_back(GetEngineCodec("VP9"));
ASSERT_TRUE(channel_->SetSendParameters(send_parameters));
const size_t kNumSpatialLayers = 3;
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
FakeVideoSendStream* stream =
AddSendStream(CreateSimStreamParams("cname", ssrcs));
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(ssrcs[0]);
ASSERT_EQ(kNumSpatialLayers, parameters.encodings.size());
ASSERT_TRUE(parameters.encodings[0].active);
ASSERT_TRUE(parameters.encodings[1].active);
ASSERT_TRUE(parameters.encodings[2].active);
// Invert value to verify copying.
parameters.encodings[1].active = false;
EXPECT_TRUE(channel_->SetRtpSendParameters(ssrcs[0], parameters).ok());
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
// number_of_streams should be 1 since all spatial layers are sent on the
// same SSRC. But encoding parameters of all layers is supposed to be copied
// and stored in simulcast_layers[].
EXPECT_EQ(1u, encoder_config.number_of_streams);
EXPECT_EQ(encoder_config.simulcast_layers.size(), kNumSpatialLayers);
EXPECT_TRUE(encoder_config.simulcast_layers[0].active);
EXPECT_FALSE(encoder_config.simulcast_layers[1].active);
EXPECT_TRUE(encoder_config.simulcast_layers[2].active);
}
class Vp9SettingsTestWithFieldTrial
: public Vp9SettingsTest,
public ::testing::WithParamInterface<
::testing::tuple<const char*, int, int, webrtc::InterLayerPredMode>> {
protected:
Vp9SettingsTestWithFieldTrial()
: Vp9SettingsTest(::testing::get<0>(GetParam())),
num_spatial_layers_(::testing::get<1>(GetParam())),
num_temporal_layers_(::testing::get<2>(GetParam())),
inter_layer_pred_mode_(::testing::get<3>(GetParam())) {}
void VerifySettings(int num_spatial_layers,
int num_temporal_layers,
webrtc::InterLayerPredMode interLayerPred) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP9"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = SetUpSimulcast(false, false);
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
webrtc::VideoCodecVP9 vp9_settings;
ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
EXPECT_EQ(num_spatial_layers, vp9_settings.numberOfSpatialLayers);
EXPECT_EQ(num_temporal_layers, vp9_settings.numberOfTemporalLayers);
EXPECT_EQ(inter_layer_pred_mode_, vp9_settings.interLayerPred);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
const uint8_t num_spatial_layers_;
const uint8_t num_temporal_layers_;
const webrtc::InterLayerPredMode inter_layer_pred_mode_;
};
TEST_P(Vp9SettingsTestWithFieldTrial, VerifyCodecSettings) {
VerifySettings(num_spatial_layers_, num_temporal_layers_,
inter_layer_pred_mode_);
}
INSTANTIATE_TEST_SUITE_P(
All,
Vp9SettingsTestWithFieldTrial,
Values(
std::make_tuple("", 1, 1, webrtc::InterLayerPredMode::kOnKeyPic),
std::make_tuple("WebRTC-SupportVP9SVC/Default/",
1,
1,
webrtc::InterLayerPredMode::kOnKeyPic),
std::make_tuple("WebRTC-SupportVP9SVC/EnabledByFlag_2SL3TL/",
2,
3,
webrtc::InterLayerPredMode::kOnKeyPic),
std::make_tuple("WebRTC-Vp9InterLayerPred/Default/",
1,
1,
webrtc::InterLayerPredMode::kOnKeyPic),
std::make_tuple("WebRTC-Vp9InterLayerPred/Disabled/",
1,
1,
webrtc::InterLayerPredMode::kOnKeyPic),
std::make_tuple(
"WebRTC-Vp9InterLayerPred/Enabled,inter_layer_pred_mode:off/",
1,
1,
webrtc::InterLayerPredMode::kOff),
std::make_tuple(
"WebRTC-Vp9InterLayerPred/Enabled,inter_layer_pred_mode:on/",
1,
1,
webrtc::InterLayerPredMode::kOn),
std::make_tuple(
"WebRTC-Vp9InterLayerPred/Enabled,inter_layer_pred_mode:onkeypic/",
1,
1,
webrtc::InterLayerPredMode::kOnKeyPic)));
TEST_F(WebRtcVideoChannelTest, VerifyMinBitrate) {
std::vector<webrtc::VideoStream> streams = AddSendStream()->GetVideoStreams();
ASSERT_EQ(1u, streams.size());
EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps, streams[0].min_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest, VerifyMinBitrateWithForcedFallbackFieldTrial) {
RTC_DCHECK(!override_field_trials_);
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-VP8-Forced-Fallback-Encoder-v2/Enabled-1,2,34567/");
std::vector<webrtc::VideoStream> streams = AddSendStream()->GetVideoStreams();
ASSERT_EQ(1u, streams.size());
EXPECT_EQ(34567, streams[0].min_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest,
BalancedDegradationPreferenceNotSupportedWithoutFieldtrial) {
RTC_DCHECK(!override_field_trials_);
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-Video-BalancedDegradation/Disabled/");
const bool kResolutionScalingEnabled = true;
const bool kFpsScalingEnabled = false;
TestDegradationPreference(kResolutionScalingEnabled, kFpsScalingEnabled);
}
TEST_F(WebRtcVideoChannelTest,
BalancedDegradationPreferenceSupportedBehindFieldtrial) {
RTC_DCHECK(!override_field_trials_);
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-Video-BalancedDegradation/Enabled/");
const bool kResolutionScalingEnabled = true;
const bool kFpsScalingEnabled = true;
TestDegradationPreference(kResolutionScalingEnabled, kFpsScalingEnabled);
}
TEST_F(WebRtcVideoChannelTest, AdaptsOnOveruse) {
TestCpuAdaptation(true, false);
}
TEST_F(WebRtcVideoChannelTest, DoesNotAdaptOnOveruseWhenDisabled) {
TestCpuAdaptation(false, false);
}
TEST_F(WebRtcVideoChannelTest, DoesNotAdaptWhenScreeensharing) {
TestCpuAdaptation(false, true);
}
TEST_F(WebRtcVideoChannelTest, DoesNotAdaptOnOveruseWhenScreensharing) {
TestCpuAdaptation(true, true);
}
TEST_F(WebRtcVideoChannelTest, PreviousAdaptationDoesNotApplyToScreenshare) {
cricket::VideoCodec codec = GetEngineCodec("VP8");
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
MediaConfig media_config = GetMediaConfig();
media_config.video.enable_cpu_adaptation = true;
channel_.reset(engine_.CreateMediaChannel(
fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
ASSERT_TRUE(channel_->SetSendParameters(parameters));
AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
ASSERT_TRUE(channel_->SetSend(true));
cricket::VideoOptions camera_options;
camera_options.is_screencast = false;
channel_->SetVideoSend(last_ssrc_, &camera_options, &frame_forwarder);
ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size());
FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_TRUE(send_stream->resolution_scaling_enabled());
// Dont' expect anything on framerate_scaling_enabled, since the default is
// transitioning from MAINTAIN_FRAMERATE to BALANCED.
// Switch to screen share. Expect no resolution scaling.
cricket::VideoOptions screenshare_options;
screenshare_options.is_screencast = true;
channel_->SetVideoSend(last_ssrc_, &screenshare_options, &frame_forwarder);
ASSERT_EQ(2, fake_call_->GetNumCreatedSendStreams());
send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_FALSE(send_stream->resolution_scaling_enabled());
// Switch back to the normal capturer. Expect resolution scaling to be
// reenabled.
channel_->SetVideoSend(last_ssrc_, &camera_options, &frame_forwarder);
send_stream = fake_call_->GetVideoSendStreams().front();
ASSERT_EQ(3, fake_call_->GetNumCreatedSendStreams());
send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_TRUE(send_stream->resolution_scaling_enabled());
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
// TODO(asapersson): Remove this test when the balanced field trial is removed.
void WebRtcVideoChannelTest::TestDegradationPreference(
bool resolution_scaling_enabled,
bool fps_scaling_enabled) {
cricket::VideoCodec codec = GetEngineCodec("VP8");
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
MediaConfig media_config = GetMediaConfig();
media_config.video.enable_cpu_adaptation = true;
channel_.reset(engine_.CreateMediaChannel(
fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
EXPECT_TRUE(channel_->SetSend(true));
FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
EXPECT_EQ(resolution_scaling_enabled,
send_stream->resolution_scaling_enabled());
EXPECT_EQ(fps_scaling_enabled, send_stream->framerate_scaling_enabled());
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
void WebRtcVideoChannelTest::TestCpuAdaptation(bool enable_overuse,
bool is_screenshare) {
cricket::VideoCodec codec = GetEngineCodec("VP8");
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
MediaConfig media_config = GetMediaConfig();
if (enable_overuse) {
media_config.video.enable_cpu_adaptation = true;
}
channel_.reset(engine_.CreateMediaChannel(
fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
options.is_screencast = is_screenshare;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
EXPECT_TRUE(channel_->SetSend(true));
FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
if (!enable_overuse) {
EXPECT_FALSE(send_stream->resolution_scaling_enabled());
EXPECT_FALSE(send_stream->framerate_scaling_enabled());
} else if (is_screenshare) {
EXPECT_FALSE(send_stream->resolution_scaling_enabled());
EXPECT_TRUE(send_stream->framerate_scaling_enabled());
} else {
EXPECT_TRUE(send_stream->resolution_scaling_enabled());
EXPECT_FALSE(send_stream->framerate_scaling_enabled());
}
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, EstimatesNtpStartTimeCorrectly) {
// Start at last timestamp to verify that wraparounds are estimated correctly.
static const uint32_t kInitialTimestamp = 0xFFFFFFFFu;
static const int64_t kInitialNtpTimeMs = 1247891230;
static const int kFrameOffsetMs = 20;
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
FakeVideoReceiveStream* stream = AddRecvStream();
cricket::FakeVideoRenderer renderer;
EXPECT_TRUE(channel_->SetSink(last_ssrc_, &renderer));
webrtc::VideoFrame video_frame =
webrtc::VideoFrame::Builder()
.set_video_frame_buffer(CreateBlackFrameBuffer(4, 4))
.set_timestamp_rtp(kInitialTimestamp)
.set_timestamp_us(0)
.set_rotation(webrtc::kVideoRotation_0)
.build();
// Initial NTP time is not available on the first frame, but should still be
// able to be estimated.
stream->InjectFrame(video_frame);
EXPECT_EQ(1, renderer.num_rendered_frames());
// This timestamp is kInitialTimestamp (-1) + kFrameOffsetMs * 90, which
// triggers a constant-overflow warning, hence we're calculating it explicitly
// here.
fake_clock_.AdvanceTime(webrtc::TimeDelta::Millis(kFrameOffsetMs));
video_frame.set_timestamp(kFrameOffsetMs * 90 - 1);
video_frame.set_ntp_time_ms(kInitialNtpTimeMs + kFrameOffsetMs);
stream->InjectFrame(video_frame);
EXPECT_EQ(2, renderer.num_rendered_frames());
// Verify that NTP time has been correctly deduced.
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1u, info.receivers.size());
EXPECT_EQ(kInitialNtpTimeMs, info.receivers[0].capture_start_ntp_time_ms);
}
TEST_F(WebRtcVideoChannelTest, SetDefaultSendCodecs) {
AssignDefaultAptRtxTypes();
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
VideoCodec codec;
EXPECT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_TRUE(codec.Matches(engine_.send_codecs()[0]));
// Using a RTX setup to verify that the default RTX payload type is good.
const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1);
const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1);
FakeVideoSendStream* stream = AddSendStream(
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
// Make sure NACK and FEC are enabled on the correct payload types.
EXPECT_EQ(1000, config.rtp.nack.rtp_history_ms);
EXPECT_EQ(GetEngineCodec("ulpfec").id, config.rtp.ulpfec.ulpfec_payload_type);
EXPECT_EQ(GetEngineCodec("red").id, config.rtp.ulpfec.red_payload_type);
EXPECT_EQ(1u, config.rtp.rtx.ssrcs.size());
EXPECT_EQ(kRtxSsrcs1[0], config.rtp.rtx.ssrcs[0]);
VerifySendStreamHasRtxTypes(config, default_apt_rtx_types_);
// TODO(juberti): Check RTCP, PLI, TMMBR.
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithoutPacketization) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream();
const webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_FALSE(config.rtp.raw_payload);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithPacketization) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.back().packetization = kPacketizationParamRaw;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream();
const webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_TRUE(config.rtp.raw_payload);
}
// The following four tests ensures that FlexFEC is not activated by default
// when the field trials are not enabled.
// TODO(brandtr): Remove or update these tests when FlexFEC _is_ enabled by
// default.
TEST_F(WebRtcVideoChannelTest, FlexfecSendCodecWithoutSsrcNotExposedByDefault) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
EXPECT_EQ(0U, config.rtp.flexfec.ssrc);
EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty());
}
TEST_F(WebRtcVideoChannelTest, FlexfecSendCodecWithSsrcNotExposedByDefault) {
FakeVideoSendStream* stream = AddSendStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
EXPECT_EQ(0U, config.rtp.flexfec.ssrc);
EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty());
}
TEST_F(WebRtcVideoChannelTest, FlexfecRecvCodecWithoutSsrcNotExposedByDefault) {
AddRecvStream();
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
EXPECT_TRUE(streams.empty());
}
TEST_F(WebRtcVideoChannelTest, FlexfecRecvCodecWithSsrcExposedByDefault) {
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
EXPECT_EQ(1U, streams.size());
}
// TODO(brandtr): When FlexFEC is no longer behind a field trial, merge all
// tests that use this test fixture into the corresponding "non-field trial"
// tests.
class WebRtcVideoChannelFlexfecRecvTest : public WebRtcVideoChannelTest {
public:
WebRtcVideoChannelFlexfecRecvTest()
: WebRtcVideoChannelTest("WebRTC-FlexFEC-03-Advertised/Enabled/") {}
};
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
DefaultFlexfecCodecHasTransportCcAndRembFeedbackParam) {
EXPECT_TRUE(cricket::HasTransportCc(GetEngineCodec("flexfec-03")));
EXPECT_TRUE(cricket::HasRemb(GetEngineCodec("flexfec-03")));
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetDefaultRecvCodecsWithoutSsrc) {
AddRecvStream();
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
EXPECT_TRUE(streams.empty());
const std::vector<FakeVideoReceiveStream*>& video_streams =
fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1U, video_streams.size());
const FakeVideoReceiveStream& video_stream = *video_streams.front();
const webrtc::VideoReceiveStream::Config& video_config =
video_stream.GetConfig();
EXPECT_FALSE(video_config.rtp.protected_by_flexfec);
EXPECT_EQ(video_config.rtp.packet_sink_, nullptr);
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetDefaultRecvCodecsWithSsrc) {
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
ASSERT_EQ(1U, streams.size());
const auto* stream = streams.front();
const webrtc::FlexfecReceiveStream::Config& config = stream->GetConfig();
EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.payload_type);
EXPECT_EQ(kFlexfecSsrc, config.rtp.remote_ssrc);
ASSERT_EQ(1U, config.protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0], config.protected_media_ssrcs[0]);
const std::vector<FakeVideoReceiveStream*>& video_streams =
fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1U, video_streams.size());
const FakeVideoReceiveStream& video_stream = *video_streams.front();
const webrtc::VideoReceiveStream::Config& video_config =
video_stream.GetConfig();
EXPECT_TRUE(video_config.rtp.protected_by_flexfec);
EXPECT_NE(video_config.rtp.packet_sink_, nullptr);
}
// Test changing the configuration after a video stream has been created and
// turn on flexfec. This will result in the video stream being recreated because
// the flexfec stream pointer is injected to the video stream at construction.
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
EnablingFlexfecRecreatesVideoReceiveStream) {
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
const std::vector<FakeVideoReceiveStream*>& video_streams =
fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1U, video_streams.size());
const FakeVideoReceiveStream* video_stream = video_streams.front();
const webrtc::VideoReceiveStream::Config* video_config =
&video_stream->GetConfig();
EXPECT_FALSE(video_config->rtp.protected_by_flexfec);
EXPECT_EQ(video_config->rtp.packet_sink_, nullptr);
// Enable FlexFEC.
recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
// Now the count of created streams will be 3 since the video stream was
// recreated and a flexfec stream was created.
EXPECT_EQ(3, fake_call_->GetNumCreatedReceiveStreams())
<< "Enabling FlexFEC should create FlexfecReceiveStream.";
EXPECT_EQ(1U, fake_call_->GetVideoReceiveStreams().size())
<< "Enabling FlexFEC should not create VideoReceiveStream.";
EXPECT_EQ(1U, fake_call_->GetFlexfecReceiveStreams().size())
<< "Enabling FlexFEC should create a single FlexfecReceiveStream.";
video_stream = video_streams.front();
video_config = &video_stream->GetConfig();
EXPECT_TRUE(video_config->rtp.protected_by_flexfec);
EXPECT_NE(video_config->rtp.packet_sink_, nullptr);
}
// Test changing the configuration after a video stream has been created with
// flexfec enabled and then turn off flexfec. This will result in the video
// stream being recreated because the flexfec stream pointer is injected to the
// video stream at construction and that config needs to be torn down.
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
DisablingFlexfecRecreatesVideoReceiveStream) {
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
EXPECT_EQ(2, fake_call_->GetNumCreatedReceiveStreams());
EXPECT_EQ(1U, fake_call_->GetFlexfecReceiveStreams().size());
const std::vector<FakeVideoReceiveStream*>& video_streams =
fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1U, video_streams.size());
const FakeVideoReceiveStream* video_stream = video_streams.front();
const webrtc::VideoReceiveStream::Config* video_config =
&video_stream->GetConfig();
EXPECT_TRUE(video_config->rtp.protected_by_flexfec);
EXPECT_NE(video_config->rtp.packet_sink_, nullptr);
// Disable FlexFEC.
recv_parameters.codecs.clear();
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
// Now the count of created streams will be 3 since the video stream had to
// be recreated on account of the flexfec stream being deleted.
EXPECT_EQ(3, fake_call_->GetNumCreatedReceiveStreams())
<< "Disabling FlexFEC should not recreate VideoReceiveStream.";
EXPECT_EQ(1U, fake_call_->GetVideoReceiveStreams().size())
<< "Disabling FlexFEC should not destroy VideoReceiveStream.";
EXPECT_TRUE(fake_call_->GetFlexfecReceiveStreams().empty())
<< "Disabling FlexFEC should destroy FlexfecReceiveStream.";
video_stream = video_streams.front();
video_config = &video_stream->GetConfig();
EXPECT_FALSE(video_config->rtp.protected_by_flexfec);
EXPECT_EQ(video_config->rtp.packet_sink_, nullptr);
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest, DuplicateFlexfecCodecIsDropped) {
constexpr int kUnusedPayloadType1 = 127;
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
cricket::VideoCodec duplicate = GetEngineCodec("flexfec-03");
duplicate.id = kUnusedPayloadType1;
recv_parameters.codecs.push_back(duplicate);
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
ASSERT_EQ(1U, streams.size());
const auto* stream = streams.front();
const webrtc::FlexfecReceiveStream::Config& config = stream->GetConfig();
EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.payload_type);
}
// TODO(brandtr): When FlexFEC is no longer behind a field trial, merge all
// tests that use this test fixture into the corresponding "non-field trial"
// tests.
class WebRtcVideoChannelFlexfecSendRecvTest : public WebRtcVideoChannelTest {
public:
WebRtcVideoChannelFlexfecSendRecvTest()
: WebRtcVideoChannelTest(
"WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/") {
}
};
TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetDefaultSendCodecsWithoutSsrc) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type);
EXPECT_EQ(0U, config.rtp.flexfec.ssrc);
EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty());
}
TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetDefaultSendCodecsWithSsrc) {
FakeVideoSendStream* stream = AddSendStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type);
EXPECT_EQ(kFlexfecSsrc, config.rtp.flexfec.ssrc);
ASSERT_EQ(1U, config.rtp.flexfec.protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0], config.rtp.flexfec.protected_media_ssrcs[0]);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithoutFec) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.ulpfec.ulpfec_payload_type);
EXPECT_EQ(-1, config.rtp.ulpfec.red_payload_type);
}
TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetSendCodecsWithoutFec) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetRecvCodecsWithFec) {
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
const std::vector<FakeFlexfecReceiveStream*>& flexfec_streams =
fake_call_->GetFlexfecReceiveStreams();
ASSERT_EQ(1U, flexfec_streams.size());
const FakeFlexfecReceiveStream* flexfec_stream = flexfec_streams.front();
const webrtc::FlexfecReceiveStream::Config& flexfec_stream_config =
flexfec_stream->GetConfig();
EXPECT_EQ(GetEngineCodec("flexfec-03").id,
flexfec_stream_config.payload_type);
EXPECT_EQ(kFlexfecSsrc, flexfec_stream_config.rtp.remote_ssrc);
ASSERT_EQ(1U, flexfec_stream_config.protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0], flexfec_stream_config.protected_media_ssrcs[0]);
const std::vector<FakeVideoReceiveStream*>& video_streams =
fake_call_->GetVideoReceiveStreams();
const FakeVideoReceiveStream* video_stream = video_streams.front();
const webrtc::VideoReceiveStream::Config& video_stream_config =
video_stream->GetConfig();
EXPECT_EQ(video_stream_config.rtp.local_ssrc,
flexfec_stream_config.rtp.local_ssrc);
EXPECT_EQ(video_stream_config.rtp.rtcp_mode, flexfec_stream_config.rtcp_mode);
EXPECT_EQ(video_stream_config.rtcp_send_transport,
flexfec_stream_config.rtcp_send_transport);
// TODO(brandtr): Update this EXPECT when we set `transport_cc` in a
// spec-compliant way.
EXPECT_EQ(video_stream_config.rtp.transport_cc,
flexfec_stream_config.rtp.transport_cc);
EXPECT_EQ(video_stream_config.rtp.rtcp_mode, flexfec_stream_config.rtcp_mode);
EXPECT_EQ(video_stream_config.rtp.extensions,
flexfec_stream_config.rtp.extensions);
}
// We should not send FlexFEC, even if we advertise it, unless the right
// field trial is set.
// TODO(brandtr): Remove when FlexFEC is enabled by default.
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
SetSendCodecsWithoutSsrcWithFecDoesNotEnableFec) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
EXPECT_EQ(0u, config.rtp.flexfec.ssrc);
EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty());
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
SetSendCodecsWithSsrcWithFecDoesNotEnableFec) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
EXPECT_EQ(0u, config.rtp.flexfec.ssrc);
EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty());
}
TEST_F(WebRtcVideoChannelTest,
SetSendCodecRejectsRtxWithoutAssociatedPayloadType) {
const int kUnusedPayloadType = 127;
EXPECT_FALSE(FindCodecById(engine_.send_codecs(), kUnusedPayloadType));
cricket::VideoSendParameters parameters;
cricket::VideoCodec rtx_codec(kUnusedPayloadType, "rtx");
parameters.codecs.push_back(rtx_codec);
EXPECT_FALSE(channel_->SetSendParameters(parameters))
<< "RTX codec without associated payload type should be rejected.";
}
TEST_F(WebRtcVideoChannelTest,
SetSendCodecRejectsRtxWithoutMatchingVideoCodec) {
const int kUnusedPayloadType1 = 126;
const int kUnusedPayloadType2 = 127;
EXPECT_FALSE(FindCodecById(engine_.send_codecs(), kUnusedPayloadType1));
EXPECT_FALSE(FindCodecById(engine_.send_codecs(), kUnusedPayloadType2));
{
cricket::VideoCodec rtx_codec = cricket::VideoCodec::CreateRtxCodec(
kUnusedPayloadType1, GetEngineCodec("VP8").id);
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(rtx_codec);
ASSERT_TRUE(channel_->SetSendParameters(parameters));
}
{
cricket::VideoCodec rtx_codec = cricket::VideoCodec::CreateRtxCodec(
kUnusedPayloadType1, kUnusedPayloadType2);
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(rtx_codec);
EXPECT_FALSE(channel_->SetSendParameters(parameters))
<< "RTX without matching video codec should be rejected.";
}
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithChangedRtxPayloadType) {
const int kUnusedPayloadType1 = 126;
const int kUnusedPayloadType2 = 127;
EXPECT_FALSE(FindCodecById(engine_.send_codecs(), kUnusedPayloadType1));
EXPECT_FALSE(FindCodecById(engine_.send_codecs(), kUnusedPayloadType2));
// SSRCs for RTX.
cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]);
AddSendStream(params);
// Original payload type for RTX.
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
cricket::VideoCodec rtx_codec(kUnusedPayloadType1, "rtx");
rtx_codec.SetParam("apt", GetEngineCodec("VP8").id);
parameters.codecs.push_back(rtx_codec);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
ASSERT_EQ(1U, fake_call_->GetVideoSendStreams().size());
const webrtc::VideoSendStream::Config& config_before =
fake_call_->GetVideoSendStreams()[0]->GetConfig();
EXPECT_EQ(kUnusedPayloadType1, config_before.rtp.rtx.payload_type);
ASSERT_EQ(1U, config_before.rtp.rtx.ssrcs.size());
EXPECT_EQ(kRtxSsrcs1[0], config_before.rtp.rtx.ssrcs[0]);
// Change payload type for RTX.
parameters.codecs[1].id = kUnusedPayloadType2;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
ASSERT_EQ(1U, fake_call_->GetVideoSendStreams().size());
const webrtc::VideoSendStream::Config& config_after =
fake_call_->GetVideoSendStreams()[0]->GetConfig();
EXPECT_EQ(kUnusedPayloadType2, config_after.rtp.rtx.payload_type);
ASSERT_EQ(1U, config_after.rtp.rtx.ssrcs.size());
EXPECT_EQ(kRtxSsrcs1[0], config_after.rtp.rtx.ssrcs[0]);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithoutFecDisablesFec) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("ulpfec"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(GetEngineCodec("ulpfec").id, config.rtp.ulpfec.ulpfec_payload_type);
parameters.codecs.pop_back();
ASSERT_TRUE(channel_->SetSendParameters(parameters));
stream = fake_call_->GetVideoSendStreams()[0];
ASSERT_TRUE(stream != nullptr);
config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.ulpfec.ulpfec_payload_type)
<< "SetSendCodec without ULPFEC should disable current ULPFEC.";
}
TEST_F(WebRtcVideoChannelFlexfecSendRecvTest,
SetSendCodecsWithoutFecDisablesFec) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = AddSendStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type);
EXPECT_EQ(kFlexfecSsrc, config.rtp.flexfec.ssrc);
ASSERT_EQ(1U, config.rtp.flexfec.protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0], config.rtp.flexfec.protected_media_ssrcs[0]);
parameters.codecs.pop_back();
ASSERT_TRUE(channel_->SetSendParameters(parameters));
stream = fake_call_->GetVideoSendStreams()[0];
ASSERT_TRUE(stream != nullptr);
config = stream->GetConfig().Copy();
EXPECT_EQ(-1, config.rtp.flexfec.payload_type)
<< "SetSendCodec without FlexFEC should disable current FlexFEC.";
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsChangesExistingStreams) {
cricket::VideoSendParameters parameters;
cricket::VideoCodec codec(100, "VP8");
codec.SetParam(kCodecParamMaxQuantization, kDefaultQpMax);
parameters.codecs.push_back(codec);
ASSERT_TRUE(channel_->SetSendParameters(parameters));
channel_->SetSend(true);
FakeVideoSendStream* stream = AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
std::vector<webrtc::VideoStream> streams = stream->GetVideoStreams();
EXPECT_EQ(kDefaultQpMax, streams[0].max_qp);
parameters.codecs.clear();
codec.SetParam(kCodecParamMaxQuantization, kDefaultQpMax + 1);
parameters.codecs.push_back(codec);
ASSERT_TRUE(channel_->SetSendParameters(parameters));
streams = fake_call_->GetVideoSendStreams()[0]->GetVideoStreams();
EXPECT_EQ(kDefaultQpMax + 1, streams[0].max_qp);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithBitrates) {
SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
200000);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithHighMaxBitrate) {
SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "10000", 10000000);
std::vector<webrtc::VideoStream> streams = AddSendStream()->GetVideoStreams();
ASSERT_EQ(1u, streams.size());
EXPECT_EQ(10000000, streams[0].max_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest,
SetSendCodecsWithoutBitratesUsesCorrectDefaults) {
SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "", -1);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsCapsMinAndStartBitrate) {
SetSendCodecsShouldWorkForBitrates("-1", 0, "-100", -1, "", -1);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsRejectsMaxLessThanMinBitrate) {
send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "300";
send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "200";
EXPECT_FALSE(channel_->SetSendParameters(send_parameters_));
}
// Test that when both the codec-specific bitrate params and max_bandwidth_bps
// are present in the same send parameters, the settings are combined correctly.
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithBitratesAndMaxSendBandwidth) {
send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "100";
send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200";
send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300";
send_parameters_.max_bandwidth_bps = 400000;
// We expect max_bandwidth_bps to take priority, if set.
ExpectSetBitrateParameters(100000, 200000, 400000);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
// Since the codec isn't changing, start_bitrate_bps should be -1.
ExpectSetBitrateParameters(100000, -1, 350000);
// Decrease max_bandwidth_bps.
send_parameters_.max_bandwidth_bps = 350000;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
// Now try again with the values flipped around.
send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "400";
send_parameters_.max_bandwidth_bps = 300000;
ExpectSetBitrateParameters(100000, 200000, 300000);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
// If we change the codec max, max_bandwidth_bps should still apply.
send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "350";
ExpectSetBitrateParameters(100000, 200000, 300000);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
}
TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthShouldPreserveOtherBitrates) {
SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
200000);
send_parameters_.max_bandwidth_bps = 300000;
// Setting max bitrate should keep previous min bitrate.
// Setting max bitrate should not reset start bitrate.
ExpectSetBitrateParameters(100000, -1, 300000);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
}
TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthShouldBeRemovable) {
send_parameters_.max_bandwidth_bps = 300000;
ExpectSetMaxBitrate(300000);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
// -1 means to disable max bitrate (set infinite).
send_parameters_.max_bandwidth_bps = -1;
ExpectSetMaxBitrate(-1);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
}
TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthAndAddSendStream) {
send_parameters_.max_bandwidth_bps = 99999;
FakeVideoSendStream* stream = AddSendStream();
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
ASSERT_EQ(1u, stream->GetVideoStreams().size());
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
send_parameters_.max_bandwidth_bps = 77777;
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
}
// Tests that when the codec specific max bitrate and VideoSendParameters
// max_bandwidth_bps are used, that it sets the VideoStream's max bitrate
// appropriately.
TEST_F(WebRtcVideoChannelTest,
MaxBitratePrioritizesVideoSendParametersOverCodecMaxBitrate) {
send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "100";
send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200";
send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300";
send_parameters_.max_bandwidth_bps = -1;
AddSendStream();
ExpectSetMaxBitrate(300000);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
std::vector<FakeVideoSendStream*> video_send_streams = GetFakeSendStreams();
ASSERT_EQ(1u, video_send_streams.size());
FakeVideoSendStream* video_send_stream = video_send_streams[0];
ASSERT_EQ(1u, video_send_streams[0]->GetVideoStreams().size());
// First the max bitrate is set based upon the codec param.
EXPECT_EQ(300000,
video_send_streams[0]->GetVideoStreams()[0].max_bitrate_bps);
// The VideoSendParameters max bitrate overrides the codec's.
send_parameters_.max_bandwidth_bps = 500000;
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
ASSERT_EQ(1u, video_send_stream->GetVideoStreams().size());
EXPECT_EQ(500000, video_send_stream->GetVideoStreams()[0].max_bitrate_bps);
}
// Tests that when the codec specific max bitrate and RtpParameters
// max_bitrate_bps are used, that it sets the VideoStream's max bitrate
// appropriately.
TEST_F(WebRtcVideoChannelTest,
MaxBitratePrioritizesRtpParametersOverCodecMaxBitrate) {
send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "100";
send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200";
send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300";
send_parameters_.max_bandwidth_bps = -1;
AddSendStream();
ExpectSetMaxBitrate(300000);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
std::vector<FakeVideoSendStream*> video_send_streams = GetFakeSendStreams();
ASSERT_EQ(1u, video_send_streams.size());
FakeVideoSendStream* video_send_stream = video_send_streams[0];
ASSERT_EQ(1u, video_send_stream->GetVideoStreams().size());
// First the max bitrate is set based upon the codec param.
EXPECT_EQ(300000, video_send_stream->GetVideoStreams()[0].max_bitrate_bps);
// The RtpParameter max bitrate overrides the codec's.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(1u, parameters.encodings.size());
parameters.encodings[0].max_bitrate_bps = 500000;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
ASSERT_EQ(1u, video_send_stream->GetVideoStreams().size());
EXPECT_EQ(parameters.encodings[0].max_bitrate_bps,
video_send_stream->GetVideoStreams()[0].max_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest,
MaxBitrateIsMinimumOfMaxSendBandwidthAndMaxEncodingBitrate) {
send_parameters_.max_bandwidth_bps = 99999;
FakeVideoSendStream* stream = AddSendStream();
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
ASSERT_EQ(1u, stream->GetVideoStreams().size());
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1u, parameters.encodings.size());
parameters.encodings[0].max_bitrate_bps = 99999 - 1;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_EQ(parameters.encodings[0].max_bitrate_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
parameters.encodings[0].max_bitrate_bps = 99999 + 1;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest, SetMaxSendBitrateCanIncreaseSenderBitrate) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
channel_->SetSend(true);
FakeVideoSendStream* stream = AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
std::vector<webrtc::VideoStream> streams = stream->GetVideoStreams();
int initial_max_bitrate_bps = streams[0].max_bitrate_bps;
EXPECT_GT(initial_max_bitrate_bps, 0);
parameters.max_bandwidth_bps = initial_max_bitrate_bps * 2;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
// Insert a frame to update the encoder config.
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
streams = stream->GetVideoStreams();
EXPECT_EQ(initial_max_bitrate_bps * 2, streams[0].max_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
SetMaxSendBitrateCanIncreaseSimulcastSenderBitrate) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
channel_->SetSend(true);
FakeVideoSendStream* stream = AddSendStream(
cricket::CreateSimStreamParams("cname", MAKE_VECTOR(kSsrcs3)));
// Send a frame to make sure this scales up to >1 stream (simulcast).
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(kSsrcs3[0], nullptr, &frame_forwarder));
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
std::vector<webrtc::VideoStream> streams = stream->GetVideoStreams();
ASSERT_GT(streams.size(), 1u)
<< "Without simulcast this test doesn't make sense.";
int initial_max_bitrate_bps = GetTotalMaxBitrate(streams).bps();
EXPECT_GT(initial_max_bitrate_bps, 0);
parameters.max_bandwidth_bps = initial_max_bitrate_bps * 2;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
// Insert a frame to update the encoder config.
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
streams = stream->GetVideoStreams();
int increased_max_bitrate_bps = GetTotalMaxBitrate(streams).bps();
EXPECT_EQ(initial_max_bitrate_bps * 2, increased_max_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(kSsrcs3[0], nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithMaxQuantization) {
static const char* kMaxQuantization = "21";
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs[0].params[kCodecParamMaxQuantization] = kMaxQuantization;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_EQ(atoi(kMaxQuantization),
AddSendStream()->GetVideoStreams().back().max_qp);
VideoCodec codec;
EXPECT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_EQ(kMaxQuantization, codec.params[kCodecParamMaxQuantization]);
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsRejectBadPayloadTypes) {
// TODO(pbos): Should we only allow the dynamic range?
static const int kIncorrectPayloads[] = {-2, -1, 128, 129};
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
for (size_t i = 0; i < arraysize(kIncorrectPayloads); ++i) {
parameters.codecs[0].id = kIncorrectPayloads[i];
EXPECT_FALSE(channel_->SetSendParameters(parameters))
<< "Bad payload type '" << kIncorrectPayloads[i] << "' accepted.";
}
}
TEST_F(WebRtcVideoChannelTest, SetSendCodecsAcceptAllValidPayloadTypes) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
for (int payload_type = 96; payload_type <= 127; ++payload_type) {
parameters.codecs[0].id = payload_type;
EXPECT_TRUE(channel_->SetSendParameters(parameters))
<< "Payload type '" << payload_type << "' rejected.";
}
}
// Test that setting the a different set of codecs but with an identical front
// codec doesn't result in the stream being recreated.
// This may happen when a subsequent negotiation includes fewer codecs, as a
// result of one of the codecs being rejected.
TEST_F(WebRtcVideoChannelTest,
SetSendCodecsIdenticalFirstCodecDoesntRecreateStream) {
cricket::VideoSendParameters parameters1;
parameters1.codecs.push_back(GetEngineCodec("VP8"));
parameters1.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetSendParameters(parameters1));
AddSendStream();
EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
cricket::VideoSendParameters parameters2;
parameters2.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetSendParameters(parameters2));
EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithOnlyVp8) {
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
}
// Test that we set our inbound RTX codecs properly.
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithRtx) {
const int kUnusedPayloadType1 = 126;
const int kUnusedPayloadType2 = 127;
EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kUnusedPayloadType1));
EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kUnusedPayloadType2));
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
cricket::VideoCodec rtx_codec(kUnusedPayloadType1, "rtx");
parameters.codecs.push_back(rtx_codec);
EXPECT_FALSE(channel_->SetRecvParameters(parameters))
<< "RTX codec without associated payload should be rejected.";
parameters.codecs[1].SetParam("apt", kUnusedPayloadType2);
EXPECT_FALSE(channel_->SetRecvParameters(parameters))
<< "RTX codec with invalid associated payload type should be rejected.";
parameters.codecs[1].SetParam("apt", GetEngineCodec("VP8").id);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
cricket::VideoCodec rtx_codec2(kUnusedPayloadType2, "rtx");
rtx_codec2.SetParam("apt", rtx_codec.id);
parameters.codecs.push_back(rtx_codec2);
EXPECT_FALSE(channel_->SetRecvParameters(parameters))
<< "RTX codec with another RTX as associated payload type should be "
"rejected.";
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithPacketization) {
cricket::VideoCodec vp8_codec = GetEngineCodec("VP8");
vp8_codec.packetization = kPacketizationParamRaw;
cricket::VideoRecvParameters parameters;
parameters.codecs = {vp8_codec, GetEngineCodec("VP9")};
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
const cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
AddRecvStream(params);
ASSERT_THAT(fake_call_->GetVideoReceiveStreams(), testing::SizeIs(1));
const webrtc::VideoReceiveStream::Config& config =
fake_call_->GetVideoReceiveStreams()[0]->GetConfig();
ASSERT_THAT(config.rtp.raw_payload_types, testing::SizeIs(1));
EXPECT_EQ(config.rtp.raw_payload_types.count(vp8_codec.id), 1U);
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithPacketizationRecreatesStream) {
cricket::VideoRecvParameters parameters;
parameters.codecs = {GetEngineCodec("VP8"), GetEngineCodec("VP9")};
parameters.codecs.back().packetization = kPacketizationParamRaw;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
const cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
AddRecvStream(params);
ASSERT_THAT(fake_call_->GetVideoReceiveStreams(), testing::SizeIs(1));
EXPECT_EQ(fake_call_->GetNumCreatedReceiveStreams(), 1);
parameters.codecs.back().packetization.reset();
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_EQ(fake_call_->GetNumCreatedReceiveStreams(), 2);
}
TEST_F(WebRtcVideoChannelTest, DuplicateUlpfecCodecIsDropped) {
constexpr int kFirstUlpfecPayloadType = 126;
constexpr int kSecondUlpfecPayloadType = 127;
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(
cricket::VideoCodec(kFirstUlpfecPayloadType, cricket::kUlpfecCodecName));
parameters.codecs.push_back(
cricket::VideoCodec(kSecondUlpfecPayloadType, cricket::kUlpfecCodecName));
ASSERT_TRUE(channel_->SetRecvParameters(parameters));
FakeVideoReceiveStream* recv_stream = AddRecvStream();
EXPECT_EQ(kFirstUlpfecPayloadType,
recv_stream->GetConfig().rtp.ulpfec_payload_type);
}
TEST_F(WebRtcVideoChannelTest, DuplicateRedCodecIsDropped) {
constexpr int kFirstRedPayloadType = 126;
constexpr int kSecondRedPayloadType = 127;
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(
cricket::VideoCodec(kFirstRedPayloadType, cricket::kRedCodecName));
parameters.codecs.push_back(
cricket::VideoCodec(kSecondRedPayloadType, cricket::kRedCodecName));
ASSERT_TRUE(channel_->SetRecvParameters(parameters));
FakeVideoReceiveStream* recv_stream = AddRecvStream();
EXPECT_EQ(kFirstRedPayloadType,
recv_stream->GetConfig().rtp.red_payload_type);
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithChangedRtxPayloadType) {
const int kUnusedPayloadType1 = 126;
const int kUnusedPayloadType2 = 127;
EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kUnusedPayloadType1));
EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kUnusedPayloadType2));
// SSRCs for RTX.
cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]);
AddRecvStream(params);
// Original payload type for RTX.
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
cricket::VideoCodec rtx_codec(kUnusedPayloadType1, "rtx");
rtx_codec.SetParam("apt", GetEngineCodec("VP8").id);
parameters.codecs.push_back(rtx_codec);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
ASSERT_EQ(1U, fake_call_->GetVideoReceiveStreams().size());
const webrtc::VideoReceiveStream::Config& config_before =
fake_call_->GetVideoReceiveStreams()[0]->GetConfig();
EXPECT_EQ(1U, config_before.rtp.rtx_associated_payload_types.size());
const int* payload_type_before = FindKeyByValue(
config_before.rtp.rtx_associated_payload_types, GetEngineCodec("VP8").id);
ASSERT_NE(payload_type_before, nullptr);
EXPECT_EQ(kUnusedPayloadType1, *payload_type_before);
EXPECT_EQ(kRtxSsrcs1[0], config_before.rtp.rtx_ssrc);
// Change payload type for RTX.
parameters.codecs[1].id = kUnusedPayloadType2;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
ASSERT_EQ(1U, fake_call_->GetVideoReceiveStreams().size());
const webrtc::VideoReceiveStream::Config& config_after =
fake_call_->GetVideoReceiveStreams()[0]->GetConfig();
EXPECT_EQ(1U, config_after.rtp.rtx_associated_payload_types.size());
const int* payload_type_after = FindKeyByValue(
config_after.rtp.rtx_associated_payload_types, GetEngineCodec("VP8").id);
ASSERT_NE(payload_type_after, nullptr);
EXPECT_EQ(kUnusedPayloadType2, *payload_type_after);
EXPECT_EQ(kRtxSsrcs1[0], config_after.rtp.rtx_ssrc);
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRtxWithRtxTime) {
const int kUnusedPayloadType1 = 126;
const int kUnusedPayloadType2 = 127;
EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kUnusedPayloadType1));
EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kUnusedPayloadType2));
// SSRCs for RTX.
cricket::StreamParams params =
cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]);
AddRecvStream(params);
// Payload type for RTX.
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
cricket::VideoCodec rtx_codec(kUnusedPayloadType1, "rtx");
rtx_codec.SetParam("apt", GetEngineCodec("VP8").id);
parameters.codecs.push_back(rtx_codec);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
ASSERT_EQ(1U, fake_call_->GetVideoReceiveStreams().size());
const webrtc::VideoReceiveStream::Config& config =
fake_call_->GetVideoReceiveStreams()[0]->GetConfig();
const int kRtxTime = 343;
// Assert that the default value is different from the ones we test
// and store the default value.
EXPECT_NE(config.rtp.nack.rtp_history_ms, kRtxTime);
int default_history_ms = config.rtp.nack.rtp_history_ms;
// Set rtx-time.
parameters.codecs[1].SetParam(kCodecParamRtxTime, kRtxTime);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_EQ(fake_call_->GetVideoReceiveStreams()[0]
->GetConfig()
.rtp.nack.rtp_history_ms,
kRtxTime);
// Negative values are ignored so the default value applies.
parameters.codecs[1].SetParam(kCodecParamRtxTime, -1);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_NE(fake_call_->GetVideoReceiveStreams()[0]
->GetConfig()
.rtp.nack.rtp_history_ms,
-1);
EXPECT_EQ(fake_call_->GetVideoReceiveStreams()[0]
->GetConfig()
.rtp.nack.rtp_history_ms,
default_history_ms);
// 0 is ignored so the default applies.
parameters.codecs[1].SetParam(kCodecParamRtxTime, 0);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_NE(fake_call_->GetVideoReceiveStreams()[0]
->GetConfig()
.rtp.nack.rtp_history_ms,
0);
EXPECT_EQ(fake_call_->GetVideoReceiveStreams()[0]
->GetConfig()
.rtp.nack.rtp_history_ms,
default_history_ms);
// Values larger than the default are clamped to the default.
parameters.codecs[1].SetParam(kCodecParamRtxTime, default_history_ms + 100);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
EXPECT_EQ(fake_call_->GetVideoReceiveStreams()[0]
->GetConfig()
.rtp.nack.rtp_history_ms,
default_history_ms);
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsDifferentPayloadType) {
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs[0].id = 99;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsAcceptDefaultCodecs) {
cricket::VideoRecvParameters parameters;
parameters.codecs = engine_.recv_codecs();
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
FakeVideoReceiveStream* stream = AddRecvStream();
const webrtc::VideoReceiveStream::Config& config = stream->GetConfig();
EXPECT_EQ(engine_.recv_codecs()[0].name,
config.decoders[0].video_format.name);
EXPECT_EQ(engine_.recv_codecs()[0].id, config.decoders[0].payload_type);
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRejectUnsupportedCodec) {
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(VideoCodec(101, "WTF3"));
EXPECT_FALSE(channel_->SetRecvParameters(parameters));
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsAcceptsMultipleVideoCodecs) {
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithoutFecDisablesFec) {
cricket::VideoSendParameters send_parameters;
send_parameters.codecs.push_back(GetEngineCodec("VP8"));
send_parameters.codecs.push_back(GetEngineCodec("red"));
send_parameters.codecs.push_back(GetEngineCodec("ulpfec"));
ASSERT_TRUE(channel_->SetSendParameters(send_parameters));
FakeVideoReceiveStream* stream = AddRecvStream();
EXPECT_EQ(GetEngineCodec("ulpfec").id,
stream->GetConfig().rtp.ulpfec_payload_type);
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
stream = fake_call_->GetVideoReceiveStreams()[0];
ASSERT_TRUE(stream != nullptr);
EXPECT_EQ(-1, stream->GetConfig().rtp.ulpfec_payload_type)
<< "SetSendCodec without ULPFEC should disable current ULPFEC.";
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetRecvParamsWithoutFecDisablesFec) {
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
ASSERT_EQ(1U, streams.size());
const FakeFlexfecReceiveStream* stream = streams.front();
EXPECT_EQ(GetEngineCodec("flexfec-03").id, stream->GetConfig().payload_type);
EXPECT_EQ(kFlexfecSsrc, stream->rtp_config().remote_ssrc);
ASSERT_EQ(1U, stream->GetConfig().protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0], stream->GetConfig().protected_media_ssrcs[0]);
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
EXPECT_TRUE(streams.empty())
<< "SetSendCodec without FlexFEC should disable current FlexFEC.";
}
TEST_F(WebRtcVideoChannelTest, SetSendParamsWithFecEnablesFec) {
FakeVideoReceiveStream* stream = AddRecvStream();
EXPECT_EQ(GetEngineCodec("ulpfec").id,
stream->GetConfig().rtp.ulpfec_payload_type);
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
recv_parameters.codecs.push_back(GetEngineCodec("red"));
recv_parameters.codecs.push_back(GetEngineCodec("ulpfec"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
stream = fake_call_->GetVideoReceiveStreams()[0];
ASSERT_TRUE(stream != nullptr);
EXPECT_EQ(GetEngineCodec("ulpfec").id,
stream->GetConfig().rtp.ulpfec_payload_type)
<< "ULPFEC should be enabled on the receive stream.";
cricket::VideoSendParameters send_parameters;
send_parameters.codecs.push_back(GetEngineCodec("VP8"));
send_parameters.codecs.push_back(GetEngineCodec("red"));
send_parameters.codecs.push_back(GetEngineCodec("ulpfec"));
ASSERT_TRUE(channel_->SetSendParameters(send_parameters));
stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(GetEngineCodec("ulpfec").id,
stream->GetConfig().rtp.ulpfec_payload_type)
<< "ULPFEC should be enabled on the receive stream.";
}
TEST_F(WebRtcVideoChannelFlexfecSendRecvTest,
SetSendRecvParamsWithFecEnablesFec) {
AddRecvStream(
CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
const std::vector<FakeFlexfecReceiveStream*>& streams =
fake_call_->GetFlexfecReceiveStreams();
cricket::VideoRecvParameters recv_parameters;
recv_parameters.codecs.push_back(GetEngineCodec("VP8"));
recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
ASSERT_EQ(1U, streams.size());
const FakeFlexfecReceiveStream* stream_with_recv_params = streams.front();
EXPECT_EQ(GetEngineCodec("flexfec-03").id,
stream_with_recv_params->GetConfig().payload_type);
EXPECT_EQ(kFlexfecSsrc, stream_with_recv_params->GetConfig().rtp.remote_ssrc);
EXPECT_EQ(1U,
stream_with_recv_params->GetConfig().protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0],
stream_with_recv_params->GetConfig().protected_media_ssrcs[0]);
cricket::VideoSendParameters send_parameters;
send_parameters.codecs.push_back(GetEngineCodec("VP8"));
send_parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
ASSERT_TRUE(channel_->SetSendParameters(send_parameters));
ASSERT_EQ(1U, streams.size());
const FakeFlexfecReceiveStream* stream_with_send_params = streams.front();
EXPECT_EQ(GetEngineCodec("flexfec-03").id,
stream_with_send_params->GetConfig().payload_type);
EXPECT_EQ(kFlexfecSsrc, stream_with_send_params->GetConfig().rtp.remote_ssrc);
EXPECT_EQ(1U,
stream_with_send_params->GetConfig().protected_media_ssrcs.size());
EXPECT_EQ(kSsrcs1[0],
stream_with_send_params->GetConfig().protected_media_ssrcs[0]);
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRejectDuplicateFecPayloads) {
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("red"));
parameters.codecs[1].id = parameters.codecs[0].id;
EXPECT_FALSE(channel_->SetRecvParameters(parameters));
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
SetRecvCodecsRejectDuplicateFecPayloads) {
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
parameters.codecs[1].id = parameters.codecs[0].id;
EXPECT_FALSE(channel_->SetRecvParameters(parameters));
}
TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRejectDuplicateCodecPayloads) {
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
parameters.codecs[1].id = parameters.codecs[0].id;
EXPECT_FALSE(channel_->SetRecvParameters(parameters));
}
TEST_F(WebRtcVideoChannelTest,
SetRecvCodecsAcceptSameCodecOnMultiplePayloadTypes) {
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs[1].id += 1;
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
}
// Test that setting the same codecs but with a different order
// doesn't result in the stream being recreated.
TEST_F(WebRtcVideoChannelTest,
SetRecvCodecsDifferentOrderDoesntRecreateStream) {
cricket::VideoRecvParameters parameters1;
parameters1.codecs.push_back(GetEngineCodec("VP8"));
parameters1.codecs.push_back(GetEngineCodec("red"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters1));
AddRecvStream(cricket::StreamParams::CreateLegacy(123));
EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
cricket::VideoRecvParameters parameters2;
parameters2.codecs.push_back(GetEngineCodec("red"));
parameters2.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters2));
EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
}
TEST_F(WebRtcVideoChannelTest, SendStreamNotSendingByDefault) {
EXPECT_FALSE(AddSendStream()->IsSending());
}
TEST_F(WebRtcVideoChannelTest, ReceiveStreamReceivingByDefault) {
EXPECT_TRUE(AddRecvStream()->IsReceiving());
}
TEST_F(WebRtcVideoChannelTest, SetSend) {
FakeVideoSendStream* stream = AddSendStream();
EXPECT_FALSE(stream->IsSending());
// false->true
EXPECT_TRUE(channel_->SetSend(true));
EXPECT_TRUE(stream->IsSending());
// true->true
EXPECT_TRUE(channel_->SetSend(true));
EXPECT_TRUE(stream->IsSending());
// true->false
EXPECT_TRUE(channel_->SetSend(false));
EXPECT_FALSE(stream->IsSending());
// false->false
EXPECT_TRUE(channel_->SetSend(false));
EXPECT_FALSE(stream->IsSending());
EXPECT_TRUE(channel_->SetSend(true));
FakeVideoSendStream* new_stream = AddSendStream();
EXPECT_TRUE(new_stream->IsSending())
<< "Send stream created after SetSend(true) not sending initially.";
}
// This test verifies DSCP settings are properly applied on video media channel.
TEST_F(WebRtcVideoChannelTest, TestSetDscpOptions) {
std::unique_ptr<cricket::FakeNetworkInterface> network_interface(
new cricket::FakeNetworkInterface);
MediaConfig config;
std::unique_ptr<cricket::WebRtcVideoChannel> channel;
webrtc::RtpParameters parameters;
channel.reset(
static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel(
call_.get(), config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get())));
channel->SetInterface(network_interface.get());
// Default value when DSCP is disabled should be DSCP_DEFAULT.
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
channel->SetInterface(nullptr);
// Default value when DSCP is enabled is also DSCP_DEFAULT, until it is set
// through rtp parameters.
config.enable_dscp = true;
channel.reset(
static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel(
call_.get(), config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get())));
channel->SetInterface(network_interface.get());
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
// Create a send stream to configure
EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
parameters = channel->GetRtpSendParameters(kSsrc);
ASSERT_FALSE(parameters.encodings.empty());
// Various priorities map to various dscp values.
parameters.encodings[0].network_priority = webrtc::Priority::kHigh;
ASSERT_TRUE(channel->SetRtpSendParameters(kSsrc, parameters).ok());
EXPECT_EQ(rtc::DSCP_AF41, network_interface->dscp());
parameters.encodings[0].network_priority = webrtc::Priority::kVeryLow;
ASSERT_TRUE(channel->SetRtpSendParameters(kSsrc, parameters).ok());
EXPECT_EQ(rtc::DSCP_CS1, network_interface->dscp());
// Packets should also self-identify their dscp in PacketOptions.
const uint8_t kData[10] = {0};
EXPECT_TRUE(static_cast<webrtc::Transport*>(channel.get())
->SendRtcp(kData, sizeof(kData)));
EXPECT_EQ(rtc::DSCP_CS1, network_interface->options().dscp);
channel->SetInterface(nullptr);
// Verify that setting the option to false resets the
// DiffServCodePoint.
config.enable_dscp = false;
channel.reset(
static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel(
call_.get(), config, VideoOptions(), webrtc::CryptoOptions(),
video_bitrate_allocator_factory_.get())));
channel->SetInterface(network_interface.get());
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
channel->SetInterface(nullptr);
}
// This test verifies that the RTCP reduced size mode is properly applied to
// send video streams.
TEST_F(WebRtcVideoChannelTest, TestSetSendRtcpReducedSize) {
// Create stream, expecting that default mode is "compound".
FakeVideoSendStream* stream1 = AddSendStream();
EXPECT_EQ(webrtc::RtcpMode::kCompound, stream1->GetConfig().rtp.rtcp_mode);
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_FALSE(rtp_parameters.rtcp.reduced_size);
// Now enable reduced size mode.
send_parameters_.rtcp.reduced_size = true;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
stream1 = fake_call_->GetVideoSendStreams()[0];
EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream1->GetConfig().rtp.rtcp_mode);
rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_TRUE(rtp_parameters.rtcp.reduced_size);
// Create a new stream and ensure it picks up the reduced size mode.
FakeVideoSendStream* stream2 = AddSendStream();
EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream2->GetConfig().rtp.rtcp_mode);
}
// This test verifies that the RTCP reduced size mode is properly applied to
// receive video streams.
TEST_F(WebRtcVideoChannelTest, TestSetRecvRtcpReducedSize) {
// Create stream, expecting that default mode is "compound".
FakeVideoReceiveStream* stream1 = AddRecvStream();
EXPECT_EQ(webrtc::RtcpMode::kCompound, stream1->GetConfig().rtp.rtcp_mode);
// Now enable reduced size mode.
// TODO(deadbeef): Once "recv_parameters" becomes "receiver_parameters",
// the reduced_size flag should come from that.
send_parameters_.rtcp.reduced_size = true;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
stream1 = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream1->GetConfig().rtp.rtcp_mode);
// Create a new stream and ensure it picks up the reduced size mode.
FakeVideoReceiveStream* stream2 = AddRecvStream();
EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream2->GetConfig().rtp.rtcp_mode);
}
TEST_F(WebRtcVideoChannelTest, OnReadyToSendSignalsNetworkState) {
EXPECT_EQ(webrtc::kNetworkUp,
fake_call_->GetNetworkState(webrtc::MediaType::VIDEO));
EXPECT_EQ(webrtc::kNetworkUp,
fake_call_->GetNetworkState(webrtc::MediaType::AUDIO));
channel_->OnReadyToSend(false);
EXPECT_EQ(webrtc::kNetworkDown,
fake_call_->GetNetworkState(webrtc::MediaType::VIDEO));
EXPECT_EQ(webrtc::kNetworkUp,
fake_call_->GetNetworkState(webrtc::MediaType::AUDIO));
channel_->OnReadyToSend(true);
EXPECT_EQ(webrtc::kNetworkUp,
fake_call_->GetNetworkState(webrtc::MediaType::VIDEO));
EXPECT_EQ(webrtc::kNetworkUp,
fake_call_->GetNetworkState(webrtc::MediaType::AUDIO));
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsSentCodecName) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
AddSendStream();
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ("VP8", info.senders[0].codec_name);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsEncoderImplementationName) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.encoder_implementation_name = "encoder_implementation_name";
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.encoder_implementation_name,
info.senders[0].encoder_implementation_name);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsCpuOveruseMetrics) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.avg_encode_time_ms = 13;
stats.encode_usage_percent = 42;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.avg_encode_time_ms, info.senders[0].avg_encode_ms);
EXPECT_EQ(stats.encode_usage_percent, info.senders[0].encode_usage_percent);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsFramesEncoded) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.frames_encoded = 13;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.frames_encoded, info.senders[0].frames_encoded);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsKeyFramesEncoded) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.substreams[123].frame_counts.key_frames = 10;
stats.substreams[456].frame_counts.key_frames = 87;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(info.senders.size(), 2u);
EXPECT_EQ(10u, info.senders[0].key_frames_encoded);
EXPECT_EQ(87u, info.senders[1].key_frames_encoded);
EXPECT_EQ(97u, info.aggregated_senders[0].key_frames_encoded);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsPerLayerQpSum) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.substreams[123].qp_sum = 15;
stats.substreams[456].qp_sum = 11;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(info.senders.size(), 2u);
EXPECT_EQ(stats.substreams[123].qp_sum, info.senders[0].qp_sum);
EXPECT_EQ(stats.substreams[456].qp_sum, info.senders[1].qp_sum);
EXPECT_EQ(*info.aggregated_senders[0].qp_sum, 26u);
}
webrtc::VideoSendStream::Stats GetInitialisedStats() {
webrtc::VideoSendStream::Stats stats;
stats.encoder_implementation_name = "vp";
stats.input_frame_rate = 1;
stats.encode_frame_rate = 2;
stats.avg_encode_time_ms = 3;
stats.encode_usage_percent = 4;
stats.frames_encoded = 5;
stats.total_encode_time_ms = 6;
stats.frames_dropped_by_capturer = 7;
stats.frames_dropped_by_encoder_queue = 8;
stats.frames_dropped_by_rate_limiter = 9;
stats.frames_dropped_by_congestion_window = 10;
stats.frames_dropped_by_encoder = 11;
stats.target_media_bitrate_bps = 13;
stats.media_bitrate_bps = 14;
stats.suspended = true;
stats.bw_limited_resolution = true;
stats.cpu_limited_resolution = true;
// Not wired.
stats.bw_limited_framerate = true;
// Not wired.
stats.cpu_limited_framerate = true;
stats.quality_limitation_reason = webrtc::QualityLimitationReason::kCpu;
stats.quality_limitation_durations_ms[webrtc::QualityLimitationReason::kCpu] =
15;
stats.quality_limitation_resolution_changes = 16;
stats.number_of_cpu_adapt_changes = 17;
stats.number_of_quality_adapt_changes = 18;
stats.has_entered_low_resolution = true;
stats.content_type = webrtc::VideoContentType::SCREENSHARE;
stats.frames_sent = 19;
stats.huge_frames_sent = 20;
return stats;
}
TEST_F(WebRtcVideoChannelTest, GetAggregatedStatsReportWithoutSubStreams) {
FakeVideoSendStream* stream = AddSendStream();
auto stats = GetInitialisedStats();
stream->SetStats(stats);
cricket::VideoMediaInfo video_media_info;
ASSERT_TRUE(channel_->GetStats(&video_media_info));
EXPECT_EQ(video_media_info.aggregated_senders.size(), 1u);
auto& sender = video_media_info.aggregated_senders[0];
// MediaSenderInfo
EXPECT_EQ(sender.payload_bytes_sent, 0);
EXPECT_EQ(sender.header_and_padding_bytes_sent, 0);
EXPECT_EQ(sender.retransmitted_bytes_sent, 0u);
EXPECT_EQ(sender.packets_sent, 0);
EXPECT_EQ(sender.retransmitted_packets_sent, 0u);
EXPECT_EQ(sender.packets_lost, 0);
EXPECT_EQ(sender.fraction_lost, 0.0f);
EXPECT_EQ(sender.rtt_ms, 0);
EXPECT_EQ(sender.codec_name, DefaultCodec().name);
EXPECT_EQ(sender.codec_payload_type, DefaultCodec().id);
EXPECT_EQ(sender.local_stats.size(), 1u);
EXPECT_EQ(sender.local_stats[0].ssrc, last_ssrc_);
EXPECT_EQ(sender.local_stats[0].timestamp, 0.0f);
EXPECT_EQ(sender.remote_stats.size(), 0u);
EXPECT_EQ(sender.report_block_datas.size(), 0u);
// VideoSenderInfo
EXPECT_EQ(sender.ssrc_groups.size(), 0u);
EXPECT_EQ(sender.encoder_implementation_name,
stats.encoder_implementation_name);
// Comes from substream only.
EXPECT_EQ(sender.firs_rcvd, 0);
EXPECT_EQ(sender.plis_rcvd, 0);
EXPECT_EQ(sender.nacks_rcvd, 0u);
EXPECT_EQ(sender.send_frame_width, 0);
EXPECT_EQ(sender.send_frame_height, 0);
EXPECT_EQ(sender.framerate_input, stats.input_frame_rate);
EXPECT_EQ(sender.framerate_sent, stats.encode_frame_rate);
EXPECT_EQ(sender.nominal_bitrate, stats.media_bitrate_bps);
EXPECT_NE(sender.adapt_reason & WebRtcVideoChannel::ADAPTREASON_CPU, 0);
EXPECT_NE(sender.adapt_reason & WebRtcVideoChannel::ADAPTREASON_BANDWIDTH, 0);
EXPECT_EQ(sender.adapt_changes, stats.number_of_cpu_adapt_changes);
EXPECT_EQ(sender.quality_limitation_reason, stats.quality_limitation_reason);
EXPECT_EQ(sender.quality_limitation_durations_ms,
stats.quality_limitation_durations_ms);
EXPECT_EQ(sender.quality_limitation_resolution_changes,
stats.quality_limitation_resolution_changes);
EXPECT_EQ(sender.avg_encode_ms, stats.avg_encode_time_ms);
EXPECT_EQ(sender.encode_usage_percent, stats.encode_usage_percent);
EXPECT_EQ(sender.frames_encoded, stats.frames_encoded);
// Comes from substream only.
EXPECT_EQ(sender.key_frames_encoded, 0u);
EXPECT_EQ(sender.total_encode_time_ms, stats.total_encode_time_ms);
EXPECT_EQ(sender.total_encoded_bytes_target,
stats.total_encoded_bytes_target);
// Comes from substream only.
EXPECT_EQ(sender.total_packet_send_delay_ms, 0u);
EXPECT_EQ(sender.qp_sum, absl::nullopt);
EXPECT_EQ(sender.has_entered_low_resolution,
stats.has_entered_low_resolution);
EXPECT_EQ(sender.content_type, webrtc::VideoContentType::SCREENSHARE);
EXPECT_EQ(sender.frames_sent, stats.frames_encoded);
EXPECT_EQ(sender.huge_frames_sent, stats.huge_frames_sent);
EXPECT_EQ(sender.rid, absl::nullopt);
}
TEST_F(WebRtcVideoChannelTest, GetAggregatedStatsReportForSubStreams) {
FakeVideoSendStream* stream = AddSendStream();
auto stats = GetInitialisedStats();
const uint32_t ssrc_1 = 123u;
const uint32_t ssrc_2 = 456u;
auto& substream = stats.substreams[ssrc_1];
substream.frame_counts.key_frames = 1;
substream.frame_counts.delta_frames = 2;
substream.width = 3;
substream.height = 4;
substream.total_bitrate_bps = 5;
substream.retransmit_bitrate_bps = 6;
substream.avg_delay_ms = 7;
substream.max_delay_ms = 8;
substream.total_packet_send_delay_ms = 9;
substream.rtp_stats.transmitted.header_bytes = 10;
substream.rtp_stats.transmitted.padding_bytes = 11;
substream.rtp_stats.retransmitted.payload_bytes = 12;
substream.rtp_stats.retransmitted.packets = 13;
substream.rtcp_packet_type_counts.fir_packets = 14;
substream.rtcp_packet_type_counts.nack_packets = 15;
substream.rtcp_packet_type_counts.pli_packets = 16;
webrtc::RTCPReportBlock report_block;
report_block.packets_lost = 17;
report_block.fraction_lost = 18;
webrtc::ReportBlockData report_block_data;
report_block_data.SetReportBlock(report_block, 0);
report_block_data.AddRoundTripTimeSample(19);
substream.report_block_data = report_block_data;
substream.encode_frame_rate = 20.0;
substream.frames_encoded = 21;
substream.qp_sum = 22;
substream.total_encode_time_ms = 23;
substream.total_encoded_bytes_target = 24;
substream.huge_frames_sent = 25;
stats.substreams[ssrc_2] = substream;
stream->SetStats(stats);
cricket::VideoMediaInfo video_media_info;
ASSERT_TRUE(channel_->GetStats(&video_media_info));
EXPECT_EQ(video_media_info.aggregated_senders.size(), 1u);
auto& sender = video_media_info.aggregated_senders[0];
// MediaSenderInfo
EXPECT_EQ(
sender.payload_bytes_sent,
static_cast<int64_t>(2u * substream.rtp_stats.transmitted.payload_bytes));
EXPECT_EQ(sender.header_and_padding_bytes_sent,
static_cast<int64_t>(
2u * (substream.rtp_stats.transmitted.header_bytes +
substream.rtp_stats.transmitted.padding_bytes)));
EXPECT_EQ(sender.retransmitted_bytes_sent,
2u * substream.rtp_stats.retransmitted.payload_bytes);
EXPECT_EQ(sender.packets_sent,
static_cast<int>(2 * substream.rtp_stats.transmitted.packets));
EXPECT_EQ(sender.retransmitted_packets_sent,
2u * substream.rtp_stats.retransmitted.packets);
EXPECT_EQ(sender.packets_lost,
2 * substream.report_block_data->report_block().packets_lost);
EXPECT_EQ(sender.fraction_lost,
static_cast<float>(
substream.report_block_data->report_block().fraction_lost) /
(1 << 8));
EXPECT_EQ(sender.rtt_ms, 0);
EXPECT_EQ(sender.codec_name, DefaultCodec().name);
EXPECT_EQ(sender.codec_payload_type, DefaultCodec().id);
EXPECT_EQ(sender.local_stats.size(), 1u);
EXPECT_EQ(sender.local_stats[0].ssrc, last_ssrc_);
EXPECT_EQ(sender.local_stats[0].timestamp, 0.0f);
EXPECT_EQ(sender.remote_stats.size(), 0u);
EXPECT_EQ(sender.report_block_datas.size(), 2u * 1);
// VideoSenderInfo
EXPECT_EQ(sender.ssrc_groups.size(), 0u);
EXPECT_EQ(sender.encoder_implementation_name,
stats.encoder_implementation_name);
EXPECT_EQ(
sender.firs_rcvd,
static_cast<int>(2 * substream.rtcp_packet_type_counts.fir_packets));
EXPECT_EQ(
sender.plis_rcvd,
static_cast<int>(2 * substream.rtcp_packet_type_counts.pli_packets));
EXPECT_EQ(sender.nacks_rcvd,
2 * substream.rtcp_packet_type_counts.nack_packets);
EXPECT_EQ(sender.send_frame_width, substream.width);
EXPECT_EQ(sender.send_frame_height, substream.height);
EXPECT_EQ(sender.framerate_input, stats.input_frame_rate);
EXPECT_EQ(sender.framerate_sent, stats.encode_frame_rate);
EXPECT_EQ(sender.nominal_bitrate, stats.media_bitrate_bps);
EXPECT_NE(sender.adapt_reason & WebRtcVideoChannel::ADAPTREASON_CPU, 0);
EXPECT_NE(sender.adapt_reason & WebRtcVideoChannel::ADAPTREASON_BANDWIDTH, 0);
EXPECT_EQ(sender.adapt_changes, stats.number_of_cpu_adapt_changes);
EXPECT_EQ(sender.quality_limitation_reason, stats.quality_limitation_reason);
EXPECT_EQ(sender.quality_limitation_durations_ms,
stats.quality_limitation_durations_ms);
EXPECT_EQ(sender.quality_limitation_resolution_changes,
stats.quality_limitation_resolution_changes);
EXPECT_EQ(sender.avg_encode_ms, stats.avg_encode_time_ms);
EXPECT_EQ(sender.encode_usage_percent, stats.encode_usage_percent);
EXPECT_EQ(sender.frames_encoded, 2u * substream.frames_encoded);
EXPECT_EQ(sender.key_frames_encoded, 2u * substream.frame_counts.key_frames);
EXPECT_EQ(sender.total_encode_time_ms, 2u * substream.total_encode_time_ms);
EXPECT_EQ(sender.total_encoded_bytes_target,
2u * substream.total_encoded_bytes_target);
EXPECT_EQ(sender.total_packet_send_delay_ms,
2u * substream.total_packet_send_delay_ms);
EXPECT_EQ(sender.has_entered_low_resolution,
stats.has_entered_low_resolution);
EXPECT_EQ(sender.qp_sum, 2u * *substream.qp_sum);
EXPECT_EQ(sender.content_type, webrtc::VideoContentType::SCREENSHARE);
EXPECT_EQ(sender.frames_sent, 2u * substream.frames_encoded);
EXPECT_EQ(sender.huge_frames_sent, stats.huge_frames_sent);
EXPECT_EQ(sender.rid, absl::nullopt);
}
TEST_F(WebRtcVideoChannelTest, GetPerLayerStatsReportForSubStreams) {
FakeVideoSendStream* stream = AddSendStream();
auto stats = GetInitialisedStats();
const uint32_t ssrc_1 = 123u;
const uint32_t ssrc_2 = 456u;
auto& substream = stats.substreams[ssrc_1];
substream.frame_counts.key_frames = 1;
substream.frame_counts.delta_frames = 2;
substream.width = 3;
substream.height = 4;
substream.total_bitrate_bps = 5;
substream.retransmit_bitrate_bps = 6;
substream.avg_delay_ms = 7;
substream.max_delay_ms = 8;
substream.total_packet_send_delay_ms = 9;
substream.rtp_stats.transmitted.header_bytes = 10;
substream.rtp_stats.transmitted.padding_bytes = 11;
substream.rtp_stats.retransmitted.payload_bytes = 12;
substream.rtp_stats.retransmitted.packets = 13;
substream.rtcp_packet_type_counts.fir_packets = 14;
substream.rtcp_packet_type_counts.nack_packets = 15;
substream.rtcp_packet_type_counts.pli_packets = 16;
webrtc::RTCPReportBlock report_block;
report_block.packets_lost = 17;
report_block.fraction_lost = 18;
webrtc::ReportBlockData report_block_data;
report_block_data.SetReportBlock(report_block, 0);
report_block_data.AddRoundTripTimeSample(19);
substream.report_block_data = report_block_data;
substream.encode_frame_rate = 20.0;
substream.frames_encoded = 21;
substream.qp_sum = 22;
substream.total_encode_time_ms = 23;
substream.total_encoded_bytes_target = 24;
substream.huge_frames_sent = 25;
stats.substreams[ssrc_2] = substream;
stream->SetStats(stats);
cricket::VideoMediaInfo video_media_info;
ASSERT_TRUE(channel_->GetStats(&video_media_info));
EXPECT_EQ(video_media_info.senders.size(), 2u);
auto& sender = video_media_info.senders[0];
// MediaSenderInfo
EXPECT_EQ(
sender.payload_bytes_sent,
static_cast<int64_t>(substream.rtp_stats.transmitted.payload_bytes));
EXPECT_EQ(
sender.header_and_padding_bytes_sent,
static_cast<int64_t>(substream.rtp_stats.transmitted.header_bytes +
substream.rtp_stats.transmitted.padding_bytes));
EXPECT_EQ(sender.retransmitted_bytes_sent,
substream.rtp_stats.retransmitted.payload_bytes);
EXPECT_EQ(sender.packets_sent,
static_cast<int>(substream.rtp_stats.transmitted.packets));
EXPECT_EQ(sender.retransmitted_packets_sent,
substream.rtp_stats.retransmitted.packets);
EXPECT_EQ(sender.packets_lost,
substream.report_block_data->report_block().packets_lost);
EXPECT_EQ(sender.fraction_lost,
static_cast<float>(
substream.report_block_data->report_block().fraction_lost) /
(1 << 8));
EXPECT_EQ(sender.rtt_ms, 0);
EXPECT_EQ(sender.codec_name, DefaultCodec().name);
EXPECT_EQ(sender.codec_payload_type, DefaultCodec().id);
EXPECT_EQ(sender.local_stats.size(), 1u);
EXPECT_EQ(sender.local_stats[0].ssrc, ssrc_1);
EXPECT_EQ(sender.local_stats[0].timestamp, 0.0f);
EXPECT_EQ(sender.remote_stats.size(), 0u);
EXPECT_EQ(sender.report_block_datas.size(), 1u);
// VideoSenderInfo
EXPECT_EQ(sender.ssrc_groups.size(), 0u);
EXPECT_EQ(sender.encoder_implementation_name,
stats.encoder_implementation_name);
EXPECT_EQ(sender.firs_rcvd,
static_cast<int>(substream.rtcp_packet_type_counts.fir_packets));
EXPECT_EQ(sender.plis_rcvd,
static_cast<int>(substream.rtcp_packet_type_counts.pli_packets));
EXPECT_EQ(sender.nacks_rcvd, substream.rtcp_packet_type_counts.nack_packets);
EXPECT_EQ(sender.send_frame_width, substream.width);
EXPECT_EQ(sender.send_frame_height, substream.height);
EXPECT_EQ(sender.framerate_input, stats.input_frame_rate);
EXPECT_EQ(sender.framerate_sent, substream.encode_frame_rate);
EXPECT_EQ(sender.nominal_bitrate, stats.media_bitrate_bps);
EXPECT_NE(sender.adapt_reason & WebRtcVideoChannel::ADAPTREASON_CPU, 0);
EXPECT_NE(sender.adapt_reason & WebRtcVideoChannel::ADAPTREASON_BANDWIDTH, 0);
EXPECT_EQ(sender.adapt_changes, stats.number_of_cpu_adapt_changes);
EXPECT_EQ(sender.quality_limitation_reason, stats.quality_limitation_reason);
EXPECT_EQ(sender.quality_limitation_durations_ms,
stats.quality_limitation_durations_ms);
EXPECT_EQ(sender.quality_limitation_resolution_changes,
stats.quality_limitation_resolution_changes);
EXPECT_EQ(sender.avg_encode_ms, stats.avg_encode_time_ms);
EXPECT_EQ(sender.encode_usage_percent, stats.encode_usage_percent);
EXPECT_EQ(sender.frames_encoded,
static_cast<uint32_t>(substream.frames_encoded));
EXPECT_EQ(sender.key_frames_encoded,
static_cast<uint32_t>(substream.frame_counts.key_frames));
EXPECT_EQ(sender.total_encode_time_ms, substream.total_encode_time_ms);
EXPECT_EQ(sender.total_encoded_bytes_target,
substream.total_encoded_bytes_target);
EXPECT_EQ(sender.total_packet_send_delay_ms,
substream.total_packet_send_delay_ms);
EXPECT_EQ(sender.has_entered_low_resolution,
stats.has_entered_low_resolution);
EXPECT_EQ(sender.qp_sum, *substream.qp_sum);
EXPECT_EQ(sender.content_type, webrtc::VideoContentType::SCREENSHARE);
EXPECT_EQ(sender.frames_sent,
static_cast<uint32_t>(substream.frames_encoded));
EXPECT_EQ(sender.huge_frames_sent, substream.huge_frames_sent);
EXPECT_EQ(sender.rid, absl::nullopt);
}
TEST_F(WebRtcVideoChannelTest, MediaSubstreamMissingProducesEmpyStats) {
FakeVideoSendStream* stream = AddSendStream();
const uint32_t kRtxSsrc = 123u;
const uint32_t kMissingMediaSsrc = 124u;
// Set up a scenarios where we have a substream that is not kMedia (in this
// case: kRtx) but its associated kMedia stream does not exist yet. This
// results in zero GetPerLayerVideoSenderInfos despite non-empty substreams.
// Covers https://crbug.com/1090712.
auto stats = GetInitialisedStats();
auto& substream = stats.substreams[kRtxSsrc];
substream.type = webrtc::VideoSendStream::StreamStats::StreamType::kRtx;
substream.referenced_media_ssrc = kMissingMediaSsrc;
stream->SetStats(stats);
cricket::VideoMediaInfo video_media_info;
ASSERT_TRUE(channel_->GetStats(&video_media_info));
EXPECT_TRUE(video_media_info.senders.empty());
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsUpperResolution) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.substreams[17].width = 123;
stats.substreams[17].height = 40;
stats.substreams[42].width = 80;
stats.substreams[42].height = 31;
stats.substreams[11].width = 20;
stats.substreams[11].height = 90;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1u, info.aggregated_senders.size());
ASSERT_EQ(3u, info.senders.size());
EXPECT_EQ(123, info.senders[1].send_frame_width);
EXPECT_EQ(40, info.senders[1].send_frame_height);
EXPECT_EQ(80, info.senders[2].send_frame_width);
EXPECT_EQ(31, info.senders[2].send_frame_height);
EXPECT_EQ(20, info.senders[0].send_frame_width);
EXPECT_EQ(90, info.senders[0].send_frame_height);
EXPECT_EQ(123, info.aggregated_senders[0].send_frame_width);
EXPECT_EQ(90, info.aggregated_senders[0].send_frame_height);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsCpuAdaptationStats) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.number_of_cpu_adapt_changes = 2;
stats.cpu_limited_resolution = true;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1U, info.senders.size());
EXPECT_EQ(WebRtcVideoChannel::ADAPTREASON_CPU, info.senders[0].adapt_reason);
EXPECT_EQ(stats.number_of_cpu_adapt_changes, info.senders[0].adapt_changes);
}
TEST_F(WebRtcVideoChannelTest, GetStatsReportsAdaptationAndBandwidthStats) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.number_of_cpu_adapt_changes = 2;
stats.cpu_limited_resolution = true;
stats.bw_limited_resolution = true;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1U, info.senders.size());
EXPECT_EQ(WebRtcVideoChannel::ADAPTREASON_CPU |
WebRtcVideoChannel::ADAPTREASON_BANDWIDTH,
info.senders[0].adapt_reason);
EXPECT_EQ(stats.number_of_cpu_adapt_changes, info.senders[0].adapt_changes);
}
TEST(WebRtcVideoChannelHelperTest, MergeInfoAboutOutboundRtpSubstreams) {
const uint32_t kFirstMediaStreamSsrc = 10;
const uint32_t kSecondMediaStreamSsrc = 20;
const uint32_t kRtxSsrc = 30;
const uint32_t kFlexfecSsrc = 40;
std::map<uint32_t, webrtc::VideoSendStream::StreamStats> substreams;
// First kMedia stream.
substreams[kFirstMediaStreamSsrc].type =
webrtc::VideoSendStream::StreamStats::StreamType::kMedia;
substreams[kFirstMediaStreamSsrc].rtp_stats.transmitted.header_bytes = 1;
substreams[kFirstMediaStreamSsrc].rtp_stats.transmitted.padding_bytes = 2;
substreams[kFirstMediaStreamSsrc].rtp_stats.transmitted.payload_bytes = 3;
substreams[kFirstMediaStreamSsrc].rtp_stats.transmitted.packets = 4;
substreams[kFirstMediaStreamSsrc].rtp_stats.retransmitted.header_bytes = 5;
substreams[kFirstMediaStreamSsrc].rtp_stats.retransmitted.padding_bytes = 6;
substreams[kFirstMediaStreamSsrc].rtp_stats.retransmitted.payload_bytes = 7;
substreams[kFirstMediaStreamSsrc].rtp_stats.retransmitted.packets = 8;
substreams[kFirstMediaStreamSsrc].referenced_media_ssrc = absl::nullopt;
substreams[kFirstMediaStreamSsrc].width = 1280;
substreams[kFirstMediaStreamSsrc].height = 720;
// Second kMedia stream.
substreams[kSecondMediaStreamSsrc].type =
webrtc::VideoSendStream::StreamStats::StreamType::kMedia;
substreams[kSecondMediaStreamSsrc].rtp_stats.transmitted.header_bytes = 10;
substreams[kSecondMediaStreamSsrc].rtp_stats.transmitted.padding_bytes = 11;
substreams[kSecondMediaStreamSsrc].rtp_stats.transmitted.payload_bytes = 12;
substreams[kSecondMediaStreamSsrc].rtp_stats.transmitted.packets = 13;
substreams[kSecondMediaStreamSsrc].rtp_stats.retransmitted.header_bytes = 14;
substreams[kSecondMediaStreamSsrc].rtp_stats.retransmitted.padding_bytes = 15;
substreams[kSecondMediaStreamSsrc].rtp_stats.retransmitted.payload_bytes = 16;
substreams[kSecondMediaStreamSsrc].rtp_stats.retransmitted.packets = 17;
substreams[kSecondMediaStreamSsrc].referenced_media_ssrc = absl::nullopt;
substreams[kSecondMediaStreamSsrc].width = 640;
substreams[kSecondMediaStreamSsrc].height = 480;
// kRtx stream referencing the first kMedia stream.
substreams[kRtxSsrc].type =
webrtc::VideoSendStream::StreamStats::StreamType::kRtx;
substreams[kRtxSsrc].rtp_stats.transmitted.header_bytes = 19;
substreams[kRtxSsrc].rtp_stats.transmitted.padding_bytes = 20;
substreams[kRtxSsrc].rtp_stats.transmitted.payload_bytes = 21;
substreams[kRtxSsrc].rtp_stats.transmitted.packets = 22;
substreams[kRtxSsrc].rtp_stats.retransmitted.header_bytes = 23;
substreams[kRtxSsrc].rtp_stats.retransmitted.padding_bytes = 24;
substreams[kRtxSsrc].rtp_stats.retransmitted.payload_bytes = 25;
substreams[kRtxSsrc].rtp_stats.retransmitted.packets = 26;
substreams[kRtxSsrc].referenced_media_ssrc = kFirstMediaStreamSsrc;
// kFlexfec stream referencing the second kMedia stream.
substreams[kFlexfecSsrc].type =
webrtc::VideoSendStream::StreamStats::StreamType::kFlexfec;
substreams[kFlexfecSsrc].rtp_stats.transmitted.header_bytes = 19;
substreams[kFlexfecSsrc].rtp_stats.transmitted.padding_bytes = 20;
substreams[kFlexfecSsrc].rtp_stats.transmitted.payload_bytes = 21;
substreams[kFlexfecSsrc].rtp_stats.transmitted.packets = 22;
substreams[kFlexfecSsrc].rtp_stats.retransmitted.header_bytes = 23;
substreams[kFlexfecSsrc].rtp_stats.retransmitted.padding_bytes = 24;
substreams[kFlexfecSsrc].rtp_stats.retransmitted.payload_bytes = 25;
substreams[kFlexfecSsrc].rtp_stats.retransmitted.packets = 26;
substreams[kFlexfecSsrc].referenced_media_ssrc = kSecondMediaStreamSsrc;
auto merged_substreams =
MergeInfoAboutOutboundRtpSubstreamsForTesting(substreams);
// Only kMedia substreams remain.
EXPECT_TRUE(merged_substreams.find(kFirstMediaStreamSsrc) !=
merged_substreams.end());
EXPECT_EQ(merged_substreams[kFirstMediaStreamSsrc].type,
webrtc::VideoSendStream::StreamStats::StreamType::kMedia);
EXPECT_TRUE(merged_substreams.find(kSecondMediaStreamSsrc) !=
merged_substreams.end());
EXPECT_EQ(merged_substreams[kSecondMediaStreamSsrc].type,
webrtc::VideoSendStream::StreamStats::StreamType::kMedia);
EXPECT_FALSE(merged_substreams.find(kRtxSsrc) != merged_substreams.end());
EXPECT_FALSE(merged_substreams.find(kFlexfecSsrc) != merged_substreams.end());
// Expect kFirstMediaStreamSsrc's rtp_stats to be merged with kRtxSsrc.
webrtc::StreamDataCounters first_media_expected_rtp_stats =
substreams[kFirstMediaStreamSsrc].rtp_stats;
first_media_expected_rtp_stats.Add(substreams[kRtxSsrc].rtp_stats);
EXPECT_EQ(merged_substreams[kFirstMediaStreamSsrc].rtp_stats.transmitted,
first_media_expected_rtp_stats.transmitted);
EXPECT_EQ(merged_substreams[kFirstMediaStreamSsrc].rtp_stats.retransmitted,
first_media_expected_rtp_stats.retransmitted);
// Expect kSecondMediaStreamSsrc' rtp_stats to be merged with kFlexfecSsrc.
webrtc::StreamDataCounters second_media_expected_rtp_stats =
substreams[kSecondMediaStreamSsrc].rtp_stats;
second_media_expected_rtp_stats.Add(substreams[kFlexfecSsrc].rtp_stats);
EXPECT_EQ(merged_substreams[kSecondMediaStreamSsrc].rtp_stats.transmitted,
second_media_expected_rtp_stats.transmitted);
EXPECT_EQ(merged_substreams[kSecondMediaStreamSsrc].rtp_stats.retransmitted,
second_media_expected_rtp_stats.retransmitted);
// Expect other metrics to come from the original kMedia stats.
EXPECT_EQ(merged_substreams[kFirstMediaStreamSsrc].width,
substreams[kFirstMediaStreamSsrc].width);
EXPECT_EQ(merged_substreams[kFirstMediaStreamSsrc].height,
substreams[kFirstMediaStreamSsrc].height);
EXPECT_EQ(merged_substreams[kSecondMediaStreamSsrc].width,
substreams[kSecondMediaStreamSsrc].width);
EXPECT_EQ(merged_substreams[kSecondMediaStreamSsrc].height,
substreams[kSecondMediaStreamSsrc].height);
}
TEST_F(WebRtcVideoChannelTest,
GetStatsReportsTransmittedAndRetransmittedBytesAndPacketsCorrectly) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
// Simulcast layer 1, RTP stream. header+padding=10, payload=20, packets=3.
stats.substreams[101].type =
webrtc::VideoSendStream::StreamStats::StreamType::kMedia;
stats.substreams[101].rtp_stats.transmitted.header_bytes = 5;
stats.substreams[101].rtp_stats.transmitted.padding_bytes = 5;
stats.substreams[101].rtp_stats.transmitted.payload_bytes = 20;
stats.substreams[101].rtp_stats.transmitted.packets = 3;
stats.substreams[101].rtp_stats.retransmitted.header_bytes = 0;
stats.substreams[101].rtp_stats.retransmitted.padding_bytes = 0;
stats.substreams[101].rtp_stats.retransmitted.payload_bytes = 0;
stats.substreams[101].rtp_stats.retransmitted.packets = 0;
stats.substreams[101].referenced_media_ssrc = absl::nullopt;
// Simulcast layer 1, RTX stream. header+padding=5, payload=10, packets=1.
stats.substreams[102].type =
webrtc::VideoSendStream::StreamStats::StreamType::kRtx;
stats.substreams[102].rtp_stats.retransmitted.header_bytes = 3;
stats.substreams[102].rtp_stats.retransmitted.padding_bytes = 2;
stats.substreams[102].rtp_stats.retransmitted.payload_bytes = 10;
stats.substreams[102].rtp_stats.retransmitted.packets = 1;
stats.substreams[102].rtp_stats.transmitted =
stats.substreams[102].rtp_stats.retransmitted;
stats.substreams[102].referenced_media_ssrc = 101;
// Simulcast layer 2, RTP stream. header+padding=20, payload=40, packets=7.
stats.substreams[201].type =
webrtc::VideoSendStream::StreamStats::StreamType::kMedia;
stats.substreams[201].rtp_stats.transmitted.header_bytes = 10;
stats.substreams[201].rtp_stats.transmitted.padding_bytes = 10;
stats.substreams[201].rtp_stats.transmitted.payload_bytes = 40;
stats.substreams[201].rtp_stats.transmitted.packets = 7;
stats.substreams[201].rtp_stats.retransmitted.header_bytes = 0;
stats.substreams[201].rtp_stats.retransmitted.padding_bytes = 0;
stats.substreams[201].rtp_stats.retransmitted.payload_bytes = 0;
stats.substreams[201].rtp_stats.retransmitted.packets = 0;
stats.substreams[201].referenced_media_ssrc = absl::nullopt;
// Simulcast layer 2, RTX stream. header+padding=10, payload=20, packets=4.
stats.substreams[202].type =
webrtc::VideoSendStream::StreamStats::StreamType::kRtx;
stats.substreams[202].rtp_stats.retransmitted.header_bytes = 6;
stats.substreams[202].rtp_stats.retransmitted.padding_bytes = 4;
stats.substreams[202].rtp_stats.retransmitted.payload_bytes = 20;
stats.substreams[202].rtp_stats.retransmitted.packets = 4;
stats.substreams[202].rtp_stats.transmitted =
stats.substreams[202].rtp_stats.retransmitted;
stats.substreams[202].referenced_media_ssrc = 201;
// FlexFEC stream associated with the Simulcast layer 2.
// header+padding=15, payload=17, packets=5.
stats.substreams[301].type =
webrtc::VideoSendStream::StreamStats::StreamType::kFlexfec;
stats.substreams[301].rtp_stats.transmitted.header_bytes = 13;
stats.substreams[301].rtp_stats.transmitted.padding_bytes = 2;
stats.substreams[301].rtp_stats.transmitted.payload_bytes = 17;
stats.substreams[301].rtp_stats.transmitted.packets = 5;
stats.substreams[301].rtp_stats.retransmitted.header_bytes = 0;
stats.substreams[301].rtp_stats.retransmitted.padding_bytes = 0;
stats.substreams[301].rtp_stats.retransmitted.payload_bytes = 0;
stats.substreams[301].rtp_stats.retransmitted.packets = 0;
stats.substreams[301].referenced_media_ssrc = 201;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(info.senders.size(), 2u);
EXPECT_EQ(15u, info.senders[0].header_and_padding_bytes_sent);
EXPECT_EQ(30u, info.senders[0].payload_bytes_sent);
EXPECT_EQ(4, info.senders[0].packets_sent);
EXPECT_EQ(10u, info.senders[0].retransmitted_bytes_sent);
EXPECT_EQ(1u, info.senders[0].retransmitted_packets_sent);
EXPECT_EQ(45u, info.senders[1].header_and_padding_bytes_sent);
EXPECT_EQ(77u, info.senders[1].payload_bytes_sent);
EXPECT_EQ(16, info.senders[1].packets_sent);
EXPECT_EQ(20u, info.senders[1].retransmitted_bytes_sent);
EXPECT_EQ(4u, info.senders[1].retransmitted_packets_sent);
}
TEST_F(WebRtcVideoChannelTest,
GetStatsTranslatesBandwidthLimitedResolutionCorrectly) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.bw_limited_resolution = true;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
EXPECT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1U, info.senders.size());
EXPECT_EQ(WebRtcVideoChannel::ADAPTREASON_BANDWIDTH,
info.senders[0].adapt_reason);
}
TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesSendRtcpPacketTypesCorrectly) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.substreams[17].rtcp_packet_type_counts.fir_packets = 2;
stats.substreams[17].rtcp_packet_type_counts.nack_packets = 3;
stats.substreams[17].rtcp_packet_type_counts.pli_packets = 4;
stats.substreams[42].rtcp_packet_type_counts.fir_packets = 5;
stats.substreams[42].rtcp_packet_type_counts.nack_packets = 7;
stats.substreams[42].rtcp_packet_type_counts.pli_packets = 9;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(2, info.senders[0].firs_rcvd);
EXPECT_EQ(3u, info.senders[0].nacks_rcvd);
EXPECT_EQ(4, info.senders[0].plis_rcvd);
EXPECT_EQ(5, info.senders[1].firs_rcvd);
EXPECT_EQ(7u, info.senders[1].nacks_rcvd);
EXPECT_EQ(9, info.senders[1].plis_rcvd);
EXPECT_EQ(7, info.aggregated_senders[0].firs_rcvd);
EXPECT_EQ(10u, info.aggregated_senders[0].nacks_rcvd);
EXPECT_EQ(13, info.aggregated_senders[0].plis_rcvd);
}
TEST_F(WebRtcVideoChannelTest,
GetStatsTranslatesReceiveRtcpPacketTypesCorrectly) {
FakeVideoReceiveStream* stream = AddRecvStream();
webrtc::VideoReceiveStream::Stats stats;
stats.rtcp_packet_type_counts.fir_packets = 2;
stats.rtcp_packet_type_counts.nack_packets = 3;
stats.rtcp_packet_type_counts.pli_packets = 4;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.rtcp_packet_type_counts.fir_packets,
rtc::checked_cast<unsigned int>(info.receivers[0].firs_sent));
EXPECT_EQ(stats.rtcp_packet_type_counts.nack_packets,
info.receivers[0].nacks_sent);
EXPECT_EQ(stats.rtcp_packet_type_counts.pli_packets,
rtc::checked_cast<unsigned int>(info.receivers[0].plis_sent));
}
TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesDecodeStatsCorrectly) {
FakeVideoReceiveStream* stream = AddRecvStream();
webrtc::VideoReceiveStream::Stats stats;
stats.decoder_implementation_name = "decoder_implementation_name";
stats.decode_ms = 2;
stats.max_decode_ms = 3;
stats.current_delay_ms = 4;
stats.target_delay_ms = 5;
stats.jitter_buffer_ms = 6;
stats.jitter_buffer_delay_seconds = 60;
stats.jitter_buffer_emitted_count = 6;
stats.min_playout_delay_ms = 7;
stats.render_delay_ms = 8;
stats.width = 9;
stats.height = 10;
stats.frame_counts.key_frames = 11;
stats.frame_counts.delta_frames = 12;
stats.frames_rendered = 13;
stats.frames_decoded = 14;
stats.qp_sum = 15;
stats.total_decode_time_ms = 16;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.decoder_implementation_name,
info.receivers[0].decoder_implementation_name);
EXPECT_EQ(stats.decode_ms, info.receivers[0].decode_ms);
EXPECT_EQ(stats.max_decode_ms, info.receivers[0].max_decode_ms);
EXPECT_EQ(stats.current_delay_ms, info.receivers[0].current_delay_ms);
EXPECT_EQ(stats.target_delay_ms, info.receivers[0].target_delay_ms);
EXPECT_EQ(stats.jitter_buffer_ms, info.receivers[0].jitter_buffer_ms);
EXPECT_EQ(stats.jitter_buffer_delay_seconds,
info.receivers[0].jitter_buffer_delay_seconds);
EXPECT_EQ(stats.jitter_buffer_emitted_count,
info.receivers[0].jitter_buffer_emitted_count);
EXPECT_EQ(stats.min_playout_delay_ms, info.receivers[0].min_playout_delay_ms);
EXPECT_EQ(stats.render_delay_ms, info.receivers[0].render_delay_ms);
EXPECT_EQ(stats.width, info.receivers[0].frame_width);
EXPECT_EQ(stats.height, info.receivers[0].frame_height);
EXPECT_EQ(rtc::checked_cast<unsigned int>(stats.frame_counts.key_frames +
stats.frame_counts.delta_frames),
info.receivers[0].frames_received);
EXPECT_EQ(stats.frames_rendered, info.receivers[0].frames_rendered);
EXPECT_EQ(stats.frames_decoded, info.receivers[0].frames_decoded);
EXPECT_EQ(rtc::checked_cast<unsigned int>(stats.frame_counts.key_frames),
info.receivers[0].key_frames_decoded);
EXPECT_EQ(stats.qp_sum, info.receivers[0].qp_sum);
EXPECT_EQ(stats.total_decode_time_ms, info.receivers[0].total_decode_time_ms);
}
TEST_F(WebRtcVideoChannelTest,
GetStatsTranslatesInterFrameDelayStatsCorrectly) {
FakeVideoReceiveStream* stream = AddRecvStream();
webrtc::VideoReceiveStream::Stats stats;
stats.total_inter_frame_delay = 0.123;
stats.total_squared_inter_frame_delay = 0.00456;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.total_inter_frame_delay,
info.receivers[0].total_inter_frame_delay);
EXPECT_EQ(stats.total_squared_inter_frame_delay,
info.receivers[0].total_squared_inter_frame_delay);
}
TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesReceivePacketStatsCorrectly) {
FakeVideoReceiveStream* stream = AddRecvStream();
webrtc::VideoReceiveStream::Stats stats;
stats.rtp_stats.packet_counter.payload_bytes = 2;
stats.rtp_stats.packet_counter.header_bytes = 3;
stats.rtp_stats.packet_counter.padding_bytes = 4;
stats.rtp_stats.packet_counter.packets = 5;
stats.rtp_stats.packets_lost = 6;
stream->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_EQ(stats.rtp_stats.packet_counter.payload_bytes,
rtc::checked_cast<size_t>(info.receivers[0].payload_bytes_rcvd));
EXPECT_EQ(stats.rtp_stats.packet_counter.packets,
rtc::checked_cast<unsigned int>(info.receivers[0].packets_rcvd));
EXPECT_EQ(stats.rtp_stats.packets_lost, info.receivers[0].packets_lost);
}
TEST_F(WebRtcVideoChannelTest, TranslatesCallStatsCorrectly) {
AddSendStream();
AddSendStream();
webrtc::Call::Stats stats;
stats.rtt_ms = 123;
fake_call_->SetStats(stats);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(2u, info.senders.size());
EXPECT_EQ(stats.rtt_ms, info.senders[0].rtt_ms);
EXPECT_EQ(stats.rtt_ms, info.senders[1].rtt_ms);
}
TEST_F(WebRtcVideoChannelTest, TranslatesSenderBitrateStatsCorrectly) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
stats.target_media_bitrate_bps = 156;
stats.media_bitrate_bps = 123;
stats.substreams[17].total_bitrate_bps = 1;
stats.substreams[17].retransmit_bitrate_bps = 2;
stats.substreams[42].total_bitrate_bps = 3;
stats.substreams[42].retransmit_bitrate_bps = 4;
stream->SetStats(stats);
FakeVideoSendStream* stream2 = AddSendStream();
webrtc::VideoSendStream::Stats stats2;
stats2.target_media_bitrate_bps = 200;
stats2.media_bitrate_bps = 321;
stats2.substreams[13].total_bitrate_bps = 5;
stats2.substreams[13].retransmit_bitrate_bps = 6;
stats2.substreams[21].total_bitrate_bps = 7;
stats2.substreams[21].retransmit_bitrate_bps = 8;
stream2->SetStats(stats2);
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(2u, info.aggregated_senders.size());
ASSERT_EQ(4u, info.senders.size());
BandwidthEstimationInfo bwe_info;
channel_->FillBitrateInfo(&bwe_info);
// Assuming stream and stream2 corresponds to senders[0] and [1] respectively
// is OK as std::maps are sorted and AddSendStream() gives increasing SSRCs.
EXPECT_EQ(stats.media_bitrate_bps,
info.aggregated_senders[0].nominal_bitrate);
EXPECT_EQ(stats2.media_bitrate_bps,
info.aggregated_senders[1].nominal_bitrate);
EXPECT_EQ(stats.target_media_bitrate_bps + stats2.target_media_bitrate_bps,
bwe_info.target_enc_bitrate);
EXPECT_EQ(stats.media_bitrate_bps + stats2.media_bitrate_bps,
bwe_info.actual_enc_bitrate);
EXPECT_EQ(1 + 3 + 5 + 7, bwe_info.transmit_bitrate)
<< "Bandwidth stats should take all streams into account.";
EXPECT_EQ(2 + 4 + 6 + 8, bwe_info.retransmit_bitrate)
<< "Bandwidth stats should take all streams into account.";
}
TEST_F(WebRtcVideoChannelTest, DefaultReceiveStreamReconfiguresToUseRtx) {
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1);
const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1);
ASSERT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
RtpPacket packet;
packet.SetSsrc(ssrcs[0]);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size())
<< "No default receive stream created.";
FakeVideoReceiveStream* recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(0u, recv_stream->GetConfig().rtp.rtx_ssrc)
<< "Default receive stream should not have configured RTX";
EXPECT_TRUE(channel_->AddRecvStream(
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)));
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size())
<< "AddRecvStream should have reconfigured, not added a new receiver.";
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_FALSE(
recv_stream->GetConfig().rtp.rtx_associated_payload_types.empty());
EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
<< "RTX should be mapped for all decoders/payload types.";
EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
GetEngineCodec("red").id))
<< "RTX should be mapped also for the RED payload type";
EXPECT_EQ(rtx_ssrcs[0], recv_stream->GetConfig().rtp.rtx_ssrc);
}
TEST_F(WebRtcVideoChannelTest, RejectsAddingStreamsWithMissingSsrcsForRtx) {
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1);
const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1);
StreamParams sp =
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs);
sp.ssrcs = ssrcs; // Without RTXs, this is the important part.
EXPECT_FALSE(channel_->AddSendStream(sp));
EXPECT_FALSE(channel_->AddRecvStream(sp));
}
TEST_F(WebRtcVideoChannelTest, RejectsAddingStreamsWithOverlappingRtxSsrcs) {
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1);
const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1);
StreamParams sp =
cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs);
EXPECT_TRUE(channel_->AddSendStream(sp));
EXPECT_TRUE(channel_->AddRecvStream(sp));
// The RTX SSRC is already used in previous streams, using it should fail.
sp = cricket::StreamParams::CreateLegacy(rtx_ssrcs[0]);
EXPECT_FALSE(channel_->AddSendStream(sp));
EXPECT_FALSE(channel_->AddRecvStream(sp));
// After removing the original stream this should be fine to add (makes sure
// that RTX ssrcs are not forever taken).
EXPECT_TRUE(channel_->RemoveSendStream(ssrcs[0]));
EXPECT_TRUE(channel_->RemoveRecvStream(ssrcs[0]));
EXPECT_TRUE(channel_->AddSendStream(sp));
EXPECT_TRUE(channel_->AddRecvStream(sp));
}
TEST_F(WebRtcVideoChannelTest,
RejectsAddingStreamsWithOverlappingSimulcastSsrcs) {
static const uint32_t kFirstStreamSsrcs[] = {1, 2, 3};
static const uint32_t kOverlappingStreamSsrcs[] = {4, 3, 5};
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
StreamParams sp =
cricket::CreateSimStreamParams("cname", MAKE_VECTOR(kFirstStreamSsrcs));
EXPECT_TRUE(channel_->AddSendStream(sp));
EXPECT_TRUE(channel_->AddRecvStream(sp));
// One of the SSRCs is already used in previous streams, using it should fail.
sp = cricket::CreateSimStreamParams("cname",
MAKE_VECTOR(kOverlappingStreamSsrcs));
EXPECT_FALSE(channel_->AddSendStream(sp));
EXPECT_FALSE(channel_->AddRecvStream(sp));
// After removing the original stream this should be fine to add (makes sure
// that RTX ssrcs are not forever taken).
EXPECT_TRUE(channel_->RemoveSendStream(kFirstStreamSsrcs[0]));
EXPECT_TRUE(channel_->RemoveRecvStream(kFirstStreamSsrcs[0]));
EXPECT_TRUE(channel_->AddSendStream(sp));
EXPECT_TRUE(channel_->AddRecvStream(sp));
}
TEST_F(WebRtcVideoChannelTest, ReportsSsrcGroupsInStats) {
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
static const uint32_t kSenderSsrcs[] = {4, 7, 10};
static const uint32_t kSenderRtxSsrcs[] = {5, 8, 11};
StreamParams sender_sp = cricket::CreateSimWithRtxStreamParams(
"cname", MAKE_VECTOR(kSenderSsrcs), MAKE_VECTOR(kSenderRtxSsrcs));
EXPECT_TRUE(channel_->AddSendStream(sender_sp));
static const uint32_t kReceiverSsrcs[] = {3};
static const uint32_t kReceiverRtxSsrcs[] = {2};
StreamParams receiver_sp = cricket::CreateSimWithRtxStreamParams(
"cname", MAKE_VECTOR(kReceiverSsrcs), MAKE_VECTOR(kReceiverRtxSsrcs));
EXPECT_TRUE(channel_->AddRecvStream(receiver_sp));
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
ASSERT_EQ(1u, info.senders.size());
ASSERT_EQ(1u, info.receivers.size());
EXPECT_NE(sender_sp.ssrc_groups, receiver_sp.ssrc_groups);
EXPECT_EQ(sender_sp.ssrc_groups, info.senders[0].ssrc_groups);
EXPECT_EQ(receiver_sp.ssrc_groups, info.receivers[0].ssrc_groups);
}
TEST_F(WebRtcVideoChannelTest, MapsReceivedPayloadTypeToCodecName) {
FakeVideoReceiveStream* stream = AddRecvStream();
webrtc::VideoReceiveStream::Stats stats;
cricket::VideoMediaInfo info;
// Report no codec name before receiving.
stream->SetStats(stats);
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_STREQ("", info.receivers[0].codec_name.c_str());
// Report VP8 if we're receiving it.
stats.current_payload_type = GetEngineCodec("VP8").id;
stream->SetStats(stats);
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_STREQ(kVp8CodecName, info.receivers[0].codec_name.c_str());
// Report no codec name for unknown playload types.
stats.current_payload_type = 3;
stream->SetStats(stats);
ASSERT_TRUE(channel_->GetStats(&info));
EXPECT_STREQ("", info.receivers[0].codec_name.c_str());
}
// Tests that when we add a stream without SSRCs, but contains a stream_id
// that it is stored and its stream id is later used when the first packet
// arrives to properly create a receive stream with a sync label.
TEST_F(WebRtcVideoChannelTest, RecvUnsignaledSsrcWithSignaledStreamId) {
const char kSyncLabel[] = "sync_label";
cricket::StreamParams unsignaled_stream;
unsignaled_stream.set_stream_ids({kSyncLabel});
ASSERT_TRUE(channel_->AddRecvStream(unsignaled_stream));
channel_->OnDemuxerCriteriaUpdatePending();
channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
// The stream shouldn't have been created at this point because it doesn't
// have any SSRCs.
EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
// Create and deliver packet.
RtpPacket packet;
packet.SetSsrc(kIncomingUnsignalledSsrc);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
// The stream should now be created with the appropriate sync label.
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
EXPECT_EQ(kSyncLabel,
fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group);
// Reset the unsignaled stream to clear the cache. This deletes the receive
// stream.
channel_->ResetUnsignaledRecvStream();
channel_->OnDemuxerCriteriaUpdatePending();
EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
// Until the demuxer criteria has been updated, we ignore in-flight ssrcs of
// the recently removed unsignaled receive stream.
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
// After the demuxer criteria has been updated, we should proceed to create
// unsignalled receive streams. This time when a default video receive stream
// is created it won't have a sync_group.
channel_->OnDemuxerCriteriaUpdateComplete();
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
EXPECT_TRUE(
fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group.empty());
}
TEST_F(WebRtcVideoChannelTest,
ResetUnsignaledRecvStreamDeletesAllDefaultStreams) {
// No receive streams to start with.
EXPECT_TRUE(fake_call_->GetVideoReceiveStreams().empty());
// Packet with unsignaled SSRC is received.
RtpPacket packet;
packet.SetSsrc(kIncomingUnsignalledSsrc);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
// Default receive stream created.
const auto& receivers1 = fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(receivers1.size(), 1u);
EXPECT_EQ(receivers1[0]->GetConfig().rtp.remote_ssrc,
kIncomingUnsignalledSsrc);
// Stream with another SSRC gets signaled.
channel_->ResetUnsignaledRecvStream();
constexpr uint32_t kIncomingSignalledSsrc = kIncomingUnsignalledSsrc + 1;
ASSERT_TRUE(channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kIncomingSignalledSsrc)));
// New receiver is for the signaled stream.
const auto& receivers2 = fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(receivers2.size(), 1u);
EXPECT_EQ(receivers2[0]->GetConfig().rtp.remote_ssrc, kIncomingSignalledSsrc);
}
TEST_F(WebRtcVideoChannelTest,
RecentlyAddedSsrcsDoNotCreateUnsignalledRecvStreams) {
const uint32_t kSsrc1 = 1;
const uint32_t kSsrc2 = 2;
// Starting point: receiving kSsrc1.
EXPECT_TRUE(channel_->AddRecvStream(StreamParams::CreateLegacy(kSsrc1)));
channel_->OnDemuxerCriteriaUpdatePending();
channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u);
// If this is the only m= section the demuxer might be configure to forward
// all packets, regardless of ssrc, to this channel. When we go to multiple m=
// sections, there can thus be a window of time where packets that should
// never have belonged to this channel arrive anyway.
// Emulate a second m= section being created by updating the demuxer criteria
// without adding any streams.
channel_->OnDemuxerCriteriaUpdatePending();
// Emulate there being in-flight packets for kSsrc1 and kSsrc2 arriving before
// the demuxer is updated.
{
// Receive a packet for kSsrc1.
RtpPacket packet;
packet.SetSsrc(kSsrc1);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
}
{
// Receive a packet for kSsrc2.
RtpPacket packet;
packet.SetSsrc(kSsrc2);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
}
// No unsignaled ssrc for kSsrc2 should have been created, but kSsrc1 should
// arrive since it already has a stream.
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 1u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 0u);
// Signal that the demuxer update is complete. Because there are no more
// pending demuxer updates, receiving unknown ssrcs (kSsrc2) should again
// result in unsignalled receive streams being created.
channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
// Receive packets for kSsrc1 and kSsrc2 again.
{
// Receive a packet for kSsrc1.
RtpPacket packet;
packet.SetSsrc(kSsrc1);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
}
{
// Receive a packet for kSsrc2.
RtpPacket packet;
packet.SetSsrc(kSsrc2);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
}
// An unsignalled ssrc for kSsrc2 should be created and the packet counter
// should increase for both ssrcs.
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 2u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 2u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 1u);
}
TEST_F(WebRtcVideoChannelTest,
RecentlyRemovedSsrcsDoNotCreateUnsignalledRecvStreams) {
const uint32_t kSsrc1 = 1;
const uint32_t kSsrc2 = 2;
// Starting point: receiving kSsrc1 and kSsrc2.
EXPECT_TRUE(channel_->AddRecvStream(StreamParams::CreateLegacy(kSsrc1)));
EXPECT_TRUE(channel_->AddRecvStream(StreamParams::CreateLegacy(kSsrc2)));
channel_->OnDemuxerCriteriaUpdatePending();
channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 2u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 0u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 0u);
// Remove kSsrc1, signal that a demuxer criteria update is pending, but not
// completed yet.
EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc1));
channel_->OnDemuxerCriteriaUpdatePending();
// We only have a receiver for kSsrc2 now.
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u);
// Emulate there being in-flight packets for kSsrc1 and kSsrc2 arriving before
// the demuxer is updated.
{
// Receive a packet for kSsrc1.
RtpPacket packet;
packet.SetSsrc(kSsrc1);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
}
{
// Receive a packet for kSsrc2.
RtpPacket packet;
packet.SetSsrc(kSsrc2);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
}
// No unsignaled ssrc for kSsrc1 should have been created, but the packet
// count for kSsrc2 should increase.
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 0u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 1u);
// Signal that the demuxer update is complete. This means we should stop
// ignorning kSsrc1.
channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
// Receive packets for kSsrc1 and kSsrc2 again.
{
// Receive a packet for kSsrc1.
RtpPacket packet;
packet.SetSsrc(kSsrc1);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
}
{
// Receive a packet for kSsrc2.
RtpPacket packet;
packet.SetSsrc(kSsrc2);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
}
// An unsignalled ssrc for kSsrc1 should be created and the packet counter
// should increase for both ssrcs.
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 2u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 1u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 2u);
}
TEST_F(WebRtcVideoChannelTest, MultiplePendingDemuxerCriteriaUpdates) {
const uint32_t kSsrc = 1;
// Starting point: receiving kSsrc.
EXPECT_TRUE(channel_->AddRecvStream(StreamParams::CreateLegacy(kSsrc)));
channel_->OnDemuxerCriteriaUpdatePending();
channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
ASSERT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u);
// Remove kSsrc...
EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc));
channel_->OnDemuxerCriteriaUpdatePending();
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 0u);
// And then add it back again, before the demuxer knows about the new
// criteria!
EXPECT_TRUE(channel_->AddRecvStream(StreamParams::CreateLegacy(kSsrc)));
channel_->OnDemuxerCriteriaUpdatePending();
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u);
// In-flight packets should arrive because the stream was recreated, even
// though demuxer criteria updates are pending...
{
RtpPacket packet;
packet.SetSsrc(kSsrc);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
}
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc), 1u);
// Signal that the demuxer knows about the first update: the removal.
channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
// This still should not prevent in-flight packets from arriving because we
// have a receive stream for it.
{
RtpPacket packet;
packet.SetSsrc(kSsrc);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
}
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc), 2u);
// Remove the kSsrc again while previous demuxer updates are still pending.
EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc));
channel_->OnDemuxerCriteriaUpdatePending();
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 0u);
// Now the packet should be dropped and not create an unsignalled receive
// stream.
{
RtpPacket packet;
packet.SetSsrc(kSsrc);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
}
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 0u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc), 2u);
// Signal that the demuxer knows about the second update: adding it back.
channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
// The packets should continue to be dropped because removal happened after
// the most recently completed demuxer update.
{
RtpPacket packet;
packet.SetSsrc(kSsrc);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
}
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 0u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc), 2u);
// Signal that the demuxer knows about the last update: the second removal.
channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
// If packets still arrive after the demuxer knows about the latest removal we
// should finally create an unsignalled receive stream.
{
RtpPacket packet;
packet.SetSsrc(kSsrc);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
}
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc), 3u);
}
TEST_F(WebRtcVideoChannelTest, UnsignalledSsrcHasACooldown) {
const uint32_t kSsrc1 = 1;
const uint32_t kSsrc2 = 2;
// Send packets for kSsrc1, creating an unsignalled receive stream.
{
// Receive a packet for kSsrc1.
RtpPacket packet;
packet.SetSsrc(kSsrc1);
channel_->OnPacketReceived(packet.Buffer(), /* packet_time_us */ -1);
}
rtc::Thread::Current()->ProcessMessages(0);
fake_clock_.AdvanceTime(
webrtc::TimeDelta::Millis(kUnsignalledReceiveStreamCooldownMs - 1));
// We now have an unsignalled receive stream for kSsrc1.
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 1u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 0u);
{
// Receive a packet for kSsrc2.
RtpPacket packet;
packet.SetSsrc(kSsrc2);
channel_->OnPacketReceived(packet.Buffer(), /* packet_time_us */ -1);
}
rtc::Thread::Current()->ProcessMessages(0);
// Not enough time has passed to replace the unsignalled receive stream, so
// the kSsrc2 should be ignored.
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 1u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 0u);
// After 500 ms, kSsrc2 should trigger a new unsignalled receive stream that
// replaces the old one.
fake_clock_.AdvanceTime(webrtc::TimeDelta::Millis(1));
{
// Receive a packet for kSsrc2.
RtpPacket packet;
packet.SetSsrc(kSsrc2);
channel_->OnPacketReceived(packet.Buffer(), /* packet_time_us */ -1);
}
rtc::Thread::Current()->ProcessMessages(0);
// The old unsignalled receive stream was destroyed and replaced, so we still
// only have one unsignalled receive stream. But tha packet counter for kSsrc2
// has now increased.
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 1u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 1u);
}
// Test BaseMinimumPlayoutDelayMs on receive streams.
TEST_F(WebRtcVideoChannelTest, BaseMinimumPlayoutDelayMs) {
// Test that set won't work for non-existing receive streams.
EXPECT_FALSE(channel_->SetBaseMinimumPlayoutDelayMs(kSsrc + 2, 200));
// Test that get won't work for non-existing receive streams.
EXPECT_FALSE(channel_->GetBaseMinimumPlayoutDelayMs(kSsrc + 2));
EXPECT_TRUE(AddRecvStream());
// Test that set works for the existing receive stream.
EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(last_ssrc_, 200));
auto* recv_stream = fake_call_->GetVideoReceiveStream(last_ssrc_);
EXPECT_TRUE(recv_stream);
EXPECT_EQ(recv_stream->base_mininum_playout_delay_ms(), 200);
EXPECT_EQ(channel_->GetBaseMinimumPlayoutDelayMs(last_ssrc_).value_or(0),
200);
}
// Test BaseMinimumPlayoutDelayMs on unsignaled receive streams.
TEST_F(WebRtcVideoChannelTest, BaseMinimumPlayoutDelayMsUnsignaledRecvStream) {
absl::optional<int> delay_ms;
const FakeVideoReceiveStream* recv_stream;
// Set default stream with SSRC 0
EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(0, 200));
EXPECT_EQ(200, channel_->GetBaseMinimumPlayoutDelayMs(0).value_or(0));
// Spawn an unsignaled stream by sending a packet, it should inherit
// default delay 200.
RtpPacket packet;
packet.SetSsrc(kIncomingUnsignalledSsrc);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
recv_stream = fake_call_->GetVideoReceiveStream(kIncomingUnsignalledSsrc);
EXPECT_EQ(recv_stream->base_mininum_playout_delay_ms(), 200);
delay_ms = channel_->GetBaseMinimumPlayoutDelayMs(kIncomingUnsignalledSsrc);
EXPECT_EQ(200, delay_ms.value_or(0));
// Check that now if we change delay for SSRC 0 it will change delay for the
// default receiving stream as well.
EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(0, 300));
EXPECT_EQ(300, channel_->GetBaseMinimumPlayoutDelayMs(0).value_or(0));
delay_ms = channel_->GetBaseMinimumPlayoutDelayMs(kIncomingUnsignalledSsrc);
EXPECT_EQ(300, delay_ms.value_or(0));
recv_stream = fake_call_->GetVideoReceiveStream(kIncomingUnsignalledSsrc);
EXPECT_EQ(recv_stream->base_mininum_playout_delay_ms(), 300);
}
void WebRtcVideoChannelTest::TestReceiveUnsignaledSsrcPacket(
uint8_t payload_type,
bool expect_created_receive_stream) {
// kRedRtxPayloadType must currently be unused.
EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kRedRtxPayloadType));
// Add a RED RTX codec.
VideoCodec red_rtx_codec =
VideoCodec::CreateRtxCodec(kRedRtxPayloadType, GetEngineCodec("red").id);
recv_parameters_.codecs.push_back(red_rtx_codec);
EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
ASSERT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
RtpPacket packet;
packet.SetPayloadType(payload_type);
packet.SetSsrc(kIncomingUnsignalledSsrc);
ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1);
if (expect_created_receive_stream) {
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size())
<< "Should have created a receive stream for payload type: "
<< payload_type;
} else {
EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size())
<< "Shouldn't have created a receive stream for payload type: "
<< payload_type;
}
}
class WebRtcVideoChannelDiscardUnknownSsrcTest : public WebRtcVideoChannelTest {
public:
WebRtcVideoChannelDiscardUnknownSsrcTest()
: WebRtcVideoChannelTest(
"WebRTC-Video-DiscardPacketsWithUnknownSsrc/Enabled/") {}
};
TEST_F(WebRtcVideoChannelDiscardUnknownSsrcTest, NoUnsignalledStreamCreated) {
TestReceiveUnsignaledSsrcPacket(GetEngineCodec("VP8").id,
false /* expect_created_receive_stream */);
}
TEST_F(WebRtcVideoChannelTest, Vp8PacketCreatesUnsignalledStream) {
TestReceiveUnsignaledSsrcPacket(GetEngineCodec("VP8").id,
true /* expect_created_receive_stream */);
}
TEST_F(WebRtcVideoChannelTest, Vp9PacketCreatesUnsignalledStream) {
TestReceiveUnsignaledSsrcPacket(GetEngineCodec("VP9").id,
true /* expect_created_receive_stream */);
}
TEST_F(WebRtcVideoChannelTest, RtxPacketDoesntCreateUnsignalledStream) {
AssignDefaultAptRtxTypes();
const cricket::VideoCodec vp8 = GetEngineCodec("VP8");
const int rtx_vp8_payload_type = default_apt_rtx_types_[vp8.id];
TestReceiveUnsignaledSsrcPacket(rtx_vp8_payload_type,
false /* expect_created_receive_stream */);
}
TEST_F(WebRtcVideoChannelTest, UlpfecPacketDoesntCreateUnsignalledStream) {
TestReceiveUnsignaledSsrcPacket(GetEngineCodec("ulpfec").id,
false /* expect_created_receive_stream */);
}
TEST_F(WebRtcVideoChannelFlexfecRecvTest,
FlexfecPacketDoesntCreateUnsignalledStream) {
TestReceiveUnsignaledSsrcPacket(GetEngineCodec("flexfec-03").id,
false /* expect_created_receive_stream */);
}
TEST_F(WebRtcVideoChannelTest, RedRtxPacketDoesntCreateUnsignalledStream) {
TestReceiveUnsignaledSsrcPacket(kRedRtxPayloadType,
false /* expect_created_receive_stream */);
}
// Test that receiving any unsignalled SSRC works even if it changes.
// The first unsignalled SSRC received will create a default receive stream.
// Any different unsignalled SSRC received will replace the default.
TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) {
// Allow receiving VP8, VP9, H264 (if enabled).
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
#if defined(WEBRTC_USE_H264)
cricket::VideoCodec H264codec(126, "H264");
parameters.codecs.push_back(H264codec);
#endif
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
// No receive streams yet.
ASSERT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
cricket::FakeVideoRenderer renderer;
channel_->SetDefaultSink(&renderer);
// Receive VP8 packet on first SSRC.
RtpPacket rtp_packet;
rtp_packet.SetPayloadType(GetEngineCodec("VP8").id);
rtp_packet.SetSsrc(kIncomingUnsignalledSsrc + 1);
ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1);
// VP8 packet should create default receive stream.
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
FakeVideoReceiveStream* recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(rtp_packet.Ssrc(), recv_stream->GetConfig().rtp.remote_ssrc);
// Verify that the receive stream sinks to a renderer.
webrtc::VideoFrame video_frame =
webrtc::VideoFrame::Builder()
.set_video_frame_buffer(CreateBlackFrameBuffer(4, 4))
.set_timestamp_rtp(100)
.set_timestamp_us(0)
.set_rotation(webrtc::kVideoRotation_0)
.build();
recv_stream->InjectFrame(video_frame);
EXPECT_EQ(1, renderer.num_rendered_frames());
// Receive VP9 packet on second SSRC.
rtp_packet.SetPayloadType(GetEngineCodec("VP9").id);
rtp_packet.SetSsrc(kIncomingUnsignalledSsrc + 2);
ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1);
// VP9 packet should replace the default receive SSRC.
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(rtp_packet.Ssrc(), recv_stream->GetConfig().rtp.remote_ssrc);
// Verify that the receive stream sinks to a renderer.
webrtc::VideoFrame video_frame2 =
webrtc::VideoFrame::Builder()
.set_video_frame_buffer(CreateBlackFrameBuffer(4, 4))
.set_timestamp_rtp(200)
.set_timestamp_us(0)
.set_rotation(webrtc::kVideoRotation_0)
.build();
recv_stream->InjectFrame(video_frame2);
EXPECT_EQ(2, renderer.num_rendered_frames());
#if defined(WEBRTC_USE_H264)
// Receive H264 packet on third SSRC.
rtp_packet.SetPayloadType(126);
rtp_packet.SetSsrc(kIncomingUnsignalledSsrc + 3);
ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1);
// H264 packet should replace the default receive SSRC.
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
recv_stream = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(rtp_packet.Ssrc(), recv_stream->GetConfig().rtp.remote_ssrc);
// Verify that the receive stream sinks to a renderer.
webrtc::VideoFrame video_frame3 =
webrtc::VideoFrame::Builder()
.set_video_frame_buffer(CreateBlackFrameBuffer(4, 4))
.set_timestamp_rtp(300)
.set_timestamp_us(0)
.set_rotation(webrtc::kVideoRotation_0)
.build();
recv_stream->InjectFrame(video_frame3);
EXPECT_EQ(3, renderer.num_rendered_frames());
#endif
}
// This test verifies that when a new default stream is created for a new
// unsignaled SSRC, the new stream does not overwrite any old stream that had
// been the default receive stream before being properly signaled.
TEST_F(WebRtcVideoChannelTest,
NewUnsignaledStreamDoesNotDestroyPreviouslyUnsignaledStream) {
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
ASSERT_TRUE(channel_->SetRecvParameters(parameters));
// No streams signaled and no packets received, so we should not have any
// stream objects created yet.
EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
// Receive packet on an unsignaled SSRC.
RtpPacket rtp_packet;
rtp_packet.SetPayloadType(GetEngineCodec("VP8").id);
rtp_packet.SetSsrc(kSsrcs3[0]);
ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1);
// Default receive stream should be created.
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
FakeVideoReceiveStream* recv_stream0 =
fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(kSsrcs3[0], recv_stream0->GetConfig().rtp.remote_ssrc);
// Signal the SSRC.
EXPECT_TRUE(
channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrcs3[0])));
ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
recv_stream0 = fake_call_->GetVideoReceiveStreams()[0];
EXPECT_EQ(kSsrcs3[0], recv_stream0->GetConfig().rtp.remote_ssrc);
// Receive packet on a different unsignaled SSRC.
rtp_packet.SetSsrc(kSsrcs3[1]);
ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1);
// New default receive stream should be created, but old stream should remain.
ASSERT_EQ(2u, fake_call_->GetVideoReceiveStreams().size());
EXPECT_EQ(recv_stream0, fake_call_->GetVideoReceiveStreams()[0]);
FakeVideoReceiveStream* recv_stream1 =
fake_call_->GetVideoReceiveStreams()[1];
EXPECT_EQ(kSsrcs3[1], recv_stream1->GetConfig().rtp.remote_ssrc);
}
TEST_F(WebRtcVideoChannelTest, CanSetMaxBitrateForExistingStream) {
AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
EXPECT_TRUE(channel_->SetSend(true));
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
int default_encoder_bitrate = GetMaxEncoderBitrate();
EXPECT_GT(default_encoder_bitrate, 1000);
// TODO(skvlad): Resolve the inconsistency between the interpretation
// of the global bitrate limit for audio and video:
// - Audio: max_bandwidth_bps = 0 - fail the operation,
// max_bandwidth_bps = -1 - remove the bandwidth limit
// - Video: max_bandwidth_bps = 0 - remove the bandwidth limit,
// max_bandwidth_bps = -1 - remove the bandwidth limit
SetAndExpectMaxBitrate(1000, 0, 1000);
SetAndExpectMaxBitrate(1000, 800, 800);
SetAndExpectMaxBitrate(600, 800, 600);
SetAndExpectMaxBitrate(0, 800, 800);
SetAndExpectMaxBitrate(0, 0, default_encoder_bitrate);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, CannotSetMaxBitrateForNonexistentStream) {
webrtc::RtpParameters nonexistent_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(0u, nonexistent_parameters.encodings.size());
nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters());
EXPECT_FALSE(
channel_->SetRtpSendParameters(last_ssrc_, nonexistent_parameters).ok());
}
TEST_F(WebRtcVideoChannelTest,
SetLowMaxBitrateOverwritesVideoStreamMinBitrate) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_FALSE(parameters.encodings[0].max_bitrate_bps.has_value());
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Note that this is testing the behavior of the FakeVideoSendStream, which
// also calls to CreateEncoderStreams to get the VideoStreams, so essentially
// we are just testing the behavior of
// EncoderStreamFactory::CreateEncoderStreams.
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps,
stream->GetVideoStreams()[0].min_bitrate_bps);
// Set a low max bitrate & check that VideoStream.min_bitrate_bps is limited
// by this amount.
parameters = channel_->GetRtpSendParameters(last_ssrc_);
int low_max_bitrate_bps = webrtc::kDefaultMinVideoBitrateBps - 1000;
parameters.encodings[0].max_bitrate_bps = low_max_bitrate_bps;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(low_max_bitrate_bps, stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(low_max_bitrate_bps, stream->GetVideoStreams()[0].max_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest,
SetHighMinBitrateOverwritesVideoStreamMaxBitrate) {
FakeVideoSendStream* stream = AddSendStream();
// Note that this is testing the behavior of the FakeVideoSendStream, which
// also calls to CreateEncoderStreams to get the VideoStreams, so essentially
// we are just testing the behavior of
// EncoderStreamFactory::CreateEncoderStreams.
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
int high_min_bitrate_bps = stream->GetVideoStreams()[0].max_bitrate_bps + 1;
// Set a high min bitrate and check that max_bitrate_bps is adjusted up.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
parameters.encodings[0].min_bitrate_bps = high_min_bitrate_bps;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(high_min_bitrate_bps, stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(high_min_bitrate_bps, stream->GetVideoStreams()[0].max_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest,
SetMinBitrateAboveMaxBitrateLimitAdjustsMinBitrateDown) {
send_parameters_.max_bandwidth_bps = 99999;
FakeVideoSendStream* stream = AddSendStream();
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps,
stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
// Set min bitrate above global max bitrate and check that min_bitrate_bps is
// adjusted down.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
parameters.encodings[0].min_bitrate_bps = 99999 + 1;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest, SetMaxFramerateOneStream) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_FALSE(parameters.encodings[0].max_framerate.has_value());
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Note that this is testing the behavior of the FakeVideoSendStream, which
// also calls to CreateEncoderStreams to get the VideoStreams, so essentially
// we are just testing the behavior of
// EncoderStreamFactory::CreateEncoderStreams.
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(kDefaultVideoMaxFramerate,
stream->GetVideoStreams()[0].max_framerate);
// Set max framerate and check that VideoStream.max_framerate is set.
const int kNewMaxFramerate = kDefaultVideoMaxFramerate - 1;
parameters = channel_->GetRtpSendParameters(last_ssrc_);
parameters.encodings[0].max_framerate = kNewMaxFramerate;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(kNewMaxFramerate, stream->GetVideoStreams()[0].max_framerate);
}
TEST_F(WebRtcVideoChannelTest, SetNumTemporalLayersForSingleStream) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_FALSE(parameters.encodings[0].num_temporal_layers.has_value());
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Note that this is testing the behavior of the FakeVideoSendStream, which
// also calls to CreateEncoderStreams to get the VideoStreams, so essentially
// we are just testing the behavior of
// EncoderStreamFactory::CreateEncoderStreams.
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_FALSE(stream->GetVideoStreams()[0].num_temporal_layers.has_value());
// Set temporal layers and check that VideoStream.num_temporal_layers is set.
parameters = channel_->GetRtpSendParameters(last_ssrc_);
parameters.encodings[0].num_temporal_layers = 2;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
ASSERT_EQ(1UL, stream->GetVideoStreams().size());
EXPECT_EQ(2UL, stream->GetVideoStreams()[0].num_temporal_layers);
}
TEST_F(WebRtcVideoChannelTest,
CannotSetRtpSendParametersWithIncorrectNumberOfEncodings) {
AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
// Two or more encodings should result in failure.
parameters.encodings.push_back(webrtc::RtpEncodingParameters());
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Zero encodings should also fail.
parameters.encodings.clear();
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
}
TEST_F(WebRtcVideoChannelTest,
CannotSetSimulcastRtpSendParametersWithIncorrectNumberOfEncodings) {
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
StreamParams sp = CreateSimStreamParams("cname", ssrcs);
AddSendStream(sp);
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
// Additional encodings should result in failure.
parameters.encodings.push_back(webrtc::RtpEncodingParameters());
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Zero encodings should also fail.
parameters.encodings.clear();
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
}
// Changing the SSRC through RtpParameters is not allowed.
TEST_F(WebRtcVideoChannelTest, CannotSetSsrcInRtpSendParameters) {
AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
parameters.encodings[0].ssrc = 0xdeadbeef;
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
}
// Tests that when RTCRtpEncodingParameters.bitrate_priority gets set to
// a value <= 0, setting the parameters returns false.
TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersInvalidBitratePriority) {
AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_EQ(webrtc::kDefaultBitratePriority,
parameters.encodings[0].bitrate_priority);
parameters.encodings[0].bitrate_priority = 0;
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
parameters.encodings[0].bitrate_priority = -2;
EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
}
// Tests when the the RTCRtpEncodingParameters.bitrate_priority gets set
// properly on the VideoChannel and propogates down to the video encoder.
TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPriorityOneStream) {
AddSendStream();
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_EQ(webrtc::kDefaultBitratePriority,
parameters.encodings[0].bitrate_priority);
// Change the value and set it on the VideoChannel.
double new_bitrate_priority = 2.0;
parameters.encodings[0].bitrate_priority = new_bitrate_priority;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the encoding parameters bitrate_priority is set for the
// VideoChannel.
parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1UL, parameters.encodings.size());
EXPECT_EQ(new_bitrate_priority, parameters.encodings[0].bitrate_priority);
// Verify that the new value propagated down to the encoder.
std::vector<FakeVideoSendStream*> video_send_streams =
fake_call_->GetVideoSendStreams();
EXPECT_EQ(1UL, video_send_streams.size());
FakeVideoSendStream* video_send_stream = video_send_streams.front();
// Check that the WebRtcVideoSendStream updated the VideoEncoderConfig
// appropriately.
EXPECT_EQ(new_bitrate_priority,
video_send_stream->GetEncoderConfig().bitrate_priority);
// Check that the vector of VideoStreams also was propagated correctly. Note
// that this is testing the behavior of the FakeVideoSendStream, which mimics
// the calls to CreateEncoderStreams to get the VideoStreams.
EXPECT_EQ(absl::optional<double>(new_bitrate_priority),
video_send_stream->GetVideoStreams()[0].bitrate_priority);
}
// Tests that the RTCRtpEncodingParameters.bitrate_priority is set for the
// VideoChannel and the value propogates to the video encoder with all simulcast
// streams.
TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPrioritySimulcastStreams) {
// Create the stream params with multiple ssrcs for simulcast.
const size_t kNumSimulcastStreams = 3;
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
StreamParams stream_params = CreateSimStreamParams("cname", ssrcs);
AddSendStream(stream_params);
uint32_t primary_ssrc = stream_params.first_ssrc();
// Using the FrameForwarder, we manually send a full size
// frame. This creates multiple VideoStreams for all simulcast layers when
// reconfiguring, and allows us to test this behavior.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame(
1920, 1080, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / 30));
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters =
channel_->GetRtpSendParameters(primary_ssrc);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_EQ(webrtc::kDefaultBitratePriority,
parameters.encodings[0].bitrate_priority);
// Change the value and set it on the VideoChannel.
double new_bitrate_priority = 2.0;
parameters.encodings[0].bitrate_priority = new_bitrate_priority;
EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok());
// Verify that the encoding parameters priority is set on the VideoChannel.
parameters = channel_->GetRtpSendParameters(primary_ssrc);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_EQ(new_bitrate_priority, parameters.encodings[0].bitrate_priority);
// Verify that the new value propagated down to the encoder.
std::vector<FakeVideoSendStream*> video_send_streams =
fake_call_->GetVideoSendStreams();
EXPECT_EQ(1UL, video_send_streams.size());
FakeVideoSendStream* video_send_stream = video_send_streams.front();
// Check that the WebRtcVideoSendStream updated the VideoEncoderConfig
// appropriately.
EXPECT_EQ(kNumSimulcastStreams,
video_send_stream->GetEncoderConfig().number_of_streams);
EXPECT_EQ(new_bitrate_priority,
video_send_stream->GetEncoderConfig().bitrate_priority);
// Check that the vector of VideoStreams also propagated correctly. The
// FakeVideoSendStream calls CreateEncoderStreams, and we are testing that
// these are created appropriately for the simulcast case.
EXPECT_EQ(kNumSimulcastStreams, video_send_stream->GetVideoStreams().size());
EXPECT_EQ(absl::optional<double>(new_bitrate_priority),
video_send_stream->GetVideoStreams()[0].bitrate_priority);
// Since we are only setting bitrate priority per-sender, the other
// VideoStreams should have a bitrate priority of 0.
EXPECT_EQ(absl::nullopt,
video_send_stream->GetVideoStreams()[1].bitrate_priority);
EXPECT_EQ(absl::nullopt,
video_send_stream->GetVideoStreams()[2].bitrate_priority);
EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
GetAndSetRtpSendParametersScaleResolutionDownByVP8) {
VideoSendParameters parameters;
parameters.codecs.push_back(VideoCodec(kVp8CodecName));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
webrtc::test::FrameForwarder frame_forwarder;
FakeFrameSource frame_source(1280, 720, rtc::kNumMicrosecsPerSec / 30);
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
// Try layers in natural order (smallest to largest).
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 4.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 1.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(320u, video_streams[0].width);
EXPECT_EQ(180u, video_streams[0].height);
EXPECT_EQ(640u, video_streams[1].width);
EXPECT_EQ(360u, video_streams[1].height);
EXPECT_EQ(1280u, video_streams[2].width);
EXPECT_EQ(720u, video_streams[2].height);
}
// Try layers in reverse natural order (largest to smallest).
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 1.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(1280u, video_streams[0].width);
EXPECT_EQ(720u, video_streams[0].height);
EXPECT_EQ(640u, video_streams[1].width);
EXPECT_EQ(360u, video_streams[1].height);
EXPECT_EQ(320u, video_streams[2].width);
EXPECT_EQ(180u, video_streams[2].height);
}
// Try layers in mixed order.
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 10.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(128u, video_streams[0].width);
EXPECT_EQ(72u, video_streams[0].height);
EXPECT_EQ(640u, video_streams[1].width);
EXPECT_EQ(360u, video_streams[1].height);
EXPECT_EQ(320u, video_streams[2].width);
EXPECT_EQ(180u, video_streams[2].height);
}
// Try with a missing scale setting, defaults to 1.0 if any other is set.
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 1.0;
rtp_parameters.encodings[1].scale_resolution_down_by.reset();
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(1280u, video_streams[0].width);
EXPECT_EQ(720u, video_streams[0].height);
EXPECT_EQ(1280u, video_streams[1].width);
EXPECT_EQ(720u, video_streams[1].height);
EXPECT_EQ(320u, video_streams[2].width);
EXPECT_EQ(180u, video_streams[2].height);
}
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
GetAndSetRtpSendParametersScaleResolutionDownByVP8WithOddResolution) {
// Ensure that the top layer has width and height divisible by 2^3,
// so that the bottom layer has width and height divisible by 2.
// TODO(bugs.webrtc.org/8785): Remove this field trial when we fully trust
// the number of simulcast layers set by the app.
webrtc::test::ScopedFieldTrials field_trial(
"WebRTC-NormalizeSimulcastResolution/Enabled-3/");
// Set up WebRtcVideoChannel for 3-layer VP8 simulcast.
VideoSendParameters parameters;
parameters.codecs.push_back(VideoCodec(kVp8CodecName));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, /*options=*/nullptr,
&frame_forwarder));
channel_->SetSend(true);
// Set `scale_resolution_down_by`'s.
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(rtp_parameters.encodings.size(), 3u);
rtp_parameters.encodings[0].scale_resolution_down_by = 1.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
const auto result =
channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
// Use a capture resolution whose width and height are not divisible by 2^3.
// (See field trial set at the top of the test.)
FakeFrameSource frame_source(2007, 1207, rtc::kNumMicrosecsPerSec / 30);
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Ensure the scaling is correct.
const auto video_streams = stream->GetVideoStreams();
ASSERT_EQ(video_streams.size(), 3u);
// Ensure that we round the capture resolution down for the top layer...
EXPECT_EQ(video_streams[0].width, 2000u);
EXPECT_EQ(video_streams[0].height, 1200u);
EXPECT_EQ(video_streams[1].width, 1000u);
EXPECT_EQ(video_streams[1].height, 600u);
// ...and that the bottom layer has a width/height divisible by 2.
EXPECT_EQ(video_streams[2].width, 500u);
EXPECT_EQ(video_streams[2].height, 300u);
// Tear down.
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
GetAndSetRtpSendParametersScaleResolutionDownByH264) {
encoder_factory_->AddSupportedVideoCodecType(kH264CodecName);
VideoSendParameters parameters;
parameters.codecs.push_back(VideoCodec(kH264CodecName));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
webrtc::test::FrameForwarder frame_forwarder;
FakeFrameSource frame_source(1280, 720, rtc::kNumMicrosecsPerSec / 30);
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
// Try layers in natural order (smallest to largest).
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 4.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 1.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(320u, video_streams[0].width);
EXPECT_EQ(180u, video_streams[0].height);
EXPECT_EQ(640u, video_streams[1].width);
EXPECT_EQ(360u, video_streams[1].height);
EXPECT_EQ(1280u, video_streams[2].width);
EXPECT_EQ(720u, video_streams[2].height);
}
// Try layers in reverse natural order (largest to smallest).
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 1.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(1280u, video_streams[0].width);
EXPECT_EQ(720u, video_streams[0].height);
EXPECT_EQ(640u, video_streams[1].width);
EXPECT_EQ(360u, video_streams[1].height);
EXPECT_EQ(320u, video_streams[2].width);
EXPECT_EQ(180u, video_streams[2].height);
}
// Try layers in mixed order.
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 10.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(128u, video_streams[0].width);
EXPECT_EQ(72u, video_streams[0].height);
EXPECT_EQ(640u, video_streams[1].width);
EXPECT_EQ(360u, video_streams[1].height);
EXPECT_EQ(320u, video_streams[2].width);
EXPECT_EQ(180u, video_streams[2].height);
}
// Try with a missing scale setting, defaults to 1.0 if any other is set.
{
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
rtp_parameters.encodings[0].scale_resolution_down_by = 1.0;
rtp_parameters.encodings[1].scale_resolution_down_by.reset();
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(3u, video_streams.size());
EXPECT_EQ(1280u, video_streams[0].width);
EXPECT_EQ(720u, video_streams[0].height);
EXPECT_EQ(1280u, video_streams[1].width);
EXPECT_EQ(720u, video_streams[1].height);
EXPECT_EQ(320u, video_streams[2].width);
EXPECT_EQ(180u, video_streams[2].height);
}
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
GetAndSetRtpSendParametersScaleResolutionDownByH264WithOddResolution) {
// Ensure that the top layer has width and height divisible by 2^3,
// so that the bottom layer has width and height divisible by 2.
// TODO(bugs.webrtc.org/8785): Remove this field trial when we fully trust
// the number of simulcast layers set by the app.
webrtc::test::ScopedFieldTrials field_trial(
"WebRTC-NormalizeSimulcastResolution/Enabled-3/");
// Set up WebRtcVideoChannel for 3-layer H264 simulcast.
encoder_factory_->AddSupportedVideoCodecType(kH264CodecName);
VideoSendParameters parameters;
parameters.codecs.push_back(VideoCodec(kH264CodecName));
ASSERT_TRUE(channel_->SetSendParameters(parameters));
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, /*options=*/nullptr,
&frame_forwarder));
channel_->SetSend(true);
// Set `scale_resolution_down_by`'s.
auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(rtp_parameters.encodings.size(), 3u);
rtp_parameters.encodings[0].scale_resolution_down_by = 1.0;
rtp_parameters.encodings[1].scale_resolution_down_by = 2.0;
rtp_parameters.encodings[2].scale_resolution_down_by = 4.0;
const auto result =
channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
ASSERT_TRUE(result.ok());
// Use a capture resolution whose width and height are not divisible by 2^3.
// (See field trial set at the top of the test.)
FakeFrameSource frame_source(2007, 1207, rtc::kNumMicrosecsPerSec / 30);
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
// Ensure the scaling is correct.
const auto video_streams = stream->GetVideoStreams();
ASSERT_EQ(video_streams.size(), 3u);
// Ensure that we round the capture resolution down for the top layer...
EXPECT_EQ(video_streams[0].width, 2000u);
EXPECT_EQ(video_streams[0].height, 1200u);
EXPECT_EQ(video_streams[1].width, 1000u);
EXPECT_EQ(video_streams[1].height, 600u);
// ...and that the bottom layer has a width/height divisible by 2.
EXPECT_EQ(video_streams[2].width, 500u);
EXPECT_EQ(video_streams[2].height, 300u);
// Tear down.
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, GetAndSetRtpSendParametersMaxFramerate) {
const size_t kNumSimulcastStreams = 3;
SetUpSimulcast(true, false);
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
for (const auto& encoding : parameters.encodings) {
EXPECT_FALSE(encoding.max_framerate);
}
// Change the value and set it on the VideoChannel.
parameters.encodings[0].max_framerate = 10;
parameters.encodings[1].max_framerate = 20;
parameters.encodings[2].max_framerate = 25;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the bitrates are set on the VideoChannel.
parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_EQ(10, parameters.encodings[0].max_framerate);
EXPECT_EQ(20, parameters.encodings[1].max_framerate);
EXPECT_EQ(25, parameters.encodings[2].max_framerate);
}
TEST_F(WebRtcVideoChannelTest,
SetRtpSendParametersNumTemporalLayersFailsForInvalidRange) {
const size_t kNumSimulcastStreams = 3;
SetUpSimulcast(true, false);
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
// Num temporal layers should be in the range [1, kMaxTemporalStreams].
parameters.encodings[0].num_temporal_layers = 0;
EXPECT_EQ(webrtc::RTCErrorType::INVALID_RANGE,
channel_->SetRtpSendParameters(last_ssrc_, parameters).type());
parameters.encodings[0].num_temporal_layers = webrtc::kMaxTemporalStreams + 1;
EXPECT_EQ(webrtc::RTCErrorType::INVALID_RANGE,
channel_->SetRtpSendParameters(last_ssrc_, parameters).type());
}
TEST_F(WebRtcVideoChannelTest, GetAndSetRtpSendParametersNumTemporalLayers) {
const size_t kNumSimulcastStreams = 3;
SetUpSimulcast(true, false);
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
for (const auto& encoding : parameters.encodings)
EXPECT_FALSE(encoding.num_temporal_layers);
// Change the value and set it on the VideoChannel.
parameters.encodings[0].num_temporal_layers = 3;
parameters.encodings[1].num_temporal_layers = 3;
parameters.encodings[2].num_temporal_layers = 3;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the number of temporal layers are set on the VideoChannel.
parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_EQ(3, parameters.encodings[0].num_temporal_layers);
EXPECT_EQ(3, parameters.encodings[1].num_temporal_layers);
EXPECT_EQ(3, parameters.encodings[2].num_temporal_layers);
}
TEST_F(WebRtcVideoChannelTest, NumTemporalLayersPropagatedToEncoder) {
const size_t kNumSimulcastStreams = 3;
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Get and set the rtp encoding parameters.
// Change the value and set it on the VideoChannel.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
parameters.encodings[0].num_temporal_layers = 3;
parameters.encodings[1].num_temporal_layers = 2;
parameters.encodings[2].num_temporal_layers = 1;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the new value is propagated down to the encoder.
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
EXPECT_EQ(2, stream->num_encoder_reconfigurations());
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams);
EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size());
EXPECT_EQ(3UL, encoder_config.simulcast_layers[0].num_temporal_layers);
EXPECT_EQ(2UL, encoder_config.simulcast_layers[1].num_temporal_layers);
EXPECT_EQ(1UL, encoder_config.simulcast_layers[2].num_temporal_layers);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately for the simulcast case.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
EXPECT_EQ(3UL, stream->GetVideoStreams()[0].num_temporal_layers);
EXPECT_EQ(2UL, stream->GetVideoStreams()[1].num_temporal_layers);
EXPECT_EQ(1UL, stream->GetVideoStreams()[2].num_temporal_layers);
// No parameter changed, encoder should not be reconfigured.
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_EQ(2, stream->num_encoder_reconfigurations());
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
DefaultValuePropagatedToEncoderForUnsetNumTemporalLayers) {
const size_t kDefaultNumTemporalLayers = 3;
const size_t kNumSimulcastStreams = 3;
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Change rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
parameters.encodings[0].num_temporal_layers = 2;
parameters.encodings[2].num_temporal_layers = 1;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that no value is propagated down to the encoder.
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams);
EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size());
EXPECT_EQ(2UL, encoder_config.simulcast_layers[0].num_temporal_layers);
EXPECT_FALSE(encoder_config.simulcast_layers[1].num_temporal_layers);
EXPECT_EQ(1UL, encoder_config.simulcast_layers[2].num_temporal_layers);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately for the simulcast case.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
EXPECT_EQ(2UL, stream->GetVideoStreams()[0].num_temporal_layers);
EXPECT_EQ(kDefaultNumTemporalLayers,
stream->GetVideoStreams()[1].num_temporal_layers);
EXPECT_EQ(1UL, stream->GetVideoStreams()[2].num_temporal_layers);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
DefaultValuePropagatedToEncoderForUnsetFramerate) {
const size_t kNumSimulcastStreams = 3;
const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Get and set the rtp encoding parameters.
// Change the value and set it on the VideoChannel.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
parameters.encodings[0].max_framerate = 15;
parameters.encodings[2].max_framerate = 20;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the new value propagated down to the encoder.
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams);
EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size());
EXPECT_EQ(15, encoder_config.simulcast_layers[0].max_framerate);
EXPECT_EQ(-1, encoder_config.simulcast_layers[1].max_framerate);
EXPECT_EQ(20, encoder_config.simulcast_layers[2].max_framerate);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately for the simulcast case.
// The maximum `max_framerate` is used, kDefaultVideoMaxFramerate: 60.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
EXPECT_EQ(15, stream->GetVideoStreams()[0].max_framerate);
EXPECT_EQ(kDefaultVideoMaxFramerate,
stream->GetVideoStreams()[1].max_framerate);
EXPECT_EQ(20, stream->GetVideoStreams()[2].max_framerate);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, GetAndSetRtpSendParametersMinAndMaxBitrate) {
const size_t kNumSimulcastStreams = 3;
SetUpSimulcast(true, false);
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
for (const auto& encoding : parameters.encodings) {
EXPECT_FALSE(encoding.min_bitrate_bps);
EXPECT_FALSE(encoding.max_bitrate_bps);
}
// Change the value and set it on the VideoChannel.
parameters.encodings[0].min_bitrate_bps = 100000;
parameters.encodings[0].max_bitrate_bps = 200000;
parameters.encodings[1].min_bitrate_bps = 300000;
parameters.encodings[1].max_bitrate_bps = 400000;
parameters.encodings[2].min_bitrate_bps = 500000;
parameters.encodings[2].max_bitrate_bps = 600000;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the bitrates are set on the VideoChannel.
parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_EQ(100000, parameters.encodings[0].min_bitrate_bps);
EXPECT_EQ(200000, parameters.encodings[0].max_bitrate_bps);
EXPECT_EQ(300000, parameters.encodings[1].min_bitrate_bps);
EXPECT_EQ(400000, parameters.encodings[1].max_bitrate_bps);
EXPECT_EQ(500000, parameters.encodings[2].min_bitrate_bps);
EXPECT_EQ(600000, parameters.encodings[2].max_bitrate_bps);
}
TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersFailsWithIncorrectBitrate) {
const size_t kNumSimulcastStreams = 3;
SetUpSimulcast(true, false);
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
// Max bitrate lower than min bitrate should fail.
parameters.encodings[2].min_bitrate_bps = 100000;
parameters.encodings[2].max_bitrate_bps = 100000 - 1;
EXPECT_EQ(webrtc::RTCErrorType::INVALID_RANGE,
channel_->SetRtpSendParameters(last_ssrc_, parameters).type());
}
// Test that min and max bitrate values set via RtpParameters are correctly
// propagated to the underlying encoder, and that the target is set to 3/4 of
// the maximum (3/4 was chosen because it's similar to the simulcast defaults
// that are used if no min/max are specified).
TEST_F(WebRtcVideoChannelTest, MinAndMaxSimulcastBitratePropagatedToEncoder) {
const size_t kNumSimulcastStreams = 3;
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Get and set the rtp encoding parameters.
// Change the value and set it on the VideoChannel.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
parameters.encodings[0].min_bitrate_bps = 100000;
parameters.encodings[0].max_bitrate_bps = 200000;
parameters.encodings[1].min_bitrate_bps = 300000;
parameters.encodings[1].max_bitrate_bps = 400000;
parameters.encodings[2].min_bitrate_bps = 500000;
parameters.encodings[2].max_bitrate_bps = 600000;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the new value propagated down to the encoder.
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
EXPECT_EQ(2, stream->num_encoder_reconfigurations());
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams);
EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size());
EXPECT_EQ(100000, encoder_config.simulcast_layers[0].min_bitrate_bps);
EXPECT_EQ(200000, encoder_config.simulcast_layers[0].max_bitrate_bps);
EXPECT_EQ(300000, encoder_config.simulcast_layers[1].min_bitrate_bps);
EXPECT_EQ(400000, encoder_config.simulcast_layers[1].max_bitrate_bps);
EXPECT_EQ(500000, encoder_config.simulcast_layers[2].min_bitrate_bps);
EXPECT_EQ(600000, encoder_config.simulcast_layers[2].max_bitrate_bps);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately for the simulcast case.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
// Target bitrate: 200000 * 3 / 4 = 150000.
EXPECT_EQ(100000, stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(150000, stream->GetVideoStreams()[0].target_bitrate_bps);
EXPECT_EQ(200000, stream->GetVideoStreams()[0].max_bitrate_bps);
// Target bitrate: 400000 * 3 / 4 = 300000.
EXPECT_EQ(300000, stream->GetVideoStreams()[1].min_bitrate_bps);
EXPECT_EQ(300000, stream->GetVideoStreams()[1].target_bitrate_bps);
EXPECT_EQ(400000, stream->GetVideoStreams()[1].max_bitrate_bps);
// Target bitrate: 600000 * 3 / 4 = 450000, less than min -> max.
EXPECT_EQ(500000, stream->GetVideoStreams()[2].min_bitrate_bps);
EXPECT_EQ(600000, stream->GetVideoStreams()[2].target_bitrate_bps);
EXPECT_EQ(600000, stream->GetVideoStreams()[2].max_bitrate_bps);
// No parameter changed, encoder should not be reconfigured.
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_EQ(2, stream->num_encoder_reconfigurations());
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
// Test to only specify the min or max bitrate value for a layer via
// RtpParameters. The unspecified min/max and target value should be set to the
// simulcast default that is used if no min/max are specified.
TEST_F(WebRtcVideoChannelTest, MinOrMaxSimulcastBitratePropagatedToEncoder) {
const size_t kNumSimulcastStreams = 3;
const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
// Change the value and set it on the VideoChannel.
// Layer 0: only configure min bitrate.
const int kMinBpsLayer0 = kDefault[0].min_bitrate_bps + 1;
parameters.encodings[0].min_bitrate_bps = kMinBpsLayer0;
// Layer 1: only configure max bitrate.
const int kMaxBpsLayer1 = kDefault[1].max_bitrate_bps - 1;
parameters.encodings[1].max_bitrate_bps = kMaxBpsLayer1;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the new value propagated down to the encoder.
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams);
EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size());
EXPECT_EQ(kMinBpsLayer0, encoder_config.simulcast_layers[0].min_bitrate_bps);
EXPECT_EQ(-1, encoder_config.simulcast_layers[0].max_bitrate_bps);
EXPECT_EQ(-1, encoder_config.simulcast_layers[1].min_bitrate_bps);
EXPECT_EQ(kMaxBpsLayer1, encoder_config.simulcast_layers[1].max_bitrate_bps);
EXPECT_EQ(-1, encoder_config.simulcast_layers[2].min_bitrate_bps);
EXPECT_EQ(-1, encoder_config.simulcast_layers[2].max_bitrate_bps);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately for the simulcast case.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
// Layer 0: min configured bitrate should overwrite min default.
EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(kDefault[0].target_bitrate_bps,
stream->GetVideoStreams()[0].target_bitrate_bps);
EXPECT_EQ(kDefault[0].max_bitrate_bps,
stream->GetVideoStreams()[0].max_bitrate_bps);
// Layer 1: max configured bitrate should overwrite max default.
// And target bitrate should be 3/4 * max bitrate or default target
// which is larger.
EXPECT_EQ(kDefault[1].min_bitrate_bps,
stream->GetVideoStreams()[1].min_bitrate_bps);
const int kTargetBpsLayer1 =
std::max(kDefault[1].target_bitrate_bps, kMaxBpsLayer1 * 3 / 4);
EXPECT_EQ(kTargetBpsLayer1, stream->GetVideoStreams()[1].target_bitrate_bps);
EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].max_bitrate_bps);
// Layer 2: min and max bitrate not configured, default expected.
EXPECT_EQ(kDefault[2].min_bitrate_bps,
stream->GetVideoStreams()[2].min_bitrate_bps);
EXPECT_EQ(kDefault[2].target_bitrate_bps,
stream->GetVideoStreams()[2].target_bitrate_bps);
EXPECT_EQ(kDefault[2].max_bitrate_bps,
stream->GetVideoStreams()[2].max_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
// Test that specifying the min (or max) bitrate value for a layer via
// RtpParameters above (or below) the simulcast default max (or min) adjusts the
// unspecified values accordingly.
TEST_F(WebRtcVideoChannelTest, SetMinAndMaxSimulcastBitrateAboveBelowDefault) {
const size_t kNumSimulcastStreams = 3;
const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Get and set the rtp encoding parameters.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
// Change the value and set it on the VideoChannel.
// For layer 0, set the min bitrate above the default max.
const int kMinBpsLayer0 = kDefault[0].max_bitrate_bps + 1;
parameters.encodings[0].min_bitrate_bps = kMinBpsLayer0;
// For layer 1, set the max bitrate below the default min.
const int kMaxBpsLayer1 = kDefault[1].min_bitrate_bps - 1;
parameters.encodings[1].max_bitrate_bps = kMaxBpsLayer1;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Verify that the new value propagated down to the encoder.
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately for the simulcast case.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
// Layer 0: Min bitrate above default max (target/max should be adjusted).
EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].target_bitrate_bps);
EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].max_bitrate_bps);
// Layer 1: Max bitrate below default min (min/target should be adjusted).
EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].min_bitrate_bps);
EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].target_bitrate_bps);
EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].max_bitrate_bps);
// Layer 2: min and max bitrate not configured, default expected.
EXPECT_EQ(kDefault[2].min_bitrate_bps,
stream->GetVideoStreams()[2].min_bitrate_bps);
EXPECT_EQ(kDefault[2].target_bitrate_bps,
stream->GetVideoStreams()[2].target_bitrate_bps);
EXPECT_EQ(kDefault[2].max_bitrate_bps,
stream->GetVideoStreams()[2].max_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest, BandwidthAboveTotalMaxBitrateGivenToMaxLayer) {
const size_t kNumSimulcastStreams = 3;
const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Set max bitrate for all but the highest layer.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
parameters.encodings[0].max_bitrate_bps = kDefault[0].max_bitrate_bps;
parameters.encodings[1].max_bitrate_bps = kDefault[1].max_bitrate_bps;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Set max bandwidth equal to total max bitrate.
send_parameters_.max_bandwidth_bps =
GetTotalMaxBitrate(stream->GetVideoStreams()).bps();
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
// No bitrate above the total max to give to the highest layer.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
EXPECT_EQ(kDefault[2].max_bitrate_bps,
stream->GetVideoStreams()[2].max_bitrate_bps);
// Set max bandwidth above the total max bitrate.
send_parameters_.max_bandwidth_bps =
GetTotalMaxBitrate(stream->GetVideoStreams()).bps() + 1;
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
// The highest layer has no max bitrate set -> the bitrate above the total
// max should be given to the highest layer.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
EXPECT_EQ(send_parameters_.max_bandwidth_bps,
GetTotalMaxBitrate(stream->GetVideoStreams()).bps());
EXPECT_EQ(kDefault[2].max_bitrate_bps + 1,
stream->GetVideoStreams()[2].max_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
BandwidthAboveTotalMaxBitrateNotGivenToMaxLayerIfMaxBitrateSet) {
const size_t kNumSimulcastStreams = 3;
const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
EXPECT_EQ(kNumSimulcastStreams, kDefault.size());
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
// Send a full size frame so all simulcast layers are used when reconfiguring.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame());
// Set max bitrate for the highest layer.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
parameters.encodings[2].max_bitrate_bps = kDefault[2].max_bitrate_bps;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Set max bandwidth above the total max bitrate.
send_parameters_.max_bandwidth_bps =
GetTotalMaxBitrate(stream->GetVideoStreams()).bps() + 1;
ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
// The highest layer has the max bitrate set -> the bitrate above the total
// max should not be given to the highest layer.
EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
EXPECT_EQ(*parameters.encodings[2].max_bitrate_bps,
stream->GetVideoStreams()[2].max_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
// Test that min and max bitrate values set via RtpParameters are correctly
// propagated to the underlying encoder for a single stream.
TEST_F(WebRtcVideoChannelTest, MinAndMaxBitratePropagatedToEncoder) {
FakeVideoSendStream* stream = AddSendStream();
EXPECT_TRUE(channel_->SetSend(true));
EXPECT_TRUE(stream->IsSending());
// Set min and max bitrate.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(1u, parameters.encodings.size());
parameters.encodings[0].min_bitrate_bps = 80000;
parameters.encodings[0].max_bitrate_bps = 150000;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(1u, encoder_config.number_of_streams);
EXPECT_EQ(1u, encoder_config.simulcast_layers.size());
EXPECT_EQ(80000, encoder_config.simulcast_layers[0].min_bitrate_bps);
EXPECT_EQ(150000, encoder_config.simulcast_layers[0].max_bitrate_bps);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately.
EXPECT_EQ(1u, stream->GetVideoStreams().size());
EXPECT_EQ(80000, stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(150000, stream->GetVideoStreams()[0].target_bitrate_bps);
EXPECT_EQ(150000, stream->GetVideoStreams()[0].max_bitrate_bps);
}
// Test the default min and max bitrate value are correctly propagated to the
// underlying encoder for a single stream (when the values are not set via
// RtpParameters).
TEST_F(WebRtcVideoChannelTest, DefaultMinAndMaxBitratePropagatedToEncoder) {
FakeVideoSendStream* stream = AddSendStream();
EXPECT_TRUE(channel_->SetSend(true));
EXPECT_TRUE(stream->IsSending());
// Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
EXPECT_EQ(1u, encoder_config.number_of_streams);
EXPECT_EQ(1u, encoder_config.simulcast_layers.size());
EXPECT_EQ(-1, encoder_config.simulcast_layers[0].min_bitrate_bps);
EXPECT_EQ(-1, encoder_config.simulcast_layers[0].max_bitrate_bps);
// FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
// VideoStreams are created appropriately.
EXPECT_EQ(1u, stream->GetVideoStreams().size());
EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps,
stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_GT(stream->GetVideoStreams()[0].max_bitrate_bps,
stream->GetVideoStreams()[0].min_bitrate_bps);
EXPECT_EQ(stream->GetVideoStreams()[0].max_bitrate_bps,
stream->GetVideoStreams()[0].target_bitrate_bps);
}
// Test that a stream will not be sending if its encoding is made inactive
// through SetRtpSendParameters.
TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersOneEncodingActive) {
FakeVideoSendStream* stream = AddSendStream();
EXPECT_TRUE(channel_->SetSend(true));
EXPECT_TRUE(stream->IsSending());
// Get current parameters and change "active" to false.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(1u, parameters.encodings.size());
ASSERT_TRUE(parameters.encodings[0].active);
parameters.encodings[0].active = false;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_FALSE(stream->IsSending());
// Now change it back to active and verify we resume sending.
parameters.encodings[0].active = true;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_TRUE(stream->IsSending());
}
// Tests that when active is updated for any simulcast layer then the send
// stream's sending state will be updated and it will be reconfigured with the
// new appropriate active simulcast streams.
TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersMultipleEncodingsActive) {
// Create the stream params with multiple ssrcs for simulcast.
const size_t kNumSimulcastStreams = 3;
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
StreamParams stream_params = CreateSimStreamParams("cname", ssrcs);
FakeVideoSendStream* fake_video_send_stream = AddSendStream(stream_params);
uint32_t primary_ssrc = stream_params.first_ssrc();
// Using the FrameForwarder, we manually send a full size
// frame. This allows us to test that ReconfigureEncoder is called
// appropriately.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame(
1920, 1080, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / 30));
// Check that all encodings are initially active.
webrtc::RtpParameters parameters =
channel_->GetRtpSendParameters(primary_ssrc);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_TRUE(parameters.encodings[0].active);
EXPECT_TRUE(parameters.encodings[1].active);
EXPECT_TRUE(parameters.encodings[2].active);
EXPECT_TRUE(fake_video_send_stream->IsSending());
// Only turn on only the middle stream.
parameters.encodings[0].active = false;
parameters.encodings[1].active = true;
parameters.encodings[2].active = false;
EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok());
// Verify that the active fields are set on the VideoChannel.
parameters = channel_->GetRtpSendParameters(primary_ssrc);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_FALSE(parameters.encodings[0].active);
EXPECT_TRUE(parameters.encodings[1].active);
EXPECT_FALSE(parameters.encodings[2].active);
// Check that the VideoSendStream is updated appropriately. This means its
// send state was updated and it was reconfigured.
EXPECT_TRUE(fake_video_send_stream->IsSending());
std::vector<webrtc::VideoStream> simulcast_streams =
fake_video_send_stream->GetVideoStreams();
EXPECT_EQ(kNumSimulcastStreams, simulcast_streams.size());
EXPECT_FALSE(simulcast_streams[0].active);
EXPECT_TRUE(simulcast_streams[1].active);
EXPECT_FALSE(simulcast_streams[2].active);
// Turn off all streams.
parameters.encodings[0].active = false;
parameters.encodings[1].active = false;
parameters.encodings[2].active = false;
EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok());
// Verify that the active fields are set on the VideoChannel.
parameters = channel_->GetRtpSendParameters(primary_ssrc);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_FALSE(parameters.encodings[0].active);
EXPECT_FALSE(parameters.encodings[1].active);
EXPECT_FALSE(parameters.encodings[2].active);
// Check that the VideoSendStream is off.
EXPECT_FALSE(fake_video_send_stream->IsSending());
simulcast_streams = fake_video_send_stream->GetVideoStreams();
EXPECT_EQ(kNumSimulcastStreams, simulcast_streams.size());
EXPECT_FALSE(simulcast_streams[0].active);
EXPECT_FALSE(simulcast_streams[1].active);
EXPECT_FALSE(simulcast_streams[2].active);
EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, nullptr, nullptr));
}
// Tests that when some streams are disactivated then the lowest
// stream min_bitrate would be reused for the first active stream.
TEST_F(WebRtcVideoChannelTest,
SetRtpSendParametersSetsMinBitrateForFirstActiveStream) {
// Create the stream params with multiple ssrcs for simulcast.
const size_t kNumSimulcastStreams = 3;
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
StreamParams stream_params = CreateSimStreamParams("cname", ssrcs);
FakeVideoSendStream* fake_video_send_stream = AddSendStream(stream_params);
uint32_t primary_ssrc = stream_params.first_ssrc();
// Using the FrameForwarder, we manually send a full size
// frame. This allows us to test that ReconfigureEncoder is called
// appropriately.
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, &options, &frame_forwarder));
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame(
1920, 1080, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / 30));
// Check that all encodings are initially active.
webrtc::RtpParameters parameters =
channel_->GetRtpSendParameters(primary_ssrc);
EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
EXPECT_TRUE(parameters.encodings[0].active);
EXPECT_TRUE(parameters.encodings[1].active);
EXPECT_TRUE(parameters.encodings[2].active);
EXPECT_TRUE(fake_video_send_stream->IsSending());
// Only turn on the highest stream.
parameters.encodings[0].active = false;
parameters.encodings[1].active = false;
parameters.encodings[2].active = true;
EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok());
// Check that the VideoSendStream is updated appropriately. This means its
// send state was updated and it was reconfigured.
EXPECT_TRUE(fake_video_send_stream->IsSending());
std::vector<webrtc::VideoStream> simulcast_streams =
fake_video_send_stream->GetVideoStreams();
EXPECT_EQ(kNumSimulcastStreams, simulcast_streams.size());
EXPECT_FALSE(simulcast_streams[0].active);
EXPECT_FALSE(simulcast_streams[1].active);
EXPECT_TRUE(simulcast_streams[2].active);
EXPECT_EQ(simulcast_streams[2].min_bitrate_bps,
simulcast_streams[0].min_bitrate_bps);
EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, nullptr, nullptr));
}
// Test that if a stream is reconfigured (due to a codec change or other
// change) while its encoding is still inactive, it doesn't start sending.
TEST_F(WebRtcVideoChannelTest,
InactiveStreamDoesntStartSendingWhenReconfigured) {
// Set an initial codec list, which will be modified later.
cricket::VideoSendParameters parameters1;
parameters1.codecs.push_back(GetEngineCodec("VP8"));
parameters1.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetSendParameters(parameters1));
FakeVideoSendStream* stream = AddSendStream();
EXPECT_TRUE(channel_->SetSend(true));
EXPECT_TRUE(stream->IsSending());
// Get current parameters and change "active" to false.
webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(1u, parameters.encodings.size());
ASSERT_TRUE(parameters.encodings[0].active);
parameters.encodings[0].active = false;
EXPECT_EQ(1u, GetFakeSendStreams().size());
EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
EXPECT_FALSE(stream->IsSending());
// Reorder the codec list, causing the stream to be reconfigured.
cricket::VideoSendParameters parameters2;
parameters2.codecs.push_back(GetEngineCodec("VP9"));
parameters2.codecs.push_back(GetEngineCodec("VP8"));
EXPECT_TRUE(channel_->SetSendParameters(parameters2));
auto new_streams = GetFakeSendStreams();
// Assert that a new underlying stream was created due to the codec change.
// Otherwise, this test isn't testing what it set out to test.
EXPECT_EQ(1u, GetFakeSendStreams().size());
EXPECT_EQ(2, fake_call_->GetNumCreatedSendStreams());
// Verify that we still are not sending anything, due to the inactive
// encoding.
EXPECT_FALSE(new_streams[0]->IsSending());
}
// Test that GetRtpSendParameters returns the currently configured codecs.
TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersCodecs) {
AddSendStream();
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(2u, rtp_parameters.codecs.size());
EXPECT_EQ(GetEngineCodec("VP8").ToCodecParameters(),
rtp_parameters.codecs[0]);
EXPECT_EQ(GetEngineCodec("VP9").ToCodecParameters(),
rtp_parameters.codecs[1]);
}
// Test that GetRtpSendParameters returns the currently configured RTCP CNAME.
TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersRtcpCname) {
StreamParams params = StreamParams::CreateLegacy(kSsrc);
params.cname = "rtcpcname";
AddSendStream(params);
webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrc);
EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str());
}
// Test that RtpParameters for send stream has one encoding and it has
// the correct SSRC.
TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersSsrc) {
AddSendStream();
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_EQ(last_ssrc_, rtp_parameters.encodings[0].ssrc);
}
TEST_F(WebRtcVideoChannelTest, DetectRtpSendParameterHeaderExtensionsChange) {
AddSendStream();
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
rtp_parameters.header_extensions.emplace_back();
EXPECT_NE(0u, rtp_parameters.header_extensions.size());
webrtc::RTCError result =
channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
EXPECT_EQ(webrtc::RTCErrorType::INVALID_MODIFICATION, result.type());
}
TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersDegradationPreference) {
AddSendStream();
webrtc::test::FrameForwarder frame_forwarder;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder));
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_FALSE(rtp_parameters.degradation_preference.has_value());
rtp_parameters.degradation_preference =
webrtc::DegradationPreference::MAINTAIN_FRAMERATE;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters).ok());
webrtc::RtpParameters updated_rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
EXPECT_EQ(updated_rtp_parameters.degradation_preference,
webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
// Remove the source since it will be destroyed before the channel
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
// Test that if we set/get parameters multiple times, we get the same results.
TEST_F(WebRtcVideoChannelTest, SetAndGetRtpSendParameters) {
AddSendStream();
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetSendParameters(parameters));
webrtc::RtpParameters initial_params =
channel_->GetRtpSendParameters(last_ssrc_);
// We should be able to set the params we just got.
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, initial_params).ok());
// ... And this shouldn't change the params returned by GetRtpSendParameters.
EXPECT_EQ(initial_params, channel_->GetRtpSendParameters(last_ssrc_));
}
// Test that GetRtpReceiveParameters returns the currently configured codecs.
TEST_F(WebRtcVideoChannelTest, GetRtpReceiveParametersCodecs) {
AddRecvStream();
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpReceiveParameters(last_ssrc_);
ASSERT_EQ(2u, rtp_parameters.codecs.size());
EXPECT_EQ(GetEngineCodec("VP8").ToCodecParameters(),
rtp_parameters.codecs[0]);
EXPECT_EQ(GetEngineCodec("VP9").ToCodecParameters(),
rtp_parameters.codecs[1]);
}
#if defined(WEBRTC_USE_H264)
TEST_F(WebRtcVideoChannelTest, GetRtpReceiveFmtpSprop) {
#else
TEST_F(WebRtcVideoChannelTest, DISABLED_GetRtpReceiveFmtpSprop) {
#endif
cricket::VideoRecvParameters parameters;
cricket::VideoCodec kH264sprop1(101, "H264");
kH264sprop1.SetParam(kH264FmtpSpropParameterSets, "uvw");
parameters.codecs.push_back(kH264sprop1);
cricket::VideoCodec kH264sprop2(102, "H264");
kH264sprop2.SetParam(kH264FmtpSpropParameterSets, "xyz");
parameters.codecs.push_back(kH264sprop2);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
FakeVideoReceiveStream* recv_stream = AddRecvStream();
const webrtc::VideoReceiveStream::Config& cfg = recv_stream->GetConfig();
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpReceiveParameters(last_ssrc_);
ASSERT_EQ(2u, rtp_parameters.codecs.size());
EXPECT_EQ(kH264sprop1.ToCodecParameters(), rtp_parameters.codecs[0]);
ASSERT_EQ(2u, cfg.decoders.size());
EXPECT_EQ(101, cfg.decoders[0].payload_type);
EXPECT_EQ("H264", cfg.decoders[0].video_format.name);
const auto it0 =
cfg.decoders[0].video_format.parameters.find(kH264FmtpSpropParameterSets);
ASSERT_TRUE(it0 != cfg.decoders[0].video_format.parameters.end());
EXPECT_EQ("uvw", it0->second);
EXPECT_EQ(102, cfg.decoders[1].payload_type);
EXPECT_EQ("H264", cfg.decoders[1].video_format.name);
const auto it1 =
cfg.decoders[1].video_format.parameters.find(kH264FmtpSpropParameterSets);
ASSERT_TRUE(it1 != cfg.decoders[1].video_format.parameters.end());
EXPECT_EQ("xyz", it1->second);
}
// Test that RtpParameters for receive stream has one encoding and it has
// the correct SSRC.
TEST_F(WebRtcVideoChannelTest, GetRtpReceiveParametersSsrc) {
AddRecvStream();
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpReceiveParameters(last_ssrc_);
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_EQ(last_ssrc_, rtp_parameters.encodings[0].ssrc);
}
// Test that if we set/get parameters multiple times, we get the same results.
TEST_F(WebRtcVideoChannelTest, SetAndGetRtpReceiveParameters) {
AddRecvStream();
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
webrtc::RtpParameters initial_params =
channel_->GetRtpReceiveParameters(last_ssrc_);
// ... And this shouldn't change the params returned by
// GetRtpReceiveParameters.
EXPECT_EQ(initial_params, channel_->GetRtpReceiveParameters(last_ssrc_));
}
// Test that GetDefaultRtpReceiveParameters returns parameters correctly when
// SSRCs aren't signaled. It should always return an empty
// "RtpEncodingParameters", even after a packet is received and the unsignaled
// SSRC is known.
TEST_F(WebRtcVideoChannelTest,
GetDefaultRtpReceiveParametersWithUnsignaledSsrc) {
// Call necessary methods to configure receiving a default stream as
// soon as it arrives.
cricket::VideoRecvParameters parameters;
parameters.codecs.push_back(GetEngineCodec("VP8"));
parameters.codecs.push_back(GetEngineCodec("VP9"));
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
// Call GetRtpReceiveParameters before configured to receive an unsignaled
// stream. Should return nothing.
EXPECT_EQ(webrtc::RtpParameters(),
channel_->GetDefaultRtpReceiveParameters());
// Set a sink for an unsignaled stream.
cricket::FakeVideoRenderer renderer;
channel_->SetDefaultSink(&renderer);
// Call GetDefaultRtpReceiveParameters before the SSRC is known.
webrtc::RtpParameters rtp_parameters =
channel_->GetDefaultRtpReceiveParameters();
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
// Receive VP8 packet.
RtpPacket rtp_packet;
rtp_packet.SetPayloadType(GetEngineCodec("VP8").id);
rtp_packet.SetSsrc(kIncomingUnsignalledSsrc);
ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1);
// The `ssrc` member should still be unset.
rtp_parameters = channel_->GetDefaultRtpReceiveParameters();
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
}
// Test that if a default stream is created for a non-primary stream (for
// example, RTX before we know it's RTX), we are still able to explicitly add
// the stream later.
TEST_F(WebRtcVideoChannelTest,
AddReceiveStreamAfterReceivingNonPrimaryUnsignaledSsrc) {
// Receive VP8 RTX packet.
RtpPacket rtp_packet;
const cricket::VideoCodec vp8 = GetEngineCodec("VP8");
rtp_packet.SetPayloadType(default_apt_rtx_types_[vp8.id]);
rtp_packet.SetSsrc(2);
ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1);
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
cricket::StreamParams params = cricket::StreamParams::CreateLegacy(1);
params.AddFidSsrc(1, 2);
EXPECT_TRUE(channel_->AddRecvStream(params));
}
void WebRtcVideoChannelTest::TestReceiverLocalSsrcConfiguration(
bool receiver_first) {
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
const uint32_t kSenderSsrc = 0xC0FFEE;
const uint32_t kSecondSenderSsrc = 0xBADCAFE;
const uint32_t kReceiverSsrc = 0x4711;
const uint32_t kExpectedDefaultReceiverSsrc = 1;
if (receiver_first) {
AddRecvStream(StreamParams::CreateLegacy(kReceiverSsrc));
std::vector<FakeVideoReceiveStream*> receive_streams =
fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1u, receive_streams.size());
// Default local SSRC when we have no sender.
EXPECT_EQ(kExpectedDefaultReceiverSsrc,
receive_streams[0]->GetConfig().rtp.local_ssrc);
}
AddSendStream(StreamParams::CreateLegacy(kSenderSsrc));
if (!receiver_first)
AddRecvStream(StreamParams::CreateLegacy(kReceiverSsrc));
std::vector<FakeVideoReceiveStream*> receive_streams =
fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1u, receive_streams.size());
EXPECT_EQ(kSenderSsrc, receive_streams[0]->GetConfig().rtp.local_ssrc);
// Removing first sender should fall back to another (in this case the second)
// local send stream's SSRC.
AddSendStream(StreamParams::CreateLegacy(kSecondSenderSsrc));
ASSERT_TRUE(channel_->RemoveSendStream(kSenderSsrc));
receive_streams = fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1u, receive_streams.size());
EXPECT_EQ(kSecondSenderSsrc, receive_streams[0]->GetConfig().rtp.local_ssrc);
// Removing the last sender should fall back to default local SSRC.
ASSERT_TRUE(channel_->RemoveSendStream(kSecondSenderSsrc));
receive_streams = fake_call_->GetVideoReceiveStreams();
ASSERT_EQ(1u, receive_streams.size());
EXPECT_EQ(kExpectedDefaultReceiverSsrc,
receive_streams[0]->GetConfig().rtp.local_ssrc);
}
TEST_F(WebRtcVideoChannelTest, ConfiguresLocalSsrc) {
TestReceiverLocalSsrcConfiguration(false);
}
TEST_F(WebRtcVideoChannelTest, ConfiguresLocalSsrcOnExistingReceivers) {
TestReceiverLocalSsrcConfiguration(true);
}
TEST_F(WebRtcVideoChannelTest, Simulcast_QualityScalingNotAllowed) {
FakeVideoSendStream* stream = SetUpSimulcast(true, true);
EXPECT_FALSE(stream->GetEncoderConfig().is_quality_scaling_allowed);
}
TEST_F(WebRtcVideoChannelTest, Singlecast_QualityScalingAllowed) {
FakeVideoSendStream* stream = SetUpSimulcast(false, true);
EXPECT_TRUE(stream->GetEncoderConfig().is_quality_scaling_allowed);
}
TEST_F(WebRtcVideoChannelTest,
SinglecastScreenSharing_QualityScalingNotAllowed) {
SetUpSimulcast(false, true);
webrtc::test::FrameForwarder frame_forwarder;
VideoOptions options;
options.is_screencast = true;
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder));
// Fetch the latest stream since SetVideoSend() may recreate it if the
// screen content setting is changed.
FakeVideoSendStream* stream = fake_call_->GetVideoSendStreams().front();
EXPECT_FALSE(stream->GetEncoderConfig().is_quality_scaling_allowed);
EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
}
TEST_F(WebRtcVideoChannelTest,
SimulcastSingleActiveStream_QualityScalingAllowed) {
FakeVideoSendStream* stream = SetUpSimulcast(true, false);
webrtc::RtpParameters rtp_parameters =
channel_->GetRtpSendParameters(last_ssrc_);
ASSERT_EQ(3u, rtp_parameters.encodings.size());
ASSERT_TRUE(rtp_parameters.encodings[0].active);
ASSERT_TRUE(rtp_parameters.encodings[1].active);
ASSERT_TRUE(rtp_parameters.encodings[2].active);
rtp_parameters.encodings[0].active = false;
rtp_parameters.encodings[1].active = false;
EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters).ok());
EXPECT_TRUE(stream->GetEncoderConfig().is_quality_scaling_allowed);
}
class WebRtcVideoChannelSimulcastTest : public ::testing::Test {
public:
WebRtcVideoChannelSimulcastTest()
: fake_call_(),
encoder_factory_(new cricket::FakeWebRtcVideoEncoderFactory),
decoder_factory_(new cricket::FakeWebRtcVideoDecoderFactory),
mock_rate_allocator_factory_(
std::make_unique<webrtc::MockVideoBitrateAllocatorFactory>()),
engine_(std::unique_ptr<cricket::FakeWebRtcVideoEncoderFactory>(
encoder_factory_),
std::unique_ptr<cricket::FakeWebRtcVideoDecoderFactory>(
decoder_factory_),
field_trials_),
last_ssrc_(0) {}
void SetUp() override {
encoder_factory_->AddSupportedVideoCodecType("VP8");
decoder_factory_->AddSupportedVideoCodecType("VP8");
channel_.reset(engine_.CreateMediaChannel(
&fake_call_, GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(),
mock_rate_allocator_factory_.get()));
channel_->OnReadyToSend(true);
last_ssrc_ = 123;
}
protected:
void VerifySimulcastSettings(const VideoCodec& codec,
int capture_width,
int capture_height,
size_t num_configured_streams,
size_t expected_num_streams,
bool screenshare,
bool conference_mode) {
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
parameters.conference_mode = conference_mode;
ASSERT_TRUE(channel_->SetSendParameters(parameters));
std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
RTC_DCHECK(num_configured_streams <= ssrcs.size());
ssrcs.resize(num_configured_streams);
AddSendStream(CreateSimStreamParams("cname", ssrcs));
// Send a full-size frame to trigger a stream reconfiguration to use all
// expected simulcast layers.
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(capture_width, capture_height,
rtc::kNumMicrosecsPerSec / 30);
VideoOptions options;
if (screenshare)
options.is_screencast = screenshare;
EXPECT_TRUE(
channel_->SetVideoSend(ssrcs.front(), &options, &frame_forwarder));
// Fetch the latest stream since SetVideoSend() may recreate it if the
// screen content setting is changed.
FakeVideoSendStream* stream = fake_call_.GetVideoSendStreams().front();
channel_->SetSend(true);
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
auto rtp_parameters = channel_->GetRtpSendParameters(kSsrcs3[0]);
EXPECT_EQ(num_configured_streams, rtp_parameters.encodings.size());
std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
ASSERT_EQ(expected_num_streams, video_streams.size());
EXPECT_LE(expected_num_streams, stream->GetConfig().rtp.ssrcs.size());
std::vector<webrtc::VideoStream> expected_streams;
if (num_configured_streams > 1 || conference_mode) {
expected_streams = GetSimulcastConfig(
/*min_layers=*/1, num_configured_streams, capture_width,
capture_height, webrtc::kDefaultBitratePriority, kDefaultQpMax,
screenshare && conference_mode, true, field_trials_);
if (screenshare && conference_mode) {
for (const webrtc::VideoStream& stream : expected_streams) {
// Never scale screen content.
EXPECT_EQ(stream.width, rtc::checked_cast<size_t>(capture_width));
EXPECT_EQ(stream.height, rtc::checked_cast<size_t>(capture_height));
}
}
} else {
webrtc::VideoStream stream;
stream.width = capture_width;
stream.height = capture_height;
stream.max_framerate = kDefaultVideoMaxFramerate;
stream.min_bitrate_bps = webrtc::kDefaultMinVideoBitrateBps;
stream.target_bitrate_bps = stream.max_bitrate_bps =
GetMaxDefaultBitrateBps(capture_width, capture_height);
stream.max_qp = kDefaultQpMax;
expected_streams.push_back(stream);
}
ASSERT_EQ(expected_streams.size(), video_streams.size());
size_t num_streams = video_streams.size();
int total_max_bitrate_bps = 0;
for (size_t i = 0; i < num_streams; ++i) {
EXPECT_EQ(expected_streams[i].width, video_streams[i].width);
EXPECT_EQ(expected_streams[i].height, video_streams[i].height);
EXPECT_GT(video_streams[i].max_framerate, 0);
EXPECT_EQ(expected_streams[i].max_framerate,
video_streams[i].max_framerate);
EXPECT_GT(video_streams[i].min_bitrate_bps, 0);
EXPECT_EQ(expected_streams[i].min_bitrate_bps,
video_streams[i].min_bitrate_bps);
EXPECT_GT(video_streams[i].target_bitrate_bps, 0);
EXPECT_EQ(expected_streams[i].target_bitrate_bps,
video_streams[i].target_bitrate_bps);
EXPECT_GT(video_streams[i].max_bitrate_bps, 0);
EXPECT_EQ(expected_streams[i].max_bitrate_bps,
video_streams[i].max_bitrate_bps);
EXPECT_GT(video_streams[i].max_qp, 0);
EXPECT_EQ(expected_streams[i].max_qp, video_streams[i].max_qp);
EXPECT_EQ(num_configured_streams > 1 || conference_mode,
expected_streams[i].num_temporal_layers.has_value());
if (conference_mode) {
EXPECT_EQ(expected_streams[i].num_temporal_layers,
video_streams[i].num_temporal_layers);
}
if (i == num_streams - 1) {
total_max_bitrate_bps += video_streams[i].max_bitrate_bps;
} else {
total_max_bitrate_bps += video_streams[i].target_bitrate_bps;
}
}
EXPECT_TRUE(channel_->SetVideoSend(ssrcs.front(), nullptr, nullptr));
}
FakeVideoSendStream* AddSendStream() {
return AddSendStream(StreamParams::CreateLegacy(last_ssrc_++));
}
FakeVideoSendStream* AddSendStream(const StreamParams& sp) {
size_t num_streams = fake_call_.GetVideoSendStreams().size();
EXPECT_TRUE(channel_->AddSendStream(sp));
std::vector<FakeVideoSendStream*> streams =
fake_call_.GetVideoSendStreams();
EXPECT_EQ(num_streams + 1, streams.size());
return streams[streams.size() - 1];
}
std::vector<FakeVideoSendStream*> GetFakeSendStreams() {
return fake_call_.GetVideoSendStreams();
}
FakeVideoReceiveStream* AddRecvStream() {
return AddRecvStream(StreamParams::CreateLegacy(last_ssrc_++));
}
FakeVideoReceiveStream* AddRecvStream(const StreamParams& sp) {
size_t num_streams = fake_call_.GetVideoReceiveStreams().size();
EXPECT_TRUE(channel_->AddRecvStream(sp));
std::vector<FakeVideoReceiveStream*> streams =
fake_call_.GetVideoReceiveStreams();
EXPECT_EQ(num_streams + 1, streams.size());
return streams[streams.size() - 1];
}
webrtc::RtcEventLogNull event_log_;
FakeCall fake_call_;
cricket::FakeWebRtcVideoEncoderFactory* encoder_factory_;
cricket::FakeWebRtcVideoDecoderFactory* decoder_factory_;
std::unique_ptr<webrtc::MockVideoBitrateAllocatorFactory>
mock_rate_allocator_factory_;
webrtc::FieldTrialBasedConfig field_trials_;
WebRtcVideoEngine engine_;
std::unique_ptr<VideoMediaChannel> channel_;
uint32_t last_ssrc_;
};
TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsWith2SimulcastStreams) {
VerifySimulcastSettings(cricket::VideoCodec("VP8"), 640, 360, 2, 2, false,
true);
}
TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsWith3SimulcastStreams) {
VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 3, false,
true);
}
// Test that we normalize send codec format size in simulcast.
TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsWithOddSizeInSimulcast) {
VerifySimulcastSettings(cricket::VideoCodec("VP8"), 541, 271, 2, 2, false,
true);
}
TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsForScreenshare) {
VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 3, true,
false);
}
TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsForSimulcastScreenshare) {
VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 2, true,
true);
}
TEST_F(WebRtcVideoChannelSimulcastTest, SimulcastScreenshareWithoutConference) {
VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 3, true,
false);
}
TEST_F(WebRtcVideoChannelBaseTest, GetSources) {
EXPECT_THAT(channel_->GetSources(kSsrc), IsEmpty());
channel_->SetDefaultSink(&renderer_);
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(renderer_.num_rendered_frames(), 0);
// Send and receive one frame.
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_THAT(channel_->GetSources(kSsrc - 1), IsEmpty());
EXPECT_THAT(channel_->GetSources(kSsrc), SizeIs(1));
EXPECT_THAT(channel_->GetSources(kSsrc + 1), IsEmpty());
webrtc::RtpSource source = channel_->GetSources(kSsrc)[0];
EXPECT_EQ(source.source_id(), kSsrc);
EXPECT_EQ(source.source_type(), webrtc::RtpSourceType::SSRC);
int64_t rtp_timestamp_1 = source.rtp_timestamp();
int64_t timestamp_ms_1 = source.timestamp_ms();
// Send and receive another frame.
SendFrame();
EXPECT_FRAME_WAIT(2, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_THAT(channel_->GetSources(kSsrc - 1), IsEmpty());
EXPECT_THAT(channel_->GetSources(kSsrc), SizeIs(1));
EXPECT_THAT(channel_->GetSources(kSsrc + 1), IsEmpty());
source = channel_->GetSources(kSsrc)[0];
EXPECT_EQ(source.source_id(), kSsrc);
EXPECT_EQ(source.source_type(), webrtc::RtpSourceType::SSRC);
int64_t rtp_timestamp_2 = source.rtp_timestamp();
int64_t timestamp_ms_2 = source.timestamp_ms();
EXPECT_GT(rtp_timestamp_2, rtp_timestamp_1);
EXPECT_GT(timestamp_ms_2, timestamp_ms_1);
}
TEST_F(WebRtcVideoChannelTest, SetsRidsOnSendStream) {
StreamParams sp = CreateSimStreamParams("cname", {123, 456, 789});
std::vector<std::string> rids = {"f", "h", "q"};
std::vector<cricket::RidDescription> rid_descriptions;
for (const auto& rid : rids) {
rid_descriptions.emplace_back(rid, cricket::RidDirection::kSend);
}
sp.set_rids(rid_descriptions);
ASSERT_TRUE(channel_->AddSendStream(sp));
const auto& streams = fake_call_->GetVideoSendStreams();
ASSERT_EQ(1u, streams.size());
auto stream = streams[0];
ASSERT_NE(stream, nullptr);
const auto& config = stream->GetConfig();
EXPECT_THAT(config.rtp.rids, ElementsAreArray(rids));
}
} // namespace cricket