Implement NACK over RTX for VideoSendStream.
BUG=2231
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2197008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4751 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 76d8bc3..5adf8b9 100644
--- a/webrtc/video_engine/internal/video_send_stream.cc
+++ b/webrtc/video_engine/internal/video_send_stream.cc
@@ -105,6 +105,18 @@
}
rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
rtp_rtcp_->SetTransmissionSmoothingStatus(channel_, config_.pacing);
+ if (!config_.rtp.rtx.ssrcs.empty()) {
+ assert(config_.rtp.rtx.ssrcs.size() == config_.rtp.ssrcs.size());
+ for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
+ rtp_rtcp_->SetLocalSSRC(
+ channel_, config_.rtp.rtx.ssrcs[i], kViEStreamTypeRtx, i);
+ }
+
+ if (config_.rtp.rtx.rtx_payload_type != 0) {
+ rtp_rtcp_->SetRtxSendPayloadType(channel_,
+ config_.rtp.rtx.rtx_payload_type);
+ }
+ }
for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
const std::string& extension = config_.rtp.extensions[i].name;
diff --git a/webrtc/video_engine/new_include/config.h b/webrtc/video_engine/new_include/config.h
index 5eff973..d19c8d9 100644
--- a/webrtc/video_engine/new_include/config.h
+++ b/webrtc/video_engine/new_include/config.h
@@ -12,6 +12,7 @@
#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_
#include <string>
+#include <vector>
namespace webrtc {
@@ -60,9 +61,9 @@
// Settings for RTP retransmission payload format, see RFC 4588 for details.
struct RtxConfig {
- RtxConfig() : ssrc(0), rtx_payload_type(0), video_payload_type(0) {}
- // SSRC to use for the RTX stream.
- uint32_t ssrc;
+ RtxConfig() : rtx_payload_type(0), video_payload_type(0) {}
+ // SSRCs to use for the RTX streams.
+ std::vector<uint32_t> ssrcs;
// Payload type to use for the RTX stream.
int rtx_payload_type;
diff --git a/webrtc/video_engine/test/send_stream_tests.cc b/webrtc/video_engine/test/send_stream_tests.cc
index 1753b43..879a3e4 100644
--- a/webrtc/video_engine/test/send_stream_tests.cc
+++ b/webrtc/video_engine/test/send_stream_tests.cc
@@ -47,6 +47,7 @@
protected:
static const uint32_t kSendSsrc;
+ static const uint32_t kSendRtxSsrc;
void RunSendTest(Call* call,
const VideoSendStream::Config& config,
SendTransportObserver* observer) {
@@ -75,10 +76,13 @@
return config;
}
+ void TestNackRetransmission(uint32_t retransmit_ssrc);
+
test::FakeEncoder fake_encoder_;
};
const uint32_t VideoSendStreamTest::kSendSsrc = 0xC0FFEE;
+const uint32_t VideoSendStreamTest::kSendRtxSsrc = 0xBADCAFE;
TEST_F(VideoSendStreamTest, SendsSetSsrc) {
class SendSsrcObserver : public SendTransportObserver {
@@ -215,15 +219,15 @@
RunSendTest(call.get(), send_config, &observer);
}
-TEST_F(VideoSendStreamTest, RespondsToNack) {
+void VideoSendStreamTest::TestNackRetransmission(uint32_t retransmit_ssrc) {
class NackObserver : public SendTransportObserver, webrtc::Transport {
public:
- NackObserver()
+ NackObserver(uint32_t retransmit_ssrc)
: SendTransportObserver(30 * 1000),
thread_(ThreadWrapper::CreateThread(NackProcess, this)),
send_call_receiver_(NULL),
send_count_(0),
- ssrc_(0),
+ retransmit_ssrc_(retransmit_ssrc),
nacked_sequence_number_(0) {}
~NackObserver() {
@@ -247,7 +251,7 @@
EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(this));
rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
- rtcp_sender.SetRemoteSSRC(ssrc_);
+ rtcp_sender.SetRemoteSSRC(kSendSsrc);
RTCPSender::FeedbackState feedback_state;
EXPECT_EQ(0, rtcp_sender.SendRTCP(
@@ -277,14 +281,23 @@
// Nack second packet after receiving the third one.
if (++send_count_ == 3) {
- ssrc_ = header.ssrc;
nacked_sequence_number_ = header.sequenceNumber - 1;
unsigned int id;
EXPECT_TRUE(thread_->Start(id));
}
- if (header.sequenceNumber == nacked_sequence_number_)
+ uint16_t sequence_number = header.sequenceNumber;
+
+ if (header.ssrc == retransmit_ssrc_ && retransmit_ssrc_ != kSendSsrc) {
+ // Not kSendSsrc, assume correct RTX packet. Extract sequence number.
+ const uint8_t* rtx_header = packet + header.headerLength;
+ sequence_number = (rtx_header[0] << 8) + rtx_header[1];
+ }
+
+ if (sequence_number == nacked_sequence_number_) {
+ EXPECT_EQ(retransmit_ssrc_, header.ssrc);
send_test_complete_->Set();
+ }
return true;
}
@@ -292,9 +305,9 @@
scoped_ptr<ThreadWrapper> thread_;
PacketReceiver* send_call_receiver_;
int send_count_;
- uint32_t ssrc_;
+ uint32_t retransmit_ssrc_;
uint16_t nacked_sequence_number_;
- } observer;
+ } observer(retransmit_ssrc);
Call::Config call_config(&observer);
scoped_ptr<Call> call(Call::Create(call_config));
@@ -302,8 +315,20 @@
VideoSendStream::Config send_config = GetSendTestConfig(call.get());
send_config.rtp.nack.rtp_history_ms = 1000;
+ if (retransmit_ssrc != kSendSsrc)
+ send_config.rtp.rtx.ssrcs.push_back(retransmit_ssrc);
RunSendTest(call.get(), send_config, &observer);
}
+TEST_F(VideoSendStreamTest, RetransmitsNack) {
+ // Normal NACKs should use the send SSRC.
+ TestNackRetransmission(kSendSsrc);
+}
+
+TEST_F(VideoSendStreamTest, RetransmitsNackOverRtx) {
+ // NACKs over RTX should use a separate SSRC.
+ TestNackRetransmission(kSendRtxSsrc);
+}
+
} // namespace webrtc
