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Enabling and testing RTCP CNAME in new API.

Browse Source 
BUG=2232
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2076004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4592 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2013-08-22 09:42:17 +00:00
parent 360e376872
commit 013d994583
2 changed files with 69 additions and 14 deletions
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9
webrtc/video_engine/internal/video_send_stream.cc
View File

@ -10,6 +10,8 @@
#include "webrtc/video_engine/internal/video_send_stream.h"
#include <string.h>
#include <vector>
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
@ -105,6 +107,13 @@ VideoSendStream::VideoSendStream(newapi::Transport* transport,
rtp_rtcp_->SetTransmissionSmoothingStatus(channel_, config_.pacing);
rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, 1);
char rtcp_cname[ViERTP_RTCP::KMaxRTCPCNameLength];
assert(config_.rtp.c_name.length() < ViERTP_RTCP::KMaxRTCPCNameLength);
strncpy(rtcp_cname, config_.rtp.c_name.c_str(), sizeof(rtcp_cname) - 1);
rtcp_cname[sizeof(rtcp_cname) - 1] = '\0';
rtp_rtcp_->SetRTCPCName(channel_, rtcp_cname);
capture_ = ViECapture::GetInterface(video_engine);
capture_->AllocateExternalCaptureDevice(capture_id_, external_capture_);
capture_->ConnectCaptureDevice(capture_id_, channel_);

74
webrtc/video_engine/test/send_stream_tests.cc
View File

@ -9,6 +9,7 @@
*/
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/video_engine/test/common/frame_generator.h"
@ -19,7 +20,6 @@
namespace webrtc {
class VideoSendStreamTest : public ::testing::Test {};
class SendTransportObserver : public test::NullTransport {
public:
explicit SendTransportObserver(unsigned long timeout_ms)
@ -39,8 +39,32 @@ class SendTransportObserver : public test::NullTransport {
unsigned long timeout_ms_;
};
class VideoSendStreamTest : public ::testing::Test {
protected:
static const uint32_t kSendSsrc;
void RunSendTest(newapi::VideoCall* call,
const newapi::VideoSendStream::Config& config,
SendTransportObserver* observer) {
newapi::VideoSendStream* send_stream = call->CreateSendStream(config);
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(
send_stream->Input(),
test::FrameGenerator::Create(320, 240, Clock::GetRealTimeClock()),
30));
send_stream->StartSend();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, observer->Wait());
frame_generator_capturer->Stop();
send_stream->StopSend();
call->DestroySendStream(send_stream);
}
};
const uint32_t VideoSendStreamTest::kSendSsrc = 0xC0FFEE;
TEST_F(VideoSendStreamTest, SendsSetSsrc) {
static const uint32_t kSendSsrc = 0xC0FFEE;
class SendSsrcObserver : public SendTransportObserver {
public:
SendSsrcObserver() : SendTransportObserver(30 * 1000) {}
@ -62,20 +86,42 @@ TEST_F(VideoSendStreamTest, SendsSetSsrc) {
newapi::VideoSendStream::Config send_config = call->GetDefaultSendConfig();
send_config.rtp.ssrcs.push_back(kSendSsrc);
newapi::VideoSendStream* send_stream = call->CreateSendStream(send_config);
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(
send_stream->Input(),
test::FrameGenerator::Create(320, 240, Clock::GetRealTimeClock()),
30));
send_stream->StartSend();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, observer.Wait());
RunSendTest(call.get(), send_config, &observer);
}
frame_generator_capturer->Stop();
send_stream->StopSend();
call->DestroySendStream(send_stream);
TEST_F(VideoSendStreamTest, SupportsCName) {
static std::string kCName = "PjQatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo=";
class CNameObserver : public SendTransportObserver {
public:
CNameObserver() : SendTransportObserver(30 * 1000) {}
virtual bool SendRTCP(const uint8_t* packet, size_t length) OVERRIDE {
RTCPUtility::RTCPParserV2 parser(packet, length, true);
EXPECT_TRUE(parser.IsValid());
RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
while (packet_type != RTCPUtility::kRtcpNotValidCode) {
if (packet_type == RTCPUtility::kRtcpSdesChunkCode) {
EXPECT_EQ(parser.Packet().CName.CName, kCName);
send_test_complete_->Set();
}
packet_type = parser.Iterate();
}
return true;
}
} observer;
newapi::VideoCall::Config call_config(&observer);
scoped_ptr<newapi::VideoCall> call(newapi::VideoCall::Create(call_config));
newapi::VideoSendStream::Config send_config = call->GetDefaultSendConfig();
send_config.rtp.ssrcs.push_back(kSendSsrc);
send_config.rtp.c_name = kCName;
RunSendTest(call.get(), send_config, &observer);
}
} // namespace webrtc