webrtc_m130/webrtc/test/call_test.cc
pbos@webrtc.org 2bb1bdab8d Preserve RTP states for restarted VideoSendStreams.
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.

Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.

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

Review URL: https://webrtc-codereview.appspot.com/20819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 13:06:48 +00:00

227 lines
6.9 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 {
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_, &video_streams_);
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;
video_streams_ = test::CreateVideoStreams(num_streams);
for (size_t i = 0; i < num_streams; ++i)
send_config_.rtp.ssrcs.push_back(kSendSsrcs[i]);
}
void CallTest::CreateMatchingReceiveConfigs() {
assert(!send_config_.rtp.ssrcs.empty());
assert(receive_configs_.empty());
VideoReceiveStream::Config config;
VideoCodec codec =
test::CreateDecoderVideoCodec(send_config_.encoder_settings);
config.codecs.push_back(codec);
if (send_config_.encoder_settings.encoder == &fake_encoder_) {
ExternalVideoDecoder decoder;
decoder.decoder = &fake_decoder_;
decoder.payload_type = send_config_.encoder_settings.payload_type;
config.external_decoders.push_back(decoder);
}
config.rtp.local_ssrc = kReceiverLocalSsrc;
for (size_t i = 0; i < send_config_.rtp.ssrcs.size(); ++i) {
config.rtp.remote_ssrc = send_config_.rtp.ssrcs[i];
receive_configs_.push_back(config);
}
}
void CallTest::CreateFrameGeneratorCapturer() {
VideoStream stream = video_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_, video_streams_, NULL);
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();
}
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 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;
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,
std::vector<VideoStream>* video_streams) {
}
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