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gecko-dev/third_party/libwebrtc/pc/rtp_sender_receiver_unittes...

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/*
* Copyright 2012 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 <stddef.h>
#include <cstdint>
#include <iterator>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "api/audio_options.h"
#include "api/crypto/crypto_options.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/dtmf_sender_interface.h"
#include "api/media_stream_interface.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/test/fake_frame_decryptor.h"
#include "api/test/fake_frame_encryptor.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "api/video/video_codec_constants.h"
#include "media/base/codec.h"
#include "media/base/delayable.h"
#include "media/base/fake_media_engine.h"
#include "media/base/media_channel.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "media/base/rid_description.h"
#include "media/base/stream_params.h"
#include "media/base/test_utils.h"
#include "media/engine/fake_webrtc_call.h"
#include "p2p/base/dtls_transport_internal.h"
#include "p2p/base/fake_dtls_transport.h"
#include "p2p/base/p2p_constants.h"
#include "pc/audio_rtp_receiver.h"
#include "pc/audio_track.h"
#include "pc/channel.h"
#include "pc/dtls_srtp_transport.h"
#include "pc/local_audio_source.h"
#include "pc/media_stream.h"
#include "pc/rtp_sender.h"
#include "pc/rtp_transport_internal.h"
#include "pc/test/fake_video_track_source.h"
#include "pc/video_rtp_receiver.h"
#include "pc/video_track.h"
#include "rtc_base/checks.h"
#include "rtc_base/gunit.h"
#include "rtc_base/thread.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/run_loop.h"
#include "test/scoped_key_value_config.h"
using ::testing::_;
using ::testing::ContainerEq;
using ::testing::Exactly;
using ::testing::InvokeWithoutArgs;
using ::testing::Return;
using RidList = std::vector<std::string>;
namespace {
static const char kStreamId1[] = "local_stream_1";
static const char kVideoTrackId[] = "video_1";
static const char kAudioTrackId[] = "audio_1";
static const uint32_t kVideoSsrc = 98;
static const uint32_t kVideoSsrc2 = 100;
static const uint32_t kAudioSsrc = 99;
static const uint32_t kAudioSsrc2 = 101;
static const uint32_t kVideoSsrcSimulcast = 102;
static const uint32_t kVideoSimulcastLayerCount = 2;
static const int kDefaultTimeout = 10000; // 10 seconds.
class MockSetStreamsObserver
: public webrtc::RtpSenderBase::SetStreamsObserver {
public:
MOCK_METHOD(void, OnSetStreams, (), (override));
};
} // namespace
namespace webrtc {
class RtpSenderReceiverTest
: public ::testing::Test,
public ::testing::WithParamInterface<std::pair<RidList, RidList>> {
public:
RtpSenderReceiverTest()
: network_thread_(rtc::Thread::Current()),
worker_thread_(rtc::Thread::Current()),
video_bitrate_allocator_factory_(
webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
// Create fake media engine/etc. so we can create channels to use to
// test RtpSenders/RtpReceivers.
media_engine_(std::make_unique<cricket::FakeMediaEngine>()),
fake_call_(worker_thread_, network_thread_),
local_stream_(MediaStream::Create(kStreamId1)) {
rtp_dtls_transport_ = std::make_unique<cricket::FakeDtlsTransport>(
"fake_dtls_transport", cricket::ICE_CANDIDATE_COMPONENT_RTP);
rtp_transport_ = CreateDtlsSrtpTransport();
// Create the channels, discard the result; we get them later.
// Fake media channels are owned by the media engine.
media_engine_->voice().CreateMediaChannel(
&fake_call_, cricket::MediaConfig(), cricket::AudioOptions(),
webrtc::CryptoOptions());
media_engine_->video().CreateMediaChannel(
&fake_call_, cricket::MediaConfig(), cricket::VideoOptions(),
webrtc::CryptoOptions(), video_bitrate_allocator_factory_.get());
voice_media_channel_ = absl::WrapUnique(media_engine_->GetVoiceChannel(0));
video_media_channel_ = absl::WrapUnique(media_engine_->GetVideoChannel(0));
RTC_CHECK(voice_media_channel());
RTC_CHECK(video_media_channel());
// Create sender channel objects
voice_send_channel_ =
std::make_unique<cricket::VoiceMediaSendChannel>(voice_media_channel());
video_send_channel_ =
std::make_unique<cricket::VideoMediaSendChannel>(video_media_channel());
// Create streams for predefined SSRCs. Streams need to exist in order
// for the senders and receievers to apply parameters to them.
// Normally these would be created by SetLocalDescription and
// SetRemoteDescription.
voice_media_channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kAudioSsrc));
voice_media_channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kAudioSsrc));
voice_media_channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kAudioSsrc2));
voice_media_channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kAudioSsrc2));
video_media_channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kVideoSsrc));
video_media_channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kVideoSsrc));
video_media_channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kVideoSsrc2));
video_media_channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kVideoSsrc2));
}
~RtpSenderReceiverTest() {
audio_rtp_sender_ = nullptr;
video_rtp_sender_ = nullptr;
audio_rtp_receiver_ = nullptr;
video_rtp_receiver_ = nullptr;
local_stream_ = nullptr;
video_track_ = nullptr;
audio_track_ = nullptr;
}
std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() {
auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
/*rtcp_mux_required=*/true, field_trials_);
dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport_.get(),
/*rtcp_dtls_transport=*/nullptr);
return dtls_srtp_transport;
}
// Needed to use DTMF sender.
void AddDtmfCodec() {
cricket::AudioSendParameters params;
const cricket::AudioCodec kTelephoneEventCodec(106, "telephone-event", 8000,
0, 1);
params.codecs.push_back(kTelephoneEventCodec);
voice_media_channel()->SetSendParameters(params);
}
void AddVideoTrack() { AddVideoTrack(false); }
void AddVideoTrack(bool is_screencast) {
rtc::scoped_refptr<VideoTrackSourceInterface> source(
FakeVideoTrackSource::Create(is_screencast));
video_track_ =
VideoTrack::Create(kVideoTrackId, source, rtc::Thread::Current());
EXPECT_TRUE(local_stream_->AddTrack(video_track_));
}
void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); }
void CreateAudioRtpSender(
const rtc::scoped_refptr<LocalAudioSource>& source) {
audio_track_ = AudioTrack::Create(kAudioTrackId, source);
EXPECT_TRUE(local_stream_->AddTrack(audio_track_));
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
std::make_unique<MockSetStreamsObserver>();
audio_rtp_sender_ =
AudioRtpSender::Create(worker_thread_, audio_track_->id(), nullptr,
set_streams_observer.get());
ASSERT_TRUE(audio_rtp_sender_->SetTrack(audio_track_.get()));
EXPECT_CALL(*set_streams_observer, OnSetStreams());
audio_rtp_sender_->SetStreams({local_stream_->id()});
audio_rtp_sender_->SetMediaChannel(voice_send_channel_.get());
audio_rtp_sender_->SetSsrc(kAudioSsrc);
VerifyVoiceChannelInput();
}
void CreateAudioRtpSenderWithNoTrack() {
audio_rtp_sender_ =
AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr, nullptr);
audio_rtp_sender_->SetMediaChannel(
voice_media_channel()->AsVoiceSendChannel());
}
void CreateVideoRtpSender(uint32_t ssrc) {
CreateVideoRtpSender(false, ssrc);
}
void CreateVideoRtpSender() { CreateVideoRtpSender(false); }
cricket::StreamParams CreateSimulcastStreamParams(int num_layers) {
std::vector<uint32_t> ssrcs;
ssrcs.reserve(num_layers);
for (int i = 0; i < num_layers; ++i) {
ssrcs.push_back(kVideoSsrcSimulcast + i);
}
return cricket::CreateSimStreamParams("cname", ssrcs);
}
uint32_t CreateVideoRtpSender(const cricket::StreamParams& stream_params) {
video_media_channel_->AddSendStream(stream_params);
uint32_t primary_ssrc = stream_params.first_ssrc();
CreateVideoRtpSender(primary_ssrc);
return primary_ssrc;
}
uint32_t CreateVideoRtpSenderWithSimulcast(
int num_layers = kVideoSimulcastLayerCount) {
return CreateVideoRtpSender(CreateSimulcastStreamParams(num_layers));
}
uint32_t CreateVideoRtpSenderWithSimulcast(
const std::vector<std::string>& rids) {
cricket::StreamParams stream_params =
CreateSimulcastStreamParams(rids.size());
std::vector<cricket::RidDescription> rid_descriptions;
absl::c_transform(
rids, std::back_inserter(rid_descriptions), [](const std::string& rid) {
return cricket::RidDescription(rid, cricket::RidDirection::kSend);
});
stream_params.set_rids(rid_descriptions);
return CreateVideoRtpSender(stream_params);
}
void CreateVideoRtpSender(bool is_screencast, uint32_t ssrc = kVideoSsrc) {
AddVideoTrack(is_screencast);
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
std::make_unique<MockSetStreamsObserver>();
video_rtp_sender_ = VideoRtpSender::Create(
worker_thread_, video_track_->id(), set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_.get()));
EXPECT_CALL(*set_streams_observer, OnSetStreams());
video_rtp_sender_->SetStreams({local_stream_->id()});
video_rtp_sender_->SetMediaChannel(
video_media_channel()->AsVideoSendChannel());
video_rtp_sender_->SetSsrc(ssrc);
VerifyVideoChannelInput(ssrc);
}
void CreateVideoRtpSenderWithNoTrack() {
video_rtp_sender_ =
VideoRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
video_rtp_sender_->SetMediaChannel(
video_media_channel()->AsVideoSendChannel());
}
void DestroyAudioRtpSender() {
audio_rtp_sender_ = nullptr;
VerifyVoiceChannelNoInput();
}
void DestroyVideoRtpSender() {
video_rtp_sender_ = nullptr;
VerifyVideoChannelNoInput();
}
void CreateAudioRtpReceiver(
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
audio_rtp_receiver_ = rtc::make_ref_counted<AudioRtpReceiver>(
rtc::Thread::Current(), kAudioTrackId, streams,
/*is_unified_plan=*/true);
audio_rtp_receiver_->SetMediaChannel(
voice_media_channel()->AsVoiceReceiveChannel());
audio_rtp_receiver_->SetupMediaChannel(kAudioSsrc);
audio_track_ = audio_rtp_receiver_->audio_track();
VerifyVoiceChannelOutput();
}
void CreateVideoRtpReceiver(
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
video_rtp_receiver_ = rtc::make_ref_counted<VideoRtpReceiver>(
rtc::Thread::Current(), kVideoTrackId, streams);
video_rtp_receiver_->SetMediaChannel(
video_media_channel()->AsVideoReceiveChannel());
video_rtp_receiver_->SetupMediaChannel(kVideoSsrc);
video_track_ = video_rtp_receiver_->video_track();
VerifyVideoChannelOutput();
}
void CreateVideoRtpReceiverWithSimulcast(
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {},
int num_layers = kVideoSimulcastLayerCount) {
std::vector<uint32_t> ssrcs;
ssrcs.reserve(num_layers);
for (int i = 0; i < num_layers; ++i)
ssrcs.push_back(kVideoSsrcSimulcast + i);
cricket::StreamParams stream_params =
cricket::CreateSimStreamParams("cname", ssrcs);
video_media_channel_->AddRecvStream(stream_params);
uint32_t primary_ssrc = stream_params.first_ssrc();
video_rtp_receiver_ = rtc::make_ref_counted<VideoRtpReceiver>(
rtc::Thread::Current(), kVideoTrackId, streams);
video_rtp_receiver_->SetMediaChannel(
video_media_channel()->AsVideoReceiveChannel());
video_rtp_receiver_->SetupMediaChannel(primary_ssrc);
video_track_ = video_rtp_receiver_->video_track();
}
void DestroyAudioRtpReceiver() {
if (!audio_rtp_receiver_)
return;
audio_rtp_receiver_->SetMediaChannel(nullptr);
audio_rtp_receiver_ = nullptr;
VerifyVoiceChannelNoOutput();
}
void DestroyVideoRtpReceiver() {
if (!video_rtp_receiver_)
return;
video_rtp_receiver_->Stop();
video_rtp_receiver_->SetMediaChannel(nullptr);
video_rtp_receiver_ = nullptr;
VerifyVideoChannelNoOutput();
}
void VerifyVoiceChannelInput() { VerifyVoiceChannelInput(kAudioSsrc); }
void VerifyVoiceChannelInput(uint32_t ssrc) {
// Verify that the media channel has an audio source, and the stream isn't
// muted.
EXPECT_TRUE(voice_media_channel()->HasSource(ssrc));
EXPECT_FALSE(voice_media_channel()->IsStreamMuted(ssrc));
}
void VerifyVideoChannelInput() { VerifyVideoChannelInput(kVideoSsrc); }
void VerifyVideoChannelInput(uint32_t ssrc) {
// Verify that the media channel has a video source,
EXPECT_TRUE(video_media_channel_->HasSource(ssrc));
}
void VerifyVoiceChannelNoInput() { VerifyVoiceChannelNoInput(kAudioSsrc); }
void VerifyVoiceChannelNoInput(uint32_t ssrc) {
// Verify that the media channel's source is reset.
EXPECT_FALSE(voice_media_channel()->HasSource(ssrc));
}
void VerifyVideoChannelNoInput() { VerifyVideoChannelNoInput(kVideoSsrc); }
void VerifyVideoChannelNoInput(uint32_t ssrc) {
// Verify that the media channel's source is reset.
EXPECT_FALSE(video_media_channel_->HasSource(ssrc));
}
void VerifyVoiceChannelOutput() {
// Verify that the volume is initialized to 1.
double volume;
EXPECT_TRUE(voice_media_channel()->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(1, volume);
}
void VerifyVideoChannelOutput() {
// Verify that the media channel has a sink.
EXPECT_TRUE(video_media_channel_->HasSink(kVideoSsrc));
}
void VerifyVoiceChannelNoOutput() {
// Verify that the volume is reset to 0.
double volume;
EXPECT_TRUE(voice_media_channel()->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0, volume);
}
void VerifyVideoChannelNoOutput() {
// Verify that the media channel's sink is reset.
EXPECT_FALSE(video_media_channel_->HasSink(kVideoSsrc));
}
// Verifies that the encoding layers contain the specified RIDs.
bool VerifyEncodingLayers(const VideoRtpSender& sender,
const std::vector<std::string>& rids) {
bool has_failure = HasFailure();
RtpParameters parameters = sender.GetParameters();
std::vector<std::string> encoding_rids;
absl::c_transform(
parameters.encodings, std::back_inserter(encoding_rids),
[](const RtpEncodingParameters& encoding) { return encoding.rid; });
EXPECT_THAT(rids, ContainerEq(encoding_rids));
return has_failure || !HasFailure();
}
// Runs a test for disabling the encoding layers on the specified sender.
void RunDisableEncodingLayersTest(
const std::vector<std::string>& all_layers,
const std::vector<std::string>& disabled_layers,
VideoRtpSender* sender) {
std::vector<std::string> expected;
absl::c_copy_if(all_layers, std::back_inserter(expected),
[&disabled_layers](const std::string& rid) {
return !absl::c_linear_search(disabled_layers, rid);
});
EXPECT_TRUE(VerifyEncodingLayers(*sender, all_layers));
sender->DisableEncodingLayers(disabled_layers);
EXPECT_TRUE(VerifyEncodingLayers(*sender, expected));
}
// Runs a test for setting an encoding layer as inactive.
// This test assumes that some layers have already been disabled.
void RunSetLastLayerAsInactiveTest(VideoRtpSender* sender) {
auto parameters = sender->GetParameters();
if (parameters.encodings.size() == 0) {
return;
}
RtpEncodingParameters& encoding = parameters.encodings.back();
auto rid = encoding.rid;
EXPECT_TRUE(encoding.active);
encoding.active = false;
auto error = sender->SetParameters(parameters);
ASSERT_TRUE(error.ok());
parameters = sender->GetParameters();
RtpEncodingParameters& result_encoding = parameters.encodings.back();
EXPECT_EQ(rid, result_encoding.rid);
EXPECT_FALSE(result_encoding.active);
}
// Runs a test for disabling the encoding layers on a sender without a media
// channel.
void RunDisableSimulcastLayersWithoutMediaEngineTest(
const std::vector<std::string>& all_layers,
const std::vector<std::string>& disabled_layers) {
auto sender = VideoRtpSender::Create(rtc::Thread::Current(), "1", nullptr);
RtpParameters parameters;
parameters.encodings.resize(all_layers.size());
for (size_t i = 0; i < all_layers.size(); ++i) {
parameters.encodings[i].rid = all_layers[i];
}
sender->set_init_send_encodings(parameters.encodings);
RunDisableEncodingLayersTest(all_layers, disabled_layers, sender.get());
RunSetLastLayerAsInactiveTest(sender.get());
}
// Runs a test for disabling the encoding layers on a sender with a media
// channel.
void RunDisableSimulcastLayersWithMediaEngineTest(
const std::vector<std::string>& all_layers,
const std::vector<std::string>& disabled_layers) {
uint32_t ssrc = CreateVideoRtpSenderWithSimulcast(all_layers);
RunDisableEncodingLayersTest(all_layers, disabled_layers,
video_rtp_sender_.get());
auto channel_parameters = video_media_channel_->GetRtpSendParameters(ssrc);
ASSERT_EQ(channel_parameters.encodings.size(), all_layers.size());
for (size_t i = 0; i < all_layers.size(); ++i) {
EXPECT_EQ(all_layers[i], channel_parameters.encodings[i].rid);
bool is_active = !absl::c_linear_search(disabled_layers, all_layers[i]);
EXPECT_EQ(is_active, channel_parameters.encodings[i].active);
}
RunSetLastLayerAsInactiveTest(video_rtp_sender_.get());
}
// Check that minimum Jitter Buffer delay is propagated to the underlying
// `media_channel`.
void VerifyRtpReceiverDelayBehaviour(cricket::Delayable* media_channel,
RtpReceiverInterface* receiver,
uint32_t ssrc) {
receiver->SetJitterBufferMinimumDelay(/*delay_seconds=*/0.5);
absl::optional<int> delay_ms =
media_channel->GetBaseMinimumPlayoutDelayMs(ssrc); // In milliseconds.
EXPECT_DOUBLE_EQ(0.5, delay_ms.value_or(0) / 1000.0);
}
protected:
cricket::FakeVideoMediaChannel* video_media_channel() {
return video_media_channel_.get();
}
cricket::FakeVoiceMediaChannel* voice_media_channel() {
return voice_media_channel_.get();
}
test::RunLoop run_loop_;
rtc::Thread* const network_thread_;
rtc::Thread* const worker_thread_;
webrtc::RtcEventLogNull event_log_;
// The `rtp_dtls_transport_` and `rtp_transport_` should be destroyed after
// the `channel_manager`.
std::unique_ptr<cricket::DtlsTransportInternal> rtp_dtls_transport_;
std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport_;
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory_;
std::unique_ptr<cricket::FakeMediaEngine> media_engine_;
rtc::UniqueRandomIdGenerator ssrc_generator_;
cricket::FakeCall fake_call_;
std::unique_ptr<cricket::FakeVoiceMediaChannel> voice_media_channel_;
std::unique_ptr<cricket::FakeVideoMediaChannel> video_media_channel_;
std::unique_ptr<cricket::VoiceMediaSendChannel> voice_send_channel_;
std::unique_ptr<cricket::VideoMediaSendChannel> video_send_channel_;
rtc::scoped_refptr<AudioRtpSender> audio_rtp_sender_;
rtc::scoped_refptr<VideoRtpSender> video_rtp_sender_;
rtc::scoped_refptr<AudioRtpReceiver> audio_rtp_receiver_;
rtc::scoped_refptr<VideoRtpReceiver> video_rtp_receiver_;
rtc::scoped_refptr<MediaStreamInterface> local_stream_;
rtc::scoped_refptr<VideoTrackInterface> video_track_;
rtc::scoped_refptr<AudioTrackInterface> audio_track_;
webrtc::test::ScopedKeyValueConfig field_trials_;
};
// Test that `voice_channel_` is updated when an audio track is associated
// and disassociated with an AudioRtpSender.
TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpSender) {
CreateAudioRtpSender();
DestroyAudioRtpSender();
}
// Test that `video_channel_` is updated when a video track is associated and
// disassociated with a VideoRtpSender.
TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpSender) {
CreateVideoRtpSender();
DestroyVideoRtpSender();
}
// Test that `voice_channel_` is updated when a remote audio track is
// associated and disassociated with an AudioRtpReceiver.
TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpReceiver) {
CreateAudioRtpReceiver();
DestroyAudioRtpReceiver();
}
// Test that `video_channel_` is updated when a remote video track is
// associated and disassociated with a VideoRtpReceiver.
TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpReceiver) {
CreateVideoRtpReceiver();
DestroyVideoRtpReceiver();
}
TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpReceiverWithStreams) {
CreateAudioRtpReceiver({local_stream_});
DestroyAudioRtpReceiver();
}
TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpReceiverWithStreams) {
CreateVideoRtpReceiver({local_stream_});
DestroyVideoRtpReceiver();
}
// Test that the AudioRtpSender applies options from the local audio source.
TEST_F(RtpSenderReceiverTest, LocalAudioSourceOptionsApplied) {
cricket::AudioOptions options;
options.echo_cancellation = true;
auto source = LocalAudioSource::Create(&options);
CreateAudioRtpSender(source);
EXPECT_EQ(true, voice_media_channel()->options().echo_cancellation);
DestroyAudioRtpSender();
}
// Test that the stream is muted when the track is disabled, and unmuted when
// the track is enabled.
TEST_F(RtpSenderReceiverTest, LocalAudioTrackDisable) {
CreateAudioRtpSender();
audio_track_->set_enabled(false);
EXPECT_TRUE(voice_media_channel()->IsStreamMuted(kAudioSsrc));
audio_track_->set_enabled(true);
EXPECT_FALSE(voice_media_channel()->IsStreamMuted(kAudioSsrc));
DestroyAudioRtpSender();
}
// Test that the volume is set to 0 when the track is disabled, and back to
// 1 when the track is enabled.
TEST_F(RtpSenderReceiverTest, RemoteAudioTrackDisable) {
CreateAudioRtpReceiver();
double volume;
EXPECT_TRUE(voice_media_channel()->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(1, volume);
// Handling of enable/disable is applied asynchronously.
audio_track_->set_enabled(false);
run_loop_.Flush();
EXPECT_TRUE(voice_media_channel()->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0, volume);
audio_track_->set_enabled(true);
run_loop_.Flush();
EXPECT_TRUE(voice_media_channel()->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(1, volume);
DestroyAudioRtpReceiver();
}
// Currently no action is taken when a remote video track is disabled or
// enabled, so there's nothing to test here, other than what is normally
// verified in DestroyVideoRtpSender.
TEST_F(RtpSenderReceiverTest, LocalVideoTrackDisable) {
CreateVideoRtpSender();
video_track_->set_enabled(false);
video_track_->set_enabled(true);
DestroyVideoRtpSender();
}
// Test that the state of the video track created by the VideoRtpReceiver is
// updated when the receiver is destroyed.
TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) {
CreateVideoRtpReceiver();
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track_->state());
EXPECT_EQ(webrtc::MediaSourceInterface::kLive,
video_track_->GetSource()->state());
DestroyVideoRtpReceiver();
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track_->state());
EXPECT_EQ(webrtc::MediaSourceInterface::kEnded,
video_track_->GetSource()->state());
DestroyVideoRtpReceiver();
}
// Currently no action is taken when a remote video track is disabled or
// enabled, so there's nothing to test here, other than what is normally
// verified in DestroyVideoRtpReceiver.
TEST_F(RtpSenderReceiverTest, RemoteVideoTrackDisable) {
CreateVideoRtpReceiver();
video_track_->set_enabled(false);
video_track_->set_enabled(true);
DestroyVideoRtpReceiver();
}
// Test that the AudioRtpReceiver applies volume changes from the track source
// to the media channel.
TEST_F(RtpSenderReceiverTest, RemoteAudioTrackSetVolume) {
CreateAudioRtpReceiver();
double volume;
audio_track_->GetSource()->SetVolume(0.5);
run_loop_.Flush();
EXPECT_TRUE(voice_media_channel()->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0.5, volume);
// Disable the audio track, this should prevent setting the volume.
audio_track_->set_enabled(false);
RTC_DCHECK_EQ(worker_thread_, run_loop_.task_queue());
run_loop_.Flush();
audio_track_->GetSource()->SetVolume(0.8);
EXPECT_TRUE(voice_media_channel()->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0, volume);
// When the track is enabled, the previously set volume should take effect.
audio_track_->set_enabled(true);
run_loop_.Flush();
EXPECT_TRUE(voice_media_channel()->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0.8, volume);
// Try changing volume one more time.
audio_track_->GetSource()->SetVolume(0.9);
run_loop_.Flush();
EXPECT_TRUE(voice_media_channel()->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0.9, volume);
DestroyAudioRtpReceiver();
}
TEST_F(RtpSenderReceiverTest, AudioRtpReceiverDelay) {
CreateAudioRtpReceiver();
VerifyRtpReceiverDelayBehaviour(
voice_media_channel()->AsVoiceReceiveChannel(), audio_rtp_receiver_.get(),
kAudioSsrc);
DestroyAudioRtpReceiver();
}
TEST_F(RtpSenderReceiverTest, VideoRtpReceiverDelay) {
CreateVideoRtpReceiver();
VerifyRtpReceiverDelayBehaviour(
video_media_channel()->AsVideoReceiveChannel(), video_rtp_receiver_.get(),
kVideoSsrc);
DestroyVideoRtpReceiver();
}
// Test that the media channel isn't enabled for sending if the audio sender
// doesn't have both a track and SSRC.
TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) {
CreateAudioRtpSenderWithNoTrack();
rtc::scoped_refptr<AudioTrackInterface> track =
AudioTrack::Create(kAudioTrackId, nullptr);
// Track but no SSRC.
EXPECT_TRUE(audio_rtp_sender_->SetTrack(track.get()));
VerifyVoiceChannelNoInput();
// SSRC but no track.
EXPECT_TRUE(audio_rtp_sender_->SetTrack(nullptr));
audio_rtp_sender_->SetSsrc(kAudioSsrc);
VerifyVoiceChannelNoInput();
}
// Test that the media channel isn't enabled for sending if the video sender
// doesn't have both a track and SSRC.
TEST_F(RtpSenderReceiverTest, VideoSenderWithoutTrackAndSsrc) {
CreateVideoRtpSenderWithNoTrack();
// Track but no SSRC.
EXPECT_TRUE(video_rtp_sender_->SetTrack(video_track_.get()));
VerifyVideoChannelNoInput();
// SSRC but no track.
EXPECT_TRUE(video_rtp_sender_->SetTrack(nullptr));
video_rtp_sender_->SetSsrc(kVideoSsrc);
VerifyVideoChannelNoInput();
}
// Test that the media channel is enabled for sending when the audio sender
// has a track and SSRC, when the SSRC is set first.
TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupSsrcThenTrack) {
CreateAudioRtpSenderWithNoTrack();
rtc::scoped_refptr<AudioTrackInterface> track =
AudioTrack::Create(kAudioTrackId, nullptr);
audio_rtp_sender_->SetSsrc(kAudioSsrc);
audio_rtp_sender_->SetTrack(track.get());
VerifyVoiceChannelInput();
DestroyAudioRtpSender();
}
// Test that the media channel is enabled for sending when the audio sender
// has a track and SSRC, when the SSRC is set last.
TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupTrackThenSsrc) {
CreateAudioRtpSenderWithNoTrack();
rtc::scoped_refptr<AudioTrackInterface> track =
AudioTrack::Create(kAudioTrackId, nullptr);
audio_rtp_sender_->SetTrack(track.get());
audio_rtp_sender_->SetSsrc(kAudioSsrc);
VerifyVoiceChannelInput();
DestroyAudioRtpSender();
}
// Test that the media channel is enabled for sending when the video sender
// has a track and SSRC, when the SSRC is set first.
TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) {
AddVideoTrack();
CreateVideoRtpSenderWithNoTrack();
video_rtp_sender_->SetSsrc(kVideoSsrc);
video_rtp_sender_->SetTrack(video_track_.get());
VerifyVideoChannelInput();
DestroyVideoRtpSender();
}
// Test that the media channel is enabled for sending when the video sender
// has a track and SSRC, when the SSRC is set last.
TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc) {
AddVideoTrack();
CreateVideoRtpSenderWithNoTrack();
video_rtp_sender_->SetTrack(video_track_.get());
video_rtp_sender_->SetSsrc(kVideoSsrc);
VerifyVideoChannelInput();
DestroyVideoRtpSender();
}
// Test that the media channel stops sending when the audio sender's SSRC is set
// to 0.
TEST_F(RtpSenderReceiverTest, AudioSenderSsrcSetToZero) {
CreateAudioRtpSender();
audio_rtp_sender_->SetSsrc(0);
VerifyVoiceChannelNoInput();
}
// Test that the media channel stops sending when the video sender's SSRC is set
// to 0.
TEST_F(RtpSenderReceiverTest, VideoSenderSsrcSetToZero) {
CreateAudioRtpSender();
audio_rtp_sender_->SetSsrc(0);
VerifyVideoChannelNoInput();
}
// Test that the media channel stops sending when the audio sender's track is
// set to null.
TEST_F(RtpSenderReceiverTest, AudioSenderTrackSetToNull) {
CreateAudioRtpSender();
EXPECT_TRUE(audio_rtp_sender_->SetTrack(nullptr));
VerifyVoiceChannelNoInput();
}
// Test that the media channel stops sending when the video sender's track is
// set to null.
TEST_F(RtpSenderReceiverTest, VideoSenderTrackSetToNull) {
CreateVideoRtpSender();
video_rtp_sender_->SetSsrc(0);
VerifyVideoChannelNoInput();
}
// Test that when the audio sender's SSRC is changed, the media channel stops
// sending with the old SSRC and starts sending with the new one.
TEST_F(RtpSenderReceiverTest, AudioSenderSsrcChanged) {
CreateAudioRtpSender();
audio_rtp_sender_->SetSsrc(kAudioSsrc2);
VerifyVoiceChannelNoInput(kAudioSsrc);
VerifyVoiceChannelInput(kAudioSsrc2);
audio_rtp_sender_ = nullptr;
VerifyVoiceChannelNoInput(kAudioSsrc2);
}
// Test that when the audio sender's SSRC is changed, the media channel stops
// sending with the old SSRC and starts sending with the new one.
TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) {
CreateVideoRtpSender();
video_rtp_sender_->SetSsrc(kVideoSsrc2);
VerifyVideoChannelNoInput(kVideoSsrc);
VerifyVideoChannelInput(kVideoSsrc2);
video_rtp_sender_ = nullptr;
VerifyVideoChannelNoInput(kVideoSsrc2);
}
TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParametersAsync) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
absl::optional<webrtc::RTCError> result;
audio_rtp_sender_->SetParametersAsync(
params, [&result](webrtc::RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParametersBeforeNegotiation) {
audio_rtp_sender_ =
AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr, nullptr);
RtpParameters params = audio_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
params.encodings[0].max_bitrate_bps = 90000;
EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000);
EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest,
AudioSenderCanSetParametersAsyncBeforeNegotiation) {
audio_rtp_sender_ =
AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr, nullptr);
absl::optional<webrtc::RTCError> result;
RtpParameters params = audio_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
params.encodings[0].max_bitrate_bps = 90000;
audio_rtp_sender_->SetParametersAsync(
params, [&result](webrtc::RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000);
audio_rtp_sender_->SetParametersAsync(
params, [&result](webrtc::RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioSenderInitParametersMovedAfterNegotiation) {
audio_track_ = AudioTrack::Create(kAudioTrackId, nullptr);
EXPECT_TRUE(local_stream_->AddTrack(audio_track_));
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
std::make_unique<MockSetStreamsObserver>();
audio_rtp_sender_ = AudioRtpSender::Create(
worker_thread_, audio_track_->id(), nullptr, set_streams_observer.get());
ASSERT_TRUE(audio_rtp_sender_->SetTrack(audio_track_.get()));
EXPECT_CALL(*set_streams_observer, OnSetStreams());
audio_rtp_sender_->SetStreams({local_stream_->id()});
std::vector<RtpEncodingParameters> init_encodings(1);
init_encodings[0].max_bitrate_bps = 60000;
audio_rtp_sender_->set_init_send_encodings(init_encodings);
RtpParameters params = audio_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 60000);
// Simulate the setLocalDescription call
std::vector<uint32_t> ssrcs(1, 1);
cricket::StreamParams stream_params =
cricket::CreateSimStreamParams("cname", ssrcs);
voice_media_channel()->AddSendStream(stream_params);
audio_rtp_sender_->SetMediaChannel(
voice_media_channel()->AsVoiceSendChannel());
audio_rtp_sender_->SetSsrc(1);
params = audio_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 60000);
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest,
AudioSenderMustCallGetParametersBeforeSetParametersBeforeNegotiation) {
audio_rtp_sender_ =
AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr, nullptr);
RtpParameters params;
RTCError result = audio_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest,
AudioSenderMustCallGetParametersBeforeSetParameters) {
CreateAudioRtpSender();
RtpParameters params;
RTCError result = audio_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest,
AudioSenderSetParametersInvalidatesTransactionId) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
RTCError result = audio_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest,
AudioSenderSetParametersAsyncInvalidatesTransactionId) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
absl::optional<webrtc::RTCError> result;
audio_rtp_sender_->SetParametersAsync(
params, [&result](webrtc::RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
audio_rtp_sender_->SetParametersAsync(
params, [&result](webrtc::RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_EQ(RTCErrorType::INVALID_STATE, result->type());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioSenderDetectTransactionIdModification) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
params.transaction_id = "";
RTCError result = audio_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioSenderCheckTransactionIdRefresh) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_NE(params.transaction_id.size(), 0U);
auto saved_transaction_id = params.transaction_id;
params = audio_rtp_sender_->GetParameters();
EXPECT_NE(saved_transaction_id, params.transaction_id);
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioSenderSetParametersOldValueFail) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
RtpParameters second_params = audio_rtp_sender_->GetParameters();
RTCError result = audio_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioSenderCantSetUnimplementedRtpParameters) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
// Unimplemented RtpParameters: mid
params.mid = "dummy_mid";
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
audio_rtp_sender_->SetParameters(params).type());
params = audio_rtp_sender_->GetParameters();
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) {
CreateAudioRtpSender();
EXPECT_EQ(-1, voice_media_channel()->max_bps());
webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
params.encodings[0].max_bitrate_bps = 1000;
EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
// Read back the parameters and verify they have been changed.
params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
// Verify that the audio channel received the new parameters.
params = voice_media_channel()->GetRtpSendParameters(kAudioSsrc);
EXPECT_EQ(1U, params.encodings.size());
EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
// Verify that the global bitrate limit has not been changed.
EXPECT_EQ(-1, voice_media_channel()->max_bps());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, SetAudioBitratePriority) {
CreateAudioRtpSender();
webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
EXPECT_EQ(webrtc::kDefaultBitratePriority,
params.encodings[0].bitrate_priority);
double new_bitrate_priority = 2.0;
params.encodings[0].bitrate_priority = new_bitrate_priority;
EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
params = voice_media_channel()->GetRtpSendParameters(kAudioSsrc);
EXPECT_EQ(1U, params.encodings.size());
EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParametersAsync) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
absl::optional<webrtc::RTCError> result;
video_rtp_sender_->SetParametersAsync(
params, [&result](webrtc::RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParametersBeforeNegotiation) {
video_rtp_sender_ =
VideoRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
RtpParameters params = video_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
params.encodings[0].max_bitrate_bps = 90000;
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
params = video_rtp_sender_->GetParameters();
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest,
VideoSenderCanSetParametersAsyncBeforeNegotiation) {
video_rtp_sender_ =
VideoRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
absl::optional<webrtc::RTCError> result;
RtpParameters params = video_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
params.encodings[0].max_bitrate_bps = 90000;
video_rtp_sender_->SetParametersAsync(
params, [&result](webrtc::RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
params = video_rtp_sender_->GetParameters();
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000);
video_rtp_sender_->SetParametersAsync(
params, [&result](webrtc::RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderInitParametersMovedAfterNegotiation) {
AddVideoTrack(false);
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
std::make_unique<MockSetStreamsObserver>();
video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, video_track_->id(),
set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_.get()));
EXPECT_CALL(*set_streams_observer, OnSetStreams());
video_rtp_sender_->SetStreams({local_stream_->id()});
std::vector<RtpEncodingParameters> init_encodings(2);
init_encodings[0].max_bitrate_bps = 60000;
init_encodings[1].max_bitrate_bps = 900000;
video_rtp_sender_->set_init_send_encodings(init_encodings);
RtpParameters params = video_rtp_sender_->GetParameters();
ASSERT_EQ(2u, params.encodings.size());
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 60000);
EXPECT_EQ(params.encodings[1].max_bitrate_bps, 900000);
// Simulate the setLocalDescription call
std::vector<uint32_t> ssrcs;
ssrcs.reserve(2);
for (int i = 0; i < 2; ++i)
ssrcs.push_back(kVideoSsrcSimulcast + i);
cricket::StreamParams stream_params =
cricket::CreateSimStreamParams("cname", ssrcs);
video_media_channel()->AddSendStream(stream_params);
video_rtp_sender_->SetMediaChannel(
video_media_channel()->AsVideoSendChannel());
video_rtp_sender_->SetSsrc(kVideoSsrcSimulcast);
params = video_rtp_sender_->GetParameters();
ASSERT_EQ(2u, params.encodings.size());
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 60000);
EXPECT_EQ(params.encodings[1].max_bitrate_bps, 900000);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest,
VideoSenderInitParametersMovedAfterManualSimulcastAndNegotiation) {
AddVideoTrack(false);
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
std::make_unique<MockSetStreamsObserver>();
video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, video_track_->id(),
set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_.get()));
EXPECT_CALL(*set_streams_observer, OnSetStreams());
video_rtp_sender_->SetStreams({local_stream_->id()});
std::vector<RtpEncodingParameters> init_encodings(1);
init_encodings[0].max_bitrate_bps = 60000;
video_rtp_sender_->set_init_send_encodings(init_encodings);
RtpParameters params = video_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 60000);
// Simulate the setLocalDescription call as if the user used SDP munging
// to enable simulcast
std::vector<uint32_t> ssrcs;
ssrcs.reserve(2);
for (int i = 0; i < 2; ++i)
ssrcs.push_back(kVideoSsrcSimulcast + i);
cricket::StreamParams stream_params =
cricket::CreateSimStreamParams("cname", ssrcs);
video_media_channel()->AddSendStream(stream_params);
video_rtp_sender_->SetMediaChannel(
video_media_channel()->AsVideoSendChannel());
video_rtp_sender_->SetSsrc(kVideoSsrcSimulcast);
params = video_rtp_sender_->GetParameters();
ASSERT_EQ(2u, params.encodings.size());
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 60000);
DestroyVideoRtpSender();
}
#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
using RtpSenderReceiverDeathTest = RtpSenderReceiverTest;
TEST_F(RtpSenderReceiverDeathTest,
VideoSenderManualRemoveSimulcastFailsDeathTest) {
AddVideoTrack(false);
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
std::make_unique<MockSetStreamsObserver>();
video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, video_track_->id(),
set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_.get()));
EXPECT_CALL(*set_streams_observer, OnSetStreams());
video_rtp_sender_->SetStreams({local_stream_->id()});
std::vector<RtpEncodingParameters> init_encodings(2);
init_encodings[0].max_bitrate_bps = 60000;
init_encodings[1].max_bitrate_bps = 120000;
video_rtp_sender_->set_init_send_encodings(init_encodings);
RtpParameters params = video_rtp_sender_->GetParameters();
ASSERT_EQ(2u, params.encodings.size());
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 60000);
// Simulate the setLocalDescription call as if the user used SDP munging
// to disable simulcast.
std::vector<uint32_t> ssrcs;
ssrcs.reserve(2);
for (int i = 0; i < 2; ++i)
ssrcs.push_back(kVideoSsrcSimulcast + i);
cricket::StreamParams stream_params =
cricket::StreamParams::CreateLegacy(kVideoSsrc);
video_media_channel()->AddSendStream(stream_params);
video_rtp_sender_->SetMediaChannel(
video_media_channel()->AsVideoSendChannel());
EXPECT_DEATH(video_rtp_sender_->SetSsrc(kVideoSsrcSimulcast), "");
}
#endif
TEST_F(RtpSenderReceiverTest,
VideoSenderMustCallGetParametersBeforeSetParametersBeforeNegotiation) {
video_rtp_sender_ =
VideoRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
RtpParameters params;
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest,
VideoSenderMustCallGetParametersBeforeSetParameters) {
CreateVideoRtpSender();
RtpParameters params;
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest,
VideoSenderSetParametersInvalidatesTransactionId) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest,
VideoSenderSetParametersAsyncInvalidatesTransactionId) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
absl::optional<webrtc::RTCError> result;
video_rtp_sender_->SetParametersAsync(
params, [&result](webrtc::RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
video_rtp_sender_->SetParametersAsync(
params, [&result](webrtc::RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_EQ(RTCErrorType::INVALID_STATE, result->type());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderDetectTransactionIdModification) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
params.transaction_id = "";
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderCheckTransactionIdRefresh) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_NE(params.transaction_id.size(), 0U);
auto saved_transaction_id = params.transaction_id;
params = video_rtp_sender_->GetParameters();
EXPECT_NE(saved_transaction_id, params.transaction_id);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderSetParametersOldValueFail) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
RtpParameters second_params = video_rtp_sender_->GetParameters();
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderCantSetUnimplementedRtpParameters) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
// Unimplemented RtpParameters: mid
params.mid = "dummy_mid";
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
params = video_rtp_sender_->GetParameters();
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderCanSetScaleResolutionDownBy) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
params.encodings[0].scale_resolution_down_by = 2;
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
params = video_rtp_sender_->GetParameters();
EXPECT_EQ(2, params.encodings[0].scale_resolution_down_by);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderDetectInvalidScaleResolutionDownBy) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
params.encodings[0].scale_resolution_down_by = 0.5;
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_RANGE, result.type());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderCanSetNumTemporalLayers) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
params.encodings[0].num_temporal_layers = 2;
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
params = video_rtp_sender_->GetParameters();
EXPECT_EQ(2, params.encodings[0].num_temporal_layers);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderDetectInvalidNumTemporalLayers) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
params.encodings[0].num_temporal_layers = webrtc::kMaxTemporalStreams + 1;
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_RANGE, result.type());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderCanSetMaxFramerate) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
params.encodings[0].max_framerate = 20;
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
params = video_rtp_sender_->GetParameters();
EXPECT_EQ(20., params.encodings[0].max_framerate);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderCanSetMaxFramerateZero) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
params.encodings[0].max_framerate = 0.;
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
params = video_rtp_sender_->GetParameters();
EXPECT_EQ(0., params.encodings[0].max_framerate);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderDetectInvalidMaxFramerate) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
params.encodings[0].max_framerate = -5.;
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_RANGE, result.type());
DestroyVideoRtpSender();
}
// A video sender can have multiple simulcast layers, in which case it will
// contain multiple RtpEncodingParameters. This tests that if this is the case
// (simulcast), then we can't set the bitrate_priority, or max_bitrate_bps
// for any encodings besides at index 0, because these are both implemented
// "per-sender."
TEST_F(RtpSenderReceiverTest, VideoSenderCantSetPerSenderEncodingParameters) {
// Add a simulcast specific send stream that contains 2 encoding parameters.
CreateVideoRtpSenderWithSimulcast();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(kVideoSimulcastLayerCount, params.encodings.size());
params.encodings[1].bitrate_priority = 2.0;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
params = video_rtp_sender_->GetParameters();
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderCantSetReadOnlyEncodingParameters) {
// Add a simulcast specific send stream that contains 2 encoding parameters.
CreateVideoRtpSenderWithSimulcast();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(kVideoSimulcastLayerCount, params.encodings.size());
for (size_t i = 0; i < params.encodings.size(); i++) {
params.encodings[i].ssrc = 1337;
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION,
video_rtp_sender_->SetParameters(params).type());
params = video_rtp_sender_->GetParameters();
}
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrate) {
CreateVideoRtpSender();
EXPECT_EQ(-1, video_media_channel()->max_bps());
webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
EXPECT_FALSE(params.encodings[0].min_bitrate_bps);
EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
params.encodings[0].min_bitrate_bps = 100;
params.encodings[0].max_bitrate_bps = 1000;
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
// Read back the parameters and verify they have been changed.
params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
EXPECT_EQ(100, params.encodings[0].min_bitrate_bps);
EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
// Verify that the video channel received the new parameters.
params = video_media_channel()->GetRtpSendParameters(kVideoSsrc);
EXPECT_EQ(1U, params.encodings.size());
EXPECT_EQ(100, params.encodings[0].min_bitrate_bps);
EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
// Verify that the global bitrate limit has not been changed.
EXPECT_EQ(-1, video_media_channel()->max_bps());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrateSimulcast) {
// Add a simulcast specific send stream that contains 2 encoding parameters.
CreateVideoRtpSenderWithSimulcast();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(kVideoSimulcastLayerCount, params.encodings.size());
params.encodings[0].min_bitrate_bps = 100;
params.encodings[0].max_bitrate_bps = 1000;
params.encodings[1].min_bitrate_bps = 200;
params.encodings[1].max_bitrate_bps = 2000;
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
// Verify that the video channel received the new parameters.
params = video_media_channel()->GetRtpSendParameters(kVideoSsrcSimulcast);
EXPECT_EQ(kVideoSimulcastLayerCount, params.encodings.size());
EXPECT_EQ(100, params.encodings[0].min_bitrate_bps);
EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
EXPECT_EQ(200, params.encodings[1].min_bitrate_bps);
EXPECT_EQ(2000, params.encodings[1].max_bitrate_bps);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, SetVideoBitratePriority) {
CreateVideoRtpSender();
webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
EXPECT_EQ(webrtc::kDefaultBitratePriority,
params.encodings[0].bitrate_priority);
double new_bitrate_priority = 2.0;
params.encodings[0].bitrate_priority = new_bitrate_priority;
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
params = video_media_channel()->GetRtpSendParameters(kVideoSsrc);
EXPECT_EQ(1U, params.encodings.size());
EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoReceiverCanGetParametersWithSimulcast) {
CreateVideoRtpReceiverWithSimulcast({}, 2);
RtpParameters params = video_rtp_receiver_->GetParameters();
EXPECT_EQ(2u, params.encodings.size());
DestroyVideoRtpReceiver();
}
TEST_F(RtpSenderReceiverTest, GenerateKeyFrameWithAudio) {
CreateAudioRtpSender();
auto error = audio_rtp_sender_->GenerateKeyFrame({});
EXPECT_FALSE(error.ok());
EXPECT_EQ(error.type(), RTCErrorType::UNSUPPORTED_OPERATION);
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, GenerateKeyFrameWithVideo) {
CreateVideoRtpSenderWithSimulcast({"1", "2", "3"});
auto error = video_rtp_sender_->GenerateKeyFrame({});
EXPECT_TRUE(error.ok());
error = video_rtp_sender_->GenerateKeyFrame({"1"});
EXPECT_TRUE(error.ok());
error = video_rtp_sender_->GenerateKeyFrame({""});
EXPECT_FALSE(error.ok());
EXPECT_EQ(error.type(), RTCErrorType::INVALID_PARAMETER);
error = video_rtp_sender_->GenerateKeyFrame({"no-such-rid"});
EXPECT_FALSE(error.ok());
EXPECT_EQ(error.type(), RTCErrorType::INVALID_PARAMETER);
DestroyVideoRtpSender();
}
// Test that makes sure that a video track content hint translates to the proper
// value for sources that are not screencast.
TEST_F(RtpSenderReceiverTest, PropagatesVideoTrackContentHint) {
CreateVideoRtpSender();
video_track_->set_enabled(true);
// `video_track_` is not screencast by default.
EXPECT_EQ(false, video_media_channel()->options().is_screencast);
// No content hint should be set by default.
EXPECT_EQ(VideoTrackInterface::ContentHint::kNone,
video_track_->content_hint());
// Setting detailed should turn a non-screencast source into screencast mode.
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);
EXPECT_EQ(true, video_media_channel()->options().is_screencast);
// Removing the content hint should turn the track back into non-screencast
// mode.
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kNone);
EXPECT_EQ(false, video_media_channel()->options().is_screencast);
// Setting fluid should remain in non-screencast mode (its default).
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kFluid);
EXPECT_EQ(false, video_media_channel()->options().is_screencast);
// Setting text should have the same effect as Detailed
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kText);
EXPECT_EQ(true, video_media_channel()->options().is_screencast);
DestroyVideoRtpSender();
}
// Test that makes sure that a video track content hint translates to the proper
// value for screencast sources.
TEST_F(RtpSenderReceiverTest,
PropagatesVideoTrackContentHintForScreencastSource) {
CreateVideoRtpSender(true);
video_track_->set_enabled(true);
// `video_track_` with a screencast source should be screencast by default.
EXPECT_EQ(true, video_media_channel()->options().is_screencast);
// No content hint should be set by default.
EXPECT_EQ(VideoTrackInterface::ContentHint::kNone,
video_track_->content_hint());
// Setting fluid should turn a screencast source into non-screencast mode.
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kFluid);
EXPECT_EQ(false, video_media_channel()->options().is_screencast);
// Removing the content hint should turn the track back into screencast mode.
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kNone);
EXPECT_EQ(true, video_media_channel()->options().is_screencast);
// Setting detailed should still remain in screencast mode (its default).
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);
EXPECT_EQ(true, video_media_channel()->options().is_screencast);
// Setting text should have the same effect as Detailed
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kText);
EXPECT_EQ(true, video_media_channel()->options().is_screencast);
DestroyVideoRtpSender();
}
// Test that makes sure any content hints that are set on a track before
// VideoRtpSender is ready to send are still applied when it gets ready to send.
TEST_F(RtpSenderReceiverTest,
PropagatesVideoTrackContentHintSetBeforeEnabling) {
AddVideoTrack();
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
std::make_unique<MockSetStreamsObserver>();
// Setting detailed overrides the default non-screencast mode. This should be
// applied even if the track is set on construction.
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);
video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, video_track_->id(),
set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_.get()));
EXPECT_CALL(*set_streams_observer, OnSetStreams());
video_rtp_sender_->SetStreams({local_stream_->id()});
video_rtp_sender_->SetMediaChannel(
video_media_channel()->AsVideoSendChannel());
video_track_->set_enabled(true);
// Sender is not ready to send (no SSRC) so no option should have been set.
EXPECT_EQ(absl::nullopt, video_media_channel()->options().is_screencast);
// Verify that the content hint is accounted for when video_rtp_sender_ does
// get enabled.
video_rtp_sender_->SetSsrc(kVideoSsrc);
EXPECT_EQ(true, video_media_channel()->options().is_screencast);
// And removing the hint should go back to false (to verify that false was
// default correctly).
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kNone);
EXPECT_EQ(false, video_media_channel()->options().is_screencast);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioSenderHasDtmfSender) {
CreateAudioRtpSender();
EXPECT_NE(nullptr, audio_rtp_sender_->GetDtmfSender());
}
TEST_F(RtpSenderReceiverTest, VideoSenderDoesNotHaveDtmfSender) {
CreateVideoRtpSender();
EXPECT_EQ(nullptr, video_rtp_sender_->GetDtmfSender());
}
// Test that the DTMF sender is really using `voice_channel_`, and thus returns
// true/false from CanSendDtmf based on what `voice_channel_` returns.
TEST_F(RtpSenderReceiverTest, CanInsertDtmf) {
AddDtmfCodec();
CreateAudioRtpSender();
auto dtmf_sender = audio_rtp_sender_->GetDtmfSender();
ASSERT_NE(nullptr, dtmf_sender);
EXPECT_TRUE(dtmf_sender->CanInsertDtmf());
}
TEST_F(RtpSenderReceiverTest, CanNotInsertDtmf) {
CreateAudioRtpSender();
auto dtmf_sender = audio_rtp_sender_->GetDtmfSender();
ASSERT_NE(nullptr, dtmf_sender);
// DTMF codec has not been added, as it was in the above test.
EXPECT_FALSE(dtmf_sender->CanInsertDtmf());
}
TEST_F(RtpSenderReceiverTest, InsertDtmf) {
AddDtmfCodec();
CreateAudioRtpSender();
auto dtmf_sender = audio_rtp_sender_->GetDtmfSender();
ASSERT_NE(nullptr, dtmf_sender);
EXPECT_EQ(0U, voice_media_channel()->dtmf_info_queue().size());
// Insert DTMF
const int expected_duration = 90;
dtmf_sender->InsertDtmf("012", expected_duration, 100);
// Verify
ASSERT_EQ_WAIT(3U, voice_media_channel()->dtmf_info_queue().size(),
kDefaultTimeout);
const uint32_t send_ssrc =
voice_media_channel()->send_streams()[0].first_ssrc();
EXPECT_TRUE(CompareDtmfInfo(voice_media_channel()->dtmf_info_queue()[0],
send_ssrc, 0, expected_duration));
EXPECT_TRUE(CompareDtmfInfo(voice_media_channel()->dtmf_info_queue()[1],
send_ssrc, 1, expected_duration));
EXPECT_TRUE(CompareDtmfInfo(voice_media_channel()->dtmf_info_queue()[2],
send_ssrc, 2, expected_duration));
}
// Validate that the default FrameEncryptor setting is nullptr.
TEST_F(RtpSenderReceiverTest, AudioSenderCanSetFrameEncryptor) {
CreateAudioRtpSender();
rtc::scoped_refptr<FrameEncryptorInterface> fake_frame_encryptor(
new FakeFrameEncryptor());
EXPECT_EQ(nullptr, audio_rtp_sender_->GetFrameEncryptor());
audio_rtp_sender_->SetFrameEncryptor(fake_frame_encryptor);
EXPECT_EQ(fake_frame_encryptor.get(),
audio_rtp_sender_->GetFrameEncryptor().get());
}
// Validate that setting a FrameEncryptor after the send stream is stopped does
// nothing.
TEST_F(RtpSenderReceiverTest, AudioSenderCannotSetFrameEncryptorAfterStop) {
CreateAudioRtpSender();
rtc::scoped_refptr<FrameEncryptorInterface> fake_frame_encryptor(
new FakeFrameEncryptor());
EXPECT_EQ(nullptr, audio_rtp_sender_->GetFrameEncryptor());
audio_rtp_sender_->Stop();
audio_rtp_sender_->SetFrameEncryptor(fake_frame_encryptor);
// TODO(webrtc:9926) - Validate media channel not set once fakes updated.
}
// Validate that the default FrameEncryptor setting is nullptr.
TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetFrameDecryptor) {
CreateAudioRtpReceiver();
rtc::scoped_refptr<FrameDecryptorInterface> fake_frame_decryptor(
rtc::make_ref_counted<FakeFrameDecryptor>());
EXPECT_EQ(nullptr, audio_rtp_receiver_->GetFrameDecryptor());
audio_rtp_receiver_->SetFrameDecryptor(fake_frame_decryptor);
EXPECT_EQ(fake_frame_decryptor.get(),
audio_rtp_receiver_->GetFrameDecryptor().get());
DestroyAudioRtpReceiver();
}
// Validate that the default FrameEncryptor setting is nullptr.
TEST_F(RtpSenderReceiverTest, AudioReceiverCannotSetFrameDecryptorAfterStop) {
CreateAudioRtpReceiver();
rtc::scoped_refptr<FrameDecryptorInterface> fake_frame_decryptor(
rtc::make_ref_counted<FakeFrameDecryptor>());
EXPECT_EQ(nullptr, audio_rtp_receiver_->GetFrameDecryptor());
audio_rtp_receiver_->SetMediaChannel(nullptr);
audio_rtp_receiver_->SetFrameDecryptor(fake_frame_decryptor);
// TODO(webrtc:9926) - Validate media channel not set once fakes updated.
DestroyAudioRtpReceiver();
}
// Validate that the default FrameEncryptor setting is nullptr.
TEST_F(RtpSenderReceiverTest, VideoSenderCanSetFrameEncryptor) {
CreateVideoRtpSender();
rtc::scoped_refptr<FrameEncryptorInterface> fake_frame_encryptor(
new FakeFrameEncryptor());
EXPECT_EQ(nullptr, video_rtp_sender_->GetFrameEncryptor());
video_rtp_sender_->SetFrameEncryptor(fake_frame_encryptor);
EXPECT_EQ(fake_frame_encryptor.get(),
video_rtp_sender_->GetFrameEncryptor().get());
}
// Validate that setting a FrameEncryptor after the send stream is stopped does
// nothing.
TEST_F(RtpSenderReceiverTest, VideoSenderCannotSetFrameEncryptorAfterStop) {
CreateVideoRtpSender();
rtc::scoped_refptr<FrameEncryptorInterface> fake_frame_encryptor(
new FakeFrameEncryptor());
EXPECT_EQ(nullptr, video_rtp_sender_->GetFrameEncryptor());
video_rtp_sender_->Stop();
video_rtp_sender_->SetFrameEncryptor(fake_frame_encryptor);
// TODO(webrtc:9926) - Validate media channel not set once fakes updated.
}
// Validate that the default FrameEncryptor setting is nullptr.
TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetFrameDecryptor) {
CreateVideoRtpReceiver();
rtc::scoped_refptr<FrameDecryptorInterface> fake_frame_decryptor(
rtc::make_ref_counted<FakeFrameDecryptor>());
EXPECT_EQ(nullptr, video_rtp_receiver_->GetFrameDecryptor());
video_rtp_receiver_->SetFrameDecryptor(fake_frame_decryptor);
EXPECT_EQ(fake_frame_decryptor.get(),
video_rtp_receiver_->GetFrameDecryptor().get());
DestroyVideoRtpReceiver();
}
// Validate that the default FrameEncryptor setting is nullptr.
TEST_F(RtpSenderReceiverTest, VideoReceiverCannotSetFrameDecryptorAfterStop) {
CreateVideoRtpReceiver();
rtc::scoped_refptr<FrameDecryptorInterface> fake_frame_decryptor(
rtc::make_ref_counted<FakeFrameDecryptor>());
EXPECT_EQ(nullptr, video_rtp_receiver_->GetFrameDecryptor());
video_rtp_receiver_->SetMediaChannel(nullptr);
video_rtp_receiver_->SetFrameDecryptor(fake_frame_decryptor);
// TODO(webrtc:9926) - Validate media channel not set once fakes updated.
DestroyVideoRtpReceiver();
}
// Checks that calling the internal methods for get/set parameters do not
// invalidate any parameters retreived by clients.
TEST_F(RtpSenderReceiverTest,
InternalParameterMethodsDoNotInvalidateTransaction) {
CreateVideoRtpSender();
RtpParameters parameters = video_rtp_sender_->GetParameters();
RtpParameters new_parameters = video_rtp_sender_->GetParametersInternal();
new_parameters.encodings[0].active = false;
video_rtp_sender_->SetParametersInternal(new_parameters, nullptr, true);
new_parameters.encodings[0].active = true;
video_rtp_sender_->SetParametersInternal(new_parameters, nullptr, true);
parameters.encodings[0].active = false;
EXPECT_TRUE(video_rtp_sender_->SetParameters(parameters).ok());
}
// Checks that the senders SetStreams eliminates duplicate stream ids.
TEST_F(RtpSenderReceiverTest, SenderSetStreamsEliminatesDuplicateIds) {
AddVideoTrack();
video_rtp_sender_ =
VideoRtpSender::Create(worker_thread_, video_track_->id(), nullptr);
video_rtp_sender_->SetStreams({"1", "2", "1"});
EXPECT_EQ(video_rtp_sender_->stream_ids().size(), 2u);
}
// Helper method for syntactic sugar for accepting a vector with '{}' notation.
std::pair<RidList, RidList> CreatePairOfRidVectors(
const std::vector<std::string>& first,
const std::vector<std::string>& second) {
return std::make_pair(first, second);
}
// These parameters are used to test disabling simulcast layers.
const std::pair<RidList, RidList> kDisableSimulcastLayersParameters[] = {
// Tests removing the first layer. This is a special case because
// the first layer's SSRC is also the 'primary' SSRC used to associate the
// parameters to the media channel.
CreatePairOfRidVectors({"1", "2", "3", "4"}, {"1"}),
// Tests removing some layers.
CreatePairOfRidVectors({"1", "2", "3", "4"}, {"2", "4"}),
// Tests simulcast rejected scenario all layers except first are rejected.
CreatePairOfRidVectors({"1", "2", "3", "4"}, {"2", "3", "4"}),
// Tests removing all layers.
CreatePairOfRidVectors({"1", "2", "3", "4"}, {"1", "2", "3", "4"}),
};
// Runs test for disabling layers on a sender without a media engine set.
TEST_P(RtpSenderReceiverTest, DisableSimulcastLayersWithoutMediaEngine) {
auto parameter = GetParam();
RunDisableSimulcastLayersWithoutMediaEngineTest(parameter.first,
parameter.second);
}
// Runs test for disabling layers on a sender with a media engine set.
TEST_P(RtpSenderReceiverTest, DisableSimulcastLayersWithMediaEngine) {
auto parameter = GetParam();
RunDisableSimulcastLayersWithMediaEngineTest(parameter.first,
parameter.second);
}
INSTANTIATE_TEST_SUITE_P(
DisableSimulcastLayersInSender,
RtpSenderReceiverTest,
::testing::ValuesIn(kDisableSimulcastLayersParameters));
} // namespace webrtc
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