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Unified Diff: webrtc/video/rampup_tests.cc

Issue 1362923002: Revert of Wire up send-side bandwidth estimation. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Manual merge of revert Created 5 years, 3 months ago
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Index: webrtc/video/rampup_tests.cc
diff --git a/webrtc/video/rampup_tests.cc b/webrtc/video/rampup_tests.cc
index fd60fc4d64e1cf7897a098f04d353cc6c050d3f5..d308f2ddb72619bd12145c2fd32ac4983b9de6f4 100644
--- a/webrtc/video/rampup_tests.cc
+++ b/webrtc/video/rampup_tests.cc
@@ -11,18 +11,14 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
-#include "webrtc/base/event.h"
-#include "webrtc/modules/pacing/include/packet_router.h"
#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
-#include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h"
#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "webrtc/video/rampup_tests.h"
@@ -74,22 +70,14 @@ StreamObserver::StreamObserver(const SsrcMap& rtx_media_ssrcs,
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
rtp_rtcp_->SetREMBStatus(true);
rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
- packet_router_.reset(new PacketRouter());
- packet_router_->AddRtpModule(rtp_rtcp_.get());
rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsSendTimeExtensionId);
rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
- rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
- kTransportSequenceNumberExtensionId);
payload_registry_->SetRtxPayloadType(RampUpTest::kSendRtxPayloadType,
RampUpTest::kFakeSendPayloadType);
}
-StreamObserver::~StreamObserver() {
- packet_router_->RemoveRtpModule(rtp_rtcp_.get());
-}
-
void StreamObserver::set_expected_bitrate_bps(
unsigned int expected_bitrate_bps) {
rtc::CritScope lock(&crit_);
@@ -175,10 +163,6 @@ void StreamObserver::SetRemoteBitrateEstimator(RemoteBitrateEstimator* rbe) {
remote_bitrate_estimator_.reset(rbe);
}
-PacketRouter* StreamObserver::GetPacketRouter() {
- return packet_router_.get();
-}
-
void StreamObserver::ReportResult(const std::string& measurement,
size_t value,
const std::string& units) {
@@ -387,48 +371,6 @@ EventTypeWrapper LowRateStreamObserver::Wait() {
return test_done_->Wait(test::CallTest::kLongTimeoutMs);
}
-class SendBitrateAdapter {
- public:
- static const int64_t kPollIntervalMs = 250;
-
- SendBitrateAdapter(const Call& call,
- const std::vector<uint32_t>& ssrcs,
- RemoteBitrateObserver* bitrate_observer)
- : event_(false, false),
- call_(call),
- ssrcs_(ssrcs),
- bitrate_observer_(bitrate_observer) {
- RTC_DCHECK(bitrate_observer != nullptr);
- poller_thread_ = ThreadWrapper::CreateThread(&SendBitrateAdapterThread,
- this, "SendBitratePoller");
- RTC_DCHECK(poller_thread_->Start());
- }
-
- virtual ~SendBitrateAdapter() {
- event_.Set();
- poller_thread_->Stop();
- }
-
- private:
- static bool SendBitrateAdapterThread(void* obj) {
- return static_cast<SendBitrateAdapter*>(obj)->PollStats();
- }
-
- bool PollStats() {
- Call::Stats stats = call_.GetStats();
-
- bitrate_observer_->OnReceiveBitrateChanged(ssrcs_,
- stats.send_bandwidth_bps);
- return !event_.Wait(kPollIntervalMs);
- }
-
- rtc::Event event_;
- rtc::scoped_ptr<ThreadWrapper> poller_thread_;
- const Call& call_;
- const std::vector<uint32_t> ssrcs_;
- RemoteBitrateObserver* const bitrate_observer_;
-};
-
void RampUpTest::RunRampUpTest(size_t num_streams,
unsigned int start_bitrate_bps,
const std::string& extension_type,
@@ -449,8 +391,6 @@ void RampUpTest::RunRampUpTest(size_t num_streams,
CreateSendConfig(num_streams, &stream_observer);
send_config_.rtp.extensions.clear();
- rtc::scoped_ptr<SendBitrateAdapter> send_bitrate_adapter_;
-
if (extension_type == RtpExtension::kAbsSendTime) {
stream_observer.SetRemoteBitrateEstimator(
new RemoteBitrateEstimatorAbsSendTime(
@@ -458,11 +398,6 @@ void RampUpTest::RunRampUpTest(size_t num_streams,
kRemoteBitrateEstimatorMinBitrateBps));
send_config_.rtp.extensions.push_back(RtpExtension(
extension_type.c_str(), kAbsSendTimeExtensionId));
- } else if (extension_type == RtpExtension::kTransportSequenceNumber) {
- stream_observer.SetRemoteBitrateEstimator(new RemoteEstimatorProxy(
- Clock::GetRealTimeClock(), stream_observer.GetPacketRouter()));
- send_config_.rtp.extensions.push_back(RtpExtension(
- extension_type.c_str(), kTransportSequenceNumberExtensionId));
} else {
stream_observer.SetRemoteBitrateEstimator(
new RemoteBitrateEstimatorSingleStream(
@@ -514,18 +449,10 @@ void RampUpTest::RunRampUpTest(size_t num_streams,
CreateStreams();
CreateFrameGeneratorCapturer();
- if (extension_type == RtpExtension::kTransportSequenceNumber) {
- send_bitrate_adapter_.reset(
- new SendBitrateAdapter(*sender_call_.get(), ssrcs, &stream_observer));
- }
Start();
EXPECT_EQ(kEventSignaled, stream_observer.Wait());
- // Destroy the SendBitrateAdapter (if any) to stop the poller thread in it,
- // otherwise we might get a data race with the destruction of the call.
- send_bitrate_adapter_.reset();
-
Stop();
DestroyStreams();
}
@@ -636,25 +563,4 @@ TEST_F(RampUpTest, AbsSendTimeSingleStreamWithHighStartBitrate) {
RunRampUpTest(1, 0.9 * kSingleStreamTargetBps, RtpExtension::kAbsSendTime,
false, false);
}
-
-TEST_F(RampUpTest, TransportSequenceNumberSingleStream) {
- RunRampUpTest(1, 0, RtpExtension::kTransportSequenceNumber, false, false);
-}
-
-TEST_F(RampUpTest, TransportSequenceNumberSimulcast) {
- RunRampUpTest(3, 0, RtpExtension::kTransportSequenceNumber, false, false);
-}
-
-TEST_F(RampUpTest, TransportSequenceNumberSimulcastWithRtx) {
- RunRampUpTest(3, 0, RtpExtension::kTransportSequenceNumber, true, false);
-}
-
-TEST_F(RampUpTest, TransportSequenceNumberSimulcastByRedWithRtx) {
- RunRampUpTest(3, 0, RtpExtension::kTransportSequenceNumber, true, true);
-}
-
-TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) {
- RunRampUpTest(1, 0.9 * kSingleStreamTargetBps,
- RtpExtension::kTransportSequenceNumber, false, false);
-}
} // namespace webrtc
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