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Fixes some pacer/padding issues found while testing.

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- A bug was introduced in r4234 causing no paced packets to be sent.
- Only update the sequence number counter if a padding packet is actually going to be sent, to avoid packet loss.
- Have all packets go through the pacer if pacing is enabled to avoid reordering.
- Fix race condition on reading capture_time_ms_/timestamp_ in rtp_sender.cc.

BUG=1837
TEST=trybots and vie_auto_test --automated
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1682004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4246 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2013-06-19 14:13:42 +00:00
parent 2d7617afce
commit 8ccb9f9716
9 changed files with 218 additions and 159 deletions
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30
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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@ -165,7 +165,8 @@ TEST_F(RtpSenderTest, BuildRTPPacket) {
int32_t length = rtp_sender_->BuildRTPheader(packet_,
kPayload,
kMarkerBit,
kTimestamp);
kTimestamp,
0);
EXPECT_EQ(12, length);
// Verify
@ -193,7 +194,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) {
int32_t length = rtp_sender_->BuildRTPheader(packet_,
kPayload,
kMarkerBit,
kTimestamp);
kTimestamp,
0);
EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
// Verify
@ -230,7 +232,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) {
int32_t length = rtp_sender_->BuildRTPheader(packet_,
kPayload,
kMarkerBit,
kTimestamp);
kTimestamp,
0);
EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
// Verify
@ -257,7 +260,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) {
int32_t length = rtp_sender_->BuildRTPheader(packet_,
kPayload,
kMarkerBit,
kTimestamp);
kTimestamp,
0);
EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
// Verify
@ -295,7 +299,8 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
int32_t length = rtp_sender_->BuildRTPheader(packet_,
kPayload,
kMarkerBit,
kTimestamp);
kTimestamp,
0);
EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
// Verify
@ -337,12 +342,12 @@ TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
rtp_sender_->SetTargetSendBitrate(300000);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_,
kPayload,
kMarkerBit,
kTimestamp);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
kTimestamp,
capture_time_ms);
// Packet should be stored in a send bucket.
EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_,
@ -391,12 +396,12 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
rtp_sender_->SetTargetSendBitrate(300000);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
int32_t rtp_length = rtp_sender_->BuildRTPheader(packet_,
kPayload,
kMarkerBit,
kTimestamp);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
kTimestamp,
capture_time_ms);
// Packet should be stored in a send bucket.
EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_,
@ -521,7 +526,8 @@ TEST_F(RtpSenderAudioTest, BuildRTPPacketWithAudioLevelExtension) {
int32_t length = rtp_sender_->BuildRTPheader(packet_,
kAudioPayload,
kMarkerBit,
kTimestamp);
kTimestamp,
0);
EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
// Currently, no space is added by for header extension by BuildRTPHeader().