Add tests for verifying transport feedback for audio and video.
BUG=webrtc:5263
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1589523002 .
Cr-Commit-Position: refs/heads/master@{#11255}
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index 850e487..e9651e3 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -31,7 +31,7 @@
audio_send_config_(nullptr),
audio_send_stream_(nullptr),
fake_encoder_(clock_),
- num_video_streams_(0),
+ num_video_streams_(1),
num_audio_streams_(0),
fake_send_audio_device_(nullptr),
fake_recv_audio_device_(nullptr) {}
@@ -60,9 +60,9 @@
}
CreateReceiverCall(recv_config);
}
+ test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
send_transport_.reset(test->CreateSendTransport(sender_call_.get()));
receive_transport_.reset(test->CreateReceiveTransport());
- test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
if (test->ShouldCreateReceivers()) {
send_transport_->SetReceiver(receiver_call_->Receiver());
@@ -97,8 +97,10 @@
test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
}
- CreateFrameGeneratorCapturer();
- test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
+ if (num_video_streams_ > 0) {
+ CreateFrameGeneratorCapturer();
+ test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
+ }
Start();
test->PerformTest();
@@ -179,17 +181,20 @@
RTC_DCHECK(num_video_streams <= kNumSsrcs);
RTC_DCHECK_LE(num_audio_streams, 1u);
RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
- video_send_config_ = VideoSendStream::Config(send_transport);
- video_send_config_.encoder_settings.encoder = &fake_encoder_;
- video_send_config_.encoder_settings.payload_name = "FAKE";
- video_send_config_.encoder_settings.payload_type = kFakeVideoSendPayloadType;
- video_send_config_.rtp.extensions.push_back(
- RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
- video_encoder_config_.streams = test::CreateVideoStreams(num_video_streams);
- for (size_t i = 0; i < num_video_streams; ++i)
- video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
- video_send_config_.rtp.extensions.push_back(
- RtpExtension(RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
+ if (num_video_streams > 0) {
+ video_send_config_ = VideoSendStream::Config(send_transport);
+ video_send_config_.encoder_settings.encoder = &fake_encoder_;
+ video_send_config_.encoder_settings.payload_name = "FAKE";
+ video_send_config_.encoder_settings.payload_type =
+ kFakeVideoSendPayloadType;
+ video_send_config_.rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
+ video_encoder_config_.streams = test::CreateVideoStreams(num_video_streams);
+ for (size_t i = 0; i < num_video_streams; ++i)
+ video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
+ video_send_config_.rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
+ }
if (num_audio_streams > 0) {
audio_send_config_ = AudioSendStream::Config(send_transport);
@@ -199,27 +204,29 @@
}
void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
- RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
RTC_DCHECK(video_receive_configs_.empty());
RTC_DCHECK(allocated_decoders_.empty());
- RTC_DCHECK(num_audio_streams_ == 0 || voe_send_.channel_id >= 0);
- VideoReceiveStream::Config video_config(rtcp_send_transport);
- video_config.rtp.remb = true;
- video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
- for (const RtpExtension& extension : video_send_config_.rtp.extensions)
- video_config.rtp.extensions.push_back(extension);
- for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
- VideoReceiveStream::Decoder decoder =
- test::CreateMatchingDecoder(video_send_config_.encoder_settings);
- allocated_decoders_.push_back(decoder.decoder);
- video_config.decoders.clear();
- video_config.decoders.push_back(decoder);
- video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
- video_receive_configs_.push_back(video_config);
+ if (num_video_streams_ > 0) {
+ RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
+ VideoReceiveStream::Config video_config(rtcp_send_transport);
+ video_config.rtp.remb = true;
+ video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
+ for (const RtpExtension& extension : video_send_config_.rtp.extensions)
+ video_config.rtp.extensions.push_back(extension);
+ for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
+ VideoReceiveStream::Decoder decoder =
+ test::CreateMatchingDecoder(video_send_config_.encoder_settings);
+ allocated_decoders_.push_back(decoder.decoder);
+ video_config.decoders.clear();
+ video_config.decoders.push_back(decoder);
+ video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
+ video_receive_configs_.push_back(video_config);
+ }
}
RTC_DCHECK(num_audio_streams_ <= 1);
if (num_audio_streams_ == 1) {
+ RTC_DCHECK(voe_send_.channel_id >= 0);
AudioReceiveStream::Config audio_config;
audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
audio_config.rtcp_send_transport = rtcp_send_transport;
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 51d1d2c..48dc3e8 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -1542,82 +1542,115 @@
tester.RunTest();
}
-void TransportFeedbackTest(bool feedback_enabled) {
+class TransportFeedbackTester : public test::EndToEndTest {
+ public:
+ explicit TransportFeedbackTester(bool feedback_enabled,
+ size_t num_video_streams,
+ size_t num_audio_streams)
+ : EndToEndTest(::webrtc::EndToEndTest::kDefaultTimeoutMs),
+ feedback_enabled_(feedback_enabled),
+ num_video_streams_(num_video_streams),
+ num_audio_streams_(num_audio_streams) {
+ // Only one stream of each supported for now.
+ EXPECT_LE(num_video_streams, 1u);
+ EXPECT_LE(num_audio_streams, 1u);
+ }
+
+ protected:
+ Action OnSendRtcp(const uint8_t* data, size_t length) override {
+ EXPECT_FALSE(HasTransportFeedback(data, length));
+ return SEND_PACKET;
+ }
+
+ Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
+ if (HasTransportFeedback(data, length))
+ observation_complete_.Set();
+ return SEND_PACKET;
+ }
+
+ bool HasTransportFeedback(const uint8_t* data, size_t length) const {
+ RTCPUtility::RTCPParserV2 parser(data, length, true);
+ EXPECT_TRUE(parser.IsValid());
+
+ RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
+ while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
+ if (packet_type == RTCPUtility::RTCPPacketTypes::kTransportFeedback)
+ return true;
+ packet_type = parser.Iterate();
+ }
+
+ return false;
+ }
+
+ void PerformTest() override {
+ const int64_t kDisabledFeedbackTimeoutMs = 5000;
+ EXPECT_EQ(feedback_enabled_,
+ observation_complete_.Wait(feedback_enabled_
+ ? test::CallTest::kDefaultTimeoutMs
+ : kDisabledFeedbackTimeoutMs));
+ }
+
+ void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
+ receiver_call_ = receiver_call;
+ }
+
+ size_t GetNumVideoStreams() const override { return num_video_streams_; }
+ size_t GetNumAudioStreams() const override { return num_audio_streams_; }
+
+ void ModifyVideoConfigs(
+ VideoSendStream::Config* send_config,
+ std::vector<VideoReceiveStream::Config>* receive_configs,
+ VideoEncoderConfig* encoder_config) override {
+ send_config->rtp.extensions.clear();
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+ (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
+ (*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
+ }
+
+ void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ send_config->rtp.extensions.clear();
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+ (*receive_configs)[0].rtp.extensions.clear();
+ (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
+ (*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
+ (*receive_configs)[0].combined_audio_video_bwe = true;
+ }
+
+ private:
static const int kExtensionId = 5;
- class TransportFeedbackObserver : public test::DirectTransport {
- public:
- TransportFeedbackObserver(Call* receiver_call, rtc::Event* done_event)
- : DirectTransport(receiver_call), done_(done_event) {}
- virtual ~TransportFeedbackObserver() {}
+ const bool feedback_enabled_;
+ const size_t num_video_streams_;
+ const size_t num_audio_streams_;
+ Call* receiver_call_;
+};
- bool SendRtcp(const uint8_t* data, size_t length) override {
- RTCPUtility::RTCPParserV2 parser(data, length, true);
- EXPECT_TRUE(parser.IsValid());
-
- RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
- while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
- if (packet_type == RTCPUtility::RTCPPacketTypes::kTransportFeedback) {
- done_->Set();
- break;
- }
- packet_type = parser.Iterate();
- }
-
- return test::DirectTransport::SendRtcp(data, length);
- }
-
- rtc::Event* done_;
- };
-
- class TransportFeedbackTester : public MultiStreamTest {
- public:
- explicit TransportFeedbackTester(bool feedback_enabled)
- : feedback_enabled_(feedback_enabled), done_(false, false) {}
- virtual ~TransportFeedbackTester() {}
-
- protected:
- void Wait() override {
- const int64_t kDisabledFeedbackTimeoutMs = 5000;
- EXPECT_EQ(feedback_enabled_, done_.Wait(feedback_enabled_
- ? test::CallTest::kDefaultTimeoutMs
- : kDisabledFeedbackTimeoutMs));
- }
-
- void UpdateSendConfig(
- size_t stream_index,
- VideoSendStream::Config* send_config,
- VideoEncoderConfig* encoder_config,
- test::FrameGeneratorCapturer** frame_generator) override {
- send_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
- }
-
- void UpdateReceiveConfig(
- size_t stream_index,
- VideoReceiveStream::Config* receive_config) override {
- receive_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
- receive_config->rtp.transport_cc = feedback_enabled_;
- }
-
- test::DirectTransport* CreateReceiveTransport(
- Call* receiver_call) override {
- return new TransportFeedbackObserver(receiver_call, &done_);
- }
-
- private:
- const bool feedback_enabled_;
- rtc::Event done_;
- } tester(feedback_enabled);
- tester.RunTest();
+TEST_F(EndToEndTest, VideoReceivesTransportFeedback) {
+ TransportFeedbackTester test(true, 1, 0);
+ RunBaseTest(&test);
}
-TEST_F(EndToEndTest, ReceivesTransportFeedback) {
- TransportFeedbackTest(true);
+TEST_F(EndToEndTest, VideoTransportFeedbackNotConfigured) {
+ TransportFeedbackTester test(false, 1, 0);
+ RunBaseTest(&test);
}
-TEST_F(EndToEndTest, TransportFeedbackNotConfigured) {
- TransportFeedbackTest(false);
+TEST_F(EndToEndTest, AudioReceivesTransportFeedback) {
+ TransportFeedbackTester test(true, 0, 1);
+ RunBaseTest(&test);
+}
+
+TEST_F(EndToEndTest, AudioTransportFeedbackNotConfigured) {
+ TransportFeedbackTester test(false, 0, 1);
+ RunBaseTest(&test);
+}
+
+TEST_F(EndToEndTest, AudioVideoReceivesTransportFeedback) {
+ TransportFeedbackTester test(true, 1, 1);
+ RunBaseTest(&test);
}
TEST_F(EndToEndTest, ObserversEncodedFrames) {
