Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(208)

Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

Issue 2649913002: Count FlexFEC packets in |fec_bitrate_| in RTPSenderVideo. (Closed)
Patch Set: Unit test. Created 3 years, 11 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « no previous file | webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index ddf6422a9f16822f1b873f34f99b9e9872b2537e..a4ab7b7c882eed91cce5fe4dddaccea121fbe41c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -891,6 +891,49 @@ TEST_F(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
EXPECT_EQ(kFlexfecSsrc, flexfec_packet.Ssrc());
}
+TEST_F(RtpSenderTest, FecOverheadRate) {
+ constexpr int kMediaPayloadType = 127;
+ constexpr int kFlexfecPayloadType = 118;
+ constexpr uint32_t kMediaSsrc = 1234;
+ constexpr uint32_t kFlexfecSsrc = 5678;
+ const std::vector<RtpExtension> kNoRtpExtensions;
+ FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
+ kNoRtpExtensions, &fake_clock_);
+
+ // Reset |rtp_sender_| to use FlexFEC.
+ rtp_sender_.reset(new RTPSender(
+ false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
+ &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
+ &mock_rtc_event_log_, &send_packet_observer_,
+ &retransmission_rate_limiter_, nullptr));
+ rtp_sender_->SetSSRC(kMediaSsrc);
+ rtp_sender_->SetSequenceNumber(kSeqNum);
+ rtp_sender_->SetSendPayloadType(kMediaPayloadType);
+
+ // Parameters selected to generate a single FEC packet per media packet.
+ FecProtectionParams params;
+ params.fec_rate = 15;
+ params.max_fec_frames = 1;
+ params.fec_mask_type = kFecMaskRandom;
+ rtp_sender_->SetFecParameters(params, params);
+
+ // Send 10 media packet, and therefore 10 FEC packets.
+ EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, false))
+ .WillRepeatedly(testing::Return());
danilchap 2017/01/24 09:56:27 if it doesn't matter how many times InsertPacket w
brandtr 2017/01/24 10:28:13 Done.
+ for (int i = 0; i < 10; ++i) {
+ SendGenericPayload();
+ fake_clock_.AdvanceTimeMilliseconds(10);
+ }
+ constexpr size_t kRtpHeaderLength = 12;
+ constexpr size_t kFlexfecHeaderLength = 20;
+ constexpr size_t kGenericCodecHeaderLength = 1;
+ constexpr size_t kPayloadLength = sizeof(kPayloadData);
+ constexpr size_t kPacketLength = kRtpHeaderLength + kFlexfecHeaderLength +
+ kGenericCodecHeaderLength + kPayloadLength;
+ EXPECT_EQ(static_cast<uint32_t>(10 * 8 * kPacketLength / 0.101f),
danilchap 2017/01/24 09:56:27 may be EXPECT_NEAR to avoid cast and magic constan
brandtr 2017/01/24 10:28:13 Done. WDYT?
danilchap 2017/01/24 10:35:13 Nice!
+ rtp_sender_->FecOverheadRate());
+}
+
TEST_F(RtpSenderTest, FrameCountCallbacks) {
class TestCallback : public FrameCountObserver {
public:
« no previous file with comments | « no previous file | webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698