Implementation and testing of PLI in new API.

BUG=2174
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2011004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4567 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2013-08-19 16:09:34 +00:00
parent d4f607e70a
commit e7f056ec45
4 changed files with 413 additions and 155 deletions

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@ -41,6 +41,8 @@ VideoReceiveStream::VideoReceiveStream(
// TODO(pbos): This is not fine grained enough...
rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
rtp_rtcp_->SetKeyFrameRequestMethod(channel_,
kViEKeyFrameRequestPliRtcp);
assert(config_.rtp.ssrc != 0);

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@ -0,0 +1,137 @@
/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
#include <map>
#include <vector>
#include "webrtc/typedefs.h"
#include "webrtc/video_engine/new_include/video_send_stream.h"
namespace webrtc {
namespace test {
class RtpRtcpObserver {
public:
newapi::Transport* SendTransport() {
return &send_transport_;
}
newapi::Transport* ReceiveTransport() {
return &receive_transport_;
}
void SetReceivers(newapi::PacketReceiver* send_transport_receiver,
newapi::PacketReceiver* receive_transport_receiver) {
send_transport_.SetReceiver(send_transport_receiver);
receive_transport_.SetReceiver(receive_transport_receiver);
}
void StopSending() {
send_transport_.StopSending();
receive_transport_.StopSending();
}
protected:
RtpRtcpObserver()
: lock_(CriticalSectionWrapper::CreateCriticalSection()),
send_transport_(lock_.get(),
this,
&RtpRtcpObserver::OnSendRtp,
&RtpRtcpObserver::OnSendRtcp),
receive_transport_(lock_.get(),
this,
&RtpRtcpObserver::OnReceiveRtp,
&RtpRtcpObserver::OnReceiveRtcp) {}
enum Action {
SEND_PACKET,
DROP_PACKET,
};
virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
return SEND_PACKET;
}
virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
return SEND_PACKET;
}
virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) {
return SEND_PACKET;
}
virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length) {
return SEND_PACKET;
}
private:
class PacketTransport : public test::DirectTransport {
public:
typedef Action (RtpRtcpObserver::*PacketTransportAction)(const uint8_t*,
size_t);
PacketTransport(CriticalSectionWrapper* lock,
RtpRtcpObserver* observer,
PacketTransportAction on_rtp,
PacketTransportAction on_rtcp)
: lock_(lock),
observer_(observer),
on_rtp_(on_rtp),
on_rtcp_(on_rtcp) {}
private:
virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
Action action;
{
CriticalSectionScoped crit_(lock_);
action = (observer_->*on_rtp_)(packet, length);
}
switch (action) {
case DROP_PACKET:
// Drop packet silently.
return true;
case SEND_PACKET:
return test::DirectTransport::SendRTP(packet, length);
}
}
virtual bool SendRTCP(const uint8_t* packet, size_t length) OVERRIDE {
Action action;
{
CriticalSectionScoped crit_(lock_);
action = (observer_->*on_rtcp_)(packet, length);
}
switch (action) {
case DROP_PACKET:
// Drop packet silently.
return true;
case SEND_PACKET:
return test::DirectTransport::SendRTCP(packet, length);
}
}
// Pointer to shared lock instance protecting on_rtp_/on_rtcp_ calls.
CriticalSectionWrapper* lock_;
RtpRtcpObserver* observer_;
PacketTransportAction on_rtp_, on_rtcp_;
};
protected:
scoped_ptr<CriticalSectionWrapper> lock_;
private:
PacketTransport send_transport_, receive_transport_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_

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@ -7,6 +7,8 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <assert.h>
#include <map>
#include "testing/gtest/include/gtest/gtest.h"
@ -21,86 +23,131 @@
#include "webrtc/video_engine/test/common/frame_generator.h"
#include "webrtc/video_engine/test/common/frame_generator_capturer.h"
#include "webrtc/video_engine/test/common/generate_ssrcs.h"
#include "webrtc/video_engine/test/common/rtp_rtcp_observer.h"
namespace webrtc {
class NackObserver {
struct EngineTestParams {
size_t width, height;
struct {
unsigned int min, start, max;
} bitrate;
};
class EngineTest : public ::testing::TestWithParam<EngineTestParams> {
public:
class SenderTransport : public test::DirectTransport {
public:
explicit SenderTransport(NackObserver* observer) : observer_(observer) {}
EngineTest() : send_stream_(NULL), receive_stream_(NULL) {}
virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
{
CriticalSectionScoped lock(observer_->crit_.get());
if (observer_->DropSendPacket(packet, length))
return true;
++observer_->sent_rtp_packets_;
}
~EngineTest() {
EXPECT_EQ(NULL, send_stream_);
EXPECT_EQ(NULL, receive_stream_);
}
return test::DirectTransport::SendRTP(packet, length);
}
protected:
void CreateCalls(newapi::Transport* sender_transport,
newapi::Transport* receiver_transport) {
newapi::VideoCall::Config sender_config(sender_transport);
newapi::VideoCall::Config receiver_config(receiver_transport);
sender_call_.reset(newapi::VideoCall::Create(sender_config));
receiver_call_.reset(newapi::VideoCall::Create(receiver_config));
}
NackObserver* observer_;
} sender_transport_;
void CreateTestConfigs() {
EngineTestParams params = GetParam();
send_config_ = sender_call_->GetDefaultSendConfig();
receive_config_ = receiver_call_->GetDefaultReceiveConfig();
class ReceiverTransport : public test::DirectTransport {
public:
explicit ReceiverTransport(NackObserver* observer) : observer_(observer) {}
test::GenerateRandomSsrcs(&send_config_, &reserved_ssrcs_);
send_config_.codec.width = static_cast<uint16_t>(params.width);
send_config_.codec.height = static_cast<uint16_t>(params.height);
send_config_.codec.minBitrate = params.bitrate.min;
send_config_.codec.startBitrate = params.bitrate.start;
send_config_.codec.maxBitrate = params.bitrate.max;
bool SendRTCP(const uint8_t* packet, size_t length) {
{
CriticalSectionScoped lock(observer_->crit_.get());
receive_config_.rtp.ssrc = send_config_.rtp.ssrcs[0];
}
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
void CreateStreams() {
assert(send_stream_ == NULL);
assert(receive_stream_ == NULL);
bool received_nack = false;
RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
while (packet_type != RTCPUtility::kRtcpNotValidCode) {
if (packet_type == RTCPUtility::kRtcpRtpfbNackCode)
received_nack = true;
send_stream_ = sender_call_->CreateSendStream(send_config_);
receive_stream_ = receiver_call_->CreateReceiveStream(receive_config_);
}
packet_type = parser.Iterate();
}
void CreateFrameGenerator() {
EngineTestParams params = GetParam();
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
send_stream_->Input(),
test::FrameGenerator::Create(
params.width, params.height, Clock::GetRealTimeClock()),
30));
}
if (received_nack) {
observer_->ReceivedNack();
} else {
observer_->RtcpWithoutNack();
}
}
return DirectTransport::SendRTCP(packet, length);
}
void StartSending() {
receive_stream_->StartReceive();
send_stream_->StartSend();
frame_generator_capturer_->Start();
}
NackObserver* observer_;
} receiver_transport_;
void StopSending() {
frame_generator_capturer_->Stop();
send_stream_->StopSend();
receive_stream_->StopReceive();
}
void DestroyStreams() {
sender_call_->DestroySendStream(send_stream_);
receiver_call_->DestroyReceiveStream(receive_stream_);
send_stream_= NULL;
receive_stream_ = NULL;
}
void ReceivesPliAndRecovers(int rtp_history_ms);
scoped_ptr<newapi::VideoCall> sender_call_;
scoped_ptr<newapi::VideoCall> receiver_call_;
newapi::VideoSendStream::Config send_config_;
newapi::VideoReceiveStream::Config receive_config_;
newapi::VideoSendStream* send_stream_;
newapi::VideoReceiveStream* receive_stream_;
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
std::map<uint32_t, bool> reserved_ssrcs_;
};
// TODO(pbos): What are sane values here for bitrate? Are we missing any
// important resolutions?
EngineTestParams video_1080p = {1920, 1080, {300, 600, 800}};
EngineTestParams video_720p = {1280, 720, {300, 600, 800}};
EngineTestParams video_vga = {640, 480, {300, 600, 800}};
EngineTestParams video_qvga = {320, 240, {300, 600, 800}};
EngineTestParams video_4cif = {704, 576, {300, 600, 800}};
EngineTestParams video_cif = {352, 288, {300, 600, 800}};
EngineTestParams video_qcif = {176, 144, {300, 600, 800}};
class NackObserver : public test::RtpRtcpObserver {
static const int kNumberOfNacksToObserve = 4;
static const int kInverseProbabilityToStartLossBurst = 20;
static const int kMaxLossBurst = 10;
public:
NackObserver()
: sender_transport_(this),
receiver_transport_(this),
crit_(CriticalSectionWrapper::CreateCriticalSection()),
received_all_retransmissions_(EventWrapper::Create()),
: received_all_retransmissions_(EventWrapper::Create()),
rtp_parser_(RtpHeaderParser::Create()),
drop_burst_count_(0),
sent_rtp_packets_(0),
nacks_left_(4) {}
nacks_left_(kNumberOfNacksToObserve) {}
EventTypeWrapper Wait() {
// 2 minutes should be more than enough time for the test to finish.
return received_all_retransmissions_->Wait(2 * 60 * 1000);
}
void StopSending() {
sender_transport_.StopSending();
receiver_transport_.StopSending();
}
private:
// Decides whether a current packet should be dropped or not. A retransmitted
// packet will never be dropped. Packets are dropped in short bursts. When
// enough NACKs have been received, no further packets are dropped.
bool DropSendPacket(const uint8_t* packet, size_t length) {
virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, static_cast<int>(length)));
RTPHeader header;
@ -110,29 +157,56 @@ class NackObserver {
if (dropped_packets_.find(header.sequenceNumber) !=
dropped_packets_.end()) {
retransmitted_packets_.insert(header.sequenceNumber);
return false;
return SEND_PACKET;
}
// Enough NACKs received, stop dropping packets.
if (nacks_left_ == 0)
return false;
if (nacks_left_ == 0) {
++sent_rtp_packets_;
return SEND_PACKET;
}
// Still dropping packets.
if (drop_burst_count_ > 0) {
--drop_burst_count_;
dropped_packets_.insert(header.sequenceNumber);
return true;
return DROP_PACKET;
}
if (sent_rtp_packets_ > 0 && rand() % 20 == 0) {
drop_burst_count_ = rand() % 10;
// Should we start dropping packets?
if (sent_rtp_packets_ > 0 &&
rand() % kInverseProbabilityToStartLossBurst == 0) {
drop_burst_count_ = rand() % kMaxLossBurst;
dropped_packets_.insert(header.sequenceNumber);
return true;
return DROP_PACKET;
}
return false;
++sent_rtp_packets_;
return SEND_PACKET;
}
virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length) OVERRIDE {
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
bool received_nack = false;
RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
while (packet_type != RTCPUtility::kRtcpNotValidCode) {
if (packet_type == RTCPUtility::kRtcpRtpfbNackCode)
received_nack = true;
packet_type = parser.Iterate();
}
if (received_nack) {
ReceivedNack();
} else {
RtcpWithoutNack();
}
return SEND_PACKET;
}
private:
void ReceivedNack() {
if (nacks_left_ > 0)
--nacks_left_;
@ -151,7 +225,6 @@ class NackObserver {
}
}
scoped_ptr<CriticalSectionWrapper> crit_;
scoped_ptr<EventWrapper> received_all_retransmissions_;
scoped_ptr<RtpHeaderParser> rtp_parser_;
@ -164,111 +237,156 @@ class NackObserver {
static const int kRequiredRtcpsWithoutNack = 2;
};
struct EngineTestParams {
size_t width, height;
struct {
unsigned int min, start, max;
} bitrate;
};
class EngineTest : public ::testing::TestWithParam<EngineTestParams> {
public:
virtual void SetUp() {
reserved_ssrcs_.clear();
}
protected:
newapi::VideoCall* CreateTestCall(newapi::Transport* transport) {
newapi::VideoCall::Config call_config(transport);
return newapi::VideoCall::Create(call_config);
}
newapi::VideoSendStream::Config CreateTestSendConfig(
newapi::VideoCall* call,
EngineTestParams params) {
newapi::VideoSendStream::Config config = call->GetDefaultSendConfig();
test::GenerateRandomSsrcs(&config, &reserved_ssrcs_);
config.codec.width = static_cast<uint16_t>(params.width);
config.codec.height = static_cast<uint16_t>(params.height);
config.codec.minBitrate = params.bitrate.min;
config.codec.startBitrate = params.bitrate.start;
config.codec.maxBitrate = params.bitrate.max;
return config;
}
test::FrameGeneratorCapturer* CreateTestFrameGeneratorCapturer(
newapi::VideoSendStream* target,
EngineTestParams params) {
return test::FrameGeneratorCapturer::Create(
target->Input(),
test::FrameGenerator::Create(
params.width, params.height, Clock::GetRealTimeClock()),
30);
}
std::map<uint32_t, bool> reserved_ssrcs_;
};
// TODO(pbos): What are sane values here for bitrate? Are we missing any
// important resolutions?
EngineTestParams video_1080p = {1920, 1080, {300, 600, 800}};
EngineTestParams video_720p = {1280, 720, {300, 600, 800}};
EngineTestParams video_vga = {640, 480, {300, 600, 800}};
EngineTestParams video_qvga = {320, 240, {300, 600, 800}};
EngineTestParams video_4cif = {704, 576, {300, 600, 800}};
EngineTestParams video_cif = {352, 288, {300, 600, 800}};
EngineTestParams video_qcif = {176, 144, {300, 600, 800}};
TEST_P(EngineTest, ReceivesAndRetransmitsNack) {
EngineTestParams params = GetParam();
// Set up a video call per sender and receiver. Both send RTCP, and have a set
// RTP history > 0 to enable NACK and retransmissions.
NackObserver observer;
scoped_ptr<newapi::VideoCall> sender_call(
CreateTestCall(&observer.sender_transport_));
scoped_ptr<newapi::VideoCall> receiver_call(
CreateTestCall(&observer.receiver_transport_));
CreateCalls(observer.SendTransport(), observer.ReceiveTransport());
observer.receiver_transport_.SetReceiver(sender_call->Receiver());
observer.sender_transport_.SetReceiver(receiver_call->Receiver());
observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver());
newapi::VideoSendStream::Config send_config =
CreateTestSendConfig(sender_call.get(), params);
send_config.rtp.nack.rtp_history_ms = 1000;
CreateTestConfigs();
int rtp_history_ms = 1000;
send_config_.rtp.nack.rtp_history_ms = rtp_history_ms;
receive_config_.rtp.nack.rtp_history_ms = rtp_history_ms;
newapi::VideoReceiveStream::Config receive_config =
receiver_call->GetDefaultReceiveConfig();
receive_config.rtp.ssrc = send_config.rtp.ssrcs[0];
receive_config.rtp.nack.rtp_history_ms = send_config.rtp.nack.rtp_history_ms;
CreateStreams();
CreateFrameGenerator();
newapi::VideoSendStream* send_stream =
sender_call->CreateSendStream(send_config);
newapi::VideoReceiveStream* receive_stream =
receiver_call->CreateReceiveStream(receive_config);
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
CreateTestFrameGeneratorCapturer(send_stream, params));
ASSERT_TRUE(frame_generator_capturer.get() != NULL);
receive_stream->StartReceive();
send_stream->StartSend();
frame_generator_capturer->Start();
StartSending();
// Wait() waits for an event triggered when NACKs have been received, NACKed
// packets retransmitted and frames rendered again.
EXPECT_EQ(kEventSignaled, observer.Wait());
frame_generator_capturer->Stop();
send_stream->StopSend();
receive_stream->StopReceive();
StopSending();
DestroyStreams();
receiver_call->DestroyReceiveStream(receive_stream);
receiver_call->DestroySendStream(send_stream);
observer.StopSending();
}
class PliObserver : public test::RtpRtcpObserver {
static const int kInverseDropProbability = 16;
public:
PliObserver(bool nack_enabled) :
renderer_(this),
rtp_header_parser_(RtpHeaderParser::Create()),
nack_enabled_(nack_enabled),
first_retransmitted_timestamp_(0),
last_send_timestamp_(0),
rendered_frame_(false),
received_pli_(false) {}
virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
RTPHeader header;
EXPECT_TRUE(rtp_header_parser_->Parse(packet, length, &header));
// Drop all NACK retransmissions. This is to force transmission of a PLI.
if (header.timestamp < last_send_timestamp_)
return DROP_PACKET;
if (received_pli_) {
if (first_retransmitted_timestamp_ == 0) {
first_retransmitted_timestamp_ = header.timestamp;
}
} else if (rendered_frame_ && rand() % kInverseDropProbability == 0) {
return DROP_PACKET;
}
last_send_timestamp_ = header.timestamp;
return SEND_PACKET;
}
virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length) OVERRIDE {
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
packet_type != RTCPUtility::kRtcpNotValidCode;
packet_type = parser.Iterate()) {
if (!nack_enabled_)
EXPECT_NE(packet_type, RTCPUtility::kRtcpRtpfbNackCode);
if (packet_type == RTCPUtility::kRtcpPsfbPliCode) {
received_pli_ = true;
break;
}
}
return SEND_PACKET;
}
class ReceiverRenderer : public newapi::VideoRenderer {
public:
ReceiverRenderer(PliObserver* observer)
: rendered_retransmission_(EventWrapper::Create()),
observer_(observer) {}
virtual void RenderFrame(const I420VideoFrame& video_frame,
int time_to_render_ms) {
CriticalSectionScoped crit_(observer_->lock_.get());
if (observer_->first_retransmitted_timestamp_ != 0 &&
video_frame.timestamp() > observer_->first_retransmitted_timestamp_) {
EXPECT_TRUE(observer_->received_pli_);
rendered_retransmission_->Set();
}
observer_->rendered_frame_ = true;
}
scoped_ptr<EventWrapper> rendered_retransmission_;
PliObserver* observer_;
} renderer_;
EventTypeWrapper Wait() {
// 120 seconds should be plenty of time.
return renderer_.rendered_retransmission_->Wait(2 * 60 * 1000);
}
private:
scoped_ptr<RtpHeaderParser> rtp_header_parser_;
bool nack_enabled_;
uint32_t first_retransmitted_timestamp_;
uint32_t last_send_timestamp_;
bool rendered_frame_;
bool received_pli_;
};
void EngineTest::ReceivesPliAndRecovers(int rtp_history_ms) {
PliObserver observer(rtp_history_ms > 0);
CreateCalls(observer.SendTransport(), observer.ReceiveTransport());
observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver());
CreateTestConfigs();
send_config_.rtp.nack.rtp_history_ms = rtp_history_ms;
receive_config_.rtp.nack.rtp_history_ms = rtp_history_ms;
receive_config_.renderer = &observer.renderer_;
CreateStreams();
CreateFrameGenerator();
StartSending();
// Wait() waits for an event triggered when Pli has been received and frames
// have been rendered afterwards.
EXPECT_EQ(kEventSignaled, observer.Wait());
StopSending();
DestroyStreams();
observer.StopSending();
}
TEST_P(EngineTest, ReceivesPliAndRecoversWithNack) {
ReceivesPliAndRecovers(1000);
}
// TODO(pbos): Enable this when 2250 is resolved.
TEST_P(EngineTest, DISABLED_ReceivesPliAndRecoversWithoutNack) {
ReceivesPliAndRecovers(0);
}
INSTANTIATE_TEST_CASE_P(EngineTest, EngineTest, ::testing::Values(video_vga));
} // namespace webrtc

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@ -32,6 +32,7 @@
'common/mac/video_renderer_mac.h',
'common/mac/video_renderer_mac.mm',
'common/null_platform_renderer.cc',
'common/rtp_rtcp_observer.h',
'common/run_tests.cc',
'common/run_tests.h',
'common/run_loop.cc',