Implement 'toffset' extension in VideoSendStream.
BUG=2229
R=holmer@google.com, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4722 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/video_engine/internal/video_send_stream.cc b/webrtc/video_engine/internal/video_send_stream.cc
index 0a4e738..979631b 100644
--- a/webrtc/video_engine/internal/video_send_stream.cc
+++ b/webrtc/video_engine/internal/video_send_stream.cc
@@ -105,7 +105,17 @@
}
rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
rtp_rtcp_->SetTransmissionSmoothingStatus(channel_, config_.pacing);
- rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, 1);
+
+ for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
+ const std::string& extension = config_.rtp.extensions[i].name;
+ int id = config_.rtp.extensions[i].id;
+ if (extension == "toffset") {
+ if (rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, id) != 0)
+ abort();
+ } else {
+ abort(); // Unsupported extension.
+ }
+ }
char rtcp_cname[ViERTP_RTCP::KMaxRTCPCNameLength];
assert(config_.rtp.c_name.length() < ViERTP_RTCP::KMaxRTCPCNameLength);
@@ -142,7 +152,7 @@
new ResolutionAdaptor(codec_, channel_, config_.codec.width,
config_.codec.height));
video_engine_base_->RegisterCpuOveruseObserver(channel_,
- overuse_observer_.get());
+ overuse_observer_.get());
}
}
diff --git a/webrtc/video_engine/new_include/config.h b/webrtc/video_engine/new_include/config.h
index 45c74b9..5eff973 100644
--- a/webrtc/video_engine/new_include/config.h
+++ b/webrtc/video_engine/new_include/config.h
@@ -73,7 +73,7 @@
// RTP header extension to use for the video stream, see RFC 5285.
struct RtpExtension {
- RtpExtension() : id(0) {}
+ RtpExtension(const char* name, int id) : name(name), id(id) {}
// TODO(mflodman) Add API to query supported extensions.
std::string name;
int id;
diff --git a/webrtc/video_engine/test/rampup_tests.cc b/webrtc/video_engine/test/rampup_tests.cc
index 18c1688..8e3e027 100644
--- a/webrtc/video_engine/test/rampup_tests.cc
+++ b/webrtc/video_engine/test/rampup_tests.cc
@@ -31,6 +31,10 @@
namespace webrtc {
+namespace {
+ static const int kTOffsetExtensionId = 7;
+}
+
class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
public:
typedef std::map<uint32_t, int> BytesSentMap;
@@ -55,7 +59,7 @@
rtp_rtcp_->SetREMBStatus(true);
rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
- 1);
+ kTOffsetExtensionId);
AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory;
remote_bitrate_estimator_.reset(rbe_factory.Create(this, clock));
}
@@ -150,6 +154,8 @@
send_config.internal_source = false;
test::FakeEncoder::SetCodecSettings(&send_config.codec, 3);
send_config.pacing = GetParam();
+ send_config.rtp.extensions.push_back(
+ RtpExtension("toffset", kTOffsetExtensionId));
test::GenerateRandomSsrcs(&send_config, &reserved_ssrcs_);
diff --git a/webrtc/video_engine/test/send_stream_tests.cc b/webrtc/video_engine/test/send_stream_tests.cc
index 8117e75..645cda0 100644
--- a/webrtc/video_engine/test/send_stream_tests.cc
+++ b/webrtc/video_engine/test/send_stream_tests.cc
@@ -13,6 +13,7 @@
#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/system_wrappers/interface/sleep.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/video_engine/test/common/fake_encoder.h"
#include "webrtc/video_engine/test/common/frame_generator.h"
@@ -137,6 +138,52 @@
RunSendTest(call.get(), send_config, &observer);
}
+TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
+ static const uint8_t kTOffsetExtensionId = 13;
+ class DelayedEncoder : public test::FakeEncoder {
+ public:
+ DelayedEncoder(Clock* clock) : test::FakeEncoder(clock) {}
+ virtual int32_t Encode(
+ const I420VideoFrame& input_image,
+ const CodecSpecificInfo* codec_specific_info,
+ const std::vector<VideoFrameType>* frame_types) OVERRIDE {
+ // A delay needs to be introduced to assure that we get a timestamp
+ // offset.
+ SleepMs(5);
+ return FakeEncoder::Encode(input_image, codec_specific_info, frame_types);
+ }
+ } encoder(Clock::GetRealTimeClock());
+
+ class TransmissionTimeOffsetObserver : public SendTransportObserver {
+ public:
+ TransmissionTimeOffsetObserver() : SendTransportObserver(30 * 1000) {
+ EXPECT_TRUE(rtp_header_parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionTransmissionTimeOffset, kTOffsetExtensionId));
+ }
+
+ virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
+ RTPHeader header;
+ EXPECT_TRUE(
+ rtp_header_parser_->Parse(packet, static_cast<int>(length), &header));
+
+ EXPECT_GT(header.extension.transmissionTimeOffset, 0);
+ send_test_complete_->Set();
+
+ return true;
+ }
+ } observer;
+
+ Call::Config call_config(&observer);
+ scoped_ptr<Call> call(Call::Create(call_config));
+
+ VideoSendStream::Config send_config = GetSendTestConfig(call.get());
+ send_config.encoder = &encoder;
+ send_config.rtp.extensions.push_back(
+ RtpExtension("toffset", kTOffsetExtensionId));
+
+ RunSendTest(call.get(), send_config, &observer);
+}
+
TEST_F(VideoSendStreamTest, RespondsToNack) {
class NackObserver : public SendTransportObserver, webrtc::Transport {
public:
