Files
platform-external-webrtc/webrtc/test/call_test.cc
Erik Språng 9526187dde Default enable abs send time bwe for CallTest
Using the single stream bwe is really bad for the screenshare
test case in particular, but would probably help in other
cases as well so enabling it by default in CallTest setup.

BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43089004

Cr-Commit-Position: refs/heads/master@{#8971}
2015-04-10 09:58:51 +00:00

238 lines
7.4 KiB
C++

/*
* Copyright (c) 2014 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.
*/
#include "webrtc/test/call_test.h"
#include "webrtc/test/encoder_settings.h"
namespace webrtc {
namespace test {
namespace {
const int kVideoRotationRtpExtensionId = 4;
}
CallTest::CallTest()
: clock_(Clock::GetRealTimeClock()),
send_stream_(NULL),
fake_encoder_(clock_) {
}
CallTest::~CallTest() {
}
void CallTest::RunBaseTest(BaseTest* test) {
CreateSenderCall(test->GetSenderCallConfig());
if (test->ShouldCreateReceivers())
CreateReceiverCall(test->GetReceiverCallConfig());
test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
if (test->ShouldCreateReceivers()) {
test->SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver());
} else {
// Sender-only call delivers to itself.
test->SetReceivers(sender_call_->Receiver(), NULL);
}
CreateSendConfig(test->GetNumStreams());
if (test->ShouldCreateReceivers()) {
CreateMatchingReceiveConfigs();
}
test->ModifyConfigs(&send_config_, &receive_configs_, &encoder_config_);
CreateStreams();
test->OnStreamsCreated(send_stream_, receive_streams_);
CreateFrameGeneratorCapturer();
test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
Start();
test->PerformTest();
test->StopSending();
Stop();
DestroyStreams();
}
void CallTest::Start() {
send_stream_->Start();
for (size_t i = 0; i < receive_streams_.size(); ++i)
receive_streams_[i]->Start();
if (frame_generator_capturer_.get() != NULL)
frame_generator_capturer_->Start();
}
void CallTest::Stop() {
if (frame_generator_capturer_.get() != NULL)
frame_generator_capturer_->Stop();
for (size_t i = 0; i < receive_streams_.size(); ++i)
receive_streams_[i]->Stop();
send_stream_->Stop();
}
void CallTest::CreateCalls(const Call::Config& sender_config,
const Call::Config& receiver_config) {
CreateSenderCall(sender_config);
CreateReceiverCall(receiver_config);
}
void CallTest::CreateSenderCall(const Call::Config& config) {
sender_call_.reset(Call::Create(config));
}
void CallTest::CreateReceiverCall(const Call::Config& config) {
receiver_call_.reset(Call::Create(config));
}
void CallTest::CreateSendConfig(size_t num_streams) {
assert(num_streams <= kNumSsrcs);
send_config_ = VideoSendStream::Config();
send_config_.encoder_settings.encoder = &fake_encoder_;
send_config_.encoder_settings.payload_name = "FAKE";
send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
encoder_config_.streams = test::CreateVideoStreams(num_streams);
for (size_t i = 0; i < num_streams; ++i)
send_config_.rtp.ssrcs.push_back(kSendSsrcs[i]);
send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
}
void CallTest::CreateMatchingReceiveConfigs() {
assert(!send_config_.rtp.ssrcs.empty());
assert(receive_configs_.empty());
assert(allocated_decoders_.empty());
VideoReceiveStream::Config config;
config.rtp.local_ssrc = kReceiverLocalSsrc;
for (const RtpExtension& extension : send_config_.rtp.extensions)
config.rtp.extensions.push_back(extension);
for (size_t i = 0; i < send_config_.rtp.ssrcs.size(); ++i) {
VideoReceiveStream::Decoder decoder =
test::CreateMatchingDecoder(send_config_.encoder_settings);
allocated_decoders_.push_back(decoder.decoder);
config.decoders.clear();
config.decoders.push_back(decoder);
config.rtp.remote_ssrc = send_config_.rtp.ssrcs[i];
receive_configs_.push_back(config);
}
}
void CallTest::CreateFrameGeneratorCapturer() {
VideoStream stream = encoder_config_.streams.back();
frame_generator_capturer_.reset(
test::FrameGeneratorCapturer::Create(send_stream_->Input(),
stream.width,
stream.height,
stream.max_framerate,
clock_));
}
void CallTest::CreateStreams() {
assert(send_stream_ == NULL);
assert(receive_streams_.empty());
send_stream_ =
sender_call_->CreateVideoSendStream(send_config_, encoder_config_);
for (size_t i = 0; i < receive_configs_.size(); ++i) {
receive_streams_.push_back(
receiver_call_->CreateVideoReceiveStream(receive_configs_[i]));
}
}
void CallTest::DestroyStreams() {
if (send_stream_ != NULL)
sender_call_->DestroyVideoSendStream(send_stream_);
send_stream_ = NULL;
for (size_t i = 0; i < receive_streams_.size(); ++i)
receiver_call_->DestroyVideoReceiveStream(receive_streams_[i]);
receive_streams_.clear();
allocated_decoders_.clear();
}
const unsigned int CallTest::kDefaultTimeoutMs = 30 * 1000;
const unsigned int CallTest::kLongTimeoutMs = 120 * 1000;
const uint8_t CallTest::kSendPayloadType = 100;
const uint8_t CallTest::kFakeSendPayloadType = 125;
const uint8_t CallTest::kSendRtxPayloadType = 98;
const uint8_t CallTest::kRedPayloadType = 118;
const uint8_t CallTest::kUlpfecPayloadType = 119;
const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE,
0xBADCAFF};
const uint32_t CallTest::kSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE, 0xC0FFEF};
const uint32_t CallTest::kReceiverLocalSsrc = 0x123456;
const int CallTest::kNackRtpHistoryMs = 1000;
const int CallTest::kAbsSendTimeExtensionId = 7;
BaseTest::BaseTest(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) {
}
BaseTest::BaseTest(unsigned int timeout_ms,
const FakeNetworkPipe::Config& config)
: RtpRtcpObserver(timeout_ms, config) {
}
BaseTest::~BaseTest() {
}
Call::Config BaseTest::GetSenderCallConfig() {
return Call::Config(SendTransport());
}
Call::Config BaseTest::GetReceiverCallConfig() {
return Call::Config(ReceiveTransport());
}
void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
}
size_t BaseTest::GetNumStreams() const {
return 1;
}
void BaseTest::ModifyConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) {
}
void BaseTest::OnStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) {
}
void BaseTest::OnFrameGeneratorCapturerCreated(
FrameGeneratorCapturer* frame_generator_capturer) {
}
SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
}
SendTest::SendTest(unsigned int timeout_ms,
const FakeNetworkPipe::Config& config)
: BaseTest(timeout_ms, config) {
}
bool SendTest::ShouldCreateReceivers() const {
return false;
}
EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
}
EndToEndTest::EndToEndTest(unsigned int timeout_ms,
const FakeNetworkPipe::Config& config)
: BaseTest(timeout_ms, config) {
}
bool EndToEndTest::ShouldCreateReceivers() const {
return true;
}
} // namespace test
} // namespace webrtc