media/cast/test/end2end_unittest.cc - Issue 147993010: Cast: end2end bug fixes

Unified Diff: media/cast/test/end2end_unittest.cc

Issue 147993010: Cast: end2end bug fixes (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src

Patch Set: Disable (some) tests on win Created 6 years, 10 months ago

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Index: media/cast/test/end2end_unittest.cc

diff --git a/media/cast/test/end2end_unittest.cc b/media/cast/test/end2end_unittest.cc

index f42c4c5e07150761f140766d88804c827d3c05b9..5b517c842581eed6541d3e161e5118aae445533f 100644

--- a/media/cast/test/end2end_unittest.cc

+++ b/media/cast/test/end2end_unittest.cc

@@ -44,8 +44,10 @@ static const int64 kStartMillisecond = GG_INT64_C(1245);

static const int kAudioChannels = 2;

static const double kSoundFrequency = 314.15926535897; // Freq of sine wave.

static const float kSoundVolume = 0.5f;

-static const int kVideoWidth = 1280;

-static const int kVideoHeight = 720;

+static const int kVideoHdWidth = 1280;

+static const int kVideoHdHeight = 720;

+static const int kVideoQcifWidth = 176;

+static const int kVideoQcifHeight = 144;

static const int kCommonRtpHeaderLength = 12;

static const uint8 kCastReferenceFrameIdBitReset = 0xDF; // Mask is 0x40.

@@ -61,7 +63,11 @@ static const int kFrameTimerMs = 33;

// The packets pass through the pacer which can delay the beginning of the

// frame by 10 ms if there is packets belonging to the previous frame being

// retransmitted.

-static const int kTimerErrorMs = 15;

+static const int kTimerErrorMs = 20;

pwestin 2014/02/12 21:16:43 please add your findings to where these 20 ms orig

mikhal1 2014/02/12 23:21:05 Done.

+// Start the video synthetic start value to medium range value, to avoid edge

+// effects cause by encoding and quantization.

+static const int kVideoStart = 100;

std::string ConvertFromBase16String(const std::string base_16) {

std::string compressed;

@@ -95,7 +101,8 @@ std::map<RtpTimestamp, LoggingEventCounts> GetEventCountForFrameEvents(

const std::vector<FrameEvent>& frame_events) {

std::map<RtpTimestamp, LoggingEventCounts> event_counter_for_frame;

for (std::vector<FrameEvent>::const_iterator it = frame_events.begin();

- it != frame_events.end(); ++it) {

+ it != frame_events.end();

+ ++it) {

std::map<RtpTimestamp, LoggingEventCounts>::iterator map_it =

event_counter_for_frame.find(it->rtp_timestamp);

if (map_it == event_counter_for_frame.end()) {

@@ -117,7 +124,8 @@ std::map<uint16, LoggingEventCounts> GetEventCountForPacketEvents(

const std::vector<PacketEvent>& packet_events) {

std::map<uint16, LoggingEventCounts> event_counter_for_packet;

for (std::vector<PacketEvent>::const_iterator it = packet_events.begin();

- it != packet_events.end(); ++it) {

+ it != packet_events.end();

+ ++it) {

std::map<uint16, LoggingEventCounts>::iterator map_it =

event_counter_for_packet.find(it->packet_id);

if (map_it == event_counter_for_packet.end()) {

@@ -253,7 +261,8 @@ class TestReceiverAudioCallback

return; // No more checks needed.

EXPECT_NEAR(CountZeroCrossings(expected_audio_frame.audio_frame.samples),

- CountZeroCrossings(audio_frame->samples), 1);

+ CountZeroCrossings(audio_frame->samples),

+ 1);

}

void CheckCodedPcmAudioFrame(

@@ -291,7 +300,8 @@ class TestReceiverAudioCallback

}

EXPECT_NEAR(CountZeroCrossings(expected_audio_frame.audio_frame.samples),

- CountZeroCrossings(output_audio_samples), 1);

+ CountZeroCrossings(output_audio_samples),

+ 1);

}

int number_times_called() const { return num_called_; }

@@ -320,7 +330,9 @@ class TestReceiverVideoCallback

TestReceiverVideoCallback() : num_called_(0) {}

- void AddExpectedResult(int start_value, int width, int height,

+ void AddExpectedResult(int start_value,

+ int width,

+ int height,

const base::TimeTicks& capture_time) {

ExpectedVideoFrame expected_video_frame;

expected_video_frame.start_value = start_value;

@@ -379,8 +391,6 @@ class End2EndTest : public ::testing::Test {

: start_time_(),

testing_clock_(new base::SimpleTestTickClock()),

task_runner_(new test::FakeSingleThreadTaskRunner(testing_clock_)),

- transport_task_runner_(

- new test::FakeSingleThreadTaskRunner(testing_clock_)),

cast_environment_(new CastEnvironment(

scoped_ptr<base::TickClock>(testing_clock_).Pass(),

task_runner_,

@@ -431,8 +441,8 @@ class End2EndTest : public ::testing::Test {

video_sender_config_.incoming_feedback_ssrc = 4;

video_sender_config_.rtp_config.payload_type = 97;

video_sender_config_.use_external_encoder = false;

- video_sender_config_.width = kVideoWidth;

- video_sender_config_.height = kVideoHeight;

+ video_sender_config_.width = kVideoHdWidth;

+ video_sender_config_.height = kVideoHdHeight;

video_sender_config_.max_bitrate = 5000000;

video_sender_config_.min_bitrate = 1000000;

video_sender_config_.start_bitrate = 5000000;

@@ -472,7 +482,7 @@ class End2EndTest : public ::testing::Test {

testing_clock_,

transport_config_,

base::Bind(&UpdateCastTransportStatus),

- transport_task_runner_,

+ task_runner_,

&sender_to_receiver_));

cast_sender_.reset(CastSender::CreateCastSender(

@@ -509,10 +519,9 @@ class End2EndTest : public ::testing::Test {

void SendVideoFrame(int start_value, const base::TimeTicks& capture_time) {

if (start_time_.is_null())

- start_time_ = testing_clock_->NowTicks();

- base::TimeDelta time_diff = testing_clock_->NowTicks() - start_time_;

- gfx::Size size(kVideoWidth, kVideoHeight);

+ start_time_ = capture_time;

+ base::TimeDelta time_diff = capture_time - start_time_;

+ gfx::Size size(video_sender_config_.width, video_sender_config_.height);

EXPECT_TRUE(VideoFrame::IsValidConfig(

VideoFrame::I420, size, gfx::Rect(size), size));

scoped_refptr<media::VideoFrame> video_frame =

@@ -527,7 +536,6 @@ class End2EndTest : public ::testing::Test {

// Call process the timers every 1 ms.

testing_clock_->Advance(base::TimeDelta::FromMilliseconds(1));

task_runner_->RunTasks();

- transport_task_runner_->RunTasks();

}

@@ -542,9 +550,8 @@ class End2EndTest : public ::testing::Test {

transport::CastTransportConfig transport_config_;

base::TimeTicks start_time_;

- base::SimpleTestTickClock* testing_clock_; // Owned by CastEnvironment.

+ base::SimpleTestTickClock* testing_clock_;

scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;

- scoped_refptr<test::FakeSingleThreadTaskRunner> transport_task_runner_;

scoped_refptr<CastEnvironment> cast_environment_;

LoopBackTransport receiver_to_sender_;

@@ -567,33 +574,36 @@ class End2EndTest : public ::testing::Test {

std::vector<GenericEvent> generic_events_;

};

-// Audio and video test without packet loss using raw PCM 16 audio "codec";

-// This test is too slow. Disabled for now: crbug.com/329333.

-TEST_F(End2EndTest, DISABLED_LoopNoLossPcm16) {

+#if defined(OS_WIN)

+#define MAYBE_LoopNoLossPcm16 DISABLED_LoopNoLossPcm16

+#else

+#define MAYBE_LoopNoLossPcm16 LoopNoLossPcm16

+#endif

+// TODO(mikhal): Crashes in win bots (http://crbug.com/329563)

+TEST_F(End2EndTest, MAYBE_LoopNoLossPcm16) {

SetupConfig(transport::kPcm16, 32000, false, 1);

+ // Reduce video resolution to allow processing multiple frames within a

+ // reasonable time frame.

+ video_sender_config_.width = kVideoQcifWidth;

+ video_sender_config_.height = kVideoQcifHeight;

Create();

- int video_start = 1;

+ int video_start = kVideoStart;

int audio_diff = kFrameTimerMs;

int i = 0;

- for (; i < 10; ++i) {

+ for (; i < 300; ++i) {

int num_10ms_blocks = audio_diff / 10;

audio_diff -= num_10ms_blocks * 10;

- base::TimeTicks send_time = testing_clock_->NowTicks();

- test_receiver_video_callback_->AddExpectedResult(

- video_start,

- video_sender_config_.width,

- video_sender_config_.height,

- send_time);

scoped_ptr<AudioBus> audio_bus(audio_bus_factory_->NextAudioBus(

base::TimeDelta::FromMilliseconds(10) * num_10ms_blocks));

+ base::TimeTicks send_time = testing_clock_->NowTicks();

if (i != 0) {

// Due to the re-sampler and NetEq in the webrtc AudioCodingModule the

- // first samples will be 0 and then slowly ramp up to its real amplitude;

+ // first samples will be 0 and then slowly ramp up to its real

+ // amplitude;

// ignore the first frame.

test_receiver_audio_callback_->AddExpectedResult(

ToPcmAudioFrame(*audio_bus, audio_sender_config_.frequency),

@@ -607,11 +617,13 @@ TEST_F(End2EndTest, DISABLED_LoopNoLossPcm16) {

send_time,

base::Bind(&OwnThatAudioBus, base::Passed(&audio_bus)));

+ test_receiver_video_callback_->AddExpectedResult(

+ video_start,

+ video_sender_config_.width,

+ video_sender_config_.height,

+ send_time);

SendVideoFrame(video_start, send_time);

- RunTasks(kFrameTimerMs);

- audio_diff += kFrameTimerMs;

if (i == 0) {

frame_receiver_->GetRawAudioFrame(

num_10ms_blocks,

@@ -630,6 +642,8 @@ TEST_F(End2EndTest, DISABLED_LoopNoLossPcm16) {

base::Bind(&TestReceiverVideoCallback::CheckVideoFrame,

test_receiver_video_callback_));

+ RunTasks(kFrameTimerMs);

+ audio_diff += kFrameTimerMs;

video_start++;

}

@@ -737,7 +751,7 @@ TEST_F(End2EndTest, DISABLED_StartSenderBeforeReceiver) {

SetupConfig(transport::kOpus, kDefaultAudioSamplingRate, false, 1);

Create();

- int video_start = 1;

+ int video_start = kVideoStart;

int audio_diff = kFrameTimerMs;

sender_to_receiver_.SetSendPackets(false);

@@ -828,7 +842,7 @@ TEST_F(End2EndTest, GlitchWith3Buffers) {

video_receiver_config_.rtp_max_delay_ms = 67;

Create();

- int video_start = 50;

+ int video_start = kVideoStart;

base::TimeTicks send_time = testing_clock_->NowTicks();

SendVideoFrame(video_start, send_time);

RunTasks(kFrameTimerMs);

@@ -884,7 +898,7 @@ TEST_F(End2EndTest, DropEveryOtherFrame3Buffers) {

Create();

sender_to_receiver_.DropAllPacketsBelongingToOddFrames();

- int video_start = 50;

+ int video_start = kVideoStart;

base::TimeTicks send_time;

int i = 0;

@@ -1010,7 +1024,8 @@ TEST_F(End2EndTest, MAYBE_CryptoAudio) {

if (frames_counter != 0) {

// Due to the re-sampler and NetEq in the webrtc AudioCodingModule the

- // first samples will be 0 and then slowly ramp up to its real amplitude;

+ // first samples will be 0 and then slowly ramp up to its real

+ // amplitude;

// ignore the first frame.

test_receiver_audio_callback_->AddExpectedResult(

ToPcmAudioFrame(*audio_bus, audio_sender_config_.frequency),

@@ -1044,13 +1059,14 @@ TEST_F(End2EndTest, MAYBE_CryptoAudio) {

test_receiver_audio_callback_->number_times_called());

}

-// Video test without packet loss; This test is targeted at testing the logging

+// Video test without packet loss; This test is targeted at testing the

+// logging

// aspects of the end2end, but is basically equivalent to LoopNoLossPcm16.

TEST_F(End2EndTest, VideoLogging) {

SetupConfig(transport::kPcm16, 32000, false, 1);

Create();

- int video_start = 1;

+ int video_start = kVideoStart;

const int num_frames = 1;

for (int i = 0; i < num_frames; ++i) {

base::TimeTicks send_time = testing_clock_->NowTicks();

@@ -1094,7 +1110,8 @@ TEST_F(End2EndTest, VideoLogging) {

// Verify that each frame have the expected types of events logged.

for (std::map<RtpTimestamp, LoggingEventCounts>::iterator map_it =

event_counter_for_frame.begin();

- map_it != event_counter_for_frame.end(); ++map_it) {

+ map_it != event_counter_for_frame.end();

+ ++map_it) {

int total_event_count_for_frame = 0;

for (int i = 0; i < kNumOfLoggingEvents; ++i) {

total_event_count_for_frame += map_it->second.counter[i];

@@ -1121,7 +1138,8 @@ TEST_F(End2EndTest, VideoLogging) {

expected_event_count_for_frame +=

map_it->second.counter[kVideoFrameDecoded];

- // Verify that there were no other events logged with respect to this frame.

+ // Verify that there were no other events logged with respect to this

+ // frame.

// (i.e. Total event count = expected event count)

EXPECT_EQ(total_event_count_for_frame, expected_event_count_for_frame);

}

@@ -1135,7 +1153,8 @@ TEST_F(End2EndTest, VideoLogging) {

// Verify that each packet have the expected types of events logged.

for (std::map<uint16, LoggingEventCounts>::iterator map_it =

event_count_for_packet.begin();

- map_it != event_count_for_packet.end(); ++map_it) {

+ map_it != event_count_for_packet.end();

+ ++map_it) {

int total_event_count_for_packet = 0;

for (int i = 0; i < kNumOfLoggingEvents; ++i) {

total_event_count_for_packet += map_it->second.counter[i];

@@ -1158,7 +1177,8 @@ TEST_F(End2EndTest, VideoLogging) {

#else

#define MAYBE_AudioLogging AudioLogging

#endif

-// Audio test without packet loss; This test is targeted at testing the logging

+// Audio test without packet loss; This test is targeted at testing the

+// logging

// aspects of the end2end, but is basically equivalent to LoopNoLossPcm16.

pwestin 2014/02/12 21:16:43 weird sentence break

mikhal1 2014/02/12 23:21:05 Done.

TEST_F(End2EndTest, MAYBE_AudioLogging) {

SetupConfig(transport::kPcm16, 32000, false, 1);

@@ -1176,7 +1196,8 @@ TEST_F(End2EndTest, MAYBE_AudioLogging) {

if (i != 0) {

// Due to the re-sampler and NetEq in the webrtc AudioCodingModule the

- // first samples will be 0 and then slowly ramp up to its real amplitude;

+ // first samples will be 0 and then slowly ramp up to its real

+ // amplitude;

// ignore the first frame.

test_receiver_audio_callback_->AddExpectedResult(

ToPcmAudioFrame(*audio_bus, audio_sender_config_.frequency),

@@ -1229,12 +1250,16 @@ TEST_F(End2EndTest, MAYBE_AudioLogging) {

// TODO(imcheng): This only checks the first frame. This doesn't work

// properly for all frames for two reasons:

- // 1. There is a loopback of kAudioPlayoutDelay and kAudioFrameDecoded events

+ // 1. There is a loopback of kAudioPlayoutDelay and kAudioFrameDecoded

+ // events

pwestin 2014/02/12 21:16:43 weird sentence break

mikhal1 2014/02/12 23:21:05 Done.

// due to shared CastEnvironment between sender and

- // receiver in the test setup. We will need to create separate CastEnvironment

+ // receiver in the test setup. We will need to create separate

+ // CastEnvironment

pwestin 2014/02/12 21:16:43 weird sentence break

mikhal1 2014/02/12 23:21:05 Done.

// once again to fix this.

- // 2. kAudioPlayoutDelay and kAudioFrameDecoded RTP timestamps aren't exactly

- // aligned with those of kAudioFrameReceived and kAudioFrameEncoded. Note that

+ // 2. kAudioPlayoutDelay and kAudioFrameDecoded RTP timestamps aren't

+ // exactly

+ // aligned with those of kAudioFrameReceived and kAudioFrameEncoded. Note

+ // that

pwestin 2014/02/12 21:16:43 weird sentence break

mikhal1 2014/02/12 23:21:05 Done.

// these RTP timestamps are output from webrtc::AudioCodingModule which are

// different from RTP timestamps that the cast library generates during the

// encode step (and which are sent to receiver). The first frame just happen

@@ -1243,25 +1268,20 @@ TEST_F(End2EndTest, MAYBE_AudioLogging) {

for (int j = 0; j < kNumOfLoggingEvents; ++j)

total_event_count_for_frame += map_it->second.counter[j];

int expected_event_count_for_frame = 0;

EXPECT_GT(map_it->second.counter[kAudioFrameReceived], 0);

- expected_event_count_for_frame +=

- map_it->second.counter[kAudioFrameReceived];

+ expected_event_count_for_frame += map_it->second.counter[kAudioFrameReceived];

EXPECT_GT(map_it->second.counter[kAudioFrameEncoded], 0);

- expected_event_count_for_frame +=

- map_it->second.counter[kAudioFrameEncoded];

+ expected_event_count_for_frame += map_it->second.counter[kAudioFrameEncoded];

// Note that this is a big positive number instead of just 1 due to loopback

// described in TODO above.

EXPECT_GT(map_it->second.counter[kAudioPlayoutDelay], 0);

- expected_event_count_for_frame +=

- map_it->second.counter[kAudioPlayoutDelay];

+ expected_event_count_for_frame += map_it->second.counter[kAudioPlayoutDelay];

EXPECT_GT(map_it->second.counter[kAudioFrameDecoded], 0);

- expected_event_count_for_frame +=

- map_it->second.counter[kAudioFrameDecoded];

+ expected_event_count_for_frame += map_it->second.counter[kAudioFrameDecoded];

// Verify that there were no other events logged with respect to this frame.

// (i.e. Total event count = expected event count)

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