webrtc/call/call_perf_tests.cc - Issue 1419193002: Call OnSentPacket for all packets sent in the test framework.

Unified Diff: webrtc/call/call_perf_tests.cc

Issue 1419193002: Call OnSentPacket for all packets sent in the test framework. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: . Created 5 years, 2 months ago

Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.

Jump to:

View side-by-side diff with in-line comments

Download patch

Index: webrtc/call/call_perf_tests.cc

diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc

index cab39144503e36a098fe7610bbbf4fba9a831c13..6c28b45cd00a63fe9d97eb43b4685baabe0ebdd1 100644

--- a/webrtc/call/call_perf_tests.cc

+++ b/webrtc/call/call_perf_tests.cc

@@ -58,8 +58,7 @@ class CallPerfTest : public test::CallTest {

class SyncRtcpObserver : public test::RtpRtcpObserver {

public:

- explicit SyncRtcpObserver(const FakeNetworkPipe::Config& config)

- : test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs, config) {}

+ SyncRtcpObserver() : test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs) {}

Action OnSendRtcp(const uint8_t* packet, size_t length) override {

RTCPUtility::RTCPParserV2 parser(packet, length, true);

@@ -127,7 +126,7 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {

int voe_channel,

VoEVideoSync* voe_sync,

SyncRtcpObserver* audio_observer)

- : SyncRtcpObserver(FakeNetworkPipe::Config()),

+ : SyncRtcpObserver(),

pbos-webrtc 2015/10/27 14:06:29 Remove from ctor list

stefan-webrtc 2015/10/27 14:33:16 Done.

clock_(clock),

voe_channel_(voe_channel),

voe_sync_(voe_sync),

@@ -182,10 +181,10 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {

private:

Clock* const clock_;

- int voe_channel_;

- VoEVideoSync* voe_sync_;

- SyncRtcpObserver* audio_observer_;

- int64_t creation_time_ms_;

+ const int voe_channel_;

+ VoEVideoSync* const voe_sync_;

+ SyncRtcpObserver* const audio_observer_;

+ const int64_t creation_time_ms_;

int64_t first_time_in_sync_;

};

@@ -232,14 +231,7 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {

EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr));

int channel = voe_base->CreateChannel();

- FakeNetworkPipe::Config net_config;

- net_config.queue_delay_ms = 500;

- net_config.loss_percent = 5;

- SyncRtcpObserver audio_observer(net_config);

- VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(),

- channel,

- voe_sync,

- &audio_observer);

+ SyncRtcpObserver audio_observer;

Call::Config receiver_config;

receiver_config.voice_engine = voice_engine;

@@ -249,19 +241,41 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {

EXPECT_EQ(0, voe_codec->SetSendCodec(channel, isac));

AudioPacketReceiver voe_packet_receiver(channel, voe_network);

- audio_observer.SetReceivers(&voe_packet_receiver, &voe_packet_receiver);

- internal::TransportAdapter transport_adapter(audio_observer.SendTransport());

+ FakeNetworkPipe::Config net_config;

+ net_config.queue_delay_ms = 500;

+ net_config.loss_percent = 5;

+ test::PacketTransport audio_send_transport(

+ nullptr, &audio_observer, test::PacketTransport::kSender, net_config);

+ audio_send_transport.SetReceiver(&voe_packet_receiver);

pbos-webrtc 2015/10/27 14:06:29 consider adding this to ctor if possible in a foll

stefan-webrtc 2015/10/27 14:33:16 TODO for now.

+ test::PacketTransport audio_receive_transport(

+ nullptr, &audio_observer, test::PacketTransport::kReceiver, net_config);

+ audio_receive_transport.SetReceiver(&voe_packet_receiver);

+ audio_observer.SetTransports(&audio_send_transport, &audio_receive_transport);

+ internal::TransportAdapter transport_adapter(&audio_send_transport);

transport_adapter.Enable();

EXPECT_EQ(0,

voe_network->RegisterExternalTransport(channel, transport_adapter));

- observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver());

+ VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), channel,

+ voe_sync, &audio_observer);

+ test::PacketTransport sync_send_transport(sender_call_.get(), &observer,

+ test::PacketTransport::kSender,

+ FakeNetworkPipe::Config());

+ sync_send_transport.SetReceiver(receiver_call_->Receiver());

+ test::PacketTransport sync_receive_transport(receiver_call_.get(), &observer,

+ test::PacketTransport::kReceiver,

+ FakeNetworkPipe::Config());

+ sync_receive_transport.SetReceiver(sender_call_->Receiver());

+ observer.SetTransports(&sync_send_transport, &sync_receive_transport);

test::FakeDecoder fake_decoder;

- CreateSendConfig(1, observer.SendTransport());

- CreateMatchingReceiveConfigs(observer.ReceiveTransport());

+ CreateSendConfig(1, &sync_send_transport);

+ CreateMatchingReceiveConfigs(&sync_receive_transport);

send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;

if (fec) {

@@ -308,8 +322,10 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {

fake_audio_device.Stop();

Stop();

- observer.StopSending();

- audio_observer.StopSending();

+ sync_send_transport.StopSending();

+ sync_receive_transport.StopSending();

+ audio_send_transport.StopSending();

+ audio_receive_transport.StopSending();

voe_base->DeleteChannel(channel);

voe_base->Release();

@@ -345,11 +361,8 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,

class CaptureNtpTimeObserver : public test::EndToEndTest,

public VideoRenderer {

public:

- CaptureNtpTimeObserver(const FakeNetworkPipe::Config& config,

- int threshold_ms,

- int start_time_ms,

- int run_time_ms)

- : EndToEndTest(kLongTimeoutMs, config),

+ CaptureNtpTimeObserver(int threshold_ms, int start_time_ms, int run_time_ms)

+ : EndToEndTest(kLongTimeoutMs),

clock_(Clock::GetRealTimeClock()),

threshold_ms_(threshold_ms),

start_time_ms_(start_time_ms),

@@ -362,6 +375,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,

private:

void RenderFrame(const VideoFrame& video_frame,

int time_to_render_ms) override {

+ rtc::CritScope lock(&crit_);

if (video_frame.ntp_time_ms() <= 0) {

// Haven't got enough RTCP SR in order to calculate the capture ntp

// time.

@@ -402,6 +416,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,

bool IsTextureSupported() const override { return false; }

virtual Action OnSendRtp(const uint8_t* packet, size_t length) {

+ rtc::CritScope lock(&crit_);

RTPHeader header;

EXPECT_TRUE(parser_->Parse(packet, length, &header));

@@ -440,6 +455,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,

"within bounds.";

}

+ rtc::CriticalSection crit_;

Clock* clock_;

pbos-webrtc 2015/10/27 14:06:29 const

int threshold_ms_;

pbos-webrtc 2015/10/27 14:06:29 thread annotate what you need

stefan-webrtc 2015/10/27 14:33:16 Done.

int start_time_ms_;

@@ -450,9 +466,9 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,

uint32_t rtp_start_timestamp_;

typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList;

FrameCaptureTimeList capture_time_list_;

- } test(net_config, threshold_ms, start_time_ms, run_time_ms);

+ } test(threshold_ms, start_time_ms, run_time_ms);

- RunBaseTest(&test);

+ RunBaseTest(&test, net_config);

}

TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {

@@ -508,7 +524,7 @@ void CallPerfTest::TestCpuOveruse(LoadObserver::Load tested_load,

test::DelayedEncoder encoder_;

} test(tested_load, encode_delay_ms);

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(CallPerfTest, ReceivesCpuUnderuse) {

@@ -526,28 +542,18 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {

static const int kMinTransmitBitrateBps = 150000;

static const int kMinAcceptableTransmitBitrate = 130;

static const int kMaxAcceptableTransmitBitrate = 170;

- static const int kNumBitrateObservationsInRange = 100;

+ static const int kNumBitrateObservationsInRange = 10;

static const int kAcceptableBitrateErrorMargin = 15; // +- 7

- class BitrateObserver : public test::EndToEndTest, public PacketReceiver {

+ class BitrateObserver : public test::EndToEndTest {

public:

explicit BitrateObserver(bool using_min_transmit_bitrate)

: EndToEndTest(kLongTimeoutMs),

send_stream_(nullptr),

- send_transport_receiver_(nullptr),

pad_to_min_bitrate_(using_min_transmit_bitrate),

num_bitrate_observations_in_range_(0) {}

private:

- void SetReceivers(PacketReceiver* send_transport_receiver,

- PacketReceiver* receive_transport_receiver) override {

- send_transport_receiver_ = send_transport_receiver;

- test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver);

- }

- DeliveryStatus DeliverPacket(MediaType media_type,

- const uint8_t* packet,

- size_t length,

- const PacketTime& packet_time) override {

+ Action OnSendRtcp(const uint8_t* packet, size_t length) override {

pbos-webrtc 2015/10/27 14:06:29 Make a TODO to make this into something that runs

VideoSendStream::Stats stats = send_stream_->GetStats();

if (stats.substreams.size() > 0) {

RTC_DCHECK_EQ(1u, stats.substreams.size());

@@ -581,8 +587,7 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {

observation_complete_->Set();

}

- return send_transport_receiver_->DeliverPacket(media_type, packet, length,

- packet_time);

+ return SEND_PACKET;

}

void OnStreamsCreated(

@@ -607,13 +612,12 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {

}

VideoSendStream* send_stream_;

- PacketReceiver* send_transport_receiver_;

const bool pad_to_min_bitrate_;

int num_bitrate_observations_in_range_;

} test(pad_to_min_bitrate);

fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps);

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(CallPerfTest, PadsToMinTransmitBitrate) { TestMinTransmitBitrate(true); }

@@ -708,7 +712,7 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {

VideoEncoderConfig encoder_config_;

} test;

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

} // namespace webrtc

« no previous file with comments | « webrtc/call/call.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender.cc » ('j') | webrtc/test/call_test.cc » ('J')