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Diffstat (limited to 'webrtc/call/bitrate_estimator_tests.cc')
-rw-r--r--webrtc/call/bitrate_estimator_tests.cc201
1 files changed, 94 insertions, 107 deletions
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc
index 685f3fd665..4b24bbd5ef 100644
--- a/webrtc/call/bitrate_estimator_tests.cc
+++ b/webrtc/call/bitrate_estimator_tests.cc
@@ -13,66 +13,54 @@
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/audio_state.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/call.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
-#include "webrtc/test/fake_voice_engine.h"
+#include "webrtc/test/mock_voice_engine.h"
#include "webrtc/test/frame_generator_capturer.h"
namespace webrtc {
namespace {
// Note: If you consider to re-use this class, think twice and instead consider
-// writing tests that don't depend on the trace system.
-class TraceObserver {
+// writing tests that don't depend on the logging system.
+class LogObserver {
public:
- TraceObserver() {
- Trace::set_level_filter(kTraceTerseInfo);
-
- Trace::CreateTrace();
- Trace::SetTraceCallback(&callback_);
-
- // Call webrtc trace to initialize the tracer that would otherwise trigger a
- // data-race if left to be initialized by multiple threads (i.e. threads
- // spawned by test::DirectTransport members in BitrateEstimatorTest).
- WEBRTC_TRACE(kTraceStateInfo,
- kTraceUtility,
- -1,
- "Instantiate without data races.");
- }
+ LogObserver() { rtc::LogMessage::AddLogToStream(&callback_, rtc::LS_INFO); }
- ~TraceObserver() {
- Trace::SetTraceCallback(nullptr);
- Trace::ReturnTrace();
- }
+ ~LogObserver() { rtc::LogMessage::RemoveLogToStream(&callback_); }
void PushExpectedLogLine(const std::string& expected_log_line) {
callback_.PushExpectedLogLine(expected_log_line);
}
- EventTypeWrapper Wait() {
- return callback_.Wait();
- }
+ bool Wait() { return callback_.Wait(); }
private:
- class Callback : public TraceCallback {
+ class Callback : public rtc::LogSink {
public:
- Callback() : done_(EventWrapper::Create()) {}
+ Callback() : done_(false, false) {}
- void Print(TraceLevel level, const char* message, int length) override {
+ void OnLogMessage(const std::string& message) override {
rtc::CritScope lock(&crit_sect_);
- std::string msg(message);
- if (msg.find("BitrateEstimator") != std::string::npos) {
- received_log_lines_.push_back(msg);
+ // Ignore log lines that are due to missing AST extensions, these are
+ // logged when we switch back from AST to TOF until the wrapping bitrate
+ // estimator gives up on using AST.
+ if (message.find("BitrateEstimator") != std::string::npos &&
+ message.find("packet is missing") == std::string::npos) {
+ received_log_lines_.push_back(message);
}
+
int num_popped = 0;
while (!received_log_lines_.empty() && !expected_log_lines_.empty()) {
std::string a = received_log_lines_.front();
@@ -80,19 +68,17 @@ class TraceObserver {
received_log_lines_.pop_front();
expected_log_lines_.pop_front();
num_popped++;
- EXPECT_TRUE(a.find(b) != std::string::npos);
+ EXPECT_TRUE(a.find(b) != std::string::npos) << a << " != " << b;
}
if (expected_log_lines_.size() <= 0) {
if (num_popped > 0) {
- done_->Set();
+ done_.Set();
}
return;
}
}
- EventTypeWrapper Wait() {
- return done_->Wait(test::CallTest::kDefaultTimeoutMs);
- }
+ bool Wait() { return done_.Wait(test::CallTest::kDefaultTimeoutMs); }
void PushExpectedLogLine(const std::string& expected_log_line) {
rtc::CritScope lock(&crit_sect_);
@@ -104,7 +90,7 @@ class TraceObserver {
rtc::CriticalSection crit_sect_;
Strings received_log_lines_ GUARDED_BY(crit_sect_);
Strings expected_log_lines_ GUARDED_BY(crit_sect_);
- rtc::scoped_ptr<EventWrapper> done_;
+ rtc::Event done_;
};
Callback callback_;
@@ -118,13 +104,13 @@ class BitrateEstimatorTest : public test::CallTest {
public:
BitrateEstimatorTest() : receive_config_(nullptr) {}
- virtual ~BitrateEstimatorTest() {
- EXPECT_TRUE(streams_.empty());
- }
+ virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
virtual void SetUp() {
+ AudioState::Config audio_state_config;
+ audio_state_config.voice_engine = &mock_voice_engine_;
Call::Config config;
- config.voice_engine = &fake_voice_engine_;
+ config.audio_state = AudioState::Create(audio_state_config);
receiver_call_.reset(Call::Create(config));
sender_call_.reset(Call::Create(config));
@@ -133,18 +119,19 @@ class BitrateEstimatorTest : public test::CallTest {
receive_transport_.reset(new test::DirectTransport(receiver_call_.get()));
receive_transport_->SetReceiver(sender_call_->Receiver());
- send_config_ = VideoSendStream::Config(send_transport_.get());
- send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
+ video_send_config_ = VideoSendStream::Config(send_transport_.get());
+ video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
// Encoders will be set separately per stream.
- send_config_.encoder_settings.encoder = nullptr;
- send_config_.encoder_settings.payload_name = "FAKE";
- send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
- encoder_config_.streams = test::CreateVideoStreams(1);
+ video_send_config_.encoder_settings.encoder = nullptr;
+ video_send_config_.encoder_settings.payload_name = "FAKE";
+ video_send_config_.encoder_settings.payload_type =
+ kFakeVideoSendPayloadType;
+ video_encoder_config_.streams = test::CreateVideoStreams(1);
receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
// receive_config_.decoders will be set by every stream separately.
- receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0];
- receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
+ receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
+ receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
receive_config_.rtp.remb = true;
receive_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
@@ -154,7 +141,7 @@ class BitrateEstimatorTest : public test::CallTest {
virtual void TearDown() {
std::for_each(streams_.begin(), streams_.end(),
- std::mem_fun(&Stream::StopSending));
+ std::mem_fun(&Stream::StopSending));
send_transport_->StopSending();
receive_transport_->StopSending();
@@ -165,6 +152,7 @@ class BitrateEstimatorTest : public test::CallTest {
}
receiver_call_.reset();
+ sender_call_.reset();
}
protected:
@@ -181,23 +169,21 @@ class BitrateEstimatorTest : public test::CallTest {
frame_generator_capturer_(),
fake_encoder_(Clock::GetRealTimeClock()),
fake_decoder_() {
- test_->send_config_.rtp.ssrcs[0]++;
- test_->send_config_.encoder_settings.encoder = &fake_encoder_;
+ test_->video_send_config_.rtp.ssrcs[0]++;
+ test_->video_send_config_.encoder_settings.encoder = &fake_encoder_;
send_stream_ = test_->sender_call_->CreateVideoSendStream(
- test_->send_config_, test_->encoder_config_);
- RTC_DCHECK_EQ(1u, test_->encoder_config_.streams.size());
+ test_->video_send_config_, test_->video_encoder_config_);
+ RTC_DCHECK_EQ(1u, test_->video_encoder_config_.streams.size());
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
- send_stream_->Input(),
- test_->encoder_config_.streams[0].width,
- test_->encoder_config_.streams[0].height,
- 30,
+ send_stream_->Input(), test_->video_encoder_config_.streams[0].width,
+ test_->video_encoder_config_.streams[0].height, 30,
Clock::GetRealTimeClock()));
send_stream_->Start();
frame_generator_capturer_->Start();
if (receive_audio) {
AudioReceiveStream::Config receive_config;
- receive_config.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
+ receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0];
// Bogus non-default id to prevent hitting a RTC_DCHECK when creating
// the AudioReceiveStream. Every receive stream has to correspond to
// an underlying channel id.
@@ -211,12 +197,13 @@ class BitrateEstimatorTest : public test::CallTest {
VideoReceiveStream::Decoder decoder;
decoder.decoder = &fake_decoder_;
decoder.payload_type =
- test_->send_config_.encoder_settings.payload_type;
+ test_->video_send_config_.encoder_settings.payload_type;
decoder.payload_name =
- test_->send_config_.encoder_settings.payload_name;
+ test_->video_send_config_.encoder_settings.payload_name;
+ test_->receive_config_.decoders.clear();
test_->receive_config_.decoders.push_back(decoder);
test_->receive_config_.rtp.remote_ssrc =
- test_->send_config_.rtp.ssrcs[0];
+ test_->video_send_config_.rtp.ssrcs[0];
test_->receive_config_.rtp.local_ssrc++;
video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
test_->receive_config_);
@@ -262,8 +249,8 @@ class BitrateEstimatorTest : public test::CallTest {
test::FakeDecoder fake_decoder_;
};
- test::FakeVoiceEngine fake_voice_engine_;
- TraceObserver receiver_trace_;
+ testing::NiceMock<test::MockVoiceEngine> mock_voice_engine_;
+ LogObserver receiver_log_;
rtc::scoped_ptr<test::DirectTransport> send_transport_;
rtc::scoped_ptr<test::DirectTransport> receive_transport_;
rtc::scoped_ptr<Call> sender_call_;
@@ -278,89 +265,89 @@ static const char* kSingleStreamLog =
"RemoteBitrateEstimatorSingleStream: Instantiating.";
TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForAudio) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, true));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToASTForAudio) {
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, true));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, true));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
- send_config_.rtp.extensions[0] =
+ video_send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToASTThenBackToTOFForVideo) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
- send_config_.rtp.extensions[0] =
+ video_send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
- send_config_.rtp.extensions[0] =
+ video_send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
- receiver_trace_.PushExpectedLogLine(
+ receiver_log_.PushExpectedLogLine(
"WrappingBitrateEstimator: Switching to transmission time offset RBE.");
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
streams_[0]->StopSending();
streams_[1]->StopSending();
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
}
} // namespace webrtc
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-13 05:00:05 +0000