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Fixed sending Rtp packets with non zero csrcs and certain extensions.

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Added test that fails because of given issue.

BUG=webrtc:5413
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1586523003

Cr-Commit-Position: refs/heads/master@{#11258}
This commit is contained in:
danilchap
2016-01-14 14:55:19 -08:00
committed by Commit bot
parent 67b1e1ab0b
commit d9e62f5837
2 changed files with 47 additions and 14 deletions
Download Patch File Download Diff File Expand all files Collapse all files

17
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -1456,8 +1456,9 @@ bool RTPSender::FindHeaderExtensionPosition(RTPExtensionType type,
HeaderExtension header_extension(type);
size_t block_pos =
kRtpHeaderLength + rtp_header.numCSRCs + extension_block_pos;
size_t extension_pos =
kRtpHeaderLength + rtp_header.numCSRCs * sizeof(uint32_t);
size_t block_pos = extension_pos + extension_block_pos;
if (rtp_packet_length < block_pos + header_extension.length ||
rtp_header.headerLength < block_pos + header_extension.length) {
LOG(LS_WARNING) << "Failed to find extension position for " << type
@ -1466,8 +1467,8 @@ bool RTPSender::FindHeaderExtensionPosition(RTPExtensionType type,
}
// Verify that header contains extension.
if (!((rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs] == 0xBE) &&
(rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs + 1] == 0xDE))) {
if (!(rtp_packet[extension_pos] == 0xBE &&
rtp_packet[extension_pos + 1] == 0xDE)) {
LOG(LS_WARNING) << "Failed to find extension position for " << type
<< "as hdr extension not found.";
return false;
@ -1494,14 +1495,6 @@ RTPSender::ExtensionStatus RTPSender::VerifyExtension(
rtp_packet_length, rtp_header, &block_pos))
return ExtensionStatus::kError;
// Verify that header contains extension.
if (!((rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs] == 0xBE) &&
(rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs + 1] == 0xDE))) {
LOG(LS_WARNING)
<< "Failed to update absolute send time, hdr extension not found.";
return ExtensionStatus::kError;
}
// Verify first byte in block.
const uint8_t first_block_byte = (id << 4) + (extension_length_bytes - 2);
if (rtp_packet[block_pos] != first_block_byte)

44
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -146,16 +146,22 @@ class RtpSenderTest : public ::testing::Test {
uint8_t packet_[kMaxPacketLength];
void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) {
VerifyRTPHeaderCommon(rtp_header, kMarkerBit);
VerifyRTPHeaderCommon(rtp_header, kMarkerBit, 0);
}
void VerifyRTPHeaderCommon(const RTPHeader& rtp_header, bool marker_bit) {
VerifyRTPHeaderCommon(rtp_header, marker_bit, 0);
}
void VerifyRTPHeaderCommon(const RTPHeader& rtp_header,
bool marker_bit,
uint8_t number_of_csrcs) {
EXPECT_EQ(marker_bit, rtp_header.markerBit);
EXPECT_EQ(payload_, rtp_header.payloadType);
EXPECT_EQ(kSeqNum, rtp_header.sequenceNumber);
EXPECT_EQ(kTimestamp, rtp_header.timestamp);
EXPECT_EQ(rtp_sender_->SSRC(), rtp_header.ssrc);
EXPECT_EQ(0, rtp_header.numCSRCs);
EXPECT_EQ(number_of_csrcs, rtp_header.numCSRCs);
EXPECT_EQ(0U, rtp_header.paddingLength);
}
@ -550,6 +556,40 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
}
TEST_F(RtpSenderTestWithoutPacer,
BuildRTPPacketWithCSRCAndAudioLevelExtension) {
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
kAudioLevelExtensionId));
std::vector<uint32_t> csrcs;
csrcs.push_back(0x23456789);
rtp_sender_->SetCsrcs(csrcs);
size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
packet_, kPayload, kMarkerBit, kTimestamp, 0));
// Verify
webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
webrtc::RTPHeader rtp_header;
// Updating audio level is done in RTPSenderAudio, so simulate it here.
rtp_parser.Parse(&rtp_header);
EXPECT_TRUE(rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true,
kAudioLevel));
RtpHeaderExtensionMap map;
map.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
ASSERT_TRUE(valid_rtp_header);
ASSERT_FALSE(rtp_parser.RTCP());
VerifyRTPHeaderCommon(rtp_header, kMarkerBit, csrcs.size());
EXPECT_EQ(length, rtp_header.headerLength);
EXPECT_TRUE(rtp_header.extension.hasAudioLevel);
EXPECT_TRUE(rtp_header.extension.voiceActivity);
EXPECT_EQ(kAudioLevel, rtp_header.extension.audioLevel);
EXPECT_EQ(1u, rtp_header.numCSRCs);
EXPECT_EQ(csrcs[0], rtp_header.arrOfCSRCs[0]);
}
TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime));